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LanScape VOIP Media Engine™ - Pre-Sales Technical Support
 LanScape Support Forum -> LanScape VOIP Media Engine™ - Pre-Sales Technical Support
Subject Topic: Authenticating with Asterisk PBX Post ReplyPost New Topic
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mfitzgerald
Vetran
Vetran


Joined: June 14 2006
Location: United States
Posts: 142
Posted: June 14 2006 at 3:21pm | IP Logged Quote mfitzgerald
I have successfully tested this softphone implementation across a network from peer to peer via IP successfully. Now however I am attempting to authenticate with an Asterisk PBX box and am having problems authenticating. I saw the will’s message board posting about authentication problems and my problem does not appear to be a naming problem, I am getting an “SIP/2.0 401 Unauthorized” reply from the Asterisk logs.

As per will’s posting I have setup I have placed the function to add authentication in this order.

1: call "AddAuthorizationCredentials" with the username, password and their server's IP address.
2. call "EnableSipDomain"
3. call "EnableSipProxyServer"
4. call "EnableSipRegisterServer"

Any help would be great.

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mfitzgerald
Vetran
Vetran


Joined: June 14 2006
Location: United States
Posts: 142
Posted: June 14 2006 at 3:33pm | IP Logged Quote mfitzgerald
<-- SIP read from 172.26.253.176:5060:
REGISTER sip:lbb-vmail SIP/2.0
Via: SIP/2.0/UDP 172.26.253.176:5060
From: <sip:4307@lbb-vmail>;tag=b5e9e00
To: <sip:4307@lbb-vmail>
Call-Id: a67937ac-9e4f-4cd3-9721-67f6d3639ea9-00000a48@172.26.253.176
CSeq: 190738565 REGISTER
Expires: 20
Max-Forwards: 70
Contact: <sip:4307@172.26.253.176:5060>;user=phone
User-Agent: LanScape VOIP Media Engine/5.11.0205 (www.LanScapeCorp.com - This is a trial version not for general deployment)
Content-Length: 0



--- (11 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 172.26.253.176 : 5060 (non-NAT)
Transmitting (no NAT) to 172.26.253.176:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.26.253.176:5060;received=172.26.253.176
From: <sip:4307@lbb-vmail>;tag=b5e9e00
To: <sip:4307@lbb-vmail>
Call-ID: a67937ac-9e4f-4cd3-9721-67f6d3639ea9-00000a48@172.26.253.176
CSeq: 190738565 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4307@172.26.254.151>
Content-Length: 0


---

Transmitting (no NAT) to 172.26.253.176:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.26.253.176:5060;received=172.26.253.176
From: <sip:4307@lbb-vmail>;tag=b5e9e00
To: <sip:4307@lbb-vmail>;tag=as4d6ab59c
Call-ID: a67937ac-9e4f-4cd3-9721-67f6d3639ea9-00000a48@172.26.253.176
CSeq: 190738565 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4307@172.26.254.151>
WWW-Authenticate: Digest realm="asterisk", nonce="68f37571"
Content-Length: 0
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support
Administrator
Administrator


Joined: January 26 2005
Location: United States
Posts: 1666
Posted: June 14 2006 at 4:58pm | IP Logged Quote support
Hi Fitz,

You can download a new PhoneBase.cpp module that has all the code changes made and should make things a little easier for you. Download it from:

http://www.lanscapecorp.com/support/VoipMediaEngine/5.11/Pho neBasev511d.zip

It contains all the code changes for proxy, registrar and authentication support.

Overwrite your existing PhoneBase.cpp module with the new one in the above ZIP file.
There are a bunch of macros defined at the head of the PhoneBase.cpp module. Of special importance are the following:

ENABLE_AUTHENTICATION
ENABLE_SIP_REGISTRAR_AND_PROXY_SUPPORT

Set both of these to non zero to enable these features.

Also, there are some other macros at the head of the module you will have to change for your particular test setup.

One important thing you will have to do is go to line#1141 and specify proper challenge credential settings for the call to the AddAuthorizationCredentials() API proc.

If you make the changes above, you should be good to go. If you have further difficulty, post a complete SIP log from the Media Engine.


Note: We have updated the software examples a lot in the upcoming v5.12 release so that all these settings can now be specified via the GUI and make evaluation and testing simpler. V5.12 will be released very soon.

Repost as required,

Support
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mfitzgerald
Vetran
Vetran


Joined: June 14 2006
Location: United States
Posts: 142
Posted: June 15 2006 at 10:04am | IP Logged Quote mfitzgerald
Thank you for the fast response.

I have replaced PhoneBase.cpp with the new version in your zip and updated the registration and Asterisk connection data accordingly.

Eureka, the registration does work, however there is another problem. I have tried to make an invite between the LandScape Soft Phone to a test VoIP phone with no success. However an invite does work from the VoIP phone to the Soft Phone.

Here’s the log:
Code:

************* Log Opened (Jun 15 09:38:08) *************
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (0 Ms, To: 172.26.254.151:5060) >>>>
REGISTER sip:lbb-vmail SIP/2.0
Via: SIP/2.0/UDP 172.26.254.57:5060
From: <sip:4227@lbb-vmail>;tag=579e2e2
To: <sip:4227@lbb-vmail>
Call-Id: 7356f4e1-7abb-42d7-89a9-ec6bfd0a2a3c-000014b8@172.26.254.57
CSeq: 91878874 REGISTER
Expires: 180
Max-Forwards: 70
Contact: <sip:4227@172.26.254.57:5060>;user=phone
User-Agent: LanScape VOIP Media Engine/5.11.0205 (www.LanScapeCorp.com - This is a trial version not for general deployment)
Content-Length: 0




<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (0 Ms, From: 172.26.254.151:5060) <<<<
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.26.254.57:5060;received=172.26.254.57
From: <sip:4227@lbb-vmail>;tag=579e2e2
To: <sip:4227@lbb-vmail>
Call-ID: 7356f4e1-7abb-42d7-89a9-ec6bfd0a2a3c-000014b8@172.26.254.57
CSeq: 91878874 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4227@172.26.254.151>
Content-Length: 0




<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (0 Ms, From: 172.26.254.151:5060) <<<<
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.26.254.57:5060;received=172.26.254.57
From: <sip:4227@lbb-vmail>;tag=579e2e2
To: <sip:4227@lbb-vmail>;tag=as28a49729
Call-ID: 7356f4e1-7abb-42d7-89a9-ec6bfd0a2a3c-000014b8@172.26.254.57
CSeq: 91878874 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4227@172.26.254.151>
WWW-Authenticate: Digest realm="asterisk", nonce="4f66002e"
Content-Length: 0




>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (15 Ms, To: 172.26.254.151:5060) >>>>
REGISTER sip:lbb-vmail SIP/2.0
Via: SIP/2.0/UDP 172.26.254.57:5060
From: <sip:4227@lbb-vmail>;tag=57a4352
To: <sip:4227@lbb-vmail>
Call-Id: 7356f4e1-7abb-42d7-89a9-ec6bfd0a2a3c-000014b8@172.26.254.57
CSeq: 91878875 REGISTER
Authorization: Digest algorithm=md5, nonce="4f66002e", realm="asterisk", response="39966c0ae3951aecc5db8a41cb2e5467", uri="sip:lbb-vmail", username="4227"
Expires: 180
Max-Forwards: 70
Contact: <sip:4227@172.26.254.57:5060>;user=phone
User-Agent: LanScape VOIP Media Engine/5.11.0205 (www.LanScapeCorp.com - This is a trial version not for general deployment)
Content-Length: 0




<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (0 Ms, From: 172.26.254.151:5060) <<<<
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.26.254.57:5060;received=172.26.254.57
From: <sip:4227@lbb-vmail>;tag=57a4352
To: <sip:4227@lbb-vmail>
Call-ID: 7356f4e1-7abb-42d7-89a9-ec6bfd0a2a3c-000014b8@172.26.254.57
CSeq: 91878875 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4227@172.26.254.151>
Content-Length: 0




<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (0 Ms, From: 172.26.254.151:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.26.254.57:5060;received=172.26.254.57
From: <sip:4227@lbb-vmail>;tag=57a4352
To: <sip:4227@lbb-vmail>;tag=as28a49729
Call-ID: 7356f4e1-7abb-42d7-89a9-ec6bfd0a2a3c-000014b8@172.26.254.57
CSeq: 91878875 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Expires: 180
Contact: <sip:4227@172.26.254.57:5060>;expires=180
Date: Thu, 15 Jun 2006 14:38:08 GMT
Content-Length: 0




<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (7031 Ms, From: 172.26.254.151:5060) <<<<
NOTIFY sip:4227@172.26.254.57:5060 SIP/2.0
Via: SIP/2.0/UDP 172.26.254.151:5060;branch=z9hG4bK03c08a92
From: "Unknown" <sip:Unknown@172.26.254.151>;tag=as12282b3d
To: <sip:4227@172.26.254.57:5060>
Contact: <sip:Unknown@172.26.254.151>
Call-ID: 0845151a325d42554049ac9b46160088@172.26.254.151
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 80

Messages-Waiting: no
Message-Account: sip:asterisk@
Voice-Message: 0/0 (0/0)



>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (7031 Ms, To: 172.26.254.151:5060) >>>>
SIP/2.0 481 Transaction Does Not Exist
Via: SIP/2.0/UDP 172.26.254.151:5060;branch=z9hG4bK03c08a92
From: "Unknown" <sip:Unknown@172.26.254.151>;tag=as12282b3d
To: <sip:4227@172.26.254.57>;tag=3bb50e00
Call-Id: 0845151a325d42554049ac9b46160088@172.26.254.151
CSeq: 102 NOTIFY
User-Agent: LanScape VOIP Media Engine/5.11.0205 (www.LanScapeCorp.com - This is a trial version not for general deployment)
Content-Length: 0




>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (13672 Ms, To: 172.26.254.151:5060) >>>>
INVITE sip:4991@lbb-vmail SIP/2.0
Via: SIP/2.0/UDP 172.26.254.57:5060
From: "LanScape Phone" <sip:4227@lbb-vmail>;tag=57a9410
To: <sip:4991@lbb-vmail>
Contact: <sip:4227@lbb-vmail:5060>
Call-Id: 3a521e75-6685-4394-80e2-11ef53f66699-000014b8@172.26.254.57
CSeq: 91892594 INVITE
Max-Forwards: 70
Organization: CB2F325D-0BF5-442B-BDC4-37965875C824
Content-Length: 203
User-Agent: LanScape VOIP Media Engine/5.11.0205 (www.LanScapeCorp.com - This is a trial version not for general deployment)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp

v=0
o=4227 91891843 91891843 IN IP4 172.26.254.57
s=LanScape
c=IN IP4 172.26.254.57
t=0 0
m=audio 8006 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15




<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (13672 Ms, From: 172.26.254.151:5060) <<<<
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 172.26.254.57:5060;received=172.26.254.57
From: "LanScape Phone" <sip:4227@lbb-vmail>;tag=57a9410
To: <sip:4991@lbb-vmail>;tag=as0427c478
Call-ID: 3a521e75-6685-4394-80e2-11ef53f66699-000014b8@172.26.254.57
CSeq: 91892594 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4991@172.26.254.151>
Proxy-Authenticate: Digest realm="asterisk", nonce="6945f1d2"
Content-Length: 0




>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (32 Ms, To: 172.26.254.151:5060) >>>>
ACK sip:4991@lbb-vmail SIP/2.0
Via: SIP/2.0/UDP 172.26.254.57:5060;received=172.26.254.57
From: "LanScape Phone" <sip:4227@lbb-vmail>;tag=57a9410
To: <sip:4991@lbb-vmail>;tag=as0427c478
Call-Id: 3a521e75-6685-4394-80e2-11ef53f66699-000014b8@172.26.254.57
CSeq: 91892594 ACK
Max-Forwards: 70
Route: <sip:4991@172.26.254.151>
User-Agent: LanScape VOIP Media Engine/5.11.0205 (www.LanScapeCorp.com - This is a trial version not for general deployment)
Content-Length: 0




>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (0 Ms, To: 172.26.254.151:5060) >>>>
INVITE sip:4991@lbb-vmail SIP/2.0
Via: SIP/2.0/UDP 172.26.254.57:5060
From: "LanScape Phone" <sip:4227@lbb-vmail>;tag=57a9430
To: <sip:4991@lbb-vmail>
Contact: <sip:4227@lbb-vmail:5060>
Call-Id: 8c226a3d-b93d-45e5-82bb-fe49cbe8bbf1-000014b8@172.26.254.57
CSeq: 91892626 INVITE
Max-Forwards: 70
Organization: CB2F325D-0BF5-442B-BDC4-37965875C824
Proxy-Authorization: Digest algorithm=md5, nonce="6945f1d2", realm="asterisk", response="cae36832678f9d212eaea7ebb6fcca0d", uri="sip:4991@lbb-vmail", username="4227"
Content-Length: 203
User-Agent: LanScape VOIP Media Engine/5.11.0205 (www.LanScapeCorp.com - This is a trial version not for general deployment)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp

v=0
o=4227 91891875 91891875 IN IP4 172.26.254.57
s=LanScape
c=IN IP4 172.26.254.57
t=0 0
m=audio 8006 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15




<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (32 Ms, From: 172.26.254.151:5060) <<<<
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 172.26.254.57:5060;received=172.26.254.57
From: "LanScape Phone" <sip:4227@lbb-vmail>;tag=57a9430
To: <sip:4991@lbb-vmail>;tag=as03c5e5bf
Call-ID: 8c226a3d-b93d-45e5-82bb-fe49cbe8bbf1-000014b8@172.26.254.57
CSeq: 91892626 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4991@172.26.254.151>
Proxy-Authenticate: Digest realm="asterisk", nonce="53c52525"
Content-Length: 0




>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (0 Ms, To: 172.26.254.151:5060) >>>>
ACK sip:4991@lbb-vmail SIP/2.0
Via: SIP/2.0/UDP 172.26.254.57:5060;received=172.26.254.57
From: "LanScape Phone" <sip:4227@lbb-vmail>;tag=57a9430
To: <sip:4991@lbb-vmail>;tag=as03c5e5bf
Call-Id: 8c226a3d-b93d-45e5-82bb-fe49cbe8bbf1-000014b8@172.26.254.57
CSeq: 91892626 ACK
Max-Forwards: 70
Route: <sip:4991@172.26.254.151>
User-Agent: LanScape VOIP Media Engine/5.11.0205 (www.LanScapeCorp.com - This is a trial version not for general deployment)
Content-Length: 0




>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (0 Ms, To: 172.26.254.151:5060) >>>>
INVITE sip:4991@lbb-vmail SIP/2.0
Via: SIP/2.0/UDP 172.26.254.57:5060
From: "LanScape Phone" <sip:4227@lbb-vmail>;tag=57a9430
To: <sip:4991@lbb-vmail>
Contact: <sip:4227@lbb-vmail:5060>
Call-Id: 1ce740bf-2189-4f5b-8ad8-6e7b73e7486c-000014b8@172.26.254.57
CSeq: 91892626 INVITE
Max-Forwards: 70
Organization: CB2F325D-0BF5-442B-BDC4-37965875C824
Proxy-Authorization: Digest algorithm=md5, nonce="53c52525", realm="asterisk", response="5db544ab8c2ffbd40446b6beab7cfebe", uri="sip:4991@lbb-vmail", username="4227"
Content-Length: 203
User-Agent: LanScape VOIP Media Engine/5.11.0205 (www.LanScapeCorp.com - This is a trial version not for general deployment)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp

v=0
o=4227 91891875 91891875 IN IP4 172.26.254.57
s=LanScape
c=IN IP4 172.26.254.57
t=0 0
m=audio 8006 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15




<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (0 Ms, From: 172.26.254.151:5060) <<<<
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 172.26.254.57:5060;received=172.26.254.57
From: "LanScape Phone" <sip:4227@lbb-vmail>;tag=57a9430
To: <sip:4991@lbb-vmail>;tag=as5177fe4d
Call-ID: 1ce740bf-2189-4f5b-8ad8-6e7b73e7486c-000014b8@172.26.254.57
CSeq: 91892626 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4991@172.26.254.151>
Proxy-Authenticate: Digest realm="asterisk", nonce="367506b7"
Content-Length: 0




>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (0 Ms, To: 172.26.254.151:5060) >>>>
ACK sip:4991@lbb-vmail SIP/2.0
Via: SIP/2.0/UDP 172.26.254.57:5060;received=172.26.254.57
From: "LanScape Phone" <sip:4227@lbb-vmail>;tag=57a9430
To: <sip:4991@lbb-vmail>;tag=as5177fe4d
Call-Id: 1ce740bf-2189-4f5b-8ad8-6e7b73e7486c-000014b8@172.26.254.57
CSeq: 91892626 ACK
Max-Forwards: 70
Route: <sip:4991@172.26.254.151>
User-Agent: LanScape VOIP Media Engine/5.11.0205 (www.LanScapeCorp.com - This is a trial version not for general deployment)
Content-Length: 0




>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (3500 Ms, To: 172.26.254.151:5060) >>>>
REGISTER sip:lbb-vmail SIP/2.0
Via: SIP/2.0/UDP 172.26.254.57:5060
From: <sip:4227@lbb-vmail>;tag=57a3678
To: <sip:4227@lbb-vmail>
Call-Id: 7356f4e1-7abb-42d7-89a9-ec6bfd0a2a3c-000014b8@172.26.254.57
CSeq: 91878876 REGISTER
Expires: 0
Max-Forwards: 70
Contact: <sip:4227@172.26.254.57:5060>;user=phone
User-Agent: LanScape VOIP Media Engine/5.11.0205 (www.LanScapeCorp.com - This is a trial version not for general deployment)
Content-Length: 0




<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (3500 Ms, From: 172.26.254.151:5060) <<<<
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.26.254.57:5060;received=172.26.254.57
From: <sip:4227@lbb-vmail>;tag=57a3678
To: <sip:4227@lbb-vmail>
Call-ID: 7356f4e1-7abb-42d7-89a9-ec6bfd0a2a3c-000014b8@172.26.254.57
CSeq: 91878876 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4227@172.26.254.151>
Content-Length: 0




<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (0 Ms, From: 172.26.254.151:5060) <<<<
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.26.254.57:5060;received=172.26.254.57
From: <sip:4227@lbb-vmail>;tag=57a3678
To: <sip:4227@lbb-vmail>;tag=as3638eedb
Call-ID: 7356f4e1-7abb-42d7-89a9-ec6bfd0a2a3c-000014b8@172.26.254.57
CSeq: 91878876 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4227@172.26.254.151>
WWW-Authenticate: Digest realm="asterisk", nonce="6d9f929d"
Content-Length: 0




>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (0 Ms, To: 172.26.254.151:5060) >>>>
REGISTER sip:lbb-vmail SIP/2.0
Via: SIP/2.0/UDP 172.26.254.57:5060
From: <sip:4227@lbb-vmail>;tag=57a7e72
To: <sip:4227@lbb-vmail>
Call-Id: 7356f4e1-7abb-42d7-89a9-ec6bfd0a2a3c-000014b8@172.26.254.57
CSeq: 91878877 REGISTER
Authorization: Digest algorithm=md5, nonce="6d9f929d", realm="asterisk", response="b1364870f0b6f3911ab287e4ea91213f", uri="sip:lbb-vmail", username="4227"
Expires: 0
Max-Forwards: 70
Contact: <sip:4227@172.26.254.57:5060>;user=phone
User-Agent: LanScape VOIP Media Engine/5.11.0205 (www.LanScapeCorp.com - This is a trial version not for general deployment)
Content-Length: 0




<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (0 Ms, From: 172.26.254.151:5060) <<<<
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.26.254.57:5060;received=172.26.254.57
From: <sip:4227@lbb-vmail>;tag=57a7e72
To: <sip:4227@lbb-vmail>
Call-ID: 7356f4e1-7abb-42d7-89a9-ec6bfd0a2a3c-000014b8@172.26.254.57
CSeq: 91878877 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4227@172.26.254.151>
Content-Length: 0




<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (15 Ms, From: 172.26.254.151:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.26.254.57:5060;received=172.26.254.57
From: <sip:4227@lbb-vmail>;tag=57a7e72
To: <sip:4227@lbb-vmail>;tag=as3638eedb
Call-ID: 7356f4e1-7abb-42d7-89a9-ec6bfd0a2a3c-000014b8@172.26.254.57
CSeq: 91878877 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Expires: 0
Date: Thu, 15 Jun 2006 14:38:32 GMT
Content-Length: 0




************* Log Closed (Jun 15 09:38:32) *************

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support
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Posted: June 15 2006 at 4:31pm | IP Logged Quote support
Fitz,

We are looking into the problem you are having with authenticating. The last time we tested everything using Asterisk, it all worked as expected. The Asterisk box in our lab needs to be reconfigured with the latest software and set up. After that we should be able to get you an answer. Hang on.

Let us know what version of Asterisk you are testing with.

One thing we noticed is that your Asterisk domain name and the authentication realm are not configured the same. It shouldn't matter if they are different but I would set the authentication realm to be the same as the host machine name.

Support
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Posted: June 16 2006 at 11:00am | IP Logged Quote support
Update notice:

We are seeing the same thing here in our lab on our Asterisk server. Please be patient while we track down the problem.

Support
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mfitzgerald
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Posted: June 16 2006 at 1:47pm | IP Logged Quote mfitzgerald
The Asterisk version is Version 1.2.7.1

Thank you.
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Posted: June 16 2006 at 1:56pm | IP Logged Quote support
Hi fitz,

Thanks for the info. We are looking into the issue as we speak. You might have uncovered a subtle bug or incompatibility since we last performed inter-op testing with Asterisk.

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Posted: June 19 2006 at 7:42am | IP Logged Quote support
Hi Fitz,

Thanks for waiting as we look into this issue.

There definitely seems to be something going on with your version of Asterisk's authentication of SIP messages coming from the VOIP Media Engine.

We have verified with other SIP equipment here in our lab (various Cisco products and the open source Sip Express Router) and all authentication works as expected. The last time we performed testing with Asterisk using some previous version, we had no authentication issues.

We have also verified based on the same "nonce" value used in MD5 Digest authentication that the VOIP Media Engine computed its hash response value the same way as other SIP devices from Grandstream, Cisco, Polycom, Avaya, SwissVoice, etc).

What appears to be happening is the fields in the SIP "Proxy-Authorization:" header the VOIP Media Engine sends may be in a different order and we are suspecting a SIP header parsing problem in Asterisk. We say this because all the basic header information is present and computed correctly.

Our engineers are looking into this as we speak and should have an answer shortly. The order of fields in a "Proxy-Authorization:" header should not matter but something is definitely going on.

For the time being, we ask that you continue to perform your evaluation with authentication disabled on your Asterisk server.

If you are curious, the problem seems to be located in the chan_sip.c module.

Support
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Posted: June 19 2006 at 1:07pm | IP Logged Quote support
Hi Fitz,

We have identified the issue concerning authentication retries using v5.11 of the VOIP Media Engine and version v1.2.7.1 or later of Asterisk PBX.

In summary, authentication retries for SIP INVITE messages use a different call ID each time a new INVITE transaction is sent to Asterisk. Because of the way Asterisk handles authentication retires for INVITE requests, Asterisk will never properly authenticate the user event if proper authentication information is computed and placed into the INVITE SIP message.

INVITE authentication previously functioned properly with Asterisk so we are not sure when this issue crept into the picture.

For now we ask that you continue your evaluation with Asterisk authentication disabled. Version 5.12 of the VOIP Media Engine will be released by the end of this month. This issue will be fixed in that release. You can then request a new v5.12 trial image. There are some nice changes in v5.12 in addition to updated sample source code that makes evaluation and testing simpler.

Here is a SIP log of v5.12 fully authenticating with Asterisk using Asterisk host machine name "linuxsip" and realm "linuxsip". The Media Engine registers with Asterisk, places a call to extension 6301 and then hangs up:

Code:


************* Log Opened (Jun 19 13:06:46) *************

>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (3250 Ms, To: 192.168.1.181:5060) >>>>
REGISTER sip:linuxsip SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5069;rport;branch=z9hG4bK01849f5b
From: <sip:6300@linuxsip:5069>;tag=1845761
To: <sip:6300@linuxsip:5069>
Call-Id: efc3d598-7d7a-4439-9592-6ea4d63d0433-00000cac@192.168.1.2
CSeq: 8675868 REGISTER
Expires: 36000
Max-Forwards: 70
Contact: <sip:6300@192.168.1.2:5069>;user=phone
User-Agent: LanScape VOIP Media Engine/5.12.0301 (www.LanScapeCorp.com)
Content-Length: 0



<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRx (3218 Ms, From: 192.168.1.181:5060) <<<<
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5069;branch=z9hG4bK01849f5b
From: <sip:6300@linuxsip:5069>;tag=1845761
To: <sip:6300@linuxsip:5069>;tag=as585f8f7a
Call-ID: efc3d598-7d7a-4439-9592-6ea4d63d0433-00000cac@192.168.1.2
CSeq: 8675868 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 3600
Contact: <sip:6300@192.168.1.181>;expires=3600
Content-Length: 0



<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRx (0 Ms, From: 192.168.1.181:5060) <<<<
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5069;branch=z9hG4bK01849f5b
From: <sip:6300@linuxsip:5069>;tag=1845761
To: <sip:6300@linuxsip:5069>;tag=as585f8f7a
Call-ID: efc3d598-7d7a-4439-9592-6ea4d63d0433-00000cac@192.168.1.2
CSeq: 8675868 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 3600
Contact: <sip:6300@192.168.1.181>;expires=3600
WWW-Authenticate: Digest realm="linuxsip", nonce="10913936"
Content-Length: 0



>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (0 Ms, To: 192.168.1.181:5060) >>>>
REGISTER sip:linuxsip SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5069;rport;branch=z9hG4bK0184bebc
From: <sip:6300@linuxsip:5069>;tag=1846ff6
To: <sip:6300@linuxsip:5069>
Call-Id: efc3d598-7d7a-4439-9592-6ea4d63d0433-00000cac@192.168.1.2
CSeq: 8675869 REGISTER
Authorization: Digest algorithm=md5, nonce="10913936", realm="linuxsip", response="5904db1f31704c9856dc29f33e95f04d", uri="sip:linuxsip", username="6300"
Expires: 36000
Max-Forwards: 70
Contact: <sip:6300@192.168.1.2:5069>;user=phone
User-Agent: LanScape VOIP Media Engine/5.12.0301 (www.LanScapeCorp.com)
Content-Length: 0



<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRx (0 Ms, From: 192.168.1.181:5060) <<<<
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5069;branch=z9hG4bK0184bebc
From: <sip:6300@linuxsip:5069>;tag=1846ff6
To: <sip:6300@linuxsip:5069>;tag=as585f8f7a
Call-ID: efc3d598-7d7a-4439-9592-6ea4d63d0433-00000cac@192.168.1.2
CSeq: 8675869 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 3600
Contact: <sip:6300@192.168.1.181>;expires=3600
Content-Length: 0



<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRx (63 Ms, From: 192.168.1.181:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5069;branch=z9hG4bK0184bebc
From: <sip:6300@linuxsip:5069>;tag=1846ff6
To: <sip:6300@linuxsip:5069>;tag=as585f8f7a
Call-ID: efc3d598-7d7a-4439-9592-6ea4d63d0433-00000cac@192.168.1.2
CSeq: 8675869 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 3600
Contact: <sip:6300@192.168.1.2:5069>;expires=3600
Date: Mon, 19 Jun 2006 18:03:52 GMT
Content-Length: 0



>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (6750 Ms, To: 192.168.1.181:5060) >>>>
INVITE sip:6301@linuxsip SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5069;rport;branch=z9hG4bK018480ae
From: "extension 6300" <sip:6300@linuxsip>;tag=1847e35;x-UaId=xxxxx-yyyy-zzzz zz
To: <sip:6301@linuxsip>
Contact: <sip:6300@192.168.1.2:5069>;x-inst="VGVzdCBDYWxsIERhdG EgZnJvbSB0aGUgVlBob25lIGFwcC4="
Call-Id: 290b0ee7-8c5b-4d3f-8582-879b6a3ceb36-00000cac@192.168.1.2
CSeq: 8688024 INVITE
Max-Forwards: 70
Organization: 1D05B88F-516D-483E-B5A1-E22654099440
Content-Length: 221
User-Agent: LanScape VOIP Media Engine/5.12.0301 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp

v=0
o=6300 25457046 25457046 IN IP4 192.168.1.2
s=LanScape
c=IN IP4 192.168.1.2
t=0 0
m=audio 8042 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=ptime:20
a=fmtp:101 0-15



<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRx (6687 Ms, From: 192.168.1.181:5060) <<<<
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.2:5069;branch=z9hG4bK018480ae
From: "extension 6300" <sip:6300@linuxsip>;tag=1847e35;x-UaId=xxxxx-yyyy-zzzz zz
To: <sip:6301@linuxsip>;tag=as2553f3f4
Call-ID: 290b0ee7-8c5b-4d3f-8582-879b6a3ceb36-00000cac@192.168.1.2
CSeq: 8688024 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:6301@192.168.1.181>
Proxy-Authenticate: Digest realm="linuxsip", nonce="3bba930d"
Content-Length: 0



>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (79 Ms, To: 192.168.1.181:5060) >>>>
ACK sip:6301@linuxsip SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5069;branch=z9hG4bK018480ae
From: "extension 6300" <sip:6300@linuxsip>;tag=1847e35;x-UaId=xxxxx-yyyy-zzzz zz
To: <sip:6301@linuxsip>;tag=as2553f3f4
Call-Id: 290b0ee7-8c5b-4d3f-8582-879b6a3ceb36-00000cac@192.168.1.2
CSeq: 8688024 ACK
Max-Forwards: 70
Route: <sip:6301@192.168.1.181>
User-Agent: LanScape VOIP Media Engine/5.12.0301 (www.LanScapeCorp.com)
Content-Length: 0



>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (0 Ms, To: 192.168.1.181:5060) >>>>
INVITE sip:6301@linuxsip SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5069;rport;branch=z9hG4bK018480fd
From: "extension 6300" <sip:6300@linuxsip>;tag=1847e84;x-UaId=xxxxx-yyyy-zzzz zz
To: <sip:6301@linuxsip>
Contact: <sip:6300@192.168.1.2:5069>;x-inst="VGVzdCBDYWxsIERhdG EgZnJvbSB0aGUgVlBob25lIGFwcC4="
Call-Id: 290b0ee7-8c5b-4d3f-8582-879b6a3ceb36-00000cac@192.168.1.2
CSeq: 8688103 INVITE
Max-Forwards: 70
Organization: 1D05B88F-516D-483E-B5A1-E22654099440
Proxy-Authorization: Digest algorithm=md5, nonce="3bba930d", realm="linuxsip", response="3b7af95fe43afa57a72f55a58942d9cb", uri="sip:6301@linuxsip", username="6300"
Content-Length: 221
User-Agent: LanScape VOIP Media Engine/5.12.0301 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp

v=0
o=6300 25457125 25457125 IN IP4 192.168.1.2
s=LanScape
c=IN IP4 192.168.1.2
t=0 0
m=audio 8042 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=ptime:20
a=fmtp:101 0-15



<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRx (94 Ms, From: 192.168.1.181:5060) <<<<
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5069;branch=z9hG4bK018480fd
From: "extension 6300" <sip:6300@linuxsip>;tag=1847e84;x-UaId=xxxxx-yyyy-zzzz zz
To: <sip:6301@linuxsip>;tag=as1a67664e
Call-ID: 290b0ee7-8c5b-4d3f-8582-879b6a3ceb36-00000cac@192.168.1.2
CSeq: 8688103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:6301@192.168.1.181>
Content-Length: 0



<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRx (250 Ms, From: 192.168.1.181:5060) <<<<
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.2:5069;branch=z9hG4bK018480fd
From: "extension 6300" <sip:6300@linuxsip>;tag=1847e84;x-UaId=xxxxx-yyyy-zzzz zz
To: <sip:6301@linuxsip>;tag=as1a67664e
Call-ID: 290b0ee7-8c5b-4d3f-8582-879b6a3ceb36-00000cac@192.168.1.2
CSeq: 8688103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:6301@192.168.1.181>
Content-Length: 0



<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRx (969 Ms, From: 192.168.1.181:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5069;branch=z9hG4bK018480fd
From: "extension 6300" <sip:6300@linuxsip>;tag=1847e84;x-UaId=xxxxx-yyyy-zzzz zz
To: <sip:6301@linuxsip>;tag=as1a67664e
Call-ID: 290b0ee7-8c5b-4d3f-8582-879b6a3ceb36-00000cac@192.168.1.2
CSeq: 8688103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:6301@192.168.1.181>
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 8956 8956 IN IP4 192.168.1.181
s=session
c=IN IP4 192.168.1.181
t=0 0
m=audio 17144 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -


>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (1234 Ms, To: 192.168.1.181:5060) >>>>
ACK sip:6301@linuxsip SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5069;branch=z9hG4bK018480fd
From: "extension 6300" <sip:6300@linuxsip>;tag=1847e84;x-UaId=xxxxx-yyyy-zzzz zz
To: <sip:6301@linuxsip>;tag=as1a67664e
Call-Id: 290b0ee7-8c5b-4d3f-8582-879b6a3ceb36-00000cac@192.168.1.2
CSeq: 8688103 ACK
Max-Forwards: 70
Route: <sip:6301@192.168.1.181>
User-Agent: LanScape VOIP Media Engine/5.12.0301 (www.LanScapeCorp.com)
Content-Length: 0



>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (15297 Ms, To: 192.168.1.181:5060) >>>>
BYE sip:6301@linuxsip SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5069;rport;branch=z9hG4bK0184b402
From: "extension 6300" <sip:6300@linuxsip>;tag=1847e84;x-UaId=xxxxx-yyyy-zzzz zz
To: <sip:6301@linuxsip>;tag=as1a67664e
Call-Id: 290b0ee7-8c5b-4d3f-8582-879b6a3ceb36-00000cac@192.168.1.2
CSeq: 8688104 BYE
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.0301 (www.LanScapeCorp.com)
Content-Length: 0



<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRx (15297 Ms, From: 192.168.1.181:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5069;branch=z9hG4bK0184b402
From: "extension 6300" <sip:6300@linuxsip>;tag=1847e84;x-UaId=xxxxx-yyyy-zzzz zz
To: <sip:6301@linuxsip>;tag=as1a67664e
Call-ID: 290b0ee7-8c5b-4d3f-8582-879b6a3ceb36-00000cac@192.168.1.2
CSeq: 8688104 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:6301@192.168.1.181>
Content-Length: 0

************* Log Closed (Jun 19 13:07:15) *************


For further information, see the following post:

Asterisk v1.2.7.1 or higher not authenticating INVITE Fixed

Support
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mfitzgerald
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Posts: 142
Posted: June 20 2006 at 1:30pm | IP Logged Quote mfitzgerald
A very quick question.

I’m trying to find out how to disable authentication in asterisk. Anyone know how? I would think it is in the iax.conf file but I'm not certain.
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mfitzgerald
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Posted: June 20 2006 at 1:45pm | IP Logged Quote mfitzgerald
sorry. rather the sip.conf
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Posts: 1666
Posted: June 20 2006 at 3:38pm | IP Logged Quote support
Fitz,

When you define an extension that uses authentication in sip.conf, it will look like the following:

;define extension 6300. uses authentication.
[6300]
type=friend
username=6300
host=dynamic
canreinvite=no
auth=md5
secret=eddiez

The above extension will have to authenticate when it tries to register and when it initiates calls (INVITEs).

If you want to turn off all authentication challenges for an extension, simply comment out the two lines as below:

[6300]
type=friend
username=6300
host=dynamic
canreinvite=no
;auth=md5
;secret=eddiez


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Posted: June 30 2006 at 3:26pm | IP Logged Quote support
Hi Fitz,

v5.12 of the VOIP Media Engine™ has now been released.

See the following post:

LanScape VOIP Media Engine™ v5.12 Released

Request a new trial and we can move on from here.


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mfitzgerald
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Location: United States
Posts: 142
Posted: July 12 2006 at 12:05pm | IP Logged Quote mfitzgerald
I have downloaded the new version 5.12 and have two single-line phone apps running. They do register correctly and it is very helpful to allow for multiple instances of the LanScape application for testing.

However there still seem to be some authentication issues remaining.

Though the single-line phone application will receive calls, it fails to correctly make an authenticated invite.

Failed:
LanScape single-line phone app to single-line phone app:
Invite Authorization line from the Asterisk PBX
Code:

Proxy-Authorization: Digest realm=lbb-vmail,algorithm=md5, ,nonce="41b59669", ,realm="lbb-vmail", ,response="3a9d51d2d05f3b3c7e87672319204399", ,uri="sip:4307@lbb-vmail", ,username="4306"


Worked:
with another softphone app to single-line phone app:
Invite Authorization line from the Asterisk PBX
Code:

Proxy-Authorization: Digest username="4306",realm="lbb-vmail",nonce="06c9cd33",response= "a44fa0f097953e59dee1d319058290ac",uri="sip:4307@lbb-vmail:5 060",algorithm=MD5



I would assume Asterisk would ignore the double commas, though I do not know if the realm=lbb-vmail bit will cause problems w/o quotation marks. However, I noticed the port number was missing on the URI. I don’t know if this could be the issue or if it is some combination of these.

Here is the better representation of the PBX log regarding this issue:
The Failed Single-line phone app to Single-line phone app
Code:

<-- SIP read from 172.26.253.176:5061:


--- (0 headers 0 lines) Nat keepalive ---
lbb-vmail*CLI>
<-- SIP read from 172.26.253.176:5061:
INVITE sip:4307@lbb-vmail SIP/2.0
Via: SIP/2.0/UDP 172.26.253.176:5061;rport;branch=z9hG4bK00657990
From: 4306 <sip:4306@lbb-vmail>;tag=659218
To: <sip:4307@lbb-vmail>
Contact: <sip:4306@172.26.253.176:5061>
Call-Id: 4f1d5a62-745f-407f-a45f-00170140f799-00000ac8@172.26.253.176
CSeq: 6660290 INVITE
Max-Forwards: 70
Organization: BF73E56A-76AC-40F1-B8A8-4E0B4C8641C9
x-MyCustomHeader: "This is a modified transmitted SIP message."
Content-Length: 227
User-Agent: LanScape VOIP Media Engine/5.12.0301 (www.LanScapeCorp.com - This is a trial version not for general deployment)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp

v=0
o=4306 6639046 6639046 IN IP4 172.26.253.176
s=LanScape
c=IN IP4 172.26.253.176
t=0 0
m=audio 8590 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=ptime:20
a=fmtp:101 0-15


--- (14 headers 12 lines)---
Using INVITE request as basis request - 4f1d5a62-745f-407f-a45f-00170140f799-00000ac8@172.26.253.176
Sending to 172.26.253.176 : 5061 (non-NAT)
Reliably Transmitting (no NAT) to 172.26.253.176:5061:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 172.26.253.176:5061;rport;branch=z9hG4bK00657990;received=17 2.26.253.176
From: 4306 <sip:4306@lbb-vmail>;tag=659218
To: <sip:4307@lbb-vmail>;tag=as5a6bd9f4
Call-ID: 4f1d5a62-745f-407f-a45f-00170140f799-00000ac8@172.26.253.176
CSeq: 6660290 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4307@172.26.254.151>
Proxy-Authenticate: Digest realm="lbb-vmail", nonce="260eec2e"
Content-Length: 0


---
Scheduling destruction of call '4f1d5a62-745f-407f-a45f-00170140f799-00000ac8@172.26.253.17 6' in 15000 ms
Found user '4306'
lbb-vmail*CLI>
<-- SIP read from 172.26.253.176:5061:
ACK sip:4307@lbb-vmail SIP/2.0
Via: SIP/2.0/UDP 172.26.253.176:5061;rport;received=172.26.253.176;branch=z9h G4bK00657990
From: 4306 <sip:4306@lbb-vmail>;tag=659218
To: <sip:4307@lbb-vmail>;tag=as5a6bd9f4
Call-Id: 4f1d5a62-745f-407f-a45f-00170140f799-00000ac8@172.26.253.176
CSeq: 6660290 ACK
Max-Forwards: 70
Route: <sip:4307@172.26.254.151>
User-Agent: LanScape VOIP Media Engine/5.12.0301 (www.LanScapeCorp.com - This is a trial version not for general deployment)
x-MyCustomHeader: "This is a modified transmitted SIP message."
Content-Length: 0


--- (11 headers 0 lines)---
lbb-vmail*CLI>
<-- SIP read from 172.26.253.176:5061:
INVITE sip:4307@lbb-vmail SIP/2.0
Via: SIP/2.0/UDP 172.26.253.176:5061;rport;branch=z9hG4bK006579b0
From: 4306 <sip:4306@lbb-vmail>;tag=659238
To: <sip:4307@lbb-vmail>
Contact: <sip:4306@172.26.253.176:5061>
Call-Id: 4f1d5a62-745f-407f-a45f-00170140f799-00000ac8@172.26.253.176
CSeq: 6660322 INVITE
Max-Forwards: 70
Organization: BF73E56A-76AC-40F1-B8A8-4E0B4C8641C9
Proxy-Authorization: Digest realm=lbb-vmail,algorithm=md5, ,nonce="260eec2e", ,realm="lbb-vmail", ,response="6bb7b292ef234dc3fc618b78950dd304", ,uri="sip:4307@lbb-vmail", ,username="4306"
x-MyCustomHeader: "This is a modified transmitted SIP message."
Content-Length: 227
User-Agent: LanScape VOIP Media Engine/5.12.0301 (www.LanScapeCorp.com - This is a trial version not for general deployment)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp

v=0
o=4306 6639078 6639078 IN IP4 172.26.253.176
s=LanScape
c=IN IP4 172.26.253.176
t=0 0
m=audio 8590 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=ptime:20
a=fmtp:101 0-15


--- (15 headers 12 lines)---
Using INVITE request as basis request - 4f1d5a62-745f-407f-a45f-00170140f799-00000ac8@172.26.253.176
Sending to 172.26.253.176 : 5061 (non-NAT)
Reliably Transmitting (no NAT) to 172.26.253.176:5061:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 172.26.253.176:5061;rport;branch=z9hG4bK006579b0;received=17 2.26.253.176
From: 4306 <sip:4306@lbb-vmail>;tag=659238
To: <sip:4307@lbb-vmail>;tag=as5a6bd9f4
Call-ID: 4f1d5a62-745f-407f-a45f-00170140f799-00000ac8@172.26.253.176
CSeq: 6660322 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4307@172.26.254.151>
Proxy-Authenticate: Digest realm="lbb-vmail", nonce="0ebe68fc"
Content-Length: 0


---
Scheduling destruction of call '4f1d5a62-745f-407f-a45f-00170140f799-00000ac8@172.26.253.17 6' in 15000 ms
Found user '4306'
lbb-vmail*CLI>
<-- SIP read from 172.26.253.176:5061:
ACK sip:4307@lbb-vmail SIP/2.0
Via: SIP/2.0/UDP 172.26.253.176:5061;rport;received=172.26.253.176;branch=z9h G4bK006579b0
From: 4306 <sip:4306@lbb-vmail>;tag=659238
To: <sip:4307@lbb-vmail>;tag=as5a6bd9f4
Call-Id: 4f1d5a62-745f-407f-a45f-00170140f799-00000ac8@172.26.253.176
CSeq: 6660322 ACK
Max-Forwards: 70
Route: <sip:4307@172.26.254.151>
User-Agent: LanScape VOIP Media Engine/5.12.0301 (www.LanScapeCorp.com - This is a trial version not for general deployment)
x-MyCustomHeader: "This is a modified transmitted SIP message."
Content-Length: 0


--- (11 headers 0 lines)---
lbb-vmail*CLI>
<-- SIP read from 172.26.253.176:5061:
INVITE sip:4307@lbb-vmail SIP/2.0
Via: SIP/2.0/UDP 172.26.253.176:5061;rport;branch=z9hG4bK006579b0
From: 4306 <sip:4306@lbb-vmail>;tag=659238
To: <sip:4307@lbb-vmail>
Contact: <sip:4306@172.26.253.176:5061>
Call-Id: 4f1d5a62-745f-407f-a45f-00170140f799-00000ac8@172.26.253.176
CSeq: 6660322 INVITE
Max-Forwards: 70
Organization: BF73E56A-76AC-40F1-B8A8-4E0B4C8641C9
Proxy-Authorization: Digest realm=lbb-vmail,algorithm=md5, ,nonce="0ebe68fc", ,realm="lbb-vmail", ,response="78422c9643cf16ecc7ff7b8363d4fee4", ,uri="sip:4307@lbb-vmail", ,username="4306"
x-MyCustomHeader: "This is a modified transmitted SIP message."
Content-Length: 227
User-Agent: LanScape VOIP Media Engine/5.12.0301 (www.LanScapeCorp.com - This is a trial version not for general deployment)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp

v=0
o=4306 6639078 6639078 IN IP4 172.26.253.176
s=LanScape
c=IN IP4 172.26.253.176
t=0 0
m=audio 8590 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=ptime:20
a=fmtp:101 0-15


--- (15 headers 12 lines)---
Ignoring this INVITE request
lbb-vmail*CLI>
<-- SIP read from 172.26.253.176:5060:


--- (0 headers 0 lines) Nat keepalive ---
lbb-vmail*CLI>
<-- SIP read from 172.26.253.176:5061:


--- (0 headers 0 lines) Nat keepalive ---
lbb-vmail*CLI>
<-- SIP read from 172.26.253.176:5061:
CANCEL sip:4307@lbb-vmail SIP/2.0
Via: SIP/2.0/UDP 172.26.253.176:5061;rport;branch=z9hG4bK006579b0
From: 4306 <sip:4306@lbb-vmail>;tag=659238
To: <sip:4307@lbb-vmail>
Call-Id: 4f1d5a62-745f-407f-a45f-00170140f799-00000ac8@172.26.253.176
CSeq: 6660322 CANCEL
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.0301 (www.LanScapeCorp.com - This is a trial version not for general deployment)
x-MyCustomHeader: "This is a modified transmitted SIP message."
Content-Length: 0


--- (10 headers 0 lines)---
Sending to 172.26.253.176 : 5061 (non-NAT)
Reliably Transmitting (no NAT) to 172.26.253.176:5061:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.26.253.176:5061;rport;branch=z9hG4bK006579b0;received=17 2.26.253.176
From: 4306 <sip:4306@lbb-vmail>;tag=659238
To: <sip:4307@lbb-vmail>;tag=as5a6bd9f4
Call-ID: 4f1d5a62-745f-407f-a45f-00170140f799-00000ac8@172.26.253.176
CSeq: 6660322 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4307@172.26.254.151>
Content-Length: 0


---
Transmitting (no NAT) to 172.26.253.176:5061:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.26.253.176:5061;rport;branch=z9hG4bK006579b0;received=17 2.26.253.176
From: 4306 <sip:4306@lbb-vmail>;tag=659238
To: <sip:4307@lbb-vmail>;tag=as5a6bd9f4
Call-ID: 4f1d5a62-745f-407f-a45f-00170140f799-00000ac8@172.26.253.176
CSeq: 6660322 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4307@172.26.254.151>
Content-Length: 0


---
lbb-vmail*CLI>
<-- SIP read from 172.26.253.176:5061:
ACK sip:4307@lbb-vmail SIP/2.0
Via: SIP/2.0/UDP 172.26.253.176:5061;rport;received=172.26.253.176;branch=z9h G4bK006579b0
From: 4306 <sip:4306@lbb-vmail>;tag=659238
To: <sip:4307@lbb-vmail>;tag=as5a6bd9f4
Call-Id: 4f1d5a62-745f-407f-a45f-00170140f799-00000ac8@172.26.253.176
CSeq: 6660322 ACK
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.0301 (www.LanScapeCorp.com - This is a trial version not for general deployment)
x-MyCustomHeader: "This is a modified transmitted SIP message."
Content-Length: 0


--- (10 headers 0 lines)---
Retransmitting #1 (no NAT) to 172.26.253.176:5061:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.26.253.176:5061;rport;branch=z9hG4bK006579b0;received=17 2.26.253.176
From: 4306 <sip:4306@lbb-vmail>;tag=659238
To: <sip:4307@lbb-vmail>;tag=as5a6bd9f4
Call-ID: 4f1d5a62-745f-407f-a45f-00170140f799-00000ac8@172.26.253.176
CSeq: 6660322 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4307@172.26.254.151>
Content-Length: 0


---
Retransmitting #2 (no NAT) to 172.26.253.176:5061:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.26.253.176:5061;rport;branch=z9hG4bK006579b0;received=17 2.26.253.176
From: 4306 <sip:4306@lbb-vmail>;tag=659238
To: <sip:4307@lbb-vmail>;tag=as5a6bd9f4
Call-ID: 4f1d5a62-745f-407f-a45f-00170140f799-00000ac8@172.26.253.176
CSeq: 6660322 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4307@172.26.254.151>
Content-Length: 0


---
lbb-vmail*CLI>
<-- SIP read from 172.26.253.176:5060:


--- (0 headers 0 lines) Nat keepalive ---
Retransmitting #3 (no NAT) to 172.26.253.176:5061:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.26.253.176:5061;rport;branch=z9hG4bK006579b0;received=17 2.26.253.176
From: 4306 <sip:4306@lbb-vmail>;tag=659238
To: <sip:4307@lbb-vmail>;tag=as5a6bd9f4
Call-ID: 4f1d5a62-745f-407f-a45f-00170140f799-00000ac8@172.26.253.176
CSeq: 6660322 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4307@172.26.254.151>
Content-Length: 0


---
lbb-vmail*CLI>
<-- SIP read from 172.26.253.176:5061:


--- (0 headers 0 lines) Nat keepalive ---
Destroying call '4f1d5a62-745f-407f-a45f-00170140f799-00000ac8@172.26.253.17 6'
lbb-vmail*CLI>



Success other Softphone to Single-line phone app
Code:

--- (0 headers 0 lines) Nat keepalive ---
lbb-vmail*CLI>
<-- SIP read from 172.26.253.176:5061:
INVITE sip:4307@lbb-vmail:5060 SIP/2.0
Via: SIP/2.0/UDP 172.26.253.176:5061;branch=z9hG4bKBA7F4E53F24C4B30A51D1EBBE3 AAAA94
From: "Micaiah Line 2" <sip:4306@lbb-vmail>;tag=F1C4540A15244A84A8B64F84D63FC A73
To: <sip:4307@lbb-vmail:5060>
Contact: <sip:4306@172.26.253.176:5061>
Call-ID: 4100E68FE2324CF783C719032B033105@172.26.253.176
CSeq: 2240 INVITE
User-Agent: WOSISIP 1.0
Max-Forwards: 70
Content-Length: 345
Content-Type: application/sdp

v=0
o=4306 632882999550911650 632882999550911668 IN IP4 172.26.253.176
s=WOSI SIP 0.5
c=IN IP4 172.26.253.176
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 7 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15


--- (11 headers 15 lines)---
Using INVITE request as basis request - 4100E68FE2324CF783C719032B033105@172.26.253.176
Sending to 172.26.253.176 : 5061 (non-NAT)
Reliably Transmitting (no NAT) to 172.26.253.176:5061:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 172.26.253.176:5061;branch=z9hG4bKBA7F4E53F24C4B30A51D1EBBE3 AAAA94;received=172.26.253.176
From: "Micaiah Line 2" <sip:4306@lbb-vmail>;tag=F1C4540A15244A84A8B64F84D63FC A73
To: <sip:4307@lbb-vmail:5060>;tag=as40b64260
Call-ID: 4100E68FE2324CF783C719032B033105@172.26.253.176
CSeq: 2240 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4307@172.26.254.151>
Proxy-Authenticate: Digest realm="lbb-vmail", nonce="7a3ef812"
Content-Length: 0


---
Scheduling destruction of call '4100E68FE2324CF783C719032B033105@172.26.253.176' in 15000 ms
Found user '4306'
lbb-vmail*CLI>
<-- SIP read from 172.26.253.176:5061:
ACK sip:4307@lbb-vmail:5060 SIP/2.0
Via: SIP/2.0/UDP 172.26.253.176:5061;branch=z9hG4bKBA7F4E53F24C4B30A51D1EBBE3 AAAA94
From: "Micaiah Line 2" <sip:4306@lbb-vmail>;tag=F1C4540A15244A84A8B64F84D63FC A73
To: <sip:4307@lbb-vmail:5060>;tag=as40b64260
Call-ID: 4100E68FE2324CF783C719032B033105@172.26.253.176
CSeq: 2240 ACK
User-Agent: WOSISIP 1.0
Max-Forwards: 70
Content-Length: 0


--- (9 headers 0 lines)---
lbb-vmail*CLI>
<-- SIP read from 172.26.253.176:5061:
INVITE sip:4307@lbb-vmail:5060 SIP/2.0
Via: SIP/2.0/UDP 172.26.253.176:5061;branch=z9hG4bKAF907CA6C9344DACBB1A3CDE02 B7B3C1
From: "Micaiah Line 2" <sip:4306@lbb-vmail>;tag=F1C4540A15244A84A8B64F84D63FC A73
To: <sip:4307@lbb-vmail:5060>
Contact: <sip:4306@172.26.253.176:5061>
Call-ID: 4100E68FE2324CF783C719032B033105@172.26.253.176
CSeq: 2241 INVITE
User-Agent: WOSISIP 1.0
Max-Forwards: 70
Content-Length: 345
Content-Type: application/sdp
Proxy-Authorization: Digest username="4306",realm="lbb-vmail",nonce="7a3ef812",response= "1e7b3630a252467395172a736e72a3a0",uri="sip:4307@lbb-vmail:5 060",algorithm=MD5

v=0
o=4306 632882999550911650 632882999550911668 IN IP4 172.26.253.176
s=WOSI SIP 0.5
c=IN IP4 172.26.253.176
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 7 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15


--- (12 headers 15 lines)---
Using INVITE request as basis request - 4100E68FE2324CF783C719032B033105@172.26.253.176
Sending to 172.26.253.176 : 5061 (non-NAT)
Found user '4306'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 7
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 172.26.253.176:8000
Found description format PCMU
Found description format PCMA
Found description format GSM
Found description format iLBC
Found description format LPC
Found description format G729
Found description format telephone-event
Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x58e (gsm|ulaw|alaw|lpc10|g729|ilbc)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 4307 in from-internal (domain lbb-vmail)
list_route: hop: <sip:4306@172.26.253.176:5061>
Transmitting (no NAT) to 172.26.253.176:5061:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.26.253.176:5061;branch=z9hG4bKAF907CA6C9344DACBB1A3CDE02 B7B3C1;received=172.26.253.176
From: "Micaiah Line 2" <sip:4306@lbb-vmail>;tag=F1C4540A15244A84A8B64F84D63FC A73
To: <sip:4307@lbb-vmail:5060>
Call-ID: 4100E68FE2324CF783C719032B033105@172.26.253.176
CSeq: 2241 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4307@172.26.254.151>
Content-Length: 0


---
    -- Executing Macro("SIP/4306-121e", "exten-vm|novm|4307") in new stack
    -- Executing Macro("SIP/4306-121e", "user-callerid") in new stack
    -- Executing DBget("SIP/4306-121e", "AMPUSER=DEVICE/4306/user") in new stack
    -- DBget: varname=AMPUSER, family=DEVICE, key=4306/user
    -- DBget: set variable AMPUSER to 4306
    -- Executing DBget("SIP/4306-121e", "AMPUSERCIDNAME=AMPUSER/4306/cidname") in new stack
    -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=4306/cidname
    -- DBget: set variable AMPUSERCIDNAME to Micaiah Line 2
    -- Executing GotoIf("SIP/4306-121e", "0?5") in new stack
    -- Executing SetCallerID("SIP/4306-121e", ""Micaiah Line 2" <4306>") in new stack
    -- Executing NoOp("SIP/4306-121e", "Using CallerID "Micaiah Line 2" <4306>") in new stack
    -- Executing SetVar("SIP/4306-121e", "FROMCONTEXT=exten-vm") in new stack
    -- Executing Macro("SIP/4306-121e", "record-enable|4307|IN") in new stack
    -- Executing GotoIf("SIP/4306-121e", "0 > 0?2:4") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing AGI("SIP/4306-121e", "recordingcheck|20060712-111915|1152721155.781") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060712-111915|1152721155.781: Inbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing NoOp("SIP/4306-121e", "No recording needed") in new stack
    -- Executing SetVar("SIP/4306-121e", "X-Subject=") in new stack
    -- Executing NoOp("SIP/4306-121e", "X-Subject: ") in new stack
    -- Executing SIPAddHeader("SIP/4306-121e", "X-NewSubject: ") in new stack
    -- Executing Macro("SIP/4306-121e", "dial|15|tr|4307") in new stack
    -- Executing GotoIf("SIP/4306-121e", "0?4:2") in new stack
    -- Goto (macro-dial,s,2)
    -- Executing GotoIf("SIP/4306-121e", "0?5:4") in new stack
    -- Goto (macro-dial,s,4)
    -- Executing AGI("SIP/4306-121e", "dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
    -- dialparties.agi: priority = 4
    -- dialparties.agi: callingani2 = 0
    -- dialparties.agi: accountcode =
    -- dialparties.agi: channel = SIP/4306-121e
    -- dialparties.agi: callerid = 4306
    -- dialparties.agi: context = macro-dial
    -- dialparties.agi: callington = 0
    -- dialparties.agi: dnid = 4307
    -- dialparties.agi: request = dialparties.agi
    -- dialparties.agi: calleridname = Micaiah Line 2
    -- dialparties.agi: extension = s
    -- dialparties.agi: language = en
    -- dialparties.agi: uniqueid = 1152721155.781
    -- dialparties.agi: callingpres = 0
    -- dialparties.agi: type = SIP
    -- dialparties.agi: rdnis = unknown
    -- dialparties.agi: callingtns = 0
    -- dialparties.agi: enhanced = 0.0
dialparties.agi: Caller ID name and number are '4306'
dialparties.agi: Methodology of ring is 'none'
    -- dialparties.agi: Added extension 4307 to extension map
    -- dialparties.agi: Extension 4307 cf is disabled
    -- dialparties.agi: Extension 4307 do not disturb is disabled
       > dialparties.agi: extnum: 4307
       > dialparties.agi: exthascw: 0
       > dialparties.agi: exthascfb: 0
       > dialparties.agi: extcfb:
    -- dialparties.agi: Checking CW and CFB status for extension 4307
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
    -- dialparties.agi: Correct AMPMGRUSER and AMPMGRPASS
== Manager 'admin' logged off from 127.0.0.1
       > dialparties.agi: extstate: 0
dialparties.agi: Extension 4307 is available...skipping checks
    -- dialparties.agi: DbSet CALLTRACE/4307 to 4306
    -- AGI Script dialparties.agi completed, returning 0
    -- Executing Dial("SIP/4306-121e", "SIP/4307|15|tr") in new stack
We're at 172.26.254.151 port 13122
Adding codec 0x100 (g729) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
14 headers, 14 lines
Reliably Transmitting (no NAT) to 172.26.253.176:5060:
INVITE sip:4307@172.26.253.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.26.254.151:5060;branch=z9hG4bK03dc34b2
From: "Micaiah Line 2" <sip:4306@172.26.254.151>;tag=as12be7a80
To: <sip:4307@172.26.253.176:5060>
Contact: <sip:4306@172.26.254.151>
Call-ID: 696cb3f429e6fd50253727e40beb0380@172.26.254.151
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 12 Jul 2006 16:19:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-NewSubject:
Content-Type: application/sdp
Content-Length: 311

v=0
o=root 424 424 IN IP4 172.26.254.151
s=session
c=IN IP4 172.26.254.151
t=0 0
m=audio 13122 RTP/AVP 18 4 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Called 4307
Transmitting (no NAT) to 172.26.253.176:5061:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.26.253.176:5061;branch=z9hG4bKAF907CA6C9344DACBB1A3CDE02 B7B3C1;received=172.26.253.176
From: "Micaiah Line 2" <sip:4306@lbb-vmail>;tag=F1C4540A15244A84A8B64F84D63FC A73
To: <sip:4307@lbb-vmail:5060>;tag=as044407af
Call-ID: 4100E68FE2324CF783C719032B033105@172.26.253.176
CSeq: 2241 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4307@172.26.254.151>
Content-Length: 0


---
lbb-vmail*CLI>
<-- SIP read from 172.26.253.176:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.26.254.151:5060;received=172.26.254.151;branch=z9hG4bK03 dc34b2
x-MyCustomHeader: "This is a modified transmitted SIP message."
From: "Micaiah Line 2" <sip:4306@172.26.254.151>;tag=as12be7a80
To: <sip:4307@172.26.253.176:5060>
Call-ID: 696cb3f429e6fd50253727e40beb0380@172.26.254.151
CSeq: 102 INVITE
User-Agent: LanScape VOIP Media Engine/5.12.0301 (www.LanScapeCorp.com - This is a trial version not for general deployment)
Content-Length: 0



--- (9 headers 1 lines)---

<-- SIP read from 172.26.253.176:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.26.254.151:5060;received=172.26.254.151;branch=z9hG4bK03 dc34b2
x-MyCustomHeader: "This is a modified transmitted SIP message."
From: "Micaiah Line 2" <sip:4306@172.26.254.151>;tag=as12be7a80
To: <sip:4307@172.26.253.176:5060>;tag=6a6f6e
Call-ID: 696cb3f429e6fd50253727e40beb0380@172.26.254.151
CSeq: 102 INVITE
User-Agent: LanScape VOIP Media Engine/5.12.0301 (www.LanScapeCorp.com - This is a trial version not for general deployment)
Content-Length: 0



--- (9 headers 1 lines)---
    -- SIP/4307-67d4 is ringing
lbb-vmail*CLI>
<-- SIP read from 172.26.253.176:5060:


--- (0 headers 0 lines) Nat keepalive ---
lbb-vmail*CLI>
<-- SIP read from 172.26.253.176:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.26.254.151:5060;received=172.26.254.151;branch=z9hG4bK03 dc34b2
From: "Micaiah Line 2" <sip:4306@172.26.254.151>;tag=as12be7a80
To: <sip:4307@172.26.253.176>;tag=6a6f6e
Call-Id: 696cb3f429e6fd50253727e40beb0380@172.26.254.151
CSeq: 102 INVITE
Contact: <sip:4307@172.26.253.176:5060>
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
User-Agent: LanScape VOIP Media Engine/5.12.0301 (www.LanScapeCorp.com - This is a trial version not for general deployment)
Content-Length: 179
Content-Type: application/sdp

v=0
o=LanScape 3361709964 3361709964 IN IP4 172.26.253.176
s=LanScape
c=IN IP4 172.26.253.176
t=0 0
m=audio 8894 RTP/AVP 18
a=rtpmap:18 G729/8000/1
a=sendrecv
a=ptime:20

--- (11 headers 9 lines)---
Found RTP audio format 18
Peer audio RTP is at port 172.26.253.176:8894
Found description format G729
Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
list_route: hop: <sip:4307@172.26.253.176:5060>
set_destination: Parsing <sip:4307@172.26.253.176:5060> for address/port to send to
set_destination: set destination to 172.26.253.176, port 5060
Transmitting (no NAT) to 172.26.253.176:5060:
ACK sip:4307@172.26.253.176:5060 SIP/2.0
Via: SIP/2.0/UDP 172.26.254.151:5060;branch=z9hG4bK5dda6284
From: "Micaiah Line 2" <sip:4306@172.26.254.151>;tag=as12be7a80
To: <sip:4307@172.26.253.176:5060>;tag=6a6f6e
Contact: <sip:4306@172.26.254.151>
Call-ID: 696cb3f429e6fd50253727e40beb0380@172.26.254.151
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/4307-67d4 answered SIP/4306-121e
We're at 172.26.254.151 port 19570
Adding codec 0x100 (g729) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.26.253.176:5061:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.26.253.176:5061;branch=z9hG4bKAF907CA6C9344DACBB1A3CDE02 B7B3C1;received=172.26.253.176
From: "Micaiah Line 2" <sip:4306@lbb-vmail>;tag=F1C4540A15244A84A8B64F84D63FC A73
To: <sip:4307@lbb-vmail:5060>;tag=as044407af
Call-ID: 4100E68FE2324CF783C719032B033105@172.26.253.176
CSeq: 2241 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4307@172.26.254.151>
Content-Type: application/sdp
Content-Length: 311

v=0
o=root 424 424 IN IP4 172.26.254.151
s=session
c=IN IP4 172.26.254.151
t=0 0
m=audio 19570 RTP/AVP 18 4 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Attempting native bridge of SIP/4306-121e and SIP/4307-67d4
lbb-vmail*CLI>
<-- SIP read from 172.26.253.176:5061:
ACK sip:4307@172.26.254.151 SIP/2.0
Via: SIP/2.0/UDP 172.26.253.176:5061;branch=z9hG4bK20931358DB8B4F8E938F6838E9 99FE7D
From: "Micaiah Line 2" <sip:4306@lbb-vmail>;tag=F1C4540A15244A84A8B64F84D63FC A73
To: <sip:4307@lbb-vmail:5060>;tag=as044407af
Call-ID: 4100E68FE2324CF783C719032B033105@172.26.253.176
CSeq: 2241 ACK
User-Agent: WOSISIP 1.0
Max-Forwards: 70
Content-Length: 0

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support
Administrator
Administrator


Joined: January 26 2005
Location: United States
Posts: 1666
Posted: July 12 2006 at 1:23pm | IP Logged Quote support
Fitz,

We will look at it immediately. The double commas in the 2nd INVITE response are not correct.

Support
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support
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Administrator


Joined: January 26 2005
Location: United States
Posts: 1666
Posted: July 12 2006 at 1:33pm | IP Logged Quote support
Fitz,

For whatever reason, we are seeing the same behavior here with our support test applications and Asterisk.

We are going to talk to our development people right now because this issue was verified as being resolved.

We will try to get back with you asap.

Support
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support
Administrator
Administrator


Joined: January 26 2005
Location: United States
Posts: 1666
Posted: July 12 2006 at 3:02pm | IP Logged Quote support
Fitz,

Sorry for the delay. We think we have isolated the culprit.

We would like for you to perform a quick test. It means you will have to rebuild the sample applications. We assume you are using Visual Studio 6, 2003 or 2005.

In the PhoneBase.cpp module, search for the enumeration value: SipModifySipMessage. it should be around line 408.

Conditionally compile out the entire case that performs example SIP message modifications like the following:


Code:


     if(NotificationType == IMMEDIATE_NOTIFICATION)
     {
           switch(TelephonyEvent)
           {
#if 0
                case SipModifySipMessage:
                      {
                     
                           blah, blah, blah....
                           
                      }
                      break;
#endif
                     
                ////////////////////////////// ////////////
                // Inbound Event subscriptions.
                ////////////////////////////// ////////////
                case SipSubscriptionReceived:
                
                etc...
                
           }
     }


It appears that new functionality to modify outgoing SIP messages has a side effect. The side effect is that it is causing errors in SIP message construction just before SIP messages are transmitted by the media engine.

Perform this quick test and post your results back to this forum thread.

We performed the same test here and found that "Proxy-Authenticate:" headers in transmitted SIP messages were successfully being constructed.


Support
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mfitzgerald
Vetran
Vetran


Joined: June 14 2006
Location: United States
Posts: 142
Posted: July 12 2006 at 3:34pm | IP Logged Quote mfitzgerald
That seems to have adjusted the issue.

Code:

Proxy-Authorization: Digest algorithm=md5, nonce="59a65ad5", realm="lbb-vmail", response="7df57e0ee96199f1ee7340ec37c2f290", uri="sip:4307@lbb-vmail", username="4306"

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mfitzgerald
Vetran
Vetran


Joined: June 14 2006
Location: United States
Posts: 142
Posted: July 12 2006 at 3:35pm | IP Logged Quote mfitzgerald
A single-line app invite to a single-line app was successful with this code commented out.
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support
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Location: United States
Posts: 1666
Posted: July 12 2006 at 3:56pm | IP Logged Quote support
Fitz,

Your test results are good. You should work for LanScape’s QA department. :)

We QA and test these VOIP products in just about every conceivable fashion before we perform a product release. It takes a lot of time and effort. However, sometimes these issues creep in unexpectedly. Sorry for that.

Posting SIP logs the way you did really helps us get to the bottom of problems quickly. Thanks for that.

We will be performing an interim 5.12.3.3 release to resolve this latest issue. It should be available by this Friday.

This current support thread is getting pretty long so, if you happen to identify any other authentication issues, we should start a new thread.

Support
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