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Jalal
Vetran
Joined: April 24 2006
Location: Iran
Posts: 188
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| Posted: March 11 2010 at 4:46am | IP Logged
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Hi,
I am going to use StartEventSubscription to subscribe some events to Asterisk v1.6 . But it seems asterisk needs authorization for Subscribe events, and LMEVoip does not response to "401 Unauthorized" which is sent by asterisk.
I receive SipSubscriptionRequiresAuthentication immediate notification from LMEVoip in such state.
Is there any way to tell LMEVoip to Authorize automatically for Subscribe events? If no then how can I resend Subscribe with authorization parameters manually?
Thanks,
Jalal
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: March 11 2010 at 8:55pm | IP Logged
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Hi Jalal,
If the media engine (LME) has been given proper authentication credentials and a “far end” SIP device challenges any SIP message transmitted by the media engine, the challenges will be handled automatically when the 401/407 are received (in other words, the media engine will automatically retransmit the challenged SIP messages with proper authentication information).
If your VOIP application tells the media engine to transmit SUBSCRIBE requests to Asterisk and Asterisk then challenges them, make sure your application calls the AddAuthorizationCredentials() API procedure. If you tell the LME the proper auth credentials, all SIP exchanges should be automatic.
I wanted to actually test this today with Asterisk v1.6 but time ran out.
Randal
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Jalal
Vetran
Joined: April 24 2006
Location: Iran
Posts: 188
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| Posted: March 13 2010 at 12:55am | IP Logged
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Hi Randal,
So I think this is a bug in LME. Because I have used AddAuthorizationCredentials correctly, so that Registeration to Asterisk PBX is done as expected. But when SIP/Subscribe is sent and asterisk returns 401 Unauthorized, LME answers with a new Subscribe which does not contain Authorization SIP segment.
This is the line used to Subscribe presense to TrixBox 2.8.0.3 (using Asterisk 1.6):
Code:
StartEventSubscription(hSipEngine, '600','704','10.10.10.100 ',5060,True,'presence',#13#10'Accept: application/pidf+xml', 60,10000,20,EventSubscribeHandle)
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Code:
>>>> Tx:
SUBSCRIBE sip:704@10.10.10.100 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.15:5062;rport;branch=z9hG4bK005c6d b7
From: <sip:600@10.10.10.100>;tag=5c28cd
To: <sip:704@10.10.10.100>
Call-ID: 2293110e-23aa-4b99-83e5-2b1906d03809-00001780@10.10 .10.15
CSeq: 6048918 SUBSCRIBE
Expires: 60
Max-Forwards: 70
Contact: <sip:600@10.10.10.100:5060>
User-Agent: LanScape VOIP Media Engine/5.12.8.1 (www.LanScap eCorp.com)
Event: presence;param="
Accept: application/pidf+xml"
Content-Length: 0
<<<< Rx:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.10.10.15:5062;branch=z9hG4bK005c6db7;rec eived=10.10.10.15;rport=5062
From: <sip:600@10.10.10.100>;tag=5c28cd
To: <sip:704@10.10.10.100>;tag=as565465e4
Call-ID: 2293110e-23aa-4b99-83e5-2b1906d03809-00001780@10.10 .10.15
CSeq: 6048918 SUBSCRIBE
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", no nce="702d8d05"
Content-Length: 0
>>>> Tx:
SUBSCRIBE sip:704@10.10.10.100 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.15:5062;rport;branch=z9hG4bK005c88 03
From: <sip:600@10.10.10.100>;tag=5c4923
To: <sip:704@10.10.10.100>
Call-ID: 294292de-3ab5-4ab7-ae56-c523f7d93acc-00001780@10.10 .10.15
CSeq: 6048379 SUBSCRIBE
Expires: 60
Max-Forwards: 70
Contact: <sip:600@10.10.10.100:5060>
User-Agent: LanScape VOIP Media Engine/5.12.8.1 (www.LanScap eCorp.com)
Event: presence;param="
Accept: application/pidf+xml"
Content-Length: 0
<<<< Rx:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.10.10.15:5062;branch=z9hG4bK005c8803;rec eived=10.10.10.15;rport=5062
From: <sip:600@10.10.10.100>;tag=5c4923
To: <sip:704@10.10.10.100>;tag=as57558832
Call-ID: 294292de-3ab5-4ab7-ae56-c523f7d93acc-00001780@10.10 .10.15
CSeq: 6048379 SUBSCRIBE
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", no nce="1b504d12"
Content-Length: 0
...
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Thanks,
Jalal
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: March 16 2010 at 4:04pm | IP Logged
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Hi Jalal,
Thanks for pointing this out. It took a bit longer that expected to look into this….
There is a media engine + asterisk SIP interoperability issue – I see what is occurring. Internally the LME is actually computing the challenge response for the SUBSCRIBE but it is not putting it into the transmitted SUBSCRIBE that gets sent later on.
Looking at the LME source code, you might be able to get around this issue if you make the SUBSCRIBE challenge responses of “SIP/2.0 401 Unauthorized” look like “SIP/2.0 407 Proxy Authentication Required”. You can do this by using the SipModifySipMessage event for received SUBSCRIBE responses.
For Received SIP SUBSCRIBE challenges:
1)
Trap received SUBSCRIBE challenges from Asterisk via SipModifySipMessage.
2)
Change the SIP request line (the first SIP line) to “SIP/2.0 407 Proxy Authentication Required”.
3)
Change the “WWW-Authenticate:” header name to “Proxy-Authenticate:” but leave all the challenge data the same.
4)
Allow this modified received SIP message to be processed by the LME.
For Transmitted SIP SUBSCRIBE requests that contain authentication info:
1)
Trap ready-to-be transmitted SUBSCRIBE SIP messages using the SipModifySipMessage event.
2)
If the SUBSCRIBE request contains a “Proxy-Authorization:” header, change the header name to an “Authorization:” header.
3)
Allow the modified SIP message to be transmitted to Asterisk.
Note: pEventParameter of StartEventSubscription() API call:
If you need to add the SIP header line:
Accept: application/pidf+xml
To your transmitted SUBSCRIBE SIP messages, you should process the SipModifySipMessage event from the LME and add that SIP header line there.
The pEventParameter of StartEventSubscription() API call is meant to contain simple string values and not contain <cr><lf> characters. The two SIP headers in your SIP log are malformed:
Code:
Event: presence;param="
Accept: application/pidf+xml"
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Question:
By the way – how are you enabling presence in your asterisk setup?
Thanks,
Randal
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Jalal
Vetran
Joined: April 24 2006
Location: Iran
Posts: 188
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| Posted: March 17 2010 at 1:04am | IP Logged
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Hi,
Thanks for your detailed answer.
From what you described, this is not a media engine + asterisk SIP interoperability issue. But this is a media engine + any proxy interoperability issue which uses WWW-Authenticate for Subscribe events. So I think it's better to be fixed in your next release.
Currently we have defined a SIP Extension with no password, to be used for SIP presence subscription. So we don't receive 401 Unauthorized anymore. And we don't see this problem. This is a more simple solution to this problem till we can have your next release.
Thanks for your guidance about pEventParameter. I did know about it, but sometimes when you are so tired, using such tricks would help ;). It seems asterisk has no problem with " character at the end of accept line.
But about your question: As I said we are using TrixBox CE version 2.8.0.3. The freepbx web interface generates SIP hint parameters in SIP.conf automatically, so that let us monitor extensions using SIP presence. But if you want to use asterisk directly see
http://www.voip-info.org/wiki/view/Asterisk+presence . If you did not get result, I have good experince on asterisk configurations :).
Thanks,
Jalal
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: March 17 2010 at 9:33am | IP Logged
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Jalal,
Yes - This issue is not specific to LME+Asterisk. It will be fixed.
Yes - using a SIP extension with no password is a good workaround.
LME has been updated to better handle this situation. Below is a SIP log from a test. Still, the SUBSCRIBE is not being authenticated. I assume because I do not have asterisk setup properly.
We have the same exact PBX setup here: Trixbox 2.8.0.3. Please tell me how to setup presence via the web interface - I don't have time to figure out the proper asterisk setup myself and I would prefer to test this update completely this week so everything works in the future. Spoon feed me: I need the proper web based configuration steps so I can test the solution. Assume I will be running extensions 111 and 333 on the PBX.
Note:
I saw the same link to asterisk presence as you posted above but I couldn’t get it to work in the short amount of time that was available.
Code:
************* Log Opened (Mar 17 09:16:37) *************
>>>> TxTxTxTxTx (#1, [09:16:40.968] 0 Ms, To: 192.168.1.122:5060) >>>>
REGISTER sip:asterisk122.lslab.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5062;rport;branch=z9hG4bK007f3ca1
From: <sip:333@asterisk122.lslab.com>;tag=7f1750
To: <sip:333@asterisk122.lslab.com>
Call-ID: 15d9e575-cc81-4001-9b1a-e98593b2cdfd-000014e4@192.168.1.2
CSeq: 8323278 REGISTER
Expires: 3600
Max-Forwards: 70
Contact: <sip:333@192.168.1.2:5062>
User-Agent: LanScape Utility Softphone/5.10.0.8
x-CustomHeader-Extension-333: "Modified transmitted SIP message."
x-VOIP-SDK: LanScape VOIP Media Engine/6.0.0.17 (www.LanScapeCorp.com)
Content-Length: 0
<<<< RxRxRxRxRx (#1, [09:16:40.968] 0 Ms, From: 192.168.1.122:5060) <<<<
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5062;branch=z9hG4bK007f3ca1;received=192.168.1.2;rport=5062
From: <sip:333@asterisk122.lslab.com>;tag=7f1750
To: <sip:333@asterisk122.lslab.com>;tag=as3184d401
Call-ID: 15d9e575-cc81-4001-9b1a-e98593b2cdfd-000014e4@192.168.1.2
CSeq: 8323278 REGISTER
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="194802a9"
Content-Length: 0
>>>> TxTxTxTxTx (#2, [09:16:40.968] 0 Ms, To: 192.168.1.122:5060) >>>>
REGISTER sip:asterisk122.lslab.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5062;rport;branch=z9hG4bK007f020f
From: <sip:333@asterisk122.lslab.com>;tag=7f1750
To: <sip:333@asterisk122.lslab.com>
Call-ID: 15d9e575-cc81-4001-9b1a-e98593b2cdfd-000014e4@192.168.1.2
CSeq: 8323279 REGISTER
Authorization: Digest algorithm=md5,nonce="194802a9",realm="asterisk",
response="5bde3becc2bbdbf12f8ccbe10c909ac0",
uri="sip:asterisk122.lslab.com",username="333"
Expires: 3600
Max-Forwards: 70
Contact: <sip:333@192.168.1.2:5062>
User-Agent: LanScape Utility Softphone/5.10.0.8
x-CustomHeader-Extension-333: "Modified transmitted SIP message."
x-VOIP-SDK: LanScape VOIP Media Engine/6.0.0.17 (www.LanScapeCorp.com)
Content-Length: 0
<<<< RxRxRxRxRx (#2, [09:16:40.968] 0 Ms, From: 192.168.1.122:5060) <<<<
OPTIONS sip:333@192.168.1.2:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.122:5060;branch=z9hG4bK7aaf010b;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.122>;tag=as6b7e963e
To: <sip:333@192.168.1.2:5062>
Contact: <sip:Unknown@192.168.1.122>
Call-ID: 6b42d45e3f1f9f1124554e751b4e7288@192.168.1.122
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Date: Tue, 16 Mar 2010 09:26:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0
<<<< RxRxRxRxRx (#3, [09:16:40.968] 0 Ms, From: 192.168.1.122:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5062;branch=z9hG4bK007f020f;received=192.168.1.2;rport=5062
From: <sip:333@asterisk122.lslab.com>;tag=7f1750
To: <sip:333@asterisk122.lslab.com>;tag=as3184d401
Call-ID: 15d9e575-cc81-4001-9b1a-e98593b2cdfd-000014e4@192.168.1.2
CSeq: 8323279 REGISTER
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Expires: 3600
Contact: <sip:333@192.168.1.2:5062>;expires=3600
Date: Tue, 16 Mar 2010 09:26:41 GMT
Content-Length: 0
>>>> TxTxTxTxTx (#3, [09:16:40.984] 16 Ms, To: 192.168.1.122:5060) >>>>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.122:5060;rport;branch=z9hG4bK7aaf010b
From: "Unknown" <sip:Unknown@192.168.1.122>;tag=as6b7e963e
To: <sip:333@192.168.1.2:5062>;tag=ea860800
Call-ID: 6b42d45e3f1f9f1124554e751b4e7288@192.168.1.122
CSeq: 102 OPTIONS
User-Agent: LanScape Utility Softphone/5.10.0.8
x-CustomHeader-Extension-333: "Modified transmitted SIP message."
x-VOIP-SDK: LanScape VOIP Media Engine/6.0.0.17 (www.LanScapeCorp.com)
Content-Length: 0
>>>> TxTxTxTxTx (#4, [09:16:41.031] 47 Ms, To: 192.168.1.122:5060) >>>>
SUBSCRIBE sip:333@asterisk122.lslab.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5062;rport;branch=z9hG4bK007efd21
From: <sip:333@asterisk122.lslab.com>;tag=7f2c85
To: <sip:333@asterisk122.lslab.com>
Call-ID: c756838e-1271-4305-82c3-5f983ba26a58-000014e4@192.168.1.2
CSeq: 8316043 SUBSCRIBE
Expires: 3600
Max-Forwards: 70
Contact: <sip:333@asterisk122.lslab.com:5060>
User-Agent: LanScape Utility Softphone/5.10.0.8
Event: presence
Accept: application/pidf+xml
x-CustomHeader-Extension-333: "Modified transmitted SIP message."
x-VOIP-SDK: LanScape VOIP Media Engine/6.0.0.17 (www.LanScapeCorp.com)
Content-Length: 0
<<<< RxRxRxRxRx (#4, [09:16:41.031] 63 Ms, From: 192.168.1.122:5060) <<<<
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5062;branch=z9hG4bK007efd21;received=192.168.1.2;rport=5062
From: <sip:333@asterisk122.lslab.com>;tag=7f2c85
To: <sip:333@asterisk122.lslab.com>;tag=as270f4836
Call-ID: c756838e-1271-4305-82c3-5f983ba26a58-000014e4@192.168.1.2
CSeq: 8316043 SUBSCRIBE
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6363123b"
Content-Length: 0
>>>> TxTxTxTxTx (#5, [09:16:41.031] 0 Ms, To: 192.168.1.122:5060) >>>>
SUBSCRIBE sip:333@asterisk122.lslab.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5062;rport;branch=z9hG4bK007f21d0
From: <sip:333@asterisk122.lslab.com>;tag=7f2f44
To: <sip:333@asterisk122.lslab.com>
Call-ID: c397bcb9-b831-4e7a-be7a-c24f1d518349-000014e4@192.168.1.2
CSeq: 8342503 SUBSCRIBE
Expires: 3600
Max-Forwards: 70
Contact: <sip:333@asterisk122.lslab.com:5060>
User-Agent: LanScape Utility Softphone/5.10.0.8
Authorization: Digest algorithm=md5,nonce="6363123b",realm="asterisk",
response="6f713614f2fdf03177ae3988f5107298",
uri="sip:333@asterisk122.lslab.com",username="333"
Event: presence
Accept: application/pidf+xml
x-CustomHeader-Extension-333: "Modified transmitted SIP message."
x-VOIP-SDK: LanScape VOIP Media Engine/6.0.0.17 (www.LanScapeCorp.com)
Content-Length: 0
<<<< RxRxRxRxRx (#5, [09:16:41.031] 0 Ms, From: 192.168.1.122:5060) <<<<
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5062;branch=z9hG4bK007f21d0;received=192.168.1.2;rport=5062
From: <sip:333@asterisk122.lslab.com>;tag=7f2f44
To: <sip:333@asterisk122.lslab.com>;tag=as44ed47b0
Call-ID: c397bcb9-b831-4e7a-be7a-c24f1d518349-000014e4@192.168.1.2
CSeq: 8342503 SUBSCRIBE
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3ad840a8"
Content-Length: 0
>>>> TxTxTxTxTx (#6, [09:16:41.046] 15 Ms, To: 192.168.1.122:5060) >>>>
SUBSCRIBE sip:333@asterisk122.lslab.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5062;rport;branch=z9hG4bK007f4db7
From: <sip:333@asterisk122.lslab.com>;tag=7f5712
To: <sip:333@asterisk122.lslab.com>
Call-ID: 4a33ea27-d277-4ea1-9629-97c043abe490-000014e4@192.168.1.2
CSeq: 8327482 SUBSCRIBE
Expires: 3600
Max-Forwards: 70
Contact: <sip:333@asterisk122.lslab.com:5060>
User-Agent: LanScape Utility Softphone/5.10.0.8
Authorization: Digest algorithm=md5,nonce="3ad840a8",realm="asterisk",
response="5bb70d0983969fd6c648f367774b0b0c",
uri="sip:333@asterisk122.lslab.com",username="333"
Event: presence
Accept: application/pidf+xml
x-CustomHeader-Extension-333: "Modified transmitted SIP message."
x-VOIP-SDK: LanScape VOIP Media Engine/6.0.0.17 (www.LanScapeCorp.com)
Content-Length: 0
<<<< RxRxRxRxRx (#6, [09:16:41.046] 15 Ms, From: 192.168.1.122:5060) <<<<
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5062;branch=z9hG4bK007f4db7;received=192.168.1.2;rport=5062
From: <sip:333@asterisk122.lslab.com>;tag=7f5712
To: <sip:333@asterisk122.lslab.com>;tag=as6a52d6d4
Call-ID: 4a33ea27-d277-4ea1-9629-97c043abe490-000014e4@192.168.1.2
CSeq: 8327482 SUBSCRIBE
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4fe308fa"
Content-Length: 0
>>>> TxTxTxTxTx (#7, [09:16:42.984] 1938 Ms, To: 192.168.1.122:5060) >>>>
REGISTER sip:asterisk122.lslab.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5062;rport;branch=z9hG4bK007f342a
From: <sip:333@asterisk122.lslab.com>;tag=7eec2f
To: <sip:333@asterisk122.lslab.com>
Call-ID: 15d9e575-cc81-4001-9b1a-e98593b2cdfd-000014e4@192.168.1.2
CSeq: 8323280 REGISTER
Expires: 0
Max-Forwards: 70
Contact: <sip:333@192.168.1.2:5062>
User-Agent: LanScape Utility Softphone/5.10.0.8
x-CustomHeader-Extension-333: "Modified transmitted SIP message."
x-VOIP-SDK: LanScape VOIP Media Engine/6.0.0.17 (www.LanScapeCorp.com)
Content-Length: 0
<<<< RxRxRxRxRx (#7, [09:16:42.984] 1938 Ms, From: 192.168.1.122:5060) <<<<
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5062;branch=z9hG4bK007f342a;received=192.168.1.2;rport=5062
From: <sip:333@asterisk122.lslab.com>;tag=7eec2f
To: <sip:333@asterisk122.lslab.com>;tag=as3184d401
Call-ID: 15d9e575-cc81-4001-9b1a-e98593b2cdfd-000014e4@192.168.1.2
CSeq: 8323280 REGISTER
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="32acf2d8"
Content-Length: 0
>>>> TxTxTxTxTx (#8, [09:16:42.984] 0 Ms, To: 192.168.1.122:5060) >>>>
REGISTER sip:asterisk122.lslab.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5062;rport;branch=z9hG4bK007f04c6
From: <sip:333@asterisk122.lslab.com>;tag=7eec2f
To: <sip:333@asterisk122.lslab.com>
Call-ID: 15d9e575-cc81-4001-9b1a-e98593b2cdfd-000014e4@192.168.1.2
CSeq: 8323281 REGISTER
Authorization: Digest algorithm=md5,nonce="32acf2d8",realm="asterisk",
response="dba4558e36d5af544830dcb7f29f5286",
uri="sip:asterisk122.lslab.com",username="333"
Expires: 0
Max-Forwards: 70
Contact: <sip:333@192.168.1.2:5062>
User-Agent: LanScape Utility Softphone/5.10.0.8
x-CustomHeader-Extension-333: "Modified transmitted SIP message."
x-VOIP-SDK: LanScape VOIP Media Engine/6.0.0.17 (www.LanScapeCorp.com)
Content-Length: 0
<<<< RxRxRxRxRx (#8, [09:16:43.000] 16 Ms, From: 192.168.1.122:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5062;branch=z9hG4bK007f04c6;received=192.168.1.2;rport=5062
From: <sip:333@asterisk122.lslab.com>;tag=7eec2f
To: <sip:333@asterisk122.lslab.com>;tag=as3184d401
Call-ID: 15d9e575-cc81-4001-9b1a-e98593b2cdfd-000014e4@192.168.1.2
CSeq: 8323281 REGISTER
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Expires: 0
Date: Tue, 16 Mar 2010 09:26:43 GMT
Content-Length: 0
************* Log Closed (Mar 17 09:16:46) *************
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Jalal
Vetran
Joined: April 24 2006
Location: Iran
Posts: 188
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| Posted: March 18 2010 at 3:25am | IP Logged
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Hi,
Indeed there is no special settings needed to configure for SIP presence using TrixBox 2.8.0.3 . All default settings to define an extension using FreePBX web interface should be enough.
To monitor an extension using SIP/Presence a line like "exten => 100,hint,SIP/peername" is needed for each extension in extensions_additional.conf which FreePBX web config does this for us automatically. If for any reason this line is not added to this file in "[ext-local]" section, asterisk will answer with "SIP/2.0 404 Not Found" whenever a subscribe/presence event is sent (even if there is no www-authenticate parameter existing in SIP/Subscribe message).
So as asterisk is answering with "401 Unauthorized", I'm almost sure that your asterisk configuration for SIP/presence is complete and correct. In my opinion you should recheck your second SIP/Subscribe message which contains WWW-Authenticate. I think it should have something missing or wrong that causes asterisk to answer again with "401 Unauthorized".
If still you have doubt on your asterisk configuration, you may set "secret" parameter to empty, and test Subscribe/presence again.
Some quick notes to use trixbox web interface:
1- after opening first page use "switch" link in the up-right-corner of page to switch to admin mode. Use maint user and password to switch to admin mode.
2- To define a new extension from menu go to PBX->PBX Settings, then select Extensions from left pane. Select "Generic SIP device" and then submit button. Only enter "User Extension","Display Name","secret" and then press submit button. I usually change default value for following parameters but I don't think if they are related.
Call Waiting=disabled
canreinvite=yes
nat=no
3- After defining or changing extension parameters, you should select "Apply Configuration Changes" button from almost center of upper menu. You will find it in orange background. This button will restart your asterisk, till your changes take effect.
4- If you want to see SIP/hint line in extensions_additional.conf you would open PBX->Config File Editor from main menu. Then select extensions_additional.conf. This settings is located at "ext-local" section.
I hope it helps to find it soon :-).
Thanks,
Jalal
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: March 18 2010 at 8:27am | IP Logged
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Jalal,
Fantastic explanation – good work. OK – the focus is now on the LME authentication computation.
When you state:
To monitor an extension using SIP/Presence a line like "exten => 100,hint,SIP/peername" is needed for each extension in extensions_additional.conf…
Are you referring to the following RED lines:
(This text is from the [ext-local] section of the extensions_additional.conf on the PBX box here)
Code:
[ext-local]
include => ext-local-custom
exten => 111,1,Macro(exten-vm,novm,111)
exten => 111,n,Goto(${IVR_CONTEXT},return,1)
exten => 111,hint,SIP/111&Custom:DND111
exten => 333,1,Macro(exten-vm,333,333)
exten => 333,n,Goto(vmret,1)
exten => 333,hint,SIP/333&Custom:DND333
exten => ${VM_PREFIX}333,1,Macro(vm,333,DIRECTDIAL,${IVR_RETVM})
exten => ${VM_PREFIX}333,n,Goto(vmret,1)
exten => vmb333,1,Macro(vm,333,BUSY,${IVR_RETVM})
exten => vmb333,n,Goto(vmret,1)
exten => vmu333,1,Macro(vm,333,NOANSWER,${IVR_RETVM})
exten => vmu333,n,Goto(vmret,1)
exten => vms333,1,Macro(vm,333,NOMESSAGE,${IVR_RETVM})
exten => vms333,n,Goto(vmret,1)
exten => vmret,1,GotoIf($["${IVR_RETVM}" = "RETURN" & "${IVR_CONTEXT}" != ""]?playret)
exten => vmret,n,Hangup
exten => vmret,n(playret),Playback(exited-vm-will-be-transfered&silence/1)
exten => vmret,n,Goto(${IVR_CONTEXT},return,1)
; end of [ext-local]
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I will repost when we have further progress on this…
Thanks,
Randal
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Jalal
Vetran
Joined: April 24 2006
Location: Iran
Posts: 188
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| Posted: March 28 2010 at 12:24am | IP Logged
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Hi,
Sorry for late reply. We have been on holidays for our new year ceremony (Nowrouz).
Yes, Red lines are exactly what I meant FreePBX adds for hints.
Any progress on this issue?
Thanks,
Jalal
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: March 31 2010 at 10:19am | IP Logged
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Jala,
Happy new year.
I have not had time to look into this further. Assuming the authentication is what is causing the issue, I will have to set up a debug Asterisk setup to see why the authenticated SUBSCRIBE requests being sent to Asterisk are not being accepted. I see no glaring error in the SUBSCRIBE SIP challenge requests the media engine is transmitting (format wise).
The challenge authentication computation the media engine uses for the SUBSCRIBE requests is the same as the computation that is used for all other SIP message challenges.
I did not test it but can the media engine REGISTER using authentication?
How about initiating a call where the media engine transmitted INVITE is challenged. Does that work?
Randal
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Jalal
Vetran
Joined: April 24 2006
Location: Iran
Posts: 188
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| Posted: April 03 2010 at 12:53am | IP Logged
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Hi,
Yes, as I said in my first post, Register authentication is done correctly. INVITE authentication challenge is also done without problem. I have done successfuly outgoing calls using LME.
For more tests, I have found Zoiper free softphone. This softphone can show Contact's status using SIP/presence. Here is the detailed wireshark capture of all sip messages passed between Asterisk (10.10.10.100) and Zoiper (Ext 703 on 10.10.10.15).
Please see Frames 235 to 238. I hope it can help.
Code:
Frame 188 (646 bytes on wire, 646 bytes captured)
Ethernet II, Src: 00:24:01:02:6c:ac (00:24:01:02:6c:ac), Dst: Vmware_d1:1e:d7 (00:0c:29:d1:1e:d7)
Internet Protocol, Src: 10.10.10.15 (10.10.10.15), Dst: 10.10.10.100 (10.10.10.100)
User Datagram Protocol, Src Port: sip-tls (5061), Dst Port: sip (5060)
Session Initiation Protocol
Request-Line: SUBSCRIBE sip:703@10.10.10.100;transport=UDP SIP/2.0
Message Header
Via: SIP/2.0/UDP 10.10.10.15:5061;branch=z9hG4bK-d8754z-5d9d3358ecd75c40-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:703@10.10.10.15:5061>
To: "Jalal"<sip:703@10.10.10.100;transport=UDP>
From: "Jalal"<sip:703@10.10.10.100;transport=UDP>;tag=c72d9f25
Call-ID: ZDlkMjhiZGRmMTI0MGFlNDRjYzQzYmFkZmUyYmQxM2E.
CSeq: 1 SUBSCRIBE
Expires: 3600
Accept: application/simple-message-summary
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Zoiper rev.6848
Event: message-summary
Content-Length: 0
Frame 189 (637 bytes on wire, 637 bytes captured)
Ethernet II, Src: 00:24:01:02:6c:ac (00:24:01:02:6c:ac), Dst: Vmware_d1:1e:d7 (00:0c:29:d1:1e:d7)
Internet Protocol, Src: 10.10.10.15 (10.10.10.15), Dst: 10.10.10.100 (10.10.10.100)
User Datagram Protocol, Src Port: sip-tls (5061), Dst Port: sip (5060)
Session Initiation Protocol
Request-Line: REGISTER sip:10.10.10.100;transport=UDP SIP/2.0
Message Header
Via: SIP/2.0/UDP 10.10.10.15:5061;branch=z9hG4bK-d8754z-4c7828af0c7867cf-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:703@10.10.10.15:5061;rinstance=e9bd877fcf923b0e;transport=UDP>
To: "Jalal"<sip:703@10.10.10.100;transport=UDP>
From: "Jalal"<sip:703@10.10.10.100;transport=UDP>;tag=194edc4c
Call-ID: NDYxYzk4YjkzMGY5MDQxNmVjNzYyN2I4MjE5ZThjZWI.
CSeq: 1 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Zoiper rev.6848
Allow-Events: presence
Content-Length: 0
Frame 190 (628 bytes on wire, 628 bytes captured)
Ethernet II, Src: Vmware_d1:1e:d7 (00:0c:29:d1:1e:d7), Dst: 00:24:01:02:6c:ac (00:24:01:02:6c:ac)
Internet Protocol, Src: 10.10.10.100 (10.10.10.100), Dst: 10.10.10.15 (10.10.10.15)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip-tls (5061)
Session Initiation Protocol
Status-Line: SIP/2.0 401 Unauthorized
Message Header
Via: SIP/2.0/UDP 10.10.10.15:5061;branch=z9hG4bK-d8754z-5d9d3358ecd75c40-1---d8754z-;received=10.10.10.15;rport=5061
From: "Jalal"<sip:703@10.10.10.100;transport=UDP>;tag=c72d9f25
To: "Jalal"<sip:703@10.10.10.100;transport=UDP>;tag=as70f847d9
Call-ID: ZDlkMjhiZGRmMTI0MGFlNDRjYzQzYmFkZmUyYmQxM2E.
CSeq: 1 SUBSCRIBE
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="33ddef00"
Content-Length: 0
Frame 191 (627 bytes on wire, 627 bytes captured)
Ethernet II, Src: Vmware_d1:1e:d7 (00:0c:29:d1:1e:d7), Dst: 00:24:01:02:6c:ac (00:24:01:02:6c:ac)
Internet Protocol, Src: 10.10.10.100 (10.10.10.100), Dst: 10.10.10.15 (10.10.10.15)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip-tls (5061)
Session Initiation Protocol
Status-Line: SIP/2.0 401 Unauthorized
Message Header
Via: SIP/2.0/UDP 10.10.10.15:5061;branch=z9hG4bK-d8754z-4c7828af0c7867cf-1---d8754z-;received=10.10.10.15;rport=5061
From: "Jalal"<sip:703@10.10.10.100;transport=UDP>;tag=194edc4c
To: "Jalal"<sip:703@10.10.10.100;transport=UDP>;tag=as4ea3024f
Call-ID: NDYxYzk4YjkzMGY5MDQxNmVjNzYyN2I4MjE5ZThjZWI.
CSeq: 1 REGISTER
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2052c6cb"
Content-Length: 0
Frame 192 (817 bytes on wire, 817 bytes captured)
Ethernet II, Src: 00:24:01:02:6c:ac (00:24:01:02:6c:ac), Dst: Vmware_d1:1e:d7 (00:0c:29:d1:1e:d7)
Internet Protocol, Src: 10.10.10.15 (10.10.10.15), Dst: 10.10.10.100 (10.10.10.100)
User Datagram Protocol, Src Port: sip-tls (5061), Dst Port: sip (5060)
Session Initiation Protocol
Request-Line: SUBSCRIBE sip:703@10.10.10.100;transport=UDP SIP/2.0
Message Header
Via: SIP/2.0/UDP 10.10.10.15:5061;branch=z9hG4bK-d8754z-e02df0957dbc0809-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:703@10.10.10.15:5061>
To: "Jalal"<sip:703@10.10.10.100;transport=UDP>
From: "Jalal"<sip:703@10.10.10.100;transport=UDP>;tag=c72d9f25
Call-ID: ZDlkMjhiZGRmMTI0MGFlNDRjYzQzYmFkZmUyYmQxM2E.
CSeq: 2 SUBSCRIBE
Expires: 3600
Accept: application/simple-message-summary
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Zoiper rev.6848
Authorization: Digest username="703",realm="asterisk",nonce="33ddef00",uri="sip:703@10.10.10.100;transport=UDP",response="f555473db41b27f62db939d515e5bafc",algorithm=MD5
Event: message-summary
Content-Length: 0
Frame 193 (603 bytes on wire, 603 bytes captured)
Ethernet II, Src: Vmware_d1:1e:d7 (00:0c:29:d1:1e:d7), Dst: 00:24:01:02:6c:ac (00:24:01:02:6c:ac)
Internet Protocol, Src: 10.10.10.100 (10.10.10.100), Dst: 10.10.10.15 (10.10.10.15)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip-tls (5061)
Session Initiation Protocol
Status-Line: SIP/2.0 200 OK
Message Header
Via: SIP/2.0/UDP 10.10.10.15:5061;branch=z9hG4bK-d8754z-e02df0957dbc0809-1---d8754z-;received=10.10.10.15;rport=5061
From: "Jalal"<sip:703@10.10.10.100;transport=UDP>;tag=c72d9f25
To: "Jalal"<sip:703@10.10.10.100;transport=UDP>;tag=as70f847d9
Call-ID: ZDlkMjhiZGRmMTI0MGFlNDRjYzQzYmFkZmUyYmQxM2E.
CSeq: 2 SUBSCRIBE
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Expires: 3600
Contact: <sip:703@10.10.10.100>;expires=3600
Content-Length: 0
Frame 194 (804 bytes on wire, 804 bytes captured)
Ethernet II, Src: 00:24:01:02:6c:ac (00:24:01:02:6c:ac), Dst: Vmware_d1:1e:d7 (00:0c:29:d1:1e:d7)
Internet Protocol, Src: 10.10.10.15 (10.10.10.15), Dst: 10.10.10.100 (10.10.10.100)
User Datagram Protocol, Src Port: sip-tls (5061), Dst Port: sip (5060)
Session Initiation Protocol
Request-Line: REGISTER sip:10.10.10.100;transport=UDP SIP/2.0
Message Header
Via: SIP/2.0/UDP 10.10.10.15:5061;branch=z9hG4bK-d8754z-d01c53604c67871d-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:703@10.10.10.15:5061;rinstance=e9bd877fcf923b0e;transport=UDP>
To: "Jalal"<sip:703@10.10.10.100;transport=UDP>
From: "Jalal"<sip:703@10.10.10.100;transport=UDP>;tag=194edc4c
Call-ID: NDYxYzk4YjkzMGY5MDQxNmVjNzYyN2I4MjE5ZThjZWI.
CSeq: 2 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Zoiper rev.6848
Authorization: Digest username="703",realm="asterisk",nonce="2052c6cb",uri="sip:10.10.10.100;transport=UDP",response="74a6517e021b4425994d2e3211a18dff",algorithm=MD5
Allow-Events: presence
Content-Length: 0
Frame 195 (631 bytes on wire, 631 bytes captured)
Ethernet II, Src: Vmware_d1:1e:d7 (00:0c:29:d1:1e:d7), Dst: 00:24:01:02:6c:ac (00:24:01:02:6c:ac)
Internet Protocol, Src: 10.10.10.100 (10.10.10.100), Dst: 10.10.10.15 (10.10.10.15)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip-tls (5061)
Session Initiation Protocol
Request-Line: OPTIONS sip:703@10.10.10.15:5061;rinstance=e9bd877fcf923b0e;transport=UDP SIP/2.0
Message Header
Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK09a54fdc;rport
Max-Forwards: 70
From: "" <sip:@10.10.10.100>;tag=as5addd7d0
To: <sip:703@10.10.10.15:5061;rinstance=e9bd877fcf923b0e;transport=UDP>
Contact: <sip:10.10.10.100>
Call-ID: 4698d33d3c010f146cf2609749b53b6e@10.10.10.100
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Date: Sun, 28 Mar 2010 06:49:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0
Frame 196 (583 bytes on wire, 583 bytes captured)
Ethernet II, Src: 00:24:01:02:6c:ac (00:24:01:02:6c:ac), Dst: Vmware_d1:1e:d7 (00:0c:29:d1:1e:d7)
Internet Protocol, Src: 10.10.10.15 (10.10.10.15), Dst: 10.10.10.100 (10.10.10.100)
User Datagram Protocol, Src Port: sip-tls (5061), Dst Port: sip (5060)
Session Initiation Protocol
Status-Line: SIP/2.0 200 OK
Message Header
Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK09a54fdc;rport=5060
Contact: <sip:10.10.10.15:5061>
To: <sip:703@10.10.10.15:5061;rinstance=e9bd877fcf923b0e;transport=UDP>;tag=36686f17
From: <sip:10.10.10.100>;tag=as5addd7d0
Call-ID: 4698d33d3c010f146cf2609749b53b6e@10.10.10.100
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Zoiper rev.6848
Allow-Events: presence
Content-Length: 0
Frame 197 (684 bytes on wire, 684 bytes captured)
Ethernet II, Src: Vmware_d1:1e:d7 (00:0c:29:d1:1e:d7), Dst: 00:24:01:02:6c:ac (00:24:01:02:6c:ac)
Internet Protocol, Src: 10.10.10.100 (10.10.10.100), Dst: 10.10.10.15 (10.10.10.15)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip-tls (5061)
Session Initiation Protocol
Status-Line: SIP/2.0 200 OK
Message Header
Via: SIP/2.0/UDP 10.10.10.15:5061;branch=z9hG4bK-d8754z-d01c53604c67871d-1---d8754z-;received=10.10.10.15;rport=5061
From: "Jalal"<sip:703@10.10.10.100;transport=UDP>;tag=194edc4c
To: "Jalal"<sip:703@10.10.10.100;transport=UDP>;tag=as4ea3024f
Call-ID: NDYxYzk4YjkzMGY5MDQxNmVjNzYyN2I4MjE5ZThjZWI.
CSeq: 2 REGISTER
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Expires: 3600
Contact: <sip:703@10.10.10.15:5061;rinstance=e9bd877fcf923b0e;transport=UDP>;expires=3600
Date: Sun, 28 Mar 2010 06:49:48 GMT
Content-Length: 0
Frame 235 (618 bytes on wire, 618 bytes captured)
Ethernet II, Src: 00:24:01:02:6c:ac (00:24:01:02:6c:ac), Dst: Vmware_d1:1e:d7 (00:0c:29:d1:1e:d7)
Internet Protocol, Src: 10.10.10.15 (10.10.10.15), Dst: 10.10.10.100 (10.10.10.100)
User Datagram Protocol, Src Port: sip-tls (5061), Dst Port: sip (5060)
Session Initiation Protocol
Request-Line: SUBSCRIBE sip:600@10.10.10.100;transport=UDP SIP/2.0
Message Header
Via: SIP/2.0/UDP 10.10.10.15:5061;branch=z9hG4bK-d8754z-a397542f062a6408-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:703@10.10.10.15:5061>
To: <sip:600@10.10.10.100;transport=UDP>
From: "Jalal"<sip:703@10.10.10.100;transport=UDP>;tag=2e272141
Call-ID: ZjE2NmQyNDY5NjI2NzY5MjdhNTUyNDhjZDExY2VmZjE.
CSeq: 1 SUBSCRIBE
Expires: 1800
Accept: application/pidf+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Zoiper rev.6848
Event: presence
Content-Length: 0
Frame 236 (621 bytes on wire, 621 bytes captured)
Ethernet II, Src: Vmware_d1:1e:d7 (00:0c:29:d1:1e:d7), Dst: 00:24:01:02:6c:ac (00:24:01:02:6c:ac)
Internet Protocol, Src: 10.10.10.100 (10.10.10.100), Dst: 10.10.10.15 (10.10.10.15)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip-tls (5061)
Session Initiation Protocol
Status-Line: SIP/2.0 401 Unauthorized
Message Header
Via: SIP/2.0/UDP 10.10.10.15:5061;branch=z9hG4bK-d8754z-a397542f062a6408-1---d8754z-;received=10.10.10.15;rport=5061
From: "Jalal"<sip:703@10.10.10.100;transport=UDP>;tag=2e272141
To: <sip:600@10.10.10.100;transport=UDP>;tag=as74136443
Call-ID: ZjE2NmQyNDY5NjI2NzY5MjdhNTUyNDhjZDExY2VmZjE.
CSeq: 1 SUBSCRIBE
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="490086c4"
Content-Length: 0
Frame 237 (789 bytes on wire, 789 bytes captured)
Ethernet II, Src: 00:24:01:02:6c:ac (00:24:01:02:6c:ac), Dst: Vmware_d1:1e:d7 (00:0c:29:d1:1e:d7)
Internet Protocol, Src: 10.10.10.15 (10.10.10.15), Dst: 10.10.10.100 (10.10.10.100)
User Datagram Protocol, Src Port: sip-tls (5061), Dst Port: sip (5060)
Session Initiation Protocol
Request-Line: SUBSCRIBE sip:600@10.10.10.100;transport=UDP SIP/2.0
Message Header
Via: SIP/2.0/UDP 10.10.10.15:5061;branch=z9hG4bK-d8754z-3ab773384a5c1a9b-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:703@10.10.10.15:5061>
To: <sip:600@10.10.10.100;transport=UDP>
From: "Jalal"<sip:703@10.10.10.100;transport=UDP>;tag=2e272141
Call-ID: ZjE2NmQyNDY5NjI2NzY5MjdhNTUyNDhjZDExY2VmZjE.
CSeq: 2 SUBSCRIBE
Expires: 1800
Accept: application/pidf+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Zoiper rev.6848
Authorization: Digest username="703",realm="asterisk",nonce="490086c4",uri="sip:600@10.10.10.100;transport=UDP",response="1049e5b9dfdd459ba90604d06c5ee6e6",algorithm=MD5
Event: presence
Content-Length: 0
Frame 238 (596 bytes on wire, 596 bytes captured)
Ethernet II, Src: Vmware_d1:1e:d7 (00:0c:29:d1:1e:d7), Dst: 00:24:01:02:6c:ac (00:24:01:02:6c:ac)
Internet Protocol, Src: 10.10.10.100 (10.10.10.100), Dst: 10.10.10.15 (10.10.10.15)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip-tls (5061)
Session Initiation Protocol
Status-Line: SIP/2.0 200 OK
Message Header
Via: SIP/2.0/UDP 10.10.10.15:5061;branch=z9hG4bK-d8754z-3ab773384a5c1a9b-1---d8754z-;received=10.10.10.15;rport=5061
From: "Jalal"<sip:703@10.10.10.100;transport=UDP>;tag=2e272141
To: <sip:600@10.10.10.100;transport=UDP>;tag=as74136443
Call-ID: ZjE2NmQyNDY5NjI2NzY5MjdhNTUyNDhjZDExY2VmZjE.
CSeq: 2 SUBSCRIBE
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Expires: 1800
Contact: <sip:600@10.10.10.100>;expires=1800
Content-Length: 0
Frame 239 (1069 bytes on wire, 1069 bytes captured)
Ethernet II, Src: Vmware_d1:1e:d7 (00:0c:29:d1:1e:d7), Dst: 00:24:01:02:6c:ac (00:24:01:02:6c:ac)
Internet Protocol, Src: 10.10.10.100 (10.10.10.100), Dst: 10.10.10.15 (10.10.10.15)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip-tls (5061)
Session Initiation Protocol
Request-Line: NOTIFY sip:703@10.10.10.15:5061 SIP/2.0
Message Header
Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0ff5ffc0;rport
Max-Forwards: 70
From: <sip:600@10.10.10.100;transport=UDP>;tag=as74136443
To: "Jalal"<sip:703@10.10.10.100;transport=UDP>;tag=2e272141
Contact: <sip:600@10.10.10.100>
Call-ID: ZjE2NmQyNDY5NjI2NzY5MjdhNTUyNDhjZDExY2VmZjE.
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Event: presence
Content-Type: application/pidf+xml
Subscription-State: active
Content-Length: 525
Message Body
Frame 240 (409 bytes on wire, 409 bytes captured)
Ethernet II, Src: 00:24:01:02:6c:ac (00:24:01:02:6c:ac), Dst: Vmware_d1:1e:d7 (00:0c:29:d1:1e:d7)
Internet Protocol, Src: 10.10.10.15 (10.10.10.15), Dst: 10.10.10.100 (10.10.10.100)
User Datagram Protocol, Src Port: sip-tls (5061), Dst Port: sip (5060)
Session Initiation Protocol
Status-Line: SIP/2.0 200 OK
Message Header
Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0ff5ffc0;rport=5060
Contact: <sip:703@10.10.10.15:5061>
To: "Jalal"<sip:703@10.10.10.100;transport=UDP>;tag=2e272141
From: <sip:600@10.10.10.100;transport=UDP>;tag=as74136443
Call-ID: ZjE2NmQyNDY5NjI2NzY5MjdhNTUyNDhjZDExY2VmZjE.
CSeq: 102 NOTIFY
User-Agent: Zoiper rev.6848
Content-Length: 0
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Thanks,
Jalal
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: April 03 2010 at 12:17pm | IP Logged
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Thanks Jalal.
The Wireshark info you posted will definitely help. I will look into this and find the solution.
If you have the actual Wireshark capture file, I would like to look at it too. Upload it to your support FTP account if you want.
I am sure this is a simple fix.
Randal
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Jalal
Vetran
Joined: April 24 2006
Location: Iran
Posts: 188
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| Posted: April 04 2010 at 5:52am | IP Logged
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Hi,
Indeed I did not have the wireshark of my previous post. But I have saved another one with exactly the same configuration, and uploaded to our support FTP account.
Thanks for your efforts,
Jalal
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: April 05 2010 at 8:34am | IP Logged
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Thank you Jalal - I see it.
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Jalal
Vetran
Joined: April 24 2006
Location: Iran
Posts: 188
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| Posted: June 08 2010 at 7:27am | IP Logged
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Hi Randal,
Any progress on this issue?
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: June 09 2010 at 5:55am | IP Logged
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Hi Jalal,
There is a current change that has been implemented/tested that appears to be what we need to get the SUBSCRIBE requests properly authorized. I have not merged this change back into the main LME code base because there are other code branches/updates that are taking precedence at the moment.
I surely intend to have this SUBSCRIBE authorize update merged back into the main code branch and included in the next media engine product release. I do not have a hard release date scheduled. We are working tirelessly to get out the next release and simply have adopted the “as soon as possible” approach.
Best regards,
Randal
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LanScape Support Forum -> LanScape VOIP Media Engine™ - Technical Support
Topic: Subscribe Authorization
