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speedvoip
Vetran
Joined: August 07 2008
Location: Canada
Posts: 156
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| Posted: December 16 2008 at 10:40am | IP Logged
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Hi Support:
With SingleLine Phone example within latest image v6.0.0.3, DTMF does not work with IVR server even though setting related to DTMF is correct(Both/Simultaneous). Pls try the following server for troubleshooting.
IVR Server: sw2.sp.ring-fone.com Port: 6060
user: 77880002 password: 785236
Pls dial 1008 or 1006 directly for accessing IVR(G.729). When hearing voice prompt, press 4 for english information.
Thanks,
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: December 16 2008 at 1:44pm | IP Logged
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Which single line phone sample app did you use – the managed code version or the native code version?
Using v6.0.0.3 media engine with another softphone app here and accessing your server using works good using RFC2833 DTMF signaling. Your server however does not seem to respond to in-band DTMF signaling.
Its probably a bug in the sample source code. If you tell us which single line phone sample you used, we will check further.
Support
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speedvoip
Vetran
Joined: August 07 2008
Location: Canada
Posts: 156
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| Posted: December 17 2008 at 1:59am | IP Logged
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Hi support:
Sample used for test is native code version. I think there exists a bug in handling DTMF within example code. Pls fix it to allow me test again.
Thanks
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speedvoip
Vetran
Joined: August 07 2008
Location: Canada
Posts: 156
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| Posted: December 17 2008 at 10:12am | IP Logged
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Hi Randal:
Urgent to get u resolve DTMF bug at native singleline phone example for accessing IVR server.
Thanks,
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: December 17 2008 at 5:27pm | IP Logged
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Hi George,
Hmmm… something must have changed in your sample source code maybe???
I just performed the following test:
1)
Installed V6.0.0.3 on a development machine here in the lab.
2)
Started the native code single line soft phone.
3)
Configured the phone settings for the sw2 server at your location.
4)
Set the phone line codec to G729.
5)
Set the DTMF type for all codecs to be “Both” and then selected all the checkboxes for “Simultaneous DTMF and audio output”.
6)
Made a test call to extension 1008 and played around with the DTMF menus (once I got it to change from Chinese prompts to English by pressing DTMF keypad 2).
All seems to be ok. Is there some change in your code that is causing the issue?
Here is the SIP log for the test call:
Code:
************* Log Opened (Dec 17 17:19:45) *************
>>>> TxTxTxTxTx (#1, [17:19:50.468] 0 Ms, To: 203.167.54.180:6060) >>>>
REGISTER sip:sw2.sp.ring-fone.com:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5068;rport;branch=z9hG4bK01cc01a7
From: <sip:77880002@sw2.sp.ring-fone.com:6060>;tag=1cbb779
To: <sip:77880002@sw2.sp.ring-fone.com:6060>
Call-ID: 12c4b207-306e-45d1-8e9f-832132e20983-00001730@192.168.1.2
CSeq: 13369951 REGISTER
Expires: 3600
Max-Forwards: 70
Contact: <sip:77880002@192.168.1.2:5068>;user=phone
User-Agent: LanScape VOIP Media Engine/6.0.0.3 (www.LanScapeCorp.com)
x-MyCustomHeader: "This is a modified transmitted SIP message."
x-VOIP-SDK: LanScape VOIP Media Engine/6.0.0.3 (www.LanScapeCorp.com)
Content-Length: 0
<<<< RxRxRxRxRx (#1, [17:19:50.765] 0 Ms, From: 203.167.54.180:6060) <<<<
SIP/2.0 401 Unauthorized
From: <sip:77880002@sw2.sp.ring-fone.com:6060>;tag=1cbb779
To: <sip:77880002@sw2.sp.ring-fone.com:6060>;tag=b436a7cb-17ac-49498df8-327a879-1f520de1
Call-ID: 12c4b207-306e-45d1-8e9f-832132e20983-00001730@192.168.1.2
CSeq: 13369951 REGISTER
Contact: <sip:77880002@192.168.1.2:5068>;user=phone
Expires: 60
WWW-Authenticate: Digest realm="Subcentrex", nonce="49498DF8",
stale=false, algorithm=MD5, qop="auth,auth-int"
Via: SIP/2.0/UDP 192.168.1.2:5068;received=75.73.107.63;rport=5068;branch=z9hG4bK01cc01a7
Supported: replaces,ACK,INFO,CANCEL,BYE,OPTIONS,REFER,SUBSCRIBE,NOTIFY
Content-Length: 0
>>>> TxTxTxTxTx (#2, [17:19:50.765] 297 Ms, To: 203.167.54.180:6060) >>>>
REGISTER sip:sw2.sp.ring-fone.com:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5068;rport;branch=z9hG4bK01cbf829
From: <sip:77880002@sw2.sp.ring-fone.com:6060>;tag=1cbb779
To: <sip:77880002@sw2.sp.ring-fone.com:6060>
Call-ID: 12c4b207-306e-45d1-8e9f-832132e20983-00001730@192.168.1.2
CSeq: 13369952 REGISTER
Authorization: Digest algorithm=md5,cnonce="00cba9ba",nc=00000001,
nonce="49498DF8",qop=auth,realm="Subcentrex",
response="32e870af67105eea85108b8cd007dbeb",
uri="sip:sw2.sp.ring-fone.com:6060",username="77880002"
Expires: 3600
Max-Forwards: 70
Contact: <sip:77880002@192.168.1.2:5068>;user=phone
User-Agent: LanScape VOIP Media Engine/6.0.0.3 (www.LanScapeCorp.com)
x-MyCustomHeader: "This is a modified transmitted SIP message."
x-VOIP-SDK: LanScape VOIP Media Engine/6.0.0.3 (www.LanScapeCorp.com)
Content-Length: 0
<<<< RxRxRxRxRx (#2, [17:19:51.031] 266 Ms, From: 203.167.54.180:6060) <<<<
SIP/2.0 200 OK
From: <sip:77880002@sw2.sp.ring-fone.com:6060>;tag=1cbb779
To: <sip:77880002@sw2.sp.ring-fone.com:6060>;tag=b436a7cb-17ac-49498df8-327a984-533f21c7
Call-ID: 12c4b207-306e-45d1-8e9f-832132e20983-00001730@192.168.1.2
CSeq: 13369952 REGISTER
Contact: <sip:77880002@192.168.1.2:5068>;user=phone
Expires: 60
Via: SIP/2.0/UDP 192.168.1.2:5068;received=75.73.107.63;rport=5068;branch=z9hG4bK01cbf829
Supported: replaces,ACK,INFO,CANCEL,BYE,OPTIONS,REFER,SUBSCRIBE,NOTIFY
Content-Length: 0
>>>> TxTxTxTxTx (#3, [17:20:09.828] 19063 Ms, To: 203.167.54.180:6060) >>>>
INVITE sip:1008@sw2.sp.ring-fone.com:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5068;rport;branch=z9hG4bK01cc5a4e
From: "LanScape Phone 77880002" <sip:77880002@sw2.sp.ring-fone.com:6060>;tag=1cc1050
To: <sip:1008@sw2.sp.ring-fone.com:6060>
Contact: <sip:77880002@75.73.107.63:5068>
Call-ID: ab826382-a8c6-4447-8f30-3fe3f9bd4db2-00001730@75.73.107.63
CSeq: 13378317 INVITE
Max-Forwards: 70
Organization: BA323A86-E294-4D2F-8991-49AA949E753C
x-MyCustomHeader: "This is a modified transmitted SIP message."
x-VOIP-SDK: LanScape VOIP Media Engine/6.0.0.3 (www.LanScapeCorp.com)
Content-Length: 250
User-Agent: LanScape VOIP Media Engine/6.0.0.3 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
v=0
o=77880002 30140796 30140796 IN IP4 75.73.107.63
s=LanScape
c=IN IP4 75.73.107.63
t=0 0
m=audio 8594 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=fmtp:18 annexb=no
a=ptime:20
a=fmtp:101 0-16
<<<< RxRxRxRxRx (#3, [17:20:10.000] 18969 Ms, From: 203.167.54.180:6060) <<<<
SIP/2.0 100 Trying
From: "LanScape Phone 77880002"<sip:77880002@sw2.sp.ring-fone.com:6060>;tag=1cc1050
To: <sip:1008@sw2.sp.ring-fone.com:6060>;tag=b436a7cb-17ac-49498e0b-327f3a5-56467678
Call-ID: ab826382-a8c6-4447-8f30-3fe3f9bd4db2-00001730@75.73.107.63
CSeq: 13378317 INVITE
Via: SIP/2.0/UDP 192.168.1.2:5068;received=75.73.107.63;rport=5068;branch=z9hG4bK01cc5a4e
Supported: replaces,ACK,INFO,CANCEL,BYE,OPTIONS,REFER,SUBSCRIBE,NOTIFY
Contact: <sip:1008@sw2.sp.ring-fone.com:6060>
Content-Length: 0
<<<< RxRxRxRxRx (#4, [17:20:10.203] 203 Ms, From: 203.167.54.180:6060) <<<<
SIP/2.0 200 OK
From: "LanScape Phone 77880002"<sip:77880002@sw2.sp.ring-fone.com:6060>;tag=1cc1050
To: <sip:1008@sw2.sp.ring-fone.com:6060>;tag=b436a7cb-17ac-49498e0b-327f3a5-56467678
Call-ID: ab826382-a8c6-4447-8f30-3fe3f9bd4db2-00001730@75.73.107.63
CSeq: 13378317 INVITE
Content-Type: application/sdp
Supported: replaces,ACK,INFO,CANCEL,BYE,OPTIONS,REFER,SUBSCRIBE,NOTIFY
Via: SIP/2.0/UDP 192.168.1.2:5068;received=75.73.107.63;rport=5068;branch=z9hG4bK01cc5a4e
Contact: <sip:1008@203.167.54.180:6060>
Content-Length: 230
v=0
o=1008 1229557260 1229557260 IN IP4 203.167.54.180
s=session-name
c=IN IP4 203.167.54.180
t=0 0
m=audio 31062 RTP/AVP 18 101
c=IN IP4 203.167.54.180
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
>>>> TxTxTxTxTx (#4, [17:20:10.203] 375 Ms, To: 203.167.54.180:6060) >>>>
ACK sip:1008@sw2.sp.ring-fone.com:6060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5068;received=75.73.107.63;rport=5068;branch=z9hG4bK01cc5a4e
From: "LanScape Phone 77880002" <sip:77880002@sw2.sp.ring-fone.com:6060>;tag=1cc1050
To: <sip:1008@sw2.sp.ring-fone.com:6060>;tag=b436a7cb-17ac-49498e0b-327f3a5-56467678
Call-ID: ab826382-a8c6-4447-8f30-3fe3f9bd4db2-00001730@75.73.107.63
CSeq: 13378317 ACK
Max-Forwards: 70
Route: <sip:1008@203.167.54.180:6060>
User-Agent: LanScape VOIP Media Engine/6.0.0.3 (www.LanScapeCorp.com)
x-MyCustomHeader: "This is a modified transmitted SIP message."
x-VOIP-SDK: LanScape VOIP Media Engine/6.0.0.3 (www.LanScapeCorp.com)
Content-Length: 0
>>>> TxTxTxTxTx (#5, [17:20:37.031] 26828 Ms, To: 203.167.54.180:6060) >>>>
BYE sip:1008@sw2.sp.ring-fone.com:6060 SIP/2.0
Via: SIP/2.0/UDP 75.73.107.63:5068;rport;branch=z9hG4bK01cc899c
From: "LanScape Phone 77880002" <sip:77880002@sw2.sp.ring-fone.com:6060>;tag=1cc1050
To: <sip:1008@sw2.sp.ring-fone.com:6060>;tag=b436a7cb-17ac-49498e0b-327f3a5-56467678
Call-ID: ab826382-a8c6-4447-8f30-3fe3f9bd4db2-00001730@75.73.107.63
CSeq: 13378318 BYE
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/6.0.0.3 (www.LanScapeCorp.com)
x-MyCustomHeader: "This is a modified transmitted SIP message."
x-VOIP-SDK: LanScape VOIP Media Engine/6.0.0.3 (www.LanScapeCorp.com)
Content-Length: 0
<<<< RxRxRxRxRx (#5, [17:20:37.218] 27015 Ms, From: 203.167.54.180:6060) <<<<
SIP/2.0 200 OK
From: "LanScape Phone 77880002"<sip:77880002@sw2.sp.ring-fone.com:6060>;tag=1cc1050
To: <sip:1008@sw2.sp.ring-fone.com:6060>;tag=b436a7cb-17ac-49498e0b-327f3a5-56467678
Call-ID: ab826382-a8c6-4447-8f30-3fe3f9bd4db2-00001730@75.73.107.63
CSeq: 13378318 BYE
Via: SIP/2.0/UDP 75.73.107.63:5068;rport=5068;branch=z9hG4bK01cc899c
Supported: replaces,ACK,INFO,CANCEL,BYE,OPTIONS,REFER,SUBSCRIBE,NOTIFY
Content-Length: 0
************* Log Closed (Dec 17 17:20:47) *************
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Thanks,
Randal
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speedvoip
Vetran
Joined: August 07 2008
Location: Canada
Posts: 156
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| Posted: December 18 2008 at 3:29am | IP Logged
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Hi Support:
Very pity I don't get same testing result as u with SingleLine Phone example. Frankly speaking, no any modification to existing example source code with v6.0.0.3 product image. So in order to clarify root cause of IVR access issue, can u mail me your used full example source code?
It is urgent issue which must be resolved and I am pushed very much.
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: December 18 2008 at 9:40am | IP Logged
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Hi George,
Please do not take offense, but these problems you are having are a bit ridiculous. Maybe your group should consider entering into a short term paid support agreement with us so that we can assist you with a higher level of support instead of this free support forum stuff. If your VOIP project is important to you and you are rushed for time, the free support we offer may not be adequate for you. If you obtain paid support from us, I will personally hold your hand to get you the results you need. If that’s what it takes - Not a problem.
It is rare that a customer struggles as much as you have struggled. It is not good for you and it is not good for us.
I just submitted this post to the forum in answer to another question you had:
Why My Normal license NOT Work?
http://www.lanscapecorp.com/forum/forum_posts.asp?TID=555&PN =1
...and all the issues you are reporting we cannot reproduce. Maybe we are having a language communication issue or something similar. Have you read any of the VOIP Media Engine documentation or at least looked at the Software Developer’s Reference on how to use the media engine? Hmm……..
Your product image and the sample source code for the v6.0.0.3 product you have and the “single line phone” native app is ok. Your problems lie elsewhere.
Thanks,
Randal
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LanScape Support Forum -> LanScape VOIP Media Engine™ - Technical Support
Topic: DTMF Failure for Accessing IVR Server
