Folks,
Sorry to bother you all with such a silly question, but I am facing some issues with ringback tones. I am using SJphones with SER. They work perfectly well, I can make calls to other extensions and talk. I am maintaining a list of valid users in a database table, and have a small module which checks the authennticity of the dialed extension when SER receives INVITEs, and if it is valid, the call is t_relayed to the called extension. My problem is when an extension is dialed, I cannot hear the ringing tone at the callers end. I searched on the archives, but could not find mails addressing such a problem. Obviously I am missing something here. Can anyone help?
Thanks for your time...
Regards, Girish
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The ringback tone is generated locally by the callers user agent - typically as soon as the 180 ringing is received.
If a 183 Early media is received, the callers UA will use the incoming RTP stream as ringback tone.
klaus
GR S wrote:
Folks,
Sorry to bother you all with such a silly question, but I am facing some issues with ringback tones. I am using SJphones with SER. They work perfectly well, I can make calls to other extensions and talk. I am maintaining a list of valid users in a database table, and have a small module which checks the authennticity of the dialed extension when SER receives INVITEs, and if it is valid, the call is t_relayed to the called extension. My problem is when an extension is dialed, I cannot hear the ringing tone at the callers end. I searched on the archives, but could not find mails addressing such a problem. Obviously I am missing something here. Can anyone help?
Thanks for your time...
Regards, Girish
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Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Thanks to all.. my setup for domain works!
Here is my setup ser.cfg
Thanks All.
I have an issue with ringback :
For specific users (AT&T cell phones) When they call something on my network, the person on the AT&T cell phone does not hear a ring. The call looks like:
AT&TCELL <-> PSTN <-> CISCOSIP <-> SER <-> UAC
But, any other cell phone, and any other phone for that matter, hears a ring when calling.
---greg
Greg Fausak www.AddaBrand.com (US) 469-546-1265 On May 4, 2004, at 2:58 PM, Klaus Darilion wrote:
The ringback tone is generated locally by the callers user agent - typically as soon as the 180 ringing is received.
If a 183 Early media is received, the callers UA will use the incoming RTP stream as ringback tone.
klaus
GR S wrote:
Folks, Sorry to bother you all with such a silly question, but I am facing some issues with ringback tones. I am using SJphones with SER. They work perfectly well, I can make calls to other extensions and talk. I am maintaining a list of valid users in a database table, and have a small module which checks the authennticity of the dialed extension when SER receives INVITEs, and if it is valid, the call is t_relayed to the called extension. My problem is when an extension is dialed, I cannot hear the ringing tone at the callers end. I searched on the archives, but could not find mails addressing such a problem. Obviously I am missing something here. Can anyone help? Thanks for your time... Regards, Girish
__________________________________ Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs http://hotjobs.sweepstakes.yahoo.com/careermakeover _______________________________________________ Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
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Hello,
--- Klaus Darilion klaus.mailinglists@pernau.at wrote:
The ringback tone is generated locally by the callers user agent - typically as soon as the 180 ringing is received.
If a 183 Early media is received, the callers UA will use the incoming RTP stream as ringback tone.
klaus
Thanks for the reply. The user agent is getting the 180 ringing, but it does not generate the ringing tone. I have tested it with SJphone and another softphone which is built using Windows and RTC, but both behave the same way. I have captured the SIP trafic using ethreal and did not find anything weird. However, i am curious about one of the lines in that trace:
Warning: 392 192.168.68.20:5060 "Noisy feedback tells: pid=23603 req_src_ip=192.168.68.16 req_src_port=5060 in_uri=sip:9000@192.168.68.20 out_uri=sip:shiva-smarttest-com@192.168.68.20 via_cnt==1"
I am attaching the relevent parts of the ethreal trace with this.
Thoughts?
Regards, Girish
INVITE sip:9000@192.168.68.20 SIP/2.0 Content-Length: 116 Contact: sip:192.168.68.16 Call-ID: 4E7D7393-F25F-453B-A136-B26F0FCEB7AC@192.168.68.16 Content-Type: application/sdp Max-Forwards: 70 From: " "sip:krishna-smarttest-com@192.168.68.20;tag=7678651 CSeq: 1 INVITE To: sip:9000@192.168.68.20 Via: SIP/2.0/UDP 192.168.68.16:5060 User-Agent: SJLabs-SJPhone/1.0
SIP/2.0 100 trying -- your call is important to us Call-ID: 4E7D7393-F25F-453B-A136-B26F0FCEB7AC@192.168.68.16 From: " "sip:krishna-smarttest-com@192.168.68.20;tag=7678651 CSeq: 1 INVITE To: sip:9000@192.168.68.20 Via: SIP/2.0/UDP 192.168.68.16:5060 Server: Sip EXpress router (0.8.13-dev-27-unixsock (i386/linux)) Content-Length: 0 Warning: 392 192.168.68.20:5060 "Noisy feedback tells: pid=23603 req_src_ip=192.168.68.16 req_src_port=5060 in_uri=sip:9000@192.168.68.20 out_uri=sip:shiva-smarttest-com@192.168.68.20 via_cnt==1"
INVITE sip:192.168.68.5 SIP/2.0 Record-Route: sip:192.168.68.20;ftag=7678651;lr Record-Route: sip:192.168.68.20;ftag=7678651;lr Content-Length: 116 Contact: sip:192.168.68.16 Call-ID: 4E7D7393-F25F-453B-A136-B26F0FCEB7AC@192.168.68.16 Content-Type: application/sdp Max-Forwards: 68 From: " "sip:krishna-smarttest-com@192.168.68.20;tag=7678651 CSeq: 1 INVITE To: sip:9000@192.168.68.20 Via: SIP/2.0/UDP 192.168.68.20;branch=z9hG4bK30f5.b7e81e43.0 Via: SIP/2.0/UDP 192.168.68.20;branch=z9hG4bK30f5.a7e81e43.0 Via: SIP/2.0/UDP 192.168.68.16:5060 User-Agent: SJLabs-SJPhone/1.0
SIP/2.0 100 Trying Content-Length: 0 Contact: sip:192.168.68.5:5060 Call-ID: 4E7D7393-F25F-453B-A136-B26F0FCEB7AC@192.168.68.16 CSeq: 1 INVITE From: sip:krishna-smarttest-com@192.168.68.20;tag=7678651 Record-Route: sip:192.168.68.20;ftag=7678651;lr,sip:192.168.68.20;ftag=7678651;lr To: "Suresh"sip:9000@192.168.68.20;tag=8359859 Server: SJLabs-SJPhone/1.0 Via: SIP/2.0/UDP 192.168.68.20:5060;branch=z9hG4bK30f5.b7e81e43.0,SIP/2.0/UDP 192.168.68.20:5060;branch=z9hG4bK30f5.a7e81e43.0,SIP/2.0/UDP 192.168.68.16:5060
SIP/2.0 180 Ringing Content-Length: 0 Contact: sip:192.168.68.5:5060 Call-ID: 4E7D7393-F25F-453B-A136-B26F0FCEB7AC@192.168.68.16 CSeq: 1 INVITE From: sip:krishna-smarttest-com@192.168.68.20;tag=7678651 Record-Route: sip:192.168.68.20;ftag=7678651;lr,sip:192.168.68.20;ftag=7678651;lr To: "Suresh"sip:9000@192.168.68.20;tag=8359859 Server: SJLabs-SJPhone/1.0 Via: SIP/2.0/UDP 192.168.68.20:5060;branch=z9hG4bK30f5.b7e81e43.0,SIP/2.0/UDP 192.168.68.20:5060;branch=z9hG4bK30f5.a7e81e43.0,SIP/2.0/UDP 192.168.68.16:5060
SIP/2.0 180 Ringing Content-Length: 0 Contact: sip:192.168.68.5:5060 Call-ID: 4E7D7393-F25F-453B-A136-B26F0FCEB7AC@192.168.68.16 CSeq: 1 INVITE From: sip:krishna-smarttest-com@192.168.68.20;tag=7678651 Record-Route: sip:192.168.68.20;ftag=7678651;lr,sip:192.168.68.20;ftag=7678651;lr To: "Suresh"sip:9000@192.168.68.20;tag=8359859 Server: SJLabs-SJPhone/1.0 Via: SIP/2.0/UDP 192.168.68.16:5060
SIP/2.0 200 OK
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Windows Messenger does not create a ringback tone. I don't know the behaviour of SJphone as I never used it. But I can recommend x-lite which generates local ringback.
The warning header is nothing to worry about - it's just for debugging purposes (IMO very useful). You can turn it off if you like (http://iptel.org/ser/doc/seruser/seruser.html#AEN709)
klaus
GR S wrote:
Hello,
--- Klaus Darilion klaus.mailinglists@pernau.at wrote:
The ringback tone is generated locally by the callers user agent - typically as soon as the 180 ringing is received.
If a 183 Early media is received, the callers UA will use the incoming RTP stream as ringback tone.
klaus
Thanks for the reply. The user agent is getting the 180 ringing, but it does not generate the ringing tone. I have tested it with SJphone and another softphone which is built using Windows and RTC, but both behave the same way. I have captured the SIP trafic using ethreal and did not find anything weird. However, i am curious about one of the lines in that trace:
Warning: 392 192.168.68.20:5060 "Noisy feedback tells: pid=23603 req_src_ip=192.168.68.16 req_src_port=5060 in_uri=sip:9000@192.168.68.20 out_uri=sip:shiva-smarttest-com@192.168.68.20 via_cnt==1"
I am attaching the relevent parts of the ethreal trace with this.
Thoughts?
Regards, Girish
INVITE sip:9000@192.168.68.20 SIP/2.0 Content-Length: 116 Contact: sip:192.168.68.16 Call-ID: 4E7D7393-F25F-453B-A136-B26F0FCEB7AC@192.168.68.16 Content-Type: application/sdp Max-Forwards: 70 From: " "sip:krishna-smarttest-com@192.168.68.20;tag=7678651 CSeq: 1 INVITE To: sip:9000@192.168.68.20 Via: SIP/2.0/UDP 192.168.68.16:5060 User-Agent: SJLabs-SJPhone/1.0
SIP/2.0 100 trying -- your call is important to us Call-ID: 4E7D7393-F25F-453B-A136-B26F0FCEB7AC@192.168.68.16 From: " "sip:krishna-smarttest-com@192.168.68.20;tag=7678651 CSeq: 1 INVITE To: sip:9000@192.168.68.20 Via: SIP/2.0/UDP 192.168.68.16:5060 Server: Sip EXpress router (0.8.13-dev-27-unixsock (i386/linux)) Content-Length: 0 Warning: 392 192.168.68.20:5060 "Noisy feedback tells: pid=23603 req_src_ip=192.168.68.16 req_src_port=5060 in_uri=sip:9000@192.168.68.20 out_uri=sip:shiva-smarttest-com@192.168.68.20 via_cnt==1"
INVITE sip:192.168.68.5 SIP/2.0 Record-Route: sip:192.168.68.20;ftag=7678651;lr Record-Route: sip:192.168.68.20;ftag=7678651;lr Content-Length: 116 Contact: sip:192.168.68.16 Call-ID: 4E7D7393-F25F-453B-A136-B26F0FCEB7AC@192.168.68.16 Content-Type: application/sdp Max-Forwards: 68 From: " "sip:krishna-smarttest-com@192.168.68.20;tag=7678651 CSeq: 1 INVITE To: sip:9000@192.168.68.20 Via: SIP/2.0/UDP 192.168.68.20;branch=z9hG4bK30f5.b7e81e43.0 Via: SIP/2.0/UDP 192.168.68.20;branch=z9hG4bK30f5.a7e81e43.0 Via: SIP/2.0/UDP 192.168.68.16:5060 User-Agent: SJLabs-SJPhone/1.0
SIP/2.0 100 Trying Content-Length: 0 Contact: sip:192.168.68.5:5060 Call-ID: 4E7D7393-F25F-453B-A136-B26F0FCEB7AC@192.168.68.16 CSeq: 1 INVITE From: sip:krishna-smarttest-com@192.168.68.20;tag=7678651 Record-Route: sip:192.168.68.20;ftag=7678651;lr,sip:192.168.68.20;ftag=7678651;lr To: "Suresh"sip:9000@192.168.68.20;tag=8359859 Server: SJLabs-SJPhone/1.0 Via: SIP/2.0/UDP 192.168.68.20:5060;branch=z9hG4bK30f5.b7e81e43.0,SIP/2.0/UDP 192.168.68.20:5060;branch=z9hG4bK30f5.a7e81e43.0,SIP/2.0/UDP 192.168.68.16:5060
SIP/2.0 180 Ringing Content-Length: 0 Contact: sip:192.168.68.5:5060 Call-ID: 4E7D7393-F25F-453B-A136-B26F0FCEB7AC@192.168.68.16 CSeq: 1 INVITE From: sip:krishna-smarttest-com@192.168.68.20;tag=7678651 Record-Route: sip:192.168.68.20;ftag=7678651;lr,sip:192.168.68.20;ftag=7678651;lr To: "Suresh"sip:9000@192.168.68.20;tag=8359859 Server: SJLabs-SJPhone/1.0 Via: SIP/2.0/UDP 192.168.68.20:5060;branch=z9hG4bK30f5.b7e81e43.0,SIP/2.0/UDP 192.168.68.20:5060;branch=z9hG4bK30f5.a7e81e43.0,SIP/2.0/UDP 192.168.68.16:5060
SIP/2.0 180 Ringing Content-Length: 0 Contact: sip:192.168.68.5:5060 Call-ID: 4E7D7393-F25F-453B-A136-B26F0FCEB7AC@192.168.68.16 CSeq: 1 INVITE From: sip:krishna-smarttest-com@192.168.68.20;tag=7678651 Record-Route: sip:192.168.68.20;ftag=7678651;lr,sip:192.168.68.20;ftag=7678651;lr To: "Suresh"sip:9000@192.168.68.20;tag=8359859 Server: SJLabs-SJPhone/1.0 Via: SIP/2.0/UDP 192.168.68.16:5060
SIP/2.0 200 OK
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Klaus,
I haven't tested this feature with x-lite, but this local ringback can be enabled in the other softphone and we need to test that. I haven't found any such settings in SJphone. In fact i had tested SJphones with Asterisk, and got the ringtone at the callers end. This must be because of the 'r' option set in the Asterisk Dial command and i expected the same with SER too.
Thanks for the comments...
Regards, Girish
--- Klaus Darilion klaus.mailinglists@pernau.at wrote:
Windows Messenger does not create a ringback tone. I don't know the behaviour of SJphone as I never used it. But I can recommend x-lite which generates local ringback.
The warning header is nothing to worry about - it's just for debugging purposes (IMO very useful). You can turn it off if you like (http://iptel.org/ser/doc/seruser/seruser.html#AEN709)
klaus
===== Regards, Girish
__________________________________ Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs http://hotjobs.sweepstakes.yahoo.com/careermakeover
Maybe asterisk generates the ringback. But the ringback can't be forced at the SIP proxy.
klaus
GR S wrote:
Klaus,
I haven't tested this feature with x-lite, but this local ringback can be enabled in the other softphone and we need to test that. I haven't found any such settings in SJphone. In fact i had tested SJphones with Asterisk, and got the ringtone at the callers end. This must be because of the 'r' option set in the Asterisk Dial command and i expected the same with SER too.
Thanks for the comments...
Regards, Girish
--- Klaus Darilion klaus.mailinglists@pernau.at wrote:
Windows Messenger does not create a ringback tone. I don't know the behaviour of SJphone as I never used it. But I can recommend x-lite which generates local ringback.
The warning header is nothing to worry about - it's just for debugging purposes (IMO very useful). You can turn it off if you like (http://iptel.org/ser/doc/seruser/seruser.html#AEN709)
klaus
===== Regards, Girish
__________________________________ Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs http://hotjobs.sweepstakes.yahoo.com/careermakeover