Race. Here are thre results of the tests:
Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x404 (ulaw|ilbc) Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing) Seems both can speak ulaw and ilbc but still, no prompt on the phone :( Any ideas? -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Race Vanderdecken Sent: Sábado, 19 de Febrero de 2005 07:15 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds Correct. -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Saturday, February 19, 2005 8:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds This is a very good place to start Race. So if I understand you correctly, Ill do the sip debug but maybe trying to force both to use ilbc or ulaw/alaw might help so I can listen to the prompts right? -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Race Vanderdecken Sent: Sábado, 19 de Febrero de 2005 06:55 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds Grasshopper, You have your first clue, the live test works. Do you understand how SIP works? During the INVITE sequence the Asterisk and the phone trade RTP CODEC information. RTP is the protocol that actually carries the sounds, SIP only does the handshaking for the call. A CODEC is what the RTP is carrying between the pones. If you do "sip debug" inside of the asterisk command line interface *CLI> sip debug Then you will see the SIP Messages and the Codec agreements. ... 16 headers, 13 lines Using latest request as basis request Sending to 192.168.1.102 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.102:10054 Found description format pcmu Found description format pcma Found description format gsm Found description format g729 Found description format telephone-event -- Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) .... ... ... m=audio 19958 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - Asterisk is the " us - 0x8000e (gsm|ulaw|alaw|h263)" Phone is the " peer - audio=0x10e (gsm|ulaw|alaw|g729)" Above is the trace of my SNOM 200 -- see the "combined - 0xe (gsm|ulaw|alaw)"? The phone can do g729, but asterisk can't, so asterisk and the phone agree on a non-g729 codec, ulaw. Eventually the phone agrees to "a=rtpmap:0 PCMU/8000", it is going to talk sound using ulaw at 8000hz. (again, I might be a little wrong on the extact details.) If the phones agree to use G729 then the playback won't work because you don't have a g729 license, $10 from Digium. Remember that asterisk is a third party to a conference and if your conference is using g729, then asterisk can't do that. In the sip.conf, Disallow=all Allow=gsm Allow=ulaw Allow=alaw This will force the phone and asterisk to speak gsm, ulaw or alaw. I had the same experience with no sound when I first connected a Cisco 7960, I could here other people, but not the prompts. Asterisk will allow G729 to pass through, but it will not allow G729 to originate and terminate without the license (I might be a little mistaken here...) I hope this helps. I have not use [EMAIL PROTECTED], it might be different. Let me know, Race "The Tyrant" Vanderdecken -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Saturday, February 19, 2005 7:07 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds Hi Race.. In this case, the asterisk|home comes preconfigured with some stuff different than the asterisk tar file. I check and the phone supports all mentioned codecs, I also made a test by using the phone and sjphone to do a live test directly, conversation was successful using gsm, ulaw and ilbc. Any other ideas? -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Race Vanderdecken Sent: Sábado, 19 de Febrero de 2005 05:43 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] No Sounds Okay, A couple of things could be happening so let's run through a list. Your questions are a little vague so I shall make my answer also vague. 1. Codec. Are you allowing for and does the "phone" support the codec that the sounds are in? (I.e. do you have a G.729 license for your Minkey?) 2. Does the phone support the G.711a/u law that I think the sounds come in? 3. Test first the voicemail playback demo and make sure you can here the sound there before jumping into a conference. Dial extension 500, do you hear anything? 4. Is your phone working? Have you made a call to a live person? Race -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Saturday, February 19, 2005 2:53 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] No Sounds I just installed [EMAIL PROTECTED] to see how it works and seems there is a problem with sounds... I dont hear any announcements or recordings... sounds are on /var/lib/asterisk/sounds and the logs show this: -- Created MeetMe conference 1023 for conference '8200' -- Playing 'conf-onlyperson' (language 'en') But I dont hear anything... any experiences with this kind of errors and/or [EMAIL PROTECTED] Thx! __________________________________________________________________ Anton Krall _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users