[Asterisk-Users] Problems with chan_sip: random calls have no sound withouth any errors
Hi All, I have been busy with this problem for a while now, but I can't find any solution. First I thought this was a problem with the phones, but all my phones have this problem. (2 SNOM 200, 2 GS 102, 2 Cisco 7960). I tried all firmware versions I could find for the phones. First, my situation: - No NAT, No Firewall, same subnet - Codec configuration: In general: disallow=all disallow=g723.1 disallow=g729 disallow=gsm allow=ulaw allow=alaw In the phones: disallow=all disallow=g723.1 disallow=g729 disallow=gsm allow=ulaw allow=alaw But I also tried other codec configs. (allow=gsm, etc). Same problem. I'm testing from the Cisco 7960, as this phone seems to work best. I could also test from another phone with the same results. The S is for Success (can talk), the F is for Failure(Call gets setup but no speech/sound). Cisco 7960 to SNOM S,S,F,S,F,F,F,S,S,S,S,S,F Cisco 7960 to GS S,F,S,S,F,S,S,F,S,F, I placed a sip debug from asterisk for each situation at the following URL: http://audix.noc.ams-ix.net/asterisk/dumps/ - cisco_to_gs_failure.txt - cisco_to_gs_success.txt - cisco_to_snom_success.txt - cisco_to_snom_failure.txt Somebody have a clue? I'm thinking of filing a bug but I want to make sure this is no configuration or other problem at my side. Thanks and kind regards, Geert Nijpels ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with chan_sip: random calls have no sound withouth any errors
Asterisk is still saying it accepts G729. That is prolly the problem. Try updating to the latest CVS of Asterisk or to Asterisk version 0.7.1. If there any reason you are allowing both ulaw AND alaw. On Tue, 2004-02-03 at 08:48, Geert Nijpels wrote: Hi All, I have been busy with this problem for a while now, but I can't find any solution. First I thought this was a problem with the phones, but all my phones have this problem. (2 SNOM 200, 2 GS 102, 2 Cisco 7960). I tried all firmware versions I could find for the phones. First, my situation: - No NAT, No Firewall, same subnet - Codec configuration: In general: disallow=all disallow=g723.1 disallow=g729 disallow=gsm allow=ulaw allow=alaw In the phones: disallow=all disallow=g723.1 disallow=g729 disallow=gsm allow=ulaw allow=alaw But I also tried other codec configs. (allow=gsm, etc). Same problem. I'm testing from the Cisco 7960, as this phone seems to work best. I could also test from another phone with the same results. The S is for Success (can talk), the F is for Failure(Call gets setup but no speech/sound). Cisco 7960 to SNOM S,S,F,S,F,F,F,S,S,S,S,S,F Cisco 7960 to GS S,F,S,S,F,S,S,F,S,F, I placed a sip debug from asterisk for each situation at the following URL: http://audix.noc.ams-ix.net/asterisk/dumps/ - cisco_to_gs_failure.txt - cisco_to_gs_success.txt - cisco_to_snom_success.txt - cisco_to_snom_failure.txt Somebody have a clue? I'm thinking of filing a bug but I want to make sure this is no configuration or other problem at my side. Thanks and kind regards, Geert Nijpels ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the Asterisk Resource Pages. BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with chan_sip: random calls have no sound withouth any errors
Eric Wieling wrote: Asterisk is still saying it accepts G729. That is prolly the problem. Try updating to the latest CVS of Asterisk or to Asterisk version 0.7.1. If there any reason you are allowing both ulaw AND alaw. Sorry forgot to mention it. I'm already at latest CVS, but I have this problem also with 0.7.1. Well I use alaw and ulaw because all my phones support these codecs. But I get this problem with other codec configurations too. Kind regards, Geert On Tue, 2004-02-03 at 08:48, Geert Nijpels wrote: Hi All, I have been busy with this problem for a while now, but I can't find any solution. First I thought this was a problem with the phones, but all my phones have this problem. (2 SNOM 200, 2 GS 102, 2 Cisco 7960). I tried all firmware versions I could find for the phones. First, my situation: - No NAT, No Firewall, same subnet - Codec configuration: In general: disallow=all disallow=g723.1 disallow=g729 disallow=gsm allow=ulaw allow=alaw In the phones: disallow=all disallow=g723.1 disallow=g729 disallow=gsm allow=ulaw allow=alaw But I also tried other codec configs. (allow=gsm, etc). Same problem. I'm testing from the Cisco 7960, as this phone seems to work best. I could also test from another phone with the same results. The S is for Success (can talk), the F is for Failure(Call gets setup but no speech/sound). Cisco 7960 to SNOM S,S,F,S,F,F,F,S,S,S,S,S,F Cisco 7960 to GS S,F,S,S,F,S,S,F,S,F, I placed a sip debug from asterisk for each situation at the following URL: http://audix.noc.ams-ix.net/asterisk/dumps/ - cisco_to_gs_failure.txt - cisco_to_gs_success.txt - cisco_to_snom_success.txt - cisco_to_snom_failure.txt Somebody have a clue? I'm thinking of filing a bug but I want to make sure this is no configuration or other problem at my side. Thanks and kind regards, Geert Nijpels ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
