We're having a horrid problem with our asterisk setup. Sometimes calls just go dead - we can't hear what the other end is saying. (I think they can't hear us either). The call doesn't hang up until one of the callers gets bored. Internaly we use Thomson ST2030 SIP phones. Externaly we have 3 ISDN BRI lines (6 channels total), connected to an Eicon Diver server card (4BRI). We're using Asterisk 1.4.11 (1.4.11-BRIstuffed-0.4.0-test4) on a Debian system, with chan-capi 1.0.1. Any idea what could be going wrong, where to look &c?
Hi John! I have the same problem, the system contains 1 port Billion ISDN BRI card, and 1 sip trunk. This is a trixbox with Asterisk 1.2.22-BRIstuffed-0.3.0-PRE-1y-i The ISDN call is forwarded to a ring-group. The 6 sip phones are welltech lp399 series. If incoming the call get wrong, we can not hear the other side, but they hear us. In my case the rtp debug shows there are no incoming rtp packets from asterisk to SIP phone. If somebody experienced this problem, please help US! Thanks! 2007/9/20, John Hughes <john at calva.com>:> We're having a horrid problem with our asterisk setup. > > Sometimes calls just go dead - we can't hear what the other end is > saying. (I think they can't hear us either). The call doesn't hang up > until one of the callers gets bored. > > Internaly we use Thomson ST2030 SIP phones. > > Externaly we have 3 ISDN BRI lines (6 channels total), connected to an > Eicon Diver server card (4BRI). > > We're using Asterisk 1.4.11 (1.4.11-BRIstuffed-0.4.0-test4) on a Debian > system, with chan-capi 1.0.1. > > Any idea what could be going wrong, where to look &c? > > > _______________________________________________ > > Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- _________________ T?th P?ter Tel.: +36703834578 _________________
Tzafrir Cohen
2007-Sep-20 11:33 UTC
[asterisk-users] Horrible problem - calls losing sound
On Thu, Sep 20, 2007 at 01:24:52PM +0200, P?ter T?th wrote:> Hi John! > > I have the same problem, the system contains 1 port Billion ISDN BRI > card, and 1 sip trunk. This is a trixbox with Asterisk > 1.2.22-BRIstuffed-0.3.0-PRE-1y-i > > The ISDN call is forwarded to a ring-group. The 6 sip phones are > welltech lp399 series. > > If incoming the call get wrong, we can not hear the other side, but > they hear us. In my case the rtp debug shows there are no incoming rtp > packets from asterisk to SIP phone. > > If somebody experienced this problem, please help US!Just as you have rtp debug, you have bri debug . bri debug span 1 and hope for a friendly ISDN guru on the list... -- Tzafrir Cohen icq#16849755 jabber:tzafrir at jabber.org +972-50-7952406 mailto:tzafrir.cohen at xorcom.com http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
On Thu, 2007-09-20 at 12:49 +0200, John Hughes wrote:> We're having a horrid problem with our asterisk setup. > > Sometimes calls just go dead - we can't hear what the other end is > saying. (I think they can't hear us either). The call doesn't hang up > until one of the callers gets bored. > > Internaly we use Thomson ST2030 SIP phones. > > Externaly we have 3 ISDN BRI lines (6 channels total), connected to an > Eicon Diver server card (4BRI). > > We're using Asterisk 1.4.11 (1.4.11-BRIstuffed-0.4.0-test4) on a Debian > system, with chan-capi 1.0.1. > > Any idea what could be going wrong, where to look &c?I'm don't know what is causing your problem but afaik you don't need bristuff with your Eicon Diva Server card. I have several asterisk 1.2.24 boxes with 1x BRI and 4x BRI Eicon Diva Server cards and I only use asterisk + chan_capi. This setup is very stable and works great. Perhaps you could try rebuilding asterisk without the bristuff patch then rebuild and install chan_capi and see if the problem goes away? Regards, Patrick
Tzafrir Cohen wrote:> On Thu, Sep 20, 2007 at 01:24:52PM +0200, P?ter T?th wrote: > > Just as you have rtp debug, you have bri debug . > > bri debug span 1 > > and hope for a friendly ISDN guru on the list... >Does nothing for me - my isdn is connected via chan-capi.
Tzafrir Cohen wrote:> On Thu, Sep 20, 2007 at 01:24:52PM +0200, P?ter T?th wrote: > >> Hi John! >> >> I have the same problem, the system contains 1 port Billion ISDN BRI >> card, and 1 sip trunk. This is a trixbox with Asterisk >> 1.2.22-BRIstuffed-0.3.0-PRE-1y-i >> >> The ISDN call is forwarded to a ring-group. The 6 sip phones are >> welltech lp399 series. >> >> If incoming the call get wrong, we can not hear the other side, but >> they hear us. In my case the rtp debug shows there are no incoming rtp >> packets from asterisk to SIP phone. >> >> If somebody experienced this problem, please help US! >> > Just as you have rtp debug, you have bri debug . > > bri debug span 1 > > and hope for a friendly ISDN guru on the list... >Ah, I can get the same (excessive!) info by set verbose 8 capi debug