Hi everyone, Does anyone know the answer for this situation? I have Asterisk with E1 PRI links, with SIP phones registered to Asterisk and with h.323 connection to Cisco CallManager. I am using oh323. I think I have a problem with codecs but I do not know exactly what is wrong. this is working ok: -------------------------- Call from CallManager (7960) to SIP phone on Asterisk (X-Lite or 7960 with SIP image) - working OK Call from CallManager (7960) to E1 PRI trunk to PSTN through Asterisk - working OK Call from E1 PRI trunk from PSTN through Asterisk to CallManager (7960) - working OK here is the problem --------------------------- Call from SIP phone to CallManager - rings the phone, in the moment when called party picks the receiver Asterisk crashes with core dump Interesting is that if you establish a call in opposite direction (from CallManager to SIP phone) prior to that one, Asterisk wouldn't crash sometimes I will appreciate if anyone can help Tomica -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040214/921d37d6/attachment.htm
Like so many here I too still seek a small, affordable FXO gate way for 2-4 lines. I just stumbled upon the Teliann 210 online at www.teliann.com. Anyone here every use these? They seem very new as their online shop web site lists them as launched Dec 2003. They're h.323 and not SIP, but I suppose I could get around that as they're only $220 for the two port model. Michael Graves -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com "If the gods had meant us to vote they'd have given us candidates!" - Jim Hightower, Texan Populist ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704
Michael Manousos
2004-Feb-16 02:59 UTC
[Asterisk-Users] Asterisk - oh323 - Cisco CallManager
Did you try the latest version (0.5.9)? Michael. Tomica Crnek wrote:> Hi everyone, > > Does anyone know the answer for this situation? I have Asterisk with E1 > PRI links, with SIP phones registered to Asterisk and with h.323 > connection to Cisco CallManager. I am using oh323. I think I have a > problem with codecs but I do not know exactly what is wrong. > > this is working ok: > -------------------------- > Call from CallManager (7960) to SIP phone on Asterisk (X-Lite or > 7960 with SIP image) - working OK > Call from CallManager (7960) to E1 PRI trunk to PSTN through Asterisk - > working OK > Call from E1 PRI trunk from PSTN through Asterisk to CallManager (7960) > - working OK > > here is the problem > --------------------------- > Call from SIP phone to CallManager - rings the phone, in the moment when > called party picks the receiver Asterisk crashes with core dump > Interesting is that if you establish a call in opposite direction (from > CallManager to SIP phone) prior to that one, Asterisk wouldn't crash > sometimes > > I will appreciate if anyone can help > > Tomica >
I have just installed 0.5.9. Up to this moment it didn't crash :P -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Michael Manousos Sent: Monday, February 16, 2004 11:00 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk - oh323 - Cisco CallManager Did you try the latest version (0.5.9)? Michael. Tomica Crnek wrote:> Hi everyone, > > Does anyone know the answer for this situation? I have Asterisk with > E1 PRI links, with SIP phones registered to Asterisk and with h.323 > connection to Cisco CallManager. I am using oh323. I think I have a > problem with codecs but I do not know exactly what is wrong. > > this is working ok: > -------------------------- > Call from CallManager (7960) to SIP phone on Asterisk (X-Lite or 7960 > with SIP image) - working OK Call from CallManager (7960) to E1 PRI > trunk to PSTN through Asterisk - working OK Call from E1 PRI trunk > from PSTN through Asterisk to CallManager (7960) > - working OK > > here is the problem > --------------------------- > Call from SIP phone to CallManager - rings the phone, in the moment > when called party picks the receiver Asterisk crashes with core dump > Interesting is that if you establish a call in opposite direction > (from CallManager to SIP phone) prior to that one, Asterisk wouldn't > crash sometimes > > I will appreciate if anyone can help > > Tomica >_______________________________________________ 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