[Asterisk-bsd] sip & iax2 not agreeing on the numbers of channels

Hendrik Visage hvjunk at gmail.com
Tue Jun 19 01:34:27 CDT 2007


Hi there,

Outgoing call centre:
Sip-phones, => pbx1 =IAX2(trunk(s)=>vrouter=>Sip/Voip/provider

I'm experiencing asterisk crashes (1.4.3 on FreeBSD 6.2 + G729 codecs
elsewhere downloaded), but the strange part is that I see this on the
vrouter:

192.168.123.142  0720653046  3d9ef7915f5  00102/558531892  g729  No
   Rx: ACK
192.168.123.142  0720653046  4fc72983748  00102/558564206  g729  No
   Rx: ACK
192.168.123.142  0725285584  0daf511d483  00102/00000  unkn  No
Init: INVITE
192.168.123.142  0725285584  2f71adff548  00102/00000  unkn  No
Init: INVITE
192.168.123.142  0725285584  350f2426641  00102/00000  unkn  No
Init: INVITE
192.168.123.142  0725285584  6a5e17da29c  00102/00000  unkn  No
Init: INVITE
192.168.123.142  0725285584  6b85b3847f6  00102/00000  unkn  No
Init: INVITE

Thus according to the "sip show channels" I have ~43 calls, but
according to "iax2 show channels" I only see 14 calls.

Is there something I'm doing wrong in the dialplan? (I just set the
call recording on and Dial(Sip/Provider)) Or is this "expected"??


-- 
Hendrik Visage



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