[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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