[asterisk-users] Call doesn't disconnect in SIP

velusamy velu velu.technical at gmail.com
Mon Feb 8 00:19:38 CST 2010


Dear All,
   I am using asterisk 1.4.21.2. I have used Originate manager application
to to call the two persons. I have called AGI application to call another
person. There I have used GET FULL VARIABLE AGI command to get the value. I
am able to call another person form AGI. But when one end cut the call
another one didn't disconnected.

 The following errors are displayed in Asterisk console,

[Feb  8 11:12:16] ERROR[4115]: chan_sip.c:15553 sipsock_read: We could NOT
get the channel lock for SIP/700-081da948!
[Feb  8 11:12:16] ERROR[4115]: chan_sip.c:15554 sipsock_read: SIP
transaction failed: 458b7e8f26d86ee10ec99cba526059b3 at 192.168.1.44

 Could you please help me to solve this problem?

Thanks.
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