"Unable to create channel of type 'SIP'

[ghurty]

I am not sure what happened, but I started getting:
"Unable to create channel of type 'SIP' (cause 3 - No route to destination)"

errors, and no outgoing calls work. inbound does but not outgoing.

I tried this with different trunks.


Thanks

[2010-04-16 18:08:34] DEBUG[21106] app_macro.c: Executed application: Macro
[2010-04-16 18:08:34] VERBOSE[21106] logger.c: -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/102-097f20f8", "0?bypass|1") in new stack
[2010-04-16 18:08:34] DEBUG[21106] app_macro.c: Executed application: GotoIf
[2010-04-16 18:08:34] VERBOSE[21106] logger.c: -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/102-097f20f8", "0?customtrunk") in new stack
[2010-04-16 18:08:34] DEBUG[21106] app_macro.c: Executed application: GotoIf
[2010-04-16 18:08:34] VERBOSE[21106] logger.c: -- Executing [s@macro-dialout-trunk:19] Dial("SIP/102-097f20f8", "SIP/jivregen/8005551212|300|") in new stack
[2010-04-16 18:08:34] WARNING[21106] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
[2010-04-16 18:08:34] VERBOSE[21106] logger.c:   == Everyone is busy/congested at this time (1:0/0/1)
[2010-04-16 18:08:34] DEBUG[21106] app_macro.c: Executed application: Dial

 

[dswartz]

what does 'sip show registry' show you? (from an "asterisk -r" session.)

 

[blanchae]

Looks like a networking problem and possibly DNS. Calls can come in but can't get out. Check your DNS server.

 

[blanchae]

Are you using MOR? Advanced Softswitch and Billing? Here's a link that mentions that if this error happens trying to call through some provider - that means that device with same username exists. MOR 0.7+ and does not allow this, for lower versions change login value for provider or device so usernames do not match. It does give some solutions for

In iax2.conf set
rtignoreregexpire = yes

and in sip.conf set
ignoreregexpire = yes

Reload Asterisk