Auto fallthrough, channel 'SIP/101-00000003' status is 'UNKN

Asterisk Asterisk Support
1
  • Executing [200@default:1] NoOp(“SIP/101-00000003”, ““101” <9718485393>”) in new stack
    – Executing [200@default:2] AGI(“SIP/101-00000003”, “agi-sayani.agi”) in new stack
    – Launched AGI Script /var/lib/asterisk/agi-bin/agi-sayani.agi
    AGI Tx >> agi_request: agi-sayani.agi
    AGI Tx >> agi_channel: SIP/101-00000003
    AGI Tx >> agi_language: en
    AGI Tx >> agi_type: SIP
    AGI Tx >> agi_uniqueid: 1380690988.148
    AGI Tx >> agi_version: 11.5.1
    AGI Tx >> agi_callerid: 9718485393
    AGI Tx >> agi_calleridname: 101
    AGI Tx >> agi_callingpres: 0
    AGI Tx >> agi_callingani2: 0
    AGI Tx >> agi_callington: 0
    AGI Tx >> agi_callingtns: 0
    AGI Tx >> agi_dnid: 200
    AGI Tx >> agi_rdnis: unknown
    AGI Tx >> agi_context: default
    AGI Tx >> agi_extension: 200
    AGI Tx >> agi_priority: 2
    AGI Tx >> agi_enhanced: 0.0
    AGI Tx >> agi_accountcode:
    AGI Tx >> agi_threadid: 140559224530688
    AGI Tx >>
    – <SIP/101-00000003>AGI Script agi-sayani.agi completed, returning 0
    – Auto fallthrough, channel ‘SIP/101-00000003’ status is ‘UNKNOWN’

please help me…how to create a proper sip.conf…my system has dhcp

2

On the basis of the information that you have provided, the answer is read and follow the documentation, which can be found at asteriskdocs.org/ and in the configs/sip.conf.sample file that should have come with your Asterisk code.

3

hai david55,

still am facing this issue…

== Using SIP RTP CoS mark 5
– Executing [100@default:1] NoOp(“SIP/2001-00000008”, ““2001” <2001>”) in new stack
– Executing [100@default:2] Answer(“SIP/2001-00000008”, “”) in new stack
> 0x7fbb5c0306f0 – Probation passed - setting RTP source address to 192.168.1.22:40020
– Executing [100@default:3] AGI(“SIP/2001-00000008”, “agi-sayani2.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/agi-sayani2.agi
[Oct 2 17:45:40] NOTICE[3685][C-00000008]: channel.c:4257 __ast_read: Dropping incompatible voice frame on SIP/2001-00000008 of format ulaw since our native format has changed to (gsm|h263)
– <SIP/2001-00000008>AGI Script agi-sayani2.agi completed, returning 0
– Auto fallthrough, channel ‘SIP/2001-00000008’ status is ‘UNKNOWN’

Please help me

4

You have a broken SIP peer as it is using a codec that was not agreed in the initial codec negotiation.

You could try telling Asterisk to allow alaw, but it still looks as though the peer is broken.

See wiki.asterisk.org/wiki/display/ … nformation for more information on the detail you would need to collect.

5

dear david55,

still the problem exist.

Can u give detailed information to resolve this…issue…earler this issue was not there

6

Replace the SIP peer with one that works.