Rejected Because Extension Not Found In Context 'introutingB'


Dear All, I am trying to recieve call from inbound proxy then route to internal peer (localhost) and then route to outgoing sip proxy but it failing with subject error. Can any one please correct me what i am doing wrong in below config.


[Inbound] type=peer context=introuting host4.107.XXX.XXX disallow=all allow=all

[astinside] type=peer context=introutingB host=localhost disallow=all ;allow=speex

[voipprovider] type=peer host=XXX.X.XXX.159 disallow=all allow=g723:120


[introuting] exten => _X.,1,Dial(SIP/astinside)

[introutingB] exten => s,n,Dial(SIP/voipprovider)

I will appreciate for your help.

Asterisk Users 2.3 years ago 1 Answer

Asterisk 11, SIP. OK To BYE Goes To Wrong Ip/port Combination


Hi all,

I've read several discussions about asterisk adding 'received' parameter to the top Via header.

In our case asterisk (release 11.4) gets BYE from sip proxy (with BYE top via header containing proxy ip address and port) but added 'received' parameter contains ip address from a 2nd Via (or from "From') and OK gets lost.

I'm just trying to adjust sip configuration that used to work for simple call scenarios (in 1.4, for example) for Asterisk 11. Your input is appreciated.

Thank you.

Alex Zarubin

In sip.conf

[general] nat = no outboundproxy=PROXYipaddress:PROXYport

[CARRIER] type=peer host=CARRIERipaddress port=CARRIERport canreinvite=no

Outbound call from asterisk is established normally via outbound proxy.…

Asterisk Users 2.3 years ago 0 Answers

Blog About WebRTC + TLS + Asterisk 11


I've now prepared a blog about my experience setting up Asterisk 11 with repro as a SIP proxy for WebSocket clients:

In particular, the focus is on the use of packages because that makes it faster for more people to deploy identical working systems. To get the demo running for the WebSocket client, I really only needed to change about 5 lines in sip.conf - all other configuration is the default - the more painful step is rebuilding the packages with SRTP support.



Asterisk Users 2.4 years ago 0 Answers

And Call Drop


Hello All, I am having a bit peculiar problem with Asterisk 11 for a carrier. This carrier shares quite some information in SDP header, which should not be the problem, however what happen is as follow:

Carrier----> (INVITE) -> *SIP Proxy -> Asterisk 11 -> Answer()* -> right after answering call drops... Carrier send a BYE with (cause 79: service or option not implemented).

*NOTE: Please refer to complete SIP traces attached. * * * *Also Note:* *Carrier*: *Proxy*: 77.X.X.X:5060 *Asterisk11*: 77.X.X.X:5080

*Here is Invite SDP from Carrier -> Proxy -> Asterisk 11*

INVITE sip:69609000@77.X.X.X SIP/2.0 v=0 o=AudiocodesGW 1638819008 1638818710 IN IP4…

Asterisk Users 2.8 years ago 8 Answers