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 2.1 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 2.2 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 2.3 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 2.7 years ago 8 Answers

New How-to Guide: Using Repro SIP Proxy For TLS With Asterisk


Given the limitations around Asterisk's TLS support, and all the benefits of using a SIP proxy, I've put together a rough guide about how to use the repro SIP proxy as a front-end for Asterisk connectivity with TLS peers:

It works for TLS from phones, but also for full federated SIP with any other SIP-enabled domain on the public Internet.

* repro does all the connectivity work (certificate validation, etc) and registration service

* Asterisk sits in the background and provides applications (voicemail, queues, etc)

Any feedback, questions or discussion about this is very welcome.

Asterisk Users 3.1 years ago 1 Answer



Hi everybody, I want to use asterisk for build a SIP Proxy and REGISTAR, Is
AsteriskNow a Sip Proxy already? or, do i need to build it? How does asterisk solves the issues of a SIP USER AGENT Behind a NAT? Does asterisk implements RFC 5626? Does asterisk implements STUN, TURN and ICE? Thanks for your time. Sorry for my english.

Asterisk Users 3.3 years ago 1 Answer

twenty thousands (20, 000) users, which asterisk and how many servers?


Hi All; I need to use Asterisk for 20 000 users, so which asterisk version to be used? Is there asterisk version that supports 20,000 users on one hardware machine? Can I use one strong hardware server i7 with 64 GB RAM and fast hard desk to handle 20 000 users, and concurrent calls 2000? Or I need multiple servers, how much? If I am going to use multiple servers (until now I do not know how much, and I do not know if the barrier will be the asterisk software or the hardware), then do I have to use…

Asterisk Users 3.3 years ago 2 Answers

enabling dialing by sip uri


On 05/10/2012 09:39 AM, Arif Hossain wrote:
> I have following sip account :
> Name/username Host Dyn
> Forcerport ACL Port Status Description
> demo-alice/demo-alice D
> N 1080 Unmonitored
> demo-bob/demo-bob D
> N 5060 Unmonitored
> and i have set up the following extensions for them:
> [users]
> exten=>6001,1,Dial(SIP/demo-alice,20)
> exten=>6002,1,Dial(SIP/demo-bob,20)
> exten => _.,n,GotoIf($[${SIPDOMAIN} = ${ASTERISK_IP}]?unhandled)
> exten => _.,n,GotoIf($[${SIPDOMAIN} = ${ASTERISK_IP}:5060]?unhandled)
> exten => _.,n,Macro(uri-dial,${EXTEN}@${SIPDOMAIN})

Asterisk Users 3.4 years ago 3 Answers

Open source replacement for AudioCodes nCite 1000 SBC


List users, I have an AudioCodes nCite 1000 SBC that is end-of-life and I'm
looking to replace it with open source software. I believe one of the
SIP proxy projects will fit my needs, but I'm a bit overwhelmed by
the number of choices and I'd like the advice of experienced users
before I venture too far down any one path. The projects that came to mind first were Kamailio, OpenSIPS, SER, and
SIP-Router, but I'm aware that there are others and I'm open to
suggestions. Please keep in mind that I'm looking for…

Asterisk Users 3.4 years ago 0 Answers