* You are viewing Posts Tagged ‘sip proxy’

Cisco 7940 SIP 8.12 No Audio When Using Outbound Proxy

Hi All,

Simple scenario:

7940 SIP> Inbound/outbound calls work fine 2 way audio, features ok, no issues that I can tell so far.

7940 SIP Using Outbound SIP Proxy> w/Public IP
Phone registers, call in/out SIP Signaling traversing the proxy ok no audio on phone, SDP messaging is correct. Not using media proxy, media flows between Asterisk and NAT router to phone, no return media from phone to Asterisk. This only occurs on inbound calls to the phone, when the phone makes outbound calls, audio sets up fine 2-way. On inbound call to the phone, I can see media going to the phone but don't hear any on the speaker/handset, no media flowing out of the phone back towards Asterisk so no audio on that end either.

I’m testing various phones, the Polycom and Cisco SPA5xx lines work great using outbound proxy. So I’m certain this is a Cisco 7940
problem not accepting RTP due to some internal security check with SIP
signaling and media coming from different ip addresses or something like that.

So testing a bit more, I put the Cisco 7940 on a Public IP, seems to work fine, audio sets up 2-way inbound and outbound calls. So now I’m thinking it is a NAT issue, but only when using outbound proxy, doesn’t make sense, now I’m really confused.

Any feedback is appreciated.

Thanks.

JR

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.

SIP.conf

[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

Extentions.config

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

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


I will appreciate for your help.

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. BYE coming from the CARRIER

<--- SIP read from UDP:PROXYipaddress:PROXYport --->
BYE sip:XYZ@ASTERISKipaddress:5060 SIP/2.0
Via: SIP/2.0/UDP PROXYipaddress:PROXYport;branch=z9hG4bK-whatever-rsMNRDPbOY.0
Via: SIP/2.0/UDP CARRIERipaddress:CARRIERport;branch=z9hG4bK+17d247e1b781cd07f8d5339588fb32091+127.0.0.1+1
From: ;tag=127.0.0.1alUtKGp-07233+1+9362fc7+68f1babf To: ;tag=as1ca814af Call-ID: 146b538429a2380a68bb374543d40c6d@OURipaddress:5060
CSeq: 1035426164 BYE
Max-Forwards: 69
User-Agent: Alcatel-Lucent 5060 MGC-8 8.3.0.6.SP1.2
Content-Length: 0
Supported: replaces, 100rel

Asterisk adds received=CARRIERipaddress (taken either from 2nd Via or from ‘From’) and sends OK to CARRIERipaddress:PROXYport. This OK goes nowhere, carrier re-sends BYE several times…

<--- Transmitting (no NAT) to CARRIERipaddress:PROXYport --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP PROXYipaddress:PROXYport;branch=z9hG4bK-whatever-rsMNRDPbOY.0;received= CARRIERipaddress Via: SIP/2.0/UDP CARRIERipaddress:CARRIERport;branch=z9hG4bK+17d247e1b781cd07f8d5339588fb32091+127.0.0.1+1
From: ;tag=127.0.0.1alUtKGp-07233+1+9362fc7+68f1babf To: ;tag=as1ca814af Call-ID: 146b538429a2380a68bb374543d40c6d@OURipaddress:5060
CSeq: 1035426164 BYE
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer Content-Length: 0

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:

http://danielpocock.com/using-resiprocate-to-connect-asterisk-webrtc

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.

Regards,

Daniel

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*: 62.61.147.214
*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 62.61.147.214
s=Phone-Call c=IN IP4 77.X.X.X
t=0 0
m=audio 53372 RTP/AVP 8 118 18
a=rtpmap:8 PCMA/8000
a=rtpmap:118 PCMA/8000
a=gpmd:118 vbd=yes a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no a=ptime:20
a=sendrecv a=rtcp:53373 IN IP4 77.X.X.X
m=image 56854 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:122
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

*SDP:After Answered by Asterisk 11*
v=0
o=root 164966782 164966782 IN IP4 77.X.X.X
s=Asterisk v11.0.1
c=IN IP4 77.X.X.X
t=0 0
m=audio 12636 RTP/AVP 18 8
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv
*m=image 0 udptl t38*

I have tired by disabling/unloading fax modules as *I am not using* them but no results. Secondly, also tried tweaking of udptl ever-odd nothing worked.

The same carrier works for Asterisk 1.6.X and the only difference I have notice so far is the above underlined line in Answered SDP -> m=image 0
udptl t38. I think if I some how do not advertise udptl here i would be able to avoid this scenario. I have tried multiple ways to strip off SDP
from incoming INVITE at SIP proxy level but it is not SDP wise enough.


*Note:*

In Asterisk 1.6 => WARNING[32671]: chan_sip.c:8833 process_sdp:
Unsupported SDP media type in offer: image 59978 udptl t38
In Asterisk 11 => WARNING[18748][C-0000002f]: chan_sip.c:10277 process_sdp:
Failed to initialize UDPTL, declining image stream

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:

http://www.opentelecoms.org/using-repro-with-asterisk-or-freeswitch

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.