similar to: SIP authentication

Hi, I've searched the asterisk.org and voip-info wiki sites, but not found an answer that seems to match. Hopefully this is a simple question. COLP is working very well on our system - Unfortunately it is working a bit TOO well in some circumstances. We have some "untrusted" trunks. On these trunks, an initial CallerID can be used, but any redirected caller numbers, COLP updates

Hello- I am in the process of adding a new provider to my asterisk box (both for outbound termination as well as inbound DID). They are going to be delivering and receiving traffic via SIP only. Now, in IAX via Voicepulse or others I know that I can simply have one registration statement along with an inbound context, then in extension.conf map the outbound context. from iax.conf: register

I'm hoping someone can help me out with this. I am basically just trying to separate my outbound and inbound calls into separate contexts instead of having everything in a single context. Any help would be appreciated. Perhaps I've missed something really obvious.... Here is the network layout: <remote> <--TDM400P--> <nattedbox> <--IAX--> <liveipbox>

On Sun, Mar 15, 2015 at 8:32 AM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote: > I have setup my Asterisk 13.1.0 server with PJSIP on AWS/EC2. My basic > configuration works, and I am connected to a SIP trunk using SIP.US, and > have set up my inbound calling which works correctly (when I call my PBX > DID, the call does come into my PBX network). > > The

On Sun, Mar 15, 2015 at 10:34 AM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote: > That was the issue, thanks. I now am able to get the caller ringing on an > outbound call, but an external phone number (E164) I am dialing does not > ring. > Any error messages? If you set 'core set verbose 3' and try it, does the Dial get executed? > > On Sun, Mar

On Sun, Mar 15, 2015 at 1:33 PM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote: > Yes, I think the dial does get executed (sonny calling outbound > 202-555-1212): > > core set verbose 3 > Console verbose was OFF and is now 3. > -- Executing [912025551212 at from-internal:1] > Log("PJSIP/sonny-00000031", "NOTICE, Dialing out from

My comment, (forwarded from Bristuff list) - A few people are seeing a Cause 34 (congestion) from ISDN installs, where there clearly is an available channel. This was originally related to Bristuff as it happens to ISDN2 users, but there is at least one report of an unpatched 1.6.x user seeing the same issue. 2009/4/23 Steve Davies <davies147 at gmail.com>: > I think I have a site where

Hello, Asterisk seems to try to authenticate incoming INVITE based on the [section] in sip.conf and not the username specified. I just removed the "insecure" option from my sip.conf requesting every connection to be authenticated. I added the match_auth_username=yes in the [general] section for extra security. To make it work, I have to use the same [section] identifier as username.

Hi, First a confession - The box in question is a 1.2.35 box, so this may be solved in a newer version as I know the JB code is all hugely changed, but... It may be worth checking into. Scenario: - IAX outbound call from Asterisk, which rings okay. - Remote end sends ANSWER, which we immediately ACK. - The ANSWER control packet gets put into the JB (that's how I read the code) - The remote

[0004f2xxxxxx](poly650) defaultuser=0004f2xxxxxx callerid="Front Desk" <1600> mailbox=1600 *setvar=callidnum=1234561600* and from extensions.conf: [outgoing] ; Outbound unrestricted domestic calls exten => _1NXXXXXXXXX,1,Verbose(Outbound call from ${callidnum} to ${EXTEN} on ${STRFTIME(${EPOCH},,%D)} at ${STRFTIME(${EPOCH},,%T)}.) *exten =>

hallo everyone, fyi ... working SIP Trunk configuration between Asterisk and Mitel 3300 ICP attached. let's refine further, please test and share your feedback, regards, Joseph Okoegwale Abuja, Nigeria -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070709/4f7f15e1/attachment.htm --------------

Hi all, I'm trying to resolve a weird issue with SIP routing. I have a number of SIP trunks, from a selection of providers, all of which are registered in sip.conf: [general] context=default allowguest=no allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=yes tcpbindaddr=0.0.0.0 transport=udp bindport=15060 srvlookup=yes allowsubscribe=yes

Hello, Using 1.6.2-rc5, my settings include: [local-phone](!) context=mylocal type=friend nat=no canreinvite=no host=dynamic qualify=yes dtmf=info language=fr call-limit=5 subscribecontext=subs disallow=all allow=alaw t38pt_udptl=no setvar=accountcode=foo [168](local-phone) defaultuser=168 secret=pass168 callerid=John Doe<168>

I have setup my Asterisk 13.1.0 server with PJSIP on AWS/EC2. My basic configuration works, and I am connected to a SIP trunk using SIP.US, and have set up my inbound calling which works correctly (when I call my PBX DID, the call does come into my PBX network). The issue is that I am not able to make outbound calls, because the call fails with the error:

I have a puzzling situation, and would be grateful for any insight. I have a dialplan that forwards an incoming call out to another number via the same SIP trunk as it came in on. e.g. [from-siptrunk] exten => 0123456789,1,NoOp exten => 0123456789,n,Dial(SIP/siptrunk/0987654321) Now, if I use a different SIP trunk for the outbound call, than the inbound call came on, the call is set up

Hi All, Simple scenario: 7940 SIP><NAT Router><INTERNET><Asterisk SIP B2BUA w/Public IP Inbound/outbound calls work fine 2 way audio, features ok, no issues that I can tell so far. 7940 SIP Using Outbound SIP Proxy><NAT Router><INTERNET><Asterisk SIP w/Public IP Phone registers, call in/out SIP Signaling traversing the proxy ok no audio on phone, SDP

Hopefully this is really a generic question about IAX and doesn't turn out to be something specific to Teliax, because they haven't been too helpful so far. All they can tell me is that my login shows "status unknown" on their end, which prevents me from receiving inbound calls on my Teliax number. Outbound calls through the same server work fine, which rules out most networking

Hello, I'm trying to get calls working between websocket clients and sip clients. For clients I have sip.js based clients on chrome, Zoipers and a Grandstream phone. Challenge here is I'd like to have Kamailio and rtpengine to handle the bridging between different rtp profiles but Asterisk changes them in the sdp bodies along the way. I'm using Asterisk 11.11.0. Is there a way to

Hi, I am migrating some billing code from 1.2 to 1.6 cdr output. Mostly this is not too bad, but I have a scenario where some data appears to be "lost" Call from SIP/100 to SIP/200, but the SIP/200 device is programmed to send a redirect to extension 1234. chan_sip creates a Local/1234 at context call, which has its own CDR. In 1.2, the CDR records look something like: 1)

I have done additional testing and I haven't been able to figure out why it's failing. Since my original testing we now set the realm on the authentication section to match what we receive from NEC. It's of the format abc at xyz.com I have verified the md5_cred several times and it matches the user:realm:password formula 3016:insiph at something0a646666.com:3016 where username is