similar to: SIP dialing with authentication with dialstring and wothout sip; conf

I've been banging my head against a brick wall for the last hour and I'm sure this is one of those easy to solve things - just that I can't see the wood for the trees. I'm trying to do: ----------- [some-context] Exten => s,1,Macro(dodial,'SIP/201,15,tT',123456,MOHClass) [macro-dodial] Exten => s,1,SetCallerID(${ARG2}) Exten => s,2,SetMusicOnHold(${ARG3}) Exten

Hi, Looking at http://the-asterisk-book.com/1.6/applikationen-dial.html you should be able to dial with SIP credentials in the DP. Is this still possible in recent versions of Asterisk either with chan_sip or pj_sip? TIA. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm building a dialplan for use with a bunch of GXP2000 desk sets. During testing, we had some user issues surrounding the lack of an on-phone dialplan. Users would hit 9 and sit there waiting for a redial tone, and the GXP would time out, sending just '9' to *, which couldn't do much other than spit back a 404 or play pbx-invalid. I turned on the "early dial" option

Hello, We have 2 SIP trunks from Bandwidth.com and if both are in use and someone tries to dial out, they cause another call to get one-way audio (the caller hears us, we cannot hear them). This happens 100% of the time and Bandwidth.com doesn't offer any support. I don't see any setting that tells Asterisk that there are 2 channels available from Bandwidth.com's IP. I'm

On Tue, Jul 9, 2019 at 6:05 AM Joshua C. Colp <jcolp at digium.com> wrote: > On Tue, Jul 9, 2019, at 7:00 AM, Dovid Bender wrote: > > Hi, > > > > Looking at http://the-asterisk-book.com/1.6/applikationen-dial.html you > > should be able to dial with SIP credentials in the DP. Is this still > > possible in recent versions of Asterisk either with chan_sip or

I was wondering if anyone has an easy way to emulate dialing in a round robin fashion like when you use Zap/r1 for Zap trunks. At the moment what I do is simply make a macro that will dial the sip trunks in order so if the first one fails it goes to the second and so on. The problem with this approach is that the first few SIP trunks will always be busy because of outgoing traffic. Is there an

Update on this: I found that by changing insecure = very to insecure = invite, adding the second trunk no longer stopped calls working. I've read the documentation on this switch and still don't see how it applies/is meant to get used. Anyway, with this change in place, the following may help: asterisk*CLI> sip show registry Host Username

Hi, I am attempting to setup an asterisk server, current specs: CentOS release 5 (Final) Asterisk 1.4.11 Asterisk-gui checked out from SVN last week I started with a fairly basic setup involving one VOIP provider who provided one dial in number, and a couple of handsets. Config files are below. It was pretty much totally built by Asterisk-gui, except for the fact I had to add

Hi everyone! I couldn't find anything expressive about gtalk dialstrings. It doesn't seem to work. I'm not sure why, so I'll start at the easiest point. The syntax I found was: gtalk/my_account_name/buddys_account_name at gmail.com Is this correct? And does any of you googletalkers know, if a simple google-mail account is enough to use the talking bit, or do I have to

We have multiple entries like the one below in our users.conf file... where the username. Contact and secret changes for different customers and we register on their behalf to the Service Provider. For the trunk below: when the call is placed out, Asterisk (1.4.18) sends the username of "abc.com" in the MD5 Auth .....which obviously does not match the trunk setup for this Customer with

Having big problems and for months. Our service provider (via:talk) says they are Asterisk friendly but they are not. Here are the specifics (please read the bottom of the msg too) System: Dell SM Business server 2GB RAM, Core II Processor (should be plenty) OS: open SUSE 11 Asterisk Version: 1.4.2 Asterisk GUI Version: 2.0 The system was completely set up using the Asterisk GUI with a

Hello, I would like to have "whisper" channel spy (not private whisper) in Asterisk 1.2. I see here: http://www.the-asterisk-book.com/unstable/applikationen-chanspy.html That is only available for Asterisk 1.4. I wonder if there is any way to emulate it in Asterisk 1.2. For example, to "join" two calls: one to use a private whisper and other one to use a normal chanspy.

Hi i need to implement "Inward" SIP usring dialing in my Asterisk IPpbx, So anybody can recah me by dialing my SIP uri. same time my DNS on same server where currently Asterisk running. how ican implement this. Please help me with config details at DNS & Asterisk point of view. anybody can provide me config exmple? I am using Asterisk 1.4.9. Plz help me Regards Amit

Hello list, I try to setup an asterisk-server with different SIP-Peers to PSTN. The Peer are working and configured in sip.conf: [peer1] type=peer host=10.10.10.1 [peer2] type=peer host=10.10.10.2 Now dialout is no problem. Extensions.conf says: exten => _0Z.,1,Dial(SIP/49${EXTEN:1}@peer1,30) But how can I setup a failure-route if the SIP-Proxy "peer1" ist not

--- span_1 = DAHDI/g11 1,1,dial(${span_1}/${EXTEN:0}) --- The configuration was rsync'd from a working pair of asterisk servers in another office. The only difference was the version 1.4.22 for the original servers that were operating as expected, 1.4.24 and 1.4.24.1 for the new servers. Included in both working and non working servers is the following configuration:

Hi everyone, having a issue with asterisk and my new Voip providers service. Iv set up many asterisk systems before but never seen this and have tried to fix this with no luck.. I have used this exact same sort of setup for 5 other providers and never had this issue, If i replace the trunk login details with my works voip account and set it to IAX then it works perfect, Just not the new

Replying to myself, I've just read in 1.6.1 announcement that a new Incomplete dialplan application is the one that provides what I'm looking for ... 2008/10/3 Olivier <oza-4h07 at myamail.com> > Hi, > > If my memory serves me right, there was thread (in dev mailing list ?) > explaining how we could implement 2 stages dialing with SIP endpoints: > user dials 1234

Hi, I just wonder, if it is possible, to suppress my own number on outbound dials with chan_capi. I took a look into the sources and think it might work with toggeling the "@" in front of the outbound msn in the dialstring. (Dial(CAPI/@msn... vs. Dial(CAPI/msn... But it doesn't work. Maybee I'm wrong and misunderstood the code. Thanks for any answers! Karsten

Hi, I am trying to setup a simple home voip service w/ * I have compiled and installed the svn source as a first step I am trying to configure SIP for inside my network. I have a handful of softphones and a few hardphones that I want to all be able to call each other I have configured users.conf with a single softphone(kphone) and have tried calling itself (ext 6000) and the demo from the

Dear all, I want to setup the incoming calls, that don't use authentication in sip.conf file. My configurations as follows, [2000] type=peer host=dynamic insecure=port,invite ; (both) context=Testing But when I call '2000', I noticed the following message in Asterisk console, "NOTICE[5702]: chan_sip.c:10543 handle_request_invite: Failed to authenticate user