similar to: Problems Solved, two left

I'm using "s" extension in my dialplan: [gv-voice] exten => s,1,Verbose(callerid is "${CALLERID(all)}" or "${CALLERID(num)}") ;Set(Var_TO=${SIP_HEADER(TO)}) ; PJSIP_HEADER(read,To) same=>n,.... But when a call comes in to the gv-voice context, "s" doesn't match the extension: res_pjsip_session.c:2991 new_invite: Call from

Yes, I have tried that too (i.e, exten => +17775551212,1,Log(WARNING, TWILIO)). It does not work and NO error message in CLI. I have also tried http://orourketech.com/elastix-plus-sign-caller-id-messing-things/ since I first emailed this group, but that does not seem to work either. Here is my log: [Dec 2 15:09:28] NOTICE[26464]: res_pjsip_session.c:1920 new_invite: Call from

Hello Doug, Friday, December 13, 2019, 11:03:37 AM, you wrote: >> This is exactly what I do - “press 1 for a human” >> Works great > I do this as well, but I also do a database lookup to see if the number > is on our speeddial list and if so, pass the call directly on without > the IVR prompts. I do something similar for calls without caller ID, but I was still getting

Hello, I am running Asterisk 13.6.0 in an AWS instance, and I set it up with Twilio SIP trunk using pjsip_wizard.conf (nice feature!). I see that the calls actually "reach" the PBX, but for some reason, they are not caught by any of my extensions context. Here's what I observe when I test this from a non-PBX connected E164 number (a landline), say 555-666-1212. My Twilio number is

Asterisk 12.5 I'm using AMI to initiate a "call me now" feature from the web site. The AMI looks like: Action: Originate Channel: Local/s at callmenow Context: dial-to-customer Exten: s Priority: 1 Async: true Variable: CHANNEL_TO_CUSTOMER=SIP/voipms/1112223333 Timeout: 999999 Dial Plan: [callmenow] exten => s,1,NoOp(callmenow: Queue without answer) same

On 5/24/23 08:03, Steve Matzura wrote: > > ***  extensions.conf  *** > > > [general] > > [globals] > > ; Make sure to include inbound prior to outbound because the > _NXXNXXXXXX handler will match the incoming call and create a loop > include => voipms-inbound > include => voipms-outbound > > [voipms-outbound] > exten =>

Asterisk 11.1.0 I'm trying to use the "b" subroutine of the Dial application so that I can do some stuff with our internal applications that need to have access to the called channel information. I can see that the subroutine is being executed, but the arguments I pass don't see to make it to the subroutine. [callmenow] exten => s,1,NoOp(callmenow: Queue without answer)

Hello community, I've been running Asterisk on an embedded device for about six months, and my operation has been largely trouble-free. I'm hoping I could get some help with a minor problem: Every week or three, my PBX gets stuck in a state where it can receive calls, but it becomes completely unable to originate outgoing calls until I do a "sip reload". After doing the SIP

Hello I'm using softphones as my only 'landline' phone service for almost 3 years now (Diamondcard and now voip.ms), so far using SIP (and mostly Twinkle). Also, I'm using Linux (Debian) as my choice of desktop OS. Also, sometimes I'm in networks with badly behaving NAT routers (for some time I used openvpn to solve this unreliably, then I ended up using 3G instead of wifi

Hello, everything goes well on asterisk 1.4.30, but with asterisk 1.6.2.11 I get the following : [Sep 12 18:59:29] WARNING[2066]: chan_sip.c:12738 check_auth: username mismatch, have <329909006666>, digest has <3291119600> [Sep 12 18:59:29] NOTICE[2066]: chan_sip.c:20082 handle_request_invite: Failed to authenticate device "0473990000" <sip:0473990000 at

>Using system() you could issue a asterisk -rx 'core restart now' So I tried this exten => s,1,Playback(beep) exten => s,n,Dial(Console/default,20,g) exten => s,n,Hangup exten => s,n,System(asterisk -rx 'core restart now') But it does not continue. Last thing I see is "Exited non zero" so its not doing the hangup or the system. jerry --------------

This was supposed to go to the list. I am now thoroughly confused. In the [voipms] stanza where endpoint is defined (type=endpoint), everything points to voipms. But in the [yealink] stanzas, I tried pointing everything to Steve, one item at a time, then both of them, and nothing changed. On 5/24/2023 10:00 AM, Stefan Tichy wrote: block quote Am Wed, May 24, 2023 at 09:40:18AM -0400 schrieb

Doug from this list got me to change my connectivity to my DID provider from SIP to IAX, and bingo, it all just worked instantly. For my next trick: setting up voicemail. The book does it all with smoke and mirrors (SQL), but I'm fresh outa those, so I'll be doing it the old-fashioned way, by editing the voicemail.conf and users.conf files with some hopefully helpful hints from our

I saw this thread which is very similar to my issue, though I cannot solve mine with iptables. http://lists.digium.com/pipermail/asterisk-users/2013-September/280429.html Using asterisk 11.5, IAX does not seem to be able to receive any packets. My IP tables looks like this: root at dlaptop:/home/darryl# iptables -L Chain INPUT (policy ACCEPT) target prot opt source

I am attending a training course covering rspec. I am using a MS WinXPpro SP3 machine. I have installed the cygwin environment. I am using git 1.5.4. I am using ruby 1.8.6 (2007-09-24 patchlevel 111) [i386-mswin32] and rails 2.0.2 (albeit rails 2.1.0 is installed as well) I had to install rspec-rails and rspec via git and as a gem. Problem 1. If one goes to the rspec.info website and links

Hey; I figured I would ask here as I seem to get better results. I am using the voip.ms <http://voip.ms/> VoIP service. I have no problem configuring my Asterisk server 1.8x to dial out with my Softphone. HOWEVER, for some reason, I cannot get inbound. All that I hear is a busy signal. I know this is not much for you folks to go on, but what would be a good place to start

On Mon, Jun 5, 2017, at 04:26 PM, Michael Maier wrote: > On 06/05/2017 at 06:29 PM, Joshua Colp wrote: > > On Mon, Jun 5, 2017, at 01:22 PM, Michael Maier wrote: > >> > >> Do you have any idea where to start to look at? Adding additional output > >> in the source code? Which functions could be interesting? I may add own > >> debug code to see why things

07.03.2015 0:24, Kevin Harwell ?????: > On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov <serov.d.p at gmail.com > <mailto:serov.d.p at gmail.com>> wrote: > > Hello. > > Asterisk 13.2. > I transfer configs from chan_sip to res_pjsip. > In chan_sip i have "match_auth_username=yes" and have nothing in > pjsip. > > I have a

Hello. Asterisk 13.2. I transfer configs from chan_sip to res_pjsip. In chan_sip i have "match_auth_username=yes" and have nothing in pjsip. I have a lot of endpoints and registrations on same SIP server. And it's problem in pjsip now. Is not it? I requesting to add new value for endpoint option identify_by. The value 'uri'. Simple config (cutted): [siptrunk]

On 06/14/2017 at 05:53 PM Joshua Colp wrote: > On Wed, Jun 14, 2017, at 12:47 PM, Michael Maier wrote: > > <snip> > >> >> I added this patch to see, if really all packages are are freed after >> they have been processed: >> >> --- b/res/res_pjsip/pjsip_distributor.c 2017-05-30 19:44:16.000000000 >> +0200 >> +++