similar to: Sequence of transfers fail

Hello, I have the following scenario: 1. VoIP Gateway G400 connected to PSTN 2. Asterisk server 1 (working as IVR) 3. Asterisk server 2 (working as ACD, with several agents connected) I have incoming calls coming from PSTN through the VoIP Gateway to Asterisk server 1 (IVR). When the IVR ends working with the call, transfers it to the Asterisk server 2 (ACD). In Asterisk server 1

Hello Everyone, We are looking for solutions where the transcoding is abstracted away from our * box (i.e., to the network layer) using some carrier grade gateway, or router. The reason for my post is to know about solutions people have used in the past, and how it fits into their overall architecture. Our transcoding needs consists mainly of u/alaw <-> g729, and gsm would also be good....

Unsubscribe Elvin G. Nodalo -----Original Message----- From: asterisk-users-request at lists.digium.com Sent: 7/10/2013 1:00 AM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 108, Issue 14 Send asterisk-users mailing list submissions to asterisk-users at lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit

Unsubscribe Elvin G. Nodalo -----Original Message----- From: asterisk-users-request at lists.digium.com Sent: 7/10/2013 1:00 AM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 108, Issue 14 Send asterisk-users mailing list submissions to asterisk-users at lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit

When I set CONNECTEDLINE() info for an incoming ISDN call, the calling party sees only CONNECTEDLINE(num) and the name does not get displayed. Some time ago I called a number, where I did get back a name and a number and everything was displayed correctly. So I think the calling site should basically be able to handle all connected line info. Looking at a pcap trace of the D-channel data, I

Thanks a lot Dona and jg for your inputs. I'll try to find some way to do this from Dialplan or AMI and let you guys know soon. Please share if you have some more ideas. Regards, Rajib Date: Tue, 11 Jun 2013 18:34:46 +0200 From: jg <webaccounts at jgoettgens.de> Subject: Re: [asterisk-users] announcement to be played for attended transfer call To: Asterisk Users Mailing List -

yes I called On Mon, Apr 13, 2015 at 1:27 PM, jg <webaccounts173 at jgoettgens.de> wrote: > > >> I'm not able to install asterisk whenever I hit make command I get below >> error: >> >> make[1]: *** No rule to make target `../main/modules.link', needed by >> `asterisk'. Stop. >> make: *** [main] Error 2 >> >> > Just

SNOM phones can be configured using files on a TFTP server. On Thu, Apr 9, 2015 at 11:14 AM, jg <webaccounts173 at jgoettgens.de> wrote: > > Does anyone know how to program Snom phones using a Mac addresses like > what is done with the Ciscos. I have about 50 extensions to be programmed > and I am hoping there is a way on Asterisk to program extensions on the > snom

I'm working with GXP2130. About 12 phone on gigabit with PC after phone. With Vlans on CISCO switch is stable and not so difficult. This configuration running without problems since July 2013. Quoting jg <webaccounts173 at jgoettgens.de>: >> I have a customer looking to deploy about 20 Grandstream GXP2140 >> phones. Normally they would deploy Yealink brand phones but

Is there a recommended way to find out the cause of DIALSTATUS = CONGESTION for PRI/BRI channels? Currently I am evaluating the DIALSTATUS variable and I also count the active ISDN channels for the ISDN trunk in question. Counting the active ISDN channels seems somewhat clumsy as the mapping to a specific trunk must be done by hand (or write even more code). I have a setup where outgoing calls

On 10 February 2015 at 09:02, jg <webaccounts173 at jgoettgens.de> wrote: > > >> >> I get an occasional similar problem, we have Mikrotik firewalls and from >> tcpdump monitoring on the asterisk boxes I can see that the firewall >> (unbidden) has changed the IAX port. Usually a firewall reset and sometimes >> PBX reset combination fixes it. >>

jg <webaccounts173 at jgoettgens.de> schrieb: > It doesn't really depend on your sip.conf and Asterisk. Your gateway/router > will be the major problem. My summer project will be to look at session Are you sure? Right now I'm using an italian SIP-Provider (Messagenet), configured in my sip.conf and I can receive calls without any problem... So, I don't think, I have to

4 Port PRI sangoma a104 From: jg [mailto:webaccounts173 at jgoettgens.de] Sent: Wednesday, March 18, 2015 2:09 PM To: Andrew Colin; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 4 Port PRI I have a 4 port PRI card that I need to setup each port in their own group. In chan_dahdi.conf I have the following which works for one port How do I add the rest

<div>Thanks for the tip. Our goal is to know that call-limit is triggered. And later analyze this info, maybe do some action.</div><div>Yes, we can parse CDRs or execute AGI script but we do not want inmplement this logic on Asterisk because it can affectš<span>performance.</span></div><div>š</div><div>02.07.2015, 15:31, "jg"

I have 2 ISDN BRI boxes, each with 4 spans, where the first one is configured as CPE, the second one as NET(so I don't need real lines for developing and testing). Once in a while I do see the following libpri error messages simultaneously on both boxes: PRI Span: 1 TEI=0 MDL-ERROR (A): Got supervisory frame with F=1 in state 7(Multi-frame established) PRI Span: 2 TEI=0 MDL-ERROR (A): Got

Hi! Sometimes IAX peers are not reachable and with "iax2 set debug on" I get something like this Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 00014ms SCall: 00001 DCall: 01200 79.233.155.174:49153 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00014ms SCall: 01200 DCall: 00001 79.233.155.174:49153

On 10 Feb 2015, at 12:22, Jose Flores Galicia wrote: > 2015-02-09 14:36 GMT-06:00 jg <webaccounts173 at jgoettgens.de>: >> Hi! >> >> Sometimes IAX peers are not reachable and with "iax2 set debug on" I >> get >> something like this >> >> Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: >> PONG >>

Zitat von jg <webaccounts173 at jgoettgens.de>: > Yes, it is called "core set verbose 42", the other options is "core > set debug 42". Enjoy the show! OK, thanks, but with this option I can just debug what happens if I call an extension right now... I'd like to have a command to ask Asterisk how it will handle a call... > Once you are more familiar

Dear R community, I am trying to fit an nlme model where I want to estimate the fixed effects of two treatments on the parameters on the following equation Photo~(a*(1-exp(-c*PARi/a)))-b I was able to fit a simple model without covariates following the method described in Mixed-Effects Methods and Classes for S and S-PLUS, version 3.0, but when I add the covariates, I get the error "

Hi! Maybe I have overlooked something, but I am sort of facing the following problem. I always used the AMI interface to allow (older) client programs on Windows to use their TAPI client code in order to communicate with Asterisk servers. The functionality is basically minimal as only incoming calls need to get detected and there are occasional outgoing ones. For the outgoing calls, a regular