similar to: Network based transcoding

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

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

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

<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"

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

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

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

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

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

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

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. >>

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

Hi Guys 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 of the ports in their own groups so that I can have different signaling on each? [channels] language=en switchtype=euroisdn pridialplan=unknown resetinterval=600 echocancel=yes echotraining=yes

Hi all! :) I search a function or option for application Dail(). My situations: I have two or more Dial()s with multiple devices (Handgroups). Level1: Dial(SIP/device1,20) Level2: Dial(SIP/device1&SIP/device2,20) Level3: Dial(SIP/device1&SIP/device2&SIP/device3,20) When in level one, no one accept the call until the timeout, they have a missed call on device. When in level two,

Hi Guys We have a strange a strange issue at a client they have 3 panels on their phone and every so often the panels reboot themselves yet the phone stays on. We decided to replace the T26 for a T28 to see if it fixes the issue and still have the exact same issue. Has anyone seen this before? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello Everyone, We are getting some rather poor results (relative) with our Asterisk setup. Not sure if we are using the sipp correctly etc.. but nevertheless, is there any documentation that describes how we can get the most our of our Asterisk box. For example when we hit the "too many file" error, and fixing it using ulimit..... Also, is there any way we can allocate sufficient

Hi Everyone. I have a customer looking to deploy about 20 Grandstream GXP2140 phones. Normally they would deploy Yealink brand phones but they want a phone with gigabit pass through and the Yealinks with gigabit are too expensive for their budget. Does anyone on the list have experience with the GXP2140? Is it a reliable phone? Does anyone have recommendations for other phones with gigabit

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 >>