similar to: PSTN connectivity

Hi All, I have searched the various post and not able to find the solution. I am using Asterisk 1.4.21.2 for outgoing calls. Earlier i used ZAP trunk and it works fine. Now i need to move to SIP trunk and configured the same. When i try from softphone i got error as "Call rejected" and in the asterisk i got error as

hi,all when i send a call to other server use SIP trunk, i got this below, <--- SIP read from 222.46.18.52:5060 ---> SIP/2.0 403 Forbidden what's rong with is? > Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others. > Created by Mark Spencer <markster at digium.com> > Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for

i have a prablom here, i want to send a call from A to B use sip trunk , the call can sended B,but not work to PSTN. the message from B server. help pls,what's rong? > > <--- SIP read from 192.168.0.176:5060 ---> > INVITE sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151>SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport

Hello everyone! I've had this problem for a while and cant figure it out. When an outside caller calls an extension on my asterisk system, they do not hear any sort of ringing. Inside extensions calling other extensions do hear ringing. We have 3 other asterisk systems that are configured the same way, but do not have this problem. We think it has something to do with asterisk 1.6. The other

Hi this message give me when I calling a number than actually not busy: "Dial failed due to trunk reporting BUSY - giving up" max channel is unlimited and sometimes it dial number ok but most of the time it gives me this error. Please inform me how can solve this problem. thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

You would need to give more information really. Your sip.conf file listing the entries for the phones especially which codecs are permitted. A copy of the 'asterisk -rvvv' console output when you make the call. On 27/03/15 17:05, Salaheddine Elharit wrote: > please no body has som with aastra can help me in this issue > > 2015-03-26 11:02 GMT+00:00 Salaheddine Elharit >

hello list i have an issue related to outbound calls i can contact all the number except on number given by our provider in trunk the issue just when i configure my trunk in our server but when i configure the trunk directly in x-lite i can contact this number without issue below the cli == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [0149xxxxxx at

I am trying to record calls on demand both inbound and outbound calls. I can record outbound calls just fine but not inbound calls or calls from an internally between extensions. I am using the latest asterisk 1.8.x certified version. On an outbound call I see: == Using SIP RTP CoS mark 5 -- Called SIP/ BVTrunk /7190000000 -- SIP/BVTrunk-00000163 is making progress passing it to

I just had a user report that they called out to someone on a cell phone this morning, and after a short conversation, the call was dropped/lost. The person on the cell phone says that this is very rare, and would not suspect the dropped/lost call to be on their side. I have looked at the asterisk/full log as thoroughly as I can, and have pasted the lines which seem relevant to that call below.

Hi people, I have a problem with some extensions. The calls are ended after 31/35 seconds, also, it depends on the number which I call. This is the log, but I've not been able to find something wrong... Any ideas? [Jun 11 15:50:46] DEBUG[26071] app_macro.c: Executed application: ExecIf [Jun 11 15:50:46] VERBOSE[26071] logger.c: -- Executing [s at macro-dialout-trunk:16]

I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make this thing into MGCP gateway or even a SIP gateway for asterisk? Seems like it should bee useful for something! I'm perfectly happy to do my homework, but also don't feel thee need to reinvent the wheel! So, links with relevant info would be appreciated. If there is a config for a 2621 being used as a gateway

Could someone please comment on the current state of chan_capi, chan_misdn and chan_modem channel drivers in terms of functionality and stability. Specifically, which channel driver would be best for a passive PCI HFC or W6692 ISDN card. The chan_misdn wiki claims that chan_capi distinguishes between junghanns and non-junghans cards, and that chan_misdn is better suited for general misdn

Hi All, I am a new bee to *. I just installed Asterisk@home on FC3. I hv a FXO card. I hv configured two extensions one x-lite and other iaxComm. I configured * using AMP. The following setup works - x-lite (x 200) to iaxComm (x 201) - PSTN to x-lite - PSTN to iaxComm Voice mail, weather etc work fine. When i try to make an external call i am getting message "All routes are busy". In

Hi, We are using VOIP-SIP gateway to route outbound PSTN calls. Recently, I am getting == No one is available to answer at this time message, after making 5 SIP attempts (Retransmitting #5 (no NAT):), and the calls are going out through alternate Zap-trunk. I do not see any hit (sip-debug traffic) on the voip-gateway for the failed calls. Strange thing is that this is happening randomly,

How many times do you ask for help here before getting a respone? Every single thing I do No matter what I get busy extensions. I am willing to pay someone to help here. Anybody got a clue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm

Greetings *`s, I am manually creating call files and dropping them into /var/spool/asterisk/outgoing to be picked up by *. Presently, when I use local/internal parameters using SIP it works..ie I make an internal call from device to device. However, when I try dial an outside number which I have set up in a custom conf file, it bombs out with the following message :

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

Hello, My client customers complaining that their calls suddenly get hung-up, I am just investigating if the problem from my side, I had a log of a hang-up case, Does it help to know if there is a problem that can be resolved from my side? elastix*CLI> -- Hungup 'IAX2/99999-6813' == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on

hello list, i have asterisk 11.15.0 and i have some trunks sip from my provider we have some ip phone astra 6731i each Ip-phone is configured with trunk and we call no ihave configured another trunk from the same provider in my asterisk i can call all numbers just the numbers are configured in thses ip phones. but when i configured the same trunk in x-lite i can call theses ip-phones without

Hello everybody, and first of all sorry for my poor English. I'm having trouble with Quadbri installed on Asterisk 1.2.17-BRIstuffed-0.3.0-PRE-1y-e. Everything is working fine, except calling to switched off or "out of coverage" cell phones. In this case I have to wait 40 seconds until Asterisk tell me that "all circuits are busy now" instead of receive cell phone