similar to: incoming number information

I'm trying to call from XLite phone to PSTN (I've tried this from internet and from local network the same) The Xlite doesn't write that it is connected but receives excelent audio. At the other end comes only noise. Some times only for a second you can here the caller voice , but this was only one time :) I saw with ethereal that UDP packets are coming and going to the asterisk

Hi all: I've no response for the last question with the same subject. Please excuse me for the extreme length of this mail, but I send 2 SIP traces. I have problem with * and 5300, when the incoming and outgoing call are routed thru the same SIP gateway (AS5300). Do I need to set an special things in sip.conf? First all, the * printout. Second, the 5300 trace. Thanks in advace, Gus

Hello: I'm newbie in asterisk, please help me. My context is as follows: 192.168.4.2 --> Asterisk 11.13.1 complied from source 192.168.4.4 --> Yeastar NeoGate TG100 GSM gateway When I call from a GSM cell phone, my TG100 GSM gateway answers and dials extension 7777 (configured as a hotline on TG100) to asterisk server, but asterisk server sends me "SIP/2.0 401

2014-11-12 2:45 GMT-02:00 Luis Eduardo Cortes <luedcortes at gmail.com>: > Hello: > > I'm newbie in asterisk, please help me. > > My context is as follows: > > 192.168.4.2 --> Asterisk 11.13.1 complied from source > > 192.168.4.4 --> Yeastar NeoGate TG100 GSM gateway > > When I call from a GSM cell phone, my TG100 GSM gateway answers and > dials

Dear all, I have small question in sip.conf I added [service] type=friend ;username= ;secret= qualify=900 host=X.X.X.X dtmfmode = rfc2833 disallow=all ;allow=g729 allow=gsm allow=alaw allow=ulaw and I can proccess incoming call from soft phone only I calling on number that is used in extensions.conf(in example below it is 1) exten => 1,1,Answer; exten => 1,2,Playback(hello-world,skip);

Hello, I am looking for ideas and suggestions. I want to use a 16 port GSM gateway as a trunk for outbound/inbound. I will also have two PSTN phone lines coming into the Asterisk server. All calls will go to an IVR on the Asterisk PBX. Outbound from extensions will route to GSM-GW when the dialed number matches a pattern set in the GSM-GW trunk. And accordingly for the PSTN trunks. But that's

Greetings to all, I ma having a problem with channel variables on a couple of our Asterisk boxes. Here is the setup. Asterisk on customer's site (1.2.5), using IAX to our external GW (1.2.5), IAX to PSTN GW (1.0.10), E1/PRI to PSTN. On the External GW, we also have an IAX trunk to a VOIP provider if for some reason the E1 is down. If the DIALSTATUS is CHANUNAVAIL, which should be

Hello, I have found asterisk is dialing a random number when it recieves a call, would anyone know why? The first thing I noticed found peer 4563 (this is a n Xlite Client) Many thanks, Spencer SIP Debugging Enabled spitfire*CLI> <-- SIP read from 82.70.154.145:5060: INVITE sip:448715046363@iptel.tgfslp.dalmany.co.uk SIP/2.0 Max-Forwards: 10 Record-Route:

Hello guys, i have a question about a voip gateway we use. I saw those values typing in cli: core show translation g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 slin16 g723 - - - - - - - - - - - - - - gsm - - 2001 2001 6000 2001 2000 16000 - 34002 - 6000

I've installed Asterisk from the Asterisk @home distribution. Ultimately I will be using Asterisk for voicemail for about 150 users. Calls are (mostly) handled by a legacy PBX although we do have a couple of Cisco 1760 routers that connect a remote office. I've setup a SIP trunk that routes calls from Asterisk to the 1760, and that works fine. I've also configured one of the 1760s to

Hi I am a newbie to * and I am having a problem which appears strange as I did not find any mention of it anywhere in my search. Simply speaking, I have an external SIP proxy server which I am trying to configure for incoming and outgoing calls from my asterisk installation. So here is my configuration in sip.conf [general] register =>

According to the wiki at voip-info.org, the dial structure for using oh323 without a gatekeeper is: OH323/<exten>@<host>:<port> or OH323/<exten> The second option is valid only in the case where a gatekeeper is used. NOTE: OpenH323 library v1.12.0 has a bug in the parsing of the destination host. When this version is used then the above syntax should be:

Dear All I want to use asterisk for the following Senario and Need help to find a SAMPLE extension.conf Incoming call >>>>>>>>>>>Asterisk >>>>>>>>>>>>>>GSM Termination Gw first leg second leg What I want to do is putting first call leg on

Hi list, I try to connect to a GW which have one domain eg sip.mydomain.com and have few IPs related to this domain. I register * to this domain with host=sip.mydomain.com and type=user. So DNS will decide on which IP of my domain I will register (or redirection on the GW side). If an incoming call arrive, I would guess that, as type=user, it will not try to match the IP from INVITE as I

Hi All, Sorry if this has been covered in the past; I've tried searching the archives, but haven't had any luck finding a similar problem. Basically I have problems when using IAX2 (which I now understand is just IAX). I have three IAX connections setup - VoicePulse, IAXtel, and an Asterisk IAX<->PSTN termination provider here in Sydney (ATP) If I try to use the CVS STABLE version

Hi All When I dial a number it reaches the asterisk switch as abc at xyz@123.com what I need to do is to parse the abc and send it to my pstn gateway as in exten => _.,2,Dial(SIP/${EXTEN}@pstn.gw) this does work but I do have a varying number of numbers before the @ exten => _.,1,Dial(SIP/${EXTEN:0:12}@pstn.gw) Well can I use some kind of regular expression to take all numbers before

Hi All, I;m using asterisk 1.4 with FreePBX and a Grandstream 4108. I am noticing a delay calling in and out via the FXO, but calls to local extension are ok. What i noticed when i used ngrep is that, it sends invite but got no response from the server, send another invite but got no response again, then again until it finally gets it. but if you will notice on the 2nd ngrep, the asterisk

Hello. I'm trying to use Asterisk in combination with SER, to make the routing proccess to my PSTN-Gateways. I made a simple test defining some extension in my extension.conf, when i made a call my SER (SIP) Server forward the call to Asterisk, this proccess is ok, but when the call is answered i see an INVITE going out from Asterisk to my SER Server, this invite is then passed to my

Hi I've problem with incoming call quality to GSM gateway connected to beronet card (BN8S0), -----> [ GSM Gateway ] -------> [ BN8S0 ] ==== asterisk Port connected to GSM gatway is in TE mode , gateway is in NT mode , When I dialin to cellphone numer , call goes to 'from-eragsm' context, to Echo application. [from-eragsm] exten => 700,1,Goto(600,1) exten

Hi, I tried to play the standard demo-echotest file !. It works when i use an ip-phone (like x-lite or kphone), but as far as i use an PSTN Gateway (from an VOIP Provider) to call my phone - i get the following error: Sep 8 14:58:33 NOTICE[-182461520]: channel.c:1691 ast_set_write_format: Unable to find a path from GSM to G729A Sep 8 14:58:33 WARNING[-182461520]: file.c:779 ast_streamfile: