similar to: cisco voice gw / cisco call manager /asterisk for voice record, ivr

I need some extensions logic assistance, I'm trying to dial out one of multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only allow 1 call per trunk) and roll over to a second or third depending on that busy status Here's what I've got for a macro thusfar, but it's not working(fails if the 1st trunk is busy) extensions.conf: [globals] trunk_1 => SIP/trunk1

Hello, For lab testing, I'm trying to build two differents PJSIP trunks between two Asterisk 13.8.0enabled boxes. I thought I could set up both trunks like this: Box A/port 5060 <------ Trunk1 -----> Box B/port 5060 Box A/port 5062 <------ Trunk2 -----> Box B/port 5062 and declare trunks like this: [foobar1] type=endpoint transport=simpletrans context=from-customer

I just thought i'd post a small sample that uses the new chan_local to show one way of doing variable callfwding This sample extension.conf uses's the ast DB to store a users current extension, in a db family of CallFWD and the unique Key is based on the current channel the user is assigned. In the globals var section each key is hardcoded EXT1, EXT2 this is used in the [incoming] context

Hi! I am having difficultly in having users of various SIP devices obtain the correct behaviour when they call a busy number ie. only hearing the Congestion/Busy tone. I assume this might be because the SIP device itself generates the 'ring' tone? With my current setup in the dialplan extract (below) the user of the SIP device hears one 'ring' and then the busy tone if a number

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi there. I've done some googling around to try and find an example of what I'm trying to do, but it's one of those things that just seems hard to find the right terms to search for. If there's some documentation out there on this, I'd appreciate being pointed in the right direction. If not, then if someone has some

Hi, In my dial plan I've configured two trunks to make outbound calls (trunk1 and trunk2) to same service provider but I want when any of my exten starts with _2. should goto trunk2 and there should be some kind of disturbance (like some noise or some background noise) when my calls goes to trunk2 to make the call quality bad. Mainly I want to achieve bad call quality on trunk2 by adding

Before I start, I just want to say this is not necessarily a problem with ASTCC, but may be a problem the way I have set up ASTCC (and possibly the way others have set it up as well). The issue is that ASTCC tries to match the pattern *anywhere* in the called number, not necessarily only at the beginning. I have set up ASTCC Routes like this: 1800 Tollfree Trunk1 0 0 100 1416 Canada Trunk2 0 0

Hello guys. Have 2 external numbers that required registration on provider server, trunk1: 73432260005 at 80.75.132.66 trunk2: 73432260050 at 80.75.132.66 Thing is I can?t figure out how to route them to different IVRs by default Asterisk can?t match endpoint Request from '<sip:+ 73432260005 at 80.75.132.66>' failed for '80.75.132.66:5060' (callid:

It gets hard to read my logs when every time someone makes a phone call it displays long pages of "Dropping voice frame". Anyone encounter this before? Asterisk is bridging two SIP lines together, so the technology should be the same. Maybe I'll try allowing only ULAW. ************************************** Asterisk Standard debug (level 3)

Hi, I want to do so that extension 501 will always use trunk1 for outbound calls and 502 will use trunk2 for outboud calls. How do I do this. Right now all extensions use the same trunk for outbound calls. -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is there anyway to get the following scenario to work... I have 3 IAX trunks that I want to setup to peer with other * boxes. I have 1 physical interface, eth0. I also have 2 sub interfaces, eth0:1 & eth0:2. I want to setup a single IAX trunk on each of the interfaces. All 3 interfaces are going to have separate publicly routable IPs, and for this purpose, let's say that because of

Hello, I want to make an agi script to match incoming DIDs with usernames. I tried to do such entry in incoming trunk. [DID_diddw] include = from-didww [from-didww] exten = 3130XXXXXXX,1,AGI("did.php") exten = 3130XXXXXXX,n,DIAL(SIP/${yup_no},20) but when i run the rule it says chan_sip.c:20152 handle_request_invite: Call from '81.85.224.41' to extension

Hello list, I need a hand to find the best dialplan failover solution when using two SIP Trunks. My reasons to do failover are: a) one of the two providers could be unreachable b) both providers may be UP but one of them could return a SIP error message (maybe caused by DOWN E1s) Googling I found a few possible solutions: 1.

Hi, I am looking to integrate Asterisk with a Siemens HiCom 150e via BRI and wondered if anyone is able to offer any advice. In simplistic terms, my goal is to pass calls from the HiCom to the Asterisk box. e.g: HiCom user dials access code and can call Asterisk extension or establish SIP call over Internet. Likewise, I'd like Asterisk to be able to present a call to the Hicom, either

I have a strange problem with overlap dialing. I installed an asterisk server between a Siemens HiCom PBX and our telephony provider. Everything is working fine except some strange problems with the dialing of the fax (connected to the HiCom PBX). It seems to me that if dialing takes too long Asterisk just hangs up the channel without recognizing that the fax machine is still dialing: (Fax gets

Dan et al; This looks like a perfect solution. However, I have one issue. If I initiate the macro manually (put it in the proper context/dialplan) it works. I see the *.wav file being created and growing in the /var/spool/asterisk/monitor directory. If I try to implement it adding the MixMonApp => *1,self/both,Macro,mixmon to the [applicationmap] in features.conf, I cannot get it to

Hello, I can not seem to be able to get StripMSD and Prefix to work for me in extensions.conf. This is an example of what I have: exten => _050.,1,StripMSD,1 exten => _50.,Prefix,01051 exten => _001051.,1,Dial(${TRUNK2}/${EXTEN}) exten => _001051.,2,Busy exten => _001051.,102,Busy What I want to achieve is to call 001051501657887 on TRUNK2 when dialing 0501657887. dialing

Hello, we have a PRI (E1) to a carrier and a second one to a legacy PBX: DTAG ---pri---- * ------ Hicmo (PSTN) | | Sip and more Many normal inbound calls are direcly routed to the hicom. Outbound calls from the Hicom go through LCR and then to PSTN. Inbound faxes are working, but outbound faxes from hicom to pstn are

hi, we are currently planning are large site which will migrate from an old siemens hicom pbx to asterisk. it will be a slow migration, the asterisk server will be inserted between the telco E1 and the hicom. new phones will be sip ones. the customer has several fax machines and analog phones (some of them have to be explosion-proof). around 50 analog ports in total are needed. as we are in

Hi, Is is possible to implement this: Hicom150 --- BRI (QSIG) ---- Asterisk I've been reading Siemens documentation and they say: "Digital nailed connections Corporate communication networks can be implemented over digital S0 or S2M nailed connections between several Hicom systems using the CorNet N protocol and between Hicom and non-Siemens systems using the QSig protocol. The