similar to: Is it possible to have two trunks between two Asterisk boxes ?

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

-----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

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

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:

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:

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

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

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

Hi all, I'm trying to resolve a weird issue with SIP routing. I have a number of SIP trunks, from a selection of providers, all of which are registered in sip.conf: [general] context=default allowguest=no allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=yes tcpbindaddr=0.0.0.0 transport=udp bindport=15060 srvlookup=yes allowsubscribe=yes

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.

Is there any way to have users be able to turn on or off call forwarding at the asterisk server, so they can configure their own forwarding number and enable/disable it? Hopefully, with the added benefit that it will remain on between server reloads and restarts? I have written a hack -- a AGI script to do various checking, and if the destination is "ok" set a database variable

Hello all, I have no problems setting up trunk groups in general, but is there a way to set up a trunk group for outbound calls that includes channels on multiple servers? I might have missed something somewhere, but I couldn't find any reading about this topic. Thanks! Sean

I''ve been wrestling with this all night, I''m hoping someone can help. I followed the exact steps in: http://wiki.rubyonrails.org/rails/pages/HowtoUseRailsWithSubversion ..but when I open a new ''Checkout project from SVN'' in RadRails, it opens up the second level dirs as the project dirs (ie. app, log, script, etc), leaving me with a mess of projects. I redid

Hi All, I'm connecting to my carrier which requires setting of outboundproxy. There has been few cases where the proxy server failed due to network issues and required us to use a secondary one. Is there a timeout or qualify setting for outboundproxy setting in sip.conf? I do appreciate if anyone can help please. Thank you -Abeed -------------- next part -------------- An HTML attachment

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

Hey List, I'm using Samba 3.0.24 and OpenLDAP 2.3.30 (with the ppolicy and smbk5pwd overlays). While testing Samba as a PDC with an OpenLDAP backend, I've hit a snag on password change. I currently have the following in my smb.conf related to password changes: passwd program = /usr/bin/ldappasswd -x -W -S -D uid=%u,ou=Users,dc=example,dc=com passwd chat = "*Enter

Using pjsip 2.7.2 on Asterisk 15.5 Really struggling to make sense of translating these old 1.8 SIP instructions into a neat pjsip_wizard conf suitable for 2018 http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk#Version-18 In pjsip_wizard.conf, I have the following, which seems to get me registered, and it responds to an incoming call, but I always get this: [Jul 28 18:32:29]

Hello list, I try to setup an asterisk-server with different SIP-Peers to PSTN. The Peer are working and configured in sip.conf: [peer1] type=peer host=10.10.10.1 [peer2] type=peer host=10.10.10.2 Now dialout is no problem. Extensions.conf says: exten => _0Z.,1,Dial(SIP/49${EXTEN:1}@peer1,30) But how can I setup a failure-route if the SIP-Proxy "peer1" ist not

On 2 February 2015 at 15:12, Joshua Colp <jcolp at digium.com> wrote: > Sunny wrote: > >> Hello, >> >> >> In Asterisk 11 it is possible to set extensions on DB table (sipppers) >> and also in sip.conf. >> >> But in Asterisk 13 apparently this is not possible: as I tried to set in >> ps_endpoints and also in pjsip.conf but only the