similar to: Modifying dialplan for DUNDi compatibility

Hello, I am new on asterisk and do some tests on freepbx. I have 2 SIP provider: Provider1: In-/Out- Flatrate, only 1 Number Provider2: Incoming Flatrate, Outgoing Cost depend on destination, 3 numbers On Asterisk site i have 3 phones (branch ??, don't know how its called in asterisk) Is it possible to do something like: Phone 1: Incoming Call: Number1/Provider1 Outgoing Call:

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 asterisk-users-bounces at lists.digium.com wrote on 12/14/2017 09:36:06 AM: > From: "basti" <mailinglist at unix-solution.de> > To: asterisk-users at lists.digium.com > Date: 12/14/2017 09:36 AM > Subject: Re: [asterisk-users] Rewrite Outgoing Number > Sent by: asterisk-users-bounces at

Hi :) I hope someone has a hint concerning Early Media. The situation: My Asterisk is connected to small local carrier who works with several SIP servers. I traced some SIP headers and find something like this: Via: SIP/2.0 UDP sip1.provider1.de In the SDP part I found something like this: o=- 2268929 0 IN IP4 sip2.provider1.de c=IN IP4 sip2.provider1.de If I send

Hello everyone. I've stumbled upon a strange networking issue with multiple interfaces on CentOS 5. The network setup is just like the diagram in http://lartc.org/howto/lartc.rpdb.multiple-links.html It looks like linux is not routing correctly outgoing packets on interfaces different from the one of the default gateway, but instead broadcasts an ARP request on the link, looking for the

Greetings list, Over the last few days I've been gearing up to replace a couple of our servers with 1.6 as something of a testbed, but I'm encountering a few problems, and wondering if anyone can help... In extensions.conf, there are a number of contexts defined for each group of users, along the lines of: [groupa] [groupb] etc. In each of those, there's a command include =>

I am using GotoIf all over the place in 1.4.8 but for some reason, the following in my dial plan: ############################################################# exten => _1NXXNXXXXXX,1,GotoIf([${EXTEN} = "15554441212"]?100) exten => _1NXXNXXXXXX,n,Dial(SIP/provider1/${EXTEN},60) exten => _1NXXNXXXXXX,n,Dial(SIP/provider2/${EXTEN},60) exten => _1NXXNXXXXXX,n,Hangup exten =>

How do you setup the timing in Meetme conference? I have a x100p and tdm4x card. When I dialing to my conference I get a request to schedule in the past error message. thanks -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Saturday, April 10, 2004 10:48 AM To:

Hello In the LARCT how-to subitem: 4.2.2. Load balancing the following phrase says: "" Instead of choosing one of the two providers as your default route, you now set up the default route to be a multipath route. In the default kernel this will balance routes over the two providers. It is done as follows (once more building on the example in the section on split-access): ip

Hi, I put on sip.conf the following line #include "sip.d/*.conf" inside I have files like that provider1.conf provider2.conf But asterisk does not want to load it This is the error Sep 1 13:18:35 VERBOSE[8756]: == Parsing '/etc/asterisk/sip.d/*.conf': Sep 1 13:18:35 VERBOSE[8756]: == Parsing '/etc/asterisk/sip.d/*.conf': Not found (No such file or directory) this

Hi, I am having a few dial plan problems which I wondered if anyone would be able to help with. Firstly, I wanted to send 0800 calls through 1 sip provider and other 08xx calls through another. I have this: exten => _0800.,1,Dial(SIP/${EXTEN}@provider1,30) exten => _0800.,2,Congestion exten => _08.,1,Dial(SIP/${EXTEN}@provider2,30) exten => _08.,2,Congestion However,

Hi! I'm really hope you can help me solve a little mystery, the mystery is probably just my misunderstanding ! sorry... I've got an iaxy talking to my * box which connects to two providers. I'm running the stable release of the pbx. The only thing is that when dialling from the iaxy the ringing tone isn't heard while calling someone - you just hear silence then, they either

Hello, I would like to known how to use DUNDi with a Lua dialplan ? In extensions.conf, we should do like these: |[lookupdundi] switch => DUNDi/priv [internal] include => dundiextens include => lookupdundi exten => _XXXX,2,NoOp(calling ${EXTEN}) exten => _XXXX,n,Dial(SIP/${EXTEN}) exten => _XXXX,n,Hangup()| priority 1 is either defined in dundiextens (local registered

Hi, I''ve been experimenting with ip route for the last few days to get load sharing accross 2 providers working. While it works most of the time, on a few occasions, packets are routed to the wrong interface. I''m not sure to understand rules and routes traversal correctly (I couldn''t find answers in the howto). So, here are my questions: 1. How does the rule

I'm looking at finding a way for my Macro(dundi-dundi-test,${ENTEN}) when I dial out on the dundi-test network to return a +101 to my [dundi-test-out] context, if the number being dialed on the dundi-test network does not exist, then I will route the call out using my pstn or voip connection i have. I have a feeling it will have to be the switch => DUNDi/dundi-test that will have to return

Hello, I have the following setup: sip phones <->SER <-> asterisk <-> voip provider1 <-> voip provider2 i got a toll-free DID from voipprovider1 to allow people from outside to call into asterisk, get authenticated, and use voipprovider2 to call out (kind of a primitive calling card app). anyway, voiprovider is giving my

Dear community, Please help. I've been looking around the internet (and in this great forum) for help with DUNDi setup between servers (I'm using Elastix) and while I can get my servers to lookup extensions on each other very well, I have not been able to successfully make calls between servers. For my test environment, I have 3 servers setup for now, and these are the steps I've

>From the reading and testing I have done it doesn't look like SIP supports a username and password in the Dial string. I currently have the following mapping. priv => dundi-extens,0,SIP, dundi:pass at 1.1.1.1/${NUMBER},nounsolicited,nocomunsolicit,nopartial On the sending side I see NOTICE[31598] chan_sip.c: Conflicting extension values given. Using 'dundi' and not

I have three Asterisk boxes. Each has the following in dundi.conf: 180net => dundi_local,0,IAX,dundi:${SECRET}@${IPADDR}/${NUMBER},nopartial 180q => dundi_q_pbx1,1,IAX,dundi:${SECRET}@${IPADDR}/${NUMBER},nopartial 180q => dundi_q_pbx2,2,IAX,dundi:${SECRET}@${IPADDR}/${NUMBER},nopartial 180q => dundi_q_pbx3,3,IAX,dundi:${SECRET}@${IPADDR}/${NUMBER},nopartial My iax.conf on all three

Hi. I am trying to pass a variable from one Asterisk PBX to another. I'm using DUNDi with IAX2. Is there a way to do it? I tried the following but it fails. On peer1: [dundi-outgoing] switch => DUNDI/priv exten => s,1,Set(CDR(userfield)=test) exten => s,2,Set(DUNDIVAR=${ARG1}#TEST) exten => s,3,NoOp(Passing ${DUNDIVAR} to DUNDi peer.) exten => s,4,Goto(${DUNDIVAR},1) On

Hi all, I'm new in asterisk and I got to set up a dundi config for my work. I have 2 PBX for the test, the two PBX are in the same local network PBX A : 192.168.199.23 PBX B : 192.168.199.21 my config files : (on PBX B , the config files on PBX A looks like it) /etc/asterisk/dundi.conf [general] bind=192.168.199.21 port=4520 cachetime=5 ttl=32 autokill=yes entityid=00:30:18:4C:33:53