similar to: voicemailmain()

Hi, I am setting up a small call center using *. I have ZAP setup for incoming calls and IAX setup for agents. Agents login using AgentCallbackLogin. When customers call, it's getting picked up and when queue is trying to call back the agents, I am getting error. I am using CVS HEAD, and updated just now. The error is: -- Executing Answer("Zap/1-1", "") in new

I have an Ubuntu system and went into Synaptic and checked asterisk for installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc and got the following output with several errors and notices. Do I need to do more or are these ok? I expected to have some conf files in /etc/asterisk but there is nothing there. Thanks! Created by Mark Spencer <markster@digium.com>

Hi, Ive been struggling with asterisk for a few days now. I understand pretty much how it works and how to tie things together (SIP -> SIP internally works fine etc), but just cannot get asterisk to work with an X100P clone (its a Ambient MD3200, if that means anything to you guys). I have tried (initially) asterisk 1.07 with zaptel 1.07, and now Asterisk CVS-HEAD with zaptel cvs. Both give

Please can someone help me with the following two error messages: Error 1. I have loaded the Zaptel dirvers and everything is ok with ztcfg. I have configured Zapata.conf and everthing looks good but it apears the Zap channels dont load when starting Asterisk. When I make a call to one of the fxs port I get the following error message. -- Executing Dial("SIP/39-b204",

Anyone have any idea what this is talking about. Here is my zapata.conf [channels] switchtype=5ess signalling=pri_cpe echocancel=yes echocancelwhenbridged=yes echotraining=400 callerid=asreceived group=1 context=default musiconhold=default faxdetect=incoming channel => 1-23 Here is my zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 # set this to 1-15,17-31 for E1 dchan=24 # set this to 16 for

I'm simply trying to forward calls to users who have the call forwarding feature enabled (FWD Status and FWD Ph Number kept in the astDB). The problem is that I want users to be able to forward calls to numbers that they would normally be allowed to dial within their own context. (I don't want a local call only person forwarding to a long dist number, for example.) I'm able to

Hi. I can't figure this one out. Hope someone can help me. root@pbx:# cat /etc/odbc.ini [Asterisk] Description=PostgreSQL asterisk Driver=PostgreSQL Trace=No TraceFile=/tmp/odbc.log Database=asterisk ServerName=localhost UserName=XXXX Password=XXXX Port=5432 Protocol=7.4 ReadOnly=No RowVersioning=No ShowSystemTables=Yes ShowOidColumn=Yes FakeOidIndex=Yes ConnSettings=

Hi all, Trying to do what should be a basic info retrieval from my asterisk network -- get the list of IAXPeers and break down by IP, name, and Status etc. Asterisk 1.0 Manager unfortunately. Using PHP. Easy to get the response -- but parsing it is impossible! That table format throws me for a loop ... Any idea of an easy way to seperate out the columns of that horrid 'table' reponse

Folks, I've just successfully set up Asterisk (as part of the Asterisk Management Portal installation). When I say "successfully", I mean that I have gone through all the steps detailed for the installation of AMP and not hit any snags there. I can connect to my asterisk server via ssh and can also connect via Http to the portal to change settings in AMP. Now I'm trying to

I'm going to try and lay out all the relevant information I have here in this one post. I can provide more info if necessary. ISSUE 1: Office A routinely looses connection to Office B. When typing IAX2 Show Peers, it will show as Unreachable. I issue IAX2 Reload and it will work again for 1-3 days (haven't narrowed the time down yet). My theory is that the DSL at Office2 is changing

Hello all, I hope this is not off-topic, if it is please let me know. I'm currently playing with an Asterisk at home, in order to get to know it's ins and outs. Very very impressive indeed. I've got it hooked up to my home phone line via a Wildcard clone board (Intel modem with Ambient chipset), and it works like a charm. Zaptel picks up the card as a generic clone, and works with it

I'm trying to isolate the IP address of inbound calls to my switch over IAX2. Is the proper way to get that information as follows: ${IAXPEER(IP)} If the caller was inbound via SIP, this works: ${SIPCHANINFO(PEERIP)} So I'm looking to return the IP address of the caller via IAX2. Thanks Lee -------------- next part -------------- An HTML attachment was

Is there an advantage to having a Queue members URI in the form: SIP/User (or indeed IAX2/User) Over Local/<number>@context ? I know that the latter will allow you to do things like set counting logic etc. through dialplan operations, but the former appears to be a more direct route to calling the party. (and if need be, there is the ability in queues to run a script on connection iirc).

I have a brand new IAXy I'm playing with. I do not get a dialtone on the phone, or any response at ll on the phone. No sound, no dialing, no ringing. The phone and wire are tested and known to be good. I think I have it setup correctly. When I give the iaxprov command I get this: #iaxyprov 192.168.1.90 iaxy.conf 02: c0 a8 01 5a 05: 11 d9 03: ff ff ff 00 04:

According to my IAX-provider, an account has been created for me on their Asterisk-server... But the Asterisk CLI tells me this : asterisk*CLI> iax2 reload == Parsing '/etc/asterisk/iax.conf': Found [Apr 30 20:51:30] NOTICE[6391]: chan_iax2.c:10124 set_config: Ignoring bindport on reload [Apr 30 20:51:30] NOTICE[6391]: chan_iax2.c:10183 set_config: Ignoring bindaddr on reload

How do I get Gnophone to register to my Asterisk server? I have set up iax. conf as follows: [tim] type=friend ;username=tstornes host=dynamic ;defaultip=207.194.60.56 secret=1111 context=from-iax callerid => "Tim" <5000> auth=plaintext qualify=10 permit=0.0.0.0/0.0.0.0 and extensions.conf includes a section in the context from-iax: exten => 5000,1,Dial(IAX/tim/s|100|r)

Hi I have been using the CHANNEL variable as a way of checking if a user is allowed to make outgoing calls, and what their source caller ID should be (these values are in a database). This works all of the time with SIP and most of the time with IAX, however sometimes with IAX the channel variable seems to be wrong. I have been using Zoiper as my IAX client and Asterisk 1.6.2.0-rc6. For the sake

I am having a really hard time getting gnophone working with asterisk. Gnophone tries to register with my server but there is no response. I can direct incoming calls to gnophone but if gnophone answers them, asterisk does not recognize it. Here is my configuration: iax.conf [jambo] type=user host=dynamic defaultip=136.159.99.100 permit=136.159.99.100 username=jambo secret=fubar

Hi All, I'm trying to achieve the following: - If <sip/iax user> logs in from home, they can dial internal extensions only (this is to avoid employees going wild on local/mobile calls from home) - If <sip/iax user> logs in from the office, they can call anyone they want. Since I have my users defined in an LDAP tree, I'd like to stick to one-account-per-user (each account is

Hello- For maintenance purposes, if possible I'd like to use the same iax.conf file in several different asterisk systems. However, on one of the systems only, I would like to include an IAX "register" command to another external system. Within iax.conf or other configuration files (other than extensions.conf), is there a way of determining what system I'm running on,