similar to: Print CALLERID in CLI during "pri debug "

Hi all, I was looking at a way to add the caller id to the outgoing calls (which are made using .call files) using asterisk. Any ideas how to do this ? Currently I get 'Unknown' number displayed on my phone when asterisk makes an outgoing call. Also using something like this is not working as it still displays unknown number. I want set the callerid on the 1.call which is made. exten

Hi Asterisk Users, I was wondering why a call that is received from Asterisk shows a caller ID 'Unknown' . So here is the scenario, 'A' calls 'Asterisk'. 'A' joins a conference on 'Asterisk'. 'Asterisk' calls 'B'. 'B' gets joined to the same conference also. 'B' somehow receives the caller ID 'Unknown' and not the

Hi * Users, What is the way from the dial-plan to get the IP address of an incoming or outgoing SIP call? I can see the IP address of the SIP call using 'sip show peers' from the CLI. Thanks Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998

Hi, I have a strange problem. I have a TE110p digium card. I want to dial 19173995791 when any incoming call comes in. What is happening is that when I dial 19173-995791. Asterisk picks up the first 5 digits assuming it is the extension and appends 212-85 (here in the university most numbers start with this) in front . Therefore I get connected to some random number 212-85-(19173) (where the

Hello * users, I know that passing variable in the AGI script is by exten => _.,1,AGI(simple_c_prgm|123|789) ; 123, 789 are arguments being passed and simple_c_prgm is C code Now how will I receive these variables within C code ? Is it by the same way arguments are passed in command line to C by using argc and argv or there is more to be done than that? Thanks Regards -- Arpit Mehta

Hi all, Does Asterisk contain a full fledged ISDN packet sniffer. By giving the command " pri intense debug span 1 " , does it debug every packet received (control and voice/data packets) ? Thanks -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998

Hi all, My scenario is such that I have three users connected to a conference. CLI> meetme list 1234 User #: 01 9176502096 <no name> Channel: Zap/23-1 (unmonitored)00:00:32 User #: 02 john john Channel: SIP/john-b7800468 (unmonitored) 00:00:28 User #: 03 6463875998 <no name> Channel: Zap/22-1 (unmonitored)00:00:19 3 users in that

Hi all, i need to study the SIP protocol. can anybody tell me about any ebook which could halp me understand the sip protocol, architecture, and how to read and understand the sip signalling when i use "sip debug" in asterisk? -- Rizwan Hisham Software Engineer AXVOICE Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I was just wondering how would the application be where the Asterisk calls a number and that number joins the conference as soon as the call connects. There would be only one conference already defined in meetme.conf and there is one person already joined the conference. Currently MeetMe requires a person dialing into it and the joining the conference. How could this be done using MeetMe or

Hello Asterisk Users, I wanted to know a simple way in which I could read some file from a console (say by using system command) and based on that either return true or false back to dialplan. Is there any built in command in Asterisk for that ? What are the options do I have ? Are there any sample code to do so ? Thanks a lot Regards -- Arpit Mehta Graduate Student Department of Computer

hi, This might be the most obvious thing to you. I need to change some parts of the source code of Asterisk. I was wondering if we have to compile the whole source code again everytime using the commands (which i think might take some time to compile again) cd /usr/src/asterisk-version make make install or is there a faster and better way to do things Thanks a lot for all the help i have

Hello, I have an ISDN-30 connection and a Digium TE-121 with VPMADT032 echo cancellation. Every 30-60 minutes I experience PRI dropping. @@@ /etc/zaptel.conf: loadzone=dk defaultzone=dk span=2,1,0,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-62 @@@ @@@ /etc/asterisk/zapata.conf [channels] switchtype=euroisdn usecallerid=yes group=2 signalling=pri_cpe context=incoming channel => 32-46

Hi, ? I'm trying to figure out what the issue is splicing Asterisk between our Telewest PRI and a GDK-186 with a PRI card. ? We're using the Digium TE405P ? Our telco provider is Telewest, and Telco directly into switch is fine. ? When I splice Asterisk in, I can make and receive calls from Asterisk extensions, I can make outbound calls from the GDK, but inbound calls do not seem to pass

Hey, I wrote yesterday about PRI dropping, which turned out to just be a regular reset of unused B-channels. This time there's a real issue. As noted earlier I have an ISDN-30 connection, a Digium TE-121 with VPMADT032 echo cancellation. These are my configurations files: == /etc/zaptel.conf loadzone=dk defaultzone=dk span=1,1,0,css,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 == ==

Hi, Is there a way to restart the DISA to the enter phone number? For instance, Bell Calling Cards let you hit # at any point which lets you enter another number to call. This is useful to reduce the number of digits dialed and to utilize per-minute calls. I was not able to find anything on the web. Thanks, -Ryan

Hi all, I have an asterisk-server (cvs-head from august) connected to a carrier's switch (DMS/Euroisdn) via a te410p, and I am having problems with DDI (standard 'official pstn' number plus extra digits for 'internal' use) Basically, when the entire number (including the extra digits) is dialled via a redial or a programmed key, I see the entire called party number (including

hello everyone, I have a trixbox server with an E1 card(not digium).It connects to an AVAYA pbx use E1. It works fine.But when i change the E1 card to digium TE220B,there is a problem. When sip extension A(on trixbox) call PSTN extension B(on avaya),A must wait longtime before B start to ring.From the log I find there are two times call. I don't know why the first request be rejected

Hello! Is it possible to connect Siemens HiPath 4000 to Asterisk? What equipment required on Siemens side? I mean IP not E1. Sorry for asking here. Siemens-related websites use "salesperson language". There is no technical information.

Is there a setting somewhere in * to define whether I am receiving callerID or true ANI? Global Crossing claims they are sending ANI but I dont think so. My understanding of ANI is that it is always sent, regardless if callerID is blocked. If I dial *67 and my DID, I get "Presentation: Presentation prohibited of network provided number" and no number. Before I call GC on Monday

I just downloaded the cvs version CVS-07/28/03-14:45:19 and now I cannot recieve the the calls from the zaptel interface which is a E100P with pri signaling. That is something with asterisk becouse rolling back to version from 06/23/03 using the new libpri and zaptel fixes the problem. Here is an exept from the config: [macro-stdexten]; ; ; Standard extension macro: ; ${ARG1} - Extension