similar to: Billing/Call Control engine : AGI scripts/ AstMan API

Hello All, Is CALLERID() setting broken in 1.4.4? My small dialplan : [testclid] exten => _0.,1,Set(CALLERID(all)=Ben Jacob <988077>) exten => _0.,n,Dial(SIP/${EXTEN}) Correct me if I am wrong, Set(CALLERID(all) above supposed to change the display name as above(Ben Jacob) and change the From URI to 988077 at myip?? As of now, only the _display name_ is being replaced, but not the

Hello ppl, Am looking at some PSTN termination providers in US. If this question has been repeated, please point me to the correct link, as I've tried searching the archives but have been unsuccesful so far. I have come across quite a few companies which provide the same, such as : Iconnecthere <http://www.iconnecthere.com> Vonage <http://www.vonage.com> Teliax

Hello good ppl, Any way of stopping asterisk writing into syslogs or any other file, if I set verbose 6 on the console? All I want is the verbose output only on the console, nowhere else. My logger.conf says : console=> notice,error ;messages => notice,warning,error Thanks in advance. - Benjamin Jacob. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential

Hello All, I am using ODBC storage for voicemail on my asterisk box. I want to have a common voicemail box for different extensions. I know how to do that, but the question troubling me is how and where do I store the the extension name for which a particular voicemail was left. e.g. extensions 1000, 1001, 1002 all using the same voicemailbox 55555. Now, when someone calls 1000, and leaves a

Hello good ppl, A couple of questions for multiple pbxes 1. Is it possible to support multiple pbxes in one Asterisk box(using contexts, etc.)? 2. Can we use the "domain" field in sip.conf to specify the different domains for sip users, having one domain for each pbx? I just tried registering two xlites, with different domain names (with the same specified in sip.conf). But, Asterisk

Dear All, I have a requirement to 'originate' a number of calls to various external users from within a conference room, so that the end users does not pay for the call. I know that within Astman I can define an extension and then originate the call from that extension. Can I define a conference room (how would I configure that on astman? What channel would it use?) and then generate a

Has anyone used Gastman/Astman successfully? I have it up and running (Gastman win32), but have a problem with the creation of end stations on the map. I'm not sure of the format of the extension to use when creating a end station icon. Services like Conference bridge and Musichonhold seem to work ok (I use 555@mainmenu and 666@mainmenu) for the Icon extensions. IAX softphone seems to work

Is this getting through?? EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions

Hello All, I've got the following warning messages a couple of days back: /chan_sip.c: Maximum retries exceeded on transmission <SIPcallId> for seqno 1 (Critical Response). /Have got the warnings repeatedly for one Callid. If maximum retries have exceeded why should it give me those warnings again n again for the same callid, with a gap 4 seconds between each warning. The callids

Hello ppl, Sorry to re-post it, but kinda these issues are getting on my nerves. I tried Set(CALLERID(num)=7329) on 1.2.12, which works fine, but not on 1.4.4. The problem : 1. I receive call from caller 'AAA' on my number, 'BBB' which is on my Asterisk box. 2. I have to redirect the call to some other number, say, my cell num - 'CCC'. 3. My PSTN provider wants the

has any body considered using astman/gastman to show a) sip/iax etc registry status with other * b) manage multiple * boxes would this be any use, i was just thinking if some tech support has to handle multiple * for one organization, it may be worthwhile to have mutiple * boxen become part of one REALM and manage that realm, or maybe some other way, but i'm sure we can find someone to

Hello ppl, Searched all over, but couldn't find anything conclusive. Does an off-the-shelf version of Asterisk run without any issues on a 64-bit machine? Does anyone have any 'conclusive' figures? Apologies if this is a repeat question. Would appreciate if I could be redirected to the appropriate link. cheerz - Ben. EMAIL DISCLAIMER : This email and any files transmitted with it

Hi All, I want to track a call that is originated using originate AMI command through AstManProxy server. I m using AstManProxy server and I developed an AstManProxy client. By using my AstManClient program I can able to login AstManProxy server. Now I can able to issue/send originate command to generate a call but I m very confuse that I cannot able to track my call. The AMI events were

Hi, I'm working on a 500 seats Asterisk project. I'm wondering whether or not I should consider using Asterisk Realtime and a database to manage phones registrations. Stories in Dev mailing list say Realtime is mis-used or should be improved. So, what's the bottom line ? Can I consider anything I can do with .conf files can be done with a combination of .conf files and Realtime.

Hi all, Any one know how to increase the Linux limit? I am hiting a wall on 200 calls playing files at the same time. From Asterisk console, I am getting messages like Sip_request_call: Unable to build sip pvt data for "asterisk1/700" Too many open files Is this a limit of my Linux box? I only have 512MB of ram. Will increase it to 2G help or I have to change some configuration in

Hello ppl, Any way to specify alphabetical exten patterns in the dialplans on Asterisk? All my users would have alpha/numerical ids. I don't want to add a line for every user in my dialplans. I searched around, but couldn't get anything useful. Any way to get around this? Thanks in advance - Benjamin Jacob. EMAIL DISCLAIMER : This email and any files transmitted with it are

Greetings everyone, I'm pleased to announce the release of Astmanproxy 1.20, the fast, flexible proxy server for Asterisk's Manager Interface. Astmanproxy allows you to communicate with multiple Asterisk boxes from a single point of contact using a variety of I/O formats, now including support for XML, HTTP, HTTPS, SSL, CSV, and the Asterisk-native standard format. Astmanproxy is

Hi, I am trying to get the extension status (weather it has dialed outgoing call via SIP or IAX2), using the following piece of code however it always returns -1 on all the extensions (valid/invalid). Am i missing something ? Any help. Thanks ----------------------------------- #!/usr/bin/perl use Asterisk::Manager; use lib './lib', '../lib'; $|++; my $astman = new

Hello, I am not an asterisk expert but i am developing a web application that is using asterisk. I would like to know if it is possible to configure a Manager to only monitor a special extension, and of course how to do that. The application is written in java and is using asterisk-java. Right now i have one manager that i am connected to and i receive all the events but i would like to have

Hi, We currently fire multiple HTTP requests (via multi-curl) to the AJAM interface in order to place calls. We are finding Asterisk has it's limits however, and I've found astmanproxy recommended for helping maintain the connections. This would prove particularly useful with multiple servers of course. However, in testing astmanproxy crashes with buffer overflows. This leads to the