similar to: different codec for different extensions

Hi everybody, >From asterisk CLI we can add extensions in dial-plan dynamically using "dialplan add extension" command. but how we can dynamically create a context in dialplan. is that possible? Nasir Iqbal

@Olivior I agree that seting up WebRTC is hard, however when done it is smooth to use. For replication you can build RPMs with working configurations. Regarding stability, it is not being used widly, so can't say it is mature. However we have no complain so far regarding audio or connectivity. sometime we provide support for "allow media / mic" type issues, but you know it is

Hi Olivior, We have recently worked on a WebRTC based agent panel. As based on my experience I think that WebRTC based phones are far better and cheaper then those soft / sip phone. the big plus is that they are easy to customize and developer can use the power of browser and web to build / offer features which are not possible with regular phones. Regarding your concern about BLF or call

We tried to use the CURL fn to POST json, but it's sent as form data and there seems no support for changing the Content-Type header. We switched to invoking curl in the shell. All the documentation I could find says there is just one parameter for the url and an optional second for POST body. Is there an undocumented way to set Content-Type? -------------- next part -------------- An HTML

Hi, folks. I'm having a heck of a time trying to get outgoing T38 faxing (I don't need inbound right now) working with FFA and Gafachi. G711 faxing works (as well as can be expected over the internet), but I want the higher reliability of T38. I'm running Asterisk 10-beta1. When I drop my callfile in to make the call, I get this: -- Attempting call on SIP/18884732963 at

Hi, I have followed steps which were mentioned on forum and given below. Still couldn't get speex working. On test calls getting error "chan_sip.c: sip_call: No audio format found to offer." # yum install speex # yum install speex-devel # cd /usr/src/asterisk # make clean # make # service asterisk stop # make install # service asterisk start Also, it is not

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 All, I setup auto dial on my asterisk server. The problem is asterisk does not wait for called party to answer the call but proceed to process the extension specifed in my .call file My sample call file : hannel: local/0124787924@outbound-reminder MaxRetries: 5 RetryTime: 300 WaitTime: 40 Account: Reminder context: remindem extension: s priority: 1 Set: MSG=0135.20070601.0124787924 Set:

I am looking for a simple way to send occasional faxes via the FXO port on my SPA3102 -- without having to connect a fax modem to an ATA. In an ideal world, this would be some sort of "softfax" that runs on my Linux desktop and talks (via Asterisk) to the SPA3102 with T.38. This is one of those things that I thought would be relatively straightforward, but a couple of hours of Googling

I understand that Asterisk 1.4 should support T.38 pass-through, but I need Asterisk (or something on the Asterisk box) to act as a T.38 endpoint. Judging from the unclaimed $12,000USD bounty, it doesn't appear that Asterisk itself can do this. http://www.voip-info.org/wiki-Asterisk+T.38+Bounty Does anyone have any experience with this, or are able to point to an example of this working?

Hi all, Can some one suggest me an IAX client for Linux and Windows? I used KIAX once, but know it seems complicated to have it working on Ubuntu. Thanks. Ronaldo.

I'm trying to add MySQL CDR recording in Asterisk 1.4.6. I'm following the instructions posted here: http://www.voip-info.org/wiki-Asterisk+cdr+mysql I have MySQL installed and it works fine - starts on stratup, I can create DBs, tables and so on and I can connect through php. rpm -qa indicates: MySQL-server-5.0.22-0 MySQL-devel-5.0.22-0 MySQL-client-5.0.22-0 However I still get XXX

Can asterisk log the registration date/time in a database? Is there a standard option to do this? I know it being logged in the asterisks 'full' (debug) log and we are probably able to script something with the API interface but there might be somewhat easier if there is a option to make asterisk log this information directly into a database. Thanks in advance, Bram

Greetings list, Whilst running through a routine check of some CDRs, I've noticed that the originating channel's accountcode isn't preserved on creating a local channel. For example, if we start with: exten => 123,1,Set(CDR(accountcode)=foo) exten => 123,n,Queue(bar,nrtw,,,) And the queue 'bar' is defined as follows: [bar] member => Local/456 at outbound member

hi does any body know which version combination of spandsp/tx_fax/rx_fax will work with * 1.2.24? i tried different combo. they're either seg fault during runtime or won't compile. very frustrated :/ p.s. i know. hylafax/iaxmodem is far more stable. but i have specific reasons to use rx_fax. -- Edwin Lam <edwin.lam at officegeneral.com> Systems Engineer, Office General, Inc.

Hello Everyone, I am trying to diagnose issue with my IAX2 extension not working. When I have iax2 set debug on all I see is this: *Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ * * Timestamp: 00003ms SCall: 00130 DCall: 00000 [64.229.229.111:64823]* * USERNAME : 100* * REFRESH : 60* * * *Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type:

Hi, I have xlite registered with a user. Now i dial an extension say 1500 which has the dial plan as follows. exten==>1500,1,AGI("localhost//hello.agi" So when i dial extenstion 1500 the script hello.agi is invoked which in turn plays a welcome message. I now want to transfer the call now to operator. How can i achieve this???Please help me in this regard as this is very urgent.

Hi Vardan I did same as you told and deleted the SIP information in Astdb and restarted asterisk. but the result was same. as you said there might be mistake in sip.conf so i am pasting both servers configuration here.. 1- nasir.server.com [abc] username=abc type=friend secret=mysecret nat=yes mailbox=12234568 incominglimit=2 outgoinglimit=2 host=dynamic dtmfmode=rfc2833 context=payasyougo

Yes this scenario works on my 2 systems which are at LAN. I made one system as server (192.168.0.20) and registered from other system... it is fine but now there is a different scene. actually there is a registered user named abc at system1 (192.168.0.20) having context [payasyougo] which is used to do outbound calls. we want to use this user's context and account so that when we register

Right, I'm baffled. I have: exten => s,1,MYSQL(Connect DB1 127.0.0.1 geraint xxx amis2) exten => s,n,MYSQL(Query NORESULT ${DB1} INSERT\ INTO\ recordings\ (caller_number\,called_number\,date_created\,date_started\,in_use\,server_id)\ VALUES\ (\'${CALLERID(number)}\'\,\'${ARG1}\'\,NOW()\,NOW()\,\'Yes\'\,12)) exten => s,n,MYSQL(Query RESULT1 ${DB1} SELECT\