similar to: asterisk-api

hi everybody, Is there SIP client which work with Asterisk and can be embedded in a HTML page ? Thanks Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031029/226a8b1b/attachment.htm

Thanks Stefan for input. I know that there is a "hangup" action in Asterisk Manager API. I am looking for "hold and retrive" commend. I search google and find that redirecting to parkslot can work. If I have a PSTN call connecting to Asterisk and then to a SIP extension, there are two connections here. If I redirect one channel to parkslot, another channel will automatically

Hi I?m a new user and I do test with my hardware . I have a x100p and telephone vozip. And when I run this command asterisk ?vvvvc for to test it . My computer show it ?warning? [chan_iax.so] => (Inter Asterisk eXchange) == Manager registered action IAX1peers == Parsing '/etc/asterisk/iax1.conf': Not found (No such file or directory) Apr 2 07:45:12 ERROR[16384]:

I figured it out. The old system (Televantage 3 and 4 I think) has limited specifications on the T1. After setting up the system, I was able to send and recieve calls. I still have some work to do like figuring out faxing and a floating receptionist, but this is a nice start. ----------------------------------------------------Televantage T1 Requirements: Framing: D4 Superframe or Extended

Hello, I have few questions about Asterisk. I installed Asterisk from CVS on FreeBSD and I made cvsup 2 days ago. 1.I couldn't find Asterisk version using "asterisk -V" command. How can I to find version information? 2.I am using Wildcard X101P (FXO) and Wildcard TDM400P REV E/F (FXS)on it. I tried Asterisk CallerID feature, but unable to get it. I tried callerid signaling V23,

Hi i want use the API on my asterisk 1.6, but i have a small problems : In extension, i start it : exten => _X.,3,AGI(My-Script.agi) The perl agi file are started without problems but i want get into this script a lot of variable: Type (SIP or IAX) src (from cdr) but that's don't work: use Asterisk::AGI; use lib "/var/lib/asterisk/agi-bin"; $AGI = new

Hi, I'm playing with the agent/queue system. Everything work well with v1.0.3. but I want the 'Action: Agents' in the manager API that is only on the CVS version. So i switched to, but now the Queue/Agent system barely work. (my agent don't get the call) Where I can get a 'stable' CVS version? Or maybe, how I can solve my Queue/Problem? here is the detail: 1. I can

Hi All! Let me explain the problem. When using the Originate? command from the manager api, the dialstatus variable returns results? for whichever phone picks up first, and in this case it is the IAX/2? connection. It doesn't matter if Zap/G2/XXXXXXX is set as the channel,? or an extension either. What I am ultimately trying to do is get the? dialstatus of the Zap/X/XXXXXXX channel, i.e.,

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I want to have the ability to have anonymous SIP calls hit my server but I want to run it on different ports and create an SRV record for my target domain. My understanding of SIP is limited, but in a nutshell I want to accomplish the following: - - run SIP signaling on port 6200 - - create RTP ports on 6201-62XX Do I really need 10k ports open for

The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available security releases are released as versions 1.8.28-cert3, 11.6-cert8, 1.8.32.1, 11.14.1, 12.7.1, and 13.0.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of these

The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available security releases are released as versions 1.8.28-cert3, 11.6-cert8, 1.8.32.1, 11.14.1, 12.7.1, and 13.0.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of these

Hello asterisk-users, I setup my asterisk to support several features like automon,blindxfer,atxfer,parkcall etc. by using features.conf and the global variable DYNAMIC_FEATURES=automon#blindxfer#atxfer#parkcall#disconnect in extension.conf. Every Dial() command in my diaplan has the appropriate parameters out of {tTkWwW}. For calls from my SIP phones everything works fine. Pressing #1 will

[moving to asterisk-users by request] On Tue, Jan 27, 2009 at 12:56 AM, John Todd <jtodd at digium.com> wrote: > > On Jan 26, 2009, at 7:38 PM, James Lamanna wrote: > >>> On Jan 26, 2009, at 8:53 PM, James Lamanna wrote: >>> >>>> Hi, >>>> Is it just me, or does DTMF queuing not work properly? >>>> I'm consistently faced with

Hello! I am having problems figuring out how to do something, and any help would be much appreciated. I would like to use the manager API to take an existing call on a specific SIP extension, dial and conference in a third party. From what I can tell, the way to do this would be to take the two original parties on the call and stick them in a meetme room using Redirect with ExtraChannel,

Hi all, I want to initiate a call from the asterisk to an extension, where I will forward the asterisk side to another extension later (to the conference extension). I can initiate a call uning originate call from an extension to the desired extension, but it would need someone from the originator extension to answer the phone. How can i register an extension to asterisk where it

Hi, I'm running Asterisk 1.4.2 on a Debian GNU/Linux and a Digium TE210P card. I have one E1 with MFCR2 Signaling. I compiled asterisk + libunicall, and I can make calls over E1, but can't receive. Here the CLI when I make a call: -- Executing [006191642208@ps5:1] Dial("SIP/23-081cbc40", "Unicall/g1/91642208|50") in new stack -- Called g1/91642208 [May 14

I am having a problem trying to do inband DTMF passthru via asterisk. My setup: PSTN gateway MAXTNT v11.0 SIP (T1 PRI/NT2) Asterisk HEAD or v1.0 makes no difference (I am using HEAD mostly) 12/10/04 and 01/17/05 (no difference) CAC ABII-T100P to/from analog lines to/from asterisk BTW, I have used a ABI and it works just like the ABII with asterisk. What I am seeing is: I make a call from a

I've setup a SIP channel between two Asterisk box, and use Manager API to generate some calls. It's working quite fine, except this message (on the caller-side) : Jan 10 18:18:09 WARNING[25046]: chan_sip.c:6805 handle_response: Forbidden - wrong password on authentication for INVITE to '"sip1" <sip:asterisk@192.168.1.200>;tag=as77e9ebbb' But the call is going

Been researching connecting over TCP\IP to Asterisk Manager API to initiate several concurrent calls to dial out. Prefer not to generate ASCII .call files. Question : I read in places that you use "originate" command and wait for an event back, does that mean you cannot place another "originate" until the event comes back ? Is it true that multiple API connections to Asterisk

Hi all This is a followup on my post "Asterisk 13 - system() dialplan app cannot call bash scripts" from yesterday I've given up trying to use system() to call BASH scripts with parameters from Asterisk 13. Turned out under Asterisk 13.22.0 System() DOES work, but only if you do NOT attempt to pass any parameters to the called script. This works, and reliably calls the script: