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Hi folks, After a fairly extensive Google trawl, I don't think the following is possible but would appreciate confirmation from anyone else who has tried something similar. I have an AGI (not particularly relevant) which is executed when someone calls into a specific extension. This AGI finds a suitable 'agent' (not actually a queuing system in the Asterisk Queue sense) and Dial()s

My users dial *120 get to an IVR menu that plays their balance and then ask them for a voucher. Ater the balance is played and the request for the voucher is played the user don't hear any other audio from the asterisk box. I can see the asterisk server playing the files to ask for the voucher again but the user cannot hear any thing. Has any one seens this issue with IVRs. I notice a

Hi all, as in last mail, i've installed Asterisk from CVS and AMP to manage it. I've made 4 extensions: moloch*CLI> sip show peers Name/username Host Dyn Nat ACL Mask Port Status 204/204 (Unspecified) D 255.255.255.255 0 UNKNOWN 203/203 192.167.125.9 D 255.255.255.255 5062 OK (3 ms) 202/202

Hi, I am facing a (for me) strange problem. When placing a SIP-Call I normally get connected and the dialplan is executed. The Call-Flow is controlled by a PHP-Agi-Script. The script answers the call (via AGI-Command) and a simple voicefile is played. SOMETIMES(!) I get disconnected immediately after the Answer - without any reason. This happens about all fifth call. Later on you will find

Hi all, i've installed Asterisk with AMP. I've created 4 extensions (for 4 SIP phones) and this is what my "sip show users" return: moloch*CLI> sip show users Username Secret Accountcode Def.Context ACL NAT 204 moira from-internal No No 203 michele from-internal No

Hi. I've found a problem when I pickup a remote sip phone with *8. There're both budgetones 102 and are both in the same group. When one sip phone is ringing, I can pickup the call from another sip phone, but the first one keeps playing a loud busy signal... that don't go away until I receive another call or go off hook and then on hook on the first phone. I think that could be a

Hi - I've recently bought a mediatrix 1204 and have had a complete nightmare getting it up and running with an asterisk@home setup. I know this isn't a mediatrix list but I'm at my wits end and the support with this product is atrocious. (mine was even shipped with firmware that was incompatible with the win32 software it came with so I wasted a day trying to work out why the SNMP

I have a box running asterisk 1.4.17 that had been working. it has 2 uniden phones connected on it. This was working and now the phones dont ring when calling each other. below is the sip debug. I cant see why the other phone does not ring? I also tried changing the canreinvite for no to yes but that made no difference after restarting. Very simple network. server, linksys router and 2 phones.

Hi, I am experiencing one-way audio from: SIP Device ----> Asterisk -----> Cisco 7200 The Cisco 7200 has a VXE+ card that will allow you to do SIP. I can pass audio from SIP Device to Asterisk through the Cisco 7200 to the other end, but the Cisco 7200 does not return any audio back to the SIP Device or Asterisk, it seems. I have tried upgrading to 12.3T IOS version, but no

REPOSTED with MORE Info and Modified Subject Line: -------------------------------------------------------- I am using one of the Minute Provider to dial out USA numbers. Now in one of my process, we need to Dial IVR and the enter DTMF digit and then it connects to the automated IVR. When I dial out the IVR directly using Xlite and VOIP Mins provider , it works perfectly. but when In try from

Hi to all i'm using Asterisk 1.4 and need to announce something like 'The operator answering to you call is XXX' to the caller, is it possible to do that using an AGI script ? The syntax in Asterisk 1.4 is Queue(queuename[|options][|URL][|announceoverride][|timeout][|AGI]) So, setting up an appropriate AGI script can i play an audio file (or create it with some tts) to the

i have a prablom here, i want to send a call from A to B use sip trunk , the call can sended B,but not work to PSTN. the message from B server. help pls,what's rong? > > <--- SIP read from 192.168.0.176:5060 ---> > INVITE sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151>SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport

Hi, Could somebody help me this SIP trasport? I'm receiving SIP "invite" with CLI of calling party from the SIP gateway, aster that my IVR has to answer the call. sip.conf: ========= [general] port = 5060 bindaddr = 0.0.0.0 context = incomingsip videosupport=yes ; Turn on support for SIP video disallow=all ; Disallow all codecs allow=g729

Hi, I have come across an odd problem. Basically I am transferring a call to an agent. The agent is logged in and set as paused. In order to find which agent to call I am using a fastagi script to just set a variable. When it falls through the agi script and dials the agent (using the variable) it doesn't connect the call properly to the agent. I get the beep but no audio (along with

Here is the CS1000 log. Again, the CS1000 using SIP accepts incoming calls just fine. However, using outgoing call files the CS1000 is hanging up after I answer the call. I dont know why? Thanks, for any assistance. Jerry my sip.conf entry is: [Nortel] type=friend dtmfmode=rfc2833 username=XXXXXXXXX disallow=all allow=ulaw allow=alaw

You may loose 'control' of the call but you can always 'get it back' Use the UnigueID of the call to track it throught Asterisk. You can palce a monitor event to redirect, bridge, drop, answer or antything else. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Sebastian

The Asterisk Development Team would like to announce the release of Asterisk 14.6.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.6.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 13.17.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.17.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

I am having a problem with the Call Queue Feature. If I let a user into the Queue prior to an agent being available for them to take the call, they experience the following: 1) they hear that they are the first in line 2) when the agent finally logs in (the caller on hold in the queue is sent to the users phone) 3) the AGENT is still in the login phase hearing that they are "successfully

major undertaking. But since you are using an AGI to control the Queue command instead of using it from the dialplan, you have more control over this problem than you realize. For simplicity of illustration, let's say your AGI simply wants to take a call and send it to the next agent in the queue. Your Agents are Agent007, AgentQ and AgentM. Because you did the Polycom transfer from