similar to: SIP Packets

I am using CVS-v1-0-04/20/05-12:31:26. Windows Messenger5.1 works MOH fine. After I stop MOH on Windows Messenger, if the hungup signal could not send to *, the sip channel(e.g.,SIP/52001-08ca) for this MOH remains. Then the user trys again MOH, a new sip channel starts. And again the hugup signal can not send to *,......... When I 'stop now' from CLI> , * cleanups the remaining sip

Hi: I have a linksys voip gateway connecting to an asterisk server ,when i dial a call from the linksys gateway to asterisk , i see repeated messages of a RTP errors ,and at same time i hear fake ring on the linksys?, This is wht i see on asterisk console?: ? -- Executing [9613070741 at direct:1] Set("SIP/03070741-088bd470", "CALLERID(number)=96170707070") in new stack ??? --

Dear Sir, I'm running Asterisk 1.4.21.2 on a CentOS machine....When running asterisk -rvvvvv I can see a lot of messages about RTCP too short... -- Remote UNIX connection disconnected [Nov 28 13:33:00] WARNING[24863]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891

Hi, I am trying to achieve 3-way conferencing taking hint from wiki link http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO Here is the scenario: 1. user "ua1" calls user "ca1" 2. "ua1" then presses the feature code "*0" to redirect "ca1" to conference room 300 3. "ua1" then dials the user "33" 4. user

Hi I'm trying to get the rebuilt parking functionality to work in Asterisk 12.0.0. In Asterisk 11.6.0 I managed to get a call to get parked by adding a dynamic feature in features.conf for the DMTF sequence *# which called a macro in extensions.conf, which then runned the ParkAndAnnounce application, and the call got parked. The syntax for ParkAndAnnounce I used was this (I don't

Hi all, We have an asterisk version v1.6.1.20 with a TDM400 board (2 FXS and 2 FXO). We want to do a transfer "blind" and "attended" from a line external connected to one FXO. We have made configuration, and transfers from internal lines (FXS) work fine but from (FXO) not. We have made 2 test, one work fine from FXS and the other form FXO no. Test 1, work fine: 1) A

Dear All, I just setup the Asterisk with E100P which it's no problem in Dial In but I have problem when outdial. The connection method is like this : E1 PRI <-SIGNAL-1-> MaxLink (PBX) <-SIGNAL-2-> E100P <-> Asterisk <--> SIP \-----> Analog PHone Now when I tried to dial out by SIP X-Lite on Windows, it shows me Connect, Trying,

I built asterisk-trunk at 92526 and asterisk-addons-trunk at 496. I have my Bluetooth cell phone connected. It almost works. In mobile.conf, I have "context=incoming-mobile" for the phone, and that looks like: context incoming-mobile { _. => { VoiceMail(9999,b); Hangup(); }; } Calls to the cell phone get directed answered by Asterisk and directed to

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer

Hello, how are you? I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer

Does anybody know why I get this error when using zapateller: WARNING[1209214400]: File rtp.c, Line 327 (ast_rtcp_read): RTP Read error: Resource temporarily unavailable It's scrolls until a sound is recived from the line, then it plays the zapateller tones. /Mike

Hello Asterisk list, I've facing a memory allocation issue that happens occasionally but on a consistent basis. The problem happens as follow, suddenly Asterisk starts consuming a lot of memory, in a rate of more than 1GB per hour. Kernel will eventually kill it via the OOM killer when memory is really exausted... This situation does not generate backtrace because Asterisk is responsive

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I just noticed that the Skype API for linux seems to be available. I've read before a number of posts where people were talking about implementing a chan_skype with the skype API. I wonder if there is any progress in that direction, and if anyone is working on it. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to

Hi! I'm using asterisk 1.4.17 with twinkle and a custom phone based on iaxclient 2.0.2 and I'm struggling a bit with DTMF and features.conf. While the twinkle client is able to initiate an attended transfer using *2 (as configured in features.conf), the iax client is not. I can see the DTMF messages showing up on the asterisk console, but asterisk does not invoke the features

Hi All, I seem to have stumbled on a bit of a problem. When trying to send a fax with Fax For Asterisk on 1.6.2.x (have tried 1.6.2.5, 1.6.2.7 and the current svn version, with FFA 1.2 I get a core dump each time. Here is an extract form the console: [May 12 22:47:09] DEBUG[22584]: app_queue.c:1084 handle_statechange: Device 'SIP/vltb-sbc01' changed to state '1' (Not in use)

Hello again all, I''m proud to say that with the help of a good friend and some serious tinkering, I have finally figured out filtering within tc. But alas, I am having an issue I know I should not be having. I am trying to filter all IRC traffic on my network so that it neither consumes large amounts of band width nor gets to little band width. I originally wanted to filter just

Hi all, I'm running a CentOS 5.2 server with a FC8 xen domU guest, and an irc server on the FC8 domU guest. For some odd reason, I can't access the irc server on the xen domU guest. >From FC8, when I run netstat -a, I can see irc is listening: netstat -a Active Internet connections (servers and established) Proto Recv-Q Send-Q Local Address Foreign Address State tcp

Hi all, I'm running a CentOS 5.2 server with a FC8 xen domU guest, and an irc server on the FC8 domU guest. For some odd reason, I can't access the irc server on the xen domU guest. >From FC8, when I run netstat -a, I can see irc is listening: netstat -a Active Internet connections (servers and established) Proto Recv-Q Send-Q Local Address Foreign Address State tcp