similar to: Show calls in progress

I am seeing some curious behaviour with a 1.2.32 system, which I do not understand and so can't work out how to fix it. I have a PRI routed to context default. Here is the complete default context: [default] exten => _9X.,1,Dial(IAX2/m1peer/${EXTEN:1}) exten => _20XX,1,Dial(IAX2/sipeer/${EXTEN}) exten => _X.,1,Dial(IAX2/m1peer/${EXTEN}) exten =>

I'm not sure if this is a -users or a -dev question, since the answer probably comes down to something in the code. I'm running the latest CVS-STABLE, and am subscribed to PSTN service using IAX2 via Voiptalk in the UK. I've just been alerted by a customer that the sending of DTMF from my asterisk box to a remote PSTN user doesn't work, although it used to. To test it, I have

Hi, Asterisk is new for me. I had a working configuration, but suddenly I can't call anymore with my voip provider. I am not aware that I changed anything in the configuration, but who knows. Can somebody explain me what is happening here? I changed username, password and number. -- Executing Dial("Zap/2-1",

Hi We are having a significant (> 1 sec) delay in a multi-asterisk conference, with IAX2 legs connecting meetme on different boxes. All the other legs are PSTN (TE410P). The example configuration Slave box 1 meetme <--- IAX2 ---> Master box meetme <--- IAX2 ---> Slave box 2 meetme The delay is between Slave box 1 and Slave box 2 The primary suspect is our iax configuration

We upgraded our asterisk server to 1.2.18 last night to pick up the security update. Since then, any calls coming in on IAX2 links get dropped if they try to enter a MeetMe conference room. The log shows this: Apr 26 08:33:16 NOTICE[27362]: chan_iax2.c:3167 iax2_read: I should never be called! Hanging up. I've temporarily worked around it by switching our inbound provider to use SIP

August 5th, 2005: Craig Southeren announced today that OPAL (http://www.voxgratia.org) now provides support for the IAX2 protocol(Written by Derek Smithies and released under the MPL). This support allows you to use chan_woomera (http://www.pbxfreeware.org) driver developed by Anthony Minessale II to interconnect your asterisk systems and use the IAX2, SIP, and H.323 protocols. I would

Hello all. I am trying to use app_mysql. It works for selects and functions, but does not want to work with procedures. Pls have a look: Calling function: CREATE FUNCTION f_1(a VARCHAR(20)) RETURNS INTEGER RETURN (SELECT count(*) from peer where name = a); Result: -- Executing Macro("IAX2/100-3", "local|100") in new stack -- Executing MYSQL("IAX2/100-3",

Hi there I am using IAX2 softphones dialing into meetme conferences. I also have jitterbuffer=yes, with typical jitterbuffer settings. The problem I am having is that as soon as there is a delay from a participant, then the delay continues until the participant hangs up and dials in again. When dialing in again the delay seems to go. It seems to me as though as soon as the server registers

Hi Can anyone provide me with a Manchester (0161) UK DID number, preferably IAX2 but SIP is ok too, that I can use for my incoming calls? Call volume will be low. The critical thing is that DTMF must be correctly passed 100% of the time, unlike Sipgate, my current (free) provider, whose DTMF detection/passing is not at all reliable, making it useless for a virtual receptionist scenario. I

Hello, Situation: I've got two asterisk 1.2.4 servers, connected to each other over the internet with IAX2 with about 20msec delay. One of the servers is hosting MeetMe. It's working fine as long as only SIP phones connected to the meetme server participate in the conference. As soon as a participant using IAX2 is connecting, lots and lots of buffer overruns and underruns are

Hi, If there are 5 asterisk servers on the local net and each server runs meetme, eg. 3311,3321,3331,3341,3351 respectively. Can I connect these 5 meetme conferences to one meetme using IAX2? Regards, Zen

Hi there I am using IAX2 softphones dialing into a meetme conference. In my softphone I was forcing uses to click on a button when they wanted to speak, enabling their microphone and disabling their speakers. This way when a user was speaking they did not hear their voice half a second later (because meetme mixes the voice and sends to everyone in the conference). Now because of requirements

Hi! When I try to load the ztdummy driver via "insmod ztdummy", I get the following errors: /lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_unregister /lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_transmit /lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_receive /lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_register I'm using zaptel-1.0.8 and

When I call meetme: exten => 1000,1,Answer exten => 1000,n,Meetme(|||d) Asterisk is complaing with: -- Executing Answer("IAX2/xxx.yyy.142.204:4569-2", "") in new stack -- Executing MeetMe("IAX2/xxx.yyy.142.204:4569-2", "|||d") in new stack -- Playing 'conf-getconfno' (language 'en') Warning, flexible rate not

Hi there I have a question regarding IAX jitter. I have 3 users on a LAN dialing into a Meetme conference on an Asterisk box which is also hosted on the LAN. I have set jitterbuffer = no and tos = lowdelay. Now, for 2 of the users the audio is fine, but for the 3rd user there is intermittent break up in the audio when they are receiving. I have had a look at "iax2 show channels" and

Hi. I'm dialling two numbers - one that's unobtainable, one that's busy. ${DIALSTATUS} is coming back ANSWER each time right before the channels hang up. Am using the following dialplan macro to dial out. [macro-advdial] exten => s,1,Dial(${ARG1},20,g) ; Ring the interface, 20 seconds maximum exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status

The Asterisk development team has released versions 1.2.28, 1.4.19.1, and 1.6.0-beta8. All of these releases contain a security patch for the vulnerability described in the AST-2008-006 security advisory. 1.6.0-beta8 is also a regular update to the 1.6.0 series with a number of bug fixes over the previous beta release. Early last year, we made some modifications to the IAX2 channel driver to

The Asterisk development team has released versions 1.2.28, 1.4.19.1, and 1.6.0-beta8. All of these releases contain a security patch for the vulnerability described in the AST-2008-006 security advisory. 1.6.0-beta8 is also a regular update to the 1.6.0 series with a number of bug fixes over the previous beta release. Early last year, we made some modifications to the IAX2 channel driver to

Hi all, I'm trying to resolve a weird issue with SIP routing. I have a number of SIP trunks, from a selection of providers, all of which are registered in sip.conf: [general] context=default allowguest=no allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=yes tcpbindaddr=0.0.0.0 transport=udp bindport=15060 srvlookup=yes allowsubscribe=yes

Can anyone shed some light on this, I have two asterisk boxes using heartbeat for failover. Sip traffic works just fine with the virtual IP but IAX does not. For example on my servers one server has the following: eth0 = 192.168.1.95 eth0:0 = 192.168.1.2 the other server has: eth0 = 192.168.1.220 if the first "Master" server goes down the second server will take that virtal IP for