similar to: cdr and edit dst field

For make outgoing call, i setup 0. However 0 is write in the cdr dst field. Is there a way to remove it when asterisk send it to cdr_mysql ? exten => _0X.,1,Dial,SIP/${EXTEN:1}@mygateway I just want have in cdr dst = ${EXTEN:1} This don't work : exten => _0X.,1,SetVar(EXTEN=${EXTEN:1}) exten => _0X.,2,Dial,SIP/${EXTEN}@mygateway Use another variable still record ${EXTEN} --

Hello, I have trouble to enable a sip trunk with a CIRPAK. CIRPAK support answer that's there parameter are unvalid : a=silenceSupp:off - - - - is not standard and not working with cirpak - to be remove m=video 13072 RTP/AVP no video, how to remove it ? my extension.conf : exten => _6X.,1,Dial,SIP/${EXTEN:1}@x.x.x.x Regards, -- Arnaud Pignard (apignard@frontier.fr) Frontier Online -

Hello, We have one of our SIP provider that's is sending incoming sip call without need of registration. Incoming call working fine (as outgoing call), but * still try to register to there sip gateway : chan_sip.c:3159 sip_reg_timeout: Registration for 'phone@50.50.50.50' timed out, trying again -- Got SIP response 404 "User Not Found in data base" back from 50.50.50.50

Hi ! I have a TDM40B and i try to use it connected to modem for incoming call data transfert. I have no problem to use it with a phone and a talk communication work fine. But when we try to use with modem, with most modem, we got data carrier for few seconds and channel hungup. < [ TYPE: Null Frame (4) SUBCLASS: N/A (3) ] [Zap/4-1] -- Zap/4-1 is ringing << [ TYPE: Null Frame

Hi, Since around a week i have one asterisk server how stop responding randomely. CVS HEAD with RealTime engine used. The debug log only write "Failed to grab lock, trying again..." until i stop Asterisk. No more activity for IAX or SIP channels (no log...). CLI still responding. When i try to stop asterisk (stop now or crtl+C) nothing happen and cli die but asterisk still running.

Hi, I have a partial working installation with zaphfc. Incoming call : For incoming call, seems work fine. But the sound is very bad with bounce short crashing sound. Same sound with echo cancel off or on. SDA work fine. Another problem, it's seems that's zaphfc don't reset correctly the line. I have one of my D channel how was busy even after stop communication. Outgoing call :

Hi Just received my spanky new TE405P today to replace my Cisco gateway... After much fiddling (I forgot to switch it to E1) I got it to work and everything "seems" to work perfectly on our ISDN PRI. If I dial-in from the PSTN to a SIP phone, the call goes through and if I hangup either the SIP phone or the remote end, the call gets disconnected and destroyed However, if I dial-in

Hello, I've got an Asterisk server with a TDM400P with 2 FXO modules and 2 FXS modules. This server is connected to 2 PSTN lines and 2 analog phones. In my Zaptel configuration, I've defined 2 groups : one for the 2 FXO's and one for the 2 FXS. The asterisk server is just used to add a little IVR and Voicemail service. Eveything works fine, but sometimes the conversation is

Hello, I am not able to use ftp and ssh on the lan. Both the port are open. I have the lines on the rules file : # Accept SSH connections from the local network for administration # ACCEPT loc fw tcp 22 # # Accept ftp connections from the local network for administration # ACCEPT loc fw tcp 21 Thanks in advance Varun

Hi All, I'm having some troubles with using call files. I'm trying to establish the following: - want to use call files to connect two (outside) extensions - want to use the outbound routes set in FreePBX - want to set the outgoing callerid for both calls - want to set a custom CDR field in MySQL ( field name 'azonosito' ) Asterisk is version 1.8.2.3 with freepbx 2.8.1. What

Hi! Trying to figure out how to rewrite calling number of an incoming call... A cell phone (0733025975) dials a X-Lite (977). X-Lite "shows" 733025975 at the display, but I want it to be 0317998975. I thought i could do something like: exten => 977/733025975,1,Set(CALLERID(number)=0317998975) exten => 977,n,Dial(SIP/0317998977) [Dec 14 19:07:43] NOTICE[20731]: chan_h323.c:2272

Hello, I can?t compile "iaxclient", because one needs to compile the new version "wiax.dll". I tried to compile it under MinGW/Cygwin, but I had the messages like: cc -I. -Igsm/inc -Iportaudio/pa_common -Iportaudio/pablio -Iportmixer/px_common -Ilibspeex/include -g -O2 -DSPEEX_PREPROCESS=1 -DNEWJB -Ilibiax2/src -IAXC_IAX -DLIBIAX -DSPEEX_EC=1 -DWIN32 -DBUILDING_DLL -c

The following errata on freeradius package was issued by redhat on october 15 and is not yet available under Centos 5 : http://rhn.redhat.com/errata/RHBA-2008-0845.html Is there any particular reason ? Regards, -- Alain RICHARD <mailto:alain.richard at equation.fr> EQUATION SA <http://www.equation.fr/> Tel : +33 477 79 48 00 Fax : +33 477 79 48 01 E-Liance, Op?rateur des

HI, I'm using the usual Set(Callerid(num) function to change the incoming from skype callerid but it's not working. Asterisk 1.4.31 and last release of skype channels This is the dialplan exten => _0X.,1,NoOP(${CALLERID(num)} - ${CALLERID(name)}) exten => _0X.,n,Set(STRINGA="Skype") exten => _0X.,n,NoOP(${STRINGA}) exten => _0X.,n,Set(CALLERID(num) = ${STRINGA})

Hi! I am searching for the description of the available dialstrin options for the DAHDI channel (and also other channel types). I am not looking for outdated voip-info links, but for the authoritative source, e.g. something like "core show application Dial" Does such thing exists? thanks Klaus

I have a time series of 1095 data corresponding to a daily data of three years. I want to know how to use ma(timeserie, order=??, centre=??) to detect the trend: which order is suitable and what is the difference between centre= true or false. How to avoid these errors: 1-Error in timeserie - trend : ? argument non num?rique pour un op?rateur binaire="non-numeric argument for a binary

I have a system with two 4 port T1 cards, with 5 PRI's configured. Each PRI is configured as an individual PRI and belongs to it's own group (groups 1-5) This system is handling roll-over from another system, where any error in processing the call on that system results in it being sent here. This mainly results in all calls resulting in a busy being sent for retry here. I then

I'm confused about something. It's the way Asterisk handles the A leg (ie the first party dialed) on an originate command via the Manager Interface. Lets say our originate commands looks like this: ACTION: Originate Async: yes Timeout: 60000 Exten: callback Channel: SIP/5551212 at provider Variable: destination=SIP/8675309 at provider Callerid: 5551212 Context: default ActionID: 849120

Jean-Luc, Please keep the communications on the list, for the benefit of others, now and in the future, via the list archive. I am adding r-devel back here. I can't speak to the rationale in some of these cases. As I noted, it may be (is likely) due to differing authors over time, and there may have been relevant use cases at the time that the code was written, resulting in the various

I have setup as per the monitor example configuration on the wiki site and all works well for an extension dialing 8 then the number. However, if I dial from an AGI script the recording stops after a few seconds. I see an extra answer in the console and suspect that is the reason. Could any kind soul help me to get around this? Extensions.conf.. exten =>