search for: macroing

I currently have the following outbound-local config in my setup.... I can call SOME of the numbers (like 337xxxx, and 998xxxx, and 323xxxx).. but when I try to dial say like 601xxxx I get a 404.. any thoughts, I can't see any difference in the config. Also, I seem to be able to dial any number that starts with a 9.. such as 977, 990, 903.. [outbound-local] ;exten =>

Hi All, I have searched the various post and not able to find the solution. I am using Asterisk 1.4.21.2 for outgoing calls. Earlier i used ZAP trunk and it works fine. Now i need to move to SIP trunk and configured the same. When i try from softphone i got error as "Call rejected" and in the asterisk i got error as

I am trying to record calls on demand both inbound and outbound calls. I can record outbound calls just fine but not inbound calls or calls from an internally between extensions. I am using the latest asterisk 1.8.x certified version. On an outbound call I see: == Using SIP RTP CoS mark 5 -- Called SIP/ BVTrunk /7190000000 -- SIP/BVTrunk-00000163 is making progress passing it to

My wife's got an appropriate Southern England (Wimbledon) accent and I'm sure she would try her hand. Does anyone have a comprehensive list of the words that need to be said? Matt, do you have them if your wife's done a set for French users? Mark, if you have the kit maybe you could chop up the file? I write a utility to chop up and compress the wave file based on some of the C

hi,all when i send a call to other server use SIP trunk, i got this below, <--- SIP read from 222.46.18.52:5060 ---> SIP/2.0 403 Forbidden what's rong with is? > Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others. > Created by Mark Spencer <markster at digium.com> > Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for

Hello, Sometimes the call gets answered for 1 second, but actually the phone has not rang, it?s just the CDR, and asterisk hangup automatically, I cought the log of 1 call like this, I hope you can help me with this. My setup is : <vendor> ----SIP--? <Asterisk> ?----IAX2---? <Asterisk with Dhadi channels> Here: -- Executing [966505103150 at from-internal:1]

Hi All, I am trying to connect my asterisk box with freepbx to PSTN. I have purchased x100p FXO card and installed in my asterisk server. My freepbx detected the x100p FXO card and i can see the card specific details in command line. I have configured the following things. 1. OUTBOUND caller id and Dialing rules in Freepbx. 2. INBOUND route When i call to the PSTN number before

Hi people, I have the following dialplan, but it doesn't have the behavior that I think it should have. [default] exten => 2001,1,Answer exten => 2001,n,Dial(local/3005) exten => 2001,n,Hangup exten => 3005,1,Set(__RINGTIMER=10) exten => 3005,n,Macro(exten-vm,novm,3005) exten => 3005,n,Hangup When I execute the Originate (AMI) with the argument Channel=local/2001, It rings

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

Using Sun Forte C Compiling smbd/server.c "/usr/include/sys/termios.h", line 38: macro CTRL redefines previous macro at "/usr/ucbinclude/sys/ttychars.h", line 76 "/usr/include/sys/termios.h", line 164: macro CEOT redefines previous macro at "/usr/ucbinclude/sys/ttychars.h", line 87 "/usr/include/sys/termios.h", line 167: macro CEOF redefines

Hi, I would like to implement macro debug info support in LLVM. Below you will find 4 parts: 1. Background on what does it mean to debug macros. 2. A brief explanation on how to represent macro debug info in DWARF 4.0. 3. The suggested design. 4. A full example: Source -> AST -> LLVM IR -> DWARF. Feel free to skip first two parts if you think you know the background.

hello list i have an issue related to outbound calls i can contact all the number except on number given by our provider in trunk the issue just when i configure my trunk in our server but when i configure the trunk directly in x-lite i can contact this number without issue below the cli == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [0149xxxxxx at

Hello everyone! I've had this problem for a while and cant figure it out. When an outside caller calls an extension on my asterisk system, they do not hear any sort of ringing. Inside extensions calling other extensions do hear ringing. We have 3 other asterisk systems that are configured the same way, but do not have this problem. We think it has something to do with asterisk 1.6. The other

Hello, My client customers complaining that their calls suddenly get hung-up, I am just investigating if the problem from my side, I had a log of a hang-up case, Does it help to know if there is a problem that can be resolved from my side? elastix*CLI> -- Hungup 'IAX2/99999-6813' == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on

Hi this message give me when I calling a number than actually not busy: "Dial failed due to trunk reporting BUSY - giving up" max channel is unlimited and sometimes it dial number ok but most of the time it gives me this error. Please inform me how can solve this problem. thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

> Do we really need to touch the AST? Or would it be reasonable to wire up the CGDebugInfo directly to the PPCallbacks, if it isn't already? (perhaps it is already wired up for other reasons?) This sound as a good idea, I will check that approach. PPCallbacks is only an interface, has nothing connected to it, but we will create a new class, which implement PPCallbacks, for macros. So we can

This is an automatic message generated by the builder on builder-ubuntu for febootstrap. Log files from the build follow below. Linux builder-ubuntu 2.6.35-22-generic #35-Ubuntu SMP Sat Oct 16 20:45:36 UTC 2010 x86_64 GNU/Linux Tue Dec 14 22:00:01 GMT 2010 ----- + git pull --rebase git://git.annexia.org/git/febootstrap.git master >From git://git.annexia.org/git/febootstrap * branch

This is an automatic message generated by the builder on builder-ubuntu for febootstrap. Log files from the build follow below. Linux builder-ubuntu 2.6.35-22-generic #35-Ubuntu SMP Sat Oct 16 20:45:36 UTC 2010 x86_64 GNU/Linux Fri Jan 14 22:00:01 GMT 2011 ----- + git pull --rebase git://git.annexia.org/git/febootstrap.git master >From git://git.annexia.org/git/febootstrap * branch

This is an automatic message generated by the builder on builder-ubuntu for febootstrap. Log files from the build follow below. Linux builder-ubuntu 2.6.35-22-generic #35-Ubuntu SMP Sat Oct 16 20:45:36 UTC 2010 x86_64 GNU/Linux Tue Feb 15 22:00:01 GMT 2011 ----- + git pull --rebase git://git.annexia.org/git/febootstrap.git master >From git://git.annexia.org/git/febootstrap * branch

This is an automatic message generated by the builder on builder-debian for febootstrap. Log files from the build follow below. Linux builder-debian.home.annexia.org 2.6.36-trunk-amd64 #1 SMP Wed Oct 27 14:28:29 UTC 2010 x86_64 GNU/Linux Fri Jan 14 20:00:01 GMT 2011 ----- + git pull --rebase git://git.annexia.org/git/febootstrap.git master >From git://git.annexia.org/git/febootstrap *