similar to: PRI hangup request, cause 18

Hello, Limiting the call duration with the L-option of the Dial()-command is working fine, however the announcement is not played. Dialplan : exten => _367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes) exten => _367,n,Dial(SIP/test6,,L(11000,5000,5000)) The call lasts for 11 seconds, but 5 minutes before time runs out an announcement should come. I hear no announcement, not on caller-side nor on

Hi, I am trying to setup a GSM-Card for Asterisk. I currently use a "vgsm I" from voismart (http://www.voismart.it/) but the driver is very bad (compile-problems and no echo cancellation). Is there anybody out there who can recommend me another piece of hardware (pci card)? I need 1 or better 4 gsm-ports. Should be stable and have an echo cancelltaion feature. And of course it

Hello, Using a PRI (E1) with the euroisdn protocol, I don't seem to get any specific message from the telco when attempting to dial a non-existent number. Asterisk returns a busy/congested code, but nothing indicating the number's real status. How do you guys manage that issue? Do you record a message ("sorry, the number dialed can't be completed") and play it when the

We have setup asterisk to handle our calls before between telco and an Avaya definity. The PRI keeps locking up every so often. In addition I keep getting this error when trying to call the avaya: -- Channel 0/2, span 1 got hangup request, cause 102 -- Hungup 'Zap/2-1' When that error happens I get a fast busy (congestion) tone. Any one can point me in the right direction? TIA

Hi! I'm writing to this list because I've got a very confusing issue with our Asterisk 1.8.3.2 installation. On high IO load on the hard drives Asterisk becomes instable and crashes after a few minutes. I tried to reproduce this by running bonnie++ on the hardware while making calls. The calls didn't get disturbed (no noises or crackles) but after about five minutes Asterisk suddenly

Hey Guy, I have quick question. I am purchasing Sangoma A102D card but i am confused between PCI and PCI Express. Which card would be good for me. Definitely PCI Express is advance but i just want to know is there any major difference, like quality, performance etc.. -Satish -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am running a service where I play full songs but MP3 files kept on crashing my server. I resorted to wav but the quality is really poor after converting..or even sometimes not audible at all! Do you guys know of a better way I can convert mp3 to wav and restore quality? Below is the script I am using, I also tried the steps at

Hi All; I need to know which version of asterisk to use, if to be 1.4 or 1.6 or 1.8? For example, when to decide that I have to go for 1.6 or I have to go for 1.8? Regards Bilal

Hello, We would like to continue a Perl AGI after a Dial() it has done completes following caller hangup. We would like to do this in the same AGI, and not using a new AGI from the 'h' extension. It works fine when the called party hangs up and the 'g' option is used, but not for caller hangup. Is this possible? If not a confirmation that this is the case would be very helpful.

hi: our current connection is below: sip phone<--->asterisk<---->alcatel PBX<---->PSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3.

Dear; Thanks a lot for guiding me. Is it possible that the installation libpri-1.4.11.5 newer than the libpri-1.4.11.5-patch? Well, when I typed (note: I am trying to apply the libpri-1.4.11.5-patch for the libpri-1.4.11.5): libpri-1.4.11.5# patch -p0 -i libpri-1.4.11.5-patch It gave me that patched detected as shown below (example of one file, and I got same for other files): patching file

Hello, I seem to be having an issue with the configuration of my PRI on a new asterisk server I've created to replace an old install that I have. The card is Digium Wildcard TE133. I continually get messages like "Primary D-Channel on span 1 down", rather irregularly: [2013-07-29 17:31:39] VERBOSE[3621] sig_pri.c: == Primary D-Channel on span 1 up [2013-07-29 17:31:39]

Hello, I'm reading this in my log files: [Mar 25 12:01:23] WARNING[1593] sig_pri.c: Span 1: Got SETUP with duplicate call ptr (0x8e3b998). Dropping call. [Mar 25 13:21:40] WARNING[1593] sig_pri.c: Span 1: Got SETUP with duplicate call ptr (0x8e3b998). Dropping call. [Mar 26 10:20:54] WARNING[2643] sig_pri.c: Span 1: Got SETUP with duplicate call ptr (0x8e06788). Dropping call. [Mar 26

Hi, I have a B410P card with span ports set up as span=3,1,0,CCS,AMI span=4,2,0,CCS,AMI span=5,3,0,CCS,AMI signalling = bri_cpe switchtype = euroisdn layer1_presence = ignore However, I keep getting these messages over and over again: [Dec 14 18:53:14] WARNING[22476]: sig_pri.c:1150 pri_find_dchan: Span 3: D-channel is down! == Primary D-Channel on span 3 up == Primary D-Channel on span 4

Hi all, one of my servers crashed last sunday just right after a startup and I need to understand what was going on. I am running Windows 2008 R2 Standard Edition on Oracle VM 3.1 - Xen 4.1.2-OVM On sunday a scheduled task shutted down cleanly the vm. the shutdown started at Sun Jun 16 10:00:33 and it ended at 2013-06-16 10:00:46.938. xend.log says: [2013-06-16 10:00:45 7228] INFO

I am trying to setup Asterisk to act like a PBX connected via a PRI gateway to a voice netowrk where Asterisk is doing outbound overlap dialing for calls that terminate via that PRI. AFter researching through the archives and online dcocs, I thought I had everyting setup right, dialplan configured for '_X!' and the 'overlapdial=yes' in the chan_dahdi.conf file, but when I try and

Hey guys, I have stdexten macro dialplan and I have to handle those who doesn't have voicemail box setup. Right now if someone call and if person unavailable the it's just hangup that call. I want it say "person doest have vm setup yet." smthing like that. How should I handle this in my dialplan ? -- Sent from my iPhone

Hi, df -h output: root at ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# df -h S.ficheros Tam. Usado Disp. % Uso Montado en /cow 14G 4,5G 8,7G 34% / udev 999M 4,0K 999M 1% /dev tmpfs 403M 860K 402M 1% /run /dev/sdb1 799M 693M 106M 87% /cdrom /dev/loop0 668M 668M 0 100% /rofs

Hi, Starting in Asterisk 1.8.0, Asterisk supports connected line updates. This is fantastic for SIP. How can I prevent them from being sent to a PRI channel? I'm having problems when a call is answered by an internal SIP extension, then transferred (blind or attended) to another internal SIP extension. One of my PRI providers can't handle the ROSE_ETSI_EctInform APDU and drops the

Hey guys, I've got a part of my dialplan that dials multiple people: exten => direct,n,Dial(${QUEUEEXTS},${RINGTIME}) Multiple extensions are in the ${QUEUEEXTS} from an external script - e.g. SIP/100&SIP/101&SIP/105 etc This works great, however I want to see if I can find a way to work out (and run an AGI script) when the call is picked up by someone. Thanks all!