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Dear ALL: I find that everybody discuss with asterisk-oh323 instead of chan_h323 of asterisk channels. Why do they choose asterisk-oh323 and not use the build-in module (chan_h323)? What different between these two componments? In my guess, the build-in module should be easy to implement rather than setup another application(asterisk-oh323) expect some bugs or failed functions on chan_h323.

Hello list, There seem to be some problem with the ACD of asterisk where when we use this parameter in queues.conf . We could not get any announcement as expected. Iam useing the latest CVS-head Even weight also doesnot seem to work properly I tried like this where we have two queues one with 100 weight and another with 200 as weight when both enter into the queue when queue is empty when

hi all, I have Asterisk 1.6.0.5 Installed and kamailio 1.0.5 version installed when i leave voicemail On Asterisk i need MWI Indication on kamailio extension there are some methods i tried but still cant get success All other feature are working fine also try voip-info.org methods can anybody suggest me for different method and have some different setting on SIP . any help appreciated

Good day Asterisk users, If this is the wrong place to post this, my apologies. However, I'm trying to see where I can get a patch for the res_musiconhold.so module. I have an issue where if someone is placed on hold, or is placed in a queue, after any announcement is played in the queue, or if someone is put on hold, the call is resumed, then is put back on hold, if the same music is still

I'm thinking about using asterisk as an IVR system with an existing avaya index system. I've got 2x PRI 30 lines coming in to the Index, and I have 4 spare PRI cards in the Index. I was thinking about using a QUAD PRI card from Digium and having the calls come into the Index then transfer to Asterisk for IVR then back to the Index. That way if we get 60 inbound calls we'd in

...to transition from AMI to ARI. Running into a small hiccup when I try to create (originate a call) with the caller id name and number I can pass the Name and Number if the name has no spaces in it and it shows up in my PhonerLite application. curl -v -u asterisk:asterisk -X POST http://asterisk:astersk at localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at 1003&app=hello-world&extension=1000&context=mycontext&priority=1&channelId=mycallerid.1&formats=ulaw&timeout=30&callerId=Dan<291> However, when the caller id name has a space in it, I can't...

...hiccup when I try to create (originate a call) with > the caller id name and number > > > > I can pass the Name and Number if the name has no spaces in it and it > shows up in my PhonerLite application. > > > > curl -v -u asterisk:asterisk -X POST > http://asterisk:astersk at localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at 1003&app=hello-world&extension=1000&context=mycontext&priority=1&channelId=mycallerid.1&formats=ulaw&timeout=30&callerId=Dan<291 > > > > > > However, when the caller id name has a...

...to transition from AMI to ARI. Running into a small hiccup when I try to create (originate a call) with the caller id name and number I can pass the Name and Number if the name has no spaces in it and it shows up in my PhonerLite application. curl -v -u asterisk:asterisk -X POST http://asterisk:astersk at localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at 1003&app=hello-world&extension=1000&context=mycontext&priority=1&channelId=mycallerid.1&formats=ulaw&timeout=30&callerId=Dan<291<http://asterisk:astersk at localhost:8088/ari/channels/mycallerid...

Hi Guys I'm conecting 2 astersk servers using this arquitecture (Ext softphone)<==sip==>(asterisk 1)<====iax2 trunk====>(asterisk 2) <===alaw==>(pstn) If i call from the Ext to the asterisk 2 the sound is perfect, but if i call from Ext to the pstn, i can hear perfect but they tell me that sound really c...

Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX exten...

...to transition from AMI to ARI. Running into a small hiccup when I try to create (originate a call) with the caller id name and number I can pass the Name and Number if the name has no spaces in it and it shows up in my PhonerLite application. curl -v -u asterisk:asterisk -X POST http://asterisk:astersk at localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at 1003&app=hello-world&extension=1000&context=mycontext&priority=1&channelId=mycallerid.1&formats=ulaw&timeout=30&callerId=Dan<291<http://asterisk:astersk at localhost:8088/ari/channels/mycallerid...

...hiccup when I try to create (originate a call) with > the caller id name and number > > > > I can pass the Name and Number if the name has no spaces in it and it > shows up in my PhonerLite application. > > > > curl -v -u asterisk:asterisk -X POST > http://asterisk:astersk at localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at 1003&app=hello-world&extension=1000&context=mycontext&priority=1&channelId=mycallerid.1&formats=ulaw&timeout=30&callerId=Dan<291 > > > > > > However, when the caller id name has a...

Hello everybody! I have some problems with my Astersk. I have an analogical OpenVox card and A Billion ISDN card (with mISDN). I load the modules with modprobe zaptel and modprobe wctdm. When I run ztcfg -vv I have this: Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configur...

When i attempt to make asterisk-oh323 I get the following error. Iam using the following versions : asterisk-oh323-0.6.3b openh323-v1_13_5 pwlib-v1_6_6 openh323_1.13.5-make.patch redhat 8 kernel 2.4.18-14 Anyone can help to get a way out ? Or suggetions for compatible versions ? -----------------------------------------------------------------------------------

Hi all, I use astersk 11.7.0 on Ubuntu 12.04.01 TLS (i386). I use cdr_adaptive_odbc to write CDR to my MySQL's cdr table. After my testing, this scenario is working well. After a long idle time, I didn't make any call to the asterisk server. When I try to make a call again after 8 hours, I found that the cdr lo...

...r to server on their end. They must have some sort of load balancing going on that is causing that. In the past we could get the DTMF to pass when we were on the initial server we registered with but when we got pushed to another server the DTMF would fail till I did a sip reload or restarted Astersk. Now we get no DTMF ever. System set up. Asterisk 1.4.22 Asterisk GUI 2.0 users.conf [trunk_1] context = DID_trunk_1 host = galvatron.vtnoc.net username = user name secret = password trunkname = via:talk - galvatron ; GUI metadata hasiax = no registeriax = no hassip = yes registersip = yes trun...

...ll) with >> the caller id name and number >> >> >> >> I can pass the Name and Number if the name has no spaces in it and it >> shows up in my PhonerLite application. >> >> >> >> curl -v -u asterisk:asterisk -X POST >> http://asterisk:astersk at localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at 1003&app=hello-world&extension=1000&context=mycontext&priority=1&channelId=mycallerid.1&formats=ulaw&timeout=30&callerId=Dan<291 >> > >> >> >> >> However, when the...

...nsion (default, 907768385144, 2) exited non-zero on 'SIP/201-083e75c0' when I make a call from a sip device to my outbound analog trunk using a Digium TDM card. My /etc/zaptel.conf file: loadzone=uk defaultzone=uk fxsks=1-4 I am in the uk by the way. Relevant part of /etc/astersk/zapata.conf: signalling=v23 ; added for UK CLI detection cidstart=polarity ; added for UK CLI detection context=frompstnanalog group=1 callgroup=1 pickupgroup=1 signalling=fxs_ks channel=>1-4 part of extensions.conf: exten => _X.,1,Dial(ZAP/g1/${EXTEN},60) exten => _X.,2,...

Hii, I want to forward calls from an asterisk server to a local gsm network. I have read the wiki pages on various forums. But the thing i want is to receive the call(Voip) from an asterisk server then it should be forwarded to a gsm network & again to either a gsm/ PSTN from the gsm network itself. Please post a help. Thanx in advance. -- Day by Day in Every Way I'm Getting Better

...diately gives "Call failed" messages on attempted outbound calls with no message coming from the asterisk console. The odd thing is that if I call from x1000 to x2000 (Cisco -> Smartphone), the call will complete. Extensions are configured exactly the same using the AMP portion of Astersk@Home. I've google'ed quite a bit but can't seem to find a solution or pointer on this one. The only thing I can come up with is that my system is a little shy on memory. I have 184MB in there, and it gets to about 93% used, although the server still isn't hitting swap. Sug...