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...ot;") in new stack == Spawn extension (a2billing, 999alijawad, 2) exited non-zero on 'SIP/alijawad-08aaf0f0' Even though I do have 999alijawad as a mailbox in voicemail_users If I do setup the mailbox in voicemail.conf it works fine. [a2billing] 999alijawad => 123456, alijawad, alijawad1 at gmail.com I did setup extconfig.conf as it should be: voicemail => mysql,mya2billing,voicemail_users Please advice -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081120/78cf0886/attachment.htm

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Guys Ive got an internet cafe on which I have a debian sarge box running. The Debian box acts as a gateway and it has masquerading on. I have 40 client PC and i do not want to assign more than 64k per pc for upload and the same is true for download too. Ive done alot of research and Ive read tutorials about CBQ and HTB. I found that CBQ.init is

oops, i forgot to reply to the list :-/ Début du message réexpédié : > De : Vincent Dautremont <vdautrem@ulb.ac.be> > Date : 30 août 2007 16:58:26 GMT+02:00 > À : Ali Jawad <alijawad1@gmail.com> > Objet : Rép : [LARTC] Priotirize SSH Traffic > > try that > #tc qdisc add dev eth0 root handle1: prio > # tc filter add dev eth0 protocol ip parent 1: prio 1 u32 match ip > dport 22 0xffff flowid 1:1 > # tc filter add dev eth0 protocol ip parent 1: prio 2 u32...

Hi All I am developing a client that uses libjingle to do xmpp stuff with ejabberd. I can also make audio calls between those clients. What I am trying to archive now is to send calls to pstn using jingle. I was told in the jingle-dev community that asterisk can do that. Is there any way to send jingle audio calls to asterisk and will it understand them ? If yes..can I forward those calls to PSTN

Dear All I am using gtalk features with my own XMPP server "OpenFire" I have setup gtalk.conf and jabber.conf on asterisk and now I can make calls from clients registered on my XMPP server to SIP devices by calling the xmpp accounts registered as clients on asterisk. So far so good. So if I want to call sip:1000 I call the xmpp account that is bound to that account in extensions.conf.

Hi All I have two asterisk servers in two different companies. I want both servers to use the same configuration. I.e. if I add an extension on server A I want this to be reflected on server B. This can be done using NFS or rsync. However this is not my main problem. My problem is as follows. Lets suppose I have the two servers and they are identical. If user A registers at Server A and User B

Hi I am trying to get H323 to run on Asterisk, basically I had Asterisk running so I followed this tutorial http://astrecipes.net/index.php?n=286 and got h323 to run on my first server on the second server it is just throwing the error: ERROR:Failed to create H323 listener The whole error is : ERROR: Could not open H.323 listener port on 1720 [Oct 13 09:20:48] ERROR[3608]: chan_h323.c:3166

Hi All I am currently learning traffic shapping and I need a script that does prioritize SSH traffic on my debian router. My Internet interface is eth1 My Lan interface is eth0 My Internet connection is 256 kbit/s down and 128 kbit per second up. I hope someone does have a well documented (or maybe not so documented) example on which I can build further rules as needed. Thx All.

Hi Iam using the script below to limit usage for the computers on my lan with respect to download and upload I have a 256kb up and 256 kb down connection, I want limit the speed of each computer to 64kbyte down and 32 up as a maximum. The script below works however it limits the up and down of the whole specified network to 64/32 ... what do I have to edit so that the script handles the requests

Hi Iam using the script below to limit usage for the computers on my lan with respect to download and upload I have a 256kb up and 256 kb down connection, I want limit the speed of each computer to 64kbyte down and 32 up as a maximum. The script below works however it limits the up and down of the whole specified network to 64/32 ... what do I have to edit so that the script handles the requests

Hi Ive been reading, testing and applying what Iam reading in the LARTC tutorial for a couple of days, I do not wish to use ready made scripts because that means I will always come back and ask the same question again. So Ive been wondering if I have 10 computers and I do want to limit the download for each of those 10 computers to 10 kbyte per second. I would create a leaf class and match the

Hi Ive been reading, testing and applying what Iam reading in the LARTC tutorial for a couple of days, I do not wish to use ready made scripts because that means I will always come back and ask the same question again. So Ive been wondering if I have 10 computers and I do want to limit the download for each of those 10 computers to 10 kbyte per second. I would create a leaf class and match the

Hi One of my softphones is supposed to support g711 , however I am getting these errors and a 404 not found when I try to make a call from it. However on xlite it works fine using g711. Below is the log of the phone that is not working. Content-Type: application/sdp Content-Length: 1123 P-hint: outbound v=0 o=- 1218448446 197568495 IN IP4 127.0.0.1 s=- c=IN IP4 192.168.0.176 t=0 0

Hi All When I dial a number it reaches the asterisk switch as abc at xyz@123.com what I need to do is to parse the abc and send it to my pstn gateway as in exten => _.,2,Dial(SIP/${EXTEN}@pstn.gw) this does work but I do have a varying number of numbers before the @ exten => _.,1,Dial(SIP/${EXTEN:0:12}@pstn.gw) Well can I use some kind of regular expression to take all numbers before