similar to: call-limit

Hi all, I'm using asterisk in real time mode...All extensions are defined in table sip_buddies...Everything looks fine and asterisk is reading extensions info from the sip_buddies table...The problem occurs as soon as any information on an extension is changed from sip_buddies table...Which mean, if I change the secret field in sip_buddies table then i should reload asterisk to read again the

Hi all.... I'm using asterisk for making PSTN calls from extensions registered on OpenSIPS...In peak hours ,number of calls Increase dramatically to a non logic number..When checking the calls using asterisk CLI I saw a lot of calls in ringing status and after 300s(rtphold timeout), asterisk release all calls...I checked the log file and found.. [Feb 28 11:34:14] NOTICE[19197] chan_sip.c:

Hi! I am proving Asterisk 1.2.24 in realtime with MySQL 5.0.27 using Idefisk softphones. I followed the steps of "how to" of voip-org but always have this error: Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime:

Hi! I am proving Asterisk 1.2.24 in realtime with MySQL 5.0.27 using Idefisk softphones. I followed the steps of "how to" of voip-org but always have this error: Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime:

Hi all, I'm planning to build a VOIP solution for handling SIP calls coming from endpoints registered on a specific SIP proxy...I made some research regarding network architecture and found out that the best solution is to use OpenSips as SIP proxy for registration and local calls between registered endpoints and use asterisk server with a2billing for PSTN calls, rating, routing and all other

Dear All, I have the following CPU info on my asterisk server: Linux switch1.domain.net 2.6.18-92.1.22.el5 #1 SMP Tue Dec 16 12:03:43 EST 2008 i686 i686 i386 GNU/Linux I need to install G729 on the asterisk server just to pass through and not for encoding...Which G729 package do you advice me to install? I tried several packages with no luck Regards -------------- next part -------------- An

Dear All, I have installed 4 asterisks on the same Centos machine..>Each Asterisk has its own installation folder and use its own libraries...Everything looks great and all asterisks are doing their jobs correctly except one thing...I faced a voice quality issue...On a specific time, and after the number of calls begin increasing, the voice quality will begin degradation... Could it be a

Hi all, I have the below extension defined under sip.conf: [2203] type=friend username=2203 secret=123456 host=192.168.0.164 mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 When trying to register from a softphone installed on a PC behind a nat with IP=192.168.0.164, I got 503 FOrbidden...Does anyone have any idea about what could be the issue? Regards -------------- next part

Dear All, I would like to ask please if someone has a AGI script that select a value from a database and dial this value as a destination number Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090220/e2aa530c/attachment.htm

Hello list, asterisk 1.4.30 2 situations in which call-limit should work, but it does not : [Jul 8 09:15:49] WARNING[11132]: app_queue.c:3272 try_calling: The device state of this queue member, test12, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. In sip.conf I have : limitonpeer = yes In my realtime sip_buddies

Dear All, I'm trying to send Fax using T.38 protocol but the FAX is not going through..I'm getting the following error om /var/log/messages [Jan 27 16:46:31] WARNING[25435] channel.c: No path to translate from SIP/80.169.210.181-0896bfd0(4) to SIP/028949469-b7703d40(256) [Jan 27 16:46:40] WARNING[24349] channel.c: Unable to find a codec translation path from 0x100 (g729) to 0x4 (ulaw)

*Dear All, I run the below tcp dump on my asterisk server tcpdump -i eth0 -n -s0 -v udp port 5060 I got the following result 20:29:48.596867 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto 17, length: 373) SIP_PROXY_IP.5060 > Asterisk_IP.5060: UDP, length 345 What i need to know please what TTL means specifically and what is the best value og TTL and what is the lengh vale mean

Dear All, I have the following scenario...When a customer dial 111 number a beep message will iplay in order to record and playback his voice...Else he'll be routed to another call flow as you can see in the context below: [a2billing] exten => _X.,1,Gotoif($[${EXTEN} = 111] ? custom-recordme,111,1) exten => _X.,2,DeadAGI,a2billing.php exten => _X.,3,Wait,2 exten => _X.,4,Hangup

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Hi all, I need to register asterisk on an OpenSIPS SIP Proxy...The Registration is OK but my asterisk is sending OPTIONS packets to OpenSIPS and the SIP Proxy is not replying back...The issue is the UNKNOWN username that reside in the OPTIONS packet as you can see in the captured packets as you can see below: 1. U Asterisk_IP:5060 -> OPENSIPS_IP:5060 2. OPTIONS sip:OPENSIPS_IP

Dear Sir, Can someone help me please to find a free ebook talking about AGI scripting through asterisk? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090218/a59fc299/attachment.htm

adding gsm or just comment out the disallow and the 2 allows. (your = recipient is using a codec that isn't ulaw or alaw). =20 _____ =20 From: asterisk-users-bounces at lists.digium.com = [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel = freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

I have followed the instructions on voip-info.org for Realtime SIP peers, but I get this notice : [Feb 22 20:05:32] NOTICE[15298]: chan_sip.c:15889 handle_request_register: Registration from '<sip:testsip at 192.168.1.150;transport=UDP>' failed for '192.168.1.105' - No matching peer found The CLI shows : [Feb 22 19:58:23] == Parsing

Dera All, I have the following scenario, A customer dial a DID number...The call is routed to a PSTN GW that send the call to asterisk... On asterisk I created an AGI Script that send the call to an extension registered on OpenSIPS server... The extension is ringing successfully, but as soon as I accept the call on OpenSIPS side the call is hangd up... I checked rhe SIP debug and it seems that I

Dear All, I have created an inbound context in SIP .conf that forward incoming call to opensips server...The problem appears as soon as I enable t38pt_udptl = yes...The Asterisk negotiate the SIP session with OpenSIPS without adding voice codec to INVITE packet...It just contains T.38 protocol...When t38pt_udptl is disabled everything looks OK and Ulaw is negotiated with OpenSIPS and cal