similar to: incoming call problem

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

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:

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 under General context...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

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

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'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 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

Dear All, I'm trying to send FAX to an endpoint Behind NAT...The scenario i the following: PSTN_GW-->Asterisk-->asterisk-->OpenSIPS-->Endpoint behind NAT.. The FAX is failed and I got the following error log on asterisk: Mar 20 09:21:09] WARNING[20388]: chan_sip.c:12409 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog

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

hi i need a link or something about asterisk load balancing i cant find any, i only found a paragraf in an email anything wiil be wolcome thanks! David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

*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

Hello, I am using OpenSIPS to register all the users and planning to use asterisk for Auto Attendant, Queues, Voicemail and Conference Bridge. I have a scenario where the signaling does not happen properly: 1) A user from Opensips dials an extension 7000 which is an auto-attendant extension. The call is routed to asterisk to play the auto attendant messages like Welcome and Dial the

I have a problem where SIP calls from some providers are dropping at 15 minutes. The topology is: Client sends calls to a host running OpenSIPS, OpenSIPS sends calls to an Asterisk server. Below, 'Client' is the IP address of the client's host (running FPBX-2.8.1(1.8.20.0) 'OpenSIPS' is the IP address of my host running OpenSIPS 1.7.2-tls 'Asterisk' is the IP

Hi, We're using Asterisk 13.17.0 with PJSIP 2.8 bundled. I've found an issue when Asterisk tries to make a SIP call out using auth, but has the wrong credentials and keeps getting returned a SIP 407, in this example to an OpenSIPs server requiring user auth. Basically this happens: 1. Asterisk sends plain INVITE to OpenSIPs 2. OpenSIPs responds with SIP 407 auth required with a

Hello Everyone, I have gone through a few really good tutorials from the OpenSIPS site, Asterisk resources etc.. The unanswered question (and final piece of our puzzle) is if it's possible to have a register free environment in an OpenSIPS/Asterisk integration. Most approaches have OpenSIPS relay the UA's REGISTER request to Asterisk which has "host=dynamic" set for the

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