similar to: Dialplan matching problem

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 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

Hi all, I'm using asterisk in real time mode...My extensions.conf table contains: [default] switch => Realtime/default at extensions I have added the following to extensions.conf table; context:micho exten: _X. priority: 1 app:Dial appdata: SIP/00XXXXXX at PSTN GAteway Asterisk server is connected succeffully to database...As soon as i make a call i got the following error message:

Dear All, I have the below context defined in extensions.con: [a2billing] exten => _X.,1,Gotoif($[${EXTEN} = 111] ? 21) exten => _X.,2,DeadAGI,a2billing.php exten => _X.,3,Wait,2 exten => _X.,4,Hangup exten => _X.,21,Wait(2) exten => _X.,22,Record(/tmp/asterisk-recording:ulaw) exten => _X.,23,Wait(2) exten => _X.,24,Playback(/tmp/asterisk-recording) exten =>

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

Hi all, Does asterisk support the following scenario? I need when a customer who own an endpoint registered on asterisk open the line, the asterisk will run a specific AGI script inside the endpoint context? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090421/4dce9c34/attachment.htm

Dear Sir, I need to configure my Voice Mail on asterisk...I made the following configuration: * extensions.conf:* exten => _999.,1,VoiceMail(${EXTEN}) exten => _999.,2,HangUp() If the customer dial 9991234 then a prompt message should ask him to enter his voice message and this what is not happening *voicemail.conf:* [a2billing] 999123456 => 123456, 123456, michofr at mm.com The

Hi all, I would like to ask please if there is a way to play a ring back tone from asterisk when the customer try to make a call...I already added the ringing function to the context in extensions .conf and it work perfectly...But the issue that the asterisk server is stoping playing back his own ring back tone as soon as it detect a ring back tone coming from the carrier side... Is there a way

I have this [TRONCAL-SIP] exten=>225/91,1,Answer exten=>225/91,2,Echo exten=>225/91,3,Hangup exten=>225/92,1,Answer exten=>225/92,2,Playback(conf-invalid) exten=>225/92,3,Hangup When I make a call CLI> -- Recv IAM CIC=8 ANI=91 DNI=225 RNI= redirect=no/0 complete=1 Dont work If I add this rule exten=>225,1,Answer Works ok -------------- next part --------------

Hi all, I have the following context in extensions.conf: [a2billing] exten => _X.,1,Gotoif($[${EXTEN} = 111] ? 21) exten => _X.,2,DeadAGI,a2billing.php exten => _X.,3,Wait,2 exten => _X.,4,Hangup exten => _X.,21,Playback(AR_GetGiveToID) exten => _X.,22,Wait(2) exten => _X.,23,Record(/tmp/asterisk-recording:ulaw,,5) exten => _X.,24,Wait(2) exten =>

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

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

Dear All, i have the following context defines in etensions.conf: [a2billing] exten => _X.,1,Gotoif($[${EXTEN} = 111] ? 21) exten => _X.,2,DeadAGI,a2billing.php exten => _X.,3,Wait,2 exten => _X.,4,Hangup exten => _X.,21,Playback(AR_GetGiveToID) exten => _X.,22,Wait(2) exten => _X.,23,Record(/tmp/asterisk-recording:ulaw,,5) exten => _X.,24,Wait(2) exten =>

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

Hi team, I am trying to find solution to hangup b-party call after 1 min with out disconnecting the call of a-party but following dial plan which is disconnect both the calls. Please suggest me the solution. [TB] exten => _X.,1,Wait(${INCOMING_WAIT}) exten =>_X.,2,Verbose(TB) exten =>_X.,3,Answer() exten => _X.,4,Set(mainLoop=0) exten =>

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 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

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 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 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)