Re: [Asterisk-Users] ASTCC - INUSE Flag
It works.
I terminated the call during the playback.
AGI debug
AGI Tx 200 result=-1 endpos=480
HUP received!
Allowing
setinuse() to get called
Thanks
Michael
On 10/6/05 12:13 AM, Darren Wiebe [EMAIL PROTECTED] wrote:
Edit astcc.agi and stick these lines in before sub load_config.
$SIG{HUP} = 'ignore_hup';
sub ignore_hup {
print STDERR \nHUP received!\n\n;
}
Darren Wiebe
[EMAIL PROTECTED]
Scott Wolfe wrote:
How do you you apply the patch?
-Scott
- Original Message - From: Nicolás Gudiño [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, October 05, 2005 9:31 PM
Subject: Re: [Asterisk-Users] ASTCC - INUSE Flag
On 10/5/05, Darren Wiebe [EMAIL PROTECTED] wrote:
Any developers out there that would like to look at this one? It works
fine on Asterisk CVS-v1-0-09/22/05-22:23:34 on a i686 running Linux but
it does not work on the 1.2 betas. I agree that the number should be
set aside then. I wonder what the problem is.
http://bugs.digium.com/view.php?id=5400
Seems to fix the problem... please test and give feedback.
--
Nicolás Gudiño
Buenos Aires - Argentina
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Re: [Asterisk-Users] ASTCC - INUSE Flag
This is my debug with the same issue The agi terminates during the sub tell_time() and exits without calling sub setinuse() or completing the reset of the script. AGI Tx agi_request: astcc.agi AGI Tx agi_channel: Zap/49-1 AGI Tx agi_language: en AGI Tx agi_type: Zap AGI Tx agi_uniqueid: 1128401550.162 AGI Tx agi_callerid: xx AGI Tx agi_calleridname: unknown AGI Tx agi_callingpres: 3 AGI Tx agi_callingani2: 0 AGI Tx agi_callington: 33 AGI Tx agi_callingtns: 0 AGI Tx agi_dnid: xx AGI Tx agi_rdnis: unknown AGI Tx agi_context: default AGI Tx agi_extension: xx AGI Tx agi_priority: 103 AGI Tx agi_enhanced: 0.0 AGI Tx agi_accountcode: xxx AGI Tx 0-r1*CLI AGI Rx ANSWERLI AGI Tx 200 result=0 AGI Rx GET DATA astcc-enter-card-num 6000 -- Playing 'astcc-enter-card-num' (language 'en') AGI Tx 200 result=3546 AGI Rx STREAM FILE astcc-youhave 0123456789 AGI Tx 200 result=0 endpos=4480 AGI Rx SAY NUMBER 11 0123456789 -- Playing 'digits/11' (language 'en') AGI Tx 200 result=0 AGI Rx STREAM FILE astcc-dollars 0123456789 AGI Tx 200 result=0 endpos=6720 AGI Rx STREAM FILE astcc-and 0123456789 AGI Tx 200 result=0 endpos=3680 AGI Rx SAY NUMBER 88 0123456789 -- Playing 'digits/80' (language 'en') -- Channel 0/1, span 3 got hangup request AGI Tx 200 result=-1 == Spawn extension (default, x, 103) exited non-zero on 'Zap/49-1' -- Hungup 'Zap/49-1' -Michael On 10/3/05 10:52 PM, Darren Wiebe [EMAIL PROTECTED] wrote: Can you please post the output with debug agi on ? Darren Wiebe [EMAIL PROTECTED] Scott Wolfe wrote: I download and installed ASTCC over the weekend and I am having an issue where the INUSE flag will not get set back to 0 if the user drops a call while the balance is being played. All other times it seems to reset the flag correctly. I have tried both AGI and DeadAGI with the same results. Those of you using it for a while, how did you get around this? Just for fun this is all I am doing in my astcc-exten.conf [incoming] exten = s,1,Answer ;exten = s,2,DeadAGI(astcc.agi) exten = s,2,AGI(astcc.agi) exten = s,3,Hangup I did some Google search on this issue and saw someone else had a problem but no response. -Scott ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC issues
I have been testing the ASTCC and have notice that when the caller hangs up the line while the balance is being played back the sub savedata() is not being called because the asterisk terminates the AGI and the rest of the script does not get executed thus never returning: AGI Script astcc.agi completed, returning 0 This leave the inuse set to 1 and the pin can not be used. I am using the lastest CVS HEAD asterisk-perl-0.08 Any comments -Michael ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC UPDATEproblem
Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 2005-08-18 02:46:15 UTC I am getting this error when the astcc.agi tries to UPDATE inuse = 0 LOG: unexpected EOF on client connection (postgres on Debian) I use another astcc.agi that UPDATEs to a different server (Postgres on OSX) and the inuse = 0 gets update properly Any ideas Michael Rodriguez ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1-800 number
More info On 8/17/05 3:34 AM, Christoph Eicke [EMAIL PROTECTED] wrote: Hi! I'm searching for a 1-800 number that simply plays music for a long time (3mins) and no one picks up. I've bothered the ATT lines so far when trying out my SIP-PSTN connection but then always someone answered :-) Anyone have a number? Christoph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] faxes
I have tested a fax call on asterisk with success. I used an IAXy on a broadband Time Warner connection. Faxes are much more sensitive than voice calls. If you have a good internet connection, faxes should complete fine. The only downfall it is recommended that you call to verify fax transmission after every fax. -Michael On 3/25/05 10:59 PM, AS [EMAIL PROTECTED] wrote: Is it possible and if so for a workstation user to send his fax via asterisk? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: asterisk+radius
I agree, why run to DBs. On the other hand, I have spoken with several people asking about radius support for asterisk because they have a billing solution that uses data from the radius servers to populate their billing DB. -Michael On 3/17/05 11:00 AM, Matthew Boehm [EMAIL PROTECTED] wrote: Kamran Ahmad wrote: i have written app for billing with asterisk. what is the problem in using radius. kamran Its a pain and redundant. Why run two seperate databases when 1 will do what you need? There is no native radius support for Asterisk. There is an addon, (search the wiki) but the last I heard of it, it was unstable. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI info digits question
Does anyone know how does asterisk handles INFO digit from a PRI line? Can info digit be used in extensions.conf to signal a call from a public phone? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail cuts off / hangs up
If I am not mistaken, I believe the dial command is omitted if you do not have a sound card configured on your system (loaded module). -michael On 12/2/04 1:07 AM, Matt Hess [EMAIL PROTECTED] wrote: Does cvs tag v1-0 not have a dial command? I do not seem to have one.. dial No such command 'dial' (type 'help' for help) Henry Devito wrote: Ok try this Login into console Set verbose 15 Dial (extension of VoiceMailMain app) Dial mailbox number Dial password Hangup Does it still die? See my example below asterisk*CLI dial 777 -- Executing VoiceMailMain(OSS/dsp, ) in new stack Console call has been answered -- Playing 'vm-login' (language 'en') asterisk*CLI dial 500 -- Playing 'vm-password' (language 'en') asterisk*CLI dial 1234 -- Playing 'vm-youhave' (language 'en') -- Playing 'digits/8' (language 'en') -- Playing 'vm-INBOX' (language 'en') -- Playing 'vm-messages' (language 'en') -- Playing 'vm-onefor' (language 'en') -- Playing 'vm-INBOX' (language 'en') -- Playing 'vm-messages' (language 'en') asterisk*CLI hangup -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Hess Sent: Wednesday, December 01, 2004 4:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] voicemail cuts off / hangs up Test completed successfully.. test dialplan: exten = 555,1,Answer exten = 555,2,Wait(2) exten = 555,3,Playback(digits/0) exten = 555,4,Playback(digits/1) exten = 555,5,Playback(digits/2) exten = 555,6,Playback(digits/3) exten = 555,7,Playback(digits/4) exten = 555,8,Playback(digits/5) exten = 555,9,Playback(digits/6) exten = 555,10,Playback(digits/7) exten = 555,11,Playback(digits/8) exten = 555,12,Playback(digits/9) exten = 555,13,Busy log: -- Executing Answer(SIP/3036284315-31b3, ) in new stack -- Executing Wait(SIP/3036284315-31b3, 2) in new stack -- Executing Playback(SIP/3036284315-31b3, digits/0) in new stack -- Playing 'digits/0' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/1) in new stack -- Playing 'digits/1' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/2) in new stack -- Playing 'digits/2' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/3) in new stack -- Playing 'digits/3' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/4) in new stack -- Playing 'digits/4' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/5) in new stack -- Playing 'digits/5' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/6) in new stack -- Playing 'digits/6' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/7) in new stack -- Playing 'digits/7' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/8) in new stack -- Playing 'digits/8' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/9) in new stack -- Playing 'digits/9' (language 'en') -- Executing Busy(SIP/3036284315-31b3, ) in new stack == Spawn extension (lwn, 555, 13) exited non-zero on 'SIP/3036284315-31b3' Henry Devito wrote: Try to play a number sound file by using the Playback application, I think the voicemail uses the same app to play the digits. See if that works. exten = 500,1,Playback(digits/3) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Hess Sent: Wednesday, December 01, 2004 3:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] voicemail cuts off / hangs up yup.. that's something I thought of as well.. and it's all there.. funny thing is.. I can start asterisk.. login just fine to voice mail.. I try again right away and I get that error that I had sent earlier and get cutoff.. Henry Devito wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Hess Sent: Wednesday, December 01, 2004 11:47 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] voicemail cuts off / hangs up I'm having a problem with voicemail where the system will allow me to login to the vm box no problem but when it starts tell tell me the number of messages I have it hangs up.. I get you have and it dies right there.. I'm running cvs tag v1-0.. what might be causing this? I looked through my mail list archive and didn't notice anything like this.. [*] Ok Did you check to see if the /var/lib/asterisk/sounds/digits/ directory is intact? I had an install at an earlier date from the CVS that did not download all of the sounds. Just a thought. ___ Asterisk-Users mailing list
RE: [Asterisk-Users] H.323 - NO AUDIO IN BOTH DIRECTIONS
FYI I am experiencing the same problem. I have complied asterisk from the latest CVS The call connects with no audio or DTMF to either end. I tested with ulaw and g729 with no success. -Michael On Fri, 2004-06-25 at 10:55, Scott Stingel wrote: Just checking that you have installed the proper versions of both OpenH323 and PWLib, as mentioned in ~/asterisk/channels/h232/README, and have rebuilt asterisk after those installations as specified? If so, then you are having the same problem I'm experiencing: no audio on H.323. I'm also connecting through a Cisco 5300. I'm just generating audio in one direction: outbound from asterisk - I hear nothing. This used to work I'm pretty sure... There is an outstanding bug report covering H.323 problems (#1334), not sure what the current status is. Cheers Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com http://www.evtmedia.com/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian Nocetti Sent: Friday, June 25, 2004 7:55 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] NO AUDIO IN BOTH DIRECTIONS hello all, I am having a trouble with Audio using h.323 channel... I am doing this Call comes into cisco 5300 and is sent to Asterisk, asterisk catch call with h.323 driver and send call to a SoftSwitch that routes the call, I can see log debug telling me, CALLED XXX, and then RINGING, and I can hear ring tones... but when call is answered, I DONT HEAR ANYTHING... I am using lastest ASTERISK download somebody can help me to solve this problem thanks..!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Michael K. Rodriguez Dialmex LLC Director of Network Operations 200 S. 10th Suite 1209 McAllen, TX 78501 (956) 994-0014 x107 office (956) 682-8521 fax (956) 239-0627 mobile
[Asterisk-Users] Registration Error
I am using a 7960 and it is registered to the *server, but I keep getting this error. Does anyone know why? NOTICE[5126]: File chan_sip.c, Line 3080 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '67.98.37.220' Michael K. Rodriguez DialMex LLC NOC Engineer 200 S. 10th Street Suite 1209 McAllen, TX 78501 (956) 994-0014 x107 office (956) 239-0627 mobile (956) 682-5821 fax [EMAIL PROTECTED] Escalation Procedure +++The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and destroy any copies of this document.+++
[Asterisk-Users] Asterisk to gateway
Is it possible to send a call from the asterisk server to a gateway via sipv2 protocol. I have some 7960 phones that can receive a call from a 5350 via sipv2 and the phone can send to the gateway via sipv2. Is there an exten that dials to a gateways ? Michael K. Rodriguez DialMex LLC NOC Engineer 200 S. 10th Street Suite 1209 McAllen, TX 78501 (956) 994-0014 x107 office (956) 239-0627 mobile (956) 682-5821 fax [EMAIL PROTECTED] Escalation Procedure +++The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and destroy any copies of this document.+++
[Asterisk-Users] Dialout Zap1/1
Any ideas on how to dialout exten = zap 1/1 Michael K. Rodriguez DialMex LLC NOC Engineer 200 S. 10th Street Suite 1209 McAllen, TX 78501 (956) 994-0014 x107 office (956) 239-0627 mobile (956) 682-5821 fax [EMAIL PROTECTED] Escalation Procedure +++The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and destroy any copies of this document.+++
RE: [Asterisk-Users] Dialout Zap1/1
I would like to dialout the line attached to zap/1. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Wednesday, March 26, 2003 1:19 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Dialout Zap1/1 On Wednesday 26 March 2003 12:55 pm, Michael K. Rodriguez wrote: Any ideas on how to dialout exten = zap 1/1 Do you want to Dial the station at Zap/1? Or do you want to dial out on the telephone line attached to Zap/1? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
