Re: [asterisk-users] WaitExten() always times out
As far as I can tell Asterisk recives media perfectly. For outgoing calls it looks something like this: -- Executing [...@proxy:5] WaitExten(SIP/voiptrunk-0083, 7) in new stack DEBUG[28557]: rtp.c:1032 process_rfc2833: - RTP 2833 Event: 0001 (len = 4) DEBUG[28557]: rtp.c:880 send_dtmf: Sending dtmf: 49 (1), at xx.xx.xxx.x On incoming, as far as I can tell, Asterisk does not recieve anything. I just don't know why. I have added exceptions in firewall and network to allow voip traffic, successfully allowing incoming and outgoing calls. Just no DTMF on incoming calls. My tests consist of a regular landline, I dial a DID and successfully reach my asterisk box. Everything is fine until I come to user input. None is recognized. I get a -User entered nothing and timeout. On Mon, Aug 23, 2010 at 8:07 AM, Miguel Molina mmol...@millenium.com.cowrote: El 20/08/10 16:14, Kathryn Jones escribió: Thanks for all the help, but I still can't find what's wrong. I enabled console = notice,warning,error,debug,dtmf like Miguel suggested. The output is attached. I noticed that the rtp.c session never starts, which as I understand is what catches the dtmf tone, but I could not find how to start it :s. The Answer() and waitExten(5,m) didn't fix my problem. I hope someone can help me see the problem after looking at the attached console output. The following line brought my attention: [Aug 20 16:50:04] DEBUG[5319]: channel.c:1882 __ast_answer: Didn't receive a media frame from SIP/xx.xx.xxx.xx-0026 within 500 ms of answering. Continuing anyway Are your sure that RTP audio (media) is correctly received in asterisk? I suspect network or firewall problems. Also, you said that you were going to receive calls from the PSTN, but are you testing from a SIP endpoint? Regards, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WaitExten() always times out
El 20/08/10 16:14, Kathryn Jones escribió: Thanks for all the help, but I still can't find what's wrong. I enabled console = notice,warning,error,debug,dtmf like Miguel suggested. The output is attached. I noticed that the rtp.c session never starts, which as I understand is what catches the dtmf tone, but I could not find how to start it :s. The Answer() and waitExten(5,m) didn't fix my problem. I hope someone can help me see the problem after looking at the attached console output. The following line brought my attention: [Aug 20 16:50:04] DEBUG[5319]: channel.c:1882 __ast_answer: Didn't receive a media frame from SIP/xx.xx.xxx.xx-0026 within 500 ms of answering. Continuing anyway Are your sure that RTP audio (media) is correctly received in asterisk? I suspect network or firewall problems. Also, you said that you were going to receive calls from the PSTN, but are you testing from a SIP endpoint? Regards, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WaitExten() always times out
Thanks for all the help, but I still can't find what's wrong. I enabled console = notice,warning,error,debug,dtmf like Miguel suggested. The output is attached. I noticed that the rtp.c session never starts, which as I understand is what catches the dtmf tone, but I could not find how to start it :s. The Answer() and waitExten(5,m) didn't fix my problem. I hope someone can help me see the problem after looking at the attached console output. On Thu, Aug 19, 2010 at 2:46 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Miguel Molina *Subject:* Re: [asterisk-users] WaitExten() always times out snip Til gave you the answer; When you call out the other end controls timing. Put a waitexten(5,m) in front of background(welcome) and see if that helps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Executing [...@default:1] Answer(SIP/xx.xx.xxx.xx-0026, ) in new stack [Aug 20 16:50:04] DEBUG[5319]: chan_sip.c:6197 sip_answer: SIP answering channel: SIP/xx.xx.xxx.xx-0026 [Aug 20 16:50:04] DEBUG[5319]: chan_sip.c:10426 transmit_response_with_sdp: Setting framing from config on incoming call [Aug 20 16:50:04] DEBUG[5319]: chan_sip.c:10115 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Aug 20 16:50:04] DEBUG[5319]: chan_sip.c:10116 add_sdp: ** Our prefcodec: 0x0 (nothing) [Aug 20 16:50:04] DEBUG[5319]: chan_sip.c:10227 add_sdp: -- Done with adding codecs to SDP [Aug 20 16:50:04] DEBUG[5319]: channel.c:3096 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan-timingfd=29) [Aug 20 16:50:04] DEBUG[5319]: chan_sip.c:10363 add_sdp: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Aug 20 16:50:04] DEBUG[5319]: chan_sip.c:3557 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for xx.xx.xxx.xx:5060 [Aug 20 16:50:04] DEBUG[1188]: devicestate.c:342 _ast_device_state: No provider found, checking channel drivers for SIP - xx.xx.xxx.xx [Aug 20 16:50:04] DEBUG[1188]: chan_sip.c:23000 sip_devicestate: Checking device state for peer xx.xx.xxx.xx [Aug 20 16:50:04] DEBUG[1188]: devicestate.c:460 do_state_change: Changing state for SIP/xx.xx.xxx.xx - state 2 (In use) [Aug 20 16:50:04] DEBUG[1188]: devicestate.c:440 devstate_event: device 'SIP/xx.xx.xxx.xx' state '2' [Aug 20 16:50:04] DEBUG[1188]: devicestate.c:342 _ast_device_state: No provider found, checking channel drivers for SIP - xx.xx.xxx.xx [Aug 20 16:50:04] DEBUG[1188]: chan_sip.c:23000 sip_devicestate: Checking device state for peer xx.xx.xxx.xx [Aug 20 16:50:04] DEBUG[1188]: devicestate.c:460 do_state_change: Changing state for SIP/xx.xx.xxx.xx - state 2 (In use) [Aug 20 16:50:04] DEBUG[1188]: devicestate.c:440 devstate_event: device 'SIP/xx.xx.xxx.xx' state '2' [Aug 20 16:50:04] DEBUG[1251]: app_queue.c:1084 handle_statechange: Device 'SIP/xx.xx.xxx.xx' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 20 16:50:04] DEBUG[1251]: app_queue.c:1084 handle_statechange: Device 'SIP/xx.xx.xxx.xx' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 20 16:50:04] DEBUG[5319]: channel.c:1882 __ast_answer: Didn't receive a media frame from SIP/xx.xx.xxx.xx-0026 within 500 ms of answering. Continuing anyway [Aug 20 16:50:04] DEBUG[5319]: pbx.c:3692 pbx_extension_helper: Launching 'WaitExten' -- Executing [...@default:2] WaitExten(SIP/xx.xx.xxx.xx-0026, 10,m) in new stack -- Started music on hold, class 'default', on SIP/xx.xx.xxx.xx-0026 [Aug 20 16:50:04] DEBUG[5319]: channel.c:2426 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 20 16:50:04] DEBUG[5319]: channel.c:3727 set_format: Set channel SIP/xx.xx.xxx.xx-0026 to write format slin [Aug 20 16:50:04] DEBUG[5319]: res_musiconhold.c:303 ast_moh_files_next: SIP/xx.xx.xxx.xx-0026 Opened file 1 '/var/lib/asterisk/moh/manolo_camp-morning_coffee' [Aug 20 16:50:04] DEBUG[5319]: rtp.c:3832 ast_rtp_write: Ooh, format changed from unknown to ulaw [Aug 20 16:50:04] DEBUG[5319]: rtp.c:3858 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Aug 20 16:50:05] DEBUG[1232]: chan_sip.c:3758 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #6725)) [Aug 20 16:50:05] DEBUG[1232]: chan_sip.c:3557 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for xx.xx.xxx.xx:5060 [Aug 20 16:50:06] DEBUG[1232]: chan_sip.c:3758
Re: [asterisk-users] WaitExten() always times out
On Wednesday 18 August 2010 16:52:38 Kathryn Jones wrote: I must not be receiving them properly, since I can't make it work. I just can't see why :P. My asterisk version is 1.6.2.6. Like I said before, for outgoing .call files WaitExten works fine, it's on incoming calls that I cannot receive the number I need. There's your answer. On outgoing calls, the other end signals the line into answered state, whereas on incoming calls, you must explicitly answer the channel before listening for DTMF. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WaitExten() always times out
Thanks for your reply :)
I added Answer to my dialplan:
exten = xxx,1,Answer()
exten = xxx,n,Background(welcome)
exten = xxx,n,WaitExten(7)
exten = _X,1,AGI(agi.php)
exten = _X,n,PlayBack(vm-tocallnumber)
exten = _X,n,Dial(SIP/voiptrunk/${NUM})
exten = t,1,Noop(*timeout*)
exten = t,n,Playback(pbx-invalid)
exten = t,n,Hangup()
cli output:
-- Executing [...@default:1] Answer(SIP/xx.xx.xx.xx-0004, ) in new
stack
-- Executing [...@default:2] BackGround(SIP/xx.xx.xx.xx-0004,
welcome) in new stack
-- SIP/xx.xx.xx.xx-0004 Playing 'welcome.slin' (language 'en')
-- Executing [...@default:3] WaitExten(SIP/xx.xx.xx.xx-0004, 7)
in new stack
-- Timeout on SIP/xx.xx.xx.xx-0004, going to 't'
-- Executing [...@default:1] NoOp(SIP/xx.xx.xx.xx-0004,
*timeout*) in new stack
-- Executing [...@default:2] Playback(SIP/xx.xx.xx.xx-0004,
pbx-invalid) in new stack
-- SIP/xx.xx.xx.xx-0004 Playing 'pbx-invalid.gsm' (language 'en')
-- Executing [...@default:3] Hangup(SIP/xx.xx.xx.xx-0004, ) in new
stack
== Spawn extension (default, t, 3) exited non-zero on
'SIP/xx.xx.xx.xx-0004'
[] WARNING[1235]: chan_sip.c:3780 retrans_pkt: Maximum retries exceeded on
transmission 0ef328f40a5fd6ca31a68dae2af75...@xx.xx.xx.xx for seqno 102
(Critical Response) -- See doc/sip-retransmit.txt.
[] WARNING[1235]: chan_sip.c:3780 retrans_pkt: Maximum retries exceeded on
transmission 0ef328f40a5fd6ca31a68dae2af75...@xx.xx.xx.xx for seqno 102
(Critical Response) -- See doc/sip-retransmit.txt.
I still can't read the DTMF input :(
I also tried adding:
dtmfmode = rfc2833
rfc2833compensate = yes
relaxdmtf = no ; should be no because setting it to yes cause talkoff
to sip.conf and chan_dahdi.conf
and increasing rxgain=20 (I wasn't sure how much was appropriate)
Nothing seems to help.
ANY tips or ideas will be apreciated.
On Thu, Aug 19, 2010 at 1:19 PM, Tilghman Lesher tles...@digium.com wrote:
On Wednesday 18 August 2010 16:52:38 Kathryn Jones wrote:
I must not be receiving them properly, since I can't make it work. I just
can't see why :P.
My asterisk version is 1.6.2.6. Like I said before, for outgoing .call
files WaitExten works fine, it's on incoming calls that I cannot receive
the number I need.
There's your answer. On outgoing calls, the other end signals the line
into
answered state, whereas on incoming calls, you must explicitly answer the
channel before listening for DTMF.
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com www.asterisk.org
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] WaitExten() always times out
El 19/08/10 15:07, Kathryn Jones escribió:
Thanks for your reply :)
I added Answer to my dialplan:
exten = xxx,1,Answer()
exten = xxx,n,Background(welcome)
exten = xxx,n,WaitExten(7)
exten = _X,1,AGI(agi.php)
exten = _X,n,PlayBack(vm-tocallnumber)
exten = _X,n,Dial(SIP/voiptrunk/${NUM})
exten = t,1,Noop(*timeout*)
exten = t,n,Playback(pbx-invalid)
exten = t,n,Hangup()
cli output:
-- Executing [...@default:1] Answer(SIP/xx.xx.xx.xx-0004, ) in
new stack
-- Executing [...@default:2]
BackGround(SIP/xx.xx.xx.xx-0004, welcome) in new stack
-- SIP/xx.xx.xx.xx-0004 Playing 'welcome.slin' (language 'en')
-- Executing [...@default:3] WaitExten(SIP/xx.xx.xx.xx-0004,
7) in new stack
-- Timeout on SIP/xx.xx.xx.xx-0004, going to 't'
-- Executing [...@default:1] NoOp(SIP/xx.xx.xx.xx-0004,
*timeout*) in new stack
-- Executing [...@default:2] Playback(SIP/xx.xx.xx.xx-0004,
pbx-invalid) in new stack
-- SIP/xx.xx.xx.xx-0004 Playing 'pbx-invalid.gsm' (language
'en')
-- Executing [...@default:3] Hangup(SIP/xx.xx.xx.xx-0004, )
in new stack
== Spawn extension (default, t, 3) exited non-zero on
'SIP/xx.xx.xx.xx-0004'
[] WARNING[1235]: chan_sip.c:3780 retrans_pkt: Maximum retries
exceeded on transmission 0ef328f40a5fd6ca31a68dae2af75...@xx.xx.xx.xx
for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt.
[] WARNING[1235]: chan_sip.c:3780 retrans_pkt: Maximum retries
exceeded on transmission 0ef328f40a5fd6ca31a68dae2af75...@xx.xx.xx.xx
for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt.
I still can't read the DTMF input :(
I also tried adding:
dtmfmode = rfc2833
rfc2833compensate = yes
relaxdmtf = no ; should be no because setting it to yes cause talkoff
to sip.conf and chan_dahdi.conf
and increasing rxgain=20 (I wasn't sure how much was appropriate)
Nothing seems to help.
ANY tips or ideas will be apreciated.
On Thu, Aug 19, 2010 at 1:19 PM, Tilghman Lesher tles...@digium.com
mailto:tles...@digium.com wrote:
On Wednesday 18 August 2010 16:52:38 Kathryn Jones wrote:
I must not be receiving them properly, since I can't make it
work. I just
can't see why :P.
My asterisk version is 1.6.2.6. Like I said before, for outgoing
.call
files WaitExten works fine, it's on incoming calls that I cannot
receive
the number I need.
There's your answer. On outgoing calls, the other end signals the
line into
answered state, whereas on incoming calls, you must explicitly
answer the
channel before listening for DTMF.
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com http://www.digium.com
www.asterisk.org http://www.asterisk.org
I suggest you to debug DTMF and core, enabling them in logger.conf:
console = notice,warning,error,debug,dtmf
And issuing a logger reload command in asterisk CLI.
A rxgain of 20 is too much for me, leave them in rxgain = 0.0 and
txgain= 0.0. Maybe 20dB gain is high enough to distort the audio signal
and make DTMF detection more difficult.
Look at the DTMF events in your CLI, that way you can debug better. You
can enable core debug if you want issuing the CLI command core set
debug X, with X on 1 or 2, and setting it off when you want.
If your call is received from the PSTN, asterisk will detect the inband
DTMF tones in the audio signal. The rfc2833 configurations are only for
VoIP endpoints.
Good luck in your debugging,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
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Re: [asterisk-users] WaitExten() always times out
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel Molina Subject: Re: [asterisk-users] WaitExten() always times out snip Til gave you the answer; When you call out the other end controls timing. Put a waitexten(5,m) in front of background(welcome) and see if that helps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WaitExten() always times out
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kathryn Jones Subject: [asterisk-users] WaitExten() always times out Hi, My WaitExten() is not working as I expect it to. This is the relevant part of my context. It is meant to receive incoming calls. [incoming] exten = xxx,1,Background(hello-world) exten = xxx,2,WaitExten(7) exten = _X,1,AGI(myAGI.php) When I send the call from a .call, it works perfect, but when receiving an incoming call WaitExten() times out no matter what. snip I experimented changing autofallthrough to no and got the same result. Any ideas about this strange behavior? My best guess is that your problem is that _X isn't happy for whatever reason. Generally I use Waitexten for single digit processing like this Exten = 1234,1,goto(waitdtmf,s,1) [waitdtmf] Exten = s,1,background(hello-world) Exten = s,n,waitexten(7) Exten = 1,1,AGI(myAGI.php) Exten = 2,1,AGI(myAGI.php) Exten = I,1,playback(invalid) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WaitExten() always times out
Thanks for you reply :). I thought of that and tried replacing _X with a numbers it should match (9), and it didn't work. It still times out as if no number was entered. On Wed, Aug 18, 2010 at 2:11 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Kathryn Jones *Subject:* [asterisk-users] WaitExten() always times out Hi, My WaitExten() is not working as I expect it to. This is the relevant part of my context. It is meant to receive incoming calls. [incoming] exten = xxx,1,Background(hello-world) exten = xxx,2,WaitExten(7) exten = _X,1,AGI(myAGI.php) When I send the call from a .call, it works perfect, but when receiving an incoming call WaitExten() times out no matter what. snip I experimented changing autofallthrough to no and got the same result. Any ideas about this strange behavior? My best guess is that your problem is that _X isn’t happy for whatever reason. Generally I use Waitexten for single digit processing like this Exten = 1234,1,goto(waitdtmf,s,1) [waitdtmf] Exten = s,1,background(hello-world) Exten = s,n,waitexten(7) Exten = 1,1,AGI(myAGI.php) Exten = 2,1,AGI(myAGI.php) Exten = I,1,playback(invalid) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WaitExten() always times out
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kathryn Jones Subject: Re: [asterisk-users] WaitExten() always times out Thanks for you reply :). I thought of that and tried replacing _X with a numbers it should match (9), and it didn't work. It still times out as if no number was entered. When you do the .call, it is probably on a local, SIP or IAX channel. When you hit the incoming, are you on a DAHDI channel? Also, a workaround would be to do Exten = t,1,AGI(myagi.php) So when the DTMF doesn't work it just drops through anyway. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WaitExten() always times out
My .call file goes out to a pstn number. That work around would be perfect :D, but I need the number given by the caller. On Wed, Aug 18, 2010 at 2:49 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Kathryn Jones *Subject:* Re: [asterisk-users] WaitExten() always times out Thanks for you reply :). I thought of that and tried replacing _X with a numbers it should match (9), and it didn't work. It still times out as if no number was entered. When you do the .call, it is probably on a local, SIP or IAX channel. When you hit the incoming, are you on a DAHDI channel? Also, a workaround would be to do Exten = t,1,AGI(myagi.php) So when the DTMF doesn’t work it just drops through anyway. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WaitExten() always times out
Hi, Are you sure asterisk is receiving and processing DMTF correctly? Are you using rfc2833, SIP INFO or inband DMTF? What is your asterisk version? I use WaitExten(5) all the time, no matter if they are single-digit or multiple-digit extensions. Regards, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 El 18/08/10 15:39, Kathryn Jones escribió: Thanks for you reply :). I thought of that and tried replacing _X with a numbers it should match (9), and it didn't work. It still times out as if no number was entered. On Wed, Aug 18, 2010 at 2:11 PM, Danny Nicholas da...@debsinc.com mailto:da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Kathryn Jones *Subject:* [asterisk-users] WaitExten() always times out Hi, My WaitExten() is not working as I expect it to. This is the relevant part of my context. It is meant to receive incoming calls. [incoming] exten = xxx,1,Background(hello-world) exten = xxx,2,WaitExten(7) exten = _X,1,AGI(myAGI.php) When I send the call from a .call, it works perfect, but when receiving an incoming call WaitExten() times out no matter what. snip I experimented changing autofallthrough to no and got the same result. Any ideas about this strange behavior? My best guess is that your problem is that _X isn’t happy for whatever reason. Generally I use Waitexten for single digit processing like this Exten = 1234,1,goto(waitdtmf,s,1) [waitdtmf] Exten = s,1,background(hello-world) Exten = s,n,waitexten(7) Exten = 1,1,AGI(myAGI.php) Exten = 2,1,AGI(myAGI.php) Exten = I,1,playback(invalid) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WaitExten() always times out
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kathryn Jones Subject: Re: [asterisk-users] WaitExten() always times out My .call file goes out to a pstn number. That work around would be perfect :D, but I need the number given by the caller. My bet is that the pstn/DAHDI delay is eating part of your message (it takes 3-7 seconds from Dial to actually connect). Try putting a wait(5) in front of the Background command. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WaitExten() always times out
I must not be receiving them properly, since I can't make it work. I just can't see why :P. My asterisk version is 1.6.2.6. Like I said before, for outgoing .call files WaitExten works fine, it's on incoming calls that I cannot receive the number I need. I had not checked my dtmf mode, this is new to me. So I was using asterisk default rfc2833. I am making pstn calls from regular telephones, through asterisk. What dtmfmode should I use? Could that be my problem? On Wed, Aug 18, 2010 at 2:57 PM, Miguel Molina mmol...@millenium.com.cowrote: Hi, Are you sure asterisk is receiving and processing DMTF correctly? Are you using rfc2833, SIP INFO or inband DMTF? What is your asterisk version? I use WaitExten(5) all the time, no matter if they are single-digit or multiple-digit extensions. Regards, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 El 18/08/10 15:39, Kathryn Jones escribió: Thanks for you reply :). I thought of that and tried replacing _X with a numbers it should match (9), and it didn't work. It still times out as if no number was entered. On Wed, Aug 18, 2010 at 2:11 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Kathryn Jones *Subject:* [asterisk-users] WaitExten() always times out Hi, My WaitExten() is not working as I expect it to. This is the relevant part of my context. It is meant to receive incoming calls. [incoming] exten = xxx,1,Background(hello-world) exten = xxx,2,WaitExten(7) exten = _X,1,AGI(myAGI.php) When I send the call from a .call, it works perfect, but when receiving an incoming call WaitExten() times out no matter what. snip I experimented changing autofallthrough to no and got the same result. Any ideas about this strange behavior? My best guess is that your problem is that _X isn’t happy for whatever reason. Generally I use Waitexten for single digit processing like this Exten = 1234,1,goto(waitdtmf,s,1) [waitdtmf] Exten = s,1,background(hello-world) Exten = s,n,waitexten(7) Exten = 1,1,AGI(myAGI.php) Exten = 2,1,AGI(myAGI.php) Exten = I,1,playback(invalid) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
