[asterisk-users] Strange problem with zap channel.
I am trying to help a guy out with his Atcom IP04. He has set it up like this. He has a handful of IP phones all connecting via SIP. He has two phone lines connected to the FXO ports one from telecom, another from vodaphone. He has set up the dialplan so that one of the trunks fails over to the other trunk. Everything seems to be working OK except for outgoing calls. He can call from extension to extension without problems. If I call in on either of the trunk lines we can have a normal conversation. If he calls out to me he can hear me but I can't hear him. The status on GUI shows the phone as still ringing even though I picked up and he can hear me. Here is a log of one of the calls. If anybody can offer a clue as to what the problem might be I'd be grateful. I looked at the port definitions and they are set up for NZ signaling (kewl loop). [Jun 6 13:24:29] DEBUG[4667]: app_macro.c:337 _macro_exec: Executed application: Dial -- Executing [1-d...@macro-trunkdial-failover-0.3:2] GotoIf(SIP/6006-015d0004, 16 0 ?1-BUSY|1:1-out|1) in new stack -- Goto (macro-trunkdial-failover-0.3,1-BUSY,1) [Jun 6 13:24:29] DEBUG[4667]: app_macro.c:337 _macro_exec: Executed application: Gotoif == Auto fallthrough, channel 'SIP/6006-015d0004' status is 'BUSY' -- Executing [9075763...@dlpn_dialplan1:1] Macro(SIP/6006-015d0004, trunkdial-failover-0.3|Zap/g1/075763441|Zap/g2/075763441|trunk_1|trunk_2) in new stack -- Executing [...@macro-trunkdial-failover-0.3:1] Set(SIP/6006-015d0004, CALLERID(num)=6498287700) in new stack [Jun 6 13:31:41] DEBUG[4825]: app_macro.c:337 _macro_exec: Executed application: Set -- Executing [...@macro-trunkdial-failover-0.3:2] GotoIf(SIP/6006-015d0004, 1?1-dial|1) in new stack -- Goto (macro-trunkdial-failover-0.3,1-dial,1) [Jun 6 13:31:41] DEBUG[4825]: app_macro.c:337 _macro_exec: Executed application: GotoIf -- Executing [1-d...@macro-trunkdial-failover-0.3:1] Dial(SIP/6006-015d0004, Zap/g1/075763441) in new stack [Jun 6 13:31:41] DEBUG[4825]: dsp.c:1787 ast_dsp_set_busy_pattern: dsp busy pattern set to 0,0 [Jun 6 13:31:41] DEBUG[4825]: chan_zap.c:1952 zt_call: Dialing '075763441' [Jun 6 13:31:41] DEBUG[4825]: chan_zap.c:2028 zt_call: Deferring dialing... -- Called g1/075763441 [Jun 6 13:31:42] DEBUG[4825]: chan_zap.c: zt_handle_event: Sent deferred digit string: T075763441w [Jun 6 13:31:44] DEBUG[4825]: chan_zap.c:3788 zt_handle_event: Done dialing, but waiting for progress detection before doing more... At this point I have picked up the phone and am speaking, he can hear me but I can't hear him. After I hang up I get this. [Jun 6 13:32:02] DEBUG[4825]: dsp.c:1445 ast_dsp_busydetect: ast_dsp_busydetect detected busy, avgtone: 255, avgsilence 240 -- Zap/1-1 is busy -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:1/0/0) [Jun 6 13:32:02] DEBUG[4825]: app_macro.c:337 _macro_exec: Executed application: Dial -- Executing [1-d...@macro-trunkdial-failover-0.3:2] GotoIf(SIP/6006-015d0004, 16 0 ?1-BUSY|1:1-out|1) in new stack -- Goto (macro-trunkdial-failover-0.3,1-BUSY,1) [Jun 6 13:32:02] DEBUG[4825]: app_macro.c:337 _macro_exec: Executed application: Gotoif == Auto fallthrough, channel 'SIP/6006-015d0004' status is 'BUSY' -- _ -- 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] IVR for asterisk
On Wed, Nov 25, 2009 at 1:12 AM, B.Masoud @ SH i...@saudihome.com wrote: Anyone can recommend a commercial large scale IVR with easy + pro management for asterisk? I don't know what you mean by pro management but you can write IVR applications in any language you want. Personally I like ruby but you can do it java, pyton, php, perl, erlang etc. Drop me a private email if you want to know more. Writing IVRs is not difficult but there are issues you need to think about especially if you are concerned about uptime, load balancing etc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can't get pap2 to register from outside the LAN.
On Tue, Nov 24, 2009 at 4:16 PM, Landy Landy landysacco...@yahoo.com wrote: How about adding: insecure=invite,port That didn't work. How weird. I have reset the device to factory settings too. Nothing seems to work. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can't get pap2 to register from outside the LAN.
On Tue, Nov 24, 2009 at 3:38 PM, Michael Wyres mwy...@cdm.com.au wrote: I would without the deny and permit directives in the SIP, and rule out some sort of clash there that is rejecting the address the registration is coming from, and take it from there. It made no difference to remove those entries. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] can't get pap2 to register from outside the LAN.
I am having a hell of a problem trying to get a linksys pap2t to register with my asterisk from outside the LAN. I have tried every combination of NAT, outbound proxy, stun, specify external IP address etc and it just won't work. Here are the relevant details. In asterisk I have set the following. externip=my.ip.address localnet=192.168.0.0/255.255.0.0 nat=yes bindport=5060 here is the sip user deny=0.0.0.0/0.0.0.0 type=friend secret=blah qualify=yes port=5060 pickupgroup= permit=0.0.0.0/0.0.0.0 nat=yes mailbox=...@device host=dynamic dtmfmode=rfc2833 dial=SIP/372 context=from-internal canreinvite=no callgroup= callerid=device 372 accountcode= call-limit=50 I have tried nat = no, nat=never, nat=route, and leaving out the nat no difference. On the linksys end I have tried everything I can think of. Nat, no nat, stun, hard coded external IP address etc. I have read dozens of web sites and have tried every suggestion given but no joy. I know other people have had the same problem but none of the links I ran into had a solution that worked for me. This device connects perfectly when inside the lan, take it out and it won't connect no matter what I do. Here is the sip debug trace. What truly puzzles me is the 401 not authorized packets. The password is correct, it connects fine inside the lan but the same username and password fails outside the LAN. [Nov 24 14:18:41] --- Transmitting (NAT) to 218.101.6.157:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d;received=218.101.6.157 From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108;tag=as1f31845b Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=0dc307da Content-Length: 0 [Nov 24 14:18:41] Scheduling destruction of SIP dialog 'f4e6d9bc-59a7c...@192.168.50.183' in 32000 ms (Method: REGISTER) [Nov 24 14:18:42] ip --- SIP read from 218.101.6.157:5060 --- REGISTER sip:203.109.148.108 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108 Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER Max-Forwards: 70 Contact: 372 sip:3...@192.168.50.183:5060;expires=3600 User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces - [Nov 24 14:18:42] --- (12 headers 0 lines) --- [Nov 24 14:18:42] Using latest REGISTER request as basis request [Nov 24 14:18:42] Sending to 218.101.6.157 : 5060 (NAT) [Nov 24 14:18:42] --- Transmitting (NAT) to 218.101.6.157:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d;received=218.101.6.157 From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108 Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:3...@203.109.148.108 Content-Length: 0 [Nov 24 14:18:42] --- Transmitting (NAT) to 218.101.6.157:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d;received=218.101.6.157 From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108;tag=as1f31845b Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=0dc307da Content-Length: 0 [Nov 24 14:18:42] Scheduling destruction of SIP dialog 'f4e6d9bc-59a7c...@192.168.50.183' in 32000 ms (Method: REGISTER) [Nov 24 14:18:44] ip --- SIP read from 218.101.6.157:5060 --- REGISTER sip:203.109.148.108 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108 Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER Max-Forwards: 70 Contact: 372 sip:3...@192.168.50.183:5060;expires=3600 User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces - [Nov 24 14:18:44] --- (12 headers 0 lines) --- [Nov 24 14:18:44] Using latest REGISTER request as basis request [Nov 24 14:18:44] Sending to 218.101.6.157 : 5060 (NAT) [Nov 24 14:18:44] --- Transmitting (NAT) to 218.101.6.157:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d;received=218.101.6.157 From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108 Call-ID:
Re: [asterisk-users] Re: unable to get channel lock BAD BAD BAD
Pre 1.2.8, chan_sip would try to get the lock indefinitely, and would basically hang the thread until the lock was fetched. Since 1.2.8, it gives up after 100 tries and logs that message. Of course that does not explain why the lock is failing... Judging by the lack of response here it seems like this is broken and nobody knows how to fix it. It sucks when you are having a problem and there is no answer. I googled and asked everywhere but nobody knows the answer. It's probably a bug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: unable to get channel lock BAD BAD BAD
98% of the people here don't use Trixbox. I don't think this is something with trixbox. I asked the person having the same problem as me if he was using trixbox to see if that would narrow down the realm of the problem. Anyway the error message is in the asterisk log. Googling around I see that other people have posted the problem and nobody has gotten an answer. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: unable to get channel lock BAD BAD BAD
I can confirm the BAD! BAD! BAD! message on both our servers running 1.2.13. Our servers running 1.2.7.1 do not exhibit the problem. Pre 1.2.8, chan_sip would try to get the lock indefinitely, and would basically hang the thread until the lock was fetched. Since 1.2.8, it gives up after 100 tries and logs that message. I am running Asterisk 1.2.12.1 svn rev 42879. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: unable to get channel lock BAD BAD BAD
I'm sure I'm not the only one, that when seeing the message posted, I'm having x issues with Trixbox I press the delete button most of the time. Your lack of response might be because of this. On the on the one hand I can understand this because trixbox has it's own configuration schema and it's full of software any one of which can be causing the problem. On the other hand I am very disapointed to hear this. The trixbox forums are not much help either because most of the users are newbies (like me) who are attracted to the project because it's easy to get up and going with asterisk. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] trixbox + agi
For the Asterisk side of things, are you using Asterisk directly or Trixbox? I'm just trying to get a prototype working so don't want to spend a lot of time on the initial asterisk setup. If Trixbox will allow me to do the php+agi integration, I'll do that, if not, will try to just try to install Asterisk from source. You can use trixbox but be aware of the following. Trixbox scatters it's config files. Some stuff is kept in the database, some in the conf files. You have to keep your configuration in specific files that won't be overrritten. Trixbox has it's own contexts for everything so when people give you instructions that work on a plain jane asterisk box it won't work. Trixbox does not have a mailing list. The forums suck. There is no real support from anybody. Everybody is asking questions and maybe somebody will answer your question maybe they wont. People who use trixbox are not writing AGI scripts by and large so I don't think you will get any help in that regard at all. I am using trixbox and I am starting to feel like I would have been better off with just a plain asterisk box for my agi work. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: unable to get channel lock BAD BAD BAD
Yes, I get same error message in my log. Anybody has any info on this one? Are you using trixbox? It would be nice to try and isolate this problem by ruling out a bad config in trixbox. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] unable to get channel lock BAD BAD BAD
I am seeing the following in my log file (standard trixbox install).
One seems to be complaining about an error in the dialplan but it
won't tell me what file or what line. The other (maybe related) is
complaining about a channel lock.
How to do go about trying to figure out what the problem is and how to solve it?
---Logfile
Nov 14 07:20:19 WARNING[14067] ast_expr2.fl: ast_yyerror(): syntax
error: syntax error, unexpected $end, expecting TOK_MINUS or TOK_COMPL
or TOK_LP or TOKEN; Input:
!=
^
Nov 14 07:20:20 WARNING[14067] ast_expr2.fl: If you have questions,
please refer to doc/README.variables in the asterisk source.
Nov 14 07:20:44 ERROR[24091] chan_sip.c: We could NOT get the channel
lock for SIP/101-082695f0!
Nov 14 07:20:44 ERROR[24091] chan_sip.c: SIP MESSAGE JUST IGNORED: ACK
Nov 14 07:20:44 ERROR[24091] chan_sip.c: BAD! BAD! BAD!
Nov 14 07:20:45 ERROR[24091] chan_sip.c: We could NOT get the channel
lock for SIP/101-082695f0!
Nov 14 07:20:45 ERROR[24091] chan_sip.c: SIP MESSAGE JUST IGNORED: ACK
Nov 14 07:20:45 ERROR[24091] chan_sip.c: BAD! BAD! BAD!
Nov 14 07:20:46 ERROR[24091] chan_sip.c: We could NOT get the channel
lock for SIP/101-082695f0!
Nov 14 07:20:46 ERROR[24091] chan_sip.c: SIP MESSAGE JUST IGNORED: ACK
Nov 14 07:20:46 ERROR[24091] chan_sip.c: BAD! BAD! BAD!
--LInes in the dialplan with != in them--
asterisk -r -x 'show dialplan' | grep !=
's' =1. GotoIf($[${REALCALLERIDNUM:1:2} != ]?start)
[pbx_config]
2. GotoIf($[${REALCALLERIDNUM:1:2} != ]?start)
[pbx_config]
2.
GotoIf($[${CALLERID(name):0:${LEN(${RGPREFIX})}} !=
${RGPREFIX}]?4:3) [pbx_config]
2.
Set(VMGAIN=${IF($[foo${VM_GAIN}!=foo]?g(${VM_GAIN}):)})
[pbx_config]
'docfu' =1.
Set(RTCFU=${IF($[${VMBOX}!=novm]?${RINGTIMER}:)}) [pbx_config]
6. Set(RT=${IF($[$[${VMBOX}!=novm] |
$[foo${CFUEXT}!=foo]]?${RINGTIMER}:)}) [pbx_config]
10. GosubIf($[$[${DIALSTATUS}=NOANSWER]
$[foo${CFUEXT}!=foo]]?docfu|1) [pbx_config]
31. GotoIf($[$[${CALLTRACE_HUNT} != ]
$[${RingGroupMethod} = hunt ]]?32:35 ) [pbx_config]
35. GotoIf($[$[${CALLTRACE_HUNT} != ]
$[${RingGroupMethod} = memoryhunt ]]?36:50 ) [pbx_config]
2.
GotoIf($[${CALLERID(name):0:${LEN(${RGPREFIX})}} !=
${RGPREFIX}]?NEWPREFIX) [pbx_config]
2.
GotoIf($[${CALLERID(name):0:${LEN(${RGPREFIX})}} !=
${RGPREFIX}]?NEWPREFIX) [pbx_config]
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] trixbox + agi
On 11/15/06, blackwater dev [EMAIL PROTECTED] wrote: I need to write an app which takes a phone call, asks for the user to input a number and then queries a db via a webservice and reads the results a row at a time back to the caller. First, is this beyond Asterisk? Second, can I do this if I use the Trixbox implementation? Third, any good tutorials on doing just this? There are numerous AGI toolkits in different languages. I have just started fooling around with RAGI which is integrated with ruby on rails. From my experiments so far it seems to work OK. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] trixbox + agi
If I were you I would go the AGI way. Use ruby, python, php, perl, java, c# or even erlang. Aything but the asterisk dialplan commands. There is no sense in putting yourself through that pain. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trixbox dialout problems
Hello All.
I am trying to use RAGI the ruby agi framework with trixbox. I am
having a problem
with the dialout part. The RAGI framework creates a file in the
/var/spool/asterisk/outgoing directory and routes the call to an
extension (I have listed the relevent portion of the file below). The
problem is that the initial dial command does not execute properly in
trixbox. I am hoping somebody who has expertise in trixbox could help
me debug this problem. I can post an asterisk log snippet if anybody
is interested.
Here is the extension. The callout file is below too.
Extension_custom.conf
[dialout]
exten = outbound,1,Answer ; switches to outbound-handler
exten = outbound,2,Wait(60)
exten = outbound,3,Hangup
exten =
outbound-handler,1,Dial(${CallInitiate_phonenumber},50,gM(outbound-connect^${AGI_SERVER}${AGI_URL}^${CallInitiate_hashdata}^${MACHINE_STATUS_UNKNOWN}))
exten = outbound-handler,2,GotoIf($[${DIALSTATUS} = ANSWER]?104)
exten = outbound-handler,3,NoOp(status=${DIALSTATUS},
DIALEDTIME=${DIALEDTIME}, ANSWEREDTIME=${ANSWEREDTIME})
exten =
outbound-handler,4,SetVar(CallInitiate_hashdata=${CallInitiate_hashdata})
exten = outbound-handler,5,deadagi(agi://${AGI_SERVER}${AGI_URL})
;DIAL_STATUS is busy, etc.
exten = outbound-handler,6,Goto(104)
exten =
outbound-handler,102,SetVar(CallInitiate_hashdata=${CallInitiate_hashdata})
exten = outbound-handler,103,deadagi(agi://${AGI_SERVER}${AGI_URL})
;DIAL_STATUS is busy, etc.
exten = outbound-handler,104,Hangup()
[macro-outbound-connect]
exten = s,1,Answer()
exten = s,2,SetVar(CallInitiate_hashdata=${ARG2})
exten = s,3,SetVar(machinestatus=${ARG3})
exten = s,4,deadagi(agi://${ARG1})
exten = s,5,Hangup
Callout file-
;This file was generated by RAGI's callInitiate class
;File generated date: 11-07-2006 at 12:47 -- Tuesday
;Call date: 11-07-2006 at 12:47 -- Tuesday
Channel: Local/[EMAIL PROTECTED]
Callerid: 10
MaxRetries: 0
RetryTime: 5
WaitTime: 45
;magic extension for outbound calls via RAGI callInitiate
Context: dialout
Extension: outbound-handler
Priority: 1
SetVar: CallInitiate_phonenumber=90275524911
SetVar: CallInitiate_callerid=1000
SetVar: AGI_URL=/test_outbound/dialup
SetVar: AGI_SERVER=harborreach.panztel.biz:4573
SetVar: CallInitiate_hashdata=---+%0A
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
