[Asterisk-Users] Annoying Notice Message: Don't know what to do with control frame 15
Hi, I haven't found anything about the message below on the mailing list, Does anyones knows why this notice is being appearing? -- Executing Dial(Local/[EMAIL PROTECTED],2, IAX2/CallOut/12365533643|30|otT) in new stack -- Called CallOut/12365533643 -- Call accepted by 12.11.11.11 (format ulaw) -- Format for call is ulaw -- IAX2/10.11.240.110:4569-3 is proceeding passing it to Local/[EMAIL PROTECTED],2Jan 6 13:20:41 NOTICE[26911]: channel.c:2416 __ast_request_and_dial: Don't know what to do with control frame 15 -- IAX2/10.11.240.110:4569-3 is circuit-busy -- Hungup 'IAX2/12.11.11.11:4569-3' == Everyone is busy/congested at this time (1:0/1/0) -- Executing Goto(Local/[EMAIL PROTECTED],2, s-CONGESTION|1) in new stack -- Goto (default,s-CONGESTION,1) -- Executing NoOp(Local/[EMAIL PROTECTED],2, CONG) in new stack -- Executing Congestion( Local/[EMAIL PROTECTED],2, ) in new stack Channel Local/[EMAIL PROTECTED],1 was never answered. == Spawn extension (default, s-CONGESTION, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' My calling scenario is like this:server01 server02 pstn server --IAX trunking-- agents/sip server server01: Asterisk 1.2.1server02: Asterisk 1.2.1 ___ --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] ${DIALSTATUS} problems
I'm not expert on this matter,but base on experience that is a normal situation on SIP/IAX channels since the call made is answered by the other end regardless of the situation you might found.
I'm on PRI ISDN and for me dialstatus and hangupcause works pretty good.
Regards
Jb
On 9/15/05, Mark Edwards [EMAIL PROTECTED] wrote:
Hi.I'm dialling two numbers - one that's unobtainable, one that's busy.${DIALSTATUS} is coming back ANSWER each time right before the channels hang up.
Am using the following dialplan macro to dial out.[macro-advdial]exten = s,1,Dial(${ARG1},20,g) ; Ring the interface, 20 seconds maximumexten = s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL
,CONGESTION,ANSWER)exten = s-CHANUNAVAIL,1,NoOp(CHANUNAVAIL)exten = s-CHANUNAVAIL,2,UserEvent(ChannelUnavailable|Account: ${ACCOUNTCODE}^${CALLERIDNUM})exten = s-CONGESTION,1,NoOp(CONGESTION)
exten = s-CONGESTION,2,UserEvent(Congestion|Account: ${ACCOUNTCODE}^${CALLERIDNUM})exten = s-ANSWER,1,NoOp(ANSWER)exten = s-ANSWER,2,UserEvent(Answer|Account: ${ACCOUNTCODE}^${CALLERIDNUM})
exten = s-BUSY,1,NoOp(BUSY)exten = s-BUSY,2,UserEvent(Busy|Account: ${ACCOUNTCODE}^${CALLERIDNUM})exten = s-NOANSWER,1,NoOp(NOANSWER)exten = s-NOANSWER,2,UserEvent(NoAnswer|Account: ${ACCOUNTCODE}^${CALLERIDNUM})
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answerOutbound calls are made using Manager originate interface from a meetme room channel Local/4000/n where 4000 is an extension which accesses the meetme room.
ITSP is terminating outbound calls to me via IAX2.I need to be able to see the CAUSE CODE status of the call if it is answered, CONGESTED or BUSY.my ITSP is in Australia - as am I.the IAX2 debug clearly indicates a zero CAUSE CODE on most call cases.
Any idea what I might be able to do to make the CAUSE CODE a little more meaningful?Also, does ${DIALSTATUS} or ${HANGUPCAUSE} work better on PRI?Cheers,Mark.
-- regards,Mark P. EdwardsFWD: 667917___--Bandwidth and Colocation sponsored by
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Re: [Asterisk-Users] Multiple PCI cards
Did you make any special configuration with the switch on the card? I have 2 TE400P that I haven't being able to use on 1 server. jb On 8/28/05, Asterisk [EMAIL PROTECTED] wrote: I have 2 TE410P's and a TDM400P in same machine without issuesBart-Original Message- From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] ]On Behalf Of Damon EstepSent: Sunday, August 28, 2005 10:59 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Multiple PCI cards -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED]] On Behalf Of Garth van Sittert Sent: Sunday, August 28, 2005 11:46 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Multiple PCI cards Hi All Does anyone know if multiple Digium cards on a single machine will bea problem. Machine specs:Dual Zeon 3.0GHz on Intel server board. Cards: TE411P, TDM400P, TDM400P I will turn off all unnecessary PCI devices; USB, parallel, serial,etc... ThanksHave not tried it since November 2004, but at that time I ended upreplacing the FXO/FXS cards with sipura SPA3000 ?(check model number, its been awhile). Each one gave 1FXO/1FXS port so 4 of them replaced 2 4port TDM cads. Works well.Again, this was almost a year ago, so look for more feedback for usersthat have tried it with current hardware/firmware/software. ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://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] Extensions Puzzle: Contexts Confligting with each other.
I have a setup for 2 companies. Both ones should have separate
schedules and differents menu options. here is an example:
[default]
include= company1
include= company2
[company1]
include = optionscompany1
[company2]
include = optionscompany2
[optionscompany1]
; option 0 Operator
exten = 0,1,Playback,CallTransfer_SP
exten = 0,3,SetCIDName,Operator
exten = 0,4,Goto,CMP1|1100|1
; option 1 Spanish
;exten = 1,1,SetLanguage(sp)
;exten = 1,2,Goto(day1|${CENTRAL1}|7))
; option 2 Addresss
exten = 2,1,Directory,default
exten = 2,2,Goto(day1|${CENTRAL1}|7))
; option 3
exten = 3,1,Playback,CallTransfer_SP
exten = 3,2,Goto(day1|${CENTRAL1}|7))
[optionscompany2]
; option 0 Operator
exten = 0,1,Playback,CallTransfer_SP
exten = 0,3,SetCIDName,Operator
exten = 0,4,Goto,CMP2|1100|1
; option 1 Spanish
;exten = 1,1,SetLanguage(sp)
;exten = 1,2,Goto(day2|${CENTRAL2}|7))
; option 2 Addresss
exten = 2,1,Directory,default
exten = 2,2,Goto(day2|${CENTRAL2}|7))
; option 3
exten = 3,1,Playback,CallTransfer_SP
exten = 3,2,Goto(day2|${CENTRAL2}|7))
The problem I have is that if a valid comes in for Company2 and the
caller select any available options it goes to the context for options
on Company1. Is there any way to correct or prevent this from
happening?
Thanks
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Re: [Asterisk-Users] SpanDSP+astfax with multiple fax pages
Hi, Daniel Would you explain how you are converting your images to tiff in order to send your faxes. Gracias jb On 6/21/05, Daniel Cubero Salas, Ing. [EMAIL PROTECTED] wrote: Hi, all I'd installing asterisk 1.0.7 with spanDSP 0.0.2pre18+astfax 1.0 on Fedora core 2. Fax reception (using RxFax) is working well. I have problems when sending a fax (it's an image in TIFF G3 format, using TxFax) composed of 2 parts/pages to a fax machine on PSTN, only receive first page but second page is a blank piece of paper. Fax machine reports LINE ERROR and hangs up. I try to send an email attached 2 images but the same result. If anyone had figured it out or resolved it, please let me know. Regards Daniel ___ 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] Fwd: Transfer of Calls Between Legacy PBX and Asterisk
Hi, We have a scenario where we receive calls from 2 different places: 1- Avaya IP Office 2- CIC Interactive Intelligence PBX and the calls are transfer automatically to an Asterisk Box. The problem we are experiencing is that more that half of those calls come with Echo and Jitter. For outbounds calls everything works perfect. I'm on RH9 Stable v1.4 Version, using an quadspad TMD400P digium card and have attached 2 pri lines. Calls transfers to my Asterisk Box are received directly to my T1 lines. My server is configure accordingly Digiums standards, however I found this issue annoying and don't have a clue was going on. We someone has experienced the same problem I maybe would guide me in solving this problem I would be more than glad. ___ 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] Transfer of Calls Between Legacy PBX and Asterisk
Hi, We have a scenario where we receive calls from 2 different places: 1- Avaya IP Office 2- CIC Interactive Intelligence PBX and the calls are transfer automatically to an Asterisk Box. The problem we are experiencing is that more that half of those calls come with Echo and Jitter. For outbounds calls everything works perfect. I'm on RH9 Stable v1.4 Version, using an quadspad TMD400P digium card and have attached 2 pri lines. Calls transfers to my Asterisk Box are received directly to my T1 lines. My server is configure accordingly Digiums standards, however I found this issue annoying and don't have a clue was going on. We someone has experienced the same problem I maybe would guide me in solving this problem I would be more than glad. ___ 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] Transfer of Calls Between Legacy PBX and Asterisk
Hi, We have a scenario where we receive calls from 2 different places: 1- Avaya IP Office 2- CIC Interactive Intelligence PBX and the calls are transfer automatically to an Asterisk Box. The problem we are experiencing is that more that half of those calls come with Echo and Jitter. For outbounds calls everything works perfect. I'm on RH9 Stable v1.4 Version, using an quadspad TMD400P digium card and have attached 2 pri lines. Calls transfers to my Asterisk Box are received directly to my T1 lines. My server is configure accordingly Digiums standards, however I found this issue annoying and don't have a clue was going on. We someone has experienced the same problem I maybe would guide me in solving this problem I would be more than glad. ___ 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] slight echo on incoming call
I'm on RH9 Stable v1.4 Version, using an old TMD400P digium card. They are some situations when I received echo on my cisco phones: 1- Some users or PC is dowloading some files from the network. 2- A call comes in from a cell phone. 3- A international call is received on our Asterisk Box. 4. We have a separate legacy PBX that on some circumstances need to tranfers call to asterisk. Most of those calls are received with echo. Besides this, is about tuning the rxgain,txgain usually on a range of 1 - 3. and to be sure your Digium uses it's own IRQ. using LSPCI If someone could explain to me a way to improve or solve the above items, would really appreciate. Thanks Jb On 5/12/05, Gary Carr [EMAIL PROTECTED] wrote: Running CVS-HEAD-04/12/05-16:39:24 on CentOS 4.0 final installation. I am hearing a brief echo on our Cisco 7960 phones when a incoming call is answered. After a few seconds of conversation the echo disappears. There is no echo on outbound calls or transferred calls. After a search of the mailing list, wikki, and google I have tried the following to no avail. echocanel=yes as well as 16, 32, 128, and 256 echotraining=yes as well as 800 echocancelwhenbridged=yes I have also modified the rxgain= and txgain= settings. Short of recompiling the zaptel with agreessive echotraining can anyone suggest some other things I may be missing? Thanks, Gary ___ 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
