[asterisk-users] voicemailmain
When I call to VoicemailMain it just sits. ; Retrieve Voice Mail exten = 2500,1,Wait(2) exten = 2500,2,VoicemailMain(s100) exten = 2500,3,Macro(endcall) 1.4.3 latest SVN. voicemail(100) works and the mwi systems works. I am not using ODBC or SQL. Voice mail to email works ok. I just cannot retrieve it by the application. I'm not sure when this quite we get little voice mail traffic. Thanks --john ___ --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] re:voicemailmain
I looked at the voicemail.c code and you must have the res.adsi module loaded. I was not loading it. Now it works. Something to remember. Thanks --john ___ --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] callback on busy
I looked at this. Iguess I willuse dialstatus busy and create a .call file And see what happens. Thanks --john From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giovanni MianoSent: Thursday, January 05, 2006 4:05 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] callback on busy www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+outCheers,Giovanni Miano 2006/1/5, Hill, John [EMAIL PROTECTED]: I was looking for a way to catch the zap busy return and do a redial.I would dial out on a zap channel. If the call is busy it would then hangupthe zap channel and ask if I wanted to redial "press 1 to redial or hangup to quit".On the 1 it would hangup the extension redial the number and call back theextension after x number of tries. If the number still was busy then itplays a message that it was busy would I like to continue or quit. Thanks--John Hill--This mail was scanned by AntiVir Milter.This product is licensed for non-commercial use.See www.antivir.de for details. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Giovanni Miano -- This mail was scanned by AntiVir Milter. This product is licensed for non-commercial use. See www.antivir.de for details. -- This mail was scanned by AntiVir Milter. This product is licensed for non-commercial use. See www.antivir.de for details.___ --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] smp
Tzafrir, Thanks for the reply. This is a 2.6.13 kernel. Runs very well. It really is not hurting anything memory usage is ok and it is responsive. Just my old school resource attitude. Shana Tova --john -- This mail was scanned by AntiVir Milter. This product is licensed for non-commercial use. See www.antivir.de for details. ___ --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] smp
I have a small test system -- 6 phones. It is a dual processor server. I noticed that asterisk spawns 12 child processes. Can this be controlled? I would think 2-4 would be plenty for this test site. Thanks --john -- This mail was scanned by AntiVir Milter. This product is licensed for non-commercial use. See www.antivir.de for details. ___ --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] ? In CLI not working
Has anyone noticed that a ? Entered at the root CLI does not work any longer? Petty I know but I did use it. --john -- This mail was scanned by AntiVir Milter. This product is licensed for non-commercial use. See www.antivir.de for details. ___ --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
RE: Re: [Asterisk-Users] custom ring tone
Look at the dial app. I think it has several options. Most custom 'TONES' are wav, acc, mp3 etc. files. If you can set a different MOH class or perhaps a playback file in the dial app that plays a file that is a 'RING TONE' that may work. -John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Domjan Attila Sent: Thursday, September 22, 2005 2:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: Re: [Asterisk-Users] custom ring tone Dial(SIP/1234|90|m) the caller will hear music on hold while SIP/1234 is ringing On Thu, 2005-09-22 at 20:18 +0200, Marko Rakar wrote: yes, but I want this feature to be turned on for people who are calling my asterisk from PSTN Two atoms bump into each other. One says I think I lost an electron! The other asks, Are you sure?, to which the first replies, I'm positive. mailto:[EMAIL PROTECTED] http://printel.hr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Domjan Attila Sent: Thursday, September 22, 2005 7:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: PS: Re: [Asterisk-Users] custom ring tone Hi, Dial application with m option, if the telco accept from you. On Thu, 2005-09-22 at 19:10 +0200, Marko Rakar wrote: Few weeks back local telco introduced option of custom ring tones. I am not talking about your phone ring tones but about ring tones you hear in your headset while phone is ringing on the other end. If I understand correctly, ringing tone is generated localy on asterisk if you are connected to phone network with E1/T1 connection. Which means that instead of regular beep-beep tone we could send something else to the caller in PSTN (like mp3 music). Is there a way of customizing ring tone in asterisk and if yes how? ___ --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 -- Domjan Attila [EMAIL PROTECTED] ___ --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
RE: Re: [Asterisk-Users] custom ring tone
I have a [pstn-inbound] that calls a dial app that plays music to the pstn caller. --john -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marko Rakar Sent: Thursday, September 22, 2005 2:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: Re: [Asterisk-Users] custom ring tone I am not interested in Dial app, I want the callers who are calling FROM pstn TO asterisk to hear different kind of ring tone (wav, mp3, gsm or whatever) For users within asterisk domain who actually use Dial command it does not matter and I know that I can have full control over them Two atoms bump into each other. One says I think I lost an electron! The other asks, Are you sure?, to which the first replies, I'm positive. mailto:[EMAIL PROTECTED] http://printel.hr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Hill Sent: Thursday, September 22, 2005 9:26 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: Re: [Asterisk-Users] custom ring tone Look at the dial app. I think it has several options. Most custom 'TONES' are wav, acc, mp3 etc. files. If you can set a different MOH class or perhaps a playback file in the dial app that plays a file that is a 'RING TONE' that may work. ___ --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
RE: Re: [Asterisk-Users] custom ring tone
Forgot. You must answer the line first. Other than that Asterisk is not involved with the external pstn until it is answered. --john -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marko Rakar Sent: Thursday, September 22, 2005 2:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: Re: [Asterisk-Users] custom ring tone I am not interested in Dial app, I want the callers who are calling FROM pstn TO asterisk to hear different kind of ring tone (wav, mp3, gsm or whatever) For users within asterisk domain who actually use Dial command it does not matter and I know that I can have full control over them Two atoms bump into each other. One says I think I lost an electron! The other asks, Are you sure?, to which the first replies, I'm positive. mailto:[EMAIL PROTECTED] http://printel.hr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Hill Sent: Thursday, September 22, 2005 9:26 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: Re: [Asterisk-Users] custom ring tone Look at the dial app. I think it has several options. Most custom 'TONES' are wav, acc, mp3 etc. files. If you can set a different MOH class or perhaps a playback file in the dial app that plays a file that is a 'RING TONE' that may work. ___ --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
RE: [Asterisk-Users] custom ring tone
I was thinking of PSTN over FXO cards. When I see PSTN I think pots. You mentioned BRI whould PRI do as well? --john -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fernando Herrera Sent: Thursday, September 22, 2005 3:07 PM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] custom ring tone John, Ringback is provided by your PSTN provider until answer by asterisk. You have no control until you answer Generally the ringback tone is sent by the last ClassV/Class IV switch in the telephony path. This is for Telco's to send inband error/progress/information announcements. However, some telcos just send back the relase indicating a certain Release Cause Value and letting you (in case you are another Telco) decide whether to play an announcement or not. Marko, I think that the DIAL command will match your needs. When you get an incoming call to your asterisk (through any channel, let's say, just as an example, the incoming call comes from an ITSP through a SIP channel) you configure the Asterisk to send the Music On Hold as a ring back tone (Dial(SIP/1234|90|m)). Though, when you got an incoming call, this will happen: 1. The ITSP sends an INVITE to your asterisk 2. Asterisk answers with a TRYING 3. Then. Asterisk will send a 183 (Session Progress) and you start transmiting RTP. Normally, you will send the RTP for ring back tone (tuuu tuuu). Here, you will send music on hold through the RTP channel. 4. At this very same moment, the asterisk's end user's phone starts ringing. You will be able to implement such thing with SIP or H.323 channels if you connect to PSTN through an ITSP. In case your asterisk is connected to PSTN through POTS, you will only be able to do it if you use ISDN. If you are using FXS/FXO, you won't be able to do it, since in this case the ringback tone is generated by the TELCO's Class V switch. Kind regards, Fernando Herrera -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de John Novack Enviado el: Jueves, 22 de Septiembre de 2005 16:46 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] custom ring tone Marko Rakar wrote: I am not interested in Dial app, I want the callers who are calling FROM pstn TO asterisk to hear different kind of ring tone (wav, mp3, gsm or whatever) ?? Ringback is provided by your PSTN provider until answer by asterisk. You have no control until you answer Then you go to IVR, VM or ?? John Novack For users within asterisk domain who actually use Dial command it does not matter and I know that I can have full control over them ___ --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
RE: [Asterisk-Users] Callerid fails in any release after beta1 fails
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Wednesday, September 14, 2005 4:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Callerid fails in any release after beta1 fails Richard Kashdan wrote: On Mon Sep 12 10:32:47 CDT 2005 John Hill wrote: I am having the identical problem. I use the CVSHEAD Asterisk and do an update every couple of weeks or so. I did one last week and the caller id quit working on my two lines that have x100p cards. I didn't make any changes to my configuration files at that time, simply updated Asterisk. In the meantime I checked my configuration files carefully and don't see anything wrong. Callerid has stoped working for us as well from the SIP phones to the PRI. PRI to the SIP phones work fine. Doug Today I did a make update for zaptel, libpri and asterisk. Then recompiled. I no longer get an error message. Callerid is still blank. The log and cli return this line: Sep 14 08:22:51 NOTICE[13266]: chan_zap.c:5946 ss_thread: Got event 18 (Ring Begin)... I was getting a checksum error and a mylen 0 error. It would say callerid failed: success. I deleted all modules and did a make install of the beta1 source using the cvshead of zaptel and libpri. Caller id then works fine? Something has changed in the asterisk code that is not seeing callerid from of my x101p. I'm stumpted! --John ___ --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] R1.502 of chan_zap.c kills callerid on a x101p
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Wednesday, September 14, 2005 10:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Callerid fails in any release after beta1 fails John Hill wrote: I deleted all modules and did a make install of the beta1 source using the cvshead of zaptel and libpri. Caller id then works fine? Something has changed in the asterisk code that is not seeing callerid from of my x101p. I was thinking about doing a fresh install this weekend as well to see if that makes any difference. Doug R1.502 of chan_zap.c kills callerid on a x101p You might want to wait. I'm trying to figure out how to report this as a bug. --john ___ --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] PLEASE HELP!! CALLERID FAILS!!
I have 1.2.0 beta 1 running and it works fine. My x100p returns caller id with no problems. When I test the CVSHEAD callerid fails with checksum and len 0 errors. I can run with the cvshead of zaptel and libpri with beta1 but only the beta1 source works for caller id. Any source after beta1 fails. Thanks --john ___ --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] Callerid fails in any release after beta1 fails
I have 1 x100p. Caller id works fine with the beta1 release. Cvshead releases fail with a combination of checksum and ss_thread errors? I'm concerned when beta2 or the 1.2 release comes out it will not work. I have been through the configs I can't find and changes that need to be made to get CVSHEAD to work. Thanks John Hill ___ --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] New CUT()
I store my speed dial numbers in the astdb key speeddial with the number and
then name separated by a -.
This dial plan works fine:
[speed-dial]
exten = _*0XX,1,Set(temp=${DB(speeddial/${EXTEN:2})})
exten = _*0XX,2,Cut(number=temp,,1)
exten = _*0XX,3,Goto(house-phones,${number},1)
The log informs me that cut is replaced with CUT.
I rewrote the dial plan using CUT (as best I can figure out) The plan below
returns the entire string number-name and fails?
[speed-dial]
exten = _*0XX,1,Set(temp=${DB(speeddial/${EXTEN:2})})
exten = _*0XX,2,CUT(temp,,1)
exten = _*0XX,3,Goto(house-phones,${temp},1)
What am I missing.
Thanks
--john
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[Asterisk-Users] RE:NewCUT()
In article 200509091317.j89DGtY3019393 at commserver.noach.com,
John Hill jhill at noach.com wrote:
I store my speed dial numbers in the astdb key speeddial with the number
and
then name separated by a -.
This dial plan works fine:
[speed-dial]
exten = _*0XX,1,Set(temp=${DB(speeddial/${EXTEN:2})})
exten = _*0XX,2,Cut(number=temp,,1)
exten = _*0XX,3,Goto(house-phones,${number},1)
The log informs me that cut is replaced with CUT.
I rewrote the dial plan using CUT (as best I can figure out) The plan
below
returns the entire string number-name and fails?
[speed-dial]
exten = _*0XX,1,Set(temp=${DB(speeddial/${EXTEN:2})})
exten = _*0XX,2,CUT(temp,,1)
exten = _*0XX,3,Goto(house-phones,${temp},1)
What am I missing.
The new CUT is a function, and should be used within a Set command.
Something approximating (please check the detail):
exten = _*0XX,1,Set(temp=${DB(speeddial/${EXTEN:2})})
exten = _*0XX,2,Set(number=CUT(temp,,1))
exten = _*0XX,3,Goto(house-phones,${number},1)
Your second example is calling the same Cut command as the first.
Cheers
Tony
exten = _*0XX,2,Set(number=${CUT(temp,,1)})
That fixed it. I only had to add the ${} to get the string returned.
Thanks
--john
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[Asterisk-Users] CVSHEAD callerid not working
The 1.2 beta1 works fine. When I install the current cvshead it gives me different errors: I have seen checksum errors, Got event ring 18, etc. all give empty callerid. I have an x100p. Thanks --john ___ --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] Sip reg problem
I get this error message in my syslog. I have searched the list but I can't seem to find a answer that solves the problem. chan_sip.c: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '192.168.100.2' The asterisk server is running dev head. The server is the 100.1 ip. I have 3 ata186's numbered 110.2-4. In sip.conf I have 6 entries all the same except for the ip and extension number. Username=2238 [-2243] Password= host=dynamic default ip=192.168.100.2 [-4] (2 phones at each ip) Etc. It works fine calls in and out. What am I missing? Thanks --john ___ 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] CDR question
I use the CDR CVS file for logging my home phone system. Can I force write data to a CDR Field from an extensions macro? Say if the callerid was empty and I dumped the call to put data in the CDR to let me know that is what happened. Thanks --John ___ 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] Gotoif question
Is there a way to combine these lines into one?
exten = s,2,GotoIf($[${CALLERIDNUM:0:3} = 800]?s|108)
exten = s,3,GotoIf($[${CALLERIDNUM:0:3} = 866]?s|108)
exten = s,4,GotoIf($[${CALLERIDNUM:0:3} = 877]?s|108)
exten = s,5,GotoIf($[${CALLERIDNUM:0:3} = 888]?s|108)
Thanks
--John
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RE: [Asterisk-Users] Gotoif question
That did it.
Thanks
--John
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Diego Aguirre
Sent: Thursday, January 06, 2005 8:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Gotoif question
Try this:
exten = s,2,GotoIf($[${CALLERIDNUM:0:3} == 800] ||
$[${CALLERIDNUM:0:3} = 866] || $[${CALLERIDNUM:0:3} = 877] ||
$[${CALLERIDNUM:0:3} = 888]?s|108)
Diego Aguirre
FWD# 459696
- Original Message -
From: John Hill [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Thursday, January 06, 2005 11:30 AM
Subject: [Asterisk-Users] Gotoif question
Is there a way to combine these lines into one?
exten = s,2,GotoIf($[${CALLERIDNUM:0:3} = 800]?s|108)
exten = s,3,GotoIf($[${CALLERIDNUM:0:3} = 866]?s|108)
exten = s,4,GotoIf($[${CALLERIDNUM:0:3} = 877]?s|108)
exten = s,5,GotoIf($[${CALLERIDNUM:0:3} = 888]?s|108)
Thanks
--John
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RE: [Asterisk-Users] Why use 'Answer'?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Wednesday, December 22, 2004 8:12 AM
To: Asterisk-a-users-list
Subject: [Asterisk-Users] Why use 'Answer'?
Why is it that newcomers always feel like inserting 'Answer' is a
necessary step in their extension.conf entries?
[voiptalk.org]
;forwards any calls starting with an 8 thru voiptalk.org
exten = _8.,1,Answer
exten = _8.,3,SetCIDNum()
exten = _8.,4,SetCIDName(My Name And Surname)
exten = _8.,5,Dial(SIP/${EXTEN:[EMAIL PROTECTED],,g)
exten = _8.,6,HangUp
I fully understand that incoming pstn calls have to be answered (in
most cases) before executing a Playback(invalid) type statement. But
there must be some examples, documentation, or somthing that is
suggesting to newcomers that all sequences have to start with an Answer.
Question:
Do you need to answer to detect a fax?
Excerpt from my conf file.
I would like to tune this. I have tried putting the 800 service checks on a
single line but it fails. Any advice would be useful.
[inbound-pstn]
exten = s,1,NoOp(${CALLERID}) ; log callerID string
exten = s,2,GotoIf($[${CALLERIDNUM:0:3} = 800]?s|108)
exten = s,3,GotoIf($[${CALLERIDNUM:0:3} = 866]?s|108)
exten = s,4,GotoIf($[${CALLERIDNUM:0:3} = 877]?s|108)
exten = s,5,GotoIf($[${CALLERIDNUM:0:3} = 888]?s|108)
exten = s,6,GotoIf($[${CALLERIDNUM} = ]?s|109)
exten = s,7,LookupBlacklist
exten = s,8,Answer
exten = s,9,Ringing
exten = fax,1,Macro(faxreceive)
exten = s,10,Macro(ringphones)
exten = s,11,Wait(2)
exten = s,12,PlayBack(im-sorry)
exten = s,13,Voicemail(u100)
exten = s,14,Hangup
exten = s,108,Macro(noservice)
exten = s,109,Macro(nocallerid)
exten = h,1,Hangup
--John
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RE: [Asterisk-Users] Why use 'Answer'?
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Seth Remington Sent: Wednesday, December 22, 2004 10:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Why use 'Answer'? On Wed, 2004-12-22 at 08:41, John Hill wrote: Question: Do you need to answer to detect a fax? Yes. You need to answer the line so the calling fax will start sending the fax tones and * can detect them. -Seth -- That's what I thought. My dial plan works but it looks a bit messy. --John ___ 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] upgraded source now ata's ring but stop silence on inbound calls
I was doing a daily make update for asterisk. On the 19th the new version compiled fine. I installed it. All of my ata 186's can call out to pstn etc. All inbound calls, the phones ring but when you pickup, just silence both local and remote with no complaints in the cli. I backed down to the r 1.0 1.0.3 on the 20th CVS-v1-0-12/20/04-18:24:52 it worked ok. Yesterday I did a cvs update on the 1.0 source code, recompiled and now it does the same thing as the dev source. Calls go out no problem but inbound rings but are dead upon answer. I had to return to the CVS-v1-0-12/20/04-18:24:52 code to get it to work again. Have I missed a configuration change somewhere? Thanks John Hill ___ 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] upgraded source now ata's ring but stop silence oninbound calls
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of John Hill Sent: Tuesday, December 21, 2004 8:40 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] upgraded source now ata's ring but stop silence oninbound calls I was doing a daily make update for asterisk. On the 19th the new version compiled fine. I installed it. All of my ata 186's can call out to pstn etc. All inbound calls, the phones ring but when you pickup, just silence both local and remote with no complaints in the cli. I backed down to the r 1.0 1.0.3 on the 20th CVS-v1-0-12/20/04-18:24:52 it worked ok. Yesterday I did a cvs update on the 1.0 source code, recompiled and now it does the same thing as the dev source. Calls go out no problem but inbound rings but are dead upon answer. I had to return to the CVS-v1-0-12/20/04-18:24:52 code to get it to work again. Have I missed a configuration change somewhere? Thanks John Hill I had canreinvite=yes in sip.conf I changed it to no. Still the same new stable source has no audio on inbound calls. Stable source from the 20th works? --John Hill ___ 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] upgraded source now ata's ring but stop silenceon inbound calls
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Hess Sent: Tuesday, December 21, 2004 10:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] upgraded source now ata's ring but stop silenceon inbound calls I reported this on dev yesterday.. I thought I saw it fixed in dev but not stable according to the cvs list.. Modified Files: chan_sip.c Log Message: Minor ACk fix (bug #2687, again) So the stable version is still borked.. but head should be cleared up..heh, stable ain't that stable right now ;) John Hill wrote: I was doing a daily make update for asterisk. On the 19th the new version compiled fine. I installed it. All of my ata 186's can call out to pstn etc. All inbound calls, the phones ring but when you pickup, just silence both local and remote with no complaints in the cli. I backed down to the r 1.0 1.0.3 on the 20th CVS-v1-0-12/20/04-18:24:52 it worked ok. Yesterday I did a cvs update on the 1.0 source code, recompiled and now it does the same thing as the dev source. Calls go out no problem but inbound rings but are dead upon answer. I had to return to the CVS-v1-0-12/20/04-18:24:52 code to get it to work again. Have I missed a configuration change somewhere? Thanks John Hill I downloaded the tar release 1.0.3 and it works. The stable cvs yesterday failed. I have not tried the cvs dev or stable. I'll just sit on the release. --john ___ 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] Yet another faxing issue..
; Zap Fax ; exten = 8021,1,Dial(SIP/8021,20) exten = 8021,2,Hangup [incoming] exten = s,1,Answer exten = s,2,Wait,1 exten = s,3,DigitTimeout(10) exten = s,4,ResponseTimeout(20) exten = s,5,Background(vm-extension) exten = fax,1,Goto(8021,30) I thinkg this should be goto(8021,1) Goto(extension,priority) --John ___ 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
RE: [Asterisk-Users] Problem with fax tone (CNG) from TxFax and busydetect
I'm losing call files in /var/spool/asterisk/outgoing because * isn't able to detect the busy signal. The call file looks like this: Channel: Zap/g2/3036701917 MaxRetries: 1000 RetryTime: 60 WaitTime: 45 Application: TxFAX Data: filename.tiff|caller Using the |caller parameter, TxFax injects the fax tone (CNG) onto the line. With the CNG tone, asterisk is unable to detect the busy tones. If I were to remove |caller then the receiving station wouldn't receive the CNG tone and possibly not direct the call to the fax machine. Is there a way for * to detect busy tones while ignoring (filtering) the fax tones? I have a similar problem: I dial out but no CNG signal is sent at all. It is as if the auto-dial program never sees the line has been answered. It never calls the application. I set it up to dial my cell phone. I answer and nothing, dead. I hang up and the script hangs up. My call file is set up as yours. John ___ 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
[Asterisk-Users] txfax
Trying to send a fax using a call file and txfax. Phone dials the remote fax answers but * gives me: Call failed to go through, reason 3 And hangs up. Any help. Thanks --John ___ 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
RE: [Asterisk-Users] Pause during dial
I may be wrong but after looking around all I could find was an email about
w and p. It said w is to wait for a tone and p was for a pause. I can't find
anything to verify this.
--john
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Henry Devito
Sent: Wednesday, November 10, 2004 2:15 PM
To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE: [Asterisk-Users] Pause during dial
Is there a way to put pauses in a dial string? I need * to dial a
number then pause for 6 seconds and dial a second string of numbers.
search the list.
This question has been answered tons of time before.
Matteo.
Good day,
I did search the list before I posted and found several answers all saying
to use the w prompt. I have not been able to get this to work, so I was
hoping to get an example from someone on the list. I need to dial one
variable pause and the dial the second variable. I tried to use the D
prompt also, but I must be making a formatting error.
exten = 3,1,DigitTimeout(4) ; Set Digit Timeout 4 seconds
exten = 3,2,ResponseTimeout(5) ; Set Response Timeout 5 sec
exten = 3,3,Read(destination,enter_destination,4)
exten = 3,4,NoOp(${destination})
exten = 3,5,Wait(2)
exten = 3,6,Read(pin,Enter_pin,6)
exten = 3,7,NoOp(${pin})
exten = 3,8,Dial(sip/${destination}D{$pin})
exten = 3,9,Hangup
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RE: [Asterisk-Users] Reject a call if no callerID
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Walt Reed
Sent: Wednesday, November 03, 2004 8:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Reject a call if no callerID
On Wed, Nov 03, 2004 at 04:45:03PM +0900, Hermann Wecke said:
I couldn't think any recipe to reject a call if no callerID is presented.
PrivacyManager and Zapateller are not an option, as the call will be
answered before I can drop it. I just want to silent drop the call: no
callerID, no answer.
See example 3 in:
http://www.voip-info.org/wiki-Asterisk+cmd+GotoIf
Here is what I do:
exten = s,1,GotoIf($[${CALLERIDNUM} = ]?s|5)
exten = s,2,yadayada
exten = s,3,yadayada
exten = s,4,yadayada
exten = s,5,Hangup
;
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RE: [Asterisk-Users] Wireless VOIP Phone suggestions
Use an ATA the plug in any cordless phone. Works fine. --john -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Christopher TenHarmsel Sent: Tuesday, November 02, 2004 10:50 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Wireless VOIP Phone suggestions Hi all, We're using Asterisk in our office to run our phone system (right now about 5 SIP phones, various Cisco 7912's and 7960's), but we are in desperate need for cordless phones. We don't need 802.11b/g phones, but just something that is wireless and does SIP. I've done some searching around, and we've even tried out the one from Pulver Innovations (with no luck), so I wondered if someone could make some suggestions? Thanks, Chris -- Chris TenHarmsel Software Journeyman Atomic Object, LLC ___ 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 ___ 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
RE: [Asterisk-Users] Wireless VOIP Phone suggestions
I have 3 ata186's from Cisco. I have never used any other ATA but my guess is they are all ok. The 186's have 2 analog ports. I use one box for two different cordless phones. The only thing I could not get to work was the *8. It must trap this string. I had to change *8 to an extension number (features.conf) to get group pickup to work. We use them every day. --john -Original Message- From: Paradise Dove [mailto:[EMAIL PROTECTED] Sent: Tuesday, November 02, 2004 11:45 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Wireless VOIP Phone suggestions Use an ATA the plug in any cordless phone. Works fine. which brand/model cordless phone you suggest? i'm looking for one with long distance coverege. - shabanip ___ 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
[Asterisk-Users] gotoif regex?
I have a test for all tool free numbers.
This works but would using regex in one statement be more efficient?
exten = s,4,GotoIf($[${CALLERIDNUM:0:3} = 888]?s|108)
exten = s,5,GotoIf($[${CALLERIDNUM:0:3} = 877]?s|108)
exten = s,6,GotoIf($[${CALLERIDNUM:0:3} = 866]?s|108)
exten = s,7,GotoIf($[${CALLERIDNUM:0:3} = 800]?s|108)
Would this be the proper syntax?
exten = s,4,GotoIf($[${CALLERIDNUM:0:3} : 888|877|866|800]?s|108)
I tried this but I'm not sure it works.
Thanks
--John
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RE: [Asterisk-Users] Cisco 7905/7912 SIP image location (onCisco'ssite)
Use this url: If you have a valid userid. http://www.cisco.com/kobayashi/sw-center/sw-voice.shtml -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Vallee Sent: Tuesday, September 21, 2004 4:27 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Cisco 7905/7912 SIP image location (onCisco'ssite) Hi, I don't know if it the only way to do that, but it is how I did it. You need to have a valid cco account and your Smartnet contract have to be associated with your cco login. Next go to http://www.cisco.com/public/sw-center/ and when you are logged in, you can find the firmware of your choice under Voice Software section. If you have a smartnet account, don't hesitate to contact cisco technical support. Regards Christian Vallee -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Jeb Campbell Envoyé : 21 septembre 2004 14:17 À : [EMAIL PROTECTED] Objet : [Asterisk-Users] Cisco 7905/7912 SIP image location (on Cisco'ssite) Hello all, I feel dumb asking this, but does anyone have a link to the SIP firmware for the 7912 on Cisco's site? I have a SmartNet contract, but I just can't find the link (you can search for 7960 sip firmware and find that fast). Thanks for the help, Jeb Campbell [EMAIL PROTECTED] ___ 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 ___ 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 ___ 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
RE: [Asterisk-Users] Cisco 7905/7912 SIP image location (onCisco'ssite)
Her is the 7905-12 page http://www.cisco.com/cgi-bin/tablebuild.pl/ip-phone-7905 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Vallee Sent: Tuesday, September 21, 2004 4:27 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Cisco 7905/7912 SIP image location (onCisco'ssite) Hi, I don't know if it the only way to do that, but it is how I did it. You need to have a valid cco account and your Smartnet contract have to be associated with your cco login. Next go to http://www.cisco.com/public/sw-center/ and when you are logged in, you can find the firmware of your choice under Voice Software section. If you have a smartnet account, don't hesitate to contact cisco technical support. Regards Christian Vallee -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Jeb Campbell Envoyé : 21 septembre 2004 14:17 À : [EMAIL PROTECTED] Objet : [Asterisk-Users] Cisco 7905/7912 SIP image location (on Cisco'ssite) Hello all, I feel dumb asking this, but does anyone have a link to the SIP firmware for the 7912 on Cisco's site? I have a SmartNet contract, but I just can't find the link (you can search for 7960 sip firmware and find that fast). Thanks for the help, Jeb Campbell [EMAIL PROTECTED] ___ 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 ___ 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 ___ 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
RE: [Asterisk-Users] IAX- FAX
I sent a test fax last night from home to my office. I sent it through an ata186 out Asterisk to NUFONE on an iax2 connection. It reported failed. I resent it with success. When I arrived at my office the fax had been received both times. I was not expecting this so I don't have any debug or log information. I also sent one out to the (x100p) pots, it worked the first time. How can I monitor this process? Thanks --John Hill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, September 16, 2004 2:49 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] IAX- FAX Nothing special about my config, I am not doing any fax detection just have a DID Set up with a triple ring that my fax unit is set to pick up on. Get this some of my outbound faxes seem to go thru even though it is reporting an error. I think its messing up on disconnect. Paul Seniuk -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: September 16, 2004 12:56 PM To: asterisk-users Subject: RE: [Asterisk-Users] IAX- FAX We suffer the same from with outbound using a mediatrix sip/fx box The connected fax machine dials and during handshake drops the call. The Iax link is set to use ULAW Im trying to get asterisk to handle inbound natively, i.e asterisk answer listens and dumps into a file on the linux box, I read voip-info but can get it to work. Have you got a config I can read over? Thanks d -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 16 September 2004 18:13 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] IAX- FAX D, I have a IAX2 gateway that connects to our remote asterisk gateway that has a PRI. Inbound seems to work without a hitch. Make sure your iax.conf allows ULAW as well, Since fax cannot be compressed. Outbound is a different story. My fax seems to ring thru, but it never seems to establish A carrier. Have you been able to get outbound working? Paul Seniuk -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: September 16, 2004 10:40 AM To: asterisk-users Subject: [Asterisk-Users] IAX- FAX Has anyone had any success using iax for inbound fax into asterisk. I tried this but can seem to get asterisk to listen for fax, is it zap specific ? d ___ 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 ___ 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 ___ 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 ___ 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 ___ 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
Re: [Asterisk-Users] Problems installing x100p
Make sure you have the bios updated to recognise the card. If the hardware does not see it the OS wont either. I just built Asterisk on an old single 450 PII Compaq Prolient 800 with an x100p and 3 ata186's. It works just fine. Hope this helps. --john - Original Message - From: Rodolfo Grave [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, September 12, 2004 6:50 AM Subject: [Asterisk-Users] Problems installing x100p Hi. I have succesfuly installed asterisk and after I added a x100p card, but the system doesn't seem to know the card is there. This is what I've done: compiled and installed zaptel, libpri and asterisk in that order using make clean ; make install commands. also, make samples for asterisk. shutdown the PC and installed the x100p PCI card. execute the following commands (I also include the output): linux# ] modprobe zaptel linux# ] modprobe wcfxs /lib/modules/2.4.21-99-default/misc/wcfxs.o: init_module: No such device /lib/modules/2.4.21-99-default/misc/wcfxs.o: insmod /lib/modules/2.4.21-99-default/misc/wcfxs.o failed /lib/modules/2.4.21-99-default/misc/wcfxs.o: insmod wcfxs failed Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg linux# ] lspci 00:00.0 Host bridge: VIA Technologies, Inc. VT82C693A/694x [Apollo PRO133x] (rev 22) 00:01.0 PCI bridge: VIA Technologies, Inc. VT82C598/694x [Apollo MVP3/Pro133x AGP] 00:07.0 ISA bridge: VIA Technologies, Inc. VT82C596 ISA [Mobile South] (rev 09) 00:07.1 IDE interface: VIA Technologies, Inc. VT82C586A/B/VT82C686/A/B/VT8233/A/C/VT8235 PIPC Bus Master IDE (rev 06) 00:07.2 USB Controller: VIA Technologies, Inc. USB (rev 02) 00:07.3 Host bridge: VIA Technologies, Inc. VT82C596 Power Management 00:0a.0 Ethernet controller: MYSON Technology Inc SURECOM EP-320X-S 100/10M Ethernet PCI Adapter 00:0c.0 Multimedia audio controller: C-Media Electronics Inc CM8738 (rev 10) 00:0c.1 Communication controller: C-Media Electronics Inc CM8738 (rev 10) 01:00.0 VGA compatible controller: Silicon Integrated Systems [SiS] 86C326 5598/6326 (rev 0b) It seems that the PCI card is not detected... any ideas? Thanks in advance. RODOLFO --- avast! Antivirus: Outbound message clean. Virus Database (VPS): 0437-1, 09/09/2004 Tested on: 12/09/2004 13:50:10 avast! - copyright (c) 2000-2004 ALWIL Software. http://www.avast.com ___ 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 ___ 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
[Asterisk-Users] call park question
I can part a call (dial #700 it is parked on 701) but ifI dial 701 I am told it is not a valid extension? I have include = parkedcalls in my local extension context. I have Ttr on all extensions and the incoming pots line. It parks, plays MOH but I can't retrieve it. --john ___ 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
