RE: [Asterisk-Users] automon - one touch record
Hello Doug,
As previously discussed. Using the wW option in Queue() is currently not
functional, even though it is documented to work.
Kind regards
Jennifer
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Tuesday, January 17, 2006 5:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] automon - one touch record
Jennifer, yes I know but call recording is done differently with ACD
queues. Do a 'show application queue' on the console and you will see what I
mean.
-Original Message-
From: Jennifer Hales [mailto:[EMAIL PROTECTED]
Sent: Mon 1/16/2006 10:53 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc:
Subject: RE: [Asterisk-Users] automon - one touch record
Hi Doug,
On touch record and Monitor are two different ways of recording. This is
how I have input the 'w' (for one touch) into my dial plan and it works. So
if a staff member receiving a call wants to record a conversation they input
*1 (as per features.conf)and it starts recording. When they are finished
they press *1 again and the recording stops.
exten =_3xxx,2,Dial(SIP/${EXTEN},30,wth).
Hope this is of some help.
Regards
Jenn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Tuesday, January 17, 2006 11:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] automon - one touch record
Actually the docs for the Queue application say:
'w' -- allow the called user to write the conversation to disk via
Monitor
'W' -- allow the calling user to write the conversation to disk via
Monitor
couldn't get these to work tho. Does this mean I can do one touch recording
with agents, or does it mean I can use the monitor() command? Very
confusing...
Doug.
-Original Message-
From: Jennifer Hales [mailto:[EMAIL PROTECTED]
Sent: Sunday, January 15, 2006 4:01 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] automon - one touch record
Hello Kevin,
Thank you for your response. I commented out DYNAMIC_FEATURES and moved the
'Ww' option to the Dial() instead of Queue() and now it works.
Regards
Jenn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Friday, January 13, 2006 3:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] automon - one touch record
Jennifer Hales wrote:
I am unable to get automon recording to work; can someone advise me what I
am doing wrong? When I do *1 all I see in the CLI screen is attempting
native bridge of SIP/3006-291b and SIP/3153-6fdd, and there is no call
record generated in /var/spool/asterisk/monitor/.
I can't explain why it's not working, but DYNAMIC_FEATURES is not
necessary if you are providing the 'wW' options to the Queue application
as you are.
Can you try this with a regular Dial() call instead, to eliminate the
queue application?
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[Asterisk-Users] Dell PowerEdge 830 server
Hello all, We are looking at using a Dell PowerEdge 830 Server for an Asterisk installation. Does anyone have experience using this server with Asterisk? Any feed back would be appreciated. Kind regards Jenn ___ --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] Dell PowerEdge 830 server
Hello Kerry, Many thanks for your information. Do you mind giving some more details on your setup? What version of Asterisk are you using? How many users do you have? Are you using real-time? And what Asterisk features are you providing? Feel free to reply off list if you wish. Kind regards Jenn Hales -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Wednesday, January 18, 2006 1:29 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Dell PowerEdge 830 server Go into the BIOS, disable all unneeded peripherals like floppy controller, serial ports, parallel ports, etc. It should work fine, I have one at a decent sized installation. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jennifer Hales Sent: Tuesday, January 17, 2006 5:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Dell PowerEdge 830 server Hello all, We are looking at using a Dell PowerEdge 830 Server for an Asterisk installation. Does anyone have experience using this server with Asterisk? Any feed back would be appreciated. Kind regards Jenn ___ --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 ___ --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 ___ --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] Dell PowerEdge 830 server
Hello Kerry, Thank you so much. I am extremely gratefully for all your assistance. Kind regards Jenn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Wednesday, January 18, 2006 3:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Dell PowerEdge 830 server To be specific, I installed [EMAIL PROTECTED] 2.2 which is CentOS 4.2, Asterisk 1.2.1, Asterisk Management Portal, Flash Operator Panel, etc etc. That site has about 15 users with half of them having both on-site and off-site extensions (setup using AMP's Users and Devices mode). This site is not using any real-time functions. They do use the meet-me rooms fairly heavily. The system has a TDM400 with 4 FXO ports on it and the phone lines are in a hunt group that does a rollover on the 5th call to Teliax on the pay as you go plan which provides 10 additional channels. Does that help? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jennifer Hales Sent: Tuesday, January 17, 2006 7:55 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Dell PowerEdge 830 server Hello Kerry, Many thanks for your information. Do you mind giving some more details on your setup? What version of Asterisk are you using? How many users do you have? Are you using real-time? And what Asterisk features are you providing? Feel free to reply off list if you wish. Kind regards Jenn Hales -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Wednesday, January 18, 2006 1:29 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Dell PowerEdge 830 server Go into the BIOS, disable all unneeded peripherals like floppy controller, serial ports, parallel ports, etc. It should work fine, I have one at a decent sized installation. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jennifer Hales Sent: Tuesday, January 17, 2006 5:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Dell PowerEdge 830 server Hello all, We are looking at using a Dell PowerEdge 830 Server for an Asterisk installation. Does anyone have experience using this server with Asterisk? Any feed back would be appreciated. Kind regards Jenn ___ --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 ___ --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 ___ --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 ___ --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 ___ --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] automon - one touch record
Hello Kevin, Thank you for your response. I commented out DYNAMIC_FEATURES and moved the 'Ww' option to the Dial() instead of Queue() and now it works. Regards Jenn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Friday, January 13, 2006 3:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] automon - one touch record Jennifer Hales wrote: I am unable to get automon recording to work; can someone advise me what I am doing wrong? When I do *1 all I see in the CLI screen is attempting native bridge of SIP/3006-291b and SIP/3153-6fdd, and there is no call record generated in /var/spool/asterisk/monitor/. I can't explain why it's not working, but DYNAMIC_FEATURES is not necessary if you are providing the 'wW' options to the Queue application as you are. Can you try this with a regular Dial() call instead, to eliminate the queue application? ___ --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 ___ --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] automon - one touch record
Hello Francesco, Many thanks for your response. It originally appeared that not all the features.conf options worked, however further investigation determined that the problem was with the Grandstream phone I was testing with. My other phone a Polycom IP300 works fine. We got automon working by removing DYNAMIC_FEATURES=automon and putting the wW in Dial() not Queue(). I checked out canreinvite, it has had no impact on automon working or not. Does anybody know a way around this on a GXP2000? Regards Jenn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Francesco Peeters (Asterisk) Sent: Friday, January 13, 2006 5:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] automon - one touch record On Fri, January 13, 2006 5:15, Jennifer Hales said: Hello all, I am unable to get automon recording to work; can someone advise me what I am doing wrong? When I do *1 all I see in the CLI screen is attempting native bridge of SIP/3006-291b and SIP/3153-6fdd, and there is no call record generated in /var/spool/asterisk/monitor/. Here are my settings: SNIP Does transferring with # or *2 work? (Or whatever sequences you assigned to those functions in feastures.conf...) That way you can get an idea whether it is just automon, or whether there's a more generic issue... Also: What are the SIP CanReinvite settings for these phones? Good luck! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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 ___ --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] automon - one touch record
Hi Doug,
On touch record and Monitor are two different ways of recording. This is
how I have input the 'w' (for one touch) into my dial plan and it works. So
if a staff member receiving a call wants to record a conversation they input
*1 (as per features.conf)and it starts recording. When they are finished
they press *1 again and the recording stops.
exten =_3xxx,2,Dial(SIP/${EXTEN},30,wth).
Hope this is of some help.
Regards
Jenn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Tuesday, January 17, 2006 11:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] automon - one touch record
Actually the docs for the Queue application say:
'w' -- allow the called user to write the conversation to disk via
Monitor
'W' -- allow the calling user to write the conversation to disk via
Monitor
couldn't get these to work tho. Does this mean I can do one touch recording
with agents, or does it mean I can use the monitor() command? Very
confusing...
Doug.
-Original Message-
From: Jennifer Hales [mailto:[EMAIL PROTECTED]
Sent: Sunday, January 15, 2006 4:01 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] automon - one touch record
Hello Kevin,
Thank you for your response. I commented out DYNAMIC_FEATURES and moved the
'Ww' option to the Dial() instead of Queue() and now it works.
Regards
Jenn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Friday, January 13, 2006 3:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] automon - one touch record
Jennifer Hales wrote:
I am unable to get automon recording to work; can someone advise me what I
am doing wrong? When I do *1 all I see in the CLI screen is attempting
native bridge of SIP/3006-291b and SIP/3153-6fdd, and there is no call
record generated in /var/spool/asterisk/monitor/.
I can't explain why it's not working, but DYNAMIC_FEATURES is not
necessary if you are providing the 'wW' options to the Queue application
as you are.
Can you try this with a regular Dial() call instead, to eliminate the
queue application?
___
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[Asterisk-Users] automon - one touch record
Hello all, I am unable to get automon recording to work; can someone advise me what I am doing wrong? When I do *1 all I see in the CLI screen is attempting native bridge of SIP/3006-291b and SIP/3153-6fdd, and there is no call record generated in /var/spool/asterisk/monitor/. Here are my settings: Asterisk Version 1.2.1 Sox Version 12.17.5 features.conf [featuremap] automon = *1 extensions.conf [globals] DYNAMIC_FEATURES=automon [test] exten = s,4,Queue(test|twW|||1800) Regards Jenn ___ --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] voicemail locking
I had the same issue here and found when I went to the file /var/spool/asterisk/voicemail/default//INBOX that there were multiple empty files that were not in other voicemail accounts. I deleted these empty files and everything started working again for that person. Hope this helps Regards Jenn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sharon Sent: Tuesday, November 15, 2005 9:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] voicemail locking Hello, We are running asterisk cvs head on our servers. We are having issues with the voicemail getting locked for some users when opposite person tries to leave a voicemail. This happens randomly .Error message seen on the server: ast_lock_path: Failed to lock path /var/spool/asterisk/voicemail/default/1234/INBOX: File exists please help. Thank you, ___ --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] Choppy Audio in Echo Test and Music On Hold(1.2.0-b2)
We had problems with music on hold and finally decided to move to option 2 on the faking it document. We have not had any trouble since. Good luck. http://www.voip-info.org/wiki-Asterisk+mpg123+faking+it Regards Jenn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Tracy Sent: Tuesday, November 08, 2005 2:42 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Choppy Audio in Echo Test and Music On Hold(1.2.0-b2) I recently resurrected an old athlon system and put CentOS 4.2 on it to play with asterisk. First I tried asterisk-1.0.9, now I'm using 1.2.0-b2. Both have the same audio issues that have me stumped. I looked through all the lists and forums and the closest I could get were some messages from 2003: http://lists.digium.com/pipermail/asterisk-users/2003-August/017171.html I've got asterisk set up with my xten-lite softphone on extension 200 over SIP. I've configured extension 611 as an echo test and 612 will play 30 seconds of MusicOnHold. I can connect to both just fine, however, they sound rather bad when they work (quite muddy) and periodically they just drop out for as much as 5 seconds before coming back. Enabling all the debugging and verbosity options, I've found a few messages that occur during each drop. During the MOH run, every time there's a drop, the console scrolls: res_musiconhold.c:535 monmp3thread: Only wrote -1 of 640 bytes to pipe over and over until the sound comes back, at which point, the console message: rtp.c:1247 ast_rtp_raw_write: Difference is 33824, ms is 4248 is displayed. (Not always the same numbers in that one, obviously) In the echo test, again, after a drop, the audio returns and a message similar to: rtp.c:1247 ast_rtp_raw_write: Difference is 12496, ms is 1582 is displayed. The asterisk server is on a single Athlon MP 1600+ (1.4GHz) with 512MB of RAM. It's got a K7D-Master mobo, and is connected to the system running the softphone through a 100Mbit LAN. I've not enabled any of the MMX optimizations as there were warnings that they didn't play nice with AMD chips. If there's any further info I can provide, I'd be happy to. Thanks, Chris ___ --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] Testing with X101P
Hello Carlos, Try putting in a exten = s,5,WaitExten,5 after your background,welcome. It will give you 5 seconds to input your extension. Regards Jenn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Medina Sent: Monday, November 07, 2005 11:56 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Testing with X101P Hi there, im testing my asterisk box using a Modem Intel 56K which on the documentation says it must have the same behavior as an X101P. So im trying to configure just a simple line with 6 extensions. Asterisk loads fine and when im testing an incoming call the welcome message answers but when im trying to dial to any extension, anything happens is like asterisk dont recognice any digit after the welcome. Im using Asterisk version 1.0.9 and im attaching my extensions.conf which is very simple. Any clue will be very helpful. Thanks a lot for your help. Carlos Andres Medina __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --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] escaping to an extension whilelisteningtovoicemail message
Hope this helps.
exten = s,1,Dial(${ARG1},30,t)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG2})
exten = s-NOANSWER,2,Hangup
exten = s-BUSY,1,Voicemail(b${ARG2})
exten = s-BUSY,2,Hangup
exten = s-CHANUNAVAIL,1,Voicemail(u${ARG2})
exten = s-CHANUNAVAIL,2,Hangup
exten = s-.,1,Goto(s-NOANSWER,1)
exten = o,1,Hangup
Regards
Jenn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Monday, November 07, 2005 12:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] escaping to an extension
whilelisteningtovoicemail message
Can you post an example?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|[EMAIL PROTECTED]
|Sent: Sunday, November 06, 2005 3:32 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] escaping to an extension while
|listeningtovoicemail message
|
|The 'o' works well - especially with the attended transfer function.
|
|PaulH
|
|- Original Message -
|From: Eric ManxPower Wieling [EMAIL PROTECTED]
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|asterisk-users@lists.digium.com
|Sent: Monday, November 07, 2005 4:58 AM
|Subject: Re: [Asterisk-Users] escaping to an extension while
|listening tovoicemail message
|
|
| Anton Krall wrote:
| Guys.
|
| I was wondering, some voicemail systems let you escape to another
|extension
| or context while listening to the voicemail greeting, for
|example, for
| leaving faxes, like Hi, you have reached XXX, if you want
|to leave a
|fax,
| press 5 now, otherwise stay to leave voicemail.
|
| Can this be done on asterisk?
|
| See show application voicemail Pay special attention to the notes
| about the o and a extensions.
| ___
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|
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RE: [Asterisk-Users] Meetme Conference-reg
Yep,If you do not have a card installed, then you will need to use ztdummy. http://www.voip-info.org/wiki-Asterisk+timer+ztdummy Regards Jenn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici Sent: Monday, November 07, 2005 3:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Meetme Conference-reg If my memory serves you need some kind of zaptel device for meetme to work -- I think even dummy will do, but you need something. on Sunday 11/06/2005 nr k([EMAIL PROTECTED]) wrote Hi I configured the meetme number in the area where i specified the other extensions but still i am having pbm. herewith i am sending the error i got in the asterisk console. Nov 6 19:07:35 WARNING[4952]: chan_zap.c:770 zt_open: Unable to open '/dev/zap/pseudo': No such device or address Nov 6 19:07:35 ERROR[4952]: chan_zap.c:6731 chandup: Unable to dup channel: No such device or address Nov 6 19:07:35 WARNING[4952]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device Nov 6 19:07:35 WARNING[4952]: app_meetme.c:230 build_conf: Unable to open pseudo device regards ramakrishnan.n --- Rich Adamson [EMAIL PROTECTED] wrote: I am having Asterisk 1.0.9. now i configured the meetme conference with conference number 1234 and also i add the extension 1234 in extension.conf.if i call to 1234 asterisk says it's invalid conference number. i am having both sccp and sip devices. [room] ; Usage is conf = confno[,pin] conf = 1234 I assume you put the above in meetme.conf file? extension.conf [default] exten = 1234,1,Meetme(1234) Is the [default] section of extensions.conf where all of your other extensions are defined? If not, move the above entry to whatever section you have your other extensions defined. Then stop and restart asterisk. If the above doesn't address your issue, then copy/paste the CLI stuff so we can see what it is telling you. ___ --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 __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [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
[Asterisk-Users] Phone lines
Hello all, We have a situation where our 30 lines are maxing out, but no one is on a call. We are currently running CVS head downloaded on 15/8/2005 on a Dell Power Edge 2850. Our office mainly functions on a queue system. At the time this happened all our agents were logged in and no one was taking a call. Does anyone have a similar experience and ideas on how to fix this problem? This is our Cli printout. -- Playing 'queue-thank you' (language 'en') Sep 21 14:53:59 WARNING[3762]: chan_zap.c:8107 pri_dchannel: Ring requested on channel 0/9 already in use on span 1. Hanging up owner. shSep 21 14:54:00 WARNING[3762]: chan_zap.c:8107 pri_dchannel: Ring requested on channel 0/10 already in use on span 1. Hanging up owner. -- Started music on hold, class 'classic', on Zap/24-1 ow Sep 21 14:54:01 WARNING[3762]: chan_zap.c:8107 pri_dchannel: Ring requested on channel 0/11 already in use on span 1. Hanging up owner. -- Started music on hold, class 'classic', on Zap/22-1 Sep 21 14:54:01 WARNING[3762]: chan_zap.c:8107 pri_dchannel: Ring requested on channel 0/12 already in use on span 1. Hanging up owner. chaSep 21 14:54:01 WARNING[3762]: chan_zap.c:8107 pri_dchannel: Ring requested on channel 0/13 already in use on span 1. Hanging up owner. nnelsSep 21 14:54:02 WARNING[3762]: chan_zap.c:8107 pri_dchannel: Ring requested on channel 0/14 already in use on span 1. Hanging up owner. No such command 'showshow' (type 'help' for help) *CLI Sep 21 14:54:03 WARNING[3762]: chan_zap.c:8107 pri_dchannel: Ring requested on channel 0/17 already in use on span 1. Hanging up owner. show channels Channel Location State Application(Data) Zap/1-1 [EMAIL PROTECTED]:4 Up Queue(other|t|||1800) Zap/4-1 [EMAIL PROTECTED]:4 Up Queue(cts|t|||1800) Zap/8-1 [EMAIL PROTECTED]:4 Up Queue(csales|t|||30) Zap/20-1 [EMAIL PROTECTED]:4 Up Queue(cts|t|||1800) Zap/21-1 [EMAIL PROTECTED]:4 Up Queue(cts|t|||1800) Zap/19-1 [EMAIL PROTECTED]:4 Up Queue(cts|t|||1800) Zap/7-1 [EMAIL PROTECTED]:4 Up Queue(csales|t|||30) Zap/27-1 [EMAIL PROTECTED]:4 Up Queue(cts|t|||1800) Zap/18-1 [EMAIL PROTECTED]:4 Up Queue(cts|t|||1800) Zap/17-1 [EMAIL PROTECTED]:4 Up Queue(accounts|t|||1800) Zap/5-1 [EMAIL PROTECTED]:4 Up Queue(onetech|t|||1800) Zap/14-1 [EMAIL PROTECTED]:4 Up Queue(sales|t|||1800) Zap/3-1 [EMAIL PROTECTED]:4 Up Queue(onetech|t|||1800) Zap/13-1 [EMAIL PROTECTED]:4 Up Queue(other|t|||1800) Zap/12-1 [EMAIL PROTECTED]:4 Up Queue(cts|t|||1800) Zap/11-1 [EMAIL PROTECTED]:4 Up Queue(rts|t|||1800) Zap/28-1 [EMAIL PROTECTED]:4 Up Queue(cts|t|||1800) Zap/25-1 [EMAIL PROTECTED]:4 Up Queue(cts|t|||1800) Zap/6-1 [EMAIL PROTECTED]:4 Up Queue(rts|t|||1800) Zap/22-1 [EMAIL PROTECTED]:4 Up Queue(cts|t|||1800) Zap/10-1 [EMAIL PROTECTED]:4 Up Queue(other|t|||1800) Zap/9-1 [EMAIL PROTECTED]:4 Up Queue(rts|t|||1800) Zap/15-1 [EMAIL PROTECTED]:4 Up Queue(cts|t|||1800) Zap/2-1 [EMAIL PROTECTED]:4 Up Queue(onetech|t|||1800) Zap/24-1 [EMAIL PROTECTED]:4 Up Queue(cts|t|||1800) Zap/31-1 [EMAIL PROTECTED]:4 Up Queue(onetech|t|||1800) Zap/23-1 [EMAIL PROTECTED]:4 Up Queue(cts|t|||1800) Zap/30-1 [EMAIL PROTECTED]:4 Up Queue(cts|t|||1800) Zap/26-1 [EMAIL PROTECTED]:4 Up Queue(onetech|t|||1800) Zap/29-1 [EMAIL PROTECTED]:4 Up Queue(onetech|t|||1800) 30 active channels 30 active calls -- Stopped music on hold on Zap/15-1 -- Playing 'queue-thereare' (language 'en') Kind regards Jennifer Hales ___ --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] ChanSpy
Hi all,
Does any one know how to make the g option
work with Chanspy? I have done this and it does not work.
[snoop]
include = restricted
exten
=756,1,Set(${SPYGROUP}=1)
exten
=756,2,ChanSpy(Agent,qg)
exten =756,3,Hangup
Regards
Jenn Hales
___
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[Asterisk-Users] Unable to hear.
Hi all, Has anyone had problems with not being able to hear callers and them not being able to hear you? And had any success on how to fix it? Our call centre staff are complaining that this is a continual problem. Appreciate any thoughts on this. Regards Jennifer Hales ___ --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] Dell 2850 anyone ...
Hello, We have experienced problems with our Dell 2850 as well. Asterisk would slow down and people calling in would not be able to get through (receive an engaged signal). I had a chat to Digium and they advised that they experienced occasional issues where the Intel gigabit ethernet module for some reason (occasionally) causes the interrupt for the zaptel driver to be removed. What we captured was our network cards stopping and starting again when these symptoms occurred. We have now replaced our network cards in the machine and are still waiting to see if this has fixed our problem. I hope this is of some help. Regards Jennifer Hales -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of The VoIP Connection Sent: Friday, August 26, 2005 9:27 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Dell 2850 anyone ... Hi Bill, We just built one for a customer with Fedora Core 3 and a TE210. We get PCI parity errors and the machine shuts down. I'm sure we'll get it working, but it hasn't exactly been the smoothest install ever. I agree that the second CPU and GB of RAM is probably overkill and as you know, I also share your bias towards two smaller servers as opposed to one big one. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: William Boehlke [mailto:[EMAIL PROTECTED] Sent: Thursday, August 25, 2005 5:32 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Dell 2850 anyone ... We successfully use 2850s with Digium T1 cards, though I don't think we've installed a TE411P. It'll handle two T1s with ease. You don't need the second processor or the second GB of RAM for the expected load. For your configuration we would usually use two single processor 1u servers with RAID 1 for roughly the same cost so we're not vulnerable to a motherboard failure. William Boehlke Signate -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alan Bunch Sent: Thursday, August 25, 2005 2:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Dell 2850 anyone ... Can anyone comment or share experences with using Dell 2850's with Asterisk. Proposed config is 2850, 2 x 3.6g procs, 2 g's of ram, 4 x 36g 15k rpm drives raid 10, Digium TE411P ( the echo cancelling cards ). Expected load is 1 or 2 pri's (most likely 1 ) 100 Polycom phones on the local network, 15 phone on a remote T1. 6 phone remote via the internet using IAX, Voicemail for 125 users. As little transcoding as possible. G.729 licenses. If Dell is not the answer how about sharing what works. I do need a natinal brand that I can take to managment. Alan ___ --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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.15/81 - Release Date: 8/24/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.15/81 - Release Date: 8/24/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cvs Head
Hello Asterisk Users, Does anyone have a stable cvs head release date you can recommend? It will need to be deemed stable in a queue environment. We are currently running Centos 2.6 kernal and have implemented different versions of cvs head with varying results. I am currenly using cvs head 20/05/2005 however it is not utilizing the wrapuptime function in Agents.conf. We are real close as this appears to be our only problem. Appreciate your help Kind regards Jennifer Hales ___ 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] Ringing a few phones
If you want to dial a number of phones at the same time do exten = 5000,1,Dial(SIP/5000SIP/5001SIP?5002). The value is what does the job. Kind regards Jenn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shidan Sent: Thursday, June 09, 2005 11:01 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Ringing a few phones I have a client requirement that multiple phones can be dialed, however they don't want the pstn phone to pick up automatically because of voicemail etc, nothing can be changed on the phones, how can I handle this requirement, by the way no zap channels are involved, all the pstn phones are behing another sip gateway. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Built-In Transfer Questions
Hello Matthew, You need to put exten = o,1,Hangup underneath your voicemail macro, then if your dial zero the call will come back to you, however it does read back an error in your ear. It still works. Jenn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Wednesday, June 01, 2005 1:14 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Built-In Transfer Questions I've read the Wiki on using asterisk's built-in transfer options (#8 and #6). They work fine but how does one cancle an attended transfer? Example: I have person on phone, I hit #6 to being att-transfer. I enter Sally's extension. I let it ring for a few seconds. Sally never picks up but her voicemail does. How do I hangup her voicemail and resume the previous call? The example on the wiki assumes the transferee picks up the phone. :/ -Matthew -- Matthew Boehm, IT DirectorCypress Telecommunications [EMAIL PROTECTED] 3838 N. Sam Houston Parkway E #400 T: 832-200-8640 x3044 Houston, TX 77032 My girlfriend was recently diagnosed with multiple personality disorder; When she called yesterday, my CallerID box exploded. ___ 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] App_odbcexec
Hello everyone, I have just gone through the installation process to add commands ODBCexec and ODBCquery to the extensions.conf. However I am receiving an error in asterisk No application ODBCquery for extension (incoming,33,2). I have installed unixODBC-2.2.11 and myODBC 3.51. I have gone through the instructions as per http://www.loligo.com/asterisk/misc/apps/odbc/app_odbcexec twice. Does anyone know what I may have missed? Or if there are issues with these versions? Kind regards Jenn ___ 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] Realtime
Hello Mark, My employer has asked me to email you in regard to the development of Realtime. We are hoping to obtain an overall idea of where you ultimately see realtime going, as well as the status on the current development. Our guys here are exited about the prospects of Realtime and we want to get an idea as to whether their ideas have merit. I understand that you are very busy and would appreciate it if you can not personally respond maybe you can put me in touch with someone who can. Kind regards Jenn Hales ___ 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] Queue/Agent recording and configuration
Good Morning, Does anyone know a way around my problem? The call is from a queue. I need to know how to play a message to the customer (terms conditions) keep the agent with the call while a message is played and record only a small portion of the call (the callers acceptance of the terms and conditions). Kind regards Jenn Hales ___ 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 from/to a queue
Good Morning, Does anyone know if it is possible to transfer a caller from queue 1 to queue 2? Regards Jenn Hales ___ 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
