Re: [asterisk-users] What do you use? .conf or AEL?
On Wed, 11 Feb 2009, Tilghman Lesher wrote: On Wednesday 11 February 2009 13:07:16 Gordon Henderson wrote: In my world, they go to the web interface, create the extension, tick a selection of existing extensions and the code then writes out a new segment of dialplan to create the new extension, issues an extensions reload command to asterisk and off it goes... The dialplan was static before, and is static after, it's just that I wrote some php to write dialplan based on user input... How do others do it? I'd have a range of extensions, when dialled, it goes to the database, retrieves the list of channels, and dials those channels. The web frontend would look exactly the same, but the data would go directly into a database, not taking an extra step to go into a dialplan, then reload the text file. The advantage is that I'd never have the possibility of two people colliding on the regeneration of a text file. While it is possible for two people to select the same number when defining new extensions, that can be very easily worked around, given that database updates are atomic. Intersting, but I have no database. (well, not in the sense of a relational or SQL based one) I build systems that boot off flash and run them from RAM. Configuration is stored back on flash, but not using a relational database. Thats just too heavy IMO for embedded systems, and storing data in flat-files is just as efficient, if not more-so for a few 100 entries. The other issues, locking, etc. these are all classic computer science problems which have been long-solved so shouldn't really be an issue (and my background is real-time control, robotics, parallel processing, etc. so from my point of view nothing more than an academic excercise) Cheers, Gordon (Off to dine with some philosophers now and travel round Exeter, selling my wares before going to book an airline ticket ;-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday the 13th Muhahaha Allison Smith and more on the Polycom Applications
Hi, Allison Smith continues to contribute to the open source asterisk resources and she is launching a new site that will make it even easier to grab sound files. Allison joins us to talk about that and whatever else comes up. She says: I have been the Voice of Asterisk -- the world's fastest-growing telephony platform -- since its inception (which for me, was marked by an animated e-mail I received around 2002 from some hyperactive young guys in Alabama, asking me to do their quite unusual and offbeat prompts -- I was more than happy to voice them, but pretty much dismissed it as a one-off. Just one of those fun sessions in which I was voicing things like Weasels have eaten our phone system! instead of the usual Press One for Technical Support IVR fare which I was accustomed to voicing.) Up until that point, I'd never heard of Digium (owners and maintainers of Asterisk), and who could anticipate how big and rapidly Asterisk would grow? Allison is always fun to have on board, be there live to welcome her. After Polycom last week, we need to kick around ideas for the applications contest, so a little later in the call we will be doing just that. Join us for this weekly event, it's basically a large Asterisk User Group: http://www.voipusersconference.org Permatime: http://permatime.com/CET/2009-02-13/18:00/VoIP_Users_Conference SIP: 7463#2262...@proxy.ideasip.com (Please use your PIN in the place of the 1) IRC #voip-users-conference on Freenode.net /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Keep your passwords secure .. (VoIP hacker news)
http://www.theregister.co.uk/2009/02/11/fugitive_voip_hacker_arrested/ Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Siemens Hipath PRI to Asterisk Call Routing?
Hi all, I have a connect between a siemens hipath Asterisk system over PRI The connection works perfectly I can call from the Hipath to an Asterisk Extension. I want to allow the hipath extensions to dial out over a SIP trunk on asterisk but I keep getting The number you have dialed is not in service In this CLI I'm dialing from hipath extension 9 (prefix for sip trunk) then the number 1905 (Freefone number in Ireland) Please help I cant figure this one out. Thanks, Joe CLI - [Feb 11 17:45:25] VERBOSE[4526] logger.c: -- Accepting overlap call from '0339' to 'unspecified' on channel 0/31, span 1 [Feb 11 17:45:25] VERBOSE[5764] logger.c: -- Starting simple switch on 'Zap/31-1' [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [91...@from-pstn:1] Set(Zap/31-1, __FROM_DID=91905) in new stack [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [91...@from-pstn:2] NoOp(Zap/31-1, Received an unknown call with DID set to 91905) in new stack [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [91...@from-pstn:3] Goto(Zap/31-1, s|a2) in new stack [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Goto (from-pstn,s,2) [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [...@from-pstn:2] Answer(Zap/31-1, ) in new stack [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [...@from-pstn:3] Wait(Zap/31-1, 2) in new stack [Feb 11 17:45:33] VERBOSE[5764] logger.c: -- Executing [...@from-pstn:4] Playback(Zap/31-1, ss-noservice) in new stack [Feb 11 17:45:33] VERBOSE[5764] logger.c: -- Zap/31-1 Playing 'ss-noservice' (language 'en') [Feb 11 17:45:38] VERBOSE[5764] logger.c: -- Executing [...@from-pstn:5] SayAlpha(Zap/31-1, 91905) in new stack [Feb 11 17:45:38] VERBOSE[5764] logger.c: -- Zap/31-1 Playing 'digits/9' (language 'en') [Feb 11 17:45:39] VERBOSE[4526] logger.c: -- Channel 0/31, span 1 got hangup request, cause 16 [Feb 11 17:45:39] WARNING[5764] file.c: Failed to write frame [Feb 11 17:45:39] VERBOSE[5764] logger.c: == Spawn extension (from-pstn, s, 5) exited non-zero on 'Zap/31-1' [Feb 11 17:45:39] VERBOSE[5764] logger.c: -- Executing [...@from-pstn:1] Hangup(Zap/31-1, ) in new stack [Feb 11 17:45:39] VERBOSE[5764] logger.c: == Spawn extension (from-pstn, h, 1) exited non-zero on 'Zap/31-1' [Feb 11 17:45:39] DEBUG[5764] chan_dahdi.c: Set option AUDIO MODE, value: ON(1) on Zap/31-1 [Feb 11 17:45:39] DEBUG[5764] chan_dahdi.c: Not yet hungup... Calling hangup once with icause, and clearing call [Feb 11 17:45:39] DEBUG[5764] chan_dahdi.c: Set option AUDIO MODE, value: OFF(0) on Zap/31-1 [Feb 11 17:45:39] VERBOSE[5764] logger.c: -- Hungup 'Zap/31-1' ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siemens Hipath PRI to Asterisk Call Routing?
Which line of code is generating this log entry? [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [91...@from-pstn:3] Goto(Zap/31-1, s|a2) in new stack ...because this appears to be where your problem lies. joek...@gmail.com wrote: Hi all, I have a connect between a siemens hipath Asterisk system over PRI The connection works perfectly I can call from the Hipath to an Asterisk Extension. I want to allow the hipath extensions to dial out over a SIP trunk on asterisk but I keep getting The number you have dialed is not in service In this CLI I'm dialing from hipath extension 9 (prefix for sip trunk) then the number 1905 (Freefone number in Ireland) Please help I cant figure this one out. Thanks, Joe CLI - [Feb 11 17:45:25] VERBOSE[4526] logger.c: -- Accepting overlap call from '0339' to 'unspecified' on channel 0/31, span 1 [Feb 11 17:45:25] VERBOSE[5764] logger.c: -- Starting simple switch on 'Zap/31-1' [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [91...@from-pstn:1] Set(Zap/31-1, __FROM_DID=91905) in new stack [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [91...@from-pstn:2] NoOp(Zap/31-1, Received an unknown call with DID set to 91905) in new stack [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [91...@from-pstn:3] Goto(Zap/31-1, s|a2) in new stack [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Goto (from-pstn,s,2) [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [...@from-pstn:2] Answer(Zap/31-1, ) in new stack [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [...@from-pstn:3] Wait(Zap/31-1, 2) in new stack [Feb 11 17:45:33] VERBOSE[5764] logger.c: -- Executing [...@from-pstn:4] Playback(Zap/31-1, ss-noservice) in new stack [Feb 11 17:45:33] VERBOSE[5764] logger.c: -- Zap/31-1 Playing 'ss-noservice' (language 'en') [Feb 11 17:45:38] VERBOSE[5764] logger.c: -- Executing [...@from-pstn:5] SayAlpha(Zap/31-1, 91905) in new stack [Feb 11 17:45:38] VERBOSE[5764] logger.c: -- Zap/31-1 Playing 'digits/9' (language 'en') [Feb 11 17:45:39] VERBOSE[4526] logger.c: -- Channel 0/31, span 1 got hangup request, cause 16 [Feb 11 17:45:39] WARNING[5764] file.c: Failed to write frame [Feb 11 17:45:39] VERBOSE[5764] logger.c: == Spawn extension (from-pstn, s, 5) exited non-zero on 'Zap/31-1' [Feb 11 17:45:39] VERBOSE[5764] logger.c: -- Executing [...@from-pstn:1] Hangup(Zap/31-1, ) in new stack [Feb 11 17:45:39] VERBOSE[5764] logger.c: == Spawn extension (from-pstn, h, 1) exited non-zero on 'Zap/31-1' [Feb 11 17:45:39] DEBUG[5764] chan_dahdi.c: Set option AUDIO MODE, value: ON(1) on Zap/31-1 [Feb 11 17:45:39] DEBUG[5764] chan_dahdi.c: Not yet hungup... Calling hangup once with icause, and clearing call [Feb 11 17:45:39] DEBUG[5764] chan_dahdi.c: Set option AUDIO MODE, value: OFF(0) on Zap/31-1 [Feb 11 17:45:39] VERBOSE[5764] logger.c: -- Hungup 'Zap/31-1' ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siemens Hipath PRI to Asterisk Call Routing?
I thing, you have bad routing configuration in extensions.conf. Send me from-pstn context configuration. turby joek...@gmail.com napsal(a): Hi all, I have a connect between a siemens hipath Asterisk system over PRI The connection works perfectly I can call from the Hipath to an Asterisk Extension. I want to allow the hipath extensions to dial out over a SIP trunk on asterisk but I keep getting The number you have dialed is not in service In this CLI I'm dialing from hipath extension 9 (prefix for sip trunk) then the number 1905 (Freefone number in Ireland) Please help I cant figure this one out. Thanks, Joe CLI - [Feb 11 17:45:25] VERBOSE[4526] logger.c: -- Accepting overlap call from '0339' to 'unspecified' on channel 0/31, span 1 [Feb 11 17:45:25] VERBOSE[5764] logger.c: -- Starting simple switch on 'Zap/31-1' [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [91...@from-pstn:1] Set(Zap/31-1, __FROM_DID=91905) in new stack [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [91...@from-pstn:2] NoOp(Zap/31-1, Received an unknown call with DID set to 91905) in new stack [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [91...@from-pstn:3] Goto(Zap/31-1, s|a2) in new stack [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Goto (from-pstn,s,2) [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [...@from-pstn:2] Answer(Zap/31-1, ) in new stack [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [...@from-pstn:3] Wait(Zap/31-1, 2) in new stack [Feb 11 17:45:33] VERBOSE[5764] logger.c: -- Executing [...@from-pstn:4] Playback(Zap/31-1, ss-noservice) in new stack [Feb 11 17:45:33] VERBOSE[5764] logger.c: -- Zap/31-1 Playing 'ss-noservice' (language 'en') [Feb 11 17:45:38] VERBOSE[5764] logger.c: -- Executing [...@from-pstn:5] SayAlpha(Zap/31-1, 91905) in new stack [Feb 11 17:45:38] VERBOSE[5764] logger.c: -- Zap/31-1 Playing 'digits/9' (language 'en') [Feb 11 17:45:39] VERBOSE[4526] logger.c: -- Channel 0/31, span 1 got hangup request, cause 16 [Feb 11 17:45:39] WARNING[5764] file.c: Failed to write frame [Feb 11 17:45:39] VERBOSE[5764] logger.c: == Spawn extension (from-pstn, s, 5) exited non-zero on 'Zap/31-1' [Feb 11 17:45:39] VERBOSE[5764] logger.c: -- Executing [...@from-pstn:1] Hangup(Zap/31-1, ) in new stack [Feb 11 17:45:39] VERBOSE[5764] logger.c: == Spawn extension (from-pstn, h, 1) exited non-zero on 'Zap/31-1' [Feb 11 17:45:39] DEBUG[5764] chan_dahdi.c: Set option AUDIO MODE, value: ON(1) on Zap/31-1 [Feb 11 17:45:39] DEBUG[5764] chan_dahdi.c: Not yet hungup... Calling hangup once with icause, and clearing call [Feb 11 17:45:39] DEBUG[5764] chan_dahdi.c: Set option AUDIO MODE, value: OFF(0) on Zap/31-1 [Feb 11 17:45:39] VERBOSE[5764] logger.c: -- Hungup 'Zap/31-1' ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OPTIONS packets
11 feb 2009 kl. 10.43 skrev michel freiha: Hi all, I need to register asterisk on an OpenSIPS SIP Proxy...The Registration is OK but my asterisk is sending OPTIONS packets to OpenSIPS and the SIP Proxy is not replying back...The issue is the UNKNOWN username that reside in the OPTIONS packet as you can see in the captured packets as you can see below: OpenSIPS/kamailio/OpenSER/xxx proxy won't answer unless you tell it to. Check the options module and then change your routing script configuration so that the proxy starts answering. Make sure you only answer to requests addressed to the actual proxy. The proxy will need to proxy requests to other URIs. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup extensions via CLI?
Asterisk 1.6 implements the hangup on the channel you just made the call
and I used it with this command (apparently)
asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep SIP/7000|
awk '{ print $1 '} )
In my asterisk system:
debian*CLI core show channels
Channel Location State Application(Data)
SIP/7000-09c63a30(None) Up AppDial((Outgoing Line))
SIP/-09c599387...@internos:5 Up Dial(SIP/7000)
2 active channels
1 active call
6 calls processed
debian*CLI
debian:~# asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep
SIP/7000|awk '{ print $1 '} )
SIP/7000-09c63a30
SIP/-09c59938 is not a known channel
But, with the channel SIP/-09c59938 is OK.
asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep SIP/|
awk '{ print $1 '} )
Requested Hangup on channel 'SIP/-09c59938'
I use asterisk 1.6.1 beta4
On Wed 11 Feb 2009 09:34:12 Lenz Emilitri wrote:
This is a bit of trickery, but could not resist :)
This will kill a channel that is connected to SIP/201
asterisk -rx soft hangup $(asterisk -rx 'show channels' | grep SIP/201 |
awk '{ print $1 '} )
It basically calls *, gets the list of channels, filters them out to get
the channel name and hangs it up.
OK, using AMI and a real programming language and hadling multiple lines
would be better.
Thanks
l.
2009/2/9 Tim Nelson tnel...@rockbochs.com
Greetings list-
I'd like the ability to hangup all calls for a particular extension from
the system CLI. I understand this can probably be scripted using the AMI
but I'm not familiar on how to do it. Help!
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
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Re: [asterisk-users] OPTIONS packets
Thanks Johansson, Everything is OK now and you were right Regards On Thu, Feb 12, 2009 at 12:30 PM, Johansson Olle E o...@edvina.net wrote: 11 feb 2009 kl. 10.43 skrev michel freiha: Hi all, I need to register asterisk on an OpenSIPS SIP Proxy...The Registration is OK but my asterisk is sending OPTIONS packets to OpenSIPS and the SIP Proxy is not replying back...The issue is the UNKNOWN username that reside in the OPTIONS packet as you can see in the captured packets as you can see below: OpenSIPS/kamailio/OpenSER/xxx proxy won't answer unless you tell it to. Check the options module and then change your routing script configuration so that the proxy starts answering. Make sure you only answer to requests addressed to the actual proxy. The proxy will need to proxy requests to other URIs. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] g723 llicense
Hi, Looking to buy at least 20 license for g723. May i know where to purchase please? Regards, Nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disabling Echo Cancellation on a per Call basis
Hello, just want to bring this question up again. Is someone able to tell me, yes you can disable echo can on a per call basis and it is done this way, or not it is not possible, go away ... If it is not possible, why is that so ? Is there really no need to do this and i am totally mistaken? Generally receiving faxes works fine, but sometimes they break and i assume i might have something to do with echo can. Kind Regards, Tobias Tobias Wolf schrieb: Olivier schrieb: 2009/2/10 Tobias Wolf tobias.w...@evision.de mailto:tobias.w...@evision.de Hello all, i was just made aware on the Bristuff-Mailing list, that it is possible to disable echo cancellation per dialplan application. This comes in very handy, for terminating faxes. But the application seems only to be existing in the bristuff patches. Does there exist a solution for Asterisk 1.6.0.3 Digium Wildcard TE110P T1/E1 DAHDI Version: 2.1.0.3 Echo Canceller: MG2 without any Bristuff? At the Moment i have fax detection enabled. Do you mean a given DID receives voice or fax calls ? If positive, which app is detecting faxes ? Since i have a dedicated DID for fax calls, i don't really need the fax detection. For this number i simply start the ReceiveFAX-Application and have some voodoo around it to name the file correctly. But if i do this, and look into the channel information from Dahdi i see that the fax handled flag is set to no. And this seems wrong to me. I have the feeling that the percentage of failed faxes is higher is this flag is set to no (or false, can't remember) ... Since i have a PRI connected to my Asterisk, i use the built-in fax detection of DAHDI. I have enabled it for incoming fax calls, in chan_dahdi.conf faxdetect=incoming The incoming call is answered and with an included Wait(4) the fax is detected and switched to the fax extension, where the ReceiveFAX-App is executed. Now the fax handled flag is set to yes and i am able to receive most of the fax calls. But i have massive problems receiving fax calls from certain people, especially from UK (i am in germany). I am not quite sure, if the echo canceller is automatically disabled if DAHDI knows that the call is a fax and the channel info doesn't indicate otherwise, since it says that echo canceller is active even if it says that it handles an fax. This is the reason why i was so happy to hear, that there seems to be the option to control the echo canceller with an dialplan app. But since this seems to be an Bristuff-only feature i am a little bit stuck. Kind regards, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Keep your passwords secure .. (VoIP hacker news)
On Thu, Feb 12, 2009 at 3:27 AM, Gordon Henderson gordon+aster...@drogon.net wrote: http://www.theregister.co.uk/2009/02/11/fugitive_voip_hacker_arrested/ Gordon Of course the following only applies if his is, in fact, guilty. It seems that the author and probably everyone reading this has already convicted this guy. I bet the only reason he got caught in the first place was his financial activities. What a scumbag to skip out on bail secured by his girlfriend's mother's house, I hope he either bought that house or had some some funds set aside to pay her back. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] After Monitor() files disappear
Hello list, Using Asterisk 1.2.29 I use the Monitor() application. In extensions.conf I have set MONITOR_EXEC to my script (for mixing files together and convert to mp3) and I set TOUCH_MONITOR on every new channel which has to be recorded. But sometimes I'm missing the recording files. I had a look to the Asterisk-Log and saw those lines: Feb 10 15:18:57 NOTICE[16772] res_monitor.c: monitor executing /root/recordings //var/spool/asterisk/monitor/X_20090210-151258-in.wav //var/spool/asterisk/monitor/X_20090210-151258-out.wav //var/spool/asterisk/monitor/X_20090210-151258.wav X is the phone-number. Exactly these recordings with two slashes at the beginning of the parameters (//var/spool...) I'm missing. My script does not delete them (for testing I cleared my script so it does nothing). While Monitor is running there are files in /var/spool/asterisk/monitor. Asterisk definitly records the call. Seems the files disappear after the channel is hung up. It must be an Asterisk problem. Anyone out there with an idea or a hint? I knew Asterisk 1.2 is out of date. But I can't believe that it is an Asterisk problem and no one else had the same problem. Thanks, Gunnar Schaller ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange dialplan matching issue
We use extensions like plant201 and tunnel12 so it does work in 1.4
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jose P.
Espinal
Sent: February 11, 2009 10:16 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Strange dialplan matching issue
On extensions.conf.sample I see this:
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal. In patterns, some characters have special meanings:
;
; X - any digit from 0-9
; Z - any digit from 1-9
; N - any digit from 2-9
; [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
; . - wildcard, matches anything remaining (e.g. _9011. matches
; anything starting with 9011 excluding 9011 itself)
; ! - wildcard, causes the matching process to complete as soon as
; it can unambiguously determine that no other matches are possible
Maybe after using '_' Asterisk is waiting for one of the above pattern
matching characters.
a. The 'hilton-' part of your dialplan might not being considered valid,
and Asterisk *might* be trying to match the 'XX' part LITERALLY, and
would be trying to reach extension '2XX'
exten = _hilton-2XX,1,Goto(hilton,${EXTEN:7},1)
b. then, in:
exten = hilton-203,1,Goto(hilton,${EXTEN:7},1)
You provided the real extension number (after you take out the fist 7
digits).
So, Asterisk reaches '203', etc.
Try only using valid pattern matching characters in your dialplan to see
if it works.
Chris Bagnall wrote:
Greetings list,
Wondering if anyone has come across this strange dialplan pattern matching
issue before:
I have a context defined as follows (the plus simply implies it follows on
from an existing context in another #include - which, yes, has been included
first):
[privatedundi](+)
exten = _hilton-2XX,1,Goto(hilton,${EXTEN:7},1)
When dialling hilton-202 from another box via IAX2, I get:
NOTICE[3727]: chan_iax2.c:8085 socket_process: Rejected connect attempt
from ip masked, request 'hilton-...@privatedundi' does not exist
Changing the context to read as follows solves the problem immediately:
[privatedundi](+)
exten = hilton-201,1,Goto(hilton,${EXTEN:7},1)
exten = hilton-202,1,Goto(hilton,${EXTEN:7},1)
exten = hilton-203,1,Goto(hilton,${EXTEN:7},1)
Dialling hilton-202 now works every time.
The *really* strange thing is that I have a number of similar pattern
matches, and all the others work fine, it's just this one that doesn't.
The box in question is running 1.4.22, but I have had a similar issue in
the past with a 1.2 box, so it does not appear to be version specific.
Any thoughts?
TIA.
Regards,
Chris
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--
Jose P. Espinal
http://www.eSlackware.com
IRC: [OFTC|FreeNode]
Khratos @ #slackware | #asterisk/-doc/-bugs
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Re: [asterisk-users] Keep your passwords secure .. (VoIP hacker news)
On Thu, 12 Feb 2009, Steve Totaro wrote: What a scumbag to skip out on bail secured by his girlfriend's mother's house, I hope he either bought that house or had some some funds set aside to pay her back. That won't help if he is convicted - his assets will be seized to pay the telcos. I'm betting he doesn't have much cash laying around anymore :) j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g723 llicense
Nhadie wrote: Looking to buy at least 20 license for g723. May i know where to purchase please? I don't believe any company sells licenses for software G.723.1 transcoding support for Asterisk. The only solution Digium offers is the TC400B hardware transcoder, which supports G.729 and G.723.1. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disabling Echo Cancellation on a per Call basis
Tobias Wolf wrote: If it is not possible, why is that so ? Is there really no need to do this and i am totally mistaken? This is generally true. Any standards-compliant FAX machine or modem will generate a CED tone during the beginning of the call process, and any standards compliant echo canceler (including the ones in Zaptel/DAHDI) will respond to this tone by disabling the echo canceler. With Zaptel/DAHDI and a software echo canceler, you can see the evidence of this by watching the kernel message log for messages of the form 'Disabled echo canceler because of tone (..) on channel If a hardware echo canceler is in use, there won't be any messages generated, but it is still happening. Generally receiving faxes works fine, but sometimes they break and i assume i might have something to do with echo can. Unfortunately that is not likely to be the cause of your problems. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup extensions via CLI?
Here's an improved hack to this bit of trickery:
Exten = _86,1,system('/usr/sbin/asterisk -rx soft hangup
$(/usr/sbin/asterisk -rx 'core show channels' | grep SIP/${EXTEN(2)| awk '{
print $1 '} ))
Where dialing 861234 would hangup extension 1234
If this needs refinement, will repost:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Helius
Ferreira
Sent: Thursday, February 12, 2009 4:42 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Hangup extensions via CLI?
Asterisk 1.6 implements the hangup on the channel you just made the call
and I used it with this command (apparently)
asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep
SIP/7000|
awk '{ print $1 '} )
In my asterisk system:
debian*CLI core show channels
Channel Location State Application(Data)
SIP/7000-09c63a30(None) Up AppDial((Outgoing Line))
SIP/-09c599387...@internos:5 Up Dial(SIP/7000)
2 active channels
1 active call
6 calls processed
debian*CLI
debian:~# asterisk -rx soft hangup $(asterisk -rx 'core show channels' |
grep
SIP/7000|awk '{ print $1 '} )
SIP/7000-09c63a30
SIP/-09c59938 is not a known channel
But, with the channel SIP/-09c59938 is OK.
asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep
SIP/|
awk '{ print $1 '} )
Requested Hangup on channel 'SIP/-09c59938'
I use asterisk 1.6.1 beta4
On Wed 11 Feb 2009 09:34:12 Lenz Emilitri wrote:
This is a bit of trickery, but could not resist :)
This will kill a channel that is connected to SIP/201
asterisk -rx soft hangup $(asterisk -rx 'show channels' | grep SIP/201 |
awk '{ print $1 '} )
It basically calls *, gets the list of channels, filters them out to get
the channel name and hangs it up.
OK, using AMI and a real programming language and hadling multiple lines
would be better.
Thanks
l.
2009/2/9 Tim Nelson tnel...@rockbochs.com
Greetings list-
I'd like the ability to hangup all calls for a particular extension from
the system CLI. I understand this can probably be scripted using the AMI
but I'm not familiar on how to do it. Help!
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
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[asterisk-users] Multiple caller id ...
If I have the following in the dialplan exten = foo,n,Dial(SIP/1234Zap/G1c/55443322) and SIP/5432 calls this extension, is it possible to show different callerid numbers to each of the target numbers ? The reason I ask is that if the call is from an internal sip phone, I want to show the internal callerid (5432) to the SIP phone on 1234, and the DDI of the 5432 extension (01702444555) to the zap line. Julian __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with parking
Hi,
I'm having problem with call parking.
When I park call, either via transfer to xten or park digit sequence from
features.conf, I hear the parking lot number read to me and the user gets
transferred.
However, MOH stops for the caller the moment user is transferred.
The user can be retrieved by dialing the parked extension and voice resumes.
If the parked user hangs up, the channel state does not update (nor the
hint) and call seems to be live still.
If the timeout for the park is reached, the user is not transferred back
(nothing happens actually -- he is just able to stay parked forever).
Even after the user hangs up, the parking lot extension exists and is
callable, but when that is done (the parking extension is called), it
answers and promptly hangs up and then is available for parking again.
Yesterday I had everything working like a charm, and I don't think I have
changed anything, although that seems increasingly unlikely since things
don't usually break on their own. I've restarted asterisk numerous times and
tried changing things I think are relevant, but with no avail.
It does not matter whether the caller is coming in from a SIP trunk or a SIP
peer.
The SIP messages look normal. When the caller hangs up while parked he sends
a BYE to the asterisk to the callee's number, and the asterisk replies with
ACK.
Any help would be appreciated.
My comments in asterisk CLI output below prefixed by #
Configuration files
== features.conf ==
[general]
parkext = 10
parkpos = 11-14
parkinghints = yes
== sip.conf ==
[general]
context=deadend
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
subscribecontext=parkedcalls
allowsubscribe=yes
notifyringing=yes
notifyhold=yes
limitonpeers=yes
canreinvite=no
[Tal]
context=incoming
type=friend
host=XXX.XXX.XXX.XXX
canreinvite=no
port=5060
dtmfmode=rfc2833
disallow=all
allow=alaw
[smg01]
type=friend
context=incoming
host=XXX.XXX.XXX.XXX
canreinvite=no
port=5060
dtmfmode=rfc2833
disallow=all
allow=alaw
[2552]
type=friend
context=dialplan-1
host=dynamic
call-limit=10
defaultuser=2552
secret=xxx
[2556]
type=friend
context=dialplan-1
host=dynamic
call-limit=10
defaultuser=2556
secret=xxx
== extensions.conf ==
[globals]
CID = 5822550
OCODE = 582
XTENS = 255X
DTRK = smg01
CONF = 2559
ADALNUMER = 2550
ADALDIAL = SIP/2551SIP/2552SIP/2553SIP/2554
BAKVAKT = 7712555
[general]
static=yes
writeprotect=yes
clearglobalvars=no
userscontext=default
[dialplan-1]
include = conferences
include = ringgroups
include = internal
include = landlines
include = gsm
include = special
include = international
include = parkedcalls
[dialplan-2]
include = conferences
include = ringgroups
include = internal
include = landlines
include = parkedcalls
[dialplan-3]
include = conferences
include = ringgroups
include = internal
include = landlines
include = gsm
include = parkedcalls
[dialplan-4]
include = conferences
include = ringgroups
include = internal
include = landlines
include = international
include = parkedcalls
[incoming]
exten = 4891001,1,Set(CALLERID(name)=SMG01)
exten = 4891001,n,Dial(${ADALDIAL})
exten =
${CID},1,Macro(ringgroup,${ADALDIAL},ADALNUMER,${ADALNUMER},ringgroup)
exten = ${OCODE}${CONf},1,Goto(conferences,${EXTEN:3},1)
exten = _${OCODE}${XTENS},1,Set(CALLERID(name)=BEIN HRINGING)
exten = _${OCODE}${XTENS},n,Macro(internal-call,${EXTEN:3},incoming)
[internal]
exten = _${XTENS},1,Macro(internal-call,${EXTEN},internal)
exten = *80,1,Page(SIP/2553)
exten = *97,1,VoiceMailMain(${CALLERID(num)},s)
exten = *98,1,Playback(templokun)
exten = *99,1,Record(templokun.wav)
exten = _9X.,1,Macro(trunkdial,Tal,${EXTEN:1},${CALLERID(num)},landline)
[landlines]
exten =
_800,1,Macro(trunkdial,${DTRK},${EXTEN},${CALLERID(num)},tollfree)
exten =
_[4-5]XX,1,Macro(trunkdial,${DTRK},${EXTEN},${CALLERID(num)},landline)
exten = _177X,1,Macro(trunkdial,${DTRK},${EXTEN},${CALLERID(num)},landline)
exten = 1817,1,Macro(trunkdial,${DTRK},${EXTEN},${CALLERID(num)},landline)
exten = 1414,1,Macro(trunkdial,${DTRK},${EXTEN},${CALLERID(num)},landline)
[gsm]
exten =
_[6-8]XX,1,Macro(trunkdial,${DTRK},${EXTEN},${CALLERID(num)},gsm)
[special]
exten = 112,1,Macro(trunkdial,${DTRK},${EXTEN},$CALLERID(num)},emergency)
exten = _11X,1,Macro(trunkdial,${DTRK},${EXTEN},${CALLERID(num)},special)
exten = _15X,1,Macro(trunkdial,${DTRK},${EXTEN},${CALLERID(num)},special)
exten =
_9XX,1,Macro(trunkdial,${DTRK},${EXTEN},${CALLERID(num)},special)
[international]
exten =
_00x.,1,Macro(trunkdial,${DTRK},${EXTEN},${CALLERID(num)},international)
[conferences]
exten = ${CONF},1,MeetMe(1,MsI)
[ringgroups]
exten =
${ADALNUMER},1,Macro(ringgroup,${ADALDIAL},ADALNUMER,${ADALNUMER},internal)
[forwarding]
exten = _X.,1,Set(CALLERID(all)=${origcid})
exten = _X.,n,Set(GLOBAL(origcid)=)
exten = _X.,n,Goto(dialplan-1,${DB(CF/${EXTEN})},1)
[2550]
exten = s,1,Answer()
exten = s,n,Background(templokun)
exten = s,n,WaitExten(4)
exten = 1,1,VoiceMail(2550)
exten =
Re: [asterisk-users] Keep your passwords secure .. (VoIP hacker news)
On Thu, Feb 12, 2009 at 9:16 AM, Jeff LaCoursiere j...@jeff.net wrote: On Thu, 12 Feb 2009, Steve Totaro wrote: What a scumbag to skip out on bail secured by his girlfriend's mother's house, I hope he either bought that house or had some some funds set aside to pay her back. That won't help if he is convicted - his assets will be seized to pay the telcos. I'm betting he doesn't have much cash laying around anymore :) His assets are long gone, he has been on the lamb for two or three years and it doesn't sound like he was very good at laundering money, reminds me of Office Space. http://www.youtube.com/watch?v=xvJSpwspils FUNNY! I would also bet if he doesn't fold to the Feds and cop a plea, many witnesses and evidence is not around anymore. A true jury of his Peers probably would realize that besides the money trail, there is very evidence little to implicate him. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to make the Asterisk-GUIworkwithDAHDI..please??
Probably not a Real term (apologies especially to the foreign language
readers)
In * 1.4.22* there is a Zaptel-to-DAHDI.txt file; here is the excerpt I
derived my term from
However, in spite of the file name changes, the channels and
applications provided by these modules can still be used with 'Zap'
style names; see below for more information.
Second, there are have been a number of efforts made to ensure that
existing systems will not have to have any major configuration changes
made solely because Asterisk was built against DAHDI instead of
Zaptel. This includes:
chan_dahdi.so:
This module will determine which channel name ('Zap' or 'DAHDI')
should be used for incoming and outgoing channels based on the
build-time choice of telephony drivers. However, if you wish to
continue using the 'Zap' channel name even though you built Asterisk
against the DAHDI drivers, you can add the following line to the
[options] section of your /etc/asterisk/asterisk.conf file:
dahdichanname = no
All CLI commands that begin with 'zap' are now available as 'dahdi'
commands as well; the 'zap' variants will report that they are
deprecated the first time you use each one in an Asterisk instance
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovid Bender
Sent: Wednesday, February 11, 2009 11:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to make the
Asterisk-GUIworkwithDAHDI..please??
What is Zap mirroring ?
- Original Message -
From: Danny Nicholas da...@debsinc.com
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Monday, February 09, 2009 10:09 PM
Subject: Re: [asterisk-users] How to make the Asterisk-GUI
workwithDAHDI..please??
Assuming you're still in the 1.4 set, enable the Zap mirroring. I'm not
sure the fine folks at Digium have accounted for DAHDI in the current
interfaces (but I'm probably wrong).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Enrique
Sent: Monday, February 09, 2009 12:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to make the Asterisk-GUI work
withDAHDI..please??
How to make the Asterisk-GUI work with DAHDI..please??
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Re: [asterisk-users] Hangup extensions via CLI?
asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep SIP/7000 Hi, I used this way of processing output from asterisk 1.2 and found out that it is not 100% safe because there can appear unprintable characters in the output. This will cause the following grep command to show message similar to Binary content: matched instead of expected line. It is necessary to use strings -a to filter output. So your example should be: asterisk -rx 'core show channels' | strings -a | grep SIP/7000 Hope it helps Lukas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WiFi SIP phone w/VPN?
Hi, all. My subject line says it all: is there a WiFi SIP phone with VPN abilities? Failing that, a WiFi phone that runs Linux? I already know one phone that does meet my requirements -- the iPhone. The new software comes with a Cisco VPN client, and a SIP client can be had from third-party vendors for jailbroken phones. And, while I'm not averse to the idea, a) it ain't cheap, and b) it's a bit hack. I've googled my heart out, but haven't found anything else that (I'm sure) meets all three requirements. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with parking
Update on the matter:
I have reduced the config to one context and two peers:
extensions.conf:
[internal]
include = parkedcalls
exten = 2552,1,Dial(SIP/2552,,t)
exten = 2556,1,Dial(SIP/2556,,t)
sip.conf:
[general]
context=deadend
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
[2552]
type=friend
context=dialplan-1
host=dynamic
call-limit=10
defaultuser=2552
secret=xxx
[2556]
type=friend
context=dialplan-1
host=dynamic
call-limit=10
defaultuser=2556
secret=xxx
features.conf:
[general]
parkext = 10
parkpos = 11-14
parkingtime = 45
[featuremap]
blindxfer = #
That's all. I am starting to think that this must be an asterisk bug...
version is 1.6.0.1.
Regards,
Örn
On Thu, Feb 12, 2009 at 3:05 PM, Örn Arnarson o...@arnarson.net wrote:
Hi,
I'm having problem with call parking.
When I park call, either via transfer to xten or park digit sequence from
features.conf, I hear the parking lot number read to me and the user gets
transferred.
However, MOH stops for the caller the moment user is transferred.
The user can be retrieved by dialing the parked extension and voice
resumes.
If the parked user hangs up, the channel state does not update (nor the
hint) and call seems to be live still.
If the timeout for the park is reached, the user is not transferred back
(nothing happens actually -- he is just able to stay parked forever).
Even after the user hangs up, the parking lot extension exists and is
callable, but when that is done (the parking extension is called), it
answers and promptly hangs up and then is available for parking again.
Yesterday I had everything working like a charm, and I don't think I have
changed anything, although that seems increasingly unlikely since things
don't usually break on their own. I've restarted asterisk numerous times and
tried changing things I think are relevant, but with no avail.
It does not matter whether the caller is coming in from a SIP trunk or a
SIP peer.
The SIP messages look normal. When the caller hangs up while parked he
sends a BYE to the asterisk to the callee's number, and the asterisk replies
with ACK.
Any help would be appreciated.
My comments in asterisk CLI output below prefixed by #
Configuration files
== features.conf ==
[general]
parkext = 10
parkpos = 11-14
parkinghints = yes
== sip.conf ==
[general]
context=deadend
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
subscribecontext=parkedcalls
allowsubscribe=yes
notifyringing=yes
notifyhold=yes
limitonpeers=yes
canreinvite=no
[Tal]
context=incoming
type=friend
host=XXX.XXX.XXX.XXX
canreinvite=no
port=5060
dtmfmode=rfc2833
disallow=all
allow=alaw
[smg01]
type=friend
context=incoming
host=XXX.XXX.XXX.XXX
canreinvite=no
port=5060
dtmfmode=rfc2833
disallow=all
allow=alaw
[2552]
type=friend
context=dialplan-1
host=dynamic
call-limit=10
defaultuser=2552
secret=xxx
[2556]
type=friend
context=dialplan-1
host=dynamic
call-limit=10
defaultuser=2556
secret=xxx
== extensions.conf ==
[globals]
CID = 5822550
OCODE = 582
XTENS = 255X
DTRK = smg01
CONF = 2559
ADALNUMER = 2550
ADALDIAL = SIP/2551SIP/2552SIP/2553SIP/2554
BAKVAKT = 7712555
[general]
static=yes
writeprotect=yes
clearglobalvars=no
userscontext=default
[dialplan-1]
include = conferences
include = ringgroups
include = internal
include = landlines
include = gsm
include = special
include = international
include = parkedcalls
[dialplan-2]
include = conferences
include = ringgroups
include = internal
include = landlines
include = parkedcalls
[dialplan-3]
include = conferences
include = ringgroups
include = internal
include = landlines
include = gsm
include = parkedcalls
[dialplan-4]
include = conferences
include = ringgroups
include = internal
include = landlines
include = international
include = parkedcalls
[incoming]
exten = 4891001,1,Set(CALLERID(name)=SMG01)
exten = 4891001,n,Dial(${ADALDIAL})
exten =
${CID},1,Macro(ringgroup,${ADALDIAL},ADALNUMER,${ADALNUMER},ringgroup)
exten = ${OCODE}${CONf},1,Goto(conferences,${EXTEN:3},1)
exten = _${OCODE}${XTENS},1,Set(CALLERID(name)=BEIN HRINGING)
exten = _${OCODE}${XTENS},n,Macro(internal-call,${EXTEN:3},incoming)
[internal]
exten = _${XTENS},1,Macro(internal-call,${EXTEN},internal)
exten = *80,1,Page(SIP/2553)
exten = *97,1,VoiceMailMain(${CALLERID(num)},s)
exten = *98,1,Playback(templokun)
exten = *99,1,Record(templokun.wav)
exten = _9X.,1,Macro(trunkdial,Tal,${EXTEN:1},${CALLERID(num)},landline)
[landlines]
exten =
_800,1,Macro(trunkdial,${DTRK},${EXTEN},${CALLERID(num)},tollfree)
exten =
_[4-5]XX,1,Macro(trunkdial,${DTRK},${EXTEN},${CALLERID(num)},landline)
exten =
_177X,1,Macro(trunkdial,${DTRK},${EXTEN},${CALLERID(num)},landline)
exten = 1817,1,Macro(trunkdial,${DTRK},${EXTEN},${CALLERID(num)},landline)
exten = 1414,1,Macro(trunkdial,${DTRK},${EXTEN},${CALLERID(num)},landline)
[asterisk-users] OSLEC not being loaded on Ubuntu Intrepid
I wonder if anyone has any ideas on this.
I have recently migrated my server from a custom built Linux with
Asterisk, Zaptel and Oslec running fine, to Ubuntu Server 8.10.
I have Asterisk installed via synaptic at it works fine.
I have built and installed the zaptel package by doing the following
commands:
sudo m-a -t build zaptel
cd /usr/src
sudo dpkg -i zaptel-modules-{version}.deb
sudo modprobe zaptel
sudo modprobe wcfxo
But when the wcfxo module is loaded, it is not loading the oslec module.
There is an oslec.ko in /lib/modules/my-running-kernel-version/misc/oslec/
According to launchpad, oslec should be the default ec now for zaptel.
Anyone got any ideas please?
Thanks
Alan
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Re: [asterisk-users] What do you use? .conf or AEL?
On Thu, 2009-02-12 at 08:20 +0100, Louis-David Mitterrand wrote: On Tue, Feb 10, 2009 at 01:56:16PM -0800, Mik Cheez wrote: I use them both; my legacy dialplan is all .conf and new stuff is .ael. I find AEL to be the better option when jumping around, but that's just my opinion. But isn't AEL just converted into .conf language anyway? Or has this evolved with 1.4.x ? AEL is converted to the same internal representation inside of asterisk that traditional .conf diaplan is converted to. -- Matthew Nicholson Digium, Inc. | Software Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuring Patton SmartNode with ISDN2e and Asterisk
2009/2/2 Phil Knighton phil.knigh...@mjog.com Hello Does anyone have any experience with configuring BT (British Telecom) ISDN2e lines to work with Patton SmartNodes - and then Asterisk? I have a Patton SmartNode 4638, which is now connected to 3 x ISDN2e lines - and in turn connected to our internal LAN. I'm having huge issues configuring the SmartNode to successfully see the ISDN channels - and to be honest, I'm lost as to how to then route those calls to Asterisk? The instructions, user guide and website all assume a knowledge of the technology/terminology that I just don't have :-( Has anyone had to do anything similar, and if so would you be able to provide some guidance in plain English? A kind list member has given me the outline steps of which bits need configuring, but I'm hoping someone may have a little more info on the actual settings for ISDN2e - as the Patton website (or BT) is not hugely helpful. Any further information on configuring Patton or other gateway boxes with Asterisk would also be extremely helpful. Thanks in advance Hi, Did overcome the issues you met when setting this Patton ? I can't speak ISDN2e as I'm not living in the UK but I would happy to help if you think it could be relevant. Using Patton embeded web server, what does : (hostname of your patton box)/Ports/BRI shows ? How many interfaces are enabled ? For each enabled BRI Port, have you declared an ISDN interface (can be checked reading Call-router ISDN interfaces panel) ? Cheers Phil ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF tones mid conversation
Hi Francois, I am using the latest *, dahdi/zaptel and libpri (1.4-current). This happens with both Zaptel and Dahdi and various versions of * (1.4.22.1 and 1.4.23). So, even the latest 'stable' would seem to have a problem. Cheers Andy -- -Original Message- -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- -- boun...@lists.digium.com] On Behalf Of F6HQZ -- Sent: 11 February 2009 16:49 -- To: Asterisk Users Mailing List - Non-Commercial Discussion -- Subject: Re: [asterisk-users] DTMF tones mid conversation -- -- Hi men, -- -- Resolved for one of my customers by upgrading Asterisk/Libpri/Zaptel. -- I don't remember what wer the versions, sorry. -- Check and advise us the results, please. -- -- Best Regards, -- Francois -- -- No virus found in this outgoing message. -- Checked by AVG - www.avg.com -- Version: 8.0.233 / Virus Database: 270.10.19/1941 - Release Date: -- 02/09/09 06:50:00 -- -- -- ___ -- -- Bandwidth and Colocation Provided by http://www.api-digital.com -- -- -- asterisk-users mailing list -- To UNSUBSCRIBE or update options visit: -- http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Caller ID replacement
I'm working on building a pbx that will allow us to use our cellphones as extensions (to some extent) The dialout is working fine. What I would like to do is have an inbound cellphone call appear as if it were an extension. So right now if I call in from cell #9995551212 the caller id is 9995551212 but if I dial extension 30013 it will call cell #9995551212. I would like to change the caller id so 9995551212 is changed to 30013 on the inbound call. Doing one is simple enough, but I would like have an easy (more or less) way of setting up some global variables that link the cell phone #'s and extensions and have this done somewhat automagically. Any suggestions? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID replacement
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 David Ruggles wrote: global variables that link the cell phone #'s and extensions and have this done somewhat automagically. Load your cross-reference in AstDB and do the lookup that way. If the cell number exists in the database, replace the callerID with the extension number. If it doesn't exist then it must be from someone else so don't change the callerId. BK -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJlFGwCFu3bIiwtTARAlRSAJ48FS53xS4u0eIeJ63VrZulPZxMMQCffFHw 7riqdRkR6vq5tGT9Z78FpiQ= =SuKH -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID replacement
On Thu, 2009-02-12 at 11:27 -0500, David Ruggles wrote: I'm working on building a pbx that will allow us to use our cellphones as extensions (to some extent) The dialout is working fine. What I would like to do is have an inbound cellphone call appear as if it were an extension. So right now if I call in from cell #9995551212 the caller id is 9995551212 but if I dial extension 30013 it will call cell #9995551212. I would like to change the caller id so 9995551212 is changed to 30013 on the inbound call. Doing one is simple enough, but I would like have an easy (more or less) way of setting up some global variables that link the cell phone #'s and extensions and have this done somewhat automagically. I would implement this using the a database (astdb or odbc) containing the mapping from cell number to extension. Then for each call that may need callerid modification, you can check the database for the proper mapping. With this method it is also easy to add new mappings. -- Matthew Nicholson Digium, Inc. | Software Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Queue and URL Calling
Dear All
I want to integrate sugarcrm and asterisk , so when customer call the call
center the agent or extension which answers the call , before pickup the
phone and talk to customer , view his/her information if it is available.
I do this as described below :
1-Setup login username for sugarcrm for each extension
2-Extension Users will login to the sugarcrm.
3-Develop php script to handle new call ( this script will execute from
asterisk )
4-The path to the script is like this :
http://sugarcrmIP/popup.php?number=CALLERIDextension=EXT
i.e. consider the extension number is 105 and the callerid is 7891234.
so these numbers will replace the CALLERID and EXT as follows
http://sugarcrmIP/popup?number=7891234extension=105
5-In asterisk in extensions.conf File I have edited the Macro named
std-exten ( [macro-stdexten] )
Just before Dial I put the following :
[macro-stdexten]
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten = s,1,Gotoif($[{LEN(${CALLERID(num)})} 3]?3)
exten = s,2,Goto(s,4)
exten = s,3,TrySystem(wget -qb -O /dev/null -o /dev/null
http://sugarcrmIP/popup.php?number=${CALLERID(num)}extension=${ARG1})
exten = s,4,Dial(${ARG2},35,rt) ; Ring the interface, 20 seconds maximum
exten = s,5,Goto(s-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send to
voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to start
;exten = s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to voicemail w/
busy announce
;exten = s-BUSY,2,Goto(default,s,1) ; If they press #, return to start
exten = s-BUSY,1,Playtones(busy)
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
exten = a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into
VoicemailMain
so when calls come in to asterisk before execute the dial in the macro it
will execute the
TrySystem(wget -qb -O /dev/null -o /dev/null
http://sugarcrmIP/popup.php?number=${CALLERID(num)}extension=${ARG1})
and replace ${CALLERID(num)} with the callerid of customer and ${ARG1} with
user extension
as I stated in step 4 as an example the following will execute
TrySystem(wget -qb -O /dev/null -o /dev/null
http://sugarcrmIP/popup.php?number=7891234extension=105;)
and this will cause a screen popup on the user screen which is logged in the
sugarcrm and his/her extension number is 105.
Everything work fine with this implementation but when it comes to Queue it
fails , cause in the Queue the [macro-stdexten] will not be executed for
dialing.
I have a Queue ( huntgroup ) which all the extensions are its member and
with random algorithm the calls will be distributed to extensions.
I need the popup but don't know how to execute the URL with the replacement
of the ${CALLERID(num)} and ${ARG1} with value.
Please someone help me.
Best Regards,
Mohsen
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Re: [asterisk-users] OSLEC not being loaded on Ubuntu Intrepid [SOLVED]
Alan Lord (News) wrote:
I wonder if anyone has any ideas on this.
I have recently migrated my server from a custom built Linux with
Asterisk, Zaptel and Oslec running fine, to Ubuntu Server 8.10.
I have Asterisk installed via synaptic at it works fine.
I have built and installed the zaptel package by doing the following
commands:
sudo m-a -t build zaptel
cd /usr/src
sudo dpkg -i zaptel-modules-{version}.deb
sudo modprobe zaptel
sudo modprobe wcfxo
But when the wcfxo module is loaded, it is not loading the oslec module.
There is an oslec.ko in /lib/modules/my-running-kernel-version/misc/oslec/
According to launchpad, oslec should be the default ec now for zaptel.
Answering my own question - the file zconfig.h is still declaring MG2 as
the default ec.
I edited this file to define OSLEC instead, zipped up the archive and
then rebuilt the zaptel module as above. Now works.
Maybe this will help someone else too.
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Re: [asterisk-users] Caller ID replacement
Could you give me an example of how this would look in the dialplan? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Nicholson Sent: Thursday, February 12, 2009 11:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Caller ID replacement On Thu, 2009-02-12 at 11:27 -0500, David Ruggles wrote: I'm working on building a pbx that will allow us to use our cellphones as extensions (to some extent) The dialout is working fine. What I would like to do is have an inbound cellphone call appear as if it were an extension. So right now if I call in from cell #9995551212 the caller id is 9995551212 but if I dial extension 30013 it will call cell #9995551212. I would like to change the caller id so 9995551212 is changed to 30013 on the inbound call. Doing one is simple enough, but I would like have an easy (more or less) way of setting up some global variables that link the cell phone #'s and extensions and have this done somewhat automagically. I would implement this using the a database (astdb or odbc) containing the mapping from cell number to extension. Then for each call that may need callerid modification, you can check the database for the proper mapping. With this method it is also easy to add new mappings. -- Matthew Nicholson Digium, Inc. | Software Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.233 / Virus Database: 270.10.18/1936 - Release Date: 02/05/09 11:34:00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disabling Echo Cancellation on a per Call basis
Hi Kevin, Kevin P. Fleming schrieb: Tobias Wolf wrote: If it is not possible, why is that so ? Is there really no need to do this and i am totally mistaken? This is generally true. Any standards-compliant FAX machine or modem will generate a CED tone during the beginning of the call process, and any standards compliant echo canceler (including the ones in Zaptel/DAHDI) will respond to this tone by disabling the echo canceler. With Zaptel/DAHDI and a software echo canceler, you can see the evidence of this by watching the kernel message log for messages of the form 'Disabled echo canceler because of tone (..) on channel Does this only take place if fax detection is enabled in DAHDI or is it something that happens everytime a CED tone is send over the line? Since i have only deticated fax lines, i like to get rid of the fax detection for that i need to add an Wait(4) to the dialplan, after Answer(). Unfortunatly my Linux Machine seems not to log messages from DAHDI. I have looked into /var/log/kern.log There are messages from the kernel module like; Feb 11 17:38:12 officepbx kernel: wcte1xxp: Setting yellow alarm Feb 11 17:38:21 officepbx kernel: wcte1xxp: Clearing yellow alarm But no messages about echo can. Maybe i am missing configuration for the kernel module to enable logging ? Generally receiving faxes works fine, but sometimes they break and i assume i might have something to do with echo can. Unfortunately that is not likely to be the cause of your problems. Well, this may be the case ... But thanks anyway for your helpful informations, they help me a lot to get a better understanding. Cheers, -- Tobias Wolf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID replacement
Some googling lead me to this: http://hans.fugal.net/blog/tag/astdb Which looks like it has an answer. Thanks all! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Ruggles Sent: Thursday, February 12, 2009 12:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Caller ID replacement Could you give me an example of how this would look in the dialplan? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Nicholson Sent: Thursday, February 12, 2009 11:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Caller ID replacement On Thu, 2009-02-12 at 11:27 -0500, David Ruggles wrote: I'm working on building a pbx that will allow us to use our cellphones as extensions (to some extent) The dialout is working fine. What I would like to do is have an inbound cellphone call appear as if it were an extension. So right now if I call in from cell #9995551212 the caller id is 9995551212 but if I dial extension 30013 it will call cell #9995551212. I would like to change the caller id so 9995551212 is changed to 30013 on the inbound call. Doing one is simple enough, but I would like have an easy (more or less) way of setting up some global variables that link the cell phone #'s and extensions and have this done somewhat automagically. I would implement this using the a database (astdb or odbc) containing the mapping from cell number to extension. Then for each call that may need callerid modification, you can check the database for the proper mapping. With this method it is also easy to add new mappings. -- Matthew Nicholson Digium, Inc. | Software Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.233 / Virus Database: 270.10.18/1936 - Release Date: 02/05/09 11:34:00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.233 / Virus Database: 270.10.18/1936 - Release Date: 02/05/09 11:34:00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] reinvite
On Mon, 9 Feb 2009, Jeff LaCoursiere wrote: I've never used reinvite in systems I have installed to date, and I have finally run across a situation where it would be preferred. A remote office has a flaky Internet connection. With G729 encoding the calls to the central office over the 'net are tolerable. One Linksys 2102 drives two phones at this location, and when the first one calls the second one it travels to the central office and back, which is no longer tolerable. For each sip peer I have canreinvite=yes, but I am a bit confused as to the correct options on the 2102 to use this feature. Is anyone doing this with 2102s that can give me some pointers? I have been playing around with this in my lab and cannot seem to make it work as expected. I have a remote asterisk server on a public IP - 1.4.22-3 on Centos 5. I have two Polycom IP501s on a local LAN behind a NAT gateway. Both Polycom's register with the remote server and can call each other without issues. Both SIP contexts have nat=yes, canreinvite=yes. The caller is 223, the callee is 222. eth0 is the outside (public) interface, XXX is my dynamic IP. I trapped a conversation on the asterisk server with: tcpdump -nli eth0 -s 0 -w /tmp/reinvite.debug host XXX and not port 22 While this was running I made a call between the two extensions for a few seconds then hungup. I opened this capture in etherreal and can see the following: 223-AST INVITE 2...@ast AST-223 407 Proxy auth required 223-AST ACK 223-AST INVITE 2...@ast, with proxy-auth info AST-223 100 Trying AST-223 200 OK 223-AST ACK Then I see the RTP traffic begin back and forth. I am confused on two fronts - first where is the INVITE from AST to 222? Not sure how I missed capturing that side of the conversation. And of course where is the AST reinvite? It isn't occurring since I can clearly see the RTP traffic flowing via the asterisk server. Any ideas? Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] reinvite
Jeff LaCoursiere wrote: On Mon, 9 Feb 2009, Jeff LaCoursiere wrote: I've never used reinvite in systems I have installed to date, and I have finally run across a situation where it would be preferred. A remote office has a flaky Internet connection. With G729 encoding the calls to the central office over the 'net are tolerable. One Linksys 2102 drives two phones at this location, and when the first one calls the second one it travels to the central office and back, which is no longer tolerable. For each sip peer I have canreinvite=yes, but I am a bit confused as to the correct options on the 2102 to use this feature. Is anyone doing this with 2102s that can give me some pointers? I have been playing around with this in my lab and cannot seem to make it work as expected. I have a remote asterisk server on a public IP - 1.4.22-3 on Centos 5. I have two Polycom IP501s on a local LAN behind a NAT gateway. Both Polycom's register with the remote server and can call each other without issues. Both SIP contexts have nat=yes, canreinvite=yes. The caller is 223, the callee is 222. eth0 is the outside (public) interface, XXX is my dynamic IP. I trapped a conversation on the asterisk server with: tcpdump -nli eth0 -s 0 -w /tmp/reinvite.debug host XXX and not port 22 While this was running I made a call between the two extensions for a few seconds then hungup. I opened this capture in etherreal and can see the following: 223-AST INVITE 2...@ast AST-223 407 Proxy auth required 223-AST ACK 223-AST INVITE 2...@ast, with proxy-auth info AST-223 100 Trying AST-223 200 OK 223-AST ACK Then I see the RTP traffic begin back and forth. I am confused on two fronts - first where is the INVITE from AST to 222? Not sure how I missed capturing that side of the conversation. And of course where is the AST reinvite? It isn't occurring since I can clearly see the RTP traffic flowing via the asterisk server. Any ideas? Cheers, j Asterisk may not be sending reinvites to the phones due to options you have passed to the Dial application. If Asterisk needs to intercept DTMF for a feature, then Asterisk will not send reinvites to the endpoints to redirect the media. For instance, if you have the 't' or 'T' options enabled in your Dial application, then Asterisk will not send reinvites to the endpoints even if you have configured chan_sip to allow reinvites to be sent. Other factors which can contribute are use of applications like Monitor and MixMonitor which require the media to go through Asterisk. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OSLEC not being loaded on Ubuntu Intrepid
On Thu, Feb 12, 2009 at 03:59:26PM +, Alan Lord (News) wrote:
I wonder if anyone has any ideas on this.
I have recently migrated my server from a custom built Linux with
Asterisk, Zaptel and Oslec running fine, to Ubuntu Server 8.10.
I have Asterisk installed via synaptic at it works fine.
I have built and installed the zaptel package by doing the following
commands:
sudo m-a -t build zaptel
cd /usr/src
sudo dpkg -i zaptel-modules-{version}.deb
sudo modprobe zaptel
sudo modprobe wcfxo
But when the wcfxo module is loaded, it is not loading the oslec module.
There is an oslec.ko in /lib/modules/my-running-kernel-version/misc/oslec/
According to launchpad, oslec should be the default ec now for zaptel.
Anyone got any ideas please?
http://bugs.debian.org/510858
Fixed in SVN: http://svn.debian.org/viewsvn/pkg-voip?rev=6684view=rev
As mentioned there, the workaround is to set ECHO_CANC_NAME explicitly:
ECHO_CAN_NAME=oslec m-a a-i zaptel
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[asterisk-users] 1.6.1-rc1 errors
I am getting the following warnings on the CLI when loading Asterisk 1.6.1-rc1: [Feb 12 12:32:34] NOTICE[22261]: timing.c:59 ast_install_timing_functions: Multiple timing modules are loaded. You should only load one. [Feb 12 12:32:34] ERROR[22261]: codec_dahdi.c:398 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory [Feb 12 12:32:33] WARNING[22261]: chan_iax2.c:10939 build_user: Unable to support trunking on user 'telecomab' without DAHDI timing [Feb 12 12:32:33] WARNING[22261]: chan_iax2.c:10679 build_peer: Unable to support trunking on peer 'telecomab' without a timing interface I am using DAHDI 2.1.0.4, Asterisk 1.6.1-rc1 on a CentOS 5.2 machine with a TDM04 card. These are the modules: Module Size Used by dahdi_echocan_mg2 9608 0 wctdm 39884 4 dahdi 190728 2 dahdi_echocan_mg2,wctdm Where do I have to specify the timing module? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disabling Echo Cancellation on a per Call basis
Tobias Wolf wrote: Does this only take place if fax detection is enabled in DAHDI or is it something that happens everytime a CED tone is send over the line? FAX detection is not done in DAHDI, it's done in chan_dahdi (in Asterisk). CED detection is done in the echo canceler itself, so it is completely independent of Asterisk (or any application, for that matter). Correctly responding to CED is something that an echo canceler must do just to be compliant with various specifications. Unfortunatly my Linux Machine seems not to log messages from DAHDI. I have looked into /var/log/kern.log There are messages from the kernel module like; Feb 11 17:38:12 officepbx kernel: wcte1xxp: Setting yellow alarm Feb 11 17:38:21 officepbx kernel: wcte1xxp: Clearing yellow alarm But no messages about echo can. Maybe i am missing configuration for the kernel module to enable logging ? It's logged as a LOG_NOTICE message and is always generated, unless DAHDI was built with NO_ECHOCAN_DISABLE defined, which would be uncommon. However, /var/log/kern.log on your system might be only from the boot process. Have you checked 'dmesg' as the system is running? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.1-rc1 errors
Carlos Chavez wrote: I am getting the following warnings on the CLI when loading Asterisk 1.6.1-rc1: [Feb 12 12:32:34] NOTICE[22261]: timing.c:59 ast_install_timing_functions: Multiple timing modules are loaded. You should only load one. [Feb 12 12:32:34] ERROR[22261]: codec_dahdi.c:398 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory [Feb 12 12:32:33] WARNING[22261]: chan_iax2.c:10939 build_user: Unable to support trunking on user 'telecomab' without DAHDI timing [Feb 12 12:32:33] WARNING[22261]: chan_iax2.c:10679 build_peer: Unable to support trunking on peer 'telecomab' without a timing interface I am using DAHDI 2.1.0.4, Asterisk 1.6.1-rc1 on a CentOS 5.2 machine with a TDM04 card. These are the modules: Module Size Used by dahdi_echocan_mg2 9608 0 wctdm 39884 4 dahdi 190728 2 dahdi_echocan_mg2,wctdm Where do I have to specify the timing module? Timing may be provided from one of two sources in Asterisk 1.6.1: res_timing_dahdi.so (get timing from DAHDI), and res_timing_pthread.so (use pthread library for timing). There are a couple of ways to fix your problem, assuming that the timing module you want to use is res_timing_dahdi.so. 1) Remove res_timing_pthread.so from /usr/lib/asterisk/modules and restart Asterisk 2) In modules.conf, add noload = res_timing_pthread.so 3) While not a requirement, you can also make menuselect and disable res_timing_pthread.so from being built at all. The module can be found under the Resource Modules menu. It looks as though the timing modules for 1.6.1 are not well-documented, and Menuselect should be altered to not allow for both modules to be built. We'll get to work getting this documented better. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype beta news ?
I am curious as to if there are any updates on this? Olivier wrote: Hi, Has anyone any return to share about Skype-Digium beta program ? I would be very curious to know how things are going on this. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disabling Echo Cancellation on a per Call basis
Kevin P. Fleming schrieb: Tobias Wolf wrote: Does this only take place if fax detection is enabled in DAHDI or is it something that happens everytime a CED tone is send over the line? FAX detection is not done in DAHDI, it's done in chan_dahdi (in Asterisk). CED detection is done in the echo canceler itself, so it is completely independent of Asterisk (or any application, for that matter). Correctly responding to CED is something that an echo canceler must do just to be compliant with various specifications. Alright, understood. Unfortunatly my Linux Machine seems not to log messages from DAHDI. I have looked into /var/log/kern.log There are messages from the kernel module like; Feb 11 17:38:12 officepbx kernel: wcte1xxp: Setting yellow alarm Feb 11 17:38:21 officepbx kernel: wcte1xxp: Clearing yellow alarm But no messages about echo can. Maybe i am missing configuration for the kernel module to enable logging ? It's logged as a LOG_NOTICE message and is always generated, unless DAHDI was built with NO_ECHOCAN_DISABLE defined, which would be uncommon. However, /var/log/kern.log on your system might be only from the boot process. Have you checked 'dmesg' as the system is running? Have done that, no messages appear while receiving a fax. I know that i have seen such messages before the upgrade. I have another asterisk server which is bristuffed, and there i can see this message with 'dmesg'. I am absolutly positive that DAHDI was not built with NO_ECHOCAN_DISABLE, since i have only downloaded the package and done a 'make; make install' without touching anything of the source code. By the way, i am using DAHDI-linux 2.1.0.3. Regards -- Tobias Wolf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WiFi SIP phone w/VPN?
Hi, all. My subject line says it all: is there a WiFi SIP phone with VPN abilities? Failing that, a WiFi phone that runs Linux? I already know one phone that does meet my requirements -- the iPhone. The new software comes with a Cisco VPN client, and a SIP client can be had from third-party vendors for jailbroken phones. And, while I'm not averse to the idea, a) it ain't cheap, and b) it's a bit hack. I've googled my heart out, but haven't found anything else that (I'm sure) meets all three requirements. The Nokia N810 internet tablet might fit your requirements. It runs Linux, much of the software kit is open-source, it has WiFi, it has a built-in SIP phone application, and it has an OpenVPN client available. The SIP phone app will support multiple SIP accounts. I use mine fairly regularly to connect with my home Asterisk server when in restaurants and stores that have WiFi access for their customers. The use of the OpenVPN connection makes life *much* simpler, as the VPN can successfully create a tunnel through most NAT routers, and doesn't require STUN support. I have two different account definitions on my N810 - one for my own Asterisk server (via the tunnel) and another which registers directly with my telco origination provider. The latter will establish a more direct connection when I dial out onto the PSTN (since the traffic doesn't go through my home-DSL line twice) but is somewhat less certain to work at any given wireless site (since it's dependent on STUN and on the settings of that site's firewall/router). Getting the OpenVPN/SIP setup working requires a bit of fiddling, as it's not straightforward: - You must add one or two additional Maemo software repositories to the Application Manager, - You must use the blue pill mode of the installer to add OpenVPN to the system (install the OpenSSH or Dropbear SSH client and server at the same time) - You must create your OpenVPN certs on your OpenVPN server and then download them to the N810 and install them in the right directories. - Accessing the Asterisk server via the OpenVPN tunnel requires changing the SIP-phone account definition via a shell command line tool, to force the SIP phone to use the tunnel's IP address rather than that of whatever WiFi connection you are using at the time. Fortunately, this can be done automagically when the tunnel starts up, via some up and down shell scripts... I can provide samples upon request. - If your OpenVPN tunnel doesn't terminate on the same machine that runs your Asterisk server, you may need a SIP proxy running on the tunnel-termination server. I wouldn't have bought the N810 for use solely as a WiFi phone, but having this feature added to an otherwise-very-useful lightweight Internet access device / GPS is extremely handy. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What t38pt_udptl is ? Explain T.38 in 1.4
2009/2/10 David Backeberg dbackeb...@gmail.com On Mon, Feb 9, 2009 at 4:23 PM, Olivier oza-4...@myamail.com wrote: Hi, I would like to improve my understanding of T.38. I recommend you try out Asterisk 1.6 if you want to play with T.38. I DID get asterisk-1.4 working with fax, but I was having a lot of issues with faxes dropping in weird ways. All of those issues went away when I upgraded to 1.6. Don't waste your time like I did. I can see what you mean, now, since I'm trying hard to understand how Asterisk 1.4 is working with T.38. At the moment, not only fax is not really working but I even get aborted voice calls because of 415 unsupported media ! It seems Asterisk keeps advertising T.38 support to callee endpoint though this endpoint is configured in sip.conf to accept only alaw !! Endpoint is configured with : t38pt_udptl=no t38pt_rtp=no t38pt_tcp=no disallow=all allow=alaw canreinvite=no So I must be misunderstanding what t38pt_udptl=no means. Maybe it simply means if t38pt_udptl=yes, then use this type of T.38, else use another type of T.38 and setting t38pt_udptl=no is not enough to prevent an endpoint to receive T.38 offers ... Or maybe it's a bug ... app_fax in asterisk-1.6 is very, very nice and worked great for me. It works with traditional analog fax as well as T.38. I have to say I'm impressed and grateful for all who contributed to this. asterisk-1.6 also has a better SIP stack and other improvements. Rather than trying to get 1.4 working with this, I strongly recommend reading the directions at: http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 to build asterisk-1.6 with app_fax support ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disabling Echo Cancellation on a per Call basis
Tobias Wolf wrote: I am absolutly positive that DAHDI was not built with NO_ECHOCAN_DISABLE, since i have only downloaded the package and done a 'make; make install' without touching anything of the source code. By the way, i am using DAHDI-linux 2.1.0.3. Can you post your /etc/dahdi/system.conf? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype beta news ?
On Thursday 12 February 2009 12:58:03 Casey Boone wrote: I am curious as to if there are any updates on this? Olivier wrote: Has anyone any return to share about Skype-Digium beta program ? I would be very curious to know how things are going on this. The beta is going well. I really can't say much more than that, as I am not more than tangentially involved with it. Stay tuned to the press releases, as I'm sure we'll have an announcement coming out in the next few months. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CISCO 2950 - 4 connections - Cap of 512 Kbps - How to bond ?
The ISP giving net access at our office has installed a 24 port CISCO 2950 switch in our server room. I can buy 24 connections from them and get 12Mbps of Upload but each individual connection is restricted to 512Kbps. Currently we have requirement of 20 simulataneous calls so we purchased 4 connections from the ISP. Giving us a total of 2 Mbps of upload b/w but spread over 4 different connections from the ISP. Each connection we buy from the ISP gives us the right to use one port on this CISCO 2950 switch. So curretly we have purchased 4 connections from the ISP and hence we have the right to use 4 ports on this switch. My three questions are: 1. Is there any technical reason behind why the ISP will not sell more then 512 Kbps of b/w on a single port to us ? 2. Can I do something to over come the restriction put by the ISP. 3. Is there an automated script that can load balance the asterisk calls across these 4 connections ? Thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple caller id ...
Julian Lyndon-Smith aster...@dotr.com writes: exten = foo,n,Dial(SIP/1234Zap/G1c/55443322) and SIP/5432 calls this extension, is it possible to show different callerid numbers to each of the target numbers ? No, but you can do Dial(Local/1...@sipcallsLocal/55443...@zapg1c), and then change callerid as appropriate in the [sipcalls] and [zapg1c] contexts. Naming can obviously be improved... /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Queue and URL Calling
Mohsen mohsen1...@gmail.com writes:
I have a Queue ( huntgroup ) which all the extensions are its member and
with random algorithm the calls will be distributed to extensions.
I need the popup but don't know how to execute the URL with the
replacement of the ${CALLERID(num)} and ${ARG1} with value.
You can do almost anything if you use Local/1...@somecontext instead
of SIP/1234 in your queues.
/Benny
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Re: [asterisk-users] Asterisk Queue and URL Calling
Mohsen,
Here is one way you can do this:
1. In queues.conf add setinterfacevar=yes
2. Pass an AGI script to the queue application which reads in the
MEMBERINTERFACE channel variable and makes the web request for you.
Hope this helps,
-Dave
Mohsen wrote:
Dear All
I want to integrate sugarcrm and asterisk , so when customer call the
call center the agent or extension which answers the call , before
pickup the phone and talk to customer , view his/her information if it
is available.
I do this as described below :
1-Setup login username for sugarcrm for each extension
2-Extension Users will login to the sugarcrm.
3-Develop php script to handle new call ( this script will execute
from asterisk )
4-The path to the script is like this :
http://sugarcrmIP/popup.php?number=CALLERIDextension=EXT
http://sugarcrmIP/popup.php?number=CALLERIDextension=EXT
i.e. consider the extension number is 105 and the callerid is 7891234.
so these numbers will replace the CALLERID and EXT as follows
http://sugarcrmIP/popup?number=7891234extension=105
http://sugarcrmIP/popup?number=7891234extension=105
5-In asterisk in extensions.conf File I have edited the Macro named
std-exten ( [macro-stdexten] )
Just before Dial I put the following :
[macro-stdexten]
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten = s,1,Gotoif($[{LEN(${CALLERID(num)})} 3]?3)
exten = s,2,Goto(s,4)
exten = s,3,TrySystem(wget -qb -O /dev/null -o /dev/null
http://sugarcrmIP/popup.php?number=${CALLERID(num)}extension=${ARG1}
http://sugarcrmIP/popup.php?number=$%7BCALLERID%28num%29%7Dextension=$%7BARG1%7D)
exten = s,4,Dial(${ARG2},35,rt) ; Ring the interface, 20 seconds maximum
exten = s,5,Goto(s-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send to
voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to
start
;exten = s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to voicemail
w/ busy announce
;exten = s-BUSY,2,Goto(default,s,1) ; If they press #, return to start
exten = s-BUSY,1,Playtones(busy)
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
exten = a,1,VoicemailMain(${ARG1}) ; If they press *, send the user
into VoicemailMain
so when calls come in to asterisk before execute the dial in the macro
it will execute the
TrySystem(wget -qb -O /dev/null -o /dev/null
http://sugarcrmIP/popup.php?number=${CALLERID(num)}extension=${ARG1}
http://sugarcrmIP/popup.php?number=$%7BCALLERID%28num%29%7Dextension=$%7BARG1%7D)
and replace ${CALLERID(num)} with the callerid of customer and
${ARG1} with user extension
as I stated in step 4 as an example the following will execute
TrySystem(wget -qb -O /dev/null -o /dev/null
http://sugarcrmIP/popup.php?number=7891234extension=105
http://sugarcrmIP/popup.php?number=7891234extension=105)
and this will cause a screen popup on the user screen which is logged
in the sugarcrm and his/her extension number is 105.
Everything work fine with this implementation but when it comes to
Queue it fails , cause in the Queue the [macro-stdexten] will not be
executed for dialing.
I have a Queue ( huntgroup ) which all the extensions are its member
and with random algorithm the calls will be distributed to extensions.
I need the popup but don't know how to execute the URL with the
replacement of the ${CALLERID(num)} and ${ARG1} with value.
Please someone help me.
Best Regards,
Mohsen
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Re: [asterisk-users] asterisk across a firewall
On Wed, Feb 11, 2009 at 1:56 PM, Gordon Henderson gordon+aster...@drogon.net wrote: On Wed, 11 Feb 2009, Erick Perez wrote: Excuse my ignorance but if i have an asterisk in a LAN, and i have users in their homes/internet (dozens), in order to correctly connect those users across my firewall, what is the technology that i need to buy, called? secure border gateway? session controller? secure gateway? the audiocodes site seems to have many names for the same thing...but i better ask here and learn before i make a big mistake. my customer has a dumb firewall (not SIP aware) that will not replace. he wants another box to do the magic. I have many customers like that, and working from home is gaining momenting where I live... So the scenario (if I interpret it correctly): Asterisk at HQ is behind a NAT firewall with remote users (who themselves may be behing a NAT firewall) HQ needs a static IP address on the outside and plenty of bandwidth. The dumb router at HQ needs to port-forward external port 5060 and 1-2 into the asterisk box (you can limit this range - see rtp.conf) Most dumb routers can port-forward. Asterisk needs to know it's LAN and extneral ip address - sip.conf, externip= and localnet= remote extensions need nat=yes in sip.conf and that's basically it. If the remote extensions are themselves behind a NAT firewall, then the easiest way to get them through it is by using a stun server - ether run your own, or use someone elses... Do not do any port-forwarding at the remote users sites. Yes, you can fiddle about with proxies, gateways, etc. but keep it simple to start with and I have many installations doing it this way and it just works. One day I'm sure I'll trip up, but until then... Pitfalls - the same with all VoIP - bandwidth, espeically outgoing b/w from HQ. Broken NAT gateways, and routers which have SIP ALGs built in which are also broken. (Turn them off!) Routers with broken SIP ALG are the biggest PITA to work round. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thank you all for the excellent responses. I will do some test here to decide on a method/technology to use. -- Erick Perez Cel +(507) 6675-5083 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What t38pt_udptl is ? Explain T.38 in 1.4
Olivier oza-4...@myamail.com writes: So I must be misunderstanding what t38pt_udptl=no means. Maybe it simply means if t38pt_udptl=yes, then use this type of T.38, else use another type of T.38 and setting t38pt_udptl=no is not enough to prevent an endpoint to receive T.38 offers ... There are two facts you need to bear in mind when doing T.38 in 1.4 1) Asterisk 1.4 won't transcode between T.38 and ulaw/Alaw 2) The two call legs negotiate codecs independently So what happens is that a call comes in from a T.38-capable device, and Asterisk negotiates T.38. The Dial ends up going to a non-T.38-capable device, but Asterisk won't transcode, so it sends out T.38 anyway. This problem makes T.38-support in Asterisk 1.4 fairly useless IMHO, but maybe you can find a workaround. Luckily it works brilliantly in 1.6. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WiFi SIP phone w/VPN?
Not in the form factor that you would expect. Can I ask why? Most modern VoFi phones support WPA2. Frank -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio Sent: Wednesday, February 11, 2009 5:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] WiFi SIP phone w/VPN? Hi, all. My subject line says it all: is there a WiFi SIP phone with VPN abilities? Failing that, a WiFi phone that runs Linux? I already know one phone that does meet my requirements -- the iPhone. The new software comes with a Cisco VPN client, and a SIP client can be had from third-party vendors for jailbroken phones. And, while I'm not averse to the idea, a) it ain't cheap, and b) it's a bit hack. I've googled my heart out, but haven't found anything else that (I'm sure) meets all three requirements. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue problem
Hi, I am using Asterisk 1.4.23.1. I have the following context: [queue_context] exten = s,1,Queue(test) exten = s,n,Verbose(1|test) exten = s,n,Voicemail(5...@test) exten = s,n,Hangup() After 60 seconds, the call always hangs up. No Verbose, no Voicemail, nothing shows up in the CLI. Just a hangup. Any ideas? Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CISCO 2950 - 4 connections - Cap of 512 Kbps - How to bond ?
On Thu, Feb 12, 2009 at 4:04 PM, Vikas topg...@gmail.com wrote: The ISP giving net access at our office has installed a 24 port CISCO 2950 switch in our server room. I can buy 24 connections from them and get 12Mbps of Upload but each individual connection is restricted to 512Kbps. My three questions are: 3. Is there an automated script that can load balance the asterisk calls across these 4 connections ? This is crazy. Just tell the ISP that you want the port rate limit on a single port to be 2M. -- Heath Roberts htrobe...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue problem
What's the verbose setting on your CLI? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Thursday, February 12, 2009 3:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Queue problem Hi, I am using Asterisk 1.4.23.1. I have the following context: [queue_context] exten = s,1,Queue(test) exten = s,n,Verbose(1|test) exten = s,n,Voicemail(5...@test) exten = s,n,Hangup() After 60 seconds, the call always hangs up. No Verbose, no Voicemail, nothing shows up in the CLI. Just a hangup. Any ideas? Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CISCO 2950 - 4 connections - Cap of 512 Kbps - How to bond ?
I have asked the ISP to rate limit a single port to 2M but my requests have got me no where, I would really appreciate any suggestions on what I can do at my end since I have given up hope of the ISP co-operating with me, Thanks, On Thu, Feb 12, 2009 at 3:36 PM, Heath Roberts htrobe...@gmail.com wrote: On Thu, Feb 12, 2009 at 4:04 PM, Vikas topg...@gmail.com wrote: The ISP giving net access at our office has installed a 24 port CISCO 2950 switch in our server room. I can buy 24 connections from them and get 12Mbps of Upload but each individual connection is restricted to 512Kbps. My three questions are: 3. Is there an automated script that can load balance the asterisk calls across these 4 connections ? This is crazy. Just tell the ISP that you want the port rate limit on a single port to be 2M. -- Heath Roberts htrobe...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CISCO 2950 - 4 connections - Cap of 512 Kbps - How to bond ?
On Thu, Feb 12, 2009 at 4:04 PM, Vikas topg...@gmail.com wrote: My three questions are: 1. Is there any technical reason behind why the ISP will not sell more then 512 Kbps of b/w on a single port to us ? Yes. Somebody programmed their equipment that way and didn't train anybody else on Cisco before they got a better job. A Cisco 2950 can do 100Mbps per port (or 1000Mbps if it's a 2950G), and while you can't send all of that upstream, you can send way more than 12Mbps upstream. 2. Can I do something to over come the restriction put by the ISP. Yep, lots of things, none of which are going to be as direct as telling them that you've found another ISP who will give you what you want, and either they can remain your ISP and rehire the guy who knows how to program Cisco gear, or you are terminating your contract. Unless you live truly in the middle of nowhere, you will be able to find somebody else who can provide your phone service. Also, twenty simultaneous connections sounds a lot like a traditional T1. Call your phone company and compare the price of getting a T1 versus what these clowns are charging you. Just because you have voip now, doesn't mean it's cheaper than POTS. Asterisk does a great job of acting as a T1 to voip gateway. You can even get appliances for that task. 3. Is there an automated script that can load balance the asterisk calls across these 4 connections ? It will be way easier to write your termination letter than to write that script. This is a human problem, and not an asterisk problem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CISCO 2950 - 4 connections - Cap of 512 Kbps - How to bond ?
Vikas topg...@gmail.com writes: My three questions are: 1. Is there any technical reason behind why the ISP will not sell more then 512 Kbps of b/w on a single port to us ? The copper to your location only handles 512kpbs per pair, so they add an extra modem every time they open a new port? 2. Can I do something to over come the restriction put by the ISP. Most likely not. 3. Is there an automated script that can load balance the asterisk calls across these 4 connections ? If you get a different IP address on each port, it's hard. You'd need a device which could do flow-based NAT, and that would only work for outbound calls. On the other hand, if the ISP cooperates, there are lots of options: a) Multi-pair SDSL modems b) Multi-link PPP c) Equal-cost multipath /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disabling Echo Cancellation on a per Call basis
Kevin P. Fleming schrieb: Tobias Wolf wrote: I am absolutly positive that DAHDI was not built with NO_ECHOCAN_DISABLE, since i have only downloaded the package and done a 'make; make install' without touching anything of the source code. By the way, i am using DAHDI-linux 2.1.0.3. Can you post your /etc/dahdi/system.conf? Sure, here you are: See Attachment ... # # DAHDI Configuration File # # This file is parsed by the DAHDI Configurator, dahdi_cfg # # Span Configuration # ^^ # First come the span definitions, in the format # # span=span num,timing source,line build out (LBO),framing,coding[,yellow] # # All T1/E1/BRI spans generate a clock signal on their transmit side. The # timing source parameter determines whether the clock signal from the far # end of the T1/E1/BRI is used as the master source of clock timing. If it is, our # own clock will synchronise to it. T1/E1/BRI connected directly or indirectly to # a PSTN provider (telco) should generally be the first choice to sync to. The # PSTN will never be a slave to you. You must be a slave to it. # # Choose 1 to make the equipment at the far end of the E1/T1/BRI link the preferred # source of the master clock. Choose 2 to make it the second choice for the master # clock, if the first choice port fails (the far end dies, a cable breaks, or # whatever). Choose 3 to make a port the third choice, and so on. If you have, say, # 2 ports connected to the PSTN, mark those as 1 and 2. The number used for each # port should be different. # # If you choose 0, the port will never be used as a source of timing. This is # appropriate when you know the far end should always be a slave to you. If # the port is connected to a channel bank, for example, you should always be # its master. Likewise, BRI TE ports should always be configured as a slave. # Any number of ports can be marked as 0. # # Incorrect timing sync may cause clicks/noise in the audio, poor quality or failed # faxes, unreliable modem operation, and is a general all round bad thing. # # The line build-out (or LBO) is an integer, from the following table: # # 0: 0 db (CSU) / 0-133 feet (DSX-1) # 1: 133-266 feet (DSX-1) # 2: 266-399 feet (DSX-1) # 3: 399-533 feet (DSX-1) # 4: 533-655 feet (DSX-1) # 5: -7.5db (CSU) # 6: -15db (CSU) # 7: -22.5db (CSU) # # If the span is a BRI port the line build-out is not used and should be set # to 0. # # framing:: # one of 'd4' or 'esf' for T1 or 'cas' or 'ccs' for E1. Use 'ccs' for BRI. # 'd4' could be referred to as 'sf' or 'superframe' # # coding:: # one of 'ami' or 'b8zs' for T1 or 'ami' or 'hdb3' for E1. Use 'ami' for # BRI. # # * For E1 there is the optional keyword 'crc4' to enable CRC4 checking. # * If the keyword 'yellow' follows, yellow alarm is transmitted when no # channels are open. # #span=1,0,0,esf,b8zs #span=2,1,0,esf,b8zs #span=3,0,0,ccs,hdb3,crc4 # # DOKOM span=1,1,0,ccs,hdb3,crc4,yellow bchan=1-15 dchan=16 bchan=17-31 # Dynamic Spans # ^ # Next come the dynamic span definitions, in the form: # # dynamic=driver,address,numchans,timing # # Where driver is the name of the driver (e.g. eth), address is the # driver specific address (like a MAC for eth), numchans is the number # of channels, and timing is a timing priority, like for a normal span. # use 0 to not use this as a timing source, or prioritize them as # primary, secondard, etc. Note that you MUST have a REAL DAHDI device # if you are not using external timing. # # dynamic=eth,eth0/00:02:b3:35:43:9c,24,0 # # If a non-zero timing value is used, as above, only the last span should # have the non-zero value. # # Channel Configuration # ^ # Next come the definitions for using the channels. The format is: # device=channel list # # Valid devices are: # # em:: # Channel(s) are signalled using EM signalling (specific # implementation, such as Immediate, Wink, or Feature Group D # are handled by the userspace library). # fxsls:: # Channel(s) are signalled using FXS Loopstart protocol. # fxsgs:: # Channel(s) are signalled using FXS Groundstart protocol. # fxsks:: # Channel(s) are signalled using FXS Koolstart protocol. # fxols:: # Channel(s) are signalled using FXO Loopstart protocol. # fxogs:: # Channel(s) are signalled using FXO Groundstart protocol. # fxoks:: # Channel(s) are signalled using FXO Koolstart protocol. # sf:: # Channel(s) are signalled using in-band single freq tone. # Syntax as follows: # # channel# = sf:rxfreq,rxbw,rxflag,txfreq,txlevel,txflag # # rxfreq is rx tone freq in Hz, rxbw is rx notch (and decode) # bandwith in hz (typically 10.0), rxflag is either 'normal' or # 'inverted', txfreq is tx tone freq in hz, txlevel is tx tone # level in dbm, txflag is either 'normal' or 'inverted'. Set # rxfreq or txfreq to 0.0 if that tone is not desired. # # unused:: # No signalling is performed, each channel in the list remains
Re: [asterisk-users] Disabling Echo Cancellation on a per Call basis
Tobias Wolf wrote: Sure, here you are: See Attachment ... Well, that looks perfectly normal. I'm not sure what to tell you, other than that your system might be configured (via klogd) to suppress NOTICE-level kernel messages after boot time or something like that. If you are comfortable editing code, you can find the 'Disabled echo...' line in drivers/dahdi/dahdi-base.c and change the KERN_NOTICE to KERN_WARNING or KERN_ERROR to see if that makes it appear. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CISCO 2950 - 4 connections - Cap of 512 Kbps - How to bond ?
Get a Cisco with five ethernet ports. Use one for your connection to asterisk. Use the other four as your connection to the ISP, and MUX them. Great way to spend 5K :) j On Thu, 12 Feb 2009, Vikas wrote: I have asked the ISP to rate limit a single port to 2M but my requests have got me no where, I would really appreciate any suggestions on what I can do at my end since I have given up hope of the ISP co-operating with me, Thanks, On Thu, Feb 12, 2009 at 3:36 PM, Heath Roberts htrobe...@gmail.com wrote: On Thu, Feb 12, 2009 at 4:04 PM, Vikas topg...@gmail.com wrote: The ISP giving net access at our office has installed a 24 port CISCO 2950 switch in our server room. I can buy 24 connections from them and get 12Mbps of Upload but each individual connection is restricted to 512Kbps. My three questions are: 3. Is there an automated script that can load balance the asterisk calls across these 4 connections ? This is crazy. Just tell the ISP that you want the port rate limit on a single port to be 2M. -- Heath Roberts htrobe...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CISCO 2950 - 4 connections - Cap of 512 Kbps - How to bond ?
On Fri, 13 Feb 2009 12:41:51 Jeff LaCoursiere wrote: Get a Cisco with five ethernet ports. Use one for your connection to asterisk. Use the other four as your connection to the ISP, and MUX them. Great way to spend 5K :) j On Thu, 12 Feb 2009, Vikas wrote: I have asked the ISP to rate limit a single port to 2M but my requests have got me no where, I would really appreciate any suggestions on what I can do at my end since I have given up hope of the ISP co-operating with me, As someone who works for an ISP, the best advice I can give you is to tell them where to go (*after* fully setting up and testing a new ISP that is). With the economies of the world tighter then usual at present, and ISP's a plenty, I can only suggest they are idiots or for some reason they don't want your business Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call picking and transfers
On Wed, 11 Feb 2009, Philipp Kempgen wrote: Jeff LaCoursiere schrieb: Working on some niche requests from one of my hotel clients. asterisk 1.4.20-1 on CentOS, Polycom 501s. The first request is for the Polycom's screen to show the CID of the inbound caller when a call pick is executed, so the picker knows if the call is internal or external. I have already worked around this issue by using ALERT info to give seperate ring tones for outside and inside, but they are used to their old Nortel switch which apparently showed the CID immediately after the pick, and they then knew how to answer the phone. The second is to show CID information on the screen when a call has been answered by the front desk, then a blind transfer sent to an internal phone. Today they simply see Front Desk and there is no indication of who the actual caller is to distinguish internal staff, internal guest room, or outside caller. Has anyone attacked these things with Polycom that might share their approach? These bugs cover the functionality you need I guess: http://bugs.digium.com/view.php?id=5014 http://bugs.digium.com/view.php?id=13827 http://bugs.digium.com/view.php?id=8824 However none of the patches are likely to be merged into 1.4. Thats quite a lot of reading - am still trying to digest it all. You are right - it does seem to be what I am looking for and several people have reported success with Polycom and the patches. It looks like as of the 10th of Feb gareth has posted patches against 1.4.23.1 . I will try to tackle this first... thanks for the tips! j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype beta news ?
On Thu, Feb 12, 2009 at 2:58 PM, Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: On Thursday 12 February 2009 12:58:03 Casey Boone wrote: I am curious as to if there are any updates on this? Olivier wrote: Has anyone any return to share about Skype-Digium beta program ? I would be very curious to know how things are going on this. The beta is going well. I really can't say much more than that, as I am not more than tangentially involved with it. Stay tuned to the press releases, as I'm sure we'll have an announcement coming out in the next few months. -- Tilghman Coming to a SwitchVox near you. Same SwitchVox time, same SwitchVox channel. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zaptel for asterisk
Hi every one... Any body can tell me that what kernel in CentOS is better for asterisk calls. Xen is worked with asterisk conference calling is that make halting problem? Thanks for your supports. -- This message has been scanned for viruses and dangerous content by Orbit Mail Server, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: Manager Interface Originate (ASYNC) - How to get the Originate Status
Dear All,
I am originating the call directly to the SIP Provider using the maganger
interface + originate (ASYNC) command. Here is the PHP-AGI Script.
$call = $asm-send_request('Originate',
array('Channel'=SIP/416...@abc/n,
'Context'='ORIG',
'Exten'='s',
* 'Async'='1',*
'MaxRetries' = '1',
'RetryTime' = '10',
'Priority'=1,
'Account'=$phonenumber,
'Callerid'=$callid)
*extensions.conf*
[ORIG]
exten = s,1,Answer
exten = s,2,Playback(ivrfile)
exten = s,n,Hangup
How Can I get the Originate Status using Async? ANSWER, BUSY, NOANSWER
etc..
Thank you.
Lloyd
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[asterisk-users] MINNESOTA: TwinCities Asterisk Users Group Meeting, this Saturday. Valentine's Day Feb/14/2009 11:30am
Hello Asterisk Users and those with an Interest in VoIP Tech, The agenda is open for our next meeting. I think we'll plan on an open discussion of anyone's choosing. If we're lacking a topic, we'll give a demo of installing fail2ban for your asterisk system. Bring your questions, ideas and projects and we will help you work through them. Jimmy John's is just a block away or we can order pizza. The meeting will begin at 11:30am on Saturday. There is a large parking lot in the rear of the building. Please note, the doors at the front of the property are not available for use, you must enter on the South side of the building (next to the parking area). Building is on the corner of Raymond Ave and University Ave. The address is 2356 University Ave West. Saint Paul Minnesota, 55114. We will meet in suite 401. We'll have another book to give away as a door pirze, Nagios second edition. A special thank you again to No Starch Press and O'Reilly Publishing. Hope you can make it! -Eric Osterberg Sound Choice Communications LLC ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MINNESOTA: TwinCities Asterisk Users Group Meeting, this Saturday. Valentine's Day Feb/14/2009 11:30am
On Thu, 12 Feb 2009 23:04:50 -0600 (CST), asterisk_h...@iwishi.nu wrote: Jimmy John's is just a block away or we can order pizza. The meeting will begin at 11:30am on Saturday. You folks have Jimmy John's up there? Word! -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Queue and URL Calling
You can do almost anything if you use Local/1...@somecontext instead of SIP/1234 in your queues. /Benny Thanks Benny I will check it. Best Regards, Mohsen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re : Asterisk Queue and URL Calling
Mohsen, Here is one way you can do this: 1. In queues.conf add setinterfacevar=yes 2. Pass an AGI script to the queue application which reads in the MEMBERINTERFACE channel variable and makes the web request for you. Hope this helps, -Dave Thanks Dave , I will check it. Best Regards, Mohsen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ExitIf() convention?
I want the first line of my dialplan to check and expression, and exit
from the dailplan if it is true - is there a convention for this?
My goal is to exit from the dialplan before calling Answer() if the
callerid is null. By this means I hope to work around this issue:
http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/217794
- I noticed that the callerid is never null for incomming calls - even
blocked numbers are PRIVATE, while on occasions when Asterisk
incorrectly answers during an in progress conversation, the callerid is
null.
Is it correct to use:
exten = s,1,GotoIf(${ISNULL(CALLERID())}?h)
- or is there a more commonly used convention?
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Re: [asterisk-users] Fwd: Manager Interface Originate (ASYNC) - How to get the Originate Status
2009/2/13 Aloysius Thevarajah Lloyd (SunTel Technologies)
lloyd.aloys...@sunteltech.ca
Dear All,
I am originating the call directly to the SIP Provider using the maganger
interface + originate (ASYNC) command. Here is the PHP-AGI Script.
$call = $asm-send_request('Originate',
array('Channel'=SIP/416...@abc/n,
'Context'='ORIG',
'Exten'='s',
* 'Async'='1',*
'MaxRetries' = '1',
'RetryTime' = '10',
'Priority'=1,
'Account'=$phonenumber,
'Callerid'=$callid)
*extensions.conf*
[ORIG]
exten = s,1,Answer
exten = s,2,Playback(ivrfile)
exten = s,n,Hangup
How Can I get the Originate Status using Async? ANSWER, BUSY, NOANSWER
etc..
Thank you.
Lloyd
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Hi.,
You can use 'failed' extension on the ORIG .
--
Pagarbiai / Best Regards,
Giedrius Augys
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Re: [asterisk-users] ExitIf() convention?
On Thursday 12 February 2009 23:35:42 Jack Bates wrote:
I want the first line of my dialplan to check and expression, and exit
from the dailplan if it is true - is there a convention for this?
My goal is to exit from the dialplan before calling Answer() if the
callerid is null. By this means I hope to work around this issue:
http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/217794
- I noticed that the callerid is never null for incomming calls - even
blocked numbers are PRIVATE, while on occasions when Asterisk
incorrectly answers during an in progress conversation, the callerid is
null.
Is it correct to use:
exten = s,1,GotoIf(${ISNULL(CALLERID())}?h)
This syntax will universally fail.
- or is there a more commonly used convention?
exten = s,1,ExecIf($[${CALLERID(num)}=],Hangup)
--
Tilghman
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Re: [asterisk-users] Fwd: Manager Interface Originate (ASYNC) - How to get the Originate Status
Can you explain what do you mean failed extension ?
Regards
Lloyd
On Fri, Feb 13, 2009 at 2:00 AM, Giedrius Augys voi...@gmail.com wrote:
2009/2/13 Aloysius Thevarajah Lloyd (SunTel Technologies)
lloyd.aloys...@sunteltech.ca
Dear All,
I am originating the call directly to the SIP Provider using the maganger
interface + originate (ASYNC) command. Here is the PHP-AGI Script.
$call = $asm-send_request('Originate',
array('Channel'=SIP/416...@abc/n,
'Context'='ORIG',
'Exten'='s',
* 'Async'='1',*
'MaxRetries' = '1',
'RetryTime' = '10',
'Priority'=1,
'Account'=$phonenumber,
'Callerid'=$callid)
*extensions.conf*
[ORIG]
exten = s,1,Answer
exten = s,2,Playback(ivrfile)
exten = s,n,Hangup
How Can I get the Originate Status using Async? ANSWER, BUSY, NOANSWER
etc..
Thank you.
Lloyd
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Hi.,
You can use 'failed' extension on the ORIG .
--
Pagarbiai / Best Regards,
Giedrius Augys
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Re: [asterisk-users] Fwd: Manager Interface Originate (ASYNC) - How to get the Originate Status
You mean
exten = failed,
Regards,
Lloyd
On Fri, Feb 13, 2009 at 2:00 AM, Giedrius Augys voi...@gmail.com wrote:
2009/2/13 Aloysius Thevarajah Lloyd (SunTel Technologies)
lloyd.aloys...@sunteltech.ca
Dear All,
I am originating the call directly to the SIP Provider using the maganger
interface + originate (ASYNC) command. Here is the PHP-AGI Script.
$call = $asm-send_request('Originate',
array('Channel'=SIP/416...@abc/n,
'Context'='ORIG',
'Exten'='s',
* 'Async'='1',*
'MaxRetries' = '1',
'RetryTime' = '10',
'Priority'=1,
'Account'=$phonenumber,
'Callerid'=$callid)
*extensions.conf*
[ORIG]
exten = s,1,Answer
exten = s,2,Playback(ivrfile)
exten = s,n,Hangup
How Can I get the Originate Status using Async? ANSWER, BUSY, NOANSWER
etc..
Thank you.
Lloyd
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Hi.,
You can use 'failed' extension on the ORIG .
--
Pagarbiai / Best Regards,
Giedrius Augys
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Re: [asterisk-users] Asterisk Queue and URL Calling
If you use the free version of QueueMetrics, you can have the queue URL
parameter passed along and each agent can open up an external app using the
web interface. As this part is not linked to the main stats module, it works
just fine for all of your agents with no limitations.
Thanks
l.
2009/2/12 Mohsen mohsen1...@gmail.com
Dear All
I want to integrate sugarcrm and asterisk , so when customer call the call
center the agent or extension which answers the call , before pickup the
phone and talk to customer , view his/her information if it is available.
I do this as described below :
1-Setup login username for sugarcrm for each extension
2-Extension Users will login to the sugarcrm.
3-Develop php script to handle new call ( this script will execute from
asterisk )
4-The path to the script is like this :
http://sugarcrmIP/popup.php?number=CALLERIDextension=EXT
i.e. consider the extension number is 105 and the callerid is 7891234.
so these numbers will replace the CALLERID and EXT as follows
http://sugarcrmIP/popup?number=7891234extension=105
5-In asterisk in extensions.conf File I have edited the Macro named
std-exten ( [macro-stdexten] )
Just before Dial I put the following :
[macro-stdexten]
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten = s,1,Gotoif($[{LEN(${CALLERID(num)})} 3]?3)
exten = s,2,Goto(s,4)
exten = s,3,TrySystem(wget -qb -O /dev/null -o /dev/null
http://sugarcrmIP/popup.php?number=${CALLERID(num)}extension=${ARG1}http://sugarcrmIP/popup.php?number=$%7BCALLERID(num)%7Dextension=$%7BARG1%7D
)
exten = s,4,Dial(${ARG2},35,rt) ; Ring the interface, 20 seconds maximum
exten = s,5,Goto(s-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send to
voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to start
;exten = s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to voicemail w/
busy announce
;exten = s-BUSY,2,Goto(default,s,1) ; If they press #, return to start
exten = s-BUSY,1,Playtones(busy)
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
exten = a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into
VoicemailMain
so when calls come in to asterisk before execute the dial in the macro it
will execute the
TrySystem(wget -qb -O /dev/null -o /dev/null
http://sugarcrmIP/popup.php?number=${CALLERID(num)}extension=${ARG1}http://sugarcrmIP/popup.php?number=$%7BCALLERID(num)%7Dextension=$%7BARG1%7D
)
and replace ${CALLERID(num)} with the callerid of customer and ${ARG1}
with user extension
as I stated in step 4 as an example the following will execute
TrySystem(wget -qb -O /dev/null -o /dev/null
http://sugarcrmIP/popup.php?number=7891234extension=105;)
and this will cause a screen popup on the user screen which is logged in
the sugarcrm and his/her extension number is 105.
Everything work fine with this implementation but when it comes to Queue it
fails , cause in the Queue the [macro-stdexten] will not be executed for
dialing.
I have a Queue ( huntgroup ) which all the extensions are its member and
with random algorithm the calls will be distributed to extensions.
I need the popup but don't know how to execute the URL with the replacement
of the ${CALLERID(num)} and ${ARG1} with value.
Please someone help me.
Best Regards,
Mohsen
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Re: [asterisk-users] Hangup extensions via CLI?
I have a feeling we're overdoing it :) l. 2009/2/12 Lukas Rypl r...@marconi.ttc.cz asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep SIP/7000 Hi, I used this way of processing output from asterisk 1.2 and found out that it is not 100% safe because there can appear unprintable characters in the output. This will cause the following grep command to show message similar to Binary content: matched instead of expected line. It is necessary to use strings -a to filter output. So your example should be: asterisk -rx 'core show channels' | strings -a | grep SIP/7000 Hope it helps Lukas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: Manager Interface Originate (ASYNC) - How to get the Originate Status
2009/2/13 Aloysius Lloyd lloyd.aloys...@sunteltech.ca
Can you explain what do you mean failed extension ?
Regards
Lloyd
On Fri, Feb 13, 2009 at 2:00 AM, Giedrius Augys voi...@gmail.com wrote:
2009/2/13 Aloysius Thevarajah Lloyd (SunTel Technologies)
lloyd.aloys...@sunteltech.ca
Dear All,
I am originating the call directly to the SIP Provider using the maganger
interface + originate (ASYNC) command. Here is the PHP-AGI Script.
$call = $asm-send_request('Originate',
array('Channel'=SIP/416...@abc/n,
'Context'='ORIG',
'Exten'='s',
* 'Async'='1',*
'MaxRetries' = '1',
'RetryTime' = '10',
'Priority'=1,
'Account'=$phonenumber,
'Callerid'=$callid)
*extensions.conf*
[ORIG]
exten = s,1,Answer
exten = s,2,Playback(ivrfile)
exten = s,n,Hangup
How Can I get the Originate Status using Async? ANSWER, BUSY, NOANSWER
etc..
Thank you.
Lloyd
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Hi.,
You can use 'failed' extension on the ORIG .
--
Pagarbiai / Best Regards,
Giedrius Augys
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I hope this help you:
context autodialer {
_X. = {
Answer();
Wait(1);
Playback(${PROMPT});
Hangup();
};
failed = {
Noop(Unsuccessfull call);
};
};
--
Pagarbiai / Best Regards,
Giedrius Augys
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Re: [asterisk-users] Fwd: Manager Interface Originate (ASYNC) - How to get the Originate Status
Thank you.
Lloyd
On Fri, Feb 13, 2009 at 2:43 AM, Giedrius Augys voi...@gmail.com wrote:
2009/2/13 Aloysius Lloyd lloyd.aloys...@sunteltech.ca
Can you explain what do you mean failed extension ?
Regards
Lloyd
On Fri, Feb 13, 2009 at 2:00 AM, Giedrius Augys voi...@gmail.comwrote:
2009/2/13 Aloysius Thevarajah Lloyd (SunTel Technologies)
lloyd.aloys...@sunteltech.ca
Dear All,
I am originating the call directly to the SIP Provider using the
maganger interface + originate (ASYNC) command. Here is the PHP-AGI
Script.
$call = $asm-send_request('Originate',
array('Channel'=SIP/416...@abc/n,
'Context'='ORIG',
'Exten'='s',
* 'Async'='1',*
'MaxRetries' = '1',
'RetryTime' = '10',
'Priority'=1,
'Account'=$phonenumber,
'Callerid'=$callid)
*extensions.conf*
[ORIG]
exten = s,1,Answer
exten = s,2,Playback(ivrfile)
exten = s,n,Hangup
How Can I get the Originate Status using Async? ANSWER, BUSY, NOANSWER
etc..
Thank you.
Lloyd
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Hi.,
You can use 'failed' extension on the ORIG .
--
Pagarbiai / Best Regards,
Giedrius Augys
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I hope this help you:
context autodialer {
_X. = {
Answer();
Wait(1);
Playback(${PROMPT});
Hangup();
};
failed = {
Noop(Unsuccessfull call);
};
};
--
Pagarbiai / Best Regards,
Giedrius Augys
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Re: [asterisk-users] CISCO 2950 - 4 connections - Cap of 512 Kbps - How to bond ?
In my opinion the only strategy that has a high probability of success is: Get a Cisco with five ethernet ports. Use one for your connection to asterisk. Use the other four as your connection to the ISP, and MUX them. Can you please point me to some resource on how to MUX ? All the other suggestions have a very low probability of success since: As someone who works for an ISP, the best advice I can give you is to tell them where to go (*after* fully setting up and testing a new ISP that is). In this town in Asia this is the only ISP that would work given the requirements of low latency to the VOIP server on the west coast and their ability to keep the connection up. Call your phone company and compare the price of getting a T1 versus what these clowns are charging you. Each 512Kbps of upload costs $40 but a T1 to handle 20 calls will cost much more then $160 a month. And to answer a question that was asked: The copper to your location only handles 512kpbs per pair, so they add an extra modem every time they open a new port? The ISP said that they ran a fiber optic wire to a media box at our office and from there there is a RJ45 to the switch. They bring no new equipment to our premises each time we provison a new port. Hence this upload speed limitation is not due to the copper wire. Any suggestions on what to do from this point on, Thanks for your time, On Thu, Feb 12, 2009 at 5:53 PM, Michael mich...@networkstuff.co.nz wrote: On Fri, 13 Feb 2009 12:41:51 Jeff LaCoursiere wrote: Get a Cisco with five ethernet ports. Use one for your connection to asterisk. Use the other four as your connection to the ISP, and MUX them. Great way to spend 5K :) j On Thu, 12 Feb 2009, Vikas wrote: I have asked the ISP to rate limit a single port to 2M but my requests have got me no where, I would really appreciate any suggestions on what I can do at my end since I have given up hope of the ISP co-operating with me, As someone who works for an ISP, the best advice I can give you is to tell them where to go (*after* fully setting up and testing a new ISP that is). With the economies of the world tighter then usual at present, and ISP's a plenty, I can only suggest they are idiots or for some reason they don't want your business Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
