[Asterisk-Users] Re: MeetMe 'i' option not working correctly?
In article [EMAIL PROTECTED], Jon Webster [EMAIL PROTECTED] wrote: I'm running 2.4.5 and app_meetme never plays conf-hasleft or conf-hasjoined with user names. I looked at app_meetme.c, but couldn't determine the cause. Any suggestions are greatly appreciated. exten = 600,1,MeetMe(600|i) I get the following: -- Executing MeetMe(SIP/jon-21f8, 600|aciMps) in new stack == Parsing '/etc/asterisk/meetme.conf': Found Mar 8 06:13:53 WARNING[5197]: channel.c:2535 ast_request: No channel type registered for 'zap' Mar 8 06:13:53 WARNING[5197]: app_meetme.c:461 build_conf: Unable to open pseudo channel - trying device The above messages indicate that chan_zap.so isn't loaded. Possibly it isn't even built. You need to build *and install* zaptel before starting to build Asterisk. Asterisk will find the zaptel libraries and will build chan_zap. Or maybe you just did what someone else did the other day: added a noload statement for chan_zap in modules.conf because you didn't have any zaptel hardware. You need chan_zap for MeetMe, and if you haven't got any zaptel hardware, you also need to make sure ztdummy is loaded and running correctly. Hope this helps. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can't hear busy tone
HI Group! I have strange problem. Since I started to use H323 with my VoIP provider when I dial the person that is currently busy, I can't hear busy tone on my handset. What could be the problem? What should I look for? How is this exactly called (because I even don't know what to look for). Hopefully someone will be able at least to give me some starting point. -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Attended transfer returns invalid extension
Hello, I am trying to set up a slim Asterisk. Blind transfer works, and attended transfer always plays pbx-transfer, then I dial the extension and it plays pbx-invalid. Here is my modules.conf : ; * Resources * load = res_features.so load = res_adsi.so load = res_monitor.so load = res_musiconhold.so ; * PBX * load = pbx_config.so ; * Channels * load = chan_sip.so ; * Codecs * load = codec_gsm.so load = codec_ulaw.so ; * Formats * load = format_gsm.so ; * Applications * load = app_db.so load = app_dial.so load = app_macro.so load = app_playback.so load = app_transfer.so My extensions.conf is quite bigger, I use some macros and different contexts. But I do not understand why blind transfer works while attended transfer returns invalid extension. Do I miss a module ? Please help, Alexis Fécourt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: ON DEMAND call Recording
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: 8. ozujak 2006 23:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: ON DEMAND call Recording I would find two possibilities: 1. on demand. Dial another extension number after the call, what executes a system command 2. automatically. Add in the dialplan the system command after hanging up. Hi Ronald! The second option is weary interesting. I think I have enough knowledge to make it work. -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] REGISTER headers changed
Hasn't anybody ever come accross these changes before? Jason Jason Frisch wrote: Can someone help me with upgrading to the lastest version. I am using the same sip.conf file, but the headers have changed and registration fails. Has something change in the conf file that would cause this? Notice 1.2.5 has no Authoization at all... Regards, Jason Version 1.0.9 --- REGISTER sip:voip-ca35323.ocn.ne.jp SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK035070d8 From: sip:[EMAIL PROTECTED];tag=as1cdfeadf To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Authorization: Digest username=someusername, realm=nc01.ipp.biglobe.ne.jp, algorithm=MD5, uri=sip:2 10.227.109.232, nonce=1141805370, response=016070a49b3caa88a3fb76e8b7a91aa1, opaque= Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 (no NAT) to 210.227.109.232:5060 denwa*CLI Sip read: SIP/2.0 200 OK v: SIP/2.0/UDP xxx:5060;branch=z9hG4bK035070d8 From: sip:[EMAIL PROTECTED];tag=as1cdfeadf To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Expires: 7200 m: sip:[EMAIL PROTECTED]:5060;expires=7200 Event: registration Content-Length: 0 Date: Wed, 08 Mar 2006 08:55:11 GMT -- Version 1.2.5 -- REGISTER sip:voip-ca35323.ocn.ne.jp SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK08b2597f;rport From: sip:[EMAIL PROTECTED];tag=as6d23ff7d To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 -- SIP read from :5060: SIP/2.0 400 Bad Request v: SIP/2.0/UDP :5060;branch=z9hG4bK08b2597f From: sip:[EMAIL PROTECTED];tag=as6d23ff7d To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Expires: 120 Event: registration l: 0 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe
Best of luck :-D I would be interested in your progress on this. I am having very little problem in convincing ppl to upgrade their multiple BRI cricuits for a single pri. The cost difference between a te110 (or a Sangoma A101) MORE than covers the difference from the customer stand point, especially once you are up to 3 ISDN-2 Interfaces. A single port E1 is cheaper than any multi BRI adapter I've seen, and based on Telstra pricing, 3.5 BRI services is about the point where the PRI is the cheaper option in terms of monthly rental. Installation cost is another matter but after a year or so it doesn't matter so much. One use for the multi BRI card though, especially one that can do NT mode, is that you can use it to trunk to a legacy BRI PBX, which is why I'm still interested in finding one for use in Australia. James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk code help
I'm very new to Asterisk, I'm tracing the Asterisk code, i'm feeling difficulty in understanding the code, so please tell me where i can get the documentation of the code and, design and architecture of the code. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MultiBRI in Australia - found one - maybe
James Harper wrote: One use for the multi BRI card though, especially one that can do NT mode, is that you can use it to trunk to a legacy BRI PBX, which is why I'm still interested in finding one for use in Australia. I'm using the Eicon Diva Server V-4BRI (~$2,200 each). They are awesome. Onboard hardware echo cancellation, native CAPI drivers for Linux (source available) and chan_capi compatibility. They can do both NT and TE mode, so you can use them to connect to legacy BRI PBXs as well. cYa, Avi -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Festival tts
Hi I have installed Festival on the same box as asterisk and followed the instructions to integrate it with asterisk. Festival seems to work fine on its own performing text to speech from the command line or via a file. Asterisk answers the call but there is no speech. I can see no errors in the Festival log file The asterisk console shows --Executing Answer(SIP/81801-c091, ) in a new stack --Executing Festival(SIP/81801-c091, mary had a little lamb) in a new stack ==Parsing '/etc/asterisk/festival.conf':Found there is nothing else after this If I start festival as festival --server I can see the output Server 11:39:14 : Festival server started on port 1314 Client(1) 11:39:21 : accepted from localhost Client(1) 11:39:21 : disconnected Initially I added the code to festival.scm for * but later patched the Festival code and re-complied it. For every test I have restarted * after Festival Any help appreciated Thanks Steven Steven Jack Videoconferencing Manager University of Glasgow Computing Service Glasgow G12 8QQ UK Tel +44(0)1413303828 Fax +44(0)1413303820 Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Real Time Asterisk
Olle E Johansson wrote: 8 mar 2006 kl. 17.07 skrev Fernando Lujan: There is a realtime PostgreSQL driver for testing in the bug tracker, please test it! http://bugs.digium.com/view.php?id=5637 Hi Olle, I want something read for production. Is there such driver, even using the mysql database? I found this article: http://www.asteriskguru.com/tutorials/realtime_pgsql.html But it uses a old version of asterisk. Does the current stable version support this features? I don't want to put a cvs head version in a production system. Thanks in advance. Fernando Lujan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No DTMF
already tried it and it didnt work. Could there be any other files that may have been messed with that is causing this ? Dovid --- Mark Edwards [EMAIL PROTECTED] wrote: Try dtmfmode=info and see if that works. Mark -Original Message- From: Dovid Bender [mailto:[EMAIL PROTECTED] Sent: Thursday, 9 March 2006 6:08 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] No DTMF Some one was on my server making changes to my sip.conf files. I am now having trouble with DTMF. No matter what I use (inband,auto,rfc2833) the dtmf tones seem to not come thru. I compared it to the wiki and all the configs seem to be in order. Here is my sip.conf [general] disallow=all ;allow=g729 ; requires license for g729 allow=ulaw port = 5060 nat=yes context=from-sip bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) maxexpirey=4800 ; Maximum expiration for registrations defaultexpirey=1800 ; Default expiration for registrations canreinvite=no ; Allow clients to directly connect if set to yes. Set to no if behind NAT. tos=reliability srvlookup=yes ; Enable DNS SRV lookups on outbound calls videosupport=no ; Turn on support for SIP video dtmfmode=rfc2833 ;rfc2833 ;inband ;rfc2833 ; DTMF inband need to be set here. pedantic=no externip=..XXX ;Sip Media register = XX:[EMAIL PROTECTED]/7322761368 [sipmedia6] type=friend user=XX ;(Phone Number) username=XX ;(Phone Number) fromuser=XX ;(Phone Number) authname=XX ;(Phone Number) secret= ;(SIP Password) host=sip.sipmedia.com disallow=all allow=ulaw context=ServerHighway realm=sip1.xchangetele.com fromdomain=sip.sipmedia.com dtmfmode=rfc2833 canreinvite=no insecure=very Here is my extensions.conf [general] static=yes writeprotect=yes [ServerHighway] ;Play Server Highway IVR Exten = s,1,Background(server-highway-ivr) Exten = s,2,Background(blank-file-10) Exten = 1,1,Ringing() Exten = 1,2,Wait(15) Exten = 1,3,Macro(stdexten,9511,9511) Exten = 2,1,Ringing() Exten = 2,2,Wait(15) Exten = 2,3,Macro(stdexten,9512,9512) Exten = 3,1,Ringing() Exten = 3,2,Wait(15) Exten = 3,3,Macro(stdexten,9513,9513) Exten = 4,1,Ringing() Exten = 4,2,Wait(15) Exten = 4,3,Macro(stdexten,9514,9514) Exten = i,1,Background(invalid) Exten = i,2,Goto(s,1) Exten = t,1,Goto(s,1) exten = 9,1,Goto(s,1) ;Extension To Record Main IVR Message exten = 500,1,Authenticate(XXX) exten = 500,2,Record(ServerHighwayIvr:gsm) _ Yahoo! Mail Bring photos to life! New http://pa.yahoo.com/*http:/us.rd.yahoo.com/evt=39174/*http:/photomail.mail. yahoo.com PhotoMail makes sharing a breeze. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to check if ztdummy is working properly?
Hi, As I am having problems with MoH and have tried everything to solve it and nothing worked, I was thinking maybe the timing source, i.e. ztdummy, is not working properly and that is what is causing problem. Is there any test to check whether ztdummy is working properly? Thanks, Zach A ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax behind ATA
Hi, I have installed a fax machine on a HT 486 ATA in my office, and it works perfectly, to send and receive faxes. When I install the same ATA on a fax machine at home (behind a NAT, in case it matters) faxes are received correctly, but I cannot send. Asterisk keeps showing a message "Unknown RTP codec 96 received" I am using g711u as the default codec. Does anyone have any idea or have already been through this problem? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Music On Hold playback
Hello I have a rather limit hardware where i try to run Asterisk, 1 GHz VIA C3. Everything works fine except music playback on MOH. I have encoded the music in same codac as as the voicestream (ulaw), ulaw is used end to end. MOH fades in and out with varius volume, and is very choppy. What I dont quite understand is when I put the music file as you are first in line it plays perfectly, not choppy or anything. Does anyone know what this can be? Below is my musiconhold.conf //File musiconhold.conf [classes] [default] mode=files directory=/var/lib/asterisk/mohmp3/default random=yes // BR Amund Nygaard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [asterisk-biz] Professional Recordings
Waldo Rubinstein wrote: Can anyone recommend a company that does professional Asterisk recordings for things like IVR, greetings, MOH, announcements, etc? Thanks, Waldo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz I use cheapradiospots.com usually. Mike charges $15 for a page (12 point, double spaced) of dry voice. You'll need to edit it down (I use Audacity/OSX) into individual prompts. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: [asterisk-biz] Professional Recordings
Try Allison at theivrvoice.com. She is the voice of Asterisk. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, March 08, 2006 11:06 PM To: Commercial and Business-Oriented Asterisk Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re: [asterisk-biz] Professional Recordings http://www.mikesullivan.com/ http://thevoice.digium.com/ On Wed, 8 Mar 2006, Waldo Rubinstein wrote: Can anyone recommend a company that does professional Asterisk recordings for things like IVR, greetings, MOH, announcements, etc? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Festival tts
I tried doing the same things as you to make Festival
work with Asterisk,
but I had a small problem with Festival only prducing
the sound if the text
was tess than 14 characters
So I used the other approach and used the text2wave
utility instead (I saw
on some postings that people recommended it) and it
wrols like a charm now.
Here is the complete macro I used for TTS:
[macro-sandtts]
exten = s,1,Set(FNAME=${EPOCH})
exten = s,2,System(echo ${ARG1} |
/usr/bin/text2wave -scale 1.5 -F
000 -o /tmp/${FNAME}.wav)
exten = s,3,Playback(/tmp/${FNAME})
exten = s,4,System(rm /tmp/${FNAME}.wav)
First we creat ann (almost) unique file name
Next we call the text2wave utility with correct
switches and passing the
text we need to pronounce as input to the utility.
then we playback the generated wave file.
Finally we remove the generated wave file.
Just call the macro with the text you want to say and
it will work for you.
Message: 28
Date: Thu, 9 Mar 2006 11:43:56 -
From: Steven [EMAIL PROTECTED]
Subject: [Asterisk-Users] Festival tts
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
Hi I have installed Festival on the same box as
asterisk and followed the
instructions to integrate it with asterisk.
Festival seems to work fine on its own performing
text to speech from the
command line or via a file.
Asterisk answers the call but there is no speech. I
can see no errors in
the
Festival log file
The asterisk console shows
--Executing Answer(SIP/81801-c091, ) in a new
stack
--Executing Festival(SIP/81801-c091, mary had a
little lamb) in a new
stack
==Parsing '/etc/asterisk/festival.conf':Found
there is nothing else after this
If I start festival as festival --server I can see
the output
Server 11:39:14 : Festival server started on port
1314
Client(1) 11:39:21 : accepted from localhost
Client(1) 11:39:21 : disconnected
Initially I added the code to festival.scm for * but
later patched the
Festival code and re-complied it.
For every test I have restarted * after Festival
Any help appreciated
Thanks
Steven
Steven Jack
Videoconferencing Manager
University of Glasgow
Computing Service
Glasgow G12 8QQ
UK
Tel +44(0)1413303828 Fax +44(0)1413303820
Email: [EMAIL PROTECTED]
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[Asterisk-Users] ruby-agi-1.1.2 released
Release notes of ruby-agi-1.1.2 March 07, 2006 In this release bug # 3722 has been fixed Details of this can be found at http://rubyforge.org/tracker/index.php?func=detailaid=3733group_id=883atid=3477 http://rubyforge.org/tracker/index.php?func=detailaid=3733group_id=883atid=3477 Feedback, suggestion, feature request, bug report is always appreciated. For more information, please visit projects homepage: http://rubyforge.org/projects/ruby-agi/ To install ruby-agi, % gem install ruby-agi and to update exiting ruby-agi % gem update ruby-agi Thanks, Mohammad Khan info AT beeplove DOT com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HW Echo Cancellers
Darren Wright wrote: Forget the orion.lots of DTMF problemstech support is not Terribly well spoken. Look for ANY of the 257* series... Just ebay for t1 echo -D Do *not* forget the Orion. We have 3 in place now working beautifully. Can't speak on Darren's problems, but our units have all installed in less than 5 minutes, totally eradicated echo, have had zero DTMF issues and Orion has been helpful when needed (though we have not needed much tech support). Mike Clark ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] digium certification for Europe
Title: digium certification for Europe Im little bit confused which Digium hardware is certificated for use in Europe. It looks like new cards are certificated, like TE4XX series. What about TE110 or TDM400P? Can someone confirm that? Thanks, David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is extension.conf documentation wrong?
In extension.conf i read this: ; ! - wildcard, causes the matching process to complete as soon as ; it can unambiguously determine that no other matches are possible I tried to define: exten = _50,1,Dial(...) exten = _5!,1,Dial(...) If i dial 50, due to asterisk reordering _5! is exectued but in the comment above it says ! is matchaed only if unambiguous... so what's wrong with my test? Thanks in advance Mario ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Real Time Asterisk
I am using the odbc set up with postgres right now and it works fine. http://www.voip-info.org/wiki/view/Asterisk+RealTime+PostgreSQL has most of the info to get you running. As for meetme, I took the app_cbmysql stuff for webmeetme and rewrote it for postgres. I am still testing it, but it seems to work great right now... if dan is out there anywhere... I would like to help move the webmeetme part of this to db independant and make it so it can run register_globals off :) Sean On Thu, 2006-03-09 at 09:09 -0300, Fernando Lujan wrote: production. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 7940/60 SIP 8.2
So has anybody tried installing the new SIP version? It seems nobody has had luck with the 7970 and it's new SIP image and the description for the 7940/60 specifically says for CCM v5.0. _ Mobilcom http://www.mobilcom.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] impact of qualify=yes
Anyone have any information on the performance impact of using qualify=yes for hundreds (500ish) of SIP UAs? I have seen tidbits on qualifyspreading=yes, but not enough to understand what it does. I assume lessens the peak load of qualify sip options queries? Thx! Qualify=yes means we send one SIP packet to the sip user and receive one packet back, and calculate a round trip time. And I think this happens around once a minute. I can't imagine the performance impact being very big. The PC on my desk can do 2000 ICMP pings in 10 seconds with no impact whatsoever.qualifying SIP agents can't be much worse. But I am not an expert on the matter. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Jitter buffer for SIP channels (OT?)
This might be a better question for the dev list, but I don't think they want to be bothered by my silly questions. Does anyone know when we can expect to see a jitter buffer for SIP channels? I know they've been working on a generic jitter buffer since around last summer, just wondering if there's been any progress. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7940/60 SIP 8.2
Mailing List wrote: So has anybody tried installing the new SIP version? It seems nobody has had luck with the 7970 and it's new SIP image and the description for the 7940/60 specifically says for CCM v5.0. I'm planning on trying early this afternoon. (EST) Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] paly sound when we Start and stop recording
Hii have enabled on demand recording on my asterisk server.i want to have voice playing when i start recording saying recording has started and when i press *1 to stop the recording it should play recording stopped . is that possible with the latest version [EMAIL PROTECTED] 2.6 . or is there way we can tweak some module and get the desired output .please let me know if some one have some solution thanksGiridhar Bandi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Stress Tests from AsteriskGur with [EMAIL PROTECTED]
Hi all, I'm planning to test my two [EMAIL PROTECTED] one is 1.5 and another is 2.5 Does any one got already Astertest - asterisk stress testing tool working one? I've red Asterisk Guru, http://www.asteriskguru.com/tutorials/astertest.html and after all the tutorial still remaining questions from users with problems ( in fact i didn't find any sucessfull feedback). I'm a bit afraid of doing all the tutorial and get in troubles with my stable asterisks Any one has tried it? Best regards, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Real Time Asterisk
Sean Cook wrote: I am using the odbc set up with postgres right now and it works fine. http://www.voip-info.org/wiki/view/Asterisk+RealTime+PostgreSQL has most of the info to get you running. As for meetme, I took the app_cbmysql stuff for webmeetme and rewrote it for postgres. I am still testing it, but it seems to work great right now... if dan is out there anywhere... I would like to help move the webmeetme part of this to db independant and make it so it can run register_globals off :) Thanks for your reply Sean. This is indeed what I need. I should have installed unixodbc before asterisk? Will asterisk automatically be compiled using it in the 1.2.5 version? Thanks in advance. Fernando Lujan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Festival tts
Can someone tell me what I'm doing wrong here? I'm trying this from the
command prompt.
# echo Hello World | /usr/bin/text2wave -scale 1.5 -F 000 -o
/tmp/1141915933.wav
rateconv: failed to convert from 16000 to 0
doing v
#
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Antoine
Megalla
Sent: Thursday, March 09, 2006 8:27 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Festival tts
I tried doing the same things as you to make Festival work with
Asterisk, but I had a small problem with Festival only prducing the
sound if the text was tess than 14 characters
So I used the other approach and used the text2wave utility instead (I
saw on some postings that people recommended it) and it wrols like a
charm now.
Here is the complete macro I used for TTS:
[macro-sandtts]
exten = s,1,Set(FNAME=${EPOCH})
exten = s,2,System(echo ${ARG1} |
/usr/bin/text2wave -scale 1.5 -F
000 -o /tmp/${FNAME}.wav)
exten = s,3,Playback(/tmp/${FNAME})
exten = s,4,System(rm /tmp/${FNAME}.wav)
First we creat ann (almost) unique file name Next we call the text2wave
utility with correct switches and passing the text we need to pronounce
as input to the utility.
then we playback the generated wave file.
Finally we remove the generated wave file.
Just call the macro with the text you want to say and it will work for
you.
Message: 28
Date: Thu, 9 Mar 2006 11:43:56 -
From: Steven [EMAIL PROTECTED]
Subject: [Asterisk-Users] Festival tts
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
Hi I have installed Festival on the same box as
asterisk and followed the
instructions to integrate it with asterisk.
Festival seems to work fine on its own performing
text to speech from the
command line or via a file.
Asterisk answers the call but there is no speech. I
can see no errors in
the
Festival log file
The asterisk console shows
--Executing Answer(SIP/81801-c091, ) in a new
stack
--Executing Festival(SIP/81801-c091, mary had a
little lamb) in a new
stack
==Parsing '/etc/asterisk/festival.conf':Found
there is nothing else after this
If I start festival as festival --server I can see
the output
Server 11:39:14 : Festival server started on port
1314
Client(1) 11:39:21 : accepted from localhost
Client(1) 11:39:21 : disconnected
Initially I added the code to festival.scm for * but
later patched the
Festival code and re-complied it.
For every test I have restarted * after Festival
Any help appreciated
Thanks
Steven
Steven Jack
Videoconferencing Manager
University of Glasgow
Computing Service
Glasgow G12 8QQ
UK
Tel +44(0)1413303828 Fax +44(0)1413303820
Email: [EMAIL PROTECTED]
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[Asterisk-Users] DTFM or FSK
Hi everybody! Does anyone know what is the exactly modulation that the Digium TDM400P works? DTMF or FSK? If anyone know where to get a good material about it, please let me know. Thanks for any help. Regards, Filipe Mordhorst Brazil-SC smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cdr data
Hello,
I have an E1 and the possibility to use different
caller ids in this E1, so, before a Dial, I always have a SetCallerIDNum("User",
number).
When I check the CDR, the originator of the calls
appears to be this "number" I set in the caller id, but not the actual user that
originated the call.
Is there a way to set a callerid for the outgoing
call, but on cdr records to leave the originator id?
I know I could use the CDR user field, but I am
already using it for other purposes!
Thank you very much
Dov
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RE: [Asterisk-Users] System Design
Thanks for all of your replies! I was thinking the server was a little overboard, but I want this to last and also be expandable. We might be adding users/offices within the next year so I wanted to plan ahead. The DSL speed at the remote office is 1.5 to 6.0 Down and 384 to 608 up. The DSL does have a static IP address and it's pretty rock solid in regards to stability. I was planning on buying the G729a codec from Digium for use on all calls. In regards to: snip For normal incoming and outgoing calls, just have the asterisk box at that particular location handle it (no need for the remote office to connect to the main office's asterisk box, then call out via iax or sip for a long-distance phone call). /snip Would the remote office * need a couple of POTS lines to make those local calls? Once again thanks for all of your replies! They are definitely clearing things up for me. - Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Tanner Sent: Wednesday, March 08, 2006 6:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] System Design Lot of questions, lots of variables, but I'll touch base on a few things. 5-10 concurrent calls is hardly anything. A plain T1 will more than handle that, even at ulaw or alaw (non)compression. Throw in a decent codec, and 10 calls won't even put a dent in your T1. Heck, it'd handle all 20 users in your main office, and the 5 users in your remote office with G729, no problem. How reliable is the remote office's DSL connection? I'd make sure you have a static ip for it (dynamic ips are just slightly problematic, especially if you have slightly flaky service, coupled with a slightly flaky modem). If it's reliable, then just keep that. What's the connection speed? Need to know the upload and download. If it's ADSL, then the upload will be a fraction of the download, and will be the limiting factor. Since I don't know your specific setup, I can't tell you specifically what to do. I'll make some guesses though. Keep DSL. No need to use VPN just for asterisk. Make sure each end has a static ip (dynamic ip will work, but is harder to setup and more prone to errors). Have each asterisk box register to the other. For normal incoming and outgoing calls, just have the asterisk box at that particular location handle it (no need for the remote office to connect to the main office's asterisk box, then call out via iax or sip for a long-distance phone call). You can create local extensions that when dialed, will ring a person on the other asterisk box. I.e., a user at the main office can dial 2001, and get a user at the remote office. If you deal with call queues you can group users from both offices together, no problem. A T1 or a point to point connection at the remote office would work, but is probably unecessary. If their DSL connection is flaky and unreliable, then start looking at both options. I'd probably go with whichever is cheapest, be sure to factor in equipment costs (you can generally lease equipment with a T1 line, but not with a point to point connection). As far as server specs, if all it's going to run is asterisk, then that's overkill even if it was handling all the calls. If you think you need that much server but are on a budget, then get one setup for dual processors but with just one installed, and less ram but that has room to add more. If budget's not a problem, I say go for it! That system should last you for quite a while. As for QOS, sorry I can't help you there. You could get a cheap router that has QOS built-in, or run a separate low-end server just for QOS. Personally my asterisk box also serves as my nat server, so I just run QOS directly on it. It's probably not something you want to do in an office environment, but it's better than no QOS at all. Hopefully someone else will give you some good advice on QOS equipment. Joseph Tanner On 3/7/06, Jason Adams [EMAIL PROTECTED] wrote: Hey Everyone, We are in the works of planning a new * installation for our company. We have 20 users in our main office and 5 users in a remote office a couple of states away. Our call volume for the main office will be anywhere from 5-10 concurrent calls. The remote office will have about 3 heavy users with two users making calls occasionally. Right now we have an existing PBX. We have a T-1/PRI coming into the main office and a DSL connection at the remote office. We have a Cisco 2610/PIX 501 at the main office a cheesy linksys router at the remote site. We are planning on purchasing new Cisco IP phones for everyone. My main question is this: What type of hardware/network design would be best for this situation? Would a full T-1 at the remote site work with a VPN between the offices? Or would a higher bandwidth DSL work with a VPN? Or should we move to a Point-to-Point connection? What type of hardware
RE: [Asterisk-Users] DTFM or FSK
Can you be more specific? All digital cards (regardless of manufacture) use the same modulation, it is a standard and you can probably google it. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Filipe MordhorstSent: Thursday, March 09, 2006 10:09 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] DTFM or FSK Hi everybody! Does anyone know what is the exactly modulation that the Digium TDM400P works? DTMF or FSK? If anyone know where to get a good material about it, please let me know. Thanks for any help. Regards, Filipe MordhorstBrazil-SC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 7940/60 SIP 8.2
So has anybody tried installing the new SIP version? It seems nobody has had luck with the 7970 and it's new SIP image and the description for the 7940/60 specifically says for CCM v5.0. Just downloaded it after your email and got it working on the first try. Give me a few minutes to write up the procedure. Nabeel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Festival tts
Steven [EMAIL PROTECTED] wrote: Hi I have installed Festival on the same box as asterisk and followed the instructions to integrate it with asterisk. Festival seems to work fine on its own performing text to speech from the command line or via a file. Asterisk answers the call but there is no speech. I can see no errors in the Festival log file I asked the same question to this list a while back but got no replies. What OS are you using? How did you install Festival? What version of *? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Getting to the last old voicemail message
If you have many old voicemail messages, to get to the most recent one, you have to keep hitting 6 until you reach the last one. It would be better if you could hit 4 from the first message to get to the last message and/or have a digit that takes you the first and last messages respectively. Anyone have any patches for this? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Real Time Asterisk
Yes you do need unixODBC before you compile asterisk. Once you have installed unixODBC , asterisk will compile and offer you the following modules: cdr_odbc.so res_config_odbc.so res_odbc.so res_odbc.conf and cdr_odbc.conf are the related config files... Sean On Thu, 2006-03-09 at 11:57 -0300, Fernando Lujan wrote: using ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Festival tts
hey, how i do to do that with php agi's?
Este Mensaje Esta Hecho 100% con Electrones Reciclados
- Original Message -
From: Adam Robins [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, March 09, 2006 10:04 AM
Subject: RE: [Asterisk-Users] Festival tts
Can someone tell me what I'm doing wrong here? I'm trying this from the
command prompt.
# echo Hello World | /usr/bin/text2wave -scale 1.5 -F 000 -o
/tmp/1141915933.wav
rateconv: failed to convert from 16000 to 0
doing v
#
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Antoine
Megalla
Sent: Thursday, March 09, 2006 8:27 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Festival tts
I tried doing the same things as you to make Festival work with
Asterisk, but I had a small problem with Festival only prducing the
sound if the text was tess than 14 characters
So I used the other approach and used the text2wave utility instead (I
saw on some postings that people recommended it) and it wrols like a
charm now.
Here is the complete macro I used for TTS:
[macro-sandtts]
exten = s,1,Set(FNAME=${EPOCH})
exten = s,2,System(echo ${ARG1} |
/usr/bin/text2wave -scale 1.5 -F
000 -o /tmp/${FNAME}.wav)
exten = s,3,Playback(/tmp/${FNAME})
exten = s,4,System(rm /tmp/${FNAME}.wav)
First we creat ann (almost) unique file name Next we call the text2wave
utility with correct switches and passing the text we need to pronounce
as input to the utility.
then we playback the generated wave file.
Finally we remove the generated wave file.
Just call the macro with the text you want to say and it will work for
you.
Message: 28
Date: Thu, 9 Mar 2006 11:43:56 -
From: Steven [EMAIL PROTECTED]
Subject: [Asterisk-Users] Festival tts
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
Hi I have installed Festival on the same box as
asterisk and followed the
instructions to integrate it with asterisk.
Festival seems to work fine on its own performing
text to speech from the
command line or via a file.
Asterisk answers the call but there is no speech. I
can see no errors in
the
Festival log file
The asterisk console shows
--Executing Answer(SIP/81801-c091, ) in a new
stack
--Executing Festival(SIP/81801-c091, mary had a
little lamb) in a new
stack
==Parsing '/etc/asterisk/festival.conf':Found
there is nothing else after this
If I start festival as festival --server I can see
the output
Server 11:39:14 : Festival server started on port
1314
Client(1) 11:39:21 : accepted from localhost
Client(1) 11:39:21 : disconnected
Initially I added the code to festival.scm for * but
later patched the
Festival code and re-complied it.
For every test I have restarted * after Festival
Any help appreciated
Thanks
Steven
Steven Jack
Videoconferencing Manager
University of Glasgow
Computing Service
Glasgow G12 8QQ
UK
Tel +44(0)1413303828 Fax +44(0)1413303820
Email: [EMAIL PROTECTED]
__
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Tired of spam? Yahoo! Mail has the best spam protection around
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To UNSUBSCRIBE or update options visit:
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The contents of this email message and any attachments are confidential and
are intended solely for addressee. The information may also be legally
privileged. This transmission is sent in trust, for the sole purpose of
delivery to the intended recipient. If you have received this transmission
in error, any use, reproduction or dissemination of this transmission is
strictly prohibited. If you are not the intended recipient, please
immediately notify the sender by reply email and delete this message and its
attachments, if any.
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RE: [Asterisk-Users] cdr data
That is what the accountcode field is for, you can set
a unique accountcode for each devcice if you want to.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dov
BigioSent: Thursday, March 09, 2006 10:05 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] cdr
data
Hello,
I have an E1 and the possibility to use different
caller ids in this E1, so, before a Dial, I always have a
SetCallerIDNum("User", number).
When I check the CDR, the originator of the calls
appears to be this "number" I set in the caller id, but not the actual user
that originated the call.
Is there a way to set a callerid for the outgoing
call, but on cdr records to leave the originator id?
I know I could use the CDR user field, but I am
already using it for other purposes!
Thank you very much
Dov
___
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Re: [Asterisk-Users] Jitter buffer for SIP channels (OT?)
9 mar 2006 kl. 15.26 skrev Adam Moffett: This might be a better question for the dev list, but I don't think they want to be bothered by my silly questions. Does anyone know when we can expect to see a jitter buffer for SIP channels? I know they've been working on a generic jitter buffer since around last summer, just wondering if there's been any progress. You can see it right now and on your server in a few moments! It's in the bug tracker ready for testing. Either in the jitterbuffer branch or in the test-this-branch branch - both includes the same code. You need to enable it in the makefile. You can find the information about this patch here: http://bugs.digium.com/view.php?id=3854 Download a complete development version including the jitterbuffer with this command: svn checkout http://svn.digium.com/svn/asterisk/team/oej/jitterbuffer jitterbuffer (Change jitterbuffer to test-this-branch if you want more goodies to test). We need a lot of tests to be able to judge if and when we can move this forward into an Asterisk release version. Thanks for your help! Regards, /O --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HW Echo Cancellers
Doug Lytle wrote: Matt wrote: Tellabs looks a little too up-scale for what I need :). $1k for a single port orion unit might be worth considering for really stubborn installs though. http://cgi.ebay.com/Tellabs-2572-64ms-T1-echo-canceller_W0QQitemZ5863816619QQcategoryZ51279QQssPageNameZWDVWQQrdZ1QQcmdZViewItem Just a note: This vendor is selling cards with local side echo cancellation. Most of the cards that I purchased didn't have it. The 3 that I've purchased from him did. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 SIP - Displaying Time
Is there a way to display the time of the 7960 running firmware 7.4? Im unable to find any information. Add the following to SIPDefault.cnf or SIPMAC.cnf: sntp_server: time.nrc.ca sntp_mode: unicast time_zone: EST You should of course change your NTP server and/or time zone. Nabeel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is extension.conf documentation wrong?
exten = _50,1,Dial(...) exten = _5!,1,Dial(...) Remove the _ from the first line. Nabeel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 7940/60 SIP 8.2
So has anybody tried installing the new SIP version? It seems nobody has had luck with the 7970 and it's new SIP image and the description for the 7940/60 specifically says for CCM v5.0. Just downloaded it after your email and got it working on the first try. Give me a few minutes to write up the procedure. OK, I had a 7960 running SIP 7.4 and managed to upgrade to SIP 8.2. I haven't tried any of the new features, but can make and receive calls fine. I put the following on the TFTP server that the phone has always polled for configuration files: OS79XX.txt - containing the line P003-08-2-00, although it does not appear this file was even needed (see TFTP log below) SIPDefault.cnf - containing the line image_version: P0S3-08-2-00 POS3-08-2-00.loads - from the .cop file after renaming it .tar.gz and extracting using WinRAR POO3-08-2-00.sbn - same as previous POS3-08-2-00.sb2 - same as previous POO3-08-2-00.bin - same as previous, although it does not appear this file was even needed (see TFTP log below) In addition, I had my SIPMAC.cnf, RINGLIST.DAT and dialplan.xml - the same ones I was using with SIP 7.4. This is the TFTP log after the phone was rebooted: 09/03/2006 10:11 :TFTP Error from 192.168.1.60 requesting CTLSEP003094C2A192.tlv : File does not exist 09/03/2006 10:11 :TFTP Error from 192.168.1.60 requesting SEP003094C2A192.cnf.xml : File does not exist 09/03/2006 10:11 :Sending SIP.cnf to (192.168.1.60) 09/03/2006 10:11 :Sent SIP.cnf to (192.168.1.60), 6173 bytes 09/03/2006 10:11 :Sending SIPDefault.cnf to (192.168.1.60) 09/03/2006 10:11 :Sent SIPDefault.cnf to (192.168.1.60), 46 bytes 09/03/2006 10:11 :Sending P0S3-08-2-00.loads to (192.168.1.60) 09/03/2006 10:11 :Sent P0S3-08-2-00.loads to (192.168.1.60), 461 bytes 09/03/2006 10:11 :Sending P003-08-2-00.sbn to (192.168.1.60) 09/03/2006 10:11 :Sent P003-08-2-00.sbn to (192.168.1.60), 129644 bytes 09/03/2006 10:11 :TFTP Error from 192.168.1.60 requesting CTLSEP003094C2A192.tlv : File does not exist 09/03/2006 10:11 :TFTP Error from 192.168.1.60 requesting SEP003094C2A192.cnf.xml : File does not exist 09/03/2006 10:11 :Sending SIP.cnf to (192.168.1.60) 09/03/2006 10:11 :Sent SIP.cnf to (192.168.1.60), 6173 bytes 09/03/2006 10:11 :Sending SIPDefault.cnf to (192.168.1.60) 09/03/2006 10:11 :Sent SIPDefault.cnf to (192.168.1.60), 46 bytes 09/03/2006 10:11 :Sending P0S3-08-2-00.loads to (192.168.1.60) 09/03/2006 10:11 :Sent P0S3-08-2-00.loads to (192.168.1.60), 461 bytes 09/03/2006 10:11 :Sending P0S3-08-2-00.sb2 to (192.168.1.60) 09/03/2006 10:11 :Sent P0S3-08-2-00.sb2 to (192.168.1.60), 785338 bytes 09/03/2006 10:12 :Sending SIPDefault.cnf to (192.168.1.60) 09/03/2006 10:12 :Sent SIPDefault.cnf to (192.168.1.60), 46 bytes 09/03/2006 10:12 :Sending SIP.cnf to (192.168.1.60) 09/03/2006 10:12 :Sent SIP.cnf to (192.168.1.60), 6173 bytes 09/03/2006 10:12 :Sending RINGLIST.DAT to (192.168.1.60) 09/03/2006 10:12 :Sent RINGLIST.DAT to (192.168.1.60), 19 bytes 09/03/2006 10:12 :Sending dialplan.xml to (192.168.1.60) 09/03/2006 10:12 :Sent dialplan.xml to (192.168.1.60), 1046 bytes I hope this works for everyone else as well. Nabeel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Merlin Magix Integration
Hi List,
Merlin Magix hardware v02
I'm trying to get asterisk to act as a voicemail server for a lucent
merlin magix PBX that we purchased used. We have 4 FXO channels between
the two PBXs on a Sangoma A200 card. The 770 dialgroup is working
properly, in that calls to 770 are answered by Asterisk. The magix is
sending mode codes in the format #XX#XXX#, where the 2nd block of digits
is the calling extension. I'm stripping off the unneeded pound signs
and digits, and calling voicemailmain. The problem I'm having is that
the asterisk is starting to play vm-password and then interrupts
immediately and errors with an incorrect password. Then it works
normally. Below is the relevant asterisk config and the asterisk log.
Zaptel is configured to start inbound calls in the inbound context. The
voicemail accounts and sip accounts are all in the default context.
Asterisk log
-- Starting simple switch on 'Zap/3-1'
Mar 9 10:26:35 NOTICE[4211]: chan_zap.c:6063 ss_thread: Got event 18
(Ring Begin)...
-- Executing Answer(Zap/3-1, ) in new stack
-- Executing WaitExten(Zap/3-1, 1) in new stack
== CDR updated on Zap/3-1
-- Executing NoOp(Zap/3-1, #00#219#) in new stack
-- Executing Set(Zap/3-1, [EMAIL PROTECTED]) in new stack
-- Executing NoOp(Zap/3-1, [EMAIL PROTECTED]) in new stack
-- Executing VoiceMailMain(Zap/3-1, [EMAIL PROTECTED]) in new stack
-- Playing 'vm-password' (language 'en')
-- Incorrect password '' for user '219' (context = default)
-- Playing 'vm-incorrect' (language 'en')
-- Playing 'vm-password' (language 'en')
||| Caller hangs up here |||
Mar 9 10:26:41 WARNING[4211]: app_voicemail.c:4998 vm_authenticate:
Unable to read password
-- Hungup 'Zap/3-1'
extensions.conf
[inbound]
exten = s,1,Answer()
exten = s,2,WaitExten(1) ; Allow time for mode code digits to
come across
; The following extensions grab the mode code
; coming from the Avaya PBX and route the
; call appropriately via the Voicemail()
; and VoiceMailMain() apps.
;
; someone pressed vmail check
; #00#243#
exten = _#XX#XXX#,1,noop(${EXTEN})
exten = _#XX#XXX#,2,Set(CVAR=${EXTEN:4:[EMAIL PROTECTED])
exten = _#XX#XXX#,3,NoOp(${CVAR})
exten = _#XX#XXX#,4,VoicemailMain(${CVAR})
;exten = _#XX#XXX#,2,VoicemailMain(${EXTEN:4:[EMAIL PROTECTED])
exten = _#XX#XXX#,5,Hangup()
As can be seen, I've tried calling voicemailmain with the ${EXTEN:4:3}
digit stripping as part of the command, and also I've tried moving the
digit stripping to a variable.
I'd very much appreciate any help you folks can offer.
Thanks much.
Darren Ellis
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[Asterisk-Users] Asterisk @ Home 2.6 Call hangs up
I have installed asterisk @
home 2.6. I am using a Telasip VOIP account. When I make outbound or
inbound calls the calls seem to connect and then get hung up. I was
wondering if there was something that I am misisng. I have tried several
different sip.conf configurations. Here is what they are
currently.
telasip-gw
context=telasip-indtmfmode=rfc2833fromuser=jrasxxxhost=gw4.telasip.cominsecure=verynat=yessecret=xyztype=peerusername=jrasxxx
551212
context=from-pstndtmfmode=rfc2833host=gw4.telasip.cominsecure=verynat=yesqualify=yessecret=xyztype=peerusername=jrasxxx
The odd thing is it worked once or twice then stopped. If anyone
could shed some light it would be greatly apperciated.
Here is what the asterisk output looks like:
-- AGI Script fixlocalprefix completed, returning
0 -- Executing SetVar("IAX2/100-2", "OUTNUM=770555")
in new stack -- Executing Cut("IAX2/100-2",
"custom=OUT_2|:|1") in new stack -- Executing
GotoIf("IAX2/100-2", "0?16") in new stack -- Executing
Dial("IAX2/100-2", "SIP/telasip-gw/770555") in new
stack -- Called
telasip-gw/770555 -- SIP/telasip-gw-3091 is
ringing -- SIP/telasip-gw-3091 answered
IAX2/100-2 == Spawn extension (macro-dialout-trunk, s, 14) exited
non-zero on 'IAX2/100-2' in macro 'dialout-trunk' == Spawn extension
(from-internal, 770555, 1) exited non-zero on
'IAX2/100-2' -- Executing Macro("IAX2/100-2",
"hangupcall") in new stack -- Executing
ResetCDR("IAX2/100-2", "w") in new stack -- Executing
NoCDR("IAX2/100-2", "") in new stack -- Executing
Wait("IAX2/100-2", "5") in new stack -- Executing
Hangup("IAX2/100-2", "") in new stack == Spawn extension
(macro-hangupcall, s, 4) exited non-zero on 'IAX2/100-2' in macro
'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero
on 'IAX2/100-2' -- Hungup 'IAX2/100-2'
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[Asterisk-Users] TDM11B Hang up detection not working in France ?
Hello, my config : aah 2.6 (asterisk 1.2.4) , centos 4.2, 1 TDM11B (1 Fxo / 1 fxs ), 1 phone, 1 softphone I'm in France When someone from PSTN calls and hangs up before the call is answered, internal extension keeps ringing until timeout occurs. PSTN line keeps busy. Hangup detection doesn't work. I've played with different paremeters (callprogress, busydetect, busycount, hanguponpolarityswitch) without success. I've googled around and it seems this problem is specific to France. Is there any French people in this list that has a TDM11B that hangs up correctly ? regards Pascal OFFREDO ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] System Design
The DSL speed at the remote office is 1.5 to 6.0 Down and 384 to 608 up. The DSL does have a static IP address and it's pretty rock solid in regards to stability. Curious, why the huge range in numbers? I have 1.5mb/s down and 512kb/s up, it's always been that. Or do you mean you have 6.0mb/s down and 608kb/s up, but in testing sometimes the actual speed tests lower? Anyways, just curious. If you could keep the upload at 608 that'd be great. 384 is a tad on the low side, but even handling uncompressed calls you could handle 3-4 calls. Using compression will help out a lot, especially if you're using that link for non-voip purposes. You'll definitely need some kind of qos. I'd go with another box, preferably one that doesn't require cooling and has no hard drive. I'm just picky, but I hate having multiple points of failure. Another server thrown in could have a fan fail (either locking it up or burning up the processor) or have the hard drive fail (and your whole network is brought down until you take the qos server out of the way). That's another reason I run qos on the asterisk box (I'm cheap, and it's one less possible point of failure, and one less thing to plug into my ups). I'm guessing you're going to run some kind of nat? Whatever you run the nat server on, have that handle qos too. Actually, throw on a decent firewall too. Any low-end cpu should be able to handle the load no problem, heck a 486 would do (again, personally I'd look at a newer cpu that needs no fan to keep cool, feel free to put one on it, but you'll know if it konks out your network is a-ok). Would the remote office * need a couple of POTS lines to make those local calls? It all depends. How many local calls do you plan on making at a time? If generally you need four total incoming/outgoing calls via a local line (that's incoming and outgoing combined, not separate), but will very rarely need say, 5-6 or more, it may be cheaper to just get four lines and any time a fifth call needs to go out, make it as a long-distance call. 1.1cents/minute for a few calls will be cheaper than paying for that fifth or sixth line. Even if you have enough pstn lines to handle all local calls, I'd still have it setup to automatically let them make the call as a long-distance call, never know when that important call needs to be made. You can do the same for incoming calls btw, get a feature called Call Forward Busy and do NOT get call-waiting on the line. When someone calls in, and the line's busy, it'll forward to another number you have via voip (whether it's a local number, or a toll-free number, doesn't matter). Now on those incoming calls, you may get the callerid of the original caller, or the callerid of your regular line (since in effect it's calling your other number, forwarding it on). In fact, you can get this to simulate your own PRI with just a few cheap PSTN lines. It'd be setup something like this: 555-1000: If busy, forward to 555-1001 555-1001: If busy, forward to 555-1002 555-1002: If busy, forward to 555-1003 555-1003: If busy, forward to 555-1004 555:1004: If busy, forward to 555-2000 (a voip number) 555-2000: Unlimited inbound calls Actually, you may want 555-1000 to immediately forward to 555-2000, if bandwidth isn't a concern and the number you're forwarding to is a local call. In my case, there's no local voip providers and I have to forward to a toll-free number, so I would want to keep the calls on the pstn line. Other than the possible caller-id issue (callerid may be of your own pstn line, or of the caller), this setup should work fine. Once again thanks for all of your replies! They are definitely clearing things up for me. - Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Tanner Sent: Wednesday, March 08, 2006 6:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] System Design Lot of questions, lots of variables, but I'll touch base on a few things. 5-10 concurrent calls is hardly anything. A plain T1 will more than handle that, even at ulaw or alaw (non)compression. Throw in a decent codec, and 10 calls won't even put a dent in your T1. Heck, it'd handle all 20 users in your main office, and the 5 users in your remote office with G729, no problem. How reliable is the remote office's DSL connection? I'd make sure you have a static ip for it (dynamic ips are just slightly problematic, especially if you have slightly flaky service, coupled with a slightly flaky modem). If it's reliable, then just keep that. What's the connection speed? Need to know the upload and download. If it's ADSL, then the upload will be a fraction of the download, and will be the limiting factor. Since I don't know your specific setup, I can't tell you specifically what to do. I'll make some guesses though. Keep DSL. No need to use VPN just for asterisk. Make sure
Re: [Asterisk-Users] pap2 Dial plan
Hi thanks for the help .vocie mail problem has been fixed but the delay is still there i have changed Interdigit Long Timer =2 and Interdigit short Timer=1thanksGiridhar Bandi On 3/8/06, Filipe Mordhorst [EMAIL PROTECTED] wrote: You're almost right. The PAP2 has some features that are factory default. I don't remember the section in the web interface, but here's what you going to do: Find the section that contains a lot of features name with values like this *56 or *78. Erase all of them. Letting' this filled you'll not be able to implement your asterisk features, cause' they are conflicting with the (factory defaults) PAP2 commands. About the long time waiting for start to call, the problem is that the PAP2 waits 10 or 15 (I don't remember de default) seconds after a digit is pressed to start the "send" procedure. To change these settings, go to Regional/Control Timer Values/ Interdigit Long Times and change the value to any other (this is expressed in seconds). Hope it helps. Regards, Filipe Mordhorst Brazil-SC De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] Em nome de Giridhar Bandi Enviada em: terça-feira, 7 de março de 2006 14:47 Para: asterisk-users@lists.digium.com Assunto: [Asterisk-Users] pap2 Dial plan Hi i am using pap2 phone adaptors as clients to connect to asterisk server i am able to make calls but i cannot access voice mail using phone or start recording while call is in progress and when i place a call to local sip extension there is a long pause ( 15 sec ) before the call gets dialled i assume that the problem would be due to the dial plan in PAP2 if so please help me changing it thanks Giridhar Bandi ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] broken pipe, restart asterisk
hi i'm running asterisk since 2 weeks and sometimes it crashes reporting some ouch ... broken pipe error i wolud like to write a script shell that check if asterisk is correctly started and, if not, it restart it, can i do it? how? i'm using asterisk 1.2.4 on slackware 10.2 thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7940/60 SIP 8.2
- Original Message - From: Nabeel Jafferali [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, March 09, 2006 10:42 AM Subject: RE: [Asterisk-Users] 7940/60 SIP 8.2 So has anybody tried installing the new SIP version? It seems nobody has had luck with the 7970 and it's new SIP image and the description for the 7940/60 specifically says for CCM v5.0. Just downloaded it after your email and got it working on the first try. Give me a few minutes to write up the procedure. OK, I had a 7960 running SIP 7.4 and managed to upgrade to SIP 8.2. I haven't tried any of the new features, but can make and receive calls fine. Sweet, guess I'll give it a go. _ Mobilcom http://www.mobilcom.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SPA3000 and callerID
Hi, I'm using a Sipura SPA 3000 to receive calls from a PSTN line and to send them to Asterisk. But on Asterisk, I don't see the PSTN caller ID! I only have User ID of the SPA 3000 as caller number. The caller number is present on the PSTN line. I'm in france, maybe the SPA 3000 is not compatible? If you have an idea, please tell me! Thanks, Mickaël ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HW Echo Cancellers
Just a note: This vendor is selling cards with local side echo cancellation. Most of the cards that I purchased didn't have it. The 3 that I've purchased from him did. Two questions. One, why the need for local side echo cancellation? I thought you could just reverse the connection and it would now disable echo in the opposite direction? Just curious, I don't have a T1, and this is just based on what I've read. Two, is there any way to tell what cards have this option just by looking at them? I bought a large lot (40+) and intend to resell them, probably on ebay. I would like to know what extras they have or don't have, so I can list them appropriately. Thanks! Joseph Tanner Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7940/60 SIP 8.2
Yeah, this is the same procedure I went through with mine, worked like a charm, zero problems whatsoever... Anyone have any idea what if any the new features are of this firmware? Aaron Nabeel Jafferali wrote: So has anybody tried installing the new SIP version? It seems nobody has had luck with the 7970 and it's new SIP image and the description for the 7940/60 specifically says for CCM v5.0. Just downloaded it after your email and got it working on the first try. Give me a few minutes to write up the procedure. OK, I had a 7960 running SIP 7.4 and managed to upgrade to SIP 8.2. I haven't tried any of the new features, but can make and receive calls fine. I put the following on the TFTP server that the phone has always polled for configuration files: OS79XX.txt - containing the line P003-08-2-00, although it does not appear this file was even needed (see TFTP log below) SIPDefault.cnf - containing the line image_version: P0S3-08-2-00 POS3-08-2-00.loads - from the .cop file after renaming it .tar.gz and extracting using WinRAR POO3-08-2-00.sbn - same as previous POS3-08-2-00.sb2 - same as previous POO3-08-2-00.bin - same as previous, although it does not appear this file was even needed (see TFTP log below) In addition, I had my SIPMAC.cnf, RINGLIST.DAT and dialplan.xml - the same ones I was using with SIP 7.4. This is the TFTP log after the phone was rebooted: 09/03/2006 10:11 :TFTP Error from 192.168.1.60 requesting CTLSEP003094C2A192.tlv : File does not exist 09/03/2006 10:11 :TFTP Error from 192.168.1.60 requesting SEP003094C2A192.cnf.xml : File does not exist 09/03/2006 10:11 :Sending SIP.cnf to (192.168.1.60) 09/03/2006 10:11 :Sent SIP.cnf to (192.168.1.60), 6173 bytes 09/03/2006 10:11 :Sending SIPDefault.cnf to (192.168.1.60) 09/03/2006 10:11 :Sent SIPDefault.cnf to (192.168.1.60), 46 bytes 09/03/2006 10:11 :Sending P0S3-08-2-00.loads to (192.168.1.60) 09/03/2006 10:11 :Sent P0S3-08-2-00.loads to (192.168.1.60), 461 bytes 09/03/2006 10:11 :Sending P003-08-2-00.sbn to (192.168.1.60) 09/03/2006 10:11 :Sent P003-08-2-00.sbn to (192.168.1.60), 129644 bytes 09/03/2006 10:11 :TFTP Error from 192.168.1.60 requesting CTLSEP003094C2A192.tlv : File does not exist 09/03/2006 10:11 :TFTP Error from 192.168.1.60 requesting SEP003094C2A192.cnf.xml : File does not exist 09/03/2006 10:11 :Sending SIP.cnf to (192.168.1.60) 09/03/2006 10:11 :Sent SIP.cnf to (192.168.1.60), 6173 bytes 09/03/2006 10:11 :Sending SIPDefault.cnf to (192.168.1.60) 09/03/2006 10:11 :Sent SIPDefault.cnf to (192.168.1.60), 46 bytes 09/03/2006 10:11 :Sending P0S3-08-2-00.loads to (192.168.1.60) 09/03/2006 10:11 :Sent P0S3-08-2-00.loads to (192.168.1.60), 461 bytes 09/03/2006 10:11 :Sending P0S3-08-2-00.sb2 to (192.168.1.60) 09/03/2006 10:11 :Sent P0S3-08-2-00.sb2 to (192.168.1.60), 785338 bytes 09/03/2006 10:12 :Sending SIPDefault.cnf to (192.168.1.60) 09/03/2006 10:12 :Sent SIPDefault.cnf to (192.168.1.60), 46 bytes 09/03/2006 10:12 :Sending SIP.cnf to (192.168.1.60) 09/03/2006 10:12 :Sent SIP.cnf to (192.168.1.60), 6173 bytes 09/03/2006 10:12 :Sending RINGLIST.DAT to (192.168.1.60) 09/03/2006 10:12 :Sent RINGLIST.DAT to (192.168.1.60), 19 bytes 09/03/2006 10:12 :Sending dialplan.xml to (192.168.1.60) 09/03/2006 10:12 :Sent dialplan.xml to (192.168.1.60), 1046 bytes I hope this works for everyone else as well. Nabeel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk @ Home 2.6 Call hangs up
My guess, is nat problems. Just for fun, try dialing your inbound number from something not connected to that asterisk box, say a cellphone. I know you're using IAX and SIP, so you'd think you wouldn't run into a double-nat problem (nat going out, nat coming in), but you never know. I have odd issues pop up sometimes when I try calling from my asterisk box right back into it, and I don't even have any nat in the way. Do outgoing calls generally work fine? How do incoming calls work when dialing from an outside line? For the heck of it, try calling out normally, and use a cellphone (or whatever) to dial into the asterisk box. Can it handle an outgoing AND incoming call at the same time, as long as it's not calling itself? If incoming calls still fail, then look into nat issues. Perhaps you can permanently forward port 5060 or 5061 (whichever you use, probably 5060) to your asterisk box, see if that helps any. May need to forward ports 1000-2000 as well. Joseph Tanner On 3/9/06, Jerry Rasmussen [EMAIL PROTECTED] wrote: I have installed asterisk @ home 2.6. I am using a Telasip VOIP account. When I make outbound or inbound calls the calls seem to connect and then get hung up. I was wondering if there was something that I am misisng. I have tried several different sip.conf configurations. Here is what they are currently. telasip-gw context=telasip-in dtmfmode=rfc2833 fromuser=jrasxxx host=gw4.telasip.com insecure=very nat=yes secret=xyz type=peer username=jrasxxx 551212 context=from-pstn dtmfmode=rfc2833 host=gw4.telasip.com insecure=very nat=yes qualify=yes secret=xyz type=peer username=jrasxxx The odd thing is it worked once or twice then stopped. If anyone could shed some light it would be greatly apperciated. Here is what the asterisk output looks like: -- AGI Script fixlocalprefix completed, returning 0 -- Executing SetVar(IAX2/100-2, OUTNUM=770555) in new stack -- Executing Cut(IAX2/100-2, custom=OUT_2|:|1) in new stack -- Executing GotoIf(IAX2/100-2, 0?16) in new stack -- Executing Dial(IAX2/100-2, SIP/telasip-gw/770555) in new stack -- Called telasip-gw/770555 -- SIP/telasip-gw-3091 is ringing -- SIP/telasip-gw-3091 answered IAX2/100-2 == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'IAX2/100-2' in macro 'dialout-trunk' == Spawn extension (from-internal, 770555, 1) exited non-zero on 'IAX2/100-2' -- Executing Macro(IAX2/100-2, hangupcall) in new stack -- Executing ResetCDR(IAX2/100-2, w) in new stack -- Executing NoCDR(IAX2/100-2, ) in new stack -- Executing Wait(IAX2/100-2, 5) in new stack -- Executing Hangup(IAX2/100-2, ) in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'IAX2/100-2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/100-2' -- Hungup 'IAX2/100-2' ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 SIP - Displaying Time
On Thu, 2006-03-09 at 10:36 -0500, Nabeel Jafferali wrote: Is there a way to display the time of the 7960 running firmware 7.4? Im unable to find any information. Add the following to SIPDefault.cnf or SIPMAC.cnf: sntp_server: time.nrc.ca sntp_mode: unicast time_zone: EST You should of course change your NTP server and/or time zone. On my 7960 with 7.4 firmware, the time automagically disappears for some unknown reason. The phone still functions, but the time goes away until I reboot it. Not a big deal to me, so I have not investigated it further. -Greg ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Location of MeetMe Recordings
Gavin Adams wrote: In Asterisk 1.2.4 is love being able to recording conferences. However, using the default variables, the files are being written to /var/lib/asterisk/sounds instead of /var/spool/asterisk/meetme. If I change MEETME_RECORDINGFILE variable to something different in works, bit I lose the ability to define CONFNO as part of the file name, which is handy when sorting for users to review. I call meetme using (,r,) so the conference number is not defined yet. My /etc/asterisk/asterisk.conf file is set to point to /var/spool/asterisk for recording related bits, and voicemail and general recordings are being stored in the appropriate subdirectories. It's only meetme that is going to a different place. Gavin: It doesn't appear that you can do this by simply changing an option via the meetme command. If you are comfortable enough with c code, you can change the following line in app_meetme.c (it is line number 2247 in my copy) and then rebuild Asterisk: snprintf(recordingtmp, sizeof(recordingtmp), meetme-conf-rec-%s-%s, conf-confno, chan-uniqueid); change to snprintf(recordingtmp, sizeof(recordingtmp), %s/meetme/meetme-conf-rec-%s-%s, ast_config_AST_SPOOL_DIR, conf-confno, chan-uniqueid); I just tested and this does work on my system. Thanks, Mike Clark ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HW Echo Cancellers
You sure it's not the Zaptel hardware creating the DTMF issues? what Digium card you using? On 3/8/06, Darren Wright [EMAIL PROTECTED] wrote: Forget the orion.lots of DTMF problemstech support is not Terribly well spoken. Look for ANY of the 257* series... Just ebay for t1 echo -D -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Wednesday, March 08, 2006 2:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] HW Echo Cancellers Tellabs looks a little too up-scale for what I need :). $1k for a single port orion unit might be worth considering for really stubborn installs though. Why? they go for around $100.00 on eBay. What goes for $100 on eBay? Tellabs? or Orion? I can't find any Orion equipment on eBay. What model Tellabs am I looking for? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7940/60 SIP 8.2
Why can't I find an 8.2 on Ciscos site? the latest I can find is 7.5. Are you guys talking about SIP? On 3/9/06, Mailing List [EMAIL PROTECTED] wrote: - Original Message - From: Nabeel Jafferali [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, March 09, 2006 10:42 AM Subject: RE: [Asterisk-Users] 7940/60 SIP 8.2 So has anybody tried installing the new SIP version? It seems nobody has had luck with the 7970 and it's new SIP image and the description for the 7940/60 specifically says for CCM v5.0. Just downloaded it after your email and got it working on the first try. Give me a few minutes to write up the procedure. OK, I had a 7960 running SIP 7.4 and managed to upgrade to SIP 8.2. I haven't tried any of the new features, but can make and receive calls fine. Sweet, guess I'll give it a go. _ Mobilcom http://www.mobilcom.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA3000 and callerID
On Thu, 2006-03-09 at 17:16 +0100, Mickaël Cissé wrote: Hi, I'm using a Sipura SPA 3000 to receive calls from a PSTN line and to send them to Asterisk. But on Asterisk, I don't see the PSTN caller ID! I only have User ID of the SPA 3000 as caller number. The caller number is present on the PSTN line. I'm in france, maybe the SPA 3000 is not compatible? I can assure you that it is very compatible. In PSTN-Line in Advanced mode set PSTN CID For VoIP CID: = yes -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time
We had that problem for a while. You have to configure the ntp server in the phone so it'll pull the time otherwise it just randomly loses it. Aaron Greg Oliver wrote: On Thu, 2006-03-09 at 10:36 -0500, Nabeel Jafferali wrote: Is there a way to display the time of the 7960 running firmware 7.4? Im unable to find any information. Add the following to SIPDefault.cnf or SIPMAC.cnf: sntp_server: time.nrc.ca sntp_mode: unicast time_zone: EST You should of course change your NTP server and/or time zone. On my 7960 with 7.4 firmware, the time automagically disappears for some unknown reason. The phone still functions, but the time goes away until I reboot it. Not a big deal to me, so I have not investigated it further. -Greg ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SPA-3000 vs Linksys SPA3000
the Linksys and Sipura SPA-3000 are the same, just the plastic box is different - Original Message - From: John Jensen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, March 02, 2006 1:53 PM Subject: Re: [Asterisk-Users] Sipura SPA-3000 vs Linksys SPA3000 Hi Frank, Linksys bought Sipura some time ago (and Cisco owns Linksys btw). I'd say it's a pretty safe bet that it's the same box. I just recieved word from Sipura that the following products are being 'end-of-life'd ': * SPA841, EOL per December 2005 * SPA2002, Limited supply till Mid April and will be EOL by May 1st and replaced by PAP2T * SPA2100, Limited supply till Mid April and will be EOL by May 1st and replaced by SPA2102 * SPA3000, Limited supply till Mid April and will be EOL by May 1st and replaced by SPA3102 * RTP31P2, Limited supply till Mid May and will be EOL by June 1st and replaced by RTP300 * WRT54GP2, Limited supply till Mid March and will be EOL by April 1st and replaced by WRTP54G You might want to get hold of the SPA3102 if you can. /John [EMAIL PROTECTED] 03/02/06 1:43 pm Hallo! I had ordered a Sipura SPA-3000 in the UK, but the supplier turned out to be unreliable and never shipped. Yesterday I went looking for alternative suppliers and found the Linksys SPA3000 device. It's a different box, but the specs look very similar. Is this the same device? Has anyone used this Linksys SPA3000 successfully with Asterisk? Thanks, Frank ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time
On 3/9/06, Greg Oliver [EMAIL PROTECTED] wrote: On Thu, 2006-03-09 at 10:36 -0500, Nabeel Jafferali wrote: Is there a way to display the time of the 7960 running firmware 7.4? Im unable to find any information. Add the following to SIPDefault.cnf or SIPMAC.cnf: sntp_server: time.nrc.ca sntp_mode: unicast time_zone: ESTOn my 7960 with 7.4 firmware, the time automagically disappears for someunknown reason. The phone still functions, but the time goes awayuntil I reboot it.Not a big deal to me, so I have not investigated it further.-Greg I use anycast. Seems like I read something about directbroadcast not working in recent SIP versions. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7940/60 SIP 8.2
OK, I found it now, it's under the NONSIP link on Ciscos site. Acording to the docs it's meant only for Cisco Call Manager, does it work with Asterisk? On 3/9/06, C F [EMAIL PROTECTED] wrote: Why can't I find an 8.2 on Ciscos site? the latest I can find is 7.5. Are you guys talking about SIP? On 3/9/06, Mailing List [EMAIL PROTECTED] wrote: - Original Message - From: Nabeel Jafferali [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, March 09, 2006 10:42 AM Subject: RE: [Asterisk-Users] 7940/60 SIP 8.2 So has anybody tried installing the new SIP version? It seems nobody has had luck with the 7970 and it's new SIP image and the description for the 7940/60 specifically says for CCM v5.0. Just downloaded it after your email and got it working on the first try. Give me a few minutes to write up the procedure. OK, I had a 7960 running SIP 7.4 and managed to upgrade to SIP 8.2. I haven't tried any of the new features, but can make and receive calls fine. Sweet, guess I'll give it a go. _ Mobilcom http://www.mobilcom.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk @ Home 2.6 Call hangs up
Thanks for the response Joseph. It ended up that Telasip needed to make a change on there end. They needed to disable re-invites. BTW, I wanted to give a big plug for Telasip. I thought when I called they would simply tell me it was my problem and they did not support asterisk. This was not the case at all. I recieved promt friendly curtious service. These guys had my problem fixed within 15 min of sending them my log file. I cannot say enought good things about them right now. From: [EMAIL PROTECTED] on behalf of Joseph Tanner Sent: Thu 3/9/2006 11:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk @ Home 2.6 Call hangs up My guess, is nat problems. Just for fun, try dialing your inbound number from something not connected to that asterisk box, say a cellphone. I know you're using IAX and SIP, so you'd think you wouldn't run into a double-nat problem (nat going out, nat coming in), but you never know. I have odd issues pop up sometimes when I try calling from my asterisk box right back into it, and I don't even have any nat in the way. Do outgoing calls generally work fine? How do incoming calls work when dialing from an outside line? For the heck of it, try calling out normally, and use a cellphone (or whatever) to dial into the asterisk box. Can it handle an outgoing AND incoming call at the same time, as long as it's not calling itself? If incoming calls still fail, then look into nat issues. Perhaps you can permanently forward port 5060 or 5061 (whichever you use, probably 5060) to your asterisk box, see if that helps any. May need to forward ports 1000-2000 as well. Joseph Tanner On 3/9/06, Jerry Rasmussen [EMAIL PROTECTED] wrote: I have installed asterisk @ home 2.6. I am using a Telasip VOIP account. When I make outbound or inbound calls the calls seem to connect and then get hung up. I was wondering if there was something that I am misisng. I have tried several different sip.conf configurations. Here is what they are currently. telasip-gw context=telasip-in dtmfmode=rfc2833 fromuser=jrasxxx host=gw4.telasip.com insecure=very nat=yes secret=xyz type=peer username=jrasxxx 551212 context=from-pstn dtmfmode=rfc2833 host=gw4.telasip.com insecure=very nat=yes qualify=yes secret=xyz type=peer username=jrasxxx The odd thing is it worked once or twice then stopped. If anyone could shed some light it would be greatly apperciated. Here is what the asterisk output looks like: -- AGI Script fixlocalprefix completed, returning 0 -- Executing SetVar(IAX2/100-2, OUTNUM=770555) in new stack -- Executing Cut(IAX2/100-2, custom=OUT_2|:|1) in new stack -- Executing GotoIf(IAX2/100-2, 0?16) in new stack -- Executing Dial(IAX2/100-2, SIP/telasip-gw/770555) in new stack -- Called telasip-gw/770555 -- SIP/telasip-gw-3091 is ringing -- SIP/telasip-gw-3091 answered IAX2/100-2 == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'IAX2/100-2' in macro 'dialout-trunk' == Spawn extension (from-internal, 770555, 1) exited non-zero on 'IAX2/100-2' -- Executing Macro(IAX2/100-2, hangupcall) in new stack -- Executing ResetCDR(IAX2/100-2, w) in new stack -- Executing NoCDR(IAX2/100-2, ) in new stack -- Executing Wait(IAX2/100-2, 5) in new stack -- Executing Hangup(IAX2/100-2, ) in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'IAX2/100-2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/100-2' -- Hungup 'IAX2/100-2' ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T1 GSM or DECT ? Searching for a complete microcell solution
Hi gentlemen :-) I am searching a radio base GSM or DECT with high power for long range, and the terminal units (handy). This equipment must be connected to a T1 port from an Asterisk. The number of simultaneous channels must be 7 to 10. Do you know a manufacturer with nice equipments at correct price ? Thanks in Advance. Best Regards, Francois BERGERET, France. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7940/60 SIP 8.2
The image is located in the non-sip section, go figure. They're harping that this is for their new sip ccm... Aaron C F wrote: Why can't I find an 8.2 on Ciscos site? the latest I can find is 7.5. Are you guys talking about SIP? On 3/9/06, Mailing List [EMAIL PROTECTED] wrote: - Original Message - From: Nabeel Jafferali [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, March 09, 2006 10:42 AM Subject: RE: [Asterisk-Users] 7940/60 SIP 8.2 So has anybody tried installing the new SIP version? It seems nobody has had luck with the 7970 and it's new SIP image and the description for the 7940/60 specifically says for CCM v5.0. Just downloaded it after your email and got it working on the first try. Give me a few minutes to write up the procedure. OK, I had a 7960 running SIP 7.4 and managed to upgrade to SIP 8.2. I haven't tried any of the new features, but can make and receive calls fine. Sweet, guess I'll give it a go. _ Mobilcom http://www.mobilcom.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] res_musiconhold.c: Only wrote -1 of 640 bytes to pipe // no queue music
Hi all, I setup a new asterisk machine and all is working fine for maybe 10 days Today there was the problem that nobody can hear the music during you are waiting in a/the queue/s. Only silence was the answer. Then I want to shutdown asterisk with stop now and nothing happends After killing asterisk and restarting everything is ok but I'm realy interested in what happends . Maybe someone has an idea Asterisk 1.2.4, kernel 2.6.14, zaptel-1.2.4,... This is what I see in the log: -- snip - Mar 9 15:30:26 DEBUG[31301] res_musiconhold.c: Only wrote -1 of 640 bytes to pipe Mar 9 15:30:26 DEBUG[31301] res_musiconhold.c: Only wrote -1 of 640 bytes to pipe Mar 9 15:30:26 DEBUG[31301] res_musiconhold.c: Only wrote -1 of 640 bytes to pipe -- snip - Thanks for any help or ideas Best, Morel -- - Morel Mosolff - Network-/System-Technician - NATIVE INSTRUMENTS GmbH - [EMAIL PROTECTED] - Schlesische Strasse 28- http://www.native-instruments.de/ - D-10997 Berlin- Tel. +49-30-61 10 35-1712 - Germany - Fax +49-30-61 10 35-2712 --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Festival tts
No, I did not install Festival, but I saw that the text2wave module is in the usr/bin directory. I'm running RH Ent 2.4 kernel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert La Ferla Sent: Thursday, March 09, 2006 10:17 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Festival tts Steven [EMAIL PROTECTED] wrote: Hi I have installed Festival on the same box as asterisk and followed the instructions to integrate it with asterisk. Festival seems to work fine on its own performing text to speech from the command line or via a file. Asterisk answers the call but there is no speech. I can see no errors in the Festival log file I asked the same question to this list a while back but got no replies. What OS are you using? How did you install Festival? What version of *? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7940/60 SIP 8.2
This should get you where you need to go as long as you have a login: http://www.cisco.com/cgi-bin/tablebuild.pl/ip-7900ser On 3/9/06, C F [EMAIL PROTECTED] wrote: Why can't I find an 8.2 on Ciscos site? the latest I can find is 7.5.Are you guys talking about SIP?On 3/9/06, Mailing List [EMAIL PROTECTED] wrote: - Original Message - From: Nabeel Jafferali [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, March 09, 2006 10:42 AM Subject: RE: [Asterisk-Users] 7940/60 SIP 8.2 So has anybody tried installing the new SIP version? It seems nobody has had luck with the 7970 and it's new SIP image and the description for the 7940/60 specifically says for CCM v5.0. Just downloaded it after your email and got it working on the first try. Give me a few minutes to write up the procedure. OK, I had a 7960 running SIP 7.4 and managed to upgrade to SIP 8.2. I haven't tried any of the new features, but can make and receive calls fine. Sweet, guess I'll give it a go. _ Mobilcom http://www.mobilcom.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time
This issue has been fixed in SIP firmware 7.5 Omar A. Sabek On 3/9/06, Nathan Bowyer [EMAIL PROTECTED] wrote: On 3/9/06, Greg Oliver [EMAIL PROTECTED] wrote: On Thu, 2006-03-09 at 10:36 -0500, Nabeel Jafferali wrote: Is there a way to display the time of the 7960 running firmware 7.4? Im unable to find any information. Add the following to SIPDefault.cnf or SIPMAC.cnf: sntp_server: time.nrc.ca sntp_mode: unicast time_zone: EST On my 7960 with 7.4 firmware, the time automagically disappears for some unknown reason. The phone still functions, but the time goes away until I reboot it. Not a big deal to me, so I have not investigated it further. -Greg I use anycast. Seems like I read something about directbroadcast not working in recent SIP versions. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Getting to the last old voicemail message
I made a small change to apps/app_voicemail.c to permit circular
navigation when listening to messages. If you are at the first message,
and press 4, it takes you to the last message. If you are already at
the last message and press 6, it takes you to the first message. I
did a quick test and it seems to work. If you apply it and find any
problems, please let me know and I'll fix it.
I don't have a diff but here's the code in function vm_execmain():
case '4':
if (vms.curmsg) {
vms.curmsg--;
cmd = play_message(chan, vmu, vms);
} else {
/* cmd = ast_play_and_wait(chan, vm-nomore); */
vms.curmsg = vms.lastmsg;
cmd = play_message(chan, vmu, vms);
}
break;
case '6':
if (vms.curmsg vms.lastmsg) {
vms.curmsg++;
cmd = play_message(chan, vmu, vms);
} else {
/* cmd = ast_play_and_wait(chan, vm-nomore); */
vms.curmsg = 0;
cmd = play_message(chan, vmu, vms);
}
break;
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[Asterisk-Users] Oneway voice
Hi allI have installed AAH 2.6created extension,and created Trunkcreated outbound routingiam able to make calls outand configured incoming, also working finewith the extension I have problem hereI ahve extension sitting in same network where the AAH installedMy provider support canreinvite=yeswhen iam making calls, its not consuming any b/wand voice quality is good in sip_additional.confi have made in extension also canreinvite=yesanother extension sitting another Countryand he is behind nathere also made extension caninvite=yesi get one way Voice, later i have made the extension config( out side country extension) canreinvite=nothe voice quality is good, but its taking 128Kb b/whow can i resolve this problem using g729 codecand save b/w thanks any suggestionsram ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Merlin Magix Integration
Do you have a timeout set somewhere? Try Set(TIMEOUT(digit)=3), and/or
Set(TIMEOUT(response)=5)
On 3/9/06, Darren Ellis [EMAIL PROTECTED] wrote:
Hi List,
Merlin Magix hardware v02
I'm trying to get asterisk to act as a voicemail server for a lucent
merlin magix PBX that we purchased used. We have 4 FXO channels between
the two PBXs on a Sangoma A200 card. The 770 dialgroup is working
properly, in that calls to 770 are answered by Asterisk. The magix is
sending mode codes in the format #XX#XXX#, where the 2nd block of digits
is the calling extension. I'm stripping off the unneeded pound signs
and digits, and calling voicemailmain. The problem I'm having is that
the asterisk is starting to play vm-password and then interrupts
immediately and errors with an incorrect password. Then it works
normally. Below is the relevant asterisk config and the asterisk log.
Zaptel is configured to start inbound calls in the inbound context. The
voicemail accounts and sip accounts are all in the default context.
Asterisk log
-- Starting simple switch on 'Zap/3-1'
Mar 9 10:26:35 NOTICE[4211]: chan_zap.c:6063 ss_thread: Got event 18
(Ring Begin)...
-- Executing Answer(Zap/3-1, ) in new stack
-- Executing WaitExten(Zap/3-1, 1) in new stack
== CDR updated on Zap/3-1
-- Executing NoOp(Zap/3-1, #00#219#) in new stack
-- Executing Set(Zap/3-1, [EMAIL PROTECTED]) in new stack
-- Executing NoOp(Zap/3-1, [EMAIL PROTECTED]) in new stack
-- Executing VoiceMailMain(Zap/3-1, [EMAIL PROTECTED]) in new stack
-- Playing 'vm-password' (language 'en')
-- Incorrect password '' for user '219' (context = default)
-- Playing 'vm-incorrect' (language 'en')
-- Playing 'vm-password' (language 'en')
||| Caller hangs up here |||
Mar 9 10:26:41 WARNING[4211]: app_voicemail.c:4998 vm_authenticate:
Unable to read password
-- Hungup 'Zap/3-1'
extensions.conf
[inbound]
exten = s,1,Answer()
exten = s,2,WaitExten(1) ; Allow time for mode code digits to
come across
; The following extensions grab the mode code
; coming from the Avaya PBX and route the
; call appropriately via the Voicemail()
; and VoiceMailMain() apps.
;
; someone pressed vmail check
; #00#243#
exten = _#XX#XXX#,1,noop(${EXTEN})
exten = _#XX#XXX#,2,Set(CVAR=${EXTEN:4:[EMAIL PROTECTED])
exten = _#XX#XXX#,3,NoOp(${CVAR})
exten = _#XX#XXX#,4,VoicemailMain(${CVAR})
;exten = _#XX#XXX#,2,VoicemailMain(${EXTEN:4:[EMAIL PROTECTED])
exten = _#XX#XXX#,5,Hangup()
As can be seen, I've tried calling voicemailmain with the ${EXTEN:4:3}
digit stripping as part of the command, and also I've tried moving the
digit stripping to a variable.
I'd very much appreciate any help you folks can offer.
Thanks much.
Darren Ellis
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Re: [Asterisk-Users] 7940/60 SIP 8.2
Does that mean that since CCM supports SIP, Cisco will just make sure that their SIP images work with CCM? On 3/9/06, Aaron Daniel [EMAIL PROTECTED] wrote: The image is located in the non-sip section, go figure. They're harping that this is for their new sip ccm... Aaron C F wrote: Why can't I find an 8.2 on Ciscos site? the latest I can find is 7.5. Are you guys talking about SIP? On 3/9/06, Mailing List [EMAIL PROTECTED] wrote: - Original Message - From: Nabeel Jafferali [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, March 09, 2006 10:42 AM Subject: RE: [Asterisk-Users] 7940/60 SIP 8.2 So has anybody tried installing the new SIP version? It seems nobody has had luck with the 7970 and it's new SIP image and the description for the 7940/60 specifically says for CCM v5.0. Just downloaded it after your email and got it working on the first try. Give me a few minutes to write up the procedure. OK, I had a 7960 running SIP 7.4 and managed to upgrade to SIP 8.2. I haven't tried any of the new features, but can make and receive calls fine. Sweet, guess I'll give it a go. _ Mobilcom http://www.mobilcom.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RES: [Asterisk-Users] DTFM or FSK
Thanks. Its an analog card and I really didnt find anything with good explanations (at least no for me). The problem is that the people who give me support for my actual PABX, asked me the standard tone signaling. Im trying to get in my actual PABX from asterisk through the PABX fxo port and I want to do it in a transparent way to the user. The guys form the PABX support said they will try to catch the incoming digits to make the PABX internal routing decision and to do that, they need to know the signaling standard used by my Digium card (TDM400P) Maybe Im just saying bullshit. If that is the case, please tell me, so I can have a god talk with that support team. Thanks. Regards, Filipe Mordhorst Brazil-SC De: [EMAIL PROTECTED] [mailto:asterisk-users-bounces@lists.digium.com] Em nome de Wai Wu Enviada em: quinta-feira, 9 de março de 2006 12:11 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: RE: [Asterisk-Users] DTFM or FSK Can you be more specific? All digital cards (regardless of manufacture) use the same modulation, it is a standard and you can probably google it. smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7940/60 SIP 8.2
I haven't been through everything line by line but I did notice a new Security Configuration where you can set an Encrypt Key _ Mobilcom http://www.mobilcom.net - Original Message - From: Aaron Daniel [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 09, 2006 11:20 AM Subject: Re: [Asterisk-Users] 7940/60 SIP 8.2 Yeah, this is the same procedure I went through with mine, worked like a charm, zero problems whatsoever... Anyone have any idea what if any the new features are of this firmware? Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic
Ladies and Gentlemen, this is way, way, way off topic at this point. Douglas's point was raised and a valid counter point was offered, let's please just move on. No amount of additional discussion is going to add this feature into Asterisk. If this is a deal breaker for you, Douglas, you are aware of other solutions to try. What other productive conversation can there be from this? Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Collins Sent: Wednesday, March 08, 2006 7:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Oh this is bad bindaddr and rtp traffic For the record, Douglas is correct on this point of enterprise-grade being on ABE: http://www.digium.com/index.php?menu=product_categorycategory=software Copied and pasted right from the website, it says: Asterisk Business Edition(tm) Digium(tm), the leader in open source telephony, offers Asterisk Business Edition, an enterprise-grade version of its acclaimed open source PBX for the Linux operating system. This version provides tested reliability of critical functions and features, tailored for small- and medium-sized business applications. Now, as to the debate about what is and is not available in an enterprise-grade product, I will have to defer to those who actually use Asterisk in the enterprise - I only use it for tinkering and minor voice broadcasting campaigns. -MC -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Wednesday, March 08, 2006 7:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Oh this is bad bindaddr and rtp traffic I can't be bothered looking for the link right now, but it's definitely stated somewhere on Digium's website. -Original Message- From: Alexander Lopez [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 07, 2006 3:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Oh this is bad bindaddr and rtp traffic To retort, Digium has ever to my knowledge, stamped an 'Enterprise Grade' mark on the product. If you are worried about a single point of failure you may want to replace your toaster. Asterisk is missing a 'few features' no doubt about it, but it is open source, it will be a welcome addition if you would like to add multi-homing support in, might as well do media multi-homing with call diversity. This will definably be a non-trivial re-architecture of the core. The 'missing a few features' way of thinking is what has made Asterisk what it is today. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Tuesday, March 07, 2006 11:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Oh this is bad bindaddr and rtp traffic Pardon my candour, but for a product Digium calls 'enterprise grade' it sure seems to be missing a few features. -Original Message- From: Alexander Lopez [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 07, 2006 9:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Oh this is bad bindaddr and rtp traffic Asterisk does not like multiple interfaces in the way you are configured. You can either: A) use the bindaddr in the sip.conf to limit where the packsge come and go. B) use an outside traffic manager Look up the archives, kpf explained why this would not work, as asterisk can't do load balancing at this time -Original Message- From: Robert Webb [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk- [EMAIL PROTECTED] Sent: 3/7/06 11:27 AM Subject: Re: [Asterisk-Users] Oh this is bad bindaddr and rtp traffic On Tue, 7 Mar 2006 09:12:25 -0700 Douglas Garstang [EMAIL PROTECTED] wrote: I have a configuration where RTP traffic is going out interface pub0, and coming back into through pub1. I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an shows: udp0788 0.0.0.0:50600.0.0.0:* which means that Asterisk is listening on all addresses (on all interfaces?). Anyway, when the RTP traffic comes back in on interface pub0, Asterisk does nothing with it. A 'rtp debug' shows it's receiving the RTP packets, it just seems it does nothing with them. Anyone seen this? Doug. I thought all RTP was controlled through rtp.conf and only the SIP traffic was controlled through SIP.conf. I am not sure what settings, beside the RTP port range, you can out into the rtp.conf though. Robert ___ --Bandwidth and Colocation provided by Easynews.com --
Re: [Asterisk-Users] T1 GSM or DECT ? Searching for a complete microcell solution
Please contact Globetel Communications http://www.globetel.net/ +1 954 241 0590. they have division that handles DECT solution that can interoperate with asterisk On 3/9/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi gentlemen:-)I am searching a radio base GSM or DECT with high power for long range, andthe terminal units (handy).This equipment must be connected to a T1 port from an Asterisk.The number of simultaneous channels must be 7 to 10. Do you know a manufacturer with nice equipments at correct price ?Thanks in Advance.Best Regards,Francois BERGERET,France.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HW Echo Cancellers
Joseph Tanner wrote: Just a note: Two questions. One, why the need for local side echo cancellation? I thought you could just reverse the connection and it would now disable echo in the opposite direction? Just curious, I don't have a T1, and this is just based on what I've read. I have E.C. turned off in Asterisk, I want the card to have the ability to cancel in both directions. Two, is there any way to tell what cards have this option just by looking at them? I bought a large lot (40+) and intend to resell them, probably on ebay. I would like to know what extras they have or don't have, so I can list them appropriately. If you've looked at the picture in the ebaY auction, you should notice the daughter board. Without the daughter board, the area is quite absent of chips. The 1 board that I have without the daughter board and the 3 with had send side E.C. You can verify by powering a unit up and scrolling though the options until you see 38. If there is no 38, then the board doesn't have send side E.C. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AMD64 x2 and asterisk 1.2.4 not hearing demo-congrats
I have an AMD64 x2 and running asterisk 1.2.4 When I call in to the dialplan all I have is: exten = 11,1,Playback(demo-congrats) exten = 11,n,Hangup The console is showing the demo-congrats playing but no audio. I can call phone to phone and hear audio just fine. Is there an issue with 64 bits and audio files? Is there a special compile to be done or something? Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HW Echo Cancellers
Ditto on our installation of an Orion solution here in South Africa! works like a charm.CheersRobOn 09/03/06, Mike Clark [EMAIL PROTECTED] wrote:Darren Wright wrote:Forget the orion.lots of DTMF problemstech support is not Terribly well spoken.Look for ANY of the 257* series...Just ebay for t1 echo-DDo *not* forget the Orion. We have 3 in place now working beautifully. Can't speak on Darren's problems, but our units have all installed inless than 5 minutes, totally eradicated echo, have had zero DTMF issuesand Orion has been helpful when needed (though we have not needed much tech support).Mike Clark___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RES: [Asterisk-Users] pap2 Dial plan
Try to change the Short Timer field back to the default value. If this doesnt help either, use the pound key from your telephone key pad right after the last digit is pressed, this will make the pap2 start the send procedure. Interdigit long timer is the right field to change for the problem you described. If after all of this you still having the problem then probably it isnt paps fault =) Please post if you find out something new. Regards, Filipe Mordhorst Joinville - SC - Brasil De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Giridhar Bandi Enviada em: quinta-feira, 9 de março de 2006 13:06 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [Asterisk-Users] pap2 Dial plan Hi thanks for the help . vocie mail problem has been fixed but the delay is still there i have changed Interdigit Long Timer =2 and Interdigit short Timer=1 thanks Giridhar Bandi smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Festival tts
On 3/9/06, Adam Robins [EMAIL PROTECTED] wrote: Can someone tell me what I'm doing wrong here? I'm trying this from the command prompt. # echo Hello World | /usr/bin/text2wave -scale 1.5 -F 000 -o /tmp/1141915933.wav rateconv: failed to convert from 16000 to 0 doing v # I think your problem is that you are using -F 000 when you should be using -F 8000 (8KHz, not 0KHz). Give that a try and I bet it will work. -Rusty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7940/60 SIP 8.2
Yeah, I noticed that too, and there's now five call managers in the configuration you can set... not really sure how that'll help us in the asterisk community ;) but guess we'll find out soon enough... As far as I can tell, there's really no benefit to having this, other than maybe a few bug fixes. Aaron Mailing List wrote: I haven't been through everything line by line but I did notice a new Security Configuration where you can set an Encrypt Key _ Mobilcom http://www.mobilcom.net - Original Message - From: Aaron Daniel [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 09, 2006 11:20 AM Subject: Re: [Asterisk-Users] 7940/60 SIP 8.2 Yeah, this is the same procedure I went through with mine, worked like a charm, zero problems whatsoever... Anyone have any idea what if any the new features are of this firmware? Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7940/60 SIP 8.2
I just started looking at the differences between 8.2 and 7.5. In addition to the new Security Configuration, there is also compatibility added for provisioning the phone to CCM. The firmware appears to be working fine. Also, the copywright date range has been changed to 2000-2006. Omar A. Sabek On 3/9/06, C F [EMAIL PROTECTED] wrote: Does that mean that since CCM supports SIP, Cisco will just make sure that their SIP images work with CCM? On 3/9/06, Aaron Daniel [EMAIL PROTECTED] wrote: The image is located in the non-sip section, go figure. They're harping that this is for their new sip ccm... Aaron C F wrote: Why can't I find an 8.2 on Ciscos site? the latest I can find is 7.5. Are you guys talking about SIP? On 3/9/06, Mailing List [EMAIL PROTECTED] wrote: - Original Message - From: Nabeel Jafferali [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, March 09, 2006 10:42 AM Subject: RE: [Asterisk-Users] 7940/60 SIP 8.2 So has anybody tried installing the new SIP version? It seems nobody has had luck with the 7970 and it's new SIP image and the description for the 7940/60 specifically says for CCM v5.0. Just downloaded it after your email and got it working on the first try. Give me a few minutes to write up the procedure. OK, I had a 7960 running SIP 7.4 and managed to upgrade to SIP 8.2. I haven't tried any of the new features, but can make and receive calls fine. Sweet, guess I'll give it a go. _ Mobilcom http://www.mobilcom.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM11B Hang up detection not working in France ?
PascalHere in South Africa we encountered a simialr problem and wrote a patch that has been incorporated into Asterisk 1.2.x , what we do here is:add this to your zapata.conf For Cape Town: busydetect=yes busycount=4 busypattern=500,500 callprogress=no For Johannesburg: busydetect=yes busycount=2 busypattern=2500,500 callprogress=noIf polarityswitch is not an option for you in France and busypattern is then make a recording of the busy tone and put it through something like Audacity where you can measure the tone and silence and then set the busy pattern as in examples above. the busypattern is in millisecond. cheersRobOn 09/03/06, Pascal OFFREDO [EMAIL PROTECTED] wrote: Hello,my config : aah 2.6 (asterisk 1.2.4) , centos 4.2, 1 TDM11B (1 Fxo / 1fxs ), 1 phone, 1 softphoneI'm in FranceWhen someone from PSTN calls and hangs up before the call is answered,internal extension keeps ringing until timeout occurs. PSTN line keeps busy. Hangup detection doesn't work.I've played with different paremeters (callprogress, busydetect,busycount, hanguponpolarityswitch) without success.I've googled around and it seems this problem is specific to France. Is there any French people in this list that has a TDM11B that hangs upcorrectly ?regardsPascal OFFREDO___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international.Téléchargez sur http://fr.messenger.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Festival tts
I figured it out. It should read: # echo Hello World | /usr/bin/text2wave -scale 1.5 -F 8000 -o /tmp/1141915933.wav The 8 was missing in front of the 000'. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Robins Sent: Thursday, March 09, 2006 12:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Festival tts No, I did not install Festival, but I saw that the text2wave module is in the usr/bin directory. I'm running RH Ent 2.4 kernel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert La Ferla Sent: Thursday, March 09, 2006 10:17 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Festival tts Steven [EMAIL PROTECTED] wrote: Hi I have installed Festival on the same box as asterisk and followed the instructions to integrate it with asterisk. Festival seems to work fine on its own performing text to speech from the command line or via a file. Asterisk answers the call but there is no speech. I can see no errors in the Festival log file I asked the same question to this list a while back but got no replies. What OS are you using? How did you install Festival? What version of *? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T1 card Ali
Hi I have 2 port T1 card in asterisk server. I am facing following problem, Span 1 is connected to em circuit with wink-start Span 2 is connectecd to em circuit with feature group d Is I activate the span two then I faced the following problem. 1 call come on channel 26 then all of a sunnden asterisk try to run the automated menu on different channel (i.e 31) then we get the engage tone. I dont understand why asterisk switch the channel. Do any one have installed T1 circuit with feature group d on asterisk.. I have open the case with Digium but they dont have any answer for it. Thanks Ali ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G729, G729 annex A or G729 annex B?
Hello Some questions about codecs.. What's the difference between the this codecs? Which is used by asterisk? Thanks Juan Salas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Predictive Dialer
Hello all, I have a client interested in GnuDialer. My question is: Is there anyone on this list who has been using GnuDialer and I was wondering if you would be willing to share your experiences with it. Thank You Adam ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MixMonitor Problems -- sssshh, don't be too loud
It looks like there is lots of discussion already going on about it at http://bugs.digium.com/view.php?id=6457 On 3/7/06, Matt Riddell [NZ] [EMAIL PROTECTED] wrote: The only catalyst to getting it fixed will be if someone posts a bug entry with full details on bugs.digium.com If you do, post again here with the ID and discussion and testing can continue there. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chinaroby VOIP phones? SECOND TIME!
Hi all, Do anyone have experience www.Chinaroby.com VOIP phones? I am very interestedfor models:PY-60 and PB-35 Phones. Good or bad experience with sip and IAX2, please comment. I did not find any comment on google Regards Darko Sundek eLink Group Kotor-Montenegro ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] TDM11B Hang up detection not working in France ?
Hi Pascal ! France is not more difficult than other country. This is one of my channels behind France Telecom : usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=11 ; definitive level for no loss -2 dB txgain=0 group=1 callgroup=1 pickupgroup=1 immediate=no adsi=yes busydetect=yes busycount=3 busypattern=500,500 signalling = fxs_ks callerid = asreceived amaflags = documentation context=WHAT_YOU_WANT channel = 6; my current channel number for this setting I hope this could help you and some other french guys. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Pascal OFFREDO Envoyé : jeudi 9 mars 2006 16:59 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] TDM11B Hang up detection not working in France ? Hello, my config : aah 2.6 (asterisk 1.2.4) , centos 4.2, 1 TDM11B (1 Fxo / 1 fxs ), 1 phone, 1 softphone I'm in France When someone from PSTN calls and hangs up before the call is answered, internal extension keeps ringing until timeout occurs. PSTN line keeps busy. Hangup detection doesn't work. I've played with different paremeters (callprogress, busydetect, busycount, hanguponpolarityswitch) without success. I've googled around and it seems this problem is specific to France. Is there any French people in this list that has a TDM11B that hangs up correctly ? regards Pascal OFFREDO ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7940/60 SIP 8.2
I believe they've done that the entire time. I've never known them to be real supportive of competing third party solutions. _ Mobilcom http://www.mobilcom.net - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 09, 2006 12:23 PM Subject: Re: [Asterisk-Users] 7940/60 SIP 8.2 Does that mean that since CCM supports SIP, Cisco will just make sure that their SIP images work with CCM? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7940/60 SIP 8.2
Quoting Mailing List [EMAIL PROTECTED]: I believe they've done that the entire time. I've never known them to be real supportive of competing third party solutions. They support third-party partners such as Broadsoft. This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7940/60 SIP 8.2
On Thu, 2006-03-09 at 13:11 -0600, Shane Young wrote: Quoting Mailing List [EMAIL PROTECTED]: I believe they've done that the entire time. I've never known them to be real supportive of competing third party solutions. They support third-party partners such as Broadsoft. Broadsoft is entirely SIP - just like the channel in Asterisk. They utilize the Cisco XML features just like anyone else could though. -Greg ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] news-reading question
Is there some way I can follow this list from a newsgroup?? Is this the same as the gmane group gmane.comp.telephony.pbx.asterisk.user ?? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Oneway voice
If your connection to the internet is being nated you may need to add this entry to your sip.conf externip=210.x.x.x From: [EMAIL PROTECTED] on behalf of ram Sent: Thu 3/9/2006 12:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Oneway voice Hi all I have installed AAH 2.6 created extension, and created Trunk created outbound routing iam able to make calls out and configured incoming, also working fine with the extension I have problem here I ahve extension sitting in same network where the AAH installed My provider support canreinvite=yes when iam making calls, its not consuming any b/w and voice quality is good in sip_additional.conf i have made in extension also canreinvite=yes another extension sitting another Country and he is behind nat here also made extension caninvite=yes i get one way Voice, later i have made the extension config( out side country extension) canreinvite=no the voice quality is good, but its taking 128Kb b/w how can i resolve this problem using g729 codec and save b/w thanks any suggestions ram winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
