[Asterisk-Users] ATA in MGCP sometimes dropping calls
Hello, I`m using a bunch of ATA-186 with MGCP firmware, and users are complaining that sometimes, an avarage about one in 17-20 calls when they try to do a supervised transfer via FLASH, the calling party is dropped, or that could also happen when the press FLASH and then dial an extension to transfer to but it is busy, and when they return to the first call it is dropped. I have not been able to reproduce this and get some mgcp debug messages. I`ve tried the 2.15 and 3.0 firmware versions. Anybody else experincing this? Thanks -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Vegastream 50 FXO with Asterisk
I have a Vega 50 BRI working without any of the issues you mentioned, the dual SIP registrations is normal for most multi-line boxes enabled split users. Rgds, Adam -Original Message-From: Glenn Dalgliesh [mailto:[EMAIL PROTECTED]Sent: 05 February 2004 20:11To: [EMAIL PROTECTED]Subject: [Asterisk-Users] Vegastream 50 FXO with Asterisk Anyone have any experienceconfiguringVegaStream's with Asterisk. Ihave run into afew of questions. 1. It appear that after turning on registrations I am seeing two request for registration per linesip:[EMAIL PROTECTED]sip:[EMAIL PROTECTED]What is purpose and how do I handle this?2. DTMF btw Asterisk and the Unit I was unable to get rfc2833 to work successfully with inbound or outbound DTMF. Is this a known issue? 3. How is the best way to deal with dialout and selecting a free channel on the VegaStream Any general suggestions/experiences with regard to configuring a VegaStream withasteriskwould be appricated.Thanks * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person
[Asterisk-Users] Re: Execute command in shell
Marc == Marc Fargas [EMAIL PROTECTED] writes: Marc I've seen its possible to use the System applications, but what Marc about passing arguments to the command ? A quick look at app_system.c shows that it just passes the string unaltered to system(3). So, running man 3 system will show exactly what system(3) does: system() executes a command specified in string by calling /bin/sh -c string, and returns after the command has been completed. As such, System(command arg1 arg2 etc) should do what you want. -JimC PS No, I didn't know that before I looked at the src PPS Except for what was in the system(3) manpage ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vegastream 50 FXO with Asterisk
Hi Adam, Could you show us your configs on Asterisk and on Vega so everyone on the list can have a guide to get Vega working with Asterisk? Thanks! David - Original Message - From: Low, Adam To: '[EMAIL PROTECTED]' Sent: Friday, February 06, 2004 12:47 AM Subject: RE: [Asterisk-Users] Vegastream 50 FXO with Asterisk I have a Vega 50 BRI working without any of the issues you mentioned, the dual SIP registrations is normal for most multi-line boxes enabled split users. Rgds, Adam -Original Message-From: Glenn Dalgliesh [mailto:[EMAIL PROTECTED]Sent: 05 February 2004 20:11To: [EMAIL PROTECTED]Subject: [Asterisk-Users] Vegastream 50 FXO with Asterisk Anyone have any experienceconfiguringVegaStream's with Asterisk. Ihave run into afew of questions. 1. It appear that after turning on registrations I am seeing two request for registration per linesip:[EMAIL PROTECTED]sip:[EMAIL PROTECTED]What is purpose and how do I handle this?2. DTMF btw Asterisk and the Unit I was unable to get rfc2833 to work successfully with inbound or outbound DTMF. Is this a known issue? 3. How is the best way to deal with dialout and selecting a free channel on the VegaStream Any general suggestions/experiences with regard to configuring a VegaStream withasteriskwould be appricated.Thanks * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person
[Asterisk-Users] Configuring buttons on a CISCO 12SP+ Ip Phone (skinny.conf)
Hi to everybody, Is it possible with the skinny module (skinny.conf) in asterisk configuring the buttons on a CISCO 12 SP+ ? Is someone working on this ? Thank You. -- Davide Yachaya HyperGrid s.r.l. V.le Golgi 63 - 27100 Pavia - ITALY http://www.hypergrid.it Tel: +39,0382,528875 Fax: +39,0382,408005 e-mail: [EMAIL PROTECTED] -- PGP Key - http://mail.hypergrid.it/public-key/[EMAIL PROTECTED] X509 HyperGrid CA - http://mail.hypergrid.it/cacert.der ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Trouble emailing Digium
Is it just me or is everyone having problems with emailing digium? I've tried sending two emails, but they keep getting returned with the following errors:- - The following addresses had permanent fatal errors - [EMAIL PROTECTED] (reason: 554 [EMAIL PROTECTED]: Recipient address rejected: Relay access denied) - Transcript of session follows - ... while talking to digium.com.mail1.psmtp.com.: RCPT To:[EMAIL PROTECTED] 554 [EMAIL PROTECTED]: Recipient address rejected: Relay access denied 554 5.0.0 Service unavailable Thanks, Chris Lee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN update
Klaus-Peter Junghanns ha scritto: Hi BRI people, chan_capi 0.3.1 is now released, including a fix for the pipe leak. bristuff 0.0.2rc7 is available now too. Including a zaptel driver for the HFC-S PCI A based ISDN cards (with echo cancelation, TE and NT mode). We'll also have a devkit for zaptel BRI soon. enjoy! best regards kapejod -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I tried make, make install. /usr/bin/asterisk -vvvgc and what I get is: loader.c:239 ast_load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_get_group loader.c:358 load_modules: Loading module chan_capi.so failed! what's wrong? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN update
oh yes... i added callgroup support for chan_capi. That's why you have to load res_parking.so before chan_capi.so. So in modules.conf you need. load = res_parking.so load = chan_capi.so [global] chan_capi.so=yes best regards kapejod -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ I tried make, make install. /usr/bin/asterisk -vvvgc and what I get is: loader.c:239 ast_load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_get_group loader.c:358 load_modules: Loading module chan_capi.so failed! what's wrong? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN update
Klaus-Peter Junghanns ha scritto: oh yes... i added callgroup support for chan_capi. That's why you have to load res_parking.so before chan_capi.so. So in modules.conf you need. load = res_parking.so load = chan_capi.so [global] chan_capi.so=yes best regards kapejod -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ I tried make, make install. /usr/bin/asterisk -vvvgc and what I get is: loader.c:239 ast_load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_get_group loader.c:358 load_modules: Loading module chan_capi.so failed! what's wrong? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ok, Thanks it works :) -- Matteo Rancilio === COMVERT S.R.L. C.P. 211 - 20099 Sesto S. G. Centro (MI) - ITALY Tel +39.02.27006796 | Fax +39.02.26005513 [EMAIL PROTECTED] http://www.comvert.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Execute command in shell
It drives me to a new question... how can I concatenate three strings on extensions.org ? That is, the command, and the two args; The arguments are the source e164 and destination e164 numbers of the current call. Something like /bin/false + $SOURCE164 + + $DEST164 -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de James H. Cloos Jr. Enviado el: viernes, 06 de febrero de 2004 9:58 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] Re: Execute command in shell Marc == Marc Fargas [EMAIL PROTECTED] writes: Marc I've seen its possible to use the System applications, but what Marc about passing arguments to the command ? A quick look at app_system.c shows that it just passes the string unaltered to system(3). So, running man 3 system will show exactly what system(3) does: system() executes a command specified in string by calling /bin/sh -c string, and returns after the command has been completed. As such, System(command arg1 arg2 etc) should do what you want. -JimC PS No, I didn't know that before I looked at the src PPS Except for what was in the system(3) manpage ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DIAX 0.9.6b call reception
I had some users complaining that DIAX only rung twice in the call sequence for incoming calls (ring user 1 20 secs, user 2 20 secs, ...). When testing, it turned out that with DIAX 0.9.3 I got the expected result: Feb 6 11:24:36 -- Executing Dial(CAPI[contr1/313650768]/0, IAX/ha/s|20|rt) in new stack Feb 6 11:24:36 -- Calling using options 'exten=s;callerid=00621262009;language=en;formats=2;capability=2;version=1;adsicpe=0' Feb 6 11:24:36 -- Called ha/s Feb 6 11:24:36 -- Call accepted by 80.61.160.1 (format GSM) Feb 6 11:24:36 -- Format for call is GSM Feb 6 11:24:36 -- IAX[ha]/39 is ringing Feb 6 11:24:41 -- Registered 'ha' (AUTHENTICATED) at 80.61.160.1:11800 Feb 6 11:24:56 -- Registered 'bb' (AUTHENTICATED) at 62.194.134.213:10805 Feb 6 11:24:56 -- Nobody picked up in 2 ms Feb 6 11:24:56 -- Hungup 'IAX[ha]/39' But with DIAX 0.9.6b: Feb 6 11:24:56 -- Executing Dial(CAPI[contr1/313650768]/0, IAX/bb/s|20|rt) in new stack Feb 6 11:24:56 -- Calling using options 'exten=s;callerid=00621262009;language=en;formats=2;capability=2;version=1;adsicpe=0' Feb 6 11:24:56 -- Called bb/s Feb 6 11:24:56 -- Call accepted by 62.194.134.213 (format GSM) Feb 6 11:24:56 -- Format for call is GSM Feb 6 11:24:56 -- IAX[bb]/43 is ringing Feb 6 11:25:06 -- Hungup 'IAX[bb]/43' Feb 6 11:25:06 == No one is available to answer at this time In other words, only 10 secs, and it looks like DIAX is actively rejecting the call... Am I missing something? -- Cees de Groot http://www.tric.nl [EMAIL PROTECTED] tric, the new way helpdesk/ticketing software, VoIP/CTI, web applications, custom development ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX 0.9.6b call reception
Hi, - Original Message - From: Cees de Groot [EMAIL PROTECTED] I had some users complaining that DIAX only rung twice in the call sequence for incoming calls (ring user 1 20 secs, user 2 20 secs, ...). When testing, it turned out that with DIAX 0.9.3 I got the expected result: Feb 6 11:24:36 -- Executing Dial(CAPI[contr1/313650768]/0, IAX/ha/s|20|rt) in new stack Feb 6 11:24:36 -- Calling using options 'exten=s;callerid=00621262009;language=en;formats=2;capability=2;version=1;a dsicpe=0' Feb 6 11:24:36 -- Called ha/s Feb 6 11:24:36 -- Call accepted by 80.61.160.1 (format GSM) Feb 6 11:24:36 -- Format for call is GSM Feb 6 11:24:36 -- IAX[ha]/39 is ringing Feb 6 11:24:41 -- Registered 'ha' (AUTHENTICATED) at 80.61.160.1:11800 Feb 6 11:24:56 -- Registered 'bb' (AUTHENTICATED) at 62.194.134.213:10805 Feb 6 11:24:56 -- Nobody picked up in 2 ms Feb 6 11:24:56 -- Hungup 'IAX[ha]/39' But with DIAX 0.9.6b: Feb 6 11:24:56 -- Executing Dial(CAPI[contr1/313650768]/0, IAX/bb/s|20|rt) in new stack Feb 6 11:24:56 -- Calling using options 'exten=s;callerid=00621262009;language=en;formats=2;capability=2;version=1;a dsicpe=0' Feb 6 11:24:56 -- Called bb/s Feb 6 11:24:56 -- Call accepted by 62.194.134.213 (format GSM) Feb 6 11:24:56 -- Format for call is GSM Feb 6 11:24:56 -- IAX[bb]/43 is ringing Feb 6 11:25:06 -- Hungup 'IAX[bb]/43' Feb 6 11:25:06 == No one is available to answer at this time In other words, only 10 secs, and it looks like DIAX is actively rejecting the call... Am I missing something? Do you mean that it works with version 0.9.3, but not with 0.9.6? Have you tried with both IAX(1) and IAX2? Can you use debug mode in 0.9.6 and send me the log? BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] manipulating with numbers - StripMSD, Prefix
Looks like you are shy a zero
Try exten = _50.,Prefix,001051
At 12:06 07/01/04, you wrote:
Hello,
I can not seem to be able to get StripMSD and Prefix to work for me in
extensions.conf. This is an example of what I have:
exten = _050.,1,StripMSD,1
exten = _50.,Prefix,01051
exten = _001051.,1,Dial(${TRUNK2}/${EXTEN})
exten = _001051.,2,Busy
exten = _001051.,102,Busy
What I want to achieve is to call 001051501657887 on TRUNK2 when dialing
0501657887.
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[Asterisk-Users] Re: Execute command in shell
Marc == Marc Fargas [EMAIL PROTECTED] writes:
Marc It drives me to a new question... how can I concatenate three
Marc strings on extensions.org ?
Marc That is, the command, and the two args; The arguments are the
Marc source e164 and destination e164 numbers of the current call.
Marc Something like /bin/false + $SOURCE164 + + $DEST164
I've not tested this, but I'd try something like:
exten = s,1,System(/bin/false ${SOURCE164} ${DEST164})
-JimC
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Re: [Asterisk-Users] chan_sccp: incoming calls on multiple lines
Sorry folks (I know it's annoying when people reply to their own posts)... On Fri, 6 Feb 2004, Vic Cross wrote: snip my note about incoming calls on multiple lines I just wanted to advise that I've done my patch to chan_sccp to provide this behaviour -- when a call comes in on any line, not just the 'selected' line, taking the phone off-hook answers the call. It also does not change the selection status of the lines (the line that was selected before stays selected). The patch does not change the behaviour of things getting messed up when a second call comes in while the phone is already off-hook (well, at least it's broken on my phone). I guess this is related to the fact that device-initiated call control (Hold, Call Forward, Conference, etc) is not yet implemented. One thing at a time, I'm still new to hacking Asterisk! :) If anyone is interested, please let me know. I'll lodge it in the Mantis for chan_sccp if we like it. Hoo-roo, Vic Cross ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fax with wildcards
Hi Thomas, I think you'll probably need a dedicated fax board to do this. Commetrex in Atlanta has a very nice solution. Check it out at www.commetrex.com. Also Natural MicroSystems, BrookTrout and Eichon all have very capable fax boards. I have been developing voice and fax applications with these boards for over 10 years. If you need any help selecting or coding, just let me know. You can email me directly at Tvaught at ColeTechnical.com Terry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Sent: Thursday, February 05, 2004 4:46 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Fax with wildcards Hello, does anybody know how stable is OpenCall's fax sending/receiving software? Is it still in development? I see only an old version on the ftp. Does anybody have any experienci with fax sending and the PC performance needed for this? What kind of PC hardware would I need when I would like to send concurrently faxes on one/two/three/four E1? Is it possible either? Thank you in advance, Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voiceglo questions, IAX
how many simultanius calls does voiceglo permit??? Miguel On Fri, 2004-02-06 at 01:28, Cameron Palmer wrote: IAX is what they use with glophone. http://webphone.voiceglo.com. It is a seperate server from the myphone.voiceglo.com SIP gateway. The IAX gateway is msps01-nyc.voiceglo.com on port 5036. cameron. On Thu, 5 Feb 2004, Jim Flagg wrote: - Original Message - From: Michael Swan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, February 05, 2004 5:23 PM Subject: [Asterisk-Users] Voiceglo questions 1. Can someone confirm whether Voiceglo needs to use SIP or can it handle IAX? This link seems to indicate it uses SIP: http://www.mail-archive.com/[EMAIL PROTECTED]/msg20561.html although other messages on the mailing list indicate that Voiceglo is using Asterisk in its internal architecture. Brian West indicated in this post http://lists.digium.com/pipermail/asterisk-users/2003-December/029076.html that he had Asterisk registering using IAX. Can Brian or anyone else post a copy of their IAX.conf Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_sccp: incoming calls on multiple lines
Vic Cross ha scritto: If anyone is interested, please let me know. I'll lodge it in the Mantis for chan_sccp if we like it. I like it! Where can we get it? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk GUI Client - New verison 0.9
-Original Message- From: mattf [mailto:[EMAIL PROTECTED] Sent: Thursday, February 05, 2004 4:33 PM To: '[EMAIL PROTECTED]' Subject: [Asterisk-Users] Asterisk GUI Client - New verison 0.9 Hello, I have made many changes/improvements/bug fixes to the Asterisk GUI client I have written in Perl/TK and have released a third beta version on sourceforge: http://sourceforge.net/projects/astguiclient/ Here are the screen shots of the client application running on Linux and Windows: http://www.freedomphones.net/astguiclient_linux_0.9.gif http://www.freedomphones.net/astguiclient_windows_0.9.gif 0.9 - Third public release - 2004-02-05 The majority of the work in this release it to make it more stable and fix some pretty bad bugs. We created the Asterisk Central Queue System to address the problem with buffer-overflows in the manager interface of Asterisk causing total system deadlocks. We also completed and touched-up many other features that we didn't finish in previous releases. Here is the list of changes: - Several bug fixes - Inclusion of listing for active SIP/Local channels and ability to hang them up - Completely changed the method of conferencing to be more fluid - Added HELP popup screen - Added intrasystem calling funtionality - Updater changed to allow for SIP/Local channels - Recording for conferences is now able to record all audio in and out - Added ability to send DTMF tones within a conference - Changed alert window for updater being down timeout to 20 seconds - Added an option for using the new Asterisk Central Queue System(ACQS) that reduces the risk of deadlocks that occur with buffer-overflows on remote manager interface connections - Included new script to run at boot time and rotate the logs as well as a keepalive script for the new ACQS - Changed non-AGI server-side scripts to allow for a single config file - Detailed activity logging to text file option added - Activity logging added to all non-AGI server applications We have been using the same basic client for the last four months here at my company and it is running well on over 60 machines. Let me know what you think of it, especially the new Asterisk Central Queue System that is included with it. MATT--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Matt, Thanks for posting your utiliy. I would really like to use the utility you have written. Is there any installation help or instructions for win32? Pardon my ignorance. Thanks, Dustin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk GUI Client - New verison 0.9
You do have to add some extensions, copy some files and have some scripts run on your Asterisk server, as well as have a MySQL database set up on a machine somewhere before you can install the client on a machine. All of that is explained in the documentation included with the package. As for the Win32 client, it's fairly easy, just download ActivePerl from Activestate.com and install it, then copy the libs folder and AST_WINphoneAPP_0.9.pl to C:\AST_VICI then customize the C:\AST_VICI\libs\AST_VICI_conf.pl file for your phone and you're ready to go. The package also installs in relatively the same way on Linux and in-fact runs the exact same code as it does on Win32(For linux we recommend using the ActivePerl RPM instead of default perl because of a perl/TK memory leak). It's not a simple one step setup like some other Asterisk GUI clients, but this package was made to be very flexible and easy to change, in ACQS-mode it relies entirely on the MySQL server to run having no contact with the Asterisk server directly making it much harder for the average user to inadvertantly crash the Asterisk server. If you need any more detailed help setting it up let me know. MATT--- -Original Message- From: Dustin Knuttgen [mailto:[EMAIL PROTECTED] Sent: Friday, February 06, 2004 9:30 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk GUI Client - New verison 0.9 -Original Message- From: mattf [mailto:[EMAIL PROTECTED] Sent: Thursday, February 05, 2004 4:33 PM To: '[EMAIL PROTECTED]' Subject: [Asterisk-Users] Asterisk GUI Client - New verison 0.9 Hello, I have made many changes/improvements/bug fixes to the Asterisk GUI client I have written in Perl/TK and have released a third beta version on sourceforge: http://sourceforge.net/projects/astguiclient/ Here are the screen shots of the client application running on Linux and Windows: http://www.freedomphones.net/astguiclient_linux_0.9.gif http://www.freedomphones.net/astguiclient_windows_0.9.gif 0.9 - Third public release - 2004-02-05 The majority of the work in this release it to make it more stable and fix some pretty bad bugs. We created the Asterisk Central Queue System to address the problem with buffer-overflows in the manager interface of Asterisk causing total system deadlocks. We also completed and touched-up many other features that we didn't finish in previous releases. Here is the list of changes: - Several bug fixes - Inclusion of listing for active SIP/Local channels and ability to hang them up - Completely changed the method of conferencing to be more fluid - Added HELP popup screen - Added intrasystem calling funtionality - Updater changed to allow for SIP/Local channels - Recording for conferences is now able to record all audio in and out - Added ability to send DTMF tones within a conference - Changed alert window for updater being down timeout to 20 seconds - Added an option for using the new Asterisk Central Queue System(ACQS) that reduces the risk of deadlocks that occur with buffer-overflows on remote manager interface connections - Included new script to run at boot time and rotate the logs as well as a keepalive script for the new ACQS - Changed non-AGI server-side scripts to allow for a single config file - Detailed activity logging to text file option added - Activity logging added to all non-AGI server applications We have been using the same basic client for the last four months here at my company and it is running well on over 60 machines. Let me know what you think of it, especially the new Asterisk Central Queue System that is included with it. MATT--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Matt, Thanks for posting your utiliy. I would really like to use the utility you have written. Is there any installation help or instructions for win32? Pardon my ignorance. Thanks, Dustin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] question for oh323 users
Hi, Thanks for your reply. I am definite that my h323 is running on ciscoB because the below scenario is working fine. pstnciscoA-ciscoBpstn I have also eliminated access-list problem because if my access-list is applied I could see packets hiting my access-list permit tcp host 192.168.1.2 any eq 1720 (60 matches) Is my syntax below correct ?? exten = _1905XXX,1,Dial,OH323/192.168.1.3 Any help would be appreciated. Regards, Anthony - Original Message - From: Tomica Crnek [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, February 06, 2004 3:03 AM Subject: RE: [Asterisk-Users] question for oh323 users Hi, it seams to me that h.323 service on your cisco B could be down. You see packets coming to this box, but did you activate h.323. Try telnet 192.168.1.3 1720 to see if it is running. If it is, then check to see if you are allowing connections to it from 192.168.1.2 Tomica -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Law Sent: Thursday, February 05, 2004 10:41 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] question for oh323 users Hi, I am trying to forward calls from one cisco gateway to another cisco gateway using asterisk cisco(5300)A 192.168.1.1 asterisk 192.168.1.2 cisco(5300)B 192.168.1.3 pstn --ciscoA-asterisk --ciscoB--pstn I have the below in my extension.conf [default] exten = _1905XXX,1,Dial,OH323/192.168.1.3 I keep getting error and I don't know what is wrong. I am able to see in my ciscoB accesslist, tcp packets are coming from 192.168.1.2 I get below error in my asterisk CLI Feb 5 16:17:01 ERROR[29716]: chan_oh323.c:1004 oh323_call: H323:0: Could not call 192.168.1.3. Feb 5 16:17:11 WARNING[29716]: pbx.c:1829 ast_pbx_run: Timeout, but no rule 't' in context 'default' It would be much appreciated if someone could point out what I am doing wrong or to any documentations. Many thanks. Regards, Anthony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] question for oh323 users
It must be: exten = _1905XXX,1,Dial,OH323/h323:[EMAIL PROTECTED] or exten = _1905XXX,1,Dial,OH323/[EMAIL PROTECTED] Hope this helps, Gus - Original Message - From: Anthony Law [EMAIL PROTECTED] To: Mailing List Asterisk [EMAIL PROTECTED] Sent: Friday, February 06, 2004 11:56 AM Subject: Re: [Asterisk-Users] question for oh323 users Hi, Thanks for your reply. I am definite that my h323 is running on ciscoB because the below scenario is working fine. pstnciscoA-ciscoBpstn I have also eliminated access-list problem because if my access-list is applied I could see packets hiting my access-list permit tcp host 192.168.1.2 any eq 1720 (60 matches) Is my syntax below correct ?? exten = _1905XXX,1,Dial,OH323/192.168.1.3 Any help would be appreciated. Regards, Anthony - Original Message - From: Tomica Crnek [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, February 06, 2004 3:03 AM Subject: RE: [Asterisk-Users] question for oh323 users Hi, it seams to me that h.323 service on your cisco B could be down. You see packets coming to this box, but did you activate h.323. Try telnet 192.168.1.3 1720 to see if it is running. If it is, then check to see if you are allowing connections to it from 192.168.1.2 Tomica -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Law Sent: Thursday, February 05, 2004 10:41 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] question for oh323 users Hi, I am trying to forward calls from one cisco gateway to another cisco gateway using asterisk cisco(5300)A 192.168.1.1 asterisk 192.168.1.2 cisco(5300)B 192.168.1.3 pstn --ciscoA-asterisk --ciscoB--pstn I have the below in my extension.conf [default] exten = _1905XXX,1,Dial,OH323/192.168.1.3 I keep getting error and I don't know what is wrong. I am able to see in my ciscoB accesslist, tcp packets are coming from 192.168.1.2 I get below error in my asterisk CLI Feb 5 16:17:01 ERROR[29716]: chan_oh323.c:1004 oh323_call: H323:0: Could not call 192.168.1.3. Feb 5 16:17:11 WARNING[29716]: pbx.c:1829 ast_pbx_run: Timeout, but no rule 't' in context 'default' It would be much appreciated if someone could point out what I am doing wrong or to any documentations. Many thanks. Regards, Anthony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: DIAX 0.9.6b call reception
Dan [EMAIL PROTECTED] said: Do you mean that it works with version 0.9.3, but not with 0.9.6? yes. Have you tried with both IAX(1) and IAX2? no - iax1 only. Can you use debug mode in 0.9.6 and send me the log? sure. I'll gather some more data and send it to you directly. I mainly posted here because I wasn't sure this was something known. -- Cees de Groot http://www.tric.nl [EMAIL PROTECTED] tric, the new way helpdesk/ticketing software, VoIP/CTI, web applications, custom development ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Conference server
Hi, we are setting a 120-channel conference server and would like to learn if someone already did this (hardware, problems, etc...) Best regards, PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Conference server
Currently Asterisk will cause a Kernel Panic if you are using the Linux SMP kernel and have about 30 channels in conference. Here's the bug listing: http://bugs.digium.com/bug_view_page.php?bug_id=963 MATT--- -Original Message- From: Paulo Mannheimer [mailto:[EMAIL PROTECTED] Sent: Friday, February 06, 2004 11:20 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Conference server Hi, we are setting a 120-channel conference server and would like to learn if someone already did this (hardware, problems, etc...) Best regards, PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: DIAX 0.9.6b call reception
Hi, - Original Message - From: Cees de Groot [EMAIL PROTECTED] Do you mean that it works with version 0.9.3, but not with 0.9.6? yes. Have you tried with both IAX(1) and IAX2? no - iax1 only. IAX (1) will not be supported in the future anymore. Version 0.9.7 which I intend to release next week will be IAX2 based only (with the possibility to use another DLL if you still want IAX(1), without any added feature). Can you use debug mode in 0.9.6 and send me the log? sure. I'll gather some more data and send it to you directly. I mainly posted here because I wasn't sure this was something known. They are some known problems with both IAX(1) and IAX(2) client libraries, but nobody work on IAX(1) anymore. BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.2 RPMS Updated
On Thu, 2004-02-05 at 19:42, Greg Boehnlein wrote: On Wed, 4 Feb 2004, Greg Boehnlein wrote: On Wed, 4 Feb 2004, Chris Tooley wrote: Well, I don't really know all that much about SuSE either. I just installed it about 19 hours ago for the first time. Well, depending on the version of RPM that they installed, you'll either need to issue rpm -ba asterisk.spec or rpmbuild -ba asterisk.spec. For all I know, you might just be able to install the RPMS for RH9 on Suse. It should complain if dependncies aren't met. I just realized that due to a logic error in my build-asterisk-distrib script, I did not upload the kernel-modules-zaptel RPMS for the 0.7.2 release. I have corrected that error, and they are now available. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users As my SuSE box has no zaptel or pri hardware I have not rebuilt thos RPMS but I have rebuilt the asterisk rpm using the SRPM from FC1 with no changes. (I didn't even go to the effort of changing the package name). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Conference server
This seems to only apply to non-zap channels participating in the conference, incidently. On Fri, 6 Feb 2004, mattf wrote: Currently Asterisk will cause a Kernel Panic if you are using the Linux SMP kernel and have about 30 channels in conference. Here's the bug listing: http://bugs.digium.com/bug_view_page.php?bug_id=963 MATT--- -Original Message- From: Paulo Mannheimer [mailto:[EMAIL PROTECTED] Sent: Friday, February 06, 2004 11:20 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Conference server Hi, we are setting a 120-channel conference server and would like to learn if someone already did this (hardware, problems, etc...) Best regards, PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP - Native Bridge Error
Hi, Running Version 0.7.2, I receive the following error when attempting to connect two SIP Devices. WARNING[16399]:rtp.c : 1204 ast_rtp_bridge : codec0 = 524556 is not codec1 = 524558, cannot native bridge. The bridge is made but the quality of the call is bad, a lot of disturbing noises in background. Oddly enough, both devices are using only one codec G729. I also am using the demo G729 license for Asterisk. I'm not sure how 2 different codecs are being found. I saw in ast_rtp_bridge function, that the get_codec function returned these values. Could anyone tell me where the get_codec function is? Curious as to how this is happening. Should this problem be added to the bug tracker? The SIP calls are very bad, and I did not experience this problem with 0.5.0 . Thanks, Wes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trouble emailing Digium
Small error in the zone file caused this. Its fixed now. bkw On Fri, 6 Feb 2004, Isamar Maia wrote: I bought recently a G729 and didn't any response... maybe for the same reason? :-( Isamar On Fri, 6 Feb 2004, Vic Cross wrote: Chris, On Fri, 6 Feb 2004, Christopher Lee wrote: - The following addresses had permanent fatal errors - [EMAIL PROTECTED] (reason: 554 [EMAIL PROTECTED]: Recipient address rejected: Relay access denied) - Transcript of session follows - ... while talking to digium.com.mail1.psmtp.com.: RCPT To:[EMAIL PROTECTED] 554 [EMAIL PROTECTED]: Recipient address rejected: Relay access denied 554 5.0.0 Service unavailable I was affected by something similar to this recently, with a company in the US whose mail server virus/spam filter rejected anything from com.au... Folks...? ;-) Cheers, Vic Cross ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.2 RPMS Updated
Is someone going to do the v1-0-stable RPMS? Not sure if anyone knows that it was branched yet or not. Everyone was jumping up and down and chanting BRANCH BRANCH BRANCH It fially happened and nobody says a word haha.. :) bkw On Fri, 6 Feb 2004, Chris Tooley wrote: On Thu, 2004-02-05 at 19:42, Greg Boehnlein wrote: On Wed, 4 Feb 2004, Greg Boehnlein wrote: On Wed, 4 Feb 2004, Chris Tooley wrote: Well, I don't really know all that much about SuSE either. I just installed it about 19 hours ago for the first time. Well, depending on the version of RPM that they installed, you'll either need to issue rpm -ba asterisk.spec or rpmbuild -ba asterisk.spec. For all I know, you might just be able to install the RPMS for RH9 on Suse. It should complain if dependncies aren't met. I just realized that due to a logic error in my build-asterisk-distrib script, I did not upload the kernel-modules-zaptel RPMS for the 0.7.2 release. I have corrected that error, and they are now available. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users As my SuSE box has no zaptel or pri hardware I have not rebuilt thos RPMS but I have rebuilt the asterisk rpm using the SRPM from FC1 with no changes. (I didn't even go to the effort of changing the package name). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.2 RPMS Updated
I'd be happy to help. I've got several boxes in various stages of RedHat and I've a SuSE 9 box but nothing older than that. Chris On Fri, 2004-02-06 at 09:59, Brian West wrote: Is someone going to do the v1-0-stable RPMS? Not sure if anyone knows that it was branched yet or not. Everyone was jumping up and down and chanting BRANCH BRANCH BRANCH It fially happened and nobody says a word haha.. :) bkw On Fri, 6 Feb 2004, Chris Tooley wrote: On Thu, 2004-02-05 at 19:42, Greg Boehnlein wrote: On Wed, 4 Feb 2004, Greg Boehnlein wrote: On Wed, 4 Feb 2004, Chris Tooley wrote: Well, I don't really know all that much about SuSE either. I just installed it about 19 hours ago for the first time. Well, depending on the version of RPM that they installed, you'll either need to issue rpm -ba asterisk.spec or rpmbuild -ba asterisk.spec. For all I know, you might just be able to install the RPMS for RH9 on Suse. It should complain if dependncies aren't met. I just realized that due to a logic error in my build-asterisk-distrib script, I did not upload the kernel-modules-zaptel RPMS for the 0.7.2 release. I have corrected that error, and they are now available. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users As my SuSE box has no zaptel or pri hardware I have not rebuilt thos RPMS but I have rebuilt the asterisk rpm using the SRPM from FC1 with no changes. (I didn't even go to the effort of changing the package name). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.2 RPMS Updated
I don't need them but just asked because others might. bkw On Fri, 6 Feb 2004, Chris Tooley wrote: I'd be happy to help. I've got several boxes in various stages of RedHat and I've a SuSE 9 box but nothing older than that. Chris On Fri, 2004-02-06 at 09:59, Brian West wrote: Is someone going to do the v1-0-stable RPMS? Not sure if anyone knows that it was branched yet or not. Everyone was jumping up and down and chanting BRANCH BRANCH BRANCH It fially happened and nobody says a word haha.. :) bkw On Fri, 6 Feb 2004, Chris Tooley wrote: On Thu, 2004-02-05 at 19:42, Greg Boehnlein wrote: On Wed, 4 Feb 2004, Greg Boehnlein wrote: On Wed, 4 Feb 2004, Chris Tooley wrote: Well, I don't really know all that much about SuSE either. I just installed it about 19 hours ago for the first time. Well, depending on the version of RPM that they installed, you'll either need to issue rpm -ba asterisk.spec or rpmbuild -ba asterisk.spec. For all I know, you might just be able to install the RPMS for RH9 on Suse. It should complain if dependncies aren't met. I just realized that due to a logic error in my build-asterisk-distrib script, I did not upload the kernel-modules-zaptel RPMS for the 0.7.2 release. I have corrected that error, and they are now available. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users As my SuSE box has no zaptel or pri hardware I have not rebuilt thos RPMS but I have rebuilt the asterisk rpm using the SRPM from FC1 with no changes. (I didn't even go to the effort of changing the package name). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] passing variables to a macro
I was wondering if this would work
set a variable (varX) in macro-test
call another macro [macro-subroutine]
have varX available within [macro-subroutine]
[macro-test-1]
; ${ARG1} - extension
setvar(var1=foo)
macro(subroutine,${ARG1})
[macro-subroutine]
do something with varX
Or do you have to pass all variables you want to have available within a
macro when the macro is called ?
like this:
macro(subroutine,${ARG1},${varX})
thanks
--Lance
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RE: [Asterisk-Users] Re: Execute command in shell
On Fri, 2004-02-06 at 04:33, Marc Fargas wrote: It drives me to a new question... how can I concatenate three strings on extensions.org ? That is, the command, and the two args; The arguments are the source e164 and destination e164 numbers of the current call. Something like /bin/false + $SOURCE164 + + $DEST164 Covered in README.variables. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.2 RPMS Updated
Brian West wrote: Is someone going to do the v1-0-stable RPMS? Not sure if anyone knows that it was branched yet or not. Everyone was jumping up and down and chanting BRANCH BRANCH BRANCH I don't think that we've reached 1.0 stable, though, have we? branching is an essential precursor in order to allow stablisation of the current featureset to happen in a different space to the addition of new features. Personally I welcome this - both branch HEAD should benefit :) However, I think it's too early for RPMS of a snapshot of this branch of CVS ;) It finally happened and nobody says a word haha.. :) I didn't see any announcement ;) My word: Thankyou :) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] One way h323 to Cisco 7905?
I've acquired a Cisco 7905 with H323 s/w that I have connected to *. It can make calls happily enough to H323 SIP extensions and out to the PSTN, however when ever I try to call it from any destination the call fails with H323:0 Could not call 192.168.9.23 Hungup 'H323:0' Everyone is busy at this time. TCPDUMP shows a short but spirited exchange between the 7905 and *, but nothing on the console to give me a hint. Anyone got any ideas? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.2 RPMS Updated
However, I think it's too early for RPMS of a snapshot of this branch of CVS ;) If I'm going to do the RPM's I'm probably going to want to do a daily snapshot build system that builds the RPMs. Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interrupted musiconhold sound when silence suppression is enabled
George Ye wrote: Hi, I am a new player of the Asterisk. I have a strage problem with musiconhold feature. Can anyone give some clues what might be the problem? A description of the problem is as follows: 1. Call from Cisco ATA 186 without silence suppression, when I push the hold button at the Cisco 7960 IP phone, the music plays just fine. 2. Call from Cisco ATA 186 with silence suppression enable, when I push the hold button of phone, the music is annoying, it cannot play smoothly. Sometimes, it plays well for a while, then there is a pause, then plays again, then pause,... * uses the incoming RTP Stream as a timing source for sending its outgoing Stream.. If the incoming stream is interrupted due to silence suppression then musiconhold will be choppy. So in conclusion, you cannot use silence suppresion. Certainly, an obvious solution is to disable silence suppression, however, this is unpracticable, because, sometimes, you might have no control of the remote side. Thanks in advance. George Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. *Try it!* http://us.rd.yahoo.com/evt=21608/*http://webhosting.yahoo.com/ps/sb/ -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CVS Changes (NAT-SIP)
I am having the same problem with a new CVS. Patrick also has the problem here http://lists.digium.com/pipermail/asterisk-users/2004-January/035114.html Keven had a problem here http://lists.digium.com/pipermail/asterisk-users/2004-January/035262.html but was able to get it fixed. Can you post a patch?. My asterisk computer is multi-homed behind NAT so maybe that is a factor? Is Asterisk behind NAT working with a new CVS for anybody? Thanks, - Original Message - From: Asterisk User Group [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, January 19, 2004 10:16 PM Subject: [Asterisk-Users] CVS Changes (NAT-SIP) I had been running an older patched CVS to get VOIP working with NAT and everything had been running fine. I just built * on a new box with CVS-01/18/04-12:19:25. And now I can get remote SIP users to register. Has anything major changed... [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to externip = 69.132.68.17 ; Address that we're going to put in SIP messages if we're behind a NAT localnet = 192.168.1.0 ; Internal NETWORK address localmask = 255.255.255.0 ; Internal netmask context = default ; Default for incoming calls ;srvlookup = yes; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=ilbc [1001] type=friend secret=1001 host=dynamic username=1001 mailbox=1001 context=local nat=no [1006] type=friend secret=oicu812 host=dynamic username=1006 mailbox=1006 context=local nat=yes canreinvite=no qualify=500 Internal SIP users can register it just the outside users. -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.2 RPMS Updated
On Fri, 6 Feb 2004, Brian West wrote: Is someone going to do the v1-0-stable RPMS? As soon as there is a .tgz available, (assuming it isn't already on the FTP site) I will be happy to do it. Not sure if anyone knows that it was branched yet or not. Everyone was jumping up and down and chanting BRANCH BRANCH BRANCH It fially happened and nobody says a word haha.. :) I'm excited and happy to hear that the code has branched, but until there is a snapshot released, it would be better for people to build out of the CVS tree. I suppose we could provide nightly RPM builds out of CVS, but I personally would rather see the general public beating on Snapshots dropped at known time periods so that specific releases can be tagged w/ specific bugs. If we have 100 days worth of CVS snapshot RPMS out there, imagine the confusion that can occur: I have CVS version 2004-2-1.rpm and it has this bug. Yeah, that was fixed in CVS on 2004-3-2, so upgrade. But I can't upgrade because the CVS 2004-3-4 that is available now breaks the Xyz feature I personally believe it's cleaner to maintain a snapshot release for general public consumption and allow people to maintain Changelogs between versions. That way the end user can get a clear history of what works/doesn't work based on release numbers. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk setup.-
Hi, I recently received my development kit with 1 x100p and one tdm400p (1) fxs port. I installed everything from the digium disk that i received with my kit, however, i dont; know what to do next. I would like to be able to call through the internet using xten (pc2phone) and terminate the call in my gateway. anyone has a standard setup ? thanks, Francisco - Original Message - From: Steven E. Frazier [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, February 06, 2004 12:30 AM Subject: [Asterisk-Users] Adding another X100P after X100P and TDM400P is already configured History: 1. Added X100P to my system 2. Added TDM400P (2 port) Worked fine so far 3. Now I want to add an additional X100P Is the following configs files ok and is there any issue with adding the X100P (channel 4) after my 2 analog FXS channels? Thanks. Steve Here is my /etc/zaptel.conf fxsks=1,4 fxols=2-3 loadzone = us defaultzone = us Here is my /etc/asterisk/zapata.conf ; Zapata telephony interface sample configuration file ; [channels] ; ; X100P plugged into PSTN ; X100P # 1 context=incoming signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no busydetect=no callprogress=no musiconhold=default usecallerid=yes callerid=asreceived channel = 1 ; ; ; ; TDM200B Port #1 plugged into analog Phone ; ; context=toll-access signalling=fxo_ls callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no musiconhold=default usecallerid=yes callerid=Livingroom 2201 mailbox=2201 channel = 2 ; ; TDM200B Port #2 ; ; context=toll-access signalling=fxo_ls callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no musiconhold=default usecallerid=yes callerid=Kitchen 2202 mailbox=2202 channel = 3 ; X100P # 2 context=incoming signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no busydetect=no callprogress=no musiconhold=default usecallerid=yes callerid=asreceived channel = 4 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.2 RPMS Updated
No snap shot is needed! You are able to check out the 1.0 branch from cvs. Only bug fixes will go in this branch so you can automate the checkout/update and rpm build process and produce daily 1.0 rpms. To check out code from our STABLE 1.0 Branch CVS repository for Asterisk ONLY: # cd /usr/src # export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot # cvs login - the password is anoncvs. # cvs checkout -r v1-0_stable asterisk Nothing going into stable will break anything else (or it shouldln't) Everything would be tested before being applied to 1.0-stable by me or the many other bug marshals. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.2 RPMS Updated
I don't think that we've reached 1.0 stable, though, have we? branching is an essential precursor in order to allow stablisation of the current featureset to happen in a different space to the addition of new features. Personally I welcome this - both branch HEAD should benefit :) However, I think it's too early for RPMS of a snapshot of this branch of CVS ;) You guessed it 1.0 is the precursor to stablization. We still have CVS head that is open game for anything and everything (ie the kitchen sink) I think mark said he would release 0.9.x snapshots building up to 1.0's stable status. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Silencing Background App during touch tone detection
We're still have problems with the outgoing voice message interfering with the touch tone detection. Often the first touch tone pressed will be detected twice. If I configure asterisk to not play the message, or if people wait till the outgoing message stops, it works flawlessly. I've noticed that many phone system silence the outgoing message once you start pressing buttons, is there anyway to configure asterisk to do the same? My C skills are non-existent, but the touch tone detection code in the zaptel libraries appears to work this way (apologies for basic like syntax): start: Wait for tone detected waiting: wait ~250ms still hearing tone? Yes, goto waiting No.. Add that tone to the queue, goto start In the above a short dropout that falls exactly on one of the 250ms checks would be detected as a break, even is it was only a few ms long. In my experience TT detection system work more like this, require a minimum length of silence/no Touch tone (100-250ms) before advancing to detecting the next number: start: wait for tone detected waiting: wait 250ms still hearing tone? Yes, goto waiting No.. wait 100-250ms, Still no tone? yes, add current tone to queue and goto start no, ignore silence, and goto waiting Thanks Mark Farver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: DISA
John and sundry others: First thanks for your help. You have succiently summed up the problem. I do not get dialtone fast enough. The following is a test dialplan that I set up this morning after recieveing the many kind e-mails, It's very basic, but it does allow me to process a call to my phone extension, albeit I still don't get dialtone immediately when I select a line or dial into the asterisk system. (see embedded notes for details). [general] static = yes writeprotect = no ; [main2] exten = 9,1,dial(zap/g2) exten = _5012 ignorepat = 9 ; [main1] exten = s,1,DISA(2285750,main2) exten = s,2,Hangup( ) ; ;Notes on testing: ;Circuit is a full T1 provided by my in house Nortel ;SL1 to port 3 of my Digium T410p. It's identified ;in zaptel.conf as span =3,0,0,d4,ami., and configured ;in zapata.conf as group=2, signalling=em_w, ;channel = 49-72. ; ;For purposes of testing only, I have my Nortel Norstar ;system with a T1 cartridge attached to port 4 of the ;Digium T410p. It's identified in zaptel.conf as ;span=4,0,0,esf,b8zs and configured in zapata.conf as ;group=3, signalling=em_w, channel = 73-96. ; ;ztcfg -vv indicates the configuration is correct, and ;zttool indicates that there are no errors ; ;When I select line 1 on the Norstar (where I would ;normally expect to to get dialtone, in effect simply ; going off hook) . I do not get dialtone. ; ;CLI indicates Starting simple switch on 'Zap-73-1' . ;The same hold true if I dial in on this T1. ; ;after 5 seconds (the timeout), I finally recieve dialtone. ; ;At this point I dial 2285750# and I get dialtone again ; ; CLI indicates WARNING [1225991448]: ;app_disa_c:290 : disa_exec: DISA on Zap/73-1 ;password is good. ; ;The dialplan then branches to [main2] ; [main2] exten = 9,1,dial(zap/g2) exten = _5012,1,dial(zap/g2) ignorepat = 9 ; ;Since both the Norstar and the SL1 are configured with ;dial 9 access (and yes, I've tried using straight access ;with the same results). I dial 995012, and the call ;processes, ringing my extension 5012 on the SL1. ; ;CLI indicates ;'Executing dial(Zap/73-1 , Zap/g2) in new stack'. ;Called g2 ;'Zap/49-1 answered Zap/73-1' ;'attempting native bridge of Zap/73-1 and Zap/49-1' ; ;I answer the call on my extension '5012' and talk as long ;as I care and then simply hangup. ; ;CLI indicates 'Hungup 'Zap/49-1' ;'spawn extension (main2,9,1) exited non-zero on ;Zap/73-1' ;Hungup 'Zap/73-1' ; [default] exten = s,1,answer exten = s,2,disa(no-password, main2) exten = s,3,Hangup ; - Original Message - From: John Todd [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, February 05, 2004 9:55 PM Subject: Re: [Asterisk-Users] Re: DISA At 9:32 PM -0500 2/5/04, Steve Creel wrote: On Thu, 5 Feb 2004, John Todd wrote: So, to boil your problem down to what I think is the problem: When you attach an inbound call to the DISA application, it does not produce a dialtone fast enough. snip [main1] ; ; Take any number, and give it to the DISA. The DISA ; just then takes anything typed in within the (unchangeable) ; timer values, and hands it off to main2 to be post-processed. ; I include the standard i,h,t values for pedantic reasons. ; exten = _X.,1,DISA(no-password,main2) exten = _X.,2,Hangup ; exten = h,1,Hangup exten = i,1,Congestion exten = i,2,Hangup exten = t,1,Congestion exten = t,2,Hangup Not to point out the obvious, but isn't the delay he's seeing caused by the _X. and the digittimeout? Couldn't this be resolved by using a more specific match on the DISA instead of _X. ? Steve [EMAIL PROTECTED] Ah, yes, that's probably the case. Without further information from the poster about how he was getting calls into the context, I assumed that this was a PRI or something that handed a DID to the context. If this is an FXO or some type of T1 trunking, then yes, the s extension would be more appropriate if this was an immediate=yes type of situation. GIGO. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: DISA
What is your zapata.conf? Have you tried imediate = yes? quote who=Ed Devine John and sundry others: First thanks for your help. You have succiently summed up the problem. I do not get dialtone fast enough. The following is a test dialplan that I set up this morning after recieveing the many kind e-mails, It's very basic, but it does allow me to process a call to my phone extension, albeit I still don't get dialtone immediately when I select a line or dial into the asterisk system. (see embedded notes for details). [general] static = yes writeprotect = no ; [main2] exten = 9,1,dial(zap/g2) exten = _5012 ignorepat = 9 ; [main1] exten = s,1,DISA(2285750,main2) exten = s,2,Hangup( ) ; ;Notes on testing: ;Circuit is a full T1 provided by my in house Nortel ;SL1 to port 3 of my Digium T410p. It's identified ;in zaptel.conf as span =3,0,0,d4,ami., and configured ;in zapata.conf as group=2, signalling=em_w, ;channel = 49-72. ; ;For purposes of testing only, I have my Nortel Norstar ;system with a T1 cartridge attached to port 4 of the ;Digium T410p. It's identified in zaptel.conf as ;span=4,0,0,esf,b8zs and configured in zapata.conf as ;group=3, signalling=em_w, channel = 73-96. ; ;ztcfg -vv indicates the configuration is correct, and ;zttool indicates that there are no errors ; ;When I select line 1 on the Norstar (where I would ;normally expect to to get dialtone, in effect simply ; going off hook) . I do not get dialtone. ; ;CLI indicates Starting simple switch on 'Zap-73-1' . ;The same hold true if I dial in on this T1. ; ;after 5 seconds (the timeout), I finally recieve dialtone. ; ;At this point I dial 2285750# and I get dialtone again ; ; CLI indicates WARNING [1225991448]: ;app_disa_c:290 : disa_exec: DISA on Zap/73-1 ;password is good. ; ;The dialplan then branches to [main2] ; [main2] exten = 9,1,dial(zap/g2) exten = _5012,1,dial(zap/g2) ignorepat = 9 ; ;Since both the Norstar and the SL1 are configured with ;dial 9 access (and yes, I've tried using straight access ;with the same results). I dial 995012, and the call ;processes, ringing my extension 5012 on the SL1. ; ;CLI indicates ;'Executing dial(Zap/73-1 , Zap/g2) in new stack'. ;Called g2 ;'Zap/49-1 answered Zap/73-1' ;'attempting native bridge of Zap/73-1 and Zap/49-1' ; ;I answer the call on my extension '5012' and talk as long ;as I care and then simply hangup. ; ;CLI indicates 'Hungup 'Zap/49-1' ;'spawn extension (main2,9,1) exited non-zero on ;Zap/73-1' ;Hungup 'Zap/73-1' ; [default] exten = s,1,answer exten = s,2,disa(no-password, main2) exten = s,3,Hangup ; -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Annoying Beeps
Every once and a while * throws a new wrinkle at me. It has started, all on its own, to make these annoying little beeps evey time a message prints at the CLI. If I bring down * and restart, they go away for a time, then seem to spontaneously reappear sometime later. It's almost as if * is starting to experience the Terrible Twos! No one else seems to be complaining about this, but I nevertheless assume that I can somehow disable this feature, I just can't seem to find out how. Maybe something like CLI stop beeping damit? Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Annoying Beeps
On Fri, 2004-02-06 at 13:46, Stephen R. Besch wrote: Every once and a while * throws a new wrinkle at me. It has started, all on its own, to make these annoying little beeps evey time a message prints at the CLI. If I bring down * and restart, they go away for a time, then seem to spontaneously reappear sometime later. It's almost as if * is starting to experience the Terrible Twos! No one else seems to be complaining about this, but I nevertheless assume that I can somehow disable this feature, I just can't seem to find out how. Maybe something like CLI stop beeping damit? I think that is due to there being a character on the CLI. Try hitting enter to clear the line, or hit ctrl-l to do a screen redraw and see whats on the line. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax2 jitter stats confusion
I have been kind of tracking IAX2 calls and trying to measure performance with a given iax2 set jitter command. My default is 250ms.. When a call is in progress I'll be watching it at the console with iax2 show channels Here are my stats from one particular call: 66.225.202.72benshaw 1/16413 00048/00035 00489ms 0221ms ILBC 66.225.202.72benshaw 1/16413 00137/00125 00487ms 0270ms ILBC 66.225.202.72benshaw 1/16413 00137/00125 00491ms 0269ms ILBC 66.225.202.72benshaw 1/16413 00141/00129 00487ms 0241ms ILBC 66.225.202.72benshaw 1/16413 00141/00129 00480ms 0235ms ILBC 66.225.202.72benshaw 1/16413 00143/00131 00480ms 0256ms ILBC 66.225.202.72benshaw 1/16413 00144/00132 00492ms 0268ms ILBC 66.225.202.72benshaw 1/16413 00152/00140 00487ms 0472ms ILBC 66.225.202.72benshaw 1/16413 00154/00142 00507ms 0473ms ILBC Now I figured the guy would be coming up to my office shooting but when I asked him how the call was he said perfect. -- now he knows he's on a VOIP call but he had no idea of the jitter and lag here... So I suppose my question is huh? How can I have such poor jitter and yet have this guy (not a techie) claim the call was perfect? Neither he nor the guy on the other end (PSTN through NuFone) had any issues about the quality. I don't want to look a gift horse in the mouth, so to speak, but I would like to know how to measure call quality; I thought jitter was a pretty good indicator. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Silencing Background App during touch tone detection
Post your config and we can see whats up.. bkw On Fri, 6 Feb 2004, Mark Farver wrote: We're still have problems with the outgoing voice message interfering with the touch tone detection. Often the first touch tone pressed will be detected twice. If I configure asterisk to not play the message, or if people wait till the outgoing message stops, it works flawlessly. I've noticed that many phone system silence the outgoing message once you start pressing buttons, is there anyway to configure asterisk to do the same? My C skills are non-existent, but the touch tone detection code in the zaptel libraries appears to work this way (apologies for basic like syntax): start: Wait for tone detected waiting: wait ~250ms still hearing tone? Yes, goto waiting No.. Add that tone to the queue, goto start In the above a short dropout that falls exactly on one of the 250ms checks would be detected as a break, even is it was only a few ms long. In my experience TT detection system work more like this, require a minimum length of silence/no Touch tone (100-250ms) before advancing to detecting the next number: start: wait for tone detected waiting: wait 250ms still hearing tone? Yes, goto waiting No.. wait 100-250ms, Still no tone? yes, add current tone to queue and goto start no, ignore silence, and goto waiting Thanks Mark Farver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: DISA
Yes I have tried immediate = yes. I do get dialtone immediately when I go off-hook or dial in, but then Asterisk won't accept any further input whether dialing from the Norstar or dialing on the T1 side. Essentially, I can't break dialtone. - Original Message - From: Robert Hajime Lanning [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, February 06, 2004 1:45 PM Subject: Re: [Asterisk-Users] Re: DISA What is your zapata.conf? Have you tried imediate = yes? quote who=Ed Devine John and sundry others: First thanks for your help. You have succiently summed up the problem. I do not get dialtone fast enough. The following is a test dialplan that I set up this morning after recieveing the many kind e-mails, It's very basic, but it does allow me to process a call to my phone extension, albeit I still don't get dialtone immediately when I select a line or dial into the asterisk system. (see embedded notes for details). [general] static = yes writeprotect = no ; [main2] exten = 9,1,dial(zap/g2) exten = _5012 ignorepat = 9 ; [main1] exten = s,1,DISA(2285750,main2) exten = s,2,Hangup( ) ; ;Notes on testing: ;Circuit is a full T1 provided by my in house Nortel ;SL1 to port 3 of my Digium T410p. It's identified ;in zaptel.conf as span =3,0,0,d4,ami., and configured ;in zapata.conf as group=2, signalling=em_w, ;channel = 49-72. ; ;For purposes of testing only, I have my Nortel Norstar ;system with a T1 cartridge attached to port 4 of the ;Digium T410p. It's identified in zaptel.conf as ;span=4,0,0,esf,b8zs and configured in zapata.conf as ;group=3, signalling=em_w, channel = 73-96. ; ;ztcfg -vv indicates the configuration is correct, and ;zttool indicates that there are no errors ; ;When I select line 1 on the Norstar (where I would ;normally expect to to get dialtone, in effect simply ; going off hook) . I do not get dialtone. ; ;CLI indicates Starting simple switch on 'Zap-73-1' . ;The same hold true if I dial in on this T1. ; ;after 5 seconds (the timeout), I finally recieve dialtone. ; ;At this point I dial 2285750# and I get dialtone again ; ; CLI indicates WARNING [1225991448]: ;app_disa_c:290 : disa_exec: DISA on Zap/73-1 ;password is good. ; ;The dialplan then branches to [main2] ; [main2] exten = 9,1,dial(zap/g2) exten = _5012,1,dial(zap/g2) ignorepat = 9 ; ;Since both the Norstar and the SL1 are configured with ;dial 9 access (and yes, I've tried using straight access ;with the same results). I dial 995012, and the call ;processes, ringing my extension 5012 on the SL1. ; ;CLI indicates ;'Executing dial(Zap/73-1 , Zap/g2) in new stack'. ;Called g2 ;'Zap/49-1 answered Zap/73-1' ;'attempting native bridge of Zap/73-1 and Zap/49-1' ; ;I answer the call on my extension '5012' and talk as long ;as I care and then simply hangup. ; ;CLI indicates 'Hungup 'Zap/49-1' ;'spawn extension (main2,9,1) exited non-zero on ;Zap/73-1' ;Hungup 'Zap/73-1' ; [default] exten = s,1,answer exten = s,2,disa(no-password, main2) exten = s,3,Hangup ; -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax2 jitter stats confusion
When a call is in progress I'll be watching it at the console with iax2 show channels Here are my stats from one particular call: 66.225.202.72benshaw 1/16413 00048/00035 00489ms 0221ms ILBC 66.225.202.72benshaw 1/16413 00137/00125 00487ms 0270ms ILBC 66.225.202.72benshaw 1/16413 00137/00125 00491ms 0269ms ILBC 66.225.202.72benshaw 1/16413 00141/00129 00487ms 0241ms ILBC 66.225.202.72benshaw 1/16413 00141/00129 00480ms 0235ms ILBC 66.225.202.72benshaw 1/16413 00143/00131 00480ms 0256ms ILBC 66.225.202.72benshaw 1/16413 00144/00132 00492ms 0268ms ILBC 66.225.202.72benshaw 1/16413 00152/00140 00487ms 0472ms ILBC 66.225.202.72benshaw 1/16413 00154/00142 00507ms 0473ms ILBC Now I figured the guy would be coming up to my office shooting but when I asked him how the call was he said perfect. -- now he knows he's on a VOIP call but he had no idea of the jitter and lag here... So I suppose my question is huh? How can I have such poor jitter and yet have this guy (not a techie) claim the call was perfect? Neither he nor the guy on the other end (PSTN through NuFone) had any issues about the quality. I'll take a stab at this, but you'll probably get as many opinions as there are readers. Jitter is reflective of the variation in packet delay between end points, not a measurement of audio quality. If none/few of the packets are dropped, the user wouldn't even notice other then maybe a click or something. The fact that delay exits and the variation in the delay is rather large doesn't mean it will impact quality. However, the opposite might be true: if quality were poor and you found jitter to be very high, then jitter is likely the symptom and not the root cause. If your user would have an analog pstn call going on simultanously, he would notice the significant delay. Likewise, if the VoIP call had any echo characteristics, he'd notice the delay. It would appear the necessary data packets are arriving in such a way as to allow the jitter buffer to do what its supposed to do, and apparently doing it very well. Unless you're using satellite, it would appear the delay numbers are rather high in my opinion. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Annoying Beeps
Steven Critchfield wrote: On Fri, 2004-02-06 at 13:46, Stephen R. Besch wrote: Every once and a while * throws a new wrinkle at me. It has started, all on its own, to make these annoying little beeps evey time a message prints at the CLI. If I bring down * and restart, they go away for a time, then seem to spontaneously reappear sometime later. It's almost as if * is starting to experience the Terrible Twos! No one else seems to be complaining about this, but I nevertheless assume that I can somehow disable this feature, I just can't seem to find out how. Maybe something like CLI stop beeping damit? I think that is due to there being a character on the CLI. Try hitting enter to clear the line, or hit ctrl-l to do a screen redraw and see whats on the line. That was it. I even found out how the characters (//) get typed. I have a KVM switch to pop between systems and I also have an annoying habit of hitting // rather than ctrlctrl to switch screens. It's amazing the trouble that a little sloppy typing can get one into! Thanks Steven. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The Evil of type=friend explained, again (was Re: [Asterisk-Users] Minor Registration Problem With Polycom Soun dpoint IP 500)
Tilghman Lesher wrote: On Thursday 05 February 2004 05:50, Jeremy McNamara wrote: A type=friend is simply both a type=user and type=peer using the same set of config directives. While a type=friend makes things almost trivial to get calls working in both directions, it will limit the flexibility of your config and even hinder some of the more advanced uses of Asterisk. For example: Say you want to use the same 'user' across many different Asterisk boxes, which of course will have different IP addresses. In this situation, you cannot have a host keyword in your Asterisk config stanza for the type=user, but the type=peer requires some host keyword. Thus, if you use a type=friend you will limit the use of that one username to whatever IP address is contained in the host keyword. You only need to register to Asterisk if you have a dynamic IP address or you need to blow thru a firewall/NAT device. To register you need to have a type=peer with a host=dynamic. Since in your type=friend config directive you had host=some.ip.address, while this may be this is fine to for the type=user, this same value also gets used for the type=peer, which makes it so you cannot register since the IP address is hard coded. So, either you do not need to register and things will Just Work(tm) or you will need to use separate type=user and type=peer config directives. So, why can't you just do: [someuser] type=friend host=dynamic context=internal secret=somesecret In other words, you can have your user registered to the server AND be using a type=friend definition. This is exactly how I have some test equipment set up and it works perfectly well. Sure, but then you are not restricting to any specific IP address to authenticate users and you will request the internal context on the far end when sending them calls. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] is it possible to turn auto answer off and on in the dialplan?
Is it possible to turn auto answer for the console off and on in the dialplan? If so would someone be so kind as to post a short example. I'd like to use the same sound card for external ringing over the paging system that I'm using for overhead paging. So my idea was to put the console in the group of phones to ring when a call comes in, which would ring over the speakers. But I'd like to keep being able to do over head paging by dialing an extension. I'd to have autoanswer=no in alsa.conf, and do something like ; overhead paging exten = 4600,1,SetMusicOnHold(silence) exten = 4600,2,SetAutoAnswerOff(CONSOLE/dsp) exten = 4600,3,Dial(CONSOLE/dsp) exten = 4600,4,Hangup instead of what i have now: ; overhead paging exten = 4600,1,SetMusicOnHold(silence) exten = 4600,2,Dial(CONSOLE/dsp) exten = 4600,3,Hangup Otherwise, I'd need to add another soundcard as console2 and run both outputs into the input of the paging system, correct? Thanks ahead, Jeff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fast question on extension matching
Dear All, I have a very simple question but could not find any information from the internet. Is there anyway to match code on extensions.conf without having to specify the number of digits? For example, if I want to send 01163 (Philippines to a certain IP address), is there anyway simpler to do than exten = _01163,1,. exten = _01163XXX,1,. exten = _01163XX,1,. exten = _01163X,1,. exten = _01163,1,. exten = _01163XXX,1,. Is there any one line command that could replace having to use XX... to match exact number of digits? Thanks TC --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on ebay.
While looking around for some ISDN phones I found this auction and thought some of you may get a kick out of this. http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3075387057category=11908 Seems they are selling a 2u server a T100P card and 10 Budgetone phones for $3995. What I find funny is that the auction limits the extensions to 50. Seems that with VoIP, it is virtually unlimited. Also They show the conference rooms limited to 10. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] busy status
On the 7960's with *, when an internal sip line is dialed, is it possible for the 7960 to display a status on the lcd that 'this ext is busy', etc. if the line is in use ? Does this happen by default ? Thanks, Chris Clifton ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] question for oh323 users
Hi Gus, Thanks for your reply. I have tried below and still didn't work. exten = _1905XXX,1,Dial,OH323/h323:[EMAIL PROTECTED] or exten = _1905XXX,1,Dial,OH323/[EMAIL PROTECTED] and now asterisk gives out below error Feb 6 16:12:41 WARNING[30740]: pbx.c:1773 ast_pbx_run: Channel 'H323:8915' sent into invalid extension 's' in context 'default', but no invalid handler here is exactly what I have in extension.conf [general] static=yes writeprotect=no [default] include = demo [demo] exten = _1905XXX,1,Dial,OH323/[EMAIL PROTECTED] Any idea? Regards, Anthony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fast question on extension matching
T. Chan wrote: Dear All, I have a very simple question but could not find any information from the internet. Is there anyway to match code on extensions.conf without having to specify the number of digits? For example, if I want to send 01163 (Philippines to a certain IP address), is there anyway simpler to do than exten = _01163,1,. exten = _01163XXX,1,. exten = _01163XX,1,. exten = _01163X,1,. exten = _01163,1,. exten = _01163XXX,1,. Is there any one line command that could replace having to use XX... to match exact number of digits? Thanks TC TC, Just do something like: exten = _01163.,1,Application() Cheers, Chris. iax700.824.0300 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on ebay.
They're not the only ones doing that. Check out the IP Telephony Solutions section and open the PDF for their SIP Media Gateway and PBX product. http://www.hautespot.net/products/index.html I'm going to find out how much they're charging for it shortly... Thanks, Francois -Original Message- From: Steven Critchfield [mailto:[EMAIL PROTECTED] Sent: Friday, February 06, 2004 15:12 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk on ebay. While looking around for some ISDN phones I found this auction and thought some of you may get a kick out of this. http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3075387057category=1 1908 Seems they are selling a 2u server a T100P card and 10 Budgetone phones for $3995. What I find funny is that the auction limits the extensions to 50. Seems that with VoIP, it is virtually unlimited. Also They show the conference rooms limited to 10. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] modprobe wcfxs
OK, folks... I'm having the same problem as a few people. device not found when I do the modprobe wcfxs. I looked in the archives, and I see 4 or 5 people have had the same problem. I even foudn the reply to a post like mine that said look in the archives, others have had the same problem. Very true, but I can't find the answer. If someone can simply point me to the archive with the solution, I can go from there. :) Thanks, Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Silencing Background App during touch tone detection
On Fri, 2004-02-06 at 14:08, Brian West wrote: Post your config and we can see whats up.. Incoming lines are via a FXO card in a CAC channel bank, although we had same issue with the lines connected to X100P cards. T1 Channels 1-12 are FXS to the handsets, 13-24 are FXO, only 13 - 16 are connected to POTS lines. ---zaptel.conf--- span=1,0,0,esf,b8zs fxoks=1-12 fxsls=13-24 loadzone=us defaultzone=us ---zaptel.conf--- ---zapata.conf--- ;Example user extension, all are identical adsi=yes echocancel=yes threewaycalling=yes transfer=yes signalling=fxo_ks context=local group=1 callerid=Internal Extension 412 mailbox=12 channel=12 ;Incoming POTS group=2 adsi=no echocancel=yes threewaycalling=yes transfer=no signalling=fxs_ls context=default rxgain=0 txgain=3 callerid=POTS 1 345-xxx1 context=default channel=13 callerid=POTS 2 345-xxx2 context=default channel=14 callerid=POTS 3 345-xxx3 context=default channel=15 callerid=POTS 4 345-xxx4 context=default channel=16 group=4 channel=17-20 --extensions.conf-- [default] exten = s,1,Answer exten = s,2,DigitTimeout,15 exten = s,3,ResponseTimeout,30 exten = s,4,Wait(1) ; Please enter extension, or if you do not know extension ; a directory will follow exten = s,5,Background(ticom-intro) exten = s,6,Wait(3) exten = s,7,Background(user-1-name) exten = s,8,Background(user-2-name) exten = s,9,Background(user-3-name) exten = s,10,Background(user-4-name) exten = s,11,Background(user-5-name) exten = s,12,Background(user-6-name) exten = s,13,Background(user-7-name) ; Example user extension exten = 10,1,Dial(Zap/12,15,t) exten = 10,2,Voicemail2(u10) exten = 10,3,Hangup exten = 10,102,Voicemail2(b10) exten = 10,103,Hangup [local] ;everything else is pretty much straight from the demo --extensions.conf-- On Fri, 6 Feb 2004, Mark Farver wrote: We're still have problems with the outgoing voice message interfering with the touch tone detection. Often the first touch tone pressed will be detected twice. If I configure asterisk to not play the message, or if people wait till the outgoing message stops, it works flawlessly. I've noticed that many phone system silence the outgoing message once you start pressing buttons, is there anyway to configure asterisk to do the same? My C skills are non-existent, but the touch tone detection code in the zaptel libraries appears to work this way (apologies for basic like syntax): start: Wait for tone detected waiting: wait ~250ms still hearing tone? Yes, goto waiting No.. Add that tone to the queue, goto start In the above a short dropout that falls exactly on one of the 250ms checks would be detected as a break, even is it was only a few ms long. In my experience TT detection system work more like this, require a minimum length of silence/no Touch tone (100-250ms) before advancing to detecting the next number: start: wait for tone detected waiting: wait 250ms still hearing tone? Yes, goto waiting No.. wait 100-250ms, Still no tone? yes, add current tone to queue and goto start no, ignore silence, and goto waiting Thanks Mark Farver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP - NATIVE BRIDGE ERROR
Hi, Running Version 0.7.2, I receive the following error when attempting to connect two SIP Devices. WARNING[16399]:rtp.c : 1204 ast_rtp_bridge : codec0 = 524556 is not codec1 = 524558, cannot native bridge. The bridge is made but the quality of the call is bad, a lot of disturbing noises in background. Oddly enough, both devices are using only one codec G729. I also am using the demo G729 license for Asterisk. I'm not sure how 2 different codecs are being found. I saw in ast_rtp_bridge function, that the get_codec function returned these values. Could anyone tell me where the get_codec function is? Curious as to how this is happening. Should this problem be added to the bug tracker? The SIP calls are very bad, and I did not experience this problem with 0.5.0 . Thanks, Wes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interrupted musiconhold sound when silence supression is enabled
George Ye wrote: Hi, I am a new player of the Asterisk. I have a strage problem with musiconhold feature. Can anyone give some clues what might be the problem? A description of the problem is as follows: 1. Call from Cisco ATA 186 without silence suppression, when I push the "hold" button at the Cisco 7960 IP phone, the music plays just fine. 2. Call from Cisco ATA 186 with silence suppression enable, when I push the "hold" button of phone, the music is annoying, it cannot play smoothly. Sometimes, it plays well for a while, then there is a pause, then plays again, then pause,... * uses the incoming RTP Stream as a timing source for sending its outgoing Stream.. If the incoming stream is interrupted due to silence suppression then musiconhold will be choppy. So in conclusion, you cannot use sile nce suppresion. Thanks Andre for the clarification. However, sometimes, one might not be able to disable the silence supression at the remote endpoint. I am thinkingof palying the music independently by another thread which uses local timing and checks the channel state (musiconhold or not) during its running. Any comments are welcome. Thanks. George Certainly, an obvious solution is to disable silence suppression, however, this is unpracticable, because, sometimes, you might have no control of the remote side. Thanks in advance. George Do you Yahoo!? Yahoo! Finance: Get your refund fast by filing online
[Asterisk-Users] RE:voiceglo sip config
Hi, I am trying to get voiceglo to work with asterisk. I have tried many sip configs and cant seem to get it to register. Please if someone can look at this softphone config and let me know what I am doing wrong I would appreciated it. Thanks John Bittner Simlab.net This is my config and the softphone config listed below. [general] port=5060 bindaddr=0.0.0.0 tos=lowdelay disallow=all allow=gsm allow=ulaw allow=alaw maxexpirey=180 defaultexpirey=160 tos=reliability register=973111:[EMAIL PROTECTED] [myphone.voiceglo.com] type=friend secret=UPUIOPHXDTV username=973111 host=myphone.voiceglo.com context=incoming [HKEY_CURRENT_USER\Software\SJLabs\SJvoip Project\SJphone\Options\SIP 2.0] [HKEY_CURRENT_USER\Software\SJLabs\SJvoip Project\SJphone\Options\SIP 2.0] RedirectAutoIgnore=dword: UseApplicationSIP=dword: RedirectIgnore=dword: SeparateRegistrarAddress=172.19.1.88 UseOutboundProxy=dword:0001 SendINVITEWithoutOffer=dword: FWDBehindNAT=dword: ReRegistrationInterval=dword:0e10 RedirectDND=dword: UseSeparateRegistrarAddress=dword: RegisterOnProxy=dword:0001 QuietlyAcceptRedirect=dword:0001 RestrictCallerIdentity=dword: DisableNonProxiedCalls=dword: IgnoreRefer=dword: ConfirmTransferRequests=dword: ExposeSoftwareVersion=dword:0001 UnregisterContactOnly=dword:0001 ProxyPort=dword:13c4 TrafficDumpFileName=C:\\SIPTRAFFIC.LOG CompatibilityFlag1=dword: TrafficDumpRingBufferLength=dword:00ff SeparateRegistrarPort=dword:13c4 PreferredRegistrationTCP=dword: WorkThroughProxyOnly=dword: ProxyAddress=myphone.voiceglo.com AddressOfRecord=sip:973111.voiceglo.com ProxyUserName=973111 ProxyUserPassword=UPUIOPHXDTV ProxyDomain=myphone.voiceglo.com CallerNumber=973111 RedirectionURL= FWDNumber= FWDPassword= SeparateRegistrar= ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fast question on extension matching
Dear Chris, Thanks for your lesson, it sort of works but not perfect. I tried exten = _01163.,1,Application() exten = _011.,1,Application() because I want to send Philippines to a different IP address than the rest of the world, but if I configure that way, even 01163 calls will all go to the second IP address as per 011.,1,Application(). If I take out the 011., then calls WILL go to 01163., if I put the two together it will always go to 011. extension. Any idea please? Thanks again TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Craft Sent: Friday, February 06, 2004 4:31 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Fast question on extension matching T. Chan wrote: Dear All, I have a very simple question but could not find any information from the internet. Is there anyway to match code on extensions.conf without having to specify the number of digits? For example, if I want to send 01163 (Philippines to a certain IP address), is there anyway simpler to do than exten = _01163,1,. exten = _01163XXX,1,. exten = _01163XX,1,. exten = _01163X,1,. exten = _01163,1,. exten = _01163XXX,1,. Is there any one line command that could replace having to use XX... to match exact number of digits? Thanks TC TC, Just do something like: exten = _01163.,1,Application() Cheers, Chris. iax700.824.0300 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G.729, show command or log to confirm it's using the G.729 codec.
I installed the codec, got confirmation from the istall process. Is there show command or a log that I can use to confirm calls are using G.729. Do I need to restart asterisk or can I just reload the config? Thanks! Mark Vickers, RealNetworks Inc. Desk: (206) 674-2391 Fax: (206)674-3588 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G.729, show command or log to confirm it's using the G.729 codec.
g.729 show license usage will show you how many G.729 licenses are currently being used. Derek Samford Net Phone Blue, Inc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Fri 2/6/2004 6:19 PM To: [EMAIL PROTECTED] Cc: Subject:[Asterisk-Users] G.729, show command or log to confirm it's using the G.729 codec. I installed the codec, got confirmation from the istall process. Is there show command or a log that I can use to confirm calls are using G.729. Do I need to restart asterisk or can I just reload the config? Thanks! Mark Vickers, RealNetworks Inc. Desk: (206) 674-2391 Fax: (206)674-3588 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat
Re: [Asterisk-Users] modprobe wcfxs
On Friday 06 February 2004 16:26, Tim Sailer wrote: OK, folks... I'm having the same problem as a few people. device not found when I do the modprobe wcfxs. I looked in the archives, and I see 4 or 5 people have had the same problem. I even foudn the reply to a post like mine that said look in the archives, others have had the same problem. Very true, but I can't find the answer. If someone can simply point me to the archive with the solution, I can go from there. :) A common problem is forgetting to connect the 4-pin molex on the side of the card. If you're still having problems, you could try a greater wattage power supply or a different motherboard. It's strange, but while the TDM400P is up to the PCI spec, some motherboards are deficient. It is only when inserting a card which stresses the PCI spec to the max that you may wind up discovering this. Also, if your TDM400P does not have a molex connector, you can get a free upgrade from the company that sold you the TDM400P. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk under UML?
Does anyone know if/how well Asterisk will run under User Mode Linux? Will the ztdummy or zaprtc modules work with it? Thanks, Scott - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] modprobe wcfxs
On Fri, Feb 06, 2004 at 05:36:47PM -0600, Tilghman Lesher wrote: On Friday 06 February 2004 16:26, Tim Sailer wrote: OK, folks... I'm having the same problem as a few people. device not found when I do the modprobe wcfxs. I looked in the archives, and I see 4 or 5 people have had the same problem. I even foudn the reply to a post like mine that said look in the archives, others have had the same problem. Very true, but I can't find the answer. If someone can simply point me to the archive with the solution, I can go from there. :) A common problem is forgetting to connect the 4-pin molex on the side Nope, plugged in. I even tested it to make sure the voltages were right. of the card. If you're still having problems, you could try a greater wattage power supply or a different motherboard. Hrm. A different mother board is out of the question right now. The power supply, maybe. ATX... It's strange, but while the TDM400P is up to the PCI spec, some motherboards are deficient. It is only when inserting a card which stresses the PCI spec to the max that you may wind up discovering this. This is an older motherboard, but it never had any problems driving things like video capture which tends to stress the bus. Also, if your TDM400P does not have a molex connector, you can get a free upgrade from the company that sold you the TDM400P. Brandy new from Digium. It had better be right. :) Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fwd settings
Hi, i finally was able to getdialtone on my fxs board. !! however, i think i am missing something in the fwd setting to make work my account. i am getting an error authenticating my account could someone send me the exact settings to put on sip.conf ? to make it work ? i have my own account, password but i am getting it wrong. thanks, Francisco
[Asterisk-Users] Re: Asterisk under UML?
Scott == Scott Russ [EMAIL PROTECTED] writes: Scott Does anyone know if/how well Asterisk will run under User Mode Scott Linux? Will the ztdummy or zaprtc modules work with it? Haven't tried the modules, but an all-voip setup works well, provided there is enough ram set aside for the instance, and that the (real) cpu isn't oversubscribed. -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fast question on extension matching
T == T Chan [EMAIL PROTECTED] writes: T if I configure that way, even 01163 calls will all go to the second T IP address as per 011.,1,Application(). If I take out the 011., T then calls WILL go to 01163., if I put the two together it will T always go to 011. extension. The list archives have a lot to say about this. You need to create a context for each option and include them into the main context in the order you want them matched. So, include the context that matches _01163. before the context that matches _011. to get the ordering you want. But take a look at the archives for details. -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP - NATIVE BRIDGE ERROR
Isn't the demo codec 1 channel only? Then one side is g729 and the other is what? do a sip show channels bkw On Fri, 6 Feb 2004, Wes Marderness wrote: Hi, Running Version 0.7.2, I receive the following error when attempting to connect two SIP Devices. WARNING[16399]:rtp.c : 1204 ast_rtp_bridge : codec0 = 524556 is not codec1 = 524558, cannot native bridge. The bridge is made but the quality of the call is bad, a lot of disturbing noises in background. Oddly enough, both devices are using only one codec G729. I also am using the demo G729 license for Asterisk. I'm not sure how 2 different codecs are being found. I saw in ast_rtp_bridge function, that the get_codec function returned these values. Could anyone tell me where the get_codec function is? Curious as to how this is happening. Should this problem be added to the bug tracker? The SIP calls are very bad, and I did not experience this problem with 0.5.0 . Thanks, Wes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] busy status
Nope. bkw On Fri, 6 Feb 2004, Chris Clifton wrote: On the 7960's with *, when an internal sip line is dialed, is it possible for the 7960 to display a status on the lcd that 'this ext is busy', etc. if the line is in use ? Does this happen by default ? Thanks, Chris Clifton ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Annoying Beeps
Do you here the beeps on the phone or on the Console machine. For about the last 2 weeks I have been hearing random beeps on either of my two sip phones. I do not have a console running anywhere so I have no text printing. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Friday, February 06, 2004 3:02 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Annoying Beeps On Fri, 2004-02-06 at 13:46, Stephen R. Besch wrote: Every once and a while * throws a new wrinkle at me. It has started, all on its own, to make these annoying little beeps evey time a message prints at the CLI. If I bring down * and restart, they go away for a time, then seem to spontaneously reappear sometime later. It's almost as if * is starting to experience the Terrible Twos! No one else seems to be complaining about this, but I nevertheless assume that I can somehow disable this feature, I just can't seem to find out how. Maybe something like CLI stop beeping damit? I think that is due to there being a character on the CLI. Try hitting enter to clear the line, or hit ctrl-l to do a screen redraw and see whats on the line. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] modprobe wcfxs
Now we're getting somewhere! The TDM400P is a PCI 2.2 card. So depending on what you mean by an older motherboard, that might be your problem. -Original Message- From: Tim Sailer [mailto:[EMAIL PROTECTED] Sent: Friday, February 06, 2004 6:53 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] modprobe wcfxs On Fri, Feb 06, 2004 at 05:36:47PM -0600, Tilghman Lesher wrote: On Friday 06 February 2004 16:26, Tim Sailer wrote: OK, folks... I'm having the same problem as a few people. device not found when I do the modprobe wcfxs. I looked in the archives, and I see 4 or 5 people have had the same problem. I even foudn the reply to a post like mine that said look in the archives, others have had the same problem. Very true, but I can't find the answer. If someone can simply point me to the archive with the solution, I can go from there. :) A common problem is forgetting to connect the 4-pin molex on the side Nope, plugged in. I even tested it to make sure the voltages were right. of the card. If you're still having problems, you could try a greater wattage power supply or a different motherboard. Hrm. A different mother board is out of the question right now. The power supply, maybe. ATX... It's strange, but while the TDM400P is up to the PCI spec, some motherboards are deficient. It is only when inserting a card which stresses the PCI spec to the max that you may wind up discovering this. This is an older motherboard, but it never had any problems driving things like video capture which tends to stress the bus. Also, if your TDM400P does not have a molex connector, you can get a free upgrade from the company that sold you the TDM400P. Brandy new from Digium. It had better be right. :) Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Message Not Delivered
--- Attention: Non-Delivery Report --- This report is generated by the email server at: mantraent.com The message with subject: RE: [Asterisk-Users] modprobe wcfxs and attached to this report was not delivered to the following recipients: Address: [EMAIL PROTECTED] Reason: 5.7.1 Unable to relay for [EMAIL PROTECTED] (550) -- ---BeginMessage--- Now we're getting somewhere! The TDM400P is a PCI 2.2 card. So depending on what you mean by an older motherboard, that might be your problem. -Original Message- From: Tim Sailer [mailto:[EMAIL PROTECTED] Sent: Friday, February 06, 2004 6:53 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] modprobe wcfxs On Fri, Feb 06, 2004 at 05:36:47PM -0600, Tilghman Lesher wrote: On Friday 06 February 2004 16:26, Tim Sailer wrote: OK, folks... I'm having the same problem as a few people. device not found when I do the modprobe wcfxs. I looked in the archives, and I see 4 or 5 people have had the same problem. I even foudn the reply to a post like mine that said look in the archives, others have had the same problem. Very true, but I can't find the answer. If someone can simply point me to the archive with the solution, I can go from there. :) A common problem is forgetting to connect the 4-pin molex on the side Nope, plugged in. I even tested it to make sure the voltages were right. of the card. If you're still having problems, you could try a greater wattage power supply or a different motherboard. Hrm. A different mother board is out of the question right now. The power supply, maybe. ATX... It's strange, but while the TDM400P is up to the PCI spec, some motherboards are deficient. It is only when inserting a card which stresses the PCI spec to the max that you may wind up discovering this. This is an older motherboard, but it never had any problems driving things like video capture which tends to stress the bus. Also, if your TDM400P does not have a molex connector, you can get a free upgrade from the company that sold you the TDM400P. Brandy new from Digium. It had better be right. :) Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End Message---
Re: [Asterisk-Users] modprobe wcfxs
On Fri, Feb 06, 2004 at 07:58:09PM -0500, Sean Cheesman wrote: Now we're getting somewhere! The TDM400P is a PCI 2.2 card. So depending on what you mean by an older motherboard, that might be your problem. It's a dual Celery (A-Bit, I think) board from around 1999-2000. Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] fwd settings
SIP.CONF
[general]
; Codecs your choice
disallow=all
;allow=gsm
allow=ulaw
allow=alaw
;allow=ilbc
;allow=spx
allow=g723
allow=g729
register=1234:[EMAIL PROTECTED]/5000
[fwd.pulver.com]
type=friend
secret=password
username=1234
host=fwd.pulver.com
context=sip
nat=yes
;ext for free world dial up
fromuser=1234
fromdomain=fwd.pulver.com
reinvite=no
canreinvite=no
EXTENSIONS.CONF
[globals]
FWDPHONE=SIP/5000
FWDUSERID=1234
FWDPASSWORD=password
FWDUSERNAME=CalleID Name
[default]
; context which is in zapata.conf
include = fwd-out
[sip]
exten = 5000,1,Dial(${FWDPHONE},30,t)
exten = 5000,2,Hangup
[fwd-out]
exten =
_7.,1,SetCallerID(${FWDUSERID})
exten =
_7.,2,SetCIDName(${FWDUSERNAME})
exten =
_7.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _7.,4,Playback(invalid)
exten = _7.,5,Hangup
www.ntfs.org
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Francisco Perez-Landaeta
Sent: 05 February 2004 19:20
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] fwd
settings
Hi, i finally was able to getdialtone on my fxs board.
!! however, i think i am missing something in the fwd setting to make work my
account.
i am getting an error authenticating my account
could someone send me the exact settings to put on sip.conf
? to make it work ?
i have my own account, password but i am getting it wrong.
thanks,
Francisco
RE: [Asterisk-Users] RE:voiceglo sip config
Hi, After allot of trial and error I found what I did wrong. I was missing the port. This config works if anyone needs it. Voiceglo config [general] port=5060 bindaddr=0.0.0.0 disallow=all allow=gsm allow=ulaw allow=alaw maxexpirey=180 defaultexpirey=160 tos=reliability register=973111:[EMAIL PROTECTED]:5060 [myphone.voiceglo.com] type=friend secret=UPUIOPHXDTV username=973111 host=dynamic nat=yes port=5060 context=incoming John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Bittner Sent: Friday, February 06, 2004 6:07 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RE:voiceglo sip config Hi, I am trying to get voiceglo to work with asterisk. I have tried many sip configs and cant seem to get it to register. Please if someone can look at this softphone config and let me know what I am doing wrong I would appreciated it. Thanks John Bittner Simlab.net This is my config and the softphone config listed below. [general] port=5060 bindaddr=0.0.0.0 tos=lowdelay disallow=all allow=gsm allow=ulaw allow=alaw maxexpirey=180 defaultexpirey=160 tos=reliability register=973111:[EMAIL PROTECTED] [myphone.voiceglo.com] type=friend secret=UPUIOPHXDTV username=973111 host=myphone.voiceglo.com context=incoming [HKEY_CURRENT_USER\Software\SJLabs\SJvoip Project\SJphone\Options\SIP 2.0] [HKEY_CURRENT_USER\Software\SJLabs\SJvoip Project\SJphone\Options\SIP 2.0] RedirectAutoIgnore=dword: UseApplicationSIP=dword: RedirectIgnore=dword: SeparateRegistrarAddress=172.19.1.88 UseOutboundProxy=dword:0001 SendINVITEWithoutOffer=dword: FWDBehindNAT=dword: ReRegistrationInterval=dword:0e10 RedirectDND=dword: UseSeparateRegistrarAddress=dword: RegisterOnProxy=dword:0001 QuietlyAcceptRedirect=dword:0001 RestrictCallerIdentity=dword: DisableNonProxiedCalls=dword: IgnoreRefer=dword: ConfirmTransferRequests=dword: ExposeSoftwareVersion=dword:0001 UnregisterContactOnly=dword:0001 ProxyPort=dword:13c4 TrafficDumpFileName=C:\\SIPTRAFFIC.LOG CompatibilityFlag1=dword: TrafficDumpRingBufferLength=dword:00ff SeparateRegistrarPort=dword:13c4 PreferredRegistrationTCP=dword: WorkThroughProxyOnly=dword: ProxyAddress=myphone.voiceglo.com AddressOfRecord=sip:973111.voiceglo.com ProxyUserName=973111 ProxyUserPassword=UPUIOPHXDTV ProxyDomain=myphone.voiceglo.com CallerNumber=973111 RedirectionURL= FWDNumber= FWDPassword= SeparateRegistrar= ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] modprobe wcfxs
I believe it is a requirement. When I bought mine, I had the same issue. After talking to Digium, I was informed that the card would not be recognized in a non-PCI 2.2 slot. I put it in another (newer) box and it came right up. -Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] Sent: Friday, February 06, 2004 8:03 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] modprobe wcfxs Now we're getting somewhere! The TDM400P is a PCI 2.2 card. So depending on what you mean by an older motherboard, that might be your problem. Um, the TDM400P is PCI 2.2 compliant. PCI 2.2 is not a requirement to my knowledge. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] busy status
As a follow up, looks like the polycom ip phones support this via their 'buddy watch' presence feature. Anyone else used this on recent polycom soundpoint ip 500 or 600 phones with * ? Chris Clifton - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, February 06, 2004 7:45 PM Subject: Re: [Asterisk-Users] busy status Nope. bkw On Fri, 6 Feb 2004, Chris Clifton wrote: On the 7960's with *, when an internal sip line is dialed, is it possible for the 7960 to display a status on the lcd that 'this ext is busy', etc. if the line is in use ? Does this happen by default ? Thanks, Chris Clifton ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CVS Changes (NAT-SIP)
I was able to resolve this problem, after removing and adding back the port settings in the firewall. I changed hardware and IP's. So I can only guess that arp table was messed up. I'm sure rebooting the firewall would have given me the same result. But everything has been working fine since then. Not sure if this helps. -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Flagg Posted At: Friday, February 06, 2004 1:27 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] CVS Changes (NAT-SIP) Subject: Re: [Asterisk-Users] CVS Changes (NAT-SIP) I am having the same problem with a new CVS. Patrick also has the problem here http://lists.digium.com/pipermail/asterisk-users/2004-January/035114.htm l Keven had a problem here http://lists.digium.com/pipermail/asterisk-users/2004-January/035262.htm l but was able to get it fixed. Can you post a patch?. My asterisk computer is multi-homed behind NAT so maybe that is a factor? Is Asterisk behind NAT working with a new CVS for anybody? Thanks, - Original Message - From: Asterisk User Group [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, January 19, 2004 10:16 PM Subject: [Asterisk-Users] CVS Changes (NAT-SIP) I had been running an older patched CVS to get VOIP working with NAT and everything had been running fine. I just built * on a new box with CVS-01/18/04-12:19:25. And now I can get remote SIP users to register. Has anything major changed... [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to externip = 69.132.68.17 ; Address that we're going to put in SIP messages if we're behind a NAT localnet = 192.168.1.0 ; Internal NETWORK address localmask = 255.255.255.0 ; Internal netmask context = default ; Default for incoming calls ;srvlookup = yes; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=ilbc [1001] type=friend secret=1001 host=dynamic username=1001 mailbox=1001 context=local nat=no [1006] type=friend secret=oicu812 host=dynamic username=1006 mailbox=1006 context=local nat=yes canreinvite=no qualify=500 Internal SIP users can register it just the outside users. -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE:voiceglo sip config
What happens when you use a service like voiceglo on * with the unlimited plan? Can you make multiple calls at the same time? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Bittner Sent: Friday, February 06, 2004 8:15 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RE:voiceglo sip config Hi, After allot of trial and error I found what I did wrong. I was missing the port. This config works if anyone needs it. Voiceglo config [general] port=5060 bindaddr=0.0.0.0 disallow=all allow=gsm allow=ulaw allow=alaw maxexpirey=180 defaultexpirey=160 tos=reliability register=973111:[EMAIL PROTECTED]:5060 [myphone.voiceglo.com] type=friend secret=UPUIOPHXDTV username=973111 host=dynamic nat=yes port=5060 context=incoming John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Bittner Sent: Friday, February 06, 2004 6:07 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RE:voiceglo sip config Hi, I am trying to get voiceglo to work with asterisk. I have tried many sip configs and cant seem to get it to register. Please if someone can look at this softphone config and let me know what I am doing wrong I would appreciated it. Thanks John Bittner Simlab.net This is my config and the softphone config listed below. [general] port=5060 bindaddr=0.0.0.0 tos=lowdelay disallow=all allow=gsm allow=ulaw allow=alaw maxexpirey=180 defaultexpirey=160 tos=reliability register=973111:[EMAIL PROTECTED] [myphone.voiceglo.com] type=friend secret=UPUIOPHXDTV username=973111 host=myphone.voiceglo.com context=incoming [HKEY_CURRENT_USER\Software\SJLabs\SJvoip Project\SJphone\Options\SIP 2.0] [HKEY_CURRENT_USER\Software\SJLabs\SJvoip Project\SJphone\Options\SIP 2.0] RedirectAutoIgnore=dword: UseApplicationSIP=dword: RedirectIgnore=dword: SeparateRegistrarAddress=172.19.1.88 UseOutboundProxy=dword:0001 SendINVITEWithoutOffer=dword: FWDBehindNAT=dword: ReRegistrationInterval=dword:0e10 RedirectDND=dword: UseSeparateRegistrarAddress=dword: RegisterOnProxy=dword:0001 QuietlyAcceptRedirect=dword:0001 RestrictCallerIdentity=dword: DisableNonProxiedCalls=dword: IgnoreRefer=dword: ConfirmTransferRequests=dword: ExposeSoftwareVersion=dword:0001 UnregisterContactOnly=dword:0001 ProxyPort=dword:13c4 TrafficDumpFileName=C:\\SIPTRAFFIC.LOG CompatibilityFlag1=dword: TrafficDumpRingBufferLength=dword:00ff SeparateRegistrarPort=dword:13c4 PreferredRegistrationTCP=dword: WorkThroughProxyOnly=dword: ProxyAddress=myphone.voiceglo.com AddressOfRecord=sip:973111.voiceglo.com ProxyUserName=973111 ProxyUserPassword=UPUIOPHXDTV ProxyDomain=myphone.voiceglo.com CallerNumber=973111 RedirectionURL= FWDNumber= FWDPassword= SeparateRegistrar= ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] busy status
The next thing is getting the polycom to work.. but yes I think its the publish and subscribe stuff that will do this. Asterisk doesn't support that and neither does the 7960. bkw On Fri, 6 Feb 2004, Chris Clifton wrote: As a follow up, looks like the polycom ip phones support this via their 'buddy watch' presence feature. Anyone else used this on recent polycom soundpoint ip 500 or 600 phones with * ? Chris Clifton - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, February 06, 2004 7:45 PM Subject: Re: [Asterisk-Users] busy status Nope. bkw On Fri, 6 Feb 2004, Chris Clifton wrote: On the 7960's with *, when an internal sip line is dialed, is it possible for the 7960 to display a status on the lcd that 'this ext is busy', etc. if the line is in use ? Does this happen by default ? Thanks, Chris Clifton ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] busy status
I've posted this text before, but... SoundPoint® IP supports shared call appearances (SCA) using the SUBSCRIBE-NOTIFY method in the SIP Specific Event Notification framework (RFC 3265). This from the admin guide at http://www.polycom.com/common/pw_item_show_doc/0,1276,2545,00.pdf Would love to see somebody take a whack at getting this to work for these phones. John - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, February 06, 2004 9:13 PM Subject: Re: [Asterisk-Users] busy status The next thing is getting the polycom to work.. but yes I think its the publish and subscribe stuff that will do this. Asterisk doesn't support that and neither does the 7960. bkw On Fri, 6 Feb 2004, Chris Clifton wrote: As a follow up, looks like the polycom ip phones support this via their 'buddy watch' presence feature. Anyone else used this on recent polycom soundpoint ip 500 or 600 phones with * ? Chris Clifton - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, February 06, 2004 7:45 PM Subject: Re: [Asterisk-Users] busy status Nope. bkw On Fri, 6 Feb 2004, Chris Clifton wrote: On the 7960's with *, when an internal sip line is dialed, is it possible for the 7960 to display a status on the lcd that 'this ext is busy', etc. if the line is in use ? Does this happen by default ? Thanks, Chris Clifton ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Caller-ID is being sent wrong. How to fix it?
I'm running Asterisk 0.5.0 and using Cisco 7960 phones in a sip only configuration currently. Everything is working except that caller ID is hosed. Say for example extension 100 calls extension 200. 200 sees 100 as the name but 200 as the number. IE, it gets its own number as the supposed CLID of the calling party. This is flat out wrong. Am I doing something wrong or is Asterisk just terribly broken with respect to sending caller ID information properly? Is this something that only effects Cisco phones? Thanks, John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] busy status
Can anyone say for certain whether the polycom ip 500 / 600 work with * ? From what I've seen from googling, it appears that they would. Network Computing seemed to think highly of them - http://www.nwc.com/shared/printArticle.jhtml?article=/1416/1416f2full.htmlpub=nwc - Chris Clifton - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, February 06, 2004 10:13 PM Subject: Re: [Asterisk-Users] busy status The next thing is getting the polycom to work.. but yes I think its the publish and subscribe stuff that will do this. Asterisk doesn't support that and neither does the 7960. bkw On Fri, 6 Feb 2004, Chris Clifton wrote: As a follow up, looks like the polycom ip phones support this via their 'buddy watch' presence feature. Anyone else used this on recent polycom soundpoint ip 500 or 600 phones with * ? Chris Clifton - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, February 06, 2004 7:45 PM Subject: Re: [Asterisk-Users] busy status Nope. bkw On Fri, 6 Feb 2004, Chris Clifton wrote: On the 7960's with *, when an internal sip line is dialed, is it possible for the 7960 to display a status on the lcd that 'this ext is busy', etc. if the line is in use ? Does this happen by default ? Thanks, Chris Clifton ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: ISDN update
Klaus-Peter Junghanns [EMAIL PROTECTED] said: bristuff 0.0.2rc7 is available now too. Including a zaptel driver for the HFC-S PCI A based ISDN cards (with echo cancelation, TE and NT mode). If I followed this a bit and understood it correctly, with e.g. the quad BRI card it will be possible to put * between the NT1's and an existing ISDN PBX? This would be a great way of supplementing the functionality of existing PBXes! -- Cees de Groot http://www.tric.nl [EMAIL PROTECTED] tric, the new way helpdesk/ticketing software, VoIP/CTI, web applications, custom development ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
