Re: [Asterisk-Users] bristuff-0.2.0-RC8n problems and kernel panic
Hi, this is my /etc/modprobe.d/zaptel: options torisa base=0xd alias char-major-196 torisa install tor2 /sbin/modprobe --ignore-install tor2 /sbin/ztcfg install torisa /sbin/modprobe --ignore-install torisa /sbin/ztcfg install wcusb /sbin/modprobe --ignore-install wcusb /sbin/ztcfg install wcfxo /sbin/modprobe --ignore-install wcfxo /sbin/ztcfg install wcfxs /sbin/modprobe --ignore-install wcfxs /sbin/ztcfg install ztdynamic /sbin/modprobe --ignore-install ztdynamic /sbin/ztcfg install ztd-eth /sbin/modprobe --ignore-install ztd-eth /sbin/ztcfg install wct1xxp /sbin/modprobe --ignore-install wct1xxp /sbin/ztcfg install wct4xxp /sbin/modprobe --ignore-install wct4xxp /sbin/ztcfg install wcte11xp /sbin/modprobe --ignore-install wcte11xp /sbin/ztcfg alias wctdm wcfxs and this is my /etc/init.d/asterisk made by me: #!/bin/sh ztcfg -s # unload wcfxs module because I must load # qozap module first! /sbin/rmmod wcfxs /sbin/rmmod zaptel # Now I load all the modules in the right order /sbin/insmod /lib/modules/2.6.8-2-386/misc/zaptel.ko /sbin/insmod /lib/modules/2.6.8-2-386/misc/qozap.ko /sbin/insmod /lib/modules/2.6.8-2-386/misc/wcfxs.ko ztcfg -vv # this is to exec asterisk as asterisk user chown --recursive asterisk:asterisk /dev/zap chmod --recursive u=rwx,g=rx /dev/zap chown asterisk /dev/tty9 sudo -u asterisk /usr/sbin/safe_asterisk and it magically works (!!!) even if modifying debian zaptel and wcfxs modules loading sequence should be a better way to solve the problem but I don't know where to find that damned sequence. Giorgio Incantalupo This Tzafrir Cohen wrote: On Mon, Nov 07, 2005 at 03:43:03PM +0100, gincantalupo wrote: Hi, I had some problems to with a quadBRI with a 2.6 kernel debian distro. Have you tried to insmod the zaptel.ko module instead of modprobing? It worked for me, hope it will work for you too. Giorgio Incantalupo Could you please give more details? One thing you should try to do is remove the automatic run of ztcfg at module load time. Practically: rem-out all the lines in /etc/modprobe.d/zaptel . There is some black-magic claim that if you un ztcfg more than once it may cause a problem to a configured zaphfc module. Don't forget to run ztcfg manually (or in an init.d script) later. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MP3 or OGG
Hi Waldo, Doesn't * record to .gsm file initially and then convert these to .wav later? I may be totally off the mark here, and if I am, I welcome the correction. In that case, why not leave the files in .gsm format instead of translating them into another lossy format? Obviously if * records conversations as .wav files then I'd be leaning toward Speex (Vorbis) as it is a suited to speech compression format. Both Speex and ogg are Open Source, therefore patent issues are likely non-existent. MP3, otoh, is fine if you use one of their approved apps, and not if you use anything else. I'm steering clear of .mp3 (and have been for quite a few years now). -- Regards, Hilton Travis Phone: +61 (0)7 3344 3889 (Brisbane, Australia) Phone: +61 (0)419 792 394 Manager, Quark IT http://www.quarkit.com.au Quark Group http://quarkgroup.com.au/ Microsoft Small Business Specialists http://www.threatcode.com/ -- its now time to shame poor coders into writing code that is acceptable for use on today's networks War doesn't determine who is right. War determines who is left. This document and any attachments are for the intended recipient only. It may contain confidential, privileged or copyright material which must not be disclosed or distributed. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Waldo Rubinstein Sent: Tuesday, 8 November 2005 11:32 Wasn't aware of it, but if quality is good, it makes sense since all I'm archiving is speech. Will evaluate further. Thanks, Waldo On Nov 7, 2005, at 7:14 PM, Mark Edwards wrote: I would recommend vorbis speex for this. You can get windows drivers to read speex files directly. Vorbis are the same bunch that develops ogg. Ogg and mp3 are more suited to music rather than speech. Speex is a much better fit for speech archiving. Mark -Original Message- From: BJ Weschke [mailto:[EMAIL PROTECTED] Sent: Tuesday, 8 November 2005 5:52 AM You're probably not going to be violating any patent protections by using OGG instead of MP3. As far as compression goes, I've found the difference between the two of them to be negligible. I've always used OGG when possible to stay IP safe. On 11/7/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: I'm trying to archive out call recordings and would appreciate some feedback as to which audio compression is more recommended MP3 or OGG. In the past, I've use lame to convert to MP3, but I noticed the audio volume drops significantly. Is it just a setting on the command line of lame or is OGG better? Which achieves higher compression rates while maintaining call quality? Thanks, Waldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't make calls from Asterisk IAX to other IAX
my iax.conf: [callshopcompany] type=peer host=213.61.187.150 username=X secret=X disallow=all allow=gsm --- Angelito Manansala [EMAIL PROTECTED] wrote: can you paste you iax.conf On 11/8/05, chawki hammoud [EMAIL PROTECTED] wrote: Hi: I have been having this problem for sometime that I am not able to solve and I hope someone can help. I can make VOIP calls between my Asterisk box and my VOIP provider using sip channel without a problem. But when I attempt to make a call using IAX, the call get accepted and then get a hangup message: This is the message I get when I attempt to make an IAX call: Executing Dial(OSS/dsp, IAX2/callshopcompany/0017046872001) in new stack -- Called callshopcompany/0017046872001 -- Call accepted by 213.61.187.150 (format gsm) -- Format for call is gsm -- Hungup 'IAX2/callshopcompany/1' == No one is available to answeer at this time __ Start your day with Yahoo! - Make it your home page! http://www.yahoo.com/r/hs ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fwd: differences between chan_capi and chan_capi-cm
Ok, I hope finally it will arrive to the list...I posted it twice...-- Forwarded message --From: Gabor Horvath [EMAIL PROTECTED]Date: 2005.11.06. 10:35Subject: differences between chan_capi and chan_capi-cmTo: Asterisk-Users list asterisk-users@lists.digium.com Can you tell me what are the main differences between chan_capi (http://www.junghanns.net/en/chan_capi.html ), and chan_capi-cm ( http://sourceforge.net/projects/chan-capi) Which one I have to use when I want to use AVM Fritz ISDN PCMCIA card?Thank you.Gabor ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sill looking for a provider
Does it say I use them? I only said that voipjet comes through at 19ms, so I disagree about the TOS. (didn't know about it anyway :) Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, November 07, 2005 5:42 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] sill looking for a provider OOPPS! Looks like someone just broke voipjet's tos gw at adcomcorp.com gw at adcomcorp.com wrote on Sat Nov 5 11:36:46 CST 2005 I tend to agree with you, my experience with Teliax has been decent, and getting better. If only I could get to them at under 20ms though, right now my latency is about 75ms whereas voipjet comes through at 19ms. Greg -- https://www.voipjet.com/tos.php NON-DISCLOSURE: ALL CUSTOMERS USING VOIPJET'S SERVICE ARE SPECIFICALLY PROHIBITED FROM DISCLOSING TO OTHERS THAT THEY USE VOIPJET'S SERVICE, THIS INCLUDES BUT IS NOT LIMITED TO, END USERS. CUSTOMERS MAY NOT DISCLOSE USE OF OR PAYMENTS TO VOIPJET ON PERSONAL, CORPORATE, LEGAL, ACCOUNTING AND OTHER DOCUMENTS AND COMMUNICATIONS UNLESS SPECIFICALLY REQUIRED TO DO SO BY LAW Has anyone else read these TOS'es??? Some are pretty funny. Thomas Herlihy Scaletta Moloney Armoring Chicago, IL USA 708.924.0099 Skype VoIP @ HerlsOne Free World Dialup 647717 [EMAIL PROTECTED] www.scaletta.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk as SIP gateway
Yes but i want to enable access for all users from that ip address. I don't want to write every user in sip.conf. greetings mk 2005/11/7, Peter Petrov [EMAIL PROTECTED]: Miloš Kocbek wrote: I want to enable access to some context in asterisk without authentication. In sip.conf: [username] type=friend host=x.x.x.x context=context_for_this_user -- Regards, Peter Petrov [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bristuff 0.2.0-RC8o or 0.2.0-RC8n (* 1.0.9)
Kernel 2.6 + CentOS 4.1 All work perfectly but Hangup() dont work in log/asterisk/full Nov 5 11:58:04 DEBUG[8299]: zt_hangup(Zap/1-1) Nov 5 11:58:04 DEBUG[8299]: Set option AUDIO MODE, value: ON(1) on Zap/1-1 Nov 5 11:58:04 DEBUG[8299]: Hangup: channel: 1 index = 0, normal = 9, callwait = -1, thirdcall = -1 Nov 5 11:58:04 DEBUG[8299]: Not yet hungup... Calling hangup once with icause, and clearing call Why ? Please help me. -- Giovanni Miano ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Which Wildcard?
Hello! We consider purchasing Digium Wildcard for E1 connectivity. Wildcards are pretty expensive pieces of silicon for small shop like ours. And we have no previous experience with E1 communications. What Wildcard do we need? How can we estimate our needs? How many clients (approx.) can share one E1 in practice, for example? What about hardware echo cancelation? Do we really need it? Thanks! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Softphone to show the activate sip user and their sip number
hi all, can i have a softphone which will showing the activate users and their sip number(sort of phone book for globally use)??? does xten provides such a feature? thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CallerID via chan_capi-cm-0.6 possible?
Hello All, I have a bri and iwsh to get CID w/name, however, even though Verizon has told me that CID/Name is on the circuit, I still only get ANI. No cid or cid/name. Anyone know if it is possible to get cid over bri? I am not sure if the issue could be in the eicon firmware or something else, since the eicon logs don't mention the CID or CIDName... Btw it is on a DMS100 switch. Regards, Greg ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to detect AGI script failure?
Thanks for the replies. I have realised that I can catch the execution
after the agi statement (if it fails) in the h priority, which I then
use to play an error message to the caller.
As you suggested, I am setting a variable in the agi script so that the
h priority knows whether the agi script succeeded or not.
Alex
Matt Riddell wrote:
Alex Hutton wrote:
Hello,
I'm new to the list so I hope I'm asking the question in the right
place. In our extensions.conf, we call an AGI script using the AGI
command.
e.g.
exten = 11,1,Answer
exten = 11,2,Wait(0.5)
exten = 11,3,Playback(welcome1)
exten = 11,4,agi(agi://192.168.1.88/hello.agi?src=test|${CALLERID})
If for some reason, the AGI script fails to run (e.g. our AGI prog isn't
running), can we detect it and direct the call to a pre-recorded message?
What I personally would do is first set a variable before you run the agi
(i.e. completionstatus to beforerun) then run the AGI. Once inside the AGI,
set the variable for completion status. I.E. you could have ran well, failed
with x etc etc. Then on the next priority, you can check this variable and
via gotoif for the various statuses (including beforerun which would mean
that the AGI didn't run at all).
While this doesn't exactly answer your question, it is the best way to use
multiple statuses.
Make sense?
___
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Re: [Asterisk-Users] How to make write and read formats equal to native format?
Branko Samardzic wrote: Any idea on how to enforce native format into read and write streams? In the peer definition (iax.conf or sip.conf) put: disallow=all allow=CODEC_YOU_WANT where CODEC_YOU_WANT is something like gsm, g729, ulaw etc (keep it to one entry at both ends and you're guaranteed they'll use it). -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which Wildcard?
Hi, Digium have Wildcard for FXO/FXS connections (i.e., telephone lines) and E1/T1 cards such as the TE110p. There're a few things you might want to consider: 1) TE110p is much more expensive 2) it is too much for a small shop. Concurrently supports upto 15 incoming and 15 outgoing calls (or 30 incoming calls). Hence, the number of clients can be up to 100, depending on your service needs and configuration. 3) you do need echo cancellation or your VoIP phone users will suffer. The lastest Digium E1 card support hardware echo cancellation. The builtin software echo cancellation is quite incapable! Hope that helps! H. On 11/8/05, Dmitry Ivanov [EMAIL PROTECTED] wrote: Hello! We consider purchasing Digium Wildcard for E1 connectivity. Wildcards are pretty expensive pieces of silicon for small shop like ours. And we have no previous experience with E1 communications. What Wildcard do we need? How can we estimate our needs? How many clients (approx.) can share one E1 in practice, for example? What about hardware echo cancelation? Do we really need it? Thanks! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sensing fax with txfax
Hello, When I try to use this: http://lists.digium.com/pipermail/asterisk-users/2003-October/025094.html for sensing a fax (putting a file sample.call in the /var/spool/asterisk/outgoing/) the call is made but after picking it up, asterisk disconnects. What can be a reason? I'm using 1.0.9 version. -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error compiling asterisk addons version 1.2.0-beta2
Dear All, I am facing a problem in compiling the add-ons for the mysql, though the files are downloaded correctly and checked and I tried different mirrors even the cvs but yet I get those errors : app_addon_sql_mysql.c:23:19: mysql.h: No such file or directory cdr_addon_mysql.c:38:19: mysql.h: No such file or directory cdr_addon_mysql.c:39:20: errmsg.h: No such file or directory res_config_mysql.c:51:19: mysql.h : No such file or directory res_config_mysql.c:52:27: mysql_version.h: No such file or directory res_config_mysql.c:53:20: errmsg.h: No such file or directory anyone has a clue, I used to compile it without problems Thx MAG ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SRTP proxy
Hello, As Asterisk do not work with SRTP, i'm finding a SRTP/RTP proxy. Any idea? thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sangoma a104d install
Hi, While a104d install on asterisk 1.2 and CVS-HEAD patch for zaptel.c failed. Is it avaiable not yet? Thanks. __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip provider problem or?
Hi! We are running an * with 3 sip providers. Provider 1 works perfect, provider 2 also. But the 3:rd one is a problem. All seems normal until we try to make a call. The phone rings by the called party and picks is up and hear only silence. The caller (local extension on the *) still gets ring tone as of no one answer the call. The providers ssw treats the call as answered and get no errors Any hints where to start looking? Regards Anders Svensson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sensing fax with txfax
Bartosz Piec wrote: Hello, When I try to use this: http://lists.digium.com/pipermail/asterisk-users/2003-October/025094.html for sensing a fax (putting a file sample.call in the /var/spool/asterisk/outgoing/) the call is made but after picking it up, asterisk disconnects. What can be a reason? I'm using 1.0.9 version. Did you install spandsp? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bristuff-0.2.0-RC8n problems and kernel panic
It's a problem with bristuff that has been there for quite some time. If you load the modules in the wrong order it will kernel panic the box. I have been bitten by it many times, very frustrating if you are working on a remote box I manually load all modules now too On Tue, 8 Nov 2005, gincantalupo wrote: Hi, this is my /etc/modprobe.d/zaptel: options torisa base=0xd alias char-major-196 torisa install tor2 /sbin/modprobe --ignore-install tor2 /sbin/ztcfg install torisa /sbin/modprobe --ignore-install torisa /sbin/ztcfg install wcusb /sbin/modprobe --ignore-install wcusb /sbin/ztcfg install wcfxo /sbin/modprobe --ignore-install wcfxo /sbin/ztcfg install wcfxs /sbin/modprobe --ignore-install wcfxs /sbin/ztcfg install ztdynamic /sbin/modprobe --ignore-install ztdynamic /sbin/ztcfg install ztd-eth /sbin/modprobe --ignore-install ztd-eth /sbin/ztcfg install wct1xxp /sbin/modprobe --ignore-install wct1xxp /sbin/ztcfg install wct4xxp /sbin/modprobe --ignore-install wct4xxp /sbin/ztcfg install wcte11xp /sbin/modprobe --ignore-install wcte11xp /sbin/ztcfg alias wctdm wcfxs and this is my /etc/init.d/asterisk made by me: #!/bin/sh ztcfg -s # unload wcfxs module because I must load # qozap module first! /sbin/rmmod wcfxs /sbin/rmmod zaptel # Now I load all the modules in the right order /sbin/insmod /lib/modules/2.6.8-2-386/misc/zaptel.ko /sbin/insmod /lib/modules/2.6.8-2-386/misc/qozap.ko /sbin/insmod /lib/modules/2.6.8-2-386/misc/wcfxs.ko ztcfg -vv # this is to exec asterisk as asterisk user chown --recursive asterisk:asterisk /dev/zap chmod --recursive u=rwx,g=rx /dev/zap chown asterisk /dev/tty9 sudo -u asterisk /usr/sbin/safe_asterisk and it magically works (!!!) even if modifying debian zaptel and wcfxs modules loading sequence should be a better way to solve the problem but I don't know where to find that damned sequence. Giorgio Incantalupo This Tzafrir Cohen wrote: On Mon, Nov 07, 2005 at 03:43:03PM +0100, gincantalupo wrote: Hi, I had some problems to with a quadBRI with a 2.6 kernel debian distro. Have you tried to insmod the zaptel.ko module instead of modprobing? It worked for me, hope it will work for you too. Giorgio Incantalupo Could you please give more details? One thing you should try to do is remove the automatic run of ztcfg at module load time. Practically: rem-out all the lines in /etc/modprobe.d/zaptel . There is some black-magic claim that if you un ztcfg more than once it may cause a problem to a configured zaphfc module. Don't forget to run ztcfg manually (or in an init.d script) later. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OTAnn] Feedback
shenanigans wrote: I was interested in getting feedback from current mail group users. There is a limit to the number of times you can post this... -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which Wildcard?
On Tuesday 08 November 2005 11:34, Hugh Jackman wrote: Hi, Digium have Wildcard for FXO/FXS connections (i.e., telephone lines) and E1/T1 cards such as the TE110p. There're a few things you might want to consider: 1) TE110p is much more expensive 2) it is too much for a small shop. Concurrently supports upto 15 incoming and 15 outgoing calls (or 30 incoming calls). Hence, the number of clients can be up to 100, depending on your service needs and configuration. 3) you do need echo cancellation or your VoIP phone users will suffer. The lastest Digium E1 card support hardware echo cancellation. The builtin software echo cancellation is quite incapable! Hope that helps! H. Thank you! 100 clients is not enough. Just ordered Wildcard 406 :) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sip provider problem or?
Sorry. Forgot to say that if I connect an ip phone directly to the provider it works without problwm Anders From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anders Svensson Sent: den 8 november 2005 11:09 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Sip provider problem or? Hi! We are running an * with 3 sip providers. Provider 1 works perfect, provider 2 also. But the 3:rd one is a problem. All seems normal until we try to make a call. The phone rings by the called party and picks is up and hear only silence. The caller (local extension on the *) still gets ring tone as of no one answer the call. The providers ssw treats the call as answered and get no errors Any hints where to start looking? Regards Anders Svensson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP domain support for authentication and virtual hosting
nobody has an answer here !! --- harry gaillac [EMAIL PROTECTED] a écrit : Hello, Where may i find documentation about SIP domain support and dnsmgr.conf , Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk-1.2-bêta2 | pre sence/subscription support in the SIP channel driver
nobody has an answer here! --- harry gaillac [EMAIL PROTECTED] a écrit : Hello, I configure Polycom ip300 for presence but when status change notify is no sent to subscriber !? Why ? Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fwd: differences between chan_capi and chan_capi-cm
Can you tell me what are the main differences between chan_capi (http://www.junghanns.net/en/chan_capi.html), and chan_capi-cm ( http://sourceforge.net/projects/chan-capi) chan_capi-cm is directly derived from the last development source of chan_capi. It does contain lots of fixes and several changes. To findout more about changes there is an CHANGES file in chan-capi-cm. Which one I have to use when I want to use AVM Fritz ISDN PCMCIA card? I would suggest chan-capi-cm for any configuration. Elmar ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Detect registered peers
Hi, Is there a way to detect (in the dialplan) if a SIP peer is registered with the server ? I am using macros to dial to extension, becuase i dont want to define each extension in the dialplan, and, for example, my numbers are 8xx , i want to know if a peer exists/registered before ringing the line, i need something like Voicemailexists , but for SIP peers. any solution ? Thanks. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip provider problem or?
Anders Svensson wrote: Hi! We are running an * with 3 sip providers. Provider 1 works perfect, provider 2 also. But the 3:rd one is a problem. All seems normal until we try to make a call. The phone rings by the called party and picks is up and hear only silence. The caller (local extension on the *) still gets ring tone as of no one answer the call. The providers ssw treats the call as answered and get no errors Any hints where to start looking? Try something like: disallow=all allow=ulaw If that works then you do some trial and error to see which codecs are really supported. I remember doing this with a broadvoice account on incoming. The primary DID worked but it seemed that the tollfree virtual number did not allow the same codecs. Don't make any assumptions. Test everything. I have seen cases where termination and origination codecs allowed were different. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sensing fax with txfax
Matt Riddell napisał(a): Did you install spandsp? Yes, I have installed libtiff, spandsp, txfax and rxfax. The problem now is that asterisk doesn't disconnect but when I try to receive the fax, nothing happens. Fax (PSTN) is just waiting for receive and after some time it finishes the call. Asterisk console says only this: -- Attempting call on SIP/[EMAIL PROTECTED] for application txfax(/root/testfax.tif) (Retry 1) Channel SIP/yyy-3c49 was answered. Lauching txfax(/root/testfax.tif) on SIP/yyy-3c49 Nov 8 11:22:11 NOTICE[5079]: pbx_spool.c:239 attempt_thread: Call completed to SIP/[EMAIL PROTECTED] Is there a way to debug this somehow? Maybe there is a problem with libtiff? I have the latest 3.7.4 version. And the second question. Do you know some software fax that will work under Windows and send and receive faxes over IP? -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fwd: differences between chan_capi and chan_capi-cm
Hi, basically chan_capi-cm is a fork from chan_capi. But since chan_capi is not developed any more, chan_capi-cm has more features, is more stable and works with newer versions of Asterisk too. The main difference for the user is the change in capi.conf and the dial() syntax, which is shown in README. Armin On Tue, 8 Nov 2005, Gabor Horvath wrote: Ok, I hope finally it will arrive to the list...I posted it twice... -- Forwarded message -- From: Gabor Horvath [EMAIL PROTECTED] Date: 2005.11.06. 10:35 Subject: differences between chan_capi and chan_capi-cm To: Asterisk-Users list asterisk-users@lists.digium.com Can you tell me what are the main differences between chan_capi ( http://www.junghanns.net/en/chan_capi.html), and chan_capi-cm (http://sourceforge.net/projects/chan-capi) Which one I have to use when I want to use AVM Fritz ISDN PCMCIA card? Thank you. Gabor ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID via chan_capi-cm-0.6 possible?
There is no implementation in chan_capi-cm for CIDName yet, but if it is available via CAPI messages we can add this. Can you provide a log? (A log of chan_capi-cm with 'set verbose 5' and 'capi debug', as well as a mlog from Eicon card) Armin On Tue, 8 Nov 2005 [EMAIL PROTECTED] wrote: Hello All, I have a bri and iwsh to get CID w/name, however, even though Verizon has told me that CID/Name is on the circuit, I still only get ANI. No cid or cid/name. Anyone know if it is possible to get cid over bri? I am not sure if the issue could be in the eicon firmware or something else, since the eicon logs don't mention the CID or CIDName... Btw it is on a DMS100 switch. Regards, Greg ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Choppy Audio in Echo Test and Music On Hold (1.2.0-b2)
Do you have silence suppression enabled on your clients? Asterisk can't work with silence suppression. Take a look at http://bugs.digium.com/view.php?id=5374 , the patch works fine to me, now I'm able to set SS and save bandwidth. On Mon, 2005-11-07 at 19:41 -0800, Chris Tracy wrote: I recently resurrected an old athlon system and put CentOS 4.2 on it to play with asterisk. First I tried asterisk-1.0.9, now I'm using 1.2.0-b2. Both have the same audio issues that have me stumped. I looked through all the lists and forums and the closest I could get were some messages from 2003: http://lists.digium.com/pipermail/asterisk-users/2003-August/017171.html I've got asterisk set up with my xten-lite softphone on extension 200 over SIP. I've configured extension 611 as an echo test and 612 will play 30 seconds of MusicOnHold. I can connect to both just fine, however, they sound rather bad when they work (quite muddy) and periodically they just drop out for as much as 5 seconds before coming back. Enabling all the debugging and verbosity options, I've found a few messages that occur during each drop. During the MOH run, every time there's a drop, the console scrolls: res_musiconhold.c:535 monmp3thread: Only wrote -1 of 640 bytes to pipe over and over until the sound comes back, at which point, the console message: rtp.c:1247 ast_rtp_raw_write: Difference is 33824, ms is 4248 is displayed. (Not always the same numbers in that one, obviously) In the echo test, again, after a drop, the audio returns and a message similar to: rtp.c:1247 ast_rtp_raw_write: Difference is 12496, ms is 1582 is displayed. The asterisk server is on a single Athlon MP 1600+ (1.4GHz) with 512MB of RAM. It's got a K7D-Master mobo, and is connected to the system running the softphone through a 100Mbit LAN. I've not enabled any of the MMX optimizations as there were warnings that they didn't play nice with AMD chips. If there's any further info I can provide, I'd be happy to. Thanks, Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sensing fax with txfax
Bartosz Piec wrote: Matt Riddell napisał(a): Did you install spandsp? Yes, I have installed libtiff, spandsp, txfax and rxfax. The problem now is that asterisk doesn't disconnect but when I try to receive the fax, nothing happens. Fax (PSTN) is just waiting for receive and after some time it finishes the call. Asterisk console says only this: -- Attempting call on SIP/[EMAIL PROTECTED] for application txfax(/root/testfax.tif) (Retry 1) Channel SIP/yyy-3c49 was answered. Lauching txfax(/root/testfax.tif) on SIP/yyy-3c49 Nov 8 11:22:11 NOTICE[5079]: pbx_spool.c:239 attempt_thread: Call completed to SIP/[EMAIL PROTECTED] Maybe you could make an extension that you can dial which will run txfax for you. Then you can call it with a phone and see if you hear the fax tones. Don't forget that Fax over IP is almost impossible and only works with the ulaw/alaw codecs. And the second question. Do you know some software fax that will work under Windows and send and receive faxes over IP? Sorry, I don't. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fwd: differences between chan_capi and chan_capi-cm
Elmar Haneke wrote: I would suggest chan-capi-cm for any configuration. You know which quadbri cards it works with? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk as SIP gateway
Miloš Kocbek wrote: Yes but i want to enable access for all users from that ip address. I don't want to write every user in sip.conf. So use no secret and host=x.x.x.x where x.x.x.x is the IP address you want to allow from. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-1.2-bêta2 | presence/subscription support in the SIP channel driver
harry gaillac wrote: nobody has an answer here! Actually someone asked for you config details. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem dialling multiple SIP devices
I posted to the list with this issue a few weeks ago, but nothing really came of it. Either I'm missing something obvious (for which I apologize in advance) or this is a pretty serious issue between Asterisk and the SIP devices connected to it. I have 12 SIP phones at a particular site all connected to a local asterisk server. It's in turn connected to 2 ISDN BRIs to provide up to 4 incoming calls. At the moment, when an incoming call comes in, asterisk dials every SIP phone like so: Dial (SIP/1SIP/2etc.) If 2 calls come in only a second or two apart, the first one will cause the dial command to be executed, and when the second call comes in, it'll go to voicemail because *all* the SIP phones report themselves as busy (because they're ringing for the first call). Is there any way around this problem whilst keeping the same incoming call behaviour (i.e. call comes in, all phones ring)? Thanks in advance folks. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Choppy Audio in Echo Test and Music On Hold (1.2.0-b2)
I recently resurrected an old athlon system and put CentOS 4.2 on it to play with asterisk. First I tried asterisk-1.0.9, now I'm using 1.2.0-b2. Both have the same audio issues that have me stumped. I looked through all the lists and forums and the closest I could get were some messages from 2003: http://lists.digium.com/pipermail/asterisk-users/2003-August/017171.html I've got asterisk set up with my xten-lite softphone on extension 200 over SIP. I've configured extension 611 as an echo test and 612 will play 30 seconds of MusicOnHold. I can connect to both just fine, however, they sound rather bad when they work (quite muddy) and periodically they just drop out for as much as 5 seconds before coming back. In the xlite configuration, look for an option something like 'transmit silence' and set that to yes. (Might be called 'silence suppression', I don't remember.) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sangoma a104d install
all you have to do is manually apply the patch before Setup and it will patch fine(apply the patch in the 'zaptel' directory of wanpipe's source directory to your zaptel source). MATT--- On 11/8/05, Jason Kim [EMAIL PROTECTED] wrote: Hi, While a104d install on asterisk 1.2 and CVS-HEAD patch for zaptel.c failed. Is it avaiable not yet? Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip provider problem or?
We are running an * with 3 sip providers. Provider 1 works perfect, provider 2 also. But the 3:rd one is a problem. All seems normal until we try to make a call. The phone rings by the called party and picks is up and hear only silence. The caller (local extension on the *) still gets ring tone as of no one answer the call. The providers ssw treats the call as answered and get no errors Any hints where to start looking? Turn on sip debug from the CLI and place a test call. There should be some pretty good clues that would tell you what's happening (or not happening). Best guess... probably a codec incompatibility issue. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fwd: differences between chan_capi and chan_capi-cm
On Wed, 9 Nov 2005, Matt Riddell wrote: Elmar Haneke wrote: I would suggest chan-capi-cm for any configuration. You know which quadbri cards it works with? Any which support CAPI interface. - Eicon Diva Server (all) - AVM C4 - ... Armin -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sensing fax with txfax
Bartosz Piec wrote: Matt Riddell napisał(a): -- Attempting call on SIP/[EMAIL PROTECTED] for application txfax(/root/testfax.tif) (Retry 1) Channel SIP/yyy-3c49 was answered. Faxing on a VoIP channel is not recommended. Read the following: http://www.soft-switch.org/spandsp_faq/ar01s04.html Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk as SIP gateway
Yes but if i write [test-user] host=x.x.x.x then only users test-user will able to make calls i want that every username is allowed to call greetings mk 2005/11/8, Matt Riddell [EMAIL PROTECTED]: Miloš Kocbek wrote: Yes but i want to enable access for all users from that ip address. I don't want to write every user in sip.conf. So use no secret and host=x.x.x.x where x.x.x.x is the IP address you want to allow from. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-1.2-bêta2 | pre sence/subscription support in the SIP channel driver
Hello, Sorry here are my sip.conf and extensions.conf in fact when polycom ip300 send subscribe to buddies these one send back notify but nothing else when status change Regards Harry --- Matt Riddell [EMAIL PROTECTED] a écrit : harry gaillac wrote: nobody has an answer here! Actually someone asked for you config details. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com sip.conf Description: 3455877249-sip.conf extensions.conf Description: 3949034846-extensions.conf ?xml version=1.0 standalone=yes? directory item_list item lnbob/ln fnSINCLAR/fn ct86/ct sd1/sd bw1/bw /item /item_list /directory ?xml version=1.0 standalone=yes? directory item_list item lnalice/ln fnSPRING/fn ct84/ct sd1/sd bw1/bw /item /item_list /directory___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem dialling multiple SIP devices
Chris Bagnall wrote: If 2 calls come in only a second or two apart, the first one will cause the dial command to be executed, and when the second call comes in, it'll go to voicemail because *all* the SIP phones report themselves as busy (because they're ringing for the first call). Is there any way around this problem whilst keeping the same incoming call behaviour (i.e. call comes in, all phones ring)? Drop the incoming calls into a call queue. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with dialplan to allow breakout to DISA
Since this is my DID, I want the line to ring as normal but allow a user to breakout and ultimately get an outgoing line. In this way the DID line would function as a normal telephone line. A point lost on many responders! I don't want to have to go into voicemail to breakout since I don't want to give voicemail access to some of the people I will give targeted outgoing access to. This snippet from extensions.conf seem to work OK for internal extensions. Changing the context appears to stop the Playtones() OK. Any reasons why I shouldn't turn it lose? [incoming] exten = 1004,1,Playtones(ring) exten = 1004,2,Waitexten(20) exten = 1004,3,StopPlaytones exten = 1004,4,Goto(incoming,1002,1) exten = *,1,Goto(disa-1,s,1) [disa-1] exten = s,1,Playback(enter pin) exten = s,2,ResponseTimeout(20) exten = s,3,DigitTimeout(5) exten = s,4,DISA(no-password|outgoing) exten = s,5,Congestion Message: 21 Date: Mon, 7 Nov 2005 14:25:50 -0500 (EST) From: Frank Tarczynski [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Help with dialplan to allow breakout to DISA To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain;charset=iso-8859-1 Yes, I know. BUT, I want the line to work as normal for incoming calls AND allow the user to breakout. So how do I merge: [incoming] exten = 1000,1,Ringing exten = 1000,2,Answer exten = 1000,n,Dial(IAX,iaxy/20) exten = 1000,n,Voicemail() exten = 1000,n,Hangup AND exten = *, 1, Authenticate(PASSWORD) exten = *, 2, DISA(no-password|DESTINATION_CONTEXT) exten = *, 3, Hangup to have Asterisk answer the line as normal but also react to the user pressing *? I've tried putting' all of the above in the same context but it doesn't work when I call in and press *. Frank Message: 10 Date: Mon, 7 Nov 2005 12:45:05 -0500 From: Rusty Dekema [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Help with dialplan to allow breakout to DISA To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 I do it this way: exten = *, 1, Authenticate(PASSWORD) exten = *, 2, DISA(no-password|DESTINATION_CONTEXT) exten = *, 3, Hangup It seems to work fine... -Rusty On 11/7/05, Frank Tarczynski [EMAIL PROTECTED] wrote: I'm trying to set-up a dialplan for incoming calls that allows a breakout by pressing something like *. Users would then be able to get an inside dial tone for voicemail, outgoing calls, etc. I've been struggling with Waitexten(), Disa() in the dialplan but not having much luck. Are there any good documents out there to assist me in this? Frank --- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sangoma 102 installation problem
Hi friends, during the installation, I receive that problem, but I've installed both Flex and, of course, C/C++ libraries. My OS is CentOS 3.6, completely updated. Any ideas??? Thanks - Compiling WANPIPE WanCfg Utility ... Failed! !!! WANPIPE WanCfg Compilation Failed !!! Possible solution: FLEX Package not installed Non-standard C/C++ library (eg: ulibc) Please contact Sangoma Tech. at 905 474-1990 - -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk 1.2b2 compiling problem
Don Pobanz wrote: I just checked out asterisk 1.2b2 for zaptel, libpri, asterisk and asterisk-sounds. Zaptel and libpri compile fine with a 'make clean' and 'make install'. However even after a make clean, the asterisk 'make install' does not finish on my redhat 7.3 system. CVS-D2005.09.12.05.00.00-09/14/05-02:05:11 is currently running. Here are the last few lines before erroring out. chan_agent.c:1684: parse error before `char' chan_agent.c:1701: `agent_goodbye' undeclared (first use in this function) chan_agent.c:1701: (Each undeclared identifier is reported only once chan_agent.c:1701: for each function it appears in.) chan_agent.c:1708: `tmpoptions' undeclared (first use in this function) chan_agent.c:1714: `update_cdr' undeclared (first use in this function) chan_agent.c:1732: `context' undeclared (first use in this function) chan_agent.c:1737: `play_announcement' undeclared (first use in this function) chan_agent.c:1864: `filename' undeclared (first use in this function) make[1]: *** [chan_agent.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/channels' make: *** [subdirs] Error 1 Any ideas? What version of gcc are you using ? Robert ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fwd: differences between chan_capi and chan_capi-cm
I would suggest chan-capi-cm for any configuration. You know which quadbri cards it works with? I'm using an Eicon-Diva-Server 4BRI. Elmar ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma 102 installation problem
Hi, FaberK wrote: during the installation, I receive that problem, but I've installed both Flex and, of course, C/C++ libraries. My OS is CentOS 3.6, completely updated. Any ideas??? Thanks - Compiling WANPIPE WanCfg Utility ... Failed! !!! WANPIPE WanCfg Compilation Failed !!! Possible solution: FLEX Package not installed Non-standard C/C++ library (eg: ulibc) Please contact Sangoma Tech. at 905 474-1990 So, is FLEX available on your system ? (I don't know CentOS) Florian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hiss
Title: Hiss Whoo Hoo! I managed to get * up and running last night. My most pressing problem is that there is a considerable amount of hiss heard by the called party when using a SIP phone (xten or GXP-2000). Ive tried two different computers with two different headsets, and the hiss still remains when you should be hearing silence (or near silence). The hiss does not occur when youre on one of the extensions of the legacy PBX (Nortel). Anyone have some ideas on where I should be looking? Basics below: PSTN (via PRI) * PRI Nortel Meridian Ive got a basic dial plan setup to forward most calls through * to the legacy PBX. Thanks! --Jeffrey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] groupware + unified messagerie +Asterisk
Hello, Is it possible to add a frontend groupware with asterisk in order to Provide send receive fax to mail, sms to mail, voice messages . Asterisk or openpbx could be the server of the unified messagerie . click to dial contact in address book ,... Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem dialling multiple SIP devices
Drop the incoming calls into a call queue. Is it not the case that in order for calls to go into a queue, they must be answered first? Is it possible to drop calls into a queue before they're answered (by asterisk)? Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] groupware + unified messagerie +Asterisk
harry gaillac wrote: Is it possible to add a frontend groupware with All is possible, you're only limited by your imagination. (always wanted to say this :p) I'm not sure there's a(n Open-source) project like this already. Cheers.. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hiss
Jeffrey Macko wrote: Whoo Hoo! I managed to get * up and running last night. My most pressing problem is that there is a considerable amount of hiss heard by the called party when using a SIP phone (xten or GXP-2000). I’ve tried two different computers with two different headsets, and the hiss still remains when you should be hearing silence (or near silence). The hiss does not occur when you’re on one of the extensions of the legacy PBX (Nortel). Anyone have some ideas on where I should be looking? Basics below: PSTN (via PRI) à * à PRI à Nortel Meridian I’ve got a basic dial plan setup to forward most calls through * to the legacy PBX. Thanks! --Jeffrey I test headsets by doing record/playback with audacity sound editor on a linux workstation. So far all of them have high frequency hiss and noise. I record and then apply a low pass filter and the playback sounds fine. When I have the time I will be testing some usb headsets. I get the hiss and noise with softphones using all headsets I have tried so far. I don't get it with grandstream budgetone 101 phones or phones connected to ata's. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma 102 installation problem
On 11/08/05 20:53 FaberK said the following: Any ideas??? i believe the answer is in your email. Please contact Sangoma Tech. at 905 474-1990 -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk 1.2b2 compiling problem
On 11/08/05 20:54 Robert Stanford said the following: What version of gcc are you using ? though this is documented in the UPGRADE.txt file, i believe it should have been highlighted much more clearer. this bugbear has bitten quite a few people who're unaware that gcc 3.x is the minimum needed to compile asterisk. add this to the fact that before last week, gcc 2.95 happily compiled asterisk without problems. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Playtone on answering the phone
Is it possible to get Asterisk to issue a Playtones when an outgoing call is answered? The examples indicate what happens when an incoming call is answered. /Obelix This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma 102 installation problem
Hi Florian, yes, I have Flex available: whereis flex flex: /usr/bin/flex /usr/share/man/man1/flex.1.gz other ideas? 2005/11/8, Florian Overkamp [EMAIL PROTECTED]: Hi, FaberK wrote: during the installation, I receive that problem, but I've installed both Flex and, of course, C/C++ libraries. My OS is CentOS 3.6, completely updated. Any ideas??? Thanks - Compiling WANPIPE WanCfg Utility ... Failed! !!! WANPIPE WanCfg Compilation Failed !!! Possible solution: FLEX Package not installed Non-standard C/C++ library (eg: ulibc) Please contact Sangoma Tech. at 905 474-1990 So, is FLEX available on your system ? (I don't know CentOS) Florian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 Password Recovery
i appear to misplaced my password for my cisco 7960 SIP Phone. Does anyone know the procedure to recover this? I have read in the past that you can use cisco or Cisco but this does not appear to work. Thanks in advance. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hiss
Paul wrote: I get the hiss and noise with softphones using all headsets I have tried so far. I don't get it with grandstream budgetone 101 phones or phones connected to ata's. Then it's likely to be your sound card. Try using a nice usb headset (not the cheapest you can find) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-1.2-bêta2 | presen ce/subscription support in the SIP channel driver
Ok. What does sip show subscriptions from the CLI show you? On 11/8/05, harry gaillac [EMAIL PROTECTED] wrote: Hello, Sorry here are my sip.conf and extensions.conf in fact when polycom ip300 send subscribe to buddies these one send back notify but nothing else when status change Regards Harry --- Matt Riddell [EMAIL PROTECTED] a écrit : harry gaillac wrote: nobody has an answer here! Actually someone asked for you config details. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playtone on answering the phone
Obelix wrote: Is it possible to get Asterisk to issue a Playtones when an outgoing call is answered? The examples indicate what happens when an incoming call is answered. It would have to be done by the remote machine. Unless you want to play a sound to callee once connected: Some Dial options: 'A(x)' -- play an announcement to the called party, using x as file 'D([called][:calling])' -- Send DTMF strings *after* called party has answered, but before the call gets bridged. The 'called' DTMF string is sent to the called party, and the 'calling' DTMF string is sent to the calling party. Both parameters can be used alone. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Choppy Audio in Echo Test and Music On Hold (1.2.0-b2)
Rich Adamson wrote: I recently resurrected an old athlon system and put CentOS 4.2 on it to play with asterisk. First I tried asterisk-1.0.9, now I'm using 1.2.0-b2. Both have the same audio issues that have me stumped. I looked through all the lists and forums and the closest I could get were some messages from 2003: http://lists.digium.com/pipermail/asterisk-users/2003-August/017171.html I've got asterisk set up with my xten-lite softphone on extension 200 over SIP. I've configured extension 611 as an echo test and 612 will play 30 seconds of MusicOnHold. I can connect to both just fine, however, they sound rather bad when they work (quite muddy) and periodically they just drop out for as much as 5 seconds before coming back. In the xlite configuration, look for an option something like 'transmit silence' and set that to yes. (Might be called 'silence suppression', I don't remember.) I think it's called VAD (Voice Activity Detection) in xten products. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] groupware + unified messagerie +Asterisk
What about egroupware ! Harry --- Kristof Hardy [EMAIL PROTECTED] a écrit : harry gaillac wrote: Is it possible to add a frontend groupware with All is possible, you're only limited by your imagination. (always wanted to say this :p) I'm not sure there's a(n Open-source) project like this already. Cheers.. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BRI cards, HFC, and bristuff - a general question to clear up my understanding.
Hi Folks, After doing extensive reading it seems that I am more confused than when I started out. I have an Asustek ISDNLink (P-IN100-ST-D) BRI card. I know that it has a Winbond W6692 chipset, but there is much confusion in my head regarding whether to use bristuff (which seems to work with the HFC chipset - is this Winbond an HFC chipset or not?), mISDN, which from my reading is still somewhat unstable, ISDN4linux which seems to be feature poor or capi-channel (which seems to not be supported on this card, so that's easy!). I have, in the past, got this card working with the hisax driver to do dial-up to the ISP, but this driver seems to have been deprecated. I was wondering whether those guru's out there that have had success with their BRI cards could step forward. I am loathe to buy a AVM Fritz card as they are VERY expensive here and if I can get this card working - hey presto, since these thingies are very cheap. Thanks in advance, Hamish ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma 102 installation problem
Hi, FaberK wrote: Hi Florian, yes, I have Flex available: whereis flex flex: /usr/bin/flex /usr/share/man/man1/flex.1.gz Hmm, nope sorry :P. You can try to mail or call Sangoma, their support is pretty good from what I've seen so far. Florian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hiss
Is the ambient noise in the room high? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: Tuesday, November 08, 2005 8:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Hiss Paul wrote: I get the hiss and noise with softphones using all headsets I have tried so far. I don't get it with grandstream budgetone 101 phones or phones connected to ata's. Then it's likely to be your sound card. Try using a nice usb headset (not the cheapest you can find) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] LCDProc for Asterisk?
Anyone written an LCDProc client for Asterisk? It occurs to me that as many of these systems run headless in the back of a closet a small LCD display could tell you what's going on at a glance. -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP domain support for authentication and virtual hosting
harry gaillac wrote: nobody has an answer here !! Where may i find documentation about SIP domain support and dnsmgr.conf , The problem is that dnsmgr is new and not finished, so there is not much documentation yet. Re the SIP domain support, I don't know, there is the announcement here ( http://www.voip-forum.com/news.php?p=183 ), but it doesn't really have that much info. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't make calls from Asterisk IAX to other IAX
chawki hammoud wrote: Hi: I have been having this problem for sometime that I am not able to solve and I hope someone can help. I can make VOIP calls between my Asterisk box and my VOIP provider using sip channel without a problem. But when I attempt to make a call using IAX, the call get accepted and then get a hangup message: is this the same number format you send when using sip: 0017046872001 -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-1.2-bêta2 | presen ce/subscription support in the SIP channel dri ver
Connected to Asterisk 1.2.0-beta2 currently running on serveur1 (pid = 1553) Verbosity is at least 3 serveur1*CLI sip show subscriptions Peer UserCall ID Extension Last state Type 192.168.0.21 86 2127e5fd-5f 84 Idle xpidf+xml 192.168.0.20 84 61c23b4e-3d 86 Idle xpidf+xml 2 active SIP subscriptions --- BJ Weschke [EMAIL PROTECTED] a écrit : Ok. What does sip show subscriptions from the CLI show you? On 11/8/05, harry gaillac [EMAIL PROTECTED] wrote: Hello, Sorry here are my sip.conf and extensions.conf in fact when polycom ip300 send subscribe to buddies these one send back notify but nothing else when status change Regards Harry --- Matt Riddell [EMAIL PROTECTED] a écrit : harry gaillac wrote: nobody has an answer here! Actually someone asked for you config details. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] groupware + unified messagerie +Asterisk
harry gaillac wrote: What about egroupware ! We use it, although there is no simple click to install installation package for Asterisk integration. The idea is to use flash operator panel to load a url when each extension is dialed. And for click to dial, I use call files. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura 2000
Hello, I have a problem with my Sipura 2000. The problem is that it does not accept any change in the configuration. When I access to it, via browser or phone, and make any change, after clicking submit all changes all the changes I made dissapear and teh configuration remains with the original parameters. So I need to know how can I work it out. Thank you very much. Maxi ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stopping Asterisk from forwarding calls?
Ultimately this turned out to be a red herring as well. dialparties.agi just does a database dip to figure out which extensions are forwarded and then builds a dialstring based on whats left. It then returns to the Asterisk dialplan and the extensions are still dialed in the normal way. I stopped looking at this point but it appears that it only works if you are using AMP to manage your extensions in a database. Unless someone has a better idea it looks like the only way to do this will be a patch to Asterisk. Thanks all for your suggestions. -- John Lange On Mon, 2005-11-07 at 23:20 -0600, John Lange wrote: Thanks Tad. This might turn out the be the clue I was looking for. It appears AMP has a macro-dial which has a comment about dealing with CFWD, DND etc. It actually dials using a script: exten = s,4,AGI,dialparties.agi I'm still trying to figure out what it does exactly because the code is not commented very well but it looks promising. Thanks for pointing me in this direction. John On Mon, 2005-11-07 at 15:35 -0500, Tad Heckaman wrote: I use [EMAIL PROTECTED], and if I have one of my Cisco phones forwarded to my cell phone, when the phones ring in a ring group, it never forwards. You may want to look at the latest configs that comes with [EMAIL PROTECTED] and see if theres some special dialplans thats doing what your looking for. Keep in mind I am using the call forward on the phone, and not the built in call forward in the dialplan. On 11/7/05, Kevin P. Fleming [EMAIL PROTECTED] wrote: John Lange wrote: Reading the source code I see there are two parameters for channels, allowredir_in allowredir_out. These offer me some hope that Asterisk has the ability but I couldn't figure out what these do or how to make use of them (I'm not a C programmer so maybe its just a red herring?). Those are entirely unrelated. At this time there is no method available to make Asterisk ignore incoming '302 REDIRECT' from SIP phones. It may be possible to send those 'forward' requests to a context that has no valid extensions in it, but I don't think we even support that at this time. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tad Heckaman ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 Password Recovery
Polycom User wrote: i appear to misplaced my password for my cisco 7960 SIP Phone. Does anyone know the procedure to recover this? I have read in the past that you can use cisco or Cisco but this does not appear to work. Thanks in advance. Is this phone setup using tftp? If so, I would check in the SIPDefault.cnf file or the SIPxxx.cnf file that matches the phone's MAC address on the tftp server. Mark ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New package posted to Sourceforge
I just posted a few addons for the AMP users ... These are several routines I found necessary for my system 1: Speed Dials revised my way (AMP front end into DB), 2: Intercom in business, 3: Group Paging in business, 4: Cisco phone display (XML) of internal directory list from AMP extensions DB. Intercom Paging is not AMP dependent - tested on Cisco phones. http://sourceforge.net/projects/enhanceme/ Paul Norris Silicon Valley Products ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BRI cards, HFC, and bristuff - a general question to clear up my understanding.
Get a Duxbury PCI ISDN card that has the HFC-S chipset, its type approved TE2003/013 and there is enough support on the wiki at http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+zaphfc+installdiff=24Cant find much reference to winbond+asteriskCost is also ±R200 each.Rob On 11/8/05, Hamish Whittal [EMAIL PROTECTED] wrote: Hi Folks,After doing extensive reading it seems that I am more confused than whenI started out.I have an Asustek ISDNLink (P-IN100-ST-D) BRI card. I know that it has aWinbond W6692 chipset, but there is much confusion in my head regarding whether to use bristuff (which seems to work with the HFC chipset - isthis Winbond an HFC chipset or not?), mISDN, which from my reading isstill somewhat unstable, ISDN4linux which seems to be feature poor or capi-channel (which seems to not be supported on this card, so that'seasy!).I have, in the past, got this card working with the hisax driver to dodial-up to the ISP, but this driver seems to have been deprecated. I was wondering whether those guru's out there that have had success withtheir BRI cards could step forward.I am loathe to buy a AVM Fritz card as they are VERY expensive here andif I can get this card working - hey presto, since these thingies are very cheap.Thanks in advance,Hamish___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem dialling multiple SIP devices
Chris Bagnall wrote: Drop the incoming calls into a call queue. Is it not the case that in order for calls to go into a queue, they must be answered first? Is it possible to drop calls into a queue before they're answered (by asterisk)? Yes, But your problem is stemming from the fact that the phones are reporting busy. Asterisk itself is not the problem. Asterisk can answer and put calls into the queue. At that time, it will then send it on to the phones. It will keep trying until either a phone is not busy or a timeout occurs in the queue. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BRI cards, HFC, and bristuff - a general question to clear up my understanding.
Hamish Whittal wrote: I have an Asustek ISDNLink (P-IN100-ST-D) BRI card. [..] This is not a card compatible with the bristuff. I don't know about the availability of the hfc-cards in your part of the world, but they are very inexpensive in Germany (around 30 EUR ~ 30 USD) I am loathe to buy a AVM Fritz card as they are VERY expensive here and if I can get this card working - hey presto, since these thingies are very cheap. If you are serious about asterisk, you don't want to try the modem route with your card. Sound will be bad. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LCDProc for Asterisk?
Mark Phillips wrote: Anyone written an LCDProc client for Asterisk? It occurs to me that as many of these systems run headless in the back of a closet a small LCD display could tell you what's going on at a glance. Yes, I have :) Still ironing out some bugs, but fire away if you have questions! :D -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: Atlas 550 Caller ID interoperability with Di gium TE110P?
Last night, my Atlas 550 failed big-time. Adtran, to their great credit, is overnighting me a new one even though it is out of warranty. I am working around this on my Asterisk box by plugging in my PRI directly into my TE110P and receiving faxes with Asterisk where they used to be received with an analog fax (The Atlas was basically a very expensive analog channel bank + PRI for Asterisk, fax machines, dialup modems, fax software, and other things) Immediately after plugging in the PRI directly into the TE110P I noticed that Asterisk started receiving CallerID name AND number, whereas before, with the Atlas in-line, only the Caller ID number would be passed. Since Caller ID name was spotty before, even on our Mitel 3300 without the Atlas, I just assumed that it was the telco not passing it, and I worked around it with lookup scripts that would query first our CRM, and then Canada411.com (works great!) Now that I know that the Atlas is inhibiting the Caller ID name, I asked Adtran support about it and went into detail about switchtype etc. They basically said that they had no idea why and it was my CPE (the TE110P) that was causing the problem. Obviously false, because here I am plugged directly into my PRI and I am getting full Caller ID. Are there any Atlas users out there that have any insight on this issue? Config is Atlas 550 + 2 X Octal 8 port analog cards + 1 4 port T1 card. PRI from Telco is National switchtype and plugged into Port 1 of the 4 port T1 and CPE (TE110P) is plugged into Port 2. Year-old firmware rev on the Atlas (can't tell you exactly because I can't get it to boot). Dialplan on Atlas is to send Caller ID as presented. tia ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [Asterisk-User] Estension s don't start
Why the estension s dont' start? In extensions.conf [default] exten = s,1,Answer exten = s,2,Playback(invalid) exten = s,3,Hangup In sip.conf [general] context=default ___ Yahoo! Mail: gratis 1GB per i messaggi e allegati da 10MB http://mail.yahoo.it ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP domain support for authentication and virtual hosting
thanks Matt for your answer Does asterisk-1.2-stable will provide this features ? Harry PS: Who are the main developpers for the sip channels ? --- Matt Riddell [EMAIL PROTECTED] a écrit : harry gaillac wrote: nobody has an answer here !! Where may i find documentation about SIP domain support and dnsmgr.conf , The problem is that dnsmgr is new and not finished, so there is not much documentation yet. Re the SIP domain support, I don't know, there is the announcement here ( http://www.voip-forum.com/news.php?p=183 ), but it doesn't really have that much info. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-1.2-bêta2 | presen ce/subscription support in the SIP channel driver
Ok. It looks like you've got most of the basic configurations setup correctly. Let's setup a trace and then have you repeat your steps so we can see in better detail what might be wrong. In your logger.conf file make certain you have the following line: full = notice,warning,error,debug,verbose Then, restart Asterisk if that line wasn't there in logger.conf already. Then, from the CLI issue set debug 10 and then set verbose 10 and finally sip debug, and then repeat your steps to try and have * send notifications about state changes to the phones that are subscribed. With that complete, then please zip up and send us your /var/log/asterisk/full file so we can get a better look at what's going on behind the scenes. On 11/8/05, harry gaillac [EMAIL PROTECTED] wrote: Connected to Asterisk 1.2.0-beta2 currently running on serveur1 (pid = 1553) Verbosity is at least 3 serveur1*CLI sip show subscriptions Peer UserCall ID Extension Last state Type 192.168.0.21 86 2127e5fd-5f 84 Idle xpidf+xml 192.168.0.20 84 61c23b4e-3d 86 Idle xpidf+xml 2 active SIP subscriptions --- BJ Weschke [EMAIL PROTECTED] a écrit : Ok. What does sip show subscriptions from the CLI show you? On 11/8/05, harry gaillac [EMAIL PROTECTED] wrote: Hello, Sorry here are my sip.conf and extensions.conf in fact when polycom ip300 send subscribe to buddies these one send back notify but nothing else when status change Regards Harry --- Matt Riddell [EMAIL PROTECTED] a écrit : harry gaillac wrote: nobody has an answer here! Actually someone asked for you config details. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to use #include to all files in /etc/asterisk/customdir ?
how to use #include to all files in /etc/asterisk/customdir ? in v1.0.9 #include /etc/asterisk/customdir/*.conf doesnt work ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 Password Recovery
The only way is if you are using DHCP to get an IP address to the phone. If you are, then you can have it point the phone to a TFTP server with config files with a new password. If you are using a static IP, then you are out of luck. I opened up a TAC case about a year ago, and that is what Cisco said. Fortunately we were able to guess what it was reset to, otherwise that phone would still be locked today. B. J. From: Polycom User [mailto:[EMAIL PROTECTED] Sent: Tuesday, November 08, 2005 7:51To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Cisco 7960 Password Recovery i appear to misplaced my password for my cisco 7960 SIP Phone. Does anyone know the procedure to recover this? I have read in the past that you can use "cisco" or "Cisco" but this does not appear to work. Thanks in advance. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Change Asterisk User
i had ( or still have ) the same problem. Im running asterisk as asterisk:asterisk, but dont know why, the new voicemails are saved as root:root with 700 permissions, so i made a quick workaround, i added the following line in sudoers file: %lighttpd ALL=(root)NOPASSWD: /usr/bin/chmod -R 755 /var/spool/asterisk/voicemail/ dont need to say im using lighttpd instead of apache. Then when i want to read voicemail i execute from php the command to change the permissions in /var/spool/ what do you think of that approach? is ugly yes, but any security wholes? best regardsOn 11/7/05, Ryan Amos [EMAIL PROTECTED] wrote: Use group permissions. Add the apache user to the asterisk group andgive the group the appropriate read and/or write access. IMO this is theeasiest way to get around the apache permissions thing, and probably the Right Way (tm)-Ryan-Original Message-From: [EMAIL PROTECTED][mailto: [EMAIL PROTECTED]] On Behalf Of amauryBOSSESent: Monday, November 07, 2005 12:44 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Change Asterisk User Thanks for your answer,I am working on Debian Sarge but I have compiled Astersik 1.0.9 myselfwithout .deb Packages.I need to access to voicemail and sound files from my web-interface (phpand cgi/perl) but I can't change Apache user because of others applications.Asterisk creates files under Asterisk user and I have to access themfrom www-data user.Do you have other solution? I have tried using sudo but it doesn't seemto work.Regards,Amaury ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Lost Cisco SIP phones after reboot
After rebooting my asterisk server (1.2B2) I could still call my Sipura SIP phones from outside (via cell phone). But I have a customer with two Cisco SIP phones...I don't know the exact model...those two phones could not be reached. The message in the console: Nov 8 10:03:34 NOTICE[4207]: app_dial.c:1110 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Why would the Sipura SPA2002's keep working after a reboot but not the Cisco phones? On a somewhat related note: Is there a file where asterisk stores current SIP registrations or does it just store them in memory? Or perhaps is there a way to export current registered SIP users before a reboot, do the reboot, then import them back in? The idea is to make it transparent to the user. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 Password Recovery
It is set by your SIPMAC.cnf file. phone_password: password ; Telnet/Console Password On Tue, 2005-11-08 at 08:51 -0500, Polycom User wrote: i appear to misplaced my password for my cisco 7960 SIP Phone. Does anyone know the procedure to recover this? I have read in the past that you can use cisco or Cisco but this does not appear to work. Thanks in advance ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [Asterisk-User] Estension s don't start
Fabio Montemaggiore wrote: Why the estension s dont' start? Do you get an error in the Asterisk console? A good thing to read is the Asterisk Book which you can download for free from one of the mirrors provided here: http://www.sineapps.com/news.php?rssid=1044 -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [Asterisk-User] Estension s don't start
What do you see on asterisk console? (asterisk -vc) El mar, 08-11-2005 a las 15:38 +0100, Fabio Montemaggiore escribió: Why the estension s dont' start? In extensions.conf [default] exten = s,1,Answer exten = s,2,Playback(invalid) exten = s,3,Hangup In sip.conf [general] context=default ___ Yahoo! Mail: gratis 1GB per i messaggi e allegati da 10MB http://mail.yahoo.it ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- José Luis Gómez Qualis Information Technology Av. Rivadavia 2553, PB Of. 43 EP 0342-4565684 int 102 Bs. As. 011-51990896 www.qualis.com.ar Soporte 0810-8880022 Santa Fe - Argentina ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MP3 or OGG
Hilton, AFAIK, you can optionally record in gsm. However, I think * won't do it natively. It will do -in and -out wav files, soxmix them together and then convert them to gsm. I'm offloading all of that to a different machine and just leaving * to create the raw -in and -out wav files. Maybe I'm wrong too, so comments are welcomed. Thanks, Waldo On Nov 8, 2005, at 3:14 AM, Quark IT - Hilton Travis wrote: Hi Waldo, Doesn't * record to .gsm file initially and then convert these to .wav later? I may be totally off the mark here, and if I am, I welcome the correction. In that case, why not leave the files in .gsm format instead of translating them into another lossy format? Obviously if * records conversations as .wav files then I'd be leaning toward Speex (Vorbis) as it is a suited to speech compression format. Both Speex and ogg are Open Source, therefore patent issues are likely non-existent. MP3, otoh, is fine if you use one of their approved apps, and not if you use anything else. I'm steering clear of .mp3 (and have been for quite a few years now). -- Regards, Hilton Travis Phone: +61 (0)7 3344 3889 (Brisbane, Australia) Phone: +61 (0)419 792 394 Manager, Quark IT http://www.quarkit.com.au Quark Group http://quarkgroup.com.au/ Microsoft Small Business Specialists http://www.threatcode.com/ -- its now time to shame poor coders into writing code that is acceptable for use on today's networks War doesn't determine who is right. War determines who is left. This document and any attachments are for the intended recipient only. It may contain confidential, privileged or copyright material which must not be disclosed or distributed. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Waldo Rubinstein Sent: Tuesday, 8 November 2005 11:32 Wasn't aware of it, but if quality is good, it makes sense since all I'm archiving is speech. Will evaluate further. Thanks, Waldo On Nov 7, 2005, at 7:14 PM, Mark Edwards wrote: I would recommend vorbis speex for this. You can get windows drivers to read speex files directly. Vorbis are the same bunch that develops ogg. Ogg and mp3 are more suited to music rather than speech. Speex is a much better fit for speech archiving. Mark -Original Message- From: BJ Weschke [mailto:[EMAIL PROTECTED] Sent: Tuesday, 8 November 2005 5:52 AM You're probably not going to be violating any patent protections by using OGG instead of MP3. As far as compression goes, I've found the difference between the two of them to be negligible. I've always used OGG when possible to stay IP safe. On 11/7/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: I'm trying to archive out call recordings and would appreciate some feedback as to which audio compression is more recommended MP3 or OGG. In the past, I've use lame to convert to MP3, but I noticed the audio volume drops significantly. Is it just a setting on the command line of lame or is OGG better? Which achieves higher compression rates while maintaining call quality? Thanks, Waldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sensing fax with txfax
Matt Riddell napisał(a): Maybe you could make an extension that you can dial which will run txfax for you. Then you can call it with a phone and see if you hear the fax tones. I tried this :( I hear the fax signal but then nothing happens. These lines are in asterisk console: Nov 8 15:50:03 NOTICE[6004]: channel.c:1736 ast_set_read_format: Unable to find a path from g723 to ulaw Nov 8 15:50:03 NOTICE[6004]: channel.c:1703 ast_set_write_format: Unable to find a path from ulaw to g723 Nov 8 15:52:50 NOTICE[6429]: channel.c:1736 ast_set_read_format: Unable to find a path from g723 to slin Nov 8 15:52:50 WARNING[6429]: app_txfax.c:167 txfax_exec: Unable to set to linear read mode, giving up What is the last warning? It comes from txfax but I don't know how to correct this. -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [Asterisk-User] Estension s don't start
Asterisk in console don't show not all --- José Luis Gómez [EMAIL PROTECTED] ha scritto: What do you see on asterisk console? (asterisk -vc) El mar, 08-11-2005 a las 15:38 +0100, Fabio Montemaggiore escribió: Why the estension s dont' start? In extensions.conf [default] exten = s,1,Answer exten = s,2,Playback(invalid) exten = s,3,Hangup In sip.conf [general] context=default ___ Yahoo! Mail: gratis 1GB per i messaggi e allegati da 10MB http://mail.yahoo.it ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- José Luis Gómez Qualis Information Technology Av. Rivadavia 2553, PB Of. 43 EP 0342-4565684 int 102 Bs. As. 011-51990896 www.qualis.com.ar Soporte 0810-8880022 Santa Fe - Argentina ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Yahoo! Mail: gratis 1GB per i messaggi e allegati da 10MB http://mail.yahoo.it ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New package posted to Sourceforge
Looks good except one problem I am having. The AMP script does not store the info. It adds a blank speeddial. If I edit the database the AMP script will show the correct info, but it never updates the fields. - James Paul wrote: I just posted a few addons for the AMP users ... These are several routines I found necessary for my system 1: Speed Dials revised my way (AMP front end into DB), 2: Intercom in business, 3: Group Paging in business, 4: Cisco phone display (XML) of internal directory list from AMP extensions DB. Intercom Paging is not AMP dependent - tested on Cisco phones. http://sourceforge.net/projects/enhanceme/ Paul Norris Silicon Valley Products ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [OTAnn] Feedback
I use a newsreader pointed at gmane.org. It is agregated and only uses my internet connection when I tell it to. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- shenanigans [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] I was interested in getting feedback from current mail group users. We have mirrored your mail list in a new application that provides a more aggregated and safe environment which utilizes the power of broadband. Roomity.com v 1.5 is a web 2.01 community webapp. Our newest version adds broadcast video and social networking such as favorite authors and an html editor. It?s free to join and any feedback would be appreciated. S. -- Broadband interface (RIA) + mail box saftey = Asterisk_Users_List.roomity.com *Your* clubs, no sign up to read, ad supported; try broadband internet. ~~1131397871736~~ -- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bristuff-0.2.0-RC8n problems and kernel panic
On Tue, Nov 08, 2005 at 09:12:27AM +0100, gincantalupo wrote: Hi, this is my /etc/modprobe.d/zaptel: Those are probably useless: options torisa base=0xd alias char-major-196 torisa Those won't help you a bit if you run a ztcfg in your zaptel init script anyway. install tor2 /sbin/modprobe --ignore-install tor2 /sbin/ztcfg install torisa /sbin/modprobe --ignore-install torisa /sbin/ztcfg install wcusb /sbin/modprobe --ignore-install wcusb /sbin/ztcfg install wcfxo /sbin/modprobe --ignore-install wcfxo /sbin/ztcfg install wcfxs /sbin/modprobe --ignore-install wcfxs /sbin/ztcfg install ztdynamic /sbin/modprobe --ignore-install ztdynamic /sbin/ztcfg install ztd-eth /sbin/modprobe --ignore-install ztd-eth /sbin/ztcfg install wct1xxp /sbin/modprobe --ignore-install wct1xxp /sbin/ztcfg install wct4xxp /sbin/modprobe --ignore-install wct4xxp /sbin/ztcfg install wcte11xp /sbin/modprobe --ignore-install wcte11xp /sbin/ztcfg alias wctdm wcfxs and this is my /etc/init.d/asterisk made by me: #!/bin/sh ztcfg -s # unload wcfxs module because I must load # qozap module first! /sbin/rmmod wcfxs /sbin/rmmod zaptel To override hotplug, load the modules using /etc/modules: qozap wcfxs # Now I load all the modules in the right order /sbin/insmod /lib/modules/2.6.8-2-386/misc/zaptel.ko /sbin/insmod /lib/modules/2.6.8-2-386/misc/qozap.ko /sbin/insmod /lib/modules/2.6.8-2-386/misc/wcfxs.ko Which means you don't use the stuff from modprobe.conf anyway, so it can be safely removed. ztcfg -vv This can be left here. # this is to exec asterisk as asterisk user chown --recursive asterisk:asterisk /dev/zap chmod --recursive u=rwx,g=rx /dev/zap chown asterisk /dev/tty9 sudo -u asterisk /usr/sbin/safe_asterisk Those belong in an asterisk init.d script. BTW: in debian the /dev/zap is by default owned by group dialout and asterisk should be added to that group. and it magically works (!!!) even if modifying debian zaptel and wcfxs modules loading sequence should be a better way to solve the problem but I don't know where to find that damned sequence. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New package posted to Sourceforge
After you place one in you MUST submit. That is only when it is saved Paul -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Armstrong Sent: Tuesday, November 08, 2005 10:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] New package posted to Sourceforge Looks good except one problem I am having. The AMP script does not store the info. It adds a blank speeddial. If I edit the database the AMP script will show the correct info, but it never updates the fields. - James Paul wrote: I just posted a few addons for the AMP users ... These are several routines I found necessary for my system 1: Speed Dials revised my way (AMP front end into DB), 2: Intercom in business, 3: Group Paging in business, 4: Cisco phone display (XML) of internal directory list from AMP extensions DB. Intercom Paging is not AMP dependent - tested on Cisco phones. http://sourceforge.net/projects/enhanceme/ Paul Norris Silicon Valley Products ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New package posted to Sourceforge
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Armstrong Sent: Tuesday, November 08, 2005 10:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] New package posted to Sourceforge Looks good except one problem I am having. The AMP script does not store the info. It adds a blank speeddial. If I edit the database the AMP script will show the correct info, but it never updates the fields. - James Paul wrote: I just posted a few addons for the AMP users ... These are several routines I found necessary for my system 1: Speed Dials revised my way (AMP front end into DB), 2: Intercom in business, 3: Group Paging in business, 4: Cisco phone display (XML) of internal directory list from AMP extensions DB. Intercom Paging is not AMP dependent - tested on Cisco phones. http://sourceforge.net/projects/enhanceme/ Paul Norris Silicon Valley Products ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
