[Asterisk-Users] [Voicemail] Quota
Hi all, Is there a way to put a voicemail quota to a SIP user? I mean a quota on the user's mailbox instead of a particular message of the user like the 'maxmessage' does currently. Quata can be total file size of message or total minutes of messages of a mailbox. Any help or suggestions? Thank's Carlos ___ --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] Adding caller name / ID to outbound meetme calls
Just to follow up on my post of yesterday, the solution was simple (thanks to the asteriskTFOT book!) Simply add the following line (modified, of course!) to the call file: CallerID: Asterisk 800-555-1212 Regards, Keith - Original Message - I'm calling people on Zap interface using /var/spool/asterisk/outgoing and then putting them into a MeetMe. This works 100%, but tends to give unknown name and number on the meetme list command... eg: User #: 01unknown no nameChannel: Zap/1-1 (unmonitored) ___ --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] Blind transfer from queue into another queue
Hello, did you try using a Local/XXX channel? it should work! l. On Tue, 01 Nov 2005 15:10:03 +0100, Stefan Günther [EMAIL PROTECTED] wrote: Hi, I want to transfer a call that has come into one queue, and that I have already accepted, into another queue. When I try this asterisk tells me Transfer attempted with no appropriate bridged calls to transfer. It is possible to forward the call to another person, but forwarding into a queue fails. Is forwarding from one queue into another possible at all? Bye, Stefan -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.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
[Asterisk-Users] Installing beta2
Title: Installing beta2 Once built no matter whether I do make install or make clean I get the same output [EMAIL PROTECTED] asterisk]# make clean build_tools/make_version_h include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c build_tools/make_version_h include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c make: *** [.depend] Interrupt I am using FC3 and any help would be appreciated. Regards Lee ___ --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] Fritz!Card PCI ver2.0
What chipset that card use?? Pedro Nunes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Arulraj Sent: terça-feira, 1 de Novembro de 2005 23:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Fritz!Card PCI ver2.0 Anyone knows how I can use this ISDN card for asterisk as a BRI trunk interface? Thanks, Stephen ___ --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] lilte help please
There are lots of different ways to accomplish the same thing in *,
so there is no way to answer your should do question without
looking at what you've defined. You're obviously doing something
wrong if you can't get any provider to work, and no one is going
to be able help identify what you're doing wrong unless you post
the relavent parts of your extensions.conf.
thank you
i ill try
i have a fewwq application for voice mail, speaking clock , etc.., and one
context for phones for sip
client internal, and one outgoing context is that what i should do
*** REPLY SEPARATOR ***
On 31/10/2005 at 07:20 Rich Adamson wrote:
problem i can't get asterisk to dial to sip provider no matter what
provider i choose
the prefix and telephone format is the main problem and i cant figure it
even thoug i looked at example and
diD not work for me
i took exmple on nufone and net2phone configs !
IF I UNDERSTAND THINGS WELL, i should dial 9 then phone number !, i
always get you dialed worgn number
any ideas
[OUTGOING]
exten = _91NXXNXX,1,Answer()
exten = _91NXXNXX,n,Dial(SIP/[EMAIL PROTECTED]/${EXTEN:1})
exten = _91NXXNXX,3,Congestion
You do not want to answer a call that is in the calling process.
Remove that.
To provide any better answers, we'll need ot see the context that
your sip phones are in along with any other contexts that are
included.
In your example above for nuphone, do you have a context like [nuphone]?
If so, what statements are included in it?
Can you copy/paste what the CLI is showing when you place a call?
It would be helpful to see that.
Until you understand exactly what you're doing, get rid of the n as
a priority and simply use numeric sequential numbers. In the above
example, change to 91NXXNXX,2,Dial and watch your CLI when placing
a call.
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[Asterisk-Users] Ericsson MD evolution and asterisk
hi all; Any one have experiment with this Ericsson MD evolution and asterisk, i try to do that: Phone-PABX Ericsson MD evolutionBox with asterisk server and TE110P When i try to make call with my phone behind the Ericsson PABX, i had just 4 digit in my asterisk!!! Thanks ---zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 loadzone=fr defaultzone=fr zapata.conf: [channels] language=fr ;context=default switchtype=euroisdn ;If you connect to a hicom PBX set your ISDN Numbering Plan Identifier to unknown. pridialplan=unknown prilocaldialplan=unknown usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 immediate=no context=entrant group = 0 signalling=pri_net channel = 1-15 channel = 17-31 -- cordialement Karim AMER ___ --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] is it possible to connect to Asterisk from an external application?
Is it possible to connect to Asterisk from an external application? What I mean, to connect and execute its own extensions, created by some other program: exten = 1234567,1,txfax(/home/steveu/testfax.tif|caller) or exten = $NUMBER_I_WANT,1,txfax($FILE_I_WANT|caller) and Asterisk will dial this number and execute these extensions. If it's possible, how do I do it or where can I read more about it? -- Tomek http://wpkg.org WPKG - software deployment and upgrades with Samba ___ --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] is it possible to connect to Asterisk from an external application?
On Wed, 2005-11-02 at 11:29 +0100, Tomasz Chmielewski wrote: Is it possible to connect to Asterisk from an external application? What I mean, to connect and execute its own extensions, created by some other program: exten = 1234567,1,txfax(/home/steveu/testfax.tif|caller) or exten = $NUMBER_I_WANT,1,txfax($FILE_I_WANT|caller) and Asterisk will dial this number and execute these extensions. If it's possible, how do I do it or where can I read more about it? There are a couple different ways, the easiest and what it sounds like what you want is to create a call file. A call file is simply a text file that gets placed (mv dont cp, and mv from the same partition, mv across partitions is the same as cp effectively) into a spool dir and asterisk will check that file do what it says and poof. http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --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] Noise in Echo()
Hello! First problem with 1.2-beta2. All I hear during Echo() is noise. No matter which codec selected. However, when using ulaw noise sounds better than g723 :) My equipment is Sipura SPA-3000. Works fine with 1.0.9 amd 1.0.7. ___ --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] MTP required for CCM integration ?
Thanks for this one, Greg ! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Oliver Sent: mardi, 1. novembre 2005 16:05 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] MTP required for CCM integration ? You will probably also need to change the media exchange timers in CCM if you are going to use it as a PRI gateway - otherwise asterisk - 323 - CCM - PSTN calls will get dropped after 4 secs of ringing. On Mon, 2005-10-31 at 14:41 +0100, Patrick Zwahlen wrote: Hey Dan, and thanks a lot for your answer regarding Cisco CCM and MTP. I will continue my tests, and maybe give a try to the patch you mentionned. However, this will probably be too cutting edge for the project ;-) I have a few questions, though: - You mention that Cisco indicates that any H323 trunk with advanced features needs an MTP. Can you point me to the place where you found this ? Because as far as I can tell, this is not true for a trunk to a Cisco gateway. - I have tested ooh323c from Asterisk-Addons. Reading what you wrote, I should have better luck with the Sourceforge version... - From your experience, do you feel that a clean CCM-* integration is possible ? I am currently interested in simple feature (MoH, transfers, maybe Call Park). A friend of mine is working on the voicemail (unity) replacement/integration. Thanks again for you quick support, and sorry for my late answer ! BR, - Patrick - -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Austin Sent: vendredi, 21. octobre 2005 18:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] MTP required for CCM integration ? Is it required to use an MTP on the Cisco callmanager, when integrating with asterisk (using h323) ? As of CCM 4.X, Cisco indicates that any H.323 trunk that will support MoH/Transfer/etc need MTP resources. Annoying. I am working on a project where the goal is to interconnect Cisco Callmanager (version 4) clouds together, using either SIP or IAX between multiple * servers. Basic setup will be: PHONE - sccp - CCM (V4) - h323 - ASTERISK - iax - ASTERISK - h323 -CCM - sccp - PHONE I am working on the first half of it, which is: 7920 --- SCCP --- CALLMANAGER (V4) --- chan_oh323 --- ASTERISK 1.0.9 I want to avoid the use of a gatekeeper. In that configuration, I am trying to get call transfer working. The phone can call the DEMO app on asterisk, but then I cannot transfer the call to another Cisco phone (on the same callmanager). I have some PCAP traces if required. Basically, the 2nd phone rings, but there is no audio channel. After about 10 seconds, I see that that chan_oh323 hangs up the call. Sure will drop the call. MTP does solve this. The idea was to avoid RTP streams through the call manager. Good plan, and one that I would consider a must for scalability and quality. Now, if I define a Media Termination Point (MTP) on the Callmanager, things work much better. I have also tried the new ooh323 with 1.2.0-beta1, but I couldn't get audio at all. Odd, I am using ooh323c. I have a special test release, but the fixes for our CCM4 enviroment were added to CVS. Are you using ooh323c from Asterisk-Addons or a download from Open Systems? I have read a lot about people having success with integratin CCM and*, but without any details, especially around MTP configuration. Any help would be greatly appreciated. BR, - Patrick - http://bugs.digium.com/view.php?id=5374 has a patch that allows * to send RTP packets when it is not receiving them. I wasn't expecting this result, but applying this patch resolved the disconnect when a SCCP phone put a call on hold and allows transfers. The bug/patch got quite a bit of early attention, but seems to have languished. Try it out and provide feedback. Maybe enough success reports will help get it rolling again. Dan ___ --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
RE: [Asterisk-Users] MTP required for CCM integration ?
Hi Dan, Your comments definitely help. Thanks a lot. I'll probably have more remarks / questions early next week. BR, - Patrick - From: Dan Austin [mailto:[EMAIL PROTECTED] On Behalf Of Dan Austin Sent: mardi, 1. novembre 2005 20:18 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] MTP required for CCM integration ? Comments inline From: [EMAIL PROTECTED] on behalf of Patrick Zwahlen Sent: Mon 10/31/2005 5:41 AM To: Dan Austin Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] MTP required for CCM integration ? Hey Dan, and thanks a lot for your answer regarding Cisco CCM and MTP. I will continue my tests, and maybe give a try to the patch you mentionned. However, this will probably be too cutting edge for the project ;-) I have a few questions, though: - You mention that Cisco indicates that any H323 trunk with advanced features needs an MTP. Can you point me to the place where you found this ? Because as far as I can tell, this is not true for a trunk to a Cisco gateway. Cisco introduced this requirement when 4.0 was released. I have only found it documented in the 4.X release notes. As far as the H323 trunk to the Cisco gateways, well I suspect Cisco has a way of handling that. I prefer not to use MTP resources. The Async patch solves the only issue I had with ANY of the trunking methods betweek CCM and *, which was disconnects during transfer/hold without the MTP. - I have tested ooh323c from Asterisk-Addons. Reading what you wrote, I should have better luck with the Sourceforge version... The ooh323c mailling list just had an announcement for a new release, but the * channel driver has lagged a bit and needs to be updated. - From your experience, do you feel that a clean CCM-* integration is possible ? I am currently interested in simple feature (MoH, transfers, maybe Call Park). A friend of mine is working on the voicemail (unity) replacement/integration. I would say yes. I am using * for services and not PBX functions. I can get calls into * from SCCP phones and our H323 gateways. Thanks again for you quick support, and sorry for my late answer ! No problem, I hope it helps. Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] On Behalf Of Dan Austin Sent: vendredi, 21. octobre 2005 18:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] MTP required for CCM integration ? Is it required to use an MTP on the Cisco callmanager, when integrating with asterisk (using h323) ? As of CCM 4.X, Cisco indicates that any H.323 trunk that will support MoH/Transfer/etc need MTP resources. Annoying. I am working on a project where the goal is to interconnect Cisco Callmanager (version 4) clouds together, using either SIP or IAX between multiple * servers. Basic setup will be: PHONE - sccp - CCM (V4) - h323 - ASTERISK - iax - ASTERISK - h323 -CCM - sccp - PHONE I am working on the first half of it, which is: 7920 --- SCCP --- CALLMANAGER (V4) --- chan_oh323 --- ASTERISK 1.0.9 I want to avoid the use of a gatekeeper. In that configuration, I am trying to get call transfer working. The phone can call the DEMO app on asterisk, but then I cannot transfer the call to another Cisco phone (on the same callmanager). I have some PCAP traces if required. Basically, the 2nd phone rings, but there is no audio channel. After about 10 seconds, I see that that chan_oh323 hangs up the call. Sure will drop the call. MTP does solve this. The idea was to avoid RTP streams through the call manager. Good plan, and one that I would consider a must for scalability and quality. Now, if I define a Media Termination Point (MTP) on the Callmanager, things work much better. I have also tried the new ooh323 with 1.2.0-beta1, but I couldn't get audio at all. Odd, I am using ooh323c. I have a special test release, but the fixes for our CCM4 enviroment were added to CVS. Are you using ooh323c from Asterisk-Addons or a download from Open Systems? I have read a lot about people having success with integratin CCM and*, but without any details, especially around MTP configuration. Any help would be greatly appreciated. BR, - Patrick - http://bugs.digium.com/view.php?id=5374 http://bugs.digium.com/view.php?id=5374 has a patch that allows * to send RTP packets when it is not receiving them. I wasn't expecting this result, but applying this patch resolved the disconnect when a SCCP phone put a call on hold and allows transfers. The bug/patch got quite a bit of early attention, but seems to have languished. Try it out and provide feedback. Maybe enough success reports will help get it rolling again. Dan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users
[Asterisk-Users] REGEX() 1.2beta2
Hi, anyone has a working example of this new function ? that's all that I have found -= Info about function 'REGEX' =- [Syntax] REGEX(regular expression data) [Synopsis] Regular Expression: Returns 1 if data matches regular expression. [Description] Not available Tnx! -- Best regards, Alessio mailto:[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
[Asterisk-Users] Options for 3-way or Conference Calling
Title: Options for 3-way or Conference Calling Hi all, I wonder if someone could lend a little insight into the best way to configure either 3-way calling or conference calling. My goal is to keep this as simple for my users as it was with our legacy PBX. On our old phone system, a user could simply, during a call, press a Conference button on their phone to bring in a third party to a call. Can this be accomplished with Asterisk? My phones are all SIP devices (Cisco and Sipura). David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[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
Re: [Asterisk-Users] Options for 3-way or Conference Calling
Yes. I believe the Cisco phones do conferencing in the same fashion. I'm not 100% on whether or not the SPA-841 or the new SPA-941 does it. On 11/2/05, Dave Morrow [EMAIL PROTECTED] wrote: Hi all, I wonder if someone could lend a little insight into the best way to configure either 3-way calling or conference calling. My goal is to keep this as simple for my users as it was with our legacy PBX. On our old phone system, a user could simply, during a call, press a Conference button on their phone to bring in a third party to a call. Can this be accomplished with Asterisk? My phones are all SIP devices (Cisco and Sipura). David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://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.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] Installing beta2
Are you installing over a previous source tree? If so, please rm -rf the previous source tree and install the new source tree from scratch. On 11/2/05, Lee Archer [EMAIL PROTECTED] wrote: Once built no matter whether I do make install or make clean I get the same output [EMAIL PROTECTED] asterisk]# make clean build_tools/make_version_h include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c build_tools/make_version_h include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c make: *** [.depend] Interrupt I am using FC3 and any help would be appreciated. Regards Lee ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://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.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] Options for 3-way or Conference Calling
On Wed, 2005-11-02 at 07:16 -0500, BJ Weschke wrote: Yes. I believe the Cisco phones do conferencing in the same fashion. I'm not 100% on whether or not the SPA-841 or the new SPA-941 does it. If not there is always features.conf :) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --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] Installing beta2
Hi, I had removed all old versions before starting and downloaded from CVS. Regards Lee From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ WeschkeSent: 02 November 2005 12:20To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Installing beta2 Are you installing over a previous source tree? If so, please rm -rf the previous source tree and install the new source tree from scratch. On 11/2/05, Lee Archer [EMAIL PROTECTED] wrote: Once built no matter whether I do make install or make clean I get the same output [EMAIL PROTECTED] asterisk]# make clean build_tools/make_version_h include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c build_tools/make_version_h include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c make: *** [.depend] Interrupt I am using FC3 and any help would be appreciated. Regards Lee ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.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] Asterisk as an internal pbs for a samall company
Hello all, We'd like to use asteriek as an internal pbx connected to an external sip provider to make outbound/inbound calls to pstn. We have the provider and have installed an asterisk at the office. Does anyone have a sample config? We need 25 telephone numbers(dids), to be registerd to the provider and be able to ceceive calls. Any advice is welcome. Sorry for the noob question, Olivier ___ --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] Graphical interface
Can you please suggest me some graphical interface (like AMP)? I have tried to install AMP but I have some problems and on AMP forum and mailing list I didn't get answer. Two things I need to have are. - list of calls for every user. - some information about Linux (processor load, HDD, network load...) Other things that I will welcome - operators panel - voicemail (to listen your voicemails) Thank you for your time. -- Tomislav Parčina Lama d.o.o. www.lama.hr tparcina#lama.hr ___ --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 an internal pbs for a samall company
On Wed, 2005-11-02 at 13:36 +0100, Olivier Taylor wrote: Hello all, We'd like to use asteriek as an internal pbx connected to an external sip provider to make outbound/inbound calls to pstn. We have the provider and have installed an asterisk at the office. Does anyone have a sample config? We need 25 telephone numbers(dids), to be registerd to the provider and be able to ceceive calls. Any advice is welcome. Sorry for the noob question, Olivier What you want to do depends largely on what you want to do. While that seems like a cylic statement I will try to explain. You have said that you want to route calls between your asterisk box and the PSTN via a VoIP provider that you have. So far that seems simple, but how are those calls going to go bewteen the office workers and asterisk? You will need configurations for that. How are the inbound calls going to be routed? Via an IVR? Well you will have to configure that. There is a lot of information that is missing from this setup. www.voip-info.org has a lot of asterisk examples including configuration files. You may find something there that does what you want. I cant easily help you solve this problem (and suspect that no one else can either) until you provide more information on exactly what you want. If you wish to discuss this offl ist feel free to email me directly. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --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 an internal pbs for a samall company
Well, U right, many missing informations. The case is quite simple(I guess), we have dids, and each call to these dids has to be routed to the right handset thru Asterisk, no Ivr at this time, at least an answering machine in case of busy or not available users. For the rest, we need to be able to have external calls to pstn, or even to other sip phones form other providers. Is that enough? -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de trixter aka Bret McDanel Envoyé : mercredi 2 novembre 2005 13:48 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] Asterisk as an internal pbs for a samall company On Wed, 2005-11-02 at 13:36 +0100, Olivier Taylor wrote: Hello all, We'd like to use asteriek as an internal pbx connected to an external sip provider to make outbound/inbound calls to pstn. We have the provider and have installed an asterisk at the office. Does anyone have a sample config? We need 25 telephone numbers(dids), to be registerd to the provider and be able to ceceive calls. Any advice is welcome. Sorry for the noob question, Olivier What you want to do depends largely on what you want to do. While that seems like a cylic statement I will try to explain. You have said that you want to route calls between your asterisk box and the PSTN via a VoIP provider that you have. So far that seems simple, but how are those calls going to go bewteen the office workers and asterisk? You will need configurations for that. How are the inbound calls going to be routed? Via an IVR? Well you will have to configure that. There is a lot of information that is missing from this setup. www.voip-info.org has a lot of asterisk examples including configuration files. You may find something there that does what you want. I cant easily help you solve this problem (and suspect that no one else can either) until you provide more information on exactly what you want. If you wish to discuss this offl ist feel free to email me directly. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ --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: RE : [Asterisk-Users] Asterisk as an internal pbs for a samall company
On Wed, 2005-11-02 at 13:58 +0100, Olivier Taylor wrote:
Well,
U right, many missing informations.
The case is quite simple(I guess), we have dids, and each call to these dids
has to be routed to the right handset thru Asterisk, no Ivr at this time, at
least an answering machine in case of busy or not available users.
For the rest, we need to be able to have external calls to pstn, or even to
other sip phones form other providers.
Is that enough?
Not for 100% setup, but enoughto at least get you started. From what I
understand this is what it appears you want (I may be wrong, if I am let
me know).
You will want voicemail for each user. This is configured in
voicemail.conf
http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf
You will need to edit sip.conf for the voip provider (register and
context) and if the office workers use sip to asterisk one for each of
them as well.
http://www.voip-info.org/wiki-Asterisk+config+sip.conf
Lastly you will want to create a dialplan so that when a call comes in
from the DID it will then dial the appropriate user and if busy/no
answer goto voicemail. This is done from extensions.conf.
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf
You may want a macro like:
[macro-dialvmb]
exten = s,1,Dial(${ARG1},20,t)
exten = s,2,Voicemail(u${ARG2})
exten = s,3,Hangup
exten = s,102,Voicemail(b${ARG2})
exten = s,103,Hangup
Then for each inbound DID something like:
exten = 18005551212,1,Macro(dialvmb,SIP/user1,1234)
where user1 is the user defined in sip.conf, 1234 is the voicemail
extension defined in voicemail.conf and 18005551212 is the extension
that a given did goes to (ie last part of the register line).
Hope this helps
--
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605 Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
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RE : RE : [Asterisk-Users] Asterisk as an internal pbs for a samallcompany
It seems to be what I needed
Thanks for help.
Best regards,
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de trixter aka
Bret McDanel
Envoyé : mercredi 2 novembre 2005 14:09
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: RE : [Asterisk-Users] Asterisk as an internal pbs for a
samallcompany
On Wed, 2005-11-02 at 13:58 +0100, Olivier Taylor wrote:
Well,
U right, many missing informations.
The case is quite simple(I guess), we have dids, and each call to
these dids has to be routed to the right handset thru Asterisk, no Ivr
at this time, at least an answering machine in case of busy or not
available users. For the rest, we need to be able to have external
calls to pstn, or even to other sip phones form other providers. Is
that enough?
Not for 100% setup, but enoughto at least get you started. From what I
understand this is what it appears you want (I may be wrong, if I am let me
know).
You will want voicemail for each user. This is configured in voicemail.conf
http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf
You will need to edit sip.conf for the voip provider (register and
context) and if the office workers use sip to asterisk one for each of them
as well. http://www.voip-info.org/wiki-Asterisk+config+sip.conf
Lastly you will want to create a dialplan so that when a call comes in from
the DID it will then dial the appropriate user and if busy/no answer goto
voicemail. This is done from extensions.conf.
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf
You may want a macro like:
[macro-dialvmb]
exten = s,1,Dial(${ARG1},20,t)
exten = s,2,Voicemail(u${ARG2})
exten = s,3,Hangup
exten = s,102,Voicemail(b${ARG2})
exten = s,103,Hangup
Then for each inbound DID something like:
exten = 18005551212,1,Macro(dialvmb,SIP/user1,1234)
where user1 is the user defined in sip.conf, 1234 is the voicemail extension
defined in voicemail.conf and 18005551212 is the extension that a given did
goes to (ie last part of the register line).
Hope this helps
--
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605 Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
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RE: [Asterisk-Users] llamdas por 4 lineas a eeuu tarifa plana
Hunt, Bill wrote: While I don't disagree in principle, I think an issue is that much of the benefit of this list is the knowledge gained by reading about other people's problems and resolutions. If these discussions start being held in other languages we will not all be able to benefit from them. I agree. OTOH, this list is the canonical place for the best help and the most helpful gurus. As such, I don't think the list is overly sullied by the occasional conversation in another language--particularly given the struggles I have seen some non-native-speakers undergo when trying to express their situations clearly. If another list member is able to help them in a shared non-English language, it helps the original poster, and for indeed many people the answers are accessible, to boot. Another take is that deleting a message one doesn't want to read is a lot cheaper for the list than coping with the List Police, who if they had their way would choke the flow horribly with their incessant whining and demands for purity. My HO. B. Brian, I agree with you. English is not my native language, and, in fact, sometimes I suffer the problems you depict. But, I *hardly* try to do my best to put my thoughts in English In the other hand, a viable solution coul be that (at least the response to an non-english-language-only mail), include the same response in the 2 languages used (At least, is the way I try to procced in similar cases) The sure thing is that there are NO list similar to this in ANY other languaje, so, I think, flexibility in terms of language will be a plus. Regards. Juan. -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.12.7/156 - Release Date: 02/11/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Graphical interface
[EMAIL PROTECTED] - Original Message - From: Tomislav Parčina [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, November 02, 2005 7:41 AM Subject: [Asterisk-Users] Graphical interface Can you please suggest me some graphical interface (like AMP)? I have tried to install AMP but I have some problems and on AMP forum and mailing list I didn't get answer. Two things I need to have are. - list of calls for every user. - some information about Linux (processor load, HDD, network load...) Other things that I will welcome - operators panel - voicemail (to listen your voicemails) Thank you for your time. -- Tomislav Parčina Lama d.o.o. www.lama.hr tparcina#lama.hr ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.12.6/152 - Release Date: 10/31/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zap Polarity Reversal
Previously I would get two events on my Zap channel which indicated ringing and answered. Now I am getting polarity reversal events: Nov 2 07:01:25 NOTICE[18246]: chan_zap.c:6014 ss_thread: Got event 17 (Polarity Reversal)... Nov 2 07:01:28 NOTICE[18246]: chan_zap.c:6014 ss_thread: Got event 17 (Polarity Reversal)... I am using CVS Head from: Asterisk CVS HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 2005-11-02 05:13:32 UTC ___ --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] Installing beta2
Can you issue the following command on FC3 and let us know the results? rpm -q kernel-source zlib zlib-devel openssl openssl-devel On 11/2/05, Lee Archer [EMAIL PROTECTED] wrote: Hi, I had removed all old versions before starting and downloaded from CVS. Regards Lee From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: 02 November 2005 12:20 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Installing beta2 Are you installing over a previous source tree? If so, please rm -rf the previous source tree and install the new source tree from scratch. On 11/2/05, Lee Archer [EMAIL PROTECTED] wrote: Once built no matter whether I do make install or make clean I get the same output [EMAIL PROTECTED] asterisk]# make clean build_tools/make_version_h include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c build_tools/make_version_h include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c make: *** [.depend] Interrupt I am using FC3 and any help would be appreciated. Regards Lee ___ --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 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.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: RE : [Asterisk-Users] Asterisk as an internal pbs for a samall company
Thankyou, this was a great primer for me also.
Chris
trixter aka Bret McDanel wrote:
On Wed, 2005-11-02 at 13:58 +0100, Olivier Taylor wrote:
Well,
U right, many missing informations.
The case is quite simple(I guess), we have dids, and each call to these dids
has to be routed to the right handset thru Asterisk, no Ivr at this time, at
least an answering machine in case of busy or not available users.
For the rest, we need to be able to have external calls to pstn, or even to
other sip phones form other providers.
Is that enough?
Not for 100% setup, but enoughto at least get you started. From what I
understand this is what it appears you want (I may be wrong, if I am let
me know).
You will want voicemail for each user. This is configured in
voicemail.conf
http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf
You will need to edit sip.conf for the voip provider (register and
context) and if the office workers use sip to asterisk one for each of
them as well.
http://www.voip-info.org/wiki-Asterisk+config+sip.conf
Lastly you will want to create a dialplan so that when a call comes in
from the DID it will then dial the appropriate user and if busy/no
answer goto voicemail. This is done from extensions.conf.
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf
You may want a macro like:
[macro-dialvmb]
exten = s,1,Dial(${ARG1},20,t)
exten = s,2,Voicemail(u${ARG2})
exten = s,3,Hangup
exten = s,102,Voicemail(b${ARG2})
exten = s,103,Hangup
Then for each inbound DID something like:
exten = 18005551212,1,Macro(dialvmb,SIP/user1,1234)
where user1 is the user defined in sip.conf, 1234 is the voicemail
extension defined in voicemail.conf and 18005551212 is the extension
that a given did goes to (ie last part of the register line).
Hope this helps
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RE: [Asterisk-Users] Graphical interface
Thank you, this is definitely an option. Right now I'm trying to make something work on my Linux installation (FC4). And I like to install as much things on my own, so that I really can se how that stuff works. Tomislav -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk Sent: 2. studeni 2005 14:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Graphical interface [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
Re: [Asterisk-Users] llamdas por 4 lineas a eeuu tarifa plana
please remove me from your mailing list. [EMAIL PROTECTED]asterisk [EMAIL PROTECTED] wrote: http://lists.digium.com/mailman/create This list supports English (USA). Possibly our spanish speaking friends need their own list? Thanks, Steve - Original Message - From: Carlos Alperin To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Tuesday, November 01, 2005 9:36 AM Subject: RE: [Asterisk-Users] llamdas por 4 lineas a eeuu tarifa plana Sir, If the people had asked questions on Spanish, I dont have any problem on answer those questions. Not everybody speaks English, and I didnt know any rule that forbids any other language. Sorry, From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sergey OkhapkinSent: Tuesday, November 01, 2005 9:04 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] llamdas por 4 lineas a eeuu tarifa plana AFAIK, the official language of this mailing list is English.On Tue, 2005-11-01 at 08:54 -0500, Carlos Alperin wrote: Walter,No se acerca de que es lo mas atractivo. El servicio puede ser en el horario que tu quieras (Nosotros trabajamos 24 hs, 7 dias a la semana, 365 dias al año), pero si quieres restricción de horario, se puede hacer.No dije nada acerca del españolCarlos From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walter WillisSent: Monday, October 31, 2005 7:41 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] llamdas por 4 lineas a eeuu tarifa plana 2005/10/31, Walter Willis [EMAIL PROTECTED]:Tengo un asterisk en al oficina de un cliente , que quiere hacer llamadas ilimitadas a estados unidos; las llamadas tienen que ser al mismo tiempo.alguien ofrecio una conexion iax2 para los 4 usuarios.no tienes algo mas atractivo se supone que funcionaria eso desde las 3:00pm hasta las 8:00pm diariamente ecepto los domingos.cuanto costaria eso.y gracias por als respuestas. noes que el español sea malo sino que el teclado era malo y aparte el tiempo era corto. 2005/10/31, Manny A. Wise [EMAIL PROTECTED]: NO estoy para juegos, te conteste y te ofreci lo que pediste, cual es tu problema?-Original Message-From: Walter Willis [mailto:[EMAIL PROTECTED]] Sent: Monday, October 31, 2005 7:05 PMTo: Manny A. Wise Subject: Re: [Asterisk-Users] llamdas por 4 lineas a eeuu tarifa plana pense que ibas a decir algo mas interesante, parece que estas probando tu correo en la lista. 2005/10/31, Manny A. Wise [EMAIL PROTECTED]: Y si estas en el Peru y hablas espanol, por que no lo escribes correctamente?-Original Message-From: Walter Willis [mailto:[EMAIL PROTECTED]] Sent: Monday, October 31, 2005 4:50 PM To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] llamdas por 4 lineas a eeuu tarifa plana you provider tha service unlimited call usa i want the trunk with 4 users to unlimited USA iax or iax2 2005/10/31, Manny A. Wise [EMAIL PROTECTED]: Do you speak English?Your Spanish is bad too..Puedo ayudarte en espanol Yo puedo link tu asterisk a my asterisk Manny-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of Walter WillisSent: Monday, October 31, 2005 1:03 PM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] llamdas por 4 lineas a eeuu tarifa planatengo un asterisk, alguien conoce algun proveedor que brinde el sistema de li nkar mi asterisk a su servicio para tener tarifa plana a eeuu.para llamar por 4 conexiones al miamo tiempo desde mi asterisk?me parece haber visto que se configuraba con una troncal iax22005/10/31, [EMAIL PROTECTED] [EMAIL PROTECTED]: Does anyone know where the official release of Asterisk 1.2 is? Do we havea time-frame of when this version will be released and how much longer itwill be in BETA. ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-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 listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message.Checked by AVG Free Edition.Version: 7.1.362 / Virus Database: 267.12.6/152 - Release Date: 10/31/2005___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options
RE: [Asterisk-Users] Installing beta2
Hi it says [EMAIL PROTECTED] ~]# rpm -q kernel-source zlib zlib-devel openssl openssl-devel package kernel-source is not installed zlib-1.2.1.2-3.fc3 zlib-devel-1.2.1.2-3.fc3 openssl-0.9.7a-42.1 openssl-devel-0.9.7a-42.1 Which is odd cos the sources are installed. I'm using the 2.6.9-1.667 kernel and have all the links to the build directory. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: 02 November 2005 13:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Installing beta2 Can you issue the following command on FC3 and let us know the results? rpm -q kernel-source zlib zlib-devel openssl openssl-devel On 11/2/05, Lee Archer [EMAIL PROTECTED] wrote: Hi, I had removed all old versions before starting and downloaded from CVS. Regards Lee From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: 02 November 2005 12:20 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Installing beta2 Are you installing over a previous source tree? If so, please rm -rf the previous source tree and install the new source tree from scratch. On 11/2/05, Lee Archer [EMAIL PROTECTED] wrote: Once built no matter whether I do make install or make clean I get the same output [EMAIL PROTECTED] asterisk]# make clean build_tools/make_version_h include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c build_tools/make_version_h include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c callerid.c cdr.c channel.c chanvars.c cli.c config.c cryptostub.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c make: *** [.depend] Interrupt I am using FC3 and any help would be appreciated. Regards Lee ___ --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 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.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 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com
RE: [Asterisk-Users] Graphical interface
On Wed, 2 Nov 2005, [iso-8859-2] Tomislav Parčina wrote: Thank you, this is definitely an option. Right now I'm trying to make something work on my Linux installation (FC4). And I like to install as much things on my own, so that I really can se how that stuff works. Then I guess you'll want to install AMP. It does all the things you want. It was just that you said you couldn't install it - in which case [EMAIL PROTECTED] will install it for you. Steve ___ --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] Graphical interface
On Wed, Nov 02, 2005 at 01:41:38PM +0100, Tomislav Parčina wrote: Can you please suggest me some graphical interface (like AMP)? I have tried to install AMP but I have some problems and on AMP forum and mailing list I didn't get answer. Two things I need to have are. - list of calls for every user. keyword: cdr. The simplest interface is to load the default CDR CSV file into a spreadsheet. - some information about Linux (processor load, HDD, network load...) webmin? phpsysinfo? Other things that I will welcome - operators panel - voicemail (to listen your voicemails) Thank you for your time. -- Tomislav Parčina Lama d.o.o. www.lama.hr tparcina#lama.hr ___ --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 -- 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
[Asterisk-Users] extension
Title: extension I would like to know how to set up will be in one sipura 2002 box and have another same Extension but in different locations like bedroom and kitchen I believe I need two sipura boxes are need it. Can you help? ___ --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] feature.conf in 1.2beta2
I tried it but nothing happens, seems asterisk is not getting the dtmf or something and I get an error on the CLI saying something like this when the keys are pressed. -- Attempting native bridge of SIP/212-227a and SIP/201-dc6b -- Native bridge of SIP/212-227a and SIP/201-dc6b was unsuccessful -- Attempting native bridge of SIP/212-227a and SIP/201-dc6b -- Native bridge of SIP/212-227a and SIP/201-dc6b was unsuccessful That happened when I tried pusshing *1 as defined on features.conf for what Im trying to do... This feature is something Ive been waiting for a long time now... :) can do pretty nice things if I make it work ... |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Matt Riddell |Sent: Tuesday, November 01, 2005 11:21 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] feature.conf in 1.2beta2 | |Anton Krall wrote: | Guys. | | Can somebody explain a bit further the use of this new feature in | features.conf | | [applicationmap] | ;testfeature = #9,callee,Playback,tt-monkeys ;Play tt-monkes to | ;callee if #9 was | pressed | | I cant find more info anywhere and I suspect this is what I |have been | looking for :) | |This one's really great. It basically lets you assign any |application to a DTMF code. So for example you could play |something to one of the parties or run an agi etc. | |I guess you'll probably have to use on of the flags in the |dial command that keeps Asterisk in the loop. | |-- |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] Zap Polarity Reversal
Previously I would get two events on my Zap channel which indicated ringing and answered. Now I am getting polarity reversal events: Nov 2 07:01:25 NOTICE[18246]: chan_zap.c:6014 ss_thread: Got event 17 (Polarity Reversal)... Nov 2 07:01:28 NOTICE[18246]: chan_zap.c:6014 ss_thread: Got event 17 (Polarity Reversal)... I am using CVS Head from: Asterisk CVS HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 2005-11-02 05:13:32 UTC ___ Sounds like a new feature. Have you tried reversing the polarity or putting a butt set that indicates pol on the line? I don't know that it makes a difference but you might as well correct it if it is reversed. Thanks, Steve ___ --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] extension
If the two phones are going to have the same extension its just a matter of wiring two phones to one port on the sipura unit. If you disconnect the telco service from your house's internal phone wiring and plug one of the Sipura ports into a phone jack, that sipura device will provide power and dial tone to the rest of the phone jacks in the house. So if you really only want to have 1 extension, I would find where the line from the telco connects to the house's internal wiring and disconnect it, then run that line straight to your asterisk box. Then after making sure there's no dial tone on any of your phone jacks, connect the sipura box to one of them. Then there should be dial tone on all of them. Wiring would go like this: telco service - asterisk - SIP adapter --- house wiring --- phones Guido Amendano wrote: I would like to know how to set up will be in one sipura 2002 box and have another same Extension but in different locations like bedroom and kitchen I believe I need two sipura boxes are need it. Can you help? ___ --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] Extensions
How can I configure Asterisk to tell me if there are messages on my voice mail as soon as I hook up an internal phone? Regards, Andrea Frigo ___ --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] llamdas por 4 lineas a eeuu tarifa plana
I tried to open the spanish list, but the mailmain server didn't let me open, because didn't recognize my password. So, until we can do that JODERSE. He tratado de abrir la lista en español, pero el administrador de la lista no me lo ha permito, ya que no reconocio mi contraseña. Por lo tanto, hasta que lo podamos hacer, SO Sorry. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan Janczuk Sent: Wednesday, November 02, 2005 8:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] llamdas por 4 lineas a eeuu tarifa plana Hunt, Bill wrote: While I don't disagree in principle, I think an issue is that much of the benefit of this list is the knowledge gained by reading about other people's problems and resolutions. If these discussions start being held in other languages we will not all be able to benefit from them. I agree. OTOH, this list is the canonical place for the best help and the most helpful gurus. As such, I don't think the list is overly sullied by the occasional conversation in another language--particularly given the struggles I have seen some non-native-speakers undergo when trying to express their situations clearly. If another list member is able to help them in a shared non-English language, it helps the original poster, and for indeed many people the answers are accessible, to boot. Another take is that deleting a message one doesn't want to read is a lot cheaper for the list than coping with the List Police, who if they had their way would choke the flow horribly with their incessant whining and demands for purity. My HO. B. Brian, I agree with you. English is not my native language, and, in fact, sometimes I suffer the problems you depict. But, I *hardly* try to do my best to put my thoughts in English In the other hand, a viable solution coul be that (at least the response to an non-english-language-only mail), include the same response in the 2 languages used (At least, is the way I try to procced in similar cases) The sure thing is that there are NO list similar to this in ANY other languaje, so, I think, flexibility in terms of language will be a plus. Regards. Juan. -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.12.7/156 - Release Date: 02/11/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extensions
This works for me, your mileage may vary: in sip.conf add these two lines under the sip user: mailbox=/[EMAIL PROTECTED] /notifymimetype=application/simple-message-summary example for mailbox= [EMAIL PROTECTED] The SIP device must support this feature of course. And if you're not using SIP then I have no idea what to do :) Andrea Frigo wrote: How can I configure Asterisk to tell me if there are messages on my voice mail as soon as I hook up an internal phone? Regards, Andrea Frigo ___ --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] intel e7230 chipset
Does anyone know if the intel e7230 chipset in the new dell poweredge sc430 and poweredge 830 servers is compatible with the te110p and tdm400p cards? I know there were problems with previous generation dells, but I've read that these work fine. Can anyone confirm this? thanks robbie ___ --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 show peers
Mark Edwards wrote: This indicates that 602 is a dynamic host. It must therefore register with the pbx so that the pbx knows where to send data. In this state it is unregistered so it will be unlikely you can call it. That was I expected, that I cannot call it, but I could That gives me more the hint, that sip show peers is not telling always the truth It also did not come up at the moment I called. bye Ronald Wiplinger Regards, Mark -Original Message- From: Ronald Wiplinger [mailto:[EMAIL PROTECTED] Sent: Monday, 31 October 2005 7:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] sip show peers Sip show peers includes the line: 602/602(Unspecified)D N 0UNKNOWN However, I can call it? Should not peer means if it is reachable? bye Ronald Wiplinger ___ --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 -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com http://voip.elmit.com http://e-paper.elmit.com Tel. (M) +886.939.775.516 (O) +886.2.2835.7765 (ENUM) or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ --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] Error with one of my Zapata channels
Ever since I started playing with Beta versions of Asterisk, I've had a problem. It might just be coincidence though, since before that I didn't touch the Asterisk PC for a good 2 weeks and I had done alot playing around with config files. I have a 4 port FXS/FXO card (with 2 of each in). I asked in this mailing list about an Audio pipe broken (error message from Asteirsk) and I was told it was usually Zapata.conf that was the problem, and I confirmed it was. The problem si I don't know what the exact cause is. I have my channels defined in zaptel.conf as fxoks=1,2 fxsks=3,4 In zapata.conf, I have: (I commented out the other 2 cards) signalling = fxo_ks context = test channel = 1 signalling = fxo_ks context = test channel = 2 Now, this gives me the Audio pipe broken error. BUT, If I comment out channel 1 out of zapata.conf, my second phone (connected to channel 2) works perfectly (as far as I can tell, the dialplan works). Further investigation showed that my var/log/messages had the following two lines in reference to those cards: Module 0: FAILED FXS (FCC) Module 1: INSTALLED -- AUTO FXS/DPO But ztcfg -v shows no error. Am I dealing with a hardware issue? Can I fix it, or is this a defective card? This was seen on v-1-2-0Beta1 but I have just confirmed the same behavior on beta2. I can't tell exactly what you're doing here, so I'll try to offer a couple of suggestions that might help resolve the issue. What is defined in /etc/zaptel.conf has to exactly match what is installed on the TDM card. (Might have to check the color of the TDM modules on each slot to validate what is plugged in for each module. Red modules are fxo, green are fxs modules. Module slot #1 is the one closest to the rear of the card and nearest the rj45/rj11 jacks.) You can't leave out any definitions. If four modules are installed on the card, then four definitions have to exist in /etc/zaptel.conf. Since I don't have a clue which linux distro your using, I'll use the FC3 approach for the following. If you modprobe zaptel and wctdm, then run ztcfg -vvv, you shoud see the four modules like this: Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. (The above is showing four red fxo modules installed on my system.) If you get error messages at this point, then contact digium support to have them help with an RMA on a possible defective module. If you get a response similar to the above, then the TDM card and its modules are _likely_ okay. If you have four modules installed and they are reported correctly (as noted above) then you have to have matching entries for each defnition in /etc/asterisk/zapata.conf. You can't just arbitrarily comment those out or screw with them. If they don't match what's in /etc/zaptel.conf, you _will_ get error messages when starting asterisk. If there are four modules installed on the TDM card, then you will need four entries something like this: context=inbound-bus signalling=fxs_ks group=1 echocancel=yes echotraining=800 echocancelwhenbridged=yes rxgain=0 txgain=0 channel = 1 context=inbound-home signalling=fxs_ks Channel = 2 etc, etc, etc. The above two entries are for FXO modules in slots one and two of the TDM card. The FXS modules will need different signalling= statements, etc. In the above, note that everything defined in channel 1 is inherited in channel 2, except we changed the signalling and context. So, to make your life easier to learn/understand, don't use this inherited approach, but rather specify exactly those parameters needed for each channel. Also remember that any changes made to zapata.conf _will_ require a complete stop and restart of asterisk to become effective. A simple reload will not cut it. Once you go through the above, you should have a working system. If not, contact digium support to resolve any bad modules, etc. Rich ___ --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] Extensions
For analog phones - same thing 8-) except it is in zapata.conf mailbox=whatever ; under (er just above) the channel. Should give you a stutter dial tone. Brett On 11/2/2005, Adam Moffett [EMAIL PROTECTED] wrote: This works for me, your mileage may vary: in sip.conf add these two lines under the sip user: mailbox=/[EMAIL PROTECTED] /notifymimetype=application/simple-message-summary example for mailbox= [EMAIL PROTECTED] The SIP device must support this feature of course. And if you're not using SIP then I have no idea what to do :) Andrea Frigo wrote: How can I configure Asterisk to tell me if there are messages on my voice mail as soon as I hook up an internal phone? Regards, Andrea Frigo ___ --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] TDM dial in question
I am trying to figure out how to setup asterisk with a TDM400 (TDM04B), so that the first 3 lines incoming will be answered and the 4th line is just for outgoing calls but doesnt answer on incoming calls. The easiest way to do that is to give the channel a weird context name. For example, if module #4 / channel #4 is the one that you don't want answered, then use a definition something like this in zapata.conf: context=NoAnswer signalling=fxs_ks other defintions as needed channel = 4 Just make sure there is not a context in extensions.conf for NoAnswer. ___ --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] Extensions
You should receive a short ring every 5 or 10 minutes if you have voicemails on your box. Now, if you have an IP Phone, you can have a led (Like the Cisco 7960, or an icon like on the Swissvoice IPS-10) that reports you that condition. Regards, Carlos Alperin From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrea Frigo Sent: Wednesday, November 02, 2005 10:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Extensions How can I configure Asterisk to tell me if there are messages on my voice mail as soon as I hook up an internal phone? Regards, Andrea Frigo ___ --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] Latest CVS just noticed this warning for the first time.
Just started getting this warning message about every minute. ast_sched_runq ran 20 scheduled tasks all at once I know it's a warning but Mark/Kevin Co must have thought it worth mentioning. So its a patch from me, that I may regret. I'd strongly suggest leaving it in there for a while as its likely to help point out issues that admins didn't even know existed. (I'm sure it will generate questinos as well, but that's not necessarily a bad thing.) Rich ___ --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] Zap Polarity Reversal
Previously I would get two events on my Zap channel which indicated ringing and answered. Now I am getting polarity reversal events: Nov 2 07:01:25 NOTICE[18246]: chan_zap.c:6014 ss_thread: Got event 17 (Polarity Reversal)... Nov 2 07:01:28 NOTICE[18246]: chan_zap.c:6014 ss_thread: Got event 17 (Polarity Reversal)... I am using CVS Head from: Asterisk CVS HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 2005-11-02 05:13:32 UTC ___ Sounds like a new feature. Have you tried reversing the polarity or putting a butt set that indicates pol on the line? I don't know that it makes a difference but you might as well correct it if it is reversed. That actually sounds more like whatever telco he's connecting to is providing answer supervision in the form of polarity reversal. Without knowing more about which country / telco, there is no way to tell. Note the polarity reversal is happening _after_ asterisk gets a call, therefore not likely to have anything to do with reversed tip/ring. That same message does not occur in the US with the analog TDM card, so not sure what the OP has or is connected to. ___ --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] Voicemail in Realtime mode
Hi, I have installed the asterisk 1.2 beta version and I have created the voicemail table described on this page http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail, but when I start the asterisk server I receive the following error. Any idea ? Thank you [app_voicemail.so] = (Comedian Mail (Voicemail System)) Nov 2 16:03:58 WARNING[3118]: app_voicemail.c:6140 load_config: Error reading voicemail config Nov 2 16:03:58 WARNING[3118]: loader.c:403 __load_resource: app_voicemail.so: load_module failed, returning -1 Nov 2 16:03:58 WARNING[3118]: loader.c:543 load_modules: Loading module app_voicemail.so failed! ___ --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] Re: Anyone aware of a current Dell server model with 3PCI slots
Looking for a Current Dell model, tower or 2U rackmount, that has (3) usable PCI slots? Was just cruising Dell.com and can't find a detailed spec on any of the server offerings that tells me the number of PCI slots available. Anyone using Dell for PBX builds can point me in the right direction? IIRC, the PowerEdge 2850 is used by Digium for ABE, and Signate for some of their installations, so I think it should work well. It is a 2U rack mount, and may be available as a tower as well. I think it has 3-4 PCI slots, but don't quote me. Tom ___ --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 bridge fax from pri to fxs
I have a TE110P and a TDM10B. Via DID, I want to route calls to the fax number to the fxs port to which the fax machine will be connected. I believe this will work, but wanted to know if anyone has done this. Do I need to set faxdetect=both in zapata.conf? I am assuming that Asterisk will bridge between the two cards and that the usual fax over IP won't be a factor. Any advice is greatly appreciated. Cheers, Kevin ___ --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 bridge fax from pri to fxs
On Wednesday 02 November 2005 11:17, Kevin Hanson wrote: I have a TE110P and a TDM10B. Via DID, I want to route calls to the fax number to the fxs port to which the fax machine will be connected. Do I need to set faxdetect=both in zapata.conf? I am assuming that Asterisk will bridge between the two cards and that the usual fax over IP won't be a factor. the 'faxdetect' parameter does one thing and one thing only: when a fax tone is heard, Asterisk will jump to the 'fax' extension in the channel's context. If you are using a specific DID for faxes you don't need that at all. exten = 1234567,1,Dial(Zap/g2,,g) exten = 1234567,2,Macro(handle-hangup) is how I'd do it. I'd define your PRI channels in group 1, and your FXS ports that the fax machine(s) are hooked up to in group 2. If you only want one fxs port for one fax machine, then say Dial(Zap/25) or whatever zapata port the fxs machine's on. Works like a charm for me, anyway. -A. ___ --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] TDM0xB vs. SIP for FXO
Hi, I am planning to connect my Asterisk PBX to one or two POTS lines, and am wondering if it is better to use a TDM card for this, or one or two SIP devices with FXO ports on them (such as an SPA-3000, Grandstream 488). I am interested in voice quality and reliability of operation and am wondering if one of these options is better than the other in this regard. Thanks, Rusty ___ --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] feature.conf in 1.2beta2
Anton Krall wrote: I tried it but nothing happens, seems asterisk is not getting the dtmf or something and I get an error on the CLI saying something like this when the keys are pressed. -- Attempting native bridge of SIP/212-227a and SIP/201-dc6b -- Native bridge of SIP/212-227a and SIP/201-dc6b was unsuccessful -- Attempting native bridge of SIP/212-227a and SIP/201-dc6b -- Native bridge of SIP/212-227a and SIP/201-dc6b was unsuccessful What makes you think this is an error? ___ --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] Please Press Any Key to Accept a Call
For everyone that had inquired about the Find-Me/Follow-Me application, it's now up in the bug tracker at http://bugs.digium.com/view.php?id=5574. It should compile cleanly against a 1.2b2 install. On 10/14/05, BJ Weschke [EMAIL PROTECTED] wrote: CF - You're right. Most of this can be done with the dial plan. I wrote the app though because I wanted to be able to have the option to work with both channels at the same time through a threaded model. The dial plan doesn't let me do that, and there's no reason it should. So, in this scenario, we're actually dialing already to the first number on the list while the caller is still hearing the annunciator from Allyson about please wait while we try and connect your call. The approach right now is really simple and probably could have been done in the dial plan, but I kind of envisioned expanding upon this over time and trying to get the app to act like the Wildfire follow-me application works now with some Sphinx integration down the road. It's going to be a little while before we give things like Wildfire a run for their money, but Mark, Digium, and the Asterisk community have given us a more than adequate platform and framework to get us there. On 10/14/05, C F [EMAIL PROTECTED] wrote: BJ, thanks alot for the coding but I see no reason for it as all you mention is doable currently in Asterisk using some DP magic. Will just search the list and you should find some examples on how to do it (I think there is even an example on the wiki try: http://www.voip-info.org/wiki-asterisk+cmd+dial ) On 10/14/05, BJ Weschke [EMAIL PROTECTED] wrote: I have coded a new application in Asterisk called app_followme that will do what you're looking for. The caller who made the call originally is also optionally put on hold music while the hunt is going on. There's also planned functionality for blacklisting certain callerIDs so a caller who is blacklisted will think the find-me/follow-me is working, but in reality it's just putting them in a holding pattern and then routing them to voicemail after waiting for about 20-30 seconds evil grin. The code isn't really cleaned up yet from my initial alpha / unit testing on it which is why I haven't put it on the bugtracker yet, but it's quite functional now and I'd like for more people to start testing it if they see a use for this. I'll try to get it up there in the next couple days. It's new functionality, and therefore, won't make it into the 1.2 release of Asterisk, but it doesn't really interfere with much anything else in Asterisk so you should be able to apply the patch cleanly to any recent HEAD branch and probably 1.2 once it's released. BJ On 10/14/05, Will Glass-Husain [EMAIL PROTECTED] wrote: Hi, I'd like to add a feature to my asterisk system that tries to find a user among a couple of locations, and then goes to internal voicemail if the user doesn't pick up. (e,g, an internal extension and a cell phone). The catch is that I want the user to manually accept the call to prevent it from going (for example) to the voice mail on my cell phone. Scenario * Call comes in, outside caller dials 100 * Desk phone for user Joe rings. No answer * Joe's house phone rings. * Joe's wife picks up and hears a voice Please press any key to accept a call for extension 100. * Joe's wife hangs up. * Joe's cell phone rings. * Joe picks up and hears a voice Please press any key to accept a call for extension 100. * Joe presses 1 and says Hello this is Joe. Alternately, in the penultimate step * Cell voice mail picks up. * Voice says Please press any key to accept a call for extension 100. No keys pressed since it's a voice mail * Call is routed to Asterisk voicemail. It seems straight forward to try multiple locations, but I'm not seeing how to only patch the call through if the user responds with a key press. Thanks, WILL ___ --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
Re: [Asterisk-Users] Re: Anyone aware of a current Dell server model with 3PCI slots
We have a poweredge 2850 that we use for our VoIP server and it has 3 PCI slots. On 11/2/05, Tom Rymes [EMAIL PROTECTED] wrote: Looking for a Current Dell model, tower or 2U rackmount, that has (3) usable PCI slots? Was just cruising Dell.com and can't find a detailed spec on any of the server offerings that tells me the number of PCI slots available. Anyone using Dell for PBX builds can point me in the right direction? IIRC, the PowerEdge 2850 is used by Digium for ABE, and Signate for some of their installations, so I think it should work well. It is a 2U rack mount, and may be available as a tower as well. I think it has 3-4 PCI slots, but don't quote me. Tom ___ --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] Satellite WAN
We have built an Asterisk network using an MPLS-based IP VPN. We have one location in New Brunswick Canada that consistently gives us major quality problems, whereas the others are flawless. Quality problems take the form of static, poor voice tonality, popping clicking, drops, sporadic echo, you name it. The latency of a QoS prioritized packet between the Canada site and our hub in Atlanta is 85ms (ping). I have been searching for an alternative network provider, but I'm told that they would all take the same route from the US into Canada, as there is simply no major backbone running into NB east of Toronto. So now I'm thinking about satellite. I have no idea if a) this would even be economically feasible, and b) if the latency would be any better. If anyone out there has had any such satellite network experience with VoIP, I like to hear from you. Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation 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] feature.conf in 1.2beta2
On Wednesday 02 November 2005 11:42, Kevin P. Fleming wrote: -- Attempting native bridge of SIP/212-227a and SIP/201-dc6b -- Native bridge of SIP/212-227a and SIP/201-dc6b was unsuccessful -- Attempting native bridge of SIP/212-227a and SIP/201-dc6b -- Native bridge of SIP/212-227a and SIP/201-dc6b was unsuccessful What makes you think this is an error? You have to admit that pretty much anything being unsuccessful looks like it might be a problem. It's all in the perception. I've been thinking of a way to get across the idea that a native bridge was unsuccessful in more friendly terms for a bit but nothing really concise has come to mind. Some reason code might be handy... unable due to differing codecs, unable due to necessity to listen to audio... I dunno how to concisely convey that information. -A. ___ --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] Satellite WAN
We have a few satellite trunks for VoIP in Africa and have some experience. Please mail me off list and we can discuss it [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Robins Sent: den 2 november 2005 18:01 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Satellite WAN We have built an Asterisk network using an MPLS-based IP VPN. We have one location in New Brunswick Canada that consistently gives us major quality problems, whereas the others are flawless. Quality problems take the form of static, poor voice tonality, popping clicking, drops, sporadic echo, you name it. The latency of a QoS prioritized packet between the Canada site and our hub in Atlanta is 85ms (ping). I have been searching for an alternative network provider, but I'm told that they would all take the same route from the US into Canada, as there is simply no major backbone running into NB east of Toronto. So now I'm thinking about satellite. I have no idea if a) this would even be economically feasible, and b) if the latency would be any better. If anyone out there has had any such satellite network experience with VoIP, I like to hear from you. Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation 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] Satellite WAN
Adam, I personally think that replacing hard-wired network and going with Sats is a mistake. Judging from pure round-trip delay you measured the packet round trip seems sufficient to have a good conversation, but pinging is not enough to trouble shoot the network problems. You will need to do a lot more work to identify the problem with this location. If both locations are under your control, then I would put network probes in both places to identify exactly when and how the quality problems appear. Network probes would identify the type and the amount of traffic both sides are sending and receiving. There are network probes that can even do Voice Quality Analysis and determine how well your network is performing. As a side step, I would also look at internal location in New Brunswick, because that is the only location you are having problems with. I would check to see if there are simple network problems like bad network port, network card, packet collision on the network, network card on routers, etc. I am sure you have already considered simple things like that, however you need to methodically go thru each one to see where the problems are. Replacing the network would be my last alternative. If you are at that point, well then just ignore this email. Otherwise, there are plenty of things you can do before taking such a drastic measure. HTH Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Robins Sent: Wednesday, November 02, 2005 12:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Satellite WAN We have built an Asterisk network using an MPLS-based IP VPN. We have one location in New Brunswick Canada that consistently gives us major quality problems, whereas the others are flawless. Quality problems take the form of static, poor voice tonality, popping clicking, drops, sporadic echo, you name it. The latency of a QoS prioritized packet between the Canada site and our hub in Atlanta is 85ms (ping). I have been searching for an alternative network provider, but I'm told that they would all take the same route from the US into Canada, as there is simply no major backbone running into NB east of Toronto. So now I'm thinking about satellite. I have no idea if a) this would even be economically feasible, and b) if the latency would be any better. If anyone out there has had any such satellite network experience with VoIP, I like to hear from you. Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation 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] Time based call direction
I would like to manipulate phone call direction to voicemail for lunch, after hours etc, but am unsure how to do this. Could someone point me to a howto or quickly explain the concept? Thanks Neri ___ --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] Satellite WAN
I have no experience in the matter whatsoever ;) But, I can say that long distance phone calls (non-voip) are sometimes carried over sattelite when fiber is not available. It must be possible for voip, but the latency and jitter would be tremendous and although I am not an expert on the matter, I would suggest that you would only be replacing one set of problems with a new set of problems. Adam Robins wrote: We have built an Asterisk network using an MPLS-based IP VPN. We have one location in New Brunswick Canada that consistently gives us major quality problems, whereas the others are flawless. Quality problems take the form of static, poor voice tonality, popping clicking, drops, sporadic echo, you name it. The latency of a QoS prioritized packet between the Canada site and our hub in Atlanta is 85ms (ping). I have been searching for an alternative network provider, but I'm told that they would all take the same route from the US into Canada, as there is simply no major backbone running into NB east of Toronto. So now I'm thinking about satellite. I have no idea if a) this would even be economically feasible, and b) if the latency would be any better. If anyone out there has had any such satellite network experience with VoIP, I like to hear from you. Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation 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] Satellite WAN
Are you having the same problems under terrestreal links ? which codec do you use, are you using a dedicated channel on the vsat for it to take the upstream load ? Whats your jitter settings ? On 11/2/05, Adam Moffett [EMAIL PROTECTED] wrote: I have no experience in the matter whatsoever ;) But, I can say that long distance phone calls (non-voip) are sometimes carried over sattelite when fiber is not available. It must be possible for voip, but the latency and jitter would be tremendous and although I am not an expert on the matter, I would suggest that you would only be replacing one set of problems with a new set of problems. Adam Robins wrote: We have built an Asterisk network using an MPLS-based IP VPN. We have one location in New Brunswick Canada that consistently gives us major quality problems, whereas the others are flawless. Quality problems take the form of static, poor voice tonality, popping clicking, drops, sporadic echo, you name it. The latency of a QoS prioritized packet between the Canada site and our hub in Atlanta is 85ms (ping). I have been searching for an alternative network provider, but I'm told that they would all take the same route from the US into Canada, as there is simply no major backbone running into NB east of Toronto. So now I'm thinking about satellite. I have no idea if a) this would even be economically feasible, and b) if the latency would be any better. If anyone out there has had any such satellite network experience with VoIP, I like to hear from you. Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation 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 -- Ariel Taranto 619-568.0802 ___ --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] Time based call direction
I just went through the same thing. I settled on the GoToIfTime application. One strange thing about GoToIfTime is that it doesn't allow an else argument, so you'll need a sequence of if's to get things done. try something along these lines: [yourcontext] ;lunchtime exten = s,1,GoToIfTime(12:00-13:00?yourcontext|LUNCH|1) ;after work exten = s,2,GoToIfTime(17:00-23:59?yourcontext|CLOSED|1) ;before work exten = s,3,GoToIfTime(00:00-07:59?yourcontext|CLOSED|1) ;if we got this far, must be we're open exten = s,4,GoTo(yourcontext|OPEN|1) ;;Handle lunchtime calls exten = LUNCH,1,[do something] exten = CLOSED,1,[do soemthing else] exten = OPEN,1,[do yet another thing] Rene Nelson wrote: I would like to manipulate phone call direction to voicemail for lunch, after hours etc, but am unsure how to do this. Could someone point me to a howto or quickly explain the concept? Thanks Neri ___ --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] intel e7230 chipset
Robbie Hughes wrote: Does anyone know if the intel e7230 chipset in the new dell poweredge sc430 and poweredge 830 servers is compatible with the te110p and tdm400p cards? I know there were problems with previous generation dells, but I've read that these work fine. Can anyone confirm this? thanks robbie I have a PowerEdge 830 that I am getting ready to install at a customer site. I have been testing it in my lab w/ a TDM04B and it works fine. I won't be able to test w/ the TE110P until I get on site and have access to the PRI. We are installing this weekend. I am not expecting problems, but will reply back if I do. Cheers, Kevin ___ --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] server hardware
AudioCodes is widely available. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ryan Sent: Tuesday, November 01, 2005 7:40 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [Asterisk-Users] server hardware On Tue, Nov 01, 2005 at 05:44:40PM -0800, William Boehlke exclaimed: We ship multiple Dell servers every week. Haven't tested the new cards but generally you're fine with Digium T1 if you limit yourself to one card per server. When we are less than T1, we use an external SIP gateway. Which external SIP gateway do you recommend? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.12.7/153 - Release Date: 11/1/2005 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.12.7/157 - Release Date: 11/2/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Time based call direction
BTW, show application GoToIfTime in the CLI will tell you the whole syntax. It can also take days of the week and I think months of the year as arguments, but that wasn't an issue for me since we're 7 days a week. Adam Moffett wrote: I just went through the same thing. I settled on the GoToIfTime application. One strange thing about GoToIfTime is that it doesn't allow an else argument, so you'll need a sequence of if's to get things done. try something along these lines: [yourcontext] ;lunchtime exten = s,1,GoToIfTime(12:00-13:00?yourcontext|LUNCH|1) ;after work exten = s,2,GoToIfTime(17:00-23:59?yourcontext|CLOSED|1) ;before work exten = s,3,GoToIfTime(00:00-07:59?yourcontext|CLOSED|1) ;if we got this far, must be we're open exten = s,4,GoTo(yourcontext|OPEN|1) ;;Handle lunchtime calls exten = LUNCH,1,[do something] exten = CLOSED,1,[do soemthing else] exten = OPEN,1,[do yet another thing] Rene Nelson wrote: I would like to manipulate phone call direction to voicemail for lunch, after hours etc, but am unsure how to do this. Could someone point me to a howto or quickly explain the concept? Thanks Neri ___ --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] Re: Anyone aware of a current Dell server model with 3PCI slots
Matt wrote: We have a poweredge 2850 that we use for our VoIP server and it has 3 PCI slots. On 11/2/05, Tom Rymes [EMAIL PROTECTED] wrote: Looking for a Current Dell model, tower or 2U rackmount, that has (3) usable PCI slots? Was just cruising Dell.com and can't find a detailed spec on any of the server offerings that tells me the number of PCI slots available. Anyone using Dell for PBX builds can point me in the right direction? IIRC, the PowerEdge 2850 is used by Digium for ABE, and Signate for some of their installations, so I think it should work well. It is a 2U rack mount, and may be available as a tower as well. I think it has 3-4 PCI slots, but don't quote me. Tom ___ --Bandwidth and Colocation sponsored by Easynews.com -- I'm greeting to hear this. I have installed some Digium cards into this kind of servers. I get surprised when the slots pci gets shared IRQ with ethernet devices, raid controller or VGA card. Anybody knows how get unshare the IRQ of the slots pci ? (firmware, update, some special BIOS configuration,...) We answered Dell with no response. 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] Voicemail in Realtime mode
On Wed, 2005-11-02 at 16:56 +0100, Luca Lafranchi Lists wrote: Hi, I have installed the asterisk 1.2 beta version and I have created the voicemail table described on this page http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail, but when I start the asterisk server I receive the following error. Any idea ? We are going to need more information if you want help. What database are you using? Did you configure the res_(database).conf file correctly? Provide your extconfig.conf file. -- Carlos Chavez Director de Tecnologa Telecomunicaciones Abiertas de Mxico S.A. de C.V. Tel: +52-55-91169161 Ext 2001 signature.asc Description: This is a digitally signed message part ___ --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] Time based call direction
I would like to manipulate phone call direction to voicemail for lunch, after hours etc, but am unsure how to do this. Could someone point me to a howto or quickly explain the concept? I would recommend checking a database value over the time based GoToIfTime unless you are always go to and return from lunch at EXACTLY the same time: put a value like OutToLunch=1 in the asterisk database (see dbput) write an extension to make this either 1 or 0 (using dbput) add two lines to incoming calls to forward them to Vmail IF the flag is set (using dbget and gotoif) the only problem then becomes remembering to set the falg BACK to 0 so the phone rings again :) ___ --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] firmware update polycom 500 / dial problem
Hi, sorry - I know that problem is not directly related to asterisk but mabe someone can help anyway. After updating our polycom ip 500 sip phones from 2.6.1. to 2.6.2.0032 it is mostly not possible to dial numbers with leading zeros like 0018... If you do so you see on the diplay an number like that: 1800 an the cursor is on the first position. But if you dial the number (press the buttons) without lifting the handset everything is ok...strange Thank you for any help, morel -- - Morel Mosolff- Network-/System-Technician - NATIVE INSTRUMENTS GmbH - [EMAIL PROTECTED] - Schlesische Strasse 28 - http://www.native-instruments.de/ - D-10997 Berlin - Tel. +49-30-61 10 35-1712 - Germany - Fax +49-30-61 10 35-2712 ___ --Bandwidth and Colocation 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 bridge fax from pri to fxs
Andrew Kohlsmith wrote: On Wednesday 02 November 2005 11:17, Kevin Hanson wrote: I have a TE110P and a TDM10B. Via DID, I want to route calls to the fax number to the fxs port to which the fax machine will be connected. Do I need to set faxdetect=both in zapata.conf? I am assuming that Asterisk will bridge between the two cards and that the usual fax over IP won't be a factor. the 'faxdetect' parameter does one thing and one thing only: when a fax tone is heard, Asterisk will jump to the 'fax' extension in the channel's context. If you are using a specific DID for faxes you don't need that at all. exten = 1234567,1,Dial(Zap/g2,,g) exten = 1234567,2,Macro(handle-hangup) is how I'd do it. I'd define your PRI channels in group 1, and your FXS ports that the fax machine(s) are hooked up to in group 2. If you only want one fxs port for one fax machine, then say Dial(Zap/25) or whatever zapata port the fxs machine's on. Works like a charm for me, anyway. -A. ___ Did you have to set 'echocancel=no' or fiddle w/ any other echo related settings in zapata.conf for that channel? Cheers, Kevin ___ --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: Re: intel e7230 chipset (Kevin Hanson)
That would be great. Thank you. Robbie Hughes wrote: Does anyone know if the intel e7230 chipset in the new dell poweredge sc430 and poweredge 830 servers is compatible with the te110p and tdm400p cards? I know there were problems with previous generation dells, but I've read that these work fine. Can anyone confirm this? thanks robbie I have a PowerEdge 830 that I am getting ready to install at a customer site. I have been testing it in my lab w/ a TDM04B and it works fine. I won't be able to test w/ the TE110P until I get on site and have access to the PRI. We are installing this weekend. I am not expecting problems, but will reply back if I do. Cheers, Kevin ___ --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] Time based call direction
Rene Nelson wrote: I would like to manipulate phone call direction to voicemail for lunch, after hours etc, but am unsure how to do this. Could someone point me to a howto or quickly explain the concept? include = atlunchcontext|11:00-11:59|mon-fri|* include = notatlunchcontext|09:00-10:59|mon-fri|* include = notatlunchcontext|12:00-18:00|mon-fri|* include = afterhourscontext|18:01--8:59|mon-fri|* Use something like that. Kyle ___ --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] Double DTMF with tdm card
Did you ever find a solution for this problem? I have it on latest Beta 2 Bart - Original Message - From: Walt Reed [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, October 21, 2005 7:26 AM Subject: [Asterisk-Users] Double DTMF with tdm card I have a TDM22B (latest rev), Sipura 2000, and Cisco ATA 186. Running CVS HEAD from about a week ago. Calls made from a SIP device on either the cisco or sipura work fine. Call made from an analog phone hooked up to one of the FXS ports on the TDM22B sound fine, but any DTMF entered after the call is bridged to an outside number (like entering a PIN for a bank or external conference bridge) is frequently doubled. Entering 1234 may be recognized as 112344 for example. I ran fxotune, and played with the rx and tx gains a little, but have been unable to resolve the problem... * has no problem dialing outside numbers. It's just DTMf after the call is bridged between zap channels... Any ideas? ___ --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] Re: Anyone aware of a current Dell server model with 3PCI slots
We have a poweredge 2850 that we use for our VoIP server and it has 3 PCI slots. I'm greeting to hear this. I have installed some Digium cards into this kind of servers. I get surprised when the slots pci gets shared IRQ with ethernet devices, raid controller or VGA card. Anybody knows how get unshare the IRQ of the slots pci ? (firmware, update, some special BIOS configuration,...) We answered Dell with no response. I can't say that I've had this problem with the 2850 we have. I also can't take the server down to look at it right now, however we just got another digium card which I need to put in at some point over the next few days, so I'll be taking it down sometimes soon. As far as sharing, make sure you have disabled everything you don't need USB, SERIAL, PARALLEL, etc. You can then set the PCI IRQ in the BIOS, I believe. CPU0 CPU1 0: 584222710 584230408IO-APIC-edge timer 1: 0 7IO-APIC-edge keyboard 2: 0 0 XT-PIC cascade 8: 0 1IO-APIC-edge rtc 14: 0 2IO-APIC-edge ide0 38:67120198751310 IO-APIC-level megaraid 48: 318573642 37 IO-APIC-level eth0 77: 1014625786 2080170691 IO-APIC-level t1xxp ___ --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] Double DTMF sent on T1 to T1 Native Bridge
Bump - I'm stuck until I can find a solutions Please help - I'll try anything! Bart - Original Message - From: Bart Fisher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, November 01, 2005 5:37 PM Subject: Re: [Asterisk-Users] Double DTMF sent on T1 to T1 Native Bridge An update: If I dial with an internal single FXS phone or inbound to TDM400 (FXO) it works correctly. It appears that the Telco T1 is regenerating the DTMF as received at the same time the audio DTMF is past though the bridged connection. So, the effect is I hear two tones on Legacy PBX connection. - Make sense? This is a new problem since Asterisk 1.0.9, so I guess it's a bug? Seems there should be some way to make the Telco T1 stop listening and sending DTMF after connection Bart - Original Message - From: Bart Fisher [EMAIL PROTECTED] To: Asterisk-Users@lists.digium.com Sent: Tuesday, November 01, 2005 1:41 PM Subject: [Asterisk-Users] Double DTMF sent on T1 to T1 Native Bridge I have asterisk sitting in the middle with Telco on one side and Legacy PBX on the other using two T1 ports on a TE410P. I also have the latest Beta 2 installed. My problem is after a call is connected (port to port T1) and the outside user presses a touch tone, asterisk is repeating the digit. So if I press 1234 the PBX hears 11223344 - really messes up accessing the voice mail on PBX. If I dial into PBX from an internal phone it works correctly. This is a new problem since my upgrade from Asterisk 1.0.9, so I guess there is some keyword to disable this feature zapata? Bart ___ --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] Satellite WAN
On Wed, Nov 02, 2005 at 12:31:48PM -0500, Adam Moffett wrote: I have no experience in the matter whatsoever ;) But, I can say that long distance phone calls (non-voip) are sometimes carried over sattelite when fiber is not available. It must be possible for voip, but the latency and jitter would be tremendous and although I am not an expert on the matter, I would suggest that you would only be replacing one set of problems with a new set of problems. Using TDM delays are horrid (I remember calls to the US before TAT8 was installed and most calls went via satellite). Geostationary orbit 30K miles (approx), therefore up-leg plus down-leg is 60K miles. Light travels at 186K miles/s, so that's a 1/3 of a second delay in one direction (ignoring any delays through the satelitte to reduce interference etc), so that's 2/3s there and back. Steve -- NetTek Ltd Fax +44-(0)20 7483 2455 Skype / In stevekennedyuk / UK +442088167166 / US +13106518226 Vonage UK +442079932612 / US +13108577715 / UK mob 07775 755503 Personal Blog http://stevekennedy.blogspot.com Euro Tech News Blog http://eurotechnews.blogspot.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] firmware update polycom 500 / dial problem
Morel Mosolff wrote: Hi, sorry - I know that problem is not directly related to asterisk but mabe someone can help anyway. After updating our polycom ip 500 sip phones from 2.6.1. to 2.6.2.0032 it is mostly not possible to dial numbers with leading zeros like 0018... If you do so you see on the diplay an number like that: 1800 an the cursor is on the first position. But if you dial the number (press the buttons) without lifting the handset everything is ok...strange Thank you for any help, morel Check out the digitmap in sip.cfg for the phone. This is the default: dialplan.digitmap=[2-9]11|0T|011xxx.T|[0-1][2-9]x|[2-9]x|[2-9]xxxT You'll notice there is no match for 00. You can either modify this to include a pattern that starts with 00 or disable this completely by setting: dialplan dialplan.impossibleMatchHandling=3 If you do this you will always have to hit the 'send' key to send the digits to the server. More info is in the polycom admin guide. Cheers, Kevin ___ --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] feature.conf in 1.2beta2
Andrew Kohlsmith wrote: I've been thinking of a way to get across the idea that a native bridge was unsuccessful in more friendly terms for a bit but nothing really concise has come to mind. Some reason code might be handy... unable due to differing codecs, unable due to necessity to listen to audio... I dunno how to concisely convey that information. For now I have removed the message (I added it recently), since it isn't accomplishing what it was supposed to. ___ --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] A few Zaptel BRI questions...
I'm having some issues, and thought it wise to check with the list before putting in any more time Here we go: 1) Do Zaptel BRI (Cologne based cards) support DID routing (I believe they do, but the behavior of my (*) server is making me doubt, and I want to be sure before attempting any more permutations) 2) The (*) is parallel to my current Siemens Gigaset4135. Incoming calls on all MSNs show up on the display for a split second. I assume that means (*) steals them away before anybody would be able to answer. If all worked fine, that would not be an issue, but until then, I would prefer parallel rings, or at least for (*) to leave the main MSN alone, and only capture the others. Is this possible? 3) Are MSN's the same as DIDs for (*)? That's it for now, any help is appreciated... -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --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 bridge fax from pri to fxs
On Wednesday 02 November 2005 13:09, Kevin Hanson wrote: Did you have to set 'echocancel=no' or fiddle w/ any other echo related settings in zapata.conf for that channel? No; the echo canceller is automatically disabled upon reception of a 2100Hz tone (which is part of the start of all modem and fax communications). -A. ___ --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] Fax between Asterisk SIP clients
Hi all, I'm looking for a fax solution with Asterisk. I would like the users to be able to hook up regular fax machines to their SIP ATA's and send/receive fax from PSTN and/or other SIP clients. My goal is: fax machines - SIP ATA - Asterisk - T1(TE406E) - fax on PSTN It looks likeHylafax will allow me to receive fax from PSTN, but not send to PSTN. I also tried Spandsp, and it seems to receive fax ok from ATA's, but I can't figure out how to have it automatically forward the fax file to fax machines on PSTN or other SIP extensions. Can I have Spandsp dial and send the fax to the destination automatically? Are there other software / hardware solutions that can help me achieve my goal? Please advise. Thanks to any help/ideas. AK ___ --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] Time based call direction
include = atlunchcontext|11:00-11:59|mon-fri|* include = notatlunchcontext|09:00-10:59|mon-fri|* include = notatlunchcontext|12:00-18:00|mon-fri|* include = afterhourscontext|18:01--8:59|mon-fri|* I wasn't aware that include allowed a time qualifier. Does that mean that the specified context will only be included at the specified 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
Re: [Asterisk-Users] feature.conf in 1.2beta2
On Wednesday 02 November 2005 13:28, Kevin P. Fleming wrote: For now I have removed the message (I added it recently), since it isn't accomplishing what it was supposed to. No problem, but it would be very handy to see the bridge status through show channels type of output. a Bridge Type column... Normal, Native or N/A? -A. ___ --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] Satellite WAN
Sattellite links aren't cheap, and, the worst of all, you have in a idel condition, 1.4 seconds latency. Hope this help... Juan. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Adam Robins Enviado el: Miércoles, 02 de Noviembre de 2005 02:01 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [Asterisk-Users] Satellite WAN We have built an Asterisk network using an MPLS-based IP VPN. We have one location in New Brunswick Canada that consistently gives us major quality problems, whereas the others are flawless. Quality problems take the form of static, poor voice tonality, popping clicking, drops, sporadic echo, you name it. The latency of a QoS prioritized packet between the Canada site and our hub in Atlanta is 85ms (ping). I have been searching for an alternative network provider, but I'm told that they would all take the same route from the US into Canada, as there is simply no major backbone running into NB east of Toronto. So now I'm thinking about satellite. I have no idea if a) this would even be economically feasible, and b) if the latency would be any better. If anyone out there has had any such satellite network experience with VoIP, I like to hear from you. Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.12.7/156 - Release Date: 02/11/2005 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.12.7/156 - Release Date: 02/11/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] faster transcoding possible
According to http://www.extremetech.com/article2/0,1697,1880749,00.asp ATI is delivering a GPU enabled transcoding method that cuts video transcoding down to 1/5 the time it would take the cpu. This might also be applied to audio codecs in theory (I havent looked into it enough). Lets face it the video controller in an asterisk server is most likely going to be under utilizied. You can get a reasonable card for very little money. This might be an interesting approach to squeze more performance out of a system. This might make things slightly more interesting in the near future. Although I wonder how much better it would be with a high volume of calls. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --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] Zap Polarity Reversal
I am in the US, NYC using a TDM400 card. I never have never seen this issue until now. I see some code has been changed in this area recently. MARK. Rich Adamson wrote: Previously I would get two events on my Zap channel which indicated ringing and answered. Now I am getting polarity reversal events: Nov 2 07:01:25 NOTICE[18246]: chan_zap.c:6014 ss_thread: Got event 17 (Polarity Reversal)... Nov 2 07:01:28 NOTICE[18246]: chan_zap.c:6014 ss_thread: Got event 17 (Polarity Reversal)... I am using CVS Head from: Asterisk CVS HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 2005-11-02 05:13:32 UTC ___ Sounds like a new feature. Have you tried reversing the polarity or putting a butt set that indicates pol on the line? I don't know that it makes a difference but you might as well correct it if it is reversed. That actually sounds more like whatever telco he's connecting to is providing answer supervision in the form of polarity reversal. Without knowing more about which country / telco, there is no way to tell. Note the polarity reversal is happening _after_ asterisk gets a call, therefore not likely to have anything to do with reversed tip/ring. That same message does not occur in the US with the analog TDM card, so not sure what the OP has or is connected to. ___ --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] OS for ABE
Title: OS for ABE We are setting up ABE for a client of ours. This is not our first Asterisk install, far from it, but it is our first time using ABE. Here is the problem, ABE only supports Fedora 3 and Red Hat EL3, we typically use CentOS. Our problem with this scenario is that RHEL3 is an old release, we would rather use 4 if we have to, and we have not had good experiences with Fedora. We tried to use Fedora but we are running into some problems with our SCSI card. What distro do most people use with ABE? What happens if we use CentOS? Will it render our ABE purchase useless? - Eric Alexander Senior IT Systems Consultant The Uptime Group, Inc. (303) 757-4611, Ext. 402 - ___ --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] Voicemail in Realtime mode
On Wed, 2005-11-02 at 16:56 +0100, Luca Lafranchi Lists wrote: Hi, I have installed the asterisk 1.2 beta version and I have created the voicemail table described on this page http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail, but when I start the asterisk server I receive the following error. Any idea ? We are going to need more information if you want help. What database are you using? Did you configure the res_(database).conf file correctly? Provide your extconfig.conf file. Yes, the res_mysql.conf it's configured correctly because the sip_buddies in realtime works fine. This is my extconfig.conf ; ; Static and realtime external configuration ; engine configuration ; ; Please read doc/README.extconfig for basic table ; formatting information. ; [settings] ;static extensions.conf = mysql,pbx,PBX_extensions_conf queues.conf = mysql,pbx,PBX_queues_conf ;realtime sipusers = mysql,pbx,PBX_sip_buddies sippeers = mysql,pbx,PBX_sip_buddies voicemail = mysql,pbx,PBX_voicemail this is the voicemail table for realtime in pbx db CREATE TABLE `PBX_voicemail` ( `uniqueid` int(11) NOT NULL auto_increment, `customer_id` int(11) NOT NULL default '0', `context` varchar(50) NOT NULL default '', `mailbox` int(5) NOT NULL default '0', `password` varchar(4) NOT NULL default '0', `fullname` varchar(50) NOT NULL default '', `email` varchar(50) NOT NULL default '', `pager` varchar(50) NOT NULL default '', `stamp` timestamp NOT NULL default CURRENT_TIMESTAMP on update CURRENT_TIMESTAMP, PRIMARY KEY (`mailbox`), KEY `mailbox_context` (`mailbox`,`context`,`uniqueid`) ) ENGINE=MyISAM DEFAULT CHARSET=latin1; -- Carlos Chavez Director de TecnologÃa Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --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] faster transcoding possible
On Wednesday 02 November 2005 14:11, trixter aka Bret McDanel wrote: According to http://www.extremetech.com/article2/0,1697,1880749,00.asp ATI is delivering a GPU enabled transcoding method that cuts video transcoding down to 1/5 the time it would take the cpu. This might also be applied to audio codecs in theory (I havent looked into it enough). This has come up several times over the years. YES a GPU might be able to take some CPU load off but you now add latency because you're shipping data to and from main memory to the GPU and back. It's also been stated that AGP transfers are optimized for memory to the video card and not the other way around, so you may add more latency than you expect. Using the GPU for video codec work makes sense because once it's off on the video card it ain't coming back. This is most certainly not the case with audio. :-) Nobody can really truly say until there are some benchmarks run, and nobody's stopping anyone from exerting the effort. It just takes someone curious enough to acutally go do it. -A. ___ --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] Satellite WAN
On Wed, 2 Nov 2005, Juan Janczuk wrote: Sattellite links aren't cheap, and, the worst of all, you have in a idel condition, 1.4 seconds latency. I know you can get less, our client in the mid-west uses Hughes with under 600ms. But never attempted to do VOIP over it. -- -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- - - - Jason Pyeron PD Inc. http://www.pdinc.us - - Partner Sr. Manager 7 West 24th Street #100 - - +1 (443) 921-0381 Baltimore, Maryland 21218 - - - -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- This message is for the designated recipient only and may contain privileged, proprietary, or otherwise private information. If you have received it in error, purge the message from your system and notify the sender immediately. Any other use of the email by you is prohibited. ___ --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] Possible Issue With Meetme Conferencing in 1.2.0b2 and latest CVS HEAD (02/11/2005)
I'm running Asterisk 1.2.0b2 (also tried latest CVS HEAD) in my lab and i've come across a strange problem. I've setup an extension to call the meetme application, when i call that extension it functions as expected, informing me of my conference number and that i'm the only one in the conference however right after join the conference some problems start occuring: 1. If i call in with another client (both are SIP based), it does not acknowledge the DTMF tones i send to select the conference room, it acts like it never received the DTMF (it plays the please enter the conference number followed by the pound key prompt again) I have verified that the tones are being sent properly, and otherwise work as expected. (before selecting a conference room) 2. When i hang up the phone Asterisk does not clear the SIP channel in use by that phone. Before selecting a conference room calls are properly disconnected by Asterisk and removed from the sip show channels list. 3. After the RTP timeout hits (as configured in sip.conf) it prints a message every second that the call has timed out and will be disconnected. This continues on forever it seems (12 hours in one case) Before selecting a conference room, if left idle (no RTP is sent from SIP UAC), the SIP session is properly disconnected/terminated after the RTP idle timer hits. if add the de options (dynamic, select an empty conference room) the first caller hears the meetme prompts and is put into the first conference room, however the second caller hears nothing, looking at the debug output on asterisk shows that meetme was called and nothing else after that I'm running on linux kernel 2.6.13.4 (vanilla, with grsecurity patches) Zaptel drivers were compiled with make linux26 There is a T100P card in the system and the zaptel and wct1xxp modules are loaded I've tried using the ztdummy module in place of wct1xxp with the same results Asterisk and Zaptel were compiled with gcc 3.3.5 on Debian Sarge submitted bug - http://bugs.digium.com/view.php?id=5578 tavis ___ --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] OS for ABE
Weuse Fedora 3 and ABE-A.1 The pair has been workinggreat for usso far. AK On 11/2/05, Eric Alexander [EMAIL PROTECTED] wrote: We are setting up ABE for a client of ours. This is not our first Asterisk install, far from it, but it is our first time using ABE. Here is the problem, ABE only supports Fedora 3 and Red Hat EL3, we typically use CentOS. Our problem with this scenario is that RHEL3 is an old release, we would rather use 4 if we have to, and we have not had good experiences with Fedora. We tried to use Fedora but we are running into some problems with our SCSI card. What distro do most people use with ABE? What happens if we use CentOS? Will it render our ABE purchase useless? -Eric AlexanderSenior IT Systems ConsultantThe Uptime Group, Inc.(303) 757-4611, Ext. 402- ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://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.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] Time based call direction
Rene Nelson wrote: I would like to manipulate phone call direction to voicemail for lunch, after hours etc, but am unsure how to do this. Could someone point me to a howto or quickly explain the concept? Thanks Neri Hi Neri, The command GotoIfTime() if your answer here. See http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime for more info. Now, assuming we are talking about a situation with say one main voicemail extension to collect messages from callers calling the main company number The call comes inthen do a gotoiftime to branch to two places: first place is normal, second place is lunchtime. Now, for each of these, first play an appropriate message with the Playback command, then record the message left using the voicemail command with the s option. The s option means play nothing, so basically you aren't using the built-in outgoing messages that the voicemail system has and instead will have first used some custom message via the playback function e.g. exten = 4321,111,Playback(lunchtime) exten = 4321,112,voicemail,s12345 where 12345 is your main voicemail box. 4321 and 111/112 are also just numbers picked at random for use in this example. See http://www.voip-info.org/wiki-Asterisk+cmd+Voicemail for more info on using voicemail in this way. Faris. ___ --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] Satellite WAN
Price is high that is correct but latency is not correct. We have a number of Satellite VoIP Trunks in Africa and no location has more then 500 ms latency. In all locations we have 2 Mbit dedicated lines using C-band and the hub is in the US. But price is HIGH. 6000 usd per month Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan Janczuk Sent: den 2 november 2005 20:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Satellite WAN Sattellite links aren't cheap, and, the worst of all, you have in a idel condition, 1.4 seconds latency. Hope this help... Juan. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Adam Robins Enviado el: Miércoles, 02 de Noviembre de 2005 02:01 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [Asterisk-Users] Satellite WAN We have built an Asterisk network using an MPLS-based IP VPN. We have one location in New Brunswick Canada that consistently gives us major quality problems, whereas the others are flawless. Quality problems take the form of static, poor voice tonality, popping clicking, drops, sporadic echo, you name it. The latency of a QoS prioritized packet between the Canada site and our hub in Atlanta is 85ms (ping). I have been searching for an alternative network provider, but I'm told that they would all take the same route from the US into Canada, as there is simply no major backbone running into NB east of Toronto. So now I'm thinking about satellite. I have no idea if a) this would even be economically feasible, and b) if the latency would be any better. If anyone out there has had any such satellite network experience with VoIP, I like to hear from you. Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.12.7/156 - Release Date: 02/11/2005 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.12.7/156 - Release Date: 02/11/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OS for ABE
Your ABE purchase comes with Digium support for Installation. You should call them for the answers to your questions. On 11/2/05, Eric Alexander [EMAIL PROTECTED] wrote: We are setting up ABE for a client of ours. This is not our first Asterisk install, far from it, but it is our first time using ABE. Here is the problem, ABE only supports Fedora 3 and Red Hat EL3, we typically use CentOS. Our problem with this scenario is that RHEL3 is an old release, we would rather use 4 if we have to, and we have not had good experiences with Fedora. We tried to use Fedora but we are running into some problems with our SCSI card. What distro do most people use with ABE? What happens if we use CentOS? Will it render our ABE purchase useless? - Eric Alexander Senior IT Systems Consultant The Uptime Group, Inc. (303) 757-4611, Ext. 402 - ___ --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] Voicemail in Realtime mode
-Original Message- From: Carlos Chavez [mailto:[EMAIL PROTECTED] Sent: mercoledì, 2. novembre 2005 19:04 To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Voicemail in Realtime mode On Wed, 2005-11-02 at 16:56 +0100, Luca Lafranchi Lists wrote: Hi, I have installed the asterisk 1.2 beta version and I have created the voicemail table described on this page HYPERLINK http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemailhttp://www.v oip-info.org/wiki/view/Asterisk+RealTime+Voicemail, but when I start the asterisk server I receive the following error. Any idea ? We are going to need more information if you want help. What database are you using? Did you configure the res_(database).conf file correctly? Provide your extconfig.conf file. Yes, the res_mysql.conf it's configured correctly because the sip_buddies in realtime works fine. This is my extconfig.conf ; ; Static and realtime external configuration ; engine configuration ; ; Please read doc/README.extconfig for basic table ; formatting information. ; [settings] ;static extensions.conf = mysql,pbx,PBX_extensions_conf queues.conf = mysql,pbx,PBX_queues_conf ;realtime sipusers = mysql,pbx,PBX_sip_buddies sippeers = mysql,pbx,PBX_sip_buddies voicemail = mysql,pbx,PBX_voicemail this is the voicemail table for realtime in pbx db CREATE TABLE `PBX_voicemail` ( `uniqueid` int(11) NOT NULL auto_increment, `customer_id` int(11) NOT NULL default '0', `context` varchar(50) NOT NULL default '', `mailbox` int(5) NOT NULL default '0', `password` varchar(4) NOT NULL default '0', `fullname` varchar(50) NOT NULL default '', `email` varchar(50) NOT NULL default '', `pager` varchar(50) NOT NULL default '', `stamp` timestamp NOT NULL default CURRENT_TIMESTAMP on update CURRENT_TIMESTAMP, PRIMARY KEY (`mailbox`), KEY `mailbox_context` (`mailbox`,`context`,`uniqueid`) ) ENGINE=MyISAM DEFAULT CHARSET=latin1; -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.12.7/157 - Release Date: 02.11.2005 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.12.7/157 - Release Date: 02.11.2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fritz!Card PCI ver2.0
Stephen Arulraj wrote: Anyone knows how I can use this ISDN card for asterisk as a BRI trunk interface? Thanks, Stephen Hi Stephen, Is this a new version of the AVM card? If not (or even if it is), you may find the following pages helpful: http://www.voip-info.org/wiki/index.php?page=Asterisk+AVM+Fritz+CAPI+Driver+Install http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI Please note, however, that somewhere in the wiki it suggests that you modify the AVM driver code slightly. I found this stopped it compiling, and that simply leaving the code as it is worked fine. Faris. ___ --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
