Re: [Asterisk-Users] Voip-Info
Douglas Garstang schrieb: Is it just me or is the voip-info web site down right now? I was experiencing problems accessing voip-info, too. But i guess the problems derived from accessing http://www.google-analytics.com, because i could see that voip-info was resolved rather quickly and after that stopped. Lynx seems to have to problems getting the html pages from voip-info :) Today everything seems to be back to normal. Greetings, Tobias ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Asterisk to receive fax
Hi Gidean, look at http://soft-switch.org/ and http://www.voip-info.org/wiki/view/Asterisk+fax . Bye Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Gidean Chan Gesendet: Mittwoch, 15. März 2006 08:32 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] Asterisk to receive fax Can anyone tell me how to configure my system so that fax can be received and forward to email account? Thanks Gidean Chan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: digium.com redesign
Benny Amorsen schrieb: MG == Michael George [EMAIL PROTECTED] writes: MG I may be way behind here, but I see that digium redesigned their MG site. I cannot find the mailing list search screen. MG I have found the mailman list page, but that doesn't have have a MG nice search ability. gmane.org has this mailing list as gmane.comp.telephony.pbx.asterisk.user. That's how I read it. Have taken a look at it and it looks really nice :) But because of the issue of the Original Poster I looked at http://lists.digium.com/pipermail/asterisk-users/ and saw that Asterisk Users are indead ahead of time :) ArchiveView by:Downloadable version May 2016: [ Thread ] [ Subject ] [ Author ] [ Date ] November 2007: [ Thread ] [ Subject ] [ Author ] [ Date ] So long, Tobias ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: digium.com redesign
Tobias Wolf wrote: Have taken a look at it and it looks really nice :) But because of the issue of the Original Poster I looked at http://lists.digium.com/pipermail/asterisk-users/ and saw that Asterisk Users are indead ahead of time :) Archive View by: Downloadable version May 2016: [ Thread ] [ Subject ] [ Author ] [ Date ] November 2007: [ Thread ] [ Subject ] [ Author ] [ Date ] Its most likely because people have posted to the list with dates set in advance, and the local list-archiver uses the dates in the e-mail header and not its local time for processing. You'll find a few random e-mails in those archives - but its not digium's issue So long, Tobias ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Adrian Carter Technical Manager Leading Edge Internet Web http://www.lei.net.au http://support.lei.net.au Direct+61 2 6163 6162 Support 1 300 662 415 E-mail[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Native Sounds - in case you missed it...
On Tue, Mar 14, 2006 at 11:26:14PM -0800, Ira wrote: At 08:51 PM 03/14/2006, you wrote: In my humble opinion, EVERYONE (unless you have your own in a different voice/language) that uses Asterisk should be using these prompts. How about a direct link this time: For what it's worth, the hardest problem I had was not being able to directly FTP them from my Asterisk box. I'm a Linux newbie and had no idea how to do that. FTP *to a linux box*? /me is shocked! You have ssh access, right? Use scp/sftp. Try http://winscp.sf.net/ . If you don't one to carry one around or install on your system(s), put one statically-linked copy on your file/web server and download/run it. I downloaded them to my Windows box, set up vsftpd and uploaded them using a GUI FTP client in Windows and only then could I use them. wget http://server.name/path/to/file wget ftp://server.name/path/to/file In fact, what I normally do is copy a link from my browser to the command line in the terminal window and download it with wget. Saves me an extra file copy around the net. So for those who need exact commands, here's a two-liner: wget http://mirror.astlinux.org/sounds/asterisk-native-sounds-20060209-01-sln.tar.bz2 tar xjf asterisk-native-sounds-20060209-01-sln.tar.bz2 -C /var/lib/asterisk ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] EICON Diva 4BRI
Hello, Step by Step instructions on installing the card with Asterisk can be found here: http://www.eicon.com/support/helpweb/slnxen/asterisk.asp Let me know if this helps David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy Sent: 14 March 2006 18:34 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] EICON Diva 4BRI Are there any step by step instrunctions on how to install drivers and I guess bristuff for this card? Just need to use it to handle voice on 2 BRI circuits (UK) then utilise with Asterisk and some Digium cards handling POTS phones (and some VoIP out the back). It's the EICON card stuff and how to make it all work I'm finding confusing? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IVR weirdness
A user of mine has discovered that when you call into asterisk and get the IVR menu with options 1-5 available, if you dial 1 then immediatly dial 2 it will connect you to 2 and not 1. I expect this is due to the digit timeouts and response timeout. Is there a way to force an immediate action based on the first menu option selected? -- Robert P. McKenzie, CSTA, MBCS | GammaRay Technical Services Ltd [EMAIL PROTECTED] | [EMAIL PROTECTED] http://www.uk-experience.com | http://www.gammaray-tech.com Fancy some fun? http://www.thewetwilly.com Ecademy Profile: http://www.ecademy.com/user/robertmckenzie ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Jitter Buffer for 1.2.5
14 mar 2006 kl. 15.38 skrev Matt: The jitterbuffer branch is based on svn trunk (the same as the old CVS HEAD) The jitterbuffer-1.2 branch is based on the 1.2 branch HEAD (meaning latest 1.2 version code). Ok... so if I pull the 'jitterbuffer' branch I would get 'svn trunk code'. But if I pull 'jitterbuffer-1.2' I get the same code as I would have if I downloaded 1.2.5 and then applied a jitterbuffer patch (which I know, does not exist for 1.2.5). No, but you would get 1.2.5 + jitterbuffer patch + any changes to the 1.2 branch after we released 1.2.5. /Olle --- * Olle E Johansson - [EMAIL PROTECTED] * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] isdn out of band signalling
On Wed, Mar 15, 2006 at 02:01:50PM +1100, James Harper wrote: This is more an isdn question than an asterisk specific one, but is there any end to end signalling channel available during call setup? Eg if AParty dials BParty, can any information be conveyed (in both directions preferably, and in addition to CLI[PR]) before the call is answered? There is not such thing like A to B signaling, your phone/pbx is sending signaling messages to your provider pstn switch not to BParty. The only way I can think of doing this is for the AParty to use CLIP to present a different number to the BParty, and for the BParty to terminate the call while ringing after a certain time (the time taken to terminate forms the information from B to A). On some E1/T1 , you can spoof callerid , everything depends on your provider. /pch -- Dyslexia bug unpatched since 1977 ... exploit has been leaked to the underground. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk to receive fax
Thanks! - Original Message - From: Philipp Dreimann [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, March 15, 2006 4:18 PM Subject: AW: [Asterisk-Users] Asterisk to receive fax Hi Gidean, look at http://soft-switch.org/ and http://www.voip-info.org/wiki/view/Asterisk+fax . Bye Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Gidean Chan Gesendet: Mittwoch, 15. März 2006 08:32 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] Asterisk to receive fax Can anyone tell me how to configure my system so that fax can be received and forward to email account? Thanks Gidean Chan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Jitter Buffer for 1.2.5
14 mar 2006 kl. 19.00 skrev Robert Webb: On Tue, 14 Mar 2006 14:32:02 +0100 Olle E Johansson [EMAIL PROTECTED] wrote: 14 mar 2006 kl. 13.35 skrev Matt: Right saw that. But I'm trying to get away from using CVS-HEAD :) We all are. Every developer have switched from CVS to Subversion :-) This is not the development branch, but the release branch code, which we use to create the 1.2.x releases. The jitterbuffer itself is *not* release branch code, it's very much development. Please test it. The jitterbuffer branch is based on svn trunk (the same as the old CVS HEAD) The jitterbuffer-1.2 branch is based on the 1.2 branch HEAD (meaning latest 1.2 version code). /O Olle, Pardon this dumb question please, but where are these test located. I looked under http://svn.digium.com and do not see them. I am not fluent in where everything is located and would like to do some testing on some of the other items such as the sip jitterbuffer. It will only be minimal but I would like to help where I can. For viewing: http://svn.digium.com/view/asterisk/team/oej - then pick a branch For checking out svn checkout http://svn.digium.com/svn/asterisk/team/oej/name name Use jitterbufferfor svn trunk + jitterbuffer jitterbuffer-1.2for 1.2 + jitterbuffer test-this-branchfor the test branch with a lot of cool stuff including the jitterbuffer Thanks for testing! /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] - Asterisk Developer * Asterisk Training http://edvina.net/training/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to show called name on calling polycom display
Hi, we have an asterisk 1.2.1 box and 2 polycom SIP phones. We'd like to show the called name on the calling polycom display instead of his /her extensions as I do with the caller name on the called polycom. Is it possible? If yes, how? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX choppy sound
Hi, Are you using Trunked IAX? Currently we do not use trunking. How many calls at a time? All the test we have performed so far were with only one active call. What codecs are you using? We have set the bandwith=low, so I think that G.723.1, GSM, and LPC10 are in the play. What is the ping time between the systems? Ping stats are: Server 1: 50 packets transmitted, 50 received, 0% packet loss, time 49491ms rtt min/avg/max/mdev = 190.198/213.028/283.275/25.307 ms Server 2: 50 packets transmitted, 49 received, 2% packet loss, time 49523ms rtt min/avg/max/mdev = 190.089/214.855/544.880/59.052 ms Any error messages ? There are no error messages in the console. Regards, Stojan Sljivic -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton Sent: Tuesday, March 14, 2006 19:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX choppy sound On 14 Mar 2006, at 16:44, Stojan Sljivic - GDS wrote: Hi, We have two Asterisk servers connected over IAX, with very limited bandwidth 256Kbs. When we make calls between these two Asterisk servers the sound is very choppy, no matter whether we use jitter buffer or not. However, when we make calls using Skype, the sound is perfect. Can anyone help us troubleshoot this IAX issue that we are experiencing? Maybe if you tell us some more :-) What codecs are you using? Are you using Trunked IAX? How many calls at a time? What is the ping time between the systems? Any error messages ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to assign a specific PSTN line to a specific extension ???
Hello all! I want to assign one of the PSTN lines to a specific extension only.Expecting an earlier response. Thanks a lot.Faisal Yahoo! Travel Find great deals to the top 10 hottest destinations!___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MCC v.1.3 Released
MCC - Billing solution for Asterisk PBX Current version: 1.3 + 1.3.1 Patch MCC is a web-based, user (and admin) friendly billing interface for Asterisk and VOIP. MCC is open source software licensed under the GPL Some features of MCC: Unlimited SIP, IAX and Mobile/PSTN devices assigned to user Unlimited tariffs with different rates Rate Table viewable in Currency of choice Profit counting!!! Stats by countries Blocking of users Show Balance, Expenditure, Payments and number of Calls on each account Call Data Records (also in CSV/PDF) Advanced customer management and portal management Integrated PayPal and Hanza.net commerce modules View and Store Customers payments Manage Pre Paid and Post Paid customers. Full Credit control by User Account Concurrent calls for every user MCC Requirements: Asterisk PostgreSQL Apache + PHP Homepage: http://www.paskambink.lt/mcc ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel compile errors on x86_64
Hi, Just downloaded the latest cvs from zaptel on my sparking new Athlon64 Centos4.2 system, but hitting a stumbling block... (sorry for the long post) #make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file ./makefw pciradio.rbt radfw radfw.h Loaded 42096 bytes from file cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o libtonezone.a -lm cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o zttool.o zttool.c cc -o zttool zttool.o -lnewt cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\zttest.c -o zttest cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm /lib/modules/2.6.9-34.EL/build make -C /lib/modules/2.6.9-34.EL/build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-x86_64' CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c:372: error: syntax error before zone_lock /usr/src/zaptel/zaptel.c:372: warning: type defaults to `int' in declaration of `zone_lock' /usr/src/zaptel/zaptel.c:372: error: incompatible types in initialization /usr/src/zaptel/zaptel.c:372: error: initializer element is not constant /usr/src/zaptel/zaptel.c:372: warning: data definition has no type or storage class /usr/src/zaptel/zaptel.c:373: error: syntax error before chan_lock /usr/src/zaptel/zaptel.c:373: warning: type defaults to `int' in declaration of `chan_lock' /usr/src/zaptel/zaptel.c:373: error: incompatible types in initialization /usr/src/zaptel/zaptel.c:373: error: initializer element is not constant /usr/src/zaptel/zaptel.c:373: warning: data definition has no type or storage class /usr/src/zaptel/zaptel.c:176: warning: 'fcstab' defined but not used make[2]: *** [/usr/src/zaptel/zaptel.o] Error 1 make[1]: *** [_module_/usr/src/zaptel] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.9-34.EL-x86_64' make: *** [linux26] Error 2 [EMAIL PROTECTED] zaptel]# uname -a Linux asterisk64 2.6.9-34.EL #1 Thu Mar 9 06:03:30 GMT 2006 x86_64 x86_64 x86_64 GNU/Linux Any ideas what I can do to fix this, or what I am doing wrong? Assistance appreciated. Thanks Warmest Regards, Walter. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco phones and Linksys SRW224P
I'm having problem powering Cisco phone's (7940 and 7905) on Linksys SRW224P switch (with PoE functionality). I have tested three phone's, one is working (7905) and two aren't (7905 and 7940). I have plugged all three phones on same switch port with same cable! Do I need to change anything in phone configuration? Is there something wrong with Linksys switch? How can I troubleshoot this? Please help, I'm going insane! -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Cisco phones and Linksys SRW224P
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I'm having problem powering Cisco phone's (7940 and 7905) on Linksys SRW224P switch (with PoE functionality). I have tested three phone's, one is working (7905) and two aren't (7905 and 7940). I have plugged all three phones on same switch port with same cable! Do I need to change anything in phone configuration? Is there something wrong with Linksys switch? How can I troubleshoot this? Please help, I'm going insane! One more thing. Cisco 7905 phone that is working is 74-3092-04 Rev.F0. Cisco 7905 phone that is not working is 74-3092-08 Rev.A0. Anybody know about any hardware issue with this revisions? -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk perl commands
Hi, Im using frequently the perl api within asterisk. Now Im looking for documentation for the perl commands. Some perl commands I found on this URL: http://www.voip-info.org/wiki/view/Asterisk+PHP Does anybody got more documentation or where I can found some more documentation about perl commands Kind Regards. Arjan Kroon Mobillion B.V. Copernicuslaan 30 Postbus 554 / PO Box 554 6710 BN Ede tel: +31 (0)318-648920 fax: +31 (0)318-648839 mobile: +31 (0)6-55871460 email: [EMAIL PROTECTED] internet: www.mobillion.nl ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel compile errors on x86_64
On Wed, 2006-03-15 at 17:49 +0800, Walter Klomp wrote: Hi, Just downloaded the latest cvs from zaptel on my sparking new Athlon64 Centos4.2 system, but hitting a stumbling block... (sorry for the long post) Kernel source installed? -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] There is lacking behind in recorded calls via sox
Hi ,
I have been using, Monitor (wav, ${CALLFILENAME}) to records calls from extensions.conf and soxmix software to compiles calls. The problem I am facing that for long calls more that 2 minutes there is disturbance in sequence of calls, calls from both ends are not in sequence and there is always lack behind from one end, I have tried a lot by checking different version of Sox
but still I am facing same issue .Can any one of you will let me know the reason of this lacking in calls form end after compilation while calls conversation goes fine in live calls .
Thanks,
Mazhar
Nettechltd.com
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[Asterisk-Users] there is lack behind in recoded calls via sox
Hi ,
I have been using, Monitor (wav, ${CALLFILENAME}) to records calls from extensions.conf and soxmix software to compiles calls. The problem I am facing that for long calls more that 2 minutes there is disturbance in sequence of calls, calls from both ends are not in sequence and there is always lack behind from one end, I have tried a lot by checking different version of Sox
but still I am facing same issue .Can any one of you will let me know the reason of this lacking in calls form end after compilation while calls conversation goes fine in live calls .Also I am using Asterisk
1.2.5 version
Thanks,
Mazhar
Nettechltd.com
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[Asterisk-Users] spa 3000/2100 noise
I've a problem. I've some spa3000 and spa2100. Asterisk 1.2.4. Prefered codec g711u in both. Calleng from a fxs of spa2100 to the fxo of spa3000, all works ok. Then I call from a sip phone configured for using g729, to the fxo of spa3000, it also works ok. The problem is that after this, when, making again a new call from spa2100 to spa3000, spa2100 receives only white noise. I suspect a codec mismatch. The problem disappears by powering off and on the spa3000. ¿Any ideas on how to check? -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk to receive fax
2006/3/15, Gidean Chan [EMAIL PROTECTED]: Can anyone tell me how to configure my system so that fax can be received and forward to email account? You can install iaxfax. It acts as a software modem that connects to asterisk as a iax phone. It creates a device that can be accesed as a faxmodem. Then, you can use hylafax that is very powerfull and can be configured to forward faxes to email, convert it to pdf, etc. etc (read the documentation). -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stuck. Extenions.conf? Realtime? MySQL? Grrrrr!
On 3/15/2006, Douglas Garstang [EMAIL PROTECTED] wrote: Boy, am I stuck... [snip] My brain hurts. Doug, Whenever I have gotten to this point in a project, I use two rules for handling the situation. Rule 1. Booze Rule 2. Throw money at it. Rule 1 makes me feel better. Rule 2 takes care of the problem but... If the boss isn't happy - fall back to Rule 1. The hardest target to hit in the programming shooting gallery is the moving one. Unless you 'sold' the powers that be that Asterisk is the answer to all questions... then you made your bed... but as I remember, I think you got 'stuck' with this one. You can probably (but I doubt it) buy a system that will do all this for you. Probably not out of the box tho and probably not without a large 'programmers' bill to boot. And several third-party packages. So grab your favorite alcoholic beverage, nail down what they want, and start solving the problems. Even if it takes a year - it will be better and cheaper than anything they can purchase. Brett ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Cisco phones and Linksys SRW224P
One more thing. Cisco 7905 phone that is working is 74-3092-04 Rev.F0. Cisco 7905 phone that is not working is 74-3092-08 Rev.A0. Anybody know about any hardware issue with this revisions? Nothing for sure, and you may already know this, but some early Cisco phones only knew how to speak Cisco PoE, not the 802 standard which was defined a bit later. The Cisco web site should tell you which phone talks which protocol though. James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (unexplicable) peaks of machine load
I have strange peaks of machine load on my asterisk servers, looking at top the load is very high even if cpu usage is low and no swap memory is used. This happens on all the machines, some of them have asterisk, mysql, agi and digium cards on them, so I thought I was only asking too much, but yesterday I noticed the same behaviour on an asterisk machine with only two digium in it, no other service and a two line extension. I thought it can be a problem with digium cards but the interrupts aren't shared, and I have the same problem on a pure-voip server. Asterisk version varies from 1.2.1 to 1.2.5, the kernels are 2.4 or 2.6 (right ones for the installed cpu, not generic 386) The only things in common are : Linux debian, iax channels are used, with jitterbuffer When this ghost load becomes too high ( 3) asterisk starts losing packets, and the users starts losing patience ... Anyone experiencing a similar problem ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (unexplicable) peaks of machine load
I've noticed this as well from pre 1.0 versions through to 1.2.5 across 12 separate Asterisk servers. The severity seems to be random mostly. I still haven't figured out what is causing it. MATT--- On 3/15/06, Simone Cittadini [EMAIL PROTECTED] wrote: I have strange peaks of machine load on my asterisk servers, looking at top the load is very high even if cpu usage is low and no swap memory is used. This happens on all the machines, some of them have asterisk, mysql, agi and digium cards on them, so I thought I was only asking too much, but yesterday I noticed the same behaviour on an asterisk machine with only two digium in it, no other service and a two line extension. I thought it can be a problem with digium cards but the interrupts aren't shared, and I have the same problem on a pure-voip server. Asterisk version varies from 1.2.1 to 1.2.5, the kernels are 2.4 or 2.6 (right ones for the installed cpu, not generic 386) The only things in common are : Linux debian, iax channels are used, with jitterbuffer When this ghost load becomes too high ( 3) asterisk starts losing packets, and the users starts losing patience ... Anyone experiencing a similar problem ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Development news :: T38 passthrough support
The SPA-2100 is the only one to support T.38 at the moment though. SPA-2002 has the ability to support t.38 (i.e. it has the processing power required) but the firmware support isn't there yet. C F wrote: On 3/11/06, Kristian Kielhofner [EMAIL PROTECTED] wrote: Olle E Johansson wrote: Friends in the Asterisk.org community, There is a lot of cool stuff going on in Asterisk development, things that will change Asterisk and make it work better in your organisation, make it easier to sell in your area or give you more consulting oppurtunities - in short, functionality that will make a lot of sense for you users. However, developers can't really get anywhere without a dialog with the users. You know what you need, you know what is missing and how you would like to make Asterisk a better choice. I am planning to send out a description of new features now and then, to inform you about what is going on, but also to get some feedback. The bug tracker is not only a tool for developers, but also for testers and users to react to changes and contribute. *** ITU T.38 -- Fax over VoIP Olle, Let's say that I wanted to setup a complete environment to test this. I presume that I would need the following: Fax machine T.38 compliant ATA (Sipura claims this) Asterisk server T.38 compliant something - does this need to be a Cisco 5300 (or similar)? Can it be just another plain ATA and fax machine? Another ATA like the SPA line should work on the second end as well. Please suggest some possible hardware! Thanks! -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CALL FOR COMMENTS - Dialplan
Hello Asterisk community, I have written a document that covers an Asterisk implementation I am building. I want to place it on the lists so USERS can view and make comments on, the ideas contained within. I think it is an important issue to develop a standardised Dialplan for applications, not just for Asterisk, but for pbx systems in general. As they become cheaper and more common place, each install has its own ideas of how to implement features. This makes it very hard for users to move from one system to another. In any case, If you have time, please do review the document and make comments to the list or to me directly. The document can be found at http://www.crafted.com.au/comments I cannot post it directly to the list as its TOO BIG. Thanks, James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dipura 2002 auto dial or intercom
This called hot line or batphone (as it's like the phone the commissioner used to have in Batman that went straight through to Bruce Wayne). Set the dialplan to this: (S0:#) where is the number/SIP address you want to dial. Note, that's a zero after the S. Anton Krall wrote: Guys. Anybody using sipuras 2002 knows if there is a way to make the phones connected to it to autodial an extension when the phone is picked up? For example, if the phone is on a police booth (building entrance) and you want the guys to just pick up the phone and make the phone auto dial the receptionist extension without the guys having to dial anything (ala batphone). Is this possible with spa's? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] isdn out of band signalling
On Wed, 2006-03-15 at 14:01 +1100, James Harper wrote: This is more an isdn question than an asterisk specific one, but is there any end to end signalling channel available during call setup? Eg if AParty dials BParty, can any information be conveyed (in both directions preferably, and in addition to CLI[PR]) before the call is answered? The USER-USER information element of Q.931 might do the trick, but support for that is still disabled in Asterisk 1.2.5 (look for SUPPORT_USERUSER in chan_zap.c). Maybe it works with one of the dedicated isdn channel modules. -- Dr. Michael Neuhauserphone: +43 1 789 08 49 - 30 Firmix Software GmbH fax: +43 1 789 08 49 - 55 Vienna/Austria/Europe email: [EMAIL PROTECTED] Embedded Linux Development and Serviceshttp://www.firmix.at/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel compile errors on x86_64
Walter Klomp wrote: Just downloaded the latest cvs from zaptel on my sparking new Athlon64 Centos4.2 system, but hitting a stumbling block... (sorry for the long post) Yup, there's a typo in the latest CentOS/RHEL kernel (confirmed by Redhat). The fix is to edit the Zaptel Makefile (fix courtesy of Russ Price): Here's a quick fix. In your zaptel Makefile, add the following (line 38 for 1.2.4) - THIS SHOLD BE ALL ONE LINE: CFLAGS+=$(shell if uname -r | grep -q 2.6.9-34.EL; then echo -Drw_lock_t=\rwlock_t\; fi) This way, if this is fixed in the next kernel release, you won't need to make another change to the Makefile. -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Re: Cisco phones and Linksys SRW224P
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Nothing for sure, and you may already know this, but some early Cisco phones only knew how to speak Cisco PoE, not the 802 standard which was defined a bit later. The Cisco web site should tell you which phone talks which protocol though. Hi James! It seams that you are right. I have another phone (7940 - I have bough it the same time I bough 7905 that works) that works. I have searched over Cisco site and I can't find list which hardware revision supports PoE 802.3AF. Do you have link? For now I know. 7905 - 74-3092-04 Rev.F0 = supports PoE 802.3AF 7905 - 74-3092-08 Rev.A0 = doesn't support PoE 802.3AF 7940 - 68-1735-11 Rev.A0 = supports PoE 802.3AF 7940 - 68-2564-03 B0 = doesn't support PoE 802.3AF 7960 - 68-2563-03 B0 = doesn't support PoE 802.3AF -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX choppy sound
On 15 Mar 2006, at 09:02, Stojan Sljivic - GDS wrote: Hi, Are you using Trunked IAX? Currently we do not use trunking. How many calls at a time? All the test we have performed so far were with only one active call. What codecs are you using? We have set the bandwith=low, so I think that G.723.1, GSM, and LPC10 are in the play. If you have 256kbits/s available and want to make a maximum of 2 calls you could try something using ulaw (~80kbits/s) anyhow, I would explicitly set the codec so that you can compare them. eg: disallow=all allow=ulaw What is the ping time between the systems? Ping stats are: Server 1: 50 packets transmitted, 50 received, 0% packet loss, time 49491ms rtt min/avg/max/mdev = 190.198/213.028/283.275/25.307 ms Server 2: 50 packets transmitted, 49 received, 2% packet loss, time 49523ms rtt min/avg/max/mdev = 190.089/214.855/544.880/59.052 ms That is quite a variation, over an already longish ping time. you probably need to do some traffic shaping at your routers to give IAX priority. If you are getting good results from skype over the same link, you could try examining the TOS bits in the skype packets and setting the IAX to use the same TOS bits since that may be what is making the difference. Any error messages ? There are no error messages in the console. Just to check, can you get decent call quality between 2 IAX clients on the same (local server)? Regards, Stojan Sljivic Hope that helps Tim ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT - force Cisco phones to reboot
I dont have this cisco-check-cfg exten command in my asterisk... Did you installed some extra module or channel? Thanks Joao Pereira Aaron Daniel wrote: It really depends on the number of phones you're wanting to reboot. Whenever we do a reconfiguration of our phones, I have a script that runs that night that pulls all the names from the db that are cisco phones, and does a sip notify cisco-check-cfg exten in asterisk, which notifies the phone to reboot in 20 seconds if nothing interesting happens (phone call comes in... browsing the interface... stuff like that). In order for this to work, you have to put a file in the tftpboot folder called syncinfo.xml containing this: SYNCINFO IMAGE VERSION=* SYNC=0/ /SYNCINFO in order for the phones to actually reboot though. That's what we do anyway :) Aaron Joao Pereira wrote: Hello to all Does someone knows how to force the Cisco IP phones (7940 and 7960) to reboot weekly or monthly? I think this would be useful because sometimes we change the configuration settings in the TFTP, but the phone just check the TFTP when he restarts... Thanks João ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX choppy sound
Hi Tim, Just to check, can you get decent call quality between 2 IAX clients on the same (local server)? I have never tested that since we have no IAX phones. We use SIP phones and IAX is used for connecting two Asterisk servers. Regards, Stojan Sljivic -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton Sent: Wednesday, March 15, 2006 13:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX choppy sound On 15 Mar 2006, at 09:02, Stojan Sljivic - GDS wrote: Hi, Are you using Trunked IAX? Currently we do not use trunking. How many calls at a time? All the test we have performed so far were with only one active call. What codecs are you using? We have set the bandwith=low, so I think that G.723.1, GSM, and LPC10 are in the play. If you have 256kbits/s available and want to make a maximum of 2 calls you could try something using ulaw (~80kbits/s) anyhow, I would explicitly set the codec so that you can compare them. eg: disallow=all allow=ulaw What is the ping time between the systems? Ping stats are: Server 1: 50 packets transmitted, 50 received, 0% packet loss, time 49491ms rtt min/avg/max/mdev = 190.198/213.028/283.275/25.307 ms Server 2: 50 packets transmitted, 49 received, 2% packet loss, time 49523ms rtt min/avg/max/mdev = 190.089/214.855/544.880/59.052 ms That is quite a variation, over an already longish ping time. you probably need to do some traffic shaping at your routers to give IAX priority. If you are getting good results from skype over the same link, you could try examining the TOS bits in the skype packets and setting the IAX to use the same TOS bits since that may be what is making the difference. Any error messages ? There are no error messages in the console. Just to check, can you get decent call quality between 2 IAX clients on the same (local server)? Regards, Stojan Sljivic Hope that helps Tim ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk crash too much?
hi in my callcenter i start asterisk on server with asterisk_safe command, after 4 days i can see that it is crashed 12 times, reporting segmentation fault error...each time asterisk is correctly restarted without loss of services but, is it normal? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to forward frame
Hi,
I get this error in the log file when I call from my mobile to the Asterisk
server, but hang up the mobile before anyone picks up.
Normally I would not worry about it, but I have been having some bad
experiences (only recently, after about 9 months of good operation) with
asterisk, although there have been related issues with Telco lines /
equipment and also some Asterisk initiated CRC errors after upgrading to
1.2. So I have downgraded to 1.0.9.
So I can isolate everything I have just installed a VERY VERY simple dial
plan.
The setup is
Telco --- TDM 4 Port BRI --- Ericsspn BP250
Extensions.conf (all of it)
[default]
exten = s,1,Dial(ZAP/g4/211,45,t)
[dialstring]
exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup
[te405p-frombp250]
exten = _3XX,1,Answer
exten = _3XX,2,Dial(Sip/${EXTEN},6000,t)
exten = _3XX,3,Hangup
exten = _X.,1,Answer
exten = _X.,2,Dial(Zap/g1/${EXTEN},6000,t)
exten = _X.,3,Hangup
[te405p-intelstra]
exten = _X.,1,Answer
exten = _X.,2,Dial(Zap/g4/${EXTEN},6000,t)
exten = _X.,3,Hangup
[from-sip]
exten = s,1,Dial(SIP/3332,45,t)
exten = _0X.,1,Answer
exten = _0X.,2,Dial(Zap/g1/${EXTEN:1},6000,t)
exten = _0X.,3,Hangup
exten = _X.,1,Answer
exten = _X.,2,Dial(Zap/g4/${EXTEN},6000,t)
exten = _X.,3,Hangup
Zapata.conf
[channels]
context=default
musiconhold=default
switchtype=euroisdn
usecallerid=yes
cidsignalling=v23
cidstart=polarity
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=0.0
txgain=0.0
group=1
context=te405p-intelstra
;context=te405p-ext
pridialplan=local
signalling=pri_cpe
;overlapdial=yes
callerid=asreceived
channel=1-15, 17-31
;channel=32-46, 48-62
group=4
context=te405p-frombp250
;context=te405p-in
pridialplan=local
signalling=pri_net
overlapdial=yes
callerid=asreceived
channel=94-108, 110-124
;channel=32-46, 48-62
Zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
span=2,0,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-62
span=3,0,0,ccs,hdb3,crc4
bchan=63-77
dchan=78
bchan=79-93
span=4,0,0,ccs,hdb3,crc4
bchan=94-108
dchan=109
bchan=110-124
loadzone=au
defaultzone=au
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Re: [Asterisk-Users] Development news :: T38 passthrough support
Olle E Johansson wrote: 11 mar 2006 kl. 23.44 skrev George Pajari: *** ITU T.38 -- Fax over VoIP It's not clear from the bug tracker if the problem with a T.38 endpoint (say ATA) behind NAT is working yet (with sip.conf specifying nat=yes/qualify=yes). Is this working or do both T.38 endpoints have to have public routable IP addresses still? I wasn't aware of this problem. Please tell me more! I can see at least one report so far confirming that T.38 doesn't work for nat=yes and canreinvite=no. I have the same problem. I am using the asterisk-1.2.4-t38-20060216.tar.bz2 patch against 1.2.4. Only one of the edinpoints is behind NAT. I did some sniffing and discovered that the problem appears after the asterisk server gets reINVITE for the T.38 session. Before the reINVITE asterisk will relay all RTP packets to the public IP of the NATed endpoint. After the T.38 reINVITE it will indeed switch to T.38 but all T.38 packets for the NATed endpoint suddenly start being sent to the private IP instead of the public one, never really reching that endpoint and resulting in failed fax transmission. See http://bugs.digium.com/view.php?id=5090 -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IMACS800
I take care of a system that has Premisys IMACS units in it. They are setup in pairs and each pair uses a single T1 circuit connected to their WAN Dual 8010 cards. In an effort to create a redundat T1 link I could really use some help in configuring the units to use the second T1 port when the other one fails (LOS and/or NOS). Or do I need to put in a second WAN card for the redundancy feature to work? If so, where can I get my hands on 10 of them and for how much? Am I at the right place here? Patrick Forbes Wireless Systems Technician Information Communications Systems Toronto Fire Services Works Emergency Services 4330 Dufferin Street, Centre Block, 3rd Floor, Rm N218 Toronto, Ontario M3H 5R9 Tel : 416-338-9568 Fax: 416-338-9404 Cell: 416-779-2893 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Re: Cisco phones and Linksys SRW224P
7905/12, 7940/60 are NOT 802.3af compatible ONLY 7911, 7941/61/70 (and their gigabiteth variants), are 802.3af compliant PJ Tomislav Parčina wrote: Hi James! It seams that you are right. I have another phone (7940 - I have bough it the same time I bough 7905 that works) that works. I have searched over Cisco site and I can't find list which hardware revision supports PoE 802.3AF. Do you have link? For now I know. 7905 - 74-3092-04 Rev.F0 = supports PoE 802.3AF 7905 - 74-3092-08 Rev.A0 = doesn't support PoE 802.3AF 7940 - 68-1735-11 Rev.A0 = supports PoE 802.3AF 7940 - 68-2564-03 B0 = doesn't support PoE 802.3AF 7960 - 68-2563-03 B0 = doesn't support PoE 802.3AF -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AVM C2 chan_capi-cm-0.6.3 Error on Dial
I'm getting a strange error on one of the two controllers on an AVM C2
card under chan_capi-cm-0.6.3.
I have two ISDN controllers defined, both in the same group, both
connections are UK ISDN2e Point to Point:
On the third outbound call (both of the first two calls are handled by
the second controller ISDN2,) I get this error :
chan_capi.c conf_error 0x2001 PLCI=0x301 Command=CONNECT_B3_CONF,0x8487
Does anyone have any idea what's going on here ? BT tell me there's no
problem they can see with the ISDN line involved.
jd
This is the dialstring :
exten = _9.,1,SetCallerPres(allowed)
exten = _9.,2,SetCIDNum(252000)
exten = _9.,3,Dial(CAPI/g1/${EXTEN:1}/b)
exten = _9.,4,Congestion
/etc/capi.conf contains :
c2 c2.bin DSS1 - - - - P2P
c2 c2.bin DSS1 - - - - P2P
/etc/asterisk/capi.conf contains :
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.9
txgain=0.3
; interface sections ...
[ISDN1]
isdnmode=DID
incomingmsn=*
controller=1
group=1
softdtmf=on
relaxdtmf=on
accountcode=
context=capi-in
holdtype=hold
echocancel=yes
echotail=64
bridge=yes
callgroup=1
devices=2
[ISDN2]
isdnmode=DID
incomingmsn=*
controller=2
group=1
softdtmf=on
relaxdtmf=on
accountcode=
context=capi-in
holdtype=hold
echocancel=yes
echotail=64
bridge=yes
callgroup=1
devices=2
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[Asterisk-Users] misdn problem
I am trying to use misdn insted of zaphfc to drive two billion isdn cards zaphfc is ok, but the problem with cdr and the fact tha you always have to wait the bristuffed version of asterisk took me to try another way. so I downloaded the misdn installation script from beronet for the last version ( I am using asterisk stable 1.2, so now is 1.2.5) wget http://www.beronet.com/downloads/install-misdn-mqueue.tar.gz tar -zxvf install-misdn-mqueue.tar.gz cd /usr/src/install-misdn-mqueue make make install everything OK /etc/init.d/misdn-init scan /etc/init.d/misdn-init config /etc/init.d/misdn-init start everything OK then I modify the /etc/asterisk/misdn.conf, in a very standard way: [general] debug=0 method=standard append_digits2exten=yes bridging=yes ;tracefile=/var/log/asterisk/misdn.trace [default] immediate=yes callgroup=1 pickupgroup=1 context=default language=it ;nationalprefix=0 ;internationalprefix=00 rxgain=0 txgain=0 dialplan=0 [TEports] ports=1,2 context=from-pstn msns=* ~ then: chmod 755 /usr/lib/asterisk/modules chown asterisk /dev/mISDN* -R everything still OK amportal start (I am using AMP ) OK. when I try to access an external line, asterisk crashes with a segmentation fault; the dial string is correct ... -- Executing GotoIf(SIP/567-bb09, 1?20:21) in new stack -- Goto (macro-dialout-trunk,s,20) -- Executing SetVar(SIP/567-bb09, the_num=3481303063) in new stack -- Executing Dial(SIP/567-bb09, misdn/1/3481303063) in new stack -- Called 1/3481303063 Ouch ... error while writing audio data: : Broken pipe Segmentation fault (core dumped) I am using Suse Linux 10, and I switched to default kernel (not SMP) [EMAIL PROTECTED]:~ uname -r 2.6.13-15.8-default Any help will be gratly appreciated. by the way: I read it could be possible to use chan_capi insted of chan_misdn, laying on misdn: is it correct: ? And if it is, could anybody give me an advice on how ? I tried the 0.6.4 chan_capi version I succesfully installed on anothe box with Fritz!, but in that case the capi driver for Fritz was present. thank in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] echo cancellation
Hi there I am using asterisk version 1.2.4. I have clients based on the iax client library dialling into meetme sessions. I am experiencing echo in the case where one or more users has speakers instead of headphones. So the audio from me is fed from the other participant's speakers into their mic and back to me. What is the best way to fix this? Is there an echo cancel facility in asterisk which will sort this out? Many thanks Steven ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: RE: Re: Cisco phones and Linksys SRW224P
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... 7905/12, 7940/60 are NOT 802.3af compatible ONLY 7911, 7941/61/70 (and their gigabiteth variants), are 802.3af compliant Totally not true! I have 7905 phone that IS 802.3af compatible. Its on my table, right to my laptop from which I'm sending this message. I also have 7940 phone that is 802.3af compatible. And I also have several 7905 that don't work with 802.3af, and half dozen of 7940 that don't work with 802,3af. It's all like I wrote in previous mail. I belive it's because of hardware revision. (read my previous mail). -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] misdn problem
[EMAIL PROTECTED] wrote on 15.03.2006 14:37:27:
I am trying to use misdn insted of zaphfc to drive two billion isdn
cards
zaphfc is ok, but the problem with cdr and the fact tha you always have
to
wait the bristuffed version of asterisk took me to
try another way.
so I downloaded the misdn installation script from beronet for the last
version ( I am using asterisk stable 1.2, so now is 1.2.5)
wget http://www.beronet.com/downloads/install-misdn-mqueue.tar.gz
tar -zxvf install-misdn-mqueue.tar.gz
cd /usr/src/install-misdn-mqueue
make
make install
everything OK
/etc/init.d/misdn-init scan
/etc/init.d/misdn-init config
/etc/init.d/misdn-init start
everything OK
then I modify the /etc/asterisk/misdn.conf, in a very standard way:
[general]
debug=0
method=standard
append_digits2exten=yes
bridging=yes
;tracefile=/var/log/asterisk/misdn.trace
[default]
immediate=yes
callgroup=1
pickupgroup=1
context=default
language=it
;nationalprefix=0
;internationalprefix=00
rxgain=0
txgain=0
dialplan=0
[TEports]
ports=1,2
context=from-pstn
msns=*
~
then:
chmod 755 /usr/lib/asterisk/modules
chown asterisk /dev/mISDN* -R
everything still OK
amportal start (I am using AMP )
OK.
when I try to access an external line, asterisk crashes with a
segmentation
fault;
the dial string is correct
...
-- Executing GotoIf(SIP/567-bb09, 1?20:21) in new stack
-- Goto (macro-dialout-trunk,s,20)
-- Executing SetVar(SIP/567-bb09, the_num=3481303063) in new
stack
-- Executing Dial(SIP/567-bb09, misdn/1/3481303063) in new stack
-- Called 1/3481303063
Ouch ... error while writing audio data: : Broken pipe
Segmentation fault (core dumped)
I am using Suse Linux 10, and I switched to default kernel (not SMP)
[EMAIL PROTECTED]:~ uname -r
2.6.13-15.8-default
Any help will be gratly appreciated.
by the way: I read it could be possible to use chan_capi insted of
chan_misdn, laying on misdn: is it correct: ?
And if it is, could anybody give me an advice on how ? I tried the 0.6.4
chan_capi version I succesfully installed on anothe box with Fritz!,
but in that case the capi driver for Fritz was present.
thank in advance,
Andrea
maybe try to dial via Dial(misdn/g:TEports/${EXTEN}) inside
extensions.conf
I encountered a few asterisk-crashes with mISDN as well
as it seems misdn doesn't like digital calls at all and is crashing in
this case...
yes, it's possible to use chan_capi via misdn/capi you just have to add
entries for the hfc-cards to your /etc/capi.conf
___
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Re: [Asterisk-Users] Stuck. Extenions.conf? Realtime? MySQL? Grrrrr!
On Tue, 2006-03-14 at 21:02 -0700, Douglas Garstang wrote: Boy, am I stuck... [snip] Why don't you just hire a consultant/company to implement this on a no cure no pay basis? Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] external modem
Can Asterisk @ home receive incoming call using a external modem? Thanks Gidean Chan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel compile errors on x86_64
Yep,
kernel-devel-2.6.9-22.EL
kernel-devel-2.6.9-34.EL
kernel-2.6.9-34.EL
kernel-utils-2.4-13.1.69
kernel-smp-devel-2.6.9-34.EL
kernel-2.6.9-22.EL
glibc-kernheaders-2.4-9.1.98.EL
all installed...
--
Message: 18
Date: Wed, 15 Mar 2006 11:18:36 +0100
From: Dave Cotton [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Zaptel compile errors on x86_64
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain
On Wed, 2006-03-15 at 17:49 +0800, Walter Klomp wrote:
Hi,
Just downloaded the latest cvs from zaptel on my sparking new Athlon64
Centos4.2 system, but hitting a stumbling block... (sorry for the long
post)
Kernel source installed?
--
Dave Cotton [EMAIL PROTECTED]
--
Message: 19
Date: Wed, 15 Mar 2006 15:19:32 +0500
From: Mazhar Hussain [EMAIL PROTECTED]
Subject: [Asterisk-Users] There is lacking behind in recorded calls
via sox
To: asterisk-users@lists.digium.com,
[EMAIL PROTECTED]
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=iso-8859-1
Hi ,
I have been using, Monitor (wav, ${CALLFILENAME}) to records calls from
extensions.conf and soxmix software to compiles calls. The problem I am
facing that for long calls more that 2 minutes there is disturbance in
sequence of calls, calls from both ends are not in sequence and there is
always lack behind from one end, I have tried a lot by checking different
version of Sox but still I am facing same issue .Can any one of you will
let me know the reason of this lacking in calls form end after compilation
while calls conversation goes fine in live calls .
Thanks,
Mazhar
Nettechltd.com
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Message: 20
Date: Wed, 15 Mar 2006 15:23:08 +0500
From: Mazhar Hussain [EMAIL PROTECTED]
Subject: [Asterisk-Users] there is lack behind in recoded calls via
sox
To: [EMAIL PROTECTED],
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=iso-8859-1
Hi ,
I have been using, Monitor (wav, ${CALLFILENAME}) to records calls from
extensions.conf and soxmix software to compiles calls. The problem I am
facing that for long calls more that 2 minutes there is disturbance in
sequence of calls, calls from both ends are not in sequence and there is
always lack behind from one end, I have tried a lot by checking different
version of Sox but still I am facing same issue .Can any one of you will
let me know the reason of this lacking in calls form end after compilation
while calls conversation goes fine in live calls .Also I am using Asterisk
1.2.5 version
Thanks,
Mazhar
Nettechltd.com
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Message: 21
Date: Wed, 15 Mar 2006 11:23:29 +0100
From: Alejandro Vargas [EMAIL PROTECTED]
Subject: [Asterisk-Users] spa 3000/2100 noise
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1
I've a problem. I've some spa3000 and spa2100. Asterisk 1.2.4.
Prefered codec g711u in both. Calleng from a fxs of spa2100 to the fxo
of spa3000, all works ok. Then I call from a sip phone configured for
using g729, to the fxo of spa3000, it also works ok.
The problem is that after this, when, making again a new call from
spa2100 to spa3000, spa2100 receives only white noise. I suspect a
codec mismatch. The problem disappears by powering off and on the
spa3000.
¿Any ideas on how to check?
--
Alejandro Vargas
--
Message: 22
Date: Wed, 15 Mar 2006 11:29:37 +0100
From: Alejandro Vargas [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk to receive fax
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1
2006/3/15, Gidean Chan [EMAIL PROTECTED]:
Can anyone tell me how to configure my system so that fax can be received
and forward to email account?
You can install iaxfax. It acts as a software modem that connects to
asterisk as a iax phone. It creates a device that can be accesed as a
faxmodem. Then, you can use hylafax that is very powerfull and can be
configured to forward faxes to email, convert it to pdf, etc. etc
(read the documentation).
--
Alejandro Vargas
--
Message: 23
Date: Wed, 15 Mar 2006 05:34:08 -0500
From: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Stuck. Extenions.conf? Realtime? MySQL?
Gr!
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED
[Asterisk-Users] Meetme monitoring only bug
I am not sure but there may be a bug in the Meetme() application. The flag p (allow user to exit the conference by pressing #) does not work when the flag m (sets monitor-only mode ) is also set. I am unable to exit a conference when in monitor only mode. Can anyone tell me if this is a known issue, or what a work around is? Thanks Jeff ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: RE: Re: Cisco phones and Linksys SRW224P
Tomislav, please look at: http://powerdsine.com/Support/Certification/company_All.asp?company=Cisco also on ci$co site you can't find info, that old phones are 802.3af compliant, but pre-standard... old ci$co phones can work with some poe equipment, but you can't be sure, that will be working with all 802.3af power devices/midspans only fact, that your phone is working with your 802.3af equipment can't be guarantee for compatibility... PJ Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... 7905/12, 7940/60 are NOT 802.3af compatible ONLY 7911, 7941/61/70 (and their gigabiteth variants), are 802.3af compliant Totally not true! I have 7905 phone that IS 802.3af compatible. Its on my table, right to my laptop from which I'm sending this message. I also have 7940 phone that is 802.3af compatible. And I also have several 7905 that don't work with 802.3af, and half dozen of 7940 that don't work with 802,3af. It's all like I wrote in previous mail. I belive it's because of hardware revision. (read my previous mail). -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Native Sounds - in case you missed it...
This was non-trivial for me also. I prefer to right-click-copy the link on the website, switch over to putty type in my wget (right-click), and download the file directly to the box. The link I tried on the sounds page happily downloaded index.html (if memory serves). I did go ahead and get the ulaw files the hard way... Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Wednesday, March 15, 2006 2:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Native Sounds - in case you missed it... On Tue, Mar 14, 2006 at 11:26:14PM -0800, Ira wrote: At 08:51 PM 03/14/2006, you wrote: In my humble opinion, EVERYONE (unless you have your own in a different voice/language) that uses Asterisk should be using these prompts. How about a direct link this time: For what it's worth, the hardest problem I had was not being able to directly FTP them from my Asterisk box. I'm a Linux newbie and had no idea how to do that. FTP *to a linux box*? /me is shocked! You have ssh access, right? Use scp/sftp. Try http://winscp.sf.net/ . If you don't one to carry one around or install on your system(s), put one statically-linked copy on your file/web server and download/run it. I downloaded them to my Windows box, set up vsftpd and uploaded them using a GUI FTP client in Windows and only then could I use them. wget http://server.name/path/to/file wget ftp://server.name/path/to/file In fact, what I normally do is copy a link from my browser to the command line in the terminal window and download it with wget. Saves me an extra file copy around the net. So for those who need exact commands, here's a two-liner: wget http://mirror.astlinux.org/sounds/asterisk-native-sounds-20060209-01-sln .tar.bz2 tar xjf asterisk-native-sounds-20060209-01-sln.tar.bz2 -C /var/lib/asterisk ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk to receive fax
I would recommend this particular method as well. It's quite a project, but the end result seems to be a very solid, configurable solution. Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Vargas Sent: Wednesday, March 15, 2006 4:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk to receive fax 2006/3/15, Gidean Chan [EMAIL PROTECTED]: Can anyone tell me how to configure my system so that fax can be received and forward to email account? You can install iaxfax. It acts as a software modem that connects to asterisk as a iax phone. It creates a device that can be accesed as a faxmodem. Then, you can use hylafax that is very powerfull and can be configured to forward faxes to email, convert it to pdf, etc. etc (read the documentation). -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] echo cancellation
Steven Langley wrote: Hi there I am using asterisk version 1.2.4. I have clients based on the iax client library dialling into meetme sessions. I am experiencing echo in the case where one or more users has speakers instead of headphones. So the audio from me is fed from the other participant's speakers into their mic and back to me. Echo cancellation is usually the responsibility of the speaker phone, one without E.C. is unusable. For instance, the Grandstream BT102's speaker phone. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX choppy sound
Hi, I have downloaded an IAX softphone and tested the connection locally. The sound is perfect. How should I troubleshoot this IAX connection between these two Asterisk servers? Is there some tool that can help in determining the cause of the choppy sound? Regards, Stojan Sljivic -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stojan Sljivic - GDS Sent: Wednesday, March 15, 2006 13:14 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] IAX choppy sound Hi Tim, Just to check, can you get decent call quality between 2 IAX clients on the same (local server)? I have never tested that since we have no IAX phones. We use SIP phones and IAX is used for connecting two Asterisk servers. Regards, Stojan Sljivic -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton Sent: Wednesday, March 15, 2006 13:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX choppy sound On 15 Mar 2006, at 09:02, Stojan Sljivic - GDS wrote: Hi, Are you using Trunked IAX? Currently we do not use trunking. How many calls at a time? All the test we have performed so far were with only one active call. What codecs are you using? We have set the bandwith=low, so I think that G.723.1, GSM, and LPC10 are in the play. If you have 256kbits/s available and want to make a maximum of 2 calls you could try something using ulaw (~80kbits/s) anyhow, I would explicitly set the codec so that you can compare them. eg: disallow=all allow=ulaw What is the ping time between the systems? Ping stats are: Server 1: 50 packets transmitted, 50 received, 0% packet loss, time 49491ms rtt min/avg/max/mdev = 190.198/213.028/283.275/25.307 ms Server 2: 50 packets transmitted, 49 received, 2% packet loss, time 49523ms rtt min/avg/max/mdev = 190.089/214.855/544.880/59.052 ms That is quite a variation, over an already longish ping time. you probably need to do some traffic shaping at your routers to give IAX priority. If you are getting good results from skype over the same link, you could try examining the TOS bits in the skype packets and setting the IAX to use the same TOS bits since that may be what is making the difference. Any error messages ? There are no error messages in the console. Just to check, can you get decent call quality between 2 IAX clients on the same (local server)? Regards, Stojan Sljivic Hope that helps Tim ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel compile errors on x86_64 - DEFINE_SPINLOCK???
Hi, (sorry for my mistake in not deleting the rest of the message just now) The problem seems to be here in zaptel.c (and torisa.c) #ifdef DEFINE_SPINLOCK static DEFINE_SPINLOCK(zaptimerlock); static DEFINE_SPINLOCK(bigzaplock); #else static spinlock_t zaptimerlock = SPIN_LOCK_UNLOCKED; static spinlock_t bigzaplock = SPIN_LOCK_UNLOCKED; #endif If I remark out as follows: //#ifdef DEFINE_SPINLOCK //static DEFINE_SPINLOCK(zaptimerlock); //static DEFINE_SPINLOCK(bigzaplock); //#else static spinlock_t zaptimerlock = SPIN_LOCK_UNLOCKED; static spinlock_t bigzaplock = SPIN_LOCK_UNLOCKED; //#endif Things compile, but I don't know what it actually does. Any comments? Message: 18 Date: Wed, 15 Mar 2006 11:18:36 +0100 From: Dave Cotton [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Zaptel compile errors on x86_64 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain On Wed, 2006-03-15 at 17:49 +0800, Walter Klomp wrote: Hi, Just downloaded the latest cvs from zaptel on my sparking new Athlon64 Centos4.2 system, but hitting a stumbling block... (sorry for the long post) Kernel source installed? -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problem configuring a digium quad E1 card
Hi, I bought a Digium Quad E1 card model TE406P. Till now, I can't make it work... I mean, I have red alarm when I configure one E1. The provider is in France (France Télécom) and I use the following zaptel config : span=1,1,0,ccs,hdb3,crc4bchan=1-15,17-31dchan=16 I'm using Linux 2.6.15 and when I run ztcfg -, it seems that all channels are configured... So can someone give me an advice on that matter... maybe someone in France who already configured that type of access. Also, I would like that you confirm the type of cable which can be used to connect the card to the Telco : can I use a straight cable or use a crossed cable with pair 1-2 connect to 4-5 ? Thanks Best regards! David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Development news :: T38 passthrough support
On Wed, 2006-03-15 at 11:26 +, Paul Hayes wrote: The SPA-2100 is the only one to support T.38 at the moment though. SPA-2002 has the ability to support t.38 (i.e. it has the processing power required) but the firmware support isn't there yet. Any info on the SPA-3000 and t.38 support? Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Attended transfers timing out after 3 rings
Hi, Using 1.2.5 with attended transfers, we are finding that dialling the transferee is timing out after only 3 rings, after which the original caller is transferred back. I have searched high and low but cannot find anywhere to increase the timeout for dialling the transferee. This is a big problem for us, so any help is much appreciated! -- -Barry Flanagan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming calls
Hi,
I run an asterisk server. The configuration is very basic.
Here is my question :
When someone calls my phone line, which is connected to an FXO card,
asterisk is answering using the context :
; Incoming calls goes to this default context :
[incoming-rtc]
include = postes-sip
;
exten = s,1,Goto(menu,1)
exten = s,2,Hangup
;
exten = menu,1,SetVar(count=0)
exten = menu,2,Answer
exten = menu,3,Background(silence/1)
exten = menu,4,Background(josh/welcome-msg)
exten = menu,5,Background(silence/5)
exten = menu,6,SetVar(count=$[${count} + 1])
exten = menu,7,GotoIf($[${count} 1]?4) ; Repeat 3 times
exten = menu,8,Goto(s,2)
When a friend calls, I would like for him to enter a 4 digit password
in order to access to a sub-menu, if no password is entered, then the
welcome msg is said ...
Any hints on how to do that ??
Thanks a lot !
___
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Re: [Asterisk-Users] Development news :: T38 passthrough
I found a bug in the latest T38 passthrough patches, the effect
is that a non-SIP call after being put on hold is then lost, no
resume is possible.
The fix is to be applied in the chan_sip.c file:
} else {
/* No bridged
peer with T38 enabled*/
transmit_response_with_sdp(p, 200 OK, req, 1);
}
- }
+ } else transmit_response_with_sdp(p,
200 OK, req, 1);
}
}
#else
transmit_response_with_sdp(p, 200 OK, req, 1);
#endif
Thanks for the T38 patch to everybody, it seems to be working
quite well in the first tests, but I'll keep the list updated
on the proceedings.
Paolo Prandini
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RE: [Asterisk-Users] echo cancellation
My PocketPC with ppcIAX and/or SJPhone behaves in exactly the same way. The only resolution is to use an earbud... I'm guessing that the server's echo cancelling is intended to cancel minor echo introduced by the path, but doesn't handle 'real' echo caused by looping sound. Is that right? Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Wednesday, March 15, 2006 8:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] echo cancellation Steven Langley wrote: Hi there I am using asterisk version 1.2.4. I have clients based on the iax client library dialling into meetme sessions. I am experiencing echo in the case where one or more users has speakers instead of headphones. So the audio from me is fed from the other participant's speakers into their mic and back to me. Echo cancellation is usually the responsibility of the speaker phone, one without E.C. is unusable. For instance, the Grandstream BT102's speaker phone. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Toshiba Strata DK-280 support?
Hi everyone, Been reading up on Asterisk, and very interested in learning more. I've googled and read the archives and haven't found anything definitive on support for this phone system. We have a fairly large investment in the system itself and the phones, but would love to get away from the voicemail system it forces on us. Can anyone provide any feedback on using this system with Asterisk? Am I wasting my time even thinking about it? Thanks, -- Best regards, Charles Marcus I.T. Director Media Brokers International 678.578.2200 x224 678.578.2240 fax ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: RE: Re: Cisco phones and Linksys SRW224P
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Tomislav, please look at: http://powerdsine.com/Support/Certification/company_All.asp?company=Cisco also on ci$co site you can't find info, that old phones are 802.3af compliant, but pre-standard... old ci$co phones can work with some poe equipment, but you can't be sure, that will be working with all 802.3af power devices/midspans only fact, that your phone is working with your 802.3af equipment can't be guarantee for compatibility... PJ Thank you for link. The thing is that there are difference between different hardware revisions of Cisco phone's. I need to find out What hardware revisions support what standards. Anther thing, is there any Cisco switch that supports even oldest Cisco VoIP phones (7905 and 7940)? -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Stuck. Extenions.conf? Realtime? MySQL?
Douglas Garstang [EMAIL PROTECTED] wrote: Boy, am I stuck... I'm officially ready to toss Asterisk out the window. I have to admit it isn't necessarily all the fault of Asterisk either. It just seems that every option I turn to suddenly ends in failure. I don't know if it's me that's bitten of more than I can chew with this project, or maybe Asterisk just isn't mature enough yet. Nothing complicated really Just a carrier class solution, with advanced custom routing, incoming and outgoing number blocking (at user/company and global level) and whitelisting, findme/followme, user specific pic codes and rate centres based on number dialled, blocking of specific star code prefixed features, different caller ID based on intra company calls, outside calls, calls overriden to use alternate caller id with feature codes, and not to mention it all has to be HA. At the risk of stating the obvious it appears clear from this that you have taken on a complex software development project and you should treat it accordingly. The fact that the underlying Asterisk software is providing you with a number of telephony capabilities does not mean you do not have to develop your own application nor can it make developing your own application any easier than its own inherent complexity. Going right back to basics you have two ways forward: 1. Outsource it - probably best if software development is not your core business 2. Run it as a software development project - i.e. adopt a methodology, have a project plan etc. Most of the things you mention seem feasible enough but I doubt if any of them are simple and when you layer several non-trivial tasks on top of one another you have an amount of complexity that needs to be taken seriously. I don't think from what you have said that Asterisk is the problem, if there is a problem it may just be that OSS sometimes tempts us in to bite off more than we can chew. If you are hitting specific issues with Asterisk then I'm afraid you are going to have to deal with them one at a time, I would love to tell you that this is not the case with commercial software but that would make me a liar so I won't. You are writing a complex application that requires the integration of multiple 3rd party technologies, this is bound to be a frustrating experience. On the other hand this is what most commercial software development projects are like and most of them get there in the end so if the payback is worth it hang on in there. -- Nic ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Aastra 480i CT - multiple lines?
I just set up an Aastra 480i CT with separate registrations on my Asterisk server. The way I set it up is Line 1 on the phone is registered to 101 on the server and Line 3 is registered to 103. If Line 1 is being used and a call comes in on 101, it rings to Line 2. But, if Line 3 is being used and a call comes in on 103, the phone responds Busy Here. Is there any way of assigning groups of Line buttons to different registrations (like the snom phones)? Nabeel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AVM C2 chan_capi-cm-0.6.4 Error on Dial
Whoops ! sorry - wrong release ...
chan_capi-cm-0.6.4 !
John Daragon wrote:
I'm getting a strange error on one of the two controllers on an AVM C2
card under chan_capi-cm-0.6.3.
I have two ISDN controllers defined, both in the same group, both
connections are UK ISDN2e Point to Point:
On the third outbound call (both of the first two calls are handled by
the second controller ISDN2,) I get this error :
chan_capi.c conf_error 0x2001 PLCI=0x301 Command=CONNECT_B3_CONF,0x8487
Does anyone have any idea what's going on here ? BT tell me there's no
problem they can see with the ISDN line involved.
jd
This is the dialstring :
exten = _9.,1,SetCallerPres(allowed)
exten = _9.,2,SetCIDNum(252000)
exten = _9.,3,Dial(CAPI/g1/${EXTEN:1}/b)
exten = _9.,4,Congestion
/etc/capi.conf contains :
c2c2.bin DSS1 - - - - P2P
c2c2.bin DSS1 - - - - P2P
/etc/asterisk/capi.conf contains :
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.9
txgain=0.3
; interface sections ...
[ISDN1]
isdnmode=DID
incomingmsn=*
controller=1
group=1
softdtmf=on
relaxdtmf=on
accountcode=
context=capi-in
holdtype=hold
echocancel=yes
echotail=64
bridge=yes
callgroup=1
devices=2
[ISDN2]
isdnmode=DID
incomingmsn=*
controller=2
group=1
softdtmf=on
relaxdtmf=on
accountcode=
context=capi-in
holdtype=hold
echocancel=yes
echotail=64
bridge=yes
callgroup=1
devices=2
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Re: [Asterisk-Users] misdn problem
Thank you for your answer.
I tried that syntax with misdn/g:TEports/${EXTEN}, but nothing changes.
what should I write in the /etc/capi.conf ?
If I had a Fritz, I would have
#SuSEconfig.isdn generated
# card fileproto io irq mem cardnr options
fcpci - - - - - 1
but having 2 billion ??? what to write ?
Andrea
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Re: [Asterisk-Users] misdn problem
Please respond to
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[EMAIL PROTECTED] wrote on 15.03.2006 14:37:27:
I am trying to use misdn insted of zaphfc to drive two billion isdn
cards
zaphfc is ok, but the problem with cdr and the fact tha you always have
to
wait the bristuffed version of asterisk took me to
try another way.
so I downloaded the misdn installation script from beronet for the last
version ( I am using asterisk stable 1.2, so now is 1.2.5)
wget http://www.beronet.com/downloads/install-misdn-mqueue.tar.gz
tar -zxvf install-misdn-mqueue.tar.gz
cd /usr/src/install-misdn-mqueue
make
make install
everything OK
/etc/init.d/misdn-init scan
/etc/init.d/misdn-init config
/etc/init.d/misdn-init start
everything OK
then I modify the /etc/asterisk/misdn.conf, in a very standard way:
[general]
debug=0
method=standard
append_digits2exten=yes
bridging=yes
;tracefile=/var/log/asterisk/misdn.trace
[default]
immediate=yes
callgroup=1
pickupgroup=1
context=default
language=it
;nationalprefix=0
;internationalprefix=00
rxgain=0
txgain=0
dialplan=0
[TEports]
ports=1,2
context=from-pstn
msns=*
~
then:
chmod 755 /usr/lib/asterisk/modules
chown asterisk /dev/mISDN* -R
everything still OK
amportal start (I am using AMP )
OK.
when I try to access an external line, asterisk crashes with a
segmentation
fault;
the dial string is correct
...
-- Executing GotoIf(SIP/567-bb09, 1?20:21) in new stack
-- Goto (macro-dialout-trunk,s,20)
-- Executing SetVar(SIP/567-bb09, the_num=3481303063) in new
stack
-- Executing Dial(SIP/567-bb09, misdn/1/3481303063) in new stack
-- Called 1/3481303063
Ouch ... error while writing audio data: : Broken pipe
Segmentation fault (core dumped)
I am using Suse Linux 10, and I switched to default kernel (not SMP)
[EMAIL PROTECTED]:~ uname -r
2.6.13-15.8-default
Any help will be gratly appreciated.
by the way: I read it could be possible to use chan_capi insted of
chan_misdn, laying on misdn: is it correct: ?
And if it is, could anybody give me an advice on how ? I tried the 0.6.4
chan_capi version I succesfully installed on anothe box with Fritz!,
but in that case the capi driver for Fritz was present.
thank in advance,
Andrea
maybe try to dial via Dial(misdn/g:TEports/${EXTEN}) inside
extensions.conf
I encountered a few asterisk-crashes with mISDN as well
as it seems misdn doesn't like digital calls at all and is crashing in
this case...
yes, it's possible to use chan_capi via misdn/capi you just have to add
entries for the hfc-cards to your /etc/capi.conf
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RE: [Asterisk-Users] problem configuring a digium quad E1 card
Hi You need to use a cross-over E1 cable (not an ethernet cross-over one) Good luck Jose Manuel Cortes David X Semestre Ingenieria Electronica PONTIFICIA UNIVERSIDAD JAVERIANA De: [EMAIL PROTECTED] en nombre de David Masure Enviado el: Mié 15/03/2006 9:41 Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] problem configuring a digium quad E1 card Hi, I bought a Digium Quad E1 card model TE406P. Till now, I can't make it work... I mean, I have red alarm when I configure one E1. The provider is in France (France Télécom) and I use the following zaptel config : span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 I'm using Linux 2.6.15 and when I run ztcfg -, it seems that all channels are configured... So can someone give me an advice on that matter... maybe someone in France who already configured that type of access. Also, I would like that you confirm the type of cable which can be used to connect the card to the Telco : can I use a straight cable or use a crossed cable with pair 1-2 connect to 4-5 ? Thanks Best regards! David winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cards
Hi Im developing an ip telephony project and i need some help in order to choose the better PCI card, the options at the moment are digium, sangoma and voicetronix, the strongest ones are digium and sangoma but i dont know how justify the election Best regards Jose Manuel Cortes David XSemestre Ingenieria Electronica PONTIFICIA UNIVERSIDAD JAVERIANA___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OSHA requirement to reach a live human ??
Hi, We're using Asterisk to develop a specialized IVR system for our employees and someone is telling us there is some OSHA requirement that you have to always be able to reach a live human on such systems. I've never heard of that and google didn't turn up anything in my searches. This is *not* some kind of report a spill or crucial system of any sort. It's a bit of a hassle because it wasn't going to be connected to their main phone system. Anyone ever heard of such a requirement from OSHA, or do you think someone is pulling my leg? Scott begin:vcard fn:Scott Plante n:Plante;Scott org:Insight Systems, Inc. adr:Suite 670;;1718 Peachtree St NW;Atlanta;GA;30309;US email;internet:[EMAIL PROTECTED] title:CTO tel;work:404 873 0058 x104 tel;fax:404 873 0063 x-mozilla-html:TRUE url:http://www.zyross.com version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: RE: Re: Cisco phones and Linksys SRW224P
as I known, all ci$co switches supports pre-standard ci$co phones and mostly all todays switches also supports new 802.3af phones (and also pre-standard phones) PJ Tomislav Parčina wrote: Anther thing, is there any Cisco switch that supports even oldest Cisco VoIP phones (7905 and 7940)? -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] problem configuring a digium quad E1 card
Hi again, Can you specify the pin order for each end ? thanks -Message d'origine-De: JOSE MANUEL CORTES DAVID [mailto:[EMAIL PROTECTED]De la part de JOSE MANUEL CORTES DAVIDEnvoyé: mercredi 15 mars 2006 16:28À: Asterisk Users Mailing List - Non-Commercial DiscussionObjet: RE: [Asterisk-Users] problem configuring a digium quad E1 card Hi Youneed touse a cross-over E1 cable(not an ethernet cross-over one) Good luck Jose Manuel Cortes David XSemestre Ingenieria Electronica PONTIFICIA UNIVERSIDAD JAVERIANA De: [EMAIL PROTECTED] en nombre de David MasureEnviado el: Mié 15/03/2006 9:41Para: asterisk-users@lists.digium.comAsunto: [Asterisk-Users] problem configuring a digium quad E1 card Hi, I bought a Digium Quad E1 card model TE406P. Till now, I can't make it work... I mean, I have red alarm when I configure one E1. The provider is in France (France Télécom) and I use the following zaptel config : span=1,1,0,ccs,hdb3,crc4bchan=1-15,17-31dchan=16 I'm using Linux 2.6.15 and when I run ztcfg -, it seems that all channels are configured... So can someone give me an advice on that matter... maybe someone in France who already configured that type of access. Also, I would like that you confirm the type of cable which can be used to connect the card to the Telco : can I use a straight cable or use a crossed cable with pair 1-2 connect to 4-5 ? Thanks Best regards! David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] echo cancellation
Bob McDowell wrote: My PocketPC with ppcIAX and/or SJPhone behaves in exactly the same way. The only resolution is to use an earbud... I'm guessing that the server's echo cancelling is intended to cancel minor echo introduced by the path, but doesn't handle 'real' echo caused by looping sound. Is that right? Asterisk's echo cancel is designed to cancel out echo caused by PSTN 2-wire circuits. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cisco 7912 not taking config
Hi, I am using a cisco 7912. I setup the phone at my first location. I edited the gkMAC.txt file setup the proxy and UID etc... values. generated the gkMAC file and booted the phone and it worked... I then mailed it to its final destination. this place has other 7912 phones working there.. I copied the gkMAC.txt file to that server, changed the proxy and UID values, generated the gkMAC file and the tftp server shows the phone as asking or the file. Mar 15 14:57:21 SERVER in.tftpd[21911]: RRQ from 192.168.X.Y filename gk0015c69dfc46 Mar 15 14:58:42 SERVER in.tftpd[21916]: RRQ from 192.168.X.Y filename gk0015c69dfc46 Mar 15 15:00:03 SERVER in.tftpd[21923]: RRQ from 192.168.X.Y filename gk0015c69dfc46 however all the phone shows is the initial config from my office. Its either not picking it up, rejecting it or something??? Doing a diff between the txt files from my office and the second location shows only the proxy and UID and password fields as being different. What might it be? THanks, Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Toshiba Strata DK-280 support?
On 3/15/06, Charles Marcus [EMAIL PROTECTED] wrote: Can anyone provide any feedback on using this system with Asterisk? Am Iwasting my time even thinking about it? I run Asterisk partnered up to the 280 (424 for us). We have a 6 cabinet installation of the Toshiba so I understand you dilemma. There are some quirks with the 280 that make it a challenge to use Asterisk with, but it's do-able. Keep in mind, you can move to the CTX or the CIX and still keep a lot of your investment and those systems play much better with Asterisk. -Brian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AVM C2 chan_capi-cm-0.6.4 Error on Dial
On Wed, 15 Mar 2006, John Daragon wrote:
Whoops ! sorry - wrong release ...
chan_capi-cm-0.6.4 !
There were problems with 0.6.3 and AVM cards, but should be fixed in 0.6.4.
Can you please create a full debug log (set verbose 5; capi debug) for such
a case ?
Armin
John Daragon wrote:
I'm getting a strange error on one of the two controllers on an AVM C2
card under chan_capi-cm-0.6.3.
I have two ISDN controllers defined, both in the same group, both
connections are UK ISDN2e Point to Point:
On the third outbound call (both of the first two calls are handled by
the second controller ISDN2,) I get this error :
chan_capi.c conf_error 0x2001 PLCI=0x301 Command=CONNECT_B3_CONF,0x8487
Does anyone have any idea what's going on here ? BT tell me there's no
problem they can see with the ISDN line involved.
jd
This is the dialstring :
exten = _9.,1,SetCallerPres(allowed)
exten = _9.,2,SetCIDNum(252000)
exten = _9.,3,Dial(CAPI/g1/${EXTEN:1}/b)
exten = _9.,4,Congestion
/etc/capi.conf contains :
c2 c2.bin DSS1 - - - - P2P
c2 c2.bin DSS1 - - - - P2P
/etc/asterisk/capi.conf contains :
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.9
txgain=0.3
; interface sections ...
[ISDN1]
isdnmode=DID
incomingmsn=*
controller=1
group=1
softdtmf=on
relaxdtmf=on
accountcode=
context=capi-in
holdtype=hold
echocancel=yes
echotail=64
bridge=yes
callgroup=1
devices=2
[ISDN2]
isdnmode=DID
incomingmsn=*
controller=2
group=1
softdtmf=on
relaxdtmf=on
accountcode=
context=capi-in
holdtype=hold
echocancel=yes
echotail=64
bridge=yes
callgroup=1
devices=2
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Re: [Asterisk-Users] how to show called name on calling polycom display
On 3/15/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi, we have an asterisk 1.2.1 box and 2 polycom SIP phones. We'd like to show the called name on the calling polycom display instead of his /her extensions as I do with the caller name on the called polycom. Is it possible? If yes, how? TIA Giorgio Incantalupo I believe something like this is being worked on in the bugtracker at bugs.digium.com. I don't remember how far along the project is though. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] misdn problem
##
# mISDN (experimental) #
##
#avmfritz - - - - - -
#hfcpci - - - - - -
#hfcsusb- - - - - -
#hfcmulti - - - - - -
#sedlfax- - - - - -
#w6692pci - - - - - -
this is out of the gentoo capi.conf - simply uncomment the entry you need
I've never tried 2 hfc-cards - should be the last entry to be
changed/duplicated with 1/2 for the cardnumber as like your sample
[EMAIL PROTECTED] wrote on 15.03.2006 16:25:01:
Thank you for your answer.
I tried that syntax with misdn/g:TEports/${EXTEN}, but nothing changes.
what should I write in the /etc/capi.conf ?
If I had a Fritz, I would have
#SuSEconfig.isdn generated
# card fileproto io irq mem cardnr options
fcpci - - - - - 1
but having 2 billion ??? what to write ?
Andrea
[EMAIL PROTECTED]
de
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asterisk-users-bo Asterisk Users Mailing List -
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cc
15/03/2006 14.56 Subject
Re: [Asterisk-Users] misdn
problem
Please respond to
Asterisk Users
Mailing List -
Non-Commercial
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[EMAIL PROTECTED]
ists.digium.com
[EMAIL PROTECTED] wrote on 15.03.2006 14:37:27:
I am trying to use misdn insted of zaphfc to drive two billion isdn
cards
zaphfc is ok, but the problem with cdr and the fact tha you always
have
to
wait the bristuffed version of asterisk took me to
try another way.
so I downloaded the misdn installation script from beronet for the
last
version ( I am using asterisk stable 1.2, so now is 1.2.5)
wget http://www.beronet.com/downloads/install-misdn-mqueue.tar.gz
tar -zxvf install-misdn-mqueue.tar.gz
cd /usr/src/install-misdn-mqueue
make
make install
everything OK
/etc/init.d/misdn-init scan
/etc/init.d/misdn-init config
/etc/init.d/misdn-init start
everything OK
then I modify the /etc/asterisk/misdn.conf, in a very standard way:
[general]
debug=0
method=standard
append_digits2exten=yes
bridging=yes
;tracefile=/var/log/asterisk/misdn.trace
[default]
immediate=yes
callgroup=1
pickupgroup=1
context=default
language=it
;nationalprefix=0
;internationalprefix=00
rxgain=0
txgain=0
dialplan=0
[TEports]
ports=1,2
context=from-pstn
msns=*
~
then:
chmod 755 /usr/lib/asterisk/modules
chown asterisk /dev/mISDN* -R
everything still OK
amportal start (I am using AMP )
OK.
when I try to access an external line, asterisk crashes with a
segmentation
fault;
the dial string is correct
...
-- Executing GotoIf(SIP/567-bb09, 1?20:21) in new stack
-- Goto (macro-dialout-trunk,s,20)
-- Executing SetVar(SIP/567-bb09, the_num=3481303063) in new
stack
-- Executing Dial(SIP/567-bb09, misdn/1/3481303063) in new
stack
-- Called 1/3481303063
Ouch ... error while writing audio data: : Broken pipe
Segmentation fault (core dumped)
I am using Suse Linux 10, and I switched to default kernel (not SMP)
[EMAIL PROTECTED]:~ uname -r
2.6.13-15.8-default
Any help will be gratly appreciated.
by the way: I read it could be possible to use chan_capi insted of
chan_misdn, laying on misdn: is it correct: ?
And if it is, could anybody give me an advice on how ? I tried the
0.6.4
chan_capi version I succesfully installed on anothe box with Fritz!,
but in that case the capi driver for Fritz was present.
thank in advance,
Andrea
maybe try to dial via Dial(misdn/g:TEports/${EXTEN}) inside
extensions.conf
I encountered a few asterisk-crashes with mISDN as well
as it seems misdn doesn't like digital calls at all and is crashing in
this case...
yes, it's possible to use chan_capi via misdn/capi you just have to add
entries for the hfc-cards to your /etc/capi.conf
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[Asterisk-Users] Asterisk integration with office PBX
Forgive me if this question has been asked/answered in another post. And let me reiterate what other users have frequently said - Asterisk is great, and I really appreciate all the work you folks have put into it. How have some of you gone about integrating Asterisk with a legacy office PBX, such that the end-user can use a regular office (digital handset) and dialing is fairly seamless ? Our end-users are accustomed to picking up their office handset and just dialing a 4 digit extension to reach another staffperson in our office. I'd like to replicate that so they can reach staff in our other (international) offices (behind the scenes, the call would route over IP). For instance, we have regular NEC handsets talking to an NEC PBX, and an analog line from the PBX to the Asterisk FXO. I already had our NEC tech set up an access code/alias, such that an end-user just dials 6 and it goes to the analog line going into Asterisk. Asterisk picks up after about 2 rings, and then the end-user is prompted to enter the destination phone number (which would be an e.g. 3 digit number corresponding to a SIP destination in the dialplan). But this means the end-user has to dial 6 and then wait for Asterisk to pick up. I'd Is there a way to have Asterisk pick up sooner, e.g. without any rings ? Ultimately, I'd like to get it to the point where the end-user doesn't have to pause at all. In other words, they could dial e.g. 6123 and their call would be appropriately routed. I realize that probably involves configuring Least Cost Routing on the NEC PBX, but that still leaves the issue of having to wait for Asterisk to pick up the line. Any help is appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OSHA requirement to reach a live human ??
I wanted to investigate this myself, so I called OSHA, got VoiceMail! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Plante Sent: Wednesday, March 15, 2006 10:37 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] OSHA requirement to reach a live human ?? Hi, We're using Asterisk to develop a specialized IVR system for our employees and someone is telling us there is some OSHA requirement that you have to always be able to reach a live human on such systems. I've never heard of that and google didn't turn up anything in my searches. This is *not* some kind of report a spill or crucial system of any sort. It's a bit of a hassle because it wasn't going to be connected to their main phone system. Anyone ever heard of such a requirement from OSHA, or do you think someone is pulling my leg? Scott ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OSHA requirement to reach a live human ??
People love to blame things on acronyms. Usually it's 'HIPPA' (which, by the way is a clear indicator that they've never studied HIPAA), sometimes OSHA, etc. If it really is OSHA then it should be pretty easy to find out. If (and check first) your organization is on the up and up, call them and ask. As I understand it, if you ask them there will be no penalties if they find you in the wrong. If they 'catch' you, then come the fines... Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Plante Sent: Wednesday, March 15, 2006 9:37 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] OSHA requirement to reach a live human ?? Hi, We're using Asterisk to develop a specialized IVR system for our employees and someone is telling us there is some OSHA requirement that you have to always be able to reach a live human on such systems. I've never heard of that and google didn't turn up anything in my searches. This is *not* some kind of report a spill or crucial system of any sort. It's a bit of a hassle because it wasn't going to be connected to their main phone system. Anyone ever heard of such a requirement from OSHA, or do you think someone is pulling my leg? Scott ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AVM C2 chan_capi-cm-0.6.4 Error on Dial
Armin Schindler wrote: On Wed, 15 Mar 2006, John Daragon wrote: Whoops ! sorry - wrong release ... chan_capi-cm-0.6.4 ! There were problems with 0.6.3 and AVM cards, but should be fixed in 0.6.4. Can you please create a full debug log (set verbose 5; capi debug) for such a case ? Certainly. Would you like it off-list ? jd ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: MeetMe Listen Only flag (|m)
(top posting for brevity, but original post included below, as it was over seven weeks ago) I've at last updated the patches for both trunk and 1.2, and posted them to Mantis at http://bugs.digium.com/view.php?id=6731 Cheers Tony In article [EMAIL PROTECTED], Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], Dan Austin [EMAIL PROTECTED] wrote: Tony wrote: I should tidy it up and submit it, but haven't got round to it :-( Let us know if you can. I'm already maintaining a grocery list of patches to make MeetMe viable in my orginization, so one more won't kill me. I should be able do so this weekend. That's the plan, anyway :-) I'll post the Mantis bug# when I've submitted it. OK, just to reassure people I didn't forget, I've now produced a patch for trunk and another for the 1.2 branch, by porting my changes across from the version I had, which was based on 1.0. However, I think I ought to check they compile and run before I submit them! I've run out of time to do that which weekend, so it will be a couple of days. If anyone else would like to try them out any quicker, please email me and I'll send you copies. Just to summarise what these patches provide: 1. The muting logic in the conference loop is tidied up, so that muting and unmuting is done according to the flag states near the top of the loop, and the DTMF muting/unmuting codes just set or clear the flags. 2. The 'm' flag now means initially muted, but allows the user to be unmuted from the command line. Users cannot unmute themselves if they were muted from the command line, only if they muted themselves. 3. The new 'l' flag means listen only and is what the 'm' flag used to be - unmuting is not possible. 4. Manager API events are generated when a user is muted or unmuted by admin or themselves. 5. The code '*' in the admin or user menus generates an API event which can be used by a user to attract the attention of an operator (e.g. a muted user who wishes to speak). Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OSHA requirement to reach a live human ??
Scott Plante wrote: Hi, We're using Asterisk to develop a specialized IVR system for our employees and someone is telling us there is some OSHA requirement that you have to always be able to reach a live human on such systems. I've never heard of that and google didn't turn up anything in my searches. This is *not* some kind of report a spill or crucial system of any sort. It's a bit of a hassle because it wasn't going to be connected to their main phone system. Anyone ever heard of such a requirement from OSHA, or do you think someone is pulling my leg? Our HR department said that it may well be a rule, and they'll investigate and I'll report back. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: OSHA requirement to reach a live human ??
Hi Scott - We're using Asterisk to develop a specialized IVR system for our employees and someone is telling us there is some OSHA requirement that you have to always be able to reach a live human on such systems. I've never heard of that and google didn't turn up anything in my searches. This is *not* some kind of report a spill or crucial system of any sort. It's a bit of a hassle because it wasn't going to be connected to their main phone system. Anyone ever heard of such a requirement from OSHA, or do you think someone is pulling my leg? I was going to suggest you give OSHA a call to ask them, but then I realized that you'd never actually get a live human to talk to. My company occasionally has to deal with OSHA issues, and I would guess this is not true. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OSHA requirement to reach a live human ??
On 3/15/06, Scott Plante [EMAIL PROTECTED] wrote: Hi, We're using Asterisk to develop a specialized IVR system for our employees and someone is telling us there is some OSHA requirement that you have to always be able to reach a live human on such systems. I've never heard of that and google didn't turn up anything in my searches. This is *not* some kind of report a spill or crucial system of any sort. It's a bit of a hassle because it wasn't going to be connected to their main phone system. Anyone ever heard of such a requirement from OSHA, or do you think someone is pulling my leg? Scott Unless the person is someone you absolutely can't afford to irritate, I would ask them to supply you with the actual regulation to which they are referring. That should get them to put up or shut up. -Rusty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: how to show called name on calling polycom display
Hi Giorgio - we have an asterisk 1.2.1 box and 2 polycom SIP phones. We'd like to show the called name on the calling polycom display instead of his /her extensions as I do with the caller name on the called polycom. Is it possible? If yes, how? If this is possible, it would be quite complicated to do. This would take some tricky XML hacking on the Polycom side to read this info and display it on the phone's screen, and some even more clever way to send this info from the asterisk machine. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] res_config_mysql.so not found
I just download and compile asterisk-addons. But whne I tried to start Asterisk and I go t error as below: [res_config_mysql.so]Mar 15 09:32:24 WARNING[10597]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/res_config_mysql.so: cannot open shared object file: No such file or directoryMar 15 09:32:24 WARNING[10597]: loader.c:499 load_modules: Loading module res_config_mysql.so failed! I can not find res_config_mysql.so in /usr/lib/asterisk/modules diretory. Can some one please tell me how can I load mysql reatime addon and fix thsi problem? Appreciate any help. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OSHA requirement to reach a live human ??
Are you selling it TO osha? If so, maybe they have an internal requirement.. If not, I've never heard of that. Granted, I haven't sold a LOT of phone systems, but I've been involved with a couple into public works departments of local governments as well as private corps, and nobody has ever mentioned that... From: Scott Plante [mailto:[EMAIL PROTECTED] Sent: Wed 3/15/2006 10:37 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] OSHA requirement to reach a live human ?? Hi, We're using Asterisk to develop a specialized IVR system for our employees and someone is telling us there is some OSHA requirement that you have to always be able to reach a live human on such systems. I've never heard of that and google didn't turn up anything in my searches. This is *not* some kind of report a spill or crucial system of any sort. It's a bit of a hassle because it wasn't going to be connected to their main phone system. Anyone ever heard of such a requirement from OSHA, or do you think someone is pulling my leg? Scott winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fake Ring Tone/Compile Addon
Dear All,I am currently have this problem in which I am sending call out from the Zaptel TE405 to a VoIP gateway. But the problem that the call over to the VoIP Gateway will always have a fake ring tone. Can you please give some pointer how to fix this problem? This problem is existing in my Asterisk 1.2.1 box. Also when compiling Asterisk 1.2.5 and tried to run it with the Asterisk-addon, the CLI output would get segmentation default. The source of the problem was that res_config_mysql.so was giving problem. It was when compiling the asterisk-addon, it was using the default Libarary and includes of the Fedore Core mysql directories. But my box is now running MySQL 5, and the source directory is at /root/mysql- standard-5.0.16-linux-i686 directory. My question is how to compile using this new source code for the header files and the libarary files? Many thanks.Regards,Kengie ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OSHA requirement to reach a live human ??
But, ...your call is very important to us... :) Alexander Lopez wrote: I wanted to investigate this myself, so I called OSHA, got VoiceMail! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Plante Sent: Wednesday, March 15, 2006 10:37 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] OSHA requirement to reach a live human ?? Hi, We're using Asterisk to develop a specialized IVR system for our employees and someone is telling us there is some OSHA requirement that you have to always be able to reach a live human on such systems. I've never heard of that and google didn't turn up anything in my searches. This is *not* some kind of report a spill or crucial system of any sort. It's a bit of a hassle because it wasn't going to be connected to their main phone system. Anyone ever heard of such a requirement from OSHA, or do you think someone is pulling my leg? Scott ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ooh323 Gatekeeper Bug
Dear All,It seems that there is a bug on the ooh323 while using registering with gatekeeper. The gatekeeper is GnuGK and the problem is when the Asterisk recieves a call from the Gatekeeper and routes it back out to an SIP Phone. The call would be connected and immediately dropped after 1-2 seconds connection time. This doesn't happen when ooh323 module isn't registered to a gatekeeper. This current version i am using for the ooh323 is from Asterisk-addon-1.2.1. Is there a bug on this version for the ooh323 and also how can i get the newer version of the ooh323(0.8.1) to compile with? Many thanks to you all.Regards,Kengie ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [SPAM] Re: [Asterisk-Users] OT - force Cisco phones to reboot
In the sip_notify.conf file, there's a couple different events that will cause different phones to reboot. One of them is cisco-check-cfg. In the asterisk cli, if you run sip notify cisco-check-cfg exten with that file in your tftpboot directory, you'll send the phone a reboot command. Aaron Joao Pereira wrote: I dont have this cisco-check-cfg exten command in my asterisk... Did you installed some extra module or channel? Thanks Joao Pereira Aaron Daniel wrote: It really depends on the number of phones you're wanting to reboot. Whenever we do a reconfiguration of our phones, I have a script that runs that night that pulls all the names from the db that are cisco phones, and does a sip notify cisco-check-cfg exten in asterisk, which notifies the phone to reboot in 20 seconds if nothing interesting happens (phone call comes in... browsing the interface... stuff like that). In order for this to work, you have to put a file in the tftpboot folder called syncinfo.xml containing this: SYNCINFO IMAGE VERSION=* SYNC=0/ /SYNCINFO in order for the phones to actually reboot though. That's what we do anyway :) Aaron Joao Pereira wrote: Hello to all Does someone knows how to force the Cisco IP phones (7940 and 7960) to reboot weekly or monthly? I think this would be useful because sometimes we change the configuration settings in the TFTP, but the phone just check the TFTP when he restarts... Thanks João ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fake Ring Tone/Compile Addon
Kenige Ho wrote: Dear All, I am currently have this problem in which I am sending call out from the Zaptel TE405 to a VoIP gateway. But the problem that the call over to the VoIP Gateway will always have a fake ring tone. Can you please give some pointer how to fix this problem? Don't use the fake ring option to dial. This is the r option. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Double-ring tone
I upgraded my Cisco7960 to SIP 8-2 from 7-4. Everything seems ok, works fine. Except that when I make an outbound call, I get a double-ring sound. I also found that if the target number is engaged, I get a ring sound and at the same time get a busy sound. If I revert back to 7-4, there is no problem. Anyone else had this, or any clues on how to fix it ? All of our other phones are still on 7-4. TIA. Julian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OSHA requirement to reach a live human ??
On Wednesday 15 March 2006 11:20, Steve Jones wrote: Are you selling it TO osha? If so, maybe they have an internal requirement.. If not, I've never heard of that. Granted, I haven't sold a LOT of phone systems, but I've been involved with a couple into public works departments of local governments as well as private corps, and nobody has ever mentioned that... And speaking as someone who CALLS public works and government agencies a lot, I'd have to say that my experience seems to indicate that this is most certainly NOT a requirement. It's impossible to reach a human easily; I generally just hit the first extension I can and ask to be transferred where I want. It only gets worse if there are queues to deal with, because you can't get to a human until your number comes up in the queue. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AVM C2 chan_capi-cm-0.6.4 Error on Dial
On Wed, 15 Mar 2006, John Daragon wrote: Armin Schindler wrote: On Wed, 15 Mar 2006, John Daragon wrote: Whoops ! sorry - wrong release ... chan_capi-cm-0.6.4 ! There were problems with 0.6.3 and AVM cards, but should be fixed in 0.6.4. Can you please create a full debug log (set verbose 5; capi debug) for such a case ? Certainly. Would you like it off-list ? Yes, I think that's better. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to show called name on calling polycom display
I was looking for this exactly as well Any ideas? - Gabe - Original Message - From: Giorgio Incantalupo [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 15, 2006 12:52 AM Subject: [Asterisk-Users] how to show called name on calling polycom display Hi, we have an asterisk 1.2.1 box and 2 polycom SIP phones. We'd like to show the called name on the calling polycom display instead of his /her extensions as I do with the caller name on the called polycom. Is it possible? If yes, how? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
