[asterisk-users] SkyHost is set to expire
Hi all, I'm actually using Asterisk 1.4.26 and Skype For Asterisk (last version). Everything was going fine, but yesterday I've got this messege when I've tried to restart asterisk (and skypeforasterisk): [Dec 28 15:18:08] NOTICE[2420] core.cpp: Found license 'S4A-XXX' providing 5 concurrent calls [Dec 28 15:18:08] NOTICE[2420] core.cpp: Skype For Asterisk Host-ID: X [Dec 28 15:18:08] NOTICE[2420] core.cpp: Found a total of 5 Skype For Asterisk licenses [Dec 28 15:18:08] DEBUG[2434] core.cpp: starting skyhost as: skypeforasterisk -z -f /var/spool/asterisk/skype/data -d /var/spool/asterisk/skype/skyhost-debug [Dec 28 15:18:08] DEBUG[2434] core.cpp: skyhost environment is : HOME=/var/spool/asterisk/skype [Dec 28 15:18:08] DEBUG[2433] core.cpp: starting skypewatcher as: skypewatcher 2434 [Dec 28 15:18:09] DEBUG[2432] core.cpp: got SkyHost Copyright (C) 2003-2008 Skype Technologies S.A. [Dec 28 15:18:09] DEBUG[2432] core.cpp: got Proprietary and confidential, do not share this application. [Dec 28 15:18:09] DEBUG[2432] core.cpp: got SkyHost is set to expire on Mon Dec 28 12:36:32 2009 [Dec 28 15:18:09] DEBUG[2432] core.cpp: got SkyHost has expired, please contact Skype Technologies S.A. to get a new development version. [Dec 28 15:18:09] DEBUG[2432] core.cpp: pfd 0 had an error [Dec 28 15:18:09] DEBUG[2432] core.cpp: sending SIGTERM to 2433. [Dec 28 15:18:12] VERBOSE[2436] logger.c: == Parsing '/etc/asterisk/manager.conf': [Dec 28 15:18:12] VERBOSE[2436] logger.c: Found [Dec 28 15:18:12] VERBOSE[2436] logger.c: == Parsing '/etc/asterisk/manager_additional.conf': [Dec 28 15:18:12] VERBOSE[2436] logger.c: Found [Dec 28 15:18:12] VERBOSE[2436] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': [Dec 28 15:18:12] VERBOSE[2436] logger.c: Found [Dec 28 15:18:12] VERBOSE[2436] logger.c: == Manager 'jabber' logged on from 127.0.0.1 [Dec 28 15:18:18] ERROR[2420] core.cpp: Skype engine failed to start. [Dec 28 15:18:18] ERROR[2420] chan_skype.c: Unable to start Skype For Asterisk library. Skyhost it seems to be expired and then "skypeforasterisk" and "skypewatcher" doesn't start at all. Anyone have some information about SkyHost expiration ? Thanks, Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] codecs and volume
Hi, Does using a different codec affect the volume of the voice? i was testing g711 and g729, voice seems to be softer on g729 compared to g711. sorry not really familiar on how codecs work. regards Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR is NO ANSWER when it should be ANSWERED
Hi, I'm having trouble with dialing out on analog lines. Asterisk can't seem to detect answers. I have two zap groups. Group 1 is connected to an external analog PSTN provider. This group seems to work fine, especially with answeronpolarityswitch. Group 2 is a group of GSM gateways, ie. devices that host SIM cards so that you can dial from any PBX extension to another phone via a GSM SIM card (supposedly, calls to cellular phones are cheaper this way). These devices act just like any other analog line. Every time I dial out through these analog GSM gateways the call gets through just fine. However, if the call lasts more than the timeout set on the Dial() application then it gets cut off. Also, if the call terminates, the CDR always records NO ANSWER even if it actually has been answered. During the call (answered and active), asterisk -rx show channels shows that the calls made out through group 2 are in Ring state. Any ideas on how I can detect answers on analog GSM gateways? example CDR: ,7070,6,outbound,TEST 7070,IAX2/mgmbox-1278,Zap/6-1,Dial,ZAP/g2/6|300|TWf,2009-12-29 08:30:00,,2009-12-29 08:30:10,10,0,NO ANSWER,DOCUMENTATION,1262075400.294, zapata.conf: [trunkgroups] [channels] language=es context=from-pstn switchtype=national signalling=fxs_ks rxwink=300 usecallerid=no cidsignalling=dtmf hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no rxgain=6.0 txgain=2.0 group=0 immediate=no answeronpolarityswitch=yes hanguponpolarityswitch=yes faxdetect=both channel = 1-2 group=1 channel = 3-4 group=2 channel = 5-9 group=3 channel = 10-12 answeronpolarityswitch=no hanguponpolarityswitch=no Thanks, Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off Topic: Aastra BLF limit...
Hi Carlos, It's simply not possible due to a firmware limitation when general SIP and not Aastra proprietary mode (not enougth memory capacity). Don't lack your time by searching a non exisiting solution. Best Regards, Francois -Message d'origine- De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]de la part de Carlos Chavez Envoyé : lundi 28 décembre 2009 21:00 À : Asterisk Objet : [asterisk-users] Off Topic: Aastra BLF limit... Hi. Does anyone have a patch or workaround for the 50 BLF limit of Aastra phones? I have a couple 57i with the 560M console and only the first 50 BLF lines get registered. I am using the latest firmware from Aastra but I read that this limit was imposed because of a memory leak. Obviously my customer is complaining about these last 10 lines not showing their status. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SkyHost is set to expire
Hi Daniel, Are you using a demo/beta version of Skype for Asterisk ? If yes, this status is normal, the demo/beta program is terminated from a while. I am using the real commercial (not free) and not getting that message. Best Regards, Francois -Message d'origine- De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]de la part de Daniel Grotti Envoyé : mardi 29 décembre 2009 09:19 À : Asterisk-user list Objet : [asterisk-users] SkyHost is set to expire Hi all, I'm actually using Asterisk 1.4.26 and Skype For Asterisk (last version). Everything was going fine, but yesterday I've got this messege when I've tried to restart asterisk (and skypeforasterisk): [Dec 28 15:18:08] NOTICE[2420] core.cpp: Found license 'S4A-XXX' providing 5 concurrent calls [Dec 28 15:18:08] NOTICE[2420] core.cpp: Skype For Asterisk Host-ID: X [Dec 28 15:18:08] NOTICE[2420] core.cpp: Found a total of 5 Skype For Asterisk licenses [Dec 28 15:18:08] DEBUG[2434] core.cpp: starting skyhost as: skypeforasterisk -z -f /var/spool/asterisk/skype/data -d /var/spool/asterisk/skype/skyhost-debug [Dec 28 15:18:08] DEBUG[2434] core.cpp: skyhost environment is : HOME=/var/spool/asterisk/skype [Dec 28 15:18:08] DEBUG[2433] core.cpp: starting skypewatcher as: skypewatcher 2434 [Dec 28 15:18:09] DEBUG[2432] core.cpp: got SkyHost Copyright (C) 2003-2008 Skype Technologies S.A. [Dec 28 15:18:09] DEBUG[2432] core.cpp: got Proprietary and confidential, do not share this application. [Dec 28 15:18:09] DEBUG[2432] core.cpp: got SkyHost is set to expire on Mon Dec 28 12:36:32 2009 [Dec 28 15:18:09] DEBUG[2432] core.cpp: got SkyHost has expired, please contact Skype Technologies S.A. to get a new development version. [Dec 28 15:18:09] DEBUG[2432] core.cpp: pfd 0 had an error [Dec 28 15:18:09] DEBUG[2432] core.cpp: sending SIGTERM to 2433. [Dec 28 15:18:12] VERBOSE[2436] logger.c: == Parsing '/etc/asterisk/manager.conf': [Dec 28 15:18:12] VERBOSE[2436] logger.c: Found [Dec 28 15:18:12] VERBOSE[2436] logger.c: == Parsing '/etc/asterisk/manager_additional.conf': [Dec 28 15:18:12] VERBOSE[2436] logger.c: Found [Dec 28 15:18:12] VERBOSE[2436] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': [Dec 28 15:18:12] VERBOSE[2436] logger.c: Found [Dec 28 15:18:12] VERBOSE[2436] logger.c: == Manager 'jabber' logged on from 127.0.0.1 [Dec 28 15:18:18] ERROR[2420] core.cpp: Skype engine failed to start. [Dec 28 15:18:18] ERROR[2420] chan_skype.c: Unable to start Skype For Asterisk library. Skyhost it seems to be expired and then skypeforasterisk and skypewatcher doesn't start at all. Anyone have some information about SkyHost expiration ? Thanks, Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SkyHost is set to expire
Hi, no I'm using the real commercial once. I've installed it in November 2009. Regards, daniel F6HQZ ha scritto: Hi Daniel, Are you using a demo/beta version of Skype for Asterisk ? If yes, this status is normal, the demo/beta program is terminated from a while. I am using the real commercial (not free) and not getting that message. Best Regards, Francois -Message d'origine- *De :* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]*de la part de* Daniel Grotti *Envoyé :* mardi 29 décembre 2009 09:19 *À :* Asterisk-user list *Objet :* [asterisk-users] SkyHost is set to expire Hi all, I'm actually using Asterisk 1.4.26 and Skype For Asterisk (last version). Everything was going fine, but yesterday I've got this messege when I've tried to restart asterisk (and skypeforasterisk): [Dec 28 15:18:08] NOTICE[2420] core.cpp: Found license 'S4A-XXX' providing 5 concurrent calls [Dec 28 15:18:08] NOTICE[2420] core.cpp: Skype For Asterisk Host-ID: X [Dec 28 15:18:08] NOTICE[2420] core.cpp: Found a total of 5 Skype For Asterisk licenses [Dec 28 15:18:08] DEBUG[2434] core.cpp: starting skyhost as: skypeforasterisk -z -f /var/spool/asterisk/skype/data -d /var/spool/asterisk/skype/skyhost-debug [Dec 28 15:18:08] DEBUG[2434] core.cpp: skyhost environment is : HOME=/var/spool/asterisk/skype [Dec 28 15:18:08] DEBUG[2433] core.cpp: starting skypewatcher as: skypewatcher 2434 [Dec 28 15:18:09] DEBUG[2432] core.cpp: got SkyHost Copyright (C) 2003-2008 Skype Technologies S.A. [Dec 28 15:18:09] DEBUG[2432] core.cpp: got Proprietary and confidential, do not share this application. [Dec 28 15:18:09] DEBUG[2432] core.cpp: *got SkyHost is set to expire on Mon Dec 28 12:36:32 2009 *[Dec 28 15:18:09] DEBUG[2432] core.cpp: got SkyHost has expired, please contact Skype Technologies S.A. to get a new development version. [Dec 28 15:18:09] DEBUG[2432] core.cpp: pfd 0 had an error [Dec 28 15:18:09] DEBUG[2432] core.cpp: sending SIGTERM to 2433. [Dec 28 15:18:12] VERBOSE[2436] logger.c: == Parsing '/etc/asterisk/manager.conf': [Dec 28 15:18:12] VERBOSE[2436] logger.c: Found [Dec 28 15:18:12] VERBOSE[2436] logger.c: == Parsing '/etc/asterisk/manager_additional.conf': [Dec 28 15:18:12] VERBOSE[2436] logger.c: Found [Dec 28 15:18:12] VERBOSE[2436] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': [Dec 28 15:18:12] VERBOSE[2436] logger.c: Found [Dec 28 15:18:12] VERBOSE[2436] logger.c: == Manager 'jabber' logged on from 127.0.0.1 [Dec 28 15:18:18] ERROR[2420] core.cpp: Skype engine failed to start. [Dec 28 15:18:18] ERROR[2420] chan_skype.c: Unable to start Skype For Asterisk library. Skyhost it seems to be expired and then skypeforasterisk and skypewatcher doesn't start at all. Anyone have some information about SkyHost expiration ? Thanks, Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T.38 and Linksys SPA8000
Hello everyone. I'm trying to set up a SIP DID on a customer, which uses T.38 for faxing. Voice is working great, but I never configured anything using T.38 in Asterisk so I'm kinda lost. On my googling I found out that would be best letting the Linksys SPA8000 (for those that don't know, it's identical to the PAP2 but with 8 ports instead of 2) and the telco negotiate with each other. Tried using canreinvite=yes but I still get transmission errors. Here's the relevant parts of my sip.conf (the telco trunk and the extension connected to the fax machine). If anyone has a working, in production, setup similar to what I'm trying to do here and don't mind sharing how it was configured, I would appreciate it a lot. Many thanks in advance. [voxip] username=*** type=peer secret=*** port=5060 canreinvite=no insecure=port,invite host=gvt.com.br fromuser=* fromdomain=gvt.com.br dtmfmode=rfc2833 context=entrada-voxip disallow=all allow=g729 allow=alaw qualify=yes directrtpsetup=yes t38pt_udptl=yes t38pt_usertpsource=yes [9204] type=friend secret=** callerid= 9204 host=dynamic nat=no canreinvite=yes insecure=port,invite qualify=yes context=ddi callgroup=1 pickupgroup=1 call-limit=10 disallow=all allow=g729 allow=alaw t38pt_udptl=yes t38pt_usertpsource=yes Vinícius Fontes www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] set box IP from which send sip traffic
Hello, I've asterisk (asterisk 1.6.0.6) box with two network interfaces (two public IP: IP1 and IP2). SIP binds on 0.0.0.0 . Is it possible configure SIP peer/user to receive/send traffic from one of these IP? For example one client sends/receives traffic from IP1, other client send/receives traffic from IP2. Thanks -- Pagarbiai / Best Regards, Giedrius ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SkyHost is set to expire
Hi, I would contact customer support then. My licenses expire all 2029 (20 years after buying). regards, Philipp Daniel Grotti wrote: Hi, no I'm using the real commercial once. I've installed it in November 2009. Regards, daniel F6HQZ ha scritto: Hi Daniel, Are you using a demo/beta version of Skype for Asterisk ? If yes, this status is normal, the demo/beta program is terminated from a while. I am using the real commercial (not free) and not getting that message. Best Regards, Francois -Message d'origine- *De :* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]*de la part de* Daniel Grotti *Envoyé :* mardi 29 décembre 2009 09:19 *À :* Asterisk-user list *Objet :* [asterisk-users] SkyHost is set to expire Hi all, I'm actually using Asterisk 1.4.26 and Skype For Asterisk (last version). Everything was going fine, but yesterday I've got this messege when I've tried to restart asterisk (and skypeforasterisk): [Dec 28 15:18:08] NOTICE[2420] core.cpp: Found license 'S4A-XXX' providing 5 concurrent calls [Dec 28 15:18:08] NOTICE[2420] core.cpp: Skype For Asterisk Host-ID: X [Dec 28 15:18:08] NOTICE[2420] core.cpp: Found a total of 5 Skype For Asterisk licenses [Dec 28 15:18:08] DEBUG[2434] core.cpp: starting skyhost as: skypeforasterisk -z -f /var/spool/asterisk/skype/data -d /var/spool/asterisk/skype/skyhost-debug [Dec 28 15:18:08] DEBUG[2434] core.cpp: skyhost environment is : HOME=/var/spool/asterisk/skype [Dec 28 15:18:08] DEBUG[2433] core.cpp: starting skypewatcher as: skypewatcher 2434 [Dec 28 15:18:09] DEBUG[2432] core.cpp: got SkyHost Copyright (C) 2003-2008 Skype Technologies S.A. [Dec 28 15:18:09] DEBUG[2432] core.cpp: got Proprietary and confidential, do not share this application. [Dec 28 15:18:09] DEBUG[2432] core.cpp: *got SkyHost is set to expire on Mon Dec 28 12:36:32 2009 *[Dec 28 15:18:09] DEBUG[2432] core.cpp: got SkyHost has expired, please contact Skype Technologies S.A. to get a new development version. [Dec 28 15:18:09] DEBUG[2432] core.cpp: pfd 0 had an error [Dec 28 15:18:09] DEBUG[2432] core.cpp: sending SIGTERM to 2433. [Dec 28 15:18:12] VERBOSE[2436] logger.c: == Parsing '/etc/asterisk/manager.conf': [Dec 28 15:18:12] VERBOSE[2436] logger.c: Found [Dec 28 15:18:12] VERBOSE[2436] logger.c: == Parsing '/etc/asterisk/manager_additional.conf': [Dec 28 15:18:12] VERBOSE[2436] logger.c: Found [Dec 28 15:18:12] VERBOSE[2436] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': [Dec 28 15:18:12] VERBOSE[2436] logger.c: Found [Dec 28 15:18:12] VERBOSE[2436] logger.c: == Manager 'jabber' logged on from 127.0.0.1 [Dec 28 15:18:18] ERROR[2420] core.cpp: Skype engine failed to start. [Dec 28 15:18:18] ERROR[2420] chan_skype.c: Unable to start Skype For Asterisk library. Skyhost it seems to be expired and then skypeforasterisk and skypewatcher doesn't start at all. Anyone have some information about SkyHost expiration ? Thanks, Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cheap ip phone with auto-answer
On Mon, 28 Dec 2009, Tim Nelson wrote: - Leif Neland le...@neland.dk wrote: I want some cheap ip-phones with auto-answer, to work as paging system at dinnertime. Options, please. Leif I've had great luck using the BT201 phones from Grandstream for this purpose. In fact, this is the only situation where I still use Grandstream handsets. They support auto answer as well as auto answer by call info which allows you to auto answer based upon the SIP header in case you don't want *ALL* calls to be auto answered. I'll second that. I've a small local charity (really tight on funds) who have a couple of dozen BT201's as desk phones and as paging/intercom devices. They're also cheap enough to use as remote ringer bells, screwed to a wall above head height... The only thing to watch out for is the Grandstream power supplies *WILL* die on you, likely within 2 years. For every Grandstream device I've deployed in the last few years, the power supply died and needed to be replaced. Luckily, they're just simple 5v wall warts and easy to find. That too... Although the newer ones seem much more reliable than the older (bigger) ones. Maybe the newer UK ones are better? Failure rate seems to be 1-2% over a few 100 Grandstream devices I've installed over the past few years. I try to persuade clients to go down the PoE route for the GXP2000's, if possible. Gives me a box of spare PSU's if nothing else.. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR
If asterisk enters the answered state at any point in the call, then the call disposition becomes answered. Thank you and have a nice day, Anthony Francis From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Szasz Szabolcs Sent: Tuesday, December 29, 2009 12:24 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] CDR Hi, How does Asterisk CDR work? How can I have in CDR records calls without BYE message? I checked my wireshark traces and some calls has no BYE messages, but they appears in CDR as answered call. Thanks Szabolcs Szasz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ReceiveFAX G.711 + Realtime
Hello, We're trying to receive G.711 (aLaw) faxes on the asterisk and convert them to tif. With T.38, we have several issues, so we are trying to use G.711, since the gateway is located in the same LAN, so there's no bandwidth/packet-lose issue. We also use on the same Asterisk Real-Time process for the extensions.conf My question: Is the following syntax for disabling T.38 support correct? vm*CLI -- Executing Set(SIP/Proxy-, t38pt_udptl=no) vm*CLI -- Executing Set(SIP/Proxy-, SIP_CODEC=aLaw) vm*CLI -- Executing Answer(SIP/Proxy-, ) The aLaw Set command is taken into consideration, because the SDP of the OK that follows these lines includes only codec 8, but when the ReceiveFAX command is executed, Asterisk immediately sends a T.38 reINVITE: vm*CLI -- Executing ReceiveFAX(SIP/Proxy-, /var/spool/asterisk/fax/3/1261993891.0.tif) I must be doing something wrong, but I'm not sure what :-( Your help would be highly appreciated. Thanks, Andreas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SkyHost is set to expire
Hi, My license expires at 2029, but this isn't a license problem I think. I've downloaded my S4A from the following link: http://downloads.digium.com/pub/telephony/skypeforasterisk/asterisk-1.4/x86-32/skypeforasterisk-1.4_1.0.6-x86_32.tar.gz As Digium documentation says. Regards, daniel Philipp Kolmann ha scritto: Hi, I would contact customer support then. My licenses expire all 2029 (20 years after buying). regards, Philipp Daniel Grotti wrote: Hi, no I'm using the real commercial once. I've installed it in November 2009. Regards, daniel F6HQZ ha scritto: Hi Daniel, Are you using a demo/beta version of Skype for Asterisk ? If yes, this status is normal, the demo/beta program is terminated from a while. I am using the real commercial (not free) and not getting that message. Best Regards, Francois -Message d'origine- *De :* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]*de la part de* Daniel Grotti *Envoyé :* mardi 29 décembre 2009 09:19 *À :* Asterisk-user list *Objet :* [asterisk-users] SkyHost is set to expire Hi all, I'm actually using Asterisk 1.4.26 and Skype For Asterisk (last version). Everything was going fine, but yesterday I've got this messege when I've tried to restart asterisk (and skypeforasterisk): [Dec 28 15:18:08] NOTICE[2420] core.cpp: Found license 'S4A-XXX' providing 5 concurrent calls [Dec 28 15:18:08] NOTICE[2420] core.cpp: Skype For Asterisk Host-ID: X [Dec 28 15:18:08] NOTICE[2420] core.cpp: Found a total of 5 Skype For Asterisk licenses [Dec 28 15:18:08] DEBUG[2434] core.cpp: starting skyhost as: skypeforasterisk -z -f /var/spool/asterisk/skype/data -d /var/spool/asterisk/skype/skyhost-debug [Dec 28 15:18:08] DEBUG[2434] core.cpp: skyhost environment is : HOME=/var/spool/asterisk/skype [Dec 28 15:18:08] DEBUG[2433] core.cpp: starting skypewatcher as: skypewatcher 2434 [Dec 28 15:18:09] DEBUG[2432] core.cpp: got SkyHost Copyright (C) 2003-2008 Skype Technologies S.A. [Dec 28 15:18:09] DEBUG[2432] core.cpp: got Proprietary and confidential, do not share this application. [Dec 28 15:18:09] DEBUG[2432] core.cpp: *got SkyHost is set to expire on Mon Dec 28 12:36:32 2009 *[Dec 28 15:18:09] DEBUG[2432] core.cpp: got SkyHost has expired, please contact Skype Technologies S.A. to get a new development version. [Dec 28 15:18:09] DEBUG[2432] core.cpp: pfd 0 had an error [Dec 28 15:18:09] DEBUG[2432] core.cpp: sending SIGTERM to 2433. [Dec 28 15:18:12] VERBOSE[2436] logger.c: == Parsing '/etc/asterisk/manager.conf': [Dec 28 15:18:12] VERBOSE[2436] logger.c: Found [Dec 28 15:18:12] VERBOSE[2436] logger.c: == Parsing '/etc/asterisk/manager_additional.conf': [Dec 28 15:18:12] VERBOSE[2436] logger.c: Found [Dec 28 15:18:12] VERBOSE[2436] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': [Dec 28 15:18:12] VERBOSE[2436] logger.c: Found [Dec 28 15:18:12] VERBOSE[2436] logger.c: == Manager 'jabber' logged on from 127.0.0.1 [Dec 28 15:18:18] ERROR[2420] core.cpp: Skype engine failed to start. [Dec 28 15:18:18] ERROR[2420] chan_skype.c: Unable to start Skype For Asterisk library. Skyhost it seems to be expired and then skypeforasterisk and skypewatcher doesn't start at all. Anyone have some information about SkyHost expiration ? Thanks, Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ReceiveFAX G.711 + Realtime
Cyprus VoIP wrote: My question: Is the following syntax for disabling T.38 support correct? vm*CLI -- Executing Set(SIP/Proxy-, t38pt_udptl=no) vm*CLI -- Executing Set(SIP/Proxy-, SIP_CODEC=aLaw) vm*CLI -- Executing Answer(SIP/Proxy-, ) I have no idea where you got the idea that such a thing is possible... it's not. sip.conf settings for SIP endpoints are not channel variables, and cannot be modified from the dialplan unless the CHANNEL() dialplan function has been specifically extended to support them. If you don't want T.38 support for a SIP endpoint, don't configure it that way in sip.conf or in your Realtime SIP peers table. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SkyHost is set to expire
Daniel Grotti wrote: Hi, My license expires at 2029, but this isn't a license problem I think. I've downloaded my S4A from the following link: http://downloads.digium.com/pub/telephony/skypeforasterisk/asterisk-1.4/x86-32/skypeforasterisk-1.4_1.0.6-x86_32.tar.gz As Digium documentation says. [Dec 28 15:18:09] DEBUG[2432] core.cpp: *got SkyHost is set to expire on Mon Dec 28 12:36:32 2009 *[Dec 28 15:18:09] DEBUG[2432] core.cpp: got SkyHost has expired, please contact Skype Technologies S.A. to get a new development version. Are you sure you are running 1.0.6? I doubt that in a production version of the plugin, Digium will ship a development version of skyhost. I just unloaded the chan_skype.so and reloaded it and did not get this error. what does skype show version in asterisk cli say? It seems you have an old .so running. Philipp ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SkyHost is set to expire
Hi men, I am sure this is the demo version, not the correct actual licensed one. Fro mthe CLI, enter that : fax show version My Asterisk reply that : Fax For Asterisk Components: Applications: 1.6.1_1.0.15 Digium Fax T.38 Driver: 1.6.1_1.0.11 (optimized for c3_2_32) Digium Fax G.711 Driver: 1.6.1_1.0.11 (optimized for c3_2_32) Remove the res_fax_digium.so and res_fax.so and redo your download and place the two .so files into your /usr/lib/asterisk/modules directory. Then, return to the list the success ;-) Best Regards, Francois ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking at Asterisk for 8000sq/ft residential use
Leif Neland wrote: I can't believe anyone would use RJ-11 any more. You can multi-purpose RJ-45 jacks to work with POTS lines. Run everything down to a central panel and send pots over the jacks that you need to. That way if you decide you need/want to go IP in the future, you're all set. Darrick Better read this before recommending using RJ-45 jacks with RJ-11 line cords. The jacks gets damaged. Manufactures will not warranty them!!! http://www.patentstorm.us/patents/7125288/description.html Jonn You can get or make cords with RJ-45 in one end and RJ-11 in the other. http://www.connectworld.net/cgi-bin/dataw/L0531 7 *$5.16 Leif * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The point is that you should buy the right jacks for the application. Just remember that VIOP is only about 100k of bandwidth per call to a phone. If you are connecting a pc to that phone phone thats different. Jonn ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ReceiveFAX G.711 + Realtime
I have no idea where you got the idea that such a thing is possible...
it's not. sip.conf settings for SIP endpoints are not channel variables,
and cannot be modified from the dialplan unless the CHANNEL() dialplan
function has been specifically extended to support them.
I was actually HOPING that it was possible, while guessing it probably
isn't ;-), at least not like I did it.
If you don't
want T.38 support for a SIP endpoint, don't configure it that way in
sip.conf or in your Realtime SIP peers table.
I want this endpoint to support T.38, but what I actually want is to
check the initial INVITE's SDP and based on the IP address of the media,
make a dialplan rule to decide whether to use G.711 or to switch to
T.38. So, I found the CHANNEL() variables rtpdest and t38passthrough.
This is the dialplan real-time script I ran:
id,context,exten,priority,app,appdata
80,fax,aLaw,1,NoOp,${CHANNEL(t38passthrough)}
81,fax,aLaw,2,NoOp,${CHANNEL(rtpdest)}
82,fax,aLaw,3,Set,CHANNEL(t38passthrough)=0
83,fax,aLaw,4,GotoIf,$[${CHANNEL(rtpdest)}=xx.xxx.xxx.xxx]?5:T38,1
76,fax,aLaw,5,Set,SIP_CODEC=aLaw
77,fax,aLaw,6,Answer,
78,fax,aLaw,7,ReceiveFAX,${fax_filepath}/${UNIQUEID}.tif
79,fax,aLaw,8,Hangup,
The result was:
-- Executing NoOp(SIP/Proxy-0005, 0)
-- Executing NoOp(SIP/Proxy-0005, xx.xxx.xxx.xxx:60100)
-- Executing Set(SIP/Proxy-0005, CHANNEL(t38passthrough)=0)
[2009-12-29 16:20:19.772] WARNING[5888]: func_channel.c:161
func_channel_write: Unknown or unavailable item requested: 't38passthrough'
-- Executing GotoIf(SIP/Proxy-0005, 0?5:T38,1)
-- Goto (fax,T38,1)
Now, referring to the error above, I see (in voip-info.org) that
t38passthrough is an R/O variable and not an R/W, but in any case, I got
0 as a result, so it should have been OK, and it's not, as ReceiveFAX
still sends a T.38 reINVITE. If I can't modify it, what should I do?
Also, since the rtpdest includes also the port, how do I check in the
GotoIf if the value contains that IP and not equal to it (which it
can't be)? It seems that this will always return 0:
$[${CHANNEL(rtpdest)}=xx.xxx.xxx.xxx]
Thanks.
___
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[asterisk-users] asterisk 1.6.2.0 sip channel to sip channel call dtmf inband not work.
we tested asterisk 1.6.2.0, found that
when call from one sip_channel to another sip_channel ,
--
exten = _X.,1,Noop()
exten = _X.,n,Dial(SIP/${EXTEN},50,TtgM)
--
in asterisk 1.6.2.0 ,when sip user config to use
dtmfmode=rfc2833 , it's ok, but when both users
config to use dtmfmode=inband, cannot detect the
dtmf and trigger the feature config.
but in asterisk 1.4.26 or above. it work well.
it is a bug in 1.6.2.0 ?
--
zhao ___
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Re: [asterisk-users] Does A2Billing has mial list?
Hi Sucan, A2Billing doesn't have a mailing list but you may ask your specific question on A2billing Forum or maybe even here. This may be of intrest to you if you have an installation question: A2Billing automated install script : http://a2billing2asterisk.googlepages.com http://a2billing2asterisk.googlepages.com-Bruce On Tue, Dec 29, 2009 at 1:07 AM, Zhang Shukun bit...@gmail.com wrote: hi, Does A2Billing has mial list? -- Thanks, Sucan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Issue
You should set the ddwhome variable with the Set function or declare it on the global context. Try the Dial command with the dial string directly, before using the variable. Fro debugging purposes you should set debug and verbose at least to 10 and check the logs. Regards, Juan James A. Shigley wrote: What do you mean I should use a global function. I'm kind both well versed and a newb to * James Shigley -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan E. Rodríguez Sent: Monday, December 28, 2009 12:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Issue Is ddwhome defined in global context?? If so, then you should use global function. Paste asterisk log to check. Saludos, Juan E. Rodríguez -Original Message- From: James A. Shigley j...@answeringserv.com Date: Mon, 28 Dec 2009 12:11:35 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: [asterisk-users] SIP Issue ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does A2Billing has mial list?
A2billing forum has a lot of information and questions are answered very fast. Try searching on the forum before posting, cause the answer may be there already. forum.asterisk2billing.org/ Regards, Juan Bruce Nik wrote: Hi Sucan, A2Billing doesn't have a mailing list but you may ask your specific question on A2billing Forum or maybe even here. This may be of intrest to you if you have an installation question: A2Billing automated install script : http://a2billing2asterisk.googlepages.com -Bruce On Tue, Dec 29, 2009 at 1:07 AM, Zhang Shukun bit...@gmail.com wrote: hi, Does A2Billing has mial list? -- Thanks, Sucan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any good dialplan code out there to implementvertical service codes?
It depends on what flavor of Asterisk and trunks (SIP/Zapata/DAHDI) you are using. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joe Freeman Sent: Monday, December 28, 2009 8:14 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Any good dialplan code out there to implementvertical service codes? Greetings- I'm in the process of turning up an Asterisk box for a customer and was wondering if anyone had any good code they could share for implementing vertical service codes within Asterisk. I'd really rather not have to spend hours making new wheels if someone has one or more that will fit. One of my issues is that I've had a very hard time finding out exactly which VSC's are already implemented in what parts of Asterisk. This system will be for the most part an IP Centrex platform for this customer, who will be selling services to his end-users. I've got a pretty good line on how I'm going to handle personal LD pin codes using ODBC, but if anyone has any code for that I might get some pointers from, I'd appreciate that too. Any thoughts or suggestions are greatly appreciated. Thanks- Joe ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ReceiveFAX G.711 + Realtime
Now, referring to the error above, I see (in voip-info.org) that
t38passthrough is an R/O variable and not an R/W, but in any case, I got
0 as a result, so it should have been OK, and it's not, as ReceiveFAX
still sends a T.38 reINVITE. If I can't modify it, what should I do?
For the testing, I set the peer's t38pt_udptl to no and on the
originating gateway, left only aLaw enabled. If/when I set it back to
yes, it sent the reINVITE, so I don't have a solution for that yet.
Also, since the rtpdest includes also the port, how do I check in the
GotoIf if the value contains that IP and not equal to it (which it
can't be)? It seems that this will always return 0:
$[${CHANNEL(rtpdest)}=xx.xxx.xxx.xxx]
I changed the GotoIf command to
$[${CHANNEL(rtpdest):0:14}=xx.xxx.xxx.xxx]?5:T38,1 as I don't care about
the length of the other originating IP addresses (the specific one I
check is always 14 chars long), and now it stays in in aLaw context, but
now I get this error:
app_fax.c:292 fax_generator_generate: Only generating 240 samples, where
320 requested
I see that in app_fax.c MAX_SAMPLES is set to 240 and it doesn't seem to
accept config file values. I reduced the packet length from 40ms (320
samples) to 20ms (160 samples) for G.711 codecs and it solved this
problem. Does this mean that Asterisk supports maximum 30ms packets in
G.711 Fax?
Thanks.
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Re: [asterisk-users] ReceiveFAX G.711 + Realtime
Cyprus VoIP wrote: Now, referring to the error above, I see (in voip-info.org) that t38passthrough is an R/O variable and not an R/W, but in any case, I got 0 as a result, so it should have been OK, and it's not, as ReceiveFAX still sends a T.38 reINVITE. If I can't modify it, what should I do? There is no method currently available in Asterisk to do what you want; if the SIP endpoint that the channel is connected to is configured to support T.38, it will be used. The only suggestion I can offer to is ask someone to enhance app_fax to allow T.38 usage to be disabled via an argument to the application; that should be fairly easy to do. If you are using Fax For Asterisk, you'd need to contact Digium's support department and have them enter a feature request for this to be added. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any good dialplan code out there to implementvertical service codes?
At the moment, 1.6.0.20 realtime with Dahdi 2.2, TDM is a TE420, but that won't be customer facing. Thanks- Joe Danny Nicholas wrote: It depends on what flavor of Asterisk and trunks (SIP/Zapata/DAHDI) you are using. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joe Freeman Sent: Monday, December 28, 2009 8:14 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Any good dialplan code out there to implementvertical service codes? Greetings- I'm in the process of turning up an Asterisk box for a customer and was wondering if anyone had any good code they could share for implementing vertical service codes within Asterisk. I'd really rather not have to spend hours making new wheels if someone has one or more that will fit. One of my issues is that I've had a very hard time finding out exactly which VSC's are already implemented in what parts of Asterisk. This system will be for the most part an IP Centrex platform for this customer, who will be selling services to his end-users. I've got a pretty good line on how I'm going to handle personal LD pin codes using ODBC, but if anyone has any code for that I might get some pointers from, I'd appreciate that too. Any thoughts or suggestions are greatly appreciated. Thanks- Joe ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk billing transferred calls
Hi, I'm looking for an application to show all the calls received/made including (this is very important!) transferred calls because I need to track all the time spent on the phone by all my employees. There is a list here but they are too many to try them all: http://www.voip-info.org/wiki/view/Asterisk+billing :) Any suggestions? Thank you Giorgio ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SkyHost is set to expire
Daniel- no I'm using the real commercial once. I've installed it in November 2009. Did you have the demo version installed before the commercial version? I.e. install the commercial over the top of the demo version? -Jeff F6HQZ ha scritto: Hi Daniel, Are you using a demo/beta version of Skype for Asterisk ? If yes, this status is normal, the demo/beta program is terminated from a while. I am using the real commercial (not free) and not getting that message. Best Regards, Francois -Message d'origine- *De :* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]*de la part de* Daniel Grotti *Envoyé :* mardi 29 décembre 2009 09:19 *À :* Asterisk-user list *Objet :* [asterisk-users] SkyHost is set to expire Hi all, I'm actually using Asterisk 1.4.26 and Skype For Asterisk (last version). Everything was going fine, but yesterday I've got this messege when I've tried to restart asterisk (and skypeforasterisk): [Dec 28 15:18:08] NOTICE[2420] core.cpp: Found license 'S4A-XXX' providing 5 concurrent calls [Dec 28 15:18:08] NOTICE[2420] core.cpp: Skype For Asterisk Host-ID: X [Dec 28 15:18:08] NOTICE[2420] core.cpp: Found a total of 5 Skype For Asterisk licenses [Dec 28 15:18:08] DEBUG[2434] core.cpp: starting skyhost as: skypeforasterisk -z -f /var/spool/asterisk/skype/data -d /var/spool/asterisk/skype/skyhost-debug [Dec 28 15:18:08] DEBUG[2434] core.cpp: skyhost environment is : HOME=/var/spool/asterisk/skype [Dec 28 15:18:08] DEBUG[2433] core.cpp: starting skypewatcher as: skypewatcher 2434 [Dec 28 15:18:09] DEBUG[2432] core.cpp: got SkyHost Copyright (C) 2003-2008 Skype Technologies S.A. [Dec 28 15:18:09] DEBUG[2432] core.cpp: got Proprietary and confidential, do not share this application. [Dec 28 15:18:09] DEBUG[2432] core.cpp: *got SkyHost is set to expire on Mon Dec 28 12:36:32 2009 *[Dec 28 15:18:09] DEBUG[2432] core.cpp: got SkyHost has expired, please contact Skype Technologies S.A. to get a new development version. [Dec 28 15:18:09] DEBUG[2432] core.cpp: pfd 0 had an error [Dec 28 15:18:09] DEBUG[2432] core.cpp: sending SIGTERM to 2433. [Dec 28 15:18:12] VERBOSE[2436] logger.c: == Parsing '/etc/asterisk/manager.conf': [Dec 28 15:18:12] VERBOSE[2436] logger.c: Found [Dec 28 15:18:12] VERBOSE[2436] logger.c: == Parsing '/etc/asterisk/manager_additional.conf': [Dec 28 15:18:12] VERBOSE[2436] logger.c: Found [Dec 28 15:18:12] VERBOSE[2436] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': [Dec 28 15:18:12] VERBOSE[2436] logger.c: Found [Dec 28 15:18:12] VERBOSE[2436] logger.c: == Manager 'jabber' logged on from 127.0.0.1 [Dec 28 15:18:18] ERROR[2420] core.cpp: Skype engine failed to start. [Dec 28 15:18:18] ERROR[2420] chan_skype.c: Unable to start Skype For Asterisk library. Skyhost it seems to be expired and then skypeforasterisk and skypewatcher doesn't start at all. Anyone have some information about SkyHost expiration ? Thanks, Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime mysql extensions mutiple queries for each priority?
On Monday 28 December 2009 23:49:13 Tilghman Lesher wrote:
On Monday 28 December 2009 18:09:15 JR Richardson wrote:
I turned on console debug to see the actual mysql queries and to my
surprise and concern, I see every query for an extension priority
repeated 3 or more times prior to dialplan execution. For instance my
first dialplan activity is all extracted from the database:
context exten pri app appdata
dpdefault14 _991X 1 NoOp INBOUND CALL FROM SIPP
dpdefault14 _991X 2 NoOp TRUNK-${EXTEN:0:2} DID-${EXTEN:2}
dpdefault14 _991X 3 Set CALLERID(number)=600
dpdefault14 _991X 4 Answer
dpdefault14 _991X 5 Goto ${EXTEN:2}|1
Each priority is queried several times before executing. Here is a
sample of the first 2 priorities on a pastebin:
http://pastebin.com/m54c9c41e
I would not think this is normal activity as I can query the database
directly once and get a valid response. I don't have any realtime
mysql connections issues that I can see, no errors in the logs and
console status is:
No, that's normal. The order of queries done is 1) check if the extension
exists, 2) on spawn, retrieve the extension to populate information about
the application into the channel structure, and 3) actually execute the
application. There are 3 queries done for each extension actually executed
in order of priorities and a few more when the extension changes (or
originates). It's not optimal, but it's the way that it works.
At some point, a slight optimization could certainly be done to narrow this
down to a single query from the database, followed by a fairly short
caching period (1 second would be plenty), but that optimization has never
been done.
https://issues.asterisk.org/view.php?id=16521
Needs testing.
--
Tilghman Lesher
Tilghman,
Saying I’m a bit excited right now is an understatement. First of all,
the patch seems to work fine applied to 1.4.28 stable release. The
performance of this patch is extraordinary. Before migrating my static
dialplan to the database I could push 380 calls at 15 to 20 CPS. After
migrating to the database, I could only push a little more than 100
calls and no more than 6 to 9 CPS. With this patch applied, I am
pushing reliably 300 calls at 15 CPS. 7500+ calls without a hiccup.
Not quite as good as a static dialplan, but that is expected. MySQLd
has also decreased utilization, as expected, from 6 to 12, now 1 to 6.
This has got to be an overall performance increase by 50% or more.
I will be patching on my new 1.4 systems going forward. The sooner
this patch gets applied to Asterisk, the better.
Thanks.
JR
--
JR Richardson
Engineering for the Masses
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Re: [asterisk-users] identifying channel for softhangup
On 12/29/2009 1:01 AM, Jeremy Kister wrote: e.g., in the first call, below, the channel name is SIP/vgw1-0075 -- the second call (on the same FXO port after a soft hangup on the CLI) is SIP/vgw1-0077 How can I extract this information in the dialplan so that I can use the SoftHangup app in asterisk to disrupt an existing call ? can anyone think of a different mailing list which might have members who know the answer i'm looking for? asterisk-dev ? -- Jeremy Kister http://jeremy.kister.net./ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] identifying channel for softhangup
Most of the asterisk-dev members read this discussion (In My Experience).
${EXTEN} in the case you state would be SIP/vgw1-0075.
Perhaps this link would be helpful
http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeremy Kister
Sent: Tuesday, December 29, 2009 2:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] identifying channel for softhangup
On 12/29/2009 1:01 AM, Jeremy Kister wrote:
e.g., in the first call, below, the channel name is
SIP/vgw1-0075 -- the second call (on the same FXO port after a
soft hangup on the CLI) is SIP/vgw1-0077
How can I extract this information in the dialplan so that I can use
the SoftHangup app in asterisk to disrupt an existing call ?
can anyone think of a different mailing list which might have members
who know the answer i'm looking for? asterisk-dev ?
--
Jeremy Kister
http://jeremy.kister.net./
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Re: [asterisk-users] identifying channel for softhangup
On 12/29/2009 3:23 PM, Danny Nicholas wrote:
Most of the asterisk-dev members read this discussion (In My Experience).
${EXTEN} in the case you state would be SIP/vgw1-0075.
Perhaps this link would be helpful
http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List
Thanks for the reply,
But I think ${EXTEN} would be in the channel name space - i need
something in the global name space that can let me identify the channel.
I'm trying to set up a 911 system just like Christian Hoffmeyer's
example at http://www.voip-info.org/wiki/view/Asterisk+tips+911
So, someone could be occupying my single land line with a
non-emergency phone call. this single land-line is connected to my
Cisco 1760V on FXO port 3/0. I simply am looking for something so
that if anyone dials 911, the first thing that happens is that a
SoftHangup(SIP/vgw1-XXX) is executed, and then the call goes out the
landline. then if a second 911 call goes out, then it goes out over sip.
I have all that working, except the SoftHangup -- because the channel
name is not static. So I need to look it up somehow on the fly, or
configure the channel name to be static/predictable.
--
Jeremy Kister
http://jeremy.kister.net./
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Re: [asterisk-users] identifying channel for softhangup
You could do a System(core show channels) and grep out 911 and kill
everything else; probably easier as an AGI call that a dialplan function,
but both can be done.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeremy Kister
Sent: Tuesday, December 29, 2009 2:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] identifying channel for softhangup
On 12/29/2009 3:23 PM, Danny Nicholas wrote:
Most of the asterisk-dev members read this discussion (In My Experience).
${EXTEN} in the case you state would be SIP/vgw1-0075.
Perhaps this link would be helpful
http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List
Thanks for the reply,
But I think ${EXTEN} would be in the channel name space - i need
something in the global name space that can let me identify the channel.
I'm trying to set up a 911 system just like Christian Hoffmeyer's
example at http://www.voip-info.org/wiki/view/Asterisk+tips+911
So, someone could be occupying my single land line with a
non-emergency phone call. this single land-line is connected to my
Cisco 1760V on FXO port 3/0. I simply am looking for something so
that if anyone dials 911, the first thing that happens is that a
SoftHangup(SIP/vgw1-XXX) is executed, and then the call goes out the
landline. then if a second 911 call goes out, then it goes out over sip.
I have all that working, except the SoftHangup -- because the channel
name is not static. So I need to look it up somehow on the fly, or
configure the channel name to be static/predictable.
--
Jeremy Kister
http://jeremy.kister.net./
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Re: [asterisk-users] identifying channel for softhangup
On 12/29/2009 3:54 PM, Danny Nicholas wrote: You could do a System(core show channels) and grep out 911 and kill everything else; probably easier as an AGI call that a dialplan function, but both can be done. great idea; thanks! -- Jeremy Kister http://jeremy.kister.net./ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] identifying channel for softhangup
Un-top-posting...
On 12/29/2009 1:01 AM, Jeremy Kister wrote:
e.g., in the first call, below, the channel name is SIP/vgw1-0075
-- the second call (on the same FXO port after a soft hangup on the
CLI) is SIP/vgw1-0077
How can I extract this information in the dialplan so that I can use
the SoftHangup app in asterisk to disrupt an existing call ?
can anyone think of a different mailing list which might have members
who know the answer i'm looking for? asterisk-dev ?
On Tue, 29 Dec 2009, Danny Nicholas wrote:
${EXTEN} in the case you state would be SIP/vgw1-0075.
I think you meant ${CHANNEL}.
Perhaps this link would be helpful
http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List
I'm trying to think what it is that you are trying to accomplish (and
why).
I'm guessing it's something like My ISP only allows 1 call so if there is
a call already in progress, I want to terminate the other call so I can
place my call.
I'm thinking along the lines of:
exten = *,n,softhangup(${CHANNEL-USING-MY-ISP})
exten = *,n,setglobalvar(CHANNEL-USING-MY-ISP=${CHANNEL})
exten = *,n,dial(...)
Softhangup() doesn't object to using an invalid string, so you don't need
to check the global variable before using.
Unless you want to get into a pissing match with your other user, you'll
probably want to add some more code so they can't blow you off.
You may find pages on voip-info.org relating to gotoif and execif useful.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] Context Switches and Load Average spike - Asterisk Version 1.4.22
I am running Asterisk V 1.4.22 Twice during the last two days the Context Switches on our box has gone from about 7K to 80K in 2.5 hours. The load average would spike to 17, drop to 0.35 then spike again. When connecting to the console 'core show channels' will list the channels but not total calls. 'restart now' had no effect, the only way to stop Asterisk is to kill the process. Once Asterisk is killed, everything returned to normal, for about 20 hours, then it started again. The server is a dual - quad core machine. Linux has been up over 380 days. Has anyone experienced this before? -- -Thermal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] identifying channel for softhangup
On 12/29/2009 3:54 PM, Danny Nicholas wrote:
You could do a System(core show channels) and grep out 911 and kill
everything else; probably easier as an AGI call that a dialplan function,
but both can be done.
my end result just feels ugly. the loop is due to the fact that I
have more than one FXO port on vgw1 and I cant identify which is FXO
port 0 (my land line).
anyone have anything better?
[nineoneone]
exten = s,1,Set(SET_EMERG_FLAG=0)
exten = s,n,Set(CALLERID(num)=${EMERGENCY_FROM})
exten = s,n,GotoIf($[${EMERGENCY} = 1]?lastresort,1)
exten = s,n,Set(EMERGENCY=1,g)
exten = s,n,Set(SET_EMERG_FLAG=1)
exten = s,n(kill),Set(CHAN=${SHELL(asterisk -rx core show channels
concise | awk -F! '/^SIP\/vgw1-/ { print $1 }' | head -1)})
exten = s,n,GotoIf($[${CHAN} = ]?dial)
exten = s,n,SoftHangup(${CHAN})
exten = s,n,Goto(kill)
exten = s,n(dial),Wait(3)
exten =
s,n,Dial(SIP/${emergency_trunk}${emergency_n...@${emergency_host})
exten = s,n,GotoIf($[${DIALSTATUS} != ANSWER ${DIALSTATUS} !=
CANCEL]?lastresort,1)
exten = h,1,GotoIf($[${SET_EMERG_FLAG} = 1]?unset,1)
exten = unset,1,Set(EMERGENCY=0,g)
exten = unset,n,Set(SET_EMERG_FLAG=0)
exten = lastresort,1,Macro(SaferSIPDial,${EMERGENCY_NUM})
--
Jeremy Kister
http://jeremy.kister.net./
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Re: [asterisk-users] Looking at Asterisk for 8000sq/ft residential use
Before I start I am a Panasonic certified dealer AND I have installed over 100 Asterisk systems that are in production. That said for your application use Panasonic, DONT use Asterisk. Use the Panasonic KX-TDA50G. Supports up to around 50 ports. In addition to Analog and their proprietary Digital phones they support: * SIP * DECT Wireless with their unique Wireless CellSites (so there are no dead spots). * Door boxes with door realease * External relays * CTI software * Built in MOH port * Built in external paging port Dont use the TAW848 as it uses analog proprietary phones which in addition of having just 12 programmable buttons it must have 2 pairs to work. Its options are also limited, as well as it's been discontinued. On Mon, Dec 28, 2009 at 5:13 PM, Rick Huebner r...@rhuebner.com wrote: My brother-in-law is finishing up his McMansion and I've done all of the low voltage wiring and am starting the trimout. We are batting around what to do for a phone system and I'm torn between a Panasonic TAW824/TVA50 and using an Asterisk implementation. I'm very strong on the networking/linux/basic hacking(old school, not criminal) side. I've downloaded the Asterisk VM and have some implentation questions before we make a decision. Of course we are running out of time because I need to order either RJ-11 or RJ-45 keystones for the plates to finish the trim out. We have Cat5e run everywhere so that won't be a limiting factor. Basic info: 8000sq/ft under air, 11,000sq/ft under the roof 17 phone handset outlets 15 phone jacks for potential use behind TVs 2 fax lines 1 alarm line 3 voice POTS lines 1 fax POTS line Pentium 4 old Dell with 1gig RAM to use with Asterisk if selected Requirements 1. Page over all handsets in intercom mode. They have kids and want to be able to yell over the phone if needed to find someone. Not done easily with Asterisk from ANY phone, unless you plan on putting some expensive IP Phone with a gozillion buttons or plan on teaching the kids 2-3 digit codes. 2. Easily call from room to room. Speed dial buttons would be ideal. Again only expensive IP phones have that. 3. Multiple voice line support for the office phones. 4. Unique ring tones on the phones for internal calls versus external so you can tell by listening if it is inside or outside. 5. If possible, unique ring tones for the various external lines in the offices. Panasonic will do all that. Looking for any suggestions as I need to get the keystones ordered ASAP. Thanks Rick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking at Asterisk for 8000sq/ft residential use
On Mon, Dec 28, 2009 at 5:42 PM, John Novack jnov...@stromberg-carlson.org wrote: Rick Huebner wrote: My brother-in-law is finishing up his McMansion and I've done all of the low voltage wiring and am starting the trimout. We are batting around what to do for a phone system and I'm torn between a Panasonic TAW824/TVA50 and using an Asterisk implementation. I'm very strong on the networking/linux/basic hacking(old school, not criminal) side. I've downloaded the Asterisk VM and have some implentation questions before we make a decision. Of course we are running out of time because I need to order either RJ-11 or RJ-45 keystones for the plates to finish the trim out. You can use the 8 position modular jacks regardless ( misnamed RJ-45 ) so that should not stop you from finishing the trim out. You did remember the door boxes, didn't you? Panasonic systems have that covered, even with openers if desired. The Panasonic systems I have used over the last 20 years are rugged, hang on the wall, connect with proper protection and forget them for years on end. They all have had dual ports that will either use a POTS single line phone, or one of their multibutton phones without any rewiring, reprogramming, and many even support one of each per port. An ideal system for a large house. I assume in the US? The TVS-50 isn't much of a VM system though for a house a two port box is probably OK, but NG for even a home business application. Not familiar with the model number you mentioned. Was that a typo or a new system? It's TVA-50 now, quite good, does the job for most offices. Surely does it for homes. Although many will disagree, for most users Panasonic systems with normal requirements work well for long periods with no problems and have lots of features. For the geek who wants to play, drive the rest of the family nuts changing things, then consider Asterisk. John Novack We have Cat5e run everywhere so that won't be a limiting factor. Basic info: 8000sq/ft under air, 11,000sq/ft under the roof 17 phone handset outlets 15 phone jacks for potential use behind TVs 2 fax lines 1 alarm line 3 voice POTS lines 1 fax POTS line Pentium 4 old Dell with 1gig RAM to use with Asterisk if selected Requirements 1. Page over all handsets in intercom mode. They have kids and want to be able to yell over the phone if needed to find someone. 2. Easily call from room to room. Speed dial buttons would be ideal. 3. Multiple voice line support for the office phones. 4. Unique ring tones on the phones for internal calls versus external so you can tell by listening if it is inside or outside. 5. If possible, unique ring tones for the various external lines in the offices. Looking for any suggestions as I need to get the keystones ordered ASAP. Thanks Rick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Checked by AVG - www.avg.com Version: 9.0.722 / Virus Database: 270.14.121/2589 - Release Date: 12/27/09 04:18:00 -- Dog is my co-pilot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking at Asterisk for 8000sq/ft residential use
On Mon, Dec 28, 2009 at 11:45 PM, Doug d...@natel.net wrote: At 16:13 12/28/2009, Rick Huebner wrote: My brother-in-law is finishing up his McMansion and I've done all of the low voltage wiring and am starting the trimout. We are batting around what to do for a phone system and I'm torn between a Panasonic TAW824/TVA50 and using an Asterisk implementation. I'm very strong on the networking/linux/basic hacking(old school, not criminal) side. I've downloaded the Asterisk VM and have some implentation questions before we make a decision. Of course we are running out of time because I need to order either RJ-11 or RJ-45 keystones for the plates to finish the trim out. We have Cat5e run everywhere so that won't be a limiting factor. It won't be too long before we start kicking ourselves because we didn't string 10 gbit fiber in our new construction. Basic info: 8000sq/ft under air, 11,000sq/ft under the roof 17 phone handset outlets 15 phone jacks for potential use behind TVs 2 fax lines 1 alarm line 3 voice POTS lines 1 fax POTS line How many phones do you think you will have? How many simultaneous calls? FAX, alarm, modem, credit card, postage meter, usually need to be hard (POTS) lines. The rest can be VOIP. Pentium 4 old Dell with 1gig RAM to use with Asterisk if selected New hardware is so cheap, it might be safer to buy new rather than wondering if the problem is hardware. Requirements 1. Page over all handsets in intercom mode. They have kids and want to be able to yell over the phone if needed to find someone. 2. Easily call from room to room. Speed dial buttons would be ideal. 3. Multiple voice line support for the office phones. 4. Unique ring tones on the phones for internal calls versus external so you can tell by listening if it is inside or outside. 5. If possible, unique ring tones for the various external lines in the offices. PBX in a Flash is the simplest Asterisk to set up: http://PBXinaFlash.net/ The Polycom 601 will let you know the status of 6 other phones, more with the sidecar: of 5 other people not 6, as one button has to be the phone itself. http://images.google.com/images?q=Polycom+601 Have set up lots of these systems. You'll like it a lot. Looking for any suggestions as I need to get the keystones ordered ASAP. Thanks Rick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking at Asterisk for 8000sq/ft residential use
On Tue, Dec 29, 2009 at 11:30:21PM -0500, C F wrote: Before I start I am a Panasonic certified dealer AND I have installed over 100 Asterisk systems that are in production. That said for your application use Panasonic, DONT use Asterisk. Use the Panasonic KX-TDA50G. Supports up to around 50 ports. I initial started the email to point out that this is a non commercial project. But after a quick google, this looks like a nice unit for around the US$600 mark (101Phones.com) In addition to Analog and their proprietary Digital phones they support: * SIP * DECT Wireless with their unique Wireless CellSites (so there are no dead spots). * Door boxes with door realease * External relays * CTI software * Built in MOH port * Built in external paging port Dont use the TAW848 as it uses analog proprietary phones which in addition of having just 12 programmable buttons it must have 2 pairs to work. Its options are also limited, as well as it's been discontinued. [snip] signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Auto-provisioining Polycom 430 wth dd-wrt router
Hi all, I'm trying to use a wrt54gl router running dd-wrt as a provisioning server for a remote installation. I've got dhcp working and I have provisioning files ready to go. I understand that I need to set bootp option 66 to point to the tftp/ftp/http server. In fact, I have this working completely with the ISC dhcp server The problem is that I don't know how to get dd-wrt's dhcp server to send the right string to the phone. With ISC dhcpd, I used: option boot-server ftp://user:pas...@10.0.1.1;; option tftp-server-name ftp://user:pas...@10.0.1.1;; But what is the equivelent configuration for dnsmasq? If I can't get this working, I'll have to resort to hard-coding the information into each of 12 phones Yuck! -- Take care and have fun, Mike Diehl. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Skype for Asterisk
Hi Sir, We have integrated Skype with Asterisk (skype user id:- rexesbposolutions). Each call which is coming to skype account is getting transfered to Asterisk Queue. It has following two cases: case 1: When we call from normal skype account to skype account (rexesbposolutions), everything is working fine. case 2: This skype account (rexesbposolutions) has been assigned with a online virtual number (00 44 20 ). If somebody dial this number from their landline/cellphone, call is transfered to Asterisk queue but it shows some problem related to G729 codecs. following are Asterisk CLI log: Executing [...@skypeincoming:1] Answer(Skype/rexesbposolutions-084159e8, ) in new stack -- Executing [...@skypeincoming:2] Wait(Skype/rexesbposolutions-084159e8, 5) in new stack -- Executing [...@skypeincoming:3] GotoIfTime(Skype/rexesbposolutions-084159e8, 9:00-18:00|mon-fri|*| *?sky|s|1) in new stack -- Goto (sky,s,1) -- Executing [...@sky:1] Playback(Skype/rexesbposolutions-084159e8, enter) in new stack -- Skype/rexesbposolutions-084159e8 Playing 'enter' (language 'en') [Dec 29 17:27:14] WARNING[4997]: channel.c:3106 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x4 (ulaw) [Dec 29 17:27:14] WARNING[4997]: file.c:147 ast_stopstream: Unable to restore format back to 4 -- Executing [...@sky:2] Queue(Skype/rexesbposolutions-084159e8, markq|t|||900) in new stack -- Started music on hold, class 'default', on Skype/rexesbposolutions-084159e8 [Dec 29 17:27:14] WARNING[4997]: channel.c:3106 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x40 (slin) [Dec 29 17:27:14] WARNING[4997]: res_musiconhold.c:251 ast_moh_files_next: Unable to open file '/var/lib/asterisk/moh/manolo_camp-morning_coffee': No such file or directory -- Stopped music on hold on Skype/rexesbposolutions-084159e8 [Dec 29 17:27:14] WARNING[4997]: channel.c:3106 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm) [Dec 29 17:27:14] WARNING[4997]: res_musiconhold.c:204 moh_files_release: Unable to restore channel 'Skype/rexesbposolutions-084159e8' to format '2' -- Playing periodic announcement [Dec 29 17:27:34] WARNING[4997]: channel.c:3106 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm) -- Skype/rexesbposolutions-084159e8 Playing 'queue' (language 'en') [Dec 29 17:27:43] WARNING[4997]: channel.c:3106 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm) [Dec 29 17:27:43] WARNING[4997]: file.c:147 ast_stopstream: Unable to restore format back to 2 == Spawn extension (sky, s, 2) exited non-zero on 'Skype/rexesbposolutions-084159e8' [Dec 29 17:27:43] NOTICE[4997]: core.cpp:2133 sfa_call_hangup: ending call following are output of some commands:- *CLI core show translation Translation times between formats (in milliseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 g723- ---- -- -- -- -- gsm- -222 21 26 -- 2- ulaw- 2-12 21 26 -- 2- alaw- 21-2 21 26 -- 2- g726aal2- 222- 21 26 -- 2- adpcm- 2222 -1 26 -- 2- slin- 1111 1- 15 -- 1- lpc10- 2222 21 -6 -- 2- g729- 6666 65 6- -- 6- speex- ---- -- -- -- -- ilbc- ---- -- -- -- -- g726- 2222 21 26 -- -- g722- ---- -- -- -- -- *CLI help g729 g729 show hostid Show G.729 Host-ID g729 show licenses Show G.729 Licenses and Usage g729 show version Show G.729 Module Version *CLI g729 show hostid Host-ID: 1a:cf:bb:80:da:e8:3b:dc:8c:e0:97:fe:a1:fb:65:c5:3c:e2:0b:be *CLI g729 show licenses 0/0 encoders/decoders of 1 licensed channels are currently in use Licenses Found: File: ***-*.lic -- Key: ***-* -- Host-ID: 1a:cf:bb:80:da:e8:3b:dc:8c:e0:97:fe:a1:fb:65:c5:3c:e2:0b:be -- Channels: 1 (Expires: 2029-11-30) (OK) *CLI g729 show version Digium G.729A Module Version 1.4_3.1.4 (optimized for i686_32) *CLI core show codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INTBINARYHEX TYPE NAME DESC
Re: [asterisk-users] Inquiry:Problem with Call Parking
On Wed, Sep 9, 2009 at 4:02 AM, Jeff LaCoursiere j...@jeff.net wrote: On Wed, 9 Sep 2009, hadi motamedi wrote: Thank you for your message . But I tried to find it on my server , as the followings : #find / -name sip.cfg -print But it didn't return any result . Can you please let me know where can I find it ? You probably have not setup central provisioning for your Polycom phones. I am guessing you are configuring them from their (horribly crappy) web interface. Although this kind of works, you will not be able to unleash the true power of your phones without setting up central provisioning. Worse you may be running an old version of the firmware, which may have problems. This involves getting the firmware and XML templates from Polycom, which will include the file sip.cfg. You will have to unpack these files on a TFTP or HTTP server, create XML files for each phone, and point the phone to the server to pick it up. There are numerous howtos on the web to set this up. Time for Google! j On Tue, Sep 8, 2009 at 2:58 PM, Steve Edwards asterisk@sedwards.com wrote: On Tue, 8 Sep 2009, hadi motamedi wrote: I sent you a message regarding my problem with Asterisk Call Parking feature and you told me that needs to check the polycom sip.cfg file . But my Asterisk doesn't have sip.cfg file . Can you please let me know how can I overcome ? sip.cfg is not an Asterisk file. sip.cfg should be in the directory the phone downloads it's configuration from. Typically, /tftpboot/ on a tftp server. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Dear All Further to this issue that I asked you before , please be informed that I setup for sip calls from my Asterisk console to SJPhone as sip client on an MS Windows machine . All of the configurations are working properly , I mean sip outgoing and sip incoming and voicemail but the call parking . Can you please let me know why I cannot still solve this issue ? It is appearing to me that the Polycom cfg is no longer involved here . Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
