Re: [asterisk-users] spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13
Hi Last month, people reported a crash with Asterisk 1.2.13 and spandsp-0.0.3 when receiving a fax using fax detection. Today I've tried 1.2.14 with the latest spandsp 0.0.3pre27 and with the snapshots for app_rxfax.c and app_txfax.c. The problem still happens. Has anyone found how to resolve this issue? I tried emailing Steve Underwood (with crash backtrace) but he hasn't answered... Otherwise, have you found spandsp 0.0.3 to provide better fax reception quality than 0.0.2? While I've had no problem with 0.0.2 locally, it usually fails when we receive faxes from overseas :( Thanks Regards Jean-Yves ___ --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] Linux distro + Asterisk or Trixbox?
Besides that you can use centos-plus repository which has lot of updated stuff not available in RHEL4 like php5 , mysql5 and all . On 18/12/06, Carla Schroder [EMAIL PROTECTED] wrote: On Sunday 17 December 2006 10:47 pm, Andrew Joakimsen wrote: I've used Asterisk on a bunch of RH 7.3 machines which were then replaced by RHEL 4. It is very stable, my biggest compliant is that RHEL(or CentOS, which is a direct rip-off) uses outdated packages (Linux 2.4.x, Apache 1, Mysql 4, php 4, etc) and Linux 2.4.x requires certain USB hardware to use zaptel timing without a hardware card, so we have a bunch of these dual xeon machines with the wrong USB hardware and can only run MeetMe on the one with the t1 cards. CentOS 4 was released May 2005 with a 2.6 kernel, Apache 2, and all other similarly current packages. The current kernel is 2.6.9-something. CentOS is a legal re-distribution of RHEL 4 rebuilt from source RPMs. Just like Pie Box, White Box, Tao, Lineox, and all the other Red Hat clones. -- ~ Carla Schroder Linux geek and random computer tamer check out my Linux Cookbook! http://www.oreilly.com/catalog/linuxckbk/ best book for sysadmins and power 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] sip peer name channel variable?
Or this link :
http://www.voip-info.org/wiki/index.php?page=Asterisk+Local+channels
http://www.voip-info.org/wiki/index.php?page=Asterisk+Local+channels
se the /n parameter of Local/ channels.
Cheers
Greg
_
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de William Piper
Envoyé : lundi 18 décembre 2006 06:03
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] sip peer name channel variable?
Check out this page:
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sipchaninfo
bp
On 12/17/06, Damon Estep [EMAIL PROTECTED] wrote:
Started out looking for what I thought was going to be a simple variable
name, have not found it.
Does anyone know of a variable that would contain only the SIP peer name of
the originating channel?
${CHANNEL} contains it, but it needs to be parsed and our dial plan
sometimes uses local channels, in one case it may be SIP/peer-id and in
another case local/peer-id
The peer is defined as type=friend
v1.2.13
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Re: [asterisk-users] spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13
Hi, I am using CentOS 4.4 [ asterisk-1.2.12.1 ] I too had problems with RxFax application. I tried spandsp-0.0.2pre26 spandsp-0.0.3pre23 .0.2 could install, but it crashed .0.3 doesnt install Finally I got 0.0.2pre26 running on debian sarge 3.1, without a crash ! - Danny Jean-Yves Avenard wrote: Hi Last month, people reported a crash with Asterisk 1.2.13 and spandsp-0.0.3 when receiving a fax using fax detection. Today I've tried 1.2.14 with the latest spandsp 0.0.3pre27 and with the snapshots for app_rxfax.c and app_txfax.c. The problem still happens. Has anyone found how to resolve this issue? I tried emailing Steve Underwood (with crash backtrace) but he hasn't answered... Otherwise, have you found spandsp 0.0.3 to provide better fax reception quality than 0.0.2? While I've had no problem with 0.0.2 locally, it usually fails when we receive faxes from overseas :( Thanks Regards Jean-Yves ___ --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] spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13
Hi On 12/18/06, Danny [EMAIL PROTECTED] wrote: I am using CentOS 4.4 [ asterisk-1.2.12.1 ] I too had problems with RxFax application. I tried spandsp-0.0.2pre26 spandsp-0.0.3pre23 .0.2 could install, but it crashed .0.3 doesnt install I never had any problems installing spandsp 0.0.2 with any of the version of Asterisk, and this for a few years and without a crash ever. The reason I'm looking at spandsp0.0.3 is that it's supposed to support T38 and I was also hoping it would work better with fax coming from overseas... JY ___ --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] spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13
On Mon, Dec 18, 2006 at 07:17:56PM +1100, Jean-Yves Avenard wrote: Hi Last month, people reported a crash with Asterisk 1.2.13 and spandsp-0.0.3 when receiving a fax using fax detection. Today I've tried 1.2.14 with the latest spandsp 0.0.3pre27 and with the snapshots for app_rxfax.c and app_txfax.c. The problem still happens. Can you provide a backtrace of the crash? Just saying it crashed doesn't really help. Also: what libraries are involved? ldd /usr/lib/asterisk/modules/app_rxfax.so and report what is the version and package of each library mentioned there. Any more automated way of doing this? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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 + Orion E1 GSM Gateway
Hi, I just got hold on an Orion E1 30 port GSM Gateway, and I am having problems trying to get the E1 link to come up. I am using Asteisk 1.2.12 with a Sangoma A101 card. I am quite familiar with E1's, both the Digium and Samgoma types, as I have successfully hooked up to many PBX's and such, but I just cant seem to get this one to work. None of the 30 channels 'come up'. What signailling, crc checking, should I be Master or slave? Anybody have experience on this? -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Thomson ST2030S and BLF
Hello. Once again, I came up with a problem for which I can't seem to find a solution. I'm not able to make BLF work with Thomson ST2030 phones and Asterisk (1.2.13). I've set up hints in dialplan, as well as Subscibe keys on the phone. The LED status gets updated according to the associated line status. However, when a phone is ringing, If I try to pickup the call by pressing the flashing key on the Thomson phone, I get an error, and the key keeps flashing at high rate until I reboot the phone, even if the associate line goes back to idle. I'm using firmware 1.50t3. I've also patched chan_sip as indicated on this forum: http://www.ip-phone-forum.de/showthread.php?p=590842#post590842 No success. Any help would really be appreciated. Thanks, Alberto. -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.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
Re: [asterisk-users] spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13
Hi On 12/18/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: Can you provide a backtrace of the crash? Sure. I've attached a backtrace for both 1.2.13 and 1.2.14 running the same version of spandsp and all other libraries. This is on a Fedora Core 6 machine (I can not attach the message as it makes the message over 40kB) http://www.avenard.org/asterisk/trace1-2-13.txt http://www.avenard.org/asterisk/trace1-2-14.txt Just saying it crashed doesn't really help. Well, the full backtrace was reported here last month, I was just pointing out that it was still happening with 1.2.14. Also: what libraries are involved? ldd /usr/lib/asterisk/modules/app_rxfax.so linked with spandsp 0.0.2 I get: linux-gate.so.1 = (0x00c6c000) libspandsp.so.0 = /usr/lib/libspandsp.so.0 (0x0071) libtiff.so.3 = /usr/lib/libtiff.so.3 (0x006b4000) libc.so.6 = /lib/libc.so.6 (0x001e7000) libm.so.6 = /lib/libm.so.6 (0x00e43000) libjpeg.so.62 = /usr/lib/libjpeg.so.62 (0x009c5000) libz.so.1 = /usr/lib/libz.so.1 (0x00d9e000) /lib/ld-linux.so.2 (0x00534000) I unfortunately can't try with spandsp 0.0.3 right now as I need a working asterisk ... linux-gate: spandsp: 0.0.3pre27 libtiff: 3.8.2 glibc: 2.5-3 libjpeg: 6b-37 and report what is the version and package of each library mentioned there. Any more automated way of doing this? This is standard Fedora Core 6. You can find last month, on this distribution list For the archive: http://lists.digium.com/pipermail/asterisk-users/2006-November/172652.html People mentioned this issue as well as where it was crashing. Hope that helps. Jean-Yves ___ --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] Digium TE405P with French E1 = Red Alert
Hi anyone have a idea for debug my digium TE405P card ? My zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone= fr defaultzone = fr My Zapata.conf: [channels] language=fr context=from-E1 switchtype = euroisdn pridialplan = unknown signalling = pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes immediate=no amaflags=documentation musiconhold=default group=1 callgroup=1 pickupgroup=1 channel = 1-15 channel = 17-31 a ztcfg -vv: Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: Clear channel (Default) (Slaves: 24) Channel 25: Clear channel (Default) (Slaves: 25) Channel 26: Clear channel (Default) (Slaves: 26) Channel 27: Clear channel (Default) (Slaves: 27) Channel 28: Clear channel (Default) (Slaves: 28) Channel 29: Clear channel (Default) (Slaves: 29) Channel 30: Clear channel (Default) (Slaves: 30) Channel 31: Clear channel (Default) (Slaves: 31) 31 channels configured. but with all test, i have a red alert: ipbx*CLI zap show status Description Alarms IRQ bpviol CRC4 T4XXP (PCI) Card 0 Span 1RED0 0 0 T4XXP (PCI) Card 0 Span 2UNCONFIGUR 0 0 0 T4XXP (PCI) Card 0 Span 3UNCONFIGUR 0 0 0 T4XXP (PCI) Card 0 Span 4UNCONFIGUR 0 0 0 ipbx*CLI i use a crossover cable: 1=4 2=5 4=1 5=2 to my PRI supplier My syslog: Dec 18 12:46:39 ipbx kernel: Zaptel Version: 1.2.9.1-1mdv2007.0 Echo Canceller: KB1 Dec 18 12:46:39 ipbx zaptel: Loading zaptel framework: succeeded Dec 18 12:46:39 ipbx kernel: ACPI: PCI Interrupt :00:05.0[A] - GSI 24 (level, low) - IRQ 24 Dec 18 12:46:39 ipbx kernel: Found TE4XXP at base address febffc00, remapped to f8afec00 Dec 18 12:46:39 ipbx kernel: TE4XXP version c01a016a, burst OFF, slip debug: OFF Dec 18 12:46:39 ipbx kernel: FALC version: 0005, Board ID: 00 Dec 18 12:46:39 ipbx kernel: Reg 0: 0x186ec400 Dec 18 12:46:39 ipbx kernel: Reg 1: 0x186ec000 Dec 18 12:46:39 ipbx kernel: Reg 2: 0x Dec 18 12:46:39 ipbx kernel: Reg 3: 0x Dec 18 12:46:39 ipbx kernel: Reg 4: 0x Dec 18 12:46:39 ipbx kernel: Reg 5: 0x Dec 18 12:46:39 ipbx kernel: Reg 6: 0xc01a016a Dec 18 12:46:39 ipbx kernel: Reg 7: 0x1000 Dec 18 12:46:39 ipbx kernel: Reg 8: 0x010200ff Dec 18 12:46:39 ipbx kernel: Reg 9: 0x00fd Dec 18 12:46:39 ipbx kernel: Reg 10: 0x Dec 18 12:46:39 ipbx kernel: TE4XXP: Launching card: 0 Dec 18 12:46:39 ipbx kernel: TE4XXP: Setting up global serial parameters Dec 18 12:46:39 ipbx kernel: Successfully initialized serial bus for unit 0 Dec 18 12:46:39 ipbx kernel: Successfully initialized serial bus for unit 1 Dec 18 12:46:39 ipbx kernel: Successfully initialized serial bus for unit 2 Dec 18 12:46:39 ipbx kernel: Successfully initialized serial bus for unit 3 Dec 18 12:46:39 ipbx kernel: Found a Wildcard: Wildcard TE405P (3rd Gen) Dec 18 12:46:40 ipbx kernel: About to enter spanconfig! Dec 18 12:46:40 ipbx kernel: TE4XXP: Configuring span 1 Dec 18 12:46:40 ipbx kernel: Done with spanconfig! Dec 18 12:46:40 ipbx kernel: TE4XXP: Configured channel 1 (TE4/0/1/1) sigtype 128 Dec 18 12:46:40 ipbx kernel: Unassigning channel 0/1! Dec 18 12:46:40 ipbx kernel: TE4XXP: Configured channel 2 (TE4/0/1/2) sigtype 128 Dec 18 12:46:40 ipbx kernel: Unassigning channel 0/2! Dec 18 12:46:40 ipbx kernel: TE4XXP: Configured channel 3 (TE4/0/1/3) sigtype 128 Dec 18 12:46:40 ipbx kernel: Unassigning channel 0/3! Dec 18 12:46:40 ipbx kernel: TE4XXP: Configured channel 4 (TE4/0/1/4) sigtype 128 Dec 18 12:46:40 ipbx kernel: Unassigning channel
R: [asterisk-users] Digium TE405P with French E1 = Red Alert
I'm not sure that u have to use a crossover cable. Your telco give u a network emulation, and u emulate a cpe, so i think u need a straigh cable. Giordano -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Noc Phibee Inviato: lunedì 18 dicembre 2006 12.53 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: [asterisk-users] Digium TE405P with French E1 = Red Alert Hi anyone have a idea for debug my digium TE405P card ? My zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone= fr defaultzone = fr My Zapata.conf: [channels] language=fr context=from-E1 switchtype = euroisdn pridialplan = unknown signalling = pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes immediate=no amaflags=documentation musiconhold=default group=1 callgroup=1 pickupgroup=1 channel = 1-15 channel = 17-31 a ztcfg -vv: Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: Clear channel (Default) (Slaves: 24) Channel 25: Clear channel (Default) (Slaves: 25) Channel 26: Clear channel (Default) (Slaves: 26) Channel 27: Clear channel (Default) (Slaves: 27) Channel 28: Clear channel (Default) (Slaves: 28) Channel 29: Clear channel (Default) (Slaves: 29) Channel 30: Clear channel (Default) (Slaves: 30) Channel 31: Clear channel (Default) (Slaves: 31) 31 channels configured. but with all test, i have a red alert: ipbx*CLI zap show status Description Alarms IRQ bpviol CRC4 T4XXP (PCI) Card 0 Span 1RED0 0 0 T4XXP (PCI) Card 0 Span 2UNCONFIGUR 0 0 0 T4XXP (PCI) Card 0 Span 3UNCONFIGUR 0 0 0 T4XXP (PCI) Card 0 Span 4UNCONFIGUR 0 0 0 ipbx*CLI i use a crossover cable: 1=4 2=5 4=1 5=2 to my PRI supplier My syslog: Dec 18 12:46:39 ipbx kernel: Zaptel Version: 1.2.9.1-1mdv2007.0 Echo Canceller: KB1 Dec 18 12:46:39 ipbx zaptel: Loading zaptel framework: succeeded Dec 18 12:46:39 ipbx kernel: ACPI: PCI Interrupt :00:05.0[A] - GSI 24 (level, low) - IRQ 24 Dec 18 12:46:39 ipbx kernel: Found TE4XXP at base address febffc00, remapped to f8afec00 Dec 18 12:46:39 ipbx kernel: TE4XXP version c01a016a, burst OFF, slip debug: OFF Dec 18 12:46:39 ipbx kernel: FALC version: 0005, Board ID: 00 Dec 18 12:46:39 ipbx kernel: Reg 0: 0x186ec400 Dec 18 12:46:39 ipbx kernel: Reg 1: 0x186ec000 Dec 18 12:46:39 ipbx kernel: Reg 2: 0x Dec 18 12:46:39 ipbx kernel: Reg 3: 0x Dec 18 12:46:39 ipbx kernel: Reg 4: 0x Dec 18 12:46:39 ipbx kernel: Reg 5: 0x Dec 18 12:46:39 ipbx kernel: Reg 6: 0xc01a016a Dec 18 12:46:39 ipbx kernel: Reg 7: 0x1000 Dec 18 12:46:39 ipbx kernel: Reg 8: 0x010200ff Dec 18 12:46:39 ipbx kernel: Reg 9: 0x00fd Dec 18 12:46:39 ipbx kernel: Reg 10: 0x Dec 18 12:46:39 ipbx kernel: TE4XXP: Launching card: 0 Dec 18 12:46:39 ipbx kernel: TE4XXP: Setting up global serial parameters Dec 18 12:46:39 ipbx kernel: Successfully initialized serial bus for unit 0 Dec 18 12:46:39 ipbx kernel: Successfully initialized serial bus for unit 1 Dec 18 12:46:39 ipbx kernel: Successfully initialized serial bus for unit 2 Dec 18 12:46:39 ipbx kernel: Successfully initialized serial bus for unit 3 Dec 18 12:46:39 ipbx kernel: Found a Wildcard: Wildcard TE405P (3rd Gen) Dec 18 12:46:40 ipbx kernel: About to enter spanconfig! Dec 18 12:46:40 ipbx kernel: TE4XXP: Configuring span 1 Dec 18 12:46:40 ipbx kernel: Done with spanconfig! Dec 18 12:46:40 ipbx kernel: TE4XXP: Configured channel 1 (TE4/0/1/1) sigtype 128 Dec 18 12:46:40 ipbx kernel: Unassigning
Re: [asterisk-users] Asterisk 1.4.0 B4 Sounds Directory
On 12/17/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Dec 17, 2006 at 12:35:41PM -0600, Kevin P. Fleming wrote: Samy Antoun wrote: I noticed that the sound directory is missing from asterisk-1.4.0-beta4.tar.gz. This is incorrect; the sounds directory is present and contains two files. This directory (7 M) witch existed in asterisk-1.4.0-beta3.tar.gz has GSM Core Sounds and some MOH. There was a packaging error when this tarball was created, and the sound/MOH file tarballs were not included. However, the 'make install' process will automatically download the sounds during installation anyway; adding them to the tarball just makes it slightly quicker to do the installation. Note that this breaks Debian package building. Same applies to Mandriva. Can we assume that the released tarall will include the gsm sounds? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Digium TE405P with French E1 = Red Alert
Hi what is your operator? I have some pb on orange... thx Noc Phibee a écrit : Hi anyone have a idea for debug my digium TE405P card ? My zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone= fr defaultzone = fr My Zapata.conf: [channels] language=fr context=from-E1 switchtype = euroisdn pridialplan = unknown signalling = pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes immediate=no amaflags=documentation musiconhold=default group=1 callgroup=1 pickupgroup=1 channel = 1-15 channel = 17-31 a ztcfg -vv: Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: Clear channel (Default) (Slaves: 24) Channel 25: Clear channel (Default) (Slaves: 25) Channel 26: Clear channel (Default) (Slaves: 26) Channel 27: Clear channel (Default) (Slaves: 27) Channel 28: Clear channel (Default) (Slaves: 28) Channel 29: Clear channel (Default) (Slaves: 29) Channel 30: Clear channel (Default) (Slaves: 30) Channel 31: Clear channel (Default) (Slaves: 31) 31 channels configured. but with all test, i have a red alert: ipbx*CLI zap show status Description Alarms IRQ bpviol CRC4 T4XXP (PCI) Card 0 Span 1RED0 0 0 T4XXP (PCI) Card 0 Span 2UNCONFIGUR 0 0 0 T4XXP (PCI) Card 0 Span 3UNCONFIGUR 0 0 0 T4XXP (PCI) Card 0 Span 4UNCONFIGUR 0 0 0 ipbx*CLI i use a crossover cable: 1=4 2=5 4=1 5=2 to my PRI supplier My syslog: Dec 18 12:46:39 ipbx kernel: Zaptel Version: 1.2.9.1-1mdv2007.0 Echo Canceller: KB1 Dec 18 12:46:39 ipbx zaptel: Loading zaptel framework: succeeded Dec 18 12:46:39 ipbx kernel: ACPI: PCI Interrupt :00:05.0[A] - GSI 24 (level, low) - IRQ 24 Dec 18 12:46:39 ipbx kernel: Found TE4XXP at base address febffc00, remapped to f8afec00 Dec 18 12:46:39 ipbx kernel: TE4XXP version c01a016a, burst OFF, slip debug: OFF Dec 18 12:46:39 ipbx kernel: FALC version: 0005, Board ID: 00 Dec 18 12:46:39 ipbx kernel: Reg 0: 0x186ec400 Dec 18 12:46:39 ipbx kernel: Reg 1: 0x186ec000 Dec 18 12:46:39 ipbx kernel: Reg 2: 0x Dec 18 12:46:39 ipbx kernel: Reg 3: 0x Dec 18 12:46:39 ipbx kernel: Reg 4: 0x Dec 18 12:46:39 ipbx kernel: Reg 5: 0x Dec 18 12:46:39 ipbx kernel: Reg 6: 0xc01a016a Dec 18 12:46:39 ipbx kernel: Reg 7: 0x1000 Dec 18 12:46:39 ipbx kernel: Reg 8: 0x010200ff Dec 18 12:46:39 ipbx kernel: Reg 9: 0x00fd Dec 18 12:46:39 ipbx kernel: Reg 10: 0x Dec 18 12:46:39 ipbx kernel: TE4XXP: Launching card: 0 Dec 18 12:46:39 ipbx kernel: TE4XXP: Setting up global serial parameters Dec 18 12:46:39 ipbx kernel: Successfully initialized serial bus for unit 0 Dec 18 12:46:39 ipbx kernel: Successfully initialized serial bus for unit 1 Dec 18 12:46:39 ipbx kernel: Successfully initialized serial bus for unit 2 Dec 18 12:46:39 ipbx kernel: Successfully initialized serial bus for unit 3 Dec 18 12:46:39 ipbx kernel: Found a Wildcard: Wildcard TE405P (3rd Gen) Dec 18 12:46:40 ipbx kernel: About to enter spanconfig! Dec 18 12:46:40 ipbx kernel: TE4XXP: Configuring span 1 Dec 18 12:46:40 ipbx kernel: Done with spanconfig! Dec 18 12:46:40 ipbx kernel: TE4XXP: Configured channel 1 (TE4/0/1/1) sigtype 128 Dec 18 12:46:40 ipbx kernel: Unassigning channel 0/1! Dec 18 12:46:40 ipbx kernel: TE4XXP: Configured channel 2 (TE4/0/1/2) sigtype 128 Dec 18 12:46:40 ipbx kernel: Unassigning channel 0/2! Dec 18 12:46:40 ipbx kernel: TE4XXP: Configured channel 3 (TE4/0/1/3) sigtype 128 Dec 18 12:46:40 ipbx kernel: Unassigning channel 0/3! Dec 18 12:46:40 ipbx kernel: TE4XXP:
Re: [asterisk-users] Digium TE405P with French E1 = Red Alert
Hi it's Colt-Telecom. you have a TE405P ? bye pixiesfr a écrit : Hi what is your operator? I have some pb on orange... thx Noc Phibee a écrit : Hi anyone have a idea for debug my digium TE405P card ? My zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone= fr defaultzone = fr My Zapata.conf: [channels] language=fr context=from-E1 switchtype = euroisdn pridialplan = unknown signalling = pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes immediate=no amaflags=documentation musiconhold=default group=1 callgroup=1 pickupgroup=1 channel = 1-15 channel = 17-31 a ztcfg -vv: Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: Clear channel (Default) (Slaves: 24) Channel 25: Clear channel (Default) (Slaves: 25) Channel 26: Clear channel (Default) (Slaves: 26) Channel 27: Clear channel (Default) (Slaves: 27) Channel 28: Clear channel (Default) (Slaves: 28) Channel 29: Clear channel (Default) (Slaves: 29) Channel 30: Clear channel (Default) (Slaves: 30) Channel 31: Clear channel (Default) (Slaves: 31) 31 channels configured. but with all test, i have a red alert: ipbx*CLI zap show status Description Alarms IRQ bpviol CRC4 T4XXP (PCI) Card 0 Span 1RED0 0 0 T4XXP (PCI) Card 0 Span 2UNCONFIGUR 0 0 0 T4XXP (PCI) Card 0 Span 3UNCONFIGUR 0 0 0 T4XXP (PCI) Card 0 Span 4UNCONFIGUR 0 0 0 ipbx*CLI i use a crossover cable: 1=4 2=5 4=1 5=2 to my PRI supplier My syslog: Dec 18 12:46:39 ipbx kernel: Zaptel Version: 1.2.9.1-1mdv2007.0 Echo Canceller: KB1 Dec 18 12:46:39 ipbx zaptel: Loading zaptel framework: succeeded Dec 18 12:46:39 ipbx kernel: ACPI: PCI Interrupt :00:05.0[A] - GSI 24 (level, low) - IRQ 24 Dec 18 12:46:39 ipbx kernel: Found TE4XXP at base address febffc00, remapped to f8afec00 Dec 18 12:46:39 ipbx kernel: TE4XXP version c01a016a, burst OFF, slip debug: OFF Dec 18 12:46:39 ipbx kernel: FALC version: 0005, Board ID: 00 Dec 18 12:46:39 ipbx kernel: Reg 0: 0x186ec400 Dec 18 12:46:39 ipbx kernel: Reg 1: 0x186ec000 Dec 18 12:46:39 ipbx kernel: Reg 2: 0x Dec 18 12:46:39 ipbx kernel: Reg 3: 0x Dec 18 12:46:39 ipbx kernel: Reg 4: 0x Dec 18 12:46:39 ipbx kernel: Reg 5: 0x Dec 18 12:46:39 ipbx kernel: Reg 6: 0xc01a016a Dec 18 12:46:39 ipbx kernel: Reg 7: 0x1000 Dec 18 12:46:39 ipbx kernel: Reg 8: 0x010200ff Dec 18 12:46:39 ipbx kernel: Reg 9: 0x00fd Dec 18 12:46:39 ipbx kernel: Reg 10: 0x Dec 18 12:46:39 ipbx kernel: TE4XXP: Launching card: 0 Dec 18 12:46:39 ipbx kernel: TE4XXP: Setting up global serial parameters Dec 18 12:46:39 ipbx kernel: Successfully initialized serial bus for unit 0 Dec 18 12:46:39 ipbx kernel: Successfully initialized serial bus for unit 1 Dec 18 12:46:39 ipbx kernel: Successfully initialized serial bus for unit 2 Dec 18 12:46:39 ipbx kernel: Successfully initialized serial bus for unit 3 Dec 18 12:46:39 ipbx kernel: Found a Wildcard: Wildcard TE405P (3rd Gen) Dec 18 12:46:40 ipbx kernel: About to enter spanconfig! Dec 18 12:46:40 ipbx kernel: TE4XXP: Configuring span 1 Dec 18 12:46:40 ipbx kernel: Done with spanconfig! Dec 18 12:46:40 ipbx kernel: TE4XXP: Configured channel 1 (TE4/0/1/1) sigtype 128 Dec 18 12:46:40 ipbx kernel: Unassigning channel 0/1! Dec 18 12:46:40 ipbx kernel: TE4XXP: Configured channel 2 (TE4/0/1/2) sigtype 128 Dec 18 12:46:40 ipbx kernel: Unassigning channel 0/2! Dec 18 12:46:40 ipbx kernel: TE4XXP: Configured channel 3 (TE4/0/1/3) sigtype 128 Dec 18 12:46:40 ipbx
[asterisk-users] calls interrupted by music on hold
Hi, I have Asterisk 1.2.9.1 on a Debian box with a beronet BRI card (install-misdn-mqueue driver). Sometimes, calls are interrupted by music on hold without any reason: the caller and the callee are put on hold for few seconds (they both listen to moh) and then the call is established againand it happens more frequently during long calls. Asterisk logs report nothing. Anybody has any ideas about this problem? 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
[asterisk-users] Shared Line Appearances (SLA) in 1.4
Greetings, Back in September someone asked about documentation for the new SLA feature in 1.4, however they received no replies. I thought I might ask the same question now in December. Apart from sla.conf.sample and a few comments in app_meetme.c I have been unable to find useful documentation. Is anyone using this feature right now? Is there a helpful source for information this highly desired capability? Regards, -- Anthony Kava Senior Network Administrator Pottawattamie County, Iowa Sheep are slow and tasty, and therefore must remain constantly alert. -- Bruce Schneier, Beyond Fear smime.p7s Description: S/MIME cryptographic signature ___ --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 to asterisk - to zap
Hello that might would be an easy question for someone, but im in doubt Is there any possibility to pass a call from one asterisk to another and then to ZAP channel. For instance I have A asterisk with numbering 45670 B asterisk with numbering 45680 second asterisk has TE110P card with single PRI port connected to Siemens EWSD. When I originate call from asterisk B I reach the world thru ZAP, when I call from asterisk A I reach numbering of asterisk B but cant get to the PSTN network. ASTERISK---ASTERISK-ZAP-PSTN Should I have OpenSER for that and terminate my call on CISCO AS5350 or something? 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
Re: [asterisk-users] Shared Line Appearances (SLA) in 1.4
That would be great if Antony's demand could be satisfied. Cheers ___ --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 asterisk - to zap
DUNDi can do this for you. Advertise the routes you can terminate on Box A. When you place a call on Box B, have it check your DUNDi cloud, and Box A will provide the route and terminate the call via zap for you. Alex On 12/18/06, Pryakhin Dimitry [EMAIL PROTECTED] wrote: Hello that might would be an easy question for someone, but im in doubt Is there any possibility to pass a call from one asterisk to another and then to ZAP channel. For instance I have A asterisk with numbering 45670 B asterisk with numbering 45680 second asterisk has TE110P card with single PRI port connected to Siemens EWSD. When I originate call from asterisk B I reach the world thru ZAP, when I call from asterisk A I reach numbering of asterisk B but cant get to the PSTN network. ASTERISK---ASTERISK-ZAP-PSTN Should I have OpenSER for that and terminate my call on CISCO AS5350 or something? 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 -- Alex Robar [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 to asterisk - to zap
I may be making this easier than it is but something like this should
work:
A:
DIAL(IAX2/${ASTERISKB}/[EMAIL PROTECTED])
B:
[context]
exten = EXTEN,1,DIAL(Zap/${EXTEN})
I have this scenario also except we have numerous A servers connecting
via the PRI lines on B servers.
-Jonathan
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pryakhin
Dimitry
Sent: Monday, December 18, 2006 8:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] asterisk to asterisk - to zap
Hello
that might would be an easy question for someone, but im in doubt
Is there any possibility to pass a call from one asterisk to another and
then to ZAP channel.
For instance
I have
A asterisk with numbering 45670
B asterisk with numbering 45680
second asterisk has TE110P card with single PRI port connected to
Siemens EWSD.
When I originate call from asterisk B I reach the world thru ZAP,
when I call from asterisk A I reach numbering of asterisk B but cant
get to the PSTN network.
ASTERISK---ASTERISK-ZAP-PSTN
Should I have OpenSER for that and terminate my call on CISCO AS5350 or
something?
Thanks
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Re: [asterisk-users] Thomson ST2030S and BLF
Alberto, Call pickup is not implemented yet within Thomson ST2030 (1.50 firmware). More precisely, call pickup current implementation is not Asterisk compliant. A new release is scheduled for February (I've got this confirmed by Thomson 10 minutes ago) but we don't know if call pickup will be included. Regards ___ --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] Day/night service and indications on the phone
I'm happy to report that with a very litte change to app_devstate.c (just in the way ast_device_state_changed_literal() is called) that module just compiles and works fine even without bristuffing anything. BTW I'm using a Thomson ST2030S phone with a status key subscribed to a DS/xxx hint. Thanks again for your precious help! Could you elaborate ? How is it working now ? How you extensions.conf file looks like ? Regards ___ --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 peer name channel variable?
Perfect.
I guess I could not find it on my own because I was searching for a
variable, but a function is fine with me!
Thank you,
Damon
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of William
Piper
Sent: Sunday, December 17, 2006 10:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] sip peer name channel variable?
Check out this page:
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sipchaninfo
bp
On 12/17/06, Damon Estep [EMAIL PROTECTED] wrote:
Started out looking for what I thought was going to be a simple variable
name, have not found it.
Does anyone know of a variable that would contain only the SIP peer name
of the originating channel?
${CHANNEL} contains it, but it needs to be parsed and our dial plan
sometimes uses local channels, in one case it may be SIP/peer-id and in
another case local/peer-id
The peer is defined as type=friend
v1.2.13
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Re: [asterisk-users] Fast Busy Followup
Any idea what actually causes this problem? Is it an error with the zaptel programs or asterisk? Or does this problem lay with the telco providers. Seems odd that if you restart the driver, everything is good again (mine does as well). This leads me to think its either asterisk being unable to clear a channel or an error with zaptel. Ron McLeod wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Rob Schall Sent: Friday, December 15, 2006 11:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Fast Busy Followup So I might have a bit of a more narrow question from my earlier one. Previous, I had been wondering what would cause a phone dialing into a DID that connects to the asterisk box to get a fast busy. I've noticed the following message: chan_zap.c: Ring requested on unconfigured channel 0/1 span 2 Any idea what would give me this error? And would this cause a fast busy? Thanks again everyone for your help with this matter, Rob ___ --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 I have the same problem with a span from Bell Canada. After time, calls begin to fail with the same Ring requested ... error message. I found that if I restart Zaptel and Asterisk, that the problem goes away for a while. Ron ___ --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] Shared Line Appearances (SLA) in 1.4
On 12/18/06, Anthony Kava [EMAIL PROTECTED] wrote: Greetings, Back in September someone asked about documentation for the new SLA feature in 1.4, however they received no replies. I thought I might ask the same question now in December. Apart from sla.conf.sample and a few comments in app_meetme.c I have been unable to find useful documentation. Is anyone using this feature right now? Is there a helpful source for information this highly desired capability? The last beta has been posted, and supposedly next week or so the release, and all this time we were supposed to be testing the new features. I never could figure out SLA so I gave up. I tried finding info about it in the bug tracker, but it obviously takes a much smarter person than I to figure it out. ___ --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] Fast Busy
Is this fixable? It seems as though the channels aren't clearing up after use, and after 2 or 3 incoming calls, i get the fast busy. if I wait a while, or if i restart zaptel, the channels clear up again. Any ideas? Thanks, Rob Henry.L.Coleman wrote: Sounds like you have a disconnect supervision problem. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada We currently have a pri coming into our asterisk system. Most of the time, the did numbers that we call into it work great. However, occationally, we get fast busies, but we noticed those busies were not due to anyone being on the line, etc... Any ideas what could cause this? Is this a congestion thing? Is there something I should add to the dial plan or configuration of the card to fix this? Thanks, Rob ___ --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] BLF on GXP2000
Here's what I have, it's to early for me to think so hopefully looking
at mine helps :D
extensions.conf:
[ext-local]
exten = 701,1,Macro(exten-vm,701,701)
exten = 701,n,Hangup
exten = 701,hint,SIP/701
sip.conf:
[701]
type=friend
secret=1234
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
nat=yes
host=dynamic
dtmfmode=rfc2833
dial=SIP/701
context=from-internal
canreinvite=no
callerid=device 701
mailbox=701
If this doesn't help in some fashion let me know and I'll think it
through a little later...off to get some coffee.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Johnson
Sent: Sunday, December 17, 2006 4:50 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] BLF on GXP2000
I am trying to set up the BLF on a GXP2000.
Currently what I have is
extensions.conf:
[globals]
polycom430=SIP/101
[internal]
exten = 101,1,Macro(voicemail,${polycom430})
[macro-voicemail]
exten = s,1,Dial(${ARG1},10,tT)
exten = s,2,VoiceMail([EMAIL PROTECTED] )
exten = s,102,VoiceMail([EMAIL PROTECTED])
[ext-local-custom]
exten = 101,hint,${polycom430}
sip.conf:
[general]
subscribecontext=ext-local-custom
And have set up the key to Asterisk BLF with UserID101
When I reload the phone, I get the following error:
[Dec 17 13:34:33] ERROR[2551]: chan_sip.c:14064
handle_request_subscribe: Got SUBSCRIBE for extension [EMAIL PROTECTED] from
192.168.1.248 http://192.168.1.248/ , but there is no hint for that
extension
Any help is greatly appreciated.
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Re: [asterisk-users] Asterisk + Orion E1 GSM Gateway
yusuf wrote: Hi, I just got hold on an Orion E1 30 port GSM Gateway, and I am having problems trying to get the E1 link to come up. I am using Asteisk 1.2.12 with a Sangoma A101 card. I am quite familiar with E1's, both the Digium and Samgoma types, as I have successfully hooked up to many PBX's and such, but I just cant seem to get this one to work. None of the 30 channels 'come up'. What signailling, crc checking, should I be Master or slave? Sanity check: Have you read the fine manual :)? I understand Orion makes both ISDN PRI/Q.SIG and MFC/R2 type E1 channel banks. If it's the PRI type, standard zaptel with the appropriate NET/CPE setting on the CB should be ok. If it's a MFC/R2, then you'll have to try unicall. Leo ___ --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] Wait command
Hi
I've got a script like this
exten =
s,1,SetVar(CALLFILENAME=/var/www/recordings/${TIMESTAMP:0:8:7}/${UNIQUEID})
exten = s,2,AGI(recordstart.py,${ARG1},${CALLERIDNUM},${CALLFILENAME},Ind)
exten = s,3,DIAL(ZAP/g2/${ARG1},70)
exten =
s,4,AGI(logerror.py,${ARG1},${CALLERIDNUM},${CHANNEL},${DIALSTATUS},${DATETIME},
${CAUSECODE})
exten = s,5,hangup
exten =
s,104,AGI(logerror.py,${ARG1},${CALLERIDNUM},${CHANNEL},${DIALSTATUS},${DATETIME},
${CAUSECODE})
exten = h,1,stopmonitor
exten = h,2,SetVar(CALLFILEDIR=/var/www/recordings/${TIMESTAMP:0:8:7})
exten = h,3,System(/etc/asterisk/agi-bin/filexfer ${CALLFILENAME}
${CALLFILEDIR})
It causes me some problems occausionsly and I want to pause the scipt by
wait in 5 s.
exten =
s,1,SetVar(CALLFILENAME=/var/www/recordings/${TIMESTAMP:0:8:7}/${UNIQUEID})
exten = s,2,AGI(recordstart.py,${ARG1},${CALLERIDNUM},${CALLFILENAME},Ind)
exten = s,3,DIAL(ZAP/g2/${ARG1},70)
exten =
s,4,AGI(logerror.py,${ARG1},${CALLERIDNUM},${CHANNEL},${DIALSTATUS},${DATETIME},
${CAUSECODE})
exten = s,5,hangup
exten =
s,104,AGI(logerror.py,${ARG1},${CALLERIDNUM},${CHANNEL},${DIALSTATUS},${DATETIME},
${CAUSECODE})
exten = h,1,stopmonitor
exten = h,2,wait(5)
exten = h,3,SetVar(CALLFILEDIR=/var/www/recordings/${TIMESTAMP:0:8:7})
exten = h,4,System(/etc/asterisk/agi-bin/filexfer ${CALLFILENAME}
${CALLFILEDIR})
but it doesn't work h,1 and h,2 is OK, but then it stops, any goood
suggetions
/RC
_
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[asterisk-users] Re: queue member refresh
On 12/7/06, nik600 [EMAIL PROTECTED] wrote: I am experiencing this: 1 - A,B,C are SIP users logged on QUEUEA with ringall strategy 2 - I call QUEUEA 3 - A,B,C start ringing 4 - nobody answer 5 - D logs on the QUEUEA 6 - D doen's receive any call, but A,B,C are still ringing How can i avoid that? I'd like that when D joins the QUEUEA it will immediately receive the call that is still ringing on other users... Thanks in advance, nik any ideas? How can i fix it? 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
Re: [asterisk-users] 1.4beta3 help
On Sat, 2006-12-02 at 02:07 -0500, Doug Crompton wrote: I am running an old SUSE 7.3 system, 2.4 kernel and glibc 2.2 I picked up the ncurses-devel rpm and it now requires glibc 2.3 I found a glibc 2.4 rpm but I am a little reluctent to install it. It would be a disaster to lose this system. Any reason for trying a 'state-of-the-art' asterisk version (1.4-beta) on an ancient (7.3) version of SuSE? I would recommend you download SuSE-10.2 and re-install your platform. Perhaps not the latest kernel (Suse has 2.6.18.2 versus 2.9.19.1) but all other part gets up-to-date as well... (stability, security) Hans ___ --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 + Orion E1 GSM Gateway
Leo Ann Boon wrote: yusuf wrote: Hi, I just got hold on an Orion E1 30 port GSM Gateway, and I am having problems trying to get the E1 link to come up. I am using Asteisk 1.2.12 with a Sangoma A101 card. I am quite familiar with E1's, both the Digium and Samgoma types, as I have successfully hooked up to many PBX's and such, but I just cant seem to get this one to work. None of the 30 channels 'come up'. What signailling, crc checking, should I be Master or slave? Sanity check: Have you read the fine manual :)? I understand Orion makes both ISDN PRI/Q.SIG and MFC/R2 type E1 channel banks. If it's the PRI type, standard zaptel with the appropriate NET/CPE setting on the CB should be ok. If it's a MFC/R2, then you'll have to try unicall. Leo Hi, crazy thing is I dont have any manual or anything, just the Gateway. From reading the 'sales' doc on the Orion site, this is a PRI/Q.SIg type. But I dont have anything else besides that. I dont even know how to get the Serial cable to work to configure the Gateway (through Minicom/Hyperterminal, there is a configuration on Orion, or so I'm told.) Can you help? -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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 and outlook
Network Configurations Block D, Surrey Park, Barham Road, Westville, 3610 Helpdesk: (086) 163-8266 Tel: (031) 266-1563 Fax: (031) 266-4206 Hi list. Has anyone used any commercial or open source application to integrate Asterisk into MS Outlook 2003 which can be used to place calls directly to contacts from Outlook? And if so how well does it work? Thanks, Richard Best Regards Richard Soderblom Network Configurations Cell: E-Mail: [EMAIL PROTECTED] Number of Attachments: 0 This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ --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] GXP2000 and BLF
On Sun, 2006-12-17 at 13:51 -0600, Chris Johnson wrote:
I am trying to set up the BLF on a GXP2000.
Currently what I have is
extesions.conf:
[globals]
polycom430=SIP/101
[internal]
;exten = 101,1,Dial(SIP/101,10,)
;exten = 101,2,VoiceMail([EMAIL PROTECTED] )
;exten = 101,102,VoiceMail([EMAIL PROTECTED])
exten = 101,1,Macro(voicemail,${polycom430})
[macro-voicemail]
exten = s,1,Dial(${ARG1},10,tT)
exten = s,2,VoiceMail([EMAIL PROTECTED] )
exten = s,102,VoiceMail([EMAIL PROTECTED])
[ext-local-custom]
exten = 101,hint,${polycom430}
You cannot use variables in hints. You have to put the specific
channel like:
exten = 101,hint,SIP/101
--
Telecomunicaciones Abiertas de Mexico S.A. de C.V.
Carlos Chàvez Prats
Director de Tecnologìa
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RE: [asterisk-users] Asterisk and outlook
I think Thirdlane has a software plugin for Asterisk that does this. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice direct - 716.250.3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Soderblom Sent: Monday, December 18, 2006 9:46 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk and outlook Network Configurations Block D, Surrey Park, Barham Road, Westville, 3610 Helpdesk: (086) 163-8266 Tel: (031) 266-1563 Fax: (031) 266-4206 Hi list. Has anyone used any commercial or open source application to integrate Asterisk into MS Outlook 2003 which can be used to place calls directly to contacts from Outlook? And if so how well does it work? Thanks, Richard Best Regards Richard Soderblom Network Configurations Cell: E-Mail: [EMAIL PROTECTED] Number of Attachments: 0 This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ --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 and outlook
Forgot linkage http://www.thirdlane.com/outlookdialer.htm Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice direct - 716.250.3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Soderblom Sent: Monday, December 18, 2006 9:46 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk and outlook Network Configurations Block D, Surrey Park, Barham Road, Westville, 3610 Helpdesk: (086) 163-8266 Tel: (031) 266-1563 Fax: (031) 266-4206 Hi list. Has anyone used any commercial or open source application to integrate Asterisk into MS Outlook 2003 which can be used to place calls directly to contacts from Outlook? And if so how well does it work? Thanks, Richard Best Regards Richard Soderblom Network Configurations Cell: E-Mail: [EMAIL PROTECTED] Number of Attachments: 0 This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ --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] Good Commercial Grade Service Provider?
Yikes! Thanks for that disturbing info. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen Sent: Friday, December 15, 2006 10:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Good Commercial Grade Service Provider? VoicePulse is the absolute worst. You can get additional channels for $25/month but that includes no usage whatsoever. That's almost double what the same capacity WITH MINUTES on a PRI port costs! Any decent provider will be able to give you an unlimited number of channels because you are paying for the usage. If you are paying per channel I would expect some sort of included usage. For example one of our PRI provider's offering boils down to about $12/channel, unlimited regional calling (more than Bell's local calling area) and some 200 minutes of LD calling extra DID cost less than a quarter each, compare that to voicepulse charging you $25/month for jack shit $11/month per DID but no additional usage. You can get 20 DID with them on one account and you get 4 calls at a time your cost is $220/month. you can open however 20 different accounts with one number on each, you pay the SAME $220/month however you get 80 calls at the same time! If you wanted the same arrangement on a single voicepulse account it would cost $620/month However don't do that, with a single account VoicePulse will charge you RANDOM amounts to your credit card, even if they say they will ONLY charge your card in $25 increments, I've asked them countless times to charge other amounts and they say NO impossible, billing system limitation, yada yada but when it comes down to it they can do and will charge your card for a random amount. And you cannot port any telephone number away from them, they have instructed their carrier (Broadview) to not allow any sort of LNP out request. Also any time there is an issue they blame you. And aulthough they sell a VoicePulse Connect! for Asterisk service where Asterisk is a LINUX PROGRAM they insist you run a WINDOWS PROGRAM on the same machine for troubleshooting, when you remind them you are running Windows they tell you to run WINE when you remind them that even Digium recommends you do not run a GUI on the same machine as Asterisk they start to ignore you. On 12/15/06, Paul Connolly [EMAIL PROTECTED] wrote: We currently have an Asterisk system with a PRI and 6 POTs lines for backup. We are looking to add service such as Voicepulse Connect as an extra level of redundancy and a cost saving alternative to PRI calls. VP Connect only allows 4 simultaneous calls; we are looking for 4 to 5 times that to support our call center. Also, in looking through the archives, it seems like VP has had their share of outages and problems. Can anyone suggest a good commercial grade package/provider? ___ --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 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] Good Commercial Grade Service Provider?
Thanks! I will check them out _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of LST Sent: Friday, December 15, 2006 7:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Good Commercial Grade Service Provider? On 12/15/06, Paul Connolly [EMAIL PROTECTED] wrote: We currently have an Asterisk system with a PRI and 6 POTs lines for backup. We are looking to add service such as Voicepulse Connect as an extra level of redundancy and a cost saving alternative to PRI calls. VP Connect only allows 4 simultaneous calls; we are looking for 4 to 5 times that to support our call center. Also, in looking through the archives, it seems like VP has had their share of outages and problems. Can anyone suggest a good commercial grade package/provider? Check at teliax.com. I think they allow at least 10, maybe more. voipstreet.com allows at least 20. ___ --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 and outlook
Hi list. Has anyone used any commercial or open source application to integrate Asterisk into MS Outlook 2003 which can be used to place calls directly to contacts from Outlook? And if so how well does it work? Here you go... Enjoy:) http://www.bicomsystems.com/products/C/P/319/288/ Senad ___ --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 1.4.0 B4 Sounds Directory
You're right. I just untarred asterisk-1.4.0-beta4.tar.gz. The sounds folder is there, but it is empty except for Makefile and sounds.xml. I am not expert, but when I looked at the Makefile, it appears that it prompts the user to pick a format for the sounds files (ulaw, wav, etc), and then it downloads the appropriate sound files. John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 12/16/2006 2:04 PM Hi, I noticed that the sound directory is missing from asterisk-1.4.0-beta4.tar.gz. This directory (7 M) witch existed in asterisk-1.4.0-beta3.tar.gz has GSM Core Sounds and some MOH. Does anyone know why it has been removed from the latest beta? Regards. Sponsored Link Mortgage rates near historic lows: $150,000 loan as low as $579/mo. Intro-*Terms https://www2.nextag.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 - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in error, please immediately notify the sender by telephone or return email and delete the original transmission and its attachments without reading or saving in any manner. The Evangelical Lutheran Good Samaritan Society. - ___ --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 and outlook
I've never used it but... http://www.snapanumber.com/ Looks ok feature-wise - plus there's a free version to take for a test drive. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Soderblom Sent: 18 December 2006 14:46 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk and outlook Network Configurations Block D, Surrey Park, Barham Road, Westville, 3610 Helpdesk: (086) 163-8266 Tel: (031) 266-1563 Fax: (031) 266-4206 Hi list. Has anyone used any commercial or open source application to integrate Asterisk into MS Outlook 2003 which can be used to place calls directly to contacts from Outlook? And if so how well does it work? Thanks, Richard Best Regards Richard Soderblom Network Configurations Cell: E-Mail: [EMAIL PROTECTED] Number of Attachments: 0 This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ --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 and outlook
We are using the snap program in both Outlook 2003 and 2007, it also handle click to dial from all Microsoft office apps, FireFox, thunderbird and I believe Internet explorer. Check it out at http://www.snapanumber.com On 12/18/06, Senad Jordanovic [EMAIL PROTECTED] wrote: Hi list. Has anyone used any commercial or open source application to integrate Asterisk into MS Outlook 2003 which can be used to place calls directly to contacts from Outlook? And if so how well does it work? Here you go... Enjoy:) http://www.bicomsystems.com/products/C/P/319/288/ Senad ___ --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 -- Bruce Nortex Networks ___ --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] Good Commercial Grade Service Provider?
Please do not take this as a flame against cyberdyne-ip.com. That is not the intention. I am just wondering how businesses like this expect to stick around when they are charging rates this low. You can find a whole list of other providers that thought this model would work at: http://www.voip-info.org/wiki/view/RIP+VOIP The fact is... if you want good quality, reliable service, and reasonable support, I think you should expect to pay a little more. I would be very cautious. Just my $0.02. 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
[asterisk-users] Cisco 7940 - NAT Option
I am thinking of turning on the NAT option in our Cisco phones (and the corresponding sip.conf modification) to allow the phones to be taken outside the LAN. Can anyone think of any reason not to just always turn on the NAT enabled option? I can't think of a reason not to always operate these phones with this enabled, since it would likely allow them to be taken outside our LAN and used. Could it hurt something when they are used inside our LAN with NAT enabled? Sincerely, Brent A. Torrenga Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 tel:+1 219 836 8918 x325 fax:+1 219 836 1138 email:[EMAIL PROTECTED] web:www.torrenga.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
Re: [asterisk-users] Repeated Digits
I am experience repeated digits when connecting a call from SIP using any codex I have tried the same things to fix this. If anyone knows why please let me know. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-866-638-1254 For Information on PBX Systems for SOHO http://www.bochterservices.com/?j=PBXt=email Need A Toll Free Number? http://www.bochterservices.com/?t=TFdidt=email For new and used security items http://www.bochterservices.com/?j=storet=email BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email Gustavo Flores wrote: Hi, Have anyone experience repeated digits when connecting a call from SIP and terminating it to a PRI Channel? On the other side of the PRI Channel is an IVR that expect a pin but the digits come repeated. For example, you dial 12345 but it is received as 12224445 ___ --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] BLF on GXP2000
While I don't see anything wrong with this, I'm no expert. I took my instructions from the following URL and they worked fine... I have the subscribecontext in General and it works fine. What is the firmware on the GXP? old firmware may be related -t- http://www.jackenhack.com/blog/archives/2005/11/22/setting-up- subscribenotify-blf-in-asteriskhome-for-grandstream-gxp-2000-phones/ ___ --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] Shared Line Appearances (SLA) in 1.4
we probably need to ask in dev- list, because seems that only developers knows, how to use/test SLA feature ;-) Anthony Kava wrote: Greetings, Back in September someone asked about documentation for the new SLA feature in 1.4, however they received no replies. I thought I might ask the same question now in December. Apart from sla.conf.sample and a few comments in app_meetme.c I have been unable to find useful documentation. Is anyone using this feature right now? Is there a helpful source for information this highly desired capability? Regards, -- Anthony Kava Senior Network Administrator Pottawattamie County, Iowa Sheep are slow and tasty, and therefore must remain constantly alert. -- Bruce Schneier, Beyond Fear ___ --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] spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13 [ Virusgeprüft]
a few weeks ago I encountered the same problem. I found out that asterisk is crashing when app_rxfax.so is calling line 327 of app_rxfax.c 'ast_frfree(int);' out of the testing tree running with actual spandsp-0.0.3 commenting this line out it doesn't crash *, but that's no solution it do work with asterisk-1.2.9 but not with 1.2.13 - not tested 1.2.14 yet [EMAIL PROTECTED] schrieb am 18.12.2006 12:32:12: Hi On 12/18/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: Can you provide a backtrace of the crash? Sure. I've attached a backtrace for both 1.2.13 and 1.2.14 running the same version of spandsp and all other libraries. This is on a Fedora Core 6 machine (I can not attach the message as it makes the message over 40kB) http://www.avenard.org/asterisk/trace1-2-13.txt http://www.avenard.org/asterisk/trace1-2-14.txt Just saying it crashed doesn't really help. Well, the full backtrace was reported here last month, I was just pointing out that it was still happening with 1.2.14. Also: what libraries are involved? ldd /usr/lib/asterisk/modules/app_rxfax.so linked with spandsp 0.0.2 I get: linux-gate.so.1 = (0x00c6c000) libspandsp.so.0 = /usr/lib/libspandsp.so.0 (0x0071) libtiff.so.3 = /usr/lib/libtiff.so.3 (0x006b4000) libc.so.6 = /lib/libc.so.6 (0x001e7000) libm.so.6 = /lib/libm.so.6 (0x00e43000) libjpeg.so.62 = /usr/lib/libjpeg.so.62 (0x009c5000) libz.so.1 = /usr/lib/libz.so.1 (0x00d9e000) /lib/ld-linux.so.2 (0x00534000) I unfortunately can't try with spandsp 0.0.3 right now as I need a working asterisk ... linux-gate: spandsp: 0.0.3pre27 libtiff: 3.8.2 glibc: 2.5-3 libjpeg: 6b-37 and report what is the version and package of each library mentioned there. Any more automated way of doing this? This is standard Fedora Core 6. You can find last month, on this distribution list For the archive: http://lists.digium.com/pipermail/asterisk-users/2006-November/172652.html People mentioned this issue as well as where it was crashing. Hope that helps. Jean-Yves ___ --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 Diese E-Mail und alle Anhänge enthalten vertrauliche und/oder rechtlich geschützte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese E-Mail und ihren Inhalt. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser E-Mail ist nicht gestattet. This e-mail and any attached files may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail by mistake) please notify the sender immediately and delete this e-mail. Any unauthorised duplication, disclosure or distribution of this e-mail and content is strictly forbidden. ___ --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] Voicemail delivery
Hi. How do I cause voicemails that land in one mailbox to be delivered to another? I.e. I have a incoming call extension that rings all the phones. If it times out, the caller drops into the general mailbox. I would like messages dropped in the general mailbox to fall into another users mailbox. Thanks, Ejay ___ --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] Voicemail delivery
On Monday 18 December 2006 9:24 am, Ejay Hire wrote: Hi. How do I cause voicemails that land in one mailbox to be delivered to another? I.e. I have a incoming call extension that rings all the phones. If it times out, the caller drops into the general mailbox. I would like messages dropped in the general mailbox to fall into another users mailbox. Maybe this is what you want: [ringallphones] exten = 100,1,Dial(SIP/200SIP/201SIP/202,30,t) exten = 100,2,VoiceMail([EMAIL PROTECTED]) This specifies which voicemail box to use if no one answers. -- ~ Carla Schroder Linux geek and random computer tamer check out my Linux Cookbook! http://www.oreilly.com/catalog/linuxckbk/ best book for sysadmins and power 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] Re: Best way to access MySQL data from dial plan
Resending as message didn't show up the first time
I need to access MySQL from the dial plan. Currently I am using the MYSQL
function:
exten = *78,n,MYSQL(Connect asterisklocal localhost asteriskuser password
asterisk)
exten = *78,n,MYSQL(Query resultid ${asterisklocal} CALL\
sp_ins_into_avp(\'/DND/${CALLERID(number)}\'\,\'YES\'))
exten = *78,n,MYSQL(Clear ${resultid})
exten = *78,n,MYSQL(Disconnect ${asterisklocal})
This shows authentication details in the Asterisk CLI which is not ideal.
What is the recommended way to access MySQL data?
Asterisk 1.2
CentOS 4.4
MySQL 5.0
Regards
Cameron
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[asterisk-users] Colomachine TE405P
I was wonder if anyone is rumming this combination of hardware: Colomachine.com: CM62 Digium Card: TE405P I need a rackmount to send to a data center and this combination fits my budget. Has anyone else used colomachine with asterisk? how has it performed? I plan to run the latest trixbox on it with the RAM upgrade to 1GB. Should I go to the 2GB RAM? Will this combination handle all 4 spans at full load? I'm mostly intrested in realworld experiences. I hear a lot of don't trust because of the cheap price but when I talk to actual users they seem quite happy with these servers. Your opnions please. Mark C. IS Director Payroll Services Hawaii, Inc. http://www.psh-inc.com smime.p7s Description: S/MIME Cryptographic Signature ___ --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] IBM Server / USB Ports
Well I don't see anything that specifically states digium.. but I do see this. which would be a problem if this is the digium card.. 04:04.0 Ethernet controller: Unknown device d161:2400 (rev 11) IRQ 3 05:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5721 IRQ 3 On 12/15/06, Alejandro Kauffmann [EMAIL PROTECTED] wrote: I see that the digium card doesn't share the IRQ however Digium has recommended diabled USB still... additionally the Digium card is on 169 which isn't a valid IRQ.. how can I find out what it is sharing with? lspci -vb will give you the irq as seen by the cards on the PCI bus -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.15.21/589 - Release Date: 12/15/2006 5:10 PM ___ --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] GXP2000, Linksys RV082 Firewall / NAT, Registrations
Hello, We have several clients with GXP2000 in their network and behind NAT. We have one particular client that has several GXP2000 behind a Linksys RV082 VPN Firewall/Router which is doing NAT services. According to SIP packet inspection, it detects it's a symmetric NAT. The problem we have is that even though we have configured Asterisk AND the GXP2000 to register every 60 minutes, they keep unregistering and re-registering every few couple of minutes (between 4 and 10 minutes). We have checked the client's Internet connection and it's not bouncing as well as their local network is working stable. So, the only thing we can think of why these phones are re-registering so often is their firewall device since it's the only difference between this client and our other clients that use the GXP2000. Does anyone have any idea why this is happening or how this can be resolved? Thanks, Daniel ___ --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] stop logging certain error messages
Hi, Is there a way I can stop logging this specific messages: Dec 18 13:24:52 ERROR[6913] chan_sip.c: Call to user 'remi' rejected due to usage limit of 1 Without having to completely stop logging all error messages in my log files. Thanks, Remi ___ --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: Best way to access MySQL data from dial plan
Ciao kjcsb,
I need to access MySQL from the dial plan. Currently I am using the
MYSQL function:
exten = *78,n,MYSQL(Connect asterisklocal localhost asteriskuser
password asterisk)
exten = *78,n,MYSQL(Query resultid ${asterisklocal} CALL\
sp_ins_into_avp(\'/DND/${CALLERID(number)}\'\,\'YES\'))
exten = *78,n,MYSQL(Clear ${resultid})
exten = *78,n,MYSQL(Disconnect ${asterisklocal})
This shows authentication details in the Asterisk CLI which is not
ideal. What is the recommended way to access MySQL data?
Well, you can easily modify the MYSQL() application in order to prevent
it from showing auth data in the logs.
If you aren't a C programmer, I can write for you a small patch that
will get the job done.
Bye,
--
Andrea Spadaccini
Multimedia Technologies Institute s.r.l.
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Re: [asterisk-users] Asterisk + Orion E1 GSM Gateway
Hi yusuf, I am working right now on a similar setup. If its the PRI type theres not so much on the syncing part. You need the PRI crossover rj45, theres info on voip-info on that and Orion has software to configure via Serial cable the E1 PRI as NET/USER and Time syncs. I setup mine via zaptel using css,hdb3,crc on the span. I am still debugging outogoing traffic but incoming is working OK. Lex On 12/18/06, yusuf [EMAIL PROTECTED] wrote: Leo Ann Boon wrote: yusuf wrote: Hi, I just got hold on an Orion E1 30 port GSM Gateway, and I am having problems trying to get the E1 link to come up. I am using Asteisk 1.2.12 with a Sangoma A101 card. I am quite familiar with E1's, both the Digium and Samgoma types, as I have successfully hooked up to many PBX's and such, but I just cant seem to get this one to work. None of the 30 channels 'come up'. What signailling, crc checking, should I be Master or slave? Sanity check: Have you read the fine manual :)? I understand Orion makes both ISDN PRI/Q.SIG and MFC/R2 type E1 channel banks. If it's the PRI type, standard zaptel with the appropriate NET/CPE setting on the CB should be ok. If it's a MFC/R2, then you'll have to try unicall. Leo Hi, crazy thing is I dont have any manual or anything, just the Gateway. From reading the 'sales' doc on the Orion site, this is a PRI/Q.SIg type. But I dont have anything else besides that. I dont even know how to get the Serial cable to work to configure the Gateway (through Minicom/Hyperterminal, there is a configuration on Orion, or so I'm told.) Can you help? -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] stop logging certain error messages
On Mon, Dec 18, 2006 at 01:55:37PM -0500, Remi Quezada wrote: Hi, Is there a way I can stop logging this specific messages: Dec 18 13:24:52 ERROR[6913] chan_sip.c: Call to user 'remi' rejected due to usage limit of 1 Without having to completely stop logging all error messages in my log files. look for the text of the error message in chan_sip.c and patch it not to print this error. However, why do you believe that this error should not be printed? Or should it be rate-limited? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] GXP2000, Linksys RV082 Firewall / NAT, Registrations
Do you have STUN Enabled? I had similar when I had STUN turned on. I found it better to turn off stun and place in sip.conf nat=route. Also use NAT Keep-Alive on the ATA that is NAT Timeout on the Router. Good Luck, Mark Coccimiglio IS Director Payroll Services Hawaii, Inc. http://www.psh-inc.com [EMAIL PROTECTED] wrote: Hello, We have several clients with GXP2000 in their network and behind NAT. We have one particular client that has several GXP2000 behind a Linksys RV082 VPN Firewall/Router which is doing NAT services. According to SIP packet inspection, it detects it's a symmetric NAT. The problem we have is that even though we have configured Asterisk AND the GXP2000 to register every 60 minutes, they keep unregistering and re-registering every few couple of minutes (between 4 and 10 minutes). We have checked the client's Internet connection and it's not bouncing as well as their local network is working stable. So, the only thing we can think of why these phones are re-registering so often is their firewall device since it's the only difference between this client and our other clients that use the GXP2000. Does anyone have any idea why this is happening or how this can be resolved? Thanks, Daniel ___ --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] Re: Best way to access MySQL data from dial plan
I'm not sure that any solution with the MySQL dialplan command is going to be
ideal. You also can't nest your queries, ie the connectid/result id seems to
only be good for one resultset at a time... try doing something like
findme/followme with that!
Doug.
-Original Message-
From: kjcsb [mailto:[EMAIL PROTECTED]
Sent: Monday, December 18, 2006 11:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Re: Best way to access MySQL data from dial
plan
Resending as message didn't show up the first time
I need to access MySQL from the dial plan. Currently I am
using the MYSQL
function:
exten = *78,n,MYSQL(Connect asterisklocal localhost
asteriskuser password
asterisk)
exten = *78,n,MYSQL(Query resultid ${asterisklocal} CALL\
sp_ins_into_avp(\'/DND/${CALLERID(number)}\'\,\'YES\'))
exten = *78,n,MYSQL(Clear ${resultid})
exten = *78,n,MYSQL(Disconnect ${asterisklocal})
This shows authentication details in the Asterisk CLI which
is not ideal.
What is the recommended way to access MySQL data?
Asterisk 1.2
CentOS 4.4
MySQL 5.0
Regards
Cameron
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___
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Re: [asterisk-users] Need help getting started with asterisk
On Wed, 2006-12-13 at 14:21 +1100, Paul Hales wrote: I would be very surprised if your modem is supported by Asterisk - but I suppose it's worth a try. What does 'zap show status' and 'zap show channels' show in the Asterisk CLI? PaulH OK. I got the Motorola X100P put in: Relevant lspci -v output: 05:01.0 Communication controller: Motorola Wildcard X100P Subsystem: Motorola Unknown device Flags: bus master, medium devsel, latency 32, IRQ 22 I/O ports at b800 [size=256] Memory at ff90 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 However, although I can modprobe wctdm successfully, ztcfg still balks: camille ~ # modprobe wctdm camille ~ # /sbin/ztcfg -vv Zaptel Configuration == Channel map: Channel 02: FXS Kewlstart (Default) (Slaves: 02) 1 channels configured. ZT_CHANCONFIG failed on channel 2: No such device or address (6) zap show channels comes back with encouraging results though: camille*CLI zap show channels Chan Extension Context Language MusicOnHold pseudodefault Do I need to moprobe something else or alter my kernel config for ztcfg not to error? -Michael Sullivan- ___ --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] Need help getting started with asterisk
On Mon, Dec 18, 2006 at 02:10:23PM -0600, Michael Sullivan wrote: On Wed, 2006-12-13 at 14:21 +1100, Paul Hales wrote: I would be very surprised if your modem is supported by Asterisk - but I suppose it's worth a try. What does 'zap show status' and 'zap show channels' show in the Asterisk CLI? PaulH OK. I got the Motorola X100P put in: Relevant lspci -v output: 05:01.0 Communication controller: Motorola Wildcard X100P Subsystem: Motorola Unknown device Flags: bus master, medium devsel, latency 32, IRQ 22 I/O ports at b800 [size=256] Memory at ff90 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 However, although I can modprobe wctdm successfully, ztcfg still balks: camille ~ # modprobe wctdm You need the module wcfxo. But maybe it was hotplugged/ camille ~ # /sbin/ztcfg -vv Zaptel Configuration == Channel map: Channel 02: FXS Kewlstart (Default) (Slaves: 02) 02? Are you sure? 1 channels configured. That is: this is what ztcfg tries to do. ZT_CHANCONFIG failed on channel 2: No such device or address (6) But it fails. zap show channels comes back with encouraging results though: camille*CLI zap show channels Chan Extension Context Language MusicOnHold pseudodefault Do I need to moprobe something else or alter my kernel config for ztcfg not to error? -Michael Sullivan- What do you see on /proc/zaptel/* Try genzaptelconf ... -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Need help getting started with asterisk
On Mon, 2006-12-18 at 22:19 +0200, Tzafrir Cohen wrote: On Mon, Dec 18, 2006 at 02:10:23PM -0600, Michael Sullivan wrote: On Wed, 2006-12-13 at 14:21 +1100, Paul Hales wrote: I would be very surprised if your modem is supported by Asterisk - but I suppose it's worth a try. What does 'zap show status' and 'zap show channels' show in the Asterisk CLI? PaulH OK. I got the Motorola X100P put in: Relevant lspci -v output: 05:01.0 Communication controller: Motorola Wildcard X100P Subsystem: Motorola Unknown device Flags: bus master, medium devsel, latency 32, IRQ 22 I/O ports at b800 [size=256] Memory at ff90 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 However, although I can modprobe wctdm successfully, ztcfg still balks: camille ~ # modprobe wctdm You need the module wcfxo. But maybe it was hotplugged/ camille ~ # /sbin/ztcfg -vv Zaptel Configuration == Channel map: Channel 02: FXS Kewlstart (Default) (Slaves: 02) 02? Are you sure? Is that not right? That's what the pdf said - to put fxsks=2 in /etc/zaptel.conf 1 channels configured. That is: this is what ztcfg tries to do. ZT_CHANCONFIG failed on channel 2: No such device or address (6) But it fails. zap show channels comes back with encouraging results though: camille*CLI zap show channels Chan Extension Context Language MusicOnHold pseudodefault Do I need to moprobe something else or alter my kernel config for ztcfg not to error? -Michael Sullivan- What do you see on /proc/zaptel/* camille ~ # cat /proc/zaptel/* Span 1: WCFXO/0 Wildcard X100P Board 1 1 WCFXO/0/0 Span 2: ZTDUMMY/1 ZTDUMMY/1 1 Try genzaptelconf ... Where do I get that? camille ~ # genzaptelconf bash: genzaptelconf: command not found ___ --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] Need help getting started with asterisk
I changed fxsks=2 to fxsks=1 and now ztcfg works: camille ~ # ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. camille ~ # I bought the card on ebay. The seller sent some configuration lines, but didn't say where to put them: From the seller's email: here is some config. [channels] busydetect=yes busycount=6 language=en context=from-pstn signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no useincomingcalleridonzaptransfer=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both ___ --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] Need help getting started with asterisk
On Mon, Dec 18, 2006 at 02:29:20PM -0600, Michael Sullivan wrote: On Mon, 2006-12-18 at 22:19 +0200, Tzafrir Cohen wrote: On Mon, Dec 18, 2006 at 02:10:23PM -0600, Michael Sullivan wrote: On Wed, 2006-12-13 at 14:21 +1100, Paul Hales wrote: I would be very surprised if your modem is supported by Asterisk - but I suppose it's worth a try. What does 'zap show status' and 'zap show channels' show in the Asterisk CLI? PaulH OK. I got the Motorola X100P put in: Relevant lspci -v output: 05:01.0 Communication controller: Motorola Wildcard X100P Subsystem: Motorola Unknown device Flags: bus master, medium devsel, latency 32, IRQ 22 I/O ports at b800 [size=256] Memory at ff90 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 However, although I can modprobe wctdm successfully, ztcfg still balks: camille ~ # modprobe wctdm You need the module wcfxo. But maybe it was hotplugged/ camille ~ # /sbin/ztcfg -vv Zaptel Configuration == Channel map: Channel 02: FXS Kewlstart (Default) (Slaves: 02) 02? Are you sure? Is that not right? That's what the pdf said - to put fxsks=2 in /etc/zaptel.conf What PDF? 1 channels configured. That is: this is what ztcfg tries to do. ZT_CHANCONFIG failed on channel 2: No such device or address (6) But it fails. zap show channels comes back with encouraging results though: camille*CLI zap show channels Chan Extension Context Language MusicOnHold pseudodefault Do I need to moprobe something else or alter my kernel config for ztcfg not to error? -Michael Sullivan- What do you see on /proc/zaptel/* camille ~ # cat /proc/zaptel/* Span 1: WCFXO/0 Wildcard X100P Board 1 1 WCFXO/0/0 Zaptel channel 1, indeed. Not 2. Span 2: ZTDUMMY/1 ZTDUMMY/1 1 Try genzaptelconf ... Where do I get that? camille ~ # genzaptelconf bash: genzaptelconf: command not found xpp/utils/genzaptelconf in the zaptel source directory. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] ZAP problem
when placing calls to the system through SIP, I got these messages, Dec 19 00:26:55 WARNING[5570]: channel.c:2571 ast_request: No translator path exists for channel type Zap (native 68) to 256 Dec 19 00:26:55 NOTICE[5570]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) any explanation for this? 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] pap2/wrt54gs/asterisk
I am having trouble setting this system up and wonder if some one help me. Does anyone know what is missing if anything to get 2 phones on my asterisk home server to be able to call each other. I have a WRT54GS running OpenWRT/asterisk connected to a PAP2 with 2 extensions 5060/5061, this is on the lan side of my gateway/router WRT54G 192.168.1.1 BusyBox v1.00 (2006.11.07-01:40+) Built-in shell (ash) Asterisk 1.2.1, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer [EMAIL PROTECTED] = Connected to Asterisk 1.2.1 currently running on OpenWrt (pid = 5084) OpenWrt*CLI sip show settings Global Settings: SIP Port: 5060 Bindaddress:0.0.0.0 Videosupport: No AutoCreatePeer: No Allow unknown access: Yes Promsic. redir: No SIP domain support: No Call to non-local dom.: Yes URI user is phone no: No Our auth realm asterisk Realm. auth:No User Agent: Asterisk PBX MWI checking interval: 10 secs Reg. context: (not set) Caller ID: asterisk From: Domain: Record SIP history: Off Call Events:Off IP ToS: 0x0 OSP Support:No SIP realtime: Disabled Global Signalling Settings: --- Codecs: none Relax DTMF: No Compact SIP headers:No RTP Timeout:0 (Disabled) RTP Hold Timeout: 0 (Disabled) MWI NOTIFY mime type: application/simple-message-summary DNS SRV lookup: Yes Pedantic SIP support: No Reg. max duration: 3600 secs Reg. default duration: 120 secs Outbound reg. timeout: 20 secs Outbound reg. attempts: 0 Notify ringing state: Yes Default Settings: - Context:default Nat:RFC3581 DTMF: rfc2833 Qualify:0 Use ClientCode: No Progress inband:Never Language: (Defaults to English) Musicclass: default Voice Mail Extension: asterisk **sip.conf file* GNU nano 1.3.8File: sip.conf [general] context=default ; Default context for incoming calls allowguest=yes ; Allow or reject guest calls (default is yes, this can also be set to 'osp' ; if asterisk was compiled with OSP support. ;realm=mydomain.tld ; Realm for digest authentication ; defaults to asterisk ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records ; Disabling DNS SRV lookups disables the ; ability to place SIP calls based on domain ; names to some other SIP users on the Internet ;domain=OpenWrt ; Set default domain for this host ; If configured, Asterisk will only allow ; INVITE and REFER to non-local domains ; Use sip show domains to list local domains ;domain=mydomain.tld,mydomain-incoming ; Add domain and configure incoming context ; for external calls to this domain ;domain=192.168.1.130 ; Add IP address as local domain ;domain=192.168.1.135 ; You can have several domain settings ;allowexternalinvites=no; Disable INVITE and REFER to non-local domains ; Default is yes ;autodomain=yes ; Turn this on to have Asterisk add local host ; name and local IP to domain list. ;pedantic=yes ; Enable slow, pedantic checking for Pingtel ; and multiline formatted headers for strict ; SIP compatibility (defaults to no) ;tos=184; Set IP QoS to either a keyword or numeric val ;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none ;maxexpiry=3600 ; Max length of incoming registration we allow ;defaultexpiry=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY ;checkmwi=10; Default time
Re: [asterisk-users] ZAP problem
On Mon, Dec 18, 2006 at 11:22:13PM +0200, O.Kamal wrote: when placing calls to the system through SIP, I got these messages, Dec 19 00:26:55 WARNING[5570]: channel.c:2571 ast_request: No translator path exists for channel type Zap (native 68) to 256 Dec 19 00:26:55 NOTICE[5570]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) boomtime*CLI show audio codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INTBINARYHEX TYPENAME DESC 1 (1 0) (0x1) audiog723 (G.723.1) 2 (1 1) (0x2) audio gsm (GSM) 4 (1 2) (0x4) audioulaw (G.711 u-law) 8 (1 3) (0x8) audioalaw (G.711 A-law) 16 (1 4) (0x10) audiog726 (G.726) 32 (1 5) (0x20) audio adpcm (ADPCM) 64 (1 6) (0x40) audioslin (16 bit Signed Linear PCM) 128 (1 7) (0x80) audio lpc10 (LPC10) 256 (1 8)(0x100) audiog729 (G.729A) 512 (1 9)(0x200) audio speex (SpeeX) 1024 (1 10)(0x400) audioilbc (iLBC) 68 is 64 + 4, that is: only the bits for ulaw and slinear are set. 256 means that only g729 is supported. Your system cannot transcode g729 to ulaw: you don't have a g729 codec installed, probably. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] ZAP problem
Why do I need g729 license?, i am not doing any transcoding in the middle. it is all g729 passthrough. softphone---asterisk---zap On 12/18/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Dec 18, 2006 at 11:22:13PM +0200, O.Kamal wrote: when placing calls to the system through SIP, I got these messages, Dec 19 00:26:55 WARNING[5570]: channel.c:2571 ast_request: No translator path exists for channel type Zap (native 68) to 256 Dec 19 00:26:55 NOTICE[5570]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) boomtime*CLI show audio codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INTBINARYHEX TYPENAME DESC 1 (1 0) (0x1) audiog723 (G.723.1) 2 (1 1) (0x2) audio gsm (GSM) 4 (1 2) (0x4) audioulaw (G.711 u-law) 8 (1 3) (0x8) audioalaw (G.711 A-law) 16 (1 4) (0x10) audiog726 (G.726) 32 (1 5) (0x20) audio adpcm (ADPCM) 64 (1 6) (0x40) audioslin (16 bit Signed Linear PCM) 128 (1 7) (0x80) audio lpc10 (LPC10) 256 (1 8)(0x100) audiog729 (G.729A) 512 (1 9)(0x200) audio speex (SpeeX) 1024 (1 10)(0x400) audioilbc (iLBC) 68 is 64 + 4, that is: only the bits for ulaw and slinear are set. 256 means that only g729 is supported. Your system cannot transcode g729 to ulaw: you don't have a g729 codec installed, probably. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] ZAP problem
O.Kamal wrote: Why do I need g729 license?, i am not doing any transcoding in the middle. it is all g729 passthrough. softphone---asterisk---zap I believe you are. Zap is ulaw. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation 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] ZAP problem
What zap device do you have that encodes/decodes g729? - Original Message - From: O.Kamal To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, December 18, 2006 4:37 PM Subject: Re: [asterisk-users] ZAP problem Why do I need g729 license?, i am not doing any transcoding in the middle. it is all g729 passthrough. softphone---asterisk---zap___ --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] Need help getting started with asterisk
Here's where I stand: camille asterisk # ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. camille*CLI zap show status Description Alarms IRQbpviol CRC4 Wildcard X100P Board 1 OK 0 0 0 ZTDUMMY/1 1 UNCONFIGUR 0 0 0 camille*CLI zap show channels Chan Extension Context Language MusicOnHold pseudofrom-pstn en camille*CLI I entered all the configuration the seller from ebay sent me into /etc/asterisk/zapata.conf and continues with the pdf file. The pdf said to add the following lines to /etc/asterisk/extensions.conf: [incoming] ; incoming calls from the FXO port are directed to this context from zapata.conf exten = s,1,Answer( ) exten = s,2,Echo( ) This seemed strange to me because there was no incoming context explicitly declared in zapata.conf, but I added it to extensions.conf and restarted asterisk so that the new options would take effect. I called my home number (the number that's going into my computer) from my cell phone. I let it ring ten times. Nothing. I went back into extensions.conf and changed [incoming] to [from-pstn], restarted asterisk and tried again. Same results. What am I doing wrong? Why won't asterisk pick up? The pdf says it should... ___ --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] Need help getting started with asterisk
On Mon, Dec 18, 2006 at 03:53:55PM -0600, Michael Sullivan wrote: Here's where I stand: camille asterisk # ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. camille*CLI zap show status Description Alarms IRQbpviol CRC4 Wildcard X100P Board 1 OK 0 0 0 ZTDUMMY/1 1 UNCONFIGUR 0 0 0 rmmod ztdummy, while you're at it. camille*CLI zap show channels Chan Extension Context Language MusicOnHold pseudofrom-pstn en camille*CLI I entered all the configuration the seller from ebay sent me into /etc/asterisk/zapata.conf and continues with the pdf file. The pdf said to add the following lines to /etc/asterisk/extensions.conf: [incoming] ; incoming calls from the FXO port are directed to this context from zapata.conf exten = s,1,Answer( ) exten = s,2,Echo( ) This seemed strange to me because there was no incoming context explicitly declared in zapata.conf, but I added it to extensions.conf and restarted asterisk so that the new options would take effect. I called my home number (the number that's going into my computer) from my cell phone. I let it ring ten times. Nothing. I went back into extensions.conf and changed [incoming] to [from-pstn], restarted asterisk and tried again. Same results. next time use 'reload' What am I doing wrong? Why won't asterisk pick up? The pdf says it should... You seem to be trying to follow installation instructions that assume a number of things about your installation. You can set the context in zapata.conf to 'incoming' context = incoming before the line 'channel = 1' Anyway: show dialplan from-pstn -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Need help getting started with asterisk
On Tue, 2006-12-19 at 00:09 +0200, Tzafrir Cohen wrote: On Mon, Dec 18, 2006 at 03:53:55PM -0600, Michael Sullivan wrote: Here's where I stand: camille asterisk # ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. camille*CLI zap show status Description Alarms IRQbpviol CRC4 Wildcard X100P Board 1 OK 0 0 0 ZTDUMMY/1 1 UNCONFIGUR 0 0 0 rmmod ztdummy, while you're at it. camille*CLI zap show channels Chan Extension Context Language MusicOnHold pseudofrom-pstn en camille*CLI I entered all the configuration the seller from ebay sent me into /etc/asterisk/zapata.conf and continues with the pdf file. The pdf said to add the following lines to /etc/asterisk/extensions.conf: [incoming] ; incoming calls from the FXO port are directed to this context from zapata.conf exten = s,1,Answer( ) exten = s,2,Echo( ) This seemed strange to me because there was no incoming context explicitly declared in zapata.conf, but I added it to extensions.conf and restarted asterisk so that the new options would take effect. I called my home number (the number that's going into my computer) from my cell phone. I let it ring ten times. Nothing. I went back into extensions.conf and changed [incoming] to [from-pstn], restarted asterisk and tried again. Same results. next time use 'reload' What am I doing wrong? Why won't asterisk pick up? The pdf says it should... You seem to be trying to follow installation instructions that assume a number of things about your installation. You can set the context in zapata.conf to 'incoming' context = incoming before the line 'channel = 1' Anyway: show dialplan from-pstn camille*CLI zap show channels Chan Extension Context Language MusicOnHold pseudoincomingen camille*CLI show dialplan incoming [ Context 'incoming' created by 'pbx_config' ] 's' =1. Answer( ) [pbx_config] 2. Echo( )[pbx_config] camille*CLI -= 1 extension (2 priorities) in 1 context. =- camille*CLI ___ --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: Best way to access MySQL data from dial plan
I'm not sure that any solution with the MySQL dialplan command is going to be ideal. You also can't nest your queries, ie the connectid/result id seems to only be good for one resultset at a time... try doing something like findme/followme with that! Thanks What is a better way to do it then in terms of performance, security, and flexibility? Using exec and a shell script, or agi or something else? Regards Cameron ___ --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] ZAP problem
i have digium TDM2404E, I was thinking that zap devices are not related to any kind of codecs. I will try setting my soft phone and asterisk server to use ulaw, to see how things will go... On 12/18/06, Mailing List [EMAIL PROTECTED] wrote: What zap device do you have that encodes/decodes g729? - Original Message - *From:* O.Kamal [EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Monday, December 18, 2006 4:37 PM *Subject:* Re: [asterisk-users] ZAP problem Why do I need g729 license?, i am not doing any transcoding in the middle. it is all g729 passthrough. softphone---asterisk---zap ___ --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] Queue Monitor not mixing if using UNIQUEID in MONITOR_FILENAME
Hello Asterisk Users,
I guess the subject says the most of it; here goes some more
detail:
- Running Asterisk 1.2.14
- Objective: record all calls managed by a specific queue
- Name those files ${TIMESTAMP}-${CALLERIDNUM}-${UNIQUEID}
Facts:
- If the UNIQUEID chan var is used in the MONITOR_FILENAME,
before calling the Queue() application, the two legs of the call are
not mixed and I end up with the two separate -in / -out files
- If it isn't used (for example: using only ${TIMESTAMP]-${CALLERIDNUM})
then, the legs are mixed together...
Note:
- In my first attempt I never managed to get the legs mixed... Only
after some experiment, I understood (well, not 100% clear why!)
that I had to also to add to include recordagentcalls=yes and
monitor-join=yes in agents.conf !
Can anyone provide some insight into this ? Thanks in advance!
(see below for config)
--
Ex Vito
queues.conf:
[general]
persistentmembers = yes
[the_queue]
musiconhold = default
announce = the_announcement
strategy = ringall
servicelevel = 20
context = the_context
wrapuptime = 10
announce-frequency = 30
announce-holdtime = once
monitor-format = wav
monitor-join = yes
eventwhencalled = yes
eventmemberstatus = no
reportholdtime = no
member = SIP/sip0001
agents.conf:
[general]
persistentagents=yes
recordagencalls=yes
monitor-join = yes
[agents]
(no agents declared, as they are directly configured in the
queues.conf file)
extensions.conf:
...
[globals]
SUPPORT_MONITOR_PATH=/var/spool/asterisk/monitor/support
[the_context]
exten = 305,1,Answer()
exten =
305,n,Set(MONITOR_FILENAME=${SUPPORT_MONITOR_PATH}/${TIMESTAMP}-${CALLERIDNUM})
exten = 305,n,Queue(the_queue,t)
exten = 305,n,Hangup()
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Re: [asterisk-users] spandsp 0.0.3 RxFax fax recepti on crashes bristuffed asterisk 1.2.13 [Virusgeprüft]
On 12/19/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: a few weeks ago I encountered the same problem. I found out that asterisk is crashing when app_rxfax.so is calling line 327 of app_rxfax.c 'ast_frfree(int);' out of the testing tree running with actual spandsp-0.0.3 commenting this line out it doesn't crash *, but that's no solution it do work with asterisk-1.2.9 but not with 1.2.13 - not tested 1.2.14 yet I tried with all version of Asterisk since 1.2.9, all crashes at the same spot as you mentioned. I guess commenting the line is one solution, provided you restart asterisk so it doesn't leak memory too much. We don't receive that many faxes anyway... JY ___ --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] Digium TE405P with French E1 = Red Alert
Hi On 12/18/06, Noc Phibee [EMAIL PROTECTED] wrote: Hi it's Colt-Telecom. you have a TE405P ? you don't mention what's wrong with it 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
[asterisk-users] MWI, Realtime SIP, Voicemail and Extensions, UAs registered with SER
I have the following setup: - UAs registered with SER/OpenSER - SIP peers (non cached), extensions, voicemail setup (not message storage) defined in Asterisk 1.2 using Realtime When a message is left in the user's mailbox, no Notify message is sent to SER. 1. If the SIP peer is defined in sip.conf with a host=ser.domain.com then the notfy is sent to SER. 2. If realtimecache=yes is set in sip.conf and the SIP peer is visible with sip show peers and the SIP peer host field is set to ser.domain.com then the notify is sent to SER. I have read numerous articles regarding this including: - the posting http://www.asteriskguru.com/board/need-help-for-voicemail-notification-vt535.html refers to a patch noted on http://www.voip-info.org/wiki-Asterisk+at+large. The patch is listed under Method 3, which relies on sip peers being defined in sip.conf i.e. it doesn't work for non cached realtime. - Method 1 of http://www.voip-info.org/wiki-Asterisk+at+large describes a way to send the Notify direct to the SIP UA. This relies on the phone contact details (e.g. IP address) being defined in sip.conf - not applicable in my case. - Method 2 of http://www.voip-info.org/wiki-Asterisk+at+large relates to SIP UAs registered with SER and states that Asterisk sends NOTIFY only to UACs that are registered at the Asterisk. This is not the case as described in 1 above and Method 5 of Asterisk-at-large. - Method 4 of http://www.voip-info.org/wiki-Asterisk+at+large assumes cached SIP realtime peers. I don't want to cache. - the posting http://forums.digium.com/viewtopic.php?t=4363highlight relates to SIP UAs registered with Asterisk, not those registered with SER. - the article http://openser.org/dokuwiki/doku.php/asterisk:realtime-integration doesn't deal with MWI. - the posting http://asterisk.mdaniel.net/?p=14 creates the msgnum.txt file on a remote Asterisk server and so is not relevant to my scenario. Can anyone advise how they are sending SIP Notify messages from Asterisk to SER for non-cached realtime SIP peers? Regards Cameron ___ --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] GXP2000, Linksys RV082 Firewall / NAT, Registrations
Did I forget to mention I had STUN enabled? :) Well, that did it. Your suggestion worked perfectly. Does anyone know what a reasonable NAT Keep-Alive to use, if you don't have access to their firewall/router configuration? Thanks, Daniel -Original Message- From: Mark Coccimiglio [EMAIL PROTECTED] Sent: Mon, December 18, 2006 2:27 pm To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] GXP2000, Linksys RV082 Firewall / NAT,Registrations Do you have STUN Enabled? I had similar when I had STUN turned on. I found it better to turn off stun and place in sip.conf nat=route. Also use NAT Keep-Alive on the ATA that is NAT Timeout on the Router. Good Luck, Mark Coccimiglio IS Director Payroll Services Hawaii, Inc. http://www.psh-inc.com [EMAIL PROTECTED] wrote: Hello, We have several clients with GXP2000 in their network and behind NAT. We have one particular client that has several GXP2000 behind a Linksys RV082 VPN Firewall/Router which is doing NAT services. According to SIP packet inspection, it detects it's a symmetric NAT. The problem we have is that even though we have configured Asterisk AND the GXP2000 to register every 60 minutes, they keep unregistering and re-registering every few couple of minutes (between 4 and 10 minutes). We have checked the client's Internet connection and it's not bouncing as well as their local network is working stable. So, the only thing we can think of why these phones are re-registering so often is their firewall device since it's the only difference between this client and our other clients that use the GXP2000. Does anyone have any idea why this is happening or how this can be resolved? Thanks, Daniel ___ --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] MWI, Realtime SIP, Voicemail and Extensions, UAs registered with SER
I have the following setup: - UAs registered with SER/OpenSER - SIP peers (non cached), extensions, voicemail setup (not message storage) defined in Asterisk 1.2 using Realtime When a message is left in the user's mailbox, no Notify message is sent to SER. 1. If the SIP peer is defined in sip.conf with a host=ser.domain.com then the notfy is sent to SER. 2. If realtimecache=yes is set in sip.conf and the SIP peer is visible with sip show peers and the SIP peer host field is set to ser.domain.com then the notify is sent to SER. I have read numerous articles regarding this including: - the posting http://www.asteriskguru.com/board/need-help-for-voicemail-notification-vt535.html refers to a patch noted on http://www.voip-info.org/wiki-Asterisk+at+large. The patch is listed under Method 3, which relies on sip peers being defined in sip.conf i.e. it doesn't work for non cached realtime. - Method 1 of http://www.voip-info.org/wiki-Asterisk+at+large describes a way to send the Notify direct to the SIP UA. This relies on the phone contact details (e.g. IP address) being defined in sip.conf - not applicable in my case. - Method 2 of http://www.voip-info.org/wiki-Asterisk+at+large relates to SIP UAs registered with SER and states that Asterisk sends NOTIFY only to UACs that are registered at the Asterisk. This is not the case as described in 1 above and Method 5 of Asterisk-at-large. - Method 4 of http://www.voip-info.org/wiki-Asterisk+at+large assumes cached SIP realtime peers. I don't want to cache. - the posting http://forums.digium.com/viewtopic.php?t=4363highlight relates to SIP UAs registered with Asterisk, not those registered with SER. - the article http://openser.org/dokuwiki/doku.php/asterisk:realtime-integration doesn't deal with MWI. - the posting http://asterisk.mdaniel.net/?p=14 creates the msgnum.txt file on a remote Asterisk server and so is not relevant to my scenario. Can anyone advise how they are sending SIP Notify messages from Asterisk to SER for non-cached realtime SIP peers? Regards Cameron ___ --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] BLF on GXP2000
Well, I am making some progress. I have made some changes as defined below
and now have a green line on the BLF, but it still does not indicate when
the extension receives a call or goes off hook.
Here are the changes:
the [ext-local-custom] context no longer exists
the subscribecontext in sip.con no longer exists
[internal]
exten = 101,1,Macro(voicemail,${polycom430})
exten = 101,hint,${polycom430}
Asterisk 1.4.0b3
*CLI show hints
-= Registered Asterisk Dial Plan Hints =-
[EMAIL PROTECTED]: SIP/101
State:Idle Watchers 1
- 1 hints registered
On 12/18/06, Ken Williams [EMAIL PROTECTED] wrote:
Here's what I have, it's to early for me to think so hopefully looking at
mine helps :D
extensions.conf:
[ext-local]
exten = 701,1,Macro(exten-vm,701,701)
exten = 701,n,Hangup
exten = 701,hint,SIP/701
sip.conf:
[701]
type=friend
secret=1234
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
nat=yes
host=dynamic
dtmfmode=rfc2833
dial=SIP/701
context=from-internal
canreinvite=no
callerid=device 701
mailbox=701
If this doesn't help in some fashion let me know and I'll think it through
a little later...off to get some coffee.
--
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Chris Johnson
*Sent:* Sunday, December 17, 2006 4:50 PM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] BLF on GXP2000
I am trying to set up the BLF on a GXP2000.
Currently what I have is
extensions.conf:
[globals]
polycom430=SIP/101
[internal]
exten = 101,1,Macro(voicemail,${polycom430})
[macro-voicemail]
exten = s,1,Dial(${ARG1},10,tT)
exten = s,2,VoiceMail([EMAIL PROTECTED] )
exten = s,102,VoiceMail([EMAIL PROTECTED])
[ext-local-custom]
exten = 101,hint,${polycom430}
sip.conf:
[general]
subscribecontext=ext-local-custom
And have set up the key to Asterisk BLF with UserID101
When I reload the phone, I get the following error:
[Dec 17 13:34:33] ERROR[2551]: chan_sip.c:14064 handle_request_subscribe:
Got SUBSCRIBE for extension [EMAIL PROTECTED] from 192.168.1.248 , but there is
no hint for that extension
Any help is greatly appreciated.
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[asterisk-users] RE: Best way to access MySQL data from dial plan
What is a better way to do it then in terms of performance, security,
and
flexibility? Using exec and a shell script, or agi or something else?
Setup extconfig to have realtime access to the database/table you want to
pull info from, then in the dialplan use the app Realtime.
-= Info about application 'RealTime' =-
[Synopsis]
Realtime Data Lookup
[Description]
Use the RealTime config handler system to read data into channel variables.
RealTime(family|colmatch|value[|prefix])
All unique column names will be set as channel variables with optional
prefix to the name.
e.g. prefix of 'var_' would make the column 'name' become the variable
${var_name}
This will not show any auth info in the asterisk cli and automatically
clears connect and fetch id's, works great and decreases the number of
priority routines within an extension.
JR
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[asterisk-users] Billing solution
Can anyone recommend a call accounting solution with rating for post paid billing that works well with asterisk using the account code or any other info from the CDR? I don't want the billing software to any phone calls for me, therefore any solution that modifies my extensions.conf is out, nor does it have to allow for customers the ability to log in to check their usage/balances. I have looked at astbill but it looks to be way overcomplicated for what I want, as well as it requires realtime. Thank you ___ --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] GXP2000, Linksys RV082 Firewall / NAT, Registrations
I prefer to keep my NAT Timeouts short ( ~5 minutes) and lets the applications be responsible for keeping the connections open. **Most** consumer grade routers use a timeout interval of 1 hour to 1 day. A safe figure to start with is 600 seconds (10 minutes) and see if anyone complains. [EMAIL PROTECTED] wrote: Did I forget to mention I had STUN enabled? :) Well, that did it. Your suggestion worked perfectly. Does anyone know what a reasonable NAT Keep-Alive to use, if you don't have access to their firewall/router configuration? Thanks, Daniel -Original Message- From: Mark Coccimiglio [EMAIL PROTECTED] Sent: Mon, December 18, 2006 2:27 pm To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] GXP2000, Linksys RV082 Firewall / NAT,Registrations Do you have STUN Enabled? I had similar when I had STUN turned on. I found it better to turn off stun and place in sip.conf nat=route. Also use NAT Keep-Alive on the ATA that is NAT Timeout on the Router. Good Luck, Mark Coccimiglio IS Director Payroll Services Hawaii, Inc. http://www.psh-inc.com [EMAIL PROTECTED] wrote: Hello, We have several clients with GXP2000 in their network and behind NAT. We have one particular client that has several GXP2000 behind a Linksys RV082 VPN Firewall/Router which is doing NAT services. According to SIP packet inspection, it detects it's a symmetric NAT. The problem we have is that even though we have configured Asterisk AND the GXP2000 to register every 60 minutes, they keep unregistering and re-registering every few couple of minutes (between 4 and 10 minutes). We have checked the client's Internet connection and it's not bouncing as well as their local network is working stable. So, the only thing we can think of why these phones are re-registering so often is their firewall device since it's the only difference between this client and our other clients that use the GXP2000. Does anyone have any idea why this is happening or how this can be resolved? Thanks, Daniel ___ --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] Need help getting started with asterisk
Could the fact that asterisk isn't aswering the phone be a firewall issue? What port(s) on TCP and UDP do I need to open for incoming calls to be allowed to go to 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
[asterisk-users] Inform callers on recorded/monitored number.
Hi, How could I possibly inform incoming callers that the number they'd dialed is monitored and recorded. I wanted that when a call-in or call-out is made, a playback will be played to inform caller callee that thier line is monitored prior to start conversation. Thanks. Angel __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
Re: [asterisk-users] Need help getting started with asterisk
camille*CLI zap show channels Chan Extension Context Language MusicOnHold pseudoincomingen This should show something like this : panoramix*CLI zap show channels Chan Extension Context Language MusicOnHold pseudofrom-pstn en 1from-pstn en so something is missing as Asterisk doesn't see your Zap channel what does your zapata.conf looks like ? ___ --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] HITBSecConf2007 - Dubai - Call for Papers now open!
The call for papers for the upcoming Hack in The Box Security Conference 2007 - Dubai is now open. HITBSecConf2007 - Dubai will take place at The Sheraton Creek hotel and will run from the 2nd till the 5th of April 2007. Keynote speakers for the conference will be Mikko Hypponen (Chief Research Officer, F-Secure Corporation) and Lance Spitzner (Founder, Honeynet Project). Security researchers based in an around the Middle East are encouraged to apply. SUBMISSION HITBSecConf is a deep-knowledge technical conference. Talks that are more technical or that discuss new and never before seen attack methods are of more interest than a subject that has been covered several times before. Summaries not exceeding 250 words should be submitted (in plain text format) to cfp -at- hackinthebox.org for review and possible inclusion in the programme. Submissions are due no later than 1st of February 2007 TOPICS Topics of interest include, but are not limited to the following: # Analysis of network and security vulnerabilities # Firewall technologies # Intrusion detection # Data Recovery and Incident Response # GPRS and CDMA Security # Identification and Entity Authentication # Network Protocol and Analysis # Smart Card Security # Virus and Worms # WLAN and Bluetooth Security. # Analysis of malicious code # Applications of cryptographic techniques, # Analysis of attacks against networks and machines # Denial-of-service attacks and countermeasures # File system security # Security in heterogeneous and large-scale environments # Techniques for developing secure systems PLEASE NOTE: We do not accept product or vendor related pitches. If your talk involves an advertisement for a new product or service your company is offering, please do not submit. Your submission should include: # Name, title, address, email and phone/contact number # Draft of the proposed presentation (in PDF or PowerPoint format), proof of concept for tools and exploits, etc. # Short biography, qualification, occupation, achievement and affiliations (limit 150 words). # Summary or abstract for your presentation (limit 250 words) # Time (45-60 minutes including time for discussion and questions) # Technical requirements (video, internet, wireless, audio, etc.) Each non-resident speaker will receive accommodation for 3 nights at The Sheraton Creek hotel in Dubai. For each non-resident speaker, HITB will cover travel expenses (through our airline partners, Emirates Airlines and Malaysia Airlines) up to USD 1,000.00. For further details please take a look at the CFP page: http://conference.hitb.org/hitbsecconf2007dubai/?page_id=72 Warm regards, The HITB Team HITBSecConf2007 - Dubai http://conference.hitb.org/hitbsecconf2007dubai/ ___ --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] openwrt wrt54gs running asterisk/pap2
I have asterisk running in a wrt54gs attached is a pap2 with 2 extensions working on it, the problem now is that there is lots of echo, some rythm in the background, and the voice is delayed by about 4 or 5 sec's between the 2 extensions. memory usage is about 15 to 20 megs so I think I can solve the problem with correct settings, anyone know where I might start to correct these issues Familyguy ___ --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: Best way to access MySQL data from dial plan
Setup extconfig to have realtime access to the database/table you want to pull info from, then in the dialplan use the app Realtime. Thanks. I didn't know that you could use RealTime in the dialplan like that. I thought is was just for sip, extensions etc. I created a wiki page at http://www.voip-info.org/wiki/view/Asterisk+cmd+RealTime. Feel free to edit if it's wrong! Cameron ___ --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 7914 with sccp
I was wondering if anyone had any experience getting a 7960+7914 working with any of the chan_sccp modules. I've got a 7960G with 6.0(5.0) and a factory fresh 7960G with 3.1(MF.G2). I've got 2 7914s fresh out of the box brand new. I hook them up and all I get is red lights on all of the buttons. When I go into the phone to see what version firmware they have it says Link State: Not Supported and that the expansion module is not connected. This is the case on both phones with either 7914 or both. I'm setup just like the 7914 howto on voip-info says to be. Any idea where I went wrong? I've searched this list back through June 2005 and I don't see anything that helps and I've spent hours searching on google only ending up with dead ends. Any pointers would be greatly appreciated. Thanks, -Ryan ___ --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] Follow-me challenge
The problem I am running into is that when the call to my cellphone is made,
it appears as though the call completes so it never rolls to asterisk
voicemail.
Here is my current config:
exten = 102,1,Dial(${sipura},10,)
exten = 102,n,playback(pls-wait-connect-call)
exten = 102,n,Dial(IAX2/asterisk1/9139275900,10,r)
exten = 102,n,VoiceMail([EMAIL PROTECTED])
exten = 102,107,VoiceMail([EMAIL PROTECTED])
Here is the log from asterisk:
-- Executing [EMAIL PROTECTED]:2] Playback(SIP/101-0a1178c0,
pls-wait-connect-call) in new stack
-- Playing 'pls-wait-connect-call' (language 'en')
-- Executing [EMAIL PROTECTED]:3] Dial(SIP/101-0a1178c0,
IAX2/asterisk1/9139275900|10|r) in new stack
-- Called asterisk1/9139275900
-- Call accepted by 192.168.1.2 (format ulaw)
-- Format for call is ulaw
-- IAX2/asterisk1-7 answered SIP/101-0a1178c0
-- Hungup 'IAX2/asterisk1-7'
The one thing I will note is that there is not an analog trunk in this
server. It hands off the outbound call to trixbox running on another server,
which I fear may be my problem. Having said that, I will also note that I
have had the same challenge trying to get follow-me set up on trixbox as
well.
Thanks
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Re: [asterisk-users] Need help getting started with asterisk
On Mon, 2006-12-18 at 22:03 -0500, Time Bandit wrote: camille*CLI zap show channels Chan Extension Context Language MusicOnHold pseudoincomingen This should show something like this : panoramix*CLI zap show channels Chan Extension Context Language MusicOnHold pseudofrom-pstn en 1from-pstn en so something is missing as Asterisk doesn't see your Zap channel what does your zapata.conf looks like ? I have removed the commented out lines for conciseness: [trunkgroups] [channels] language=en context=incoming switchtype=national signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=yes echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no busydetect=yes busycount=6 useincomingcalleridonzaptransfer = yes Is this right for my situation with the X100P? ___ --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] Inform callers on recorded/monitored number.
With the playback command? I think we are missing something here. PaulH On Mon, 2006-12-18 at 19:01 -0800, Angel Heart wrote: Hi, How could I possibly inform incoming callers that the number they'd dialed is monitored and recorded. I wanted that when a call-in or call-out is made, a playback will be played to inform caller callee that thier line is monitored prior to start conversation. Thanks. Angel __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
Re: [asterisk-users] Follow-me challenge
Is the problem just when you don¹t answer the cell phone? Many cell phones
go to a voice announcement when they¹re turned off or not answered, and
Asterisk thinks the call has been answered. The other issue could be that
your gateway (asterisk1) is answering the call before the outbound leg is
answered. One workaround would be to use a macro that requires you to press
a key to accept the call on your cell. (See the M option to the dial
command and http://www.voip-info.org/wiki/view/Asterisk+tips+findme)
Also, I see that you¹re using the r option you might want to drop that.
I¹m also not convinced that it will ever find 102,107 in your dialplan. You
might want to look at using ${DIALSTATUS} and making it a bit more explicit.
Cheers,
Eric
On 2006-12-18 23:09, Chris Johnson [EMAIL PROTECTED] wrote:
The problem I am running into is that when the call to my cellphone is made,
it appears as though the call completes so it never rolls to asterisk
voicemail.
Here is my current config:
exten = 102,1,Dial(${sipura},10,)
exten = 102,n,playback(pls-wait-connect-call)
exten = 102,n,Dial(IAX2/asterisk1/9139275900,10,r)
exten = 102,n,VoiceMail( [EMAIL PROTECTED])
exten = 102,107,VoiceMail([EMAIL PROTECTED])
Here is the log from asterisk:
-- Executing [EMAIL PROTECTED]:2] Playback(SIP/101-0a1178c0,
pls-wait-connect-call) in new stack
-- Playing 'pls-wait-connect-call' (language 'en')
-- Executing [EMAIL PROTECTED]:3] Dial(SIP/101-0a1178c0,
IAX2/asterisk1/9139275900|10|r) in new stack
-- Called asterisk1/9139275900
-- Call accepted by 192.168.1.2 http://192.168.1.2 (format ulaw)
-- Format for call is ulaw
-- IAX2/asterisk1-7 answered SIP/101-0a1178c0
-- Hungup 'IAX2/asterisk1-7'
The one thing I will note is that there is not an analog trunk in this server.
It hands off the outbound call to trixbox running on another server, which I
fear may be my problem. Having said that, I will also note that I have had the
same challenge trying to get follow-me set up on trixbox as well.
Thanks
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--
Eric Jacksch
+1 613 860-0964 Ottawa
+1 647 722-3544 Toronto
+1 514 907-0031 Montreal
They who would give up an essential liberty for temporary security deserve
neither liberty or security. -- Benjamin Franklin
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Re: [asterisk-users] Inform callers on recorded/monitored number.
exten = s,1,Answer
exten =
s,n,Set(REC=${URIENCODE(${STRFTIME(,America/Toronto,%Y%m%d-%H%M%S)}-${CALLER
ID(number)}-TESTBOARD-${UNIQUEID})})
exten = s,n,MixMonitor(${REC}.wav)
exten = s,n,Playback(this-call-may-be-monitored-or-recorded)
Note that I intentionally start the recording BEFORE advising the user that
the call may be monitored that way the first thing on the recording is the
user being advised of the recording.
On 2006-12-18 22:01, Angel Heart [EMAIL PROTECTED] wrote:
Hi,
How could I possibly inform incoming callers that the number they'd dialed is
monitored and recorded.
I wanted that when a call-in or call-out is made, a playback will be played to
inform caller callee that thier line is monitored prior to start
conversation.
Thanks.
Angel
__
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http://mail.yahoo.com
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--
Eric Jacksch
+1 613 860-0964 Ottawa
+1 647 722-3544 Toronto
+1 514 907-0031 Montreal
They who would give up an essential liberty for temporary security deserve
neither liberty or security. -- Benjamin Franklin
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RE: [asterisk-users] Cisco 7940 - NAT Option
. Could it hurt something when they are used inside our LAN with NAT enabled? The answer is no! With my test bed, I found that Asterisk can detect Endpoint behind NAT(match via and src_ip). So, once the EP is on LAN (same side of NAT) then they work as if there is no NAT. The option of nat=yes is immaterial. Thanks and Regards --Sandeep Kalra Ph: +91-120-4342000-X-2966 : +91-120-4342966 (direct) www.globallogic.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Torrenga Sent: Monday, December 18, 2006 9:37 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Cisco 7940 - NAT Option I am thinking of turning on the NAT option in our Cisco phones (and the corresponding sip.conf modification) to allow the phones to be taken outside the LAN. Can anyone think of any reason not to just always turn on the NAT enabled option? I can't think of a reason not to always operate these phones with this enabled, since it would likely allow them to be taken outside our LAN and used. Could it hurt something when they are used inside our LAN with NAT enabled? Sincerely, Brent A. Torrenga Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 tel:+1 219 836 8918 x325 fax:+1 219 836 1138 email:[EMAIL PROTECTED] web:www.torrenga.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] Changing CALLERIDNUM on the fly
Is what I am trying to do in this context possible. That is changing the
incoming CALLERIDNUM. In this case if the incoming CALLERIDNUM is not
preceeded by a 1 I want to add a 1. Often calls come in without the
preceeding 1 and this plays havoc with my redial if the 3 digit area
code matches a local 3 digit extension. All my outside calls are 10 digits
or 1+10 digits.
Doug
[from-pstn]
exten = s,1,set(TEST=${CALLERIDNUM::1}) test for first digit1
exten = s,2,GotoIf($[ ${TEST} = 1 ]?4:3)
exten = s,3,set(__CALLERIDNUM=1${CALLERIDNUM}) if not add 1
exten = s,4,noop(${CALLERIDNUM}) and this still displays without
I tried no, one and two underscores with the CALLERIDNUM variable.
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