Re: [asterisk-users] force ulaw passthrough if call from modem extension?
Tried that, it didn't work but maybe I didn't configure it right. Anyways how can I route all outgoing calls from that specific extension to use that trunk? Thanks. On 1/16/07, Tim Panton [EMAIL PROTECTED] wrote: On 16 Jan 2007, at 19:56, Victor Perez wrote: I have Teliax trunk set to ulaw and g729 and I have a modem/fax extension from a sipura forced to ulaw. When the call goes out through Teliax IAX trunk, asterisk transcodes to g729. Is there a way to tell asterisk not to transcode calls from/to a specific extension? try creating a separate (duplicate) entry in iax.conf for the teliax connection disallow 729 on that trunk and use it for fax/alarm calls. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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] force ulaw passthrough if call from modem extension?
I have Teliax trunk set to ulaw and g729 and I have a modem/fax extension from a sipura forced to ulaw. When the call goes out through Teliax IAX trunk, asterisk transcodes to g729. Is there a way to tell asterisk not to transcode calls from/to a specific extension? I'm running asterisk 1.2.4 and that extension is for my home alarm/dish network and fax calls. 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] OT: Looking for asterisk integrators in Dallas,TX
Sorry for posting this OT: If you are an asterisk integrator in the Dallas Area or are willing to travel for a Presentation please mail me to [EMAIL PROTECTED] Thank you, Victor Perez ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk success histories in business?
My company needs to replace our old phone switch and we are evaluating different options including VoIP. I am trying to sell the option of building our own asterisk server but they are not convinced on its reliability and prefer to invest in proven technology. I need success histories, any of you wearing brass in your companies that have successfully replaced phone systems with an asterisk-based VoIP solution and are willing to have a phone conversation with our management about the experience. We have around 100 stations counting desks and fax machines so any of you with similar or bigger installations would be very helpful. Please email me to [EMAIL PROTECTED] Thank you, Victor Perez ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: IAX provider for business
I would like to know if there ir any business level VoIP IAX provider with the ability to support the following: - GSM, uLaw (for FAX), G729 - 20 to 40 simultaneous phone conversations at any time - Ability to route DID numbers in Dallas area from existing providers (SBC, XO) - Flat rate for at least local calls (DFW) - National and international Toll Free DID numbers Altought OT, I believe this information would be useful for all asterisk integrators reading this forum so you can reply to this message or mail me to [EMAIL PROTECTED] Thank you, Victor Perez ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura 2000 intermitent failure to register
I have asterisk 1.0.2 and a Sipura SPA-2000 (firmware 2.0.6(c) ). Today it started to log registration failed at intermitent periods. It registers fine, after a few minutes it can no longer register, then after a few minutes it registers fine again. I am wondering if there is a known issue with either asterisk or that sipura firmware. Victor Perez ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with new sipura firmware 1.0.35a
I just tried to upgrade my sipura to firmware 1.0.35a and now I can't connect to it. It still works but any connection to ports 23 and 80 makes it reboot. Even the flash tool makes it to crash when trying to connect. Anybody else experiencing this problem? Regards, Victor Perez ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oh323 goes silent after 5 seconds
I have this problem trying to talk to an ADDPAC gateway using oh323, when I call the sound is great for the first 5 seconds then it goes almost silent... all you can hear are some clicks every once in a while. Anybody seen this can point me to some config settings to change? Regards, Victor Perez
RE: [Asterisk-Users] Sipura line 1 outgoing voice problem?
I don't know if this helps, but I started having this problem after I sent out a fax. My fax machine was connected to line 1 at that time. I tried changing the FAX detection settings but no luck. Regards, Victor Perez -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mark Musone Sent: Sunday, April 18, 2004 10:33 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Sipura line 1 outgoing voice problem? I'm not sure if anybody has determined the cause/fix for this problem, but I am getting the same problem. I turned on syslog debugging and there were some interesting results: ... My feeling is that this is a Sipura problem. I've upgraded to the firmare 2.02, but still no difference. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call transfer with sipura
I can transfer a call from my sipura using flash, *98 and number, the problem is If I hangup before the destination extension picks-up, the transfer is lost. Is there a way to transfer and hangup without having to wait for the destination extension to pickup? Regards, Victor Perez ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] too many arguments to function `ast_queue_hangup' compiling asterisk-oh323
when trying to build asterisk-oh323 I get the following: make[1]: Entering directory `/usr/src/asterisk-oh323-0.5.10/asterisk-driver' gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declara tions -D_REENTRANT -D_GNU_SOURCE -I/usr/src/asterisk/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c chan_oh323.c: In function `oh323_call': chan_oh323.c:1128: error: too many arguments to function `ast_queue_hangup' chan_oh323.c:1142: error: too many arguments to function `ast_queue_hangup' chan_oh323.c: In function `oh323_hangup': chan_oh323.c:1182: error: too many arguments to function `ast_queue_hangup' chan_oh323.c: In function `oh323_read': chan_oh323.c:1581: error: too many arguments to function `ast_dsp_process' chan_oh323.c: In function `ast_oh323_new': chan_oh323.c:2030: warning: assignment from incompatible pointer type chan_oh323.c: In function `cleanup_h323_connection': chan_oh323.c:2835: error: too many arguments to function `ast_queue_hangup' make[1]: *** [chan_oh323.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.5.10/asterisk-driver' make: *** [subdirs_all] Error 1 I checked chan_oh323.c and indeed it only takes one parameter now so I am wondering what was that old parameter for and when did they take it off so I may try pulling that version of asterisk to try with. Regards, Victor Perez ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ADDPAC 200 w/SIP
I am trying to connect an ADDPAC 200 to * using SIP protocol. Problem is the documentation for the ADDPAC is not very helpful (they don't mention SIP at all, seems to be a newer firmware). I think I got the * part working but the ADDPAC setup is the problem, I will appreciate if somebody can post a working example or point me to documentation in english (found stuff in russian and korean so far) Regards, Victor Perez ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] default caller id from X100P
Is there a way to set default caller id info to pass to * when the telco does not provide it? Regards, Victor Perez ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] small linux distro to run * in old boxes
Has anybody tried to install * in any of these minimalist linux distros like tinylinux? Which linux distro would you use to run * in old P2, P3 boxes? Regards, Victor Perez [EMAIL PROTECTED] (469) 221-4189 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] wcfxo module fail to load (Unable to request IRQ 0)
Hello, just compiled zaptel in mandrake 9.2 and this is what I get when trying
modprobe wcfxo:
Apr 9 11:35:15 localhost kernel: PCI: No IRQ known for interrupt pin A of devic
e 00:05.0. Please try using pci=biosirq.
Apr 9 11:35:15 localhost kernel: Setting hook state to 0 (08)
Apr 9 11:35:15 localhost kernel: Registered Span 1 ('WCFXO/0') with 1 channels
Apr 9 11:35:15 localhost kernel: Span ('WCFXO/0') is new master
Apr 9 11:35:15 localhost kernel: PCI: Setting latency timer of device 00:05.0 t
o 64
Apr 9 11:35:15 localhost kernel: wcfxo: Unable to request IRQ 0
I have this same setup (asterisk on mandrake 9.2) already working in other pc... this
is an old AT pc... any ideas?
Regards,
Victor Perez
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Victor Perez
Sent: Friday, April 09, 2004 11:03 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] small linux distro to run * in old boxes
Has anybody tried to install * in any of these minimalist linux distros like tinylinux?
Which linux distro would you use to run * in old P2, P3 boxes?
Regards,
Victor Perez
[EMAIL PROTECTED]
(469) 221-4189
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[Asterisk-Users] Connecting analog trunks to FXS card
We are looking for the cheapest way to show management we can use an * box as a VoIP gateway from our Merlin Legend. There are no T1 avail in the Legend but we have some analog Trunks available as backup for our T1. Since the X100P does not support DID and it would be way too complicated to set a workaround in our Merlin we are thinking on using FXS cards for incoming, so the merlin gets a dialtone from * and dials the VoIP extension. FXO trunks would be used for outgoing. Has anybody tried such a crazy setup? Regards, Victor Perez ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DID with X100P?
Is there a way to use an X100P as a trunk with DID numbers and all? We just bought one of these and want to create some VoIP extensions connected to our PBX as a trial. The PBX does not have capacity for any more T1 cards so it is the only cheap way for this trial. If not, what kind of hardware would you recommend to setup some analog extensions as DID trunks between a PBX and *? Thanks in advance, Victor Perez ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
