Re: [Asterisk-Users] Sipura g729 call quality to PSTN

uggg. Is anyone out there having any luck with the SPA-2000 or SPA-2100 using the g729 codec with decent call quality? On Tue, 15 Feb 2005 10:19:05 -0500, Mark Eissler [EMAIL PROTECTED] wrote: On Feb 14, 2005, at 1:25 PM, Pedro wrote: Is it just a bad implementation of g729

Re: [Asterisk-Users] Sip phones how to dial a # sign?

, Pedro On Tue, 15 Feb 2005 11:35:12 -0600, Eric Wieling [EMAIL PROTECTED] wrote: Pedro wrote: Is there a way to somehow do an escape # so that you can still use the # key to control devices that require a #, but still keep the T in the dial plan? We have clients that need

Re: [Asterisk-Users] Sipura g729 call quality to PSTN

Actually the SPA-2100 supports 2 g729 channels which is why I bought it. Unfortunately, the call quality is just as poor on the 2100 as it is on the 2000. - Pedro On Tue, 15 Feb 2005 23:12:51 +, Jeffrey Chan [EMAIL PROTECTED] wrote: Is it just a bad implementation of g729 compression

Re: [Asterisk-Users] Sip phones how to dial a # sign?

, 15 Feb 2005 11:35:12 -0600, Eric Wieling [EMAIL PROTECTED] wrote: Pedro wrote: Is there a way to somehow do an escape # so that you can still use the # key to control devices that require a #, but still keep the T in the dial plan? We have clients that need to check external

[Asterisk-Users] Sipura g729 call quality to PSTN

call. I even got an SPA-2100 in hopes that the g729 would sound better on that unit, but the same issue is present there as well. Is it just a bad implementation of g729 compression with the Sipura product line? Any thoughts or recommendations are appreciated :) Thanks! - Pedro

Re: [Asterisk-Users] Zombie SIP channels

. But good to know that if it becomes a problem, I can try upgrading to 1.0.3 or later. Thanks! Pedro On Thu, 10 Feb 2005 09:19:45 +0100, Florian Overkamp [EMAIL PROTECTED] wrote: Hi, -Original Message- Does anyone know how to kill a zombie channel? Here is what I see on a show

Re: [Asterisk-Users] Zombie SIP channels

What is odd is no meetme is being used. But may be related - thanks! Pedro On Thu, 10 Feb 2005 14:37:31 +0100, Florian Overkamp [EMAIL PROTECTED] wrote: Hi, -Original Message- This is the first time I have seen this so it does not appear to happen too often. Obviously would

Re: [Asterisk-Users] Zombie SIP channels

Ok this is odd - caught it again twice today. The more I thought about what has changed on the server I realized that I was not using a timing device before, but am now using ztdummy. I if that could be causing the zombies? - Pedro On Thu, 10 Feb 2005 08:50:35 -0500, Pedro [EMAIL PROTECTED

[Asterisk-Users] Zombie SIP channels

Bridged Call SIP/frontdesk-0461ZOMBIE SIP/frontdesk-0461ZOMBIE (customercontext 100 1 ) Ring Dial SIP/frontdesk|20|t 2 active channel(s) -- No one is on a call - how can I get rid of this without restarting asterisk? Thanks! Pedro

[Asterisk-Users] Zombie SIP channels

) -- No one is on a call - how can I get rid of this without restarting asterisk? Thanks! Pedro ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk

Re: [Asterisk-Users] Zombie SIP channels

). Any ideas on why a zombie sip channel would occur? Thanks in advance for any insight on this. - Pedro On Thu, 10 Feb 2005 14:57:17 +1300, Matt Riddell [EMAIL PROTECTED] wrote: Pedro wrote: No one is on a call - how can I get rid of this without restarting asterisk? soft hangup TAB

[Asterisk-Users] Using a Dual WAN Load Balancing Device

! LOL) I am not holding my breath that this is a viable solution, but was just wondering your thoughts. Thanks! Pedro ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE

[Asterisk-Users] Re: Using a Dual WAN Load Balancing Device

Noah, Thanks for your input on this. I am not sure if it handles incomng connections or not - will have to check. I don't think it will work either - worth a shot to ask though. Thanks! - Pedro On Tue, 8 Feb 2005 10:26:48 -0500, Noah Miller [EMAIL PROTECTED] wrote: We have a client

Re: [Asterisk-Users] Call forwarding

Cool idea. One question - let's say someone specifies their home phone number and their cell number. How do you take into the account if the cell VM picks up (ie. if cell is out of coverage and VM greeting is played)? On Fri, 04 Feb 2005 10:41:28 -0700, Kevin P. Fleming [EMAIL PROTECTED]

Re: [Asterisk-Users] Re: outbound 911 calling

different callerid's set. - Pedro On Wed, 2 Feb 2005 22:31:57 -0500, Jason Brown [EMAIL PROTECTED] wrote: Pedro Exactly my point. I have each company in a different context. How do I SetCallerID to a number based on the context

Re: [Asterisk-Users] outbound 911 calling

You need to create different contexts for each company. - Pedro On Wed, 2 Feb 2005 21:49:53 -0500, Jason Brown [EMAIL PROTECTED] wrote: In order to put a shared pbx in an office building for multiple businesses, I will have to make sure that the caller ID information going out

Re: [Asterisk-Users] Sipua SPA-2000 and liong delay after dialling number

You can also adjust the Interdigit Long Timer and Interdigit Short Timer values found in the Regional settings config screen. - Pedro On Fri, 28 Jan 2005 13:36:14 +0100 (CET), Remco Barende [EMAIL PROTECTED] wrote: On Fri, 28 Jan 2005, David John Walsh wrote: The delay is a time out

Re: [Asterisk-Users] Sipua SPA-2000 and liong delay after diallingnumber

understood - I use the # sign as well, but some users are not used to using the # sign so decreasing the timer helps those that may forget to use the # key. -Pedro On Fri, 28 Jan 2005 08:08:28 -0600, Michael B. Murdock [EMAIL PROTECTED] wrote: Pedro, You can also instruct your users

[Asterisk-Users] outgoing call queue.

Hi all, is it possible to make a queue for outgoing calls? That's for preventing Device '/dev/ttyI 0' is busy error when having only one line to dialout and many files in /var/spool/asterisk/outgoing folder. So it would call only one call at the time and when it's done it would move to next.

[Asterisk-Users] Cisco IP Conference 7935

Hi, I have one Cisco IP Conference 7935. Somebudy have any idea how I can config this phone to work woth *. My * server is now working with GrandStream Phone and X-Lite SoftPhone, I need to add this Cisco 7935 but I dont know how I can convert to SIP. Thanks, Pedro Mansilla

[Asterisk-Users] park announcement not working Help!

So I basically have park working but when the call gets parked it doesn't announce the line it parked on. How can I get this to work? Pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] CISCO IP Conference Station

Hi, Somebody have any idea how I can config a CISCO IP CONFERENCE STATION Model 7935 that work with * . Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To

[Asterisk-Users] Pros and cons on SIP vs H.323 vs MGCP

Just trying to get a feel for how these protocols have progressed and what is recommended from experience. Also, what phones work the best. Thanks for the info Pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman

[Asterisk-Users] pickupgroup and callgroup on zapata.conf , how they work ?

mean ? if someone could point me to somewhere I can find more information or help me to understand that I would be really happy =] . thanks, Pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Type of T1 for T100P card

I'm currently setting up a PBX system using the T100P card, and was wondering if it can handle the 2-way trunk type of T1s. Do 2-way trunk T1s use RBS signaling? Please excuse my ignorance, I have mostly dealt with PRI B and D channel type of T1s. Thanks Pedro

Re: [Asterisk-Users] Fax over IP doesn't works

Hi , I tried this a lot, but with no sucess , even in a local network , there is always some loss and you receive only chunks of the original file . Pedro. Miroslav Nachev wrote: Hi, We try to send Fax through IP Network but without success. The other party use NetCentrex SoftSwitch and our

RE: [Asterisk-Users] quadBRI FAX problem

Hi, Thanks Tim, we try this and works fine at first page but when the page graphic dense or more than one page, we have an error. Un saludo, Pedro -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Robinson Tim-W10277 Enviado el: miércoles, 13 de octubre de

RE: [Asterisk-Users] TDM400 synch issue

Hello, I have teh same problem with: QuadBRI - * - TDM400 - Modem Thanks in advance for your help. Regards, Pedro -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Carl Sempla Enviado el: jueves, 23 de septiembre de 2004 3:56 Para: [EMAIL PROTECTED] Asunto

[Asterisk-Users] quadBRI FAX problem

directly to ISDN lines and voice and fax work fine. Now, we have between ISDN lines and Panasonic PBX the Asterisk, and voice is ok but fax doesn´t work fine. What can I do? Thanks, Pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com

[Asterisk-Users] SIP Hard Disconnect Detection

am not receiving any disconnect control frames and don't know how to check if the clients are really connected. Can anyone help? Thank you, Pedro Goncalves

RE: [Asterisk-Users] hide caller id

Yes, my phone company has enabled the Caller ID hiden possibility, thats because with a Panasonic PBX works fine but with Asterisk not. Thanks for your aproach, what can I do now? Regards, Pedro -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Manuel Wenger

[Asterisk-Users] hide caller id

Hi, We try ti hide the caller id at calls trought E1 in EuroISDN (Spain) using restrictcid=yes and doesn´t work. What can I do, thaks Pedro -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de [EMAIL PROTECTED] Enviado el: miércoles, 31 de marzo de 2004 12:00

[Asterisk-Users] Record Application Problem

. Thank you, Pedro Goncalves

[Asterisk-Users] quadBRI and CallerID

Hi, I have a Junghanns.net quadBRI PCI Card with Telefonica ISDN BRI line, and we have in zapata.conf usecallerid=yes and hidecallerid=no, but we have not the caller ID. Can I make some configuration to solve this? Thanks, Pedro ___ Asterisk-Users

RE: [Asterisk-Users] cdr-csv / Zaptel BRI / full number not stored (missing leading zeroes)

Hi Alf, Have you got a Junghanns.net quadBRI PCI Card ? If yes, Have you received CallerID number ? How you have got configured zaptel and zapata ? Im collapssed at this point, thaks in advance, Pedro PD: maybe... around your question, are you using cdr in csv or in mysql?, if you are using

RE: [Asterisk-Users] quadBRI and UK ISDN2e

Hi, Maybe I have similar problem. I have a Junghanns.net quadBRI PCI Card in wiht Telefonica ISDN BRI line, and we have in zapata.conf usecallerid=yes and hidecallerid=no, but we have not the caller ID. Can I make some configuration to solve this? Thanks, Pedro -Mensaje original- De

[Asterisk-Users] quadBRI telco part hungs

information you need to help us?, very thanks. Regards, Pedro ___ 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] quadBRI ISDN telephone

group = 1 signalling = bri_net_ptmp channel = 1-2,4-5,7-8,10-11 Thanks, Pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com

RE: [Asterisk-Users] Fax problem

-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pedro Vela Sent: Friday, April 23, 2004 7:52 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Fax problem Hi, We have a machine with an *'s with Digium TDM400P and connected wit other machine with *'s an TDM400P

[Asterisk-Users] Fax problem

. Which can be the problem ?. What can I do to find the problem ? Thanks, in advance, Pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http

[Asterisk-Users] Busy error

Hi, When have a incoming call from E1 to a extension FXS, and this extension is busy, the incoming call recive ring tone, and it is wrong. What can I do? Thanks in advance Pedro Here is the trace: asterisk-1*CLI Protocol Discriminator: Q.931 (8) len=41 Call Ref: len= 2 (reference 66/0x42

RE: [Asterisk-Users] call initiation

Roger, Maybe you are using extensions like _9. try to put de complete number in your estension.conf ej; exten = _9XXX,1,Dial(. exten = 101,1,Dial(Zap/1) in that case send congestion if the 3 digits extensions are not in extensions.conf. Regards, Pedro J. Vela Ruiz

[Asterisk-Users] Fax can't pass trough alaw

?. What can I do to find the problem ? Thanks. Regards, Pedro ___ 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] H263 SIP Video Playback

Title: H263 SIP Video Playback Hi. Was anyone able to send an H263 to SIP clients through any Asterisk play function? If so, which h263 test files did you use? Thank you, Pedro Goncalves

[Asterisk-Users] indications.conf for Portugal

Title: indications.conf for Portugal Does someone have the settings for 'indications.conf' in Portugal? Thank you, Pedro Goncalves -- Pedro Goncalves PT Inovação SA - Pólo do Porto Largo de Mompilher, 22 - 4º 4050

RE: [Asterisk-Users] Need a list of asterisk built-in variables

Title: RE: [Asterisk-Users] Need a list of asterisk built-in variables The ${CALLERIDNUM} variable has the calling extens number. Regards, Pedro Goncalves -Original Message- From: Justin Carlson [mailto:[EMAIL PROTECTED]] Sent: terça-feira, 6 de Abril de 2004 17:32 To: [EMAIL

RE: [Asterisk-Users] Need a list of asterisk built-in variables

Title: RE: [Asterisk-Users] Need a list of asterisk built-in variables Suppose EXT1 makes call to EXT2. Then the ${CALLERIDNUM} is the number of EXT1 while ${EXT} is the number of EXT2. Any doubts? Regards, Pedro Goncalves From: Justin Carlson [mailto:[EMAIL PROTECTED

RE: [Asterisk-Users] Play file from an offset

be interesting and profitable if we could collaborate on this. How far have you done in your project? Did you test h263 video over SIP in Asterisk? I couldnt get it to work with various SIP video clients. Looking forward to hear from you. Best regards, Pedro Goncalves From: Yves

[Asterisk-Users] Outbound calling number problem

is wrong ? Regards, Pedro ___ 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] loading dialogic drivers

, gcdb_InsertLinedev is not defined in libicapi. It is defined in libgc, which is linked to chan_dialogic. Anyone has seen this before? [], pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Status of ISDN DTMF (AFAIK): Please add corrections and comments

with a better mic, but does not disappear. It seems that no solution to this problem is available. -my callees complain of the silence supression. I have not looked at it yet. Please correct and comment. pedro ___ Asterisk-Users mailing list [EMAIL

[Asterisk-Users] current status of i4l and dtmf stuff

. thanks, pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

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