[Asterisk-Users] Re: quadBRI in bri_net mode - t3 timer expired

* Paul Hewlett <[EMAIL PROTECTED]> wrote: > On Thursday 29 June 2006 20:08, Sebastian Kayser wrote: > > i successfully connected our old PBX to an asterisk server with a > > junghanns quadBRI, the quadBRI ports running in bri_cpe_ptmp mode > > connected to the interal PB

[Asterisk-Users] quadBRI in bri_net mode - t3 timer expired

span 3 What is it trying to tell me? My quadBRI doesn't do any powerfeeding, might that be a problem? - Sebastian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

[Asterisk-Users] RxFax & Asterisk possible bug?

Hat Enterprise servers without this problem (with Asterisk 1.0.9). So I don't believe this is an error caused by myself. -- With kind regards, Sebastian Berm ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mail

[Asterisk-Users] Re: Re: E1 hardware for asterisk

not affect CPU performance while on > the Junghanns it does. And in terms of quality? Does one of them perform noticable better than the other. - Sebastian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUB

Re: [Asterisk-Users] I can't call PSTN numbers

a Sipura directly into SER, I can get access to all PSTN numbers. So why not with Asterisk?. I can't find anything different. Thanks again for your help Sebastian On 5/30/06, Woodoo People .pGa! <[EMAIL PROTECTED]> wrote: autocreatepeer=yes [ser_box1] type=peer username=

[Asterisk-Users] Re: E1 hardware for asterisk

echo canceller performance of the Junghanns E1 cards compared to for example the Sangoma ones? http://www.junghanns.net/en/singleE1_produkt.html http://www.junghanns.net/en/doubleE1_produkt.html - Sebastian ___ --Bandwidth and Colocation provided by Easyne

Re: [Asterisk-Users] I can't call PSTN numbers

It doesn't work for me :-( How do you have the peer configuration in asterisk, to connect ot SER? Sebastian On 5/29/06, Woodoo People .pGa! <[EMAIL PROTECTED]> wrote: exten => _4XX,1,Dial(SIP/[EMAIL PROTECTED]) it works to me (my provider sends me the last 3 digits) > I h

[Asterisk-Users] I can't call PSTN numbers

ch in advance Sebastian ___ --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] Got reject for frame 0, but we only have others!

is logged. The BRI debug can be found at http://skayser.de/mls/au/reject-bri-intense-debug.txt Maybe some of you are more capable to interpret those cryptic BRI messages. asterisk*CLI> show version Asterisk 1.2.7.1-BRIstuffed-0.3.0

[Asterisk-Users] Re: Ringing indication not working as expected

However, even restart doesn't change anything about the ringing indication problem. See my other reply for further debug information i have gathered. - Sebastian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing lis

[Asterisk-Users] Re: Ringing indication not working as expected

* Sebastian Kayser <[EMAIL PROTECTED]> wrote: > * Sebastian Kayser <[EMAIL PROTECTED]> wrote: > > are there any caveats regarding ringing indication with Asterisk? > PSTN <-- 3 x BRI --> POTS

[Asterisk-Users] Re: Ringing indication not working as expected

* Sebastian Kayser <[EMAIL PROTECTED]> wrote: > are there any caveats regarding ringing indication with Asterisk? > > I have got an asterisk installation with a quadBRI driven by BRIstuff. > Internal phones are various snoms (320 / 360) connected via SIP and > Idefisk sof

[Asterisk-Users] Re: Ringing indication not working as expected

425/500,0/500,425/500,0/500,425/500,0/500,1600/100,0/900 record = 1400/500,0/15000 info = 950/330,0/200,1400/330,0/200,1800/330,0/1000 - Sebastian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or u

[Asterisk-Users] Ringing indication not working as expected

he calling party is a snom hardphone or an idefisk softphone. Am i missing something? asterisk*CLI> show version Asterisk 1.2.7.1-BRIstuffed-0.3.0-PRE-1p - Sebastian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users ma

Re: [Asterisk-Users] unable to set outgoing callerid

Hi, just answering myself: I am not allowed to send the leading 0 for my prefix with the callid, then it works well. Sebastian Sebastian Reitenbach <[EMAIL PROTECTED]>,Asterisk Users Mailing List - Non-Commercial Discussion wrote: > Hi *, > > now for a long time i am tr

[Asterisk-Users] unable to set outgoing callerid

the same result. anybody might have a clue what my problem might be? any small hint is appreciated as this is going to drive me crazy. On another machine at home I have no problem setting the callerid, but there I only have a SIP trunk. kind regards Sebastian

Re: [Asterisk-Users] app_meetme.so

//www.asteriskhelpdesk.com > > > > > -Original Message- > > From: Sebastian Milioto [mailto:[EMAIL PROTECTED] > > Sent: Thursday, April 13, 2006 10:31 AM > > To: asterisk-users@lists.digium.com > > Subject: [Asterisk-Users] app_meetme.so > > > > H

[Asterisk-Users] app_meetme.so

Hi all, I'm using Asterisk 1.2.5 and , for some reason, when I install it, the module app_meetme.so didn't install. Is there some way to download that module, and add it to asterisk without re-install it? Thanks in advance Sebastian ___ -

[Asterisk-Users] some problems with asterisk and E1

nfigs #include zapata-auto.conf ;Include AMP configs #include zapata_additional.conf channel => 1-15,17-31 here is my zapata-auto.conf: callerid=asreceived the zapata_additional.conf is empty. any help appreciated. kind regards Sebastian ___ --Bandw

Re: [Asterisk-Users] Asterisk on BSD?

://www.voip-info.org/tiki-index.php?page=Asterisk+FreeBSD > yeah, have it also Asterisk 1.0.9 from ports running on openbsd 3.8. With the upcoming openbsd 3.9 there will be an Asterisk port 1.2.X and app_conference (a MeetMe replacement) but still no zaptel. Sebastian ___

Re: [Asterisk-Users] cannot set outgoing cid

Hi, it is a PRI. With the old telephone system the extensions were transmitted. Only replaced the telephone systems and whatever I do, only the central dial-in number is transmitted. kind regards Sebastian "Tom Vile" <[EMAIL PROTECTED]> wrote: > Are you allowed to set

Re: [Asterisk-Users] cannot set outgoing cid

Hi, it is a PRI. With the old telephone system the extensions were transmitted. Only replaced the telephone systems and whatever I do, only the central dial-in number is transmitted. kind regards Sebastian "Tom Vile" <[EMAIL PROTECTED]> wrote: > Are you allowed to set your

[Asterisk-Users] cannot set outgoing cid

id i configured, and it seems to use them, but the caller will only see a 0338189040 instead of my extension. any hint to what could be wrong is greatly appreciated. kind regards Sebastian Mar 31 16:53:56 DEBUG[24358] chan_sip.c: = No match Their Call ID: [EMAIL PROTECTED] Their Tag 5jd9htv3r

Re: [Asterisk-Users] asterisk doesn't wait for whole extension

Hi, thanks for answering my question. I used AMP to setup the dial plan. I have attached the extensions*.conf files created by it. My message was too large for the list, therefore i omitted the extensions_custom.conf. let me know if you need it. thanks for looking. kind regards Sebastian

Re: [Asterisk-Users] asterisk doesn't wait for whole extension

only 12343 arrives at the asterisk. kind regards Sebastian Aaron Daniel <[EMAIL PROTECTED]> wrote: > Can you post your dialplan? We'd be much better at troubleshooting the > problem if we could follow the path that calls take. > > Aaron > > On Thu, 30 Mar 200

[Asterisk-Users] asterisk doesn't wait for whole extension

asterisk to wait until it recognizes the number? Or is there a way to tell asterisk that the extensions are all three digits long, so that it will wait the time until the whole extension was dialled? kind regards Sebastian ___ --Bandwidth and

Re: [Asterisk-Users] Little OT.. SER Question

but its still showing up with the > prefix. I need it to look like this: > >[EMAIL PROTECTED] > > > I got xxx.xxx.xxx.xxx to change to yyy.yyy.yyy.yyy I just need the prefix to > go away now > You just have to insert a strip(5) statement before the uri rewri

AW: [Asterisk-Users] Some problem with CAPI support

394,1,Dial(Local/6275/n) ; this will dial ${MARK} ;exten => 6275,1,Macro(stdexten,6275,${MARK}) ; assuming ${MARK} is something like Zap/2 ;exten => mark,1,Goto(6275|1) ; alias mark to 6275 ;exten => 6536,1,Macro(stdexten,6236,${WIL}); Ditto for wil ;exte

[Asterisk-Users] Some problem with CAPI support

for ol der eicon drivers) ;echotail=64 ;echo cancel tail setting ;bridge=yes ;native bridging (CAPI line interconnect) if available ;callgroup=1 ;Asterisk call group ;deflect=1234567 ;deflect incoming calls to 1234567 if all B channels are busy devices=2;number of concurrent calls

[Asterisk-Users] Linksys pap2 behind Linksys RT31

ready configured this toplogy? Could you help me with that, please? Thanks very much in advance, Sebastian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/ma

Re: [Asterisk-Users] AsteriskJava - Queue

answer or antything else. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian Silva Sent: Monday, September 26, 2005 6:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] AsteriskJava - Queue Hi, I am using Ast

[Asterisk-Users] AsteriskJava - Queue

the text that the agent sent. Thanks in advance, Sebas -- Sebastian Silva G R U P O G A U S S Depto. Sistemas Av. Libertador 6250 4 piso Tl.: 4 706- (int. 121) [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com

Re: [Asterisk-Users] externpass

erisk/additional_ser_voicemail.conf 4.  Restart asterisk with asterisk manager (php /usr/local/etc/ser/scripts/ast-reload/reload_asterisk.php)   It's working very good...   Thanks for your help!   Sebastian         - Original Message - From: "Tzafrir Cohen" <[EMAIL PR

[Asterisk-Users] Re: Asterisk with iptel.org

ost=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid="Sebastian" <200> [iptel] type=friend username=84565616 secret=password_iptel fromdomain=iptel.org host=iptel.org And the following is part of extensions.conf [outbound-allroutes] include => outbound-al

[Asterisk-Users] externpass

your help!! Sebastian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

[Asterisk-Users] Asterisk with iptel.org

ip-info.org. I will appreciate your help, Thanks in advance, Sebastian e-mail:[EMAIL PROTECTED] IM: [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.

Re: [Asterisk-Users] Get SIP to work over very limited network access

ver listening on port 443, he shouldn't have any problems connecting. Regards Sebastian A. Espindola. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.c

Re: [Asterisk-Users] SIP audio port usage

But, if I have Xlite running on client PC and at the same time the user is doing FTP, both service has the same QoS treatment? Is there a way to differentiate these services besides the port? Sebastian On 9/20/05, Sherwood McGowan <[EMAIL PROTECTED]> wrote: > Yes, because then

[Asterisk-Users] Fwd: Asterisk in Spanish

Great info!!!. Thank you all guys. Regards, Sebastian -- Forwarded message -- From: Sergio Serrano <[EMAIL PROTECTED]> Date: Mon, 19 Sep 2005 18:38:55 +0200 Subject: RE: [Asterisk-Users] Asterisk in Spanish To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Comm

[Asterisk-Users] Asterisk in Spanish

anish? Or may be another site which contain this kind of stuff (.wav, .gsm files for answering machines in spanish)? Thank you very much, Regards, Sebastian Milioto Telecommunications Engineer IM: [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] Mobile: 549 3571 543658

[Asterisk-Users] compile error with postgres and voicemail

make[1]: *** [app_voicemail.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-1.0.9/apps' make: *** [subdirs] Error 1 Can anyone give me a hint? Thanks! Sebastian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Aste

[Asterisk-Users] compile error with postgres and voicemail

make[1]: *** [app_voicemail.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-1.0.9/apps' make: *** [subdirs] Error 1 Can anyone give me a hint? Thanks! Sebastian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Aste

[Asterisk-Users] OpenH323-Channel Q.931-Problems with Gatekeeper

u know some problems with Asterisk and the H.323-Channel. We tried to compile and test nearly every version of openh323 and chan_oh323, but it wasn`t successful. Best regards from Germany, Sebastian. Nearby we will post our configs and logs: 1.)  chan_oh323.conf ---

[Asterisk-Users] Restricting outgoing calls by extension / Multiple providers

oceeds using the 'catchallvoip' trunk. Does this sound reasonable or can the Goto cmd not be used to switch contexts for outbound calling rules by extensions? Thanks in advance for any insights!-Sebastian ___ Asterisk-Users mailing list Asterisk-U

[Asterisk-Users] Transfer and CDR's

t;,"SIP/test20-2bdb","Zap/1-1","Hangup","","2005-07-04 15:09:09","2005-07-04 15:09:17","2005-07-04 15:09:52",43,35,"ANSWERED","DOCUMENTATION" Can somebody help me, and solve this problem? I can't bill this connections properly.   Regards   Sebastian Zaprzalski ___ 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

Re: [Asterisk-Users] How do you handle NAT?

___ 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 -- Sebastian Silva G R U P O G A U S S Depto. Si

Re: [Asterisk-Users] problem compile

at_tc/ ___ 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 -- Sebastian Silva G R U P O G

Re: [Asterisk-Users] Asterisk died - exactly every 60 minutes

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 -- Sebastian Silva G R U P O G A U S S Depto. Sistemas Av. Libertador 6250

Re: [Asterisk-Users] Console ALSA Sound

an/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sebastian Silva G R U P O G A U S S Depto. Sistemas Av. Libertador 6250 4 piso Tl.: 4 706- (int. 121) [EMAIL PROTECTED] ___

Re: [Asterisk-Users] Unable to open master device '/dev/zap/ctl'

___ 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 -- Sebastian Silva G R U P O G A U S S Depto. Sistemas

Re: [Asterisk-Users] app_dial.c:977 dial_exec_full: Unable to create channel of type 'Zap' (cause 0)

63, extension(s) h263 [format_g726.so] => (Raw G.726 (16/24/32/40kbps) data) == Registered file format g726-40, extension(s) g726-40 == Registered file format g726-32, extension(s) g726-32 == Registered file format g726-24, extension(s) g726-24 == Registered file format g726-16, extension(s) g72

[Asterisk-Users] SoftPhone - Solaris

x27;t care if it is a commercial product, I can buy it if works fine. thanks in advance. Sebas -- Sebastian Silva G R U P O G A U S S Depto. Sistemas Av. Libertador 6250 4 piso Tl.: 4 706- (int. 121) [EMAIL PROTECTED] ___ Asterisk-Users mailing lis

Re: [Asterisk-Users] Urgent Help neededt!! Asterisk 1.0.7 CPU at 99%

Hi, Are you sure the process consuming your CPU is Asterisk? Did you tried with different codecs? Andres Maduro wrote: Hi, I am using Asterisk 1.0.7 that comes with [EMAIL PROTECTED] 1.0 ISO. I have installed chan_unicall.c and MFCR2 support with latest Steve Underwood code unicall-0.0.2pr

RE: [Asterisk-Users] Problems with VIA Chipset

Try http://www.voip-info.org/wiki-Asterisk+Compile here said that do you need to change in the Makefile for a VIA. SA -Mensaje original- De: Armin Lediger [mailto:[EMAIL PROTECTED] Enviado el: Miércoles, 11 de Mayo de 2005 17:15 Para: asterisk-users@lists.digium.com Asunto: [Asterisk-U

AW: [Asterisk-Users] CAPI on ptp with variable length digits inphonenumber: SOLUTION for EICON

version of the firmware and chan_capi i tried). regards, Am Freitag, 6. Mai 2005 15:17 schrieb Sebastian Buntin: > Hello! > > I finally found a working solution. > calling > divactrl with the parameter -n [0..20] gives the DID-length > means, if you wanna have 123-XXX in digit-wi

[Asterisk-Users] CAPI on ptp with variable length digits in phonenumber: SOLUTION for EICON

Hello! I finally found a working solution. calling divactrl with the parameter -n [0..20] gives the DID-length means, if you wanna have 123-XXX in digit-wise mode, then call divactrl load -c 1 -n 3 -f ETSI and the card will wait for n digits. regards, Sebastian -Ursprüngliche

[Asterisk-Users] CAPI on ptp with variable length digits in phone number

ensions till the caller typed 123114. I can live with fixed length extensions. means, always wait for 3 digits. thanks for help.. Sebastian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/a

[Asterisk-Users] IAX2 Carriers

about. thanks a lot. Sebas -- Sebastian Silva G R U P O G A U S S Depto. Sistemas Av. Libertador 6250 4 piso Tl.: 4 706- (int. 121) [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman

[Asterisk-Users] chan_capi crashes asterisk

backtrace. The machine I´m using for asterisk is running fli4l which has no gdb available. Maybe somebody could give me a hint looking at the attatched logs? Cards are Fritz!DSL (with integrated isdn port) and Acer isdn surf 128. Thank you, Sebastian 8

[Asterisk-Users] oh323 Zone

Hi, Someone knows how can I register my Asterisk to a gatekeeper using zone parameters? I'm using asterisk 1.0.7 and oh323 0.6.5. I'm trying to register to a gatekeeper in another network and I can't reach this with a broadcast. Zone is the name who Cisco call the GK identification. Thank

Re: [Asterisk-Users] Extensions / Contexts

Perfect, that's exactly what I need. I will try that, thanks a lot. Sebas Matt Riddell wrote: Sebastian Silva wrote: Hi everybody, I am writing here because I can't find the solution to my problem (my asterisk configuration). I hope somebody can give me a hand with it: I need to pro

Re: [Asterisk-Users] Extensions / Contexts

ample: Comp-A user 2000 calls comp-B user 2000 by dialing 72000. -- Sebastian Silva G R U P O G A U S S Depto. Sistemas Av. Libertador 6250 4 piso Tl.: 4 706- (int. 121) [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] Extensions / Contexts

epending on the username? Does asterisk allows two extension sections with the same number?: [2000] username=companyA_2000 context=contextCompanyA [2000] username=companyB_2000 context=contextCompanyB Any help will be appreciated. Sebas -- Sebastian Silva G R U P O G A U S S Depto. Sistemas Av. Liberta

RE: [Asterisk-Users] Sipura SPA-841 auto-answer support [patch]

Can you send me the patch? SA -Mensaje original- De: Geoff Speicher [mailto:[EMAIL PROTECTED] Enviado el: Sábado, 29 de Enero de 2005 23:11 Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] Sipura SPA-841 auto-answer support [patch] Sipura has implemented auto-answer in v

[Asterisk-Users] "restart gracefully" fails

Dear Asterisk Users, if I do a : /usr/sbin/asterisk -r -x "restart gracefully" , asterisk just quits without any message. Any idea ? (debian 3.1 with asterisk packages from unstable : 1.0.7-BRIstuffed-0.2.0-RC7k) /sebastian ___ Asterisk-Use

[Asterisk-Users] agi script for german date / time

"the" ? /sebastian -SNIP #!/usr/bin/perl use strict; use Time::HiRes qw/sleep/; my %params; $|=1; while(1) { my $line = ; chomp($line); last if $line eq ''; (my $key, my $value) = split(/\: /,$line); $params{$key}=$value; } (undef,

[Asterisk-Users] prevent non-free calls

=_01186.,1,dial(SIP/[EMAIL PROTECTED],30) exten=_011886.,1,dial(SIP/[EMAIL PROTECTED],30) exten=_011972.,1,dial(SIP/[EMAIL PROTECTED],30) exten=_011.,2,congestion() ; No answer, nothing exten=_011.,102,busy() ; Busy Thank you very much Sebastian

[Asterisk-Users] silence suppression

Hi, how can I completely disable silence suppresion and echo cancelling in asterisk (and zaphfc) Thank you very much. Sebastian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To

[Asterisk-Users] rejected calls

] exten => 2122020683,1,Dial(SIP/[EMAIL PROTECTED]&SIP/[EMAIL PROTECTED],30) exten => 2122020683,2,Hangup ------ thank you very much sebastian ___ Asterisk-Users mailing list Aster

RE: [Asterisk-Users] Sipura SPA-841 auto-answer support [patch]

Which version of Asterisk this did work? Sebastián Atala -Mensaje original- De: Geoff Speicher [mailto:[EMAIL PROTECTED] Enviado el: Sábado, 29 de Enero de 2005 23:11 Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] Sipura SPA-841 auto-answer support [patch] Sipu

AW: [Asterisk-Users] HDLC for Dummies?

http://en.wikipedia.org/wiki/HDLC -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Eric Bishop Gesendet: Montag, 31. Januar 2005 11:40 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: [Asterisk-Users] HDLC for Dummies? Can any

[Asterisk-Users] Asterisk B2BUA

Can Asterisk only send and receive SIP packet without media proxy in any time? I am using re-invite but I don't want that the ring back is proxy by asterisk. Someone knows a way to do that? Sebastian ___ Asterisk-Users mailing list Asterisk-

RE: [Asterisk-Users] ASTCC

Here is the link http://www.voip-info.org/wiki-ASTCC SA -Mensaje original- De: Bilal Ghayad [mailto:[EMAIL PROTECTED] Enviado el: Martes, 14 de Enero de 2003 18:21 Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] ASTCC Dear Sebastian; Thanks a

RE: [Asterisk-Users] PrePaid Applications

Try with ASTCC is free. Sebastian -Mensaje original- De: Bilal Ghayad [mailto:[EMAIL PROTECTED] Enviado el: Martes, 14 de Enero de 2003 14:56 Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] PrePaid Applications Hi; Is the Prepaid Applications that we can use it with

RE: [Asterisk-Users] International area codes (incl. mobile)

I can send a list, mobile is not complete but it has a lot of numbers... -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de PHP Mechanic Enviado el: Viernes, 07 de Enero de 2005 11:57 a.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: R

[Asterisk-Users] Problems with msn's, did not find device for msn

=0.8 language=de [interfaces] msn=4132 incomingmsn=* controller=1 context=demo mode=immediate isdnmode=ptp devices=30 extensions.conf: [demo] exten => 4132,6,Dial(SIP/test) so, I'm a bit confused now. what can I do??? thanks for helping me out! greetings, Sebastian Capi Debug outpu

RE: [Asterisk-Users] OT: List of VoIP providers?

Voipproviderlist.com -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Jeromie Reeves Enviado el: Martes, 04 de Enero de 2005 03:30 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [Asterisk-Users] OT: List of VoIP providers? I have

[Asterisk-Users] sendURL

Someone know what kind of terminal I need to use for this feature? What exactly do this and what is way to use that? Sebastián Atala ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asteri

RE: [Asterisk-Users] h.323 Type=User

Sebastian Nocetti wrote: >> is h323 per user based working??? I have setup this: >> >> [User1] >> type=user >> host=xx.xx.xx.xx >> context=international >> incominglimit=30 >> >> But all calls from xx.xx.xx.xx are not routed to context >>

RE: [Asterisk-Users] h.323 Type=User

Thanks !! I will try!! -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Soren Rathje Enviado el: Martes, 21 de Diciembre de 2004 02:30 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] h.323 Type=User Sebastian

[Asterisk-Users] h.323 Type=User

is h323 per user based working??? I have setup this:   [User1]type=userhost=xx.xx.xx.xx context=international incominglimit=30   But all calls from xx.xx.xx.xx are not routed to context international, it is working?   I am using chan_h323   Thanks!! Sebastian Nocetti. --- Checked

[SPAM] Re: [Asterisk-Users] working with big blocks of msn's

On 16.12.2004 "Martin List-Petersen" Wrote <[EMAIL PROTECTED]>: > >> then the routing to SIP-Phones shall be based on the MSN-Configuration. >> >> means, if someone dials 4321-1000 the call shall go to SIP/boss >> and 4321-1001 to SIP/secretary >> and so on. >> >> is this "just" by adding an >>

[Asterisk-Users] working with big blocks of msn's

1,Dial(SIP/boss) to the context set in the /etc/asterisk/capi.conf? and what to do, so that, if the boss calls out the MSN of the secretary is shown? and if the secretary calls out also their MSN is shown? thank you for helping! Sebastian ___ Asterisk-Us

Re: [Asterisk-Users] asterisk + chan_sip2 + sipproxd + sipgate

exactly you're trying to to, maybe I can help you more if you do. /sebastian ___ 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

RE: [Asterisk-Users] Asterisk + AS5300

From: Sebastian Nocetti To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Wednesday, December 01, 2004 10:54 AM Subject: RE: [Asterisk-Users] Asterisk + AS5300 I am doing that actually, terminating calls via SIP on a Cisco AS5300, and it is working

RE: [Asterisk-Users] Asterisk + AS5300

I am doing that actually, terminating calls via SIP on a Cisco AS5300, and it is working good. De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de FranciscoEnviado el: Miércoles, 01 de Diciembre de 2004 10:43 a.m.Para: [EMAIL PROTECTED]Asunto: [Asterisk-Users] Asterisk + AS5300

Re: AW: [Asterisk-Users] zaphfc problem

d not initialize ..., due to ", this had helped alot and had saved alot of hours (-; /sebastian Pascal C. Kocher wrote: Hello Make sure you run ztcfg only once(!) per reboot. A second time seems to kill the zaphfc module (even if it doesn't state an error) Do you have any wcfx* card

[Asterisk-Users] zaphfc problem

card(s) in this box. Registered tone zone 3 (Netherlands) zaphfc: card 0 layer 1 state = G2 zaphfc: card 0 layer 1 state = G3 --- Thank you in advance ! Sebastian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listi

RE: [Asterisk-Users] cisco dial-peer voip

I think you CAN'T DO VOIP-VOIP into CISCO Equipment, it have to be POTS-VOIP or viceversa. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Brian Wilkins Enviado el: Martes, 30 de Noviembre de 2004 05:57 a.m. Para: Asterisk Users Mailing List - Non-Commercial

RE: [Asterisk-Users] Prepaid

Para: Asterisk Users Mailing List - Non-Commercial Discussion CC: Sebastian Atala Asunto: Re: [Asterisk-Users] Prepaid I use ASTCC and works perfect for Prepaid situations. Nhauel Ramos. On Mon, 29 Nov 2004 16:48:46 -0400, Sebastian Atala <[EMAIL PROTECTED]> wrote: > Is anyone succ

[Asterisk-Users] Prepaid

Is anyone successfully using asterisk-prepaid-0.3.1? I try to configure but doesn't work. It said that you need to do a few step, copy a few files and that is. Please, if someone has any tips about the configuration, answer me. Sebastian ___ Ast

RE: [Asterisk-Users] ATA186 V2.15.ms

Check what IOS ata have installed... Because by default it does not comes with H.323 - SIP IOS... If you want I can send you both ios... Contact me at: [EMAIL PROTECTED] -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Rodney Acosta Coya Enviado el: Mart

[Asterisk-Users] astcc db creation

and install it without error. I was looking for solution in list and internet but i dont find anything. What can be a problem ?   Sebastian Bojczuk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Configuring Asterisk From Postgres

I want to configure the voicemail, extension, agent, queue and sip from postgres. Someone have experience in that? Someone know how can I configure meetme without a Zaptel card? Sebastián ___ Asterisk-Users mailing list [EMAIL PROTECTED] http:/

[Asterisk-Users] H.323 Status

Hello all, somebody can tell me how h.323 status is? it is working OK?... it has implemented faststart and tunneling per peer based?...   thanks a lot!!   Sebastian from Argentina. --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.791 / Virus Database: 535

[Asterisk-Users] Problem adding zaprtc to Asterisk CVS on debian sarge

n the zaptrtc README, it's a 2.4.27 baked by myself Thanks in Advance, and excuse my bad English ;) Sebastian Mauer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

RE: [Asterisk-Users] Asterisk as PSTN gateway

Asterisk works ok, but it have a lot of errors... 1st: It ever handle audio packet, and you cant do for exacmple only SIGNALLING 2st: It cant handle more than 20 channels simultaneous ... I tested it. 3st: It does not have fully Radius support.- -Mensaje original- De: [EMAIL PROTECTED] [

RE: [Asterisk-Users] SIP termination in Brazil

I am interested too in termination using SIP to brazil, we need h.323 too... Can you contact me? Thanks Sebastian. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Daniel Bichara Enviado el: Martes, 21 de Septiembre de 2004 11:06 a.m. Para: [EMAIL

Re: [Asterisk-Users] Festival Installation - Asterisk 1.0-RC2 && Debian Woody

no longer available from the website. Only 1.95 is available. Will that work? Does it need the patch mentioned on the Wiki page? Of course it is available on theri website! Have a look here: http://festvox.org/packed/festival/ I have installed it from souce yesterday and there are also patche

RE: [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS

MATT--- -----Original Message- From: Sebastian Nocetti [mailto:[EMAIL PROTECTED] Sent: Friday, August 06, 2004 2:51 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS E1's, only G729 and from SIP to E1 or from E1 to SIP De: [EMAIL PROTECTED] [mailto:[

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