search for: ggonzalez

If Someone did that, How I connect extensions.conf with this type of Hybrid system to work with asterisk inside this schema: PSTN--->PANASONIC KX <------> Asterisk | |--------->send internal call Thanks.

Hello all! Does anyone have a good howto to setup Asterisk and Skype. Thanks Gustavo A. Gonz?lez Dto. de Infraestructura Despegar.com, Inc. ggonzalez at despegar.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080526/831b3824/attachment.htm

...stored whitout problems. Parameters that I send are: Channe:SIP/$CHANNEL CallerID:listener Application:ChanSpy' Data:SIP/AgentPhone. Variable:CDR(userfield)=listened Async=yes Thanks for any idea about that Gustavo A. Gonz?lez Dto. de Infraestructura Despegar.com, Inc. <mailto:ggonzalez at despegar.com> ggonzalez at despegar.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090423/830edd73/attachment.htm

...to Asterisk with Iaxmodem and Hylafax, I have no problem. But between Iax Trunks nothing happened and the fax machine registered on the first PBX give me a communication error. Thanks for any help or idea to setup and get it working. Gustavo A. Gonz?lez Dto. de Infraestructura Despegar.com, Inc. ggonzalez at despegar.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080701/d870f6ce/attachment.htm

...basic form I do: exten => _44XX,1,Answer exten => _44XX,n,Dial(Zap/g1/${EXTEN:2}|20|tTr) exten => _44XX,n,Hangup and the console logs for this are: Executing Dial("SIP/sipuser-081d13f0", "Zap/g1/38|20|tTr") in new stack -- Called g1/38 -- Zap/1-1 answered SIP/ggonzalez-081d13f0 Here Dial cmd do one ring and nothing more, Zap channel has answered but the number dialed never RING, what is wrong? what i have to do get this working fine?. Thanks for any help

...; charset=ISO-8859-1 > "Glare" that's what it's called, if the number you advertise as your > business number is zap/1 then use zap/G1 to dial out, otherwise use > zap/g1 to dial out. This will reduce but not eliminate the problem. > On 10/18/07, Gustavo Gonzalez <GGonzalez at grupo-gestion.com.ar> wrote: > Hello I have a question about incoming calls on TDM400P cards. I want to > know why an incoming call appear in a sorpresive way on a phone that I > pickup to call out. I am using ChanIsAvailable to check those lines ( Zap > channels )that are free. I...

...M400P card and when I get a free Zap channel to call I hear the voice of a people on the other side from an incomming call, I think that asterisk bridge my free channel with incomming calls but how do this?Thanks for any idea. Alejandro Gonz?lez Grupo Gesti?n 4384-0660 www.grupo-gestion.com.ar GGonzalez at grupo-gestion.com.ar ----------------------------------- ----------------------------------- RI 9000-1069 Sistema de Gesti?n de Calidad Certificado por IRAM Norma ISO: 9001-2000 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268...

Hi all! Im looking for 1ezphone to use as a web softphone but I?cant access to 1ezphone.com. Anyone knows what happened with this site?. Thanks! Gustavo A. Gonz?lez Dto. de Infraestructura Despegar.com, Inc. ggonzalez at despegar.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090109/7e46586f/attachment.htm

Hello all, iam setting up an asterisk box behind NAT to get SIP calls from outside or internet. In that eschema i can setup SIP calls but, while from the outside nat people can hear me, Im unable to listen anything behind NAT. Out of firewalls settings( I checked this to port fowarding) what can i do to get this working fine?. Thanks G.

Hello, I have a TDM400P with 4 FXO ports, currently using three. When sending or receiving calls on this card, there is a nearly constant popping/clicking sound, it is related to the echo cancellation?. I adjusted my gains properly, but to no avail. I even found that setting echotraining=no in zapata.conf didn't change the scenario at all. I've plugged analog handsets into the

Hello all, I post this issue thinking too that could help other people on an asterisk deployment over distributed offices considering both quality, prices, devices and so. Well, i am working on a deployment of a telephony system based in asterisk. My company have a central office with seven remote offices connected all through a VPN. To reduce and evaluate costs i consider solutions like:

Gustavo A. Gonz=E1lez Dto. de Infraestructura Despegar.com, Inc. ggonzalez at despegar.com=20 =20 ------=_NextPart_000_0452_01C8BF32.9F7C4290 Content-Type: text/html; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable <html xmlns:v=3D"urn:schemas-microsoft-com:vml" = xmlns:o=3D"urn:schemas-microsoft-com:office:office" =...

=20 Thanks!=20 =20 =20 Gustavo A. Gonz=E1lez Dto. de Infraestructura Despegar.com, Inc. ggonzalez at despegar.com=20 =20 ------=_NextPart_000_003E_01C8C00B.B3A8DA60 Content-Type: text/html; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable <html xmlns:v=3D"urn:schemas-microsoft-com:vml" = xmlns:o=3D"urn:schemas-microsoft-com:office:office" =...

...erisk box, when I call them for support or authentication data to load on my sip.conf, they say me that I don?t need such data, so, anyone knows how I would configure my Asterisk box against a Broadsoft peer? Thanks for any help. Gustavo A. Gonz?lez Dto. de Infraestructura Despegar.com, Inc. ggonzalez at despegar.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080723/5e7e3bd3/attachment.htm

...408 messages from asterisk and some time after this they start to complete calls normally, I don?t know what can be wrong. Someone has configured asterisk to wok with this Softswitch? Thanks for any help! Cheers! Gustavo A. Gonz?lez Dto. de Infraestructura Despegar.com, Inc. <mailto:ggonzalez at despegar.com> ggonzalez at despegar.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080821/2efa7d7f/attachment.htm

Hi Everyone... I am running Asterisk 1.2.22 on Debian "Etch". I installed it from sources. I have also installed tiff-v3.6.0, spandsp-0.0.3. and downloaded http://soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/app_rxfax.c http://soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/app_txfax.c and a Digium TDM card with 4 FXO ports When my dialplan

Hello! Iam configuring chan Skype on my asterisk box, doing some test calls I saw that asterisk answer the calls but hungs up before the call are stablished. Is this a license problem? Gustavo A. Gonz?lez Dto. de Infraestructura Despegar.com, Inc. ggonzalez at despegar.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080523/68ceea02/attachment.htm

...ng asterisk 1.2 that do it with the same configuration. In cdr_mysql.conf I have: userfield=1 accountcode=1 Is there a way to solve it or a way to match all my recorded files with the calls that appear in the CDR? Thanks!! Gustavo A. Gonz?lez Dto. de Infraestructura Despegar.com, Inc. ggonzalez at despegar.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081216/126d6f39/attachment.htm

...ll!, I?ve running asterisk 1.4.23.1 and I need to get working pick up from feature.conf. It does no work, only I can connect but cant send audio over the phone. Is there a bug with this feature?. Thanks for any response! Cheers! Gustavo A. Gonz?lez Dto. de Infraestructura Despegar.com, Inc. ggonzalez at despegar.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090415/48f0dcc6/attachment.htm

Hi all! I'm configuring a digium tdm card in Costa Rica, every things works well, but calls don't hangup. I've tested setting up progzone=br but dont work. Thanks for any help. Cheers! -- Gustavo A. Gonz?lez Dto. Telefon?a VoIP Despegar.com 54 (11) 5032-3500 ext. 3512