similar to: Glare on Incoming Calls

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 have four lines connected to my TDM400P 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

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

Ok.. will be there... -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of asterisk-users-request at lists.digium.com Sent: Thursday, October 04, 2007 12:50 PM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 39, Issue 12 Send asterisk-users mailing list submissions to

Hello! I?ve an issue whit CDR using asterisk 1.4.23.1. I?ve configured mysql to store cdr information, but, while I put into cdr_mysql.conf the field ?userfield=1? and doing a query I found that this field is empty in the cdr table. On the other hand I can?t find records in the cdr table that show me calls generated through AMI using Originate Action, that?s calls are not stored in the CDR, but I

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

Hello, I've tried to transfer a IAX call to a number configured on a traditional PBX, but it doesn't work. I have a traditional PBX connected with a zap channel to Asterisk in the following way: IAX/SIP client --> Asterisk (FXO) --> (FXS) traditional PBX ---> OFFICE Phones Asterisk is connected to the PBX with an internal number configured inside it. In other words i keep an

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:

Hello! I need to send Faxes from an Asterisk box to an Asterisk + Iaxmodem + Hylafax installed on other box. I have setup IAX trunks between this boxes, all works fine but can?t send faxes from one to other, Im trying with or without NVFaxDetect application but does not work. Is there a way to get it working?. If I connect a fax machine directly to Asterisk with Iaxmodem and Hylafax, I have no

I am looking for a sample sip configuration of a SIP provider that runs Broadsoft VoIP switch. My sip provider is Conecta from Brasil, that only give me a SIP IP address to configure my asterisk 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?

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.

Hola, ¿como están? Me encuentro haciendo un trabajo de Explotación de Información para la facultad y tengo un inconveniente a la hora de preparar los datos para poder operar con ellos. Tengo un atributo "Titulo Secundario" que cuenta con 139 valores diferentes (al final les dejo algunos de estos valores con la cantidad de ocurrencias de cada uno) y lo que quiero hacer es

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 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" =

Hi folks! I have a problem with our Sip provider that have a Softswitch Huawei SoftX3000 to send us SIP calls to our Asterisk PBX, we are working with G711 with them. They start sending calls to our pbx, some time after they start to receive 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

Hi to all the people again, I was writting a simply function in R, and wish to collect the results in a excel file. The work goes as follows, Ciervos<-function(K1, K0, A, R,M,Pi,Hembras) {B<-(K1-K0)/A T1<-(R*Pi*Hembras-M*Pi+B)/(Pi-M*Pi+R*Pi*Hembras) P1<-Pi-B R1<-P1*Hembras*R M1<-P1*M T2<-(R1-M1+B)/(P1-M1+R1) P2<-P1-B R2<-P2*Hembras*R M2<-P2*M

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

Darren, yes, I'd be happy to help. I'll contact you off list to sort out the arrangements. I should warn you that it may be a wasted journey for you, as I really dont know if it will exhibit the problem. Tim. "Storer, Darren" <starusers@comgate.tv> wrote: __________ >Hi Tim, > >TP> So it _may_ not be a problem for me as NTL is a patchwork >TP> of

You are correct... No glare on a PRI W. Kevin Hunt CCIE #11841 www.huntbrothers.com -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Steven Critchfield Sent: Tuesday, May 25, 2004 3:26 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] "Glare" condition - How well does asteriskhandle?