similar to: Costa Rica Hangup Detection

Displaying 20 results from an estimated 1000 matches similar to: "Costa Rica Hangup Detection"

2008 May 26
5
Skype Howto
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
2009 Apr 23
2
CDR issue
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
2008 Jul 01
4
Fax Between IAX Trunks
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
2009 Jan 09
1
Web Softphone
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:
2008 Jul 23
1
Broadsoft Sip provider
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?
2007 Jul 12
0
No subject
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" =
2007 Jul 12
0
No subject
=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" =
2008 Aug 21
2
Asterisk and Huawei SoftX3000
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
2008 May 23
0
Asterisk chan Skype
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:
2009 Apr 15
1
pickupexten *8
Hello all!, 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
2008 Dec 16
0
CDR and Agents Call recording
Hello, I am running asterisk 1.4.22 and Iam recording calls in agents.conf with the following configuration: recordagentcalls=yes recordformat=wav createlink=yes The calls are being recorded , but no entry appears in mysql cdr, and, on the other hand I have other pbx running asterisk 1.2 that do it with the same configuration. In cdr_mysql.conf I have: userfield=1 accountcode=1 Is there a
2009 Feb 24
1
COSTA RICA - E1
Does any have experience with E1 telephony support plus asterisk in costa rica ? Regards, Luis Morales -- --------------------------------------------------------------------------------- Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 --------------------------------------------------------------------------------- "Empieza por hacer lo necesario, luego lo que es posible... y
2008 Feb 10
3
usability Testing Costa Rica, SanJose asterisk PBX / dsl/cable service
Can anyone that can & is willing to do some IAX usability testing from Costa Rica, San Jose to my Asterisk PBX please pmail your contact info thx tim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080210/b0b0d6e3/attachment.htm
2010 Aug 23
2
DAHDI not detecting caller hangup
Hi, Odd problem have just noticed in that when I call into the PBX DAHDI detects the call and hands it off to the extension, if I then hang up it still continues to process through the dialplan. This is what I have in chan_dahdi.conf: [channels] language=en echocancel=yes usecallerid=yes cidsignalling=v23 sendcalleridafter = 2 hanguponpolarityswitch=yes rxgain=2.0 txgain=3.0 progzone=uk
2007 Jul 12
0
No subject
=20 =20 12:53:41.358166 IP (tos 0xb8, ttl 127, id 0, offset 0, flags [none], = proto: UDP (17), length: 856) 189.8.113.170.5060 > 189.8.126.177.5060: SIP, = length: 828 INVITE sip:7002 at 189.8.126.177:5060;user=3Dphone SIP/2.0 Via: SIP/2.0/UDP 189.8.113.170:5060;branch=3Dz9hG4bKba4h2m2070fhnc4q20k1.1 Call-ID: d6dc25017b171144f35fb9e1c9c393a3 at 10.0.0.10
2009 Jul 27
5
Asterisk core dumps files
Hello all! Im running asterisk 1.4.23 and sometimes it crashes. Because I need to look for what asterisk crashes I run asterisk with option '-g' for debugging purpose. When I search for core files in filesystem nothing happend and I have not generated core files. Which is the way to know if asterisk are generating core dump files? And Which is the directory where it saves them? Is
2007 Jan 16
0
input request: progzone and zaptel hangup
Hi I noticed that my system has three sets of data regarding telephony behaviour in different parts of the world: 1. libtonezone , part of zaptel, and the data is from the source file zaptel/zonedata.c . Zaptel seems to use it for generating some tones. 2. /etc/asterisk/indications.conf . Asterisk uses it to play tones of busy, congestion, dialtone, etc. (PlayTones). The format is pratically
2011 Jan 24
2
Outgoing FXO calls have no audio with callprogress=no
My outgoing FXO calls are answered but have no audio in either direction if I have callprogress=no in chan_dahdi.conf. If I change to callprogress=yes then the audio returns. My chan_dahdi.conf file is listed below. Can anyone point-out why callprogress=no isn't working? #cat /tmp/a [trunkgroups] [channels] language=en context=incoming toneduration=40 ;usedistinctiveringdetection=yes
2005 Oct 05
2
Zaptel tone description
Lilantha, the tones are supposed to be switched using the loadzone and defaultzone lines in /etc/zaptel.conf , and, progzone in /etc/asterisk/zapata.conf. The information about countries and frequencies/times are at zonedata.c located in the sourcecode of zaptel. As you may know, changing zonedata.c information requires a re-compilation of the zaptel module. Hope it helps, Ricardo
2007 Feb 17
3
Problem with busydetect and cell phones
I have a very strange problem I'm hoping someone has encountered already. I've scoured the internet for an answer to this one. My phone company provides no disconnect supervision. Hence I'm forced to use the busydetect feature. I have a TDM400 with two FXO ports. If I call from an internal extension to a landline and then hangup the landline Asterisk detects the busy signal