similar to: Web Softphone

Displaying 20 results from an estimated 300 matches similar to: "Web Softphone"

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
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" =
2010 Sep 22
1
Costa Rica Hangup Detection
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
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
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
2007 Oct 19
1
Glare on Incoming Calls
How I change my configuration to reduce this issue. I have this setting on my zapata.conf signalling=fxs_ks group=1 callgroup=1 pickupgroup=1 channel=1 signalling=fxs_ks group=2 callgroup=1 pickupgroup=1 channel=2; singalling=fxs_ks group=3 callgroup=1 pickupgroup=1 channel=3; singalling=fxs_ks group=4 callgroup=1 pickupgroup=1 channel=4 and for outbound calls I have this context on my
2007 Jan 17
1
transfer problem
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
2007 Oct 18
1
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
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
2006 Oct 30
6
Asterisk and Panasonic KX Model
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.
2007 Jan 12
4
Nat Question
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.
2007 Aug 15
8
TDM400P FXO click sounds
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
2007 Mar 08
4
Asterisk distributed deployment
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: