Displaying 20 results from an estimated 300 matches similar to: "Skype Howto"
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 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
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
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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?
2008 Feb 20
2
Skype Users
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found this today, I am not a skype user but have read on chan_skype
and don't like aspects of how it is implemented. My thoughts on it are
only theoretical as I haven't used it I just cringe at adding X to a
server. Anyhow there is a new project called sippyskype that appears
to do a similar sort of thing with a couple differences.
1. Its
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
2007 Jul 12
0
No subject
Gustavo A. Gonz=E1lez
Dto. de Infraestructura
Despegar.com, Inc.
ggonzalez at despegar.com=20
=20
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<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
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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
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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
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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 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 May 27
1
Divitas
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2010 Sep 15
1
Error loading skype_for_asterisk
This suddenly started appearing and I'm not sure why. Any ideas?
asterisk*CLI> module load chan_skype.so
Unable to load module chan_skype.so
Command 'module load chan_skype.so' failed.
[Sep 15 11:08:25] WARNING[12274]: loader.c:429 load_dynamic_module: Error loading module 'chan_skype.so': /usr/lib/asterisk/modules/chan_skype.so: undefined symbol: sfa_send_chat_message
2009 Mar 23
2
Skype for SIP
Anyone connected up to it yet?
http://www.skypeforsip.com/
It would seem to make Digiums chan_skype rather pointness, or am I missing
something?
Or is this Digiums chan_skype in a hosted box somewhere?
Gordon
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 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 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 Jul 25
3
Asterisk 1.4.9.tar.gz download fails
Hello Fellow Asterisk Mailing ListMembers,
When I tried to download the latest version of Asterisk this is what I get:
http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.0.tar.gz
Opening fileinfo database failed
http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.0.tar.gz
Opening fileinfo database failed
Where are all the latest Asterisk 1.4.x source files?
Thanks in advance,
-E