Displaying 20 results from an estimated 1000 matches similar to: "pickupexten *8"
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 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
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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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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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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
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"
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<html xmlns:v=3D"urn:schemas-microsoft-com:vml" =
xmlns:o=3D"urn:schemas-microsoft-com:office:office" =
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?
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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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
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
=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 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
2007 Mar 01
1
Digium S101i - pickupexten doesn't work
How to configure Digium S101i adapter to work with "pickupexten *8" ?
I have few Sipura adapters and "*8" work OK but my new Digium S101i
refuses to cooperate.
--
#Joseph
2006 Feb 27
5
res_features pickupexten
is where anyone who knows what is needed to get the pickupexten (*8)
running ?
gentoo asterisk-stable 1.2.4/zap1.2.4 with bristuff
I've activated it in features.conf (default *8) and also tested other
extensions
res_features.so is loaded
show features says:
Builtin Feature Default Current
--------------- ------- -------
Pickup *8 *8
Blind
2009 Dec 14
0
pickupexten on chan_dahdi
Hi,
I'm having trouble capturing calls using the chan_dahdi with dynamic span. Here my settings:
chan_dahdi.conf
[trunkgroups]
[channels]
context=default
switchtype=national
facilityenable=yes
rxwink=300 ; Atlas seems to use long (250ms) winks
; where the ring cadence is changed *after* the callerid spill.
usecallerid=yes
hidecallerid=no
2006 Mar 22
2
Pickupexten not working
Hi group. I have huge problem. My pickup exten #8 isn't working.
This is what I have configured.
pbx*CLI> show features
Builtin Feature Default Current
--------------- ------- -------
Pickup *8 #8
In sip.conf I have
callgroup=2
pickupgroup=2
For called party and same for person that is trying to pick up the call.
The person that is trying
2009 Feb 06
2
upgrade from 1.4.22-rc5 to 1.4.23.1: crash when transferring a call
Hi,
just upgraded my Asterisk from 1.4.22-rc5 to 1.4.23.1 keeping the same
zaptel/libpri/mISDN/add-ons.
It crashes when transferring a call.
Anybody tried it with success?
Thank you
Giorgio
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