Displaying 20 results from an estimated 300 matches similar to: "VoIP Gateway"
2003 Jul 01
2
Unable to get SetMusicOnHold working...
Hello,
I'm trying to do something really easy : transfer a PSTN call to a H323
client. This works great. Now I'm trying to use the SetMusicOnHold
function. I din't find any doc about it, I've just seen some mails in
the list archive, but it still doesn't work.
That's my extension.conf :
[incoming]
exten => s,1,SetMusicOnHold,default
exten =>
2010 Jul 29
3
T.38 fax between ATA's and Asterisk and Cisco PGW 2200
To provide a reliable fax solution for users connected to a Asterisk 1.6.2.6 server i have tested a few T.38 capable ATA's:
- Patton M-ATA
- Grandstream HandyTone 486
- Fritz!Box 7170
I have tried Asterisk 1.6.2.6 compiled with SpanDSP-0.0.6pre17 and also Asterisk 1.6.2.6 with Fax for Asterisk installed.
These Asterisk servers are connected to a Cisco PGW 2200 + AS5400XM.
Sending fax
2009 Mar 12
0
chanspy problems (asterisk 1.6.0.6) - When spying starts, the spied parties can't hear each other
Hi, I am in a predicament and any help/pointers would be appreciated.
We are using chanspy to listen in on conversations. We are doing this via a
web interface. The web interface lists all the ongoing calls. We click on a
call and then my local phone rings allowing me to spy on the session I
clicked on.
But "most" of the time, when I start listening in, the two parties that are
in
2005 Aug 15
1
Re: [Asterisk-Dev] MS Live Communications Server
Search google with "sip pstn site:www.microsoft.com"
You will find out how to configure LCS static routing to SIP Gateway,
like Asterisk
but you need patch Asterisk to support TCP.
http://bugs.digium.com/view.php?id=4903
Step1: configure LCS 2005 to let sip uri: *@pstngw.domain to route to
next hop: pstngw ip address
Step2: patch your asterisk chan_sip.c to support TCP
Step3: configure
2004 Jun 04
1
roaming profil problem
hello,
I've got a strange problem after an upgrade to samba 3. The existing
profils are now unavailable for users, depending on their local rights.
If the user is locally admin, his profil is succesfully loaded on login.
Else there's no error, but the profil isn't loaded. The samba logs don't
show anything special, but i can send them if someone want it.
If I create new users,
2004 Feb 03
0
Minor Registration Problem With Polycom Soundpoint IP 500
We recently took a few Polycom Soundpoint IP 500 to test out in Asterisk environment. So far it has been good. Call Hold, Transfer, DMTF etc.
However, I do notice every now and then the Polycom fails to register with Asterisk. Asterisk console outputs the following:
Feb 3 13:02:32 WARNING[278546]: chan_sip.c:2365 __transmit_response: Unable to determine sequence number from ''
Feb 3
2009 Feb 21
0
help needed -- chanspy
Hi, I require some urgent advice/help concerning chanspy.
We have a web interface that lists all bridged SIP calls (show channels
concise) When we click on the call, our local extension rings. When we pick
up to spy, parties involved in the call that we are spying on are suddenly
unable to hear each other. This happens as soon as we start spying...
why is this happening?
We are using asterisk
2003 Aug 12
12
IP phone recommendation
Hello,
I would like to buy a SIP IP phone, but I don't know wich one to
choose... Can you tell me wich IP Phone is known to work with Asterisk
please.
I've seen the Cisco 7940, but I don't know if it works, and how
expensive is it ?
I'm french, so if you know some french resellers, tell me.
Thanks a lot,
----------------------
Fabrice Tereszkiewicz
Sawadka.org
2005 Dec 07
2
up2date command to install kernel source
What is the correct command using up2date and yum
to install the kernel source package.
Thanks,
jerry
2012 Oct 25
5
trying ti use a function in aggregate
Hi -I am using R v 2.13.0. I am trying to use the aggregate function to
calculate the percent at length for each Trip_id and CommonName. Here is a
small subset of the data.
Trip_id Vessel CommonName Length Count
1 230 Sunlight Shad,American 19 1
2 230 Sunlight Shad,American 20 1
3 230 Sunlight Shad,American 21
2006 Oct 25
3
Quintum DX as gateway to PSTN for Asterisk
Hello,
I try configuring Quintum DX gateway as link to PSTN for *. Now, I can dial number which is connect to Quintum, and call is diverted to *. I don't know what I should set, if I want call from SIP_phone registred in Asterisk to PSTN via Quitnum. I set in sip.conf account for Quintum
[sip_proxy-out]
type=peer
outboundproxy=QUINTUM_IP
, and changed extensions.conf. When
2006 Jun 25
5
FW: Asterisk Quintum A800 SIP Mode
Hello,
I got Quintum A800 with the P5-2-1 firmware. I configure my asterisk trunk
as followed:
[SIP_BD1]
type=peer
qualify=yes
host=192.168.0.254
disallow=all
context=from-pstn
allow=h723
and inside the quantum I change the config sip useragent to 5060. Up to this
part if I run sip show peers, I got:
asterisk1*CLI> sip show peers
Name/username????????????? Host??????????? Dyn Nat ACL
2007 Jun 14
2
Linksys SPA941
Dear Group,
I have just purchased two Linksys SPA941 and flashed these to the latest
firmware.
Everything works well except for the Hold button? Has anyone else
experienced the same issue? What was the solution?
Kind Regards
Shad Mortazavi
2004 Sep 16
3
Creating conference calls from within Astman.
Dear All,
I have a requirement to 'originate' a number of calls to various external
users from within a conference room, so that the end users does not pay for
the call.
I know that within Astman I can define an extension and then originate the
call from that extension. Can I define a conference room (how would I
configure that on astman? What channel would it use?) and then generate a
2003 Jun 07
4
SIP, NAT & Asterisk
Hi all,
--------
beacause I am a newbie in the asterisk ralm and the existing documentation
could not satisfy I'd like to ask you some Questions:
1. Does somewhere in the Internet exist additional documentations for asterisk
configuration ?
2. Does Asterisk work as a standard SIP Proxy ?
3. I am just installing a Asterisk PBX in our institute and additionally I
purchased some ot the Snom
2004 Apr 08
3
Asterisk Server Crashing with New Application
Dear All,
I have been running a successful and very stable call center PBX based on
0.7.1 release. I need to be on this release because of a number of features
that I have complied from 3rd party patches, for the call center. I will not
be able to upgrade to release 1 until the patches catch up and I have done
the required testing.
The system was very stable until two days ago.
The changes made
2009 Jan 10
3
Asterisk/GXW410x IP Analog Gateway
Hello All,
I am trying to setup a small system where Nextone Softswitch will send
traffic to Asterisk and then terminate on Grandstream GXW410x IP Analog
Gateway but for some odd reasons the call are flashed back from
Grandstream to Asterisk and creating a Black loop...
I did follow the instructions provided by Grandstream support but it
doesn't seems to be working...
2006 Mar 21
2
need to make my oh323 work with quintum no gatekeeper
Hi all,
Can someone share with me his experience in making asterisk-oh323 talk to
quintum gateway without gatekeeper.
My set up is QUINTUM GATEWAY ------IP----M ASTERISK (OH323)
Both are gateways.. but I don't know what authentication I will set up in
oh323.conf and how to set it up
I will be glad if anyone can help
Goksie
2006 Apr 05
6
transforming g729 files to wav files
Hello list,
is there any open-source software that recodes g729 sound files to wav
sound files ?
The only way (at least) to do such transformation is with interactive
form on: http://www.asteriskguru.com/audio_conversion.php
Tofik Suleymanov
2004 Apr 16
2
SoundPointR IP 300
Dear Group,
Does any one have experience using SoundPoint(r) IP 300?
I have one call center on Snom 200's I'm adding a second and was looking at
the SoundPoint, but needed some input.
Thanks
Shad Mortazavi
---------------------------------------------------
Nexus Technical Manager
n|m Nexus Management Inc
Sydney
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