Displaying 20 results from an estimated 70000 matches similar to: "Call limits per server with Iax"
2004 Dec 27
0
IAX -> SIP Call Help; IAX with G729
I have 2 asterisk boxes: asterisk-alpha (running 1.0.3) and dev-asterisk
(running latest CVS).
I am the only SIP user on dev, everyone else in the office is on alpha.
If someone dials my extension, it should go IAX to the dev server and the
dev server should ring me.
Here is what I see on the dev machine's console:
-- Accepting AUTHENTICATED call from 192.168.1.25, requested format = 256,
2007 Jul 15
2
Asterisk with iax2 over satellite
Hi guys,
I'm in the process of setting up an Asterisk server over a satellite
connection to allow people on a remote island to place and receive calls
over the pstn.
What are the ideal settings I should use in iax.conf for the optimal
operation over satellite besides the normal options for the type=friend
peer?
Does anyone have this working? I an place calls as things are now, but there
is a
2005 Mar 28
0
Local/Remote * Servers, IAX/SIP mix and voice-mail notifications
We currently have an Asterisk server set-up, serving a handful of
sip-phones and sipuras, and connecting to the outside world via one FXO
and various SIP and IAX providers. In order to conserve bandwidth and
have a fall-back in case our internet connection becomes unavailable,
we're looking at putting * on a hosted server and funnel the calls to
our office, such that:
DID DID DID
2010 Feb 08
0
Help with iax.conf {tesco|freshtel} 1.6
I have something going on that I don't fully understand after a weekend
of looking for answers.
I have an iax account with Tesco that works flawlessly with the Zoiper
client - but is giving me trouble with inbound calls in Asterisk 1.6.
After some playing I have ended up with an iax.conf file that looks like
this:
[general]
calltokenoptional = 77.75.0.0/255.255.248.0
maxcallnumbers = 16382
2008 May 21
1
speex, ilbc and g729 codecs, GSM with IAX
Dears;
I do not know if any had experience in using speex or
ilbc with IAX and got good results, because I am
facing a problem with GSM.
I am facing a noise problem when I am using GSM with
IAX trunk as following:
IP Phone (G711) ---> Local Asterisk Box ---> IAX Trunk
using GSM codec ---> Remote Asterisk Box ---> Digium
Card (FXO) to terminate the call to the destination
While no
2004 Sep 29
1
Asterisk 1.00 Call quality problem
I upgraded from RC2 last night, but have a major call quality issue.
Heres our setup:
1 FXS and 1 FXO card.
Incoming/Outgoing calls via IAX trunking from our provider. G729
running between us and the VoIP provider.
Two handsets, one BudgetTone 102 and a Cisco 7940G running the 7.2
SIP firmware.
Both these phones are using ULAW to the server, and we have plenty of
G729 licenses on the server.
2009 Nov 21
3
Connect two Asterisk Server in IAX ?
Hi
My first post get no answer :=<, i post new with new elements.
I have two Asterisk server, running on Asterisk 1.6:
SRV1 = 192.168.0.5 on Asterisk 1.6.1.4
SRV2 = 192.168.0.20 on Asterisk 1.6.1.8
I want create a link for exchange call.
on Srv1:
iax.conf:
[general]
bindport=4569
bindaddr=0.0.0.0
language=fr
bandwidth=low
jitterbuffer=no
forcejitterbuffer=no
autokill=yes
2008 Nov 13
0
Problems with Licensed g729a codec from Digium
Firstly, I'm running Asterisk 1.4.4 on Solaris 10.
I have several different internal SIP phones all sharing a single IAX2
VoIP channel.
PHONES |------------- <SIP/uLAW> --------------| ASTERISK
|-------------- <IAX2/g729> ------------|VoIP/ISP
The g729 codec has been registered successfully and appears to be
detected by Asterisk
(NOTE: I have changed what I thought might have
2005 Mar 05
0
Are codec "capabilities bitmasks" different in IAX and SIP?
I didn't know how else to caption this.
I'm trying to play around with codec pass-through. I have two SIP
phones, both with g729, behind two Asterisk servers.
I set all the configs, SIP and IAX, to "disallow=all; allow=g729" on
both servers.
But the originating server won't even try to call the destination server:
-- Executing Dial("SIP/btel-c7d7",
2005 Oct 16
1
iax invtation problem
i had a sip invitation problem with my voip provider
and here the message that was shown :
Oct 16 20:23:19 WARNING[21901]: chan_sip.c:6890
handle_response: Forbidden - wrong password on
authentication for INVITE to '"asterisk"
<sip:XXXXX@195.112.214.99>;tag=as7b43dfbd'
-- SIP/callshop-3fcc is circuit-busy
== Everyone is busy/congested at this time
-- Got SIP
2006 Feb 01
1
(newby) IAX Trunk on low bandwidth connection
Hello everyone, this is my first post to the list, so hello again.
We're a small company in Romania and we're trying to set up a really small
version of "call center". That is, we want to get a few land-lines from our
telco in different countys and "bridge" all calls to our HQ, in order to
make it cheeper for our clients to call us.
Unfortunatelly there's no ISP
2005 Feb 24
1
Which Codec(s) to use..?
Hey Everyone,
I am playing around with my * box, and I have a few different phones
hanging off it it right now.
I have a Cisco 7960 capable of g729, ulaw and alaw, I have a Cisco
ATA186 with a Panasonic cordless phone attached to it, I have a Digum
IAXy with a dumb analog phone attached to it, and I have a Linksys
PAP2-NA with an AT&T 959 analog phone attached to it.
I also have several
2006 Mar 29
1
Avoiding initial deadlock on iax?
Hi,
My asterisk sometimes stop responding to iax calls.
In the log, I've found this:
Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for
'IAX2/trunkjstpcn-3'
Mar 29 13:35:45 DEBUG[13002] chan_sip.c: update_call_counter(1409) -
decrement call limit counter
Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for
'IAX2/trunkjstpcn-3'
Mar 29
2006 Jan 28
0
Re: 5, 000 concurrent calls system rollout question
What about IAX - SIP or IAX - IAX?
----
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
----- Original Message -----
From: <asterisk-users-request@lists.digium.com>
To: <asterisk-users@lists.digium.com>
Sent: Saturday, January 28, 2006 5:43 AM
Subject: Asterisk-Users Digest, Vol 18, Issue 185
> Send Asterisk-Users mailing list submissions to
>
2004 Apr 26
0
Help with connecting 2 servers via iax
I have successfully configured two servers and I am now trying to connect
via iax. When I attempt to call from one ext, 2006(server viop1) to
extension 3006 (server voip2) I receive a timeout or "call failed 403
forbidden.
The information I am receiving from the console is below.
Apr 26 10:53:32 WARNING[311313]: channel.c:1745 ast_request: No channel type
registered for 'IAX'
2005 Oct 02
0
iax invitation problem
i have opened an account with callshopcompany,and
when ive tried to send calls by the sip i had a
message show an asterisk invitation problem i had
these sip configuration:
sip.conf
[callshop]
type=peer
host=sip.callshopcompany.com
username=XXXXX
secret=XXXXXXX
Then i tried to add these lines and it worked :
sip.conf
[callshop]
2006 Nov 20
1
g729 registered
Hi guys,
I've registered some g729 licenses, during register process everything
worked fine.
astk2*CLI> show g729
0/0 encoders/decoders of 20 licensed channels are currently in use
But I'm not able to use this codec. I'm trying to use a linksys PAP2 to talk
using g729 but I got this answer from asterisk:
Got SIP response 488 "Not Acceptable Here" back from
2006 Feb 17
4
one way / irratic voice over iax and g729
Hi All,
We are experiencing a a problem when running calls over IAX with g.729.
The call flow is as follows:
Sip handset -(SIP)> Asterisk1 -(IAX)> Asterisk2 -(SIP)> Carrier
The first Asterisk system is running 1.2 and the second is running 1.0.
When using g726 from the handset all the way thru to Asterisk2(then 729
for the carrier leg) calls go thru fine, but when using g729, there
2005 Oct 07
2
Teliax users, g729 question
I am using Teliax to terminate my calls, and I have 3 licenses' for
g729 from Digium. "show translations" verifies that the registration
took place.
When I place a call, having "allow=g729" as the only allow option in
iax.conf, I get the following error:
WARNING[361]: chan_iax2.c:6017 socket_read: Call rejected by
208.139.204.228: Unable to negotiate codec
If I place a
2008 Jan 28
2
IAX Calls - One Way Audio
Hello List,
I am currently having a bit of a strange issue with a pair of asterisk servers that we recently set up.
For a bit of background, this particular business has two sites in two different towns, about 10 minutes apart. They have 3 analogue PSTN lines connected to the asterisk servers at each location, via a Sangoma A200 (with HEC). They are trying to have just the one receptionist for