Displaying 20 results from an estimated 200 matches similar to: "Call IAX2 => "Call rejected, CallToken Support required""
2009 Oct 02
1
IAX2 Call rejected, CallToken Support required
Hi All,
I am using Asterisk 1.4.26.2 and I am getting the following problem
making connections to this server. My other servers are Version 1.2.x
which have no problems and this 1.4.26.2 server can call the other 1.2.x
servers.
The error is:
chan_iax2.c:4251 handle_call_token: Call rejected, CallToken Support
required. If unexpected, resolve by placing address 192.168.25.250 in
the
2010 Mar 22
2
requirecalltoken & receiving IAX calls
Hi All;
I am configuring IAX endpoint, I just need to understand why I have to set requirecalltoken = no to be able to register because the following message is displayed:
[Mar 22 12:25:39] ERROR[2297]: chan_iax2.c:4251 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 78.154.240.146 in the calltokenignore list or setting user gwbilalkwpciax
2009 Sep 04
2
requirecalltoken and Realtime
Hi,
I've just had to enable the requirecalltoken=no option in iax.conf for
one of my IAX2 trunks, and I don't think it works properly in the
realtime version. I've created the requirecalltoken field in my
(Postgres via ODBC) database, type text, and have variously tried it
with 'yes', 'no' and 'auto' in the field. But the setting never seems
to be used and
2010 Dec 16
0
chan_iax2.c handle_call_token: Call rejected, CallToken Support required
I had two asterisk servers connected with each other, both were 1.4.22
but I've upgraded one to 1.4.37 and now I get a message when I try call from asterisk-1.4.22 to asterisk-1.4.37
ERROR[4539]: chan_iax2.c:4330 handle_call_token: Call rejected,
CallToken Support required. If unexpected, resolve by placing address
192.168.140.1 in the calltokenoptional list or setting user guest
2011 Mar 21
1
IAX Call token revisited
I just finished a fresh install of 1.8.3.2 at home using the packages
Digium hosts.
After correcting a number of typo/config'o error that had crept in
over the years, I thought I had everything working.
My wife just complained that she cannot call her mother (who is using an
old IAX hardphone I left for her).
After turning up the logging level I see-
chan_iax2.c: Call rejected, CallToken
2011 May 05
0
Could not place calls through IAX
Hello,
I have some problems in placing calls through IAX... It does not work :)
in the asterisk console I can't see nothing about dialplan enter or
so, IAX debbugging seems to be unuseful...
this is my configuration:
[612]
type=friend
secret=123456
notransfer=yes
disallow=all
allow=gsm
allow=ulaw
allow=alaw
context=from-internal
host=dynamic
requirecalltoken=no
I enabled IAX debugging, but
2011 Oct 18
0
Asterisk IAX Trunking Stops - Too much delay in IAX2 calltoken timestamp from address
Hi all,
Just hit this problem for the first time:
WARNING[17712] chan_iax2.c: Too much delay in IAX2 calltoken timestamp
from address 10.25x.xxx.160
When I ran "iax2 show peers" everything comes up as unreachable, no
calls are passed between the servers (as would be expected) but there
is no problem with DAHDI / SIP channels. At the same time there are no
network issues (can ping all
2016 Nov 03
4
Force hangup not working on stuck channel
I am unable to force a hangup on a channel that has been stuck for over
two days:
IAX2/from-CD-11006 oficina 2770 1 Up
Dial IAX2/to-CD/2883 3467130007 46:24:59 Sotelo
Sotelo IAX2/to-CD-20713
I have tried "hangup request IAX2/from-CD-11006" several times but no
joy. I also see the following in the CLI:
[Nov 3
2010 Nov 25
0
IAX inbound failing
Hi,
I'm testing an upgrade from 1.4.18 to 1.4.37 in a VM prior to putting it
into production.
Ive done this by installing 1.4.18 onto the VM, putting my config files
in place and then installing 1.4.37 over the top (which is what I'd have
to do on production).
I've found a few issues in the config files, but nothing I couldn't
handle until... I hit inbound IAX issues.
My
2020 Mar 02
2
No CID between Asterisk using IAX trunk
I am trying to troubleshoot two Asterisk servers that have an IAX2
trunk between them. Calls come and go but there is no CallerID from the
remote server either way. One of the servers is running Asterisk 16 and
the other is an older 1.8 install (I know, I am trying to get permission
to update). The trunk between servers is very simple. Something like:
Server 1 (Mexico)
[panama]
2011 Apr 20
1
[IAX] Everyone is busy/congested at this time (1:0/0/1)
Hi,
I have a problem with IAX accounts...
I set up a few months ago an Asterisk server, with mysql support to load iax
accounts.
Settings seems fine because apparently the system works as expected.
Yesterday I tried to add another iax account in the iax.conf directly. And I
have a problem with this new account (named 444).
I can authenticate from 444 to the server, and I can receive calls from
2013 Jun 11
2
A problem with IAX2
B.H.
Hello!
We have several Asterik boxes that are connected to PSTN using PRI cards
and they are interconnected using IAX2 trunks so that incoming calls are
delivered from PSTN to the servers they belong to.
In past we were using asterisk 1.4 on the server that is receiving IAX
connections and everything worked as expected. Recently, we have switched
to a newer box with asterisk 1.8.22 and
2005 Jan 21
0
Codec conversion sip peer <> Asterisk
Hi!
There's any way to set up a call using G726 (sip peer) receive it on Asterisk convert it to G711Mu to send it to PSTN broadband termination?
I've put the following in sip.conf:
disalow=all
allow=gsm
allow=g726 (my TAs use G726 32K)
best regards,
Helder
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2005 Aug 06
0
g729 pass-thru for sip provider and g711 ulaw for conference and voicemail
Hello,
I'd like to use g729 pass-thru when I dial out to a sip provider from my
IP phone but because I have no license for g729 I'd like to use g711 ulaw
for asterisk voicemail, conference bridge and other services.
When I set in [general] section of sip.conf the following:
disalow=all
allow=g729
allow=ulaw
the g279 pass-thru works fine with my SIP provider but
when I call the
2010 Feb 22
2
Load balance outgoing calls
Hello everybody.
I have a provider that has 3 asterisk boxes which I must balance my
calls against. At the moment, I route different destinations to
different boxes but this causes lots of problems.
Without resorting to OpenSER or other proxies (as my provider also
uses IAX), is there a way I can load balance outgoing channels in
Asterisk?
For example an IAX peer like:
[iax_provider]
type=peer
2024 Jan 03
1
chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow response
> On Jan 2, 2024, at 23:17, thelma at sys-concept.com wrote:
>
> On 1/2/24 15:13, asterisk at phreaknet.org wrote:
>>> On 1/2/2024 4:03 PM, thelma at sys-concept.com wrote:
>>> I'm using asterisk-16.30.1
>>>
>>> When I try to call another asterisk server over IAX I get a busy signal,
>>>
>>> chan_iax2.c:4739 __auto_congest:
2010 Apr 30
0
IAX trunks and audio codecs
Hi,
I have IAX trunks between Asterisk servers. They receive calls on ISDN cards and Dial() through the IAX trunks to the "primary" Asterisk server where all the SIP phone extensions are registered.
The IAX trunk settings are something like this (all servers have this identical except for the "host" field):
[inbound]
deny=all
allow=alaw
allow=gsm
type=friend
2009 Dec 18
0
Asterisk 1.6.0.20 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.6.0.20.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.0.20 resolved several issues reported by the
community, and would have not been possible without your participation. Thank
you!
* clarify requirecalltoken option in iax.sample.conf
2009 Dec 18
0
Asterisk 1.6.0.20 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.6.0.20.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.0.20 resolved several issues reported by the
community, and would have not been possible without your participation. Thank
you!
* clarify requirecalltoken option in iax.sample.conf
2020 Mar 02
0
No CID between Asterisk using IAX trunk
My Asterisk 13 IAX2 trunk posted below:
type=friend
trunk=yes
allowcallerid=yes
disallow=all
allow=alaw
allow=ulaw
allow=gsm
host=my.super.duper.host
username=my.super.duper.username
secret=my.super.duper.secret
context=sip
qualify=500
qualifysmoothing=yes
requirecalltoken=no
trunk=yes
jitterbuffer=yes
forcejitterbuffer=yes
maxjitterbuffer=300
maxjitterinterps=100
resyncthreshold=1500
tos=ef