Displaying 20 results from an estimated 10000 matches similar to: "sip.conf not paying attention to allow/disallow"
2004 Nov 22
3
Zap - 256 format frames
Any ideas on this warning? If I call this number, sometimes I get this error
and sometimes the call goes thru fine. Why would it work sometimes?
-- Executing Goto("SIP/3044-8d49", "cytel-outgoing|915124512424|1") in
new stack
-- Goto (cytel-outgoing,915124512424,1)
-- Executing SetCIDNum("SIP/3044-8d49", "2814494000") in new stack
--
2013 Mar 21
2
Allow/Disallow
Hello Everyone,
I have disallow=all and allow=g729 set in sip.conf however, it seems
that asterisk still thinks it support other codecs:
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw). How
can I disable gsm,ulaw,alaw.....
Thanks in Advance,
Nick.
2003 Dec 16
28
codec negotiation
Hi list,
I'm with a little problem on codec negotiation between a cisco827 and
asterisk.
My sip.conf is like that:
[general]
port = 5060
bindaddr = 0.0.0.0
context = default
amaflags = default
allow=g729
allow=gsm
allow=alaw
allow=ulaw
;disallow=all
and cisco like that:
dial-peer voice 6 voip
destination-pattern 0T
session protocol sipv2
session target ipv4:<asterisk-ip>
2006 Feb 23
3
Codec order sent wrong from Asterisk
I'm communicating a softphone (SJPhone) to a Grandstream phone GXP-2000.
The codec order on each one is the next:
SJPhone: GSM - iLBC - PCMA - PCMU
GXP2000: G729 - GSM - PCMA - PCMU
(I have a G729 license, so there's no problem with transcoding G729)
In my sip.conf, I've defined the following codec order:
disallow=all
allow=g729
allow=gsm
allow=g726
allow=alaw
allow=ulaw
And my
2005 Jul 20
4
Alternatives to Digium 729
Per my conversation below with digium, are there any legal alternatives
to digium's G729? It is out of date, and doesn't support VAD nor silence
detection.
Digium has stated that they have no plans to update it anytime soon.
VAD/Silence is a big deal with major carriers and we are having to fight
a battle to get them to make special arrangements to turn off
VAD/Silence in their
2007 Sep 26
2
My G729 problem re-visited
Ok, I built a test system to duplicate my problem and provide myself
a platform that I can mess around with to try and break any features.
My problem is G729 pass-through from a gateway to a phone. I think
I even have transcoding working, which makes me more confused on
what's wrong with my pass-through. It must be a configuration issue.
The basics...
*CLI> core show version
Asterisk
2007 Feb 14
6
Fax with T.38
Hi all,
I install the last version of Asterisk and I tried to send faxes, but
nothing works.
Here is my configuration:
Analog Fax <----> IP <----> Asterisk <----> IP <----> Patton M-ATA
<----> Analog Fax 2
I tried Analog Fax 2 -> Analog Fax but nothing works!!
In the Patton configuration I put G711 and no silence suppression.
In asterisk I have
2003 Dec 24
8
G729 troubles
Hello,
I've successfully installed Asterisk from last CVS and configured it
for using with DLINK-DG104S as mgcp CPE and PGW2200 as external sip
server.
All are work fine at G711 codecs, but then I disable all codecs except
g729 some calls failed (Not all calls. Some calls passed at g729
succesfully).
All my devices configred to use only g729 and I don't see other codecs
at mgcp or sip
2004 Mar 04
10
"Statistiques avec R"
Dear R users,
I want to share my joy with you. Please see the following
excellent introduction to R "Statistiques avec R " by
Vincent Zoonekynd
http://zoonek2.free.fr/UNIX/48_R/all.html
In paticular, you can see a lot of fascinating graphics
examples of R from which you can get many hints.
Soryy if this is already well-known, but the CRAN search
did not show nothing with the keyword
2005 Jun 28
4
Anyone using SipP to produce RTP load?
Hey gang,
I've been able to use sipp to produce some call volume on our asterisk
server. The server has no problems handling 50 simul calls. But then again,
no RTP is being done. I tried to use the rtp echo ability of sipp but that
doesn't seem to work right.
I also setup a fake number in asterisk that when called by sipp, would dial
another number via PRI, hoping that some 729
2004 Dec 07
3
:: Migrating to 1.0.3 => Attention. ::
Hello list ,
I?d like to announce possible problems with migrating any version prior to
1.0.2 to 1.0.3.
Pay attention :
1. Codecs
Codecs names/description have been changed .
For example :
versions <= 1.0.2
voip*CLI> show codecs
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.
1 (1 << 0)
2007 Apr 20
1
G.729 & Voicemail
List,
I have some cisco phones (7940) and asterisk 1.4 running nicely.. Communication
between the phones is G.729, and my sip.conf looks like this:
disallow=all ; First disallow all codecs
allow=g729 ;
allow=gsm
allow=ulaw
allow=alaw
However, I cannot call voicemail - I get the following error:
[Apr 20 14:58:31] WARNING[87184]: channel.c:2816
2005 Oct 18
1
select codec based on extension
I've the following installation :
|asterisk client| --- > |asterisk server| --- > |other asterisk server|
all the connections are made in IAX, the client and first server allows
711 and 729
the other server only allows 729 since it has low bandwidth at disposal
all the numbers but a few are routed to a digium card in the first
server, the others are routed to the other server, this
2010 Mar 24
1
G.729 Codec problem.
Hi,
I purchased a G.729 1 channel codec license from digium. And installed
as per the documentation. Then configured the sip.conf to use the new codec.
For that, I am added the following entries in sip.conf (via web interface,
as i am using asterisknow 1.5)
disallow=all
allow=g729
allow=ulaw
allow=alaw
allow=gsm
After that, when try to call through the PSTN line I can hear the voice of
2017 Nov 01
3
asterisk 13.18.0: pjsip: unnecessary 603 decline because of wrong codec decision
Hello!
I'm facing the following scenario:
- Initial call opened to asterisk: SDP g722,alaw,ulaw
- Outgoing call to provider started with Invite / SDP alaw, g726 and
g729.
- Provider sends 183 Session progress SDP: g729, alaw
- Provider sends g729 rtp packages
But: there is no license to transcode g729.
What is asterisk doing?
Asterisk decides to stop the call at all:
- Sends cancel
2004 Sep 21
4
Voicemail forward to a remote server?
Anybody ever managed to implement a solution where one could forward a
voicemail from one * server to another?
Dominique
2004 Jul 22
1
Sip -> H323 using oh323 and G729
Hi All,
I have set up a box that will be used as follows:
SIP Phone ----> Asterisk ----> Cisco H323 VoIP Server
192.168.1.5 192.168.1.50 192.168.1.80
Asterisk is running the latest CVS and oh323 driver.
The SIP phone is a Grandstream Budgetone 100.
I have everything setup and running with G.711 ALAW and ULAW and i'm able
to make calls through
2005 Aug 23
1
Can't get G729 working after buying a license.
List,
I purchased 2 g729 licenses but I can't get it to answer a g729 call
from a cisco router with a vwic card. In the debug output below you
will see that asterisk thinks it only supports: (gsm|ulaw|alaw|h263)
when it should support g729 according to the config also listed below.
The real odd thing is I can place g729 calls to the router, just not
from the router to *. Anyone have any
2009 Dec 30
2
Skype for Asterisk
Hi Sir,
We have integrated Skype with Asterisk (skype user id:-
rexesbposolutions). Each call which is coming to skype account is
getting transfered to Asterisk Queue. It has following two cases:
case 1: When we call from normal skype account to skype account
(rexesbposolutions), everything is working fine.
case 2: This skype account (rexesbposolutions) has been assigned with a
online virtual
2003 Nov 11
2
FWD codecs?
Hi. There is not much info on the FWD site about this. What codecs do they use? When I try to connect with X-Lite, it works with GSM. When I try to call out with *, it wants G729. I have disallow=all and allow=gsm in the sip.conf. I end up getting errors:
Unable to find a path from G729A to GSM
Unable to find a path from GSM to G729A
What's up with that? I was able to make a call once