Displaying 20 results from an estimated 20000 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>
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
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
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
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
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)
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
2007 Sep 14
1
g729 on 1.4.10.1
I have a fresh 1.4.10.1 installation that appears to have a problem
with g729 pass-through. I can see the gateway in question sending an
INVITE using g729b. However, the Asterisk is only sending g711 in the
INVITE to my Polycom phone.
[gateway]
disallow=all
allow=g729
[phone]
disallow=all
allow=ulaw
allow=alaw
allow=g729
There's the codec configs for the gateway and the phone in question.
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
2003 Oct 03
1
Budgettone + G729
hi there ..
I asked sometime ago regarding getting a Budgettone
working with Asterisk over G729.
My system is quite simple, Asterisk server with 1 G 729 license
installed, and 10 Grandstream phones. Only one of them needs
G729, because it's on a remote link via an ADSL bridge. The
rest run happily on G711 on a local network.
I added the lines
disallow=all
allow=g729
to the sip.conf entry
2005 Jan 02
3
Codec Selection in Asterisk
I am wondering how Asterisk selects codecs between devices. For example,
in my sip.conf I have:
disallow=all
allow=ulaw
allow=alaw
allow=g729
Does the order matter? Does it mean it will try each codec in succession
and use the first that both endpoints support?
Thanks,
Adi
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
2012 Jun 05
1
G729 and voice mail
I am trying to figure out the best way to deal with this. I want all of the
calls in the network to be G729 and this is working. I do have hardware
that provides me 30 g729 licenses. I am setting each extensions to
disallow=all and allow=g729. However when I have this setup, I get no voice
mail prompts. I tried setting to disallow=all and allow=g729,alaw and I
still have no audio when calling
2005 Mar 28
1
H323: g711-g729 transcoding
I have a connect to * via H.323/g711 from device A and want to connect
to B which want for H.323/g729
h323.conf contains
disallow=all
allow=alaw
allow=g729
but outgoing faststart/TCS contains only g711 (from h323_request(format)
i think) and so no codec negotiation and no voice.
Howto run up g711/H323 -> * -> g729/H323
PS intel's g729 was used. ast 1.0.3-6
PPS
stupid
-
2006 Jun 15
3
SIP codec preference order ineffective
Hi,
I set a preference order of the codecs to my sip.conf
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls of not registered phones
disallow = all
allow = g729
allow = g723
allow = alaw
allow = ulaw
Connected a 'Sipura SPA' sip phone to asterisk with g729 as its preferred codec.
Problem: asterisk cannot make
2015 Jul 15
2
Problem "no voice"
Hi list!
I have 4 numbers on my Asterisk 1.8.
3 work perfectly, the 4.th not.
I'm not sure, when it finish to work, since a month ago it runs without any
problem...
Well, if I will be called on this number I can't hear anything and in
Asterisk I see these:
[Jul 15 18:59:55] WARNING[8752]: channel.c:5060 ast_write: Codec mismatch on channel SIP/00493514977290-000001d1 setting write format
2004 Jan 05
2
Codec Negotiation Does not seem to work as expected ?? Help Please !!
Hello,
I have been trying to get my coders to work without a conversion. I have
read all the available asterisk documentation and support groups without
any luck. Here is my issue. (Please feel free to ask questions if you do
not understand what I am talking about.)
I am using Cisco ATA-186 set to g729 codec. (But it will switch to g711 if
sip-server request g711)
I have 2 SIP-services to