Displaying 20 results from an estimated 1000 matches similar to: "ATA 186, firmware SIP 3.1 and codec g.726"
2004 Jul 09
1
RE: ATA 186, firmware SIP 3.1 and codec g.726 + now SIPURA SPA-2000
To me it's a error if I can't complete calls using the ATA configured to use
the g726 codec.
I just tried it usign a sipura SPA-2000 (preferred codec: G726-32) and I
received NOTICES and WARNINGS, but I can't complete a call.
On a zap channel:
-- Executing Dial("SIP/2007-e4d8", "Zap/1/2217008") in new stack
-- Called 1/2217008
-- Zap/1-1 answered
2004 Jan 12
4
RFC3389 messages with ATA 186
I'm getting some warnings:
NOTICE[xxx]: File rtp.c, Line 264 (process_rfc3389): RFC3389 support
incomplete. Turn off on client if possible
Asterisk Version: CVS-01/06/04-13:50:26
Cisco ATA 186 version: v3.0.0 atasip (Build 031210A)
Is this something I should be concerned about? Anyone know how to "turn
off" the RFC3389 support on the ata 186?
Thanks!
2003 Nov 27
4
RFC3389 support incomplete
Hi
When i make a call using IAX2, the log of the remote asterisk say
Nov 17 20:20:12 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389):
RFC3389 s upport incomplete. Turn off on client if possible
Nov 17 20:20:22 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389):
RFC3389 s upport incomplete. Turn off on client if possible
Nov 17 20:20:26 NOTICE[28686]: File rtp.c, Line 263
2004 Apr 28
1
dual x100p and x-lite help for newbie
sorry to bother with this trivial issue, but i am
loosing all my hair
;-)
got 2 x100p's and * on a slakware box
x-lite to x-lite works fine!
i also have:
#ztcfg -vvv
Zaptel Configuration
======================
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
2 channels configured.
and in extensions.conf i got:
[locals]
exten
2003 Sep 20
1
sip tone question
Hello All,
We are running Asterisk on a linux server as a SIP proxy with Cisco ATA 186's at the subscriber end. For long distance we have iax2 connectivity with a ip carrier. For local calls we are routing out through a commercial VEGA voicestream pots unit to an adtran channel bank and then from there to our class 5 soft switch. The sip to sip calls and the long distance calls work great.
2005 Mar 12
1
ATA 186 Codec Question.
I have seen the list of codecs for the ATA 186's but not sure if it was
100% or not.
I want to know really is it possible to run GSM or ilbc on them or is a
G729 lic the only way to get a low bandwidth codec?
This is the list of codecs that I have seen.
RxCodec and TxCodec?Configure the codec ID.
* G.723.1?Codec ID 0
* G.711a?Codec ID 1
* G.711u?codec ID 2
* G.729a?codec
2003 Dec 16
2
Unable to Receive Fax -- RxFAX Application
Hi,
Below if the error message which I got from asterisk.
I was trying to fax to asterisk from my fax machine. I really dunno what
is the problem. I use alaw & ulaw codec only through my ATA 186. Can anyone
help me what could be the problem.
-- Executing Goto("SIP/-080ef9a0", "13732|s|1") in new stack
-- Goto (13732,s,1)
-- Executing
2004 Aug 27
1
xlite Problems
-- Attempting native bridge of SIP/1009-3df0 and SIP/101-f8f1
-- Attempting native bridge of SIP/1009-3df0 and SIP/101-f8f1
RFC3389: 5 bytes, level 0...
Aug 27 08:32:16 NOTICE[23572]: rtp.c:289 process_rfc3389: RFC3389
support incomplete. Turn off on client if possible
Killed
Whenever I make a call between extension 101 and 1009 which are both
Xten Xlite SIP clients, I get that error and
2004 Jun 03
4
miserable time with Cisco ATA186
I'm having a horrible experience getting a Cisco ATA-186 to work with *.
I can make calls from the ATA with no problems. However, incoming calls
make the ATA ring once, and then the call is disconnected. I have no
problems with my Sipura 2000 or my Grandstream phones.
I am running 2.16.1 sip code on the ATA 186. Neither * nor the ATA is
behind a NAT. They are both on public IP addresses
2007 Dec 11
1
RFC3389 message
When making or receiving a SIP call via my service provider, I get the
following message logged by Asterisk:
Dec 11 15:13:37 NOTICE[7392]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xxx.xxx.xxx.xxx
Since the "client" is at my service provider (who uses CISCO kit, I believe),
I don't have the
2003 Sep 29
2
cisco AS5300 : problem configuration
I wouldn't expect you to be using RFC3389 if your using A-law, can you include your IOS version and IOS config file ...
I have not specified any allow's or disallow's in my * config for the codecs with my 5300, I also use Cisco 79xx phones and I use the option within the phones config file to select the preffered codec and when I change this to G.729/A-law/U-law all works perfectly
2003 Aug 01
1
Musiconhold interrupted sound
Hi,
I don't seem to be able to get music on hold to play normally.
The sound gets often interrupted with a few seconds of silence
then starts playing again. I'm using mpg123-0.59r and tried
mp3 files with different sample rates with no luck. If that matters,
endpoints are SIP ata186, SIP Cisco 7960 and H.323 (over chan_h323)
Quintum Tenor.
Sometimes it may play fine for a few minutes
2003 Sep 11
1
g729 codex experimentation
Yesterday, I started to experiment with Cisco to Cisco SIP calls using
the g729 codec. According to the documentation, both the ATA-186 and
7960 are able to make use of the g729.
>From an earlier e-mail, I made a change to the configuration of the ATA,
changing the values:
LBRCodec:3
RxCodec: 3
TxCodec: 3
The first thing I noticed was that when I did a sip show channels, the
format had
2007 Jun 07
1
RFC-3389 problem
hello to all,
i am geting this NOTICE while i am running asterisk.
agents are able to here the customer voice but the customer is unable
to here agent voice
plz somebody help me
#rtp.c:331 process_rfc3389: Comfort noise support
incomplete in Asterisk (RFC 3389). Please turn off on client if possible.
Client IP: 64.34.224.230
--
M. VIDYASAGAR
-------------- next part --------------
An HTML
2004 Jun 11
15
Voicemail problem
I am trying to get asterisk to email me my voicemail as attachments.
What am I missing? Where do I tell it to go for SMTP services?
Voicemail.conf:
;
; Voicemail Configuration
;
[general]
format=wav49|gsm|wav
serveremail=pbx.agtcorp.local
attach=yes
maxmessage=180
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
append=yes
[default]
100 => 1234,Sean Garland,sean@siskiyoutech.com
2004 Apr 27
1
exten fax and capi
Hi to all,
I have configured in my * a avm fritz card and in my isdn connection I have
only 1 extensions.
I read from documentation tht is possible to use the exten,fax to redirect
an incoming fax. Is it possible also with avm card or only with digium card.
I used in this way but with no success:
exten => fax,1,Goto(fax,2202,1) 'Configured for spandsp
exten =>
2004 Jan 17
3
SS7 over Asterisk ?
Hello..
I have a customer who wants to connect 2 PBX's over IP..
The setup should look like this:
[PBX] <-- SS7 --> [Asterisk] <-- IAX --> [Asterisk] <-- SS7 --> [PBX]
Since there are no SS7 cards , I was thinking at a way of carrying the E1 data as bulk...Can I do that ? How ?
Is possible a scenario like this ? I'm thinking of IAX because I don't
2005 Jan 22
2
flashing zap using macro
I'm having problems using the following.
[sip]
exten => _*4.,1,macro(test,${EXTEN:2},${CALLERIDNUM})
[macro-test]
exten => s,1,Answer
exten => s,3,Flash
exten => s,3,Dial(SIP/${ARG2},30,t)
exten => s,4,Dial(SIP/${ARG1},30,t)
exten => s,t,Hangup
exten => s,i,Hangup
exten => s,h,Hangup
I know I must be missing something simple, but here is the output from
2007 Mar 14
1
strange things on call transfer
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi,
I'm setting up an Asterisk system which is connected to an Alcatel 4400
PBX. On the * I permit g729 and gsm as codecs. If I try to transfer a
call by hitting the # key, I get this messages and nothing happens on
the phone:
WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame
that isn't a multiple of 50 bytes long from
2003 May 18
2
G.729: Typical usage scenarios
Clicking on the "For more information, click here" link on the Digium
site nice brings back up the same page I was looking at before, without
any additional G.729 information that I can see.
I'm wondering if some kind asterisker out there could provide us
neophytes with some "typical scenarios" where that codec would be useful
to us.
For instance, I assume that it