Displaying 20 results from an estimated 1200 matches similar to: "SIP call doesn't execute the 's'-extension"
2005 Aug 22
1
Re: MWI problems on 9133i
Thank you Melissa. I love the phone but the dial keypad is a little bouncy.
I was hoping for a more solid feel like on the analog PT390's or my quality
standard, the Nortel 9417CW.
Other than the MWI problem, I'd like more documentation on the configuration
paramters. I have found little online configuration documentation other
than very basic stuff on the Sayson website. I'd
2004 Jul 08
0
Problem SIP no audio just noise
I'm trying to call from XLite phone to PSTN
(I've tried this from internet and from local network the same)
The Xlite doesn't write that it is connected but receives excelent audio.
At the other end comes only noise. Some times only for a second you can
here the
caller voice , but this was only one time :)
I saw with ethereal that UDP packets are coming and going to the
asterisk
2004 Apr 01
4
sip problems
chan_sip.c6524 reload_config= unable to get ip address from asterisk,
sip disabled
The ip address is working fine, Internet works great. Can anyone
help...Thanks
2007 Feb 04
5
Unicall/R2 for Asterisk 1.4 Available for TESTING
Im glad to let you know that finally I invested some time to make work
Unicall in Asterisk 1.4, I must say not much testing could be done
since I have no hardware available ( cards, servers ), however a
friend was able to test it with a couple of calls with success, I need
you to test this and report some feedback.
The sources are available in:
http://moy.ivsol.net/unicall/soft-switch/r1b1/
2009 Sep 10
1
g723 to wav conversion
hi everybody,
I try to record a call with *1 - one touch record feature in g723 format.
exten => _00[1-9].,1,Set(TOUCH_MONITOR_FORMAT=g723)
exten => _00[1-9].,n,Dial(SIP/${EXTEN}@ext-sip-account,,wW)
I have chosen g723 format because in my
CLI> show translation
there is no translation between g723 format and wav (default for *1
feature).
After pressing *1 sequence I have two
2008 Apr 24
1
G723 pass thru
Hi,
I have softphone with a g723 codec, my question is how do i set it as Pass
thru in Asterisk?
cheers,
Aby Azid
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2004 Apr 25
3
Grandstream Budgetone G723, G729 or any compression
Hi, does anybody made G723 or G729 to work with a GrandStream Phone ? I've
a Cisco here and it works fine with G723, but not with my asterisk. The
bandwitdh is very important, since we will have our extensions at home. I
know that I have what I pay, but the phone works with cisco.
Trying to use G723 or G729 Asterisk says no codec available.
Does anybody have it working with any compression
2009 May 19
1
Alternative to Adobe Audition 3 for G723 > G711 uLaw ? (old Cool Edit Pro)
Can anyone recommend a codec pack with G723 for use under Vista? I have
G723 prompts (about 70 prompts totaling 1MB) needing to be converted to
G711 uLaw.
I tried Audacity but it doesn't have G723 codecs. I tired some google
found adware free tools and websites with no success in converting G723.
It does appear the old Cool Edit (now Adobe Audition 3.0 for $349USD)
can do it -jason
2003 May 05
3
G723 - Has anyone gotten SIP_CODEC= to work?
FYI, asterisk DOES now support g723, but you have to pay for it:
http://store.yahoo.com/asteriskpbx/asteriskg729.html
-----Original Message-----
From: Dan Fernandez <danfernandez00@hotmail.com>
Date: Mon, 5 May 2003 17:33:05 -0300
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Has anyone gotten SIP_CODEC= to work?
Basically, since I?d like to use g723 for sip
2007 Jan 19
1
Asterisk 1.4 and g723
I am setting up Asterisk for use in a low bandwidth environment. As
bandwidth is precious and our ATA's support it, the decision was made to
use the g723 codec. I have been working on this for a few days and have
not been successful. The issue that I am having is garbled noise at the
client on calls whose RTP streams are terminated by Asterisk system.
This is the case for all the dialplan
2008 Mar 22
3
G723 on asterisk 1.4.1
Hi:
How to install and set up my asterisk server with G723 codec to send and receive calls using it.
Thanks in advance;
Wassim
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2003 Nov 19
2
g723 to g723 SIP call - warning message
Hi,
I am calling from a grandstream phone with g723 codec through * to iconnect.
Incoming context as well as outgoing context set to g723.1 codec in *.
Call get connected and I can talk. However I get the following warning,
which scrolls on my screen until I hang-up.
[root@asterisk sath]# cat g723.1
- Executing SetCallerID("SIP/-08122ae0", "1001") in new stack
--
2001 Oct 02
1
Probably broken getaddrinfo() on Solaris x86.
Hi,
I discovered a strange problem with the latest version (2.9.9p2) and previous
versions of OpenSSH when using portforwarding und Solaris 8 x86.
It seems like the getaddrinfo() function on Solaris 8 x86 is somehow broken,
instead of binding a port to 127.0.0.1, OpenSSH tried to bind it to
1.0.0.127 (1.0.0.127 was the ai->ai_addr returned by getaddrinfo() in
channel.c).
I could not
2010 Feb 08
3
High codec translation times on x64
Hi Users,
I was wondering if someone of you have the same thing on CentOS 64x?
asterisk01*CLI> core show translation
Translation times between formats (in microseconds) for one
second of data
Source Format (Rows) Destination Format (Columns)
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729
speex ilbc g726 g722 siren7 siren14 slin16
g723
2006 Nov 19
1
G723 pass-through and codec negotiation
All,
Our users have a softphone client that supports the G723 Codec which we
want to use for bandwidth reasons, however we do not wish (or have the
funds) to license the codec on our Asterisk servers. We have G723
pass-through working between the clients just fine, however calls fail
when terminating with Asterisk itself (i.e. Voicemail) or out to the
PSTN due to transcoding issues.
If it
2005 Feb 27
0
g723 issue+asterisk impropoer shutdown
Hello list,
i have a strange problem iam using the ulaw,alaw and
g729
codecs
in sip.conf i have like this
[general]
disallow=all
disallow=g723
allow=g729
allow=alaw
allow=ulaw
even though i am disabling the g723 any UA could able
to connect to the system and then suddenly asterisk
stops working gives segmentation fault and closing the
process.
in logs i have this messages
Feb 26 16:14:51
2004 Jul 15
2
sip phone configuration problem
I am configuring a sip-phone, receing calls, excellent voice quality. but it does not place calls, please, can some one sort out.
here is my debug output, and below that is sip-debug,
Jul 16 11:34:32 DEBUG[163850]: Setting NAT on RTP to 0
Jul 16 11:34:32 DEBUG[163850]: Stopping retransmission on 'iiasPlzFribMJMcW' of Response 1: Found
Jul 16 11:34:32 DEBUG[163850]: Setting NAT on RTP to
2011 Sep 30
1
Core show translation > 4000ms
Hi list,
we have 2 asterisk boxes in VM (kvm) on 2 different Dell servers, one is
Lenny kernel 2.6.26 asterisk 1.6.2.20, the second CentOS 2.6.18 asterisk
1.4.36 (Elastix). Both 64bits, no hardware involved, dahdi on both
machines for meetme timing.
Doing core show translation give on the Lenny server
Translation times between formats (in microseconds) for one
second of data
2003 May 23
2
Codec problems
hi,
hi we have G729 codec from Digium, without the G729 codec, we can do the hash transfers to other sip phones fine. but once we are using the G729 codec, the asterisk is not responding to hash transfer, ie, when we press "#" it does not detect it and says "transfer..",
is this a problem with G729 codec?
(for testing purposes we have bought licenses for 2 chs)
this also
2003 Sep 26
0
Unable to find a path from ULAW to G723
Hello,
I just CVS'd today and now I'm getting these errors when I call one
grandstream phone to another both using 711U:
NOTICE[1225991360]: File channel.c, Line 1476 (ast_set_read_format): Unable
to find a path from ULAW to G723
NOTICE[1225991360]: File channel.c, Line 1446 (ast_set_write_format): Unable
to find a path from G723 to ULAW
NOTICE[1225991360]: File channel.c, Line 1476