Displaying 20 results from an estimated 9000 matches similar to: "X-Lite & Asterisk: Speex & iLBC not working?"
2005 May 26
1
Asterisk con X-lite : Register Ok but no calls (404 Not found)
Hi all,
I'm working on an implementation of VoIP en Linux.
I have a Debian Suse (*.*.*.173) with an * and a X-lite client and a
Red Hat 9.0 (*.*.*.172) with another softphone X-lite.
Both of the softphones are registering and appear in the peers (sip
show peers) with the good parameters of address and port.
If I try to make a call, * receive the INVITE request and send a 404
NOT FOUND answer.
2005 May 06
2
Newbie *@home + Xten.
I have d/l the iso (*@home 0.9) , built the * box and followed the
directions in the * handbook and
http://www.geekgazette.com/index.php?option=com_content&task=view&id=2&Itemi
d=26.
I created extension 200 and verified that * was running fine.
Loaded Xten lite, setup the proxy for local ip (10.0.0.201) per the
handbook. After turning off the Norton Firewall protection, I am able to
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
2005 Jul 28
2
SIP Debug
Using AMP, the configuration I have used to work fine with Broadvoice.
Now it gets a busy signal every time. I've checked "sip show
registry" and it says it's registered just fine. I've tried "sip
debug" and it shows calls coming in, but they always get a busy signal
& I can't tell why.
Here's a SIP Debug output:
Sip read:
INVITE
2006 Jan 18
1
SIP RTP Negotiation
Dear All,
I am having some problems with connecting with a UA. Sometimes there is not
sound in the call made, sometimes the caller would near no sound, while the
callee can hear the caller. I have attached the rtp debug and sip debug for
you comments. Please help me. Thank you all.
Asterisk Version is 1.2.1
Asterisk RTP Range is 10000 to 20000
UA Listen RTP Port is 15000
Below is the the
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
2005 Sep 18
2
Asterisk Won't Process Call
We have a basic application that runs a SIP channel to pick up a call
and process it. We are using Broadvoice and it's been working great.
We recently rebooted the machine and now when a call comes in Asterisk
picks up the call but does not process it. Asterisk seems to send the
call back to Broadvoice. Nothing at all has been changed in the
configuration to warrant this. Below is the
2011 Mar 23
2
Problems Extension with a Call In on Asterisk 1.6
Hi
I request your help because i don't have actually a solution at my problems.
I have a Asterisk Server in 1.6
Connected at a SIP Provider
This provider supply me 2 numbers:
003318364xxxx (official number)
081169xxxx (Nddi Number)
When i receive a call on the 081169xxxx, he don't use
the extension. He use the 003318364xxxx extension.
SIP Debug:
<--- SIP read from
2003 Oct 06
2
Asterisk, X-Lite and iLBC..still..
Hi,
Has anyone managed to get X-Lite to work with Asterisk using the iLBC
codec.. I have just tried updating the the latest version 1079 (BTW this
new version supports up to 10 proxy configurations, Not that I can see a
reason to have 10 proxies setup, I would rather have the ability to
transfer calls)..
I can make a call using iLBC but the sound that I hear is just a lot of
pop's and
2006 Jan 17
2
IAX/SIP and openser problem. IAX bug?
Hello.
I am in a strange situation. I have two asterisk. Asterisk "A" makes a
call for asterisk "B" by IAX. Asterisk "B" recives the call and delivers
it to Openser by SIP. The problem is openser printing this in the screen:
ERROR: parse_to : unexpected char ["] in status 5: <<"David" <sip:>> .
ERROR:parse_from_header: bad from header
2007 Mar 29
3
Asterisk hangs up SIP call after 6 200 retransmits
I have the following scenario:
PSTN gateway (202.180.nnn.nnn) -> OpenSER 1.0.1 (147.202.nnn.nnn) -> Asterisk 1.2.16
(203.89.nnn.nnn)
When an incoming call is received, often (but not always) Asterisk repeatedly sends a SIP 200 OK message and eventually hangs up the call.
sip.conf
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all
2014 Dec 05
2
Inbound call from sip peer to internal webrtc peer fails while internal sip-webrtc calls work
Hello,
I'd appreciate your comments on the following problem I'm having, please
forgive me if this is something obvious, I've been scratching my head on
this for a while:
I have Asterisk+Kamailio setup where I'm currently testing inbound calls
from outside. I have both webrtc and sip clients, where webrtc peers are
defined according to sip.js instructions (
2011 Feb 10
2
Unable to make outgoing calls with Internode
Surely there must be someone here who can help me with this problem.
I have spent weeks trying to get this damned service to work with no
luck. I have incoming calls working, but no outgoing. If get outgoing
working then incoming don't work.
I have sent this problem to this list a couple of times with little or
no response, and I _really_ need some help to sort it out.
I have an asterisk
2005 Feb 17
1
Voicepulse Open Access & Asterisk Problems
I can't seem to dial out with Voicepulse Open Access service using *.
Incoming works fine. Another user posted a few weeks back that they
were having problems and there are some threads at dslreports.com
about this as well. Maybe someone here can figure out what the issue
is from the sip debug info below. I am at a loss.
The audible error message from Allison is 0984 (from VP server)
Here is
2008 Mar 16
1
Problem with incoming calls on Broadvoice after upgrade to 1.4.18
Hi all,
I just upgraded to Asterisk 1.4.18 a few days ago and I don't use
Broadvoice TOO often, however I have a Vermont number with them and so
my mother in law calls it to talk to my wife once in a while, so
that's why it took me so long to notice it wasn't working. Anyway,
when she calls she gets a busy signal (as I've tested when calling it
from my cell).
When I enable
2005 Jul 10
3
Incoming calls from BudgetPhone.nl
(this time with subject....)
Hello,
I?m trying to get Asterisk to accept incoming calls from budgetphone.nl.
When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy
tone.
I tried X-lite, which worked perfect, so my modem (with nat) probably is not
the problem.
I did a sip debug and got the following output.
Because I?m new to Asterisk I can?t get the error why this is not
2010 Dec 20
2
SIP 420
Hi;
I am running asterisk 1.6 from Fonality (Trixbox PRO).
I am trying to initiate a call FROM a softphone client to asterisk (either
an internal 4 digit extension call) or an outside line via a SIP trunk.
In both cases, asterisk rejects the call with a 420.
In this case, it?s a call from x3992 to x4415
Does this require a change on the softphone for x-call-detail?
<--- SIP read
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 Mar 11
2
g711 -> iLBC garbled voice in 1.4?
All,
Has anybody else experienced garbled voice between a phone using
alaw/ulaw and one using iLBC? I have a Nokia E series phone with a
preference to use iLBC and this works fine in Asterisk 1.2. However,
since moving to 1.4 - I get garbled voice on Inbound (g711->iLBC).
Outbound voice seems fine (iLBC->g711) though. It's not a 20/30ms
framing issue as the phone uses 30ms
2005 Jul 24
1
Help with Asterisk@home and Broadvoice incoming calls..
Hello everyone,
Well here is my initial posting to the list, and I will admit Asterisk is new
to me. I just got everything running here a couple days ago, so still learning
the ropes for sure.
OK, here is my problem. Currently I have it setup talking to a couple Cisco
IP phones, and some Xten softphones, this works great. I also got an account
with FreeWorld Dialup using IAX2 and that