Displaying 20 results from an estimated 3000 matches similar to: "Codec matching weirdness"
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 Nov 22
2
Granstream BT100 - only partial success
We are having many successes with Asterisk and starting to get the hang of
it.
But, I am still having problems getting my Budgetone BT100 (firmware
1.0.4.50) to work fully. I can receive calls, but cannot make them.
We have the latest version of Asterisk, Fedora Core 3, Digium TDM400P with
one FXO and one FXS card configured and working well. We have a PSTN line
going into the Digium card,
2004 Oct 07
1
Confused about NAT and Authentication with FWD
I have recently started experimenting with Asterisk. I am running the system the other side of the a NAT router and trying to connect to FWD. I have opened UDP ports and have configured sip.conf to handle NAT.
The problem:
I can call from the FWD phone and the extension on Asterisk rings and there is two way sound so no problem.
Now if in the extension.conf file I have,
exten =>
2005 Jan 31
0
Strange sip address?
Hi all,
I am struggling to make my asterisk server work. The problem is I can not
place a call from a phone outside, but I can call out from a phone in the
local network where the asterisk server sits.
I turn the debug on, and the log are shown below. I can see "REGISTER"
method is OK. ( SIP/2.0 200 OK) But Later, in the "INVITE" method, the
SIP addresses become
2004 Jul 13
1
codec issues between linphone and *
Hello
I am trying to connect linphone 0.12.2 to an * 0.9.1 box over a LAN using the
console version of linphone. both boxs are using the latest alsa drivers on a
LFS kernal 2.4. I am running into errors with codec compatability between
linphone and *.
A point to note is that I am able to connect to asterisk using other sip
phones noteably sjphone however linephone is giving me
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
2006 Feb 20
2
spa3000
I'm trying to get working a spa3000 with asterisk. My problem is I
cant get wroking the incomming calls (I installed the lastest
firmware). My problem is asterisk is rejecting the authentication from
the spa3000. Asterisk answers forbidden (SIP/2.0 403 Forbidden) and I
think I placed the username and password correctly...
Sip.conf says this:
[linea2]
username=linea2
type=peer
secret=1111
2009 Apr 08
1
Call Pickup Works w/Linksys ATA, not with Cisco 7940G
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<html>
<head>
<meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
</head>
<body bgcolor="#ffffff" text="#000000">
I have an Asterisk 1.4.18 with a mix of cordless phones connected using
Linksys SPA2102 ATAs and Cisco 7940G
2004 May 28
2
Asterisk with Draytek 2600V
I am unable to get a my Draytek working with our Asterisk server. I can
make/recieve calls but get no audio. I have tried the various codecs at the
Vigor end but still getting nothing. I looked at sip debug (below) but am
new to Asterisk and don't really know what I am looking for. Asterisk works
fine with XLITE so I know my installation is ok.
Sip read:
INVITE
2005 Sep 24
2
Asterisk returns 484 ADDRESS INCOMPLETE for incoming SIP calls
I'm new to asterisk and need some help with getting a SIP connection
working.
I am trying to establish a termination point/DID number in another
country. I am currently running Asterisk CVS-HEAD. My foreign provider
uses SIP and authenticates via IP address. I am not required to
register my SIP connection in order to send or receive calls.
Can someone help me with how to understand the
2005 Jul 02
1
Sipura SPA2000 behind NAT
Hi, I've one Sipura SPA2000 at home behind a linuxbox with two network
adapters (eth0 for WAN and eth1 for LAN) doing NAT/DHCP:
___________ HOME _______________ ____OFFICE ____
SPA2000 <---> Linux Box <--> Asterisk Box
192.168.0.253 192.168.0.1 eth1 200.93.xxx.a
200.93.xxx.b eth0
My problem is when I try to call to any trunk or extention
2009 May 22
3
No response to our critical packet problem
Hi,
I have a strange problem. At a site where there are 20+ phones, there
is one phone that cannot make outbound (to PSTN) calls.
Each call is dropped after 20s with "no response to our critical packet".
Calls to voicemail and internal extensions work fine.
I understand that everything points to a NAT problem, but I don't
understand how it could be because:
1) It does not affect
2003 Nov 05
0
SIP broken for budgtone.
I just downloaded the newest version from CVS(Tuesday@~7pm) and I am getting an error whenever I call the asterisk box. I cannot here any audio on the budgtone. This works fine with my pingtel phone and my sip 7960. Also if I call my Skinny 7960 it rings but I get that same error when I pick up. When the skinny phone calls the Budgtone it works fine. I have 2 budgtone phones and it does this on
2007 Nov 20
0
sl75 wlan not able of being pickuped?
Hello.
I have a strange problem. Its not possible to pickup a call that was placed
with a Siemens SL75 Wlan. When this phone calls an internal number and i try
to pickup (*8) the call from my phone i get nothing. It seems i have the call
for one second or so but after that the call is being cancelled. No problems
with other phones (polycom, grandstream). Attached the complete sip debug log
2005 Feb 20
0
SIP to SIP calls have no audio until put on hold and taken back off
A previous poster mentioned the same thing, with no response:
http://lists.digium.com/pipermail/asterisk-users/2004-
December/080161.html
Fresh asterisk 1.0.5 install on FC3, started with "make samples",
nothing fancy. It's so bland, I'm surprised the list isn't full of
people having the same trouble.
I have several Uniden UIP200 phones and a single Grandstream BudgetTone
2003 May 25
1
iconnecthere problem 481 "Call Leg/Transaction Does Not Exist"
Hi All,
I am trying to use iconnecthere to make outbound calls. I am behind a
linksys router. I keep getting this error
481 "Call Leg/Transaction Does Not Exist". Does anyone have any prior
experience with this problem. Any leads will be much appreciated. Attached
are the conf files and logs
#SIP.CONF
; SIP Configuration for Asterisk
[general]
port = 5060 ; Port
2003 Oct 23
0
SIP Call Seq Error (SIP/2.0 481 Invalid CSeq Number)
Hi all:
I've no response for the last question with the same subject. Please excuse
me for the extreme length of this mail, but I send 2 SIP traces.
I have problem with * and 5300, when the incoming and outgoing call are
routed thru the same SIP gateway (AS5300). Do I need to set an special
things in sip.conf?
First all, the * printout. Second, the 5300 trace.
Thanks in advace,
Gus
2006 Mar 08
4
PAP2 won't make two g729 calls at the same time
I have a Linksys PAP2. Identical setups for the two channels in both
the unit and in Asterisk. In particular, both channels enable g729 and
set it as the preferred codec, and have disallow=all and allow=g729 in
sip.conf.
If we make a call on one channel, it works (and uses g729), but if we
make a call on the other channel when the first one is still connected,
it fails. We have three g729
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
2003 Oct 23
0
WAS: Call pickup (*8) on SIP devices. Bug #116
I've attached two SIP debugs in reference to bug #116. They are from
today's CVS build.
1. pickup.txt is a call from SIP(1) to SIP(2) with SIP(3) picking up the
call. After which, SIP(2) rings for about 30 seconds then stops.
2. hangup.txt is a call from SIP(1) to SIP(2) with SIP(1) hanging up before
the call is answered.
SIP(1&3) are Cisco 7960's and SIP(2) is a Polycom