Displaying 20 results from an estimated 1000 matches similar to: "Problem with IAX and codecs"
2004 Jan 18
2
Nufone not taking GSM CALLS
Is nufone having problems taking gsm calls today
i had some issues dialing overseas to call my folks.
here's snip of what the console displayed
-- Executing Dial("SIP/2204-a279", "IAX2/fgravato@nufone/011351217907000|100|T") in new stack
Jan 18 10:30:02 WARNING[1200884528]: chan_iax2.c:5036 iax2_request: Unable to create translator path for UNKN to GSM on IAX2[NuFone]/1
2004 Jun 01
1
Stuck SIP channels? -> SIP show channels
Hello all
I've discovered that SIP channels sometimes get stuck in *.
I've read some posts from Fri 29 Aug 2003 which mentions this issue, but
there doesn't seem to be any final answers
I don't know if this is related to the 0001604 bug?
Below is a list from one of the incidents:
I know the (d) means that it is scheduled for destruction but the 10.1.1.45
channel hasn't
2007 Feb 27
1
Help understanding SIP SHOW CHANNELS
I have a high volume asterisk 1.40 installation and I ran a SIP SHOW
CHANNELS. (see partial output below). My questions are:
1. "wc-l" of the output shows 4000 lines. Does this mean 2000 active calls?
(2 channels per call)
2. The latter part of the output shows "unkn" for Form column. Why does it
not know the codec? Could it be UDPTL? Or are these calls messed up?
3.
2007 Sep 06
1
Dead SIP channels
I am using a2billing as calling card platform with asterisk 1.2.17.
After running for several days, if I issue 'sip show channels' command, I got a lot of dead sip channels although 'show channels' command only show 5 channels. What cause these dead channels? How can I clean out these dead channels? Will they pose any problem to my * server if left alone? What does this (d) mean?
2003 Sep 25
3
SIP codecs Errors
Hi all:
I recently update a system from CVS (Asterisk CVS-09/25/03-15:58:42), and I receiving the following message:
*CLI> WARNING[1187305408]: File chan_sip.c, Line 1864 (process_sdp): No compatible codecs!
The "show codecs" command shows:
*CLI> show codecs
1 (1 << 0) G.723.1
2 (1 << 1) GSM
4 (1 << 2) G.711 u-law
8 (1 << 3) G.711 A-law
16 (1 <<
2008 Feb 13
3
urgent-channels
Hi All
I am using asterisk 1.2.4
Please see the results when I execute Sip show channels
X
X
X
X
x
192.168.8.106(None) 04cddc1f5a0 00101/00000 unkn No
215.96.142.83 (None) caac0846-cf 00101/00000 unkn No
192.168.8.106(None) 94910146-46 00101/00000 unkn No
192.168.8.106(None) 793ed1eb0f2 00101/00000 unkn No
85.219.172.253 (None)
2004 Aug 26
0
Asterisk media problem behind NAT
Hello All,
I have a media problem while using sip communicator
user agent with asterisk behind NAT.I had enabled the
debug mode in asterisk and capture the results.I have
attached the results with this mail.Can any one help
me to fix the problem?
Thanks in advance,
Partha
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2005 Aug 12
2
trying to get live_xmms mounted
Hi, there. I am new at list and I would like to introduce myself: I am from
Tuxtla Guti?rrez, Chiapas, M?xico, and I am in Spain right now, studying at
university, i am majoring in new tecnology in media. Actually i am trying to
mount a radio statio with this software, Ice cast, but I don't understand how
to make my xmms player get mounted in live_xmms in ice cast. In fact, i only
can
2006 Jun 22
3
Showing Current Calls
Can someone recommend the best way to view current calls in progress on the Asterisk console?
Neither the 'show channels' or 'sip show channels' commands are easy to read.
hestia*CLI> show channels
Channel Location State Application(Data)
SIP/2944093-f9e2 (None) Up Bridged Call(SIP/2944079-e7f2)
SIP/2944079-e7f2
2007 Dec 07
2
7960 Won't Register Yet Multiple Attempts?
Hi List,
I've got a 7960 that's behind NAT (nat_enabled: 1 and
nat_received_processing: 1) and for whatever reason doesn't seem to
register, or at least hold a registration. If both the phone and the
router (netgear) are rebooted, the phone will register, take a few
incoming/outgoing calls no problems, then a few hours later, it drops the
registration and never re-registers. If the
2005 Aug 12
1
trying to get live_xmms mounted
thanks for the answer... another question...
how can i connect a microsoft pc running winamp or any sound streaming
software to Icecast server???
El Viernes, 12 de Agosto de 2005 18:07, ml escribi?:
> Jes?s Abidan Ramos Salas wrote:
> > Hi, there. I am new at list and I would like to introduce myself: I am
> > from Tuxtla Guti?rrez, Chiapas, M?xico, and I am in Spain right now,
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
2004 May 18
0
No luck using asterisk as proxy...
Still no luck using asterisk as a proxy.
48 hours solid working on this. I'm beginning to think asterisk isn't
going
to be compatible with the provider I'm using :(
Has anyone got *any* clues as to what can cause this message? It's
definately
provider specific (voiptalk works, pipecall doesn't) but confusingly
seems to
be caused by something in the client phone app.
I
2004 Sep 23
0
RE: An old problem still hanging around?
Having just run the command "sip show channels" I get a list of channels
even though there is no one on the phone (we only have 4 so it's easy to
tell).
Here is what I get:
Peer User/ANR Call ID Seq (Tx/Rx) Format
192.168.0.22 (None) 4c81ac8e90c 00101/00000 UNKN
192.168.0.22 (None) 984ee48048d 00101/00000 UNKN
192.168.0.22
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 Sep 27
1
Continuing Budgetone woes
I have spent the morning on this project, still without success.
Summary: Yesterday I inadvertently unplugged my Grandstream phone. I
might add I did a rebuild of my s/w from CVS at the same time. Since
then, the Budgetone seems to talk SIP just fine, but the RTP being sent
to it by asterisk "doesn't make any sound."
It was suggested I do a factory reset of the phone, which I
2004 Jan 06
1
ATA call
Hey all!
I'm having problems trying to set up an ATA 186 with my Asterisk box. When I
get the phone to place the call, I type the extension and I only get busy
signal after 5 seconds. So I can't call my Asterisk box from my ATA and
either call from my Asterisk to my ATA.
Does anybody know what can be happing?
Log is attached..
tks
regards
Oz
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>
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
2004 Jun 09
0
Call Pickup problem in Asterisk with SIP phones
I'm having a tough time getting call pickup to work on *. Here's my
configuration:
X100P with T-1, channels 1-4 voice <---> * <---CISCO 7960 with SIP 6.0 Image
A call comes in, and * picks up and presents a menu. Caller chooses
extension, (in this case ext 103, SIP/wsmith)
Wsmith is sitting in my office, hears his phone ringing, picks up my phone,
gets dial tone, and presses
2007 Jun 12
0
Zombie SIP channels
on sip show channels I do get a lot of entrys like
192.168.1.47 11 07ba5a490b3 00102/00000 unkn No
Init: INVITE
192.168.1.47 11 19090f115b8 00102/00000 unkn No
Init: INVITE
192.168.1.47 11 7d8b8fde46f 00102/00000 unkn No
Init: INVITE
How do they appear?
How can they be removed? "core show channels" does not list them.
Elmar