Displaying 20 results from an estimated 7000 matches similar to: ""Stopping retransmission on" messages"
2007 Jan 31
1
FreePBX/Debian Aborts Call While Connecting
I used the "FreePBX on Debian" HowTo at
http://powerontech.com/freepbx-on-debian.htm to install. I use callfiles
to initiate calls to my SIP carrier. They get my registration, but they
see that my call is interrupted before they can complete the connection.
My Asterisk log shows that the call times out after the time (45s)
specified in my dialplan Dial() command. What is wrong?
[from
2006 Jun 20
0
Provisional problem with SIP channel
Hi,
I'm using the Perl AGI interface for a prepaid card platform. And
sometimes (almost twice an hour), asterisk doesn't detect a call has
been hung up. The call is so hung up when the time limit for the call is
reached (the corresponding prepaid card is then emptied ...).
I've tried to look in the asterisk log files to find anything suspect
with these calls, and I've found a
2006 May 03
0
Vodini & *
Does anyone have working settings for Vodini and * using AMP / freepbx?
I am having a terrible time getting it to work with my Vodini DID in the
Phillipines.
Even though it looks like it is registering, these weird errors are
still in the logs. What does (provisional) mean?
Any help apprecated greatly.
-Matt
May 4 09:23:24 DEBUG[1904] chan_sip.c: Target address 202.xx.xx.130 is
not
2006 May 26
0
SIP call problem
Hello,
I have problem to make SIP calls, i have asterisk +
PC InterP4 + Digium TDM400P
here is the content of the sip.conf:
[SIP_PROVIDER]
type=peer
fromuser=testcomclient
username=testcomclient
secret=testr
host=IP_SIP_PROVIDER
;dtmfmode=rfc2833
context=interne
canreinvite=no
;allerid=Beer
disallow=all
allow=ulaw
allow=gsm
allow=g723.1 ; Asterisk only
2005 Sep 27
0
asterisk@home inbound call problem to SIP trunk. (voipfone UK)
Hi all,
I have recently installed Asterisk@home and outbound calling is
working great. However I am strugglings with inbound calls. I have
setup a trunk for my provider, voipfone and in the inbound area on AMP
I have the following :-
user context name = 3011XXXX
context=from-pstn
dtmfmode=rfc2283
fromdomain=voipfone.co.uk
host=voipfone.co.uk
insecure=very
secret=XXXXXX
type=user
user=3011XXXX
2007 Jul 08
1
Asterisk and Mitel 3300 ICP
Good day everyone,
I have Asterisk and Mitel 3300 ICP communicating via SIP. Calls to and from
extensions on both sides are completing successfully (details on config
coming soon). Problem is that calls from Asterisk to PSTN via E1 on Mitel
3300 ICP are rejected. What do I need to do to get Asterisk to route PSTN
calls through it successfully?
Here is an extract of the log on Asterisk whenever I
2006 Feb 08
1
incoming call release after 1 ring
Hello,
Can somebody please assist me with my problem.
Currently I am using a Asterisk@HOme version 2.4 with
a TE406P digium card. One the E1 is connected to a
telco switch via an ISDN. May issue is that may
incoming calls in the zap channels gets disconnected
or release after 1 ring. I really dont know what
setting should I change to increase the timeout of the
ring. I have even tried upgrading
2006 Jan 12
2
Random Disconnects
I am hoping some of you can help me troubleshoot this problem I am
having with my home asterisk machine. I have incoming POTS service
using a SPA-3000 (extension 119). Calls on that line go to an
attendant recording that offers a menu choice: press 1 for Nancy,
press 2 for the rest of us. In reality, pressing anything other than
1 sends the call to the rest of us by dialing both extensions 101
2005 Mar 19
1
Asterisk Quicknet FWD Problem - no path to translate from Phone/phone0 to SIP
I cant seem to be able to figure this out. As much as I can tell it is
a codec problem.
I can dial out to 612@fwd.pulver.com and the "Call Me" test there rings
my phone. However when the callee endpoint answers, there is a failure
to translate:
Outgoing Call for 612
612 is not a local user
-- Called 612@fwdpulvercom
No path to translate from SIP/fwdpulvercom-dd5a(2) to
2005 May 20
0
Registering with second SIP service causes error every 2 seconds - what is going on?
I had my asterisk server working fine with FWD as a SIP provider, so I
now added a second SIP provider (voctel). The addition to my sip.conf
file is almost identical to FWD, however, asterisk now generates lots of
debug messages for some strange reason! In particular, the line "#####
Testing 127.0.0.1 with 172.31.0.0" shows up every two seconds! (See my
log below).
If I comment out
2006 Jan 06
0
IAX2->SIP dropped calls
Apparently we've been having calls sporadically drop. We're using an
IAX outbound trunk and SIP adapters on the inside.
Below is a log excerpt detailing one of the calls which dropped, and it
looks largely normal to me except for this:
Jan 5 13:31:07 DEBUG[3776] channel.c: Didn't get a frame from channel:
IAX2/teliax-2
Jan 5 13:31:07 DEBUG[3776] channel.c: Bridge stops bridging
2005 Jan 03
0
queue_log wrong?
Well, I'm writing yet another queue_log analyser program in PHP, and I
have noticed the following entry in my queue_log file from today:
1104796626|1104796618.532|queue|NONE|ENTERQUEUE||no
1104796664|1104796618.532|queue|NONE|EXITWITHTIMEOUT|1
So, pretty sure that I didn't make someone wait 30 minutes in my queue.
extensions.conf snippet:
[remote-oldnum]
exten => s,1,Answer
exten =>
2005 Sep 13
1
wctdm, issue w/outbound calls
Hi all,
I've been running Asterisk with a TDM400P for about 6months, no problems.
2 in/outgoing analog lines, one analog phone. Recently I was messing with
the XTEN client, got to finagling with things, and not knowing what was
wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was
testing various things, and found everything worked except outgoing calls.
So I checked
2003 Oct 06
1
chan_zap.c - echo cancelation getting in the way of dialing????
It seems consistant after dialing dozens of times that the call that
doesn't go through is the one the gets the log message "No
echocancellation requested" (chan_zap.c) and the "Scheduleing timer"
(channel.c) in the middle of receiving the DTMF tones.
I'm now using the T400P card last week very simular problems the the T100P
(although I think I was actually loosing
2006 Jun 26
2
1.2.9.1 SIP/Local/Queue behaviours weird
Hi,
Does any one experience that SIP phone to SIP phone (Polycom phone)
calls can't hear each other, but Monitor application records both end's
voices. It also happens in group pickup calls. Zap calls to queue (Local
channel) also experience this problem (sometimes, our SIP phone can't
hear any voice from incoming Zap calls when pickup, sometimes this
happens after 10-50
2005 Feb 08
1
Can only call VoIP SIP Providers (Weird)
I'm using Asterisk 1.0.4 with AMP and Broad Voice.
I have that with only 5 XTen Lite phones.
I'm able to call / etc with internal phones just fine.
I can call outside Vonage Numbers, and other
BroadVoice Numbers. I have vonage where I live (626)
and can call that fine. However, other 626 numbers I
get similar errors as below.
However, everytime, I try to call cell phones, and or
2006 Dec 04
2
Odd queue issue
Hi,
I have 2 systems (A and B). I have an 800 number... when someone
calls the 800 number it goes:
IAX2-->A---IAX---B--->SIP PHONE
However.. if the user calling the 800 number is a SIP user that is
registered to A it goes:
SIP--->A---IAX---B--->SIP PHONE
This is the problem... when a call comes in from the IAX2 800
provider, things work fine... however if a SIP user registered to
2006 Feb 02
0
Agents, queues and zombies
Hi all,
Have been experimenting with agents and queues instead of placing calls
direct to a user's phone extension, but I've run into problems with calls to
both the agent and the extension which creates a zombie and double records
calls abandoned etc. We're using a unique queue for each agent (only a
handful of users) to try and get some agent/queue information to see what
the
2007 Sep 18
2
asterisk crash and core dump
My Asterisk installation crashes frequently.
Since it's a random event I am not able to reproduce
it so I can't say what is making it crash.
Here's a snippet of /var/log/asterisk/full just when
it crashes:
Sep 18 13:42:51 DEBUG[378] chan_zap.c: disabled echo
cancellation on channel 31
Sep 18 13:42:51 VERBOSE[378] logger.c: -- Hungup
'Zap/31-1'
Sep 18 13:42:51
2007 Feb 23
1
ooh323 hang up after the call is answered
Hi,
I'm trying to make ooh323 works with one asterisk box running 1.2.15
version.
I can ring from a h.323 to SIP and SIP to H.323, but when the call is
finished when the phone is answered.
This is the log when I call from the H.323 device to a SIP device:
Feb 23 10:57:32 VERBOSE[6096] logger.c: -- Executing
Dial("OOH323/Telconet Mantaer-c5f8", "SIP/666|30|TtrwWC")