Displaying 20 results from an estimated 400 matches similar to: "[Fwd: Libpri/chan_zap problems?]"
2010 Feb 25
1
Getting: Can't fix up channel from 5 to 7 because 7 is already in use, and pri_dchannel: Answer requested on channel 0/7 not in use on span 1
System have been working great for weeks, using an average 40 of 120
dahdi channels.
But today, I suddenly see scary things like this:
-- Moving call from channel 5 to channel 7
[Feb 25 10:18:12] WARNING[17129]: chan_dahdi.c:10608
pri_fixup_principle: Can't fix up channel from 5 to 7 because 7 is
already in use
[Feb 25 10:18:12] WARNING[17129]: chan_dahdi.c:11535 pri_dchannel:
Ringing
2005 Sep 07
0
Problem with PRI channels, restarted after every call.
Hi,
I got a problem with PRI that I'm not sure how to solve.
Asterisk sits between PABX and PRI.
PRI is span 1 and PABX is span 2.
After every single call (no matter in what direction) I get
"pri_fixup_principle: Call specified, but not found?" and "pri_dchannel:
Hangup on bad channel" messages and the channel in question is
restarted. As far as I can see, all
2007 May 24
1
PRI problem, pri_fixup_principle: Call specified, but not found?
Hi,
in a PRI setup, the receiving side is changing the B channel at
proceeding. It seems this sometimes breaks some logic
(pri_fixup_principle) and then the hangup kind of breaks, release is not
answered and a restart cycle is triggered (by remote side).
Anyone can help me debug this ? I've seen many posts with simmilar
issues but no answer/solution.
This is happening on Asterisk 1.2.16 +
2007 Jun 12
0
Warning on CLI
Hello everybody again.
I have a warning message in the CLI:
*CLI> Jun 12 17:34:29 WARNING[10593]: chan_zap.c:8463 pri_fixup_principle:
Call specified, but not found?
*CLI> Jun 12 17:34:29 WARNING[10593]: chan_zap.c:8463 pri_fixup_principle:
Call specified, but not found
I don't know what it means.
Can you help with this???
Thankyou very much.
Bye bye...
-------------- next part
2013 Oct 31
2
issue with dahdi_channels.conf
Hello list
i have an issue with my dahdi_channels.conf
in span 1 when i use it like below i can do my outband calls without issue
; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
group=0,11
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 17-31
context = default
group = 63
but when i add in channel 1-15 like: channel => 1-15,17-31
i receive all
2006 Mar 24
1
PRI Behavior
Just throwing out this question. integrating with Altiware server. PRI
appears to be okay. It keeps trying to move my call to a different
channel...usually channel 1. This is the deal here:
Moving call from channel 23 to channel 1
Then the following errors after no audio then hanging up manually:
Mar 24 17:46:17 WARNING[1315]: chan_zap.c:7792 pri_fixup_principle: Call
specified, but not
2008 Oct 06
2
Conneting Asterisk to Swyx pri
Hi all, I've done this a few times with other PBX's but swyx has stumped me!
I'm having some trouble getting Asterisk connected to a Swyx system using a
sangoma A104dx... currently the setup is:
BT <-> Swyx
The above setup works fine... what i'm trying to achieve is
BT & SIP Trunks <-> Asterisk <-> Swyx
I have connected to our BT (2 x ISDN30 UK) with
2010 Jan 25
1
Disa not fully bridging outbound call
Hello,
I have a situation where a remote worker dials in to the asterisk server, enters
the "secret code", then dials out via Disa on a PRI. This was all working great
until this morning. Now the calls is placed out, connected but there is no
voice from/to either phone. This is what shows on the CLI when the calls is
bridged at a verbose of 4 and a debug of 1:
[Jan 25 17:51:40] --
2013 Oct 21
1
issue after install dahdi
i need your help regarding some issue related to the outband calls
i have installed asterisk 1.4.32 with dahdi and i have 1 card diguim with 2
ports
when i try to call my phone number all time i receive message busy number
this error just with g1.
with g2 there is no problem i can call without issue
can anyone see the CLI and tell me what is the problem
thanks and regards
== Parsing
2005 Mar 04
0
Asterisk ---Toshiba
I set up a TE405P to go T1---*---Toshiba.
I have the channels configured, and can place calls from the Toshiba,
through * to the t1. Incoming calls work great to *, but if they go to
the Toshiba, I get a hangup. I think the * is sending the call to the
wrong span. I have 2 spans, span 1 from the T1, span 2 to the Toshiba.
The bchannels show as 0/1 through 0/23 on both spans in * when it
starts.
2006 Apr 06
0
Dial out on Zap
Hi,
I'm trying to test my dial out function so I did something like this in
extensions.conf
exten => 999,1,Dial(Zap/g1/02601591)
exten => 999,102,Congestion()
My Zapata.conf looks something like this
[channels]
context=from-pstn
group=0
switchtype=euroisdn
overlapdial=yes
faxdetect=no
; PRI port 1 (E1)
; context=1
group=1
signalling=pri_cpe
channel=>1-15,17-31
I am able to
2006 Apr 06
0
AW: Dial out on Zap
Hi,
i was able to fix this problem when i added the line pridialplan=local in the zapata.conf but it depends on your telco, i think.
marcus
-----Urspr?ngliche Nachricht-----
Von: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]
Gesendet: Donnerstag, 6. April 2006 11:50
An: asterisk-users@lists.digium.com
Betreff: [Asterisk-Users] Dial out on Zap
Hi,
2006 May 25
0
PRI Moving channels?
Hey Folks....I am on the 1.2 branch with the latest from Subversion.
I've been having a rough go for the last several months integrating
asterisk with out Altigen system.
I can get calls inward just fine. I have zero missed interrupts on the
digium 110p card. I have zero frame slips according to both sides.
Outgoing calls sometimes work, but more often than not I get the
following:
--
2009 Sep 14
0
DAHDI Dial 9 Receiving Setup Acknowledge
I have a Toshiba PBX connected via a QSIG PRI to Asterisk. I can make
calls from the Toshiba to Asterisk and internal calls from Asterisk to
the Toshiba. What I can't do is make an call with an outside
destination from Asterisk to the Toshiba. The Toshiba is looking for 9
to grab an outside line then it expects to see the 10 digits. In the
FreePBX dial plan I use 9|. which sends 9 plus the 10
2006 Apr 11
2
Dial out on Zap: Can't fix up channel from 31 to 30 because 30 is already in use??
Hi,
I still cant dial out on Zap and I really have no clue why.
I'm using Asterisk Asterisk 1.2.6 and Zaptel 1.2.5 with Digium card 4
ports, 31 channels each and able to receive incoming calls and fax
perfectly.
I've done this in my dial plan.
exten => 111,1,Answer()
exten => 111,n,Ringing()
exten => 111,n,Wait(2)
exten => 111,n,AbsoluteTimeout(30)
exten =>
2006 Apr 11
1
AW: Dial out on Zap: Can't fix up channel from 31 to 30 because 30 is already in use??
I'm not sure if it's the same problem but your error message likely the same.
after i additing pridialplan=local in the zapata.conf i'm able to make outboundcalls (located in germany)
marcus
-----Urspr?ngliche Nachricht-----
Von: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]
Gesendet: Dienstag, 11. April 2006 16:33
An:
2010 Feb 25
0
Intermittent DAHDI issue with a PRI line causing asterisk to crash!
Hello all,
I've got an intermittent issue with my asterisk set-up, and I'm pulling
my hair out!
Mostly everything works fine, but I get an error once every few days
that sometimes (but not always) causes asterisk to stop accepting new
calls and also stop responding to commands on the console (asterisk -r).
The usual errors look something like this:
[Feb 19 13:09:52] ERROR[18728]
2006 May 05
2
AW: AW: DTMF detection when outgoing call tomobilephones
Actually I am using Asterisk 1.2.7.1, zaptel 1.2.5, libpri 1.2.2
I ve tried many values for rx/txgain togeher with echocancel and relaxdtmf.
The detection is not working with call file, manager originate and not with the dial command to the mobile.
I have no ideas left.
I got it sometimes to work if I use a specific channel (i.e. Dial(ZAP/14/...)
But with the same vaules on a second call there
2003 Jun 16
4
POP daemon
What would be a good POP daemon to use? I know there are a few in the
mail ports. Are they any good?
What I mean by good is 'secure as possible' (is there really such thing as
being totally secure / invulnerable?)
Cheers
2006 Mar 31
1
Asterisk, QSIG and Tenovis PBX?
Hi,
we are still trying to properly connect a Tenovis PBX to an Asterisk server
(asterisk 1.2.6, libpri 1.2.2, zaptel 1.2.5, Digium Wildcard TE110P), this
time with QSIG.
Calling from a Tenovis phone to a SIP phone (i.e. traditional phone ->
Tenovis PBX -> QSIG -> Asterisk -> SIP phone) works with the following
messages:
---
Don't know what to do if second ROSE component is of