similar to: PRI problem, pri_fixup_principle: Call specified, but not found?

Previous mail did not go through. Following up... Carlos G Mendioroz @ 16/09/2007 13:27 -0300 dixit: > Hi, > I'm seeing a problem using AgentCallbackLogin (Asterisk 1.2.16) where > a call in queue while an agent is logging in results in the agent > getting the call without properly being logged in. > > This seems to be a race, although I've not (yet) pinpointed the code

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

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

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

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

Hi, I've had a hard time understanding what was going on in a new * setup. The deployment has a * box running on dual xeon RH9 stock 2.4.20-8 and some different versions of asterisk (1.2.10/1.2.16) + libpri + zaptel + wanpipe. Short version: audio from iaxclient clients is fine from windows but poor from linux when going to zaptel. E.g. Iaxcomm running on windows works fine, but the same

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] --

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

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

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 =>

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:

What am I doing wrong here? Why is this happening? libpri is version 1.0.7-1 (debian package) asterisk is version 1.0.7.dfsg.1-2 (debian package) zaptel is version 1.0.9.2 -- Executing Dial("SIP/739-5935", "Zap/g1/0916000739") in new stack -- Called g1/0916000739 -- Channel 0/1, span 1 got hangup Oct 10 13:14:45 WARNING[7544]: app_dial.c:412 wait_for_answer:

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

I see in this thread, http://dovecot.org/list/dovecot/2006-April/012918.html, that others had simmilar problems with Mac OS client. Anyone got a working Dovecot + Mac OS setup? In particular with RC15. Our setup default_mail_env = maildir:Maildir namespace private { separator = . prefix = INBOX. inbox = yes } Would it help to use hidden = yes ?? Reading the description doesn't say

Hi, To solve echo problems, I'm considering 2 alternatives. 1> Sangoma A104d - I can't find support for asterisk 1.2.1 2> Desktop echo canceller - http://www.oriontelecom.com/echo_canceller/desktop/e1_ec_desktop.html - I want to know where to buy and price. Any suggestion is appreciated. Thanks. Jason. p.s. : asterisk cli command "reload" can change rx_gain and

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have a linux system with 4 ethernet interfaces, eth0 goes to the internet, eth1, eth2, and eth3 are NAT''d LANs. I want to use an ingress filter to prioritize bandwidth (downstream from internet) to various IPs. I want to sett it up something like this.... eth0 <--[NAT]--> dummy0 <---> dummy1 <---> eth1,eth2,eth3 dummy1

My SIP seems to be configured correctly as I can dial out and my minutes show up on my broadvoice bill, but whenever anyone calls my broadvoice # inbound they just get a busy signal. I dont get anything in the logs saying anything came in from broadvoice at all. Has anyone had this/simmilar problem with inbound from Broadvoice? Any suggestions? Thanks Neri -------------- next part

Hi all, I need your lights :) There are many hardware provider for E1 cards on the market, what's your exeperience with E1 and what's the preferred provider for Asterisk out of Digium? Olivier

Hi all, as good? I try to install asterisk-1.2.14, zaptel-1.2.12,libpri-1.2.4,addons-1.2.5 , sounds-1.2.1 and wanpipe-2.3.4-3 and hwec-utils-beta4-2.3.4 But it is not compiling drivers of the Sangoma, why udev's for board in "/dev/zap"(1-31, channel,ctl,pseudo,timer) is not created. But when I install a board TE110P Digium, udev's is created and asterisk functions perfectly. : )

Hi all, I have a Solaris 2.7 host with a recent install of samba-2.2.2. I am using an smb.conf from a simmilar host that works perfectly. I can't seem to connect to the host with Samba. I am getting: ########## //g4syd/usr/local/samba/bin 634 # smbclient -d 4 -L g4syd added interface ip=10.1.5.7 bcast=10.1.5.255 nmask=255.255.255.0 Client started (version 2.2.2). resolve_hosts: Attempting