similar to: WARNING[11439]

Hello! I clear remarks in Makefile: DEBUG_THREADS = -DDEBUG_THREADS -DDETECT_DEADLOCKS But same things in CLI: Aug 30 18:16:31 WARNING[22292]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0x82f2fe0', 9 retries! -- Zap/32-1 is proceeding passing it to Zap/31-1 -- Zap/32-1 is ringing -- Accepting call from '2177' to '7141278' on channel

Hi All!! Prompt how to record all calls passing through certain span? --- Thanks...

Greetings to All! Help to solve a problem: There is an asterisk and two VoIP a sluice (NSGate 800 2FXS 2FXO, NSGate 800 2FXS). In sip.conf they are registered so: [3301] type=friend host=172.222.135.11 username=3301 secret=0000 defaultip=172.222.135.11 dtmfmode=rfc2833 context=it callerid="VoIPGate2Line1" <3301> allow=g723.1 [3302] type=friend host=172.222.135.11 username=3302

Anyone have any idea what this is talking about. Here is my zapata.conf [channels] switchtype=5ess signalling=pri_cpe echocancel=yes echocancelwhenbridged=yes echotraining=400 callerid=asreceived group=1 context=default musiconhold=default faxdetect=incoming channel => 1-23 Here is my zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 # set this to 1-15,17-31 for E1 dchan=24 # set this to 16 for

Hello, I can't figure out how to use the upper.panel and lower.panel options in pairs to alter the label options for either panel independently of the other. I would like to be able to show the pairwise scatter plots for the data as they are (a vanilla pairs plot?) but separately to be able to label the points according to a factor level. It is easy enough to do this independently, but I

I wrapped the LDA command in a script. I can see that Postfix passes "@@mydomain.tld" as the -d argument, without quotes. I then adapted the script to specifically replace this address with "@"@mydomain.tld, but this results in the following error message by Dovecot: ??? auth: Info: userdb(?): Username character disallowed by auth_username_chars: 0x22 (username:

I upgraded to Dovecot 2.3.0.1 as advised, but it still seems broken. In the Postfix log I now see: Mar? 6 13:49:03 myhost amavis[7165]: (07165-10) K00VtLRHdrYw FWD from <admin at mydomain.tld> -> <"@"@mydomain.tld>, BODY=7BIT 250 2.0.0 from MTA(smtp:[127.0.0.1]:10025): 250 2.0.0 Ok: queued as B8CA22DA1B37 Mar? 6 13:49:03 myhost amavis[7165]: (07165-10) Passed CLEAN

Op 7-3-2018 om 15:40 schreef Stephan Bosch: > > > Op 7-3-2018 om 13:46 schreef Philipp Berger: >> I wrapped the LDA command in a script. I can see that Postfix passes >> "@@mydomain.tld" as the -d argument, without quotes. >> I then adapted the script to specifically replace this address with >> "@"@mydomain.tld, but this results in the

Here is the situation. There are six different soccer games, developed by Konami, all straight ports from Playstation 2. These are the following: Pro Evolution Soccer 4 (http://appdb.winehq.org/objectManager.php?sClass=application&iId=2750) (PES4) Winning Eleven 8 (http://appdb.winehq.org/objectManager.php?sClass=application&iId=5152) (american version of PES4) Pro Evolution Soccer 5

Dear all, I have a working setup with Postfix + Dovecot, storing users in a MySQL table. I ran into problems setting up and using a mail address like a"@"b at mydomain.tld, which by RFC should be valid, but leads to problems in Dovecot. From my debugging, I can see that on the lookup, Dovecot replaces %n with "a" (quotes added by me) and %d with "b at mydomain.tld"

The Asterisk development team has released Asterisk version 1.2.21.1 and 1.4.7.1. These releases are minor updates to the releases that were made yesterday to fix a couple of introduced issues. One issue was related to the ODBC realtime driver. Another was related to music on hold. Thank you for your support!

The Asterisk development team has released Asterisk version 1.2.21.1 and 1.4.7.1. These releases are minor updates to the releases that were made yesterday to fix a couple of introduced issues. One issue was related to the ODBC realtime driver. Another was related to music on hold. Thank you for your support!

Yesterday I upgraded from asterisk 1.2.21.1 to 1.2.22. We are running Zaptel-1.2.17.1. After upgrading all of our calls from the phone company (DIDs trunks with wink start on a channelized T1) were not coming in. Looking in the log file '/var/log/messages'. I saw the following error message. pbx.c: Cannot find extension '' in context '(null)' This was confusing since I

If I let modules.conf autoload chan_h323.so then when I try to stop asterisk, it *does* stop (files in /var/run/asterisk/ are removed and connection via -vr from another console is not possible) but the "asterisk process" stays alive and stalled. In other words, a 'ps -ae | grep asterisk' show that the process is there after running 'stop now'. I either need to press

Hi all, i am having a major asterisk problem. I think it started around 1.2.19 but could also be 1.2.18 zaptel or 6.5.10 snom 360. basically we start getting busy signals, all our 4 line hunt group is busy, i then check the channels and there are open calls that were hung up long ago. i thought it was a zap problem but then i saw the same problem with iax2 calls. its becoming a huge issue

Hello All, I would like to upgrade my recently installed Asterisk 1.2.21.1 to Asterisk 1.4.8? My OS is CentOS 4.5 with Linux 2.6.9-55.0.2.plus.c4smp #1 SMP Fri Jul 6 05:25:07 EDT 2007 i686 i686 i386 GNU/Linux Is there any detail step by step procedure to uninstall the current version and install Asterisk 1.4.8, Zaptel 1.4.4, Libpri 1.4.1, Addons 1.4.2? Cheers, Nitesh

I am trying to set up a Grandstream GXV-3000 Video phone to Asterisk ver 1.2.21.1. The problem I'm having is that it can call other SIP phones, but not vice versa. Can someone tell me where is the problem? TIA! Here's part of my configurations: ---------- sip.conf ---------- ; 113 is the Grandstream phone [113] type=friend username=113 secret=secret context=default dtmfmode = rfc2833

I have one Digium adapter S101I on a local network and I'm losing the connection periodically. I'm using Asterisk-1.2.21.1 There is no pattern, sometimes it stays connected for a weak sometimes longer. In comparison I have few Sipura adapters and they stay connected for months (never lost single connection). -- #Joseph

Hi List, I have a client who has a TDM400P with 4 FXO. He has a problem them when some one calls, then hangs up it takes a good 10-15 seconds or more of the card to realize that the line was hung up on. The phones keep reigning After a bit it hangs up on the line. Also there has been some hanging. (After a user on the PBX side hangs up the card does not "release the line"). I am using

Greetings list, I've just been upgrading one of our servers from 1.2.17 to 1.2.21.1-r1, and noticed that it's not picking up any of my macros written in AEL. Upon further examination, it looks like pbx_ael is missing. Is this a deliberate change, or is this something I need to address in the pre-compile configuration? Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For