similar to: chan_sip deadlocks after some time

Hello everybody, I've got Asterisk 1.2.4 running with two Junghanns QuadBRI cards using the qozap driver from bristuff 0.3.0-PRE-1l. One of the cards is running in TE mode, the other one in NT mode. Whenever I call into an empty MeetMe conference room on one of the NT ports or via SIP and hang up the call during the "you are currently the only participant in this conference"

I am running asterisk 1.2.27 and it dead today. The following is the backtrace of core file. Can anybody help me to identify what is the possible cause of crash? It seems the mysql connection causing problem in Thread 2. But I can not tell what exactly happened. This asterisk is using as ACD for over hundred agents. #> thread apply all bt ........ ........ Thread 6 (process 20135): #0

Hi, I've got a printing server running Samba 3.0alpha21 (on Linux). The client machines running Windows 2000 are producing PDF files by printing to a "virtual" printer on the Samba machine. The printer driver shall be installed from the Samba server on demand. I need color output, b/w is not sufficient. Everything works fine, i.e. the PDF conversion process and the automatic driver

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm trying to access a Samba 3.0 server (running on Debian unstable) in an Active Directory environment. I successfully joined the domain, klist shows my Kerberos ticket(s) and I can use smbclient -k to access a Windows 2000 server. However, when I try to access a share on the Samba machine from a Windows 2000 client, I'm being asked

Hello everybody, I'm using CUPS 1.1.19 and Samba 2.2.8a on SuSE Linux 9.0 (I'm using SuSE's pre-built packages). Samba is running as a domain controller and the workstations are running Windows XP Professional. I manually installed cups-samba 1.1.19 Windows printer drivers because SuSE doesn't supply them. I've got three printers on the network which I would like to print on

This was happening, *very* frequently, after the upgrade to CentOS 5.2 (32 bit). Mark (MHR) also had problems with this. Neither of us could report it with BugBuddy, which requests a newer version of GNOME. Now, after having installed 15-20 updates that came out after the big upgrade to 5.2, it is *very* rare for me to have Evolution Calendar crash when I exit Evolution. I do not use Evolution

I'm still getting duplicate process but the results of gdb are different. Can anyone shed any light onto what is causing this? (gdb) info threads 1 Thread 1091845040 (LWP 31287) 0xffffe410 in __kernel_vsyscall () (gdb) thread apply all bt Thread 1 (Thread 1091845040 (LWP 31287)): #0 0xffffe410 in __kernel_vsyscall () #1 0x4004f13e in __lll_mutex_lock_wait () from

G'day. I hope this isn't off-topic for the list. I am looking at an Asterisk setup that includes cordless phones. The three choices I can see, at this stage, are: * wifi phones * an ATA and a cordless analog phone * a DECT to SIP basestation The various wifi phone options don't grab us as suitable -- they are costly, have poor battery life and even the best have pretty mixed

The Asterisk Development Team would like to announce the release of Asterisk 13.16.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.16.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 14.5.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.5.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

Good day to all-- First off let me say that I have been very pleased with the mailing list. I have learned a ton of stuff just reading other peoples questions and comments. I really enjoyed the VOIP Conference call on Friday morning. Still working on figuring out the best approach to custom voicemail emails (the reason I joined this group); however, we have more pressing issues. I

Hello ! I 'm trying T.38 faxig with spandsp using rxfax/txfax as fax terminal. As another endpoint I 'm using Grandstream HT 486 ATA and a fax machine. Has anybody success with the HT486 as T.38 terminal ? ATA as originator: I managed only onetimes a successfull T.38 fax session. The other times the HT486 did not initiate a re-invite with T.38 parameters. Or shall the Terminator

This is the core trace (gdb) bt #0 0xb7e5a231 in strcasecmp () from /lib/libc.so.6 #1 0xb7ce0a3f in local_ast_moh_start (chan=0x82496a8, mclass=0xb720f828 "default", interpclass=0x0) at res_musiconhold.c:646 #2 0x08083695 in ast_moh_start (chan=0x64, mclass=0x64 <Address 0x64 out of bounds>, interpclass=0x88 <Address 0x88 out of bounds>) at channel.c:4614 #3

Greets, I'm using dovecot 2.0.9 (also tried rolling back to 2.0.8 which does the same - the output below is from 2.0.8), and this problem is imap specific. The following is a recurring problem with a specific user - as far as we can tell, other users are working just fine (there are thousands); but I bet there are others. In a nutshell: User authenticates OK, then the imap process hangs.

I've bought my 12 yr old son a used laptop (Dell Latitude D600) and convinced him to use a version of linux unstead of Windows. The laptop came with PCLinuxOS release 2007. He really wants to be able to run some of his games, so I used the Synaptic Package Manager to install Wine. The version available for PCLinuxOS was Wine 0.9.58. So far I've tried two games, and I've struck out

Hi All; Asterisk version is: 1.8.5.0 But I see at the consol the following warning and really I did google but did not understand if it is bug or related to settings: [Sep 13 15:04:56] WARNING[2209]: chan_sip.c:19667 handle_response_invite: just did sched_add waitid(3429468) for sip_reinvite_retry for dialog 3c581fa96f2b-53yysntgjmwb in handle_response_invite But actually, we see some SNOM IP

Hello, I need help for that error message: ?chan_sip.c:14889 handle_response_invite: Failed to authenticate on INVITE to? My network is: Client1-- -----------asterisk1------asterisk2 Client2-- ? With client1, I do a call ? Asterisk1 forward the call to asterisk2 ? Asterisk2 forward the call to asterisk1 ? Asterisk1 forward the call to

I have a situation here where a user has an AAstra 480i phone, which function corectly. The phone is behing a nat-router (a linksys wrv200 for it's VPN point to point facility). The phone is plugued in a port wich has qos enabled. And when the user places a call, sometimes (not always), we get this in the console : [May 8 13:41:55] WARNING[5804]: chan_sip.c:1948 retrans_pkt: Maximum

Hello All This or similar topics have already been mentioned but without any solution yet. I have built a oneway conference system for a client using one caller's input and broadcast it to all the other participants using app_meetme. E.g. one talker multiple listeners. Unfortunately some of the talkers (I have got multiple rooms) are not loud enough (e.g. use just half the amplitude, so

Hello, I?ve been using Asterisk 1.2.18 for a while, and today, with no apparent changes, I started receiving these messages: Aug 31 13:26:57 NOTICE[27528]: chan_sip.c:5495 sip_reg_timeout: -- Registration for 'user at sipserver' timed out, trying again (Attempt #19) All trunks and extensions went to: sipserver:5060 user 120 Request Sent 011