similar to: Asterisk dies with Meetme

https://bugzilla.samba.org/show_bug.cgi?id=8162 Summary: MKDIR Fail Product: rsync Version: 3.0.7 Platform: x86 OS/Version: Linux Status: NEW Severity: blocker Priority: P5 Component: core AssignedTo: wayned at samba.org ReportedBy: yannick at magikdo.com QAContact: rsync-qa at

Hello. I just wanted to post it here to receive feedback. I have downloaded both the GIMP and the GTK+ runtime as installers. They run correctly (I think) on both Windows and Linux, but when running The CIMP under Wine, the program just crashed showing the well known dialog box from Windows. Is this a bug in Wine? Before you post an inane reply, I already have GIMP for Linux (x86 Linux native

What happens when you do "stop now" like the error states? Sean -----Original Message----- From: Ryan Parlee [mailto:listbox@jesca.com] Sent: Saturday, April 03, 2004 9:56 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Unabled to exit console No matter what I try, Asterisk won't let me out of the console. If I CTRL+C, of course, the process will terminate. I

Hi, I'm trying to get zapras do work, I had downloaded the pppd-source and the 2 patches. I succefull compiled and install the patched version of pppd, but got this error in message-log Jul 15 11:43:32 voip1 pppd[9296]: In file /etc/ppp/filters: unrecognized option 'active-filter' Jul 15 11:43:57 voip1 pppd[9299]: Plugin zaptel.so loaded. Jul 15 11:43:57 voip1 pppd[9299]: Zaptel

I have an issue where * tries to route a call over IAX to another server even if the server is down. I have included the relevant entries from my iax.conf, extensions.conf, and some debug output. If someone could tell me what I have configured incorrectly, I would appreciate it. Thanks, Stephen -----------iax.conf on voip2---------- [voip1] type=friend username=voip1 host=x.x.x.x (ip

I'm looking to use * for a HQ/branch office topology with fairly few calls over the WAN. The questions I have all pertain to the following architectural pic: http://www.45891.com/misc/arch.jpg I'm looking at a distributed architecture so users are somewhat functional when the link to HQ is down, with a centralized voicemail server to allow for transfer of voicemail messages from user to

Hello, I'm french, so excuse my poor English. I'm face to a terrible thing, with has stole a lot of my time. On the .184 machine, I've the following iax.conf : [general] rtcachefriends=yes bandwidth=high tos=reliability jitterbuffer=no autokill=yes #include "iax.voip1.conf" #include "iax.renoir.conf" The iax.voip1.conf file contains : [VOIP1] type=friend

Hi, I just upgraded from 1.2 to 1.4. In 1.2, when I did a "show uptime" I used to see a second line telling me the time since the last reload. Has this been removed in 1.4? The following is the output of my two test boxes: Connected to Asterisk 1.4.18.1 currently running on voip2 (pid = 10605) Verbosity is at least 3 voip2*CLI> show uptime System uptime: 15 hours, 55 seconds

Hi All, I'm on a middle of an asterisk installation/configuration for my company and I'm testing the TLS configuration. For this reason, I used the ast_tls_cert script to build the ssl certificates for my server. On sip.conf file: tlsenable=yes tlsbindaddr=0.0.0.0 tlscertfile=/etc/asterisk/keys/asterisk.pem tlscafile=/etc/asterisk/keys/ca.crt tlscipher=ALL tlsclientmethod=tlsv1 and on

I have successfully configured two servers and I am now trying to connect via iax. When I attempt to call from one ext, 2006(server viop1) to extension 3006 (server voip2) I receive a timeout or "call failed 403 forbidden. The information I am receiving from the console is below. Apr 26 10:53:32 WARNING[311313]: channel.c:1745 ast_request: No channel type registered for 'IAX'

Hi list, something crazy here. 2 asterisk on 2 different place (1.4 and 1.8) both having an extension [115], one as type peer (caller side 1.4) and one as friend (callee side 1.8). Phones from both location connect to Asterisk from LAN. Router are Linux boxes. Connection between the 2 sites is done like this: On the callee side [115] ;callee type=friend host=dynamic secret=otherSecret

I have the following pair of macros defined to handle outbound calls from *. Rather than specifying full dialstrings in the main body of extensions.conf, outbound dial commands are made using a macro call as follows: Macro (outbound,number_to_dial,callerid_to_present,gateway1,gateway2,gateway3,gate way4) The final gateway defined is nearly always a fallback to PSTN if none of the IAX or SIP

I'm sorry, i send mail in asterisk-bug, but asterisk-users is better for my problem Hello, During a call with canreinvite = no, at the beginning of the call I lose 2 seconds of audio. is obvious when I call autoattendant. schema: SipPhone --> Centrex (asterisk 1.4.24.1) --> Voip1 (Asterisk 1.4.24.1) --> Operator SIP capture of voip1: - Executing [0825387205 at

I'm wondering if hdlc can be the culprit (not sure what it is and what it does). Should I set hdlc to yes in misdn.conf (I'm asking before testing because this is a production system)? misdn.conf: [general] misdn_init=/etc/misdn-init.conf debug=0 ntdebugflags=0 ntdebugfile=/var/log/misdn-nt.log ntkeepcalls=no bridging=no stop_tone_after_first_digit=yes append_digits2exten=yes

I am having trouble getting Asterisk to compile the zaptel stuff. Here are the specifics: Linux Kernel 2.5.9-42.0.8.EL Asterisk 1.4.0 I compiled libpri, zaptel, asterisk and asterisk-addons (in that order). This is a fresh install of CentOS. Following the CentOS install, I did "yum -y update" until there were no updates left. Here is my src directory: drwxr-xr-x 24 root root

O, try the following.   Test this first. ldd /usr/sbin/hfaxd  if you getting libpam.so..  something, then hylafax is compiled with pam support.   Next,   apt-get install libpam-ldap   ( just to be sure, i do believe you have installed it already )   create the file :  /etc/pam.d/hylafax Add :   auth         required       pam_ldap.so account   required       pam_ldap.so

Hey, all... If this is too off-topic, I'd be grateful for directions to a more appropriate mailing list. I'm trying to set up Asterisk and some Sipura boxes. I've got an SPA-3000 which is registering twice with Asterisk - once for its FXS/Line1/VoIP1 and once for its FXO/PSTN/VoIP2. My eventual goal is to have inbound calls on its FXO ring four times on its FXS and then fail over to

Hi I have an account with mynetphone (australia), which gives me two voip (sip) accounts, which i used to have connected to a spa9000. this is behind a firewall, so on the spa9000 I would listen on another port apart from 5060. so on the firewall 5060 would go to voip1 and 5061 to voip2. I moved to asterisk (+tdm410) and the machine was also the firewall and I had no problem - well atleast it

Hi, I have two Asterisk systems that hang once every 4-6 days (more or less). One has * 1.2.31.1 and the other 1.4.26.2. The last system collapsed today and I saw several messages looping endlessly on screen: hfcmulti_rx no memory for rx_skb alloc_stack_skb(303,110): no skb size During this time the server is completely down (no ping). A forced reboot is necessary (very ugly situation in

Hi, Sometimes my asterisk server stops. (after a day or two) Last output from CLI is: -------------------------------- -- Registered SIP '000b82017eb7' at 213.237.12.125 port 11620 expires 120 -- Channel 0/26, span 1 got hangup -- Hungup 'Zap/26-1' voip1*CLI> Disconnected from Asterisk server Executing last minute cleanups Asterisk cleanly ending (0).