similar to: Iax bug ?

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

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

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

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

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

Has anyone seen this before? This channel is consistently present on both of my asterisk servers. Sometimes they disappear for a few seconds and then come back. It always has the same Call ID. voip1*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 192.168.0.102 (None) df92fb1b-8a 00101/03059 00000ms 0000ms UNKN

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).

I constantly get the following error messages in /var/log/asterisk/messages: Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 3324 (iax_ack_registry): Received unsolicited registry ack from '192.168.0.1' Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 4181 (socket_read): Registration failure Where 192.168.0.1 is another asterisk server. Below are the local and

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 Hi, I posted this also on hylafax list - maybe here is someone with a hint. System: Debian Jessie, Hylafax-Server 6.0.6, pam 1.1.8, libpam-ldapd 0.9.4, nslcd 0.9.4 (all actual debian packets from stable), sernet-samba-*-4.2.7-8 After a switch from OpenLDAP to a Samba 4.2 based LDAP Server, I cannot auth users anymore in Hylafax, everything else

Hi all, Today I got problem below and my domU become unresponsive and I should restart the pc to make it running properly again. [ 240.172092] INFO: task kworker/u:0:5 blocked for more than 120 seconds. [ 240.172110] "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. [ 240.172376] INFO: task jbd2/xvda1-8:153 blocked for more than 120 seconds. [ 240.172388]

My VoIP provider allows me to have more than one call outbound on the same line simultaneously, for some reason. I am pretty sure that they do not want this to happen, so I'd like instead to limit each line to one call. I do not want the users to have to dial another prefix to go out on another line. Is there any way to add multiple accounts for my _9. extension and have Asterisk

(Server :Mandrake 7, Samba 2.0.6) (Client : PC Win95 and Win 98) Samba's work fine since one year but yesterday We had a big problem in my college Lots of PC's client can't connect (invalid password). In the samba's log we read : > [2000/05/19 12:15:21, 1] smbd/service.c:close_cnum(568) > renoir (192.168.84.189) closed connection to service valancee > [2000/05/19

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 libdrm 2.4.102, lots of FreeBSD and modetest stuff. Boram Park (1): drm mode : fix memory leak when freeing drmModePropertyPtr Dave Airlie (1): Bump version to 2.4.102 Emil Velikov (17): tests: install drmdevice modetest: simplify "dump all" logic modetest: remove cursor/page_flipping_supported stubs

Hi, My initial setup for receiving faxes worked as follows: fax call arrives on ISDN BRI connected to a BOSCH PBX, signal sent to ALCATEL PBX via PRI QSIG then finally sent to ASTERISK via PRI EUROISDN. The Asterisk server then forwarded the call to a iaxmodem and HylaFax received the data. All worked fine. Now I got rid of both BOSCH and ALCATEL in the "fax path" and it's as

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi List I'm trying to create a conference room using H323 channels. If i start asterisk normally (service asterisk restart) and connect to cli using -vvvvvvvr options, when a user enters the Conference, asterisk says "You are the only ..." and then dies, withou any error message, nothing at all. But, if i start asterisk with cli

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

Adjust indentation from spaces to tab (+optional two spaces) as in coding style with command like: $ sed -e 's/^ /\t/' -i */Kconfig Signed-off-by: Krzysztof Kozlowski <krzk at kernel.org> --- drivers/gpu/drm/Kconfig | 10 +- drivers/gpu/drm/amd/display/Kconfig | 20 ++-- drivers/gpu/drm/bridge/Kconfig | 8 +-

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'