similar to: Page hangs up after 5 seconds

Hi all, -------- I have installed a TDM400 with one active FXS port (TDM10B) an connected it to a Siemens Euroset 2015 analogue phone. I have installed some smom IP phones to the network as well and configured them as usual (sip.conf). For configuring the TDM10B I have used FXO signalling in /etc/zaptel.conf and in /etc/asterisk/zapata.conf. I definded the TDM channel and the Snom phones to the

I am using the snom 200 with Phone type snom200-SIP Version snom200-SIP 2.04g Bootloader URL http://www.snom.com/download/snom200-boot1.9.bin Firmware URL http://www.snom.com/download/share/snom200-2.04o-SIP.bin I am using asterisk stable tree. I had to disable "Challenge Response on Phone" on my snom; I could not get it to work with

Check the Redirection on the web interface of your Snom 200. If it says "When Busy" that's your problem. It should say "Never". Also make sure on Sip->Lines your line appearance says "All" or else you will have the same problem. Hope this helps :) Greg Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original

I have a test SIP account set up with WorldCom and I have been trying to have Asterisk register to the WorldCom server with no luck. It appears that the SIP headers are different coming from Asterisk. I have included a packet capture from a successful login with a Windows Messenger client for reference. I have also copied in the SIP packet I captured with sip debug turned on. In my sip.conf file,

Samba 4.1.16, Centos 6.6 x86-64, BIND_DLZ 9.9. I have three AD DC's that were functioning normally. However, today I restarted BIND on one node, and it failed to start with this message in the log (names changed): May 10 07:02:49 benford named[6767]: Loading 'AD DNS Zone' using driver dlopen May 10 07:02:49 benford named[6767]: samba_dlz: started for DN DC=samdom,DC=example,DC=com May

Hello, i want to use asterisk as a gateway for H323-Phones. But i cant get it work. I'm using a gatekeeper on another computer. My IP-phone is registered there. Does anybody can sent me an oh323.conf and extension.conf as examples? Thanks in advance Erik Bastian -- NEU : GMX Internet.FreeDSL Ab sofort DSL-Tarif ohne Grundgebühr: http://www.gmx.net/dsl

Ubuntu 8.04LTS Dovecot 1.2.6 Postfix 2.5.1 I'm using dovecot imap/deliver in an environment where I don't want users with shell access to reach their mail folders (partly to do with IMAP ACLs requiring open group permissions and partly to do with not wanting users to mess their mail folders / sieve configs up). I specifically don't want to switch to a "single dovecot

Hi! For demonstration purposes I try to bridge an incoming video call from a 3G mobile handset to another 3G mobile handset using asterisk as "switch". On the incoming call leg I see all expected bearer capabilities (Digital, 64k Transparent, G.7xx 384k video) but on the outgoing call leg the bearer capability G.7xx 384k video get lost and therefore the call is dropped from the mobile

Hi, I recently updated from puppet 3.1.1 to 3.2.1 and noticed quite a bit of increased load on the puppetmaster machine. I''m using the Apache/passenger/rack way of puppetmastering. Main symptom is: higher load on puppetmaster machine (8 cores): - 3.1.1: around 4 - 3.2.1: around 9-10 Any idea why there''s more load on the machine with 3.2.1? -- You received this

I've tried 2 builds of dovecot, based on a amd64 rebuild of the experimental package on http://xi.rename-it.nl/debian/ I've tried to include as much useful info as possible, sorry if I've missed out anything of use. Let me know as I'm more then happy to assist in any way possible. $ telnet host 110 Escape character is '^]'. +OK Dovecot ready. USER ian at domain.com +OK

Hi, Our operator frequently gets requests to transfer a call directly to voicemail in order for the caller to leave a message without disturbing the callee. Basicly, I'm looking for a blindxfer to vm. My first thought was to prepend a digit (eg 7) to the extension but this does not fit well with our dialplan. According to an article on voip-info.org Asterisk@Home appears to implement

SNIP > > > Facing an issue with winbind not starting with below error log(taken > from /var/log/syslog): > > Jun 14 15:48:33 enpaq winbindd[15941]: [2010/06/14 15:48:33, 0] > param/loadparm.c:6767(service_ok) > Jun 14 15:48:33 enpaq winbindd[15941]: WARNING: No path in service > printers - making it unavailable! > Jun 14 15:48:33 enpaq winbindd[15942]: [2010/06/14

Hello Everyone, I've been playing with an agi script for paging sip phones. page.agi will take all available sip extensions and assign them to the global variable PAGE_GROUP. Allowing the phones to be paged from the dialplan with the new Page cmd. Extensions to be excluded are presented as arguments to the agi. Each time a page is made this agi refreshes the global variable. This works with

Hi All, I am in desperate need of this feature. I want to play background music during a call while the 2 parties are having some lovely conversation (or maybe give them a sort of cursing background if they are cursing each other). I found this post which talks about creating a ghost call with the help of queues and putting that queue in a meetme room where queue will play the song/curse and the

Grandstream GXP-2000 with firmware 1.1.4.18 (beta) fixes an issue where the phone did not send rtp keepalives when on mute (resulting in disconnect from tech support hold and concalls) A side effect seems to be that Asterisk pops the following warning on the console... Jul 26 14:06:35 WARNING[31654]: rtp.c:463 ast_rtp_read: RTP Read too short Grandstream say they are not sure what it is but

hi, i write this looking for free conference room, i checl code and don?t see any error but die at priority 7 if room 1001 have users in exten => _1NXXNXXXXXX,1,RouteCall(${EXTEN}) exten => _1NXXNXXXXXX,2,GotoIf($[${DESTINATION1:0:3} = CONF]?3:13) exten => _1NXXNXXXXXX,3,Setvar,var=0 exten => _1NXXNXXXXXX,4,MeetMeCount(1001|var) exten => _1NXXNXXXXXX,5,GotoIf($[${var} =0]?7:6)

On 10/05/15 12:18, Steve Thompson wrote: > Samba 4.1.16, Centos 6.6 x86-64, BIND_DLZ 9.9. I have three AD DC's that > were functioning normally. However, today I restarted BIND on one node, > and it failed to start with this message in the log (names changed): > > May 10 07:02:49 benford named[6767]: Loading 'AD DNS Zone' using > driver dlopen > May 10 07:02:49

Hi list, yesterday I converted my voicemail.conf to realtime voicemail and also configured to store the voicemessages in a database using odbc as described here <http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail> and here <http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage>. I am using asterisk 1.4.2 with mysql. I also installed the proper odbc driver for

Le 19/11/2018 à 12:33, Julien TEHERY via samba a écrit : > Le 19/11/2018 à 11:14, Marco Gaiarin via samba a écrit : >> Mandi! Julien TEHERY via samba >>    In chel di` si favelave... >> >>> Is there a good pratice when adding new remote DCs in terms of >>> replication >>> topology? >> I think you have to define a topology of the domain, using

I have some phones (and an ATA) that are shared between two users who each have separate voicemail but they are not behaving as desired nor expected. Incoming calls show up on the correct lines. Calls originating from the device are seen, at the terminating device, as coming from the account listed last in sip.conf, regardless of the line selected. This creates three main issues I would like