In reply to Auto Answer paging on a Cisco 7945:
Did you referred to the user guide. i once skimmed through it and saw auto answer feature on some page. if you haven't checked it yet, you can take a look on it at http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7965g_7945g/...
In reply to Polycom 331's w/ multiple SIP registrations - MWI problems:
Because the Polycom phones's MWI (Message Waiting Indicator) light gets confused when there are two SIP registrations on the phone (the main server and the fallback server), you need to make the following changes to the fallback server for EACH extension. This will disable the "unsolicited notify" sent from Asterisk.
Use "Config File Editor" to edit /etc/asterisk/sip_custom_post.conf and add the following two lines per extension.
[5601](+)
subscribemwi=yes
- then click "Re-Read Configs" or do amportal restart
dave
In reply to Keep on receving inbound...:
I have set the canreinvite=no, the incoming goes into the right context, I wonder why it sends the re-invite and my ivr starts running again
In reply to CBEYOND with asterisk 1.6:
This is trunk configuration:
Trunk Name: cbeyond
CID option block foreign CIDs
username=main number
type=peer
srvlookup=no
secret=
qualify=yes
;outboundproxy=sip-roxy.chi0.cbeyond.net
insecure=port,invite
dtmfmode=inband
dtmf=auto
host=sipconnect.chi0.cbeyond.net
fromdomain=sipconnect.chi0.cbeyond.net
disallow=all
context=from-pstn
canreinvite=yes
allow=ulaw
user context main DID
user details:
type=peer
secret=
qualify=yes
dtmfmode=inband
dtmf=auto
disallow=all
allow=ulaw
register string:
main number:password@cbeyond/main number
INBOUND ROUTE: very simple
DID NUMBER: 10 digits that belongs to us
SET destination whatever ....
If we called other number than main it is still redirected to main IVR as main number because of that:
-- Executing [MAINNUMBER@from-pstn:1] Set("SIP/cbeyond-0000001a", "__FROM_DID=MAINNUMBER") in new stack - if we used B number it should be:
-- Executing [MAINNUMBER@from-pstn:1] Set("SIP/cbeyond-0000001a", "__FROM_DID=B number") in new stack but ALWAYS shows MAIN NUMBER
In reply to TrixBox CE to Cisco IOS/CUBE to SIP Provider:
well, this is really a nice post.I really like the way you start and conclude your thoughts. Thank you so much for this information. keep posting such good stuff.spss research
In reply to DID Routing:
well, this is really a nice post.I really like the way you start and conclude your thoughts. Thank you so much for this information. keep posting such good stuff.spss research
In reply to Cant call extension to extension after upgrade:
I am calm, I have tried nicely to ask you for information yet you still insist on posting one sentence questions that are completely out of context and can't be responded to.
Not that many people are left in this forum but did you not notice you didn't get any answers?
Still no clue what system you are running or what you did.
In reply to Registrations Timing out, Peers unreachable... I'm going to lose my mind!:
so this problem has no solution ??
"pbx.c: FONALITY: This thread has already held the conlock, skip locking"
In reply to Cannot log into Trixbox by http interface:
I installed trixbox on vmware with network card bridged. then I changed its ip by this command:
"system-config-network"
then I tried to ping it from my win xp pro its working I mean replying.
but when I entered the its ip in my browser it still waiting waiting until connection was reset error raised
what I have to do ?
waiting for ur help
thanks
In reply to Sip extensions becoming unreachable:
I have the same issue now .. my TB was great then suddenly 10 days ago everything started
a hacking trial
fail2ban
everything ok ...
then SIP phones and trunks become unreachable from time to time .. it goes down then fixed by it own!!!
my Log file states that:
[Jan 10 16:00:11] ERROR[4321] /usr/src/redhat/BUILD/asterisk16-1.6.0.26/include/asterisk/lock.h: chan_sip.c line 20825 (restart_monitor): Deadlock? waited 30 sec for mutex '&monlock'?
[Jan 10 16:00:11] ERROR[4321] /usr/src/redhat/BUILD/asterisk16-1.6.0.26/include/asterisk/lock.h: chan_sip.c line 20806 (do_monitor): '&monlock' was locked here.
[Jan 10 16:00:12] ERROR[4342] /usr/src/redhat/BUILD/asterisk16-1.6.0.26/include/asterisk/lock.h: chan_sip.c line 20825 (restart_monitor): Deadlock? waited 5 sec for mutex '&monlock'?
[Jan 10 16:00:12] ERROR[4342] /usr/src/redhat/BUILD/asterisk16-1.6.0.26/include/asterisk/lock.h: chan_sip.c line 20806 (do_monitor): '&monlock' was locked here.
[Jan 10 16:00:12] ERROR[4336] /usr/src/redhat/BUILD/asterisk16-1.6.0.26/include/asterisk/lock.h: chan_sip.c line 20825 (restart_monitor): Deadlock? waited 10 sec for mutex '&monlock'?
[Jan 10 16:00:12] ERROR[4336] /usr/src/redhat/BUILD/asterisk16-1.6.0.26/include/asterisk/lock.h: chan_sip.c line 20806 (do_monitor): '&monlock' was locked here.
[Jan 10 16:00:16] ERROR[4321] /usr/src/redhat/BUILD/asterisk16-1.6.0.26/include/asterisk/lock.h: chan_sip.c line 20825 (restart_monitor): Deadlock? waited 35 sec for mutex '&monlock'?
[Jan 10 16:00:16] ERROR[4321] /usr/src/redhat/BUILD/asterisk16-1.6.0.26/include/asterisk/lock.h: chan_sip.c line 20806 (do_monitor): '&monlock' was locked here.
[Jan 10 16:00:16] NOTICE[3131] chan_sip.c: Peer '2226' is now UNREACHABLE! Last qualify: 14
[Jan 10 16:00:16] NOTICE[3131] chan_sip.c: Peer '2111' is now UNREACHABLE! Last qualify: 2891
[Jan 10 16:00:16] NOTICE[3131] chan_sip.c: Peer '2151' is now UNREACHABLE! Last qualify: 9
[Jan 10 16:00:16] NOTICE[3131] chan_sip.c: Peer '2223' is now UNREACHABLE! Last qualify: 16
[Jan 10 16:00:16] NOTICE[3131] chan_sip.c: Peer '2124' is now UNREACHABLE! Last qualify: 9
[Jan 10 16:00:16] NOTICE[3131] chan_sip.c: Peer '2258' is now UNREACHABLE! Last qualify: 8
[Jan 10 16:00:16] NOTICE[3131] chan_sip.c: Peer '2195' is now UNREACHABLE! Last qualify: 12
.
.
.
.
then
[Jan 10 16:00:16] NOTICE[3131] chan_sip.c: Peer '2136' is now UNREACHABLE! Last qualify: 12
[Jan 10 16:00:16] NOTICE[3131] chan_sip.c: Peer '2219' is now UNREACHABLE! Last qualify: 196
[Jan 10 16:00:16] NOTICE[3131] chan_sip.c: Peer '2202' is now UNREACHABLE! Last qualify: 227
[Jan 10 16:00:17] ERROR[4342] /usr/src/redhat/BUILD/asterisk16-1.6.0.26/include/asterisk/lock.h: chan_sip.c line 20825 (restart_monitor): Deadlock? waited 10 sec for mutex '&monlock'?
[Jan 10 16:00:17] ERROR[4342] /usr/src/redhat/BUILD/asterisk16-1.6.0.26/include/asterisk/lock.h: chan_sip.c line 20806 (do_monitor): '&monlock' was locked here.
[Jan 10 16:00:17] ERROR[4336] /usr/src/redhat/BUILD/asterisk16-1.6.0.26/include/asterisk/lock.h: chan_sip.c line 20825 (restart_monitor): Deadlock? waited 15 sec for mutex '&monlock'?
[Jan 10 16:00:17] ERROR[4336] /usr/src/redhat/BUILD/asterisk16-1.6.0.26/include/asterisk/lock.h: chan_sip.c line 20806 (do_monitor): '&monlock' was locked here.
[Jan 10 16:00:17] NOTICE[3131] chan_sip.c: Peer '2133' is now UNREACHABLE! Last qualify: 10
[Jan 10 16:00:17] NOTICE[3131] chan_sip.c: -- Registration for '8959264693@ippbx.net2phone.com' timed out, trying again (Attempt #1)
[Jan 10 16:00:17] DEBUG[3112] pbx.c: FONALITY: This thread has already held the conlock, skip locking
[Jan 10 16:00:17] DEBUG[3112] pbx.c: FONALITY: This thread has already held the conlock, skip locking
[Jan 10 16:00:17] DEBUG[3112] pbx.c: FONALITY: This thread has already held the conlock, skip locking
[Jan 10 16:00:17] DEBUG[3112] pbx.c: FONALITY: This thread has already held the conlock, skip locking
[Jan 10 16:00:17] DEBUG[3112] pbx.c: FONALITY: This thread has already held the conlock, skip locking
[Jan 10 16:00:17] DEBUG[3112] pbx.c: FONALITY: This thread has already held the conlock, skip locking
[Jan 10 16:00:17] DEBUG[3112] pbx.c: FONALITY: This thread has already held the conlock, skip locking
[Jan 10 16:00:17] DEBUG[3112] pbx.c: FONALITY: This thread has already held the conlock, skip locking
[Jan 10 16:00:17] DEBUG[3112] pbx.c: FONALITY: This thread has already held the conlock, skip locking
[Jan 10 16:00:17] DEBUG[3112] pbx.c: FONALITY: This thread has already held the conlock, skip locking
any idea guys ?
it was great then suddenly all this shit happened
?!!!
In reply to I want to replace Trixbox:
In my opinion there are only two:
FreePBX Distro
PBX In a Flash
In reply to Speaking clock sayings hour and minutes then hangs up:
I found solution.
In file
/var/www/html/admin/modules/infoservices/functions.inc.php
Line 120 should be
$ext->add($id, $c, '', new ext_sayunixtime('${FutureTime},,IM \\\'and\' S \\\'seconds\' p'));
Line 122 should be
$ext->add($id, $c, 'hr24format', new ext_sayunixtime('${FutureTime},,kM \\\'and\' S \\\'seconds\''));
(Remove double back slash after "and" and "seconds")
After that click on any "Submit" button in any freepbx admin interface and click to "reload". Trixbox will fix extensions_additional.conf by itself.
In reply to Unable to Trunk two trixbox:
You need to change the context to:
context=from-internal
to be able to call all extension numbers, otherwise create inbound routes for each extension number.
In reply to Trixbox Audio Timing Issue - X100P vs ztdummy:
Does it happen on internal calls as well? Have you "tuned" your FXO channels? I'd suspect the card - try replacing with something like Sangoma.
In reply to trixbox Community Today:
click the below link you can find more about trixbox. it'll help you who want learn more about voip
http://www.voip-sip-sdk.com/p_51-how-to-setup-ozeki-voip-sip-sdk-...
In reply to Trixbox Audio Timing Issue - X100P vs ztdummy:
It doesn't seem to happen during normal voice calls, both internal and external it only really happens on recorded messages like the IVR (for external calls), voice mail messages (both internal and external). I also get a clicking/beeping sound every so often (it varies but every 15-30 seconds approx).
I haven't 'tuned' my card, no. How do I do this? Is there any special drivers that I need to install for my card? The card was (physically) installed in the system when trixbox was installed and I am able to make outbound calls via the POTL so I assume it is installed correctly but I have not installed any extra drivers after the initial install of trixbox.
In reply to No clue..:
This could be due to permission problems. Those who run asterisk as non-root for the sake of safety usually witness these kind of issues. Meanwhile, this could be due to a sip peer also, which is trying to register outside to some voip service provider like Axvoice, Vonage or packet 8. check your sip.conf and comment any sip peers then do a sip reload and check if the issue persists.
