Closed GoogleCodeExporter closed 9 years ago
Original comment by pmerl...@googlemail.com
on 11 Jun 2009 at 12:34
same with me, since last update (installed 10 mins ago)
i can call some other, but an incoming call doesn't work
i got a snom190 on extension 30 and sipdroid on 32 (Asterisk 1.2.33)
the relevant snipped of asterisk -r (verbosity 100):
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 32 to extension map
-- dialparties.agi: Extension 32 cf is disabled
-- dialparties.agi: Extension 32 do not disturb is disabled
> dialparties.agi: extnum 32 has: cw: 1; hascfb: 0 [] hascfu: 0 []
> dialparties.agi: ExtensionState: 0
-- dialparties.agi: dbset CALLTRACE/32 to 30
-- dialparties.agi: Filtered ARG3: 32
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing Dial("SIP/30-010b5c90", "SIP/32||tr") in new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Set("SIP/30-010b5c90", "DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing GosubIf("SIP/30-010b5c90", "0?CHANUNAVAIL|1") in new stack
-- Executing GotoIf("SIP/30-010b5c90", "0?exit|return") in new stack
if you need any debug info please write to Christian@blank-online.eu
Original comment by Blank.c...@googlemail.com
on 14 Jul 2009 at 7:53
did you tried changing qualify=no?
what does "sip show peers" display?
Original comment by buzm...@gmail.com
on 14 Jul 2009 at 9:38
I am using my own local asterisk server and I also cannot receive a call. I
can dial
out but cannot receive. It just goes straight to voicemail for me. Help!
Original comment by nate.bar...@gmail.com
on 20 Jul 2009 at 5:21
Same here. (using trixbox)
Original comment by Coolk...@gmail.com
on 23 Jul 2009 at 2:37
same here, using FreeSWITCH (asterisk alternative)
Original comment by intel352
on 28 Jul 2009 at 5:18
Im having the same problems, I can make calls no problems but receiving is not
functional.
Using pbxes with no success.
Original comment by 21st....@gmail.com
on 8 Aug 2009 at 8:07
Same with me. My extension becomes UNREACHABLE after a minute or so in Asterisk
1.4.26.1 - my guess it would be T-Mobile's firewall closing the connection to
5060.
Any chance of adding an option to re-register the extension? Usually the
defaults
that I've seen are 3600 secs. My company (Aretta Communications) suggests to our
clients t use a value of 180 for normal firewalls/NAT or 30 for problem
firewalls
(Cisco routers are notorious for forgetting state tables in under a minute).
Original comment by acosg...@gmail.com
on 2 Sep 2009 at 10:29
Original comment by pmerl...@googlemail.com
on 4 Sep 2009 at 3:07
[deleted comment]
this issue is "resolved" for me. IF you go to the web admin, and modify the
extension
the Phone will be using to have it's option for qualify set to no.
Though Really still needs to be fixed, i can now receive incoming calls.
Original comment by Coolk...@gmail.com
on 25 Sep 2009 at 9:14
I got it work PBX Features turn on improve audio exit out of app and then
restart and
my incoming calls worked
Original comment by littlejo...@gmail.com
on 29 Sep 2009 at 8:15
Okay update to my issue I can get a call to come in right after I launch the
APP if I
wait 5 seconds I cannot get a call in only after I close the app and relaunch
it with
in the first 2 or 3 seconds I can get a call in but also after I hang up the
call I
cannot get a call back in.
Original comment by littlejo...@gmail.com
on 29 Sep 2009 at 8:36
I have this problem with a Verizon HTC Incredible Android 2.2 Sipdroid 2.4
beta. I use a Google Apps account. I can make outgoing calls just fine. When I
try to receive a call the PBXes.org Call Monitor shows "Busy". Could this have
something to do with not having in the correct SRV records for my Google Apps
domain?
Original comment by j...@jphein.com
on 7 Feb 2012 at 3:14
I’m having the same problems, I can make calls no problems but i recieve the
ringging but when answering it's not voice at all. I’m using the CUCM version
8.6
Original comment by abinjah...@gmail.com
on 16 Apr 2012 at 8:02
Original issue reported on code.google.com by
vtu...@gmail.com
on 11 Jun 2009 at 12:32