Dear all,
Do anyone know to setup asterisk's SIP channel to use an outbound proxy
outside of asterisk's network to proxy the SIP message?
Thanks
Ray
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Dear all,
we have try to limit the outgoing channel by using GROUP() and
GROUP_COUNT() to limit number of calls to a channel/trunk. but lately
we upgraded to 1.2.5, 1.2.6 or SVN 1.2 , both functions not work at
all. Is this a bug or just a misconfiguration on our part?
exten =
Tomislav ParĨina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi
What causes deadlock?
Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for
'0x82acb10', 10 retries!
Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for
'0x8298160', 10
Dear all,
we have problem with hang sip channels when lot of incoming calls to
registered users. we are using SVN 1.2 16771 version.
xxx.xxx.xxx.xxx 639489919160 14aea9430d0 00102/0 unkn No
Tx: CANCEL
xxx.xxx.xxx.xxx 639485949581 1b3dd036127 00102/0 unkn No
Dear all,
we have encounter problem when starting asterisk in the foreground,
asterisk -gc with more 100 SIP calls concurrently. we have set
ulimit to the highest value. still has this problem. Is this the
problem keeping asterisk in the foreground or this is a bug in SVN 1.2
16771?
Dear all,
Did anyone successfully test T38 fax pass thru to Cisco
as53xx? Weve tried 1.2.4 with latest patch and latest svn trunk
and T38 patch but still not work. Reinvites from Cisco are correctly passed
back to the originating gateway, but fax never able to connect.
Cisco IOS
Dear all,
Does anyone try to install 2 or multiple TE411 card into
one server? Can it be done? What about stability?
Thanks
Ray
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Dear all,
Anyone has experience using group and group_count to limit
outgoing calls in AGI/CAGI?
SET VARIABLE GROUP(${CALLERIDNUM}) OUTBOUND_GROUP
EXEC Gotoif $[${GROUP_COUNT([EMAIL PROTECTED])}
1]?BLOCK
SET VARIABLE GROUP(${CALLERIDNUM}) OUTBOUND_GROUP
But it doesnt work as
Dear all,
I have encountered problem with app_chanisavail for sip
channels. I have setup call-limit=1 in sip.conf as instructed, but when
making call to app_chanisavail, the channels did not increment correctly. I
end up dialing multiple times to the first channel only. I think the
Dear all,
I have encountered problem with app_chanisavail for sip
channels. I have setup call-limit=1 in sip.conf as instructed, but when
making call to app_chanisavail, the channels did not increment correctly.
I end up dialing multiple times to the first channel only.
I think the
Dear all,
Any one has experience in CAGI script setgroup? Please
let me know a bit of command detail.
Thanks,
Ray
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Dear all,
I would like to delete the
calleridname in the FROM URI so it will not forward to the gateway in SIP.
Ive tried everything available in SIP.conf but not able to do it.
Please help.
[test]
type=friend
context=sip-in
setvar(CALLERIDNAME =
)
callerid=123123123
123123123
Dear all,
Anyone get the lastest spandsp with udptl.c and tpkt.c
compile in Fedora 3?
tpkt.c: In function `accept_thread':
tpkt.c:140: error: `TCP_NODELAY' undeclared (first use
in this function)
tpkt.c:140: error: (Each undeclared identifier is
reported only once
tpkt.c:140:
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