Displaying 3 results from an estimated 3 matches for "trunk10".
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trunk1
2009 Jun 01
1
IAX2 trunking with Older Asterisk, version ?
my sip phone registered on 1.6, when i dial 4567 from 1.6 version, it wont go to 1.6 voice mail. it says
== Using SIP RTP CoS mark 5
-- Executing [4567 at sip:1] Dial("SIP/312-09f9a720", "IAX2/trunk10 at 147.120.203.98/4567,10,t") in new stack
-- Called trunk10 at 147.120.203.98/4567
[Jun 1 11:01:18] WARNING[8178]: chan_iax2.c:8991 socket_process: Call rejected by 147.120.203.98: No authority found
-- Hungup 'IAX2/trunk14-9738'
== Everyone is busy/congested at this time (...
2003 Oct 16
1
Prob with Ringing multiple Channels
...ultaneously, only 1 channel is actually ringing.
In our configuration, the Asterisk box is connected to an E1 channel bank,
where 15 analog extensions are conencted to channelbank inturn.
We tried following,
Dial,Zap/g4/444&Zap/g4/448|20|t
Heres the output,
-- Executing Dial("IAX2[trunk10@trunk50]/1", "Zap/g4/444&Zap/g4/448|20|t") in new stack
-- Called g4/444
-- Called g4/446
-- Zap/1-1 answered IAX2[trunk10@trunk50]/1
-- Hungup 'Zap/2-1'
the above, "Zap/1-1 answered IAX2[trunk10@trunk50]/1" line comes as soon as
that Zap/1-1 li...
2007 Nov 30
3
How to setup redundant SIP peers
Hello list,
I try to setup an asterisk-server with different SIP-Peers to PSTN.
The Peer are working and configured in sip.conf:
[peer1]
type=peer
host=10.10.10.1
[peer2]
type=peer
host=10.10.10.2
Now dialout is no problem. Extensions.conf says:
exten => _0Z.,1,Dial(SIP/49${EXTEN:1}@peer1,30)
But how can I setup a failure-route if the SIP-Proxy "peer1" ist not