Hi All,
I installed Astreisk 13beta with pjproject 2.3(2.2.1) from source code on
CentOS7.
--
https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject
The installation is OK.
But the connected SIP cilents (both Linphone on Windows7) cannot
communicate.
I hope your comment such as the testing for resolving the problem.
My status is the following(1 and 2).
Why 'Everyone is busy/congested at this time (1:0/0/1)'?
(1:0/0/1<---num.nochan is 1.)
----------
1. endpoint
*CLI> pjsip show endpoints
Endpoint: <Endpoint/CID.....................................>
<State.....> <Channels.>
I/OAuth:
<AuthId/UserName...........................................................>
Aor: <Aor............................................>
<MaxContact>
Contact: <Aor/ContactUri...............................>
<Status....> <RTT(ms)..>
Transport: <TransportId........> <Type> <cos>
<tos>
<BindAddress..................>
Identify:
<MatchList.................................................................>
Channel: <ChannelId......................................>
<State.....> <Time(sec)>
Exten: <DialedExten...........> CLCID:
<ConnectedLineCID.......>
========================================================================================
Endpoint: 9001 Not in
use 0 of inf
InAuth: auth9001/9001
Aor: 9001 10
Contact: 9001/sip:9001 at 192.168.177.180:16060
Avail 25.048
Transport: transport-udp udp 0 0 0.0.0.0:5060
Endpoint: 9002 Not in
use 0 of inf
InAuth: auth9002/9002
Aor: 9002 10
Contact: 9002/sip:9002 at 192.168.177.189
Avail 24.210
Transport: transport-udp udp 0 0 0.0.0.0:5060
----------
2. dial from 9001 to 9002
*CLI> -- Executing [9002 at internal:1]
Dial("PJSIP/9001-00000000",
"PJSIP/9002,20") in new stack
-- Called PJSIP/9002
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'PJSIP/9001-00000000' status is
'CHANUNAVAIL'
----------
Thanks,
MMEEGGAA
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Hi, can you check the Linphone Extension 9002!! The port is missing! Contact: 9002/sip:9002 at 192.168.177.189 <mailto:sip%3A9002 at 192.168.177.189>:???? Avail 24.210 Regards Rainer Am 05.09.2014 um 11:55 schrieb ?????????:> Hi All, > > I installed Astreisk 13beta with pjproject 2.3(2.2.1) from source code > on CentOS7. > --https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject > > The installation is OK. > But the connected SIP cilents (both Linphone on Windows7) cannot > communicate. > > I hope your comment such as the testing for resolving the problem. > > My status is the following(1 and 2). > Why 'Everyone is busy/congested at this time (1:0/0/1)'? > (1:0/0/1<---num.nochan is 1.) > > ---------- > 1. endpoint > *CLI> pjsip show endpoints > Endpoint: <Endpoint/CID.....................................> > <State.....> <Channels.> > I/OAuth: > <AuthId/UserName...........................................................> > Aor: <Aor............................................> > <MaxContact> > Contact: <Aor/ContactUri...............................> > <Status....> <RTT(ms)..> > Transport: <TransportId........> <Type> <cos> <tos> > <BindAddress..................> > Identify: > <MatchList.................................................................> > Channel: <ChannelId......................................> > <State.....> <Time(sec)> > Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......> > ========================================================================================> Endpoint: 9001 Not in > use 0 of inf > InAuth: auth9001/9001 > Aor: 9001 10 > Contact: 9001/sip:9001 at 192.168.177.180:16060 > <http://sip:9001 at 192.168.177.180:16060> Avail 25.048 > Transport: transport-udp udp 0 0 0.0.0.0:5060 > <http://0.0.0.0:5060> > Endpoint: 9002 Not in > use 0 of inf > InAuth: auth9002/9002 > Aor: 9002 10 > * Contact: 9002/**sip:9002 at 192.168.177.189 > <mailto:sip%3A9002 at 192.168.177.189>**Avail 24.210* > Transport: transport-udp udp 0 0 0.0.0.0:5060 > <http://0.0.0.0:5060> > > ---------- > 2. dial from 9001 to 9002 > > *CLI> -- Executing [9002 at internal:1] Dial("PJSIP/9001-00000000", > "PJSIP/9002,20") in new stack > -- Called PJSIP/9002 > == Everyone is busy/congested at this time (1:0/0/1) > -- Auto fallthrough, channel 'PJSIP/9001-00000000' status is > 'CHANUNAVAIL' > ---------- > > Thanks, > MMEEGGAA > > >-- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 <callto:004922897167161> P2P: sip:7000 at sip.soho-piper.de:5072 (pjsip-test) -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140905/4c406f47/attachment.html>
????????? wrote:> Hi All, > > I installed Astreisk 13beta with pjproject 2.3(2.2.1) from source code > on CentOS7. > --https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject<snip>> > ---------- > 2. dial from 9001 to 9002 > > *CLI> -- Executing [9002 at internal:1] Dial("PJSIP/9001-00000000", > "PJSIP/9002,20") in new stack > -- Called PJSIP/9002 > == Everyone is busy/congested at this time (1:0/0/1) > -- Auto fallthrough, channel 'PJSIP/9001-00000000' status is > 'CHANUNAVAIL'What is shown if you do "pjsip set logger on" and then try to place the call? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org