Re: [asterisk-users] SIP interconnection problem
Someone? As * is used so extensively with SIP I must've made a _glaring_
mistake in my config (!)
/Rob
Robert Bielik skrev:
Tarek Sawah skrev:
you need to post you SIP.conf and your Extensions.conf so someone can
have a look at them and see if there is anything missing
what are the contexts you are using with your peers?
what is the dial plan triggered when calling your destination number?
Machine 1 ---
iax.conf: ==
[general]
bandwidth=low
disallow=lpc10 ; Icky sound quality... Mr. Roboto.
jitterbuffer=no
forcejitterbuffer=no
autokill=yes
[2200]
type=friend
host=dynamic
context=users
username=2200
secret=none
auth=md5
sip.conf ===
[general]
port=5060
bindaddr=0.0.0.0
disallow=all
allow=alaw ; Allow codecs in order of preference
allow=ulaw
allow=gsm
allow=g726
dtmfmode=rfc2833
register = machine_1:wabo...@192.168.10.77/machine_2
[machine_2]
allow=alaw,ulaw,gsm,g726
host=dynamic
secret=wabooba
type=friend
context=sip_incoming
username=machine_2
extensions.conf ==
[general]
static=yes
writeprotect=no
clearglobalvars=no
[globals]
; The outgoing sip trunk
SIP_TRUNK=192.168.10.77
OUTGOING_PREFIX=0
[default]
include = sip-incoming
include = test
[test]
; Create an extension, 600, for evaluating echo latency.
;
exten = 600,1,Playback(demo-echotest) ; Let them know what's going on
exten = 600,n,Echo ; Do the echo test
exten = 600,n,Playback(demo-echodone) ; Let them know it's over
exten = 600,n,Goto(s,6)
[users]
include = sip-incoming
include = outgoing
include = test
[sip-incoming]
include = agi-async
include = internal
[agi-async]
exten = _01,1,Agi(agi:async)
[internal]
exten = _2XXX,1,NoOp()
exten = _2XXX,n,Dial(IAX2/${EXTEN})
exten = _2XXX,n,Hangup()
[outgoing-agi-async]
exten = _${OUTGOING_PREFIX}.,1,Dial(SIP/${ext...@${sip_trunk})
exten = _${OUTGOING_PREFIX}.,n,Set(CALLERID(name)=reason-${DIALSTATUS})
exten = _${OUTGOING_PREFIX}.,n,Agi(agi:async)
[outgoing]
exten = _${OUTGOING_PREFIX}.,1,Dial(SIP/${SIP_TRUNK}/${EXTEN:1})
exten = _${OUTGOING_PREFIX}.,n,Hangup()
Machine 2
sip.conf ===
[general]
port=5060
bindaddr=0.0.0.0
disallow=all
allow=alaw ; Allow codecs in order of preference
allow=ulaw
allow=gsm
allow=g726
dtmfmode=rfc2833
register = machine_2:wabo...@192.168.10.11/machine_1
[machine_1]
allow=alaw,ulaw,gsm,g726
host=dynamic
secret=wabooba
type=friend
context=sip_incoming
username=machine_1
extensions.conf ==
[globals]
; The outgoing sip trunk
SIP_TRUNK=192.168.10.11
Rest is exactly the same. I have a zoiper connected to each machine and I'm
trying to make a call from Machine 2 to zoiper
on Machine 1:
-- Registered IAX2 '2200' (AUTHENTICATED) at 192.168.10.113:4569
-- Accepting AUTHENTICATED call from 192.168.10.113:
requested format = gsm,
requested prefs = (),
actual format = gsm,
host prefs = (),
priority = mine
-- Executing [02...@users:1] Dial(IAX2/2200-1200,
SIP/192.168.10.11/2200) in new stack
== Using SIP RTP CoS mark 5
-- Called 192.168.10.11/2200
[Oct 26 09:20:25] NOTICE[20248]: chan_sip.c:15031 handle_response_invite:
Failed to authenticate on INVITE to '2200
sip:2...@192.168.10.77;tag=as6173091f'
-- SIP/192.168.10.11-090c2ea8 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [02...@users:2] Hangup(IAX2/2200-1200, ) in new stack
== Spawn extension (users, 02200, 2) exited non-zero on 'IAX2/2200-1200'
-- Hungup 'IAX2/2200-1200'
Besides that sip show peers on either machine shows the other one correctly
registered, and iax2 show peers shows the connected zoiper on each machine.
Ideas, please ??
TIA
/Rob
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Re: [asterisk-users] SIP interconnection problem
Lacking any response I tried to set insecure=invite on both sides. And lo and behold, the call gets through. Now, is this good or bad? /R ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP interconnection problem
Since you are doing peer-to-peer, this should be harmless. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Bielik Sent: Tuesday, October 27, 2009 11:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP interconnection problem Lacking any response I tried to set insecure=invite on both sides. And lo and behold, the call gets through. Now, is this good or bad? /R ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP interconnection problem
Tarek Sawah skrev:
you need to post you SIP.conf and your Extensions.conf so someone can
have a look at them and see if there is anything missing
what are the contexts you are using with your peers?
what is the dial plan triggered when calling your destination number?
Machine 1 ---
iax.conf: ==
[general]
bandwidth=low
disallow=lpc10 ; Icky sound quality... Mr. Roboto.
jitterbuffer=no
forcejitterbuffer=no
autokill=yes
[2200]
type=friend
host=dynamic
context=users
username=2200
secret=none
auth=md5
sip.conf ===
[general]
port=5060
bindaddr=0.0.0.0
disallow=all
allow=alaw ; Allow codecs in order of preference
allow=ulaw
allow=gsm
allow=g726
dtmfmode=rfc2833
register = machine_1:wabo...@192.168.10.77/machine_2
[machine_2]
allow=alaw,ulaw,gsm,g726
host=dynamic
secret=wabooba
type=friend
context=sip_incoming
username=machine_2
extensions.conf ==
[general]
static=yes
writeprotect=no
clearglobalvars=no
[globals]
; The outgoing sip trunk
SIP_TRUNK=192.168.10.77
OUTGOING_PREFIX=0
[default]
include = sip-incoming
include = test
[test]
; Create an extension, 600, for evaluating echo latency.
;
exten = 600,1,Playback(demo-echotest) ; Let them know what's going on
exten = 600,n,Echo ; Do the echo test
exten = 600,n,Playback(demo-echodone) ; Let them know it's over
exten = 600,n,Goto(s,6)
[users]
include = sip-incoming
include = outgoing
include = test
[sip-incoming]
include = agi-async
include = internal
[agi-async]
exten = _01,1,Agi(agi:async)
[internal]
exten = _2XXX,1,NoOp()
exten = _2XXX,n,Dial(IAX2/${EXTEN})
exten = _2XXX,n,Hangup()
[outgoing-agi-async]
exten = _${OUTGOING_PREFIX}.,1,Dial(SIP/${ext...@${sip_trunk})
exten = _${OUTGOING_PREFIX}.,n,Set(CALLERID(name)=reason-${DIALSTATUS})
exten = _${OUTGOING_PREFIX}.,n,Agi(agi:async)
[outgoing]
exten = _${OUTGOING_PREFIX}.,1,Dial(SIP/${SIP_TRUNK}/${EXTEN:1})
exten = _${OUTGOING_PREFIX}.,n,Hangup()
Machine 2
sip.conf ===
[general]
port=5060
bindaddr=0.0.0.0
disallow=all
allow=alaw ; Allow codecs in order of preference
allow=ulaw
allow=gsm
allow=g726
dtmfmode=rfc2833
register = machine_2:wabo...@192.168.10.11/machine_1
[machine_1]
allow=alaw,ulaw,gsm,g726
host=dynamic
secret=wabooba
type=friend
context=sip_incoming
username=machine_1
extensions.conf ==
[globals]
; The outgoing sip trunk
SIP_TRUNK=192.168.10.11
Rest is exactly the same. I have a zoiper connected to each machine and I'm
trying to make a call from Machine 2 to zoiper
on Machine 1:
-- Registered IAX2 '2200' (AUTHENTICATED) at 192.168.10.113:4569
-- Accepting AUTHENTICATED call from 192.168.10.113:
requested format = gsm,
requested prefs = (),
actual format = gsm,
host prefs = (),
priority = mine
-- Executing [02...@users:1] Dial(IAX2/2200-1200,
SIP/192.168.10.11/2200) in new stack
== Using SIP RTP CoS mark 5
-- Called 192.168.10.11/2200
[Oct 26 09:20:25] NOTICE[20248]: chan_sip.c:15031 handle_response_invite:
Failed to authenticate on INVITE to '2200
sip:2...@192.168.10.77;tag=as6173091f'
-- SIP/192.168.10.11-090c2ea8 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [02...@users:2] Hangup(IAX2/2200-1200, ) in new stack
== Spawn extension (users, 02200, 2) exited non-zero on 'IAX2/2200-1200'
-- Hungup 'IAX2/2200-1200'
Besides that sip show peers on either machine shows the other one correctly
registered, and iax2 show peers shows the connected zoiper on each machine.
Ideas, please ??
TIA
/Rob
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Re: [asterisk-users] SIP interconnection problem
Ooops.. forgot. The versions of * are: Machine 1: 1.6.1.4 Machine 2: 1.6.0.5 /Rob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP interconnection problem
Hi all, I've setup two * servers which are SIP interconnected ala osaka/toronto from the * book (before anyone sugggests using IAX instead, no, I NEED to have them SIP interconnected for verification/test purposes). Then I have a Zoiper connected to one of them via IAX (so that * will not reinvite (?)). As soon as I try to call (via Zoiper) an extension on the other * I get a Failed to authenticate on INVITE on the * to which the Zoiper is registered: -- Accepting AUTHENTICATED call from 192.168.10.113: Zoiper IP requested format = gsm, requested prefs = (), actual format = ulaw, host prefs = (ulaw|alaw|gsm), priority = mine -- Executing [010...@users:1] Dial(IAX2/2200-12940, SIP/010...@192.168.10.11) in new stack == Using SIP RTP CoS mark 5 -- Called 010...@192.168.10.11 Other * [Oct 23 11:08:25] NOTICE[13576]: chan_sip.c:15031 handle_response_invite: Failed to authenticate on INVITE to '2200 sip:2...@192.168.10.77;tag=as3e4fedb8' 192.168.10.77 == * for Zoiper -- SIP/192.168.10.11-0a1716f8 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'IAX2/2200-12940' status is 'CONGESTION' -- Hungup 'IAX2/2200-12940' Why does * try to authenticate on sip:2...@192.168.10.77, it is IAX for crying out loud :) ? I've set canreinvite=no on the IAX phone (not sure this has any meaning in IAX at all) Not sure that this is root of the interconnection problem, since I then get SIP/192.168.10.11.. is circuit-busy... ? TIA /R ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP interconnection problem
you need to post you SIP.conf and your Extensions.conf so someone can have a look at them and see if there is anything missing what are the contexts you are using with your peers? what is the dial plan triggered when calling your destination number? -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Sun, 25 Oct 2009 15:19:28 +0100 From: robert.bie...@xponaut.se To: asterisk-users@lists.digium.com Subject: [asterisk-users] SIP interconnection problem Hi all, I've setup two * servers which are SIP interconnected ala osaka/toronto from the * book (before anyone sugggests using IAX instead, no, I NEED to have them SIP interconnected for verification/test purposes). Then I have a Zoiper connected to one of them via IAX (so that * will not reinvite (?)). As soon as I try to call (via Zoiper) an extension on the other * I get a Failed to authenticate on INVITE on the * to which the Zoiper is registered: -- Accepting AUTHENTICATED call from 192.168.10.113: Zoiper IP requested format = gsm, requested prefs = (), actual format = ulaw, host prefs = (ulaw|alaw|gsm), priority = mine -- Executing [010...@users:1] Dial(IAX2/2200-12940, SIP/010...@192.168.10.11) in new stack == Using SIP RTP CoS mark 5 -- Called 010...@192.168.10.11 Other * [Oct 23 11:08:25] NOTICE[13576]: chan_sip.c:15031 handle_response_invite: Failed to authenticate on INVITE to '2200 sip:2...@192.168.10.77;tag=as3e4fedb8' 192.168.10.77 == * for Zoiper -- SIP/192.168.10.11-0a1716f8 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'IAX2/2200-12940' status is 'CONGESTION' -- Hungup 'IAX2/2200-12940' Why does * try to authenticate on sip:2...@192.168.10.77, it is IAX for crying out loud :) ? I've set canreinvite=no on the IAX phone (not sure this has any meaning in IAX at all) Not sure that this is root of the interconnection problem, since I then get SIP/192.168.10.11.. is circuit-busy... ? TIA /R ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Windows 7: I wanted more reliable, now it's more reliable. Wow! http://microsoft.com/windows/windows-7/default-ga.aspx?h=myidea?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_myidea:102009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
