Hi all,
I have the following setup:
UAs ------------SER ------------------------ ASTERISK
---------------------GNUGK --------------- GWs
SER is configured to route call requests from UAs to Asterisk. Asterisk is
configured to receive the call on SIP channel and dial out to GNUGK over
H323 channel. The problem I'm facing is that asterisk sends out the call
request to GNUGK and times out immediately, so call setup is never
completed. On GNUGK the call request comes in followed by a normal call
drop.
Any ideas on what could be the problem ??
My asterisk configuration, debug and console output are as follow :
SIP.CONF
=====[general]
port = 5080
bindaddr = 10.10.1.170
context = to_GNUGK
disallow=all
allow=g729
H323.CONF
======[general]
port = 1720
allow = g729
gatekeeper = 64.80.103.12
allowgkrouted = yes
context = to_SER
EXTENSIONS.CONF
===========[general]
static = yes
writeprotect = yes
[to_GNUGK]]
exten => _.,1,Dial(h323/${EXTEN}@10.10.1.12:1720,60,C)
[to_SER]
exten => _.,1,Dial(SIP/${EXTEN}@10.10.1.170:5060,60)
DEBUG File
=========Jul 15 16:14:10 DEBUG[65541]: Check for res for
Jul 15 16:14:10 DEBUG[65541]: is not a local user
Jul 15 16:14:10 DEBUG[65541]: build_route: Record-Route hop:
<sip:15613021234@10.10.1.170;ftag=661806388;lr=on>
Jul 15 16:14:10 DEBUG[65541]: build_route: Contact hop:
<sip:999012020@10.10.1.13:5060;user=phone;transport=udp>
Jul 15 16:14:10 DEBUG[311316]: SIMPLE DIAL (NO URL)
Jul 15 16:14:10 DEBUG[311316]: type=h323, format=256,
data=15613021234@10.10.1.12:1720.
Jul 15 16:14:10 DEBUG[311316]: Host: 10.10.1.12:1720 Username: 15613021234
Jul 15 16:14:10 DEBUG[311316]: dest=15613021234@10.10.1.12:1720, timeout=0.
Jul 15 16:14:13 DEBUG[213006]: Cleaning up our mess
Jul 15 16:14:23 DEBUG[311316]: SIMPLE DIAL (NO URL)
Jul 15 16:14:23 DEBUG[311316]: type=h323, format=256,
data=t@10.10.1.12:1720.
Jul 15 16:14:23 DEBUG[311316]: Host: 10.10.1.12:1720 Username: t
Jul 15 16:14:23 DEBUG[311316]: dest=t@10.10.1.12:1720, timeout=0.
Jul 15 16:14:24 DEBUG[213006]: Cleaning up our mess
Jul 15 16:14:31 DEBUG[311316]: SIMPLE DIAL (NO URL)
Jul 15 16:14:31 DEBUG[311316]: type=h323, format=256,
data=h@10.10.1.12:1720.
Jul 15 16:14:31 DEBUG[311316]: Host: 10.10.1.12:1720 Username: h
Jul 15 16:14:31 DEBUG[311316]: find_user() - decrement inUse counter
Jul 15 16:14:31 DEBUG[311316]: is not a local user
Jul 15 16:14:31 DEBUG[65541]: Stopping retransmission on
'842589597@10.10.1.13' of Response 1: Found
CONSOLE Output
=============*CLI> -- Executing Dial("SIP/-08121388",
"h323/15613021234@10.10.1.12:1720|60|C") in new stack
-- Called 15613021234@10.10.1.12:1720
== No one is available to answer at this time
-- Timeout on SIP/-08121388
== CDR updated on SIP/-08121388
_________________________________________________________________
Protect your PC - get McAfee.com VirusScan Online
http://clinic.mcafee.com/clinic/ibuy/campaign.asp?cid=3963
Hi all,
I have the following setup:
UAs ------------SER ---------- ASTERISK ----------GNUGK --------- GWs
SER is configured to route call requests from UAs to Asterisk. Asterisk is
configured to receive the call on SIP channel and dial out to GNUGK over
H323 channel. The problem I'm facing is that asterisk sends out the call
request to GNUGK and times out immediately, so call setup is never
completed. On GNUGK the call request comes in followed by a normal call
drop.
Any ideas on what could be the problem ??
My asterisk configuration, debug and console output are as follow :
SIP.CONF
=====[general]
port = 5080
bindaddr = 10.10.1.170
context = to_GNUGK
disallow=all
allow=g729
H323.CONF
======[general]
port = 1720
allow = g729
gatekeeper = 64.80.103.12
allowgkrouted = yes
context = to_SER
EXTENSIONS.CONF
===========[general]
static = yes
writeprotect = yes
[to_GNUGK]]
exten => _.,1,Dial(h323/${EXTEN}@10.10.1.12:1720,60,C)
[to_SER]
exten => _.,1,Dial(SIP/${EXTEN}@10.10.1.170:5060,60)
DEBUG File
=========Jul 15 16:14:10 DEBUG[65541]: Check for res for
Jul 15 16:14:10 DEBUG[65541]: is not a local user
Jul 15 16:14:10 DEBUG[65541]: build_route: Record-Route hop:
<sip:15613021234@10.10.1.170;ftag=661806388;lr=on>
Jul 15 16:14:10 DEBUG[65541]: build_route: Contact hop:
<sip:999012020@10.10.1.13:5060;user=phone;transport=udp>
Jul 15 16:14:10 DEBUG[311316]: SIMPLE DIAL (NO URL)
Jul 15 16:14:10 DEBUG[311316]: type=h323, format=256,
data=15613021234@10.10.1.12:1720.
Jul 15 16:14:10 DEBUG[311316]: Host: 10.10.1.12:1720 Username: 15613021234
Jul 15 16:14:10 DEBUG[311316]: dest=15613021234@10.10.1.12:1720, timeout=0.
Jul 15 16:14:13 DEBUG[213006]: Cleaning up our mess
Jul 15 16:14:23 DEBUG[311316]: SIMPLE DIAL (NO URL)
Jul 15 16:14:23 DEBUG[311316]: type=h323, format=256,
data=t@10.10.1.12:1720.
Jul 15 16:14:23 DEBUG[311316]: Host: 10.10.1.12:1720 Username: t
Jul 15 16:14:23 DEBUG[311316]: dest=t@10.10.1.12:1720, timeout=0.
Jul 15 16:14:24 DEBUG[213006]: Cleaning up our mess
Jul 15 16:14:31 DEBUG[311316]: SIMPLE DIAL (NO URL)
Jul 15 16:14:31 DEBUG[311316]: type=h323, format=256,
data=h@10.10.1.12:1720.
Jul 15 16:14:31 DEBUG[311316]: Host: 10.10.1.12:1720 Username: h
Jul 15 16:14:31 DEBUG[311316]: find_user() - decrement inUse counter
Jul 15 16:14:31 DEBUG[311316]: is not a local user
Jul 15 16:14:31 DEBUG[65541]: Stopping retransmission on
'842589597@10.10.1.13' of Response 1: Found
CONSOLE Output
=============*CLI> -- Executing Dial("SIP/-08121388",
"h323/15613021234@10.10.1.12:1720|60|C") in new stack
-- Called 15613021234@10.10.1.12:1720
== No one is available to answer at this time
-- Timeout on SIP/-08121388
== CDR updated on SIP/-08121388
_________________________________________________________________
MSN 8 with e-mail virus protection service: 2 months FREE*
http://join.msn.com/?page=features/virus
My SIP UA is an ATA186 and my H323 gateway is a Cisco 5300 and a Nextone.>From: administrator tootai <admin@tootai.net> >Reply-To: asterisk-users@lists.digium.com >To: asterisk-users@lists.digium.com >Subject: Re: [Asterisk-Users] SIP to H323 call timeout >Date: Tue, 20 Jul 2004 02:34:31 +0200 > >Fred Lee a écrit : > >> >> >>Hi all, >> >>I have the following setup: >> >>UAs ------------SER ---------- ASTERISK ----------GNUGK --------- GWs >> >>SER is configured to route call requests from UAs to Asterisk. Asterisk is >>configured to receive the call on SIP channel and dial out to GNUGK over >>H323 channel. The problem I'm facing is that asterisk sends out the call >>request to GNUGK and times out immediately, so call setup is never >>completed. On GNUGK the call request comes in followed by a normal call >>drop. >> >>Any ideas on what could be the problem ?? > >Do you use the h323 - Nufone? Is it a recent installation? If so, could be >the problem that GW need FastStart and the * h323 don't send it. > >> >>My asterisk configuration, debug and console output are as follow : >> >>SIP.CONF >>=====>>[general] >>port = 5080 >>bindaddr = 10.10.1.170 >>context = to_GNUGK >>disallow=all >>allow=g729 >> >> >>H323.CONF >>======>>[general] >>port = 1720 >>allow = g729 >>gatekeeper = 64.80.103.12 >>allowgkrouted = yes >>context = to_SER >> >>EXTENSIONS.CONF >>===========>>[general] >>static = yes >>writeprotect = yes >> >>[to_GNUGK]] >>exten => _.,1,Dial(h323/${EXTEN}@10.10.1.12:1720,60,C) >> >>[to_SER] >>exten => _.,1,Dial(SIP/${EXTEN}@10.10.1.170:5060,60) >> >> >> >>DEBUG File >>=========>>Jul 15 16:14:10 DEBUG[65541]: Check for res for >>Jul 15 16:14:10 DEBUG[65541]: is not a local user >>Jul 15 16:14:10 DEBUG[65541]: build_route: Record-Route hop: >><sip:15613021234@10.10.1.170;ftag=661806388;lr=on> >>Jul 15 16:14:10 DEBUG[65541]: build_route: Contact hop: >><sip:999012020@10.10.1.13:5060;user=phone;transport=udp> >>Jul 15 16:14:10 DEBUG[311316]: SIMPLE DIAL (NO URL) >>Jul 15 16:14:10 DEBUG[311316]: type=h323, format=256, >>data=15613021234@10.10.1.12:1720. >>Jul 15 16:14:10 DEBUG[311316]: Host: 10.10.1.12:1720 Username: >>15613021234 >>Jul 15 16:14:10 DEBUG[311316]: dest=15613021234@10.10.1.12:1720, >>timeout=0. >>Jul 15 16:14:13 DEBUG[213006]: Cleaning up our mess >>Jul 15 16:14:23 DEBUG[311316]: SIMPLE DIAL (NO URL) >>Jul 15 16:14:23 DEBUG[311316]: type=h323, format=256, >>data=t@10.10.1.12:1720. >>Jul 15 16:14:23 DEBUG[311316]: Host: 10.10.1.12:1720 Username: t >>Jul 15 16:14:23 DEBUG[311316]: dest=t@10.10.1.12:1720, timeout=0. >>Jul 15 16:14:24 DEBUG[213006]: Cleaning up our mess >>Jul 15 16:14:31 DEBUG[311316]: SIMPLE DIAL (NO URL) >>Jul 15 16:14:31 DEBUG[311316]: type=h323, format=256, >>data=h@10.10.1.12:1720. >>Jul 15 16:14:31 DEBUG[311316]: Host: 10.10.1.12:1720 Username: h >>Jul 15 16:14:31 DEBUG[311316]: find_user() - decrement inUse counter >>Jul 15 16:14:31 DEBUG[311316]: is not a local user >>Jul 15 16:14:31 DEBUG[65541]: Stopping retransmission on >>'842589597@10.10.1.13' of Response 1: Found >> >> >> >>CONSOLE Output >>=============>>*CLI> -- Executing Dial("SIP/-08121388", >>"h323/15613021234@10.10.1.12:1720|60|C") in new stack >> -- Called 15613021234@10.10.1.12:1720 >>== No one is available to answer at this time >> >> -- Timeout on SIP/-08121388 >>== CDR updated on SIP/-08121388 >> >>_________________________________________________________________ >>MSN 8 with e-mail virus protection service: 2 months FREE* >>http://join.msn.com/?page=features/virus >> >>_______________________________________________ >>Asterisk-Users mailing list >>Asterisk-Users@lists.digium.com >>http://lists.digium.com/mailman/listinfo/asterisk-users >>To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users_________________________________________________________________ STOP MORE SPAM with the new MSN 8 and get 2 months FREE* http://join.msn.com/?page=features/junkmail
Sorry the misunderstanding. I am using h323 nufone and faststart is being sent to the gateway, but SIP times out. My recent test was to register both ports of the ATA186 as SIP client directly to * and experiencing the same problem. When trying to call from one port to another, the call times out immediately, even though both ports are shown to be registered on the CLI interface in debug mode. I'm working with the latest release of *. Any ideas ?? Thanks for your response. Fred Lee>From: administrator tootai <admin@tootai.net> >Reply-To: asterisk-users@lists.digium.com >To: asterisk-users@lists.digium.com >Subject: Re: [Asterisk-Users] SIP to H323 call timeout >Date: Tue, 20 Jul 2004 13:20:54 +0200 > >Fred Lee a écrit : > >>My SIP UA is an ATA186 and my H323 gateway is a Cisco 5300 and a Nextone. > >My question was "which * h323 channel you're using?" (h323 Nufone or >oh323) Don't know about Cisco and Nextone but I also use an ATA186 as SIP >UA with GnuGK and have this problem. If you install an earlier that >20/05/04 CVS asterisk version with H323 Nufone channel it works. Don't know >how it works with the stable branch. > >Daniel > >> >> >>>From: administrator tootai <admin@tootai.net> >>>Reply-To: asterisk-users@lists.digium.com >>>To: asterisk-users@lists.digium.com >>>Subject: Re: [Asterisk-Users] SIP to H323 call timeout >>>Date: Tue, 20 Jul 2004 02:34:31 +0200 >>> >>>Fred Lee a écrit : >>> >>>> >>>> >>>>Hi all, >>>> >>>>I have the following setup: >>>> >>>>UAs ------------SER ---------- ASTERISK ----------GNUGK --------- GWs >>>> >>>>SER is configured to route call requests from UAs to Asterisk. Asterisk >>>>is configured to receive the call on SIP channel and dial out to GNUGK >>>>over H323 channel. The problem I'm facing is that asterisk sends out the >>>>call request to GNUGK and times out immediately, so call setup is never >>>>completed. On GNUGK the call request comes in followed by a normal call >>>>drop. >>>> >>>>Any ideas on what could be the problem ?? >>> >>> >>>Do you use the h323 - Nufone? Is it a recent installation? If so, could >>>be the problem that GW need FastStart and the * h323 don't send it. >>> >>>> >>>>My asterisk configuration, debug and console output are as follow : >>>> >>>>SIP.CONF >>>>=====>>>>[general] >>>>port = 5080 >>>>bindaddr = 10.10.1.170 >>>>context = to_GNUGK >>>>disallow=all >>>>allow=g729 >>>> >>>> >>>>H323.CONF >>>>======>>>>[general] >>>>port = 1720 >>>>allow = g729 >>>>gatekeeper = 64.80.103.12 >>>>allowgkrouted = yes >>>>context = to_SER >>>> >>>>EXTENSIONS.CONF >>>>===========>>>>[general] >>>>static = yes >>>>writeprotect = yes >>>> >>>>[to_GNUGK]] >>>>exten => _.,1,Dial(h323/${EXTEN}@10.10.1.12:1720,60,C) >>>> >>>>[to_SER] >>>>exten => _.,1,Dial(SIP/${EXTEN}@10.10.1.170:5060,60) >>>> >>>> >>>> >>>>DEBUG File >>>>=========>>>>Jul 15 16:14:10 DEBUG[65541]: Check for res for >>>>Jul 15 16:14:10 DEBUG[65541]: is not a local user >>>>Jul 15 16:14:10 DEBUG[65541]: build_route: Record-Route hop: >>>><sip:15613021234@10.10.1.170;ftag=661806388;lr=on> >>>>Jul 15 16:14:10 DEBUG[65541]: build_route: Contact hop: >>>><sip:999012020@10.10.1.13:5060;user=phone;transport=udp> >>>>Jul 15 16:14:10 DEBUG[311316]: SIMPLE DIAL (NO URL) >>>>Jul 15 16:14:10 DEBUG[311316]: type=h323, format=256, >>>>data=15613021234@10.10.1.12:1720. >>>>Jul 15 16:14:10 DEBUG[311316]: Host: 10.10.1.12:1720 Username: >>>>15613021234 >>>>Jul 15 16:14:10 DEBUG[311316]: dest=15613021234@10.10.1.12:1720, >>>>timeout=0. >>>>Jul 15 16:14:13 DEBUG[213006]: Cleaning up our mess >>>>Jul 15 16:14:23 DEBUG[311316]: SIMPLE DIAL (NO URL) >>>>Jul 15 16:14:23 DEBUG[311316]: type=h323, format=256, >>>>data=t@10.10.1.12:1720. >>>>Jul 15 16:14:23 DEBUG[311316]: Host: 10.10.1.12:1720 Username: t >>>>Jul 15 16:14:23 DEBUG[311316]: dest=t@10.10.1.12:1720, timeout=0. >>>>Jul 15 16:14:24 DEBUG[213006]: Cleaning up our mess >>>>Jul 15 16:14:31 DEBUG[311316]: SIMPLE DIAL (NO URL) >>>>Jul 15 16:14:31 DEBUG[311316]: type=h323, format=256, >>>>data=h@10.10.1.12:1720. >>>>Jul 15 16:14:31 DEBUG[311316]: Host: 10.10.1.12:1720 Username: h >>>>Jul 15 16:14:31 DEBUG[311316]: find_user() - decrement inUse counter >>>>Jul 15 16:14:31 DEBUG[311316]: is not a local user >>>>Jul 15 16:14:31 DEBUG[65541]: Stopping retransmission on >>>>'842589597@10.10.1.13' of Response 1: Found >>>> >>>> >>>> >>>>CONSOLE Output >>>>=============>>>>*CLI> -- Executing Dial("SIP/-08121388", >>>>"h323/15613021234@10.10.1.12:1720|60|C") in new stack >>>> -- Called 15613021234@10.10.1.12:1720 >>>>== No one is available to answer at this time >>>> >>>> -- Timeout on SIP/-08121388 >>>>== CDR updated on SIP/-08121388 >>>> >>>>_________________________________________________________________ >>>>MSN 8 with e-mail virus protection service: 2 months FREE* >>>>http://join.msn.com/?page=features/virus >>>> >>>>_______________________________________________ >>>>Asterisk-Users mailing list >>>>Asterisk-Users@lists.digium.com >>>>http://lists.digium.com/mailman/listinfo/asterisk-users >>>>To UNSUBSCRIBE or update options visit: >>>> http://lists.digium.com/mailman/listinfo/asterisk-users >>>> >>> >>>_______________________________________________ >>>Asterisk-Users mailing list >>>Asterisk-Users@lists.digium.com >>>http://lists.digium.com/mailman/listinfo/asterisk-users >>>To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >>_________________________________________________________________ >>STOP MORE SPAM with the new MSN 8 and get 2 months FREE* >>http://join.msn.com/?page=features/junkmail >> >>_______________________________________________ >>Asterisk-Users mailing list >>Asterisk-Users@lists.digium.com >>http://lists.digium.com/mailman/listinfo/asterisk-users >>To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users_________________________________________________________________ STOP MORE SPAM with the new MSN 8 and get 2 months FREE* http://join.msn.com/?page=features/junkmail
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