Hi Everyone,
I've got a Cisco 3810 rig with 6 analog FXS ports, and ethernet. I'm
wondering in anyone has got one of these suckers to work with asterisk in
such a way that each FXS port has it's own extension.
It speaks SIP, and I can send calls from asterisk out to it, but can't
figure out how to get it to pass username & pw to asterisk when I try to
configure it as a client. Eg -
Call from a Grandstream (working)-
Jul 8 20:53:57 DEBUG[229391]: chan_sip.c:3643 build_route: build_route:
Contact hop: <sip:4000@192.168.1.42>
-- Executing NoOp("SIP/4000-98ec", "") in new stack
-- Executing Goto("SIP/4000-98ec", "intern-post|4001|1")
in new stack
-- Goto (intern-post,4001,1)
-- Executing Dial("SIP/4000-98ec", "SIP/4001|30|Ttm") in
new stack
Jul 8 20:53:57 DEBUG[409616]: app_dial.c:420 dial_exec: SIMPLE DIAL (NO
URL)
Jul 8 20:53:57 DEBUG[409616]: chan_sip.c:835 create_addr: Setting NAT on
RTP to 0
Jul 8 20:53:57 DEBUG[409616]: chan_sip.c:1040 sip_call: Outgoing Call for
4001
Jul 8 20:53:57 DEBUG[409616]: chan_sip.c:1139 find_user: 4001 is not a
local user
-- Called 4001
Jul 8 20:53:57 DEBUG[409616]: channel.c:1402 ast_prod: Prodding channel
'SIP/4000-98ec'
Call from the Cisco (not working)
Jul 8 20:54:50 DEBUG[229391]: chan_sip.c:3643 build_route: build_route:
Contact hop: <sip:4002@192.168.1.9:5060>
-- Executing NoOp("SIP/192.168.1.9-08134bb8", "") in new
stack
-- Executing Goto("SIP/192.168.1.9-08134bb8",
"from-sip-post|4001|1")
in new stack
-- Goto (from-sip-post,4001,1)
Jul 8 20:54:50 WARNING[426000]: pbx.c:1802 ast_pbx_run: Channel
'SIP/192.168.1.9-08134bb8' sent into invalid extension '4001' in
context
'from-sip-post', but no invalid handler
BTW- Working with a ripped-off version of John Todd's configs... Anyone
get this working? It's kicking my ass.
Jim
I have an mc3800 working in my office with asterisk, you need the latest vertion of ios. i have the image if you want it. Sip has a lot of bugs on 12.2, I KNOW i went through hell On Fri, 2004-07-09 at 09:20, jlaing@freaksh0.net wrote:> Hi Everyone, > > I've got a Cisco 3810 rig with 6 analog FXS ports, and ethernet. I'm > wondering in anyone has got one of these suckers to work with asterisk in > such a way that each FXS port has it's own extension. > > It speaks SIP, and I can send calls from asterisk out to it, but can't > figure out how to get it to pass username & pw to asterisk when I try to > configure it as a client. Eg - > > Call from a Grandstream (working)- > > Jul 8 20:53:57 DEBUG[229391]: chan_sip.c:3643 build_route: build_route: > Contact hop: <sip:4000@192.168.1.42> > -- Executing NoOp("SIP/4000-98ec", "") in new stack > -- Executing Goto("SIP/4000-98ec", "intern-post|4001|1") in new stack > -- Goto (intern-post,4001,1) > -- Executing Dial("SIP/4000-98ec", "SIP/4001|30|Ttm") in new stack > Jul 8 20:53:57 DEBUG[409616]: app_dial.c:420 dial_exec: SIMPLE DIAL (NO > URL) > Jul 8 20:53:57 DEBUG[409616]: chan_sip.c:835 create_addr: Setting NAT on > RTP to 0 > Jul 8 20:53:57 DEBUG[409616]: chan_sip.c:1040 sip_call: Outgoing Call for > 4001 > Jul 8 20:53:57 DEBUG[409616]: chan_sip.c:1139 find_user: 4001 is not a > local user > -- Called 4001 > Jul 8 20:53:57 DEBUG[409616]: channel.c:1402 ast_prod: Prodding channel > 'SIP/4000-98ec' > > Call from the Cisco (not working) > > Jul 8 20:54:50 DEBUG[229391]: chan_sip.c:3643 build_route: build_route: > Contact hop: <sip:4002@192.168.1.9:5060> > -- Executing NoOp("SIP/192.168.1.9-08134bb8", "") in new stack > -- Executing Goto("SIP/192.168.1.9-08134bb8", "from-sip-post|4001|1") > in new stack > -- Goto (from-sip-post,4001,1) > Jul 8 20:54:50 WARNING[426000]: pbx.c:1802 ast_pbx_run: Channel > 'SIP/192.168.1.9-08134bb8' sent into invalid extension '4001' in context > 'from-sip-post', but no invalid handler > > BTW- Working with a ripped-off version of John Todd's configs... Anyone > get this working? It's kicking my ass. > > Jim > > > > _______________________________________________ > 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 > >
Are you also using it for outbound pstn connections? W. Kevin Hunt CCIE #11841 MCSE, Linux+ SME www.huntbrothers.com -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Alberto Fernandez Sent: Friday, July 09, 2004 9:03 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Cisco MC3810 -> Asterisk I have an mc3800 working in my office with asterisk, you need the latest vertion of ios. i have the image if you want it. Sip has a lot of bugs on 12.2, I KNOW i went through hell On Fri, 2004-07-09 at 09:20, jlaing@freaksh0.net wrote:> Hi Everyone, > > I've got a Cisco 3810 rig with 6 analog FXS ports, and ethernet. I'm > wondering in anyone has got one of these suckers to work with asterisk> in such a way that each FXS port has it's own extension. > > It speaks SIP, and I can send calls from asterisk out to it, but can't> figure out how to get it to pass username & pw to asterisk when I try > to configure it as a client. Eg - >http://lists.digium.com/mailman/listinfo/asterisk-users
Show us your sip.conf -- probably a config issue> -----Original Message----- > From: R. Anton Raharja [mailto:anton@ngoprek.org] > Sent: Friday, July 09, 2004 1:16 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] xlite "calls not approved" > > > asterisk 0.9.1 with regular sip.conf and extensions.conf > sjPhone able to register and make calls xlite said "logged > in" but when i start to call/dial it said "calls not > approved" n i dont see anything while my asterisk sip debug enabled > > can anyone give me a clue whats happening? > > >http://sleepless.ngoprek.org VoIP Rakyat: (0921) 20006 _______________________________________________ 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
ok, this is my sip.conf xlite cant calls, sjPhone can i wish sjPhone dont hav that popup thing :) [general] port = 5060 bindaddr = 0.0.0.0 context = intern tos=lowdelay videosupport=yes disallow=all allow=gsm allow=ulaw allow=alaw register => sleepless:pwd:sleepless@voiprakyat.net [voiprakyat.net] type=peer context=intern username=sleepless secret=pwd host=voiprakyat.net nat=yes canreinvite=no [1234] type=friend context=intern username=1234 secret=pwd host=dynamic nat=yes canreinvite=yes [5678] type=friend context=intern username=5678 secret=pwd host=dynamic nat=yes canreinvite=yes *********** REPLY SEPARATOR *********** On 09/07/2004 at 14:29 Jay Milk wrote:>Show us your sip.conf -- probably a config issue > >> -----Original Message----- >> From: R. Anton Raharja [mailto:anton@ngoprek.org] >> Sent: Friday, July 09, 2004 1:16 PM >> To: asterisk-users@lists.digium.com >> Subject: [Asterisk-Users] xlite "calls not approved" >> >> >> asterisk 0.9.1 with regular sip.conf and extensions.conf >> sjPhone able to register and make calls xlite said "logged >> in" but when i start to call/dial it said "calls not >> approved" n i dont see anything while my asterisk sip debug enabled >> >> can anyone give me a clue whats happening?http://sleepless.ngoprek.org VoIP Rakyat: (0921) 20006