[asterisk-users] Inbound Calls & SIP/2.0 404 Not Found
Mr. Jones
worldsense at gmail.com
Fri Aug 11 13:47:06 MST 2006
Here you go:
<-- SIP read from 1.2.3.4:5060:
INVITE sip:3125551212;npdi at 2.3.4.5;user=phone SIP/2.0
Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK08mh9j107ol1tace2500.1
From: <sip:9491234567 at 1.2.3.4:5060;user=phone;testplex=TESTPLEX-h7l88bktjdhsf>;tag=10000000-0-1617457931
To: <sip:3125551212 at 2.3.4.5;user=phone>
CSeq: 1 INVITE
Contact: <sip:9491234567 at 1.2.3.4:5060;testplex=TESTPLEX-h7l88bktjdhsf;transport=udp>
Call-ID: 1741B130-876A7D at 10.254.1.7
P-Asserted-Identity: <sip:9491234567 at 10.254.1.7;user=phone>
Privacy: none
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 228
v=0
o=- 3364317821 3364317821 IN IP4 1.2.3.4
s=-
c=IN IP4 1.2.3.4
t=0 0
m=audio 20042 RTP/AVP 0 8 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--- (12 headers 11 lines)---
Using INVITE request as basis request - 1741B130-876A7D at 10.254.1.7
Sending to 1.2.3.4 : 5060 (non-NAT)
Found peer 'paetec_inbound'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 1.2.3.4:20042
Found description format PCMU
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x8010e (gsm|ulaw|alaw|g729|h263), peer - audio=0xc
(ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for s in lance-test (domain 3125551212)
list_route: hop:
<sip:9491234567 at 1.2.3.4:5060;testplex=TESTPLEX-h7l88bktjdhsf;transport=udp>
Transmitting (NAT) to 1.2.3.4:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
1.2.3.4:5060;branch=z9hG4bK08mh9j107ol1tace2500.1;received=1.2.3.4
From: <sip:9491234567 at 1.2.3.4:5060;user=phone;testplex=TESTPLEX-h7l88bktjdhsf>;tag=10000000-0-1617457931
To: <sip:3125551212 at 2.3.4.5;user=phone>
Call-ID: 1741B130-876A7D at 10.254.1.7
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s at 2.3.4.5>
Content-Length: 0
---
-- Executing NoOp("SIP/5060-b7a1aa50", "Exten is: s") in new stack
== Auto fallthrough, channel 'SIP/5060-b7a1aa50' status is 'UNKNOWN'
Reliably Transmitting (NAT) to 1.2.3.4:5060:
SIP/2.0 603 Declined
Via: SIP/2.0/UDP
1.2.3.4:5060;branch=z9hG4bK08mh9j107ol1tace2500.1;received=1.2.3.4
From: <sip:9491234567 at 1.2.3.4:5060;user=phone;testplex=TESTPLEX-h7l88bktjdhsf>;tag=10000000-0-1617457931
To: <sip:3125551212 at 2.3.4.5;user=phone>;tag=as48adf352
Call-ID: 1741B130-876A7D at 10.254.1.7
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s at 2.3.4.5>
Content-Length: 0
---
asterisk*CLI>
<-- SIP read from 1.2.3.4:5060:
ACK sip:3125551212;npdi at 2.3.4.5;user=phone SIP/2.0
Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK08mh9j107ol1tace2500.1
CSeq: 1 ACK
From: <sip:9491234567 at 1.2.3.4:5060;user=phone;testplex=TESTPLEX-h7l88bktjdhsf>;tag=10000000-0-1617457931
To: <sip:3125551212 at 2.3.4.5;user=phone>;tag=as48adf352
Call-ID: 1741B130-876A7D at 10.254.1.7
Max-Forwards: 69
Content-Length: 0
--- (8 headers 0 lines)---
Destroying call '1741B130-876A7D at 10.254.1.7'
On 8/11/06, Nir Simionovich <nirs at exchange.atelis.net> wrote:
>
>
>
> Hmmm...
>
> Appears as if the SIP invite request is ill-formed. Can you send the SIP
> debug
> of the session to the list, so we may examine it?
>
> Nir S
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf
> Of Mr. Jones
> Sent: Friday, August 11, 2006 10:10 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found
>
>
>
>
> Actually it looks like I am getting the number but its coming through weird:
>
> This is what sip debug gives me:
>
> Looking for s in test-context (domain 9495551212)
>
> So clearly I am getting the number, just not sure if its formated ok?
>
>
> On 8/11/06, Mr. Jones <worldsense at gmail.com> wrote:
> > Yeah...
> >
> >
> > I tried the NoOp function someone gave me above and I'll I'm getting is
> "s"
> >
> > I'll go back to the provider
> >
> > On 8/11/06, C F <shmaltz at gmail.com> wrote:
> > > s, means that it got an incoming call, but no exten came with it.
> > >
> > > On 8/11/06, Mr. Jones <worldsense at gmail.com> wrote:
> > > > I double checked the context.
> > > >
> > > > But the "Looking for s" is a bit confusing - not sure what "s" is?
> > > >
> > > > On 8/11/06, Vadim Berezniker <VadimB at nbsvoice.com> wrote:
> > > > > Perhaps the context in sip.conf doesn't match the context in the
> dial plan.
> > > > >
> > > > > ________________________________
> > > > >
> > > > > From: asterisk-users-bounces at lists.digium.com on
> behalf of Mr. Jones
> > > > > Sent: Fri 8/11/2006 2:34 PM
> > > > > To: asterisk-users at lists.digium.com
> > > > > Subject: [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found
> > > > >
> > > > >
> > > > >
> > > > > I'm trying to get inbound DIDs working via SIP.
> > > > >
> > > > > I have 20 DIDs coming in via a single SIP profile in sip.conf.
> > > > >
> > > > > I was hoping to have these matched in extensions.conf, so I have
> setup
> > > > > lines like this:
> > > > >
> > > > > exten=>949271NNNN,1, Goto(mainmenu,s,1)
> > > > >
> > > > > Unfortunately these aren't getting matched and I'm getting this
> error:
> > > > >
> > > > > Looking for s in druid-default (domain 949271NNNN)
> > > > > SIP/2.0 404 Not Found
> > > > >
> > > > > Any hints or tips?
> > > > >
> > > > > TIA
> > > > > _______________________________________________
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> > > > >
> > > > >
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> > > > >
> > > > >
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