Here you go:
<-- SIP read from 1.2.3.4:5060:
INVITE sip:3125551212;[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK08mh9j107ol1tace2500.1
From: <sip:[EMAIL
PROTECTED]:5060;user=phone;testplex=TESTPLEX-h7l88bktjdhsf>;tag=10000000-0-1617457931
To: <sip:[EMAIL PROTECTED];user=phone>
CSeq: 1 INVITE
Contact: <sip:[EMAIL
PROTECTED]:5060;testplex=TESTPLEX-h7l88bktjdhsf;transport=udp>
Call-ID: [EMAIL PROTECTED]
P-Asserted-Identity: <sip:[EMAIL PROTECTED];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 - [EMAIL PROTECTED]
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:[EMAIL PROTECTED]: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:[EMAIL
PROTECTED]:5060;user=phone;testplex=TESTPLEX-h7l88bktjdhsf>;tag=10000000-0-1617457931
To: <sip:[EMAIL PROTECTED];user=phone>
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[EMAIL PROTECTED]>
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:[EMAIL
PROTECTED]:5060;user=phone;testplex=TESTPLEX-h7l88bktjdhsf>;tag=10000000-0-1617457931
To: <sip:[EMAIL PROTECTED];user=phone>;tag=as48adf352
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
---
asterisk*CLI>
<-- SIP read from 1.2.3.4:5060:
ACK sip:3125551212;[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK08mh9j107ol1tace2500.1
CSeq: 1 ACK
From: <sip:[EMAIL
PROTECTED]:5060;user=phone;testplex=TESTPLEX-h7l88bktjdhsf>;tag=10000000-0-1617457931
To: <sip:[EMAIL PROTECTED];user=phone>;tag=as48adf352
Call-ID: [EMAIL PROTECTED]
Max-Forwards: 69
Content-Length: 0
--- (8 headers 0 lines)---
Destroying call '[EMAIL PROTECTED]'
On 8/11/06, Nir Simionovich <[EMAIL PROTECTED]> 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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] 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 <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> wrote:
> > s, means that it got an incoming call, but no exten came with it.
> >
> > On 8/11/06, Mr. Jones <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> wrote:
> > > > Perhaps the context in sip.conf doesn't match the context in the
dial plan.
> > > >
> > > > ________________________________
> > > >
> > > > From: [EMAIL PROTECTED] on
behalf of Mr. Jones
> > > > Sent: Fri 8/11/2006 2:34 PM
> > > > To: [email protected]
> > > > 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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> > > > asterisk-users mailing list
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> > > >
> > > >
> > > >
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> > >
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> > > To UNSUBSCRIBE or update options visit:
> > >
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> >
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> >
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_______________________________________________
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