[asterisk-users] Calling '**1' through Asterisk

Wed May 28 15:10:14 CDT 2008

Thanks for the reply.

Good point with the features.conf. But I do not have any features.conf 
which conflict with **1 - and it is **2, **3 etc as well. Anyway, **10 should be 
tricked by features as well, which it does not. And features only works on a 
bridged call, and this does not even get that far..

The sip set debug peer <> did get me some extra output, but I am not able 
to get any sense from it. This is what came out (Asterisk is on 192.168.2.1 
AND 192.168.27.7 and the vopiphone is on 192.168.7.98 and has the 
username 018):

<--- SIP read from 192.168.27.98:5060 --->
INVITE sip:**1 at 192.168.2.1 SIP/2.0
From: <sip:018 at 192.168.2.1:5060>;tag=10c4a070-621ba8c0-13c4-40030-
16ba5-550488a5-16ba5
To: <sip:**1 at 192.168.2.1>
Call-ID: 10b2f2f0-621ba8c0-13c4-40030-16ba5-2f6d4897-16ba5
CSeq: 1 INVITE
Via: SIP/2.0/UDP 192.168.27.98:5060;rport;branch=z9hG4bK-16ba5-
58c7d35-4a11b393
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, 
SUBSCRIBE, NOTIFY, MESSAGE, INFO
User-Agent: ATMEL_UA v0.0.25-alpha
Max-Forwards: 70
Contact: <sip:018 at 192.168.27.98:5060>
Content-Type: application/sdp
Content-Length: 334

v=0
o=018 12119074 77112119074177 IN IP4 192.168.27.98
s=audio
c=IN IP4 192.168.27.98
t=0 0
m=audio 16426 RTP/AVP 0 8 18 4 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
a=ptime:30

<------------->
--- (12 headers 15 lines) ---
Sending to 192.168.27.98 : 5060 (NAT)
Using INVITE request as basis request - 10b2f2f0-621ba8c0-13c4-40030-
16ba5-2f6d4897-16ba5

<--- Reliably Transmitting (no NAT) to 192.168.27.98:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.27.98:5060;branch=z9hG4bK-16ba5-58c7d35-
4a11b393;received=192.168.27.98;rport=5060
From: <sip:018 at 192.168.2.1:5060>;tag=10c4a070-621ba8c0-13c4-40030-
16ba5-550488a5-16ba5
To: <sip:**1 at 192.168.2.1>;tag=as3903248a
Call-ID: 10b2f2f0-621ba8c0-13c4-40030-16ba5-2f6d4897-16ba5
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, 
NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="ostergaard.net", 
nonce="19ae9086"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '10b2f2f0-621ba8c0-13c4-40030-
16ba5-2f6d4897-16ba5' in 32000 ms (Method: INVITE)
Found user '018'
Ildvaeg*CLI> 
<--- SIP read from 192.168.27.98:5060 --->
ACK sip:**1 at 192.168.2.1 SIP/2.0
From: <sip:018 at 192.168.2.1:5060>;tag=10c4a070-621ba8c0-13c4-40030-
16ba5-550488a5-16ba5
To: <sip:**1 at 192.168.2.1>;tag=as3903248a
Call-ID: 10b2f2f0-621ba8c0-13c4-40030-16ba5-2f6d4897-16ba5
CSeq: 1 ACK
Via: SIP/2.0/UDP 192.168.27.98:5060;rport;branch=z9hG4bK-16ba5-
58c7d35-4a11b393
Max-Forwards: 70
Contact: <sip:018 at 192.168.27.98:5060>
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Ildvaeg*CLI> 
<--- SIP read from 192.168.27.98:5060 --->
INVITE sip:**1 at 192.168.2.1 SIP/2.0
From: <sip:018 at 192.168.2.1:5060>;tag=10c4a070-621ba8c0-13c4-40030-
16ba5-550488a5-16ba5
To: <sip:**1 at 192.168.2.1>
Call-ID: 10b2f2f0-621ba8c0-13c4-40030-16ba5-2f6d4897-16ba5
CSeq: 2 INVITE
Via: SIP/2.0/UDP 192.168.27.98:5060;rport;branch=z9hG4bK-16ba5-
58c7d53-6bede6e9
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, 
SUBSCRIBE, NOTIFY, MESSAGE, INFO
User-Agent: ATMEL_UA v0.0.25-alpha
Max-Forwards: 70
Contact: <sip:018 at 192.168.27.98:5060>
Proxy-Authorization: Digest 
username="018",realm="ostergaard.net",nonce="19ae9086",uri="sip:**1 at 1
92.168.2.1",response="512fdfcf3ad644a79d92e4037679eee9",algorithm=M
D5
Content-Type: application/sdp
Content-Length: 334

v=0
o=018 12119074 77112119074177 IN IP4 192.168.27.98
s=audio
c=IN IP4 192.168.27.98
t=0 0
m=audio 16426 RTP/AVP 0 8 18 4 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
a=ptime:30

<------------->
--- (13 headers 15 lines) ---
Sending to 192.168.27.98 : 5060 (NAT)
Using INVITE request as basis request - 10b2f2f0-621ba8c0-13c4-40030-
16ba5-2f6d4897-16ba5
Found user '018'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 192.168.27.98:16426
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format G729 for ID 18
Found description format G723 for ID 4
Found description format iLBC for ID 97
Found description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x50d 
(g723|ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-
event), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.27.98:16426
Looking for **1 in from-internal (domain 192.168.2.1)

<--- Reliably Transmitting (no NAT) to 192.168.27.98:5060 --->
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 192.168.27.98:5060;branch=z9hG4bK-16ba5-58c7d53-
6bede6e9;received=192.168.27.98;rport=5060
From: <sip:018 at 192.168.2.1:5060>;tag=10c4a070-621ba8c0-13c4-40030-
16ba5-550488a5-16ba5
To: <sip:**1 at 192.168.2.1>;tag=as3903248a
Call-ID: 10b2f2f0-621ba8c0-13c4-40030-16ba5-2f6d4897-16ba5
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, 
NOTIFY
Supported: replaces
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '10b2f2f0-621ba8c0-13c4-40030-
16ba5-2f6d4897-16ba5' in 32000 ms (Method: INVITE)
Ildvaeg*CLI> 
<--- SIP read from 192.168.27.98:5060 --->
ACK sip:**1 at 192.168.2.1 SIP/2.0
From: <sip:018 at 192.168.2.1:5060>;tag=10c4a070-621ba8c0-13c4-40030-
16ba5-550488a5-16ba5
To: <sip:**1 at 192.168.2.1>;tag=as3903248a
Call-ID: 10b2f2f0-621ba8c0-13c4-40030-16ba5-2f6d4897-16ba5
CSeq: 2 ACK
Via: SIP/2.0/UDP 192.168.27.98:5060;rport;branch=z9hG4bK-16ba5-
58c7d53-6bede6e9
Max-Forwards: 70
Contact: <sip:018 at 192.168.27.98:5060>
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

> Check your features.conf file and make sure that combination or a 
> similar combination (*1 or ** ) for instance isn't defined in there for 
> some reason.
> 
> 
> Matt Watson wrote:
> > Might want to do a "sip set debug peer <peer id>"
> >
> > You should then be able to see the sip packet dumps that are going between the phone and *.  Might give you some clues.
> >
> > --
> > Matt

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