AsteriskNOW and xlite

Asterisk Asterisk Support
1

I am a newbie and have installed AsteriskNOW and xlite. Though the xlite phones are registered. but calls cannot be established between the xlite phones.

The logs for the same are given here:

Reliably Transmitting (NAT) to 10.1.5.15:6516:
OPTIONS sip:2000@10.1.5.15:6516;rinstance=c496bce22ac2cced SIP/2.0
Via: SIP/2.0/UDP 10.1.5.18:5060;branch=z9hG4bK7ddd150f;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.1.5.18;tag=as69ab3161
To: sip:2000@10.1.5.15:6516;rinstance=c496bce22ac2cced
Contact: sip:asterisk@10.1.5.18:5060
Call-ID: 6c6e9453429d3793796af1d74c3bff29@10.1.5.18:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11-cert1
Date: Tue, 10 Jul 2012 04:24:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:10.1.5.15:6516 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.5.18:5060;branch=z9hG4bK7ddd150f;rport=5060
Contact: sip:10.1.5.15:6516
To: sip:2000@10.1.5.15:6516;rinstance=c496bce22ac2cced;tag=b77a0f22
From: "asterisk"sip:asterisk@10.1.5.18;tag=as69ab3161
Call-ID: 6c6e9453429d3793796af1d74c3bff29@10.1.5.18:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘6c6e9453429d3793796af1d74c3bff29@10.1.5.18:5060’ Method: OPTIONS

<— SIP read from UDP:10.1.5.30:30744 —>

<------------->

<— SIP read from UDP:10.1.5.15:6516 —>
INVITE sip:10.1.5.18 SIP/2.0
Via: SIP/2.0/UDP 10.1.5.15:6516;branch=z9hG4bK-d8754z-f9524d167f369a10-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:2000@10.1.5.15:6516
To: "2001"sip:10.1.5.18
From: "2000"sip:2000@10.1.5.18;tag=70119075
Call-ID: NTI4ZmQ4Y2IzNTRmNjQyODM0MzQ3NjlmMTVjOTNlNzQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 255

v=0
o=- 5 2 IN IP4 10.1.5.15
s=CounterPath X-Lite 3.0
c=IN IP4 10.1.5.15
t=0 0
m=audio 38296 RTP/AVP 107 0 8 101
a=alt:1 1 : 8ZRb/BEb Zol/Z9cW 10.1.5.15 38296
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
— (12 headers 11 lines) —
Sending to 10.1.5.15:6516 (NAT)
Using INVITE request as basis request - NTI4ZmQ4Y2IzNTRmNjQyODM0MzQ3NjlmMTVjOTNlNzQ.
Found peer ‘2000’ for ‘2000’ from 10.1.5.15:6516

<— Reliably Transmitting (NAT) to 10.1.5.15:6516 —>
[color=#FF0000]SIP/2.0 401 Unauthorized[/color]Via: SIP/2.0/UDP 10.1.5.15:6516;branch=z9hG4bK-d8754z-f9524d167f369a10-1—d8754z-;received=10.1.5.15;rport=6516
From: "2000"sip:2000@10.1.5.18;tag=70119075
To: "2001"sip:10.1.5.18;tag=as3e3012e4
Call-ID: NTI4ZmQ4Y2IzNTRmNjQyODM0MzQ3NjlmMTVjOTNlNzQ.
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.11-cert1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="57d13768"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘NTI4ZmQ4Y2IzNTRmNjQyODM0MzQ3NjlmMTVjOTNlNzQ.’ in 6464 ms (Method: INVITE)

<— SIP read from UDP:10.1.5.15:6516 —>
ACK sip:10.1.5.18 SIP/2.0
Via: SIP/2.0/UDP 10.1.5.15:6516;branch=z9hG4bK-d8754z-f9524d167f369a10-1—d8754z-;rport
To: "2001"sip:10.1.5.18;tag=as3e3012e4
From: "2000"sip:2000@10.1.5.18;tag=70119075
Call-ID: NTI4ZmQ4Y2IzNTRmNjQyODM0MzQ3NjlmMTVjOTNlNzQ.
CSeq: 1 ACK
Content-Length: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from UDP:10.1.5.15:6516 —>
INVITE sip:10.1.5.18 SIP/2.0
Via: SIP/2.0/UDP 10.1.5.15:6516;branch=z9hG4bK-d8754z-ff0ff461d246174a-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:2000@10.1.5.15:6516
To: "2001"sip:10.1.5.18
From: "2000"sip:2000@10.1.5.18;tag=70119075
Call-ID: NTI4ZmQ4Y2IzNTRmNjQyODM0MzQ3NjlmMTVjOTNlNzQ.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Authorization: Digest username=“2000”,realm=“asterisk”,nonce=“57d13768”,uri=“sip:10.1.5.18”,response=“96a7ff9ac7a9cf02d58c6ab48e8870fd”,algorithm=MD5
Content-Length: 255

v=0
o=- 5 2 IN IP4 10.1.5.15
s=CounterPath X-Lite 3.0
c=IN IP4 10.1.5.15
t=0 0
m=audio 38296 RTP/AVP 107 0 8 101
a=alt:1 1 : 8ZRb/BEb Zol/Z9cW 10.1.5.15 38296
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
— (13 headers 11 lines) —
Sending to 10.1.5.15:6516 (NAT)
Using INVITE request as basis request - NTI4ZmQ4Y2IzNTRmNjQyODM0MzQ3NjlmMTVjOTNlNzQ.
Found peer ‘2000’ for ‘2000’ from 10.1.5.15:6516
== Using SIP RTP CoS mark 5
Found RTP audio format 107
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found unknown media description format BV32 for ID 107
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.1.5.15:38296
Looking for s in home (domain 10.1.5.18)

<— Reliably Transmitting (NAT) to 10.1.5.15:6516 —>
[color=#FF0000]SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.1.[/color]5.15:6516;branch=z9hG4bK-d8754z-ff0ff461d246174a-1—d8754z-;received=10.1.5.15;rport=6516
From: "2000"sip:2000@10.1.5.18;tag=70119075
To: "2001"sip:10.1.5.18;tag=as3e3012e4
Call-ID: NTI4ZmQ4Y2IzNTRmNjQyODM0MzQ3NjlmMTVjOTNlNzQ.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.11-cert1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

2

Please note that answer here will be about configuring Asterisk, not about configuring AsterisnNow.

There is no called number in the calls from X-Lite, so you will need an s extension. I’m a bit surprised that X-Lite allows hot line calling (starting a call with no digits), but I would caution against using X-Lite for any serious work. (I don’t know why there is a number in the user friendly part of the To address, but that is irrelevant to routing the call.)

Either you need to dial a number into the X-Lite, before it starts the SIP call, or you need an extension “s” in your “home” context.

You might also be able to use overlap dialing, but that is an unusual thing to do. X-Lite can send calls with dialled digits.

3

Thanks for your reply.

But its still not working. we have removed secret key from sip.conf as well as xlite.
Now sip/2.0 401 unauthorized is not comming. But sip/2.0 404 not found is still comming. I am attaching the output of sip set debug on, contents on sip.conf and extentions.conf-

Output of sip set debug on-
<— SIP read from UDP:10.1.5.30:3616 —>
INVITE sip:2707@10.1.5.21 SIP/2.0
Via: SIP/2.0/UDP 10.1.5.30:3616;branch=z9hG4bK-d87543-3119d95f5b41fb18-1–d87543-;rport
Max-Forwards: 70
Contact: sip:2707@10.1.5.30:3616
To: "2707"sip:2707@10.1.5.21
From: "2707"sip:2707@10.1.5.21;tag=795bfd09
Call-ID: M2UwZmE1ZDE2Yzk2NGY4YTk0MTMwZTE2ZGJhN2U0YjU.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 316

v=0
o=- 2 2 IN IP4 10.1.5.30
s=CounterPath X-Lite 3.0
c=IN IP4 10.1.5.30
t=0 0
m=audio 34828 RTP/AVP 107 119 0 98 8 3 101
a=alt:1 1 : +HMGPOxK fGOKwUQF 10.1.5.30 34828
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
— (12 headers 13 lines) —
Sending to 10.1.5.30:3616 (NAT)
Using INVITE request as basis request - M2UwZmE1ZDE2Yzk2NGY4YTk0MTMwZTE2ZGJhN2U0YjU.
Found peer ‘2707’ for ‘2707’ from 10.1.5.30:3616
== Using SIP RTP CoS mark 5
Found RTP audio format 107
Found RTP audio format 119
Found RTP audio format 0
Found RTP audio format 98
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found unknown media description format BV32 for ID 107
Found unknown media description format BV32-FEC for ID 119
Found audio description format iLBC for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.1.5.30:34828
Looking for 2707 in users (domain 10.1.5.21)

<— Reliably Transmitting (NAT) to 10.1.5.30:3616 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.1.5.30:3616;branch=z9hG4bK-d87543-3119d95f5b41fb18-1–d87543-;received=10.1.5.30;rport=3616
From: "2707"sip:2707@10.1.5.21;tag=795bfd09
To: "2707"sip:2707@10.1.5.21;tag=as30df23cf
Call-ID: M2UwZmE1ZDE2Yzk2NGY4YTk0MTMwZTE2ZGJhN2U0YjU.
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.11-cert1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘M2UwZmE1ZDE2Yzk2NGY4YTk0MTMwZTE2ZGJhN2U0YjU.’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:10.1.5.30:3616 —>
ACK sip:2707@10.1.5.21 SIP/2.0
Via: SIP/2.0/UDP 10.1.5.30:3616;branch=z9hG4bK-d87543-3119d95f5b41fb18-1–d87543-;rport
To: "2707"sip:2707@10.1.5.21;tag=as30df23cf
From: "2707"sip:2707@10.1.5.21;tag=795bfd09
Call-ID: M2UwZmE1ZDE2Yzk2NGY4YTk0MTMwZTE2ZGJhN2U0YjU.
CSeq: 1 ACK
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘M2UwZmE1ZDE2Yzk2NGY4YTk0MTMwZTE2ZGJhN2U0YjU.’ Method: ACK

sip.conf-

[general]
register => 2707
[2707]
Nat=never
type=friend
dtmfmode=rfc2833
context=users
regexten=2707
host=dynamic

extentions.conf-

[globals]
2707=SIP/2707
[macro-oneline]
exten => s,1,Dial(${ARG1),20,t)
exten => s,2,Voicemail(u${MACRO_EXTEN})
exten => s,3,Hangup

[users]
exten => 2707,1,Macro(oneline,${2707})

Kindly tell us where we are going wrong.