4PSA sip trunk

itche

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May 3, 2010
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#1
Hello all,

I am trying to set a 4PSA sip trunk.

I have no problem to call out using this trunk but I can't get it to work for incomming calls.
below is the settings I used for this trunk

PEER Details
host=71.125.xxx.xx
username=00xx*xxx
secret=xxxxxxxxx
type=peer

USER details
secret=xxxxxxxxx
type=user
context=from-trunk

Register String
00xx*xxx:xxxxxxx@71.125.xxx.xxx/00xx*xxx

when looking in the 4PSA server I can see that elastix has a "register" status.

Anyone has any ida on how to make this work?

Thanks
Itche
 

Patrick_elx

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#2
log in to your server
type asterisk -rvvv
then sip set debug

and try to receive a call to see what's happening.
 

itche

Joined
May 3, 2010
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#3
Hi Patrick,
Here is the output for that session
I have massked few private details
Thanks
Itche


<--- SIP read from 71.125.60.205:5060 --->
INVITE sip:15164908048@24.44.140.131 SIP/2.0
Record-Route: <sip:71.125.60.205;r2=on;lr=on;ftag=as723f857a;did=923.dfa8f802>
Record-Route: <sip:10.15.1.50;r2=on;lr=on;ftag=as723f857a;did=923.dfa8f802>
Via: SIP/2.0/UDP 71.125.60.205:5060;branch=z9hG4bKccdd.a75e7661.0
Max-Forwards: 69
To: <sip:15164908048@10.15.1.50>
From: "Cell Phone CT "<sip:12032578412@71.125.60.205>;tag=as723f857a
Call-ID: 1dbd260b162f39c436668fb90a474df5@10.15.1.50
Contact: <sip:12032578412@71.125.60.205:5060>
CSeq: 102 INVITE
User-Agent: VoipNow PBX
Date: Tue, 04 May 2010 12:58:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-voipnow-did: 15164908048
X-voipnow-extension: 00xx*xxx <-- my extn on 4PSA
X-voipnow-infrastructureid: c8497ed4ff2c
X-voipnow-did: 15164908048
Content-Type: application/sdp
Content-Length: 360

v=0
o=root 476229972 476229972 IN IP4 71.125.60.205
s=Asterisk PBX 1.6.1.4
c=IN IP4 71.125.60.205
t=0 0
m=audio 15956 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:eek:ff - - - -
a=ptime:20
a=sendrecv
a=oldmediaip:10.15.1.50
a=oldmediaip:10.15.1.50

<------------->
--- (20 headers 16 lines) ---
Sending to 71.125.60.205 : 5060 (no NAT)
Using INVITE request as basis request - 1dbd260b162f39c436668fb90a474df5@10.15.1.50
Found peer 'xxxxxxxxxxxxx' <-- my peer on 4PSA

<--- Reliably Transmitting (no NAT) to 71.125.60.205:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 71.125.60.205:5060;branch=z9hG4bKccdd.a75e7661.0;received=71.125.60.205
From: "Cell Phone CT "<sip:12032578412@71.125.60.205>;tag=as723f857a
To: <sip:15164908048@10.15.1.50>;tag=as794729fd
Call-ID: 1dbd260b162f39c436668fb90a474df5@10.15.1.50
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="795641ef"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '1dbd260b162f39c436668fb90a474df5@10.15.1.50' in 32000 ms (Method: INVITE)
tdpbx*CLI>
<--- SIP read from 71.125.60.205:5060 --->
ACK sip:15164908048@24.44.140.131 SIP/2.0
Via: SIP/2.0/UDP 71.125.60.205:5060;branch=z9hG4bKccdd.a75e7661.0
From: "Cell Phone CT "<sip:12032578412@71.125.60.205>;tag=as723f857a
Call-ID: 1dbd260b162f39c436668fb90a474df5@10.15.1.50
To: <sip:15164908048@10.15.1.50>;tag=as794729fd
CSeq: 102 ACK
Max-Forwards: 70
User-Agent: VoipNow PBX
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
tdpbx*CLI>
<--- SIP read from 71.125.60.205:5060 --->
INVITE sip:15164908048@24.44.140.131 SIP/2.0
Record-Route: <sip:71.125.60.205;r2=on;lr=on;ftag=as723f857a;did=923.efa8f802>
Record-Route: <sip:10.15.1.50;r2=on;lr=on;ftag=as723f857a;did=923.efa8f802>
Via: SIP/2.0/UDP 71.125.60.205:5060;branch=z9hG4bKdcdd.f63a4141.0
Max-Forwards: 69
To: <sip:15164908048@10.15.1.50>
From: "Cell Phone CT "<sip:12032578412@71.125.60.205>;tag=as723f857a
Call-ID: 1dbd260b162f39c436668fb90a474df5@10.15.1.50
Contact: <sip:12032578412@71.125.60.205:5060>
CSeq: 103 INVITE
User-Agent: VoipNow PBX
Proxy-Authorization: Digest username="a2dc4316c1", realm="asterisk", algorithm=MD5, uri="sip:0051*001@10.15.1.50", <-- my extn on 4PSA (itche)
nonce="795641ef", response="927a25c7f9259e8ee811cc8089ccbbcf"
Date: Tue, 04 May 2010 12:58:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-voipnow-did: 15164908048
X-voipnow-extension: 00xx*xxx <-- my extn on 4PSA
X-voipnow-infrastructureid: c8497ed4ff2c
X-voipnow-did: 15164908048
Content-Type: application/sdp
Content-Length: 360

v=0
o=root 476229972 476229973 IN IP4 71.125.60.205
s=Asterisk PBX 1.6.1.4
c=IN IP4 71.125.60.205
t=0 0
m=audio 15956 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:eek:ff - - - -
a=ptime:20
a=sendrecv
a=oldmediaip:10.15.1.50
a=oldmediaip:10.15.1.50

<------------->
--- (21 headers 16 lines) ---
Sending to 71.125.60.205 : 5060 (no NAT)
Using INVITE request as basis request - 1dbd260b162f39c436668fb90a474df5@10.15.1.50
Found peer 'xxxxxxxxxxxxxx' <-- my PEER on 4PSA (itche)


<--- Reliably Transmitting (no NAT) to 71.125.60.205:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 71.125.60.205:5060;branch=z9hG4bKdcdd.f63a4141.0;received=71.125.60.205
From: "Cell Phone CT "<sip:12032578412@71.125.60.205>;tag=as723f857a
To: <sip:15164908048@10.15.1.50>;tag=as794729fd
Call-ID: 1dbd260b162f39c436668fb90a474df5@10.15.1.50
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1dbd260b162f39c436668fb90a474df5@10.15.1.50' in 32000 ms (Method: INVITE)
Really destroying SIP dialog '23a261473b610b2e10dd17972518b7a0@24.44.140.131' Method: REGISTER
tdpbx*CLI>
<--- SIP read from 71.125.60.205:5060 --->
ACK sip:15164908048@24.44.140.131 SIP/2.0
Via: SIP/2.0/UDP 71.125.60.205:5060;branch=z9hG4bKdcdd.f63a4141.0
From: "Cell Phone CT "<sip:12032578412@71.125.60.205>;tag=as723f857a
Call-ID: 1dbd260b162f39c436668fb90a474df5@10.15.1.50
To: <sip:15164908048@10.15.1.50>;tag=as794729fd
CSeq: 103 ACK
Max-Forwards: 70
User-Agent: VoipNow PBX
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
tdpbx*CLI>
<--- SIP read from 71.125.60.205:5060 --->
OPTIONS sip:00xx*xxx@24.44.140.131 SIP/2.0 <-- my extn on 4PSA
Via: SIP/2.0/UDP 10.15.1.50:5060;branch=z9hG4bK2083.0.0
From: sip:voipnow@10.15.1.50;tag=032b5f27
To: sip:00xx*xxx@24.44.140.131 <-- my extn on 4PSA (itche)
Call-ID: de20f9b6-56878d17-4ba41@10.15.1.50
CSeq: 1 OPTIONS
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
Looking for 00xx*xxx in from-sip-external (domain 24.44.140.131) <-- my extn on 4PSA (itche)

<--- Transmitting (no NAT) to 71.125.60.205:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.15.1.50:5060;branch=z9hG4bK2083.0.0;received=71.125.60.205
From: sip:voipnow@10.15.1.50;tag=032b5f27
To: sip:00xx*xxx@24.44.140.131;tag=as3ab2542b <-- my extn on 4PSA (itche)

Call-ID: de20f9b6-56878d17-4ba41@10.15.1.50
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:24.44.140.131>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'de20f9b6-56878d17-4ba41@10.15.1.50' in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog '10387e40-8401a8c0-13c4-4be019cc-16bdafc6-4be019cc' Method: REGISTER
 

giany

Joined
May 17, 2010
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#4
What version of VoipNow are you using? If its 2.X then it means VoipNow sip proxy denies connections by giving this message :"SIP/2.0 403 Forbidden". Edit /etc/kamailio/kamailio.cfg and change from
debug=1
to
debug=2

After that restart kamailio and check /var/log/kamailio.cfg
 

itche

Joined
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#5
Hi all,
Problem solved.:)
All I needed to do was to remove the username and secret fields from the PEER and the user's context.

wireshark helped alot

Thanks for your help
Itche
 

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