Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[asterisk-users] Outside call not coming through


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
ichverstehe at gmail.com
Guest
PostPosted: Sat Apr 26, 2008 9:53 am    Post subject: [asterisk-users] Outside call not coming through Reply with quote
When i try to call '36946811' from the outside the call gets through,
but is rejected and the sound file is not played, this is my conf and
sip debug output:

## sip.conf
[general]
context=incoming
register => 36946811:L0sebitch at musimi.dk/1234
port=5060
bindaddr=0.0.0.0
srvlookup=yes

## extensions.conf
[incoming]
exten => 36946811,1,Background(hello-world)

## sip debug
*CLI>
<--- SIP read from 87.54.25.114:5060 --->
INVITE sip:1234 at 67.207.147.205 SIP/2.0
Record-Route: <sip:87.54.25.114;ftag=688c7f1d;lr=on>
Via: SIP/2.0/UDP 87.54.25.114;branch=z9hG4bK70b1.34ff9f57.0
Via: SIP/2.0/UDP
192.168.2.5:5060;received=62.107.1.48;branch=z9hG4bK-d8754z-c59c0dcc191940e4-1---d8754z-;rport=5060
Max-Forwards: 16
Contact: <sip:36946731 at 62.107.1.48:5060>
To: <sip:36946811 at musimi.dk>
From: "Harry"<sip:36946731 at musimi.dk>;transport=UDP;tag=688c7f1d
Call-ID: NTRmMjg4ZGFlZDU2MDUxOTAxZjQ0OGNiYTFiOTMxOGQ.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Type: application/sdp
User-Agent: Zoiper rev.417
Content-Length: 311

v=0
o=Z 0 0 IN IP4 192.168.2.5
s=Z
c=IN IP4 192.168.2.5
t=0 0
m=audio 8000 RTP/AVP 3 110 97 8 0 101
a=fmtp:97 mode=30
a=fmtp:101 0-15
a=rtpmap:3 GSM/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=direction:active

<------------->
--- (14 headers 15 lines) ---
Sending to 87.54.25.114 : 5060 (no NAT)
Using INVITE request as basis request -
NTRmMjg4ZGFlZDU2MDUxOTAxZjQ0OGNiYTFiOTMxOGQ.
Found peer 'musimi'

<--- Reliably Transmitting (NAT) to 87.54.25.114:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
87.54.25.114;branch=z9hG4bK70b1.34ff9f57.0;received=87.54.25.114
Via: SIP/2.0/UDP
192.168.2.5:5060;received=62.107.1.48;branch=z9hG4bK-d8754z-c59c0dcc191940e4-1---d8754z-;rport=5060
From: "Harry"<sip:36946731 at musimi.dk>;transport=UDP;tag=688c7f1d
To: <sip:36946811 at musimi.dk>;tag=as0f99b309
Call-ID: NTRmMjg4ZGFlZDU2MDUxOTAxZjQ0OGNiYTFiOTMxOGQ.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="35c07307"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'NTRmMjg4ZGFlZDU2MDUxOTAxZjQ0OGNiYTFiOTMxOGQ.' in 32000 ms (Method:
INVITE)

<--- SIP read from 87.54.25.114:5060 --->
ACK sip:1234 at 67.207.147.205 SIP/2.0
Via: SIP/2.0/UDP 87.54.25.114;branch=z9hG4bK70b1.34ff9f57.0
From: "Harry"<sip:36946731 at musimi.dk>;transport=UDP;tag=688c7f1d
Call-ID: NTRmMjg4ZGFlZDU2MDUxOTAxZjQ0OGNiYTFiOTMxOGQ.
To: <sip:36946811 at musimi.dk>;tag=as0f99b309
CSeq: 2 ACK
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
Reliably Transmitting (NAT) to 62.107.1.48:5060:
OPTIONS sip:lolz at 192.168.2.5:5060;rinstance=d815b062f3a40a5e SIP/2.0
Via: SIP/2.0/UDP 67.207.147.205:5060;branch=z9hG4bK1945b111;rport
From: "asterisk" <sip:asterisk at 67.207.147.205>;tag=as00c1a604
To: <sip:lolz at 192.168.2.5:5060;rinstance=d815b062f3a40a5e>
Contact: <sip:asterisk at 67.207.147.205>
Call-ID: 4e2f2293727eb63c6d175d8726450a84 at 67.207.147.205
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 26 Apr 2008 14:46:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---

<--- SIP read from 62.107.1.48:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.207.147.205:5060;branch=z9hG4bK1945b111;rport=5060
Contact: <sip:192.168.2.5:5060>
To: <sip:lolz at 192.168.2.5:5060;rinstance=d815b062f3a40a5e>;tag=d4846e53
From: "asterisk"<sip:asterisk at 67.207.147.205>;tag=as00c1a604
Call-ID: 4e2f2293727eb63c6d175d8726450a84 at 67.207.147.205
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS
User-Agent: Zoiper rev.417
Allow-Events: message-summary
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog
'4e2f2293727eb63c6d175d8726450a84 at 67.207.147.205' Method: OPTIONS
Back to top
ichverstehe at gmail.com
Guest
PostPosted: Sat Apr 26, 2008 10:51 am    Post subject: [asterisk-users] Outside call not coming through Reply with quote
Screwed up really bad. This is the correct config and sip debug:

## sip.conf
[general]
context=incoming
register => 36946811:L0sebitch at musimi.dk/1234
port=5060
bindaddr=0.0.0.0
srvlookup=yes

## extensions.conf
[incoming]
exten => _X.,Background(hello-world)

## sip debug (updated)
<--- SIP read from 87.54.25.114:5060 --->
INVITE sip:1234 at 67.207.147.205 SIP/2.0
Record-Route: <sip:87.54.25.114;ftag=as2dae750f;lr=on>
Via: SIP/2.0/UDP 87.54.25.114;branch=z9hG4bK9588.5b1985c.0
Via: SIP/2.0/UDP 87.54.25.116:5060;branch=z9hG4bK3fdd7e37;rport=5060
From: "23864098" <sip:23864098 at 87.54.25.116>;tag=as2dae750f
To: <sip:36946811 at musimi.dk>
Contact: <sip:23864098 at 87.54.25.116>
Call-ID: 1204dee71075a8f97e6a598a4095e283 at 87.54.25.116
CSeq: 102 INVITE
User-Agent: no
Max-Forwards: 16
Date: Sat, 26 Apr 2008 15:44:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 215

v=0
o=root 19760 19760 IN IP4 87.54.25.116
s=session
c=IN IP4 87.54.25.116
t=0 0
m=audio 11114 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

<------------->
--- (15 headers 10 lines) ---
Sending to 87.54.25.114 : 5060 (no NAT)
Using INVITE request as basis request -
1204dee71075a8f97e6a598a4095e283 at 87.54.25.116
Found peer 'musimi'

<--- Reliably Transmitting (NAT) to 87.54.25.114:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 87.54.25.114;branch=z9hG4bK9588.5b1985c.0;received=87.54.25.114
Via: SIP/2.0/UDP 87.54.25.116:5060;branch=z9hG4bK3fdd7e37;rport=5060
From: "23864098" <sip:23864098 at 87.54.25.116>;tag=as2dae750f
To: <sip:36946811 at musimi.dk>;tag=as3fc8c57f
Call-ID: 1204dee71075a8f97e6a598a4095e283 at 87.54.25.116
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5dc98c57"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'1204dee71075a8f97e6a598a4095e283 at 87.54.25.116' in 32000 ms (Method:
INVITE)

<--- SIP read from 87.54.25.114:5060 --->
ACK sip:1234 at 67.207.147.205 SIP/2.0
Via: SIP/2.0/UDP 87.54.25.114;branch=z9hG4bK9588.5b1985c.0
From: "23864098" <sip:23864098 at 87.54.25.116>;tag=as2dae750f
Call-ID: 1204dee71075a8f97e6a598a4095e283 at 87.54.25.116
To: <sip:36946811 at musimi.dk>;tag=as3fc8c57f
CSeq: 102 ACK
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services