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San Singhania
2004-Sep-05 10:35 UTC
[Asterisk-Users] iconnect and Asterisk

Hello All,

I have gone thru all the resources I could find on google on asterisk + iconnect
and managed to get outgoing calls working. However,
I cannot get incoming calls to work at all. With the sip debug on, I can see
that something is happening everytime a call is received
from iconnecthere, but I get an invalid tone on the caller side. The call never
rings anywhere on the asterisk. Would appreciate any
help on this. Thanks


Below is my sip file

register=442087926805:somepassword@sipauth.deltathree.com:5060

[iconnecthere]
type=friend
secret=somepassword
username=11232634
host=sipauth.deltathree.com
canreinvite=no
;nat=yes
context=default
;dtmfmode=inband
disallow=all
;allow=all
allow=gsm
allow=ulaw
allow=alaw
allow=g726
allow=g723

This is the sip debug info when a call comes in from iconnecthere :

11 headers, 0 lines
Reliably Transmitting:
REGISTER sip:sipauth.deltathree.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK35628ab9
From: <sip:442087926805@sipauth.deltathree.com>;tag=as5c70755c
To: <sip:442087926805@sipauth.deltathree.com>
Call-ID: 6ed54db642def5322c30b4434b737f76@127.0.0.1
CSeq: 104 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:s@192.168.1.250>
Event: registration
Content-Length: 0

 (no NAT) to 213.137.73.140:5060
localhost*CLI>

Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK35628ab9
To: <sip:442087926805@sipauth.deltathree.com>
From: <sip:442087926805@sipauth.deltathree.com>;tag=as5c70755c
Call-ID: 6ed54db642def5322c30b4434b737f76@127.0.0.1
CSeq: 104 REGISTER
Content-Length: 0


7 headers, 0 lines
localhost*CLI>

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK35628ab9
From: <sip:442087926805@sipauth.deltathree.com>;tag=as5c70755c
To: <sip:442087926805@sipauth.deltathree.com>
Call-ID: 6ed54db642def5322c30b4434b737f76@127.0.0.1
CSeq: 104 REGISTER
Contact:
<sip:442087926805_202_166_50_122_5060_192_168_1_250_5060@213.137.73.173
:5060>;expires=120
Contact:
<sip:442087926805_202_166_50_122_5060_192_168_1_250_5060@213.137.73.174
:5060>;expires=14
Expires: 120
Content-Length: 0


10 headers, 0 lines
Destroying call '6ed54db642def5322c30b4434b737f76@127.0.0.1'
localhost*CLI>

Sip read:
INVITE sip:442087926805@202.166.50.122:5060 SIP/2.0
Via: SIP/2.0/UDP 213.137.73.140:5060;maddr=213.137.73.173
Via: SIP/2.0/UDP 213.137.73.176:5060;branch=34550e33-69d4c647-76eb3474-c49105c4-
1
Via: SIP/2.0/UDP 213.137.81.27:5060;received=213.137.81.27
To: <sip:442087926805@213.137.73.179>
From: <sip:44006597471958@213.137.81.27>;tag=DF81964C-1341
Call-ID: DF214E6E-FE9511D8-9BD8BD4E-2B0684DF@213.137.81.27
CSeq: 101 INVITE
Contact: <sip:44006597471958@213.137.81.27:5060>
Record-Route: <sip:442087926805@213.137.73.140:5060;maddr=213.137.73.173>
Record-Route:
<sip:44006597471958.34550e33-69d4c647-76eb3474-c49105c4@213.137.81
.27:5060;maddr=213.137.73.176>
Content-Type: application/sdp
Content-Length: 146

v=0
o=CiscoSystemsSIP-GW-UserAgent 5851 2446 IN IP4 213.137.81.27
s=SIP Call
c=IN IP4 213.137.81.27
t=0 0
m=audio 18958 RTP/AVP 4 0 8 2 101

13 headers, 6 lines
Using latest request as basis request
Sending to 213.137.73.140 : 5060 (non-NAT)
Found RTP audio format 4
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 2
Found RTP audio format 101
Peer audio RTP is at port 213.137.81.27:18958
Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x1d(G723|ULAW|ALAW
|G726)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
Found peer 'iconnecthere'
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 213.137.73.140:5060;maddr=213.137.73.173
Via: SIP/2.0/UDP 213.137.73.176:5060;branch=34550e33-69d4c647-76eb3474-c49105c4-
1
Via: SIP/2.0/UDP 213.137.81.27:5060;received=213.137.81.27
From: <sip:44006597471958@213.137.81.27>;tag=DF81964C-1341
To: <sip:442087926805@213.137.73.179>;tag=as34968f1d
Call-ID: DF214E6E-FE9511D8-9BD8BD4E-2B0684DF@213.137.81.27
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:442087926805@192.168.1.250>
Proxy-Authenticate: Digest realm="asterisk",
nonce="252c7e0a"
Content-Length: 0


 to 213.137.73.140:5060
Scheduling destruction of call
'DF214E6E-FE9511D8-9BD8BD4E-2B0684DF@213.137.81.2
7' in 15000 ms
localhost*CLI>

Sip read:
ACK sip:442087926805@202.166.50.122:5060 SIP/2.0
Via: SIP/2.0/UDP 213.137.73.140:5060;maddr=213.137.73.173
Via: SIP/2.0/UDP 213.137.73.176:5060;branch=34550e33-69d4c647-76eb3474-c49105c4-
1
From: <sip:44006597471958@213.137.81.27>;tag=DF81964C-1341
To: <sip:442087926805@213.137.73.179>;tag=as34968f1d
Call-ID: DF214E6E-FE9511D8-9BD8BD4E-2B0684DF@213.137.81.27
CSeq: 101 ACK
Content-Length: 0


8 headers, 0 lines
localhost*CLI>

Sip read:
REGISTER sip:192.168.1.250 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.60;branch=z9hG4bKda87dee87d2b42ca
From: <sip:ext100@192.168.1.250>;tag=d96a1d9a3a8eb4af
To: <sip:ext100@192.168.1.250>
Contact: <sip:ext100@192.168.1.60>
Call-ID: d1b18f3c3621e97d@192.168.1.60
CSeq: 402 REGISTER
Expires: 120
User-Agent: Grandstream BT100 1.0.4.67
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0


12 headers, 0 lines
Using latest request as basis request
Sending to 192.168.1.60 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.60;branch=z9hG4bKda87dee87d2b42ca
From: <sip:ext100@192.168.1.250>;tag=d96a1d9a3a8eb4af
To: <sip:ext100@192.168.1.250>;tag=as5926604e
Call-ID: d1b18f3c3621e97d@192.168.1.60
CSeq: 402 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:ext100@192.168.1.250>
Content-Length: 0


 to 192.168.1.60:5060
Transmitting (no NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.60;branch=z9hG4bKda87dee87d2b42ca
From: <sip:ext100@192.168.1.250>;tag=d96a1d9a3a8eb4af
To: <sip:ext100@192.168.1.250>;tag=as5926604e
Call-ID: d1b18f3c3621e97d@192.168.1.60
CSeq: 402 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:ext100@192.168.1.250>
WWW-Authenticate: Digest realm="asterisk", nonce="007140b3"
Content-Length: 0


 to 192.168.1.60:5060
Scheduling destruction of call 'd1b18f3c3621e97d@192.168.1.60' in 15000
ms
localhost*CLI>

Sip read:
REGISTER sip:192.168.1.250 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.60;branch=z9hG4bK9356e31bb2078147
From: <sip:ext100@192.168.1.250>;tag=d96a1d9a3a8eb4af
To: <sip:ext100@192.168.1.250>
Contact: <sip:ext100@192.168.1.60>
Authorization: DIGEST username="ext100", realm="asterisk",
algorithm=MD5, uri="s
ip:192.168.1.250", nonce="007140b3",
response="5d56be19a6b63ed92390724df782f89a"
Call-ID: d1b18f3c3621e97d@192.168.1.60
CSeq: 403 REGISTER
Expires: 120
User-Agent: Grandstream BT100 1.0.4.67
ax-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0


13 headers, 0 lines
Using latest request as basis request
Sending to 192.168.1.60 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.60;branch=z9hG4bK9356e31bb2078147
From: <sip:ext100@192.168.1.250>;tag=d96a1d9a3a8eb4af
To: <sip:ext100@192.168.1.250>;tag=as5926604e
Call-ID: d1b18f3c3621e97d@192.168.1.60
CSeq: 403 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:ext100@192.168.1.250>
Content-Length: 0


 to 192.168.1.60:5060
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.60;branch=z9hG4bK9356e31bb2078147
From: <sip:ext100@192.168.1.250>;tag=d96a1d9a3a8eb4af
To: <sip:ext100@192.168.1.250>;tag=as5926604e
Call-ID: d1b18f3c3621e97d@192.168.1.60
CSeq: 403 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 120
Contact: <sip:ext100@192.168.1.60>;expires=120
Date: Sun, 05 Sep 2004 17:13:34 GMT
Content-Length: 0


 to 192.168.1.60:5060
Scheduling destruction of call 'd1b18f3c3621e97d@192.168.1.60' in 15000
ms
11 headers, 2 lines
Reliably Transmitting:
NOTIFY sip:ext100@192.168.1.60 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK047cfd59
From: "asterisk" <sip:asterisk@192.168.1.250>;tag=as4757cd3d
To: <sip:ext100@192.168.1.60>
Contact: <sip:asterisk@192.168.1.250>
Call-ID: 3439dd52388de28e0a998ca671a58836@192.168.1.250
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 36

Messages-Waiting: no
Voicemail: 0/0
 (no NAT) to 192.168.1.60:5060
Scheduling destruction of call
'3439dd52388de28e0a998ca671a58836@192.168.1.250'
in 15000 ms
localhost*CLI>

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK047cfd59
From: "asterisk" <sip:asterisk@192.168.1.250>;tag=as4757cd3d
To: <sip:ext100@192.168.1.60>;tag=bcd972b14ed2943b
Call-ID: 3439dd52388de28e0a998ca671a58836@192.168.1.250
CSeq: 102 NOTIFY
User-Agent: Grandstream BT100 1.0.4.67
Contact: <sip:ext100@192.168.1.60>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0


10 headers, 0 lines
Destroying call '3439dd52388de28e0a998ca671a58836@192.168.1.250'
Destroying call 'DF214E6E-FE9511D8-9BD8BD4E-2B0684DF@213.137.81.27'
Destroying call 'd1b18f3c3621e97d@192.168.1.60'


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asterisk users - Sep 2004 - iconnect and Asterisk

[Asterisk-Users] iconnect and Asterisk

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