thr3ads.net - asterisk users - [Asterisk-Users] Incoming SIP getting in, but not ringing. [Oct 2005]

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Paul Goodyear

2005-Oct-10 09:13 UTC

[Asterisk-Users] Incoming SIP getting in, but not ringing.

Hi all.

Just as a quote note, can I thank everyone on this list. I find my
self finding pretty much every answer I am looking for on here. And a
big thanks to all thoughs helping us out. Mass Respect :)

Ok, I'm using a SIP provider (SipGate UK) to do my international
dialing etc, working great from extension 8 on phones. However some
more friends/contacts have started using SipGate also, and I want to
be able to do some SipGate to SipGate calls. As I said I can dial out
on SipGate no issues, but I cannot get my *@home box to receive
SipGate calls.

I have attached a text file with the "sip debug" option for a full
log. requests are coming in from SipGates server etc but my asterisk
box is not transfering the calls to the phones.

I have the register string in my sip.conf as so:

register=6698221:(MYSECRET)@sipgate.co.uk/6698221

Port on my IPCOP box as follows:

UDP/5060
UDP/10000:20000
UDP/8000:8012
UDP-TCP/3478

Thanks for your time.

Paul.
-------------- next part --------------
Sip read: 
INVITE sip:6698221@MY_ISP_IP:5060 SIP/2.0
Record-Route: <sip:6698221@217.10.79.219;ftag=as6a04ebdf;lr=on>
Max-Forwards:  9
Record-Route: <sip:6698221@217.10.79.8;ftag=as6a04ebdf;lr=on>
Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bKeafc.0df1fb45.0
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKeafc.dd5c7802.0
Via: SIP/2.0/UDP 217.10.79.218:5060;branch=z9hG4bK66be0c1c
From: "07976xxxxxx"
<sip:07976408760@gw02.uk.sipgate.net>;tag=as6a04ebdf
To: <sip:6698221@217.10.79.8>
Contact: <sip:07976408760@217.10.79.218>
Call-ID: 2e2350312ec6f4f457582f1530208c69@gw02.uk.sipgate.net
CSeq: 102 INVITE
User-Agent: sipgate asterisk
Date: Mon, 10 Oct 2005 15:53:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 448

v=0
o=root 5903 5903 IN IP4 217.10.79.218
s=session
c=IN IP4 217.10.79.55
t=0 0
m=audio 44214 RTP/AVP 8 0 3 97 18 2 4 5 110 7 10
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:110 speex/8000
a=rtpmap:7 LPC/8000
a=rtpmap:10 L16/8000
a=silenceSupp:off - - - -
a=direction:active
a=nortpproxy:yes

17 headers, 20 lines
Using latest request as basis request
Sending to 217.10.79.219 : 5060 (non-NAT)
Found peer 'SipGate'
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bKeafc.0df1fb45.0
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKeafc.dd5c7802.0
Via: SIP/2.0/UDP 217.10.79.218:5060;branch=z9hG4bK66be0c1c
From: "07976xxxxxx"
<sip:07976xxxxxx@gw02.uk.sipgate.net>;tag=as6a04ebdf
To: <sip:6698221@217.10.79.8>;tag=as60d08779
Call-ID: 2e2350312ec6f4f457582f1530208c69@gw02.uk.sipgate.net
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:6698221@192.168.1.100>
Proxy-Authenticate: Digest realm="asterisk",
nonce="557d3579"
Content-Length: 0


 to 217.10.79.219:5060
Scheduling destruction of call
'2e2350312ec6f4f457582f1530208c69@gw02.uk.sipgate.net' in 15000 ms
asterisk1*CLI> 

Sip read: 
ACK sip:6698221@MY_ISP_IP:5060 SIP/2.0
Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bKeafc.0df1fb45.0
From: "07976xxxxxx"
<sip:07976xxxxxx@gw02.uk.sipgate.net>;tag=as6a04ebdf
Call-ID: 2e2350312ec6f4f457582f1530208c69@gw02.uk.sipgate.net
To: <sip:6698221@217.10.79.8>;tag=as60d08779
CSeq: 102 ACK
User-Agent: sipgate ser
Content-Length: 0


8 headers, 0 lines
asterisk1*CLI> 

Sip read: 
INVITE sip:6698221@MY_ISP_IP:5060 SIP/2.0
Record-Route: <sip:6698221@217.10.79.219;ftag=as6a04ebdf;lr=on>
Max-Forwards:  9
Record-Route: <sip:6698221@217.10.79.8;ftag=as6a04ebdf;lr=on>
Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bKfafc.4aae3986.0
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKfafc.a85e7d75.0
Via: SIP/2.0/UDP 217.10.79.218:5060;branch=z9hG4bK49e9a0ad
From: "07976xxxxxx"
<sip:07976xxxxxx@gw02.uk.sipgate.net>;tag=as6a04ebdf
To: <sip:6698221@217.10.79.8>
Contact: <sip:07976xxxxxx@217.10.79.218>
Call-ID: 2e2350312ec6f4f457582f1530208c69@gw02.uk.sipgate.net
CSeq: 103 INVITE
User-Agent: sipgate asterisk
Date: Mon, 10 Oct 2005 15:53:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 448

v=0
o=root 5903 5904 IN IP4 217.10.79.218
s=session
c=IN IP4 217.10.79.55
t=0 0
m=audio 44214 RTP/AVP 8 0 3 97 18 2 4 5 110 7 10
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:110 speex/8000
a=rtpmap:7 LPC/8000
a=rtpmap:10 L16/8000
a=silenceSupp:off - - - -
a=direction:active
a=nortpproxy:yes

17 headers, 20 lines
Using latest request as basis request
Sending to 217.10.79.219 : 5060 (non-NAT)
Found peer 'SipGate'
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bKfafc.4aae3986.0
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKfafc.a85e7d75.0
Via: SIP/2.0/UDP 217.10.79.218:5060;branch=z9hG4bK49e9a0ad
From: "07976xxxxxx"
<sip:07976xxxxxx@gw02.uk.sipgate.net>;tag=as6a04ebdf
To: <sip:6698221@217.10.79.8>;tag=as60d08779
Call-ID: 2e2350312ec6f4f457582f1530208c69@gw02.uk.sipgate.net
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:6698221@192.168.1.100>
Proxy-Authenticate: Digest realm="asterisk",
nonce="70112d01"
Content-Length: 0


 to 217.10.79.219:5060
Scheduling destruction of call
'2e2350312ec6f4f457582f1530208c69@gw02.uk.sipgate.net' in 15000 ms
asterisk1*CLI> 

Sip read: 
ACK sip:6698221@MY_ISP_IP:5060 SIP/2.0
Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bKfafc.4aae3986.0
From: "07976xxxxxx"
<sip:07976xxxxxx@gw02.uk.sipgate.net>;tag=as6a04ebdf
Call-ID: 2e2350312ec6f4f457582f1530208c69@gw02.uk.sipgate.net
To: <sip:6698221@217.10.79.8>;tag=as60d08779
CSeq: 103 ACK
User-Agent: sipgate ser
Content-Length: 0


8 headers, 0 lines
asterisk1*CLI> 

Sip read: 
INVITE sip:6698221@MY_ISP_IP:5060 SIP/2.0
Record-Route: <sip:6698221@217.10.79.219;ftag=as6a04ebdf;lr=on>
Max-Forwards:  9
Record-Route: <sip:6698221@217.10.79.8;ftag=as6a04ebdf;lr=on>
Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bKcafc.67c7747.0
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKcafc.e45dd317.0
Via: SIP/2.0/UDP 217.10.79.218:5060;branch=z9hG4bK1da22991
From: "07976xxxxxx"
<sip:07976xxxxxx@gw02.uk.sipgate.net>;tag=as6a04ebdf
To: <sip:6698221@217.10.79.8>
Contact: <sip:07976xxxxxx@217.10.79.218>
Call-ID: 2e2350312ec6f4f457582f1530208c69@gw02.uk.sipgate.net
CSeq: 104 INVITE
User-Agent: sipgate asterisk
Date: Mon, 10 Oct 2005 15:53:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 448

v=0
o=root 5903 5905 IN IP4 217.10.79.218
s=session
c=IN IP4 217.10.79.55
t=0 0
m=audio 44214 RTP/AVP 8 0 3 97 18 2 4 5 110 7 10
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:110 speex/8000
a=rtpmap:7 LPC/8000
a=rtpmap:10 L16/8000
a=silenceSupp:off - - - -
a=direction:active
a=nortpproxy:yes

17 headers, 20 lines
Using latest request as basis request
Sending to 217.10.79.219 : 5060 (non-NAT)
Found peer 'SipGate'
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bKcafc.67c7747.0
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKcafc.e45dd317.0
Via: SIP/2.0/UDP 217.10.79.218:5060;branch=z9hG4bK1da22991
From: "07976xxxxxx"
<sip:07976xxxxxx@gw02.uk.sipgate.net>;tag=as6a04ebdf
To: <sip:6698221@217.10.79.8>;tag=as60d08779
Call-ID: 2e2350312ec6f4f457582f1530208c69@gw02.uk.sipgate.net
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:6698221@192.168.1.100>
Proxy-Authenticate: Digest realm="asterisk",
nonce="3d01fb4f"
Content-Length: 0


 to 217.10.79.219:5060
Scheduling destruction of call
'2e2350312ec6f4f457582f1530208c69@gw02.uk.sipgate.net' in 15000 ms
asterisk1*CLI> 

Sip read: 
ACK sip:6698221@MY_ISP_IP:5060 SIP/2.0
Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bKcafc.67c7747.0
From: "07976xxxxxx"
<sip:07976xxxxxx@gw02.uk.sipgate.net>;tag=as6a04ebdf
Call-ID: 2e2350312ec6f4f457582f1530208c69@gw02.uk.sipgate.net
To: <sip:6698221@217.10.79.8>;tag=as60d08779
CSeq: 104 ACK
User-Agent: sipgate ser
Content-Length: 0

Paul Goodyear

2005-Oct-11 02:53 UTC

head link

[Asterisk-Users] Re: Incoming SIP getting in, but not ringing.

FIXED,

FYI:

Turning the secret OFF in the incoming context fixed the problem.


On 10/10/05, Paul Goodyear <pgudge@gmail.com>
wrote:> Hi all.
>
> Just as a quote note, can I thank everyone on this list. I find my
> self finding pretty much every answer I am looking for on here. And a
> big thanks to all thoughs helping us out. Mass Respect :)
>
> Ok, I'm using a SIP provider (SipGate UK) to do my international
> dialing etc, working great from extension 8 on phones. However some
> more friends/contacts have started using SipGate also, and I want to
> be able to do some SipGate to SipGate calls. As I said I can dial out
> on SipGate no issues, but I cannot get my *@home box to receive
> SipGate calls.
>
> I have attached a text file with the "sip debug" option for a
full
> log. requests are coming in from SipGates server etc but my asterisk
> box is not transfering the calls to the phones.
>
> I have the register string in my sip.conf as so:
>
> register=6698221:(MYSECRET)@sipgate.co.uk/6698221
>
> Port on my IPCOP box as follows:
>
> UDP/5060
> UDP/10000:20000
> UDP/8000:8012
> UDP-TCP/3478
>
> Thanks for your time.
>
> Paul.
>
>
>

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asterisk users - Oct 2005 - Incoming SIP getting in, but not ringing.

[Asterisk-Users] Incoming SIP getting in, but not ringing.

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