BYE not reaching UAC

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BYE not reaching UAC

Zoho Junk
Greetings,

I have OpenSips installed on an Amazon EC2 with an elastic IP (54.242.85.140). All non REGISTER methods are passed through rewritehostport("mytwiliodomain.sip.twilio.com") to another SIP proxy.

Everything is working except when BYE is sent from a callee that is registered to my OpenSips server. The caller never receives the BYE.
I've changed my config file at least 100 times and still have no success. Please let me know if I need to include anything from my config.

My First Invite:


U 2013/12/20 12:44:21.298060 107.21.231.147:5060 -> 10.167.11.171:5060
INVITE sip:[hidden email] SIP/2.0.
Record-Route: <sip:107.21.231.147:5060;lr;ftag=31945052_6772d868_3b1eb7f5-c41b-4da5-9a97-c756d7cd8fe3>.
From: "17175554895" <sip:[hidden email]>;tag=31945052_6772d868_3b1eb7f5-c41b-4da5-9a97-c756d7cd8fe3.
To: <sip:[hidden email]>.
CSeq: 102 INVITE.
Max-Forwards: 68.
Date: Fri, 20 Dec 2013 17:42:59 GMT.
Call-ID: 6fa66f8ea2ce4b2dae83149f8b34b2e2@0.0.0.0.
Contact: "17175554895" <sip:17175554895@10.70.227.82:5060;transport=udp>.
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE.
User-Agent: Twilio Gateway.
X-Twilio-ApiVersion: 2010-04-01.
X-Twilio-AccountSid: AC7ae5d84004d4269fgfhfhfjghgjghgbb.
Content-Type: application/sdp.
X-Twilio-CallSid: CA79e7a3c2c1c418af0777467467474747.
Via: SIP/2.0/UDP 107.21.231.147:5060;branch=z9hG4bKaeaf.02902511.0.
Via: SIP/2.0/UDP 10.70.227.82:5060;branch=z9hG4bK3b1eb7f5-c41b-4da5-9a97-c756d7cd8fe3_6772d868_425297635397116.
Content-Length: 251.
.
v=0.
o=- 1274389856 1274389856 IN IP4 54.196.144.208.
s=session.
c=IN IP4 54.196.144.208.
t=0 0.
m=audio 18800 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.


My First Bye:

U 2013/12/20 12:44:24.194615 74.106.255.63:5907 -> 10.167.11.171:5060
BYE sip:17175554895@107.21.231.147:5060;transport=udp;nat=yes SIP/2.0.
Via: SIP/2.0/UDP 74.106.255.63:5907;branch=z9hG4bK-41214dcd.
From: <sip:[hidden email]>;tag=5ed7c43daadb6f7fi0.
To: "17175554895" <sip:[hidden email]>;tag=31945052_6772d868_3b1eb7f5-c41b-4da5-9a97-c756d7cd8fe3.
Call-ID:
6fa66f8ea2ce4b2dae83149f8b34b2e2@0.0.0.0.
CSeq: 101 BYE.
Max-Forwards: 70.
Route: <sip:700@54.204.19.238:5060;lr54.204.19.238;did=b1c.e6603363>, <sip:107.21.231.147:5060;lr;ftag=31945052_6772d868_3b1eb7f5-c41b-4da5-9a97-c756d7cd8fe3>.
User-Agent: Linksys/SPA942-6.1.3(a).
Content-Length: 0.


My Debug Log:
http://pastebin.com/0pAAZ68J

NGrep:
http://pastebin.com/TVYUmM58

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Re: BYE not reaching UAC

Laszlo
This one looks interesting:

Route: <sip:700@54.204.19.238:5060;lr54.204.19.238;did=b1c.e6603363>, <sip:107.21.231.147:5060;lr;ftag=31945052_6772d868_3b1eb7f5-c41b-4da5-9a97-c756d7cd8fe3>.

The lr54.204.19.238 part.

And this is in your trace when your opensips is sending out the invite:

U 2013/12/20 12:44:21.301468 10.167.11.171:5060 -> 74.106.255.63:5907
Record-Route: <sip:700@54.204.19.238:5060;lr54.204.19.238;did=b1c.e6603363>.
Record-Route: <sip:107.21.231.147:5060;lr;ftag=31945052_6772d868_3b1eb7f5-c41b-4da5-9a97-c756d7cd8fe3>.


So the rr header looks broken for me, at least for the lr54.204.19.238 part....... as it is not following the ;name=value convention
Maybe you doin something weird in your config :)

-Laszlo







2013/12/20 Zoho Junk <[hidden email]>
Greetings,

I have OpenSips installed on an Amazon EC2 with an elastic IP (54.242.85.140). All non REGISTER methods are passed through rewritehostport("mytwiliodomain.sip.twilio.com") to another SIP proxy.

Everything is working except when BYE is sent from a callee that is registered to my OpenSips server. The caller never receives the BYE.
I've changed my config file at least 100 times and still have no success. Please let me know if I need to include anything from my config.

My First Invite:


U 2013/12/20 12:44:21.298060 107.21.231.147:5060 -> 10.167.11.171:5060
INVITE sip:[hidden email] SIP/2.0.
Record-Route: <sip:107.21.231.147:5060;lr;ftag=31945052_6772d868_3b1eb7f5-c41b-4da5-9a97-c756d7cd8fe3>.
From: "17175554895" <sip:[hidden email]>;tag=31945052_6772d868_3b1eb7f5-c41b-4da5-9a97-c756d7cd8fe3.
To: <sip:[hidden email]>.
CSeq: 102 INVITE.
Max-Forwards: 68.
Date: Fri, 20 Dec 2013 17:42:59 GMT.
Call-ID: [hidden email].
Contact: "17175554895" <sip:17175554895@10.70.227.82:5060;transport=udp>.
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE.
User-Agent: Twilio Gateway.
X-Twilio-ApiVersion: 2010-04-01.
X-Twilio-AccountSid: AC7ae5d84004d4269fgfhfhfjghgjghgbb.
Content-Type: application/sdp.
X-Twilio-CallSid: CA79e7a3c2c1c418af0777467467474747.
Via: SIP/2.0/UDP 107.21.231.147:5060;branch=z9hG4bKaeaf.02902511.0.
Via: SIP/2.0/UDP 10.70.227.82:5060;branch=z9hG4bK3b1eb7f5-c41b-4da5-9a97-c756d7cd8fe3_6772d868_425297635397116.
Content-Length: 251.
.
v=0.
o=- 1274389856 1274389856 IN IP4 54.196.144.208.
s=session.
c=IN IP4 54.196.144.208.
t=0 0.
m=audio 18800 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.


My First Bye:

U 2013/12/20 12:44:24.194615 74.106.255.63:5907 -> 10.167.11.171:5060
BYE sip:17175554895@107.21.231.147:5060;transport=udp;nat=yes SIP/2.0.
Via: SIP/2.0/UDP 74.106.255.63:5907;branch=z9hG4bK-41214dcd.
From: <sip:[hidden email]>;tag=5ed7c43daadb6f7fi0.
To: "17175554895" <sip:[hidden email]>;tag=31945052_6772d868_3b1eb7f5-c41b-4da5-9a97-c756d7cd8fe3.
Call-ID:
6fa66f8ea2ce4b2dae83149f8b34b2e2@0.0.0.0.
CSeq: 101 BYE.
Max-Forwards: 70.
Route: <sip:700@54.204.19.238:5060;lr54.204.19.238;did=b1c.e6603363>, <sip:107.21.231.147:5060;lr;ftag=31945052_6772d868_3b1eb7f5-c41b-4da5-9a97-c756d7cd8fe3>.
User-Agent: Linksys/SPA942-6.1.3(a).
Content-Length: 0.


My Debug Log:
http://pastebin.com/0pAAZ68J

NGrep:
http://pastebin.com/TVYUmM58

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--

--
Kind regards,
Laszlo Bekesi
http://voipfreak.net

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Re: BYE not reaching UAC

Ali Pey
In reply to this post by Zoho Junk
I see the Bye has a route for sip:700@54.204.19.238:5060

54.204.19.238:5060 has to be your opensips server. You probably need an alias for this IP.

This is only my guess. In situations like this you always need to look at your via, record-route and route headers, plus the source IPs. This is how things are routed in SIP.


Regards,
Ali Pey



On Fri, Dec 20, 2013 at 1:23 PM, Zoho Junk <[hidden email]> wrote:
Greetings,

I have OpenSips installed on an Amazon EC2 with an elastic IP (54.242.85.140). All non REGISTER methods are passed through rewritehostport("mytwiliodomain.sip.twilio.com") to another SIP proxy.

Everything is working except when BYE is sent from a callee that is registered to my OpenSips server. The caller never receives the BYE.
I've changed my config file at least 100 times and still have no success. Please let me know if I need to include anything from my config.

My First Invite:


U 2013/12/20 12:44:21.298060 107.21.231.147:5060 -> 10.167.11.171:5060
INVITE sip:[hidden email] SIP/2.0.
Record-Route: <sip:107.21.231.147:5060;lr;ftag=31945052_6772d868_3b1eb7f5-c41b-4da5-9a97-c756d7cd8fe3>.
From: "<a href="tel:17175554895" value="+17175554895" target="_blank">17175554895" <sip:[hidden email]>;tag=31945052_6772d868_3b1eb7f5-c41b-4da5-9a97-c756d7cd8fe3.
To: <sip:[hidden email]>.
CSeq: 102 INVITE.
Max-Forwards: 68.
Date: Fri, 20 Dec 2013 17:42:59 GMT.
Call-ID: [hidden email].
Contact: "<a href="tel:17175554895" value="+17175554895" target="_blank">17175554895" <sip:<a href="tel:17175554895" value="+17175554895" target="_blank">17175554895@10.70.227.82:5060;transport=udp>.
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE.
User-Agent: Twilio Gateway.
X-Twilio-ApiVersion: 2010-04-01.
X-Twilio-AccountSid: AC7ae5d84004d4269fgfhfhfjghgjghgbb.
Content-Type: application/sdp.
X-Twilio-CallSid: CA79e7a3c2c1c418af0777467467474747.
Via: SIP/2.0/UDP 107.21.231.147:5060;branch=z9hG4bKaeaf.02902511.0.
Via: SIP/2.0/UDP 10.70.227.82:5060;branch=z9hG4bK3b1eb7f5-c41b-4da5-9a97-c756d7cd8fe3_6772d868_425297635397116.
Content-Length: 251.
.
v=0.
o=- 1274389856 1274389856 IN IP4 54.196.144.208.
s=session.
c=IN IP4 54.196.144.208.
t=0 0.
m=audio 18800 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.


My First Bye:

U 2013/12/20 12:44:24.194615 74.106.255.63:5907 -> 10.167.11.171:5060
BYE sip:<a href="tel:17175554895" value="+17175554895" target="_blank">17175554895@107.21.231.147:5060;transport=udp;nat=yes SIP/2.0.
Via: SIP/2.0/UDP 74.106.255.63:5907;branch=z9hG4bK-41214dcd.
From: <sip:[hidden email]>;tag=5ed7c43daadb6f7fi0.
To: "<a href="tel:17175554895" value="+17175554895" target="_blank">17175554895" <sip:[hidden email]>;tag=31945052_6772d868_3b1eb7f5-c41b-4da5-9a97-c756d7cd8fe3.
Call-ID:
6fa66f8ea2ce4b2dae83149f8b34b2e2@0.0.0.0.
CSeq: 101 BYE.
Max-Forwards: 70.
Route: <sip:700@54.204.19.238:5060;lr54.204.19.238;did=b1c.e6603363>, <sip:107.21.231.147:5060;lr;ftag=31945052_6772d868_3b1eb7f5-c41b-4da5-9a97-c756d7cd8fe3>.
User-Agent: Linksys/SPA942-6.1.3(a).
Content-Length: 0.


My Debug Log:
http://pastebin.com/0pAAZ68J

NGrep:
http://pastebin.com/TVYUmM58

_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users



_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users