SIPSwitch Upgrade - 6 Sep 2008

Latest news about SIP Sorcery
voovi
Posts: 13
Joined: Wed May 14, 2008 7:47 am

Post by voovi » Tue Sep 09, 2008 1:10 am

In the log on my device, I found that my VT2442 sent 2 messages (Trying and Ringing) to Voxalot but did not log any messages sent to MSS. Does it help?

ais11
Posts: 39
Joined: Mon Jul 21, 2008 8:55 am

Post by ais11 » Tue Sep 09, 2008 2:31 am

In my case, also not receiving incoming calls. My ATA (PAP2) directly logs onto MSS. Currently the ATA is telling me logged-on while monitoring page status on MSS shows NOT LOGGED-IN. I am using Ruby Dial plan.

Aaron
Site Admin
Posts: 4652
Joined: Thu Jul 12, 2007 12:13 am

Post by Aaron » Tue Sep 09, 2008 3:56 am

Hi All (or those having the issues on this thread),

I'll be able to look into this at approx 0800 UTC. At that point I will be trying to identify the cause of two issues:

1. Why some user's ATAs are not stating registered,

2. Why some user's incoming calls are not connecting even when their ATAs are registered.

To help with the first issue I will be adding a new monitoring event that writes a message when teh SIP Registrar expires a binding. For those users having the problem with point 1 it would be useful if you could connect via telnet at about 0930 and let me know what that message reports as the reason for your ATA's binding expiring.

For point 2 I'll check this thread at around 0900 UTC so if someone having an incoming call issue and who can do a few test callsat that time please post to this thread as well. I will watch the SIP traffic for those calls to try and identify why they are note getting answered.

Regards,

Aaron

teddy_b
Posts: 65
Joined: Fri Aug 15, 2008 3:56 am

Post by teddy_b » Tue Sep 09, 2008 6:11 am

Aaron wrote:in the Advanced Settings of your provider you can put 213.200.94.182:5060 as the Outbound Proxy.
One more question...
Does this proxy pass through the User-Agent header that is set in my provider's settings, or does it replace it by its own value? I cannot verify it myself since the trace only shows the traffic between the app server and the proxy...

Aaron
Site Admin
Posts: 4652
Joined: Thu Jul 12, 2007 12:13 am

Post by Aaron » Tue Sep 09, 2008 6:47 am

Hi teddy_b,

The Proxy does very little apart from act as a single point of contact for all the different SIP flows. It's job is to funnel traffic from one side to the other and it does very little in between.

The Proxy does not change the User-Agent header. The only headers it will change are the Via, Route, Record-Route and in some cases the Contact header if it contains a private IP address.

Regards,

Aaron

Aaron
Site Admin
Posts: 4652
Joined: Thu Jul 12, 2007 12:13 am

Post by Aaron » Tue Sep 09, 2008 7:39 am

For anyone experiencing failed registrations or failed incoming calls now is the time to PM me your mss username and I will take a look. I plan on being around for another hour or possibly two until 1000 UTC.

Regards,

Aaron

Aaron
Site Admin
Posts: 4652
Joined: Thu Jul 12, 2007 12:13 am

Post by Aaron » Tue Sep 09, 2008 2:53 pm

Hi All,

I believe the issue with the disappearing registrations has now been tracked down and fixed. It was a bug in the way the database updates were being handled. The SIP Registrar was still reporting the registrations as Ok but the database was removing them prematurely. Incoming calls and the monitoring page both work off the database whereas ATAs only communicate with the SIP Registrar hence the contradictory information.

Anyway that issue is now fixed and registrations to mss should be fine.

The other issue with incoming calls has not been investigated any further as I have not been able to get hold of a failing incoming number. once I do I'll keep looking but at the moment it looks like it's only happening in fairly isolated cases so could be down to a SIP integration issue between mss and a provider or two.

Assuming these two issues are resolved then the mysipswitch agent split is now complete and only minor issues are left to investigate and clean up. All 4 SIP server agents (Proxy, Registrar, Registration Agent and Application Server) are now deployed in their designated roles.

Regards,

Aaron

genesis
Posts: 14
Joined: Thu Apr 17, 2008 3:27 pm

Post by genesis » Tue Sep 09, 2008 6:10 pm

Hi,

I'm still experiencing proxy time out after 2mins...?

G

SIP3120
Posts: 38
Joined: Sun Feb 03, 2008 5:27 pm

Siptraces

Post by SIP3120 » Tue Sep 09, 2008 6:46 pm

Don't know if this will help.
SIP trace of INCOMPLETE through MSS #1
and Completed call to Softphone #2

DialPlan=> Dialplan trace commenced at 09 Sep 2008 19:16:54:825.
SIPTransaction=>Request received 77.67.57.195:5060<-213.200.94.182:5060
INVITE sip:17473598115@sip.mysipswitch.com SIP/2.0
Via: SIP/2.0/UDP 213.200.94.182:5060;branch=z9hG4bK8D59774619172BE52F87EF59030B158AEC51177
Via: SIP/2.0/UDP 192.168.9.139:52328;branch=z9hG4bK-d8754z-4208ca42d10e240c-1---d8754z-;received=76.109.000.75
To: "Suz 359.8115" <sip:17473598115@sip.mysipswitch.com>
From: "Rik SipSW" <sip:SIPSW@sip.mysipswitch.com>;tag=00572706
Call-ID: OGFjYzkzOGVmNzQ4NDk3YTI2MzU2NmM2MmUyNmMwMzA.
CSeq: 2 INVITE
Contact: <sip:SIPSW@76.109.000.75:52328>
Max-Forwards: 69
Record-Route: <sip:213.200.94.182:5060;lr>
User-Agent: eyeBeam release 1100z stamp 47739
Proxy-Authorization: Digest username="SIPSW",realm="sip.mysipswitch.com",nonce="1393484282",uri="sip:17473598115@sip.mysipswitch.com",response="b015fce3eb0d0ce4a3d026679d7c9a39",algorithm=MD5
Content-Type: application/sdp
Content-Length: 523
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO

v=0
o=- 9 2 IN IP4 76.109.000.75
s=CounterPath eyeBeam 1.5
c=IN IP4 76.109.000.75
t=0 0
m=audio 21492 RTP/AVP 107 100 106 6 0 105 8 18 3 5 101
a=alt:1 2 : OzcG8VoV hCXBhs5/ 192.168.9.139 21492
a=alt:2 1 : Zv3EnGh6 2sS6cRQ7 76.109.000.75 21492
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:C3DD0A51BA0C450785DF0B49891638C8

NewCall=> Call received sip:17473598115@sip.mysipswitch.com.
DialPlan=> New SwitchCall starting for sip:17473598115@sip.mysipswitch.com, attempting to resolve proxy01.sipphone.com.
DialPlan=> Switching sip:17473598115@sip.mysipswitch.com:5060->sip:17473598115@proxy01.sipphone.com:5060 via 198.65.166.131:5060.
SIPTransaction=> Send Request reliable 77.67.57.195:5060->198.65.166.131:5060
INVITE sip:17473598115@proxy01.sipphone.com SIP/2.0
Via: SIP/2.0/UDP 77.67.57.195:5060;branch=z9hG4bKfa6826422e494ae9b74529864cfed7a6
To: <sip:17473598115@proxy01.sipphone.com>
From: <sip:17473310000@proxy01.sipphone.com>;tag=1599120493
Call-ID: 9781c4feb8b14ddb970bc0eeb07f9ac1
CSeq: 1 INVITE
Contact: <sip:17473310000@77.67.57.195:5060>
Max-Forwards: 70
User-Agent: www.sipsorcery.com
Content-Type: application/sdp
Content-Length: 523

v=0
o=- 9 2 IN IP4 76.109.000.75
s=CounterPath eyeBeam 1.5
c=IN IP4 76.109.000.75
t=0 0
m=audio 21492 RTP/AVP 107 100 106 6 0 105 8 18 3 5 101
a=alt:1 2 : OzcG8VoV hCXBhs5/ 192.168.9.139 21492
a=alt:2 1 : Zv3EnGh6 2sS6cRQ7 76.109.000.75 21492
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:C3DD0A51BA0C450785DF0B49891638C8

SIPTransaction=> Received Response 77.67.57.195:5060<-198.65.166.131:5060
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 77.67.57.195:5060;branch=z9hG4bKfa6826422e494ae9b74529864cfed7a6
To: <sip:17473598115@proxy01.sipphone.com>
From: <sip:17473310000@proxy01.sipphone.com>;tag=1599120493
Call-ID: 9781c4feb8b14ddb970bc0eeb07f9ac1
CSeq: 1 INVITE


SIPTransaction=> Received Response 77.67.57.195:5060<-198.65.166.131:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 77.67.57.195:5060;branch=z9hG4bKfa6826422e494ae9b74529864cfed7a6
To: <sip:17473598115@proxy01.sipphone.com>;tag=0000508070EF2DEE
From: <sip:17473310000@proxy01.sipphone.com>;tag=1599120493
Call-ID: 9781c4feb8b14ddb970bc0eeb07f9ac1
CSeq: 1 INVITE
Contact: <sip:announcement+novoicemail@198.65.166.130:5060>
Record-Route: <sip:198.65.166.131;lr;ftag=1599120493>
Content-Type: application/sdp
Content-Length: 110
Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux))
Warning: 392 198.65.166.130:5060 "Noisy feedback tells: pid=14184 req_src_ip=198.65.166.131 req_src_port=5060 in_uri=sip:announce_novoicemail@sems01.sipphone.com out_uri=sip:announce_novoicemail@proxy01.sipphone.com via_cnt==0"
RemoteIP: 198.65.166.130

v=0
o=username 0 0 IN IP4 198.65.166.130
s=session
c=IN IP4 198.65.166.130
t=0 0
m=audio 1998 RTP/AVP 0

SIPTransaction=> Send Request 77.67.57.195:5060->198.65.166.131:5060
ACK sip:announcement+novoicemail@198.65.166.130:5060 SIP/2.0
Via: SIP/2.0/UDP 77.67.57.195:5060;branch=z9hG4bK617f28883bc240b598c801ca05101e93
To: <sip:17473598115@proxy01.sipphone.com>;tag=0000508070EF2DEE
From: <sip:17473310000@proxy01.sipphone.com>;tag=1599120493
Call-ID: 9781c4feb8b14ddb970bc0eeb07f9ac1
CSeq: 1 ACK
Max-Forwards: 70
Route: <sip:198.65.166.131;lr;ftag=1599120493>


DialPlan=> Response 200 OK for sip:17473598115@proxy01.sipphone.com.
SIPTransaction=> Send Final Response Reliable 77.67.57.195:5060->213.200.94.182:5060
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 213.200.94.182:5060;branch=z9hG4bK8D59774619172BE52F87EF59030B158AEC51177
Via: SIP/2.0/UDP 192.168.9.139:52328;branch=z9hG4bK-d8754z-4208ca42d10e240c-1---d8754z-;received=76.109.000.75
To: "Suz 359.8115" <sip:17473598115@sip.mysipswitch.com>;tag=1787076033
From: "Rik SipSW" <sip:SIPSW@sip.mysipswitch.com>;tag=00572706
Call-ID: OGFjYzkzOGVmNzQ4NDk3YTI2MzU2NmM2MmUyNmMwMzA.
CSeq: 2 INVITE
Contact: <sip:77.67.57.195:5060>
Record-Route: <sip:213.200.94.182:5060;lr>
Server: www.sipsorcery.com
Content-Type: application/sdp
Content-Length: 110

v=0
o=username 0 0 IN IP4 198.65.166.130
s=session
c=IN IP4 198.65.166.130
t=0 0
m=audio 1998 RTP/AVP 0

DialPlan=> Dialplan trace completed at 09 Sep 2008 19:17:25:154.


==============================================
#2
DialPlan=> Dialplan trace commenced at 09 Sep 2008 19:15:02:529.
SIPTransaction=>Request received 77.67.57.195:5060<-213.200.94.182:5060
INVITE sip:17473598115@sip.mysipswitch.com SIP/2.0
Via: SIP/2.0/UDP 213.200.94.182:5060;branch=z9hG4bKC0924F33D5C0CF2E97D7C99E44509E2D71CF8A7
Via: SIP/2.0/UDP 192.168.9.139:52328;branch=z9hG4bK-d8754z-8a223574070baf37-1---d8754z-;received=76.109.000.75
To: "Suz 359.8115" <sip:17473598115@sip.mysipswitch.com>
From: "Rik SipSW" <sip:SIPSW@sip.mysipswitch.com>;tag=8a78315b
Call-ID: MmE4NTMwYzAwNTc3MTkwYWU0MzBkZjE0YjI5YmRmNmE.
CSeq: 2 INVITE
Contact: <sip:SIPSW@76.109.000.75:52328>
Max-Forwards: 69
Record-Route: <sip:213.200.94.182:5060;lr>
User-Agent: eyeBeam release 1100z stamp 47739
Proxy-Authorization: Digest username="SIPSW",realm="sip.mysipswitch.com",nonce="1503204830",uri="sip:17473598115@sip.mysipswitch.com",response="5bfa7f7ada973810900e12b1e5d68d6e",algorithm=MD5
Content-Type: application/sdp
Content-Length: 523
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO

v=0
o=- 0 2 IN IP4 76.109.000.75
s=CounterPath eyeBeam 1.5
c=IN IP4 76.109.000.75
t=0 0
m=audio 15070 RTP/AVP 107 100 106 6 0 105 8 18 3 5 101
a=alt:1 2 : vaTVdoBh FjPZ7hqJ 192.168.9.139 15070
a=alt:2 1 : UZYZ5V0A JuUiIzN9 76.109.000.75 15070
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:10ED6D2BB2F94770BB0C425614707C6E

NewCall=> Call received sip:17473598115@sip.mysipswitch.com.
DialPlan=> New SwitchCall starting for sip:17473598115@sip.mysipswitch.com, attempting to resolve proxy01.sipphone.com.
DialPlan=> Switching sip:17473598115@sip.mysipswitch.com:5060->sip:17473598115@proxy01.sipphone.com:5060 via 198.65.166.131:5060.
SIPTransaction=> Send Request reliable 77.67.57.195:5060->198.65.166.131:5060
INVITE sip:17473598115@proxy01.sipphone.com SIP/2.0
Via: SIP/2.0/UDP 77.67.57.195:5060;branch=z9hG4bK8292b26791ea4229948e8f3194cdea00
To: <sip:17473598115@proxy01.sipphone.com>
From: <sip:17473310000@proxy01.sipphone.com>;tag=1694442836
Call-ID: 5a48d6cb460a444abd6365c9c295e6f1
CSeq: 1 INVITE
Contact: <sip:17473310000@77.67.57.195:5060>
Max-Forwards: 70
User-Agent: www.sipsorcery.com
Content-Type: application/sdp
Content-Length: 523

v=0
o=- 0 2 IN IP4 76.109.000.75
s=CounterPath eyeBeam 1.5
c=IN IP4 76.109.000.75
t=0 0
m=audio 15070 RTP/AVP 107 100 106 6 0 105 8 18 3 5 101
a=alt:1 2 : vaTVdoBh FjPZ7hqJ 192.168.9.139 15070
a=alt:2 1 : UZYZ5V0A JuUiIzN9 76.109.000.75 15070
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:10ED6D2BB2F94770BB0C425614707C6E

SIPTransaction=> Received Response 77.67.57.195:5060<-198.65.166.131:5060
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 77.67.57.195:5060;branch=z9hG4bK8292b26791ea4229948e8f3194cdea00
To: <sip:17473598115@proxy01.sipphone.com>
From: <sip:17473310000@proxy01.sipphone.com>;tag=1694442836
Call-ID: 5a48d6cb460a444abd6365c9c295e6f1
CSeq: 1 INVITE


SIPTransaction=> Received Response 77.67.57.195:5060<-198.65.166.131:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 77.67.57.195:5060;branch=z9hG4bK8292b26791ea4229948e8f3194cdea00
To: <sip:17473598115@proxy01.sipphone.com>;tag=fb04214c
From: <sip:17473310000@proxy01.sipphone.com>;tag=1694442836
Call-ID: 5a48d6cb460a444abd6365c9c295e6f1
CSeq: 1 INVITE
Contact: <sip:SuzGizmo5@76.109.000.75:22156;rinstance=1a237ead7daf11a4>
Record-Route: <sip:198.65.166.131;lr;ftag=1694442836>
User-Agent: Bria release 2.4 stamp 49274
RemoteIP: 76.109.000.75


SIPTransaction=> Send Info Response 77.67.57.195:5060->213.200.94.182:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 213.200.94.182:5060;branch=z9hG4bKC0924F33D5C0CF2E97D7C99E44509E2D71CF8A7
Via: SIP/2.0/UDP 192.168.9.139:52328;branch=z9hG4bK-d8754z-8a223574070baf37-1---d8754z-;received=76.109.000.75
To: "Suz 359.8115" <sip:17473598115@sip.mysipswitch.com>
From: "Rik SipSW" <sip:SIPSW@sip.mysipswitch.com>;tag=8a78315b
Call-ID: MmE4NTMwYzAwNTc3MTkwYWU0MzBkZjE0YjI5YmRmNmE.
CSeq: 2 INVITE


SIPTransaction=> Received Response 77.67.57.195:5060<-198.65.166.131:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 77.67.57.195:5060;branch=z9hG4bK8292b26791ea4229948e8f3194cdea00
To: <sip:17473598115@proxy01.sipphone.com>;tag=fb04214c
From: <sip:17473310000@proxy01.sipphone.com>;tag=1694442836
Call-ID: 5a48d6cb460a444abd6365c9c295e6f1
CSeq: 1 INVITE
Contact: <sip:SuzGizmo5@76.109.000.75:22156;rinstance=1a237ead7daf11a4;nat=yes>
Record-Route: <sip:198.65.166.131;lr;ftag=1694442836>
User-Agent: Bria release 2.4 stamp 49274
Content-Type: application/sdp
Content-Length: 494
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
RemoteIP: 76.109.000.75

v=0
o=- 2 2 IN IP4 76.109.000.75
s=CounterPath Bria
c=IN IP4 198.65.166.131
t=0 0
m=audio 48982 RTP/AVP 107 100 106 0 8 18 101
a=alt:1 2 : 06HtZPTW K9dW6OXd 192.168.9.139 1838
a=alt:2 1 : q5Lu+Hg3 SfpOxaoK 76.109.000.75 1838
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:4AC294708C164968AE9E6343D946CC19
a=nortpproxy:yes

SIPTransaction=> Send Request 77.67.57.195:5060->198.65.166.131:5060
ACK sip:SuzGizmo5@76.109.000.75:22156;rinstance=1a237ead7daf11a4;nat=yes SIP/2.0
Via: SIP/2.0/UDP 77.67.57.195:5060;branch=z9hG4bKbe2693bbc59046e39a6e4c7aec0af266
To: <sip:17473598115@proxy01.sipphone.com>;tag=fb04214c
From: <sip:17473310000@proxy01.sipphone.com>;tag=1694442836
Call-ID: 5a48d6cb460a444abd6365c9c295e6f1
CSeq: 1 ACK
Max-Forwards: 70
Route: <sip:198.65.166.131;lr;ftag=1694442836>


DialPlan=> Response 200 OK for sip:17473598115@proxy01.sipphone.com.
SIPTransaction=> Send Final Response Reliable 77.67.57.195:5060->213.200.94.182:5060
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 213.200.94.182:5060;branch=z9hG4bKC0924F33D5C0CF2E97D7C99E44509E2D71CF8A7
Via: SIP/2.0/UDP 192.168.9.139:52328;branch=z9hG4bK-d8754z-8a223574070baf37-1---d8754z-;received=76.109.000.75
To: "Suz 359.8115" <sip:17473598115@sip.mysipswitch.com>;tag=1171221998
From: "Rik SipSW" <sip:SIPSW@sip.mysipswitch.com>;tag=8a78315b
Call-ID: MmE4NTMwYzAwNTc3MTkwYWU0MzBkZjE0YjI5YmRmNmE.
CSeq: 2 INVITE
Contact: <sip:77.67.57.195:5060>
Record-Route: <sip:213.200.94.182:5060;lr>
Server: www.sipsorcery.com
Content-Type: application/sdp
Content-Length: 494

v=0
o=- 2 2 IN IP4 76.109.000.75
s=CounterPath Bria
c=IN IP4 198.65.166.131
t=0 0
m=audio 48982 RTP/AVP 107 100 106 0 8 18 101
a=alt:1 2 : 06HtZPTW K9dW6OXd 192.168.9.139 1838
a=alt:2 1 : q5Lu+Hg3 SfpOxaoK 76.109.000.75 1838
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:4AC294708C164968AE9E6343D946CC19
a=nortpproxy:yes

DialPlan=> Dialplan trace completed at 09 Sep 2008 19:15:11:404.

ais11
Posts: 39
Joined: Mon Jul 21, 2008 8:55 am

Post by ais11 » Tue Sep 09, 2008 6:54 pm

Just came back home and reconfigured ATA to logon to MSS. Right now I am on MSS monitoring page. The status is sometime showing logged-in sometimes showing "not logged-in" Tried to call from another line but call did not reach. So it is same atleast in my case.

Post Reply