SIPSwitch Upgrade - 6 Sep 2008

Latest news about SIP Sorcery
teddy_b
Posts: 65
Joined: Fri Aug 15, 2008 3:56 am

Post by teddy_b » Mon Sep 08, 2008 3:12 pm

[deleted]
Last edited by teddy_b on Mon Sep 08, 2008 3:22 pm, edited 1 time in total.

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

Post by Aaron » Mon Sep 08, 2008 3:12 pm

The outbound proxy setting on the Application server has been rolled back again. The reason this time is I noticed that all the Betamax servers look like they are purposefully dropping a Via header to prevent the use of SIP Proxy's on the call path to their servers.

For those that need calls to go through the same address as the register traffic, such as teddy_b, in the Advanced Settings of your provider you can put 213.200.94.182:5060 as the Outbound Proxy.

At this stage I don't believe the Outbound Proxy will be used by default since their should be few cases like teddy_b's and breaking calls to the Betamax servers would cause lots of pain.

Regards,

Aaron

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

Post by teddy_b » Mon Sep 08, 2008 3:38 pm

Aaron,

Yeah! After I set the proxy as you recommended everything is working again :)!

I suppose it would be better to use sip.mysipswitch.com:5060 for outboud proxy? Or just the IP address 213.200.94.182:5060?

Thank you very much for your help, and for taking care of even some unusual cases as mine!

ashley
Posts: 2
Joined: Mon Aug 04, 2008 6:05 pm

Post by ashley » Mon Sep 08, 2008 8:48 pm

Hello,

I am trying to use a Betamax service SMSlisto, in the forum it has been mentioned you can adjust the Advanced Call Settings and change the header to get the free calls.

Can some one tell me what needs to be adjusted?

Thanks
Ashley

dkwakkel
Posts: 2
Joined: Sun Aug 17, 2008 6:55 pm

Post by dkwakkel » Mon Sep 08, 2008 9:49 pm

I also have problems with incoming calls since yesterday. For a trace see below:

DialPlan=> Dialplan trace commenced at 08 Sep 2008 22:26:20:857.
SIPTransaction=>Request received 77.67.57.195:5060<-213.200.94.182:5060
INVITE sip:dkwakkel@sip.mysipswitch.com;switchtag=135318 SIP/2.0
Via: SIP/2.0/UDP 213.200.94.182:5060;branch=z9hG4bKQ8UJd+vTXkvSpCPps8wQJY0kR30=
Via: SIP/2.0/UDP 85.17.186.7;branch=z9hG4bK0f2d.01c15682.0
Via: SIP/2.0/UDP 81.23.228.129;branch=z9hG4bK0f2d.876e29c1.0
Via: SIP/2.0/UDP 80.252.84.190:5060;branch=z9hG4bK587266864
To: <sip:31529712016@budgetphone.nl:5060>
From: "0645161841" <sip:0645161841@80.252.84.190>;tag=1156187310
Call-ID: 845674557@80.252.84.190
CSeq: 20 INVITE
Contact: <sip:SIP_5Fd@85.17.186.7:5060>
Max-Forwards: 7
Record-Route: <sip:213.200.94.182:5060;lr>,<sip:85.17.186.7;lr=on;did=e42.53cf107;ftag=1156187310;fcd=yes>,<sip:81.23.228.129;lr=on;did=e42.d890ca5;ftag=1156187310>
Content-Type: application/sdp
Content-Length: 299
Allow: INVITE, ACK, BYE, CANCEL, INFO, UPDATE, REGISTER

v=0
o=SIP_5Fd 123456 654321 IN IP4 80.252.84.190
s=-
c=IN IP4 81.23.228.139
t=0 0
m=audio 63278 RTP/AVP 3 4 8 0 18 101
a=rtpmap:3 GSM/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15

DialPlan=> call from sip:0645161841@80.252.84.190 to dkwakkel.
SIPTransaction=> Send Info Response 77.67.57.195:5060->213.200.94.182:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 213.200.94.182:5060;branch=z9hG4bKQ8UJd+vTXkvSpCPps8wQJY0kR30=
Via: SIP/2.0/UDP 85.17.186.7;branch=z9hG4bK0f2d.01c15682.0
Via: SIP/2.0/UDP 81.23.228.129;branch=z9hG4bK0f2d.876e29c1.0
Via: SIP/2.0/UDP 80.252.84.190:5060;branch=z9hG4bK587266864
To: <sip:31529712016@budgetphone.nl:5060>
From: "0645161841" <sip:0645161841@80.252.84.190>;tag=1156187310
Call-ID: 845674557@80.252.84.190
CSeq: 20 INVITE


DialPlan=> New SwitchCall starting for sip:dkwakkel@sip.mysipswitch.com;switchtag=135318, attempting to resolve 213.200.94.182:5060.
DialPlan=> Switching sip:dkwakkel@sip.mysipswitch.com:5060->sip:dkwakkel@83.119.71.227:5060 via 213.200.94.182:5060.
SIPTransaction=> Send Request reliable 77.67.57.195:5060->213.200.94.182:5060
INVITE sip:dkwakkel@83.119.71.227:5060 SIP/2.0
Via: SIP/2.0/UDP 77.67.57.195:5060;branch=z9hG4bK5041a65ca6f04dc89f032eda26e577b6
To: <sip:dkwakkel@83.119.71.227:5060>
From: "0645161841" <sip:0645161841@80.252.84.190>;tag=1321447478
Call-ID: ac83f84638f944158bd0dc31a033a921
CSeq: 1 INVITE
Contact: <sip:77.67.57.195:5060>
Max-Forwards: 70
User-Agent: www.sipsorcery.com
Content-Type: application/sdp
Content-Length: 299

v=0
o=SIP_5Fd 123456 654321 IN IP4 80.252.84.190
s=-
c=IN IP4 81.23.228.139
t=0 0
m=audio 63278 RTP/AVP 3 4 8 0 18 101
a=rtpmap:3 GSM/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15

SIPTransaction=> Send Request retransmit 2 77.67.57.195:5060->213.200.94.182:5060
INVITE sip:dkwakkel@83.119.71.227:5060 SIP/2.0
Via: SIP/2.0/UDP 77.67.57.195:5060;branch=z9hG4bK5041a65ca6f04dc89f032eda26e577b6
To: <sip:dkwakkel@83.119.71.227:5060>
From: "0645161841" <sip:0645161841@80.252.84.190>;tag=1321447478
Call-ID: ac83f84638f944158bd0dc31a033a921
CSeq: 1 INVITE
Contact: <sip:77.67.57.195:5060>
Max-Forwards: 70
User-Agent: www.sipsorcery.com
Content-Type: application/sdp
Content-Length: 299

v=0
o=SIP_5Fd 123456 654321 IN IP4 80.252.84.190
s=-
c=IN IP4 81.23.228.139
t=0 0
m=audio 63278 RTP/AVP 3 4 8 0 18 101
a=rtpmap:3 GSM/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15

SIPTransaction=> Send Request retransmit 3 77.67.57.195:5060->213.200.94.182:5060
INVITE sip:dkwakkel@83.119.71.227:5060 SIP/2.0
Via: SIP/2.0/UDP 77.67.57.195:5060;branch=z9hG4bK5041a65ca6f04dc89f032eda26e577b6
To: <sip:dkwakkel@83.119.71.227:5060>
From: "0645161841" <sip:0645161841@80.252.84.190>;tag=1321447478
Call-ID: ac83f84638f944158bd0dc31a033a921
CSeq: 1 INVITE
Contact: <sip:77.67.57.195:5060>
Max-Forwards: 70
User-Agent: www.sipsorcery.com
Content-Type: application/sdp
Content-Length: 299

v=0
o=SIP_5Fd 123456 654321 IN IP4 80.252.84.190
s=-
c=IN IP4 81.23.228.139
t=0 0
m=audio 63278 RTP/AVP 3 4 8 0 18 101
a=rtpmap:3 GSM/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15

SIPTransaction=> Send Request retransmit 4 77.67.57.195:5060->213.200.94.182:5060
INVITE sip:dkwakkel@83.119.71.227:5060 SIP/2.0
Via: SIP/2.0/UDP 77.67.57.195:5060;branch=z9hG4bK5041a65ca6f04dc89f032eda26e577b6
To: <sip:dkwakkel@83.119.71.227:5060>
From: "0645161841" <sip:0645161841@80.252.84.190>;tag=1321447478
Call-ID: ac83f84638f944158bd0dc31a033a921
CSeq: 1 INVITE
Contact: <sip:77.67.57.195:5060>
Max-Forwards: 70
User-Agent: www.sipsorcery.com
Content-Type: application/sdp
Content-Length: 299

v=0
o=SIP_5Fd 123456 654321 IN IP4 80.252.84.190
s=-
c=IN IP4 81.23.228.139
t=0 0
m=audio 63278 RTP/AVP 3 4 8 0 18 101
a=rtpmap:3 GSM/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15

SIPTransaction=> Send Request retransmit 5 77.67.57.195:5060->213.200.94.182:5060
INVITE sip:dkwakkel@83.119.71.227:5060 SIP/2.0
Via: SIP/2.0/UDP 77.67.57.195:5060;branch=z9hG4bK5041a65ca6f04dc89f032eda26e577b6
To: <sip:dkwakkel@83.119.71.227:5060>
From: "0645161841" <sip:0645161841@80.252.84.190>;tag=1321447478
Call-ID: ac83f84638f944158bd0dc31a033a921
CSeq: 1 INVITE
Contact: <sip:77.67.57.195:5060>
Max-Forwards: 70
User-Agent: www.sipsorcery.com
Content-Type: application/sdp
Content-Length: 299

v=0
o=SIP_5Fd 123456 654321 IN IP4 80.252.84.190
s=-
c=IN IP4 81.23.228.139
t=0 0
m=audio 63278 RTP/AVP 3 4 8 0 18 101
a=rtpmap:3 GSM/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15

SIPTransaction=> Send Request retransmit 6 77.67.57.195:5060->213.200.94.182:5060
INVITE sip:dkwakkel@83.119.71.227:5060 SIP/2.0
Via: SIP/2.0/UDP 77.67.57.195:5060;branch=z9hG4bK5041a65ca6f04dc89f032eda26e577b6
To: <sip:dkwakkel@83.119.71.227:5060>
From: "0645161841" <sip:0645161841@80.252.84.190>;tag=1321447478
Call-ID: ac83f84638f944158bd0dc31a033a921
CSeq: 1 INVITE
Contact: <sip:77.67.57.195:5060>
Max-Forwards: 70
User-Agent: www.sipsorcery.com
Content-Type: application/sdp
Content-Length: 299

v=0
o=SIP_5Fd 123456 654321 IN IP4 80.252.84.190
s=-
c=IN IP4 81.23.228.139
t=0 0
m=audio 63278 RTP/AVP 3 4 8 0 18 101
a=rtpmap:3 GSM/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15

SIPTransaction=> Send Request retransmit 7 77.67.57.195:5060->213.200.94.182:5060
INVITE sip:dkwakkel@83.119.71.227:5060 SIP/2.0
Via: SIP/2.0/UDP 77.67.57.195:5060;branch=z9hG4bK5041a65ca6f04dc89f032eda26e577b6
To: <sip:dkwakkel@83.119.71.227:5060>
From: "0645161841" <sip:0645161841@80.252.84.190>;tag=1321447478
Call-ID: ac83f84638f944158bd0dc31a033a921
CSeq: 1 INVITE
Contact: <sip:77.67.57.195:5060>
Max-Forwards: 70
User-Agent: www.sipsorcery.com
Content-Type: application/sdp
Content-Length: 299

v=0
o=SIP_5Fd 123456 654321 IN IP4 80.252.84.190
s=-
c=IN IP4 81.23.228.139
t=0 0
m=audio 63278 RTP/AVP 3 4 8 0 18 101
a=rtpmap:3 GSM/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15

SIPTransaction=> Send Final Response Reliable 77.67.57.195:5060->213.200.94.182:5060
SIP/2.0 603 Unknown
Via: SIP/2.0/UDP 213.200.94.182:5060;branch=z9hG4bKQ8UJd+vTXkvSpCPps8wQJY0kR30=
Via: SIP/2.0/UDP 85.17.186.7;branch=z9hG4bK0f2d.01c15682.0
Via: SIP/2.0/UDP 81.23.228.129;branch=z9hG4bK0f2d.876e29c1.0
Via: SIP/2.0/UDP 80.252.84.190:5060;branch=z9hG4bK587266864
To: <sip:31529712016@budgetphone.nl:5060>
From: "0645161841" <sip:0645161841@80.252.84.190>;tag=1156187310
Call-ID: 845674557@80.252.84.190
CSeq: 20 INVITE

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

Post by Aaron » Mon Sep 08, 2008 10:20 pm

Hi dkwakkel,

Could you let me know how your incoming calls are being forwarded in your dialplan? Are you using the exten or Ruby syntax?

Regards,

Aaron

nd_electro
Posts: 7
Joined: Wed May 28, 2008 11:23 am

Post by nd_electro » Tue Sep 09, 2008 12:18 am

Hi Aaron,

To get incoming calls, I register the provider via the configuration menu

For registering the sipgate account I use

username: myusername
password: mypassword
server: sipgate.de

registration contact: sip:ndodel@sip.mysipswitch.com
registration expiry: 3600

auth username: myusername

thats all

in the monitor under contact uri I see

sip:ndodel@sip.mysipswitch.com;switchtag=146720


Regards

Norman

voovi
Posts: 13
Joined: Wed May 14, 2008 7:47 am

Post by voovi » Tue Sep 09, 2008 12:20 am

I also have the same problem with the incomming calls.
My dialplan is Ruby.
When I use Sipphone with MSS (device is VT2442): My ATA doesn't ring but it will ring if I register Sipphone directly.

When I use Voxalot with MSS (device is VT2442):
Line 1 (register with MSS)
Line 2 (register with Voxalot)
Only line registering with Voxalot rings if I have a incommings call to voxalot account.

Log from MSS:

Code: Select all

20:45:25:596: New SwitchCall starting for sip:v123xxx@sip.mysipswitch.com, attempting to resolve 213.200.94.182:5060. 
20:45:25:599: Switching sip:v123xxx@sip.mysipswitch.com:5060->sip:v123xxx@76.xx.xx.xxx:41279 via 213.200.94.182:5060. 
Log from VT2442 (it received 4 INVITE below). I think MSS can not receive message sent from the VT2442:

Code: Select all

SSMU: {--- Message Received: Received from addr=213.200.94.182:5060, UDP
 INVITE sip:XXXvooviXXX@67.62.47.57:4448 SIP/2.0
 Via: SIP/2.0/UDP 213.200.94.182:5060;branch=abcdef=
 Via: SIP/2.0/UDP 77.67.57.195:5060;branch=xywz
 To: {sip:XXXvooviXXX@67.62.47.57:4448}
 From: "Unkown" {sip:360XXXXXXX@66.54.140.46};tag=1234567
 Call-ID: b52630f6ac4f4d3681ba55c0c6b5c09d
 CSeq: 1 INVITE
 Contact: {sip:77.67.57.195:5060}
 Max-Forwards: 69
 Record-Route: {sip:213.200.94.182:5060;lr}
 User-Agent: www.sipsorcery.com
 Content-Type: application/sdp
 Content-Length: 379
 
 v=0
 o=root 13058 13058 IN IP4 66.54.140.46
 s=session
 c=IN IP4 66.54.140.46
 t=0 0
 m=audio 15612 RTP/AVP 0 8 3 18 97 101
 a=rtpmap:0 PCMU/8000
 
a=rtpmap:8 PCMA/8000
 a=rtpmap:3 GSM/8000
 a=rtpmap:18 G729/8000
 a=fmtp:18 annexb=no
 a=rtpmap:97 iLBC/8000
 a=fmtp:97 mode=30
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=silenceSupp:off - - - -
 a=ptime:20
 a=sendrecv

voovi
Posts: 13
Joined: Wed May 14, 2008 7:47 am

Post by voovi » Tue Sep 09, 2008 12:25 am

Hi Aaron
Log from MySipSwitch should be:

Code: Select all

User XXXvooviXXX online is True. 
Contact sip:XXXvooviXXX@sip.mysipswitch.com 
Binding 67.62.47.57:4448. 
New SwitchCall starting for sip:XXXvooviXXX@sip.mysipswitch.com, attempting to resolve 213.200.94.182:5060. 
01:07:01:217: Switching sip:XXXvooviXXX@sip.mysipswitch.com:5060->sip:XXXvooviXXX@67.62.47.57:4448 via 213.200.94.182:5060.
01:07:32:342: Cancelling forwarded call leg sip:XXXvooviXXX@67.62.47.57:4448, no repsonse from server has been received so no CANCEL request required. 

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

Post by Aaron » Tue Sep 09, 2008 1:05 am

Hi voovi,

That all looks correct. I'll do some more troubleshooting on this issue once I get home from work this evening.

Regards,

Aaron

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