brekeke re-transmit INVITE after getting 180 and 200

Discuss any topic about Brekeke SIP Server.

Moderator: Brekeke Support Team

Post Reply
breque
Posts: 9
Joined: Wed Aug 10, 2011 9:43 am

brekeke re-transmit INVITE after getting 180 and 200

Post by breque »

1. Brekeke Product Name and version:
2.4.8.6
2. Java version:
1.6.0_18
3. OS type and the version:
Linux debian 6
4. UA (phone), gateway or other hardware/software involved:
PBX and xlite
5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :
pattern 9
6. Your problem:
A is calling B
INVITE from A to Brekeke SIP Server.
Brekeke SIP Server forward the INVITE to PBX.
PBX return INVITE to Brekeke SIP Server.
Brekeke SIP Server send INVITE to B.
B reply with 180 and later with 200, however, Brekeke SIP Server is doesn't stop sending INVITE.
notes - doing the call without going to the PBX is ok, but I must have the PBX because it has all the contacts.
I have a wireshark trace if needed.

looking forward for your help.
davi
Posts: 34
Joined: Wed Jan 26, 2011 4:34 pm

Post by davi »

what kind of PBX are you using? Which product is it?
breque
Posts: 9
Joined: Wed Aug 10, 2011 9:43 am

Post by breque »

asterisk
breque
Posts: 9
Joined: Wed Aug 10, 2011 9:43 am

Post by breque »

forgot to mention the most strange thing.
the BSS sends back the 180 and the 200 correctly to the A party.
anyone encountered that behavior?
janP
Posts: 336
Joined: Sun Nov 25, 2007 2:55 pm

Post by janP »

Does A keep sending INVITE too?
Does Asterisk keep sending INVITE too?
Does Asterisk change Call-ID?
breque
Posts: 9
Joined: Wed Aug 10, 2011 9:43 am

Post by breque »

A doesn't send any more INVITE
Asterisk doesn't either.
call-id remains the same as the asterisk is configured to work as proxy.
it's like the BSS doesn't recognized that the responses belong to the INVITE but if it didn't, why did it sends back the 180 and 200 to A?
janP
Posts: 336
Joined: Sun Nov 25, 2007 2:55 pm

Post by janP »

Let you use the B2B-UA mode in Brekeke SIP Server.

Its setting is [Configuration]->[SIP]->[SIP exchanger]->[B2B-UA mode].
It will avoid the problem.
breque
Posts: 9
Joined: Wed Aug 10, 2011 9:43 am

Post by breque »

already tried it.
didn't help.
tried changing pretty much every configuration in the configuration page.
thought it might be my DP but I am using a simple one that is even shown in the documents.
it MUST be something very simple and obvious I am (sadly) missing.
janP
Posts: 336
Joined: Sun Nov 25, 2007 2:55 pm

Post by janP »

A 180 response sent back from B must have same Via, Call-D, From and To headers.
Can you check them?


If you need.. you can contact brekeke's reseller or support team to request some assistances.
breque
Posts: 9
Joined: Wed Aug 10, 2011 9:43 am

Post by breque »

here is the print of the INVITE
Request-Line: INVITE sip:6868456@62.x.x.x:5060;rinstance=b769be10c08da7ab SIP/2.0
Message Header
Via: SIP/2.0/UDP 184.x.x.x:5060;rport;branch=z9hG4bK13110541c601a14383f799-a48da33f-4323a0d4
To: <sip:6868456@184.x.x.x:5060>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Call-ID: fa610794-e3f491e6-28775069-94f1ea1e@184.x.x.x
User-Agent: X-Lite 4 release 4.1 stamp 63214
Max-Forwards: 67
From: 7675351 <sip:7675351@184.x.x.x>;tag=96044cd9
Contact: <sip:7675351@184.x.x.x:5060>
CSeq: 1 INVITE
Supported: replaces
P-Asserted-Identity: 7675351 <sip:7675351@184.x.x.x>
Content-Type: application/sdp
Content-Length: 234




and of the 180
Status-Line: SIP/2.0 180 Ringing
Message Header
Via: SIP/2.0/UDP 184.x.x.x:5060;rport=5060;branch=z9hG4bK13110541c601a14383f799-a48da33f-4323a0d4
Contact: <sip:6868456@62.x.x.x:5060;rinstance=b769be10c08da7ab>
To: <sip:6868456@184.x.x.x:5060>;tag=a02f576e
From: "7675351"<sip:7675351@184.x.x.x>;tag=96044cd9
Call-ID: fa610794-e3f491e6-28775069-94f1ea1e@184.x.x.x
CSeq: 1 INVITE
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0
janP
Posts: 336
Joined: Sun Nov 25, 2007 2:55 pm

Post by janP »

I can't find any clues from the packets.

Why do you need to send INVITE back from Asterisk to SIP Server?
Can you send INVITE directly from Asterisk to B??

Furthermore.. why do you need Asterisk ???
breque
Posts: 9
Joined: Wed Aug 10, 2011 9:43 am

Post by breque »

I am using the BSS NAT capabilities but everything is located on the asterisk.
I am surprised I am the only one facing this issue.
Seems to be very simple layout I am using.

Will appreciate any suggestion.

Thanks
janP
Posts: 336
Joined: Sun Nov 25, 2007 2:55 pm

Post by janP »

Are you sure that the Brekeke SIP Server keep sending INVITE to B?

If you capture packets between Brekeke SIP Server and Asterisk, does Asterisk resend INVITE to Brekeke SIP Server?
breque
Posts: 9
Joined: Wed Aug 10, 2011 9:43 am

Post by breque »

yep. asterisk send only one.
bss kerp sending.
got many traces
janP
Posts: 336
Joined: Sun Nov 25, 2007 2:55 pm

Post by janP »

Is B X-Lite?
Is B behind NAT?

Even if you use B on a public IP (not behind NAT), does the problem happen ?
breque
Posts: 9
Joined: Wed Aug 10, 2011 9:43 am

Post by breque »

B is Xlite and is behind NAT.
didn't try with public IP yet
janP
Posts: 336
Joined: Sun Nov 25, 2007 2:55 pm

Post by janP »

Let you try it.
Using XLite on a public IP may make a different result.
Post Reply