[OpenSIPS-Users] Working with MAX-FORWARDS in B2B topology hidding
Hello. Please tell me who faced the setting of the MAX-FORWARDS parameter pass when using the B2B module We have found that when using the module, this counter starts counting again from the initial value of 70, which leads to looping of calls in the network. example from trace in the input of opensips we have INVITE sip:543174317@94.241.67.137 SIP/2.0 *Max-Forwards: 18* Via: SIP/2.0/UDP 94.241.67.136:5062;rport;branch=z9hG4bK759843122 From: ;tag=298433023 To: Call-ID: 752394752@94.241.67.136:5062 CSeq: 3633945 INVITE User-Agent: YATE/6.2.1 Contact: Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, PRACK, INFO Supported: 100rel Content-Type: multipart/mixed;boundary=734768646_1362544590 Content-Length: 443 but on the output there is: INVITE sip:543174317@94.241.67.137 SIP/2.0 Via: SIP/2.0/UDP 94.241.67.137:5060;branch=z9hG4bK5e7e.38ff3a06.0 To: sip:543174317@94.241.67.137 From: ;tag=153c35a5fde4948908525582635dfe04 CSeq: 3633946 INVITE Call-ID: B2B.506.2825450.1617906940 *Max-Forwards: 70* Content-Length: 185 User-Agent: OpenSIPS (2.4.6 (x86_64/linux)) Content-Type: application/sdp Supported: 100rel Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, PRACK, INFO Contact: The module documentation says that this header should be passed from the dialog of one side to the other side. Max-Forwards (it is decreased by 1) Thanks in advance, with regards, Alexey Homstr ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPS 3.2 B2B tutorial pre-paid example
Hi, I tested the 3.2 Prepaid example from B2B tutorial https://www.opensips.org/Documentation/Tutorials-B2BUA-3-2 In Scenario Schema there is a first reinvite from OpenSIPS B2B to caller - this reINVITE in my tests comes without SDP. Should not there be an SDP, or am I missing something in this scenario? INVITE sip:TEST001@1.2.3.201:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 1.2.5.183:5060;branch=z9hG4bKb5b7.1ffbf7b1.0 To: ;tag=f79aacce From: ;tag=B2B.81.186.1618177460 CSeq: 3 INVITE Call-ID: 80a9a9cfbe9920f8@1.2.3.201 Max-Forwards: 70 Content-Length: 0 Contact: Thanks, Xaled ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] loose_route with opensips 3.2 B2B module
Hi,
I tried to use the extra_hdrs variable of b2b_client_new to set up the Route
header
extra_hdrs (var, optional) - AVP variable holding a list of extra headers (the
header names) to be added for any request sent to this entity.
But was confused by the "header names" part and it looks like that it will not
accept headers with values. What is then the use of the variable?
xlog("INFO: B2B $b2b_logic.ctx(next_route)");
INFO: B2B Route:
$avp(route_header) = $b2b_logic.ctx(next_route);
b2b_client_new("media", "sip:+87654321@1.1.1.1;user=phone",
"sip:$b2b_logic.ctx(server_ip)", ,$avp(route_header));
ERROR:b2b_logic:b2bl_entity_new: header names without values!
The need to have an avp for extra_hdrs is an unnecessary limitation, or is
there a specific reason for it?
Thanks,
Xaled
-Original Message-
From: Users On Behalf Of xa...@web.de
Sent: Sunday, April 11, 2021 8:33 PM
To: 'OpenSIPS users mailling list'
Subject: [OpenSIPS-Users] loose_route with opensips 3.2 B2B module
Hi,
would it be possible to respect loose routing on the caller side of B2B module
and set route header on the callee side accordingly
Something like this:
Incoming INVITE:
INVITE sip:+12345678@1.2.3.4;user=phone SIP/2.0
Via: SIP/2.0/TCP
4.3.2.1:5060;branch=z9hG4bK40365b83fb9732a054d124bb46252151.7242beeb
Route: ,
Wanted outgoing INVITE:
INVITE sip:+87654321@1.1.1.1;user=phone SIP/2.0
Via: SIP/2.0/TCP
test.com;branch=z9hG4bK40365b83fb9732a054d124bb46252151.7242beeb
Route:
I tried setting loose_route in both the original route and in the resulting b2b
route but without success.
route[b2b_test_logic] {
<-->loose_route();
if ($rm != "BYE") {
# for requests other than BYE, no special actions needs to be done,
# just pass the request to the peer
b2b_pass_request();
exit;
}
...
Thanks,
Xaled
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[OpenSIPS-Users] loose_route with opensips 3.2 B2B module
Hi,
would it be possible to respect loose routing on the caller side of B2B module
and set route header on the callee side accordingly
Something like this:
Incoming INVITE:
INVITE sip:+12345678@1.2.3.4;user=phone SIP/2.0
Via: SIP/2.0/TCP
4.3.2.1:5060;branch=z9hG4bK40365b83fb9732a054d124bb46252151.7242beeb
Route: ,
Wanted outgoing INVITE:
INVITE sip:+87654321@1.1.1.1;user=phone SIP/2.0
Via: SIP/2.0/TCP
test.com;branch=z9hG4bK40365b83fb9732a054d124bb46252151.7242beeb
Route:
I tried setting loose_route in both the original route and in the resulting b2b
route but without success.
route[b2b_test_logic] {
<-->loose_route();
if ($rm != "BYE") {
# for requests other than BYE, no special actions needs to be done,
# just pass the request to the peer
b2b_pass_request();
exit;
}
...
Thanks,
Xaled
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