Hi @miconda
Done, PR #1859
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Kamailio (SER) - Development
Type Of Change
- [x] Small bug fix (non-breaking change which fixes an issue)
- [ ] New feature (non-breaking change which adds new functionality)
- [ ] Breaking change (fix or feature that would change existing functionality)
Checklist:
- [ ] PR should be backported to stable bran
Hi All.
@duarterocha91, can you please check with change from commit
https://github.com/sergey-vb/kamailio/commit/34fb47f28948e701c3ea98c673c8fb324b2f790e
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Thank you Daniel.
Now it works as expected.
Did you will backport this change into kamailio 5.1?
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@miconda I think correct way will be:
```
tps_remove_headers(msg, HDR_RECORDROUTE_T);
if(msg->first_line.u.reply.statuscode>100
&& msg->first_line.u.reply.statuscode<200
&& msg->contact==NULL) {
@miconda Please check latest commit.
It brakes 180 and 183 responses with Contact header
```
/* keep contact without updates for redirect responses sent out */
** if(msg->first_line.u.reply.statuscode<300
** || msg->first_line.u.reply.statuscode>=400) {
Just trued with latest commit:
```
SIP/2.0 180 Ringing
From: ;tag=08001964526856
To: ;tag=vy~HzZJ
Call-ID: vqeU544121321136bcGhEfCjEbh
CSeq: 672321 INVITE
Content-Length: 0
Via: SIP/2.0/UDP
KITS1.MSS.LIFE.COM:5060;rport=5060;received=10.56.63.10;branch=z9hG4bK0017483123160717
Record-Rou
Thank you Daniel.
I think skip adding the Contact is not needed.
I have Ericsson STP, and it sends BYE even at early dialog phase.
But I think I can change this behavior by myself.
Thank you for fixing issue with "180 Ringing" without Contact.
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I agree with you.
But now I see that after
```
SIP/2.0 180 Ringing
From: ;tag=14003117305244
To: ;tag=S7nIf9Z
Call-ID: 0KaR2073911201UQbcGhEfDeMek
CSeq: 669313 INVITE Content-Length: 0
Via: SIP/2.0/UDP
1.2.3.4:5060;rport=5060;received=10.56.63.10;branch=z9hG4bK0005167007511555
Contact:
Hi Daniel.
I have try. Now "180 ringing" is sending to SBC by kamailio.
But kamailio adds "Contact" header to forwarded message.
Can you look, please?
Linphone to kamailio
```
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS
1.2.3.4:5061;branch=z9hG4bK460b.6e02acd9418a5aae97a9d5f79aeffc53.0
From: ;tag=1400
Sorry for delay.
But according to doc for textops module:
Set multipart body to a SIP message. If called with **no parameters, will
convert present body** to multipart.
So if you have look at my part of CFG:
1. I use **set_body_multipart(,"2123894789_1257887457");** to convert Corrent
current
> This may be related to log
>
> ```
> 16(27) ERROR: {1 21 INVITE 1851253425} [core/msg_translator.c:3290]:
> sip_msg_apply_changes(): cannot apply msg changes after adding record-route
> header - it breaks conditional 2nd header
> ```
>
> Think in this case must be send original INVITE.
Hi.
### Description
Kamailio with TOPOS enabled acts as a proxy between SIP client and SBC.
When client sends "180 Ringing" without Contact header - kamailio does not
forward it to SBC.
If Contact header exists in "180 Ringing" - all works as expected.
### Troubleshooting
Reproduction
Load an
@sergey-vb pushed 1 commit.
0030113 sipt: remove some debugging info
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Hi Daniel.
I have done adding new variables as a part of $sipt()
Now they are:
$sipt(redirection_info)
$sipt(redirection_number)
$sipt(redirection_number.nai)
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@sergey-vb pushed 1 commit.
2cbac29 Documenting new variables as part of $sipt(...)
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@sergey-vb pushed 1 commit.
4408cbf New variables as part of $sipt(...)
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In progress.
I have some problem with returning number from sipt_get_redirection_number
function.
I can print it in c code using
LM_INFO("REDIRECTION NUMBER : %s end \n", sb_s.s);
But when i do
pv_get_strval(msg, param, res, &sb_s);
I got some binary value in kamailio script.
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Hi Daniel,
I will try to do this.
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Best regards,
Sergey Basov e-mail: sergey.v.ba...@gmail.com
2018-04-25 15:57 GMT+03:00 Daniel-Constantin Mierla <
notificati...@github.com>:
> @sergey-vb <https://github.com/sergey-vb> - would you consider having the
>
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