date:20221207

Re: [SR-Users] Getting the name of an already set onreply_route

2022-12-07 Thread Henning Westerholt

Hello,

give this a try: 
https://www.kamailio.org/wiki/cookbooks/5.5.x/pseudovariables#cfg_key_-_config_file_attributes

Cheers,

Henning

--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com

From: sr-users  On Behalf Of Denys Pozniak
Sent: Wednesday, December 7, 2022 2:12 PM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Getting the name of an already set onreply_route

Here the question is more whether there is a variable ($T or any) into which 
the name of the onreply_route is exported.


ср, 7 дек. 2022 г. в 13:21, Patrick Karton 
mailto:patrickar...@hotmail.com>>:
Or transaction variable (xavp) is best for that.

Le 7 déc. 2022 12:00, Patrick Karton 
mailto:patrickar...@hotmail.com>> a écrit :
Hello i dont think so.

But probably you can use shared or private variable depending on your needs

Le 7 déc. 2022 11:28, Denys Pozniak 
mailto:denys.pozn...@gmail.com>> a écrit :
Hello!

Is it possible to find out the name of the onreply_route that was set before?
Something like this:

t_on_reply("MANAGE_REPLY");
...
if ( t_is_set("onreply_route") ) {
get_onreply_route_name();
...
}

--

BR,
Denys Pozniak




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--

BR,
Denys Pozniak


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Re: [SR-Users] Getting the name of an already set onreply_route

2022-12-07 Thread Denys Pozniak

Here the question is more whether there is a variable ($T or any) into
which the name of the onreply_route is exported.


ср, 7 дек. 2022 г. в 13:21, Patrick Karton :

> Or transaction variable (xavp) is best for that.
>
> Le 7 déc. 2022 12:00, Patrick Karton  a écrit :
>
> Hello i dont think so.
>
> But probably you can use shared or private variable depending on your
> needs
>
> Le 7 déc. 2022 11:28, Denys Pozniak  a écrit :
>
> Hello!
>
> Is it possible to find out the name of the onreply_route that was set
> before?
> Something like this:
>
> t_on_reply("MANAGE_REPLY");
> ...
> if ( t_is_set("onreply_route") ) {
> get_onreply_route_name();
> ...
> }
>
> --
>
> BR,
> Denys Pozniak
>
>
>
>
> __
> Kamailio - Users Mailing List - Non Commercial Discussions
> sr-users@lists.kamailio.org
> Important: keep the mailing list in the recipients, do not reply only to
> the sender!
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>


-- 

BR,
Denys Pozniak
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Re: [SR-Users] KAMCTL error

2022-12-07 Thread Olle E. Johansson

> On 7 Dec 2022, at 11:49, Govind Swamy  wrote:
> 
> 
> 
> Hi
> 
>  am unable to create using kamctl

And you got an error message. Have you checked your configuration?

kamctlrc

/O

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[SR-Users] KAMCTL error

2022-12-07 Thread Govind Swamy

[image: Error in KAMCTL.png]

Hi

 am unable to create using kamctl
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Re: [SR-Users] Getting the name of an already set onreply_route

2022-12-07 Thread Patrick Karton

Or transaction variable (xavp) is best for that.Le 7 déc. 2022 12:00, Patrick Karton  a écrit :Hello i dont think so.But probably you can use shared or private variable depending on your needs Le 7 déc. 2022 11:28, Denys Pozniak  a écrit :Hello!Is it possible to find out the name of the onreply_route that was set before?Something like this:t_on_reply("MANAGE_REPLY");...if ( t_is_set("onreply_route") ) {  get_onreply_route_name();  ...}-- BR,Denys Pozniak
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Re: [SR-Users] Getting the name of an already set onreply_route

2022-12-07 Thread Patrick Karton

Hello i dont think so.But probably you can use shared or private variable depending on your needs Le 7 déc. 2022 11:28, Denys Pozniak  a écrit :Hello!Is it possible to find out the name of the onreply_route that was set before?Something like this:t_on_reply("MANAGE_REPLY");...if ( t_is_set("onreply_route") ) {  get_onreply_route_name();  ...}-- BR,Denys Pozniak
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[SR-Users] Getting the name of an already set onreply_route

2022-12-07 Thread Denys Pozniak

Hello!

Is it possible to find out the name of the onreply_route that was set
before?
Something like this:

t_on_reply("MANAGE_REPLY");
...
if ( t_is_set("onreply_route") ) {
get_onreply_route_name();
...
}

-- 

BR,
Denys Pozniak
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[SR-Users] Fosdem 2023 - CfP

2022-12-07 Thread Daniel-Constantin Mierla

Hello,

should anyone be interested to present or just go to FOSDEM 2023, note
that it is going to take place in Brussels during February 4-5, 2023.

With a increasing demand on topics to be covered, many more devrooms
were accepted and Real Time Communication devroom has half a day
available on the afternoon of February 5, 2023. The RTC Devroom call for
presentations is open, more details at;

  - https://lists.fosdem.org/pipermail/fosdem/2022q4/003479.html

Check also the other accepted devrooms, maybe one suits better for your
work with open source nowadays and you can submit proposals to them:

  - https://fosdem.org/2023/news/2022-11-07-accepted-developer-rooms/

Cheers,
Daniel


-- 
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda


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Re: [SR-Users] Call drops after 1 minute

2022-12-07 Thread Henning Westerholt

Hello,

it looks that the 200 OK is not properly transmitted, as they seem to be 
retransmits.
Maybe you should read a bit about SIP if you have not done it yet. It will help 
you debugging this problem (and probably others in the future). 

Cheers,

Henning

--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com

From: Christian B Wiik 
Sent: Wednesday, December 7, 2022 9:13 AM
To: Henning Westerholt 
Cc: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Call drops after 1 minute

Link to entire trace:

https://docs.google.com/document/d/1yWFJ_Cv13p5cYk-d8m5HMBSeLalkutV0cKZHjHf1QHk/edit?usp=sharing

--
Regards
Christian


ons. 7. des. 2022 kl. 08:57 skrev Henning Westerholt 
mailto:h...@gilawa.com>>:
Hi Christian,

this ACK is the reply to the 407 and not the relevant one for the dialog.

Please have a look to the full SIP dialog.

Cheers,

Henning

--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com

From: Christian B Wiik mailto:c...@itf-as.no>>
Sent: Wednesday, December 7, 2022 8:14 AM
To: Henning Westerholt mailto:h...@gilawa.com>>
Cc: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Call drops after 1 minute

Thanks Henning.

These are the first 3 packets filtering on my user. I see the ACK but I'm not 
able to spot the error.

U 213.52.37.107:50336 -> 
10.1.2.10:5060 #1
  INVITE sip:k...@sip2.itf-as.com 
SIP/2.0..Via: SIP/2.0/UDP 213.52.37.107:35270;rport;branch=z9hG4bKPj398365dc9
  706413f868bdd222cadbed8..Max-Forwards: 70..From: 
mailto:sip%3acbw...@sip2.itf-as.com>>;tag=4183d760c26e4531a7a39f45d1
  4fb4c6..To: 
mailto:sip%3a...@sip2.itf-as.com>>..Contact: 
..Call-ID: b3dd380f0c1d4e
  0ebdd7fc223710d938..CSeq: 23860 INVITE..Route: 
http://sip2.itf-as.com>;lr>..Allow: PRACK, INVITE, ACK, 
BYE, CAN
  CEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS..Supported: 
replaces, 100rel, timer, norefersu
  b..Session-Expires: 1800..Min-SE: 90..User-Agent: 
MicroSIP/3.21.3..Content-Type: application/sdp..Content-Le
  ngth:   345v=0..o=- 3879388988 3879388988 IN IP4 
213.52.37.107..s=pjmedia..b=AS:84..t=0 0..a=X-nat:0..m=
  audio 35276 RTP/AVP 8 0 101..c=IN IP4 
213.52.37.107..b=TIAS:64000..a=rtcp:35277 IN IP4 213.52.37.107..a=send
  recv..a=rtpmap:8 PCMA/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:101 
telephone-event/8000..a=fmtp:101 0-16..a=ssrc
  :1053777612 cname:28d400de4b7d5918..
#
U 10.1.2.10:5060 -> 
213.52.37.107:50336 #2
  SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 
213.52.37.107:35270;rport=50336;branch=z9hG4bKPj
  398365dc9706413f868bdd222cadbed8;received=213.52.37.107..From: 
mailto:sip%3acbw...@sip2.itf-as.com>>;tag=4183d760c26e
  4531a7a39f45d14fb4c6..To: 
mailto:sip%3a...@sip2.itf-as.com>>;tag=9dd61ff61e802d8e2bef5f14621ef3c2.f003010a..Call-ID:
  b3dd380f0c1d4e0ebdd7fc223710d938..CSeq: 23860 INVITE..Proxy-Authenticate: 
Digest realm="sip2.itf-as.com", no
  nce="Y5A72WOQOq3afsXxs6AD2ihlmLAlgNOe"..Server: kamailio (5.6.2 
(x86_64/linux))..Content-Length: 0
#
U 213.52.37.107:50336 -> 
10.1.2.10:5060 #3
  ACK sip:k...@sip2.itf-as.com SIP/2.0..Via: 
SIP/2.0/UDP 213.52.37.107:35270;rport;branch=z9hG4bKPj398365dc9706
  413f868bdd222cadbed8..Max-Forwards: 70..From: 
mailto:sip%3acbw...@sip2.itf-as.com>>;tag=4183d760c26e4531a7a39f45d14fb
  4c6..To: 
mailto:sip%3a...@sip2.itf-as.com>>;tag=9dd61ff61e802d8e2bef5f14621ef3c2.f003010a..Call-ID:
 b3dd380f0c1d4e0eb
  dd7fc223710d938..CSeq: 23860 ACK..Route: 
http://sip2.itf-as.com>;lr>..Content-Length:  0

--
Regards
Christian


ons. 7. des. 2022 kl. 07:51 skrev Henning Westerholt 
mailto:h...@gilawa.com>>:
Hello,

as you’ve guessed, this can be a common problem related to the routing of the 
ACK message.

Have a look e.g. with ngrep or sngrep to the SIP signalisation on the server 
side and check if everything is correct in the SIP messages.


From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Christian B Wiik
Sent: Wednesday, December 7, 2022 7:43 AM
To: sr-users@lists.kamailio.org
Subject: [SR-Users] Call drops after 1 minute

Greetings!

I have a CentOS setup in AWS where all my calls are dropped after about a 
minute or so. I realize this typically is a NAT problem, but I can't see where 
my error is.
Sound is fine both ways.

Kamailio is set with WITH_NAT and I use rtpproxy like this:
OPTIONS="-l 10.1.2.10 -s udp:127.0.0.1:7722 -d 
INFO:LOG_LOCAL5 -m 35010 -M 35110 -A 54.171.168.48"
(10.1.2.10 is the local IP for CentOS)

Tested with MicroSIP and Linphone and tried numerous configurations. It seems 
the

Re: [SR-Users] Call drops after 1 minute

2022-12-07 Thread Christian B Wiik

Link to entire trace:

https://docs.google.com/document/d/1yWFJ_Cv13p5cYk-d8m5HMBSeLalkutV0cKZHjHf1QHk/edit?usp=sharing

-- 
Regards
Christian


ons. 7. des. 2022 kl. 08:57 skrev Henning Westerholt :

> Hi Christian,
>
>
>
> this ACK is the reply to the 407 and not the relevant one for the dialog.
>
>
>
> Please have a look to the full SIP dialog.
>
>
>
> Cheers,
>
>
>
> Henning
>
>
>
> --
>
> Henning Westerholt – https://skalatan.de/blog/
>
> Kamailio services – https://gilawa.com
>
>
>
> *From:* Christian B Wiik 
> *Sent:* Wednesday, December 7, 2022 8:14 AM
> *To:* Henning Westerholt 
> *Cc:* Kamailio (SER) - Users Mailing List 
> *Subject:* Re: [SR-Users] Call drops after 1 minute
>
>
>
> Thanks Henning.
>
>
>
> These are the first 3 packets filtering on my user. I see the ACK but I'm
> not able to spot the error.
>
>
>
> U 213.52.37.107:50336 -> 10.1.2.10:5060 #1
>   INVITE sip:k...@sip2.itf-as.com SIP/2.0..Via: SIP/2.0/UDP
> 213.52.37.107:35270;rport;branch=z9hG4bKPj398365dc9
>   706413f868bdd222cadbed8..Max-Forwards: 70..From: <
> sip:cbw...@sip2.itf-as.com>;tag=4183d760c26e4531a7a39f45d1
>   4fb4c6..To: ..Contact: <
> sip:cbwlap@213.52.37.107:35270;ob>..Call-ID: b3dd380f0c1d4e
>   0ebdd7fc223710d938..CSeq: 23860 INVITE..Route: 
> ..Allow:
> PRACK, INVITE, ACK, BYE, CAN
>   CEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE,
> OPTIONS..Supported: replaces, 100rel, timer, norefersu
>   b..Session-Expires: 1800..Min-SE: 90..User-Agent: 
> MicroSIP/3.21.3..Content-Type:
> application/sdp..Content-Le
>   ngth:   345v=0..o=- 3879388988 3879388988 IN IP4
> 213.52.37.107..s=pjmedia..b=AS:84..t=0 0..a=X-nat:0..m=
>   audio 35276 RTP/AVP 8 0 101..c=IN IP4
> 213.52.37.107..b=TIAS:64000..a=rtcp:35277 IN IP4 213.52.37.107..a=send
>   recv..a=rtpmap:8 PCMA/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:101
> telephone-event/8000..a=fmtp:101 0-16..a=ssrc
>   :1053777612 cname:28d400de4b7d5918..
> #
> U 10.1.2.10:5060 -> 213.52.37.107:50336 #2
>   SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP
> 213.52.37.107:35270;rport=50336;branch=z9hG4bKPj
>   398365dc9706413f868bdd222cadbed8;received=213.52.37.107..From: <
> sip:cbw...@sip2.itf-as.com>;tag=4183d760c26e
>   4531a7a39f45d14fb4c6..To:  >;tag=9dd61ff61e802d8e2bef5f14621ef3c2.f003010a..Call-ID:
>   b3dd380f0c1d4e0ebdd7fc223710d938..CSeq: 23860
> INVITE..Proxy-Authenticate: Digest realm="sip2.itf-as.com", no
>   nce="Y5A72WOQOq3afsXxs6AD2ihlmLAlgNOe"..Server: kamailio (5.6.2
> (x86_64/linux))..Content-Length: 0
> #
> U 213.52.37.107:50336 -> 10.1.2.10:5060 #3
>   ACK sip:k...@sip2.itf-as.com SIP/2.0..Via: SIP/2.0/UDP
> 213.52.37.107:35270;rport;branch=z9hG4bKPj398365dc9706
>   413f868bdd222cadbed8..Max-Forwards: 70..From: <
> sip:cbw...@sip2.itf-as.com>;tag=4183d760c26e4531a7a39f45d14fb
>   4c6..To: 
> ;tag=9dd61ff61e802d8e2bef5f14621ef3c2.f003010a..Call-ID:
> b3dd380f0c1d4e0eb
>   dd7fc223710d938..CSeq: 23860 ACK..Route: 
> ..Content-Length:
>  0
>
>
>
> --
>
> Regards
>
> Christian
>
>
>
>
>
> ons. 7. des. 2022 kl. 07:51 skrev Henning Westerholt :
>
> Hello,
>
>
>
> as you’ve guessed, this can be a common problem related to the routing of
> the ACK message.
>
>
>
> Have a look e.g. with ngrep or sngrep to the SIP signalisation on the
> server side and check if everything is correct in the SIP messages.
>
>
>
>
>
> *From:* sr-users  *On Behalf Of 
> *Christian
> B Wiik
> *Sent:* Wednesday, December 7, 2022 7:43 AM
> *To:* sr-users@lists.kamailio.org
> *Subject:* [SR-Users] Call drops after 1 minute
>
>
>
> Greetings!
>
>
>
> I have a CentOS setup in AWS where all my calls are dropped after about a
> minute or so. I realize this typically is a NAT problem, but I can't see
> where my error is.
>
> Sound is fine both ways.
>
>
>
> Kamailio is set with WITH_NAT and I use rtpproxy like this:
>
> OPTIONS="-l 10.1.2.10 -s udp:127.0.0.1:7722 -d INFO:LOG_LOCAL5 -m 35010
> -M 35110 -A 54.171.168.48"
>
> (10.1.2.10 is the local IP for CentOS)
>
>
>
> Tested with MicroSIP and Linphone and tried numerous configurations. It
> seems the receiving client is not able to verify the call has been set up,
> and disconnects. MicroSIP has the status "Connecting..." until it
> disconnects.
>
>
>
> All tips appreciated. Will post configuration and logs if needed.
>
> Kamailio version 5.6.2 from rpm and rtpproxy 2.1.0 compiled from source.
>
>
>
>
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Re: [SR-Users] Call drops after 1 minute

2022-12-07 Thread Henning Westerholt

Hi Christian,

this ACK is the reply to the 407 and not the relevant one for the dialog.

Please have a look to the full SIP dialog.

Cheers,

Henning

--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com

From: Christian B Wiik 
Sent: Wednesday, December 7, 2022 8:14 AM
To: Henning Westerholt 
Cc: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Call drops after 1 minute

Thanks Henning.

These are the first 3 packets filtering on my user. I see the ACK but I'm not 
able to spot the error.

U 213.52.37.107:50336 -> 
10.1.2.10:5060 #1
  INVITE sip:k...@sip2.itf-as.com 
SIP/2.0..Via: SIP/2.0/UDP 213.52.37.107:35270;rport;branch=z9hG4bKPj398365dc9
  706413f868bdd222cadbed8..Max-Forwards: 70..From: 
mailto:sip%3acbw...@sip2.itf-as.com>>;tag=4183d760c26e4531a7a39f45d1
  4fb4c6..To: 
mailto:sip%3a...@sip2.itf-as.com>>..Contact: 
..Call-ID: b3dd380f0c1d4e
  0ebdd7fc223710d938..CSeq: 23860 INVITE..Route: 
http://sip2.itf-as.com>;lr>..Allow: PRACK, INVITE, ACK, 
BYE, CAN
  CEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS..Supported: 
replaces, 100rel, timer, norefersu
  b..Session-Expires: 1800..Min-SE: 90..User-Agent: 
MicroSIP/3.21.3..Content-Type: application/sdp..Content-Le
  ngth:   345v=0..o=- 3879388988 3879388988 IN IP4 
213.52.37.107..s=pjmedia..b=AS:84..t=0 0..a=X-nat:0..m=
  audio 35276 RTP/AVP 8 0 101..c=IN IP4 
213.52.37.107..b=TIAS:64000..a=rtcp:35277 IN IP4 213.52.37.107..a=send
  recv..a=rtpmap:8 PCMA/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:101 
telephone-event/8000..a=fmtp:101 0-16..a=ssrc
  :1053777612 cname:28d400de4b7d5918..
#
U 10.1.2.10:5060 -> 
213.52.37.107:50336 #2
  SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 
213.52.37.107:35270;rport=50336;branch=z9hG4bKPj
  398365dc9706413f868bdd222cadbed8;received=213.52.37.107..From: 
mailto:sip%3acbw...@sip2.itf-as.com>>;tag=4183d760c26e
  4531a7a39f45d14fb4c6..To: 
mailto:sip%3a...@sip2.itf-as.com>>;tag=9dd61ff61e802d8e2bef5f14621ef3c2.f003010a..Call-ID:
  b3dd380f0c1d4e0ebdd7fc223710d938..CSeq: 23860 INVITE..Proxy-Authenticate: 
Digest realm="sip2.itf-as.com", no
  nce="Y5A72WOQOq3afsXxs6AD2ihlmLAlgNOe"..Server: kamailio (5.6.2 
(x86_64/linux))..Content-Length: 0
#
U 213.52.37.107:50336 -> 
10.1.2.10:5060 #3
  ACK sip:k...@sip2.itf-as.com SIP/2.0..Via: 
SIP/2.0/UDP 213.52.37.107:35270;rport;branch=z9hG4bKPj398365dc9706
  413f868bdd222cadbed8..Max-Forwards: 70..From: 
mailto:sip%3acbw...@sip2.itf-as.com>>;tag=4183d760c26e4531a7a39f45d14fb
  4c6..To: 
mailto:sip%3a...@sip2.itf-as.com>>;tag=9dd61ff61e802d8e2bef5f14621ef3c2.f003010a..Call-ID:
 b3dd380f0c1d4e0eb
  dd7fc223710d938..CSeq: 23860 ACK..Route: 
http://sip2.itf-as.com>;lr>..Content-Length:  0

--
Regards
Christian


ons. 7. des. 2022 kl. 07:51 skrev Henning Westerholt 
mailto:h...@gilawa.com>>:
Hello,

as you’ve guessed, this can be a common problem related to the routing of the 
ACK message.

Have a look e.g. with ngrep or sngrep to the SIP signalisation on the server 
side and check if everything is correct in the SIP messages.


From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Christian B Wiik
Sent: Wednesday, December 7, 2022 7:43 AM
To: sr-users@lists.kamailio.org
Subject: [SR-Users] Call drops after 1 minute

Greetings!

I have a CentOS setup in AWS where all my calls are dropped after about a 
minute or so. I realize this typically is a NAT problem, but I can't see where 
my error is.
Sound is fine both ways.

Kamailio is set with WITH_NAT and I use rtpproxy like this:
OPTIONS="-l 10.1.2.10 -s udp:127.0.0.1:7722 -d 
INFO:LOG_LOCAL5 -m 35010 -M 35110 -A 54.171.168.48"
(10.1.2.10 is the local IP for CentOS)

Tested with MicroSIP and Linphone and tried numerous configurations. It seems 
the receiving client is not able to verify the call has been set up, and 
disconnects. MicroSIP has the status "Connecting..." until it disconnects.

All tips appreciated. Will post configuration and logs if needed.
Kamailio version 5.6.2 from rpm and rtpproxy 2.1.0 compiled from source.

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