When receiving a call I get the following message:
INVITE sip:123456789@192.168.1.118:29363 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.118:65178;branch=z9hG4bK-d87543-4a4a6b6c49541b2c-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:96341@192.168.1.118:65178>
To: "123456789"<sip:123456789@192.168.1.119:29363 >
From: "96341"<sip:96341@192.168.1.119:29363 >;tag=af3c7c31
Call-ID: OTFhN2M2YmRmNmU2N2I1ZmQxNmM4ODg4YzRiZmQyNTc.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1011d stamp 40820
Content-Length: 339
v=0
o=- 3 2 IN IP4 192.168.1.118
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.1.118
t=0 0
m=audio 32170 RTP/AVP 0 8 18 101
a=alt:1 1 : +YhwgPgy ayLEx4wo 192.168.1.118 32170
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
How to print the media IP in SDP separately in the log? I didn't find the Pseudo-Variables..can you help me
Hi
There were two PUBLISH requests were sent by the Kamailio Proxy server to
the Kamailio Presence server.
Processing of the first PUBLISH request resulted in the presentity table
being updated - the "etag" changed from "a.1705053846.16611.21.1" to
"a.1705053846.16613.23.2"
Processing of the second PUBLISH failed because it also attempted to update
the etag in the presentity table but at this point in time there was no
match to the original etag value in the database
Logs in kamailio.log:
------
ps_db_update_presentity(): No E-Tag match a.1705053846.16611.21.1
DEBUG: tm [t_reply.c:637]: _reply_light(): reply sent out -
buf=0x7f81e5ca5fc0: SIP/2.0 412 Conditio... shmem=0x7f81de6da068:
SIP/2.0 412 Conditional request failed.
Will any one suggest how to resolve the issue of 412 conditional request
failed.
Dear List,
I’m trying to set up the dialog module and I keep getting the error "bad sip message or missing Contact hdr”:
---
Jan 23 11:34:17 server.example.com kamailio[2495]: 16(2511) ERROR: {1 78451036 INVITE 2ef1d082-3486-123d-0abc-02c8e9c6c24e} dialog [dlg_handlers.c:219]: populate_leg_info(): bad sip message or missing Contact hdr
Jan 23 11:34:17 server.example.com kamailio[2495]: 16(2511) ERROR: {1 78451036 INVITE 2ef1d082-3486-123d-0abc-02c8e9c6c24e} dialog [dlg_handlers.c:952]: dlg_new_dialog(): could not add further info to the dialog
---
This is for MRCP traffic and the SIP INVITEs do not contain the Contact header. They look like this:
----
INVITE sip:some.mrcp.server.example.com:8060 SIP/2.0
Via: SIP/2.0/TCP 10.10.10.10:5099;branch=z9hG4bK5Kpey3r13SBNQ
Max-Forwards: 70
From: <sip:10.10.10.10:5099>;tag=134a3eKD2a8Ua
To: <sip:some.mrcp.server.example.com:8060>
Call-ID: 051df0ff-3541-123d-f090-02f373ff014c
CSeq: 78491159 INVITE
User-Agent: google_stt
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE
Supported: timer, 100rel
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 329
v=0
o=FreeSWITCH 2474358185109188492 5282339568667506935 IN IP4 10.10.10.10
s=-
c=IN IP4 10.10.10.10
t=0 0
m=application 9 TCP/MRCPv2 1
a=setup:active
a=connection:new
a=resource:speechrecog
a=cmid:1
m=audio 12034 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 L16/8000
a=sendonly
a=mid:1
----
According to the RFC, the Contact header is not mandatory (https://datatracker.ietf.org/doc/html/rfc3261#section-8.1.1) and I don’t have an easy way of adding the Contact header to the INVITEs. Is there a way to configure Kamailio’s dialog module to do without the Contact header? Any pointers appreciated.
Thank you so much
With best wishes,
Unai Rodriguez
Hi List
Via sql_xquery I get a stacked xavp aka array.
I would like to store that stacked xavp into a hash table for later usage in other transactions. Is this
possible?
Via xarp_parameters explode or implode does not work as that would stringy the same key multiple times.
Mit freundlichen Grüssen
-Benoît Panizzon-
--
I m p r o W a r e A G - Leiter Commerce Kunden
______________________________________________________
Zurlindenstrasse 29 Tel +41 61 826 93 00
CH-4133 Pratteln Fax +41 61 826 93 01
Schweiz Web http://www.imp.ch
______________________________________________________
Hi Gang
I would like to set a variable specific to the branch. Is there a way?
append_branch($var(aor1));
$(branch(dst_uri)[-1]) = $var($dst1)
$(branch(want_plus)[-1]) = 1;
append_branch($var(aor2));
$(branch(dst_uri)[-1]) = $var($dst2)
$(branch(want_plus)[-1]) = 0;
t_on_branch("BR_T");
branch_route[BR_T]
{
if ($branch(want_plus))
{
$rU = "+" + $rU;
}
}
Or would there be a way to query the index of the added branch so I
could stack an avp with that index to access in the branch route?
Mit freundlichen Grüssen
-Benoît Panizzon-
--
I m p r o W a r e A G - Leiter Commerce Kunden
______________________________________________________
Zurlindenstrasse 29 Tel +41 61 826 93 00
CH-4133 Pratteln Fax +41 61 826 93 01
Schweiz Web http://www.imp.ch
______________________________________________________
Hey everyone,
I've set it up so that when the invite comes into Kamailio I suspend the
transaction (save the transaction info in sht) and use dispatcher to send a
request to our NSS server. On the reply from NSS I throw the identity
header into a sht and continue the original transaction. Does that seem
like the right way to do it?
The next things I need to look at are:
* if first NSS server doesn't respond quickly, try the second (change some
timeouts and use dispatcher to try the backup)
* if both NSS servers aren't working, I still want to resume the original
call and let it through. Maybe I can keep track of the failures in a
sht and if I have two failures, resume the original invite without the
identity header.
Sound like an ok plan?
--
Anthony Wittig
Mango Voice Developer
>
> ---------- Forwarded message ----------
> From: Anthony Wittig <awittig(a)mangovoice.com>
> To: sr-users(a)lists.kamailio.org, david.villasmil.work(a)gmail.com
> Cc:
> Bcc:
> Date: Tue, 23 Jan 2024 20:39:09 -0700
> Subject: [SR-Users] STIR/SHAKEN
> Thanks!
>
> This is what I've got so far:
>
> 1. Kamailio invite -> NSS
> 2. NSS 302 -> Kamailio
> 3. Kamailio ack -> NSS
>
> I can see the identity header on the 302 response in my onreply_route. How
> can I replay the same invite from step #1 with the identity header I get
> back in step #2? Should I be doing something with a transaction on step #1,
> maybe start a branch that goes out to NSS and when it comes back resume the
> transaction (these are all words I've heard smart people say, but I don't
> fully understand them...).
>
> Any pointers would be greatly appreciated, thanks!
> --
> Anthony Wittig
> Mango Voice Developer
>
> ---------- Forwarded message ----------
> From: David Villasmil <david.villasmil.work(a)gmail.com>
> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org>
> Cc:
> Bcc:
> Date: Tue, 23 Jan 2024 14:56:52 +0100
> Subject: [SR-Users] Re: STIR/SHAKEN
> Ditto, i do first approach.
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work(a)gmail.com
> phone: +34669448337
>
>
> On Tue, Jan 23, 2024 at 2:38 PM Alex Balashov via sr-users <
> sr-users(a)lists.kamailio.org> wrote:
>
>> These are all valid approaches, depending on preference. Catching 302s
>> and extracting exactly the desired info might be easier with Kamailio, so I
>> suppose, ceteris paribus, I'd recommend the first one.
>>
>> Dispatcher is a great approach!
>>
>> > On 23 Jan 2024, at 00:01, Anthony Wittig via sr-users <
>> sr-users(a)lists.kamailio.org> wrote:
>> >
>> > Hello,
>> >
>> > We're attempting to add identities to our invites. I believe we want to
>> send our invites to the NSS, and it'll reply with a 302'd invite with an
>> identity header. We currently use FreeSWITCH for dialplans and media and
>> Kamailio as our SBC.
>> >
>> > Should we be trying for something like: FreeSWITCH -> Kamailio -> NSS
>> (302) -> Kamailio -> PSTN
>> >
>> > or would it make more sense to do: FreeSWITCH -> NSS (302) ->
>> FreeSWITCH -> Kamailio -> PSTN
>> >
>> > Assuming we wanted Kamailio -> NSS, is using dispatcher the right
>> approach? We currently use it for our two Kamailios and our FreeSWITCHs.
>> We're going to have two NSS servers, one for failover.
>> >
>> > Any thoughts would be greatly appreciated, thanks!
>> >
>> > --
>> > Anthony Wittig
>> > Mango Voice Developer
>> > __________________________________________________________
>> > Kamailio - Users Mailing List - Non Commercial Discussions
>> > To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org
>> > Important: keep the mailing list in the recipients, do not reply only
>> to the sender!
>> > Edit mailing list options or unsubscribe:
>>
>> --
>> Alex Balashov
>> Principal Consultant
>> Evariste Systems LLC
>> Web: https://evaristesys.com
>> Tel: +1-706-510-6800
>>
>
Thanks!
This is what I've got so far:
1. Kamailio invite -> NSS
2. NSS 302 -> Kamailio
3. Kamailio ack -> NSS
I can see the identity header on the 302 response in my onreply_route. How
can I replay the same invite from step #1 with the identity header I get
back in step #2? Should I be doing something with a transaction on step #1,
maybe start a branch that goes out to NSS and when it comes back resume the
transaction (these are all words I've heard smart people say, but I don't
fully understand them...).
Any pointers would be greatly appreciated, thanks!
--
Anthony Wittig
Mango Voice Developer
---------- Forwarded message ----------
From: David Villasmil <david.villasmil.work(a)gmail.com>
To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org>
Cc:
Bcc:
Date: Tue, 23 Jan 2024 14:56:52 +0100
Subject: [SR-Users] Re: STIR/SHAKEN
Ditto, i do first approach.
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
On Tue, Jan 23, 2024 at 2:38 PM Alex Balashov via sr-users <
sr-users(a)lists.kamailio.org> wrote:
> These are all valid approaches, depending on preference. Catching 302s and
> extracting exactly the desired info might be easier with Kamailio, so I
> suppose, ceteris paribus, I'd recommend the first one.
>
> Dispatcher is a great approach!
>
> > On 23 Jan 2024, at 00:01, Anthony Wittig via sr-users <
> sr-users(a)lists.kamailio.org> wrote:
> >
> > Hello,
> >
> > We're attempting to add identities to our invites. I believe we want to
> send our invites to the NSS, and it'll reply with a 302'd invite with an
> identity header. We currently use FreeSWITCH for dialplans and media and
> Kamailio as our SBC.
> >
> > Should we be trying for something like: FreeSWITCH -> Kamailio -> NSS
> (302) -> Kamailio -> PSTN
> >
> > or would it make more sense to do: FreeSWITCH -> NSS (302) -> FreeSWITCH
> -> Kamailio -> PSTN
> >
> > Assuming we wanted Kamailio -> NSS, is using dispatcher the right
> approach? We currently use it for our two Kamailios and our FreeSWITCHs.
> We're going to have two NSS servers, one for failover.
> >
> > Any thoughts would be greatly appreciated, thanks!
> >
> > --
> > Anthony Wittig
> > Mango Voice Developer
> > __________________________________________________________
> > Kamailio - Users Mailing List - Non Commercial Discussions
> > To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org
> > Important: keep the mailing list in the recipients, do not reply only to
> the sender!
> > Edit mailing list options or unsubscribe:
>
> --
> Alex Balashov
> Principal Consultant
> Evariste Systems LLC
> Web: https://evaristesys.com
> Tel: +1-706-510-6800
>
> __________________________________________________________
> Kamailio - Users Mailing List - Non Commercial Discussions
> To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org
> Important: keep the mailing list in the recipients, do not reply only to
> the sender!
> Edit mailing list options or unsubscribe:
>
Hello,
We're attempting to add identities to our invites. I believe we want to
send our invites to the NSS, and it'll reply with a 302'd invite with an
identity header. We currently use FreeSWITCH for dialplans and media and
Kamailio as our SBC.
Should we be trying for something like: FreeSWITCH -> Kamailio -> NSS (302)
-> Kamailio -> PSTN
or would it make more sense to do: FreeSWITCH -> NSS (302) -> FreeSWITCH ->
Kamailio -> PSTN
Assuming we wanted Kamailio -> NSS, is using dispatcher the right approach?
We currently use it for our two Kamailios and our FreeSWITCHs. We're going
to have two NSS servers, one for failover.
Any thoughts would be greatly appreciated, thanks!
--
Anthony Wittig
Mango Voice Developer
Hi gang
location database contains multiple contacts for an AOR
The invite therefore is parallel branched to those multiple contacts.
Each reply from each branch is handled by MANAGE_REPLY.
rtpengine_manage() is called in MANAGE_REPLY to stop rtpengine
processing a failed call.
Unfortunately, if any of the contacts replies with an error
rtpengine_manage() will stop RTP processing.
Is there a way I could check if there are branches with a pending reply
from within MANAGE_REPLY so I could call rtpengine_manage() only when
the last reply (no pending replies) are present?
Hmm, realizing, if I process the last pending error reply and a prior
one got a positive reply, this would still kill media.
=> How do I solve this?
Mit freundlichen Grüssen
-Benoît Panizzon-
--
I m p r o W a r e A G - Leiter Commerce Kunden
______________________________________________________
Zurlindenstrasse 29 Tel +41 61 826 93 00
CH-4133 Pratteln Fax +41 61 826 93 01
Schweiz Web http://www.imp.ch
______________________________________________________
Hello,
a short note to inform that the Call For Presentations is now open for
Kamailio World 2024. Everyone is welcome to submit proposal for
presentations to share the knowledge about Kamailio or Real Time
Communication services, security, high availability, scalability, etc.
Submission form and more details are available at:
- https://www.kamailioworld.com/k2024/call-for-speakers/
Cheers,
Daniel
--
Daniel-Constantin Mierla (@ asipto.com)
twitter.com/miconda -- linkedin.com/in/miconda
Kamailio Consultancy, Training and Development Services -- asipto.com
Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com