Hello,
I've got an odd issue where:
1. TCP client makes call out through Kamailio.
2. Call is answered, TCP client sends e2e ACK.
3. Parser chokes on ACK, closes TCP connection (SYN+FIN).
4. Subsequent messages from the client, immediately following the ACK
(about 240 ms later), come through because the client is halfway around
the world - about 250 ms away - and is not yet aware the connection has
been closed. Otherwise, it would presumably try to re-establish the
connection before …
[View More]sending anything.
Anyway, messages from the parser concerning the ACK are:
---
Aug 4 21:55:57 sip-proxy /usr/local/kamailio/sbin/kamailio[14680]:
INFO: <core> [parser/parse_fline.c:144]: parse_first_line():
ERROR:parse_first_line: method not followed by SP
Aug 4 21:55:57 sip-proxy /usr/local/kamailio/sbin/kamailio[14680]:
ERROR: <core> [parser/parse_fline.c:257]: parse_first_line():
parse_first_line: bad message (offset: 0)
Aug 4 21:55:57 sip-proxy /usr/local/kamailio/sbin/kamailio[14680]:
ERROR: <core> [parser/msg_parser.c:690]: parse_msg(): ERROR: parse_msg:
message=<0#015#012a=rtcp:4007 IN IP4
1.1.1.1:1071#015#012a=sendrecv#015#012a=rtpmap:0
PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:96
telephone-event/8000#015#012a=fmtp:96 0-16#015#012ACK
sip:*98*0b3ff883-a89d-4317-b758-8979682f5357@10.0.2.53:5060;transport=udp
SIP/2.0#015#012Via: SIP/2.0/TCP
1.1.1.1:61979;branch=z9hG4bKPjc15bf9fdded940cab7f859634b9f036e;alias#015#012Max-Forwards:
70#015#012From:
sip:4b57be8d-267d-48c4-aa82-0f3a269d74cb@sip.evaristesys.com;tag=dd1d7fd0344d47b7b22c4017091df17a#015#012To:
sip:*98*0b3ff883-a89d-4317-b758-8979682f5357@sip.evaristesys.com;tag=2rZDKcBFmU65S#015#012Call-ID:
5ba5a84451074159aaaa931eb1f98fe5#015#012CSeq: 3071 ACK#015#012Route:
<sip:2.2.2.2;transport=tcp;lr;r2=on;ftag=dd1d7fd0344d47b7b22c4017091df17a;rtp_set_id=1>#015#012Route:
<sip:10.0.2.119;lr;r2=on;ftag=dd1d7fd0344d47b7b22c4017091df17a;rtp_set_id=1>#015#012Content-Length:
0#015#012#015#012>
Aug 4 21:55:57 sip-proxy /usr/local/kamailio/sbin/kamailio[14680]:
ERROR: <core> [receive.c:173]: receive_msg(): core parsing of SIP
message failed (1.1.1.1:61979/2)
---
That doesn't look like a clean message buffer to me. The message should
start here:
ACK
sip:*98*0b3ff883-a89d-4317-b758-8979682f5357@10.0.2.53:5060;transport=udp
SIP/2.0#015#012Via: SIP/2.0/TCP ...
But instead it's got these SDP attributes prepended:
0#015#012a=rtcp:4007 IN IP4
1.1.1.1:1071#015#012a=sendrecv#015#012a=rtpmap:0
PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:96
telephone-event/8000#015#012a=fmtp:96 0-16#015#012
These SDP attribute appear to have fallen off the back of the SDP offer
in the initial INVITE:
v=0
o=- 3679365582 3679365582 IN IP4 10.0.0.200
s=pjmedia
t=0 0
m=audio 4006 RTP/AVP 0 8 96
c=IN IP4 1.1.1.1
The relationship between the 4006 RTP port and the 4007 RTCP port
suggests that these two things belong together.
This is on kamailio 4.4.0 (x86_64/linux) d4f23c. Any help would be
greatly appreciated! The issue is consistently reproducible and this
happens every time.
Cheers,
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE, Suite 700
Atlanta, GA 30309
United States
Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
[View Less]
Hello List,
I wonder if there is a way to define synonyms. For example, I there is a
person by the name of John Smith with extention 5555 I would like to have a
user john_smith that will also be available if some INVITE 5555.
Thanks,
Nahum
Hi,
I am still struggling to get a prepaid system for my service.
After installing SIREMIS, I am able to get the CDRs updated. But even
after adding the cron settings, the call duration doesn't get updated.
And also in
http://lists.sip-router.org/pipermail/sr-users/2012-January/071517.html
it has been mentioned that
"If it is the case of 1 call per user at a time, then practically can
be done from kamailio config only. When call starts, compute the max
duration for that call based on …
[View More]caller/destination (e.g., call a
stored procedure via sqlops) and set the dialog timeout to that
duration. When call ends (BYE in main route or timeout route) update
the credit accordingly."
Can somebody let me know if my understanding is correct.
1) Add a field called "credits" in the database. And topup that field.
2) First compute max permitted duration for that user by multiplying
the cost for number prefix (say +91) and pulse.
3) And finally add this value to timeout of dialog. modparam("dialog",
"default_timeout", 100000)
$dlg_ctx(timeout_bye) = 1;
Will this work? Any help would be appreciated.
regards
Ganesh Kumar
[View Less]
Hello,
I have this scheme: SIP Carrier (external IP 22.203) < > (local ip 1.21)
mikrotik < > (local ip 1.21) Kamailio - (local ip 1.15) Asterisk.
Carrier does not know what is Route header. that is why Carrier send ACK
directly to 1.15 and fail.
I change Contact at Kamailio to 1.21 then ACK is coming to Kamailio.
But i think it's wrong way...
How can i get ACK from CArrier to Kamalio and route it in right way?
Thanks
Hello,
I have this scheme: SIP Carrier (external IP 22.203) < > (local ip 1.21)
mikrotik < > (local ip 1.21) Kamailio - (local ip 1.15) Asterisk.
Carrier does not know what is Route header. that is why Carrier send ACK
directly to 1.15 and fail.
I change Contact at Kamailio to 1.21 then ACK is coming to Kamailio.
But i think it's wrong way...
How can i get ACK from CArrier to Kamalio and route it in right way?
Thanks
Hello all
i'm having the same error on kam 4.4 when doing
set_body_multipart();
msg_apply_changes();
i see in the logs
5(17797) DEBUG: textops [textops.c:1577]: set_multibody_helper():
delimiter<17>:[unique-boundary-1]
5(17797) DEBUG: textops [textops.c:1486]: generate_boundary(): adding
final CRLF+CRLF
5(17797) DEBUG: textops [textops.c:1714]: set_multibody_helper():
content-type<44>:[multipart/mixed;boundary="unique-boundary-1"]
5(17797) DEBUG: textops [textops.c:1768]: …
[View More]set_multibody_helper(): set
flag FL_BODY_MULTIPART
5(17797) INFO: <core> [msg_translator.c:1692]: get_boundary():
Content-Type hdr has no params <application/sdp>
5(17797) WARNING: <core> [msg_translator.c:1958]:
build_req_buf_from_sip_req(): check_boundaries error
5(17797) DEBUG: <core> [msg_translator.c:422]: clen_builder():
content-length: 191 (191)
the original INVITE is like
....
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 104.
P-Asserted-Identity: "10707334" <sip:10707334@3.3.3.3>.
.
v=0.
o=user1 53655765 23536 IN IP4 79.170.68.171.
s=-.
c=IN IP4 2.2.2.2.
t=0 0.
m=audio 6001 RTP/AVP 8.
....
do you have any idea why these messages appear
thanks a lot and regards
david
[View Less]
Hello,
if you know the ip address associated with the network interfaces, you
can use keyword dst_ip to check which of the ip addresses was used to
receive the packet (or use $Ri instead of dst_ip).
Cheers,
Daniel
On 05/08/16 08:41, Marino Mileti wrote:
>
> Hi guys,
>
>
>
> I've my Kamailio box that works great J
>
>
>
> Now I would like to enable wifi on the board (in AP mode) in order to
> allow mobile devices to connect and receive and place calls.
&…
[View More]gt;
> Is it possible? Is it possible to configure Kama like a "gateway"
> between two network cards?
>
>
>
> I've find a guide regarding gateway between IPv4 and IPV6 but i can't
> use "af" to discriminate message between eth0 or wlan0. Is there a
> command to detect if a message is received from a specific interface card?
>
>
>
> Any help or any cfg example is very appreciate J
>
>
>
> /Marino Maria Mileti///
>
> /marino.mileti(a)alice.it <mailto:marino.mileti@alice.it>/
>
> / /
>
> /cid:006a01cb6b0e$67eecdae$_CDOSYS2.0//Reduce your energy consumption
> and keep polar bears on ice!/
>
>
>
>
>
> ------------------------------------------------------------------------
> Avast logo
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…>
>
>
> Questa e-mail è stata controllata per individuare virus con Avast
> antivirus.
> www.avast.com
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…>
>
>
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla
http://www.asipto.com - http://www.kamailio.orghttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
[View Less]
Hi guys,
I've my Kamailio box that works great J
Now I would like to enable wifi on the board (in AP mode) in order to allow
mobile devices to connect and receive and place calls.
Is it possible? Is it possible to configure Kama like a "gateway" between
two network cards?
I've find a guide regarding gateway between IPv4 and IPV6 but i can't use
"af" to discriminate message between eth0 or wlan0. Is there a command to
detect if a message is received from a specific interface card?
…
[View More]
Any help or any cfg example is very appreciate J
Marino Maria Mileti
<mailto:marino.mileti@alice.it> marino.mileti(a)alice.it
cid:006a01cb6b0e$67eecdae$_CDOSYS2.0Reduce your energy consumption and keep
polar bears on ice!
---
Questa e-mail è stata controllata per individuare virus con Avast antivirus.
https://www.avast.com/antivirus
[View Less]
Hi Guys,
Looking for some help on configuring RTP Engine with Dispatcher on my Kamailio Box
Please see this paste for the kamailio configuration = http://pastebin.com/yZzSEJmK
When I run a call the dispatcher does it job perfectly but the rtpengine is not been used does anyone know why
--
CONFIDENTIAL EMAIL FROM NETCALL TELECOM LIMITED
This email, and any attachments, is intended only for the above addressee.
It may contain private and/or confidential information. If you have
…
[View More]received this email in error you are on notice of its status, please
immediately notify the sender by return email then delete this message and
any attachments. If you are not the addressee, except to notify the sender,
you must not use, disclose, copy or distribute this email and/or its
attachments. Netcall Telecom accepts no responsibility for any changes made
to this message after it has been sent by the original author. Opinions or
views expressed in this email may be those of the individual sender and not
Netcall Telecom. Nothing in this email shall bind Netcall Telecom in any
contract or obligation
Netcall Telecom Ltd Registered in England 2831215. Registered Office : 3rd
Floor, Hamilton House, 111 Marlowes, Hemel Hempstead, Herts, HP1 1BB
[View Less]