Hello,
i'm encountering a matter while receiving an INVITE from a third-party
proxy module to the Kamailio s-cscf module (version: kamailio 4.1.0
(x86_64/linux) 350d2e)
Accordling to the RFC3325 section 9.1 The P-Asserted-Identity Header,
> The P-Asserted-Identity header field is used among trusted SIP
> entities (typically intermediaries) to carry the identity of the user
> sending a SIP message as it was verified by authentication.
> PAssertedID = "P-Asserted-Identity" HCOLON PAssertedID-value
> *(COMMA PAssertedID-value)
> PAssertedID-value = name-addr / addr-spec
> A P-Asserted-Identity header field value MUST consist of exactly one
> name-addr or addr-spec.
In our scenario the received INVITE contains a P-Asserted-Identity
header with an addr-spec value format, so without angle brackets, as
described in RFC2822
In Kamailio log I see that it rejects this header asserting it is malformed.
>
> Dec 9 14:22:57 IMSCore-ve /usr/sbin/kamailio[2410]: DEBUG: <core> [mem/q_malloc.c:369]: qm_malloc(): qm_malloc(0x7f271cded010, 72) called from <core>: parser/parse_rr.c: do_parse_rr_body(72)
> Dec 9 14:22:57 IMSCore-ve /usr/sbin/kamailio[2410]: DEBUG: <core> [mem/q_malloc.c:415]: qm_malloc(): qm_malloc(0x7f271cded010, 72) returns address 0x7f271cebf510 frag. 0x7f271cebf4e0 (size=72) on 1 -th hit
> Dec 9 14:22:57 IMSCore-ve /usr/sbin/kamailio[2410]: ERROR: <core> [parser/parse_rr.c:82]: do_parse_rr_body(): parse_rr(): Error while parsing name-addr (sip:+39******10@ims.example.net)
> Dec 9 14:22:57 IMSCore-ve /usr/sbin/kamailio[2410]: DEBUG: <core> [mem/q_malloc.c:439]: qm_free(): qm_free(0x7f271cded010, 0x7f271cebf510), called from <core>: parser/parse_rr.c: do_parse_rr_body(141)
> Dec 9 14:22:57 IMSCore-ve /usr/sbin/kamailio[2410]: DEBUG: <core> [mem/q_malloc.c:474]: qm_free(): qm_free: freeing frag. 0x7f271cebf4e0 alloc'ed from <core>: parser/parse_rr.c: do_parse_rr_body(72)
> Dec 9 14:22:57 IMSCore-ve /usr/sbin/kamailio[2410]: CRITICAL: <core> [ims_getters.c:408]: cscf_get_asserted_identity(): WARN:cscf_get_asserted_identity: P-Asserted-Identity header must contain a Nameaddr!!! Fix the client!
Can you confirm this issue?
--
Flavio Battimo
flavio.battimo(a)gmail.com
fbattimo(a)meetecho.com
skype: flaviobattimo
Hello,
I'm getting the following error when compiling kamailio 4.1.
CC (gcc) [M gzcompress.so] gzcompress_mod.o
gzcompress_mod.c:42:18: fatal error: zlib.h: No such file or directory
compilation terminated.
make[1]: *** [gzcompress_mod.o] Error 1
make: *** [modules] Error 1
The modules/gzcompress directory only contains the following items:
doc/
gzcompress_mod.c
makecfg.lst
Makefile
README
I installed zlibc, but that didn't work either.
Any ideas?
Hi all
I am looking for a free billing solution usable with Kamailio.
I currently use the sp from siremis - so far it works fine!
I need to get a solution where I can add credit limits for users/trunks or credit limits for carriers (eg. If we made a prepayment).
I thing JBilling could be great - but I think the Telco version will cost a monthly fee..
Other ideas or solutions?
Btw:
The invoice itself will be made in an ERP system - so I actually just need the rating and credit-management to handle the routings.
Regards,
Oli
Hi all,
I´m using EXEC Module to execute a HTTP request via curl, to comunicate
with another app like this:
exec_avp("curl --connect-timeout 1 --max-time 2
http://IP_ADDR:PORT/PATH/'$fU'/'$tU'/'$hdr(Call-ID)'",
"$avp(s:output)");
Sometimes Kamailio blocks when HTTP server does not response in time,
despite the max-time parameter.
I found this know issue on EXEC module docs:
"There is currently no guarantee that scripts ever return and stop
blocking SIP server. (There is kill.c but it is not used along with the
current mechanisms based on popen. Besides that kill.c is ugly)."
Does anyone know a better way to communicate Kamailio with a HTTP server
without blocks? I will try a python script to send the HTTP request and
call it with APP_PYTHON module. Has anyone had blocks with APP_PYTHON
module?
Thanks in advance.
--
*Victor*
I'm using the REGISTRAR module and having phones register to Kamailio. This is working properly and the phones register without problem.
The issue I have is that it takes about 60-90 seconds for the new entry to appear in the 'location' table after I do save("location"). The entry is correct, however it takes so long for it to show up.
Can anyone explain why this is happening?
-H
Hello,
It seems that when calling the function sdp_get_line_startswith, it removes
the \r (0d in hexa) from the SIP message before forwarding it. In the code
(sdpops_mod.c) from line 1402 to 1406, we see that
// remove ending \r\n if exists
if (avp_val.s.s[line.len-2] == '\r' && avp_val.s.s[line.len-1] ==
'\n')
{
avp_val.s.s[line.len-2] = '\0';
avp_val.s.len -= 2;
}
As avp_val is the pointer pointing directly to the line in the message, all
modification made to this variable will also be applied to the message.
Am I wrong?
Regards,
Hello,
I have my siptrace module configured with setflag(22) which make sure only the forwarded messages are duplicated, and not the original messages.
This works great for requests, but for replies this doesn't work. Kamailio duplicates both original and forwarded replies.
Is it possible to do the same for replies, so only forwarded replies are duplicated?
Why are replies treated differently?
Regards,
Grant
Hi,
Actually my doubt is whether jitsi will support to make a call over
the internet?
I like to explain what i have done and what i need help from you .
Steps to make jitsi client to work with kamailio.
1. I have installed kamailio server in my linux box, created global
ip to my local ip and i have installed jitsi client in my machine and my
friend machine.
2. Then i registered my SIP accounts with the local ip address like
below
3. Then i added contacts of my friends. All my friends are in online. I
could make audio call and speak with each other with local network. In
this case no problem.
4. But the problem is when i use public ip instead of local ip in
account registration wizard, it is showing all my friends are in offline
eventhough they are in online.
Actually my requirements is , I want to make audio call over
internet(i.e from other domain). I really stuck in this case.
Please help me how to achieve this?
Advance Thanks,
Balakrishnan.N