Hi Friends,
I need to send text to GSM phones from SIP client.
SIPSimple text -> GSM SMS
Does anybody know good SMS gateway service that I can use for this purpose.
Best Regards,
Roy Gandhi.
Hello,
I have a Kamailio IMS instance setup, using different hosts for each IMS
function (P-CSCF,S-CSCF,I-CSCF etc) and am able to register clients
successfully.
However, when Client-A tries to call Client-B (both registered to the same
S-CSCF), the call is failing with the "403 Forbidden - You must register
first with a S-CSCF" error. .
All hosts & clients are running on the same subnet using RFC1918 IP
addresses, there is no NAT involved.
I've seen the same issue on both 4.2 and Master branches.
I have the nathelper 'nat_uac_test' parameter set to 18.
Looking at the debug prints and SIP traces, I can see that the 403 Error is
being generated by the P-CSCF when it receives the INVITE from the S-CSCF
(for forwarding to Client-B).
Digging deeper into the debug logs, it looks like this is a result of a
failed contact match - I've experimented with different values
for "hashing_type", which do change the routing logic but still I see the
403 Error.
Here's 2 excerpts from the debug prints for different values of
hashing_type :
*modparam("ims_usrloc_pcscf", ""hashing_type",hashing_type", 0) :*
2(19636) DEBUG: ims_registrar_pcscf [service_routes.c:183]: getContactP():
> Searching in usrloc for 10.133.202.17:5060 (Proto 1)
2(19636) DEBUG: ims_usrloc_pcscf [udomain.c:618]: get_pcontact_by_src():
> Trying to find contact by src with URI: [sip:*@10.133.202.17:5060]
2(19636) DEBUG: ims_usrloc_pcscf [udomain.c:465]: get_pcontact():
> Searching for contact in P-CSCF usrloc [sip:*@10.133.202.17:5060]
2(19636) DEBUG: ims_usrloc_pcscf [usrloc.c:187]: get_aor_hash(): Returning
> hash slot: [1670705485]
2(19636) DEBUG: ims_registrar_pcscf [service_routes.c:186]: getContactP():
> No entry in usrloc for 10.133.202.17:5060 (Proto 1) found!
2(19636) DEBUG: ims_usrloc_pcscf [udomain.c:465]: get_pcontact():
> Searching for contact in P-CSCF usrloc [sip:john@10.133.202.61:64019
> ;transport=udp]
2(19636) DEBUG: ims_usrloc_pcscf [usrloc.c:187]: get_aor_hash(): Returning
> hash slot: [397265208]
2(19636) DEBUG: ims_registrar_pcscf [service_routes.c:119]:
> checkcontact(): Port 64019 (search 5060), Proto 1 (search 1), reg_state
> registered (search registered)
2(19636) ERROR: *** cfgtrace:request_route=[Orig_Initial]
> c=[/etc/kamailio/kamailio.cfg] l=926 a=26 n=send_reply
*modparam("ims_usrloc_pcscf", ""hashing_type",hashing_type", 2) :*
1(19401) DEBUG: ims_registrar_pcscf [service_routes.c:183]: getContactP():
>> Searching in usrloc for 10.133.202.17:5060 (Proto 1)
>
> 1(19401) DEBUG: ims_usrloc_pcscf [udomain.c:618]: get_pcontact_by_src():
>> Trying to find contact by src with URI: [sip:*@10.133.202.17:5060]
>
> 1(19401) DEBUG: ims_usrloc_pcscf [udomain.c:465]: get_pcontact():
>> Searching for contact in P-CSCF usrloc [sip:*@10.133.202.17:5060]
>
> 1(19401) DEBUG: ims_usrloc_pcscf [usrloc.c:97]:
>> get_alias_host_from_contact(): no params
>
> 1(19401) DEBUG: ims_usrloc_pcscf [usrloc.c:183]: get_aor_hash(): using
>> host for hash [10.133.202.17]
>
> 1(19401) DEBUG: ims_usrloc_pcscf [usrloc.c:187]: get_aor_hash():
>> Returning hash slot: [-1564914093]
>
> 1(19401) DEBUG: ims_registrar_pcscf [service_routes.c:186]:
>> getContactP(): No entry in usrloc for 10.133.202.17:5060 (Proto 1) found!
>
> 1(19401) DEBUG: ims_usrloc_pcscf [udomain.c:465]: get_pcontact():
>> Searching for contact in P-CSCF usrloc [sip:john@10.133.202.61:64013
>> ;transport=udp]
>
> 1(19401) DEBUG: ims_usrloc_pcscf [usrloc.c:176]: get_aor_hash(): Looks
>> like this contact is natted - contact URI: [10.133.202.61] but came from
>> received_host: [10.133.202.17] so will use received_host for hash
>
> 1(19401) DEBUG: ims_usrloc_pcscf [usrloc.c:187]: get_aor_hash():
>> Returning hash slot: [-1564914093]
>
> 1(19401) ERROR: *** cfgtrace:request_route=[Orig_Initial]
>> c=[/etc/kamailio/kamailio.cfg] l=926 a=26 n=send_reply
>
>
Can anyone offer any hints as to how to resolve this?
I am attaching a SIP call flow - I can supply a complete Kamailio debug
trace if needed.
Thanks
Phill
HelloI have a little project and I need to find the ip address from my SIP phone. How can I find my ip addres of the phone?Have you a command or a service whith this functions? Thanks for help !!!Best regards,Constantin
Hi Daniel
We don't have pike module enabled.
When the problem occurred on an VIP, we observed these:
* Kamailio stopped responding to any messages that were sent to the VIP,
not just OPTIONS
* The OPTIONS messages are not big. But the other SIP messages, e.g. some
of the INVITE/OK that came from some SBCs can be big
* netstat showed the Recv-Q on that VIP had a lot of bytes accumulated,
while Kamailio was not seeing/reading them
* Kamailio responded fine on its real IP, when I sent OPTIONS pings to it
using sipsak
* After restarting Kamailio it started working. But after 1 week or so the
same problem happened again
Since this only happened after running for 1 week or so, we didn't have any
traces to show what exactly happened at the particular time when it
happened. It is possible some SIP messages may have come in fragmented and
some are just too big, depending on the route they came from etc. So I was
wonder if it was possible that the UDP receive buffer was filled somehow
with messed-up messages. Is there anyway to check this? Any suggestions
where I should start looking please? Or is it generally a bad idea to use
UDP when there are messages that may be too big, either fragmented or not?
Since the it is running in the production environment, I'd like to get some
confidence that a Kamailio upgrade will fix the problem first before I
change anything there.
Cheers,
Yufei
Date: Wed, 01 Apr 2015 16:27:44 +0200
From: Daniel-Constantin Mierla <miconda(a)gmail.com>
To: "Kamailio (SER) - Users Mailing List"
<sr-users(a)lists.sip-router.org
>
Subject: Re: [SR-Users] SIP messages over UDP with sizes over MTU
Message-ID: <551C0060.3010002(a)gmail.com>
Content-Type: text/plain; charset="windows-1252"
Hello,
first, not related to the topic you reported, I would recommend at least
upgrading to latest 4.0.x, there were many fixes from v4.0.0 and there
are no changes to be done to config or database -- simply deploy latest
4.0.x and restart.
Back to this topic. Is all the other traffic handled apart of big OPTIONS?
Do you have pike module enabled? If yes, can you double check and be
sure that the SBC is not blacklisted (traffic from it should not be
handled via pike_check_req()).
Cheers,
Daniel
On 01/04/15 15:42, Yufei Tao wrote:
> Hi
>
> We've got Kamailio (v4.0.0) connected to some SBC, which sends SIP
> traffic and periodic OPTIONS pings to Kamailio's VIP. Kamailio
> responds to the OPTIONS pings with OK, i.e. in the main route block:
> if (is_method("OPTIONS"))
> {
> sl_send_reply("200","OK");
> exit;
> }
>
> All works fine for a week or so, then Kamailio stopped responding to
> the OPTIONS pings on the VIP it listens to. But it still respond to
> OPTIONS pings that are sent to its real IP. The real IP is not used
> for receiving/sending any traffic while only the VIP is. So it seems
> that Kamailio is still working, but maybe having problems with the
> receiving buffer for the VIP?
>
> We do see that some SIP messages sent to Kamailio's VIP are too big
> (sometimes over 1500 bytes). My question is, in this case, what would
> be expected to happen? Is it possible somehow the receiving buffer for
> the VIP got messed up by the big UDP messages? Any one seen similar
> problems? What is the suggested solution?
>
> We're considering moving to TCP. But since this is production
> environment, I want to get some confidence that the problem we saw was
> likely to have been caused by the UDP message being too large.
>
> Cheers,
> Yufei
Hello,
Kamailio SIP Server v4.2.4 stable release is out.
This is a maintenance release of the latest stable branch, 4.2, that
includes fixes since release of v4.2.3. There is no change to database
schema or configuration language structure that you have to do on
installations of v4.2.x. Deployments running previous v4.x.x versions
are strongly recommended to be upgraded to v4.2.4.
For more details about version 4.2.4 (including links and guidelines to
download the tarball or from GIT repository), visit:
* http://www.kamailio.org/w/2015/04/kamailio-v4-2-4-released/
RPM, Debian/Ubuntu packages will be available soon as well.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
Hi there,
I have noticed that my kamailio proxy, version 4.2.2 is originating core
dumps.
This happens when Kamailio receives an INVITE in failure route, then
execute the function async_sleep(3), after that changes request uri, then
it stops with following error message shown below:
<core> [mem/f_malloc.c:586]: fm_free(): BUG: fm_free: bad pointer
0x7f8d26605ed8 (out of memory block!), called from <core>:
parser/msg_parser.c:
set_path_vector(822) - aborting
Please see the attachment that contains the " gdb bt full" from core.dump
originated
Thank you for your attention
Best Regards
--
Cumprimentos
José Seabra
I'm looking to replace rtpproxy with rtpengine. All calls have to be forced
thru rtpengine and according to the documentation the equivalent of
rtpproxy_manage("f");
is
rtpengine_manage("force");
But rtpenginge doesn't recognize it as a valid flag. Using the legacy rtpproxy
mod it still works though. Is it a documentation failure or is a specific
version of rtpengine required for forced?
Apr 1 17:54:28 pisco rtpengine[3432]: Got valid command from 127.0.0.1:43193:
offer - { "sdp": "v=0#015#012o=tryba 8000 8001 I
N IP4 10.0.3.169#015#012s=SIP Call#015#012c=IN IP4 10.0.3.169#015#012t=0
0#015#012m=audio 5016 RTP/AVP 8 0 9 2 3 18 4 99 101 13
#015#012a=sendrecv#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:0
PCMU/8000#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:2 G726-3
2/8000#015#012a=rtpmap:3 GSM/8000#015#012a=rtpmap:18
G729/8000#015#012a=rtpmap:4 G723/8000#015#012a=rtpmap:99 iLBC/8000#015#012
a=fmtp:99 mode=20#015#012a=ptime:20#015#012a=rtpmap:101 telephone-
event/8000#015#012a=fmtp:101 0-11#015#012", "flags": [ "force
" ], "call-id": "5 ...
Apr 1 17:54:28 pisco rtpengine[3432]: ... f13ffb48e98404a(a)10.0.3.169",
"received-from": [ "IP4", "109.235.34.226" ], "from-tag
": "3819edfdf2df044a", "command": "offer" }
Apr 1 17:54:28 pisco rtpengine[3432]: Unknown flag encountered: 'force'
--
Telefoon: 088 0100 700
Sales: sales(a)pocos.nl | Service: servicedesk(a)pocos.nl
http://www.pocos.nl/ | Croy 9c, 5653 LC Eindhoven | Kamer van Koophandel
17097024
Hello,
I am considering to release a new maintenance version from latest stable
branch - to be v4.2.4 - next week, most likely on Thursday, April 2. If
there is anything important to get in and not discussed yet, bring the
topic on development mailing list.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
Using Kamailio 4.2-dev and MSILO, is it possible to "toggle" the auto-
notification reply MESSAGE using something similar to the following where
"$var(msilo_reply)" is emtpy at startup (or are there suggestions for a better
method):
modparam("msilo", "from_address", "$var(msilo_reply)")
...
if(CONDITION WHERE I WANT THE AUTO-REPLY RETURNS POSITIVE) {
$var(msilo_reply)="$rU(a)example.net";
}
m_store($rU)
...
My use case is that I would like the auto-notification reply to occur in some
instances, but not others. I also do not want to create a loop where the
auto-notification replies are also stored. I was originally using a simple
test: if(src_ip!=myself), but I have begun to use the IMC module as well as
the EXEC module and these MESSAGEs appear to originate from "myself."
Looking at the MSILO documentation I see there is "extra_hdrs(string)," but
this appears to be designed to add headers to dumped MESSAGEs. I was hoping
there would be a way to add a custom header to the auto-notification, on which
I could filter/drop generated replies earlier in the route, but it doesn't
appear that this exists.
My preference for the best-case scenario would be to control the generation of
the auto-notification reply in the first place rather than having to add a
control to detect the reply and drop it.
Thanks in advance for any recommendations.
--
Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E