Hello. I suceesfully authenticate some accounts of providers from UAc using
DB.
When I try to Call to any provider it responds me 407 or 401 reply.
To Send to It new INVITE with auth Creditans I use UAC_AUTH() with
configured modparams:
modparam("uac","credential","username:domain:password")
modparam("uac","auth_realm_avp","$avp(i:10)")
modparam("uac","auth_username_avp","$avp(i:11)")
modparam("uac","auth_password_avp","$avp(i:12)")
I think that avp`s used to get varibles from DB, but at test it is not
write.
So my question- how to get this varibles from DB (uacreg table)
Thanks
Hi, I want to realise scheme:
Asterisk->Kamailo->provider
I register providers accounts at Kamailio using UAC module.
When Generates call form Asterisk through Kamailio to provider, provider
sends back 407 response. This reponse mut be handled by KAmailio through
failure_route using uac_auth().
I try to catch this response at onreply_route by status and route this
response to failure route (t_in_failure("MY_FAILURE_ROUTE"))but Kamailio
ignores t_on_failure and forvard this pacjet to asterisk.
So My question is: How to handle this response at kamailio and not forward
to asterisk?
This is invite that forwards to provider:
INVITE sip:11234567890@my.provider.com:5060 SIP/2.0
Record-Route: <sip:my.kamailio.com:5068;nat=yes;ftag=as57a2d34c;lr=on>
Via: SIP/2.0/UDP my.kamailio.com:5068
;branch=z9hG4bK8544.480e25d0e6a328f5f455233d808cda80.0
Via: SIP/2.0/UDP 10.0.1.6:50600;branch=z9hG4bK778521b0;rport=50600
Max-Forwards: 70
From: "John" <sip:my_provider_acc@my.provider.com:50600>;tag=as57a2d34c
To: <sip:11234567890@my.provider.com:5068>
Contact:<my_provider_acc@my.kamailio.com:5068>
Call-ID: 3d8d6c357d69cdd51633b1c125729f44@10.0.1.6:50600
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.4.0
Date: Sat, 16 Aug 2014 13:18:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 540
v=0
o=root 1941643043 1941643043 IN my.kamailio.com
s=Asterisk PBX 12.4.0
c=IN IP4 my.kamailio.com
t=0 0
a=ice-lite
m=audio 30108 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp:30109
a=ice-ufrag:jOv9GFqq
a=ice-pwd:wuXGbJ3ZI7MfwM6kwto78s8reEyU
a=candidate:lzqT8la5i9wkzQzB 1 UDP 2130706431 my.kamailio.com 30108 typ host
a=candidate:lzqT8la5i9wkzQzB 2 UDP 2130706430 my.kamailio.com 30109 typ host
I've begun playing with rtpengine (git e0957d1) a little and in my testing:
CSipSimple <-> Kamailio/rtpengine <-> Asterisk 12/13
I see the following error when parsing SDP from the Asterisk side, possibly
related to the use of the FQDN in the Origin:
Got valid command from 127.0.0.1:44407: answer - {
"sdp": "v=0
o=- 3617462201 3617462204 IN IP4 hostname.example.com
s=Asterisk
<snip>
Error parsing SDP at offset 5: Error parsing o= line
Protocol error in packet from 127.0.0.1:44407: Failed to parse SDP
[d3:sdp353:v=0
o=- 3617462201 3617462204 IN IP4 hostname.example.com
s=Asterisk
<snip>
It was brought up here, https://issues.asterisk.org/jira/browse/ASTERISK-23994
and it appears that a FQDN should be supported in the SDP Origin. Is this the
case, or might it be something else?
--
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8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E
i am trying to connect to my kamailio through websocket. I added the below code in my config file:
event_route[xhttp:request] { xlog("websocket connection request"); set_reply_close(); set_reply_no_connect(); if (ws_handle_handshake()) { exit; } xhttp_reply("404", "Not found", "", "");}
However, when I try to connect from my web phone, It shows :
Aug 15 15:22:31 ubuntu /usr/local/kamailio-devel//sbin/kamailio[17202]: WARNING: xhttp [xhttp_mod.c:133]: mod_init(): event_route[xhttp:request] is empty
Can someone please tell me what wrong im doing ?
Hello,
Is there any way rtpengine can be configured to use same UDP port to
receive and transmit RTP packets?
The set up I'm trying is
SIP_Client --> Kamailio/rtpengine --> Freeswitch
Internet Internet LAN LAN 192.168.1.10
I tried *rtpengine_offer * in kamailio.cfg,
but rtpengine seems to always use different ports to Tx/Rx the packets.
(Causing Freeswitch to auto adjust its ports, which mutes the audio
channel one way).
*rtpengine_offer("force symmetric media-address=192.168.1.10 replace-origin
replace-session-connection ICE=remove");*
*Thank you,*
* - Deep N*
Hello,
As outcome to my earlier sdp/rtp challenges I've upgraded my Asterisk
version to 11.11.0 and still use a realtime integration with Kamailio. Now
I face a somewhat different problem. With my setup I also changed from
jssip client to a sip.js client in my websocket implementation. I cloned
the latest rtpengine from git today.
I had to revert my Asterisk settings a bit, Asterisk was taking over with
sdp handling, I don't know if this is relevant but that's why I got calls
seemingly working before.
When getting the 488 Not Acceptable, I arm a branch route and call
rtpengine_offer there. When trying to call rtpengine_offer, I get the
following log:
(first I print the rtpengine_offer_flags to make sure what is passed to the
function.
Aug 15 15:04:16 u363id562 kamailio[32178]: INFO: <script>:
MANAGE_RTPENGINE_BRANCH: rtpengine_offer_flags = rtcp-mux-demux
trust-address replace-origin replace-session-connection ICE=remove RTP/AVP
Aug 15 15:04:16 u363id562 kamailio[32178]: WARNING: <core> [rvalue.c:1016]:
rval_get_int(): automatic string to int conversion for "rtcp-mux-demux
trust-address replace-origin replace-session-connection ICE=remove RTP/AVP"
failed
Aug 15 15:04:16 u363id562 kamailio[32178]: WARNING: <core> [rvalue.c:1920]:
rval_expr_eval_int(): rval expression conversion to int failed
(1128,32-1128,32)
Aug 15 15:04:16 u363id562 rtpengine[32159]: Got valid command from
127.0.0.1:44292: delete - { "call-id": "k7bft3u75p5e42ib039r",
"received-from": [ "IP4", "client_public_address" ], "from-tag":
"74dovi97bi", "command": "delete" }
Aug 15 15:04:16 u363id562 rtpengine[32159]: [k7bft3u75p5e42ib039r] Call-ID
to delete not found
Aug 15 15:04:16 u363id562 rtpengine[32159]: Returning to SIP proxy:
d7:warning38:Call-ID not found or tags didn't match6:result2:oke
After this I see another 488 and the loop swirls on until Kamailio runs out
of forking capacity. Can You guys explain why is this happening?
I set the variable like this:
$avp(rtpengine_offer_flags) = "rtcp-mux-demux trust-address replace-origin
replace-session-connection ICE=remove RTP/AVP";
cheers,
Olli
Hello
i realize that if two same Invite comes to kamailio sequentially for a
call, dialog module cannot create hash data and dont'give any error
about it.
After get ACK or BYE for this call , it gives warning like 'unable to
find dialog for ACK with route param 'f3f1.0524' [7999:16976]' .
So i cannot get any data call is hook up from dialog event_route.
For prepaid account system , dialog module is weak or i made some
mistakes. is there any better solution for accounting for high volume
calls.
I just need that call is hook up and closed or failed. which module is
best option for it? (IMS ,SEAS, RADIUS)
Thanks
i am currently having problems on xmlrpc
i am using node-xmlrpc and it fails when htable.dump returns more than 1
row.
i raised this issue on the node-xmlrpc group and this is the explanation i
got.
Each <value> node inside <param> is supposed to only have one child node,
so this looks like an invalid response. The<value> should (probably, i
don't know your use case) contain an <array> with <data> wrapped around the
structs. Seehttp://xmlrpc.scripting.com/spec.html
does this mean kamailio is not following standards?
here is the working response:
<methodResponse>
<params>
<param>
<value><struct><member><name>entry</name><value><int>6</int></value></member><member><name>size</name><value><int>1</int></value></member><member><name>slot</name><value><struct><member><name>item</name><value><struct><member><name>name</name><value><string>4::num</string></value></member><member><name>value</name><value><int>1</int></value></member></struct></value></member></struct></value></member></struct>
</value>
</param>
</params>
</methodResponse>
here is the non-working response:
<methodResponse>
<params>
<param>
<value><struct><member><name>entry</name><value><int>6</int></value></member><member><name>size</name><value><int>1</int></value></member><member><name>slot</name><value><struct><member><name>item</name><value><struct><member><name>name</name><value><string>4::num</string></value></member><member><name>value</name><value><int>1</int></value></member></struct></value></member></struct></value></member></struct>
<struct><member><name>entry</name><value><int>11</int></value></member><member><name>size</name><value><int>1</int></value></member><member><name>slot</name><value><struct><member><name>item</name><value><struct><member><name>name</name><value><string>3::num</string></value></member><member><name>value</name><value><int>1</int></value></member></struct></value></member></struct></value></member></struct>
</value>
</param>
</params>
</methodResponse>
Kelvin Chua
Hello,
lately I pushed several commits to enhance the handling of remote
registrations with uac module. Among them:
- uac.reg_info - rpc command to get the details of a remote registration
record using filter on uuid, local/remote/auth username
- uac.reg_enable and uac.reg_disable - rpc commands to enable/disable
registration of a record from memory at runtime, without restarting kamailio
- uac.reg_reload - rpc command to reload remote registration records
from database (avoid restarting when adding new records in database)
- uac.reg_refresh - rpc command to load a record from database based on
l_uuid. If record is already in memory, then the password field is
updated, otherwise a new full record is added in memory
I barely had resources to test it properly, therefore I am asking people
using this feature to give it a try and report the issues -- you have to
use the git master branch.
You can read more details about the new features in the readme of the
uac module:
- http://kamailio.org/docs/modules/devel/modules/uac.html
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA
Hello,
I want to show the domain name of a callee (by which it is registered to the
domain at the time of incoming invite packet) in the logs. I have tried
adding a few lines to usrloc module. But didnot work. I am using LM_DBG
command to show the domain name in the logs. How can I add some variable
which holds the domain name and then print it in logs using LM_DBG or LM_ERR
command ??
Thanks.
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