Greetings.
We build specialized softphone for iOS. Push notifications are used to
notify user and wake up softphone app on missed calls and messages.
We used Daniel's example config script as base for own configuration
file. It has the same parts and does not differ too much. I talk about
this example: http://www.kamailio.org/events/2014-KamailioWorld/day2/26-Daniel-Constantin…
Everything was ok for few monthes - with kamailio 4.1.2.
Few weeks ago we upgraded to 4.3.3.
Now paused calls cannot be resumed properly.
Softphone receives 500 Server error (No error 0/SL) in response
This is problem part of script:
route[RELAYRESUMED] {
# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
if (is_method("INVITE|SUBSCRIBE")) {
t_on_branch("MANAGE_BRANCH");
t_on_reply("MANAGE_REPLY");
}
if (is_method("INVITE")) {
t_on_failure("MANAGE_FAILURE");
}
if ($var(rr_target) != "")
{
xlog("L_INFO","Relay to $var(rr_ip):$var(rr_port)\n");
$ru = "sip:" + $rU + "@" + $var(rr_ip) + ":" +
$var(rr_port) + ";transport=" + $var(rr_proto);
xlog("L_INFO","Current request URI is $ru\n");
if (!t_relay())
xlog("L_INFO","Cannot relay PUSHED invite.");
sl_reply_error();
}
else
{
xlog("L_WARN","Target proto is not set\n");
sl_reply_error();
}
exit;
}
It produces log
-----
Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: INFO:
<script>: resuming trasaction [23927:345343979] 1985987629
(23927:345343979)
Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: INFO:
<script>: Relay to 95.133.110.17:11791
Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: INFO:
<script>: Current request URI is
sip:1985987629@95.133.110.17:11791;transport=tls
Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: ERROR: sl
[sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: No
error (0/SL)
Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: INFO:
<script>: Checking FAILED messages for 1985987629: 0
-----
sl_reply_error() sends 500 with No error (0/SL) description. Later 180
& 200 are sent.
But client softphone terminates transaction already (as it handles 500
response code) .
There is no "Cannot relay PUSHED invite." message in the log however.
I will be happy to heary any idea what can be cause...
Thank you!
Hi,
We are working on kamailio-4.2.3 and we require B2BUA functionality to be
achieved.From kamailio official website we got to know,we can integrate
kamailio with SEMS to achieve B2BUA functionality.
We have few queries regarding this integration.
1.SEMS-1.6.0(latest version available) is been used,is this version suits
kamailio-4.2.3?
2.SEMS-1.6.0 does not support unix based communication with
kamailio-4.2.3,as SEMS-1.6.0 doesn't have unix socket interface plugin.
Since we need unix socket based communication with SEMS.can we port the
unix socket interface plugin to SEMS-1.6.0?If no,please suggest us better
approaches.
Regards,
Prashant.
Hello,
I was doing some tests with Elastix MT that use version 4.1.6 of Kamailio
and by default uses rtpproxy, I had the problem of error 488 and unable to
make calls from websockets (Incompatible SDP).
I could see that Peter Dunkley in a video explains that the problem is
solved by rewriting the SDP, but he uses the mediaproxy not rtpproxy.
What are your recommendations?
I hope you can help me.
Chreers,
Alfredo.
rtpengine readme tells:
via-branch=... - Include the “branch” value of one of the “Via” headers
in the request to the RTP proxy. Possible values are: “1” - use the
first “Via” header; “2” - use the second “Via” header; “auto” - use the
first “Via” header if this is a request, or the second one if this is a
reply; “extra” - don't take the value from a header, but instead use the
value of the “extra_id_pv” variable. This can be used to create one
media session per branch on the RTP proxy. When sending a subsequent
“delete” command to the RTP proxy, you can then stop just the session
for a specific branch when passing the flag '1' or '2' in the
“rtpengine_delete”, or stop all sessions for a call when not passing one
of those two flags there.
if via-branch=extra is used in offers of branched, should it also be
used in answers and deletes?
-- juha
Hello,
I have installed a kamailio server with rtpproxy. The server is located between 2 networks and bridge these networks. My question: Does handle rtpproxy only 1 call at the same time? Because my configuration works, but only for one call from the first to the second network. The second call is not working. But if I make internal or external call, I can do more than one call.
Do I need the multiple rtpproxy usage configuration? I thought not, because I don't use different IPs for the second call...
Cheers,
Kai
Hi All,
I have installed kamailio and using as a i-cscf,s-cscf and p-cscf server. I
have taken FHoSS(OpenImscore) for hss server Now I am able to do register
and unregister. When I am sending INVITE for call, the calling-party and
called-party were registerd but I am getting "403 Forbidden - Dialog not
found on S-CSCF or Terminating user not suitable for unregistered
services".
Could you please tell me what is the root cause of above error message ?
Best regards,
Priyaranjan
Hello,
I am looking to get some feedback on some issues I noticed more and more
lately.
Apparently some SIP devices (media servers, phones, ...) are keeping the
"a=nortpproxy:yes" line in SDP when replying to an INVITE that contains
such line.
[Alice] ------> [Kamailio+RTPProxy] ------> [Bob]
The 200ok response from Bob has "a=nortpproxy:yes" in SDP.
By default, that line in SDP makes the rtpproxy not to engage itself
anymore in rtp relaying, and as a result things like no audio or one way
audio happens.
Anyone else encountering such situations? If yes, what are the devices
with such behaviour? So far I noticed with some FreeSwitch and Snom --
none of them I can control, so there might be a specific configuration
of those devices, not something by default there.
The solution is to set:
modparam("rtpproxy", "nortpproxy_str", "")
and use flag 'r' for rtpproxy_manage() if the IP in SDP is not a private
address.
I already updated the default config for master to use flag 'r' if the
SDP media IP is not private, wondering if nortpproxy_str should be set
to empty in kamailio.cfg (or made empty as default in config).
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com
Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat
Hi all,
I'm trying to load the rtpproxy url's from database but get a error when
starting kamailio (4.2.3):
ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error
on query: Unknown column 'set_name' in 'field list'
Looks like the field set_name is missing in the database structure, but
i can't find it in the docs. If i add the field kamailio starts, but
what is the field set_name used for. Seems it needs to be a int value:
ERROR: rtpproxy [rtpproxy.c:528]: get_rtpp_set(): Invalid set name
'test' - must be integer
Thanks,
Jan
________________________________
From: Ryan Holbein <rtholbein(a)hotmail.com>
Sent: Wednesday, November 18, 2015 8:07 AM
To: miconda(a)gmail.com
Subject: Re: [SR-Users] Kamailio with Siremis
Hello,
This is what I have setup so far I know it is not much but looking for help this is my very first Proxy. If you look at my Attachments I am trying to do IP auth. I have 2 Asterisk Boxs with 192.168.89.2 and .3.. The Proxy has 2 nics the outside and the 192.168.89.0/24 ... How do I setup the LCR rule if you look at the LCR Attachment? Also is my trusted permissions setup correct(Those IPs you see are correct)?
If you are willing to help Thank you in Adv.
________________________________
From: Daniel-Constantin Mierla <miconda(a)gmail.com>
Sent: Thursday, November 12, 2015 12:05 PM
To: Ryan Holbein; Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Kamailio with Siremis
If you have a single Asterisk box, then you can simply forward using kamailio.cfg only. Look at route[PSTN] in default configuration file for an example.
If you have more Asterisk boxes, then depends on what you need/want to do:
- for load balancing look at dispatcher module - it has a complete sample config in the docs
- for least cost routing, then lcr module is indeed an option
Cheers,
Daniel
On 12/11/15 17:56, Ryan Holbein wrote:
Now I assume in LCR I will add the Asterisk boxs so it can route? Does that sound correct? Anyone able to provide an example?
Thank you
________________________________
From: sr-users <sr-users-bounces(a)lists.sip-router.org><mailto:sr-users-bounces@lists.sip-router.org> on behalf of Daniel-Constantin Mierla <miconda(a)gmail.com><mailto:miconda@gmail.com>
Sent: Thursday, November 12, 2015 3:01 AM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Kamailio with Siremis
Hello,
look at permissions module. Default kamailio can do IP auth by enabling WITH_AUTH and WITH_IPAUTH and adding the trusted IP in database table 'address' with group id 1.
Siremis has a panel for managing the records in address table.
Cheers,
Daniel
On 11/11/15 14:10, Ryan Holbein wrote:
Hello,
I have setup with my phone provider an IP endpoint to do IP Auth. How would I go about setting that up in Kamailio or even better can some tell me how to do that in Siremis?
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda<http://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com
Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com
Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat
