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
Hi Guys,
I have used the http_query function in the past, in conjunction with the json module, on version 3.3 and unbuntu.
I am now using version 4.1 on Centos 6.5, and I am using the http_query, which I'm aware records in the results the first line of the response.
Can you select the result to be recorded depending on the line in the response, for example if a number is returned only in line 4, can this be extracted from a 200 ok response using this function?
Or is this then a need to use JSON module, and if so which repository for centos 6.5 contains it! :)
Sorry for the question!
Thanks
Jon
Hello,
I am considering to release v4.1.5 sometime next week, most likely on
the 6th of August. Checking the 4.1 branch, there are not many fixes,
few are on my list for backporting. That's good, indicating a high level
of stability.
If anyone is aware of issues not reported on tracker or patches that
have not been backported, add to the tracker or write a message to
sr-dev mailing list.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Hello,
I've been experimenting with Kamailio with ws and sip clients and could
need a hand in getting a call between those two to work.
I have Kamailio 4.1.2 (using rtpproxy-ng instead of rtpproxy) on a CentOS
6.5 and a mediaproxy-ng running. I have clients wsclient(a)testers.com and
gsclient(a)testers.com and I try to make call from wsclient to gsclient. The
wsclient is a jssip client running on chrome and gsclient is a grandstream
desk phone. My config file is the default one enhanced by online examples.
I use a html5 <audio> element for the media streams, and configured my
jssip phone to accept audio options like this:
var options = {
'eventHandlers': eventHandlers,
'mediaConstraints': {'audio': true, 'video': false }
};
sipUA.call(callto, options);
I used the instructions from here:
http://www.slideshare.net/crocodilertc/webrtc-websockets
What I get is gsclient ringing, and as I answer there is no audio and call
hangs up in a few seconds. I guess this is a SDP problem, something between
Kamailio and Mediaproxy-ng but SDP is not my strong point so I'd appreciate
advice.
Question is where's my misconfiguration/problem? I would like to learn why
this problem occurs and how to fix it rather than getting a solution right
away, but please bear in mind I don't know much about SDP.
In Kamailio log I see:
kamailio[27059]: ERROR: rtpproxy-ng [rtpproxy.c:1346]:
rtpp_function_call(): proxy replied with error: Error rewriting SDP
kamailio[27058]: ERROR: rtpproxy-ng [rtpproxy.c:1346]:
rtpp_function_call(): proxy replied with error: Unknown call-id
kamailio[27057]: ERROR: rtpproxy-ng [rtpproxy.c:1346]:
rtpp_function_call(): proxy replied with error: Unknown call-id
Following are the INVITEs and 200 OKs from my SIP trace (1.1.1.1 is the ip
of my Kamailio & mediaproxy-ng box and 2.2.2.2 is the public ip behind
which both my clients are). The gsclient has port 5066.
******************************************************************************
U 2014/04/01 20:03:41.060009 1.1.1.1:5060 -> 2.2.2.2:5066
INVITE sip:gsclient@192.168.0.106:5066;transport=udp SIP/2.0.
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>.
Record-Route: <sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes>.
Via: SIP/2.0/UDP
1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.0.
Via: SIP/2.0/WS
kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267.
Max-Forwards: 16.
To: <sip:gsclient@testers.com>.
From: <sip:wsclient@testers.com>;tag=hhcd99tmvm.
Call-ID: 1dluvk38g1j22fn96t4b.
CSeq: 7237 INVITE.
Contact: <sip:wsclient@testers.com
;gr=urn:uuid:f6014564-88cb-4f57-9ae5-3b4336ef9db8;ob;alias=2.2.2.2~38986~5;alias=2.2.2.2~38986~5>.
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE.
Content-Type: application/sdp.
Supported: path, outbound, gruu.
User-Agent: JsSIP 0.3.0.
Content-Length: 2211.
.
v=0.
o=- 4897716268503406223 2 IN IP4 1.1.1.1.
s=-.
t=0 0.
a=group:BUNDLE audio.
a=msid-semantic: WMS vMh5vhUEQzvVKJYdqRkAuCcXVa2blgbEXARZ.
m=audio 30028 RTP/SAVPF 111 103 104 0 8 106 105 13 126.
c=IN IP4 1.1.1.1.
a=candidate:2999745851 1 udp 2113937151 192.168.56.1 63341 typ host
generation 0.
a=candidate:2999745851 2 udp 2113937151 192.168.56.1 63341 typ host
generation 0.
a=candidate:3350409123 1 udp 2113937151 192.168.0.101 63342 typ host
generation 0.
a=candidate:3350409123 2 udp 2113937151 192.168.0.101 63342 typ host
generation 0.
a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation
0.
a=candidate:4233069003 2 tcp 150995
T 2014/04/01 20:03:41.119806 2.2.2.2:38986 -> 1.1.1.1:5060 [A]
......
U 2014/04/01 20:03:41.159086 2.2.2.2:5066 -> 1.1.1.1:5060
SIP/2.0 488 Not Acceptable Here.
Via: SIP/2.0/UDP
1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.0.
Via: SIP/2.0/WS
kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267.
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>.
Record-Route: <sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes>.
From: <sip:wsclient@testers.com>;tag=hhcd99tmvm.
To: <sip:gsclient@testers.com>;tag=7875f08763872c34.
Call-ID: 1dluvk38g1j22fn96t4b.
CSeq: 7237 INVITE.
User-Agent: Grandstream GXP2000 1.2.2.26.
Warning: 304 GS "Media type not available".
Content-Length: 0.
.
U 2014/04/01 20:03:41.159392 1.1.1.1:5060 -> 2.2.2.2:5066
ACK sip:gsclient@192.168.0.106:5066;transport=udp SIP/2.0.
Via: SIP/2.0/UDP
1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.0.
Max-Forwards: 16.
To: <sip:gsclient@testers.com>;tag=7875f08763872c34.
From: <sip:wsclient@testers.com>;tag=hhcd99tmvm.
Call-ID: 1dluvk38g1j22fn96t4b.
CSeq: 7237 ACK.
Content-Length: 0.
.
U 2014/04/01 20:03:41.161085 1.1.1.1:5060 -> 2.2.2.2:5066
INVITE sip:gsclient@192.168.0.106:5066;transport=udp SIP/2.0.
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>.
Record-Route: <sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes>.
Via: SIP/2.0/UDP
1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.1.
Via: SIP/2.0/WS
kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267.
Max-Forwards: 16.
To: <sip:gsclient@testers.com>.
From: <sip:wsclient@testers.com>;tag=hhcd99tmvm.
Call-ID: 1dluvk38g1j22fn96t4b.
CSeq: 7237 INVITE.
Contact: <sip:wsclient@testers.com
;gr=urn:uuid:f6014564-88cb-4f57-9ae5-3b4336ef9db8;ob;alias=2.2.2.2~38986~5;alias=2.2.2.2~38986~5>.
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE.
Content-Type: application/sdp.
Supported: path, outbound, gruu.
User-Agent: JsSIP 0.3.0.
Content-Length: 3136.
.
v=0.
o=- 4897716268503406223 2 IN IP4 1.1.1.1.
s=-.
t=0 0.
a=group:BUNDLE audio.
a=msid-semantic: WMS vMh5vhUEQzvVKJYdqRkAuCcXVa2blgbEXARZ.
m=audio 30028 RTP/AVP 111 103 104 0 8 106 105 13 126.
c=IN IP4 1.1.1.1.
a=fingerprint:sha-256
72:54:87:EC:D2:4C:D1:70:C2:FE:69:08:20:5C:92:1D:E0:EA:BD:45:09:E0:90:62:27:B6:34:60:54:E2:99:28.
a=setup:actpass.
a=mid:audio.
a=sendrecv.
a=rtpmap:111 opus/48000/2.
a=fmtp:111 minptime=10.
a=rtpmap:103 ISAC/16000.
a=rtpmap:104 ISAC/32000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:106 CN/32000.
a=rtpmap:105 CN/16000.
a=rtpmap:13 CN/8000.
a=rtpmap:126 telephone-event/8000.
a=maxptime:60.
a=ssrc:3298511848 cnam
And here are the 200 OK messages when answering the call:
U 2014/04/01 20:03:46.049711 2.2.2.2:5066 -> 1.1.1.1:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.1.
Via: SIP/2.0/WS
kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267.
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>.
Record-Route: <sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes>.
From: <sip:wsclient@testers.com>;tag=hhcd99tmvm.
To: <sip:gsclient@testers.com>;tag=fb215901a251c9a0.
Call-ID: 1dluvk38g1j22fn96t4b.
CSeq: 7237 INVITE.
User-Agent: Grandstream GXP2000 1.2.2.26.
Contact: <sip:gsclient@192.168.0.106:5066;transport=udp>.
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE.
Content-Type: application/sdp.
Supported: replaces, timer.
Content-Length: 216.
.
v=0.
o=gsclient 8000 8000 IN IP4 192.168.0.106.
s=SIP Call.
c=IN IP4 192.168.0.106.
t=0 0.
m=audio 5026 RTP/AVP 0 13.
a=sendrecv.
a=rtpmap:0 PCMU/8000.
a=ptime:20.
m=audio 0 RTP/SAVPF 111 103 104 0 8 106 105 13 126.
T 2014/04/01 20:03:46.051127 1.1.1.1:5060 -> 2.2.2.2:38986 [AP]
.~.dSIP/2.0 200 OK.
Via: SIP/2.0/WS
kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267.
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>.
Record-Route: <sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes>.
From: <sip:wsclient@testers.com>;tag=hhcd99tmvm.
To: <sip:gsclient@testers.com>;tag=fb215901a251c9a0.
Call-ID: 1dluvk38g1j22fn96t4b.
CSeq: 7237 INVITE.
User-Agent: Grandstream GXP2000 1.2.2.26.
Contact: <sip:gsclient@192.168.0.106:5066;transport=udp>.
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE.
Content-Type: application/sdp.
Supported: replaces, timer.
Content-Length: 216.
.
v=0.
o=gsclient 8000 8000 IN IP4 192.168.0.106.
s=SIP Call.
c=IN IP4 192.168.0.106.
t=0 0.
m=audio 5026 RTP/AVP 0 13.
a=sendrecv.
a=rtpmap:0 PCMU/8000.
a=ptime:20.
m=audio 0 RTP/SAVPF 111 103 104 0 8 106 105 13 126.
******************************************************************************
cheers,
Olli
Hello,
I'm using Kamailio 4.1 and I'm wondering how to avoid external DNS
resolution.
I have the following config:
dns_cache_init=no
use_dns_cache=no
dns=no
rev_dns=no
Even with this config, I have many and many DNS query on SRV _sip for the
hostnames set in carrierroute module.
These requests are not useful because the A resolution is done by
/etc/hosts.
Major problem with this, is that when I have a DNS issue or IP transit
issue, Kamailio waits for resolution timeout and becomes overloaded. As a
consequence, Kamailio can't treat others SIP requests like REGISTER because
he stuck in DNS resolution.
Regards,
Igor.
I have added some XLOG and XDBG commands in registrar module of kamailio. I
want to see messages in result of these commands being printed in logs to
make sure that changes are being made surely. How can it be done ?
I am using fedora and I see kamailio related logs in /var/log/messages.log
--
View this message in context: http://sip-router.1086192.n5.nabble.com/See-XDBG-and-XERR-Messages-in-Logs-…
Sent from the Users mailing list archive at Nabble.com.
I am trying to follow the guide shown here:http://www.kamailio.org/docs/openser-performance-tests/ to load test my kamailio system.I am a little unclear as to how many instances of sipp are running. For the first part i see the command ./sipp -sf uac_msg.xml -rsa 192.168.2.102:5060 192.168.2.102:5070 -m 200000 -r 10000 -d 1 -l 70for generating the UAC part but is there another instance of sipp running on the kamailio computer acting as a UAS? I have seen come sites use for example: sipp 192.168.1.100:5060 -sn uas -p 5060 but is this not necessary?
thanks for any help
I have managed to install Kamailio and Siremis 4.x, but I have noticed that neither Kamaliio or Siremis come with a sample config that makes sense for Siremis (as far as I can tell). For instance, if I want to use LCR with dispatchers, the sample Kamilio configs don’t have the dispatcher module loaded or configured. Am I missing something? Can someone please share theirs?
Thanks!
~Noah