Dears,
I have been trying to use the dialog module to track active calls. I
was able to see the active dialogs in database before I implemented IP
authentication, now it is not saving the active dialogs to the database.
Please provide me some direction.
Module parameters :
modparam("dialog
What could I have done wrong ? I am not able to have kamailio see
rtpproxy. I am using kamailio 3.1.2 and rtpproxy 1.2
Rtpproxy is started with command :
/usr/local/bin/rtpproxy -l -s udp:localhost:7722 -u rtpproxy
Module parameter : modparam("rtpproxy", "rtpproxy_sock",
"udp:127.0.0.1:7722")
Hello,
for the second year, Kamailio is participating in GSoC program. For
2011, the proposal is to implement a signaling gateway between SIP and
JINGLE to enable voice calls between the two networks:
http://www.jitsi.org/index.php/GSOC2011/KamailioJingle
If you are a student, or you know a s
Hello,
I am again the coordinator for Kamailio's project at GSoC 2011, last
year we had very good results with other developers joining the
mentoring team. It was the case of Marius Zbihlei that really helped a
lot, now being another opportunity to thank him again.
If you want to help guidin
On 3/28/11 8:22 AM, CT Radu wrote:
Pe 27.03.2011 12:34, Daniel-Constantin Mierla a scris:
Hello,
I thought it is an useful information to announce here that the SIP
Communicator SIP softphone was renamed to Jitsi. There are several
pages mentioning Kamailio and SIP Communicator out there, amo
Hello,
On 3/28/11 9:16 AM, Linux Guy wrote:
What could I have done wrong ? I am not able to have kamailio see
rtpproxy. I am using kamailio 3.1.2 and rtpproxy 1.2
Rtpproxy is started with command :
/usr/local/bin/rtpproxy -l -s udp:localhost:7722 -u rtpproxy
Module parameter : modparam("rtpp
I forwarded the email to Jitsi mailing list and they are going to take a
look. At first sight, seems indeed a bug there.
Cheers,
Daniel
On 3/27/11 1:32 PM, Iñaki Baz Castillo wrote:
2011/3/27 Daniel-Constantin Mierla:
Presence configuration with XCAP:
* http://asipto.com/u/sp
It's being a
On 03/28/2011 11:27 AM, Daniel-Constantin Mierla wrote:
On 3/28/11 8:22 AM, CT Radu wrote:
Pe 27.03.2011 12:34, Daniel-Constantin Mierla a scris:
Hello,
I thought it is an useful information to announce here that the SIP
Communicator SIP softphone was renamed to Jitsi. There are sev
Hello,
On 3/28/11 9:12 AM, Linux Guy wrote:
Dears,
I have been trying to use the dialog module to track active calls. I
was able to see the active dialogs in database before I implemented IP
authentication, now it is not saving the active dialogs to the
database. Please provide me some dire
Daniel-Constantin Mierla writes:
> for the second year, Kamailio is participating in GSoC program. For
> 2011, the proposal is to implement a signaling gateway between SIP and
> JINGLE to enable voice calls between the two networks:
i read somewhere that googletalk now supports sip, but don't k
2011/3/28 Juha Heinanen :
> i read somewhere that googletalk now supports sip
I don't think so. GoogleTalk is XMPP and uses Jingle for voice.
Maybe you meant GoogleVoice (https://www.google.com/voice/) which is a
PSTN service in which Google uses SIP internally.
> how is jingle related to google
2011/3/28 Daniel-Constantin Mierla :
> the proposal is to implement a signaling gateway between SIP and JINGLE to
> enable voice calls between the two networks
Let me a question: Why SIP people is interested in interoperate with
XMPP while the opposite is not true?
We have also tried to build IM/p
28 mar 2011 kl. 11.29 skrev Iñaki Baz Castillo:
> 2011/3/28 Daniel-Constantin Mierla :
>> the proposal is to implement a signaling gateway between SIP and JINGLE to
>> enable voice calls between the two networks
>
> Let me a question: Why SIP people is interested in interoperate with
> XMPP whil
2011/3/28 Olle E. Johansson :
>> Let me a question: Why SIP people is interested in interoperate with
>> XMPP while the opposite is not true?
> Inaki. Do google before stating things like this ;-)
>
> THe XMPP people have written drafts about SIP XMPP interoperability but have
> gotten almost no r
Pe 28.03.2011 11:30, marius zbihlei a scris:
> On 03/28/2011 11:27 AM, Daniel-Constantin Mierla wrote:
>>
>> On 3/28/11 8:22 AM, CT Radu wrote:
>>
>>>
>> Maybe you are unlucky to connect from a network that was source of
>> scanning attacks, thus being blocked by the firewall. Try with a proxy.
2011/3/28 Martin Hoffmann :
>> Anyhow, how you ever seen a XMPP server
>> integrating some kind of interoperability/gateway with SIP? In
>> Kamailio/*SER we have some attemps to interoperate with XMPP world.
>
> Not quite sure what you are getting at. What does it matter whether the
> SIP server is
Iñaki Baz Castillo writes:
> I don't think so. GoogleTalk is XMPP and uses Jingle for voice.
> Maybe you meant GoogleVoice (https://www.google.com/voice/) which is a
> PSTN service in which Google uses SIP internally.
yes, sorry, i meant googlevoice. so i read somewhere that now
googlevoice users
Hi All .
Any other suggestion for this ?
On Sat, Mar 19, 2011 at 3:35 AM, Sameer Khan wrote:
>
> Looks Great Jeremy,
>
> I will see if there is of any help for me and let u know.
>
> Thanks for sharing
>
> On Fri, Mar 18, 2011 at 3:38 AM, Jeremy McNamara wrote:
>
>> On 3/17/11 6:23 PM, Alex Ba
Iñaki Baz Castillo wrote:
>
> Anyhow, how you ever seen a XMPP server
> integrating some kind of interoperability/gateway with SIP? In
> Kamailio/*SER we have some attemps to interoperate with XMPP world.
Not quite sure what you are getting at. What does it matter whether the
SIP server is also an
> You are right, sorry. Anyhow, how you ever seen a XMPP server
> integrating some kind of interoperability/gateway with SIP? In
> Kamailio/*SER we have some attemps to interoperate with XMPP world.
Openfire has SIP SIMPLE support.
Sebastian
___
SIP Expr
Hello Juha.
How can access this change in the code?. Do I need to update to the last
version? 3.1.2 ??
Hope you can help me.
Thanks
Ricardo.-
-Mensaje original-
De: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] En nombre de Juha Heinanen
Enviado el:
Ricardo Martinez writes:
> How can access this change in the code?. Do I need to update to the last
> version? 3.1.2 ??
> Hope you can help me.
currently the trivial change (do not check uniqueness of gateway's ip
address) is only in master branch. patch to 3.1 is included below.
-- juha
** l
HI
I did the trace using ngrep:
ngrep -T -W byline -d any port 5060
And I am confused with tm retran.
(1) case one
tm module, the| retr_timer1| default is 500 ms.
#U +0.001824 xxx.17:5060 -> xxx.16:5060
INVITE sip:224@172.16.8.49:2057;line=lnzlkxu5 SIP/2.0.
#U +0.12 xxx.17:5060 ->
Hi,
I still have the same issue, I have tested setting public ip,
localhost and 127.0.0.1 for the rtpproxy_sock.
ERROR: rtpproxy [rtpproxy.c:1517]: timeout waiting reply from a RTP proxy
ERROR: rtpproxy [rtpproxy.c:1526]: proxy does not
respond, disable it
WARNING: rtpproxy [rtpproxy.c
Does ngrep really show the difference to the previous captured packet
which mathces the filter, or to any previous packet?
Further, you have to inspect the Via branch parameter. Maybe the second
INVITE is not a retransmission (same branch parameter) but a second
branch (different branch parameter)
Hi Klaus:
thanks.
On 03/28/2011 12:15 PM, Klaus Darilion wrote:
Does ngrep really show the difference to the previous captured packet
which mathces the filter, or to any previous packet?
the ngrep man page shows:
-T Print a timestamp in the form of +S.UU, indicating the
de
HI
I guess I found the reason:
I have a bond interface (eth0_eth1), where -d any means any
interface, so I guess the ngrep will capture the traffic both from the
eth0 and bond0.
thx
min
Am 28.03.2011 17:43, schrieb Min Wang:
HI
I did the trace using ngrep:
ngrep -T -W
Hi,
you should check, if and where the rtpproxy is really listening. Just do a:
netstat -lp | grep rtpproxy (as root)
and you should see, if and where the RTP-Proxy is listening. If the
RTPProxy is really listening on "127.0.0.1:7722" you should see that
with netstat
Is there any Firewall on t
Is your iptables rules setup to default deny?
selinux enabled?
Can you manually netcat something to port 7722?
Does netstat -anp show udp port 7722 being bound?
On 3/28/11 11:45 AM, Linux Guy wrote:
Hi,
I still have the same issue, I have tested setting public ip,
localhost and 127.0.0.1 f
2011/3/28 Schumann Sebastian :
>> You are right, sorry. Anyhow, how you ever seen a XMPP server
>> integrating some kind of interoperability/gateway with SIP? In
>> Kamailio/*SER we have some attemps to interoperate with XMPP world.
>
> Openfire has SIP SIMPLE support.
What components of SIP SIMPL
Hi,
I have a kamailio 3.1.0 system.
I am using mostly the kamailio.cfg script that came with that release
but I've made a couple of changes
- The pstn.gw_ip bit doesn't work - syntax error - so I've substituted
fixed text for now
- I've expanded the PSTN section to IN_PSTN and OUT_PSTN to handle
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