Re: [OpenSIPS-Users] mediaproxy KBIn and KBOut
On 04/15/2011 02:42 PM, Dani Popa wrote:
Hi,
Mediaproxy radius request does not populate Kbin and Kbout. Also i tried
to see sessions on port 25061 and also there callee_bytes and
caller_bytes are 0.
opensips:~# telnet localhost 25061
Trying 127.0.0.1...
Connected to localhost.
Escape character is '^]'.
sessions
[]
sessions
[{"from_tag": "2a476a89", "start_time": 1302867244.13, "call_id":
"f2c9ca5f8ffe0e5ea422b57d7b93aa28@0:0:0:0:0:0:0:0", "duration": 5,
"streams": [{"status": "active", "caller_codec": "G711u",
"post_dial_delay": 3.43099498749, "callee_codec": "G711u",
"caller_bytes": 0, "start_time": 0, "callee_packets": 0, "callee_bytes":
0, "caller_packets": 0, "callee_remote": "192.5.32.12:8152", "end_time":
5, "caller_remote": "189.8.7.161:5105", "media_type": "audio",
"callee_local": "167.14.19.35:50010", "timeout_wait": 0, "caller_local":
"167.14.19.35:50008"}], "to_tag":
"a94c095b773be1dd6e8d668a785a9c8498c8c4db", "to_uri":
"cal...@domain.org", "caller_ua":
"Jitsi1.0-beta1-nightly.build.3408Linux", "callee_ua": "Cisco",
"from_uri": "cal...@domain.org"}]
[{"from_tag": "2a476a89", "start_time": 1302867244.13, "call_id":
"f2c9ca5f8ffe0e5ea422b57d7b93aa28@0:0:0:0:0:0:0:0", "duration": 11,
"streams": [{"status": "active", "caller_codec": "G711u",
"post_dial_delay": 3.43099498749, "callee_codec": "G711u",
"caller_bytes": 0, "start_time": 0, "callee_packets": 0, "callee_bytes":
0, "caller_packets": 0, "callee_remote": "192.5.32.12:8152", "end_time":
11, "caller_remote": "189.8.7.161:5105", "media_type": "audio",
"callee_local": "167.14.19.35:50010", "timeout_wait": 0, "caller_local":
"167.14.19.35:50008"}], "to_tag":
"a94c095b773be1dd6e8d668a785a9c8498c8c4db", "to_uri":
"cal...@domain.org", "caller_ua":
"Jitsi1.0-beta1-nightly.build.3408Linux", "callee_ua": "Cisco",
"from_uri": "cal...@domain.org"}]
Can someone give me a hint ?
Thanks,
Dani
Hi Dani,
What versions of the software are you using? KBIn and KBOut were renamed
quite some time ago. CDRTool will render Acct-Input-Octets and
Acct-Output-Octets fields, which are populated by MediaProxy using the
caller_bytes and callee_bytes attributes which you can see on the
statistics.
I just ran a quick test and it works here. Do the calls on the trace you
pasted have audio at all?
Regards,
--
Saúl Ibarra Corretgé
AG Projects
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[OpenSIPS-Users] Call forwarding to Android using OPenIMS core +Presence Server
Hello , I need some help regarding forwarding of call to to android phone when user is busy or don't want to receive call on their system.How can i do this using presence information. i have configured open ims core and presence server. Thanks in Advance -- *Abid khan * ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] need OpenSIPS installation
We are looking to implement OpenSIPS within our hosted VOIP environment. If you are interested in contracting out to assist us in the configuration and initial programming, please let me know. regards Jeff ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] need OpenSIPS installation
Contact bog...@opensips.org We are using them. They are the Best resource as they are the owners, developers of this project. Regards Deon On Apr 6, 2011, at 5:50 PM, Jeff Matson wrote: > We are looking to implement OpenSIPS within our hosted VOIP environment. If > you are interested in contracting out to assist us in the configuration and > initial programming, please let me know. > > regards > > Jeff > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenIMS core (localhost to real IP)
Hello everyone, I need help to configure openIMS core to real IP from local host, plz suggest me some links or documentation. Thanks in Advance. -- *Abid khan * ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] mediaproxy KBIn and KBOut
Hi,
As you can see, callee_bytes and caller_bytes are 0 during the call.
Media relay version is "media-relay 2.4.4". network hardware is Ethernet
controller: Broadcom Corporation NetXtreme II BCM5716 Gigabit Ethernet
(rev 20). The same issue i have with media-relay 2.4.3 on the same
machine. I wondering if is a network card driver issue or kernel
issue(if so, i'm dont know how to make troubleshooting, where should i
see the callee_bytes and caller_bytes in kernel stats).
Dani
On 04/18/11 10:43, Saúl Ibarra Corretgé wrote:
On 04/15/2011 02:42 PM, Dani Popa wrote:
Hi,
Mediaproxy radius request does not populate Kbin and Kbout. Also i tried
to see sessions on port 25061 and also there callee_bytes and
caller_bytes are 0.
opensips:~# telnet localhost 25061
Trying 127.0.0.1...
Connected to localhost.
Escape character is '^]'.
sessions
[]
sessions
[{"from_tag": "2a476a89", "start_time": 1302867244.13, "call_id":
"f2c9ca5f8ffe0e5ea422b57d7b93aa28@0:0:0:0:0:0:0:0", "duration": 5,
"streams": [{"status": "active", "caller_codec": "G711u",
"post_dial_delay": 3.43099498749, "callee_codec": "G711u",
"caller_bytes": 0, "start_time": 0, "callee_packets": 0, "callee_bytes":
0, "caller_packets": 0, "callee_remote": "192.5.32.12:8152", "end_time":
5, "caller_remote": "189.8.7.161:5105", "media_type": "audio",
"callee_local": "167.14.19.35:50010", "timeout_wait": 0, "caller_local":
"167.14.19.35:50008"}], "to_tag":
"a94c095b773be1dd6e8d668a785a9c8498c8c4db", "to_uri":
"cal...@domain.org", "caller_ua":
"Jitsi1.0-beta1-nightly.build.3408Linux", "callee_ua": "Cisco",
"from_uri": "cal...@domain.org"}]
[{"from_tag": "2a476a89", "start_time": 1302867244.13, "call_id":
"f2c9ca5f8ffe0e5ea422b57d7b93aa28@0:0:0:0:0:0:0:0", "duration": 11,
"streams": [{"status": "active", "caller_codec": "G711u",
"post_dial_delay": 3.43099498749, "callee_codec": "G711u",
"caller_bytes": 0, "start_time": 0, "callee_packets": 0, "callee_bytes":
0, "caller_packets": 0, "callee_remote": "192.5.32.12:8152", "end_time":
11, "caller_remote": "189.8.7.161:5105", "media_type": "audio",
"callee_local": "167.14.19.35:50010", "timeout_wait": 0, "caller_local":
"167.14.19.35:50008"}], "to_tag":
"a94c095b773be1dd6e8d668a785a9c8498c8c4db", "to_uri":
"cal...@domain.org", "caller_ua":
"Jitsi1.0-beta1-nightly.build.3408Linux", "callee_ua": "Cisco",
"from_uri": "cal...@domain.org"}]
Can someone give me a hint ?
Thanks,
Dani
Hi Dani,
What versions of the software are you using? KBIn and KBOut were
renamed quite some time ago. CDRTool will render Acct-Input-Octets and
Acct-Output-Octets fields, which are populated by MediaProxy using the
caller_bytes and callee_bytes attributes which you can see on the
statistics.
I just ran a quick test and it works here. Do the calls on the trace
you pasted have audio at all?
Regards,
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[OpenSIPS-Users] db_http timeout problem
I have inherited a 1.4 system which I just upgraded to 1.6 (current tgz
release). I am dealing with MESSAGEs and am having timeout issues on a
regex for some reason. Any pointers would be appreciated that would let
me resolve this.
I am sending from twinkle on a LAN. Opensips is running on debian
wheezy x86_64.
The first message goes through with no delay. Subsequent message to the
same or a different destination will hang for a long period (many many
minutes, unsure how long in total). If I restart either twinkle or
opensips the hang will still be present. If I restart both then it
restarts to its original "1st message is fine" behavior.
If I comment out the regex it will work perfectly, other than "composing
message" MESSAGEs get sent which I would rather avoid.
I do not see anything in the logs that indicates any problems, and was
wondering what could be done to have the regexp (or at least to filter
out non text MESSAGEs).
Additionally I am unable to escape the / in the content type. I tried
\/ and \\/ and both failed (unknown flag *). The content type (at least
in twinkle) for text messages is text/plain and for composing messages
application/something-I-forget. How does one escape / in a opensips
regex?
if (method == "MESSAGE") {
if(avp_check("$cT","re/text*/i")) {
}
}
--
Trixter aka Bret McDanel
website: http://www.0xdecafbad.com
pgp key: http://bit.ly/9XYK4b
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Re: [OpenSIPS-Users] need OpenSIPS installation
Jeff, I would be interested in working with you. I've been integrating OpenSIPS in hosted VOIP environment for 4 years. Unfortunately I will be laid off the end of this week, Apr 22 (I'm the last one to go). You can look at my resume on Linkedin ( http://www.linkedin.com/in/davesingercando ) or Monster.com or I can send you one if you are interested. Thanks, Dave On Wed, Apr 6, 2011 at 9:50 AM, Jeff Matson wrote: > We are looking to implement OpenSIPS within our hosted VOIP environment. If > you are interested in contracting out to assist us in the configuration and > initial programming, please let me know. > > > > regards > > > > Jeff > > > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Re-Invite and 491 Request Pending
Hi all
OpenSIPs seems to be sending a 491 Request Pending when a lot of RE-INVITES
are received.
I found out the following threads regarding this problem:
http://opensips.org/pipermail/users/2010-May/012724.html
http://www.mentby.com/Group/opensips-users/re-invite-problem-gt-491-request-pending.html
So, it seems that I could get rid of that error by adding a t_newtran()
before t_relay() when I handle an ACK:
http://www.opensips.org/html/docs/modules/1.6.2/tm.html#id294024
Like this:
*if ( is_method("ACK") ) {*
*if ( t_check_trans() ) {*
*# non loose-route, but stateful ACK; must be an ACK
after a 487 or e.g. 404 from upstream server*
*t_newtran();*
*t_relay();*
*exit;*
*} else {*
*# ACK without matching transaction ... ignore and
discard.\n");*
*xlog("L_WARN", "[$mi] discarding ACK\n");*
*exit;*
*};*
*};*
But I'm not sure if this is correct. I'd appreciate any comments to see if I
missed something before testing live.
Thank you and best regards,
Alejandro Rios Peña
DISCLAIMER: The opinions expressed are my own, and not necessarily those of
my employer.
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[OpenSIPS-Users] correlating Call-IDs of top-hidden dialogs
Hello, I see messages like this in the log: INFO:b2b_logic:b2b_add_dlginfo: Dialog pair: [6e13029c5fd56bdf0ef2339009984cac@74.125.91.103] - [B2B.39.5167001] I use CDRTool to examine the CDRs of the various proxies. In my configuration I've got non-B2B Opensips instances on either side of the B2B, each generating their own CDRs. With the Call-ID changing it becomes more of a challenge to follow the same call through all the relevant proxies. Is there some way to access this association of the two call IDs inside the script, and write it to db that I could later use to correlate the calls in a more automated fashion? - Jeff ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
