Unfortunately for you this will not work, for 3 reasons.
1. engage_media_proxy already calls use_media_proxy for every message in the
dialog, so by manually calling use_media_proxy on a certain message you do not
change the conditions of the problem, as engage_media_proxy already called it
for
2011/1/21 Duane Larson
> I can't remember which version of opensips 1.6.x made it to where xlog is
> no longer a extra module, but actually apart of the core.
>
http://www.opensips.org/Resources/DocsMigration162to163
> So you already have xlog when you compile opensips. Just remove the
> load
I can't remember which version of opensips 1.6.x made it to where xlog is no
longer a extra module, but actually apart of the core. So you already have
xlog when you compile opensips. Just remove the loadmodule and modparam
statements for xlog and you should be good.
Hope that is the issue you a
Hello,
In which Debian Lenny package might one find the xlog module? I'm using the
opensips.org/apt lenny repository and can't seem to find a reference to it.
Thanks,
Jeff
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org
Bogdan,
The dialogs are still showing in memory with the FIFO command and in MySQL.
I have the following in my opensips.cfg
modparam("dialog", "default_timeout", 7500)
modparam("dialog", "db_mode", 1)
One of my entries is showing this:
+++
| start_time | timeout|
+-
After looking a little more do you think this would cause the memory
issue...
Looks like my SIPP test starts the call, does whatever I want and sends the
BYE, but the dialog is still in the database after the call has ended
between the UAS and UAC. I am currently using USRLOC db_mode = 2, so I am
Bogdan,
I think I've got it now. Details inline.
On 1/20/11 3:44 PM, "Bogdan-Andrei Iancu" wrote:
>Jeff Pyle wrote:
We're looking to add a second Opensips instance on a separate server
for failover. Or, from an operational perspective, it could be
described as "active-active" si
On 01/19/2011 11:16 PM, Duane Larson wrote:
Saul,
Did you ever get a chance to make a patch?
Please, give the attachment in this ticket a try:
https://sourceforge.net/tracker/?func=detail&aid=3162970&group_id=232389&atid=1086412
--
Saúl Ibarra Corretgé
AG Projects
Jeff Pyle wrote:
We're looking to add a second Opensips instance on a separate server
for failover. Or, from an operational perspective, it could be
described as "active-active" since both will be available at any one
time. We'll control the traffic flow to the proxies with the SRV
records used b
Hi Bogdan,
On 1/20/11 2:54 PM, "Bogdan-Andrei Iancu" wrote:
>Hi Jeff,
>
>Jeff Pyle wrote:
>> Hello,
>>
>> We're looking to add a second Opensips instance on a separate server
>> for failover. Or, from an operational perspective, it could be
>> described as "active-active" since both will be avai
Yikes. Good to know; thanks.
Here's the issue. Today I use media relay in limited cases, only when I have
to have it for internal network reasons. Long story, but if I don't use it,
QoS doesn't happen right. Anyway, it's about 1% of total traffic. No big deal.
I've had no issues with the d
I do have some memory stuff in syslog. I have posted the output here
http://paste.ubuntu.com/556285/
On Thu, Jan 20, 2011 at 4:13 AM, Bogdan-Andrei Iancu wrote:
> Hi Duane,
>
> with
>
> mem_log=10
> mem_dump=1
> debug=3
>
> do you see at runtime any logs related to memory ?? (al
Hi Jeff,
Jeff Pyle wrote:
Hello,
We're looking to add a second Opensips instance on a separate server
for failover. Or, from an operational perspective, it could be
described as "active-active" since both will be available at any one
time. We'll control the traffic flow to the proxies with t
accidentally called twice in a script that is...
On Thu, Jan 20, 2011 at 1:53 PM, Duane Larson wrote:
> I think I have accidentally called use_media_proxy() in scripts and it has
> spit out syslog errors. I think it caused one way audio issues for me.
>
> On Thu, Jan 20, 2011 at 1:31 PM, Jeff Py
I think I have accidentally called use_media_proxy() in scripts and it has
spit out syslog errors. I think it caused one way audio issues for me.
On Thu, Jan 20, 2011 at 1:31 PM, Jeff Pyle wrote:
> Hello,
>
> What would happen if use_media_proxy() were called on a transaction (or
> dialog) wher
On Thu, Jan 20, 2011 at 1:31 PM, Jeff Pyle wrote:
> What would happen if use_media_proxy() were called on a transaction (or
> dialog) where it had already been used? The documented result codes don't
> cover this case.
>
>
Jeff,
I can't remember if this was with use_media_proxy() or another func
Hello,
What would happen if use_media_proxy() were called on a transaction (or dialog)
where it had already been used? The documented result codes don't cover this
case.
Would it simply start a new session with the dispatcher and close any previous
ones, modifying the SDP again? Or error out
Hello,
We're looking to add a second Opensips instance on a separate server for
failover. Or, from an operational perspective, it could be described as
"active-active" since both will be available at any one time. We'll control
the traffic flow to the proxies with the SRV records used by the
Just enable the nathelper module along with rtpproxy configured in
bridge mode and it will work fine.
Regards,
Ovidiu Sas
On Thu, Jan 20, 2011 at 10:45 AM, Alessandro Illiano
wrote:
> Hi Bogdan,
> Unfortunately i don't know... the problem is that we have many carriers
> connected and now we acce
Hi Bogdan,
Unfortunately i don't know... the problem is that we have many carriers
connected and now we accept traffic from their ip (only acl and fw rules to
prevent not authorized traffic).
May I solve configuring freeswitch servers enabling nat support?
But I don't know if this configuration ca
Thanks Bogdan,
I think this would be useful for me
Thanks!
> Message: 7
> Date: Thu, 20 Jan 2011 12:35:30 +0200
> From: Bogdan-Andrei Iancu
> Subject: Re: [OpenSIPS-Users] Gateways with SIP URIs
> To: OpenSIPS users mailling list
> Message-ID: <4d380ff2.3020...@opensips.org>
> Content-Type: t
Thank you!
I'll give it a try...
--
---
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: r...@kinetix.gr
---
On 20/1/2011 4:50 μμ, Bogdan-Andrei Iancu wrote:
ok, found the problem, a silly one . Dialog module n
ok, found the problem, a silly one . Dialog module needs loose_route to
work, so you need a more formated script (as logic) (better take a look
at the default opensips script):
route {
if(has_totag) {
#sequential request
loose_route();
t_relay();
On Thu, Jan 20, 2011 at 2:33 PM, Tijmen de Mes wrote:
> Hi,
>
> This one line is not the only thing changed in cdr_opensips.php. You should
> make sure you got the all the new files that are in the release.
>
> Did you also upgraded the database tables like instructed in the changelog
> and alter
Hi,
This one line is not the only thing changed in cdr_opensips.php. You
should make sure you got the all the new files that are in the release.
Did you also upgraded the database tables like instructed in the
changelog and alter_mysql?
From which version you upgraded?
Tijmen de Mes
AG Pro
On Thu, Jan 20, 2011 at 1:38 PM, Tijmen de Mes wrote:
> Hi,
>
> I looks like something failed in your update and you' ve stil got an old
> version of cdr_opensips.php.
>
> This bug was fixed in 8.0.7.
>
> It should be:
> $description=$this->destinations[0]["default"][$mygroup];
>
>
Yes, you are r
On 01/19/2011 09:31 PM, a.zagors...@oyster-telecom.ru wrote:
It was my conversation, and I told about
1) $Tsm is MICROseconds (6 digits)
2) It is microseconds of a current second.
Yes, that's true. I fixed it in documentation also.
--
Anca Vamanu
www.voice-system.ro
_
Enable full debugging by setting debug=6, make a complete call and send
me the full output of opensips.
Regards,
Bogdan
Apostolos Pantsiopoulos wrote:
I am using 1.6.4_notls.
I tried changing the script with no luck.
I currently have the following parameters for acc :
# - acc params
I am using 1.6.4_notls.
I tried changing the script with no luck.
I currently have the following parameters for acc :
# - acc params -
/* what sepcial events should be accounted ? */
modparam("acc", "early_media", 1)
modparam("acc", "report_ack", 1)
modparam("acc", "report_cancels", 1)
/
Hi,
I looks like something failed in your update and you' ve stil got an old
version of cdr_opensips.php.
This bug was fixed in 8.0.7.
It should be:
$description=$this->destinations[0]["default"][$mygroup];
Best regards,
Tijmen de Mes
AG Projects
Op 1/20/11 11:30 AM, Adrià Vidal schreef:
W
Hi Alessandro,
the problem is not the LB, but the presence of NAT between CARRIER and
LB - any NAT presence must be explicitly handled in SIPIn your case
is even more complicated as traffic comes from public, goes to private
and again to public..
Does your carrier support "direction:
What version of opensips are you using ? Looking at your acc content, I
guess the acc is generating START/STOP events, instead of generating
directly a CDR (per call). So, it seams that acc does not see the dialog
support...
Try to change the script as:
route {
### setflag(4); # remov
Hi Andrew,
if you are using the new CDR generation feature (acc with dialog
module), you cannot do it, as the acc module calculates the duration at
second level. So, by default you get only secondsThere is a
pseudo-variable to get the microseconds and you could use
extra_accounting to hav
Hi Bogdan,
Directly via radius_send_acct(). I imagining i should setup
Acct-Session-Type. In this scope, i belive it's more secure to use
acc_radius for accounting.
Thanks,
Dani
Bogdan-Andrei Iancu wrote:
> Hi Dani,
>
> do you use the aaa_radius directly (via radius_send_acct() ) or
> indirect
Hi all,
i'm new to opensips.
I would like to know if it's possible to implement this scenario
[CARRIER-A - internet]
---|---
---v---
[opensips LB - internet ip + LAN IP 192.168.1.1]
---|---
---v---
[FS01 - ip 192.168.1.2] or [FS02 - ip 192.168.1.3]
---|---
---v---
[OTHER PBX - internet] or [lo
Hi Dani,
do you use the aaa_radius directly (via radius_send_acct() ) or
indirectly via the ACC module ? if you use it directly, you need to
manually put all RADIUS AVPs in the requests, including the
Acct-Session-Type (see the radius sets -
http://www.opensips.org/html/docs/modules/1.6.x/a
Hi Ambert,
It is really strange as you say 1.2.1 does not work - have you tried
1.2.0 from http://rtpproxy.org/ ?
About debugging the not working version: run an ngrep to see the
communication between opensips and rtproxy (ngrep -d lo . port 8899), to
see if opensips is sending commands to r
Hi Bogdan,
Unfortunately this is not what I have.
I have an acc file (I falsely called it the CDRs file) which is in
"/var/log/acc/acc_5.log" due to the fact that I configured it like this
in acc :
modparam("acc", "db_url", "flatstore:/var/log/acc")
and the contents do not follow the "scripts
Hi,
If I understand correctly, you want to send a call to GW (to the IP of
the GW), but having as RURI the URI identifying one of the cards. Like
sending the call to 1.2.3.4 with RURI sip:4...@foo.com . Is this right ?
if so, you can do like (in route 4):
$ru = "sip:4...@foo.com"; #setting
We have upgraded our CDRtool to 8.0.13, and now when we make making a
research Grouped by Sip Destination Id
the Description field is empty
Taking a look to the code seem these part of the code is on the way
if ($this->group_byOrig==$this->DestinationIdField) {
Hi Duane,
with
mem_log=10
mem_dump=1
debug=3
do you see at runtime any logs related to memory ?? (alloc, dealloc
logs, no error) . Normally you shouldn't see anyonly when a me dump
is done (at shutdown for example).
Regards,
Bogdan
Duane Larson wrote:
I am trying to st
Hi Alan,
So those dialogs were removed from memory (you cannot see them listed by
"opensipsctl fifo dlg_list"), but they are still in DB ? If so, what are
the values for "timeout" field in DB for that those dialogs ?
Regards,
Bogdan
Alan Frisch wrote:
Using the LB module with Dialog and usi
Hi,
you can use usr_preferences avp_table and avp_load to load from db
incoming/outgoing barring rules.
in usr_preferences table you should have something like:
| 139 | | username | domain | 4122 |2 |
*barring_number* | 2011-01-11 12:19:31 |
define avp_aliases i:4122 named as y
Hi Apostolos,
There is no CDR file, only the ACC file (where the CDRs are generated by
acc + dialog modules). The description of the file (with 1.6.4):
scripts/dbtext/opensips/acc
is:
id(int,auto) method(string) from_tag(string) to_tag(string)
callid(string) sip_code(string) sip_rea
Hi,
Thanks for the prompt reply.
This is a sample of my CDRs file :
...
ACK|21912SIPpTag001|15280SIPpTag0113|1-21912@192.168.253.55|200|OK|1295513771|
BYE|21912SIPpTag001|15280SIPpTag0113|1-21912@192.168.253.55|200|OK|1295513772|
I don't think the duration is in there.
Regards,
--
--
Using the LB module with Dialog and using FreeSwitch/Asterisk to
handle my media.
Under 1.63 I had the occasional phantom call that hung around, but
would expire based on the dialog expiry value I had set.
When I upgraded to 1.64 the only change to my opensips.cfg that was
made was to remove the
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