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Regards,
Brian Southworth
Communications Developer
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I don’t think im trying to re reroute re invites not that ive noticed would it
help if you took a look on my cfg I am still new to opensips and learning as I
go along.
But I will also take another look thanks.
Regards,
Brian Southworth
From: Users [mailto:users-boun...@lists.opensips.org
back gets sent to .192 when it should go
back to .193
This doesn’t happen is I use rewritehostport but then on hold calls only work
for calls coming from that media server t_relay works across the board but
doesn’t actually work for inbound.
Regards,
Brian Southworth
From: Users
Please ignore my last email.
Both ways work it was an error in my code I wasn’t using t_relay();
Thanks for your help 😊
Regards,
Brian Southworth
From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Ovidiu Sas
Sent: 23 February 2018 13:10
To: users@lists.opensips.org
Hi Ovidiu,
I tried this in a if statement it works but same thing happens I go to take the
call off hold and it just drops the call.
I am also using v2.2 but I did check those docs too.
Regards,
Brian Southworth
From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of
Hi Ovidiu,
I tried this in a if statement it works but same thing happens I go to take the
call off hold and it just drops the call.
Regards,
Brian Southworth
Communications Developer
111 Wilmslow Road
Handforth
Wilmslow
SK9 3ER
T: 0 446677
W: www.clocom.uk <h
l (asterisk will handle all the B2B stuff)
Any help would be appreciated.
Regards,
Brian Southworth
Communications Developer
111 Wilmslow Road
Handforth
Wilmslow
SK9 3ER
T: 0 446677
W: www.clocom.uk <http://www.clocom.uk/>
tspeed.jcsnms.sbcglobal.net' (using password: YES)
Any idea how I fix this ? I want to start trying to use the EBR module which
was made available in opensips 2.3
Regards,
Brian Southworth
Communications Developer
___
Users ma
changed based on
the extension calling to match the correct call group.
$rU->accountcode (this is just an example).
I just thought being able to change this on the fly would be easier than
writing loads of new call groups into the Even based routing.
Regards,
Brian Southworth
Communicati
Hi Bogan,
Thanks for the reply, so are you saying the load balancer will send the call
over to the B2B and then to asterisk ?
Again sorry for my lack of knowledge there is still a lot I don’t understand or
know.
Regards,
Brian Southworth
Communications Developer
From: Bogdan-Andrei
ip that isn’t local.
So ive used the validate dialog code chunk and I get back the above
Regards,
Brian Southworth
Communications Developer
111 Wilmslow Road
Handforth
Wilmslow
SK9 3ER
T: 0 446677
W: www.clocom.uk <http://www.clocom
all the questions some of this is still very new to me.
Many thanks.
Regards,
Brian Southworth
From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Dragomir
Haralambiev
Sent: 08 February 2018 15:21
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Opensips to
Hi All,
How would I go about getting a opensips as a proxy to recognise the DTMF tones
or forward them onto asterisk ?
Regards,
Brian Southworth
Communications Developer
111 Wilmslow Road
Handforth
Wilmslow
SK9 3ER
T: 0 446677
W: www.clocom.uk <http://www.clocom
Hi Bogdan,
The Cisco phone, generates the refer once you press the xfer button when inside
a call.
Caller opensips asteriskCarrier
(cisco)
Regards,
Brian Southworth
Communications Developer
111 Wilmslow Road
Handforth
Wilmslow
SK9 3ER
T: 0 446677
W: www.clocom.uk <h
Regards,
Brian Southworth
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sent: 05 February 2018 15:47
To: Brian Southworth ; OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found
to match RR [1][XXX.XXX.XXX.XXX:5060]
Hi
.
Example:
Caller A receives call à caller A places call on hold and dials caller B à
caller B picks up à caller A presses cisco xfer and call is passed to caller B
I was hoping to achieve this using the proxy or asterisk box if possible.
I hope this helps.
Regards,
Brian Southworth
and out.
Regards,
Brian Southworth
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sent: 02 February 2018 14:20
To: Brian Southworth ; OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found
to match RR [1][XXX.XXX.XXX.XXX:5060]
H
Scenario:
Inbound call comes into the phone like so: carrier à ast à proxy à phone A
Phone A needs to transfer call to phone B: Phone A dials phone B à phone B
picks up à phone A presses xfer button and call is dropped.
Any help would be appreciated.
Regards,
Brian Southworth
From: Bogdan
.
Regards,
Brian Southworth
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Dns resolution issue? Do you use IPs or hostnames for your drouting
gateways?
On Monday, November 27, 2017, Callum Guy wrote:
> Hi All,
>
> This morning one of our OpenSIPs instances (2.1.4 (x86_64/linux)) had an
> issue that lasted around a minute, I'm trying to establish cause and
> wondered i
Hi Denis
We had similar issue recently with radius accounting in a single node in a
2.2 cluster. Rebuilding the node from scratch made the issue go away.
On Thursday, September 21, 2017, Denis via Users
wrote:
> In addition
>
> "
> make[2]: Entering directory '/distr/opensips-2.2/modules/aaa_
} else
exit;
}
sl_send_reply("404","Not here");
exit;
}
}
Regards,
Brian Southworth
Communications Developer
111 Wilmslow Road
Handforth
Wilmslow
SK9 3E
is active for 33 seconds until the call is cut. Due to a no reply to
critical packet 101.
The system works by using opensips as a proxy open sips then sets the dst uri
and the call is then t_relay() on to asterisk gateway.
Yeah I can confirm I am doing Record_route();
Regards,
Brian
.
Any help would be appreciated.
Regards,
Brian Southworth
Communications Developer
111 Wilmslow Road
Handforth
Wilmslow
SK9 3ER
T: 0 446677
W: www.clocom.uk <http://www.clocom.uk/>
<http://www.facebook.com/clocom.uk>
I guess this is because of unmet dependency on libfreeradius-client2 -
that appears to have been replaced by
https://packages.debian.org/stretch/libradcli4 so I guess packaging
needs to update to reflect this.
On Wed, Aug 9, 2017 at 6:27 PM, Brian : wrote:
> Hi List,
>
> Are th
Hi List,
Are the stretch packages ready for usage?
When installing opensips-radius-modules from stretch (
http://apt.opensips.org/pool/stretch//2.2-releases/o/opensips/opensips-radius-modules_2.2.5-1_amd64.deb
)
it doesn't install aaa_radius.so in modules - and if we test upgrade a
jessie box to
0:45] User-Agent: OpenSIPS (2.2.3 (x86_64/linux))
[Jun 21 10:10:45] Content-Length: 0
[Jun 21 10:10:45]
[Jun 21 10:10:45] <->
I for security reasons I have edited any IP’s if you need the unedited version
I will send it in an email directly
Regards,
Brian Southworth
Com
Hi All,
I seem to have an issue when using forward() my cisco phone says loop
detected.
If I used t_relay() my phone will say invalid destination both result in a
bye signal being sent by open sips, however the call is still passed to the
carrier
Regards,
Brian Southworth
Can you resolve the domain of the destination?
root@server# host xxx.domain.tld
xxx.domain.tld has address x.x.x.x
or Host xxx.domain.tld not found: 3(NXDOMAIN)
Brian
On Mon, Mar 14, 2016 at 6:52 PM, Travis Manson-Drake
wrote:
> Hello everyone,
>
>
>
> So I went ahead an impl
opensipsctl is just a script
on your command line do
which opensipsctl
/sbin/opensipsctl (your location maybe different)
then cat /sbin/opensipsctl or whatever the which command returns and you
will be able to see exactly what opensipsctl is doing.
On Sat, Mar 5, 2016 at 8:36 PM, Rodrigo Pimen
ou can cache it with memcache etc
> >
> > On Feb 9, 2016 1:02 PM, "Bogdan-Andrei Iancu"
> wrote:
> >>
> >> Hi Brian,
> >>
> >> Only the development version (2.2) has such a function,
> is_ip_registered() :
> >>
> http://www.ope
Hi List
When checking incoming requests with pike how would I do a lookup to see if
the IP belongs to a UA in the location table? ie - is there a function to
check if request is from registered subscriber while using pike?
Thanks
Brian
___
Users
This is not the correct way to go about what you are trying to achieve.
You *cant* edit the fifo file. You use (correctly configured) opensipsctl
fifo to send commands in realtime
On Tue, Dec 1, 2015 at 5:04 PM, Rodrigo Pimenta Carvalho
wrote:
> Dear OpenSIPS-users,
>
>
> I my opensips.cfg I ha
Could you share this in your cluster via multicast and just write down the
reports that come in periodically? Could even do unicast. Not sure why it has
to be sip at all.
--
Brian West
br...@freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH
> `tls_default_client_domain'
> /usr/local/src/opensips_1_8/cfg.y:871: undefined reference to
> `tls_default_server_domain'
> /usr/local/src/opensips_1_8/cfg.y:873: undefined reference to
> `tls_default_client_domain'
> /usr/local/src/opensips_1_8/cfg.y:881: undefined reference to
> `tls_default_server_domain'
Hi Cindy,
Just a wild guess but have you installed openssl-dev or libssl-dev ?
Brian
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from the
opensips domain table and rating works and add it back and it doesn't?
Thanks
Brian
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On Tue, Apr 10, 2012 at 1:56 PM, Brian ipt wrote:
> On Mon, Apr 9, 2012 at 7:39 AM, Adrian Georgescu wrote:
>> As the log line says "domain = " it, means no domain could be determined.
>> But why this happens for you it eludes me.
>>
>> Adrian
>>
c and uses the profiles accordingly.
Is it doing something to do with default/local domains?
Perhaps I am running into a small bug in 8.2.5?
Brian
> On Apr 7, 2012, at 11:09 AM, Brian ipt wrote:
>
>> On Fri, Apr 6, 2012 at 10:02 PM, Adrian Georgescu
>> wrote:
>>&
estId= 35678 domain= my.domain.tld Profile=900
Period=weekend Rate=902 Interval=8-18 Cost=.1000/60 Price=0.0020
PriceIn=0.0010
Thanks
Brian
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On Fri, Apr 6, 2012 at 9:02 AM, Brian ipt wrote:
> On Fri, Apr 6, 2012 at 8:55 AM, Iulian Macare wrote:
>> I had something similiar and I solved it by putting my domain in the
>> "domain" table in opensips database
>>
>
> I have the domain in the opensips
On Fri, Apr 6, 2012 at 8:55 AM, Iulian Macare wrote:
> I had something similiar and I solved it by putting my domain in the
> "domain" table in opensips database
>
I have the domain in the opensips database. And it turns out that this
is what seems to be causing the problem with CDRTool and not t
On Thu, Apr 5, 2012 at 5:17 PM, Brian ipt wrote:
> Hi,
>
> I have worked with version 6.7.1 for a long time.
> Just setting up version 8.2.5
>
> I can't seem to get it to rate by domain with this version.
>
> With all the exact same radius fields populated my domain f
e.
I can't see where I am going wrong here and I've been looking at this
for quite some time now.
Any help much appreciated.
Thanks in advance.
Brian
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On Thu, Feb 16, 2012 at 10:08 PM, J.R. Pauley wrote:
> If I rename the logfile to 'sips' same results. When I restart
> /etc/init.d/syslog restart it creates an empty logfile of the correct name
> /var/log/sips but with the log_name=sips and log_facility=LOG_LOCAL7 in
> the /etc/opensips/opensip
On Thu, Feb 16, 2012 at 9:26 PM, Brian ipt wrote:
>
>
> On Thu, Feb 16, 2012 at 9:20 PM, J.R. Pauley wrote:
>
>> linux centos 5
>>
>>
>>
>
> anything in syslog when your trying to start opensips and it fails?
>
> try changing log_name=opens
On Thu, Feb 16, 2012 at 9:20 PM, J.R. Pauley wrote:
> linux centos 5
>
>
>
anything in syslog when your trying to start opensips and it fails?
try changing log_name=opensips.log to log_name="sip"
> On Thu, Feb 16, 2012 at 4:18 PM, Brian ipt wrote:
>
>&g
On Thu, Feb 16, 2012 at 9:07 PM, J.R. Pauley wrote:
> Am looking forward to using opensips but so far spent all day trying to
> get logs to print to a file.
>
> my config file:
>
> fork=yes
> port=5060
> log_stderror=no
> debug=4
> log_name=opensips.log
> log_facility=LOG_LOCAL7
>
> I edited /etc
aproxy module is modifying SDP incorrectly for the second time..
use use_media_proxy and end_media_sessions to get around this for the
moment.
>
> On Fri, Feb 10, 2012 at 3:54 PM, Brian ipt wrote:
>
>>
>>
>> On Fri, Feb 10, 2012 at 9:21 PM, Charles Solar wrote:
>
t;, "Exhausted gateway list" );
> t_reply ("503", "Service not available");
> exit;
> }
> }
>
> Should I switch to use_media_proxy/end_media_session if I want to expec
going calls. Again, aside from the tweaks mentioned above this file
is exactly the same as the one that works perfectly on 1.6.1. Any
thoughts?
Thanks,
Brian
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Thanks everyone! I ended up using the prefix() function and it worked like a
charm.
Brian
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Dave Singer
Sent: Wednesday, February 23, 2011 3:35 PM
To: OpenSIPS users mailling list
Subject: Re
:
sip:12345#5615551212@111.222.333.444
Thanks in advance,
Brian
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=~ "^sip:1[0-9]{10}@")
{
xlog("L_INFO", "route - found a local/ld call with a 1
prefixed - removing [$fu/$tu/$ru/$ci]\n");
strip(1);
}
How would I go about adding 12345# to the given uri?
Thanks,
Brian
___
ips -c "SELECT time_stamp, method, \
convert_from(msg::bytea, 'SQL_ASCII'::name) FROM sip_trace;"
It might be nice for the one maintaining the db modules to indicate
that the text is stored UTF-8 (or ASCII) so that the user is not
forced to type the conversions.
Regards,
Brian
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What your looking for is now found under:
/var/www/opensips-cp/config/tools/system/cdrviewer, the files your looking
for are in there, cdrs.mysql and opensips_cdrs_1_6.mysql
Its just some old documentation, I had the same problem when installing it.
Hope this helps!
On 18 May 2010 20:17, zhiga
The default opensips configuration will be fine for what you want, unless
you want to enable database authentication. If not to get it started use the
opensipsctl program at /usr/local/sbin/opensipsctl start/stop/restart etc.
Hope this helps!
On May 15, 2010 7:42 AM, "Aditya Kumar" wrote:
Thank
Why are you splitting the registrations from Asterisk?
On 3 Apr 2010, at 20:31, James Lamanna wrote:
> Hi,
> I'm trying to use OpenSips as a registration server for Asterisk
> (assuming we can get presence working ok).
> Do I need to setup and use MediaProxy (or similar)? Or is the
> nathelper st
Hi List,
Is there much involved in moving a Kamailio 1.5.3 install to the current
version of OpenSIPS ?
Thanks,
Brian
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Hi Alex,
What user are you running openSips as ? Sounds like permissions problem.
Regards,
Brian
On 1 Mar 2010, at 19:42, Alejandro Recarey wrote:
> Hello everybody,
>
> I have just made a fresh install of OpenSIPS 1.6.1 using the
> http://debian.leurent.eu/ mirror.
>
>
This is awesome! Thank you and keep on good work!
Brian
-
Neotiq
www.neotiq.com
On Thu, Sep 10, 2009 at 2:45 PM, Anca Vamanu wrote:
> Hi,
>
> In the last weeks the configuration possibilities for OpenSIPS have been
> enriched with new additions related t
with point that profile at the special
rates with the default rates as fallback or is this a bad way of doing
it, ie., will rate the calls but generate a lot of errors?
Thanks,.
Brian
On 13 Feb 2009, at 09:03, Adrian Georgescu wrote:
>>>> Say a Provider operating and the normal
Hi Adrian,
On 12 Feb 2009, at 18:14, Adrian Georgescu wrote:
>
> On Feb 12, 2009, at 5:53 PM, Brian Chamberlain wrote:
>
>> Hi Adrian,
>>
>> I got my issue resolved with the rating not appearing. 2 Questions
>> for you:
>>
>
> Two? :-)
>
Yes Tw
of records you can do or some other criteria. It seems
if I try to do above 7 days it will not work.
Thanks,
Brian
Brian Chamberlain
Dot Net Solutions Ltd.
68 Parkwest Enterprise Centre,
Parkwest,
Dublin 12,
Ireland.
DDI:
[+353]
1
6296521
FAX:
[+353]
1
6237029
mobile:
[+353]
86
3883
a rate for
destination 3531, everything else seems fine. Any ideas?
Thanks,
Brian
On 12 Feb 2009, at 13:11, Brian Chamberlain wrote:
Hi All,
I have copied one of my radacct tables into radacct200807 to test my
billing and rates. It is a table from January. When I renormalise
for a day
e set billing intervals to
one minute.
How do I fix this?
Thanks,
Brian.
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different providers destination codes and
descriptions and the overlaps in between..
If it was possible to rate with the destination gateway it would make
things a lot easier.
Thanks,
Brian
On 20 Jan 2009, at 15:38, Adrian Georgescu wrote:
> If dest is 1 only rate for dest 1 is appli
and find 1617 as the rating record or am I
hoping for too much?
Regards,
Brian
On 20 Jan 2009, at 15:03, Adrian Georgescu wrote:
Hi Brian,
The logic of the rating first determines the destination then it
searches for a price for it. So for every entry in destinations
table you MUST have
though it
found the destination for 1617 in the destinations table?
Thanks,
Brian
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00 sec)
mysql>
It's quite odd as the only thing getting updated on the radacct table
for the CDR's is normalised = 1, I noted yesterday that canonicaluri
and destinationID were getting updated.
Thanks,
Brian
On 16 Jan 2009, at 14:23, Adrian Georgescu wrote:
What you desc
fresh from OpenSIPS. My
destinations table is empty.
With regard to the wiki are you talking about:
http://cdrtool.ag-projects.com/browser/doc/RATING.txt
Thanks,
Brian
On 16 Jan 2009, at 13:26, Adrian Georgescu wrote:
> You may have already normalized the CDRs when you had the
> dest
confusing as I have nothing in my destination table.. Does it
do some kind of external lookup? I did have information in the
destinations table which I purged some time ago, maybe its cached
somewhere..
Thanks,
Brian
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