Hi All
I've had a system setup for a long time, but one issue has always been
there and its come to a head.
I've always has problems with Asterisk not correctly selecting the call
route for inbound DID's because the INVITE sent to it via my core
(openSIPs) has the 'service number' not the DID in
Hi Brett
On 17/10/12 3:52 PM, Brett Nemeroff wrote:
> Well hold on a sec..
>
> First of all, the TO field is irrelevant. So whatever RURI you have
> (that's the top line INVITE URI), that's where we're sending the call
> to next. If the below invite hits asterisk it should be delivered to
> 111610
Hi Kevin
Surely there is a better solution than this, because all the Asterisk
systems I've seen have inbound routing without a registration for each DID.
The supplier of the commercially support Asterisk would need to make
changes which they are this point are not prepared to support, when
every
On 17/10/12 9:39 PM, Kevin Sandy wrote:
> Then the other providers are doing something similar to the prior suggestion
> - replacing $rU with $tU before relaying.
>
> One registration = one contact URI. So, multiple registrations, rewrite the
> user, or ditch the registrations and route to them b
Hi Guys
Does any one know how to setup CDRTool to handle B party billing. (ie
the person called pays for the call) ?
Australia has numbers a person/company can purchase which are almost
free to the person calling but are charged to the numbers owner at a per
minute rate. The numbers are in the fo
by way of a
> 6113x number (by the number owner) are charged a per minute call rate
> similar to the FreeCall 6118x numbers. Treat them the same.
>
> I don't believe changes are required to OpenSIPS.
>
> Mark
>
> On Sat, Nov 20, 2010 at 11:10 AM, Mike O'Connor wrot
Hi Jeff, Dan
My problem is the same but different, opensips 1.6.1
engage_media_proxy() works perfectly. Any version above that it does
not. Nothing in the logs even with full debug turned on.
Dan asked me to do a number of tests but I can not do them as the system
is in full production do a numbe
Hi All
I need to pre append an area code to phone numbers as configuring the
clients dial plans is driving me mad.
How do I update just the phone number in an invite. The match is
complicated as each state in Australia has a different set of prefixes
which would require the addition of a differen
> You can use regular expressions to match each state code and append
> the corresponding area code.
> These rules can be added in your database using OpenSIPS CP.
>
> [1] http://www.opensips.org/html/docs/modules/devel/dialplan.html
>
> Regards,
> --
> Răzvan Crainea
> Open
Mike
On 1/09/11 11:22 PM, Brett Nemeroff wrote:
> On Thu, Sep 1, 2011 at 4:51 AM, Mike O'Connor <mailto:m...@oeg.com.au>> wrote:
>
>
> So I need another method of rewriting the phone number, the $tU
> does not
> allow R/W so I need another function which
On 2/09/11 10:10 AM, Brett Nemeroff wrote:
> On Thu, Sep 1, 2011 at 6:22 PM, Mike O'Connor <mailto:m...@oeg.com.au>> wrote:
>
> Hi Brett
>
> So I need to rewrite the R-URI, the command which does this is
> 'rewriteuser' which only touches the act
Hi All
With the soon release of Opensips 2.4 which I understand is going to be
an LTS, I'm in the process of working though the issues of upgrading our
Opensips.
One of the major issues I currently have is some of the UACs will detect
the link quality dropping and will re-invite with an unsupport
Hi All
I'm working with a company running an old version of OpenSIPS (1.6.4),
who need to selectively block international calls. What was the
recommended method doing this back then?
I've been contracted to upgrade the whole system to current version of
everything but that is going to take t
Hi Guys
I'm having trouble finding the documentation which confirms that
OpenSIPS 1.6.x support using a text label instead of numbers for routes
and flags.
I have a working system so I did not want to play until I was sure that
I had fully understood the implementation.
A pointer to the document
Hi All
I'm trying to get drouting working, the issue is I can not work out
which tables and field its complaining about when it give the error.
'drouting:dr_load_routing_info: route <1> does not exist'
In 'dr_rules' I have a routeid of 1 for a number of rows, and if I
change this to a 2 or 5 the
'Hi All
The follow does not work, I've never seen an example of anyone trying to
use avp's to do this. All the example I've seen do a 'sethostport' to a
static address and then a t_relay.
## sethostport("192.168.2.100:5060");
## # do not set the missed call flag again
pseudo variables - only
> static strings.
>
> if you want to push data from variables, use the pseudo-vars to write into :
> $ru = $avp(i:1); # set whole URI
> $rd = $avp(i:1); # set domain
> $rU = $avp(i:1); # set username
>
> Regards,
> Bogdan
>
> M
back to opensips
with the corrected details.
Any one got a better idea ?
Thanks
Mike
On 8/01/10 7:37 PM, Mike O'Connor wrote:
> 'Hi All
>
> The follow does not work, I've never seen an example of anyone trying
> to use avp's to do this. All the example I
Hi All
Is there any way that I could have broken or changed something which
would cause a sip busy (486) to go to onreply route instead of failure
route ?
Thanks
Mike
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On 10/01/10 9:07 PM, Mike O'Connor wrote:
> Hi All
>
> Is there any way that I could have broken or changed something which
> would cause a sip busy (486) to go to onreply route instead of failure
> route ?
>
> Thanks
> Mike
>
>
>
Ok that is strange: This
ds it back again.
Thanks
Mike
On 26/01/10 10:39 PM, Bogdan-Andrei Iancu wrote:
> Hi Mike,
>
> why you say that "$ru =$avp(s:callfwdbusy); " does not work ? have you
> tried to print the $ru after the assignment to see if RURI was indeed
> changed?
>
> Regards,
> Bog
Hi All
I've been trying to get T38 faxing going and for testing I've been using
the t38 client in zoiper and asterisk 1.6.
The process is that the call goes to opensips which has a usr_pref which
indicates this is a fax number, the call is then forwarded to Asterisk.
Asterisk answers the call af
Hi Saúl
Thanks, too tell you the truth I was seeing this in the asterisk logs
but was not sure what a correct trace should look like.
I'll look into the correcting some other way.
Thanks
On 28/01/10 9:52 PM, Saúl Ibarra Corretgé wrote:
> Hi Mike,
>
> I received your trace, but let me answer
Hi All
I've a have a couple of customers who are all asking to use sipXecs,
which from my investigations does not support registrations. Instead it
excepts that the ITSP provide unauthenticated trunks for inbound and
outbound calls.
So my question is what is the recommended way of supporting this
ters just fine. Granted,
> there is a bunch of other stuff messed up, but I think that's more my
> customer's misunderstanding of how to configure it.
>
>
> - Jeff
>
>
> On Feb 2, 2010, at 1:51 AM, Mike O'Connor wrote:
>
>
>> Hi All
>>
>>
ip authentication from you as well.
>
> Again, pardon my ignorance.
>
>
>
> Mike O'Connor wrote:
>
>> I've never once been able to get sipXecs to send out a register packet,
>> the are very clear in the documentation that they expect a sip trunk. ie
>&g
}
The log message is being displayed but no calls are being sent to
mediaproxy.
Any ideas, any more information I could provide ?
Thanks
Mike O'Connor
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is sending any call details to media-dispatcher.
Apr 13 15:08:19 ser1 /usr/sbin/opensips[24945]: enage_media_proxy:
sip:0416106...@sip.*
Thanks
Mike
On 13/04/10 4:35 PM, Saúl Ibarra Corretgé wrote:
> Hi,
>
> On 13/4/10 7:49 AM, Mike O'Connor wrote:
>
>> Hi All
>>
&g
As it seems like opensips is not sending the calls to mediaproxy, should
I downgrade ?
Mike
On 13/04/10 3:19 PM, Mike O'Connor wrote:
> Hi All
>
> I just upgraded to opensips-1.6.2 and mediaproxy-2.4.2, I've made no
> changes to the opensips config but I have done the mysq
been compiling my own
64bit version.
Mike
On 15/04/10 9:41 AM, Mike O'Connor wrote:
> As it seems like opensips is not sending the calls to mediaproxy, should
> I downgrade ?
>
> Mike
>
> On 13/04/10 3:19 PM, Mike O'Connor wrote:
>
>> Hi All
>>
>>
Hi All
Has any one written a CDRTool E164 for Australia, I'm having a little
trouble getting CDRTool to rate some calls correctly because of the way
Australia has some exceptions to the rule.
Thanks in Advance.
Mike
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Hi All
I live in Australia and we have a few numbers which are outside of our
normal dialing ranges.
The following are standard numbers for land lines (10 Digits)
02
0312312332 (or 61312312332)
04
06
07
0884567532 (or 61884567532)
09
Now the special numbers are in the form of
132221 (costs a lo
Hi Adrian
>>
>> Where as a call to 132221 (ie the special) is rated as '
>> diverted-on-net' and is displayed as
>> 132...@sip.xxx.xx.xx
>
> This means that in your OpenSIPS configuration login did not set
> correctly the CanonicalURI.
>
Ok taking your explaination above I edited the CanonicalURI t
On 28/04/10 6:56 PM, Mike O'Connor wrote:
> Hi Adrian
>
>>> Where as a call to 132221 (ie the special) is rated as '
>>> diverted-on-net' and is displayed as
>>> 132...@sip.xxx.xx.xx
>>>
>> This means that in your OpenSIP
Hi All
I have a need to forward calls onward for a range of DID's, but the
other end is not going to Register. I think this is called trunking.
I need to be able to configure the DID's and the ip/port there being on
forwarded too.
What methods should I use to do this ?
I've look at a number of o
to add ENUM numbers in your DNS database and trusted
> peers in your proxy database, no need to configure much in OpenSIPS
> beside doing ENUM lookup and checking the trusted table.
>
>
> On Sat, 2010-05-01 at 16:28 +0930, Mike O'Connor wrote:
>
>> Hi All
>>
>>
em.
>
> Adrian
>
>
> On May 1, 2010, at 10:54 AM, Mike O'Connor wrote:
>
>
>> So there is no way for me to read from a db and rewrite the sip host
>> and
>> port ?
>>
>> Mike
>>
>> On 1/05/10 4:35 PM, Adrian Georgescu wrote
On 1/05/10 6:49 PM, Andreas Sikkema wrote:
> On May 1, 2010, at 9:05 AM, Adrian Georgescu wrote:
>
>
>> On Sat, 2010-05-01 at 16:28 +0930, Mike O'Connor wrote:
>>
>>> I have a need to forward calls onward for a range of DID's, but the
>>> ot
you already know
> the IP and port you want to send to so why not just do $rd=IPTOSENDTO;
> $rp=PORTTOSENDTO; ? These will accept AVP's as arguments.
>
> http://www.opensips.org/Resources/DocsCoreVar16#toc56
>
> http://www.opensips.org/Resources/DocsCoreVar16#toc61
>
&
On 2/05/10 8:36 AM, Laszlo wrote:
> $du = $avp(s:destination);
> $rd = $avp(s:reqdomain);
Hi Laszlo
Thank you for your example, the part that is most interesting to me is
the $du and $rd, they where mentioned by Richard and I was sure that my
testin
Hi All
>From my email of a last few days you will now that I'm trying to get
inbound and outbound trunking working for a system which does not
support registration.
After a lot of help, I've gotten the inbound DID forward to the equipment.
I was able to get the outbound trunking working nicely e
Hi All
I found a solution, I enabled the option in uac 'restore_mode' to auto,
and everything is good.
Cheers
Mike
On 2/05/10 2:42 PM, Mike O'Connor wrote:
> Hi All
>
> From my email of a last few days you will now that I'm trying to get
> inbound and outbou
not the
> end I have control over. If we were using Asterisk on both ends
> (rather than Cisco) we would have no issues.
>
> Dennis
>
> On Sun, May 2, 2010 at 8:11 PM, Mike O'Connor <mailto:m...@pineview.net>> wrote:
>
> Hi Dennis
>
> It would seem
Hi All
I'm running mediaproxy and opensips (current version) and I found an
error in my about mediaproxy not being able to access
'var/run/opensips/socket' so I worked out what was needed and enabled
mi_datagram. The end result of this was a load average of 9 in top.
Why would this be and what ca
, Bogdan-Andrei Iancu wrote:
> Hi Mike,
>
> In Subject you mention the mi_datagram, but in the email you talk about
> mediaproxy ?!?! what is the real report and what is the connection
> between mi_datagram and mediaproxy ??
>
> Regards,
> Bogdan
>
> Mike O'Con
On 12/05/10 4:36 PM, Saúl Ibarra Corretgé wrote:
> Hi Mike,
>
>> /etc/mediaproxy/config.ini
>> ---
>> [OpenSIPS]
>> ; Configure interaction between the media dispatcher and OpenSIPS
>>
>> ; Path to OpenSIPS's UNIX filesystem socket from the mi_datagram module.
>> socket_path = /var/run/opensips
On 12/05/10 5:00 PM, Saúl Ibarra Corretgé wrote:
> Hi,
>
>> As soon as opensips is started, and it continues for ever. I did not
>> notice it for a couple of days after I added the configuration, but it
>> had been there from that point. My munin graphs confirmed this.
>>
>
> So we can assume Media
9714.html
>
> Andrei
>
> Mike O'Connor wrote:
>
>> On 12/05/10 5:00 PM, Saúl Ibarra Corretgé wrote:
>>
>>
>>> Hi,
>>>
>>>
>>>
>>>> As soon as opensips is started, and it continues for ever. I di
Hi All
I seem to have broken my loose routing for NAT'd users, but I can not
see what I have wrong.
Any ideas what I might have wrong ?
Thanks as always.
Mike
if (client_nat_test("3")) {
fix_contact();
}
xlog("L_INFO", "New request - Request/failure/branch routes: M=$rm
RURI=$ru F=
Hi All
Just found in the mailing list history a comment about the onreply
branch needing to have a call to fix_contact().
I've just added this and the problem mentioned in my first email has
been fixed.
Thanks
Mike
On 30/05/10 11:13 AM, Mike O'Connor wrote:
> Hi All
>
> I
support UDP.
Reading the twisted docs it does not look like it would be hard to
support, but I do not have the experience to be sure.
Mike
On 13/05/10 10:40 PM, Mike O'Connor wrote:
> Yes that does look like the exact problem.
>
> I'll have to try the UDP option and see if
On 13/09/10 10:40 PM, John Khvatov wrote:
> Hello all!
>
> We are working on building geo-distributed VoIP solution with
> Mediaproxy.
>
> I think, that the current media relay selection algorithm is not perfect
> and can be improved. In the current implementation we set IP address of
> a particul
Hi Guys
I've been using OpenSIPS now for about 9 month (after upgrading from
OpenSER 1.2 used that for about 2 years) for my core SIP routing and
billing.
I'm now getting questions from customers about Virtual PBX functionality
and I would like the opinion of the group about how well this could b
er/master
capability and there is no support for multiply 'redis' masters.
For your thoughts
Mike O'Connor
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ei Iancu wrote:
> Hi Mike,
>
> That is completely new for me and checking on wikipedia, I see it is
> really new stuff - initial release in 2009
>
> As I understand this is more suitable for a new caching backend (along
> localcache and memcached), right ?
>
> Regard
On 9/11/10 3:55 AM, osiris123d wrote:
> I can't use {SIPDOMAIN} because the {SIPDOMAIN} variable is actually the IP
> address of callers phone as it appears in the location table.
>
> On a side note I was able to not use P-Asserted-Identity. because of a
> different issue I learned about the uac_r
ei Iancu wrote:
> Hi Mike,
>
> That is completely new for me and checking on wikipedia, I see it is
> really new stuff - initial release in 2009
>
> As I understand this is more suitable for a new caching backend (along
> localcache and memcached), right ?
>
> Regard
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