Re: [OpenSIPS-Users] [OpenSIPS-Devel] [Webinar] Next webinar is "Types of Routes in OpenSIPS"

[Bogdan-Andrei Iancu]

Recording available under:
  http://www.opensips.org/html/docs/video/webinar003/

Regards,
Bogdan

Bogdan-Andrei Iancu wrote:
> I apology - the correct registration link is 
> https://www2.gotomeeting.com/register/403938514
>
> Bogdan-Andrei Iancu wrote:
>   
>> Hi,
>>
>> The next free webinar is scheduled for Wednesday, August 12, 2009. The topic 
>> is "Types of Routes in OpenSIPS"
>>
>> This webinar will help you to understand how routing is done by OpenSIPS. 
>> How the SIP messages (requests/replies) are processed inside OpenSIPS and 
>> what types of scripting routes are available to help us with this.
>>
>> Also see http://www.opensips.org/Training/Webinars for future updates.
>>
>> Best regards,
>> Bogdan
>>
>>
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>>
>>   
>> 
>
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Re: [OpenSIPS-Users] OPenSIPS with SQLite3 database

[Ovidiu Sas]

Actually it works quite well on embedded systems, even with mysql enabled.
I have tested openser a while ago on ARM embedded platforms with 32M
of RAM (no swap).
If you don't want to use mysql, you can try dbtext.
If you want to use a real database, then you can run unixodbc and
connect to a db on a separate machine.

Regards,
Ovidiu Sas

On Wed, Aug 26, 2009 at 4:21 PM, Aryanto Rachmad wrote:
> Hello Bogdan,
>
> My question is actually related to one of the points listed on the
> feature list, which said:
>
> "OpenSIPS can run on embedded systems, with limited resources - the
> performances can be up to hundreds of call setups per second"
>
> As that is actually what I intended to do, which is running OpenSIPS on
> my WLAN router for my own personal use. I planned that years ago since
> OpenSER was still around, but I could never realised that :(
>
> For this purpose, I don't think MySQL would run acceptably well. So I
> think I will just give the unixodbc interface a try, as I don't think it
> is worth for my purpose to sponsor development and I am not a good coder. :)
>
> Kind regards,
>
> Anto
>
> Bogdan-Andrei Iancu wrote:
>> Hi Aryanto,
>>
>> indeed, the only way to have SQLite3 is via unixodbc. Right now only the
>> most used DBs backends got a direct implementation in OpenSIPS. On the
>> other hand if somebody is willing to do the job or sponsor the
>> development, there is no reason for adding it.
>>
>> Regards,
>> Bogdan
>>
>>
>> Aryanto Rachmad wrote:
>>
>>> Hello Everybody,
>>>
>>> This is my first post to this list, so greeting to all of you.
>>>
>>> I would like to know if anybody has successfully use OpenSIPS with
>>> SQLite3 database. I suppose we could use it through unixodbc, but is
>>> there any plan to support native interface like for MySQL?
>>>
>>> Thanks a lot in advance for your response.
>>>
>>> Kind regards,
>>>
>>> Anto
>>>
>>>
>>>
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>>>
>>>
>>
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Re: [OpenSIPS-Users] corrupted header

[Jan D.]

I filtert out the trace and altert some ip's etc, hope this will do in this
way. I uploaded it as trace_01.pcap 
http://n2.nabble.com/file/n3519219/trace_01.pcap trace_01.pcap 

Thanks,

Jan
-- 
View this message in context: 
http://n2.nabble.com/corrupted-header-tp3487025p3519219.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.

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Re: [OpenSIPS-Users] OPenSIPS with SQLite3 database

[Aryanto Rachmad]

Hello Bogdan,

My question is actually related to one of the points listed on the 
feature list, which said:

"OpenSIPS can run on embedded systems, with limited resources - the 
performances can be up to hundreds of call setups per second"

As that is actually what I intended to do, which is running OpenSIPS on 
my WLAN router for my own personal use. I planned that years ago since 
OpenSER was still around, but I could never realised that :(

For this purpose, I don't think MySQL would run acceptably well. So I 
think I will just give the unixodbc interface a try, as I don't think it 
is worth for my purpose to sponsor development and I am not a good coder. :)

Kind regards,

Anto

Bogdan-Andrei Iancu wrote:
> Hi Aryanto,
>
> indeed, the only way to have SQLite3 is via unixodbc. Right now only the 
> most used DBs backends got a direct implementation in OpenSIPS. On the 
> other hand if somebody is willing to do the job or sponsor the 
> development, there is no reason for adding it.
>
> Regards,
> Bogdan
>
>
> Aryanto Rachmad wrote:
>   
>> Hello Everybody,
>>
>> This is my first post to this list, so greeting to all of you.
>>
>> I would like to know if anybody has successfully use OpenSIPS with 
>> SQLite3 database. I suppose we could use it through unixodbc, but is 
>> there any plan to support native interface like for MySQL?
>>
>> Thanks a lot in advance for your response.
>>
>> Kind regards,
>>
>> Anto
>>
>>
>>
>> ___
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>>   
>> 
>
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Re: [OpenSIPS-Users] corrupted header

[Bogdan-Andrei Iancu]

Hi Jan,

Could you please send a masqueraded copy of the guilty message ? I want 
to run some simulations on it, in regards to logging and error detection.

Thanks and regards,
Bogdan

Jan D. wrote:
> Looks like Iñaki Baz Castillo is right, could be a buggy Sip ALG, the client
> is using a Zyxel P660.
>
> Still, I want to 'catch' the frame as a 'normal' error, also because it
> floods my log files with a lot of error lines. I just want to log one line,
> so I added a error_route at the end of my script:
>
> 
> # Error route
> 
> error_route
> {
> xlog("L_ERROR","--- error route class=$(err.class)
> level=$(err.level) info=$(err.info) rcode=$(err.rcode)
> rreason=$(err.rreason) ---\n");
> xlog("L_ERROR","--- error from [$si:$sp]\n+\n$mb\n\n\n");
> sl_send_reply("$err.rcode", "$err.rreason");
> exit;
> }
>
> But the logfile is still full of errors, the error route does not kick in:
>
> Aug 25 14:24:43 sip3 /usr/sbin/opensips[***]: ERROR:core:parse_via: bad port
> Aug 25 14:24:43 sip3 /usr/sbin/opensips[***]: ERROR:core:parse_via: 
>  145.*.*.22:23102:1;branch=z9hG4bK1505e2940e3e30927.2fc2d921ec4e7e4c7;rport#015#012Route:
> #015#012Max-Forwards:
> 70#015#012From: 31***285
> ;tag=25d7b2df4a#015#012To:
> ;tag=9D821F9C-C66#015#012Call-ID:
> 16b79f8c2c80a9cd#015#012CSeq: 18992 BYE#015#012Authorization: Digest
> username="username0023",realm="my.sipserver.com",nonce="4a93d91f0001382fafda66c1***52",uri="sip:31***...@212.*.*.245:5060",response="da7fdedfec2df8798ca3559f0b4cfd9e"#015#012Reason:
> Q.850; cause=16; text="Normal call clearing"#015#012Supported:
> timer#015#012User-Agent: HiPath 3000 V7.0 M5T SIP
> Stack/4.0.26.26#015#012Content-Length: 0#015#012#015#012>
> Aug 25 14:24:43 sip3 /usr/sbin/opensips[***]: ERROR:core:parse_via: parsed
> so far:
> Aug 25 14:24:43 sip3 /usr/sbin/opensips[***]: ERROR:core:get_hdr_field: bad
> via
> Aug 25 14:24:43 sip3 /usr/sbin/opensips[***]: ERROR:core:parse_msg:
> message= 145.*.*.22:23102:1;branch=z9hG4bK1505e2940e3e30927.2fc2d921ec4e7e4c7;rport#015#012Route:
> #015#012Max-Forwards:
> 70#015#012From: 31***285
> ;tag=25d7b2df4a#015#012To:
> ;tag=9D821F9C-C66#015#012Call-ID:
> 16b79f8c2c80a9cd#015#012CSeq: 18992 BYE#015#012Authorization: Digest
> username="username0023",realm="my.sipserver.com",nonce="4a93d91f0001382fafda66c1***52",uri="sip:31***...@212.*.*.245:5060",response="da7fdedfec2df8798ca3559f0b4cfd9e"#015#012Reason:
> Q.850; cause=16; text="Normal call clearing"#015#012Supported:
> timer#015#012User-Agent: HiPath 3000 V7.0 M5T SIP
> Stack/4.0.26.26#015#012Content-Length: 0#015#012#015#012>
> Aug 25 14:24:43 sip3 /usr/sbin/opensips[***]: ERROR:core:receive_msg:
> parse_msg failed
>
> Is there a way I can just 'drop' the INVITE (or whatever corrupt header is
> sent) and log a normal error? I hoped the error_route could do this.
>
> Jan
>   


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Re: [OpenSIPS-Users] Dialog information tracing in opensips Issue

[Bogdan-Andrei Iancu]

Hi Urmi,

some great philosopher said "patience is a virtues" ;).

I took a look at the log you sent and does not seam to be complete. I 
was looking for the processing of 200 OK reply and all I could found was:

Aug 25 16:06:43 [5488] DBG:core:parse_msg: version: 
Aug 25 16:06:43 [5488] DBG:core:parse_msg: status: <200>
Aug 25 16:06:43 [5488] DBG:core:parse_msg: reason: 


nothing following from the 5488 process...

Regards,
Bogdan

urmi lakkad wrote:
> Hello Bogdan,
>
> Thank you very much.
> Here I have attached my OpenSIPs Log. so please find the attachment.
>
> -Urmi
>
>
>
> Looks good - can you post the opensips logs (in debug=6) for this single
> call ? just to verify.
>
> Regards,
> Bogdan
>
> urmi lakkad wrote:
> > Hello Bogdan,
> >
> > Thank you for ur response.
> >
> > Here with this mail I have _attached my SIP call capture_ using ngrep.
> > So, please find the attachment. and do needful.
> >
> >
> > -Thanks
> > Urmi
> >
> > On Tue, Aug 25, 2009 at 12:41 PM, Bogdan-Andrei Iancu
> > mailto:bog...@voice-system.ro> 
> >> wrote:
> >
> > Post the SIP capture of the call you are testing with. Use "ngrep
> > -d any
> > . port 5060" to get the capture - this will solve the mystery.
> >
> > Regards,
> > Bogdan
> >
> > urmi lakkad wrote:
> > > Hello,
> > >
> > > Can u please suggest me some solution of my problem of DIALLOG
> > module ?
> > >
> > > Thank you for your attention.
> > >
> > > -Urmi
> > >
> > > 2009/8/21 urmi lakkad  
> > >
> > >  
>  > >
> > > Hello Brett,
> > >
> > > Thank you very much for quick response.
> > >
> > > My calls are working fine. I have checked through SIPp and also
> > > with Grandstream Phones. The call is working fine with out
> > > failure. At the time of call, I have started the wireshark to
> > > capture the packets, but there also I m not getting any negative
> > > reply like 400 or 300.
> > >
> > > See, my call is working fine, call dialog created successfully,
> > > but after that it destroyed, again new dialog is created n that
> > > too destroy. For a single call it creates 2 dialogs. But that
> > > dialog entry is not going to DB. Please suggest me the right thing
> > > to do.
> > >
> > > Thanks a lot for your attention.
> > >
> > >
> > > -Urmi
> > >
> > >
> > >
> > > On Thu, Aug 20, 2009 at 7:38 PM, Brett Nemeroff
> > > mailto:br...@nemeroff.com> 
> >
> >  
>  > >
> > > Urmi,
> > > You log shows the call having failed. I'm not sure why you
> > > think it runs for the proper duration. But as far as OpenSIPs
> > > is concerned, the call failed. It's likely a problem in your
> > > sipp scenario. It's very possible that sipp thinks the call is
> > > up, but the proxy does not.
> > >
> > > In any case, OpenSIPs is behaving as expected, the call fails,
> > > the dialog is destroyed.
> > > Aug 19 17:46:27 [6060] DBG:dialog:next_state_dlg: dialog
> > > 0x2d55af90 changed from state 1 to state 5, due event 1
> > > Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog
> > > 0x2d55af90
> > > failed (negative reply)
> > >
> > > BTW, a negative reply is >=400 (or may also include >= 300,
> > > can't remember). Check your traces, see where that comes from.
> > > -Brett
> > >
> > > On Thu, Aug 20, 2009 at 9:02 AM, urmi lakkad
> > > mailto:urmi.lak...@gmail.com> 
> >
> >  
>  > >
> > > Hello Stanisław Pitucha,
> > >
> > > Thank you for support.
> > >
> > > No, My call is established perfectly and is running for
> > > the specified duration without fail.
> > > I m firing the call using SIPp.
> > >
> > > Also, the dialog state gives me 1.
> > >
> > > Thanks for ur attention.
> > >
> > > -Urmi
> > >
> > > 2009/8/20 Stanisław Pitucha  
> > >
> > >  
>  > >
> > > 2009/8/20 urmi lakkad  
> > >
> > >  
>  > > > Am I doing right or not ? If not, please tell me the
> > > correct way.
> > > > One more thing, Is my configuration is correct or not ??
> > >
> > > It looks like your call doesn't even get accepted:
> > > Aug 19 17:46:27 [6060] DBG:

Re: [OpenSIPS-Users] Dialog count mismatches

[Bogdan-Andrei Iancu]

no, the dialog module count all ongoing dialog (disregarding the state 
they are). A dialog is considered onoging since the INVITE was sent out, 
till the BYE was received.

You do not have to do anything special on reply time.

Regards,
Bogdan

br...@nemeroff.com wrote:
> Is there a way to control that behavior to only count active and not early? 
> Just curious. So does that mean I shouldn't have to check for the 200 Reply 
> before I set the profile??
> -Brett
>
> Sent from my Verizon Wireless BlackBerry
>
> -Original Message-
> From: Bogdan-Andrei Iancu 
>
> Date: Wed, 26 Aug 2009 22:04:02 
> To: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] Dialog count mismatches
>
>
> Hi Brett,
>
> There was a bug in dialog module, in counting the dialogs in a profile - 
> the bug was fixed last week and it will be part of 1.5.3 that will be 
> released tomorrow.
>
> The bug consisted in counting also dialogs that were in DELETED state 
> (instead of counting only the EARLY and ACTIVE ones). The dialogs are 
> kept in DELETED state for like 5sec after the BYE.
>
> Try to upgrade from svn and see if the counting is now correct.
>
> Regards,
> Bogdan
>
> Brett Nemeroff wrote:
>   
>> All,
>> I'm running OpenSIPs 1.5.1. I use dialog profiling to "count" calls 
>> up. I notice that comparing my numbers to my providers using SBCs that 
>> my numbers are always MUCH higher than my provider for 
>> "simultaneous calls connected". For example, I may show 300 calls up, 
>> but they only show 75. The numbers are usually out of whack like this. 
>> The weird thing is, in very small quantities, 2-3 calls, I'm *sure* 
>> the numbers match.  Also, if they stop dialing entirely and I let the 
>> calls die off, the numbers return to 0 quickly enough. 
>>
>> To set my dialog profile, I do it in the onreply route like this:
>> if (t_check_status("200")) {
>> # Set my dialogs here
>> if (!is_in_profile("account","$avp(s:accountid)")) {
>> set_dlg_profile("account","$avp(s:accountid)");
>> }
>> if (!is_in_profile("trunk","$avp(s:trunkid)")) {
>> set_dlg_profile("trunk","$avp(s:trunkid)");
>> }
>>
>> }
>>
>>
>> Any thoughts on what I may be doing wrong? BTW, I do this in the 
>> onreply and not in the original INVITE because I'm trying to mimick 
>> the behavior of a nextone SBC which limits the number of CONNECTED 
>> calls, but doesn't limit the call setups. I think that's a silly way 
>> to do it, but if I don't match the behavior, nextone SBCs will show 
>> horrible ASRs for me as they'll continue to setup calls long after I 
>> start rejecting calls because too many are being set up.
>>
>> Thoughts? Thanks!
>> -Brett
>>
>>
>>
>> Thanks,
>> Brett
>>
>>
>>
>> from opensips -V: @(#) $Id: main.c 5469 2009-03-18 12:43:10Z 
>> bogdan_iancu $
>>
>>
>> 
>>
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Re: [OpenSIPS-Users] upgrade to 1.5.2 not sending Keep-Alives

[Bogdan-Andrei Iancu]

Hi Jayesh,

when you had 1.5.0 and 1.5.2 on the same machine, did the 1.5.2 used the 
same DB as the 1.5.0 ? Note that nat_traversal module store information 
(about where ping needs to be sent) on a file, and , because of 
different listen sockets on 1.5.0 instance and 1.5.2 instance, the file 
loaded by 1.5.2 produced no effect.

Regards,
Bogdan

Jayesh Nambiar wrote:
> Hello,
> I was running Opensips 1.5.0 in a prefix based installation which was 
> running fine. I have lot of NATed clients registering to it and I use 
> the nat_keepalive() function to maintain the NAT ports in REGISTER 
> requests.
> I downloaded 1.5.2 from source and compiled it in a different 
> directory and started it. Everything was running fine but only problem 
> was that Opensips stopped sending keep-alives to the NATed UAs.
> Scenario:
> /usr/local/opensips_1.1 --> Opensips 1.5.0
>
> /usr/local/opensips_1.2 --> Opensips 1.5.2
>
> I have created separate services like service opensips_1.1 for prefix 
> opensips_1.1 and service opensips_1.2 for prefix opensips_1.2 to start 
> and stop easily.
> Everything in the script is same since I didnt change the script. Only 
> when I start service opensips_1.2; the proxy just does not send any 
> keep-alives to the registered users; everything else works as expected.
>
> No errors nothing in the logs. Any ideas???
>
> Also to note; I installed Opensips 1.5.2 in a different server which 
> didnt have any previous installation of Opensips with same script; the 
> keep-alives are working as expected.
>
> Thanks,
>
> --- Jay
> 
>
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Re: [OpenSIPS-Users] Dialog count mismatches

[brett]

Is there a way to control that behavior to only count active and not early? 
Just curious. So does that mean I shouldn't have to check for the 200 Reply 
before I set the profile??
-Brett

Sent from my Verizon Wireless BlackBerry

-Original Message-
From: Bogdan-Andrei Iancu 

Date: Wed, 26 Aug 2009 22:04:02 
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Dialog count mismatches


Hi Brett,

There was a bug in dialog module, in counting the dialogs in a profile - 
the bug was fixed last week and it will be part of 1.5.3 that will be 
released tomorrow.

The bug consisted in counting also dialogs that were in DELETED state 
(instead of counting only the EARLY and ACTIVE ones). The dialogs are 
kept in DELETED state for like 5sec after the BYE.

Try to upgrade from svn and see if the counting is now correct.

Regards,
Bogdan

Brett Nemeroff wrote:
> All,
> I'm running OpenSIPs 1.5.1. I use dialog profiling to "count" calls 
> up. I notice that comparing my numbers to my providers using SBCs that 
> my numbers are always MUCH higher than my provider for 
> "simultaneous calls connected". For example, I may show 300 calls up, 
> but they only show 75. The numbers are usually out of whack like this. 
> The weird thing is, in very small quantities, 2-3 calls, I'm *sure* 
> the numbers match.  Also, if they stop dialing entirely and I let the 
> calls die off, the numbers return to 0 quickly enough. 
>
> To set my dialog profile, I do it in the onreply route like this:
> if (t_check_status("200")) {
> # Set my dialogs here
> if (!is_in_profile("account","$avp(s:accountid)")) {
> set_dlg_profile("account","$avp(s:accountid)");
> }
> if (!is_in_profile("trunk","$avp(s:trunkid)")) {
> set_dlg_profile("trunk","$avp(s:trunkid)");
> }
>
> }
>
>
> Any thoughts on what I may be doing wrong? BTW, I do this in the 
> onreply and not in the original INVITE because I'm trying to mimick 
> the behavior of a nextone SBC which limits the number of CONNECTED 
> calls, but doesn't limit the call setups. I think that's a silly way 
> to do it, but if I don't match the behavior, nextone SBCs will show 
> horrible ASRs for me as they'll continue to setup calls long after I 
> start rejecting calls because too many are being set up.
>
> Thoughts? Thanks!
> -Brett
>
>
>
> Thanks,
> Brett
>
>
>
> from opensips -V: @(#) $Id: main.c 5469 2009-03-18 12:43:10Z 
> bogdan_iancu $
>
>
> 
>
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Re: [OpenSIPS-Users] Dialog count mismatches

[Bogdan-Andrei Iancu]

Hi Brett,

There was a bug in dialog module, in counting the dialogs in a profile - 
the bug was fixed last week and it will be part of 1.5.3 that will be 
released tomorrow.

The bug consisted in counting also dialogs that were in DELETED state 
(instead of counting only the EARLY and ACTIVE ones). The dialogs are 
kept in DELETED state for like 5sec after the BYE.

Try to upgrade from svn and see if the counting is now correct.

Regards,
Bogdan

Brett Nemeroff wrote:
> All,
> I'm running OpenSIPs 1.5.1. I use dialog profiling to "count" calls 
> up. I notice that comparing my numbers to my providers using SBCs that 
> my numbers are always MUCH higher than my provider for 
> "simultaneous calls connected". For example, I may show 300 calls up, 
> but they only show 75. The numbers are usually out of whack like this. 
> The weird thing is, in very small quantities, 2-3 calls, I'm *sure* 
> the numbers match.  Also, if they stop dialing entirely and I let the 
> calls die off, the numbers return to 0 quickly enough. 
>
> To set my dialog profile, I do it in the onreply route like this:
> if (t_check_status("200")) {
> # Set my dialogs here
> if (!is_in_profile("account","$avp(s:accountid)")) {
> set_dlg_profile("account","$avp(s:accountid)");
> }
> if (!is_in_profile("trunk","$avp(s:trunkid)")) {
> set_dlg_profile("trunk","$avp(s:trunkid)");
> }
>
> }
>
>
> Any thoughts on what I may be doing wrong? BTW, I do this in the 
> onreply and not in the original INVITE because I'm trying to mimick 
> the behavior of a nextone SBC which limits the number of CONNECTED 
> calls, but doesn't limit the call setups. I think that's a silly way 
> to do it, but if I don't match the behavior, nextone SBCs will show 
> horrible ASRs for me as they'll continue to setup calls long after I 
> start rejecting calls because too many are being set up.
>
> Thoughts? Thanks!
> -Brett
>
>
>
> Thanks,
> Brett
>
>
>
> from opensips -V: @(#) $Id: main.c 5469 2009-03-18 12:43:10Z 
> bogdan_iancu $
>
>
> 
>
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Re: [OpenSIPS-Users] Regarding single user having multiple accounts

[Bogdan-Andrei Iancu]

Hi Ashwini,

Opensips has only the concept if User (subscriber). By default (if you 
do not use a billing engine), there is no concept of account, credit , 
etc...

Regards,
Bogdan

ASHWINI NAIDU wrote:
> Hi All,
>
> I need information whether a single user can have multiple accounts. 
>
> Ex: UserA having Account1, Account2. The prepaid topup/quota is 
> assigned to UserA. When Account1/Account2 calls the value should be 
> deducted from the UserA's balance.
>
> Is there a provision for this in opensips. Or is there a way for 
> working this around in opensips.
>
> -- 
> Thanking You,
> Ashwini BR Naidu
> 
>
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Re: [OpenSIPS-Users] 1.5.2 OpenSIPS and postgresql

[Bogdan-Andrei Iancu]

Hi Adolphus,

Note that the modules using DB do have a default DB URL pointing to 
mysql. My guess is you are using such a module and if you did not 
explicitly set a db_url from script, the module will try to use the 
mysql backend.

To see what module is about, see in the logs, in the error messages, 
after the message you just posted (it must be a raw of 2-3 error messages).

Regards,
Bogdan


adolphus wrote:
> Hi,
>
> I've been wrestling with postgres and OpenSIPs and have been getting the
> below error when I launch the binary that i build.  I sense that many of you
> folks have figured this out, but replys to querys seem a little scant to me
> on how the workaround is actually done.  
>
> I have db_mysql.so comment out (not used) in my opensips.cfg file and I have
> db_postgres.so commented in (so it is used) but I still get the below error
> when I supply:
>
> Does anyone have (or can recall) the series of steps the performed to get
> their systems working with postgresql?  (Any critical steps you used to get
> around the error might be helpful.)  I'm using Fedora 6 by the way.  
>
> My command line:
> opensips -f ./etc/opensips.cfg  (where my changes are in this file
> cfg file)
>
> my error:
> ERROR:core:db_check_api: module db_mysql does not export db_use_table 
>
>
> One standalone question is is db_mysql.so required whether you use
> db_postgres.so or not?
>
> Many thanks in advance
>
> -Adolphus
>   


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Re: [OpenSIPS-Users] OPenSIPS with SQLite3 database

[Bogdan-Andrei Iancu]

Hi Aryanto,

indeed, the only way to have SQLite3 is via unixodbc. Right now only the 
most used DBs backends got a direct implementation in OpenSIPS. On the 
other hand if somebody is willing to do the job or sponsor the 
development, there is no reason for adding it.

Regards,
Bogdan


Aryanto Rachmad wrote:
> Hello Everybody,
>
> This is my first post to this list, so greeting to all of you.
>
> I would like to know if anybody has successfully use OpenSIPS with 
> SQLite3 database. I suppose we could use it through unixodbc, but is 
> there any plan to support native interface like for MySQL?
>
> Thanks a lot in advance for your response.
>
> Kind regards,
>
> Anto
>
>
>
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Re: [OpenSIPS-Users] how to uninstall

[Bogdan-Andrei Iancu]

Hi Tseveendorj,

And for 1.6 we will add an uninstall option in makefile :)

Regards,
Bogdan

Tseveendorj Ochirlantuu wrote:
> SOLVED. 
>
> I removed some folders and files
>
>  /usr/local/etc/opensips
>  /usr/local/sbin/opensips*
>  /usr/local/lib/opensips
>
> and also remove source code.
>
> rebuilding it from new source code.
>
> Thank you guys.
>
>
> On Wed, Aug 26, 2009 at 3:51 PM, Tseveendorj Ochirlantuu 
> mailto:tseveend...@gmail.com>> wrote:
>
> Yes. You can see find result.
>
> eb...@beastie:/var/log$ sudo find / -name opensips
> /var/lib/mysql/opensips
> /usr/src/opensips-1.5.2-notls/scripts/dbtext/opensips
> /usr/src/opensips-1.5.2-notls/scripts/db_berkeley/opensips
> /usr/src/opensips-1.5.2-notls/opensips
>
>
>
> On Wed, Aug 26, 2009 at 2:56 PM, Saúl Ibarra  > wrote:
>
> Have you removed everything OpenSIPS related under /usr/local/tec,
> /usr/local/bin and /usr/local/lib ?
>
>
>
> --
> /Saúl
> http://www.saghul.net | http://www.sipdoc.net
>
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>
>
>
> 
>
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[OpenSIPS-Users] Dialog count mismatches

[Brett Nemeroff]

All,I'm running OpenSIPs 1.5.1. I use dialog profiling to "count" calls up.
I notice that comparing my numbers to my providers using SBCs that my
numbers are always MUCH higher than my provider for "simultaneous calls
connected". For example, I may show 300 calls up, but they only show 75. The
numbers are usually out of whack like this. The weird thing is, in very
small quantities, 2-3 calls, I'm *sure* the numbers match.  Also, if they
stop dialing entirely and I let the calls die off, the numbers return to 0
quickly enough.

To set my dialog profile, I do it in the onreply route like this:
if (t_check_status("200")) {
# Set my dialogs here
if (!is_in_profile("account","$avp(s:accountid)")) {
set_dlg_profile("account","$avp(s:accountid)");
}
if (!is_in_profile("trunk","$avp(s:trunkid)")) {
set_dlg_profile("trunk","$avp(s:trunkid)");
}

}


Any thoughts on what I may be doing wrong? BTW, I do this in the onreply and
not in the original INVITE because I'm trying to mimick the behavior of a
nextone SBC which limits the number of CONNECTED calls, but doesn't limit
the call setups. I think that's a silly way to do it, but if I don't match
the behavior, nextone SBCs will show horrible ASRs for me as they'll
continue to setup calls long after I start rejecting calls because too many
are being set up.

Thoughts? Thanks!
-Brett



Thanks,
Brett



from opensips -V: @(#) $Id: main.c 5469 2009-03-18 12:43:10Z bogdan_iancu $
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Re: [OpenSIPS-Users] init.d file?

[Bogdan-Andrei Iancu]

Hi Alex,

Typically the init file do check if the binary does exits -  something 
like : test -f $DAEMON || exit 0  

Check if the $DAEMON points to the proper binary.

Regards,
Bogdan

Alex G wrote:
> am sure i am running the start command correctly. all chmod 
> permissions are correct, default file is in etc/default, am sure the 
> daemon variable is pointing to the correct directory for opensips
>
> i am getting some ouput from the init script, just is not starting it
>
>
>
> On Wed, Aug 26, 2009 at 10:10 AM, Iñaki Baz Castillo  > wrote:
>
> 2009/8/26 Alex G  >:
> > There's no pid. Something is still missing in the init file. If
> I'm at
> > least getting output from the script, this means the daemon check is
> > succeeding.
>
> Ensure you run:
>
> /etc/init.d/openser start
>
>
>
> --
> Iñaki Baz Castillo
> mailto:i...@aliax.net>>
>
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> 
>
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Re: [OpenSIPS-Users] drouting module

[Bogdan-Andrei Iancu]

Hi Sebastian,

Sebastian Sastre wrote:
>
> Hi,
>
>  
>
> When applying the do_routing function without specifying the group id, 
> what are the matching patterns that it will look for?
>
As documented, if no group id is given to the function, it will try to 
discover the group Id by looking in the dr_group table for a record 
belonging to the FROM USER.

>  
>
> In other words If I want ALL the users behind domain test.com how 
> would I do it?
>
>  
>
> Leaving the username field empty and putting test.com in the domain 
> did not work.
>
I guess you did this is dr_group table, but this requires a full 
username and domain.

Note that do_routing() does accept the group id via a variable also, so 
you can load the group id from a DB via whatever query (see 
avp_db_query()) function. Or you can use dialplan module to have a 
translation from domain names to group ids . Or, the most simple static cfg:

$var(dr_id) = 0;
switch($rd) {
   case "test.com" : $var(dr_id) = 1; break;
   case "probe.com" : $var(dr_id) = 2; break;
   
   }

do_routing("$var(dr_id)");

Regards,
Bogdan
>
>  
>
> Thanks
>
>  
>
>  
>
> Sebastian
>
>  
>
>  
>
>  
>
>  
>
> 
>
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Re: [OpenSIPS-Users] Next OpenSIPS releases

[Bogdan-Andrei Iancu]

All pending items are completed.

Tomorrow, 27 August, 1.5.3 is planed for release.

Regards,
Bogdan

Bogdan-Andrei Iancu wrote:
> 1) Minor release 1.5.3
> ---
>
> Why: This is needed as more than 50 fixes were done on the 1.5 branch 
> since 1.5.2.
>
> Date: during this week
>
> Pending: personally I'm hunting an memory leak in SNMP module (mainly 
> design issues). If someone is aware of any other issues that requires 
> fixing in 1.5 branch, please speak up.
>   


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Re: [OpenSIPS-Users] init.d file?

[Alex G]

am sure i am running the start command correctly. all chmod permissions are
correct, default file is in etc/default, am sure the daemon variable is
pointing to the correct directory for opensips

i am getting some ouput from the init script, just is not starting it



On Wed, Aug 26, 2009 at 10:10 AM, Iñaki Baz Castillo  wrote:

> 2009/8/26 Alex G :
> > There's no pid. Something is still missing in the init file. If I'm at
> > least getting output from the script, this means the daemon check is
> > succeeding.
>
> Ensure you run:
>
> /etc/init.d/openser start
>
>
>
> --
> Iñaki Baz Castillo
> 
>
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Re: [OpenSIPS-Users] drouting module

[Sebastian Sastre]

 

Sorry bout that, I need help with this problem and I thought this was the
correct place to ask, If I’m posting incorrectly please let me know. I
resorted to asking this because i could not find enough documentation on the
group table. If anyone has extra documentation please send it over.

 

I am trying to authenticate all traffic coming from a particular domain/IP
since the phones are not directly registered to the Opensips. Therefore the
source might not be an active user but rather a caller id from a regular
pots line that comes from an access router.

 

Log 

ERROR:drouting:get_group_id: no group for user "12345"@"192.168.10.5"

ERROR:drouting:do_routing: failed to get group id

 

Table

"id","username","domain","groupid","description"

1,"","192.168.10.5",100,"test"

 

Is it possible to use an expression like % or * ?

 

As additional information this works perfect

"id","username","domain","groupid","description"

1,"12345","192.168.10.5",100,"test"

 

 

 

Thanks again.

 

 

  _  

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of David Villasmil
Sent: Tuesday, August 25, 2009 6:53 PM
To: OpenSIPS users mailling list
Cc: 
Subject: Re: [OpenSIPS-Users] drouting module

 

stop asking so much!

 

;)

 

D


El 25/08/2009, a las 18:05, "Sebastian Sastre" 
escribió:

Hi, 

 

When applying the do_routing function without specifying the group id, what
are the matching patterns that it will look for?

 

In other words If I want ALL the users behind domain test.com how would I do
it? 

 

Leaving the username field empty and putting test.com in the domain did not
work. 

 

Thanks 

 

 

Sebastian

 

 

 

 

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Re: [OpenSIPS-Users] Multiple Area Codes in Customer Area

[osiris123d]

Nice.

Thanks for the help

On Wed, Aug 26, 2009 at 10:55 AM, Bogdan-Andrei Iancu (via Nabble) <
ml-user+121376-262234...@n2.nabble.com
> wrote:

> Hi Duane,
>
> You can replace the complicated:
>
> subst_uri('/^sip:([0-9]+)@(.*)$/sip:$avp(s:areacode)\...@\2/i');
>
> with more nicer:
>
> $rU =  $avp(s:areacode)  + $rU;
>
> Regards,
> Bogdan
>
>
> osiris123d wrote:
> > Got it working!  Man OpenSIPS sure can do anything with SIP
> >
> > So here is what I did for future searchers
> >
> > So the users account is a 7 digit DID number XXX at blah.com
> >
> > I set up an AVP called areacode for the whole domain blah.com (this
> assumes
> > that the whole domain blah.com is only in one areacode)
> >
> > opensipsctl avp add -T usr_preferences 0 at blah.com areacode 1 201
> > opensipsctl avp add -T usr_preferences 0 at foo.com areacode 1 339
> >
> >
> > In the opensips.cfg file I do the following (it depends on your config as
> to
> > where you want to put this)
> > if (uri=~"^sip:[2-9][0-9]{6}@") {
> > avp_db_load("$ru/domain","$avp(s:areacode)");
> > subst_uri('/^sip:([0-9]+)@(.*)$/sip:$avp(s:areacode)\...@\2/i');
> > };
> >
> > So when someone calls a 7 digit number the avp_db_load() loads the
> variable
> > for areacode and the subst_uri adds the areacode at the beginning of the
> > Request-URI.
> >
> >
> >
> >
> >
> >
> > Bogdan-Andrei Iancu wrote:
> >
> >> Hi Duane,
> >>
> >> You can correlate AVPs you a USER, a DOAMAIN, etc - it is up to you,
> >> from the script, when loading the AVP - is a pure logical mapping.
> >>
> >> Regards,
> >> Bogdan
> >>
> >> osiris123d wrote:
> >>
> >>> I was reading Flavio's "Building Telephony Systems with OpenSER"
> chapter
> >>> about AVPOPs and he mentions that AVP's can be used for a whole domain.
>
> >>> I
> >>> was thinking that I might be able to configure a area code for Company
> >>> A's
> >>> domain and then route calls that way.  If not that then I can set the
> AVP
> >>> on
> >>> the fly within the transaction by looking at the callers Request URI's
> >>> first
> >>> 3 digits and route it appropriately.
> >>>
> >>>
> >>> Bogdan-Andrei Iancu wrote:
> >>>
> >>>
>  Hi,
> 
>  Requirements on the format of  CONTACT and TO headers are nonsense as
>  they are not used for routing at all. Only FROM (which provides info
> on
>  the caller) and RURI (request URI) (which provide info on callee).
> 
>  So, bottom line, only the normalization of the RURI should be required
>
>  on the system.
> 
>  Regards,
>  Bogdan
> 
>  osiris123d wrote:
> 
> 
> > Thanks for the info.
> >
> > I will look into this and work up a config.
> >
> > I also got this direct email about my post from someone else who
> lives
> > in
> > the US.  I figured I would go ahead and post below what he sent just
> so
> > its
> > out there.
> >
> >
> > Hello Duane --
> >
> > You have hit on one of the more difficult areas in SIP and telephony
> in
> > general -- especially here in the North American Numbering Plan.
>  Below
> > I
> > will address the problem in general, and not particularly related to
> > the
> > OpenSIPs question, because IMO you need a solution that will work in
> > any
> > architecture, not just OpenSIPs.
> >
> > In a nutshell, I recommend that for your USA users you:
> >
> > 1.) Require From: and Contact: headers to be in NANPA National (10
> > digit)
> > format.  This is method is standard in the telephone industry, and
> will
> > allow easy integration with North American ANI or Caller ID format,
> > especially when a call may eventually be handed off to the PSTN.
> >
> > 2.)  Require incoming To: headers to be in e.164 International
> format,
> > i.e.
> > NANPA-destination numbers all begin with the 1 digit, followed by the
>
> > 10
> > digit National number.   Any incoming call to 612xxx goes to
> > Sydney,
> > Austrailia, and not Minneapolis, MN.  This requirement should be
> > enforced
> > at
> > the perimeter of your network, where Customer Equipment can enforce
> the
> > "local" digit normalization policy.
> >
> > 3.)  If you can't enforce #2 above, you will need to "Normalize"
> > incoming
> > calls to the e.164 International format prior to routing.  The
> > unfortunate
> > reality here in the USA is that the requirements for how many digits
> to
> > dial
> > for a given destination (the "dialing plan") depends on where the
> call
> > comes
> > from.   Here in the Chicago area, residents of the 847 area code must
>
> > today
> > dial all calls as 11 digits.  Calls within the 312 or 773 area code
> may
> > today be dialed as 7 digits, however beginning on 07 November, all
> > calls
> > originating in 312 and 773 must be dialed as 1+10 digits, due to the
> > new
> > 872
> > ove

Re: [OpenSIPS-Users] init.d file?

[Iñaki Baz Castillo]

2009/8/26 Alex G :
> There's no pid. Something is still missing in the init file. If I'm at
> least getting output from the script, this means the daemon check is
> succeeding.

Ensure you run:

/etc/init.d/openser start



-- 
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[OpenSIPS-Users] Regarding single user having multiple accounts

[ASHWINI NAIDU]

Hi All,

I need information whether a single user can have multiple accounts.

Ex: UserA having Account1, Account2. The prepaid topup/quota is assigned to
UserA. When Account1/Account2 calls the value should be deducted from the
UserA's balance.

Is there a provision for this in opensips. Or is there a way for working
this around in opensips.

-- 
Thanking You,
Ashwini BR Naidu
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Re: [OpenSIPS-Users] init.d file?

[Alex G]

There's no pid. Something is still missing in the init file. If I'm at
least getting output from the script, this means the daemon check is
succeeding.

On 8/26/09, Saúl Ibarra  wrote:
> Look at /var/run to see if you still have an opensips pid file,
> regardless it has been stopped.
>
> 2009/8/26, Alex G :
>> not using the dpkg because i have all my opensips servers built already.
>> for
>> the sake of using monit, i require an init.d file to be there to
>> start/stop
>> opensips.
>>
>> i've changed the pointer for DAEMON and added the default file. I'm
>> getting
>> "Starting opensips: opensips already running." but its obviously not
>> running. I can see I'm getting closer, but still missing something... any
>> more tips? :)
>>
>>
>>
>> On Tue, Aug 25, 2009 at 5:14 PM, Saúl Ibarra  wrote:
>>
>>> You'll need to make it executable as well as copy the opensips.default
>>> file to /etc/default. That should do the job :)
>>>
>>> Anyway, why don't you make the debian packages so that you install
>>> opensips by just doing dpkg -i and everything is done automagically?
>>>
>>>
>>>
>>> --
>>> /Saúl
>>> http://www.saghul.net | http://www.sipdoc.net
>>>
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>>>
>>
>
> --
> Enviado desde mi dispositivo móvil
>
> /Saúl
> http://www.saghul.net | http://www.sipdoc.net
>
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[OpenSIPS-Users] upgrade to 1.5.2 not sending Keep-Alives

[Jayesh Nambiar]

Hello,I was running Opensips 1.5.0 in a prefix based installation which was
running fine. I have lot of NATed clients registering to it and I use the
nat_keepalive() function to maintain the NAT ports in REGISTER requests.
I downloaded 1.5.2 from source and compiled it in a different directory and
started it. Everything was running fine but only problem was that Opensips
stopped sending keep-alives to the NATed UAs.
Scenario:
/usr/local/opensips_1.1 --> Opensips 1.5.0

/usr/local/opensips_1.2 --> Opensips 1.5.2

I have created separate services like service opensips_1.1 for prefix
opensips_1.1 and service opensips_1.2 for prefix opensips_1.2 to start and
stop easily.
Everything in the script is same since I didnt change the script. Only when
I start service opensips_1.2; the proxy just does not send any keep-alives
to the registered users; everything else works as expected.

No errors nothing in the logs. Any ideas???

Also to note; I installed Opensips 1.5.2 in a different server which didnt
have any previous installation of Opensips with same script; the keep-alives
are working as expected.

Thanks,

--- Jay
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Re: [OpenSIPS-Users] Dialog information tracing in opensips Issue

[urmi lakkad]

Hello,

Will u please suggest me the some solution of my DIALOG module problem !!
Thank u very much in advance.


-Urmi

On Tue, Aug 25, 2009 at 4:59 PM, urmi lakkad  wrote:

> Hello Bogdan,
>
> Thank you very much.
> Here I have attached my OpenSIPs Log. so please find the attachment.
>
> -Urmi
>
>
>
>
> Looks good - can you post the opensips logs (in debug=6) for this single
> call ? just to verify.
>
> Regards,
> Bogdan
>
> urmi lakkad wrote:
> > Hello Bogdan,
> >
> > Thank you for ur response.
> >
> > Here with this mail I have _attached my SIP call capture_ using ngrep.
> > So, please find the attachment. and do needful.
> >
> >
> > -Thanks
> > Urmi
> >
> > On Tue, Aug 25, 2009 at 12:41 PM, Bogdan-Andrei Iancu
> > mailto:bog...@voice-system.ro>> wrote:
> >
> > Post the SIP capture of the call you are testing with. Use "ngrep
> > -d any
> > . port 5060" to get the capture - this will solve the mystery.
> >
> > Regards,
> > Bogdan
> >
> > urmi lakkad wrote:
> > > Hello,
> > >
> > > Can u please suggest me some solution of my problem of DIALLOG
> > module ?
> > >
> > > Thank you for your attention.
> > >
> > > -Urmi
> > >
> > > 2009/8/21 urmi lakkad  > 
> > > >>
> > >
> > > Hello Brett,
> > >
> > > Thank you very much for quick response.
> > >
> > > My calls are working fine. I have checked through SIPp and also
> > > with Grandstream Phones. The call is working fine with out
> > > failure. At the time of call, I have started the wireshark to
> > > capture the packets, but there also I m not getting any negative
> > > reply like 400 or 300.
> > >
> > > See, my call is working fine, call dialog created successfully,
> > > but after that it destroyed, again new dialog is created n that
> > > too destroy. For a single call it creates 2 dialogs. But that
> > > dialog entry is not going to DB. Please suggest me the right thing
> > > to do.
> > >
> > > Thanks a lot for your attention.
> > >
> > >
> > > -Urmi
> > >
> > >
> > >
> > > On Thu, Aug 20, 2009 at 7:38 PM, Brett Nemeroff
> > > mailto:br...@nemeroff.com>
> > >> wrote:
> > >
> > > Urmi,
> > > You log shows the call having failed. I'm not sure why you
> > > think it runs for the proper duration. But as far as OpenSIPs
> > > is concerned, the call failed. It's likely a problem in your
> > > sipp scenario. It's very possible that sipp thinks the call is
> > > up, but the proxy does not.
> > >
> > > In any case, OpenSIPs is behaving as expected, the call fails,
> > > the dialog is destroyed.
> > > Aug 19 17:46:27 [6060] DBG:dialog:next_state_dlg: dialog
> > > 0x2d55af90 changed from state 1 to state 5, due event 1
> > > Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog
> > > 0x2d55af90
> > > failed (negative reply)
> > >
> > > BTW, a negative reply is >=400 (or may also include >= 300,
> > > can't remember). Check your traces, see where that comes from.
> > > -Brett
> > >
> > > On Thu, Aug 20, 2009 at 9:02 AM, urmi lakkad
> > > mailto:urmi.lak...@gmail.com>
> > >>
> wrote:
> > >
> > > Hello Stanisław Pitucha,
> > >
> > > Thank you for support.
> > >
> > > No, My call is established perfectly and is running for
> > > the specified duration without fail.
> > > I m firing the call using SIPp.
> > >
> > > Also, the dialog state gives me 1.
> > >
> > > Thanks for ur attention.
> > >
> > > -Urmi
> > >
> > > 2009/8/20 Stanisław Pitucha  > 
> > > >>
> > >
> > > 2009/8/20 urmi lakkad  > 
> > > >>:
> > > > Am I doing right or not ? If not, please tell me the
> > > correct way.
> > > > One more thing, Is my configuration is correct or not ??
> > >
> > > It looks like your call doesn't even get accepted:
> > > Aug 19 17:46:27 [6060] DBG:dialog:next_state_dlg: dialog
> > > 0x2d55af90 changed from state 1 to state 5, due
> > > event 1
> > > Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog
> > > 0x2d55af90
> > > failed (negative reply)
> > >
> > > Maybe you require authentication, or something else?
> > > Just take care of
> > > the call not failing first. So far it's rejected
> > > before an OK answer
> > > (state 1 is "after sending an INVITE", state 5 is
> > > "deleted" - more or
> > > less).
> > > Capture the traffic and see what's going on.
> >

Re: [OpenSIPS-Users] XCAP scalability: Integrated xcap server VS xcap_client mode

[Iñaki Baz Castillo]

2009/8/26 Anca Vamanu :
> Hi Inaki,
>
> I don't know where you read that the non integrated XCAP server solution
> is better for large networks. I for sure never said that, but in fact
> always said that the integrated_xcap_server way is more efficient
> preferable if the XCAP server can act in this way.

After writing my mail, I searched again for that text (telling that
non integrated mode is better for big networks), but I didn't find
it...
Maybe I've imagined it :)

-- 
Iñaki Baz Castillo


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Re: [OpenSIPS-Users] XCAP scalability: Integrated xcap server VS xcap_client mode

[Anca Vamanu]

Hi Inaki,

I don't know where you read that the non integrated XCAP server solution 
is better for large networks. I for sure never said that, but in fact 
always said that the integrated_xcap_server way is more efficient 
preferable if the XCAP server can act in this way.

Anca

Iñaki Baz Castillo wrote:
> Hi, I've read somewhere (but cannot find it right now) that integrated xcap 
> mode (presence module getting XCAP documents vía SQL) is suitable for small 
> environments.
>
> I know that the other way, using xcap_client module, is not really suitable 
> for now as the HTTP request is blocking (the opensips process gets blocked 
> until the XCAP server replies). But let's imagine this issue is solved.
>
> So we have two options:
>
> 1) Integrated server: OpenSIPS presence module gets the documents from the 
> xcap table.
>
> 2) XCAP client mode: OpenSIPS acts as a xcap client to get the documents from 
> the XCAP server.
>
>
> In the option 1:
> - OpenSIPS does a SQL query to get the document (faster than a HTTP request).
> - So the XCAP server is not queried by the presence server (less work for the 
> XCAP server).
> - There could be various XCAP servers running at the same time (perhaps DNS 
> random or a http proxy between clients and XCAP servers), all of them storing 
> the documents in same DB. And the presence document uses directly that DB.
>
>
> In the option 2:
> - OpenSIPS must perform a HTTP request which takes more time than a SQL 
> request (even if the DB is in other host), right?
> - The XCAP server receives a XCAP request from the presence server, so the 
> XCAP server must "work".
> - The xcap client would contact just an unique XCAP server (it would learn 
> the 
> IP after the first DNS resolution, so ramdom DNS is not valid here).
>   - Solution: Using a HTTP proxy between OpenSIPS and various XCAP servers.
>
>
> Is it really the option 2 more suitable in order to favour scalability and 
> big 
> environments?
>
>
>
>   


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Re: [OpenSIPS-Users] Multiple Area Codes in Customer Area

[Bogdan-Andrei Iancu]

Hi Duane,

You can replace the complicated:

subst_uri('/^sip:([0-9]+)@(.*)$/sip:$avp(s:areacode)\...@\2/i');

with more nicer:

$rU =  $avp(s:areacode)  + $rU;

Regards,
Bogdan


osiris123d wrote:
> Got it working!  Man OpenSIPS sure can do anything with SIP
>
> So here is what I did for future searchers
>
> So the users account is a 7 digit DID number XXX at blah.com
>
> I set up an AVP called areacode for the whole domain blah.com (this assumes
> that the whole domain blah.com is only in one areacode)
>
> opensipsctl avp add -T usr_preferences 0 at blah.com areacode 1 201
> opensipsctl avp add -T usr_preferences 0 at foo.com areacode 1 339
>
>
> In the opensips.cfg file I do the following (it depends on your config as to
> where you want to put this)
> if (uri=~"^sip:[2-9][0-9]{6}@") {
> avp_db_load("$ru/domain","$avp(s:areacode)");
> subst_uri('/^sip:([0-9]+)@(.*)$/sip:$avp(s:areacode)\...@\2/i');
> };
>
> So when someone calls a 7 digit number the avp_db_load() loads the variable
> for areacode and the subst_uri adds the areacode at the beginning of the
> Request-URI.
>
>
>
>
>
>
> Bogdan-Andrei Iancu wrote:
>   
>> Hi Duane,
>>
>> You can correlate AVPs you a USER, a DOAMAIN, etc - it is up to you, 
>> from the script, when loading the AVP - is a pure logical mapping.
>>
>> Regards,
>> Bogdan
>>
>> osiris123d wrote:
>> 
>>> I was reading Flavio's "Building Telephony Systems with OpenSER" chapter
>>> about AVPOPs and he mentions that AVP's can be used for a whole domain. 
>>> I
>>> was thinking that I might be able to configure a area code for Company
>>> A's
>>> domain and then route calls that way.  If not that then I can set the AVP
>>> on
>>> the fly within the transaction by looking at the callers Request URI's
>>> first
>>> 3 digits and route it appropriately.
>>>
>>>
>>> Bogdan-Andrei Iancu wrote:
>>>   
>>>   
 Hi,

 Requirements on the format of  CONTACT and TO headers are nonsense as 
 they are not used for routing at all. Only FROM (which provides info on 
 the caller) and RURI (request URI) (which provide info on callee).

 So, bottom line, only the normalization of the RURI should be required 
 on the system.

 Regards,
 Bogdan

 osiris123d wrote:
 
 
> Thanks for the info.
>
> I will look into this and work up a config.
>
> I also got this direct email about my post from someone else who lives
> in
> the US.  I figured I would go ahead and post below what he sent just so
> its
> out there.
>
>
> Hello Duane --  
>
> You have hit on one of the more difficult areas in SIP and telephony in
> general -- especially here in the North American Numbering Plan.  Below
> I
> will address the problem in general, and not particularly related to
> the
> OpenSIPs question, because IMO you need a solution that will work in
> any
> architecture, not just OpenSIPs.
>
> In a nutshell, I recommend that for your USA users you:
>
> 1.) Require From: and Contact: headers to be in NANPA National (10
> digit)
> format.  This is method is standard in the telephone industry, and will
> allow easy integration with North American ANI or Caller ID format,
> especially when a call may eventually be handed off to the PSTN.   
>
> 2.)  Require incoming To: headers to be in e.164 International format,
> i.e. 
> NANPA-destination numbers all begin with the 1 digit, followed by the
> 10
> digit National number.   Any incoming call to 612xxx goes to
> Sydney,
> Austrailia, and not Minneapolis, MN.  This requirement should be
> enforced
> at
> the perimeter of your network, where Customer Equipment can enforce the
> "local" digit normalization policy.  
>
> 3.)  If you can't enforce #2 above, you will need to "Normalize"
> incoming
> calls to the e.164 International format prior to routing.  The
> unfortunate
> reality here in the USA is that the requirements for how many digits to
> dial
> for a given destination (the "dialing plan") depends on where the call
> comes
> from.   Here in the Chicago area, residents of the 847 area code must
> today
> dial all calls as 11 digits.  Calls within the 312 or 773 area code may
> today be dialed as 7 digits, however beginning on 07 November, all
> calls
> originating in 312 and 773 must be dialed as 1+10 digits, due to the
> new
> 872
> overlay area code.For more information, see
> http://www.nanpa.com/reports/NPA_Dialing_Plans_05_09.pdf
>
> 4.)  Finally, if you have any termination carriers who need special
> "prefixes,"  append them after you have made your route selection.  
>
> If you would like further information or discussion, please feel free
> to
> contact me.
>
> John S. Robi
>
> j...@communx

Re: [OpenSIPS-Users] openxcap installation

[Iñaki Baz Castillo]

2009/8/26 Prashant Prabhu :

> The following packages have unmet dependencies:
>   openxcap: Depends: python-support (>= 0.90.0) but 0.8.7ubuntu4 is to be
> installed
>     Depends: python-application (>= 1.1.5) but it is not going to be
> installed
>     Depends: python-gnutls (>= 1.1.8) but it is not going to be
> installed
> E: Broken packages
>
> ---
> I have added these lines to /etc/apt/sources.list
>
> deb    http://ag-projects.com/debian unstable main
> deb-src http://ag-projects.com/debian unstable main
>
> Where do i get python-support(0.90.0) and how to install it.
> I can see that this version is not available for Ubuntu 9.04 release.
>
> Thanks in advance.

Have you update the deb packages database?:
  sudo aptitude update

python-application version available in AG repositories is 1.1.5:
  http://ag-projects.com/debian/pool/main/p/python-application/

-- 
Iñaki Baz Castillo


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[OpenSIPS-Users] openxcap installation

[Prashant Prabhu]

I am stuck with a problem of installing openxcap on Ubuntu 9.04

I am trying to install openxcap as -
---
#apt-get install openxcap
Reading package lists... Done
Building dependency tree   
Reading state information... Done
Some packages could not be installed. This may mean that you have
requested an impossible situation or if you are using the unstable
distribution that some required packages have not yet been created
or been moved out of Incoming.
The following information may help to resolve the situation:

The following packages have unmet dependencies:
  openxcap: Depends: python-support (>= 0.90.0) but 0.8.7ubuntu4 is to be 
installed
Depends: python-application (>= 1.1.5) but it is not going to be 
installed
Depends: python-gnutls (>= 1.1.8) but it is not going to be 
installed
E: Broken packages

---
I have added these lines to /etc/apt/sources.list

debhttp://ag-projects.com/debian unstable main
deb-src http://ag-projects.com/debian unstable main

Where do i get python-support(0.90.0) and how to install it.
I can see that this version is not available for Ubuntu 9.04 release.

Thanks in advance.

Regards,
Prashant



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Re: [OpenSIPS-Users] how to uninstall

[Tseveendorj Ochirlantuu]

SOLVED.

I removed some folders and files

 /usr/local/etc/opensips
 /usr/local/sbin/opensips*
 /usr/local/lib/opensips

and also remove source code.

rebuilding it from new source code.

Thank you guys.


On Wed, Aug 26, 2009 at 3:51 PM, Tseveendorj Ochirlantuu <
tseveend...@gmail.com> wrote:

> Yes. You can see find result.
>
> eb...@beastie:/var/log$ sudo find / -name opensips
> /var/lib/mysql/opensips
> /usr/src/opensips-1.5.2-notls/scripts/dbtext/opensips
> /usr/src/opensips-1.5.2-notls/scripts/db_berkeley/opensips
> /usr/src/opensips-1.5.2-notls/opensips
>
>
>
> On Wed, Aug 26, 2009 at 2:56 PM, Saúl Ibarra  wrote:
>
>> Have you removed everything OpenSIPS related under /usr/local/tec,
>> /usr/local/bin and /usr/local/lib ?
>>
>>
>>
>> --
>> /Saúl
>> http://www.saghul.net | http://www.sipdoc.net
>>
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>>
>
>
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Re: [OpenSIPS-Users] how to uninstall

[Tseveendorj Ochirlantuu]

Yes. You can see find result.

eb...@beastie:/var/log$ sudo find / -name opensips
/var/lib/mysql/opensips
/usr/src/opensips-1.5.2-notls/scripts/dbtext/opensips
/usr/src/opensips-1.5.2-notls/scripts/db_berkeley/opensips
/usr/src/opensips-1.5.2-notls/opensips


On Wed, Aug 26, 2009 at 2:56 PM, Saúl Ibarra  wrote:

> Have you removed everything OpenSIPS related under /usr/local/tec,
> /usr/local/bin and /usr/local/lib ?
>
>
>
> --
> /Saúl
> http://www.saghul.net | http://www.sipdoc.net
>
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[OpenSIPS-Users] 1.5.2 OpenSIPS and postgresql

[adolphus]

Hi,

I've been wrestling with postgres and OpenSIPs and have been getting the
below error when I launch the binary that i build.  I sense that many of you
folks have figured this out, but replys to querys seem a little scant to me
on how the workaround is actually done.  

I have db_mysql.so comment out (not used) in my opensips.cfg file and I have
db_postgres.so commented in (so it is used) but I still get the below error
when I supply:

Does anyone have (or can recall) the series of steps the performed to get
their systems working with postgresql?  (Any critical steps you used to get
around the error might be helpful.)  I'm using Fedora 6 by the way.  

My command line:
opensips -f ./etc/opensips.cfg  (where my changes are in this file
cfg file)

my error:
ERROR:core:db_check_api: module db_mysql does not export db_use_table 


One standalone question is is db_mysql.so required whether you use
db_postgres.so or not?

Many thanks in advance

-Adolphus
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