Re: [OpenSIPS-Users] timeout between the invite and the sip message after the "180 trying"?

[Russell Treleaven]

ok

On Wed, Apr 25, 2018 at 9:43 AM, Abdoul Osséni <abdoul.oss...@gmail.com>
wrote:

> Yes
>
> between 100 trying or 180 ringing?
> or
> between 100 trying or 183 sdp progress?
>
>
> Abdoul
>
> 2018-04-25 15:10 GMT+02:00 Russell Treleaven <rtrelea...@bunnykick.ca>:
>
>> Do you mean 100 trying or 180 ringing?
>>
>> On Wed, Apr 25, 2018, 9:03 AM Abdoul Osséni <abdoul.oss...@gmail.com>
>> wrote:
>>
>>> Hello list,
>>>
>>> Is it possible to set the timeout between the invite and the sip message
>>> after the "180 trying"?
>>>
>>> Best regards
>>> Abdoul.
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Sincerely,

Russell Treleaven
sip:rtrelea...@sip.bunnykick.ca
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] timeout between the invite and the sip message after the "180 trying"?

[Russell Treleaven]

Do you mean 100 trying or 180 ringing?

On Wed, Apr 25, 2018, 9:03 AM Abdoul Osséni  wrote:

> Hello list,
>
> Is it possible to set the timeout between the invite and the sip message
> after the "180 trying"?
>
> Best regards
> Abdoul.
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Remote ip change in-dialog

[Russell Treleaven]

What mobile sip client supports this?


On Mon, Oct 10, 2016 at 6:50 AM, Saioa Perurena  wrote:

> Hi,
>
> I've the following problem, any advice will be welcome.
>
> A calls B, A changes his ip because of network change (3G to wifi for
> example) sends again an INVITE and the call continous ok with the same
> callid, did, tags... but if B sends and INVITE (to enable video for
> example), opensips validate_dialog returns error because the ip has
> changed.
>
> I've tried fix_route_dialog() but it doesn't work.
>
> Thanks in advance!
>
> Saioa.
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] How to overcome SIP ALG on Wi-Fi routers

[Russell Treleaven]

Look for the fragmentation flag.
On May 17, 2016 1:08 PM, "Nabeel"  wrote:

> In that case, the answer to your question seems to be that the UDP packets
> did not reach the OpenSIPS server, because nothing was added to the
> OpenSIPS logs using debug level 4. All of this seems to point to the cause
> being UDP packet fragmentation. Is this correct?
> On 17 May 2016 4:24 pm, "Bogdan-Andrei Iancu"  wrote:
>
>> The TCP/IP stack of your server may decide to drop an UDP packet if it
>> cannot re-assemble it correctly (like not all the IP fragments were
>> received).
>> In such a case, you see the IP packets (carrying the fragments) on
>> network level, but they are never delivered at application level.
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 17.05.2016 16:05, Nabeel wrote:
>>
>> The next question - is this INVITE reaching your opensips script ? to be
>>> sure that the OS delivers the UDP packet to the opensips application.
>>
>>
>> I don't have any firewall on my server. Why would the UDP packet get
>> blocked between entering the server and reaching opensips script? The
>> opensips server is running without errors. Other calls work fine.
>>
>>
>>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] How to overcome SIP ALG on Wi-Fi routers

[Russell Treleaven]

TCP works for me.
On May 2, 2016 8:43 PM, "Nabeel"  wrote:

> Thanks for the suggestions of using TLS or changing the port. I changed
> the port, but some routers are still able to mess with the SIP headers. I
> would have used TLS, if not for two reasons:
>
> 1. ICE protocol was originally designed for UDP according to RFC5245, and
> it seems to work better with UDP.
>
> 2. The SIP servers I have used (OpenSIPS and Repro) seem to be more stable
> with UDP compared to TLS (they do not randomly drop connections, throw
> unusual errors in the logs, etc.)
>
> I may try TLS again, but it would be better if there is an alternative
> workaround for UDP.
>
> On 2 May 2016 at 13:33, Patrick Wakano  wrote:
>
>> Using TLS!
>> Also configuring your systems/devices to use other port than 5060 may do
>> the trick...
>>
>> On Mon, May 2, 2016 at 9:14 AM, Nabeel  wrote:
>>
>>> Hi,
>>>
>>> Other than using rtpproxy/NAThelper modules, is there any way to
>>> bypass/workaround SIP ALG enabled on many WiFi routers? Although SIP ALG
>>> was designed to help with NAT, in most cases it does the opposite and
>>> breaks SIP.
>>>
>>> Nabeel
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] [BOOK] Building Telephony Systems with OpenSIPS - 2.1 version

[Russell Treleaven]

Never mind I see that it is now available from Oreilly.

On Tue, Feb 9, 2016 at 10:33 AM, Russell Treleaven <rtrelea...@bunnykick.ca>
wrote:

> Hi,
>
> Will this book become available through O’Reilly Media?
> I try to buy from them if possible.
>
> Sincerely,
>
> Russell Treleaven
>
> On Thu, Feb 4, 2016 at 9:37 AM, Bogdan-Andrei Iancu <bog...@opensips.org>
> wrote:
>
>> Hello all,
>>
>> Flavio Goncalves and I are happy to announce the publishing of the second
>> edition of "Building Telephony Systems with OpenSIPS", covering OpenSIPS
>> version 2.1 .
>>
>> Also many thanks to the Packt Publishing house for making it happened and
>> to all our reviewers who help us to make this book better.
>>
>>
>> https://www.packtpub.com/networking-and-servers/building-telephony-systems-opensips-second-edition
>>
>> Enjoy !
>>
>> --
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developer
>> http://www.opensips-solutions.com
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Secretary/Boss

[Russell Treleaven]

I will let others fill in the details but I think this is what you want to
achieve on a high level.

A user calls the Boss's number which is actually routed to the assistant's
phone.
Assistant answers call and transfers the call to the Boss's "real"
extension https://en.wikipedia.org/wiki/Call_transfer.

Does that sound like what you want?





On Mon, Nov 2, 2015 at 7:14 AM, Michele Pinassi 
wrote:

> Hi all,
>
> in my context, i have OpenSIPS as a SIP router and Asterisk as a media
> box for IVR, Voicemail etc etc etc...
>
> I need to implement Secretary/Boss function, like this:
>
> (USER) call--///--> (Boss phone) --- call diverted to secretary--->
> (Secretary) - divert > (Boss)
>
> User cannot call directly to Boss phone: firstly, the call will be
> diverted to secretary and only in a second time, call should be
> forwarded to boss phone (from secretary).
>
> Any hint how to do this ?
>
> Michele
>
> --
> Michele Pinassi
> Responsabile Telefonia di Ateneo
> Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di
> Siena
> tel: 0577.(23)5000 - central...@unisi.it
>
> Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ
> di Ateneo, http://www.faq.unisi.it
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] SIP and RTP Proxy without local user base

[Russell Treleaven]

As an alternative you could try bypass media.
https://freeswitch.org/confluence/display/FREESWITCH/Bypass+Media+Overview


On Fri, Apr 24, 2015 at 11:17 AM, Roman Dissauer ro...@dissauer.net wrote:

 Dear all,

 I’m running a centralized Freeswitch based PBX for use on several sites.
 All phones register against this Freeswitch instance.

 Now I want to install SIP and RTP Proxies on every site to keep RTP
 traffic locally on site. Registration should still be done by freeswitch.
 Can anybody give me a hint if this is possible with opensips and rtpproxy?

 Maybe I can clarify it a bit more:
 
 |  Freeswitch  |
 
|
|Public Internet
|
|- Phone 3 external
|
|
 --
 | Firewall / NAT |
 --
|
 
 |Proxy |   Site 1
 
|
|
|- Phone 1 internal
|- Phone 2 internal

 Phone 1 - 3 are all registered at Freeswitch
 Phone 1 calls Phone 3: SIP and RTP over Freeswitch
 Phone 1 calls Phone 2: SIP over Freeswitch but RTP over Proxy

 Does this make sense?

 Thanks,
 Roman
 ___
 Users mailing list
 Users@lists.opensips.org
 http://lists.opensips.org/cgi-bin/mailman/listinfo/users

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users