Re: [OpenSIPS-Users] Change 200 OK of BYE message

[Johan De Clercq]

Any chance to have a small capture ?
I tend to agree with Robert, if you use rr consistently there should be no
problem. It can of course also be that the client does something strange
...

2017-02-06 21:43 GMT+01:00 Daniel Zanutti :

> Hi Robert
>
> Yes, all messages are passing through the proxy, but when I receive the
> 200 OK of the BYE message, it doesn't go to Main Route or Reply Route. It
> just go to the destination and I cannot change anything on it.
>
> Any idea?
>
>
>
> On Mon, Feb 6, 2017 at 6:14 PM, Mundkowsky, Robert 
> wrote:
>
>> Did you use “record_route”?
>>
>>
>>
>> For reference:
>>
>> http://www.iptel.org/sip/intro/scenarios/rr
>>
>>
>>
>>
>>
>> Robert Mundkowsky
>>
>>
>>
>> *From:* Users [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Daniel
>> Zanutti
>> *Sent:* Monday, February 6, 2017 3:01 PM
>> *To:* OpenSIPS users mailling list 
>> *Subject:* [OpenSIPS-Users] Change 200 OK of BYE message
>>
>>
>>
>> Hi
>>
>>
>>
>> I need to change something on the 200 OK of BYE message. Tried everything
>> on Opensips but looks like this message doesn't follow standard message
>> path. Neither Main Route or Reply route pass this message.
>>
>>
>>
>> Is there any way to do it?
>>
>>
>>
>> Thanks
>>
>> --
>>
>> This e-mail and any files transmitted with it may contain privileged or
>> confidential information. It is solely for use by the individual for whom
>> it is intended, even if addressed incorrectly. If you received this e-mail
>> in error, please notify the sender; do not disclose, copy, distribute, or
>> take any action in reliance on the contents of this information; and delete
>> it from your system. Any other use of this e-mail is prohibited.
>>
>> Thank you for your compliance.
>> --
>>
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
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Re: [OpenSIPS-Users] rtpproxy and SRTP

[Sasmita Panda]

Hi ,

I have tested 1 scenario . If there is two end point which support
encrypted media (SRTP) and there is rtpproxy between them . Then rtpproxy
works as usual . It update the C line Ip in the SDP and forwards the
request and response .  This is what I tested . And its working .

   I don't know which prospective you are asking rtpproxy supports SRTP
or not . If you mean anything different then I am also curious to know .


*Thanks & Regards*
*Sasmita Panda*
*Network Testing and Software Engineer*
*3CLogic , ph:07827611765*

On Mon, Feb 6, 2017 at 7:13 PM, John Quick 
wrote:

> Please, does anyone know if rtpproxy works with SRTP?
>
> John Quick
> Smartvox Limited
>
>
>
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[OpenSIPS-Users] ds_select_dst with algorithm 6 is not using the last destination

[Pat Burke]

Hello,


It appears that 2.2.2 ds_select_dst with the parameters that I have below is 
not using the last destination in the set.  


...
if (!ds_select_dst("$avp(core_server_group)","6","FuSd")) {
...


Here is the results from ds_list full.


(with 2 destinations in the set)
PARTITION:: default
SET:: 10005
URI:: sip:216.147.191.143:5060 state=Active first_hit_counter=16
weight:: 1
priority:: 0
description:: DEV Ingress-DEV Core
URI:: sip:216.147.191.144:5060 state=Active first_hit_counter=0
weight:: 1
priority:: 0
description:: DEV Ingress-DEV Core
SET:: 10002




(with 3 destinations in the set)
PARTITION:: default
SET:: 10005
URI:: sip:216.147.191.143:5060 state=Active first_hit_counter=13
weight:: 1
priority:: 0
description:: DEV Ingress-DEV Core
URI:: sip:216.147.191.144:5060 state=Active first_hit_counter=15
weight:: 1
priority:: 0
description:: DEV Ingress-DEV Core
URI:: sip:216.147.191.142:5060 state=Active first_hit_counter=0
weight:: 1
priority:: 0
description:: DEV Ingress-DEV Core
SET:: 10002








What I am after is the functionality that I had in 1.11 which was a random use 
of all destinations in the set.  I had the following setting




modparam("dispatcher", "force_dst", 1)
modparam("dispatcher", "flags", 2)


and called using


...
    if (!ds_select_dst("$avp(core_server_group)","6")) {
...


Regards,
Pat Burke



__
Direct: (402) 403-5121   |   Cell: (402) 443-8929  |   Email: 
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Re: [OpenSIPS-Users] siptrace picks incorrect source socket

[Jeff Pyle]

Hi Bogdan,

Now it won't start.  I see the following errors on config check:

Feb  6 21:21:03 [30051] ERROR:siptrace:parse_siptrace_uri: Invalid key
 in trace id!
Feb  6 21:21:03 [30051] ERROR:siptrace:parse_siptrace_id: invalid uri
<3;transport=udp;>
Feb  6 21:21:03 [30051] ERROR:siptrace:parse_trace_id: failed to parse
siptrace uri [3;transport=udp;]
Feb  6 21:21:03 [30051] CRITICAL:core:yyerror: parse error in config file
/usr/local//etc/opensips/opensips.cfg, line 225, column 20-21: Parameter
 not found in module  - can't set
Feb  6 21:21:03 [30051] ERROR:core:main: bad config file (1 errors)
Feb  6 21:21:03 [30051] NOTICE:core:main: Exiting


The script has:

   223 loadmodule "siptrace.so"
   224 modparam("siptrace", "trace_on", 1)
   225 modparam("siptrace", "trace_id", "[tid]uri=hep:127.0.0.1:9060
;version=3;transport=udp;")


This is on 2.3/devel git revision 2bcf891 from around 01:00 UTC Feb 07.



- Jeff


On Sun, Feb 5, 2017 at 11:00 AM, Bogdan-Andrei Iancu 
wrote:

> Hi Jeff,
>
> Thank you for detailed report. I was able to reproduce and fix it. Please
> see:
> https://github.com/OpenSIPS/opensips/commit/
> b30af734cdb84991e1f906e3920a94e87c33ea04
>
> If you confirm the fix, I will do the backporting to 2.2 too.
>
> Thanks and Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 02/04/2017 04:41 AM, Jeff Pyle wrote:
>
> Hello,
>
> I recently enabled siptrace on an OpenSIPS 2.2.2 system acting as a
> registrar and a proxy.  It has one IPv4 address with several ports for UDP,
> TCP and TLS.  In a case where the INVITE is relayed from TLS to UDP, the
> replies to the UAC are incorrectly being reported as coming from the UDP
> socket.
>
> In the below scenario the UAC is at 64.65.66.67, and its ephemeral TCP
> client port for this call is 61235.  The OpenSIPS proxy is at
> 131.132.133.134 listening on UDP 5060 and TLS 5061.  Also on
> 131.132.133.134 there is a Freeswitch media server listening on UDP 5080.
> The UAC sends an INVITE to the proxy over TLS which routes it to the media
> server over UDP.  The replies relayed to the UAC are reported as having
> come from port 5060 over UDP when in reality they have come from port 5061
> over TCP (TLS).
>
> My config:
>
> listen=udp:131.132.133.134:5060
> listen=tls:131.132.133.134:5061
> listen=hep_udp:127.0.0.1:9030
>
> loadmodule "siptrace.so"
> modparam("siptrace", "trace_on", 1)
> modparam("siptrace", "trace_id", "[hep]uri=hep:127.0.0.1:9060;
> transport=udp;")
>
>
>
> Debugs:
>
>
> INVITE in from UAC over TLS
> Feb  3 21:20:22 testproxy /usr/sbin/opensips[24673]:
> DBG:siptrace:pipport2su: proto 22, host 64.65.66.67 , port 61235
> Feb  3 21:20:22 testproxy /usr/sbin/opensips[24673]:
> DBG:siptrace:pipport2su: proto 22, host 131.132.133.134 , port 5061
>
> INVITE out to media server over UDP
> Feb  3 21:20:22 testproxy /usr/sbin/opensips[24673]:
> DBG:siptrace:pipport2su: proto 17, host 131.132.133.134 , port 5060
> Feb  3 21:20:22 testproxy /usr/sbin/opensips[24673]:
> DBG:siptrace:pipport2su: proto 17, host 131.132.133.134 , port 5080
>
> 100 Trying in from media server over UDP
> Feb  3 21:20:22 testproxy /usr/sbin/opensips[24650]:
> DBG:siptrace:pipport2su: proto 17, host 131.132.133.134 , port 5080
> Feb  3 21:20:22 testproxy /usr/sbin/opensips[24650]:
> DBG:siptrace:pipport2su: proto 17, host 131.132.133.134 , port 5060
>
> 180 Ringing in from media server over UDP
> Feb  3 21:20:22 testproxy /usr/sbin/opensips[24651]:
> DBG:siptrace:pipport2su: proto 17, host 131.132.133.134 , port 5080
> Feb  3 21:20:22 testproxy /usr/sbin/opensips[24651]:
> DBG:siptrace:pipport2su: proto 17, host 131.132.133.134 , port 5060
>
> 180 Ringing out to UAC over TLS (even though it reports UDP)
> Feb  3 21:20:22 testproxy /usr/sbin/opensips[24651]:
> DBG:siptrace:pipport2su: proto 17, host 131.132.133.134 , port 5060
> Feb  3 21:20:22 testproxy /usr/sbin/opensips[24651]:
> DBG:siptrace:pipport2su: proto 22, host 64.65.66.67 , port 61235
>
> 200 OK in from media server over UDP
> Feb  3 21:20:22 testproxy /usr/sbin/opensips[24651]:
> DBG:siptrace:pipport2su: proto 17, host 131.132.133.134 , port 5080
> Feb  3 21:20:22 testproxy /usr/sbin/opensips[24651]:
> DBG:siptrace:pipport2su: proto 17, host 131.132.133.134 , port 5060
>
> 200 OK out to UAC over TLS (even though it reports udp)
> Feb  3 21:20:22 testproxy /usr/sbin/opensips[24651]:
> DBG:siptrace:pipport2su: proto 17, host 131.132.133.134 , port 5060
> Feb  3 21:20:22 testproxy /usr/sbin/opensips[24651]:
> DBG:siptrace:pipport2su: proto 22, host 64.65.66.67 , port 61235
>
> ACK in from UAC over TLS
> Feb  3 21:20:23 testproxy /usr/sbin/opensips[24673]:
> DBG:siptrace:pipport2su: proto 22, host 64.65.66.67 , port 61235
> Feb  3 21:20:23 testproxy /usr/sbin/opensips[24673]:
> DBG:siptrace:pipport2su: proto 22, host 131.132.133.134 , port 5061
>
> ACK out to media server over UDP
> Feb  3 21:20:23 testproxy 

Re: [OpenSIPS-Users] Change 200 OK of BYE message

[Mundkowsky, Robert]

I am really new to OpenSIPS and SIPS routing, so your question is beyond me 
now. ☹

I am guessing your SIP client is a miss behaving client.  If OpenSIPS is acting 
as a proxy, I think there might be a way to hide the backend topology, so that 
client has no way to know how to send messages without going thru OpenSIPS.


Robert Mundkowsky

From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Daniel 
Zanutti
Sent: Monday, February 6, 2017 3:44 PM
To: OpenSIPS users mailling list 
Subject: Re: [OpenSIPS-Users] Change 200 OK of BYE message

Hi Robert

Yes, all messages are passing through the proxy, but when I receive the 200 OK 
of the BYE message, it doesn't go to Main Route or Reply Route. It just go to 
the destination and I cannot change anything on it.

Any idea?



On Mon, Feb 6, 2017 at 6:14 PM, Mundkowsky, Robert 
> wrote:
Did you use “record_route”?

For reference:
http://www.iptel.org/sip/intro/scenarios/rr


Robert Mundkowsky

From: Users 
[mailto:users-boun...@lists.opensips.org]
 On Behalf Of Daniel Zanutti
Sent: Monday, February 6, 2017 3:01 PM
To: OpenSIPS users mailling list 
>
Subject: [OpenSIPS-Users] Change 200 OK of BYE message

Hi

I need to change something on the 200 OK of BYE message. Tried everything on 
Opensips but looks like this message doesn't follow standard message path. 
Neither Main Route or Reply route pass this message.

Is there any way to do it?

Thanks



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Re: [OpenSIPS-Users] Change 200 OK of BYE message

[Daniel Zanutti]

Hi Robert

Yes, all messages are passing through the proxy, but when I receive the 200
OK of the BYE message, it doesn't go to Main Route or Reply Route. It just
go to the destination and I cannot change anything on it.

Any idea?



On Mon, Feb 6, 2017 at 6:14 PM, Mundkowsky, Robert 
wrote:

> Did you use “record_route”?
>
>
>
> For reference:
>
> http://www.iptel.org/sip/intro/scenarios/rr
>
>
>
>
>
> Robert Mundkowsky
>
>
>
> *From:* Users [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Daniel
> Zanutti
> *Sent:* Monday, February 6, 2017 3:01 PM
> *To:* OpenSIPS users mailling list 
> *Subject:* [OpenSIPS-Users] Change 200 OK of BYE message
>
>
>
> Hi
>
>
>
> I need to change something on the 200 OK of BYE message. Tried everything
> on Opensips but looks like this message doesn't follow standard message
> path. Neither Main Route or Reply route pass this message.
>
>
>
> Is there any way to do it?
>
>
>
> Thanks
>
> --
>
> This e-mail and any files transmitted with it may contain privileged or
> confidential information. It is solely for use by the individual for whom
> it is intended, even if addressed incorrectly. If you received this e-mail
> in error, please notify the sender; do not disclose, copy, distribute, or
> take any action in reliance on the contents of this information; and delete
> it from your system. Any other use of this e-mail is prohibited.
>
> Thank you for your compliance.
> --
>
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Re: [OpenSIPS-Users] Change 200 OK of BYE message

[Mundkowsky, Robert]

Did you use “record_route”?

For reference:
http://www.iptel.org/sip/intro/scenarios/rr


Robert Mundkowsky

From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Daniel 
Zanutti
Sent: Monday, February 6, 2017 3:01 PM
To: OpenSIPS users mailling list 
Subject: [OpenSIPS-Users] Change 200 OK of BYE message

Hi

I need to change something on the 200 OK of BYE message. Tried everything on 
Opensips but looks like this message doesn't follow standard message path. 
Neither Main Route or Reply route pass this message.

Is there any way to do it?

Thanks



This e-mail and any files transmitted with it may contain privileged or 
confidential information. It is solely for use by the individual for whom it is 
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in reliance on the contents of this information; and delete it from your 
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[OpenSIPS-Users] Change 200 OK of BYE message

[Daniel Zanutti]

Hi

I need to change something on the 200 OK of BYE message. Tried everything
on Opensips but looks like this message doesn't follow standard message
path. Neither Main Route or Reply route pass this message.

Is there any way to do it?

Thanks
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Re: [OpenSIPS-Users] SIP message relay order

[Stas Kobzar]

Hello Bogdan,

In my case, ACK for previous INVITE has already been received by OpenSIPS,
but not sent yet.
In this case, will the variable $DLG_status still equals 3 ?

Thanks

On Sun, Feb 5, 2017 at 11:15 AM, Bogdan-Andrei Iancu 
wrote:

> Hi Stas,
>
> Such races may happen at application level or even at network level (when
> using UDP) - so if you have 2 packets very close as time, they may swap.
> That is SIP :)
>
> The full guilt is in the UAC device, IMHO - it should let some time gap
> between the ACK and re-INVITE, to eliminate any possible races.
>
> Now, what you can do is to use the dialog module and to check the dialog
> state when receiving the re-invite. If $DLG_status is *3* (Confirmed by a
> final reply but no ACK received yet), drop with no reply the re-INVITEs (to
> force a later retransmission) :
> http://www.opensips.org/html/docs/modules/2.2.x/dialog.html#id297400
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 02/02/2017 10:31 PM, Stas Kobzar wrote:
>
> Hello List,
>
> My call flow has initial INVITE and re-INVITE to update RTP IP/port.
> Usually everything works well, but sometimes OpenSIPS come up with
> following example:
>
> UA OpenSIPS  PSTN GW
> ---
> INV(CSeq: 100) -> | ---> INV(CSeq: 100)
> < 200 OK  | <--- 200 OK
>
> (UA sends ACK then new INVITE)
>
> ACK(CSeq: 100) -> |
> reINV(Cseq: 101) ---> |
>
> (OpenSIPS relays first INVITE then ACK)
>   | ---> reINV(CSeq: 101)
>   | --->   ACK(CSeq: 100)
>
> When PSTN gateway receives re-INVITE before ACK for previous INVITE
> it responds 500 with Retry-After header.
> This is correct behaviour which conforms to the RFC 3261 section 14.2
>
> My question is:
> Is it possible to assure order of received and relayed messages within the
> same SIP session? Is there any configuration parameter?
>
> Thank you,
> --
>
> Stas Kobzar
>
> Developeur VoIP / VoIP Developer
>
>
> Modulis­.ca Inc.
>
> # Bureau / Office: 514-284-2020 x 246 <(514)%20284-2020>
>
> Email: s tas.kob...@modulis.ca
>
> https://www.modulis.com
>
>
> ___
> Users mailing 
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>


-- 

Stas Kobzar

Developeur VoIP / VoIP Developer


Modulis­.ca Inc.

# Bureau / Office: 514-284-2020 x 246

Email: s tas.kob...@modulis.ca

https://www.modulis.com
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Re: [OpenSIPS-Users] SipTrace Module - failed to get an id for "sip" tracing!

[Ionut Ionita]

Hello Victor,

It should be fixed now. Try upgrading to latest from master

Ionut Ionita
OpenSIPS Developer

On 02/05/2017 02:46 PM, Victor Bogatyryev wrote:
Hi All, opensips 2.3 opensips.cfg  SipTrace Module loadmodule 
"siptrace.so" modparam("siptrace", "trace_on", 1) modparam("siptrace", 
"trace_id", 
"[tid]uri=mysql://opensips:opensipsrw@localhost/opensips;table=sip_trace;") 
MariaDB [opensips]> DESCRIBE sip_trace; 
+-+--+--+-+-++ 
| Field | Type | Null | Key | Default | Extra | 
+-+--+--+-+-++ 
| id | int(10) unsigned | NO | PRI | NULL | auto_increment | | 
time_stamp | datetime | NO | MUL | 1900-01-01 00:00:01 | | | callid | 
char(255) | NO | MUL | | | | trace_attrs | char(128) | YES | MUL | 
NULL | | | msg | text | NO | | NULL | | | method | char(32) | NO | | | 
| | status | char(128) | YES | | NULL | | | from_proto | char(5) | NO 
| | NULL | | | from_ip | char(50) | NO | MUL | | | | from_port | 
int(5) unsigned | NO | | NULL | | | to_proto | char(5) | NO | | NULL | 
| | to_ip | char(50) | NO | | | | | to_port | int(5) unsigned | NO | | 
NULL | | | fromtag | char(64) | NO | | | | | direction | char(4) | NO 
| | | | 
+-+--+--+-+-++ 
15 rows in set (0.00 sec) # opensipsctl start INFO: Starting OpenSIPS 
: ERROR: PID file /var/run/opensips/opensips.pid does not exist -- 
OpenSIPS start failed Degug: /local/sbin/opensips[2061]: 
ERROR:siptrace:mod_init: failed to get an id for "sip" tracing! 
/sbin/opensips[2061]: ERROR:core:init_mod: failed to initialize module 
siptrace /sbin/opensips[2061]: ERROR:core:main: error while 
initializing modules /sbin/opensips[2061]: INFO:core:cleanup: cleanup 
/local/sbin/opensips[2061]: NOTICE:core:main: Exiting Regards, 
Victor Bogatyryev



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Re: [OpenSIPS-Users] SIP message relay order

[Stas Kobzar]

Hello Bogdan and Razvan,

Thank you for sharing your ideas.

Dropping re-INVITE packet to force retransmission sounds really smart
solution for that issue! Also it is easy to implement.
I will give it a try.

Thanks again!

On Sun, Feb 5, 2017 at 11:15 AM, Bogdan-Andrei Iancu 
wrote:

> Hi Stas,
>
> Such races may happen at application level or even at network level (when
> using UDP) - so if you have 2 packets very close as time, they may swap.
> That is SIP :)
>
> The full guilt is in the UAC device, IMHO - it should let some time gap
> between the ACK and re-INVITE, to eliminate any possible races.
>
> Now, what you can do is to use the dialog module and to check the dialog
> state when receiving the re-invite. If $DLG_status is *3* (Confirmed by a
> final reply but no ACK received yet), drop with no reply the re-INVITEs (to
> force a later retransmission) :
> http://www.opensips.org/html/docs/modules/2.2.x/dialog.html#id297400
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 02/02/2017 10:31 PM, Stas Kobzar wrote:
>
> Hello List,
>
> My call flow has initial INVITE and re-INVITE to update RTP IP/port.
> Usually everything works well, but sometimes OpenSIPS come up with
> following example:
>
> UA OpenSIPS  PSTN GW
> ---
> INV(CSeq: 100) -> | ---> INV(CSeq: 100)
> < 200 OK  | <--- 200 OK
>
> (UA sends ACK then new INVITE)
>
> ACK(CSeq: 100) -> |
> reINV(Cseq: 101) ---> |
>
> (OpenSIPS relays first INVITE then ACK)
>   | ---> reINV(CSeq: 101)
>   | --->   ACK(CSeq: 100)
>
> When PSTN gateway receives re-INVITE before ACK for previous INVITE
> it responds 500 with Retry-After header.
> This is correct behaviour which conforms to the RFC 3261 section 14.2
>
> My question is:
> Is it possible to assure order of received and relayed messages within the
> same SIP session? Is there any configuration parameter?
>
> Thank you,
> --
>
> Stas Kobzar
>
> Developeur VoIP / VoIP Developer
>
>
> Modulis­.ca Inc.
>
> # Bureau / Office: 514-284-2020 x 246 <(514)%20284-2020>
>
> Email: s tas.kob...@modulis.ca
>
> https://www.modulis.com
>
>
> ___
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> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>


-- 

Stas Kobzar

Developeur VoIP / VoIP Developer


Modulis­.ca Inc.

# Bureau / Office: 514-284-2020 x 246

Email: s tas.kob...@modulis.ca

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[OpenSIPS-Users] rtpproxy and SRTP

[John Quick]

Please, does anyone know if rtpproxy works with SRTP?

John Quick
Smartvox Limited



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Re: [OpenSIPS-Users] iOS 10 Push Notifications

[Андрей Журавлёв]

Thank you very much, Adrian. So, this feature will be available from 2.3,
right? Is it possble to check it may be in beta in development 2.3 branch?
And are there any sample configurations for push notification scenario for
current version? May be you could share some, please?

I am pretty new to SIP and OpenSIPS :)

2017-02-06 16:39 GMT+04:00 Liviu Chircu :

>
> Adrian, this is exactly what I need. Is there a way to do it right now? I
> am on version 2.2
>
> I actually gave a talk about this during the RTC devroom at FOSDEM a
> couple days ago. What you need can be found on slide 16 [1]
>
> PS: note that there are some limitations, especially concerning
> successively forking branches. If you need this feature, it will be
> available starting with 2.3
>
> [1]: https://fosdem.org/2017/schedule/event/opensips/
>
> Liviu Chircu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> On 5 Feb 2017, at 12:53, Aron Podrigal  wrote:
>
> you can certainly handle this in a t_on_failure for a 408...
>
> On Thu, Feb 2, 2017, 3:38 PM Adrian Georgescu  wrote:
>
> There is more plumbing needed to replay transactions later and the future
> OpenSIPS version announced recently will help make this happen.
>
> You are not alone fighting with this problem.
>
> Regards,
> Adrian
>
>
> On 1 Feb 2017, at 19:46, Андрей Журавлёв  wrote:
>
> Hi All,
>
> I am pretty new with SIP and OpenSIPS. I have problem that affect most of
> the people who should support mobile clients for iOS 10. There must be a
> way to solve it, but it looks like I missed it. So I need your help.
>
> Background: As you probably already know in iOS 10 Apple prevented network
> connections in background mode. It mostly affected VoIP apps. In order to
> solve this issue they introduced so-called VoIP Push notifications (via
> PushKit) which should automatically wake-up application and allow it to do
> a registration and receive a call.
>
> Btw, the only thing I found capable to send VoIP push notifications
> without issus is ruby gem/binary called Houston.
>
> Now everything works fine, except one issue actually, when application
> wakes up and do a registration, it obviously missed initial INVITE message
> for a call, and it looks like server do not retry invites if no provisional
> messages returned from a client.
>
> So the question is it possible to tell OpenSIPS server to re-send INVITE
> messages (by some timer probably) if no provisional information received
> from a client.
>
> I know there is fr_timer and fr_inv_timer params from tm module, but it
> looks like they did not do the trick.
>
> Actually, I have almost default config file, except the parts, required
> for push notifications.
> I've posted it here: http://pastebin.com/tZmP320g
>
> Yours sincerely,
> Andrei Zhuravlev
> ___
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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>
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>
>
>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
> --
> С уважением,
> Андрей Журавлев
> Архитектор проекта
> MST Digital Agency
> Email: andrei.zhurav...@m-st.ru
> Skype: andrei.v.zhuravlev
> Моб: +79176187334
> http://m-st.ru
> 
> Yours sincerely,
> Andrei Zhuravlev
> Software Architect
> MST Digital Agency
> Email: andrei.zhurav...@m-st.ru
> Skype: andrei.v.zhuravlev
> Mob: +79176187334
> http://m-st.ru
>
>
>
> ___
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>
>
>
> ___
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> Users@lists.opensips.org
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>
>


-- 
С уважением,
Андрей Журавлев
Архитектор проекта
MST Digital Agency
Email: andrei.zhurav...@m-st.ru
Skype: andrei.v.zhuravlev
Моб: +79176187334
http://m-st.ru

Yours sincerely,
Andrei Zhuravlev
Software Architect
MST Digital Agency
Email: andrei.zhurav...@m-st.ru
Skype: andrei.v.zhuravlev
Mob: +79176187334
http://m-st.ru
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Re: [OpenSIPS-Users] iOS 10 Push Notifications

[Liviu Chircu]

Adrian, this is exactly what I need. Is there a way to do it right 
now? I am on version 2.2


I actually gave a talk about this during the RTC devroom at FOSDEM a 
couple days ago. What you need can be found on slide 16 [1]


PS: note that there are some limitations, especially concerning 
successively forking branches. If you need this feature, it will be 
available starting with 2.3


[1]: https://fosdem.org/2017/schedule/event/opensips/

Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

On 5 Feb 2017, at 12:53, Aron Podrigal > wrote:


you can certainly handle this in a t_on_failure for a 408...

On Thu, Feb 2, 2017, 3:38 PM Adrian Georgescu > wrote:


There is more plumbing needed to replay transactions later and
the future OpenSIPS version announced recently will help make
this happen.

You are not alone fighting with this problem.

Regards,
Adrian



On 1 Feb 2017, at 19:46, Андрей Журавлёв
> wrote:

Hi All,

I am pretty new with SIP and OpenSIPS. I have problem that
affect most of the people who should support mobile clients for
iOS 10. There must be a way to solve it, but it looks like I
missed it. So I need your help.

Background: As you probably already know in iOS 10 Apple
prevented network connections in background mode. It mostly
affected VoIP apps. In order to solve this issue they introduced
so-called VoIP Push notifications (via PushKit) which should
automatically wake-up application and allow it to do a
registration and receive a call.

Btw, the only thing I found capable to send VoIP push
notifications without issus is ruby gem/binary called Houston.

Now everything works fine, except one issue actually, when
application wakes up and do a registration, it obviously missed
initial INVITE message for a call, and it looks like server do
not retry invites if no provisional messages returned from a
client.

So the question is it possible to tell OpenSIPS server to
re-send INVITE messages (by some timer probably) if no
provisional information received from a client.

I know there is fr_timer and fr_inv_timer params from tm module,
but it looks like they did not do the trick.

Actually, I have almost default config file, except the parts,
required for push notifications.
I've posted it here: http://pastebin.com/tZmP320g

Yours sincerely,
Andrei Zhuravlev
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--
С уважением,
Андрей Журавлев
Архитектор проекта
MST Digital Agency
Email: andrei.zhurav...@m-st.ru 
Skype: andrei.v.zhuravlev
Моб: +79176187334
http://m-st.ru 

Yours sincerely,
Andrei Zhuravlev
Software Architect
MST Digital Agency
Email: andrei.zhurav...@m-st.ru 
Skype: andrei.v.zhuravlev
Mob: +79176187334
http://m-st.ru 



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Re: [OpenSIPS-Users] iOS 10 Push Notifications

[Андрей Журавлёв]

Adrian, this is exactly what I need. Is there a way to do it right now? I
am on version 2.2

2017-02-06 15:37 GMT+04:00 Adrian Georgescu :

> When the Registration expires or it is closed when the device goes to
> background, there is no 408.
>
> The idea is to attempt to wake up the device when call comes in, wait for
> it to register and then fork the original INVITE. Then timeout if no device
> registers.
>
> Adrian
>
>
> On 5 Feb 2017, at 12:53, Aron Podrigal  wrote:
>
> you can certainly handle this in a t_on_failure for a 408...
>
> On Thu, Feb 2, 2017, 3:38 PM Adrian Georgescu  wrote:
>
> There is more plumbing needed to replay transactions later and the future
> OpenSIPS version announced recently will help make this happen.
>
> You are not alone fighting with this problem.
>
> Regards,
> Adrian
>
>
> On 1 Feb 2017, at 19:46, Андрей Журавлёв  wrote:
>
> Hi All,
>
> I am pretty new with SIP and OpenSIPS. I have problem that affect most of
> the people who should support mobile clients for iOS 10. There must be a
> way to solve it, but it looks like I missed it. So I need your help.
>
> Background: As you probably already know in iOS 10 Apple prevented network
> connections in background mode. It mostly affected VoIP apps. In order to
> solve this issue they introduced so-called VoIP Push notifications (via
> PushKit) which should automatically wake-up application and allow it to do
> a registration and receive a call.
>
> Btw, the only thing I found capable to send VoIP push notifications
> without issus is ruby gem/binary called Houston.
>
> Now everything works fine, except one issue actually, when application
> wakes up and do a registration, it obviously missed initial INVITE message
> for a call, and it looks like server do not retry invites if no provisional
> messages returned from a client.
>
> So the question is it possible to tell OpenSIPS server to re-send INVITE
> messages (by some timer probably) if no provisional information received
> from a client.
>
> I know there is fr_timer and fr_inv_timer params from tm module, but it
> looks like they did not do the trick.
>
> Actually, I have almost default config file, except the parts, required
> for push notifications.
> I've posted it here: http://pastebin.com/tZmP320g
>
> Yours sincerely,
> Andrei Zhuravlev
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> ___
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>
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>
>
>
> ___
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
С уважением,
Андрей Журавлев
Архитектор проекта
MST Digital Agency
Email: andrei.zhurav...@m-st.ru
Skype: andrei.v.zhuravlev
Моб: +79176187334
http://m-st.ru

Yours sincerely,
Andrei Zhuravlev
Software Architect
MST Digital Agency
Email: andrei.zhurav...@m-st.ru
Skype: andrei.v.zhuravlev
Mob: +79176187334
http://m-st.ru
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Re: [OpenSIPS-Users] rtpengine xmlrpc integration with opensips. Teardown xmlrpc method

[Adrian Georgescu]

> On 29 Jan 2017, at 23:05, Carlos Oliva  wrote:
> 
> Hi List!
> 
> I'm using Opensips 1.11 and doing some tests to change my mediaproxy 
> rtprelays to ngcp-rtpengine. My reasons to try this change are efficiency and 
> that the mediaproxy project seems to be a little abandoned by AGProjects (not 
> really abandoned but has no new features in years)

Well, it is not abandoned. It just works. So there is nothing to fix, hence 
little activity about it.

Adrian


> After the change I started to see some dialogs in state 3 that ends at 
> timeout (6 hours in my config)
> 
> I tried to use the RTPTimeout function in rtpengine (in mediaproxy  it works 
> very well) to try to end the dialogs which don't have RTP.
> 
> To try this I used the rtpengine flags --b2b-url=http://%%:8000/RPC2 
> --xmlrpc-format=1 to send Opensips the order to end the related dialog.
> 
> It doesn't work. Doing some ngrep at xmlrpc interface seems that rtpengine 
> send some commands to opensips RPC interface that Opensips does not 
> understand. The command is: "teardown" and the callid, here is an example:
> 
> POST /RPC2 HTTP/1.1..Host: XXX.XXX.XXX.XXX:8000..Accept: */*..Content-Type: 
> text/xml..User-Agent: Xmlrpc-c/1.33.14 Curl/7.38.0. 
> .Content-Length: 204   
> ding="UTF-8"?>teardown822048991-4075...@bjc.bgi.b.ge
>  
> 
> 
> and the opensips response:
> 
> HTTP/1.1 200 OK..Connection: Keep-Alive..Content-Length: 48..Content-Type: 
> text/xml; charset=utf-8..Date: Sun, 29 Jan 2017 20:31:36 
> GMTInternal server error!
> 
> Obviously OpenSips does not implement this "teardown" method.
> 
> My questions are:
> 
> Anybody has a good idea of how to deal with this?
> 
> Devels: Do you think is a good idea to open a feature request in github about 
> this? I'll try to backport to 1.11 later.if you accept the request.
> 
> 
> Thanks and Regards,
> 
> Carlos Oliva
> 
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Re: [OpenSIPS-Users] iOS 10 Push Notifications

[Adrian Georgescu]

When the Registration expires or it is closed when the device goes to 
background, there is no 408.

The idea is to attempt to wake up the device when call comes in, wait for it to 
register and then fork the original INVITE. Then timeout if no device registers.

Adrian


> On 5 Feb 2017, at 12:53, Aron Podrigal  wrote:
> 
> you can certainly handle this in a t_on_failure for a 408...
> 
> On Thu, Feb 2, 2017, 3:38 PM Adrian Georgescu  > wrote:
> There is more plumbing needed to replay transactions later and the future 
> OpenSIPS version announced recently will help make this happen.
> 
> You are not alone fighting with this problem.
> 
> Regards,
> Adrian
> 
> 
>> On 1 Feb 2017, at 19:46, Андрей Журавлёв > > wrote:
>> 
>> Hi All,
>> 
>> I am pretty new with SIP and OpenSIPS. I have problem that affect most of 
>> the people who should support mobile clients for iOS 10. There must be a way 
>> to solve it, but it looks like I missed it. So I need your help.
>> 
>> Background: As you probably already know in iOS 10 Apple prevented network 
>> connections in background mode. It mostly affected VoIP apps. In order to 
>> solve this issue they introduced so-called VoIP Push notifications (via 
>> PushKit) which should automatically wake-up application and allow it to do a 
>> registration and receive a call.
>> 
>> Btw, the only thing I found capable to send VoIP push notifications without 
>> issus is ruby gem/binary called Houston.
>> 
>> Now everything works fine, except one issue actually, when application wakes 
>> up and do a registration, it obviously missed initial INVITE message for a 
>> call, and it looks like server do not retry invites if no provisional 
>> messages returned from a client.
>> 
>> So the question is it possible to tell OpenSIPS server to re-send INVITE 
>> messages (by some timer probably) if no provisional information received 
>> from a client.
>> 
>> I know there is fr_timer and fr_inv_timer params from tm module, but it 
>> looks like they did not do the trick.
>> 
>> Actually, I have almost default config file, except the parts, required for 
>> push notifications.
>> I've posted it here: http://pastebin.com/tZmP320g 
>> 
>> 
>> Yours sincerely,
>> Andrei Zhuravlev
>> ___
>> Users mailing list
>> Users@lists.opensips.org 
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users 
>> 
> 
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Re: [OpenSIPS-Users] rtpengine xmlrpc integration with opensips. Teardown xmlrpc method

[Carlos Oliva]

Thank you so much Bogdan. great job!

Thanks and regards,










* _ Carlos
OlivaDepartamento de Sistemas C/ Pujades, 77-79, 8a Planta 9B | 08005
Barcelona www.numintec.com  |
carlos.ol...@numintec.com  | T: 902 02 02 97
_ Talking Numintec:
Dialogando con empresarios de éxito 
 Las soluciones en la nube de
Numintec - Casos de éxito 
 Solicita una demo

_ Medio Ambiente: Antes de
imprimir este mensaje, asegúrese de que es necesario. Nota Legal: La
información contenida en la presente transmisión es confidencial y su uso
únicamente está permitido a su(s) destinatario(s). Le informamos que los
datos personales que facilite/ha facilitado pasarán/han pasado a formar
parte de un fichero responsabilidad de NUMINTEC COMUNICACIONES S.L.. y que
tiene por finalidad gestionar las relaciones. Tiene la posibilidad de
ejercitar los derechos de acceso, rectificación, cancelación y oposición
respecto a sus datos ante la empresa, en el e-mail
comunicac...@numintec.com   o bien en el
 domicilio sito en C/ Pujades, 77-79 8ª Planta 9-B 08005 de Barcelona.*

2017-02-04 16:14 GMT+01:00 Bogdan-Andrei Iancu :

> Hi Carlos,
>
> I made 2 commits on master:
>
> 1) the dlg_end_dlg MI command may identify the call also by the SIP Call-ID
> https://github.com/OpenSIPS/opensips/commit/
> 9a4f435bc1f550b0c047926fea7e9b83f71f4c7d
>
> 2) added a teardown MI command in rtpengine module - this is simple
> wrapper to dlg_end_dlg, just to make rtpengine happy
> https://github.com/OpenSIPS/opensips/commit/
> dd41b34cbcc7f5a8bfcd393016badde2c8ea6d5c
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 02/02/2017 12:35 PM, Carlos Oliva wrote:
>
> Thank you Bogdan, If you think is right I'll open a feature request about
> this.
>
> Another related question: Is there any way to end a dialog with MI using
> the callID instead of dialogID? If there is, I can make a patch for
> rtpengine and try contribute it, If they accept.
>
> thanks and regards,
>
>
>
> * _ Carlos
> OlivaDepartamento de Sistemas C/ Pujades, 77-79, 8a Planta 9B | 08005
> Barcelona www.numintec.com  |
> carlos.ol...@numintec.com  | T: 902 02 02 97
> _ Talking Numintec:
> Dialogando con empresarios de éxito 
>  Las soluciones en la nube de
> Numintec - Casos de éxito 
>  Solicita una demo
> 
> _ Medio Ambiente: Antes de
> imprimir este mensaje, asegúrese de que es necesario. Nota Legal: La
> información contenida en la presente transmisión es confidencial y su uso
> únicamente está permitido a su(s) destinatario(s). Le informamos que los
> datos personales que facilite/ha facilitado pasarán/han pasado a formar
> parte de un fichero responsabilidad de NUMINTEC COMUNICACIONES S.L.. y que
> tiene por finalidad gestionar las relaciones. Tiene la posibilidad de
> ejercitar los derechos de acceso, rectificación, cancelación y oposición
> respecto a sus datos ante la empresa, en el e-mail
> comunicac...@numintec.com   o bien en el
>  domicilio sito en C/ Pujades, 77-79 8ª Planta 9-B 08005 de Barcelona.*
> 2017-02-01 13:26 GMT+01:00 Bogdan-Andrei Iancu :
>>
>> Hi Carlos, OpenSIPS does not have any "teardown" MI command - this looks
>> like a private extension of the rtpengine. Unfortunately, the rtpengine
>> team is not so communicative (at least not with our team), so we were not
>> aware of this extension. Of course, we can do the one-way effort to align
>> OpenSIPS to the rtpengine (again). Still, if we do this, it will be for
>> OpenSIPS 2.3 - new extensions are not backported to the existing stable
>> releases. Best regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 01/30/2017 12:05 AM, Carlos Oliva wrote:
>>
>> Hi List!
>> I'm using Opensips 1.11 and doing some tests to change my mediaproxy
>> rtprelays to ngcp-rtpengine. My reasons to try this change are efficiency
>> and that the mediaproxy project seems to be a little abandoned by
>> AGProjects (not really abandoned but has no new features in years)
>> After the change I started to see some dialogs in state 3 that ends 

Re: [OpenSIPS-Users] Opensips crash

[Bogdan-Andrei Iancu]

Hi Denis,

Run "opensips -V" .

Best regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 02/06/2017 07:16 AM, Denis wrote:

Hello, Bogdan!
Could you, please, tell me how can i checked, that the version of 
Opensips is latest?

Thank you.
--
С уважением, Денис.
Best regards, Denis
05.02.2017, 19:05, "Bogdan-Andrei Iancu" :

Hi Denis,

First, be sure you use the latest 2.2 from GIT.

Now, the crash indicates a memory corruption, so please activate the 
memory debugger:  Run 'make menuconfig' from the main source dir -> 
Configure Compile Options -> Configure Compile Flags -> check 
QM_MALLOC and DBG_MALLOC. Then hit 'q' to go back to the previous 
menu, and hit 'Save Changes'.


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com 
On 02/02/2017 10:03 AM, Denis via Users wrote:

Hello!
Server:: OpenSIPS (2.2.2 (x86_64/linux))
Information from the core file: https://yadi.sk/i/IfhdM1Y83CQvPk
Thank you for any help.
--
С уважением, Денис.
Best regards, Denis
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