Re: [OpenSIPS-Users] Firing gwlist in specific order with failover

2014-01-10 Thread Nick Cameo 
Hello Everyone, I do not want to ask the same questions in a few weeks. Can
anyone give some
example or help on this please.

Thanks in Advance,

PS Ali, sorry for the typo

N.
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Re: [OpenSIPS-Users] MI_HTTP

2014-01-10 Thread Dragomir Haralambiev 
Thanks!


2014/1/10 Ovidiu Sas 

> Please point a browser to http://IP_OPENSIPS:/mi
> If you defined a specific path for the mi interface, than use that
> particular one:
> http://www.opensips.org/html/docs/modules/devel/mi_http#id248023
>
> Example:
> modparam("mi_http", "mi_http_root", "opensips_mi")
> wil result in the following valid URL:
> http://IP_OPENSIPS:/opensips_mi
>
>
> Regards,
> Ovidiu Sas
>
> On Fri, Jan 10, 2014 at 12:15 PM, Dragomir Haralambiev
>  wrote:
> > Hi all,
> >
> > When use FIFO MI I make follow:
> > /opensipsctl fifo  dlg_list_ctx
> >
> > What I do when use MI_HTTP:
> >
> > http://IP_OPENSIPS:/dlg_list_ctx
> >
> >
> > Where is available examples?
> >
> > Regards,
> > PlayMen
> >
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
>
>
> --
> VoIP Embedded, Inc.
> http://www.voipembedded.com
>
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Re: [OpenSIPS-Users] MI_HTTP

2014-01-10 Thread Ovidiu Sas 
Please point a browser to http://IP_OPENSIPS:/mi
If you defined a specific path for the mi interface, than use that
particular one:
http://www.opensips.org/html/docs/modules/devel/mi_http#id248023

Example:
modparam("mi_http", "mi_http_root", "opensips_mi")
wil result in the following valid URL:
http://IP_OPENSIPS:/opensips_mi


Regards,
Ovidiu Sas

On Fri, Jan 10, 2014 at 12:15 PM, Dragomir Haralambiev
 wrote:
> Hi all,
>
> When use FIFO MI I make follow:
> /opensipsctl fifo  dlg_list_ctx
>
> What I do when use MI_HTTP:
>
> http://IP_OPENSIPS:/dlg_list_ctx
>
>
> Where is available examples?
>
> Regards,
> PlayMen
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



-- 
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http://www.voipembedded.com

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[OpenSIPS-Users] MI_HTTP

2014-01-10 Thread Dragomir Haralambiev 
Hi all,

When use FIFO MI I make follow:
/opensipsctl fifo  dlg_list_ctx

What I do when use MI_HTTP:

http://IP_OPENSIPS:/dlg_list_ctx


Where is available examples?

Regards,
PlayMen
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[OpenSIPS-Users] IM & Presence

2014-01-10 Thread Chandra Prakash 
Hi,

 

Is it possible if we can chat with all the registered or online users
without adding them in buddies or contact list ?

 

Thanks

Chander

 

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Re: [OpenSIPS-Users] create a new call-ID

2014-01-10 Thread M.Khaled W Chehab 
Hi,
If I use b2bua can I still handle the release code ?
Or please can you guide me on how to use b2bua for call-id hiding only 

10x


-Original Message-
From: Miha [mailto:m...@softnet.si] 
Sent: Friday, January 10, 2014 3:40 PM
To: OpenSIPS users mailling list; M.Khaled W Chehab
Subject: Re: [OpenSIPS-Users] create a new call-ID

you will have to use B2BUA:)

I think there is no other way.

On Fri, 10 Jan 2014 15:32:22 +0200
 "M.Khaled W Chehab"  wrote:
> Hi,
> 
>  
> 
> User--? opensips -?trunk
> 
> How can I change the call-ID  for  the call leg from opensips to trunk 
> since I don't want to forward  the same call-id to trunk.
> 
> NB: I am not using a B2BUA
> 
>  
> 
>  
> 
> Regards
> 
>  
> 




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[OpenSIPS-Users] Opensips 1.9.1 and dialog process

2014-01-10 Thread dpa 


Hello

 

Early I used opensips1.6.4.-2. There were some fields (hash_entry and hash
id) in DB scheme of dialog module. I used this field to end dialog.

Now, in 1.9.1, the DB scheme of dialog module has been changed and there are
no fields hash_entry and hash id in шею But there is the dlg_id  field.

 

The question is, How can I bind dlg_id in mysql with hash entry and hash_id
for dialog ending.

 

Thank you for any help.

 

 

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Re: [OpenSIPS-Users] count dialog calls

2014-01-10 Thread Răzvan Crainea 

Hello!

You can try the $stat(active_dialogs) pseudo-variable. Make sure you are 
loading the statistics module.


Best regards,

Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 01/10/2014 03:29 PM, M.Khaled W Chehab wrote:

Dears,

I there a function in the script not using shell command( opensipsctl fifo 
get_statistics active_dialogs  ) to get calls in process that opensips handling 
with status from 1 to 4

Regards




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Re: [OpenSIPS-Users] create a new call-ID

2014-01-10 Thread Miha 
you will have to use B2BUA:)

I think there is no other way.

On Fri, 10 Jan 2014 15:32:22 +0200
 "M.Khaled W Chehab"  wrote:
> Hi,
> 
>  
> 
> User--? opensips -?trunk
> 
> How can I change the call-ID  for  the call leg from
> opensips to trunk since
> I don’t want to forward  the same call-id to trunk.
> 
> NB: I am not using a B2BUA
> 
>  
> 
>  
> 
> Regards
> 
>  
> 


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[OpenSIPS-Users] create a new call-ID

2014-01-10 Thread M.Khaled W Chehab 
Hi,

 

User--à opensips -àtrunk

How can I change the call-ID  for  the call leg from opensips to trunk since
I don’t want to forward  the same call-id to trunk.

NB: I am not using a B2BUA

 

 

Regards

 

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[OpenSIPS-Users] count dialog calls

2014-01-10 Thread M.Khaled W Chehab 
Dears,

I there a function in the script not using shell command( opensipsctl fifo 
get_statistics active_dialogs  ) to get calls in process that opensips handling 
with status from 1 to 4 

Regards




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Re: [OpenSIPS-Users] Caller id

2014-01-10 Thread Miha 

@Razvan tnx for you answer.

I tried with remove_hf() but I this did not work. I guess due to failure 
route as request that enter failure route does not have RPID or p-asserted.


SIP/2.0 302 Moved Temporarily.
To: ;tag=77cc99cb150aeeefi0.
From: "38618108758" ;tag=tarm9Ucep73Um.
Call-ID: 5250f421-f243-1231-5695-005056b2fe3d.
CSeq: 54214780 INVITE.
Via: SIP/2.0/UDP opensips:5060;branch=z9hG4bK96d7.e47216a1.0.
Via: SIP/2.0/UDP
RTP_IP:5080;received=RTP_IP;rport=5080;branch=z9hG4bK3X4y2N8F54H9a.
Record-Route: .
Contact: .
Diversion: "38618108753" ;reason=unconditional.
Server: Linksys/SPA922-6.1.5(a).
Content-Length: 0.

br
miha


Dne 1/9/2014 8:37 AM, piše Răzvan Crainea:

Hi, Miha!

remove_hf() is the function you should use. According to your 
scenario, you should remove the RPID and PAI in the failure route. Are 
you sure that code is reached for the second INVITE?


Best regards,

Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 01/07/2014 03:52 PM, Miha wrote:

Hi,

how can I remove/not send RPID and P-asserted identity?

Opensips sends to UAC (with RPID and p-asserted), uac sends back 302
request and that due to 302 I am doing new invite with opensips but in
this invite I can see RPID and p-asserted.

I am trying to remove it with remove_hf() but this does not works.

How can I deal with this issue.

here is a sip trace:

U opensips:5060 -> UAC_PUBLIC_IP:13647
INVITE sip:38618108753@UAC_PUBLIC_IP:13647 SIP/2.0.
Record-Route: .
Via: SIP/2.0/UDP opensips:5060;branch=z9hG4bK96d7.e47216a1.0.
Via: SIP/2.0/UDP
RTP_IP:5080;received=RTP_IP;rport=5080;branch=z9hG4bK3X4y2N8F54H9a.
Max-Forwards: 67.
From: "38618108758" ;tag=tarm9Ucep73Um.
To: .
Call-ID: 5250f421-f243-1231-5695-005056b2fe3d.
CSeq: 54214780 INVITE.
Contact: .
User-Agent:
FreeSWITCH-mod_sofia/1.2.17+git~20131230T193020Z~52377f0f65~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 207.
X-Call_id: 56c99a91-d0ef0551@172.31.1.103.
X-FS-Support: update_display,send_info.
P-Asserted-Identity: .
Remote-Party-ID: 0038618108758
;party=calling;id-type=subscriber;privacy=off;screen=yes. 


.
v=0.
o=FreeSWITCH 1389081737 1389081738 IN IP4 RTP_IP.
s=FreeSWITCH.
c=IN IP4 RTP_IP.
t=0 0.
m=audio 19952 RTP/AVP 0 8 9 101 13.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:30.


U UAC_PUBLIC_IP:13647 -> opensips:5060
SIP/2.0 302 Moved Temporarily.
To: ;tag=77cc99cb150aeeefi0.
From: "38618108758" ;tag=tarm9Ucep73Um.
Call-ID: 5250f421-f243-1231-5695-005056b2fe3d.
CSeq: 54214780 INVITE.
Via: SIP/2.0/UDP opensips:5060;branch=z9hG4bK96d7.e47216a1.0.
Via: SIP/2.0/UDP
RTP_IP:5080;received=RTP_IP;rport=5080;branch=z9hG4bK3X4y2N8F54H9a.
Record-Route: .
Contact: .
Diversion: "38618108753" 
;reason=unconditional.

Server: Linksys/SPA922-6.1.5(a).
Content-Length: 0.


U opensips:5060 -> RTP_IP:5060
INVITE sip:30238618108756@opensips SIP/2.0.
Record-Route: .
Via: SIP/2.0/UDP opensips:5060;branch=z9hG4bK96d7.e47216a1.1.
Via: SIP/2.0/UDP
RTP_IP:5080;received=RTP_IP;rport=5080;branch=z9hG4bK3X4y2N8F54H9a.
Max-Forwards: 67.
From: "38618108758" ;tag=tarm9Ucep73Um.
To: .
Call-ID: 5250f421-f243-1231-5695-005056b2fe3d.
CSeq: 54214780 INVITE.
Contact: .
User-Agent:
FreeSWITCH-mod_sofia/1.2.17+git~20131230T193020Z~52377f0f65~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 207.
X-Call_id: 56c99a91-d0ef0551@172.31.1.103.
X-FS-Support: update_display,send_info.
P-Asserted-Identity: .
Remote-Party-ID: 0038618108758
;party=calling;id-type=subscriber;privacy=off;screen=yes. 


Moved: 38618108753.
.
v=0.
o=FreeSWITCH 1389081737 1389081738 IN IP4 RTP_IP.
s=FreeSWITCH.
c=IN IP4 RTP_IP.
t=0 0.
m=audio 19952 RTP/AVP 0 8 9 101 13.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:30.

tnx!

miha







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