Hi,
Those are SIP URIs to be used as contacts by the presence server (a SIP
URI pointing back - as address- to the presence server); in a Contact
URI, the username part is no relevant (as the contact is a reference to
the IP location of the device), so you can put whatever you want as
usernam
you can count calls from/to a sip user using dialog profiling
You can read about it in opensips book (
https://books.google.es/books?id=y2wdDAAAQBAJ&pg=PA202&lpg=PA202&dq=opensips+dialog+count+calls&source=bl&ots=LFFoLxC9Gf&sig=fe_67u5d7xSNXm2gXUqbSOmTznM&hl=es&sa=X&ved=0ahUKEwiD6s3dy9DRAhXEUhQKHR
How did You check it? You checked calls from this sip login?
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Hi Marcin:
I had the same problem years ago with 1.6 version. In the first call BLF
works but the second call fails. I solved it using dialog module to check
the mumber of calls and not doing pua_set_publish if the mubers of calls
are > 1
I think this problem was solved on 1.11 version, wich is s
I think I found the reason BLF doesn't work. I made a test. I've erased
presenity table. Then:
1. I called other user, led is blinking, and presentity table shows record
with xml bodies with early state.
2. Then I pickup a call, the same record changes with confirmed state.
3. Then I end this call,
I think I found the reason BLF doesn't work. I made a test. I've erased
presenity table. Then:
1. I called other user, led is blinking, and presentity table shows record
with xml bodies with *early* state.
2. Then I pickup a call, the same record changes with *confirmed* state.
3. Then I end this
I want to ask what bolded values mean:
# -- core presence params --
modparam("presence","server_address","sip:*sa*@10.10.10.10:5060")
# -- pua and pua_dialoginfo parameters --
modparam("pua_dialoginfo", "presence_server", "sip:*sa*@10.10.10.10:5060")
I don't really know what to write there.
You are right Bogdan, I forgot about these requests. Anyway, thanks for help
- I will look at webinars.
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Hi,
It is a bit difficult to work with OpenSIPS without knowing at least the
SIP basics. I suggest to follow there webinars:
* Introduction to SIP : http://www.opensips.org/Documentation/Webinars#toc10
* Routing in SIP : http://www.opensips.org/Documentation/Webinars#toc12
A basic OpenSIPS scr
Where are the sequential requests? Can You point it in my script?
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I checked your script and you are not handling publish/subscribe for
sequential requests. You should try to move the route(handle_presence);
call for sequential requests too.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/17/2017 10:07 AM, maatohewetbi wrote:
Hello, can anybody help? I need blf to work.
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Yes, I added this route. You can check my config, it's in this post.
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It depends on where you added that block. Also, did you also add the
handle_presence route in your script?
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/09/2017 10:58 AM, maatohewetbi wrote:
Have You got any clue, why this is happnning?
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Have You got any clue, why this is happnning?
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That's right. All calls were working well before I add these lines. I
noticed, that when I erase these lines
*if (is_method("PUBLISH|SUBSCRIBE")) { **
**route(handle_presence); **
**exit; **
** }
*problem is gone, and all calls goes smoothly.*
*
Were your calls properly working befor
Were your calls properly working before adding the BLF and presence
scripting ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 03.01.2017 15:00, maatohewetbi wrote:
It doesn't work. If I call number B, it rings, and BLF LED is blinking, but
af
It doesn't work. If I call number B, it rings, and BLF LED is blinking, but
after picking up the phone, call is without audio and LED is still blinking
event I disconnect call. Then If I try to call again, call freezes like if
it's routing in loop, and B side doesn't ring at all. I heve to restart
Hello!
Basically merging the two separate configs together should do the trick.
If you follow this[1] example, change your Proxy config as it follows:
When an initial INVITE comes in, create the dialog and publish the
information:
if (is_method("INVITE")) {
create_di
I use Opensips 1.6.2 on my server. I want to add BLF function to it but I
don't know how. I have config where I route calls to asterisk. There are
tutorials where I can set Opensip Presence Server as standalone, with
OpenXcap as well, but there's nowhere config how to set Opensips to work on
one ma
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