[asterisk-users] PJSIP outgoing INVITE and "contact" value

[Dmitriy Serov]

asterisk 13.8.7, PJSIP.

One VoIP provider requires a specific value in the field "contact" of a 
INVITE.


What setting does indicate the value will be in this field (instead 
"asterisk")?


Thanks.


currect settings (with templates):

[srv_d22778](srv-auth)
username=999
password=secret
[srv_d22778](srv-aor)
contact=sip:999x...@reg.dc-tc.com
[srv_d22778](srv-endpoint)
accountcode=5100
from_domain=reg.dc-tc.com
aors=srv_d22778
outbound_auth=srv_d22778
[srv_d22778](srv-registration)
outbound_auth=srv_d22778
client_uri=sip:999x...@reg.dc-tc.com
contact_user=srv_d22778
endpoint=srv_d22778
server_uri=sip:reg.dc-tc.com

INVITE sip:79265x...@reg.dc-tc.com SIP/2.0
Via: SIP/2.0/UDP 
85.142.148.80:5060;rport;branch=z9hG4bKPj05fd6429-c1a3-4bfb-87f8-df094c6532e4
From: "999" 
;tag=2b3456b9-8adc-4413-8012-98de85f949a8

To: 
Contact: 


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Re: [asterisk-users] Strange SIP debug

[Aqs Younas]

ACK is being forwarded to contact that you have in 200 ok. You need to
check 200 contact header.

On 17 May 2016 at 17:59, Антон Сацкий  wrote:

> Hi list need your advice
> i dont understand why reply ACK goes to diferrent ip (192.168.88.32)
> SCREEN below
>
> http://tinypic.com/view.php?pic=s6m7me&s=9#.VzsVhvl96Ik
>
> THANK U ALL
>
>
>
> --
> Best regards
> Antony
> tel.   +380669197533
> tel2. +380636564340
> Paypal http://paypal.me/Satskiy
> 
> satski...@gmail.com 
>
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>
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[asterisk-users] Asterisk-Java library

[Grant Bagdasarian]

Hello,

Does the asterisk-java library (https://github.com/asterisk-java/asterisk-java) 
work with the latest LTS version of Asterisk?
I couldn't find information about the supported asterisk versions.
We're currently using the asterisk-java.1.0.0.m3 version on asterisk 1.6 and 
are planning to update to the latest version of asterisk, but I believe this 
does require asterisk-java to be updated as well.

If the library does not work with the latest version of asterisk, are there any 
other actively maintained java libraries for asterisk?

Regards,

Grant

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[asterisk-users] Strange SIP debug

[Антон Сацкий]

Hi list need your advice
i dont understand why reply ACK goes to diferrent ip (192.168.88.32)
SCREEN below

http://tinypic.com/view.php?pic=s6m7me&s=9#.VzsVhvl96Ik

THANK U ALL



-- 
Best regards
Antony
tel.   +380669197533
tel2. +380636564340
Paypal http://paypal.me/Satskiy

satski...@gmail.com 
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Re: [asterisk-users] [SOLVED] Asterisk 11 on Centos: Voicemail crashes when recording message

[asterisk]

That was it - compiling on the server & deploying has fixed the crash, 
so it must have been an architecture/cpu difference.


Cheers
Ade.

On 16/05/2016 21:08, Brian Wilson wrote:
Did you build from source on one machine and install on another? I 
ran into something like that, have to turn off optimizations in the 
build environment if you do that and the machine architecture is 
different.


On Mon, May 16, 2016 at 1:03 PM, asterisk 
> wrote:


Hi folks,

I'm running Asterisk 11 (at the moment - planning to u/grade to
v13.7 LTS), I've just configured the voicemail function, and it's
mostly working fine... except when I try to leave a voicemail!
This crashes asterisk with no entries in the messages log.






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Re: [asterisk-users] Asterisk 11 on Centos: Voicemail crashes when recording message

[asterisk]

Hi Brian,

The version I'm running right now was installed from RPMs, so I guess 
that counts as built on one machine & installed on another...


I've already compiled v13.7, next step is to try it out & see if that 
fixes the issue.


Thanks for your reply!

Cheers,
Ade.

On 16/05/2016 21:08, Brian Wilson wrote:
Did you build from source on one machine and install on another? I ran 
into something like that, have to turn off optimizations in the build 
environment if you do that and the machine architecture is different.


On Mon, May 16, 2016 at 1:03 PM, asterisk 
> wrote:


Hi folks,

I'm running Asterisk 11 (at the moment - planning to u/grade to
v13.7 LTS), I've just configured the voicemail function, and it's
mostly working fine... except when I try to leave a voicemail!
This crashes asterisk with no entries in the messages log.




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Re: [asterisk-users] asterisk admin interface

[Tzafrir Cohen]

On Mon, May 16, 2016 at 04:54:10PM -0700, John Kiniston wrote:
> You could explore using ARI with it's Push configuration.
> 
> https://wiki.asterisk.org/wiki/display/AST/ARI+Push+Configuration

With all due respect, this answer is basically like answering "use a
text editor to edit the dialplan".

The OP asked for an Asterisk administration interface. Right now the
only proper answer given (that did not involve writing one) was FreePBX.

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icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com

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