Taking a look at the DEBUG statements that are associated with the
thread processing the SIP response:
[Mar 15 13:16:05] DEBUG[27947] netsock2.c: Splitting 'FQDNz:5060' into...
[Mar 15 13:16:05] DEBUG[27947] netsock2.c: ...host 'FQDNz' and port '5060'.
[Mar 15 13:16:08] DEBUG[27947] netsock2.c: S
Taking a look at the DEBUG statements that are associated with the
thread processing the SIP response:
[Mar 15 13:16:05] DEBUG[27947] netsock2.c: Splitting 'FQDNz:5060' into...
[Mar 15 13:16:05] DEBUG[27947] netsock2.c: ...host 'FQDNz' and port '5060'.
[Mar 15 13:16:08] DEBUG[27947] netsock2.c: S
Hello!
We recently upgraded one of our customers from 1.4.44 to 1.8.15-cert1. We have
several other customers running both versions.
The customer in question does not use us as their provider as they’re located
in a different country.
When they make outgoing calls, there is a 3 second delay bet
I finally found the real culprit. The call-limit DB field was mapped to both
call-limit and callcounter in the view asterisk uses. The latter is what
caused the strange behaviour. Removed both and everything works as expected
now.
-Pan
- Original Message -
From: "Pan B. Christ
-
From: "Pan B. Christensen"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, December 10, 2012 11:54 AM
Subject: Re: [asterisk-users] BLF and call-limit in 1.8
Thanks for your reply. I just tested creating a peer in sip.conf and that
wo
- Original Message -
From: "Matthew Jordan"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Saturday, December 08, 2012 12:43 AM
Subject: Re: [asterisk-users] BLF and call-limit in 1.8
Thanks for your reply. I just tested creating a peer in sip.conf and that
wor
e the ability to the phone to set up more calls.
counteronpeer is the same as limitonpeer, just a new name.
/O
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pan B. Christensen
Sent: Thursday, December 06, 2012 9:50 AM
Hello
We have recently upgraded our internal PBX from 1.4 to 1.8. This made the BLF
lamps on our Polycom phones stop working. After a lot of googling and a lot of
testing, I have been unable to find a solution.
I did try to change the call-limit value from 4 to 1, and this actually made
BLF wo
Hello Mike,
It is possible for Polycom phones to auto-answer an incoming call with
speakerphone. I don’t have the details available right now, but it requires
changing the phone’s configuration and sending a custom sip header with the
INVITE. Great care should be taken when implementing this, a
Hello,
I have been hired to fix a large and complicated installation using several
Kamailio and Asterisk servers.
I found that I require some extra modules on some of the Asterisk servers. I
was hoping to be able to compile only the modules needed and copy them to where
they should be.
Asteri
alternatives
would be as well.
With kind regards,
Pan B. Christensen
Senior technician
Ibidium AS
http://www.ibidium.no/
- Original Message -
From: Bruce B
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tuesday, January 11, 2011 4:37 PM
Subject: Re: [asteri
rcial product?
With kind regards,
Pan B. Christensen
Ibidium AS
http://www.ibidium.no
- Original Message -
From: "Thomas Liu"
To:
Sent: Tuesday, January 11, 2011 5:14 PM
Subject: Re: [asterisk-users] Call queues on load-balanced asterisks
Hi Pan & Dhaval,
We have implemen
Hello Bruce,
Your understanding of NAT is correct, and your setup should work.
I’m not familiar with Pfsense, but I suspected that your problem was due to a
SIP ALG. Pfsense seems to have a SIP ALG and other special handling of VoIP
traffic. Hence, you are not using plain NAT. Pfsense is probab
ue name should be same for all
> server and asterisk can call same agent.
>
> you didnot mentioned that which purpose youwere use queue other wise i can
> give answer in better way.
>
> regards
> Dhaval
>
> On Fri, Jan 7, 2011 at 5:08 PM, Pan B. Christensen wrote:
>
&
Hello,
I have been asked to implement the following design:
Load-balanced Kamailio servers handling registrations and routing.
Load-balanced asterisk feature servers handling voicemail and other things
Kamailio cannot do. Plus several load-balanced gateways, but they are not
relevant to my qu
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