Displaying 20 results from an estimated 400 matches similar to: "PJSIP device_state_busy_at, how does this work?"
2017 Nov 07
4
Call preemption
Hello,
Has anyone already implemented some sort of call preemption in Asterisk
? I am trying to achieve something like this :
- I want to limit the number of calls on a given SIP peer to 10
- on the other hand, some calls have higher priority than others
- when the ceiling of 10 calls is reached and a call with a high
priority is attempted, I would like to drop a call with a lower priority
2015 Jun 28
1
Branch based on call volume
?I meant how many calls are in progress on a particular trunk. (Sorry - I didn't even think of the other interpretation).
________________________________
From: asterisk-users-bounces at lists.digium.com <asterisk-users-bounces at lists.digium.com> on behalf of Matt Riddell <lists at venturevoip.com>
Sent: Sunday, June 28, 2015 9:26 AM
To: Asterisk Users List
Subject: Re:
2013 Jul 19
2
Meetme and maxusers option
Hi all.
I'm trying to limit the number of participants in a conference room
with the realtime option "maxusers", but it doesn't work.
Someone have this option working properly?
--
thiagoc
"O povo n?o deveria temer o governo. O governo ? quem deveria temer o povo."
V de Vingan?a
2006 Mar 24
3
* Meetme Freeze patch found
Hi all
Apparently there is a patch for those 1.2.4/5 MeetMe Freezes:
http://bugs.digium.com/view.php?id=5884
Haven't tried it out yet.
Benoit Panizzon
--
I m p r o W a r e A G - System Services
______________________________________________________
Zurlindenstrasse 29 Tel +41 61 826 93 00
CH-4133 Pratteln Fax +41 61 826 93 01
Schweiz
2019 Nov 18
4
On Register, run a script, validate source IP
Hi Gang
To increase security against phished passwords and similar attacks, we
consider offering customers to define IP ranges (or GeoIP locations)
from which their dynamic registrations are being accepted.
I can already look at the source IP in the dial plan, so no issue with
validate an INVITE against a source IP.
But I would also like to prevent registrations from outside of this
2006 Mar 28
3
How to send announcement after called has picked up the phone?
Hi
I would like to send a text to the called person when he picks up the phone
before the call gets connected through. Is there a way to do this?
Example: I'm registered to multiple SIP providers. They come in to a context
each and then get through to my phone. Now I would like to send myself an
announcement about from which SIP provider this call came from.
--
Beno?t Panizzon,
2020 Jan 13
3
Solved: Re: Asterisk 13.18.3 PJSIP. Wrong Port in Contact Header in Reply to REGISTER?
Hi Joshua
Thank you for your reply.
Indeed, Ubuntu only ships with this old version. Upgraded to 16.2. via
PPA. Problem persisted.
Well, I already mentioned that this is a machine with two physical
interfaces with different routes which on the 'external' side handles
SIP customer registrations and has an 'internal' IC Trunk to a
commercial Voice Switch via private IP Range.
I
2006 Apr 24
2
User Defined VoiceMail announcement?
Hi all
I noticed that most caller are quite confused by the standard voicemail
announcement text. Especialy as the number read is the 'internal' number.
Callers often hang up because they think having called the wrong number when
they hear the announcement.
Is there a way (like in many other PBXes) that the VoiceMail user could record
his own announcement? (like, hello, this is the
2018 Jan 09
2
PJSIP: identify endpoint by authentication username?
Dear fellow list readers
This is the situation:
ISDN Devices => Patton ISDN to SIP GW => Asterisk PJSIP
The Patton GW resides on a dynamic IP address, so I cannot really use
match=ip in the identify section.
The Patton does not send a line parameter.
The ISDN Devices behind the patton have different MSN and should be
able to send them in the From: Header, so the default endpoint
2019 Nov 29
2
pjsip: How is asterisk choosing the IP address to put in the Contact header?
Hi Gang
Server, two interfaces, routing to two different networks.
Two transports defined, each bound to the corresponding ip assigned to
the interface.
But still, especially when an 183 message is sent, the Contact header
does contain the wrong IP Address.
Is this a known issue 13.18.3? Or is there a way to make absolutely
sure the IP addresses within the Contact header is corresponding to
2017 Nov 19
2
pjsip subscribe (presence) always returns: No matching endpoint found
Hi Joshua
thank you for the quick reply
> Have you checked the Asterisk console when PJSIP is loaded to see if
> the endpoint did not load for some reason? Does it show up in "pjsip
> show endpoints"?
Yes, the endpoint shows up.
Endpoint: 11/(scrubbed from mail) Not in use 0 of inf
InAuth: 11/11
Aor: 11
2012 Oct 05
3
How to log caller IP address in the CDR?
Hello
We had this situation:
Some bot-net did try to guess SIP logins and finally succeeded. The Asterisk
Server was abused to call a large number of expensive destinations.
It is clear that the sip logins have been passed to various persons (probably
posted on a forum somewhere inviting to do 'free calls').
Right after the affected password was changed, the message log shows which
2020 Jan 24
4
Perl AGI: read variable with quotes
Hi Gang
I have stumbled of this problem.
I need the P-Asserted-Identity header in an AGI scrip.
In the Dial-Plan I do:
same => n,Set(PAI=${PJSIP_HEADER(read,P-Asserted-Identity)})
In the AGI I do:
my $pai = $AGI->get_variable(PAI);
This works fine, unless the PAI contains quotes:
P-Asserted-Identity: <sip:1000 at 1.2.3.4:5060;user=phone>
I get "<sip:1000 at
2023 Dec 04
1
Asterisk 13 / chan_sip / registration after reject
Hi List
We have some CPE which run an embedded asterisk 13 with chan_sip.
Unfortunately, when a registration is rejected, those stop trying.
I am familiar with pjsip which allows to configure:
auth_rejection_permanent=no
How do I achieve the same with chan_sip?
Mit freundlichen Grüssen
-Benoît Panizzon-
--
I m p r o W a r e A G - Leiter Commerce Kunden
2017 May 22
3
SIP Trunk over Proxy (matching ip of outbound proxy in incomming calls)
Hello List
I work at an SIP Provider and we have added and SBC in front of our
Voice Switch to protect it.
This requires all our SIP Trunk customers to register via a 'proxy'.
I struggle with Asterisk to work over a proxy.
This is what I have done so far.
register => username at sip.example.com:password at sbc.example.com
This works fine, asterisk is sending registrations via the
2019 Nov 19
2
Global number rewriting rules affecting ALL headers?
Hi Joshua
I had a shot at your suggestion, bug still no success.
I fear the 181 is sent before the macro is called.
I want to change the Diversion Header in the 181 message sent back to
the caller to put the number it contains in the correct e164 format
(stripping the 0 and adding +41 for Switzerland) but just any 'dialplan
set' value would do for an example :-)
Could you please make
2004 Oct 27
1
Winbindd as NIS replacement in heterogen environement
Hi all
We have the following environement:
Microsoft ADS for Windows Users, NIS for Un*x Users.
Samba 3.x Fileservers.
Win2k/XP Clients which use CIFS to connect to the Fileserver.
FreeBSD/Linux Clients which use NFS to connect to the Fileserver.
For the moment, Windows User authenticate against the ADS and Un*x users
authenticate against a NIS Server. Everything runs fine.
But we would like
2019 Nov 15
2
pre-dial handler, how to access variables from calling channel?
Hi List
Implementing screening and routing I have stumbled over this issue:
[pbx-router]
exten => s,1,NoOp(ROUTER FROM: ${CALLERID(Number)} TO: ${DESTINATION})
same => n,Set(SOURCE=${CHANNEL(name)})
same => n,Set(PAI=${PJSIP_HEADER(read,P-Asserted-Identity)})
same => n,Set(FROM=${CALLERID(Number)})
same => n,Set(TO=${DESTINATION})
same
2018 Jul 27
1
quota-status not working in distributed environment
On 2013-06-16 21:46, Timo Sirainen wrote:
> On 14.6.2013, at 9.15, Benoit Panizzon <benoit.panizzon at imp.ch> wrote:
>
>> Is there a way to get quota-status to also use the proxy feature to
>> request
>> the quota information from the correct machine?
>
> Looks like this is a missing feature. I first thought quota-status
> would go through doveadm
2017 Nov 20
2
How to correctly set REDIRECTING to indicate diversion reason
Hello List
Next question where google did not spit out an unsable answer.
When redirecting a call with Transfer, I would like to correctly
indicate the reason.
I did try this:
exten => XX,1,NoOp(Call to ${EXTEN} from ${CALLERID(all)})
exten => XX,n,Dial(SIP/ZZ)
exten => XX,n,set(REDIRECTING(reason)=cfb)
exten => XX,n,Transfer(SIP/YY)
I did try with 'reason'