Displaying 20 results from an estimated 10000 matches similar to: "Extract Remote-Party-ID from incoming INVITE in dialplan"
2011 Mar 09
6
SIPAddHeader not working
Hello list,
I notice that the dialplan method SIPAddHeader is not working :
in dialplan :
/exten => s,n,SIPAddHeader(Privacy: id)/
in SIP invite no trace of this header :
/INVITE sip:0473 at sip.domain.be SIP/2.0
Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97
From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2
To: <sip:0473 at sip.domain.be>
2010 Jul 12
4
Remote-Party-ID party=called
Hello list,
using Asterisk 1.4.30.
I want to set the SIP-header Remote-Party-ID to display the name of the
calling party on my phone in stead of the number.
This is the dialplan :
exten => 10,1,NoOp()
exten => 10,n,SIPAddHeader(Remote-Party-ID: "eric"
<sip:10 at 192.168.1.150>;party=called )
exten => 10,n,Dial(SIP/test2)
This is what the CLI shows :
/[Jul 12
2013 Aug 20
2
Dialplan MySQL inserted ID
Hello,
how can I obtain the "inserted ID" after having inserted a row with
MySQL in the dialplan ?
exten => s,n,MYSQL(Query resultid ${connid} INSERT INTO myTable SET
C1="${ARG1}", C2="${ARG2}",
timestamp="${STRFTIME(${EPOCH},,%Y-%m-%d_%H:%M:%S)}")
I need to know the ID of the newly inserted row.
Kind regards,
Jonas.
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2023 Apr 06
1
Remote-Party-ID set to 0 on re-invite using pjsip in Asterisk 16.
We've been using Asterisk 16 for a while now, and tried turning on
send_rpid = yes in my pjsip config for end points. This solves a
problem we're having where attended transfers aren't updating the
CallerID when the transfer is complete (it would show the callerID of
the party attempting the transfer, and never update after the transfer
happened).
The side effect of this change
2014 Oct 28
1
dialplan reload context
Hello,
is it possible to reload just a context in stead of the whole dialplan ?
I see this on the tracker :
https://issues.asterisk.org/jira/browse/ASTERISK-19934
But is it possible in some Asterisk version ?
Kind regards,
Jonas.
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2011 Nov 22
1
Asterisk refuses INVITE (401) and I don't know why
Hello list,
this is the communication between an Aastra 5000 PBX and Asterisk, where
the Aastra makes a call to Asterisk. For some reason, Asterisk responds
with 401-Unauthorized and I don't know why.
Calls go well with Panasonic PBX, Avaya PBX, Alcatel-Lucent PBX but NOT
with this Aastra.
A1.A1.A1.A1 = IP-address Asterisk PBX
AS.AS.AS.AS = IP-address Aastra PBX
Aastra PBX makes a call
2015 Mar 16
1
Use dialplan variables from MySQL database and replace with value
Hello
i have the following field (text string) in a MySQL database :
"${KNUMMER} ${phone_number_to} ${phone_number_from} ${CHANNEL:4}"
I read this string form the database and want to have the dialplan
variables to be replaced with the correct content.
How can I do this ?
Currently this is not working. The variable ${PARAMS} contains the exact
string of the database field :
my
2006 May 25
1
Paging Phones stay off the hook if you dont wait long enough.
I've got one that I haven't been able to solve. Hopefully someone else
has had this issue.
I'm using the paging script in free pbx, which appears to:
Send a sipheader autoanswer,
Create a conferece
Add the phone to the conference
But if the user hits the page extension, all the phones auto answer, and
if they hang-up before the phones join the conference I end up with
dozens of
2009 Dec 21
1
Incoming calls coming into default context
My SIP-provider sends my a SIP-invite like this :
INVITE sip:329298yyy6 at 80.XX.XX.69:5060 SIP/2.0
Via: SIP/2.0/UDP 80.XX.XX.68:5060;branch=z9hG4bKf395877e02e5aa21fd8f5a0c
Max-Forwards: 70
From: <sip:321445xxx6 at 80.XX.XX.69>;tag=f395877e02bf8eb2fd8f5a0e
To: <sip:329298yyy6 at 80.XX.XX.69>
Call-ID: f395877e02187250fd8f5a0f at 80.XX.XX.68
CSeq: 1 INVITE
User-Agent: SysMaster VoIP
2007 Aug 29
3
Queue Agents on Remote Asterisk server?
Hi,
I have a main Asterisk server, and a server at a branch location
connected via a IAX2 trunk. I want to have a queue at the main
location that has people from both locations as members. I got this
working, but the trouble comes when the round-robin logic selects a
member at the branch office to call. If that user is unavailable,
their voicemail answers the call, and the main server
2010 May 31
6
Voicemail : mail attachment to multiple mail-addresses
Hello list,
google returns a discussion on the dev-list when I search for how to
mail a voicemail to multiple mail addresses.
Is there yet a seperator that actually works to define multiple mail
addresses ?
Kind regards,
Jonas.
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2016 Aug 17
4
pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol
On 16-08-16 17:45, George Joseph wrote:
>
>
> On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens
> <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote:
>
> On 16-08-16 04:38, George Joseph wrote:
>>
>>
>> On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens
>> <jonas.kellens at telenet.be <mailto:jonas.kellens at
2010 Sep 09
5
info about application not available asterisk 1.6.2.11
Hello list,
how come on my Asterisk 1.6.2.11, I have no help available ?!
asterisk*CLI> core show application Dial
-= Info about application 'Dial' =-
[Synopsis]
Not available
[Description]
Not available
[Syntax]
Not available
[Arguments]
Not available
[See Also]
Not available
Kind regards,
Jonas.
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2016 Aug 16
2
pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol
On 16-08-16 04:38, George Joseph wrote:
>
>
> On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens
> <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote:
>
> Hello
>
> using pjproject 2.5.5
> using asterisk-certified-13.8-cert1
>
>
> IIRC there were API changes in pjproject 2.5 that aren't accounted for
> in
2010 Oct 26
11
Auto provisioning from public server
Hello,
has anyone experience with auto provisioning IP-phones on different
locations through a central public provisioning server ? You use http or
https ?
Is there a danger that one uses a different MAC-address in the
provisioning link to obtain SIP username / password settings ?
Kind regards,
Jonas.
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2016 Aug 11
2
Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
On 11-08-16 18:03, Matt Fredrickson wrote:
> On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens <jonas.kellens at telenet.be> wrote:
>> My main reason not to upgrade to Ast 13 is because I'm afraid of losing
>> functionality as there are certain functions deprecated/replaced. This can
>> also cause headache :-)
>>
>> I will do so if there is no other option.
2016 Aug 12
2
Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
Hello
setting "nat=no" or omitting "nat=" in peer definition does not help
either. Still no audio.
Why do you think this is a NAT issue ? IP and port information in
SDP-body is correct.
Kind regards.
On 12-08-16 09:25, ????? ?????? wrote:
>
> Try delete nat from 770000wrtc settings ice should do the same
>
>
> On Aug 11, 2016 10:00 PM, "Jonas
2010 Oct 18
15
SIP DNS SRV
Hello list.
When using SIP DNS SRV to define a production Asterisk server with high
priority and a backup Asterisk server with a lower priority on this
DNS-server, will this work as follow :
- production server is reachable, so registration of the IP-phone goes
to this server
- production server is unreachable, so registration goes to the backup
Asterisk server
- production server is
2016 Sep 10
2
Queue show : failed to extend from 240 to 327
On 10-09-16 00:50, Richard Mudgett wrote:
>
>
> On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens
> <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote:
>
> Hello
>
> when I type on the Asterisk CLi 'queue show', I first get a list
> of my queues and then the following :
>
>
> failed to extend from 240 to 327
2012 Sep 28
1
Disconnect calls : known reasons
Hello,
are there any known reasons why Asterisk would disconnect random calls ?
My server uses 1,5 GB out of 8 GB RAM
My server uses up to 35% CPU at peak
There are about 40 concurrent calls.
I have 300 RTP-ports available.
I just see the call ending, as if one of the connected parties hung up
but that is not the case !
So what could be a bottleneck ? Any known reasons for random hangup ?