similar to: Asterisk Authentication

Displaying 20 results from an estimated 2000 matches similar to: "Asterisk Authentication"

2015 Jan 08
2
queue reload command
Hi I'm using asterisk 1.8 Does anyone know how to use the queue reload command. The built in help doesn't really help. queue reload {parameters|membe Reload queues, members, queue rules, or parameters Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w: http://www.pack-net.co.uk
2014 Jul 21
1
TLS, STRP and ARA
Hi I'm just about to upgrade to version 1.8.29.0 and have compiled with SRTP. However, we exclusively use the asterisk realtime architecture using the mysql connector. Looking at tutorials we have to set encryption=yes and transport=tls for any peer we want encrypted traffic for. Having a look at contrib/realtime/mysql/sippeers.sql from the source code shows that the encryption column is
2014 May 02
1
CDR billsec issue with calls forwarded through the Local channel
Hi I'm using asterisk 1.8.23.1 but I've seen this same issue in previous versions of 1.8. I have created some work arounds but the behaviour is incorrect. This is the scenario: Call comes in and goes to appropriate dialplan In the dialplan the call is forwarded to another number using a Local channel (and using /n ) e.g. Dial(Local/<my-number>@outbound-context/n,60) The number is
2015 Mar 13
2
ringing in queues
We use the ringall strategy for a small queue with 4 members. When a call comes in, if one of the members is busy, all the phones except the busy phone rings (as intended). While the other phones are ringing, if this busy phone becomes available again, we would like to have it start ringing. Right now it just sits idle. Is this possible? I played with ringinuse (queues.conf) and callcounter
2015 Jun 09
2
Manipulate extension state in 1.8.x
Hi Is there any way to set the presence state of a peer to in-use in asterisk 1.8? The idea is to integrate DND buttons on phones to BLF. Regards -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street
2014 May 20
2
Voicemail message to text
HI there I was wondering if anyone has implemented voicemail to text and if so, what package is being used to do so? Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester,
2013 Dec 05
1
Lync and Asterisk Realtime Architecture
Hi guys We're using asterisk 1.8.23.1 on CentOS 5 and are trying to create a trunk to MS Lync server. If I create the peer in sip.conf the trunk connects with no problem. However, we prefer to use ARA. Whenever we define the peer in our peers table, the trunk does not work, even if we use sip show peer <peer-name> load. Has anyone got any experience of connecting to Lync using ARA?
2014 Jan 10
1
CTI
Hi people I'm just mailing to see what people are using for CTI solutions with asterisk. Aslos, has anyone managed to integrate asterisk with Salesforce? Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2,
2014 Jun 10
1
Mixing res_mysql and res_odbc
Hi Is there any harm in using res_mysql for some things and res_odbc for others? We already use res_mysql for ARA but could do with having CEL logged to MySQL. Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex
2014 Oct 24
1
Call forwarding from Phones and getting the referrer IP
Hi I'm using asterisk 1.8 but I'm sure this applies to other versions. If someone puts a call divert on a handset such as a Snom phone I get this type of SIP message on receipt of an inbound call: Got SIP response 302 "Moved Temporarily" back from xxx.xxx.xxx.xxx:xxxxx Which then triggers a local channel to make the call. Is there any way I can access that IP address inside
2014 Jul 07
1
CDR dcontext not updated on FAILED and BUSY calls
Hi We're using asterisk 1.8.23.1. Our inbound calls are routed into the default context with explicit number matching. If found they are passed on to a distinct context for the number being called using the Goto application. If the call is successful or even if it has no answer, the cdr dcontext field has the correct second context. However, if the call fails or is busy, and even though we
2015 Sep 21
2
Call waiting for Queue Agents.
Hi All, I have a question about the Queues. I'm using Asterisk 11.13.0 , and I want to configure the following setup : When there is an incoming call to the queue all agents should ring even those that are already in call, they should receive a second call. Is this doable in any Asterisk version ? Thanks in advance. -------------- next part -------------- An HTML attachment was
2016 May 11
2
How is Queue avg holdtime and avg talktime calculated
2014 May 15
1
Asterisk 1.8 and calendar intergration
Hi I'm using asterisk 1.8.25.0 on CentOS 6. I have compiled it with all the calendar modules: *CLI> module show like calendar Module Description Use Count res_calendar.so Asterisk Calendar integration 4 res_calendar_ews.so Asterisk MS Exchange Web Service Calenda 0 res_calendar_caldav.so
2015 Feb 10
2
IAX port
On 10 February 2015 at 09:02, jg <webaccounts173 at jgoettgens.de> wrote: > > >> >> I get an occasional similar problem, we have Mikrotik firewalls and from >> tcpdump monitoring on the asterisk boxes I can see that the firewall >> (unbidden) has changed the IAX port. Usually a firewall reset and sometimes >> PBX reset combination fixes it. >>
2015 Jul 06
2
Asterisk how to setup alarm too many outgoing calls from same user
Hello, I would like to setup a mechanism to trigger an alarm if user is deal too many numbers within a very short period of time. Safeguard against users hacked accounts. can someone help? Thanks,
2015 Sep 22
2
How to set the global setting for each pjsip endpoint
I have many endpoints and each endpoint has some parameter in common so i wonder is there any way to config one for all endpoints? Like in my example I have two endpoints and I repeat the same thing, [100] type=endpoint aors=100 auth=100-auth allow=ulaw,alaw,gsm,g726 context=from-internal callerid=device <100> dtmf_mode=rfc4733 use_avpf=no ice_support=no
2014 Jan 07
1
Asterisk NAT friendly settings
I'm asking about this scenario: Asterisk(public IP) <--> Internet <--> Router (public IP) <--> SIP client (private IP and NAT) What settings in sip.conf will give this the best fighting chance of working? We already have nat=force_rport,comedia
2014 Mar 20
1
WebRTC and Asterisk 12
Anyone know of a tutorial for configuring WebRTC on Asterisk 12 using PJSIP? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140320/dff2c27b/attachment.html>
2014 Apr 11
1
Asterisk to Microsoft Lync2013?
Are they any gotchas to be aware of in getting Asterisk and Lync 2013 talking to each other using SIP? Or is Lync a pretty standard implementation of SIP? Cheers Tony -- Tony Mountifield Work: tony at softins.co.uk - http://www.softins.co.uk Play: tony at mountifield.org - http://tony.mountifield.org