similar to: Queues - periodic announce while ringing members

Displaying 20 results from an estimated 2000 matches similar to: "Queues - periodic announce while ringing members"

2015 Mar 13
0
ringing in queues
On 13 March 2015 at 14:04, Matt Hamilton <efes9999 at hotmail.com> wrote: > 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. >
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
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
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 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,
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 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 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
2015 Jun 09
0
Manipulate extension state in 1.8.x
You can use a custom device state to do it. [dnd] ;DND Toggle exten => *363,1,Answer() same => n,Set(CURRENT_PRESENCE=${DEVICE_STATE(Custom:DND${CHANNEL(peername)})}) same => n,GotoIf($[${CURRENT_PRESENCE}=NOT_INUSE]?*78,1:*79,1) ;DND On exten => *78,1,NoOP(Turning DND On) same => n,Set(DEVICE_STATE(Custom:DND${CHANNEL(peername)})=BUSY) same =>
2015 Jun 16
1
Variable variables
Hi Can asterisk handle asterisk variable variables? For example: If I were to set FOO300=BAR111 and I had something in a dialplan like: _3XX,1,NoOp(${FOO${EXTEN}}) And the user had entered 300, it would output BAR111 We are using asterisk 1.8 Thanks in advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: ish at
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
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 Apr 29
2
Asterisk 1.8.32.3 chan_sip deadlock
Hello asterisk-users, We've been having intermittent issues with chan_sip - it stops responding to cli requests, trying to reload chan_sip from cli doesn't seem to have any effect, initiated calls carry on for a short period, but no new SIP requests are processed ('sip show channels' hangs forever, server stops responding to SIP OPTIONS, or any other SIP messages). We have updated
2013 Nov 04
1
No matching peers message has gone (1.8.23.1)
Hi Ever since we upgraded our asterisk servers to 1.8.23.1, we no longer get the 'no matching peer' error when we get a dictionary SIP attack. Now the logs always show a 'wrong password' when there actually isn't a matching peer. We even have alwaysauthreject = yes in our sip.conf. Has anyone else noticed this phenomenon? Thanks in Advance Ish -- Ishfaq Malik
2014 Aug 13
1
Asterisk on CentOS7
Hi Is anyone using asterisk on CentOS 7? If so, is it working fine and as expected? 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 Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552
2013 Oct 15
1
PJSIP and ARA
Hi This is a bit of an exploratory question for groundwork before I start playing with asterisk 12. I've spotted the very useful looking file contrib/realtime/mysql/mysql_config.sql in the source. Are the table names starting ps_ all to do with PJSIP? Direct MySQL connection has been deprecated for quite a while, will I need to use ODBC for PJSIP or will it be supported by the old
2014 Jan 23
1
Change the preferred audio playback format
Hi Is there any way to change the preferred audio playback format in asterisk (I'm using 1.8.25.0) i.e. first check for gsm, if doesn't exits then check for slin? 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 Registered Address: PACKNET LIMITED, Duplex