similar to: Asterisk API

Hi - I figured this was probably the best place to ask this question Is there anyone that has done anything practical with the API and/or Real Time Database config? If so, I would like to pick your brains if I may. Thanks - G -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello there I'd like to use sip users and peers realtime. I think I done all I need to get asterisk works fine in realtime: res_odbc.conf configuration. extconfig.conf sippeers => odbc,asterisk,sipclient sipusers => odbc,asterisk,sipclient sip.conf [general] rtcachefriends=yes The sipclient table as suggest in this article: SIP Realtime, MySQL table structure (

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,

Hi, I have installed Elastix 1.5.2 (Barranquilla, Colombia (TELCO: METROTEL)) with a TE220P (2xE1) and TDM2400P (16FXS), openr2 is included in 1.5.2 version. The outcoming calls are ok, but with incoming call i have an error: ERROR*[*14972*]* chan_dahdi.c: Chan 2 - Protocol error. Reason = Multi Frequency Cycle Timeout, R2 State = Seize ACK Transmitted, MF state = Category Request Transmitted,

Hi, I face a problem which look like the same as David with his thread "Asterisk not receiving call from VPN address". I had an Asterisk (Elastix) working well in a VM (Debian Wheezy - KVM) having IP 192.168.111.14, my phone network is in the range 192.168.10.x I updated lately to 11.7.0 version and now no one of my phones can register anymore to the asterisk. Ngrep as well as

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

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

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

Hello ! In order to chase after a problem I implemented the following dialplan to have an answertime of exactly one minute: exten => xxxxxxxxxxx,1,NoOp(Test wait) exten => xxxxxxxxxxx,n,Answer exten => xxxxxxxxxxx,n,NoOp(Current timestamp: ${STRFTIME(${EPOCH},,%C%y%m%d%H%M%S)}) exten => xxxxxxxxxxx,n,Wait(60) exten => xxxxxxxxxxx,n,NoOp(Current timestamp:

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

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

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

Hi, I need to create something like this: <?xml version="1.0" encoding="UTF-8"?> <Container> <id>aQlfVHX+qPM</id> <lifetime>2009-09-19T08:14:55Z</lifetime> </Container> The response should contain the next headers: Content-Type=`application/vnd.3gpp+xml`

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,

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?

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,

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

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

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