similar to: Dial, RING with a digit interrupt

Displaying 20 results from an estimated 2000 matches similar to: "Dial, RING with a digit interrupt"

2005 Jun 12
0
*66 auto redial emulation?
Has anyone ever tried to roll out a *66 auto-callback-redial feature on asterisk? I'm sure that implementing this for outbound Zap calls would be a nightmare, but what about something easier, like internal extensions? On my old Panasonic key system, it used to be such that, if the called extensions were busy, you could press 6 while hearing the busy signal, it would beep twice and hangup.
2005 Sep 01
1
How to execute StopPlayTones when a SIP phone is answered
I'm trying to find a way to generate an 'internal extensions' tonelist but I can't seem to find anything on how to do this. My idea was to start a Playtones(intercom) tonelist and not indicate ringing to the line (dead air). But then, somehow StopPlayTones needs to be run once the ringing telephone picks up. This seems like a dirty way to do this. I envision an option to the
2005 May 27
1
Upgraded firmware on Polycom 500, digit-order problems
Ever since I upgraded my Polycom 500 to the newest sip.ld (kept the old bootrom), when I dial things like "*98" for voicemail, the screen shows "9*8" and doesn't dial my voicemail system! Is this user error, or errors in the new firmware? Chris Coulthurst chris@shuksan.com -------------- next part -------------- An HTML attachment was scrubbed... URL:
2010 Jan 26
0
StopPlayTones() after first digit?
I configured our SIP gateway to automatically dial extension "s" when a phone is picked up. I want Asterisk to play a dial tone, wait for an extension to be dialled, and hangup on timeout This works great, but I also want Asterisk to *stop* playing the dial tone after the first digit is pressed So far my extensions.conf contains, [internal] exten => s,1,Answer exten =>
2009 Jun 26
0
Problem with RetryDial
I issue this command: RetryDial(another-time,10,4,SIP/GXP280_18,10,ghM(cfmc_dial_private^RetryAndQ ueue^SIP/GXP280_18)) Asterisk rings the phone for 10 seconds. Asterisk then waits 10 seconds. Asterisk rings again for 10 seconds. I would expect this to happen a total of 4 times. The problem is that after the second ring for 10 seconds Asterisk exits the RetryDial step with HANGUPCAUSE=0 and
2005 May 20
0
Displayed CallerID on Polycom 500 shows CALLERNAME only
Get the new firmware - it's supposed to have changed the callerid display presentation to include name and number. _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Coulthurst Sent: Friday, May 20, 2005 1:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Displayed CallerID on
2005 Aug 02
1
AASTRA 480i Firmware 1.2.0.162 SIP ALERT_INFO problems
I have been playing with a 480i with the new firmware 1.2.0.162 I hope to get some form of paging intercom function to work. In the wiki someone post that ALERT_INFO type of paging might be in this version of firmware but I have been unable to find anything on this yet. I have tried sending the ALERT_INFO to the phone a number of ways with no results. I then hooked up my bt100 and tried to dial
2004 Nov 28
1
SetVar ALERT_INFO
Hello, I've got my dialplan configured to do a double ring when a customer service call comes in, and a normal ring when an extension is dialed directly. When a customer service call is transferred, I want to ring to revert back to normal. In the local extension macro, I have the following ; make sure ring is set to default exten => s,n,NoOp(${ALERT_INFO}) exten =>
2005 May 12
4
Sound card Line-In as MOH source
Does someone have a link to step-by-step instructions to making the Line-In on the console sound card a MOH source? I know this has to work somehow. Chris Coulthurst <mailto:chris@shuksan.com> chris@shuksan.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050512/4a3c3025/attachment.htm
2005 May 12
2
Voice mail - "Extension at" vs "Phone Number" OGM
Is there a way to make an outside call hear "The person at phone number XXXX is unavail", but when an internal extension calls another extension, they hear "The person at extension number XXXX is unavail"? I swear I've read this somewhere before but I'm not typing in the right search. I probably found it before by complete accident. Of course, we want the outside
2005 Sep 09
1
ALERT_INFO
A call comes in I set the distinctive ring by setting variable ALERT_INFO then dial a SIP channel. The channel is answered, but then the user forwards the call to another SIP channel. ALERT_INFO is still set. How can I clear the ALERT_INFO variable after the SIP channel is answered so that when the call is forwarded the ring goes back to "normal"?
2005 Jul 11
0
Forward the ALERT_INFO
Is asterisk able to forward it's ALERT_INFO data to another asterisk server ? My situation should look like the following: Call comes into asterisk1 in SIP. Asterisk1 sets the ALERT_INFO=Bellcore-r2, Asterisk1 dials Asterisk2 (SIP), Asterisk2 dials our SIP device which should ring with the Bellcore-r2 Any way to pass the ALERT_INFO through to the SIP device? Thanks -- Benjamin
2005 Sep 06
4
Working example of ALERT_INFO with Cisco ATAs?
I am wondering if there are any tricks getting the Cisco ATAs to do "distinctive rings" via the ALERT_INFO variable? I have seen some contradictory information in the Wiki, and I tried the example there. I then sniffed the connection between the server and the ATA and didn't see the header sent like it is "supposed" to be. If someone out there has a handle on this and
2005 May 20
1
Displayed CallerID on Polycom 500 shows CALLER NAME only
Does anyone know how to change the display format of caller id on the screen of a polycom 300/500/600? When I call FROM my 'shop phone 203' TO my 'office phone 201', a Polycom 500, it only says 'Shop' as the calling party. More specifically, the two lines look like this: Incoming call from: Shop I'm looking for a way to make it use both lines for caller id,
2005 Feb 08
11
More complicated huntgroups / delayed ringing
Stefan Gofferje wrote: > Hi Folks, > > on my home asterisk, I have a "huntgroup" for incoming calls on the > private line which first let ring my phones in my office and living > room, after a while then office, living room and bedroom. > I do this by simply putting two dial statements in sequence: > > > [private_huntgroup_day] > exten =>
2004 Dec 24
2
ALERT_INFO issue CVS-HEAD-12/24/04
Anyone having any problems with CVS-HEAD-12/24/04-15:59:15 and ALERT_INFO I have a system setup with polycom phones configured to auto answer on internal calls. When we upgraded to the latest CVS the auto answer stopped working. My dialplan has not changed. I did a sip debug and I dont see the alert-info tag in any of the sip traces. Any help would be appreciated. Thanks John Bittner Simlab.net
2005 Feb 03
3
Can't get Polycom auto-answer to work
Hi All - I'm trying to implement the auto-answer config from the wiki, but the result for me is that the phone just rings as normal. I'm running firmware version 1.4.1 on an IP500. I've added the following to my sip.cfg: <alertInfo voIpProt.SIP.alertInfo.2.value="Ring Answer" voIpProt.SIP.alertInfo.2.class="4"/> and this to my ipmid.cfg
2005 May 25
0
FAST BUSY on Back to back ZAP outgoing calls
Hello, I have a TDM400P with 2x2 configuration of FXOs and FXSs. I set a test extension of '444' to dial out a specific zap trunk and call a local #. First time I call out to '444' everything works fine. If I hang up the call, and within 10 seconds dial the same number again, I get a fast busy. Seems it isn't letting go of the trunk or something, and I don't have a
2003 Oct 10
2
ALERT_INFO=1/ Cisco 79x0
Hi, I've just found: http://lists.digium.com/pipermail/asterisk-users/2003-June/014475.html which talks about ALERT_INFO and Cisco phones. How do I actually get this working and what does it do? Do I need to add anything to the configs for the phone or is it just a SetVar(ALERT_INFO=1) - which I tried and it seemed to do nothing at all.. Thanks Andy
2007 Mar 12
1
deprecated ALERT_INFO var andAMI's Originate command
Since 1.4 ALERT_INFO variable has been deprecated. I used to send this via AMI: Action: Originate Channel: Sip/1234 Application: AgentLogin Data: 1234 Variable: _ALERT_INFO=info=alert-autoanswer Callerid: AutoLogin[1234] In order to send an autologin and autoanswer call to the agent 1234 on an Aastra phone at extension 1234. (just for example). Now in * 1.4 with ALERT_INFO deprecated I