search for: redialling

Anyone know how to let asterisk deal with the progress codes coming from the carrier? The problem I am having is when a customer calls an invalid number the carrier tells me the call is invalid via a progress code but doesn't route me to a recording (this number is invalid). Instead they hang up on me causing a fast busy or sometimes hold up the call with dead air for 15 to 30 seconds then a

Hello, I'm trying to figure out how to change the redial, thus far if I hit redial it will redial the last called I made that was answered, not the last call I made that was not answer. I'm using Asterisk 1.8 Thanks, Motty

Hi, What is the best way to implement Automatic Redial on No Answer ? Looking at http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples-12.txtI can see how Automatic Redial on Busy could (should) be done. How would you do it on No Answer ? Is there any event you should SUBSCRIBE to so that you're notified that you're callee is available ? What if you ask to be notified

Hello everybody, I would like to implement "callback" function. When I call a person and his extension is busy I can press, for example, 5 and get a callback when his phone is not busy anymore. When I create a call file and copy it to spool call folder asterisk makes a call. One problem is that when extension is still busy my phone rings and I get busy tone of the person who I am

I was wondering if anyone knew of a way to create a busy-redial feature in the * dialplan? For example, you try to call 12125551212 but the number is busy, so you hang up and dial *XX12125551212 and hangup again, then * would continue to retry calling the number until either it rings or a timeout is reached, if it rings * then calls back the exten that made the *XX call and bridges the two

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.

I have an issue with the Snom 360's (any firmware) and asterisk call files. When you setup a call using a call file from Asterisk and the call is connected, Asterisk will start to redial the call after about 5 minutes when the conversation is already ongoing. (Annoying and it can only be avoided by disabling call waiting) I tried to reproduce the problem with a GrandStream phone and a

Hi All, I need a softphone (PC/Mobile) which does auto redial in any case (noanswer, answer, busy, congestion etc) after a given time interval. So if the time interval was 5 secs, it would dial last number dialled after every hangup (or every failure to dial). Does anyone know such feature in a softphone? -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi

Hi, We have few systems with asterisk 1.4.22.1 and we use sip trunking for them not PRI's, one of our system is giving a problem of dtmf (rfc2833), like when we dial the number that have IVR and enter the extension or access code, it some time takes it and some times does'nt recognize the digits dialled. We also tried auto and info for dtmf but could not get the dtmf to work reliably, can

Hi list! Our CRM app is creating call files for outgoing calls which is working great I just have one problem. I am using this as my call file: Channel: SIP/228 (my phone) MaxRetries: 0 Context: from-internal (the context to dial from) Extension: 003120531234 (the phone number) Priority: 1 Callerid: Myfinecustomer <003120531234> so the external number is connected to my sip phone. However

Hi, We are experiencing this issue of redial dtmf tones generated randomly in the Voip calls, we have asterisk 1.4.21.2, dahdi 2.0.2.2 and we have dtmf as rfc 2833, we have a cisco router at the location Cisco 2431, 8FXS (only one FXS is used for Fax and rest are empy) connected to the netgear switch and all the phones are connected to this switch and there are no non sip devices in the

Hi, I would like to implement an auto-redial function in a context. The idea is about like this: Dial a number Hear busy Hangup Pick up again Dial a code like *123 => jumps into a context which redials until callresult is not busy Maybe like this: [autoredial] exten => s,1,Set(number=${CHANNEL(lastdialed)}) exten => s,2,Dial(SIP/${number}@account,60,g) exten => s,3,Wait(15) exten

I have an Asterisk 1.4.2 system installed at our office, and have a few 'on the road' sales people that want to make calls from their cell phones in transit, but they are complaining that people returning calls that they make from their cell phones are simply just using the CID that is coming from the cell phone which is causing them to get phone calls outside of business hours. What

We've just received a bill from bt where it claims that we are making numerous calls to the same number time after time. e.g. 01226xxxxxx Barnsley 20/06/2007 2115 16:00:00 01226xxxxxx Barnsley 20/06/2007 1219 08:55:32 01226xxxxxx Barnsley 21/06/2007 2115 16:00:00 01226xxxxxx Barnsley 21/06/2007 1315 08:00:00 01226xxxxxx Barnsley 21/06/2007 0515 08:00:00 01226xxxxxx Barnsley 22/06/2007 2115

Hi all, I have an external application commanding asterisk by AMI and AsyncAGI. I also have a dialplan like this: ; AsyncAGI extensions exten => _8.,1,Noop(entering in AGI loop at 8 ${EXTEN}); exten => _8.,n,AGI(agi:async); exten => _8.,n,Hangup(); ; Meetme extensions exten => _1.,1,Noop(Conference ${EXTEN} ${CONTEXT}); exten =>

I have an asterisk box and SIP / IAX2 phones. To call out, users have to add 0 (zero) before a real telephone number. That means, that if they want to call someone that has a number 123456, they have to call 0-123456. Simple, right? This has a serious drawback though - when someone calls us from the number 123456, we see the callerID 123456, and we're unable to use the callback/redial

Hi how to flush user input before READ()? I wrote a small script to ask for user password before granting access to outside, but some telefones memorize the full user input, including "#". So, when the user press redial, for instance 5556789#123, asterisk accepts the number and the password "123" and gives access to the outside word to whomever redials that terminal. Any

Hello, I have to dial prefix 9 for non local numbers however when i missed calls i Can't redial this number because of "9" is not append . I use polycom phones . What Can i do ? Harry __________________________________________________ Do You Yahoo!? En finir avec le spam? Yahoo! Mail vous offre la meilleure protection possible contre les messages non sollicit?s

Hi all, Is there a "speed dial" type application in *? The NEC PBX we currently use has a feature which allows any phone to access a system-wide speed dail database simply by keying the speed-dial number and pressing the 'redial' key from any extension. Even using a vinella phone on an sli the user can dial 77+speedial# and access this directory. Does * have a similar

Hello I have a problem where I would like to be able to send an arbitrary SIP domain when sending a call to a registered friend. By default the from domain is set to the IP of the Asterisk server, but I would like to set it to something else. The case is that when a call from a foreign domain comes in to the Asterisk, it will connect it to the callee (but with the domain changed). When