I am after using a web crm system which has a button to then get asterisk to dial the contact. For this I was looking at call files, which appear good for the job, I have one small problem with them though. 1/ file is created 2/ external number is called 3/ the external party answers 4/ the external party now hears ringing as you extension is now being called - bad! What I would like to get round this is probably the reverse - I don't want the people I am calling to hear ringing. For example as soon as it has dialled the receiving end call me, or call me first then call the other extension? It is probably something very simple I am missing! Nick -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031222/094a92c7/attachment.htm
Hi, Why don't you turn the process around: 1/ file is created 2/ internal number is called 3/ internal party answers 4/ internal party hears ringing as the external party is being called. Ofcourse everything depends on how you have built your dialplan since you'd need to have access to an context that can do outbound dialling by number. ________________________________ 1/ file is created 2/ external number is called 3/ the external party answers 4/ the external party now hears ringing as you extension is now being called - bad! What I would like to get round this is probably the reverse - I don't want the people I am calling to hear ringing. For example as soon as it has dialled the receiving end call me, or call me first then call the other extension?
Nick Knight wrote:> I am after using a web crm system which has a button to then get > asterisk to dial the contact. For this I was looking at call files, > which appear good for the job, I have one small problem with them though. > > > > 1/ file is created > > 2/ external number is called > > 3/ the external party answers > > 4/ the external party now hears ringing as you extension is now being > called ? bad! > > > > What I would like to get round this is probably the reverse ? I don?t > want the people I am calling to hear ringing. For example as soon as it > has dialled the receiving end call me, or call me first then call the > other extension? > > > > It is probably something very simple I am missing!Swap the numbers around.
> I am after using a web crm system which has a button to then get> asterisk to dial the contact. For this I was looking at call files,> which appear good for the job, I have one small problem with themthough.>>>> 1/ file is created>> 2/ external number is called>> 3/ the external party answers>> 4/ the external party now hears ringing as you extension is now being> called - bad!>>>> What I would like to get round this is probably the reverse - I don't> want the people I am calling to hear ringing. For example as soon asit> has dialled the receiving end call me, or call me first then call the> other extension?>>>> It is probably something very simple I am missing!>Swap the numbers around.Hello again, I cannot figure this out - just swap them round? I have Channel: CAPI/<isdn number>::<number> MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: internal Extension: 101 Priority: 1 But if I swap it round Channel: SIP/User MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: internal Extension: ??? Priority: 1 I have tried various things on the extension but cannot figure out how that would - works up to that point as it calls my extension but then obviously fails when it cannot figure out what to do on the return??? Thanks again Nick -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031222/ce2829e2/attachment.htm
Hi!> > What I would like to get round this is probably the reverse ? I don?t > > want the people I am calling to hear ringing. For example as soon as it> >Swap the numbers around. > > I cannot figure this out - just swap them round? > But if I swap it round > > Channel: SIP/User > MaxRetries: 2 > RetryTime: 60 > WaitTime: 30 > > Context: internal > Extension: ??? > Priority: 1Try something like this: Context: isdn_outgoing Extension: 12345678 Priority: 1 [isdn_outgoing] exten => .,1,Dial(CAPI/<yourMSN>:${EXTEN},,rT) Cheers, Philipp
I have tried this from the manager console and call files and it doesn't seem to work the other way round. It will call the sip channel but not the capi channel - in fact with capi debug this doesn't show anything getting through Asterisk monitor comes up twith Attempting call on sip/nick ofr <number>@isdnout:1 (Retry 1) Channel Sip/nick-dd98 was answered. Then the sip call is dropped (by asterisk) then nothing! Any others ideas??? Nick -----Original Message----- From: Philipp von Klitzing [mailto:klitzing@pool.informatik.rwth-aachen.de] Sent: 22 December 2003 18:17 To: Nick Knight Cc: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re: call files Hi!> > What I would like to get round this is probably the reverse " Idon(tm)t> > want the people I am calling to hear ringing. For example as soon asit> >Swap the numbers around. > > I cannot figure this out - just swap them round? > But if I swap it round > > Channel: SIP/User > MaxRetries: 2 > RetryTime: 60 > WaitTime: 30 > > Context: internal > Extension: ??? > Priority: 1Try something like this: Context: isdn_outgoing Extension: 12345678 Priority: 1 [isdn_outgoing] exten => .,1,Dial(CAPI/<yourMSN>:${EXTEN},,rT) Cheers, Philipp
Is there a way to use the call files with the dial plan rather that directly specifying the channel and number? I have a notify application that I would like to get calls form whenever I transfer calls. -- Gary White admin@netpathway.com Network Administrator Internet Pathway 105 D East Church Street Voice: 601-776-3355 P. O. Box 777 Fax: 601-776-2314 Quitman, MS 39355 Registered Linux User Number 198875 -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3182 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041109/27571a64/smime.bin
Hello, I am having trouble with call files. I want my call files to attempt only 1 time, and never retry. I am trying to bridge two calls together, one call to my office [9726172877] and the other call to my cell [2022463521] My call file looks like this: Channel: IAX2/outgoing/19726172877 SetVar: ringtime=30 Callerid: 8668954650 MaxRetries: 0 RetryTime: 0 WaitTime: 0 Context: outgoing Extension: 12022463521 Priority: 1 Everything works properly, but I get these errors: Nov 15 23:16:04 WARNING[180235]: pbx_spool.c:156 apply_outgoing: Invalid retrytime at line 5 of /var/spool/asterisk/outgoing/2d4f5de784381a423b13480003a39a6434d63f96.call Nov 15 23:16:04 WARNING[180235]: pbx_spool.c:161 apply_outgoing: Invalid retrytime at line 6 of /var/spool/asterisk/outgoing/2d4f5de784381a423b13480003a39a6434d63f96.call and sometimes Asterisk tries to call my office again while the call is bridged. What is the proper way to accomplish this?