Re: [Asterisk-Users] Release phone call

2004-02-05 Thread Glenn Dalgliesh
Title: Message



Well after thinking a little more about your 
scenario I have had situations where I have hairpinned calls back to the device 
they have come in on and in my experience without canreinvite=no the device will 
usually complain that a loop was detected. In situations were I am 
using SER(SIP EXPRESS ROUTER) I have fixed the problem by 
rewriting the headed to trick the device out of the loop detect 
scenario. Anyway in theory if the 7206 doesn't detect a loop if you have 
the sip.conf setup to the canreinvite=yes then the audio channel should not be 
proxied by *. 
 
In this calling multiple channels how do you deal 
with the cellphone voicemail not answering the call immediately if the phone is 
off or out of range which wouldn't allow for the opportunity for the other 
extension to be answered. 

  - Original Message - 
  From: 
  B. J. Bomar 
  
  To: [EMAIL PROTECTED] 
  
  Sent: Thursday, February 05, 2004 1:37 
  PM
  Subject: RE: [Asterisk-Users] Release 
  phone call
  
  The 
  way we have it setup is simply calling multiple numbers/channels.  It is 
  either setup manually in the configs, or through a very ugly menu interface I 
  constructed.
   
  B. 
  J.
   
   
   
   
  

-Original Message-From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Glenn 
DalglieshSent: Thursday, February 05, 2004 12:31To: 
[EMAIL PROTECTED]Subject: Re: [Asterisk-Users] 
    Release phone call
I don't really have a answer for you on you 
issue but have a question about what "find-me" is. I see it on the feature 
list but am unable to find any real information about it. Is this simply 
call forward or is their more to it. 
 
thanks

  - Original Message - 
  From: 
  B. J. 
  Bomar 
  To: [EMAIL PROTECTED] 
  
  Sent: Thursday, February 05, 2004 
  1:01 PM
  Subject: [Asterisk-Users] Release 
  phone call
  
  Hello all, I 
  am trying to figure out how to have * release a phone call.  We are 
  noticing some call quality issues on people who have a "find-me" feature, 
  and answer the call through a cell phone.  Here is the call path we 
  are seeing, and all VoIP connections are using SIP.
   
  PSTN ---> 
  Cisco 7206 ---> * Server
   ^---|   
  ^-|
   
  Hopefully the 
  diagram makes sense, but in case it doesn't, let me try to explain.  
  A call comes in from PSTN into our Cisco7206 with PRI card.  It then 
  goes to our * server, which then forwards the call back through the Cisco 
  to a cell phone on PSTN.  I am wanting to have * release the call to 
  the Cisco once the call is connected.  Any thoughts or 
  ideas?
   
  Thanks.
   
  B. 
  J.
   
   
   
   


RE: [Asterisk-Users] Release phone call

2004-02-05 Thread B. J. Bomar
Title: Message



The 
way we have it setup is simply calling multiple numbers/channels.  It is 
either setup manually in the configs, or through a very ugly menu interface I 
constructed.
 
B. 
J.
 
 
 
 

  
  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Glenn 
  DalglieshSent: Thursday, February 05, 2004 12:31To: 
  [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] 
  Release phone call
  I don't really have a answer for you on you issue 
  but have a question about what "find-me" is. I see it on the feature list but 
  am unable to find any real information about it. Is this simply call forward 
  or is their more to it. 
   
  thanks
  
- Original Message - 
From: 
B. J. Bomar 

To: [EMAIL PROTECTED] 

Sent: Thursday, February 05, 2004 1:01 
PM
Subject: [Asterisk-Users] Release phone 
call

Hello all, I am 
trying to figure out how to have * release a phone call.  We are 
noticing some call quality issues on people who have a "find-me" feature, 
and answer the call through a cell phone.  Here is the call path we are 
seeing, and all VoIP connections are using SIP.
 
PSTN ---> 
Cisco 7206 ---> * Server
 ^---|   
^-|
 
Hopefully the 
diagram makes sense, but in case it doesn't, let me try to explain.  A 
call comes in from PSTN into our Cisco7206 with PRI card.  It then goes 
to our * server, which then forwards the call back through the Cisco to a 
cell phone on PSTN.  I am wanting to have * release the call to the 
Cisco once the call is connected.  Any thoughts or 
ideas?
 
Thanks.
 
B. 
J.
 
 
 
 


Re: [Asterisk-Users] Release phone call

2004-02-05 Thread Glenn Dalgliesh
Title: Message



I don't really have a answer for you on you issue 
but have a question about what "find-me" is. I see it on the feature list but am 
unable to find any real information about it. Is this simply call forward or is 
their more to it. 
 
thanks

  - Original Message - 
  From: 
  B. J. Bomar 
  
  To: [EMAIL PROTECTED] 
  
  Sent: Thursday, February 05, 2004 1:01 
  PM
  Subject: [Asterisk-Users] Release phone 
  call
  
  Hello all, I am 
  trying to figure out how to have * release a phone call.  We are noticing 
  some call quality issues on people who have a "find-me" feature, and answer 
  the call through a cell phone.  Here is the call path we are seeing, and 
  all VoIP connections are using SIP.
   
  PSTN ---> Cisco 
  7206 ---> * Server
   ^---|   
  ^-|
   
  Hopefully the 
  diagram makes sense, but in case it doesn't, let me try to explain.  A 
  call comes in from PSTN into our Cisco7206 with PRI card.  It then goes 
  to our * server, which then forwards the call back through the Cisco to a cell 
  phone on PSTN.  I am wanting to have * release the call to the Cisco once 
  the call is connected.  Any thoughts or ideas?
   
  Thanks.
   
  B. 
  J.