Re: [asterisk-users] Outbound Call Transfer Problem

2007-02-12 Thread Ioan Indreias

Hello,

I see that you are using T option (allow the /calling/ user to 
transfer the call) when dialling to internal extensions and t (allow 
the /called/ user to transfer the call) when receiving calls (in home 
context). This it is why inbound transfer works fine and only one time.


So, I suggest to add t option to the Dial lines you need the transfer 
feature to be active for the called party (eg. in from-sip context).


Best regards,
## nini @ www.modulo.ro ##



Nikhil Jogia wrote:

Noah Miller wrote:

I am using Asterisk 1.2 and for the life of me, I am unable to transfer
outbound calls (eg calls I initiate from sip extensions). When I press
#, nothing happens. Inbound calls transfer fine, but only once per 
call.


Any suggestions?


I have questions:

1) what version of 1.2?


version 1.2.1

2) Anything come up in the CLI?  How about the logs?  Have you tried
turning on verbose logging in logger.conf? (be sure to turn it off
when you're done)


nothing at all.

3) What SIP phones are you using (hard and soft)?


sipura 2000 and sjphone.


- Noah
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Re: [asterisk-users] Outbound Call Transfer Problem

2007-02-09 Thread Noah Miller

I am using Asterisk 1.2 and for the life of me, I am unable to transfer
outbound calls (eg calls I initiate from sip extensions). When I press
#, nothing happens. Inbound calls transfer fine, but only once per call.

Any suggestions?


I have questions:

1) what version of 1.2?

2) Anything come up in the CLI?  How about the logs?  Have you tried
turning on verbose logging in logger.conf? (be sure to turn it off
when you're done)

3) What SIP phones are you using (hard and soft)?


- Noah
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Re: [asterisk-users] Outbound Call Transfer Problem

2007-02-09 Thread Nikhil Jogia

Noah Miller wrote:

I am using Asterisk 1.2 and for the life of me, I am unable to transfer
outbound calls (eg calls I initiate from sip extensions). When I press
#, nothing happens. Inbound calls transfer fine, but only once per call.

Any suggestions?


I have questions:

1) what version of 1.2?


version 1.2.1

2) Anything come up in the CLI?  How about the logs?  Have you tried
turning on verbose logging in logger.conf? (be sure to turn it off
when you're done)


nothing at all.

3) What SIP phones are you using (hard and soft)?


sipura 2000 and sjphone.


- Noah
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