)
This will ensure you to record the customer/caller's channel instead of
exten's channel. So no matter where you transfer the call and as long as
the caller not hangup the call, it will be always recorded.
By the way, 1.2.24 stable, we got problem with 1.2.21. 1.2.17 seems
stable.
Good luck,
Isaac Xiao
WARNING
Here is our dial plan. You need to avoid double recording as well when
you transfer the call to other extension.
exten = 7141,3,Set(CALLFILENAME=q${EXTEN}-${TIMESTAMP}-${UNIQUEID})
exten = 7141,4,Set(__FROM-EXT-QUEUES=ext-queues)
exten = 7141,5,MixMonitor(${CALLFILENAME}.gsm|b)
exten =
We have the same issue happened to all Asterisk versions
of 1.2.X (I tried all). In CLI, it shows -- Incoming call: Got SIP
response 500 Internal Server Error back from 192.168.2.104. Once
you see this msg, the buddy watch wont work any more until you reboot
the phone. I also upgrade
Here is the SIP transaction log. Caller called 7176 (Cisco 7960) from outside PSTN line, 7185(polycom 601, ip: 192.168.2.104) is the phone which monitors 7176.Reliably Transmitting (no NAT) to 192.168.2.104:5060:NOTIFY sip:[EMAIL PROTECTED] SIP/2.0Via: SIP/2.0/UDP
Hi Stephen,You said that PRI works great. We are using HiPath 3550 and Siemens digital phone which using *11, *97 etc for function keys. However Asterisk uses the the * key plus one or two digits for function keys as well(it is common key combination for functions). So is it any way to
Steve,
Our Sydney and Melbourne office both use
Verizon (MCI) E1 30 channels only AU$330/month. It is quite cheap comparing to
Telstra. The Call fee is much cheaper as well.
Isaac Xiao
___
--Bandwidth and Colocation provided
bottleneck problem for 512 simultaneous
calls.
Welcome to add your comments, do test and give
feedback.
Cheers,
Isaac Xiao
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit
to
defragment, it can handle it by itself very well. Not sure how true it
is.
I am looking a solution to record expanding simultaneous calls in the
future in a call centre which accepts calls from our global branches. If
I find the good solution, I definitely post it to the community.
Cheers,
Isaac
Would any one advice how implement Diva Server BRI or
PRI card to support fax and data modem? In Eicons website, it says that they
support these. But there is no FXS port on the card, how it can be connected to
Fax machine or data Modem?
Thanks in advance.
Isaac Xiao
, autologoff is 60
Isaac Xiao
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
. But no solution for it at that time. It is really a headache for
us.
Isaac Xiao
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman
: disabled echo cancellation on channel
13
Jun 26 16:54:02 VERBOSE[8287] logger.c: -- Hungup 'Zap/13-1'
Isaac Xiao
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit
canreinvite=no
Isaac Xiao
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
) problem if I don't reboot the server more than 2 days.
Now I am trying downgrade A104d F/W from v20 to v18 (don't know what is
new in v20 and v19) and Asterisk 1.2.6 to 1.2.4. BTW, we are using a lot
of group pickup, call transfer in queue, not sure if it is cause or not.
Isaac Xiao
Can any one help? In Toronto, we can't identify if a number is long
distance call or not. If long distance call, we have to prefix with 1.
We should hear a voice prompt as above to indicate that it is not a
local call. However, we hear the normal ring back tone (indicating the
phone had been
})
Where 222, 223 and 224 are local area codes.
On 12/4/06, Isaac Xiao [EMAIL PROTECTED] wrote:
Can any one help? In Toronto, we can't identify if a number is
long
distance call or not. If long distance call, we have to prefix with
1.
We
should hear a voice prompt as above
What version of Asterisk and Zaptel you were using? Did
you try latest Asterisk 1.2.4 and Zaptel 1.2.3? Anyone has good feedback for
TE411P?
Isaac Xiao
Stagg Shelton wrote: It was Digium's opinion that perhaps the card had a VPM. We got a replacement TE411P, I implemented it tonight
Wildcard TDM2400B? But this card is very expensive. HylaFAX or SpanDSP
seems to be a solution. But how can we fax a hardcopy document with this
fax gateway?
Thanks in advance.
Isaac Xiao
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk
We are also looking for analog port for fax and dialup modem.
Yusuf, would you pls descript what stability issues you had with
TDM400P? We are thinking about using TDM400P or Voicetronix OpenPCI-8S.
Cheers,
Isaac
Can you define a LOT of pots line?
Have you considered a channel bank. Here I'm
] -- OSeqno: 002 ISeqno: 000 Type:
IAX Subclass: PING
Timestamp: 20017ms SCall:
1 DCall: 0 [198.168.2.66:4569]
Tx-Frame Retry[002] -- OSeqno: 003 ISeqno: 000 Type:
IAX Subclass: LAGRQ
Timestamp: 20020ms SCall:
1 DCall: 0 [198.168.2.66:4569]
Thanks again.
Isaac Xiao
20 matches
Mail list logo