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How To Play Different Different Hold Music.

Dear All,

I have two server ‘A‘ and ‘B‘ . In Server ‘A‘, five different ivr (Sevices) is playing and call is *forwarding *into Server ‘B‘. Server ‘B‘ basically use for agent login(Extension). I want to play different hold music(Server ‘B‘) bases on the corresponding services which is running into server ‘A‘.

A single agent takes the call from different different services but hold music is play astrisk own by default. Is there any way to play different hold music bases on services which run into server A.

I have some changes into musiconhold.conf (server B) but problem is no solve.

please help me.

Regards

FastAGI script and DIAL execution

Hi all,

I am trying to control the whole call using a FastAGI script. To that
effect I launch a FastAGI script (written with asterisk-java).

Basically, I want to DIAL from within the FastAGI script. When the call
ends I want to control the hangup (if executed at the remote end), and
depending on the cause, dial again, play a message, or hang up. This is a
pretty standard telephony scenario. I did it before by executing the AGI,
setting variables, calling the DIAL command from the dialplan, and then
executing a second AGI script for the cleanup logic. However, now that I am
using FastAGI it seems like a better idea to keep the AGI script alive
during the duration of the call. This gives me a lot of control and
fexibility on reporting.

However, as far as I can tell, once the called party hangs up, the CDR is
generated and posted, _even though my script is still in execution_! As you
can see from the sample below, the called party hangs up, and dialplan
execution starts immediately at the h extension, even though my script is
still running. In fact, I have quite a bit of cleanup to do, adding
variables to the CDR’s, and none of them are saved! I believe this is
because the CDR is already finised.

It’s like if once you call the DIAL aplication, the dialplan forks off and
your script is running in a different place. I do not understand it. I
assumed when I called DIAL from within a script, that the script execution
would suspend, but be resumed once the DIAL command returned, but this is
not what is happening.

Is there any way to get that behaviour?

Regards,

Alex

“Entering customer extension”

IAX Trunk issue.

I’m testing a few IAX trunk scenarios in a controlled lab. From server2 extension 5000 (Server IP Address 172.16.200.212) I dial 6001 which goes across the IAX trunk to server 1 (IP address 172.16.200.210). Instead of ringing the 6001 phone, it plays tt-weasels (the s extension). When I dial 6099 it also plays tt-weasels as it’s supposed to, but it’s not the tt-weasels under its extension. It also dials the s extension.

I only placed the s extension in the dial plan to verify that the traffic was going across the IAX trunk and hitting the correct context.

Any help would be greatly appreciated.

Thanks Mitch

Asterisk-1

IP Address 172.16.200.210

SIP.CONF

[6001]
type=friend
host=dynamic
context=internal_users
secret=xxxxxxx
nat=yes

[6002]
type=friend
host=dynamic
context=internal_users
secret=xxxxxxx
nat=yes

extensions.conf

[internal_users]
exten => 6000,1,Answer()
exten => 6000,2,Playback(hello-world)
exten => 6000,3,Hangup()
exten => 6001,1,Dial(SIP/6001)
exten => 6002,1,Dial(SIP/6002)
exten => 6099,1,Playback(tt-weasels)
exten => 6099,n,HangUp
exten => _5XXX,1,Dial(${IAXTrunk}/${EXTEN})
same => n,Hangup()
exten => s,1,Answer()
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()

IAX.conf

[trunk-1]
type=friend
username=trunk-1
trunk=yes
requiretoken=no
secret=password
host=172.16.200.212
context=internal_users
auth=plaintext
disallow=all
;allow=ulaw
;allow=alaw
allow=gsm

Asterisk-2

IP Address 172.16.200.212

sip.conf

[5000]
type=friend
context=phones
host=dynamic
disallow=all
allow=ulaw
secret=xxxxxxx

extensions.conf

[phones]
exten => _60XX,1,Dial(IAX2/trunk-1)
exten => _X.,1,Dial(IAX2/trunk-1)
exten => 5000,1,Dial(SIP/${EXTEN})
exten => 5000,n,Hangup
same => n,Hangup()
exten => 5099,1,Playback(tt-monkeys)
exten => 5099,n,HangUp

iax.conf

[trunk-1]
type=friend
username=trunk-1
trunk=yes
requiretoken=no
secret=password
host=172.16.200.210
context=phones
auth=plaintext
disallow=all
;allow=ulaw
;allow=alaw
allow=gsm

ext-local and from-did-direct-ivr, how to change them?

Hi All;

Using the FreePBX, after I added the extension from the GUI, I discover that it is automatically added in the extensions_additional.conf in the context [ext-local] and [from-did-direct-ivr]

How I can change these context name? I need to determine this. How?

Regards
Bilal

Tell external number instead of internal number

Hello,

I have an internal extension, e.g. 1005 which is being called from an
external/public number like 123456789. Now when it comes to the spoken
voicemail information it says something like “number 1000 not available”,
however it should say “number 123456789 not available”. How can I configure
this? I already googled and I guess this is really easy, but I just
couldn’t figure out how to do this ): So thanks for any hint :-)

Digium IP Phones – Teleworker Capability?

We couldn’t see anything about this on the Digium site, but maybe
someone here can comment?

Do the new Digium phones provide good “teleworker” functionality?

The benchmark we’re comparing against is the capabilities of Mitel
3300 IP systems with Mitel 5330 IP phones (running their proprietary
MINET protocol), specifically:

a. A Mitel phone can be easily configured for teleworker mode (select
TW mode and the IP of the gateway server). The phone reboots and it
is ready to be used (once the Mitel border gateway is set to recognize
the unit’s ID, based on its MAC address, printed on the label on the
back of the phone). If the phone gets reallocated back to a directly
connected office environment, a simple reset procedure brings it back.

b. You can plug in the phone virtually anywhere. It has a built-in
tunnelling mechanism providing end-to-end encryption and is very
tolerant of the network configuration, routers, NAT, etc.

c. If the link between the phone and the gateway goes down, the phone
will restore itself gracefully and automatically once the network
function resumes. Absolutely hassle-free to the user.

d. Users can be configured to have hot-desk functionality. The phone
has a default extension assigned, but the user can be set up so that
they can “log in” to their normal office extension number from
wherever they are. Their office phone is automatically logged-out and
goes to its default extension when you log in to a teleworker phone
(you don’t have to log out from it first). Your phone buttons,
display settings, voicemail WMI and access, (everything) move to this
new phone, and you can work from your home office, on the road, etc.,
and inbound and outbound calls work just like you were there in the
office (callerid, etc).

These four features would be a big selling point for us to consider
moving our organization from Mitel to Digium/Asterisk/Switchvox.

How much of this can be done with Asterisk/Switchvox and, say, the
Digium D70 phone with dynamic button display?

Thanks for all comments!