[Asterisk-Users] Variable Substitution

[Justin Eckhouse]

Hi,

Can I do variable substitution in the [globals] section of extensions.conf?

For example something like this:

[globals]
EXT_BOB=4206  
PHONE_BOB=SIP/${EXT_BOB}

Thanks,
Justin

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[Asterisk-Users] Multiple Phones for 1 Extension

[Justin Eckhouse]

Hi,

I'd like to have a SIP phone at home and at the office and have them both
ring when my extension is dialed. Right now I used the same config for the
phones (Cisco 7960's). So they both register with the same login  pw. This
doesn't seem to work quiet right, where only the last phone to register
seems to get the calls.

What is the proper way to set this up?

Thanks,
Justin

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[Asterisk-Users] Cisco 7960 Transfer Call drop problem

[Justin Eckhouse]

Title: Message



Hi,

I'm having problemswith transfer from an analog line via a X100p and Cisco 
7960's running SIP.

With an 
attended transfer the a call comes in, I transfer it to another 7960, they 
answer I announce the call, press transfer again, the two parties talk for 1-2 
seconds then the analog line drops, though the Cisco phone is not aware of this, 
i.e. nothing on the screen changes. The console output for this is below. 
Interestingly enough I seem to have the same problem with an incoming SIP call, 
transferring it to another SIP ext, console output from that below as 
well.

With a blind transfer a call comes in, I 
transfer it to another extension, the analog caller hears the hold music, the 
7960 that was transferred the call acts as if it is online with the call but 
isn't. If the extension that was transferred the call puts the line on hold and 
picks it up then the lines are connected fine. 

Analog to SIP 
transfer--
 -- Zap/1-1 answered 
SIP/206-369e -- Started music on hold, class 'default', on 
Zap/1-1 -- Executing Macro("SIP/206-bcd1", 
"stdexten|SIP/202|202") in new stack -- Executing 
Dial("SIP/206-bcd1", "SIP/202|15") in new stack -- Called 
202 -- SIP/202-7264 is ringing -- 
SIP/202-7264 answered SIP/206-bcd1 -- Attempting native 
bridge of SIP/206-bcd1 and SIP/202-7264 -- Started music 
on hold, class 'default', on SIP/202-7264 -- Stopped music 
on hold on SIP/202-7264 -- Stopped music on hold on 
Zap/1-1 == Spawn extension (intern-ext, 91415XXX, 1) exited non-zero on 
'SIP/206-369e' == Spawn extension (macro-stdexten, s, 1) exited 
non-zero on 'Zap/1-1' in macro 'stdexten' == Spawn extension 
(intern-ext, 202, 1) exited non-zero on 'Zap/1-1' -- 
Hungup 'Zap/1-1'

SIP to SIP 
transfer--
 -- 
Executing Macro("SIP/206-effd", "stdexten|SIP/255|255") in new 
stack -- Executing Dial("SIP/206-effd", "SIP/255|15") in 
new stack -- Called 255 -- 
SIP/255-8cd8 is ringing -- SIP/255-8cd8 answered 
SIP/206-effd -- Attempting native bridge of SIP/206-effd 
and SIP/255-8cd8 -- Started music on hold, class 
'default', on SIP/255-8cd8 -- Executing 
Macro("SIP/206-8437", "stdexten|SIP/202|202") in new stack 
-- Executing Dial("SIP/206-8437", "SIP/202|15") in new 
stack -- Called 202 -- SIP/202-5c6b 
is ringing -- SIP/202-5c6b answered 
SIP/206-8437 -- Attempting native bridge of SIP/206-8437 
and SIP/202-5c6b -- Started music on hold, class 
'default', on SIP/202-5c6b -- Stopped music on hold on 
SIP/202-5c6b -- Stopped music on hold on 
SIP/255-8cd8 -- Attempting native bridge of SIP/206-effd 
and SIP/255-8cd8 -- Attempting native bridge of 
SIP/255-8cd8 and SIP/202-5c6b -- Got SIP response 481 
"Call Leg/Transaction Does Not Exist" back from 67.xxx.xxx.xxx == 
Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/206-effd' in 
macro 'stdexten' == Spawn extension (intern-ext, s, 1) exited non-zero 
on 'SIP/206-effd' == Spawn extension (macro-stdexten, s, 1) exited 
non-zero on 'SIP/255-8cd8' in macro 'stdexten'


Ideas?

Thanks,
Justin

[Asterisk-Users] No Sound via Sip Phone

[Justin Eckhouse]

Hi,

I just setup a box with RH 9, and latest asterisk via CVS. The box as a
T100P card in it that is currently hooked up to nothing. I did have the
sample configs in place via make samples, and the only change I made was to
add an entry to sip.conf for my Cisco 7960. When I dial 1000 to get to the
main greeting I hear nothing, though the command line output looks fine to
me.

Any ideas?

   
  -- Executing Goto(SIP/306-8509, default|s|1) in new stack
-- Goto (default,s,1)
-- Executing Wait(SIP/306-8509, 1) in new stack
-- Executing Answer(SIP/306-8509, ) in new stack
-- Executing DigitTimeout(SIP/306-8509, 5) in new stack
-- Set Digit Timeout to 5
-- Executing ResponseTimeout(SIP/306-8509, 10) in new stack
-- Set Response Timeout to 10
-- Executing BackGround(SIP/306-8509, demo-congrats) in new stack
-- Playing 'demo-congrats'
  == Spawn extension (default, s, 5) exited non-zero on 'SIP/306-8509'

Thanks,
Justin

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[Asterisk-Users] H.323 Gateway Connection

[Justin Eckhouse]

Hi,

I'm trying to setup Asterisk to allow users to dial out to the PSTN using a
remote box supporting h.323. I'm using chan_h323.so, and I'm able to make
outbound calls to a client like netmeeting with a line like this:

exten = 242,1,Dial(h323/xxx.xxx.xxx.xxx)

And I'm able to receive incoming calls to asterisk. However I'm not sure how
to route calls to the remote h.323 gateway. In my naïve state I've tried
something like this (xxx is the IP of the h.323 gw): 

exten = 244,1,Dial(h323/xxx.xxx.xxx.xxx/PSTN-NUMBER-HERE)

When I dial 244, nothing happens, this appears in the console:

-- Called xxx.xxx.xxx.xxx
  == No one is available to answer at this time

Any pointers in the right direction would be greatly appreciated.

Thanks,
Justin


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