[Asterisk-Users] get data command and Scheduled event in 0 ms?
I'm getting messages like this: NOTICE[18552]: Scheduled event in 0 ms? Has anyone gotten these errors before? I'm using the get data command to try to receive input, and I'm sometimes getting this error message. There are also times when asterisk doesn't recognize dtmf input and instead times out. I'm not sure if they're related. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 attempts native bridge when notransfer=yes
Anyone? Now I've added T to the dialstring, and the native transfer
attempts happen much less frequently, but still ocassionally happen.
- Original Message -
From: Peter Hsu [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, July 20, 2005 6:27 PM
Subject: [Asterisk-Users] IAX2 attempts native bridge when notransfer=yes
In my iax.conf, i've set:
[general]
notransfer=yes
port=5036
tos=lowdelay
jitterbuffer=no
disallow=all
allow=gsm
allow=ulaw
allow=ilbc
allow=adpcm
allow=alaw
[VoicePulse-out]
type=peer
context=outgoing
secret=XX
username=Y
host=gwiaxt01.voicepulse.com
nat=no
notransfer=yes
[Gafachi-out]
type=peer
context=outgoing
secret=XX
username=Y
host=a862sgJjAf4eZArE.iax2.gafachi.com
nat=no
notransfer=yes
My dial string looks like:
exten =
1,1,Dial(IAX2/VoicePulse-out/1${PHONE_NUMBER}|65|tm(ring)L(360))
Asterisk keeps attempting to do these native transfers..
Any ideas what I'm doing wrong? This is driving me crazy.
Peter
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[Asterisk-Users] IAX2 attempts native bridge when notransfer=yes
In my iax.conf, i've set:
[general]
notransfer=yes
port=5036
tos=lowdelay
jitterbuffer=no
disallow=all
allow=gsm
allow=ulaw
allow=ilbc
allow=adpcm
allow=alaw
[VoicePulse-out]
type=peer
context=outgoing
secret=XX
username=Y
host=gwiaxt01.voicepulse.com
nat=no
notransfer=yes
[Gafachi-out]
type=peer
context=outgoing
secret=XX
username=Y
host=a862sgJjAf4eZArE.iax2.gafachi.com
nat=no
notransfer=yes
My dial string looks like:
exten =
1,1,Dial(IAX2/VoicePulse-out/1${PHONE_NUMBER}|65|tm(ring)L(360))
Asterisk keeps attempting to do these native transfers..
Any ideas what I'm doing wrong? This is driving me crazy.
Peter
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[Asterisk-Users] Calls going out on the same channel?
I'm having some weird behavior happening with my current configuration.
I'm running asterisk 1.0.9 and I've tried placing outbound calls using the
Originate action in the Manager API.
I'm following directions from the voip-info wiki on placing a call from an
outgoing channel.
1st action:
Action: Originate
Channel: Local/[EMAIL PROTECTED] ; This now matches
[EMAIL PROTECTED] (the appended 4 digits are just so i can debug
the channels better)
Context: default ;
The context that will
Exten: 1001
Priority: 1
2nd action:
Action: Originate
Channel: Local/[EMAIL PROTECTED] ; This now matches
[EMAIL PROTECTED] (the appended 4 digits are just so i can debug
the channels better)
Context: default ;
The context that will
Exten: 1001
Priority: 1
extensions.conf snippet:
[outgoing]
exten = _1NXXNXX+,1,NoCDR()
exten = _1NXXNXX+,2,SetVar(RECEIVER_PHONE=${EXTEN:0:11})
exten = _1NXXNXX+,3,GoTo(provider_name,s,1)
[provider_name]
exten = s,1,NoCDR()
exten = s,2,Dial(IAX2/myusername:[EMAIL PROTECTED]/${RECEIVER_PHONE}|65)
[someone_picked_up]
exten = 1001,1,Festival(${message})
exten = 1001,2,Prompt(bye)
exten = 1001,3,Hangup()
(I'm using it to send myself voice alerts, triggered by various events - in
case anyone's cares)
Everything seems fine and dandy if I generate a single call to myself. It
calls me up, I get placed into the appropriate channel, and I get the TTS
message.
However, if I've generated a second call after the first call, I'll get some
strange behavior when I receive the call.
Here's the behavior I expect:
1. I generate first Manager action
2. Asterisk box calls my cell phone
3. I pick up the call
4. I generate second Manager action
5. Asterisk box calls my cell phone
6. I get another incoming call on my cell phone (allowing me to use call
waiting to answer the other call)
However, at 6, instead of the second call coming in as another call, I get
the call waiting beeping noise on my phone, but no option for call
waiting. It just keeps beeping in the background..
When the asterisk box finishes execution of the initial originate action,
instead of my phone getting disconnected, it starts execution from
[EMAIL PROTECTED] as if I'd just picked up the phone again.
However, if I hang up call the call when I'm still in the first action, My
phone will start rining.
I'm thinking it might have something to do with my phone (treo650), but I'm
not entirely sure. This might simply be because I don't really know
asterisk very well right now. Maybe this is the intended behavior? Maybe
this is a provider thing? Maybe it's my cell phone provider service?
I'd love some insight into this so I can ensure that I'm always going out on
a unique channel,
Thanks,
Peter Hsu
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[Asterisk-Users] Validating a phone number
I'm concerned about people dialing out of our asterisk server to numbers they shouldn't be dialing. Is there a concrete algorithm for determining whether a phone number is normal. i.e. calling this phone number would result in a normal long distance rate. It seems like the pattern 1NXXNXX for the U.S. is fairly commonly used, but it wouldn't catch erroneous phone numbers such as 1411XXX (and the other X11 numbers) Is it just a matter of checking for these invalid area codes? If so, is there a list anywhere that I could check against? I tried googling this topic, but it's hard to find anything with such common keywords. If anyone can direct me to a good resource, I'd appreciate it as well. On athe same topic, I'm worried about area codes like 809. Are there any other such area codes that should be avoided? Thanks, Peter Hsu ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Manager API/variable parameter
Has anyone had experience using the Manager API with asterisk?I'm using it to originate calls, and all seems fine and dandy.However, if I set the "variable" parameter to be too long (longer than 245characters), I get the following error message:Jul 14 18:35:41 WARNING[27769]: manager.c:1211 get_input: Dumping long linewithno return from xxx.xxx.xxx.xxx: variable: where xxx.xxx.xxx.xxx is the IP of the machine accessing the manager APIand is the variable parameter string truncated to 245 characters.When this happens, none of my variables are passed in. Is there a way tochange the size limit on that parameter?Thanks,Peter Hsu ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
