[asterisk-users] WaitForSilence NEVER detects silence,,Post
I have a call server that runs on a few custom AGI scripts initiating
calls and then managing the calls. I'm getting stuck on the detecting
silence functions. I wanted to use the silence detecting as a quick
method of substituting Answering Machine Detection.
However, whenever WaitForSilence is supposed to be detecting silence, it
always just ends the interval whether or not there is actual silence. If
I'm still talking, it will consider that as silence also.
My AEL dialplan associated with the calling is:
100 = {
Answer();
WaitForSilence(5000,2,60);
AGI(agi://127.0.0.1/playmessage,${CALLID});
AGI(agi://127.0.0.1/saytext,Goodbye.);
Hangup();
}
And the CLI just outputs:
== Using SIP RTP CoS mark 5
Channel SIP/twilio-006e was answered
-- Executing [100@makeCall:1] Answer(SIP/twilio-006e, ) in
new stack
-- Executing [100@makeCall:2]
WaitForSilence(SIP/twilio-006e, 5000,2,60) in new stack
-- Waiting 2 time(s) for 5000 ms silence with 60 timeout
-- Exiting with 5000ms silence = 5000ms required
-- Exiting with 5000ms silence = 5000ms required
-- Executing [100@makeCall:3] AGI(SIP/twilio-006e,
agi://127.0.0.1/playmessage,45) in new stack
-- Playing '/var/nam/data/outgoing/60' (escape_digits=#)
(sample_offset 0)
0x7f2179cf7990 -- Probation passed - setting RTP source address
to 54.172.61.251:18920
-- Playing '/var/nam/data/tts/9eccb3f2ed77972157becdfbbac7232c'
(escape_digits=1#) (sample_offset 0)
-- SIP/twilio-006eAGI Script agi://127.0.0.1/playmessage
completed, returning 4
== Spawn extension (makeCall, 100, 3) exited non-zero on
'SIP/twilio-006e'
In my test above, it waits for 5 seconds of silence twice, but even if
I'm talking for the 5 seconds it will still just figure that I'm being
silent when I'm not.
I also tried using AMD to see if that would do a good job of detecting
an answering machine, but it thinks that everything is a MACHINE.
I know Asterisk is knowledgeable abut detecting silence because I have
another AGI script that uses the RECORD FILE command that will
successfully record somebody's voice and stop recording when there is 5
seconds of silence (which is what I set).
Is there a setting somewhere that I'm missing somewhere for a silence
threshold for WaitForSilence or am I misunderstanding its use?
The Asterisk version is
Asterisk 11.7.0~dfsg-1ubuntu1
And it's Asterisk installed from an Ubuntu package.
Thanks so much!
--
Mike A. Leonetti
As warm as green tea
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WaitForSilence NEVER detects silence
I have a call server that runs on a few custom AGI scripts initiating
calls and then managing the calls. I'm getting stuck on the detecting
silence functions. I wanted to use the silence detecting as a quick
method of substituting Answering Machine Detection.
However, whenever WaitForSilence is supposed to be detecting silence, it
always just ends the interval whether or not there is actual silence. If
I'm still talking, it will consider that as silence also.
My AEL dialplan associated with the calling is:
100 = {
Answer();
WaitForSilence(5000,2,60);
AGI(agi://127.0.0.1/playmessage,${CALLID});
AGI(agi://127.0.0.1/saytext,Goodbye.);
Hangup();
}
And the CLI just outputs:
== Using SIP RTP CoS mark 5
Channel SIP/twilio-006e was answered
-- Executing [100@makeCall:1] Answer(SIP/twilio-006e, ) in
new stack
-- Executing [100@makeCall:2]
WaitForSilence(SIP/twilio-006e, 5000,2,60) in new stack
-- Waiting 2 time(s) for 5000 ms silence with 60 timeout
-- Exiting with 5000ms silence = 5000ms required
-- Exiting with 5000ms silence = 5000ms required
-- Executing [100@makeCall:3] AGI(SIP/twilio-006e,
agi://127.0.0.1/playmessage,45) in new stack
-- Playing '/var/nam/data/outgoing/60' (escape_digits=#)
(sample_offset 0)
0x7f2179cf7990 -- Probation passed - setting RTP source address
to 54.172.61.251:18920
-- Playing '/var/nam/data/tts/9eccb3f2ed77972157becdfbbac7232c'
(escape_digits=1#) (sample_offset 0)
-- SIP/twilio-006eAGI Script agi://127.0.0.1/playmessage
completed, returning 4
== Spawn extension (makeCall, 100, 3) exited non-zero on
'SIP/twilio-006e'
In my test above, it waits for 5 seconds of silence twice, but even if
I'm talking for the 5 seconds it will still just figure that I'm being
silent when I'm not.
I also tried using AMD to see if that would do a good job of detecting
an answering machine, but it thinks that everything is a MACHINE.
I know Asterisk is knowledgeable abut detecting silence because I have
another AGI script that uses the RECORD FILE command that will
successfully record somebody's voice and stop recording when there is 5
seconds of silence (which is what I set).
Is there a setting somewhere that I'm missing somewhere for a silence
threshold for WaitForSilence or am I misunderstanding its use?
The Asterisk version is
Asterisk 11.7.0~dfsg-1ubuntu1
And it's Asterisk installed from an Ubuntu package.
Thanks so much!
--
Mike A. Leonetti
As warm as green tea
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FollowMe dials numbers but can't confirm the call or hear anything
Trying to do a FollowMe test. When the extension is dialed, it dials my cellphone and my cell phone rings. But when I answer my cell phone it's just silence. When I press '1' on my cell phone, nothing happens. extensions.conf: exten = 140,1,FollowMe(mleonetti) followme.conf [general] featuredigittimeout=5000 takecall=1 declinecall=2 call_from_prompt=followme/call-from norecording_prompt=followme/no-recording options_prompt=followme/options pls_hold_prompt=followme/pls-hold-while-try status_prompt=followme/status sorry_prompt=followme/sorry [mleonetti] context=internalphones number=xx,20 Obviously I omitted my cell phone number. I don't see anything crazy when attaching to asterisk. -- Mike A. Leonetti As warm as green tea -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sending SIP credentials in INVITE
Is it possible to have Asterisk resend the SIP credentials in every INVITE? -- Mike A. Leonetti As warm as green tea -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Contact header appears incorrect on this invite Asterisk registering with another PBX
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid = 10365)
watermelon*CLI sip show registry
Host dnsmgr Username Refresh
StateReg.Time
{broadsmart_ip}:5060 N {broadsmart_user}3317
Registered Fri, 07 May 2010 11:21:41
1 SIP registrations.
It shows that I am registered. But when I go to make a call using:
exten = 706,1,Macro(broadsmart,706)
and the Macro
[macro-broadsmart]
exten = s,1,Dial(SIP/${ar...@broadsmart,60)
Asterisk reports:
[May 7 11:34:45] WARNING[10402]: chan_sip.c:17775
handle_response_invite: Received response: Forbidden from 'Mike A.
Leonetti sip:{broadsmart_us...@broadsmart.net;tag=as6376d669'
The people on the other end sent me this e-mail:
Your registration looks all wrong. The contact header appears
incorrect on this invite. Please make it read
Contact: sip:{broadsmart_us...@{our_ip}:5060
This is probably the userid or auth user id.
REGISTER sip:{broadsmart_ip} SIP/2.0
Via: SIP/2.0/UDP {our_ip}:5060;branch=z9hG4bK1e85dd83;rport
Max-Forwards: 70
From: sip:{broadsmart_us...@{broadsmart_ip};tag=as3bafb590
To: sip:{broadsmart_us...@{broadsmart_ip}
Call-ID: 13545ba119fb96b707e90636720df...@127.0.0.1
CSeq: 102 REGISTER
User-Agent: Asterisk PBX 1.6.2.5
Expires: 120
Contact: sip:s...@{our_ip}
Content-Length: 0
Please change expires to what we are configured which is 3600 seconds.
I'm not sure what it is that may be causing the Contact to show up as s.
Here are the associated configs.
sip.conf
[general]
register = {broadsmart_user}:{broadsmart_passwo...@{broadsmart_ip}
[broadsmart]
host={broadsmart_ip}
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=rfc2833
nat=no
fromuser={broadsmart_user}
secret={broadsmart_password}
fromdomain=broadsmart.net
quality=3600
canreinvite=no
Sorry for the long request. Admittedly I'm lost.
--
Mike A. Leonetti
As warm as green tea
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Contact header appears incorrect on this invite Asterisk registering with another PBX
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid = 10365)
watermelon*CLI sip show registry
Host dnsmgr Username Refresh
StateReg.Time
{broadsmart_ip}:5060 N {broadsmart_user}3317
Registered Fri, 07 May 2010 11:21:41
1 SIP registrations.
It shows that I am registered. But when I go to make a call using:
exten = 706,1,Macro(broadsmart,706)
and the Macro
[macro-broadsmart]
exten = s,1,Dial(SIP/${ar...@broadsmart,60)
Asterisk reports:
[May 7 11:34:45] WARNING[10402]: chan_sip.c:17775
handle_response_invite: Received response: Forbidden from 'Mike A.
Leonetti sip:{broadsmart_us...@broadsmart.net;tag=as6376d669'
The people on the other end sent me this e-mail:
The register command has one set of credentials but if you are dialing
using Dial(SIP/${ar...@broadsmart,60) then the credentials will be
looked up in the [broadsmart] section within sip.conf
So is there a way to dial out using what is already registered?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Contact header appears incorrect on this invite Asterisk registering with another PBX
On 05/07/10 12:14, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid = 10365)
watermelon*CLI sip show registry
Host dnsmgr Username Refresh
StateReg.Time
{broadsmart_ip}:5060 N {broadsmart_user}3317
Registered Fri, 07 May 2010 11:21:41
1 SIP registrations.
It shows that I am registered. But when I go to make a call using:
exten = 706,1,Macro(broadsmart,706)
and the Macro
[macro-broadsmart]
exten = s,1,Dial(SIP/${ar...@broadsmart,60)
Asterisk reports:
[May 7 11:34:45] WARNING[10402]: chan_sip.c:17775
handle_response_invite: Received response: Forbidden from 'Mike A.
Leonetti sip:{broadsmart_us...@broadsmart.net;tag=as6376d669'
The people on the other end sent me this e-mail:
The register command has one set of credentials but if you are dialing
using Dial(SIP/${ar...@broadsmart,60) then the credentials will be
looked up in the [broadsmart] section within sip.conf
So is there a way to dial out using what is already registered?
No. The server you register with can often be different to the one you
pass calls to so keeping them completely separate makes a lot of sense.
You can put the authentication information in the dial command itself
but that is generally not a good idea because it can expose the username
and password to other applications which integrate into asterisk or when
viewing the asterisk console.
So then where is my mistake? The credentials in broadsmart look like
the same from whats being registered.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Contact header appears incorrect on this invite Asterisk registering with another PBX
On 05/07/10 12:40, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 12:14, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register
to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid =
10365)
watermelon*CLI sip show registry
Host dnsmgr Username Refresh
StateReg.Time
{broadsmart_ip}:5060 N {broadsmart_user}3317
Registered Fri, 07 May 2010 11:21:41
1 SIP registrations.
It shows that I am registered. But when I go to make a call using:
exten = 706,1,Macro(broadsmart,706)
and the Macro
[macro-broadsmart]
exten = s,1,Dial(SIP/${ar...@broadsmart,60)
Asterisk reports:
[May 7 11:34:45] WARNING[10402]: chan_sip.c:17775
handle_response_invite: Received response: Forbidden from 'Mike A.
Leonetti sip:{broadsmart_us...@broadsmart.net;tag=as6376d669'
The people on the other end sent me this e-mail:
The register command has one set of credentials but if you are dialing
using Dial(SIP/${ar...@broadsmart,60) then the credentials will be
looked up in the [broadsmart] section within sip.conf
So is there a way to dial out using what is already registered?
No. The server you register with can often be different to the one you
pass calls to so keeping them completely separate makes a lot of sense.
You can put the authentication information in the dial command itself
but that is generally not a good idea because it can expose the username
and password to other applications which integrate into asterisk or when
viewing the asterisk console.
So then where is my mistake? The credentials in broadsmart look like
the same from whats being registered.
I cant say but just made you aware that both are separate so the
password may be wrong in one place. It would be best to do a sip debug
and that may help diagnose the problem.
I am off now so wont be back until after the weekend so hopefully
someone else will help furthur.
It turns out that it's actually on the registration end. I see that too:
[May 7 13:02:14] NOTICE[10402]: chan_sip.c:11461 sip_reregister:--
Re-registration for {broadsmart_passwo...@{broadsmart_ip}
REGISTER 12 headers, 0
lines
Reliably Transmitting (no NAT) to
{broadsmart_ip}:5060:
REGISTER sip:{broadsmart_ip}
SIP/2.0
Via: SIP/2.0/UDP
{asterisk_ip}:5060;branch=z9hG4bK6df043c0;rport
Max-Forwards:
70
From:
sip:{broadsmart_passwo...@{broadsmart_ip};tag=as59ede08c
To:
sip:{broadsmart_passwo...@{broadsmart_ip}
Call-ID:
4fd754b9115b2e1c2c17ce6d1f24b...@127.0.0.1
CSeq: 104
REGISTER
User-Agent: Asterisk PBX
1.6.2.5
Authorization: Digest username={broadsmart_password},
realm=Registered_Subscribers, algorithm=MD5, uri=sip:broadsmart.net,
nonce=c022714eff5d7016afe930e9390392a3,
response=2e14289556acb0bf2657504c9147b6c1,
opaque=e5677a6b
Expires:
3600
Contact:
sip:s...@{asterisk_ip}
Content-Length: 0
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Contact header appears incorrect on this invite Asterisk registering with another PBX
On 05/07/10 12:40, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 12:14, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register
to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid =
10365)
watermelon*CLI sip show registry
Host dnsmgr Username Refresh
StateReg.Time
{broadsmart_ip}:5060 N {broadsmart_user}3317
Registered Fri, 07 May 2010 11:21:41
1 SIP registrations.
It shows that I am registered. But when I go to make a call using:
exten = 706,1,Macro(broadsmart,706)
and the Macro
[macro-broadsmart]
exten = s,1,Dial(SIP/${ar...@broadsmart,60)
Asterisk reports:
[May 7 11:34:45] WARNING[10402]: chan_sip.c:17775
handle_response_invite: Received response: Forbidden from 'Mike A.
Leonetti sip:{broadsmart_us...@broadsmart.net;tag=as6376d669'
The people on the other end sent me this e-mail:
The register command has one set of credentials but if you are dialing
using Dial(SIP/${ar...@broadsmart,60) then the credentials will be
looked up in the [broadsmart] section within sip.conf
So is there a way to dial out using what is already registered?
No. The server you register with can often be different to the one you
pass calls to so keeping them completely separate makes a lot of sense.
You can put the authentication information in the dial command itself
but that is generally not a good idea because it can expose the username
and password to other applications which integrate into asterisk or when
viewing the asterisk console.
So then where is my mistake? The credentials in broadsmart look like
the same from whats being registered.
I cant say but just made you aware that both are separate so the
password may be wrong in one place. It would be best to do a sip debug
and that may help diagnose the problem.
I am off now so wont be back until after the weekend so hopefully
someone else will help furthur.
I see what it is. It was the contact extension value that wasn't set.
It defaults to s. Adding a / and putting that contact extension
afterwards fixed the problem. The phones still aren't working, but
thanks for all of the help.
http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP REGISTER header not containing Allow-Events or Allow
The SIP trunking service that I am trying to set up keeps saying that my
registration from Asterisk is invalid.
Asterisk registration:
REGISTER sip:{registration_ip} SIP/2.0
Via: SIP/2.0/UDP {asterisk_ip}:5060;branch=z9hG4bK5c2eb10c;rport
Max-Forwards: 70
From: sip:{registration_us...@{registration_ip};tag=as5579cc0c
To: sip:{registration_us...@{registration_ip}
Call-ID: 651194bd76e02f4d0126373c51568...@127.0.0.1
CSeq: 104 REGISTER
User-Agent: Asterisk PBX 1.6.2.5
Authorization: Digest username={registration_user},
realm=Registered_Subscribers, algorithm=MD5, uri=sip:broadsmart.net,
nonce=b47c87b5e93ba420a0cf25162fa29794,
response=98e4d21ca0f75497d7fb12a8a4914bcb, opaque=5adc3dd2
Expires: 3600
Contact: sip:{registration_us...@{asterisk_ip}
Content-Length: 0
Expected registration:
REGISTER sip:broadsmart.net SIP/2.0
Via: SIP/2.0/UDP
208.73.25.70:5060;branch=z9hG4bK-d87543-826b1b62b62ac91d-1--d87543-;rport
Record-Route: sip:2135997...@208.73.25.70;lr
From: 2135997816 sip:2135997...@broadsmart.net;tag=e944c233
To: 2135997816 sip:2135997...@broadsmart.net
Call-ID: e0576109f9699...@dgfjd3mxlmludc5uyxrlbgnvbw0uy29t
CSeq: 1 REGISTER
Contact: sip:2135997...@208.73.25.70:5060;rinstance=92c0558ad60f5de4
Max-forwards: 70
Expires: 3600
Supported: eventlist
User-agent: CounterPath eyeBeam release 3014w stamp 26275
*Allow-Events: BroadWorksSubscriberData
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO*
Content-Length: 0
They are saying that the Asterisk registration doesn't have an
Allow-Events and an Allow in the header. Would this cause any
problems and can this be set in Asterisk to send those in the header?
Thanks.
--
Mike A. Leonetti
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] E-mails from Asterisk coming from root
Ever since the upgrade to Asterisk 1.6 the e-mails from Asterisk are coming from r...@. In the voicemail.conf I have fromstring=Asterisk PBX serveremail=asterisk And in my ssmtp.conf root=asterisk However they still come from root@ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA941 WMI not lighting up when natted
Michael Leonetti wrote: I have an a bunch of SPA941 Linksys phones for users in and out of the office. When the phones are in the office (and on the same network as the asterisk server) the WMI goes on when it should and off when it should. But when the phone is on another network and natted it fails to do so. The light always stays off. Has anybody had a similar problem (and hopefully a resolve)? Never mind. It was just me being stupid. I did not define mailbox= in sip.conf. And it's MWI for Message Waiting Indicator not WMI. I was way off on this one. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Virtual machine timing (KVM)
David Backeberg wrote: On Fri, Feb 19, 2010 at 11:42 AM, Mike A. Leonetti mleone...@evolutionce.com wrote: To get MeetMe working properly, I know some sort of timing device provided by the zaptel package is required (even if it means the zt_dummy). But, on a virtual machine I know that the Linux timing won't work as expected. Is it possible to then dedicate a physical device like a USB port or something to the virtual machine to use for the timing interrupts? You could always use ConfBridge(), starting in 1.6.2.*, which does not require DAHDI/Zaptel, and therefore doesn't require a timer. Let me be the first to tell you that using a virt for a conferencing solution, especially if you want people to actually use it, sounds like a 'Bad Idea'. You could oversubscribe the resources so you don't starve the virt, but we already have a name or that. It's called not using a virt in the first place. Well, when you're right you're right. If it's really that much of a bad idea I'll just put in for a real machine. Although virtualizing seems to be all the buzz lately so I was just wondering if I could consolidate hardware (or continue to consolidate hardware). Our internal Asterisk does currently run on KVM. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Virtual machine timing (KVM)
Sean Brady wrote: To get MeetMe working properly, I know some sort of timing device provided by the zaptel package is required (even if it means the zt_dummy). But, on a virtual machine I know that the Linux timing won't work as expected. Is it possible to then dedicate a physical device like a USB port or something to the virtual machine to use for the timing interrupts? The 2.2.1 version of DAHDI using DAHDI dummy seems to be working adequately in a Xen environment on CentOS for me, although I haven't been using MeetMe. Have you run into issues with it specifically? Which version of DAHDI are you using? If there are some issues that you have found I would like to know... Thanks, Sean To be honest I haven't tried it with Asterisk version 1.4 or higher. I only tried it with 1.2 and when the DAHDI was called Zaptel. I have been a little afraid to upgrade to 1.6 from 1.2 just in case there are some incompatibilities in my config that'll bring down the phone system here at the office for a while. The issue that I had was that the even the calls were choppy. Not even specifically just the MeetMe ones. But that was on VirtualBox. I am using KVM now. I'm not sure if that matters. What is your timer frequency set to in the kernel btw? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Virtual machine timing (KVM)
To get MeetMe working properly, I know some sort of timing device provided by the zaptel package is required (even if it means the zt_dummy). But, on a virtual machine I know that the Linux timing won't work as expected. Is it possible to then dedicate a physical device like a USB port or something to the virtual machine to use for the timing interrupts? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting up only one caller at a time
A little bit of a strange request. Basically I want all calls that go to one user go to voicemail immediately if the user is on the phone. The user is using the Linsys SPA941, and even though he can be on the phone, calls will still ring his phone. I tried disabling the rest of the lines on the phone so there was only one and yet it still rings the phone when he is on it. The call comes in through a special DID. So is there a flag to set if the user is current on the phone or if there is already a call on the DID? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting up only one caller at a time
Perfect. Thanks. Mike A. Leonetti As warm as green tea Evolution CE 3468C Lawson Boulevard Oceanside, NY 11572 www.evolutionce.com 516-536-5006 ext 105 516-208-4679 (Direct) Danny Nicholas wrote: Set his call-limit to 1 in users.conf. Other than that, you could check the channel before dialing. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike A. Leonetti Sent: Wednesday, February 17, 2010 3:07 PM To: Asterisk mailing list Subject: [asterisk-users] Setting up only one caller at a time A little bit of a strange request. Basically I want all calls that go to one user go to voicemail immediately if the user is on the phone. The user is using the Linsys SPA941, and even though he can be on the phone, calls will still ring his phone. I tried disabling the rest of the lines on the phone so there was only one and yet it still rings the phone when he is on it. The call comes in through a special DID. So is there a flag to set if the user is current on the phone or if there is already a call on the DID? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
