Re: [asterisk-users] AGI Delimiter in 1.6

[James A. Shigley]

So why can't you send the Auth line into the variable and then have your script 
do the parsing to break out the segments you want. 

Or if need be two scripts. The first can accept the authline as a full string 
from a variable and break it down to its parts and save those as channel 
variables. Then your second agi script which is basically the one that worked 
in 1.2/1.4 can use the channel variables.

James Shigley


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jon Farmer
Sent: Thursday, September 16, 2010 2:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] AGI Delimiter in 1.6

> On 16 September 2010 19:50, Danny Nicholas  wrote:

> If you make the string into a dialplan Variable, you can do pretty much
> anything with it.  Let's say your dialplan is like this
>
> - exten => 1234,1,blah
> - exten => 1234,n,AGI(myagi.xx,"1234")
>
> Change line 2 to
> - exten => 1234,n,AGI(myagi.xx,${VARNAME})
>
> Then you just "do your magic" on ${VARNAME}


Yes, but the problem is I am trying to pass the whole AUTH line which
is key=value pairs seperated by commas. e.g. username=myusername,
domain=mydomain

This breaks when passing to an AGI in 1.6.



-- 
Jon Farmer
Tel 07795 118140

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Re: [asterisk-users] Asterisk Queue + Caller ID issue

[James A. Shigley]

Could you give me an example because I understand what you said, but not
sure what to put in my extensions.conf to accomplish that.

James Shigley



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Monday, July 19, 2010 11:14 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk Queue + Caller ID issue

On 07/19/2010 11:08 AM, James A. Shigley wrote:
>  
> 
> Let me rephrase this question.
> 
>  
> 
> What context does a queue use for dialing out?

It doesn't, it dials the member directly. If you need it to dial out
through the dialplan, add a Local channel as a member, instead of the
actual channel, and then do your logic in the context/extension you
specified before performing the actual dial operation.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

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[asterisk-users] Asterisk Queue + Caller ID issue

[James A. Shigley]

 

Let me rephrase this question.

 

What context does a queue use for dialing out?

 

James Shigley

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: Monday, July 19, 2010 7:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk Queue + Caller ID issue

 

 

 

 

Ok I have a queue that is working perfectly. 

 

The problem is when one of the agents is outside the building on an
external phone line (say a cell phone). My telco hangs up on the call .
I think the telco is hanging up on these calls because there is no CID
attached. (I know my telco wont connect calls without ANI, so that is
what it is my assumption)

 

So first I need to prove my assumption is right. How can I check if
those calls are being sent with caller ID. Because all I see on console
output for the phone call is this

 

 

-- Channel 0/8, span 3 got hangup, cause 50 (sometimes cause 1 or 27
instead)

-- Nobody picked up in 1000 ms

-- Hungup 'DAHDI/56-1'

 

It doesn't show where it actually tried to dial or not. I know it works
because if I sent it to the in house number it calls that number and if
someone answers it they get the person who is on hold in the queue. It
only fails on outside the building calls.

 

So where do I check to see if it is or isn't attaching caller ID.

 

Let's assume I'm right and the CID is the issue; What config and/or
context do I need to change so that the  when a queue tries to place a
call to an agent there is caller ID?

 

 

James Shigley

 

 

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[asterisk-users] Asterisk Queue + Caller ID issue

[James A. Shigley]

 

 

 

Ok I have a queue that is working perfectly. 

 

The problem is when one of the agents is outside the building on an
external phone line (say a cell phone). My telco hangs up on the call .
I think the telco is hanging up on these calls because there is no CID
attached. (I know my telco wont connect calls without ANI, so that is
what it is my assumption)

 

So first I need to prove my assumption is right. How can I check if
those calls are being sent with caller ID. Because all I see on console
output for the phone call is this

 

 

-- Channel 0/8, span 3 got hangup, cause 50 (sometimes cause 1 or 27
instead)

-- Nobody picked up in 1000 ms

-- Hungup 'DAHDI/56-1'

 

It doesn't show where it actually tried to dial or not. I know it works
because if I sent it to the in house number it calls that number and if
someone answers it they get the person who is on hold in the queue. It
only fails on outside the building calls.

 

So where do I check to see if it is or isn't attaching caller ID.

 

Let's assume I'm right and the CID is the issue; What config and/or
context do I need to change so that the  when a queue tries to place a
call to an agent there is caller ID?

 

 

James Shigley

 

 

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[asterisk-users] (no subject)

[James A. Shigley]

Ok I have a queue that is working perfectly. 

 

The problem is when one of the agents is outside the building on an
external phone line (say a cell phone). My telco hangs up on the call .
I think the telco is hanging up on these calls because there is no CID
attached. (I know my telco wont connect calls without ANI, so that is
what it is my assumption)

 

So first I need to prove my assumption is right. How can I check if
those calls are being sent with caller ID. Because all I see on console
output for the phone call is this

 

 

-- Channel 0/8, span 3 got hangup, cause 50 (sometimes cause 1 or 27
instead)

-- Nobody picked up in 1000 ms

-- Hungup 'DAHDI/56-1'

 

It doesn't show where it actually tried to dial or not. I know it works
because if I sent it to the in house number it calls that number and if
someone answers it they get the person who is on hold in the queue. It
only fails on outside the building calls.

 

So where do I check to see if it is or isn't attaching caller ID.

 

Let's assume I'm right and the CID is the issue; What config and/or
context do I need to change so that the  when a queue tries to place a
call to an agent there is caller ID?

 

 

James Shigley

 

 

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[asterisk-users] Question

[James A. Shigley]

Ok so a while back I found an example for having a number dial multiple
numbers and then whoever answers and confirms gets the call. (don't
recall who the example was from, but thank you!)

 

But Now today I've been playing with TTS and STT and came across the
BackgroundDetect command. Now If I use this allow it works fine. But
when I try and use it with this it never actually detects me talking -
or if it does it doesn't connect the caller so that the Wait time
expires and it goes on.

 

 

So my question is how can I make this work to where you can talk and it
will connect you to the caller or press 1. Not now where you just press
1. Which a lot of the time I can't get my phone out of my pocket,
unlocked, and press 1 before it is sent to VM

 

[default]

exten =>
_XX,1,Monitor(wav,/var/store/calls/PersonalLine-${STRFTIME($
{EPOCH},,%Y%m%d-%H%M%S)}-${CALLERID(num)}-${EXTEN},mb)  

exten => _XX,2,dial(${bellbu}/${EXTEN:4},40,rM(screen)) ;
without r it seems to pass a second or two of audio first 

exten => _XX,4,Hangup ; You can also substitute this with a
Voicemail destination or other alternative destination 

 

[macro-screen] 

;exten => s,1,Wait(1) 

;exten => s,n,Background(/var/lib/asterisk/sounds/press1) ; substitute a
different playback file if you need to 

;exten => s,n,WaitExten(5) ; the value is the Wait time before we assume
the call is not accepted 

;exten => 1,1,NoOp(Caller accepted) ; Do not set MACRO_RESULT to
anything to connect the caller 

;exten => i,1,Set(MACRO_RESULT=CONTINUE) 

;exten => t,1,Set(MACRO_RESULT=CONTINUE) 

 

exten => s,1,Wait(1) 

exten => s,n,BackgroundDetect(/var/lib/asterisk/sounds/press1) 

exten => s,n,WaitExten(10) ; the value is the Wait time before we assume
the call is not accepted 

exten => 1,1,NoOp(Caller accepted) ; Do not set MACRO_RESULT to anything
to connect the caller 

exten => i,1,Set(MACRO_RESULT=CONTINUE) 

exten => t,1,Set(MACRO_RESULT=CONTINUE) 

 

exten => talk,1,NoOp(Caller accepted)

 

 

[Inbound]

exten => 4095551212,1,NoOP() 

exten => 4095551212,n,Dial(LOCAL/111222&LOCAL/222333,40)


exten => 4095551212,n,Voicemail(1...@default)

 

 

 

James Shigley

 

 

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Re: [asterisk-users] Semi-Transfer

[James A. Shigley]

I've tried that as well prior to sending the initial email with no
results.

 

I'll play some with DISA today.

 

James Shigley

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Tuesday, February 02, 2010 2:50 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Semi-Transfer

 

This wiki is outdated but the group stuff still applies to DAHDI

http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels

 

Assuming that you have many available lines in group 3, changing the
option to g3 from G3 might help.

 



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: Tuesday, February 02, 2010 2:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Semi-Transfer

 

That is the PRI span there are many available lines.

 

James Shigley

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Tuesday, February 02, 2010 2:12 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Semi-Transfer

 

What lines are in your "group 3"?  It is possible that DAHDI/52 is the
only line in that group and that's why you're getting the "all
congested".

 



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: Tuesday, February 02, 2010 2:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Semi-Transfer

 

There are times when I need to call a client from my cell and I want a
recording of the call. I was trying to put into an * a way of doing
that. Below is what I'm using in my extensions.conf

 

exten=> X,1,Read(num,"/var/lib/asterisk/sounds/mtas/10digit",10,,,5)

exten=> X,2,SayDigits(${num}) 

exten=> X,3,Background(/var/lib/asterisk/sounds/mtas/verify)

exten=> X,4,WaitExten(3)

exten=>
X,5,Monitor(wav,/var/store/calls/InOutRec-${STRFTIME(${EPOCH},,%Y%m%d-%H
%M%S)}-${CALLERID(num)}-${EXTEN},mb)  

exten=> X,6,dial(${belltd}/${num})

 

 

Here is what I see in the CMD when the dial fails

 

-- Timeout on DAHDI/52-1, continuing...

-- Executing [xxx...@recout:5] Monitor("DAHDI/52-1",
"wav,/var/store/calls/InOutRec-20100202-133012-XX-XX,mb"
) in new stack

-- Executing [XX @RecOut:6] Dial("DAHDI/52-1",
"DAHDI/G3/4099819750") in new stack

-- Requested transfer capability: 0x00 - SPEECH

-- Called G3/4099819750

== Everyone is busy/congested at this time (1:0/0/1)

-- Auto fallthrough, channel 'DAHDI/52-1' status is 'CHANUNAVAIL'

-- Hungup 'DAHDI/52-1'

 

 

Now all of my lines are NOT ties up so there is available paths for the
call to go out.

 

Anyway so how would I accomplish this transfer of sorts?

James Shigley

Monroe Telephone Answering Service

409-981-9750

Infinity 5.51,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps, 

 

CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information =is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by "reply to sender only" message and destroy all
electronic and hard copies of the communication, including attachments. 

 

 

Side Note: I am James, Jim is my future father in law!

 

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Re: [asterisk-users] Semi-Transfer

[James A. Shigley]

That is the PRI span there are many available lines.

 

James Shigley

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Tuesday, February 02, 2010 2:12 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Semi-Transfer

 

What lines are in your "group 3"?  It is possible that DAHDI/52 is the
only line in that group and that's why you're getting the "all
congested".

 



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: Tuesday, February 02, 2010 2:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Semi-Transfer

 

There are times when I need to call a client from my cell and I want a
recording of the call. I was trying to put into an * a way of doing
that. Below is what I'm using in my extensions.conf

 

exten=> X,1,Read(num,"/var/lib/asterisk/sounds/mtas/10digit",10,,,5)

exten=> X,2,SayDigits(${num}) 

exten=> X,3,Background(/var/lib/asterisk/sounds/mtas/verify)

exten=> X,4,WaitExten(3)

exten=>
X,5,Monitor(wav,/var/store/calls/InOutRec-${STRFTIME(${EPOCH},,%Y%m%d-%H
%M%S)}-${CALLERID(num)}-${EXTEN},mb)  

exten=> X,6,dial(${belltd}/${num})

 

 

Here is what I see in the CMD when the dial fails

 

-- Timeout on DAHDI/52-1, continuing...

-- Executing [xxx...@recout:5] Monitor("DAHDI/52-1",
"wav,/var/store/calls/InOutRec-20100202-133012-XX-XX,mb"
) in new stack

-- Executing [XX @RecOut:6] Dial("DAHDI/52-1",
"DAHDI/G3/4099819750") in new stack

-- Requested transfer capability: 0x00 - SPEECH

-- Called G3/4099819750

== Everyone is busy/congested at this time (1:0/0/1)

-- Auto fallthrough, channel 'DAHDI/52-1' status is 'CHANUNAVAIL'

-- Hungup 'DAHDI/52-1'

 

 

Now all of my lines are NOT ties up so there is available paths for the
call to go out.

 

Anyway so how would I accomplish this transfer of sorts?

James Shigley

Monroe Telephone Answering Service

409-981-9750

Infinity 5.51,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps, 

 

CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information =is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by "reply to sender only" message and destroy all
electronic and hard copies of the communication, including attachments. 

 

 

Side Note: I am James, Jim is my future father in law!

 

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[asterisk-users] Semi-Transfer

[James A. Shigley]

There are times when I need to call a client from my cell and I want a
recording of the call. I was trying to put into an * a way of doing
that. Below is what I'm using in my extensions.conf

 

exten=> X,1,Read(num,"/var/lib/asterisk/sounds/mtas/10digit",10,,,5)

exten=> X,2,SayDigits(${num}) 

exten=> X,3,Background(/var/lib/asterisk/sounds/mtas/verify)

exten=> X,4,WaitExten(3)

exten=>
X,5,Monitor(wav,/var/store/calls/InOutRec-${STRFTIME(${EPOCH},,%Y%m%d-%H
%M%S)}-${CALLERID(num)}-${EXTEN},mb)  

exten=> X,6,dial(${belltd}/${num})

 

 

Here is what I see in the CMD when the dial fails

 

-- Timeout on DAHDI/52-1, continuing...

-- Executing [xxx...@recout:5] Monitor("DAHDI/52-1",
"wav,/var/store/calls/InOutRec-20100202-133012-XX-XX,mb"
) in new stack

-- Executing [XX @RecOut:6] Dial("DAHDI/52-1",
"DAHDI/G3/4099819750") in new stack

-- Requested transfer capability: 0x00 - SPEECH

-- Called G3/4099819750

== Everyone is busy/congested at this time (1:0/0/1)

-- Auto fallthrough, channel 'DAHDI/52-1' status is 'CHANUNAVAIL'

-- Hungup 'DAHDI/52-1'

 

 

Now all of my lines are NOT ties up so there is available paths for the
call to go out.

 

Anyway so how would I accomplish this transfer of sorts?

James Shigley

Monroe Telephone Answering Service

409-981-9750

Infinity 5.51,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps, 

 

CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information =is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by "reply to sender only" message and destroy all
electronic and hard copies of the communication, including attachments. 

 

 

Side Note: I am James, Jim is my future father in law!

 

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Re: [asterisk-users] Really Silly Question From Total Newbie

[James A. Shigley]

I can't help you two much with configuration of linux, but as to the call 
question. You will need some route for the server to be capable of 
sending/receiving calls. There is a couple of ways to do this cheaply. 

Buy a standard telephone modem (usb, pci, or serial). And plug into wall a 
jack. This will only allow you one call at a time. But if this is just a proof 
of concept that sounds like it will be plenty. Your number would be whatever 
the phone jack corresponds to.

Integrate it with something like skype.

Buy one of the products similar to a magic jack which will work with linux.

Or if you know someone who has a sip server already running. And is amiable to 
letting you piggy back off of it. Your  phone number would point to that 
person's Server (be it one of the friends that the person is lending you, or a 
number that you have forwarded/ported at/to them) with it set to forward to 
your asterisk server. from there you could use your dialplan to do whatever you 
wanted it to. And for outbound you would send the calls out thru the friends 
server via sip or iax.

James Shigley
Monroe Telephone Answering Service
409-981-9213
Infinity 5.51,UC 4.02.3803, Blink 3.0.104
Ecreator:2.21, eResponse 1.1.7
Webportal,WebApps, 
 
CONFIDENTIALITY NOTICE: This email, including any attachments, contains 
information which may be confidential or privileged. The information =is 
intended to be for the use of the individual or entity named above. If you are 
not the intended recipient, be aware that any disclosure, copying, distribution 
or use of the contents of this information is prohibited. If you have received 
this email in error, please notify the sender immediately by "reply to sender 
only" message and destroy all electronic and hard copies of the communication, 
including attachments. 

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of UIT DEVELOPMENT
Sent: Tuesday, January 05, 2010 1:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Really Silly Question From Total Newbie

.

My main question is: CAN I make call from that box to my cell phone
using a soft-phone?   If so, how can I do that?   Also, can I use my
cell phone to call into that box?   I dont know if I have to get a
phone number, or do I NEED a phone number?   At the moment, I do not
have any dollars to throw at this project.   Its purely for learning,
proof of concept sort of thing for myself on my spare time in the
evenings.  I would simply like to be able to call out and be able to
call into that box.  Later on down the road maybe I will get into
setting up an IVR using a database so I can call into that system from
wherever and get information read back to me.  But, first things
first  I'd like to know if I am heading down the wrong path here.
...

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Re: [asterisk-users] SIP Issue

[James A. Shigley]

What do you mean I should use a global function. I'm kind both well versed and 
a newb to *

James Shigley




-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan E. Rodríguez
Sent: Monday, December 28, 2009 12:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP Issue

Is ddwhome defined in global context?? If so, then you should use global 
function.

Paste asterisk log to check.
Saludos,
Juan E. Rodríguez


-Original Message-
From: "James A. Shigley" 
Date: Mon, 28 Dec 2009 12:11:35 
To: Asterisk Users Mailing List - Non-Commercial 
Discussion
Subject: [asterisk-users] SIP Issue

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[asterisk-users] SIP Issue

[James A. Shigley]

Alright I have a SIP phone located off premises with a very annoying
issue.

 

Well I say a sip phone it is actually two phones hooked to a Cisco Spa
2102 

Link: http://www.cisco.com/en/US/products/ps10026/index.html

 

Each phone being a different line/extension.

 

Alright either line can ALWAYS make outbound calls no issue. The problem
is on the Inbound side. I'm completely stumped as to why. I could make
10 back to back out bound calls and then call inbound and watch the call
come in to * and try to be passed to the sip phone only to get "Error
Message 14: Not a Working Number." So it doesn't seem to be a matter of
the phones Sip Login "Timing out"

 

And when I check sip peers it shows the correct IP address of the box so
it doesn't appear to be that it can't find the Cisco box.

 

Here is what I use for the inbound context, replacing the _X_ with the
actual extension of course.

 

[to_ddwhome]

exten=> _X_,1,wait(1)

exten=> _X_,n,Dial(${ddwhome},21)

exten=> _X_,n,Goto(dial_inf,${EXTEN},1)

 

${ddwhome}=SIP/ddwhome

 

Now the odd thing is when it gets the Error 14 message then the third
step to dial_inf does not execute. Though when it rarely does connect
with the sip phone if no one answers in 21 seconds than it will roll
over to that step.

 

Any ideas?

 

James Shigley

Monroe Telephone Answering Service

409-981-9213

Infinity 5.51,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps, 

 

CONFIDENTIALITY NOTICE: This email, including any attachments, contains
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electronic and hard copies of the communication, including attachments. 

 

 

 

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Re: [asterisk-users] G729 Pass through

[James A. Shigley]

Have you paied for and imported g729 licenses from digium so that
asterisks can use g729?

 

http://store.digium.com/productview.php?category_id=5&product_code=8G729
CODEC 

 

James Shigley

Monroe Telephone Answering Service

409-981-9213

Infinity 5.51,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps, 

 

CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information =is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by "reply to sender only" message and destroy all
electronic and hard copies of the communication, including attachments. 

 

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovey
Forman
Sent: Friday, December 11, 2009 12:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] G729 Pass through
Importance: High

 

Hi;

 

I am running Asterisk (Trix) 1.2.17 with Aastra 6731i phones as my
endpoints.

 

It seems that when I enable G729 on my peers in sip.conf and make a call
I am getting the following errors:

 

Called crp_uk/806575011971553141421

Dec 11 07:57:10 WARNING[31903] channel.c: Unable to find a codec
translation path from g729 to ulaw

Dec 11 07:57:10 WARNING[20633] chan_sip.c: Asked to transmit frame type
256, while native formats is 4 (read/write = 4/4)

 

Both my end points (Aastra phone) and my sip carrier support G729, so
this should be simple pass-through.

 

Snippet of my peer crp_uk:

 

[crp_uk]

disallow=all

allow=ulaw

allow=alaw

allow=g729

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Re: [asterisk-users] IAX2 Port issue

[James A. Shigley]

192.168.16.3 is my desk
  17.140 is *

192.168.16.0/21 is the subnet (255.255.248.0)

Firewall isn't an issue here, that I can see for sure.

James Shigley
Monroe Telephone Answering Service
409-981-9213
Infinity 5.51,UC 4.02.3803, Blink 3.0.104
Ecreator:2.21, eResponse 1.1.7
Webportal,WebApps, 
 
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including attachments. 




-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes
Sent: Friday, December 04, 2009 10:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX2 Port issue


On 4 Dec 2009, at 16:37, James A. Shigley wrote:
> egg*CLI> iax2 reload
>   == Parsing '/etc/asterisk/iax.conf':   == Found
>   == Parsing '/etc/asterisk/users.conf':   == Found
> [Dec  4 10:17:36] NOTICE[6080]: chan_iax2.c:11087 set_config:  
> Ignoring bindport on reload
> [Dec  4 10:17:36] NOTICE[6080]: chan_iax2.c:11148 set_config:  
> Ignoring bindaddr on reload
> [Dec  4 10:17:36] NOTICE[6080]: chan_iax2.c:11148 set_config:  
> Ignoring bindaddr on reload
> [Dec  4 10:17:36] NOTICE[6080]: chan_iax2.c:11148 set_config:  
> Ignoring bindaddr on reload
>   == Loaded firmware 'iaxy.bin'
> egg*CLI>
>

Its a notice rather than a warning. I doubt thats your problem.

Wireshark doesn't tell us enough, What are your network addresses? You  
at least need to tell us which end is which, and your subnet etc.  
Firewall info good too.

Steve

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[asterisk-users] IAX2 Port issue

[James A. Shigley]

 

Trying to configure IAX for use

 

I think I have everything set right. But my IAX phone wont connect.

 

When I run wireshark I'm seeing this

 

 

Note if above screenshot from wireshark does not show here is a link for
it: http://img402.imageshack.us/i/tempe.jpg/

 

I've tried a variety of setups in my IAX.conf (they all end up with the
same issue, tried just bindaddr=0.0.0.0 with bindport=4569, tried as in
the below example specifying the port for the address and using a
different once incase of conflict with something else I am unaware of.

 

[general]

bindport=4569   ; bindport and bindaddr may be specified

;   ; NOTE: bindport must be specified
BEFORE

; bindaddr or may be specified on a specific

; bindaddr if followed by colon and port

;  (e.g. bindaddr=192.168.0.1:4569)

bindaddr=192.168.17.140:4570

bindaddr=0.0.0.0; more than once to bind to multiple

;   ; addresses, but the first will be the 

;   ; default

;

 

The above being the most recent IAX.conf 

 

Below is what I get in the CLI whenever I reload for a change.

 

 

egg*CLI> iax2 reload

  == Parsing '/etc/asterisk/iax.conf':   == Found

  == Parsing '/etc/asterisk/users.conf':   == Found

[Dec  4 10:17:36] NOTICE[6080]: chan_iax2.c:11087 set_config: Ignoring
bindport on reload

[Dec  4 10:17:36] NOTICE[6080]: chan_iax2.c:11148 set_config: Ignoring
bindaddr on reload

[Dec  4 10:17:36] NOTICE[6080]: chan_iax2.c:11148 set_config: Ignoring
bindaddr on reload

[Dec  4 10:17:36] NOTICE[6080]: chan_iax2.c:11148 set_config: Ignoring
bindaddr on reload

  == Loaded firmware 'iaxy.bin'

egg*CLI>

 

Any ideas?

 

James Shigley

Monroe Telephone Answering Service

409-981-9213

Infinity 5.51,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps, 

 

CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information =is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by "reply to sender only" message and destroy all
electronic and hard copies of the communication, including attachments. 

 

 

 

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Re: [asterisk-users] Variable Name needed

[James A. Shigley]

Thank you that was it

 

James Shigley

Monroe Telephone Answering Service

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Gibbons
Sent: Wednesday, December 02, 2009 4:03 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Variable Name needed

 



My question is, Does anyone know what variable I would use to get the
information for "To" from these SIP calls, the below is the actual SIP
packet obtained from the CLI with SIP Debug On. Other than I stripped
out the IPs



 

The variable you are seeking is ${SIP_HEADER(TO)}

 

I parse the SIP headers from callcentric like this:

Set(calldest=${CUT(CUT(SIP_HEADER(To),@,1),:,2)})

 

Which gives me a real US number like 1xx.

 

Credit for the parsing syntax goes to someone else (not sure where I
found it online).

 

--Dave

 

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[asterisk-users] FW: Variable Name needed

[James A. Shigley]

It might be worth mentioning the voip call is coming from a number we
have thru bandwidth.com in case anyone uses them.

 

James Shigley

 

From: James A. Shigley 
Sent: Wednesday, December 02, 2009 3:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Variable Name needed

 

That wasn't it either. I tried a few other likely fields from that page
none of which gave the correct data

 

James Shigley

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Wednesday, December 02, 2009 2:20 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Variable Name needed

 

According to this link 
http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List I'd
go with ${SIPCALLID}

 



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: Wednesday, December 02, 2009 2:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Variable Name needed

 

Other than having stripping out IPs this is what I am receiving for my
voip calls. Now I normally use ${CALLERID(rdnis) for RDNIS, this works
fine calls that come in on my PRI. BUT at least from this VOIP source
the To field which is my RDNIS information for these calls, doesn't
actually fill into ${CALLERID(rdnis). But as you can see I'm getting the
information.

 

My question is, Does anyone know what variable I would use to get the
information for "To" from these SIP calls, the below is the actual SIP
packet obtained from the CLI with SIP Debug On. Other than I stripped
out the IPs

 

 

 

<--- Transmitting (no NAT) to:5060 --->

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP;branch=z9hG4bK6631.98b11517.0;received=

Via: SIP/2.0/UDP;branch=z9hG4bK6631.f18e9a97.0

Via: SIP/2.0/UDP :5060;branch=z9hG4bK506071629460-1256675031807

Record-Route: 

Record-Route: 

From: "BEAUMONT TX"
;tag=VPSF506071629460

 

To: ;tag=as4b59d217

 

Call-ID: DALMGC0520091202194656056692@

CSeq: 1 INVITE

User-Agent: Asterisk PBX 1.6.0.6

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces, timer

Contact: 

Content-Length: 0

<>

 

Thank You for your time, and I apologize if this is a repeat question. I
did Google, and search thru my * email archive (back thru April 09) for
an answer first.

 

James Shigley

Monroe Telephone Answering Service

409-981-9213

Infinity 5.51,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps, 

 

CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information =is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by "reply to sender only" message and destroy all
electronic and hard copies of the communication, including attachments. 

 

 

 

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Re: [asterisk-users] Variable Name needed

[James A. Shigley]

That wasn't it either. I tried a few other likely fields from that page
none of which gave the correct data

 

James Shigley

Monroe Telephone Answering Service

409-981-9213

Infinity 5.51,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps, 

 

CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information =is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by "reply to sender only" message and destroy all
electronic and hard copies of the communication, including attachments. 

 

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Wednesday, December 02, 2009 2:20 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Variable Name needed

 

According to this link 
http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List I'd
go with ${SIPCALLID}

 



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: Wednesday, December 02, 2009 2:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Variable Name needed

 

Other than having stripping out IPs this is what I am receiving for my
voip calls. Now I normally use ${CALLERID(rdnis) for RDNIS, this works
fine calls that come in on my PRI. BUT at least from this VOIP source
the To field which is my RDNIS information for these calls, doesn't
actually fill into ${CALLERID(rdnis). But as you can see I'm getting the
information.

 

My question is, Does anyone know what variable I would use to get the
information for "To" from these SIP calls, the below is the actual SIP
packet obtained from the CLI with SIP Debug On. Other than I stripped
out the IPs

 

 

 

<--- Transmitting (no NAT) to:5060 --->

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP;branch=z9hG4bK6631.98b11517.0;received=

Via: SIP/2.0/UDP;branch=z9hG4bK6631.f18e9a97.0

Via: SIP/2.0/UDP :5060;branch=z9hG4bK506071629460-1256675031807

Record-Route: 

Record-Route: 

From: "BEAUMONT TX"
;tag=VPSF506071629460

 

To: ;tag=as4b59d217

 

Call-ID: DALMGC0520091202194656056692@

CSeq: 1 INVITE

User-Agent: Asterisk PBX 1.6.0.6

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces, timer

Contact: 

Content-Length: 0

<>

 

Thank You for your time, and I apologize if this is a repeat question. I
did Google, and search thru my * email archive (back thru April 09) for
an answer first.

 

James Shigley

Monroe Telephone Answering Service

409-981-9213

Infinity 5.51,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps, 

 

CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information =is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by "reply to sender only" message and destroy all
electronic and hard copies of the communication, including attachments. 

 



 

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[asterisk-users] Variable Name needed

[James A. Shigley]

Other than having stripping out IPs this is what I am receiving for my
voip calls. Now I normally use ${CALLERID(rdnis) for RDNIS, this works
fine calls that come in on my PRI. BUT at least from this VOIP source
the To field which is my RDNIS information for these calls, doesn't
actually fill into ${CALLERID(rdnis). But as you can see I'm getting the
information.

 

My question is, Does anyone know what variable I would use to get the
information for "To" from these SIP calls, the below is the actual SIP
packet obtained from the CLI with SIP Debug On. Other than I stripped
out the IPs

 

 

 

<--- Transmitting (no NAT) to:5060 --->

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP;branch=z9hG4bK6631.98b11517.0;received=

Via: SIP/2.0/UDP;branch=z9hG4bK6631.f18e9a97.0

Via: SIP/2.0/UDP :5060;branch=z9hG4bK506071629460-1256675031807

Record-Route: 

Record-Route: 

From: "BEAUMONT TX"
;tag=VPSF506071629460

 

To: ;tag=as4b59d217

 

Call-ID: DALMGC0520091202194656056692@

CSeq: 1 INVITE

User-Agent: Asterisk PBX 1.6.0.6

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces, timer

Contact: 

Content-Length: 0

<>

 

Thank You for your time, and I apologize if this is a repeat question. I
did Google, and search thru my * email archive (back thru April 09) for
an answer first.

 

James Shigley

Monroe Telephone Answering Service

409-981-9213

Infinity 5.51,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps, 

 

CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information =is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by "reply to sender only" message and destroy all
electronic and hard copies of the communication, including attachments. 

 

 

 

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[asterisk-users] Queue Question

[James A. Shigley]

First off this is not my work for extensions.conf it is modified from

http://leifmadsen.wordpress.com/2009/07/15/migrating-from-agentcallbackl
ogin-to-standard-dialplan-methods-part-1/

So credit to Leif Madsen  

 

But as to my question

 

[AgentLogin]  

;A replaced version of AgentCallbackLogin() using a GoSub()   

; 

exten => XX,1,Verbose(2,Logging in agent)  

exten => XX,n,WaitExten(5) 

exten => XX,n,GoSub(AgentCallbackLogin,start,1)  

exten => XX,n,Hangup()

 

[AgentLogOut]

exten =>
XX,1,RemoveQueueMember(9819930,DAHDI/g1/${CALLERID(num)})   

 

; calling 'primary' queue  

;  

exten => XX,1,Verbose(2,Calling into the primary queue)  

exten => XX,n,Playback(silence/1)  

exten => XX,n,Queue(9819930)  

exten => XX,n,Hangup()

;

[AgentCallbackLogin]  

; conversion of AgentCallbackLogin() to using AddQueueMember()

;

exten => start,1,Verbose(2,Logging in agent)  

exten => start,n,Playback(silence/1)  

exten => start,n,Read(AGENT_USERID,agent-user)  

exten => start,n,VMauthenticate(${agent_user...@default)  

exten => start,n,AddQueueMember(Queue,DAHDI/g1/${CALLERID(num)})  

exten => start,n,Playback(agent-loginok)  

exten => start,n,Return()  

 

 

Queue Context from Queues.conf

 

[Queue]

music=default

strategy=linear

timeout=5

retry=5

wrapuptime=0

maxlen = 0

announce-frequency = 0

announce-holdtime = no

 

Ok Here is how I would like this queue to work

 

First try and deliver the call to the Dynamic Agents who login using the
above Setup from extensions. I of course want that to be linear in
fashion with 5 seconds to try each one. BUT if none of the dynamic
agents are available or if there are no dynamic agents I want to send
that call to an interface/context. We are an answering service. If no
agent is available in the queue I want to send it to the Interface which
goes to my TAS equipment.

 

So DAHDI/g2/Exten 

 

How do I accomplish that because I can't figure it out from googling or
http://www.voip-info.org/

 

 

James Shigley

Monroe Telephone Answering Service

409-981-9213

Infinity 5.5,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps, 

 

CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information =is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by "reply to sender only" message and destroy all
electronic and hard copies of the communication, including attachments. 

 

 

 

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Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

[James A. Shigley]

It errors the same whether I use g or G. 

 

James Shigley

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Totaro
Sent: Thursday, June 18, 2009 1:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

 

I don't feel like looking it up but does a capital G and lowercase g in
your DAHDI/group make a difference?

Just a thought.

On Thu, Jun 18, 2009 at 2:28 PM, James A. Shigley
 wrote:

I didn't have a limit set, but I put one on of 5 for testing sake that
didn't change a thing.

 

James Shigley

Monroe Telephone Answering Service

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Wednesday, June 17, 2009 2:55 PM


To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

Subject: Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

 

Is your SIP call-limit set to 1?  That might explain the busy/congest
message.

 



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: Wednesday, June 17, 2009 2:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

 

Never saw this appear on the list. So just resending it.

 

 

Alright I've been having an issue when trying to dial out locally when
coming from SIP. This used to work no problem, now it doesn't. Now the
local PRI to Bell Is working fine I have calls coming in and out of it
constantly right now. BUT if I try and make a local call from SIP (from
X-Lite or one of our Linksys SPA2102s) It fails every time with errors
like these

 

 

== Using SIP RTP CoS mark 5

-- Executing [9819...@from_test:1] Dial("SIP/test-b636a620",
"DAHDI/G3/9819213") in new stack

-- Requested transfer capability: 0x00 - SPEECH

-- Called G3/9819213

-- Channel 0/23, span 3 got hangup, cause 50

-- Hungup 'DAHDI/71-1'

  == Everyone is busy/congested at this time (1:0/0/1)

-- Auto fallthrough, channel 'SIP/test-b636a620' status is
'CHANUNAVAIL'

 

  == Using SIP RTP CoS mark 5

-- Executing [9819...@from_test:1] Dial("SIP/test-b6369010",
"DAHDI/G3/4099819213") in new stack

-- Requested transfer capability: 0x00 - SPEECH

-- Called G3/4099819213

-- Channel 0/23, span 3 got hangup, cause 50

-- Hungup 'DAHDI/71-1'

  == Everyone is busy/congested at this time (1:0/0/1)

-- Auto fallthrough, channel 'SIP/test-b6369010' status is
'CHANUNAVAIL'

 

== Using SIP RTP CoS mark 5

-- Executing [9819...@from_test:1] Set("SIP/test-09f23d18",
"CALLERID(name)=James Shigley") in new stack

-- Executing [9819...@from_test:2] Set("SIP/test-09f23d18",
"CALLERID(number)=4099819213") in new stack

-- Executing [9819...@from_test:3] Set("SIP/test-09f23d18",
"CALLERID(all)=James Shigley<4099819213>") in new stack

-- Executing [9819...@from_test:4] Dial("SIP/test-09f23d18",
"DAHDI/G3/9819213") in new stack

-- Requested transfer capability: 0x00 - SPEECH

-- Called G3/9819213

-- Channel 0/22, span 3 got hangup, cause 50

-- Hungup 'DAHDI/70-1'

  == Everyone is busy/congested at this time (1:0/0/1)

-- Auto fallthrough, channel 'SIP/test-09f23d18' status is
'CHANUNAVAIL'

 

Oh and sometimes it will also have this in the errors though no always

 

[Jun 17 10:52:53] WARNING[27549]: app_dial.c:842 wait_for_answer: Unable
to forward voice or dtmf 

 

On the second error above has the 409 added by the dialplan to see if
Bell wanted full 10 digits.

 

For the third I've tried a variety of ways of setting the CID thinking
maybe that was the issue this was just my most recent.

 

 

The odd thing is that I can send the call down one of my other PRI ports
to our Amtelco Infinity system. (via exten=>
9819213,1,Dial(${inf}/409${EXTEN}). I've tried everything I can think of
and googled for a good while trying to find an explanation for "got
hangup, cause 50". What is cause 50?

 

Sip Login information

 

[test]

username=test

type=friend

secret=X

callerid=

host=dynamic

nat=no

canreinvite=no

context=from_test

;codecs

disallow=all

allow=ulaw

 

Also had it as

 

[test]

username=test

type=friend

secret= X

callerid= "James Shigley" <4099819213>

host=dynamic

nat=no

canreinvite=no

context=from_test

;codecs

disallow=all

allow=ulaw

 

My From Context has changed several times here is some of the iterations
I've tried.

 

 

inf=DAHDI/g2  

Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

[James A. Shigley]

I didn't have a limit set, but I put one on of 5 for testing sake that
didn't change a thing.

 

James Shigley

Monroe Telephone Answering Service

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Wednesday, June 17, 2009 2:55 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

 

Is your SIP call-limit set to 1?  That might explain the busy/congest
message.

 



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: Wednesday, June 17, 2009 2:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

 

Never saw this appear on the list. So just resending it.

 

 

Alright I've been having an issue when trying to dial out locally when
coming from SIP. This used to work no problem, now it doesn't. Now the
local PRI to Bell Is working fine I have calls coming in and out of it
constantly right now. BUT if I try and make a local call from SIP (from
X-Lite or one of our Linksys SPA2102s) It fails every time with errors
like these

 

 

== Using SIP RTP CoS mark 5

-- Executing [9819...@from_test:1] Dial("SIP/test-b636a620",
"DAHDI/G3/9819213") in new stack

-- Requested transfer capability: 0x00 - SPEECH

-- Called G3/9819213

-- Channel 0/23, span 3 got hangup, cause 50

-- Hungup 'DAHDI/71-1'

  == Everyone is busy/congested at this time (1:0/0/1)

-- Auto fallthrough, channel 'SIP/test-b636a620' status is
'CHANUNAVAIL'

 

  == Using SIP RTP CoS mark 5

-- Executing [9819...@from_test:1] Dial("SIP/test-b6369010",
"DAHDI/G3/4099819213") in new stack

-- Requested transfer capability: 0x00 - SPEECH

-- Called G3/4099819213

-- Channel 0/23, span 3 got hangup, cause 50

-- Hungup 'DAHDI/71-1'

  == Everyone is busy/congested at this time (1:0/0/1)

-- Auto fallthrough, channel 'SIP/test-b6369010' status is
'CHANUNAVAIL'

 

== Using SIP RTP CoS mark 5

-- Executing [9819...@from_test:1] Set("SIP/test-09f23d18",
"CALLERID(name)=James Shigley") in new stack

-- Executing [9819...@from_test:2] Set("SIP/test-09f23d18",
"CALLERID(number)=4099819213") in new stack

-- Executing [9819...@from_test:3] Set("SIP/test-09f23d18",
"CALLERID(all)=James Shigley<4099819213>") in new stack

-- Executing [9819...@from_test:4] Dial("SIP/test-09f23d18",
"DAHDI/G3/9819213") in new stack

-- Requested transfer capability: 0x00 - SPEECH

-- Called G3/9819213

-- Channel 0/22, span 3 got hangup, cause 50

-- Hungup 'DAHDI/70-1'

  == Everyone is busy/congested at this time (1:0/0/1)

-- Auto fallthrough, channel 'SIP/test-09f23d18' status is
'CHANUNAVAIL'

 

Oh and sometimes it will also have this in the errors though no always

 

[Jun 17 10:52:53] WARNING[27549]: app_dial.c:842 wait_for_answer: Unable
to forward voice or dtmf 

 

On the second error above has the 409 added by the dialplan to see if
Bell wanted full 10 digits.

 

For the third I've tried a variety of ways of setting the CID thinking
maybe that was the issue this was just my most recent.

 

 

The odd thing is that I can send the call down one of my other PRI ports
to our Amtelco Infinity system. (via exten=>
9819213,1,Dial(${inf}/409${EXTEN}). I've tried everything I can think of
and googled for a good while trying to find an explanation for "got
hangup, cause 50". What is cause 50?

 

Sip Login information

 

[test]

username=test

type=friend

secret=X

callerid=

host=dynamic

nat=no

canreinvite=no

context=from_test

;codecs

disallow=all

allow=ulaw

 

Also had it as

 

[test]

username=test

type=friend

secret= X

callerid= "James Shigley" <4099819213>

host=dynamic

nat=no

canreinvite=no

context=from_test

;codecs

disallow=all

allow=ulaw

 

My From Context has changed several times here is some of the iterations
I've tried.

 

 

inf=DAHDI/g2  

bell=DAHDI/G3

 

[from_test] ; noted but not repaired.

exten=> _NXX,1,Dial(${belltd}/409${EXTEN})

exten=> 9819213,1,Dial(${inf}/409${EXTEN}

 

[from_test] ; noted but not repaired.

exten=> _NXX,1,Set(CALLERID(name)=James Shigley)

exten=> _NXX,2,Set(CALLERID(number)=4099819213)

exten=>
_NXX,3,Set(CALLERID(all)=${CALLERID(name)}<${CALLERID(num)}>)

exten=> _NXX,4,Dial(${bell}/${EXTEN})

 

[from_test] ; noted but not repaired.

exten=> _NXX,1,Set(CALLERID(name)=J

Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

[James A. Shigley]

A1, 0x16,
0x02, 0x01, 0xFC, 0x02, 0x01, 0x00, 0x80, 0x0E, 'CHARLOT,DANIEL' ]
PROTOCOL 1F
8B 0001 00 (CONTEXT SPECIFIC [11])
A1 0016 (CONTEXT SPECIFIC [1])
  02 0001 FC (INTEGER: 252)
  02 0001 00 (INTEGER: 0)
  80 000E 43 48 41 52 4C 4F 54 2C 44 41 4E 49 45 4C (CONTEXT SPECIFIC
[0])
< Protocol Discriminator: Q.931 (8)  len=9
< Call Ref: len= 2 (reference 27418/0x6B1A) (Terminator)
< Message type: ALERTING (1)
< [1e 02 81 88]
< Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)
0: 0  Location: Private network serving the local user (1)
<   Ext: 1  Progress Description: Inband
information or appropriate pattern now available. (8) ]
-- Processing IE 30 (cs0, Progress Indicator)
q931.c:3590 q931_receive: call 60186 on channel 2 enters state 4 (Call
Delivered)
< Protocol Discriminator: Q.931 (8)  len=9
< Call Ref: len= 2 (reference 27418/0x6B1A) (Terminator)
< Message type: CONNECT (7)
< [1e 02 81 88]
< Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)
0: 0  Location: Private network serving the local user (1)
<   Ext: 1  Progress Description: Inband
information or appropriate pattern now available. (8) ]
-- Processing IE 30 (cs0, Progress Indicator)
q931.c:3620 q931_receive: call 60186 on channel 2 enters state 10
(Active)
> Protocol Discriminator: Q.931 (8)  len=5
> Call Ref: len= 2 (reference 27418/0x6B1A) (Originator)
> Message type: CONNECT ACKNOWLEDGE (15)
!! Got reject for frame 69, but we have nothing -- resetting!
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate
Connect Request
q931.c:3009 q931_disconnect: call 60186 on channel 2 enters state 11
(Disconnect Request)
> Protocol Discriminator: Q.931 (8)  len=9
> Call Ref: len= 2 (reference 27418/0x6B1A) (Originator)
> Message type: DISCONNECT (69)
> [08 02 81 90]
> Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0
Location: Private network serving the local user (1)
>  Ext: 1  Cause: Normal Clearing (16), class = Normal
Event (1) ]
< Protocol Discriminator: Q.931 (8)  len=5
< Call Ref: len= 2 (reference 27418/0x6B1A) (Terminator)
< Message type: RELEASE (77)
q931.c:3795 q931_receive: call 60186 on channel 2 enters state 0 (Null)
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release
Request
> Protocol Discriminator: Q.931 (8)  len=9
> Call Ref: len= 2 (reference 27418/0x6B1A) (Originator)
> Message type: RELEASE COMPLETE (90)
> [08 02 81 90]
> Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0
Location: Private network serving the local user (1)
>  Ext: 1  Cause: Normal Clearing (16), class = Normal
Event (1) ]
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
!! Got reject for frame 71, but we have nothing -- resetting!

James Shigley


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andres
Sent: Wednesday, June 17, 2009 3:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

James A. Shigley wrote:

> Never saw this appear on the list. So just resending it.
>
You might get more help if you include a PRI Debug that shows the call 
being rejected.

Andres
http://www.neuroredes.com

> Alright I've been having an issue when trying to dial out locally when

> coming from SIP. This used to work no problem, now it doesn't. Now the

> local PRI to Bell Is working fine I have calls coming in and out of it

> constantly right now. BUT if I try and make a local call from SIP 
> (from X-Lite or one of our Linksys SPA2102s) It fails every time with 
> errors like these
>


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[asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

[James A. Shigley]

Never saw this appear on the list. So just resending it.

 

 

Alright I've been having an issue when trying to dial out locally when
coming from SIP. This used to work no problem, now it doesn't. Now the
local PRI to Bell Is working fine I have calls coming in and out of it
constantly right now. BUT if I try and make a local call from SIP (from
X-Lite or one of our Linksys SPA2102s) It fails every time with errors
like these

 

 

== Using SIP RTP CoS mark 5

-- Executing [9819...@from_test:1] Dial("SIP/test-b636a620",
"DAHDI/G3/9819213") in new stack

-- Requested transfer capability: 0x00 - SPEECH

-- Called G3/9819213

-- Channel 0/23, span 3 got hangup, cause 50

-- Hungup 'DAHDI/71-1'

  == Everyone is busy/congested at this time (1:0/0/1)

-- Auto fallthrough, channel 'SIP/test-b636a620' status is
'CHANUNAVAIL'

 

  == Using SIP RTP CoS mark 5

-- Executing [9819...@from_test:1] Dial("SIP/test-b6369010",
"DAHDI/G3/4099819213") in new stack

-- Requested transfer capability: 0x00 - SPEECH

-- Called G3/4099819213

-- Channel 0/23, span 3 got hangup, cause 50

-- Hungup 'DAHDI/71-1'

  == Everyone is busy/congested at this time (1:0/0/1)

-- Auto fallthrough, channel 'SIP/test-b6369010' status is
'CHANUNAVAIL'

 

== Using SIP RTP CoS mark 5

-- Executing [9819...@from_test:1] Set("SIP/test-09f23d18",
"CALLERID(name)=James Shigley") in new stack

-- Executing [9819...@from_test:2] Set("SIP/test-09f23d18",
"CALLERID(number)=4099819213") in new stack

-- Executing [9819...@from_test:3] Set("SIP/test-09f23d18",
"CALLERID(all)=James Shigley<4099819213>") in new stack

-- Executing [9819...@from_test:4] Dial("SIP/test-09f23d18",
"DAHDI/G3/9819213") in new stack

-- Requested transfer capability: 0x00 - SPEECH

-- Called G3/9819213

-- Channel 0/22, span 3 got hangup, cause 50

-- Hungup 'DAHDI/70-1'

  == Everyone is busy/congested at this time (1:0/0/1)

-- Auto fallthrough, channel 'SIP/test-09f23d18' status is
'CHANUNAVAIL'

 

Oh and sometimes it will also have this in the errors though no always

 

[Jun 17 10:52:53] WARNING[27549]: app_dial.c:842 wait_for_answer: Unable
to forward voice or dtmf 

 

On the second error above has the 409 added by the dialplan to see if
Bell wanted full 10 digits.

 

For the third I've tried a variety of ways of setting the CID thinking
maybe that was the issue this was just my most recent.

 

 

The odd thing is that I can send the call down one of my other PRI ports
to our Amtelco Infinity system. (via exten=>
9819213,1,Dial(${inf}/409${EXTEN}). I've tried everything I can think of
and googled for a good while trying to find an explanation for "got
hangup, cause 50". What is cause 50?

 

Sip Login information

 

[test]

username=test

type=friend

secret=X

callerid=

host=dynamic

nat=no

canreinvite=no

context=from_test

;codecs

disallow=all

allow=ulaw

 

Also had it as

 

[test]

username=test

type=friend

secret= X

callerid= "James Shigley" <4099819213>

host=dynamic

nat=no

canreinvite=no

context=from_test

;codecs

disallow=all

allow=ulaw

 

My From Context has changed several times here is some of the iterations
I've tried.

 

 

inf=DAHDI/g2  

bell=DAHDI/G3

 

[from_test] ; noted but not repaired.

exten=> _NXX,1,Dial(${belltd}/409${EXTEN})

exten=> 9819213,1,Dial(${inf}/409${EXTEN}

 

[from_test] ; noted but not repaired.

exten=> _NXX,1,Set(CALLERID(name)=James Shigley)

exten=> _NXX,2,Set(CALLERID(number)=4099819213)

exten=>
_NXX,3,Set(CALLERID(all)=${CALLERID(name)}<${CALLERID(num)}>)

exten=> _NXX,4,Dial(${bell}/${EXTEN})

 

[from_test] ; noted but not repaired.

exten=> _NXX,1,Set(CALLERID(name)=James Shigley)

exten=> _NXX,2,Set(CALLERID(number)=4099819213)

exten=> _NXX,3,Dial(${bell}/${EXTEN})

 

 

Note I didn't include the full context only the lines relevant to local
dialing. LD dialing which is sent out sip works just fine. Also I tried
using g3 instead of G3 thinking maybe there was an issue with the high
channels. Though when I do a core show channels there isn't near close
to all the channels used.

 

One final note. I did try calling other numbers beyond just 9819213 the
errors and issue was the same regardless of the local number dialed.

 

I think that's all the information you might need, If I forgot something
just let me know. Oh and this is on * 1.6.0.6

 

James Shigley

Monroe Telephone Answering Service

409-981-9213

 

 

 

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[asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

[James A. Shigley]

Alright I've been having an issue when trying to dial out locally when
coming from SIP. This used to work no problem, now it doesn't. Now the
local PRI to Bell Is working fine I have calls coming in and out of it
constantly right now. BUT if I try and make a local call from SIP (from
X-Lite or one of our Linksys SPA2102s) It fails every time with errors
like these

 

 

== Using SIP RTP CoS mark 5

-- Executing [9819...@from_test:1] Dial("SIP/test-b636a620",
"DAHDI/G3/9819213") in new stack

-- Requested transfer capability: 0x00 - SPEECH

-- Called G3/9819213

-- Channel 0/23, span 3 got hangup, cause 50

-- Hungup 'DAHDI/71-1'

  == Everyone is busy/congested at this time (1:0/0/1)

-- Auto fallthrough, channel 'SIP/test-b636a620' status is
'CHANUNAVAIL'

 

  == Using SIP RTP CoS mark 5

-- Executing [9819...@from_test:1] Dial("SIP/test-b6369010",
"DAHDI/G3/4099819213") in new stack

-- Requested transfer capability: 0x00 - SPEECH

-- Called G3/4099819213

-- Channel 0/23, span 3 got hangup, cause 50

-- Hungup 'DAHDI/71-1'

  == Everyone is busy/congested at this time (1:0/0/1)

-- Auto fallthrough, channel 'SIP/test-b6369010' status is
'CHANUNAVAIL'

 

== Using SIP RTP CoS mark 5

-- Executing [9819...@from_test:1] Set("SIP/test-09f23d18",
"CALLERID(name)=James Shigley") in new stack

-- Executing [9819...@from_test:2] Set("SIP/test-09f23d18",
"CALLERID(number)=4099819213") in new stack

-- Executing [9819...@from_test:3] Set("SIP/test-09f23d18",
"CALLERID(all)=James Shigley<4099819213>") in new stack

-- Executing [9819...@from_test:4] Dial("SIP/test-09f23d18",
"DAHDI/G3/9819213") in new stack

-- Requested transfer capability: 0x00 - SPEECH

-- Called G3/9819213

-- Channel 0/22, span 3 got hangup, cause 50

-- Hungup 'DAHDI/70-1'

  == Everyone is busy/congested at this time (1:0/0/1)

-- Auto fallthrough, channel 'SIP/test-09f23d18' status is
'CHANUNAVAIL'

 

Oh and sometimes it will also have this in the errors though no always

 

[Jun 17 10:52:53] WARNING[27549]: app_dial.c:842 wait_for_answer: Unable
to forward voice or dtmf 

 

On the second error above has the 409 added by the dialplan to see if
Bell wanted full 10 digits.

 

For the third I've tried a variety of ways of setting the CID thinking
maybe that was the issue this was just my most recent.

 

 

The odd thing is that I can send the call down one of my other PRI ports
to our Amtelco Infinity system. (via exten=>
9819213,1,Dial(${inf}/409${EXTEN}). I've tried everything I can think of
and googled for a good while trying to find an explanation for "got
hangup, cause 50". What is cause 50?

 

Sip Login information

 

[test]

username=test

type=friend

secret=X

callerid=

host=dynamic

nat=no

canreinvite=no

context=from_test

;codecs

disallow=all

allow=ulaw

 

Also had it as

 

[test]

username=test

type=friend

secret= X

callerid= "James Shigley" <4099819213>

host=dynamic

nat=no

canreinvite=no

context=from_test

;codecs

disallow=all

allow=ulaw

 

My From Context has changed several times here is some of the iterations
I've tried.

 

 

inf=DAHDI/g2  

bell=DAHDI/G3

 

[from_test] ; noted but not repaired.

exten=> _NXX,1,Dial(${belltd}/409${EXTEN})

exten=> 9819213,1,Dial(${inf}/409${EXTEN}

 

[from_test] ; noted but not repaired.

exten=> _NXX,1,Set(CALLERID(name)=James Shigley)

exten=> _NXX,2,Set(CALLERID(number)=4099819213)

exten=>
_NXX,3,Set(CALLERID(all)=${CALLERID(name)}<${CALLERID(num)}>)

exten=> _NXX,4,Dial(${bell}/${EXTEN})

 

[from_test] ; noted but not repaired.

exten=> _NXX,1,Set(CALLERID(name)=James Shigley)

exten=> _NXX,2,Set(CALLERID(number)=4099819213)

exten=> _NXX,3,Dial(${bell}/${EXTEN})

 

 

Note I didn't include the full context only the lines relevant to local
dialing. LD dialing which is sent out sip works just fine. Also I tried
using g3 instead of G3 thinking maybe there was an issue with the high
channels. Though when I do a core show channels there isn't near close
to all the channels used.

 

One final note. I did try calling other numbers beyond just 9819213 the
errors and issue was the same regardless of the local number dialed.

 

I think that's all the information you might need, If I forgot something
just let me know. Oh and this is on * 1.6.0.6

 

James Shigley

Monroe Telephone Answering Service

409-981-9213

 

 

 

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Re: [asterisk-users] Bandwidth management and ADSL router

[James A. Shigley]

A lot of ISP adsl modems aren't capable. But most should be. Just log in to it 
give it a private IP for your lan(192.X.X.X or whatever your using) then have 
all your computers use that local IP as their gateway address.

If you have an ADSL modem which doesn't then simple get a router (hell a 
Linksys/Dlink $50 cheapy from wallmart would work) and have the ADSL plug into 
the router and all the stations use the router for their gateway.

If you have a spare server or virtual server space you can use Vyatta 
(Vyatta.com) it is a free open source router/firewall/vpn/few other things. 
I've never used it in a virtual environment, but I see no reason why it 
wouldn't work that way. Also note that it requires almost nothing to run so you 
can put it on an old < 1Ghz machine and It would still operate just fine.

James Shigley
Monroe Telephone Answering Service
409-981-9213
Infinity 5.5,UC 4.02.3803, Blink 3.0.104
Ecreator:2.21, eResponse 1.1.7
Webportal,WebApps,
 
CONFIDENTIALITY NOTICE: This email, including any attachments, contains 
information which may be confidential or privileged. The information is 
intended to be for the use of the individual or entity named above. If you are 
not the intended recipient, be aware that any disclosure, copying, distribution 
or use of the contents of this information is prohibited. If you have received 
this email in error, please notify the sender immediately by "reply to sender 
only" message and destroy all electronic and hard copies of the communication, 
including attachments. 

"Common sense is the collection of prejudices acquired by age eighteen." -- 
Albert Einstein 
"Once you can accept the universe as matter expanding into nothing that is 
something,wearing stripes with plaid comes easy." -- Albert Einstein
"Theory is when you know something, but it doesn't work. Practice is when
something works, but you don't know why. Programmers combine theory and
practice: Nothing works and they don't know why.-Anonymous Developer"

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Tuesday, May 26, 2009 11:55 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Bandwidth management and ADSL router


Thanks for all.

But what all gave me was a software need to be installed on PC, but I am 
looking for a router (ADSL router) that can does this, because usually the ADSL 
router is the default gateway where all the traffic goes out and in. 

Any ADSL router device can do this?

About Draytek, as I understand that control can be done only at upload traffic 
and not download traffic, while 90% of the problem are coming from download 
traffic, so this is not the needed.

Any advise in that direction?

Regards
Bilal


  


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Re: [asterisk-users] Ghost ??

[James A. Shigley]

Then most likely adam is right. You have interference/crossover on an
analog line in your building or on the telco end.

 

James Shigley

Monroe Telephone Answering Service

409-981-9213

Infinity 5.5,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps,

 

CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by "reply to sender only" message and destroy all
electronic and hard copies of the communication, including attachments. 

 

"Common sense is the collection of prejudices acquired by age eighteen."
-- Albert Einstein 

"Once you can accept the universe as matter expanding into nothing that
is something,wearing stripes with plaid comes easy." -- Albert Einstein

"Theory is when you know something, but it doesn't work. Practice is
when

something works, but you don't know why. Programmers combine theory and

practice: Nothing works and they don't know why.-Anonymous Developer"

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David @ULC
Sent: Tuesday, May 19, 2009 1:32 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Ghost ??

 

No such pattern.

 

It happens ad hoc.

On Tue, May 19, 2009 at 11:46 PM, David @ULC 
wrote:

We are using asterisk and sometime when our guys are on call , they hear
some voice of person and amazingly that person is NOT from our center.

 

Any one faced this kind of thing ?

 

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Re: [asterisk-users] Ghost ??

[James A. Shigley]

Is there any pattern to when it happens. 

 

Such as when called from landlines from X company?

>From cell phones from X company?

>From X people? (then what is the same about them, but not the others.
Telco, location, ect?)

At X times of day?

 

Ect, ect.

 

It sounds like bleed over, which can be causes by some many things the
best place to start is to find a pattern if there is one.

 

James Shigley

Monroe Telephone Answering Service

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David @ULC
Sent: Tuesday, May 19, 2009 1:17 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Ghost ??

 

We are using asterisk and sometime when our guys are on call , they hear
some voice of person and amazingly that person is NOT from our center.

 

Any one faced this kind of thing ?

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Re: [asterisk-users] Is anyone keeping up with the versions?

[James A. Shigley]

Unless there is a new feature or your making a new system. Don't fix it
if it aint broke.

 

BUT do stay current on reading about new feature and things in the
releases.

 

James Shigley

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Tuesday, May 12, 2009 1:45 PM
To: 'Asterisk Users List'
Subject: Re: [asterisk-users] Is anyone keeping up with the versions?

 

Pick a release and stick with it as long as you can.  Only when you have
to jump, pick a new release, test the hell out of it, and then leave it
alone.

 

Too many people try to keep on the latest release...

 



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thermal
Wetland
Sent: Tuesday, May 12, 2009 2:32 PM
To: Asterisk Users List
Subject: [asterisk-users] Is anyone keeping up with the versions?

We are still using 1.4 and were going to start testing with 1.6.0, but
then 1.6.1 was released and now 1.6.2 is already in beta 2.

That seems like a lot of independent releases to maintain.  I read about
all the regressions ans hurried dot releases, makes us nervous.

How is everyone doing their testing?

-Matt

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[asterisk-users] AGI PHP

[James A. Shigley]

I'm just trying to make a real simple Survey via php. Just want it to
play the Question Files, wait for a response, save the response into the
correct variable and then email it all.

 

I have no issue playing the audio or emailing. But I can't get it to
wait for digits or to properly capture those digits into the variables.
I know the code is technically right since the emails have this in place
of what Q1 should be "200 result=35 endpos=2400" instead of whatever
digit I pressed during testing.

 

Anyway my questions are

 

A.  How to have it play the whole audio file before it goes to the
next command, without having to use the sleep function preferably.

B.  How to make it actually wait for digits 

C.  How to get it to actually capture the digits (I think it would
if it wasn't streaming thru the whole file without waiting for a
response.)

 

I'm just trying to capture digits 1-9 nothing fancy.

 

#!/usr/bin/php -q



 

 

James Shigley

 

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[asterisk-users] test

[James A. Shigley]

Had an inbound email server issue, just double checking it is working
again.

 

James Shigley

Monroe Telephone Answering Service

409-981-9213

Infinity 5.5,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps,

 

CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by "reply to sender only" message and destroy all
electronic and hard copies of the communication, including attachments. 

 

"Common sense is the collection of prejudices acquired by age eighteen."
-- Albert Einstein 

"Once you can accept the universe as matter expanding into nothing that
is something,wearing stripes with plaid comes easy." -- Albert Einstein

"I know a little of everything, but a lot of nothing"

 

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Re: [asterisk-users] Feature request: manager show events

[James A. Shigley]

Then a suggestion for the next version would be to have a module which has the 
core set of events that are common to most everything for listing and added 
too, but still leave it open for the custom events most everyone uses for one 
thing or another.

James Shigley
Monroe Telephone Answering Service
409-981-9213
Infinity 5.5,UC 4.02.3803, Blink 3.0.104
Ecreator:2.21, eResponse 1.1.7
Webportal,WebApps,
 
CONFIDENTIALITY NOTICE: This email, including any attachments, contains 
information which may be confidential or privileged. The information is 
intended to be for the use of the individual or entity named above. If you are 
not the intended recipient, be aware that any disclosure, copying, distribution 
or use of the contents of this information is prohibited. If you have received 
this email in error, please notify the sender immediately by "reply to sender 
only" message and destroy all electronic and hard copies of the communication, 
including attachments. 

"Common sense is the collection of prejudices acquired by age eighteen." -- 
Albert Einstein 
"Once you can accept the universe as matter expanding into nothing that is 
something,wearing stripes with plaid comes easy." -- Albert Einstein
"I know a little of everything, but a lot of nothing"


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Moises Silva
Sent: Friday, April 24, 2009 9:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Feature request: manager show events

> Hi,
>
> To further improve Asterisk documentation, would approve "manager show
> events" and "manager show event " commands to be added to CLI ?
> Today, it is possible to list available manager commands but not to list
> available events, AFAIK.
>
> Regards

The problem is that currently, manager events are not registered, any
module is free to launch events and there is no enforcement for the
events to have a clearly defined structure. Work has been done lately
in trying to make the naming of headers and order to be standard, but
there is not programming interface enforcing that behaviour. The
available events can be extracted using grep manager_event in the
asterisk source code, but I agree it would be nice to see more
structure there.

Moy

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Re: [asterisk-users] AGI PHP script

[James A. Shigley]

Actually I feel like an idiot. I had forgotten to put asterisk as an
allowed sender in the server that those emails are going out of.
(different from what * normally uses to email us)

 

James Shigley

Monroe Telephone Answering Service

409-981-9213

Infinity 5.5,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps,

 

CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by "reply to sender only" message and destroy all
electronic and hard copies of the communication, including attachments. 

 

"Common sense is the collection of prejudices acquired by age eighteen."
-- Albert Einstein 

"Once you can accept the universe as matter expanding into nothing that
is something,wearing stripes with plaid comes easy." -- Albert Einstein

"I know a little of everything, but a lot of nothing"

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy
Gbaguidi
Sent: Thursday, April 23, 2009 11:57 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] AGI PHP script

 

First run 

/var/lib/asterisk/agi-bin/newhire.php

 

>From linux command line to see if you don't have any error and that your
AGI is executable.

 

Then run 'agi debug' from the asterisk cli, place a call and see what
was send and receive from your agi

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: April-23-09 12:26 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] AGI PHP script

 

I have the below script that doesn't seem to be working. I don't know if
I have something in the script wrong that I am just missing. Or if I
don't have the php.ini set correctly for emailing

 

 

This is the CLI output

-- Executing [4099xxx...@port3_real:1] Goto("DAHDI/50-1", "newhire,s,1")
in

new stack

-- Goto (newhire,s,1)

-- Executing [...@newhire:1] Ringing("DAHDI/50-1", "") in new stack

-- Executing [...@newhire:2] Answer("DAHDI/50-1", "") in new stack

-- Executing [...@newhire:3] Monitor("DAHDI/50-1",
"wav,/var/lib/asterisk/soun

ds/NewHire/Newhire-1240503071.15148-4099819213-s,o") in new stack

-- Executing [...@newhire:4] AGI("DAHDI/50-1", "newhire.php") in new
stack

-- Launched AGI Script /var/lib/asterisk/agi-bin/newhire.php

-- AGI Script newhire.php completed, returning 0

-- Auto fallthrough, channel 'DAHDI/50-1' status is 'UNKNOWN'

-- Hungup 'DAHDI/50-1'

 

Here is the script

 

 

#!/usr/bin/php5 



 

 

So is it anything obviously wrong with the script I'm missing?

 

Besides something not being configured in php.ini correctly any other
ideas?

 

James Shigley

Monroe Telephone Answering Service

409-981-9213

Infinity 5.5,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps,

 

CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by "reply to sender only" message and destroy all
electronic and hard copies of the communication, including attachments. 

 

"Common sense is the collection of prejudices acquired by age eighteen."
-- Albert Einstein 

"Once you can accept the universe as matter expanding into nothing that
is something,wearing stripes with plaid comes easy." -- Albert Einstein

"I know a little of everything, but a lot of nothing"

 

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[asterisk-users] AGI PHP script

[James A. Shigley]

I have the below script that doesn't seem to be working. I don't know if
I have something in the script wrong that I am just missing. Or if I
don't have the php.ini set correctly for emailing

 

 

This is the CLI output

-- Executing [4099xxx...@port3_real:1] Goto("DAHDI/50-1", "newhire,s,1")
in

new stack

-- Goto (newhire,s,1)

-- Executing [...@newhire:1] Ringing("DAHDI/50-1", "") in new stack

-- Executing [...@newhire:2] Answer("DAHDI/50-1", "") in new stack

-- Executing [...@newhire:3] Monitor("DAHDI/50-1",
"wav,/var/lib/asterisk/soun

ds/NewHire/Newhire-1240503071.15148-4099819213-s,o") in new stack

-- Executing [...@newhire:4] AGI("DAHDI/50-1", "newhire.php") in new
stack

-- Launched AGI Script /var/lib/asterisk/agi-bin/newhire.php

-- AGI Script newhire.php completed, returning 0

-- Auto fallthrough, channel 'DAHDI/50-1' status is 'UNKNOWN'

-- Hungup 'DAHDI/50-1'

 

Here is the script

 

 

#!/usr/bin/php5 



 

 

So is it anything obviously wrong with the script I'm missing?

 

Besides something not being configured in php.ini correctly any other
ideas?

 

James Shigley

Monroe Telephone Answering Service

409-981-9213

Infinity 5.5,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps,

 

CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by "reply to sender only" message and destroy all
electronic and hard copies of the communication, including attachments. 

 

"Common sense is the collection of prejudices acquired by age eighteen."
-- Albert Einstein 

"Once you can accept the universe as matter expanding into nothing that
is something,wearing stripes with plaid comes easy." -- Albert Einstein

"I know a little of everything, but a lot of nothing"

 

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Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk

[James A. Shigley]

Alright again, what do you see on the CLI when you make a call to 210/211?

 

James Shigley

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens
Sent: Monday, April 13, 2009 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls using 
Asterisk

 

Tony Plack,

this is the result form Asterisk CLI :

[r...@asterisk asterisk]# /usr/sbin/asterisk -vr
Asterisk 1.4.24, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer 
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for 
details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3895)
Verbosity is at least 5
asterisk*CLI> dialplan reload
Dialplan reloaded.
  == Parsing '/etc/asterisk/extensions.conf': Found
-- Registered extension context 'intern'
-- Added extension '210' priority 1 to intern
-- Added extension '211' priority 1 to intern
  == Parsing '/etc/asterisk/users.conf': Found
asterisk*CLI> sip reload
Reloading SIP
  == Parsing '/etc/asterisk/sip.conf': Found
  == Parsing '/etc/asterisk/users.conf': Found
  == SIP Listening on 192.168.4.248:5060
  == Using SIP TOS: none
  == Parsing '/etc/asterisk/sip_notify.conf': Found

So I've changed the bindaddr... Still no change I'm afraid...

Thanks for your reply !

Please help me a bit further cause this a work I'm doing as thesis.

Jonas.


On Mon, 2009-04-13 at 11:34 -0500, Anthony Plack wrote: 

 
> bindaddr = 0.0.0.0
> 
 
I would set this to the ethernet interface IP address, I believe this may be 
your issue.
 
Registration is only for receiving calls, if you are not seeing information on 
the dial, then the phone is not talking to the server.  I would make sure of 
the settings in the web-interface as well.
 
Tony Plack
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Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls usingAsterisk

[James A. Shigley]

What do you see when you run asterisk –r and dial 210 or 211 from one of the 
phones

 

James Shigley

Monroe Telephone Answering Service

409-981-9213

Infinity 5.5,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps,

 

CONFIDENTIALITY NOTICE: This email, including any attachments, contains 
information which may be confidential or privileged. The information is 
intended to be for the use of the individual or entity named above. If you are 
not the intended recipient, be aware that any disclosure, copying, distribution 
or use of the contents of this information is prohibited. If you have received 
this email in error, please notify the sender immediately by "reply to sender 
only" message and destroy all electronic and hard copies of the communication, 
including attachments. 

 

"Common sense is the collection of prejudices acquired by age eighteen." -- 
Albert Einstein 

"Once you can accept the universe as matter expanding into nothing that is 
something,wearing stripes with plaid comes easy." -- Albert Einstein

"I know a little of everything, but a lot of nothing"

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens
Sent: Monday, April 13, 2009 11:19 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk-beginner : cannot make phonecalls 
usingAsterisk

 

Hi there,

this is the first time that I'm building an Asterisk-server.

I have compiled Asterisk together with Zaptel on an CentOS 5.3-system.
Zaptel is for later, when configuring the POTS-line. Now first internal 
communication with SIP.

Thought it would go easier...

I have 2 Grandstream IP-phones : BT-201 and GXP-1200.

These are my settings :

sip.conf : 
[r...@asterisk asterisk]# cat sip.conf
[general]
bindport=5060
bindaddr = 0.0.0.0

[BT201]
type=friend
context=intern
host=192.168.4.210
secret=testpaswoord

[GXP1200]
type=friend
context=intern
host=192.168.4.211
secret=testpaswoord 
extensions.conf : 
[r...@asterisk asterisk]# cat extensions.conf
[intern]
exten => 210,1,Dial(SIP/BT201)
exten => 211,1,Dial(SIP/GXP1200) 
Asterisk CLI shows me : 
asterisk*CLI> sip reload
Reloading SIP
  == Parsing '/etc/asterisk/sip.conf': Found
  == Parsing '/etc/asterisk/users.conf': Found
  == Parsing '/etc/asterisk/sip_notify.conf': Found
asterisk*CLI> sip show peers
Name/username  HostDyn Nat ACL Port Status  
 
GXP1200192.168.4.211   5060 Unmonitored 
  
BT201  192.168.4.210   5060 Unmonitored 
  
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]

asterisk*CLI> dialplan show intern
[ Context 'intern' created by 'pbx_config' ]
  '210' =>  1. Dial(SIP/BT201)[pbx_config]
  '211' =>  1. Dial(SIP/GXP1200)  [pbx_config] 

I pick up the phone of the BT201 and dial 211... nothing happens.
I pick up the phone of the GXP1200 and dial 210... nothing happens.

I would love to have your feedback on this. Where could this problem be 
situated ?

I notice (on the Asterisk CLI) that my SIP-phones do not register. They have a 
fixed IP and there account information is set via the web interface.

Greetingz,
Jonas. 

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Re: [asterisk-users] IVR Survey

[James A. Shigley]

Well considering php is the only language I know and even that not much. I'm 
not sure how I would accomplish the recordings and combination from within the 
php agi script.

Better question might be, how would you do it (bear in mind if AGI it would 
have to be php) and do you have a code example(s) to share..

James Shigley
Monroe Telephone Answering Service
409-981-9213
Infinity 5.5,UC 4.02.3803, Blink 3.0.104
Ecreator:2.21, eResponse 1.1.7
Webportal,WebApps,
 
CONFIDENTIALITY NOTICE: This email, including any attachments, contains 
information which may be confidential or privileged. The information is 
intended to be for the use of the individual or entity named above. If you are 
not the intended recipient, be aware that any disclosure, copying, distribution 
or use of the contents of this information is prohibited. If you have received 
this email in error, please notify the sender immediately by "reply to sender 
only" message and destroy all electronic and hard copies of the communication, 
including attachments. 

"Common sense is the collection of prejudices acquired by age eighteen." -- 
Albert Einstein 
"Once you can accept the universe as matter expanding into nothing that is 
something,wearing stripes with plaid comes easy." -- Albert Einstein
"I know a little of everything, but a lot of nothing"

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jared Smith
Sent: Friday, April 10, 2009 12:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IVR Survey

On Fri, 2009-04-10 at 11:04 -0500, James A. Shigley wrote:
> But I’m clueless as to how to combined the recordings into one file. I
> don’t want the questions in the recordings, Only the caller’s side of
> the conversation without the dead space while they listen to the
> Qs/Think on their response.

I'd use the Monitor() application to record the conversation (it records
both inbound and outbound audio, but you'll just throw the one away).
You can then use the PauseMonitor() application to pause the recording
before playing a prompt, and UnPauseMonitor() to start the recording
again after the prompt.

> And since this isn’t a vmail account and trying to avoid an AGI script
> if possible I’m not sure how to email the recording(s).  I also want
> to be able to structure the body of the email so that it reads
> something like

If you don't want to use AGI to do this, you're pretty much limited to
using the System() application and finding a way to send your email from
the system command line.  Not impossible by any stretch of the
imagination... it just takes a bit more work.



-- 
Jared Smith
Training Manager
Digium, Inc.


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[asterisk-users] IVR Survey

[James A. Shigley]

Alright I know how to do basic IVR in *. But what I'm working on trying
to do now is a survey. I've found very little things out there on google
or the archives for how to do surveys with the * ivr. 

 

Here is more or less what I'm trying to accomplish

 

1.   Call comes in Plays Greeting

2.   Starts Survey 

3.   Ask Q1, Record the answer (voice responses) repeat this step
for each Question

4.   Combined the recorded responses into one file.

5.   Email Combined Audio Fi

 

 

I know how to structure the IVR

 

1 answer

2.  Play greeting

3. Play Question 1

3. Record Response

4. Play Q2 

5. Record Response

Ect ect

 

But I'm clueless as to how to combined the recordings into one file. I
don't want the questions in the recordings, Only the caller's side of
the conversation without the dead space while they listen to the
Qs/Think on their response.

 

And since this isn't a vmail account and trying to avoid an AGI script
if possible I'm not sure how to email the recording(s).  I also want to
be able to structure the body of the email so that it reads something
like

 

You have a new call from $CallerID - "$CallerName" on 'DateTime' ...
ect, ect.

 

James Shigley

Monroe Telephone Answering Service

409-981-9213

Infinity 5.5,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps,

 

CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by "reply to sender only" message and destroy all
electronic and hard copies of the communication, including attachments. 

 

"Common sense is the collection of prejudices acquired by age eighteen."
-- Albert Einstein 

"Once you can accept the universe as matter expanding into nothing that
is something,wearing stripes with plaid comes easy." -- Albert Einstein

"I know a little of everything, but a lot of nothing"

 

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