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Warren Selby | 1 Apr 2012 02:04

Re: keep dst cdr record if context change

On Fri, Mar 30, 2012 at 4:51 PM, Daniel Knoll <daniel <at> danielknoll.de> wrote:
Looks nice, was also my first idea, but field dst is read only. I can't overwrite this and get an error like this

ERROR[2474]: cdr.c:345 ast_cdr_setvar: Attempt to set the 'dst' read-only variable!.



I was afraid of that.  Does it absolutely have to be dst that you store this information in?  You can create custom cdr fields that are both readable and writeable.  Something like:

[incoming]
exten => _X.,1,Verbose(New call coming in)
exten => _X.,n,Set(CDR(original_dst)=${EXTEN})
exten => _X.,n,Goto(mainmenu,s,1)





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--Warren Selby, dCAP
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Alec Davis | 1 Apr 2012 07:06
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Re: keep dst cdr record if context change

Create a field called 'dnid', this then is the original called number, no
matter now much you jump around with contexts.

Alec Davis

> -----Original Message-----
> From: asterisk-users-bounces <at> lists.digium.com 
> [mailto:asterisk-users-bounces <at> lists.digium.com] On Behalf Of 
> Daniel Knoll
> Sent: Saturday, 31 March 2012 7:17 a.m.
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] keep dst cdr record if context change
> 
> Hello nice group,
> 
> having a Problem with CDRs. 
> If i change the context with Goto() Asterisk write the new 
> exten in "dst" cdr field.
> 
> How can i keep the old entry? Any ideas makes me very happy.
> 
> Thanks for helping me.
> Daniel
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Todd Routhier | 1 Apr 2012 07:52
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Re: Mute DTMF

Thanks John and SamyGo,


 I tried your solution SamyGo and it does the trick fine. John, just liked doing it in the dial plan better but I may use the firewall trick in the future.

Thanks again!

--Todd


On Fri, Mar 30, 2012 at 12:45 AM, SamyGo <govoiper <at> gmail.com> wrote:
Hey,
I not sure why your dtmfmode isn't working. The way I turned off the dtmf within an IVR was:

1- fix the dtmfmode of any sip user to rfc2833, so he is able to send dtmf to navigate within the IVR.
2- For places where I wanted to ignore any user DTMF key presses, I changed the dtmfmode of channel in the dialplan.

That way I knew that the call will be negotiated on rfc2833 but changing that during the call ignores any key presses and reverting back again makes it functional again !!

I hope this helps.

Regards,
Sammy.

On Fri, Mar 30, 2012 at 2:54 AM, John Kiniston <johnkiniston <at> gmail.com> wrote:

On Thu, Mar 29, 2012 at 12:09 PM, Todd Routhier <fonemasta <at> gmail.com> wrote:
I have been breaking my head on this, can't find a solution.

Anyone know a way to mute DTMF on SIP? I have already tried changing the dtmfmode option and messing with different codec/dtmfmode settings but so far, not having any luck.


It's not an asterisk based solution but you could use RFC2833 signalling and then drop the RTP DTMF packets at your firewall.

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---Heinlein

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bilal ghayyad | 1 Apr 2012 15:04
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Telephony Card: GSM slots + Analoge

Dears;

I am looking to get a telephony card that has GSM slots (ability to place my GSM card into it) in addition to
analoge FXS and FXO. 

There is a card that I found it but really I do not know how much it is reliable: http://www.atcom.cn/AX2G4A.html

Did anyone tried atcom?

Is there a similar cards like it (but to be USA, Canada or Europe manufacturing)?

Regards
Bilal

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sean darcy | 1 Apr 2012 17:22
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10.3.0: gtalk_request: No XMPP client to talk to, us (partial JID)

Trying to use gtalk:

     -- Executing [andy <at> ipkall:2] Dial("SIP/ipkall-00000000", 
"gtalk/andy-gtalk/+1xxxyyyzzzz <at> voice.google.com") in new stack
[Apr  1 10:41:53] ERROR[2416]: chan_gtalk.c:1934 gtalk_request: No XMPP 
client to talk to, us (partial JID) : andy-gtalk

gtalk.conf

[general]
context=google-in		; Context to dump call into
allowguest=yes
stunaddr = numb.viagenie.ca
bindaddr=0.0.0.0
externip=aa.bb.cc.dd

disallow=all
allow=ulaw

[andy-gtalk]
username=<username> <at> gmail.com
context=google-in
connection=andy-jabber

gtalk show settings

Global Settings:
----------------
   UDP Bindaddress:           0.0.0.0
   Stun Address:              66.228.45.110
   External IP:               aa.bb.cc.dd
   Context:                   google-in
   Codecs:                    (ulaw)
   Parking Lot:               default
   Allow Guest:               Yes

jabber.conf:

[andy-jabber]
type=client
serverhost=talk.google.com
username=<username> <at> gmail.com/Talk
secret=<>
port=5222
usetls=yes
usesasl=yes
statusmessage=No one here
status=xaway

jabber show connections
Jabber Users and their status:
        [andy-jabber] <username> <at> gmail.com/Talk     - Connected

jabber test andy-jabber
User: <username> <at> gmail.com
Resource: gmail.675A4337
    client: http://mail.google.com/xmpp/client/caps
    version: 1.1
    Jingle Capable: 1
	Priority: 0
	Status: 3
	Message:

Oooh a working message stack!

So jabber seems to be working.

Once while trying this I got this gtalk error:

WARNING[2571]: chan_gtalk.c:1923 gtalk_request: Could not find recipient.

Thanks for any help.

sean

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Stuart Elvish | 1 Apr 2012 18:42
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404 Response to Invite - Should be 401

Hi all,

I am currently testing a new version of firmware released by an ATA
vendor and I have come across a strange problem in 1.8.9.0.

Sometimes if I dial immediately after hanging up a previous call (we
used voicemail for our testing as it has "unlimited" capacity)
Asterisk will return a 404 code instead of doing the usual INVITE -
401 - INVITE sequence. The CLI says that the call failed because the
extension was not found in context 'default'. It appears from this (as
the ATA is correctly setup and normally make calls to a different
context) that Asterisk believes the extension is unauthenticated for
some calls and sending them straight to the guest (default) context.

What should I be looking for in the initial invite from the ATA which
will be triggering a 404? Is there something wrong in the SIP header /
body which would be triggering a "safety lock down" of the call? I
have tried to look at the invite packets but I can't see anything that
is incorrect.

On one occasion the sequence was INVITE - 401 - INVITE - 404 but most
commonly it is INVITE - 404.

We have also noticed that sometimes the ATA stops responding to
OPTIONS requests (qualify) whilst this issue is happening and there
may be some other registration issues. Some diagnostics have already
been completed but so far we haven't found anything which points us to
where the issue actually is. We assume it is an ATA related issue.

Any pointers and suggestions greatly appreciated.

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Matt Riddell | 2 Apr 2012 01:37
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Re: concurrent channels limit

On 31/03/2012, at 3:28 AM, Syco wrote:
But if I change the dialplan, remove background and wait functions, add play with a g729 audio file instead, I could do again just 80 concurrent call.

How many g729 licenses do you have?  You sure you're not transcoding?

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Oguzhan Kayhan | 2 Apr 2012 08:53
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dial rule problems( on e1 interface) after upgrading 1.8

Hello,
I was using 1.6 asterisk for a long time.
My configuration is as follows. SOme of my users(analogue ones) are on
ericsson pbx which is connected to asterisk via e1 interfaces.
And asterisk is dialing out via a sip trunk.

Ericsson has a setting for prefixes as minimum digits and max(otional)
For ex, we can set 0044  min 13 max 15 as a dialing rule.
The problem is, after upgrading to 1.8.8.2 version, for example if a user
on ericsson starts to dial a 15 digit number at 13th, asterisk tries to
dial the first 13 digits without waiting the rest.

What i am suspecting is, before the default DTMF mode was inband and now
it changed to that rfc staff, which gets the first digits as a string and
tries to dial it.
There is no change in ercissons config.
So how can i make the asterisk wait for the rest of the number without
dialing it immediately??
I hope I made myself clear.

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Tzafrir Cohen | 2 Apr 2012 10:15

Re: DAHDI works, but returns CHANUNAVAIL ??

On Thu, Mar 29, 2012 at 01:58:39PM -0400, sean darcy wrote:
>  DAHDI 2.6.0, dahdi show status
> Description                              Alarms  IRQ    bpviol CRC
> Fra Codi Options  LBO
> Wildcard TDM400P REV I Board 5           OK      0      0      0 CAS
> Unk           0 db (CSU)/0-133 feet (DSX-1)
> 
> Dahdi 1 is an internal extension, dahdi 4 is pstn.
> 
> This call completes. 

How can you tell it is complete? Where is "this call"? Any trace of it?
How can you tell it was complete? Did you listen to it?

> But DAHDI comes back with CHANUNAVAIL. This a
> problem since we then test for CHANUNAVAIL to use an alternative
> provider.
> 
>    -- Executing [s <at> DialOut:17] Dial("DAHDI/1-1",
> "DAHDI/4/1XXXYYYZZZZ") in new stack
>     -- Called DAHDI/4/1XXXYYYZZZZ
>     -- Hanging up on 'DAHDI/4-1'
>     -- Hungup 'DAHDI/4-1'
>   == Everyone is busy/congested at this time (1:0/0/1)
>     -- Executing [s <at> DialOut:18] NoOp("DAHDI/1-1", ""Dialstatus is
> "CHANUNAVAIL"") in new stack

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               Tzafrir Cohen
icq#16849755              jabber:tzafrir.cohen <at> xorcom.com
+972-50-7952406           mailto:tzafrir.cohen <at> xorcom.com
http://www.xorcom.com  iax:guest <at> local.xorcom.com/tzafrir

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Anita Hall | 2 Apr 2012 12:18

fax tone testing

Hi

I suspect that my telco set-up is acting funny and I want to use spectral analysis to confirm the culprit :)

What is the best way to generate Fax tones from a dialplan and then record them at the other end? Also, where can I get a list of the all the tones and duration which are used in Fax.


Thanks.


regards,
Anita

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Gmane