[asterisk-users] Removing "Parsing /etc/asterisk/manager.conf" from CLI

[Adrian A]

Hello,

Is there any way of removing this line from showing on the console? I have a
script that logs in every few seconds to manager and it makes the CLI output
very hard to follow because of the "  == Parsing
'/etc/asterisk/manager.conf': Found". (Yes, Found! manager.conf was there 3
seconds ago, guess what it's still there.)

There is a very old feature request about this at
http://bugs.digium.com/view.php?id=3085 but I cannot see the resolution.
Mantis shows "APPLICATION ERROR #801" at the end of the page...

Adrian
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[asterisk-users] Custom multilingual VM announcements & writing VM to 'default' context

[Sergey Chernyavskiy]

Hi All,

I want callers to receive custom announcements in different languages 
and voicemail messages to be written to 'default' context.

Currently I record announcements in different contexts(en,de, etc.) and 
do something like: Voicemail([EMAIL PROTECTED]), but the message then got 
written 
in 'de' context.
I'd like to write it to the 'default' context. How do I do that?

Regards

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[asterisk-users] Maximum Include level (10) exceeded- question

[Matthew Mackes]

I have been operating an asterisk server for over a year now allowing 
this error to list several time in a row with every reload, however I 
have not had any issues in operation or dialplan.

Does anyone know if the error as an effect on operations?

Thank You,

Matt

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[asterisk-users] features on dial pad

[nhadie ramos]

Hi All,

If i were to develop a softphone, how can i add call transfer, call on
hold and 3-way conference on it? linksys Ip phone has those built-in button
to xfer, conf, on hold.
and x-lite also has those, how can i have those if i develop my own?

Thank You

Regards,
Nhadie
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Re: [asterisk-users] Jumped from 1.2.7 to 1.4.19, missing CLI colors

[Anthony Francis]

Tzafrir Cohen wrote:
> On Wed, Apr 09, 2008 at 08:00:38PM -0400, Mike wrote:
>   
>> Ah, not bad.   When I start asterisk with "/usr/sbin/asterisk -c" I get the
>> colors, but if I start it without -c and then connect to the console using
>> "/usr/sbin/asterisk -r" I get no color.
>>
>> Since I want this to be running in the background, how do I fix this so I
>> get to have my cake and eat it too?
>> 
>
> The patch is rather trivial. Just make Asterisk pretend that it is
> "vt100" (or whatever) if it is running as a service.
>
>   
I cant get color using asterisk -r on 1.2.17 or 18 either.

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Re: [asterisk-users] Jumped from 1.2.7 to 1.4.19, missing CLI colors

[Jay R. Ashworth]

On Wed, Apr 09, 2008 at 04:23:38PM -0800, Mojo with Horan & Company, LLC wrote:
> Mike wrote:
> > Ah, not bad.   When I start asterisk with "/usr/sbin/asterisk -c" I get the
> > colors, but if I start it without -c and then connect to the console using
> > "/usr/sbin/asterisk -r" I get no color.
> >
> > Since I want this to be running in the background, how do I fix this so I
> > get to have my cake and eat it too?
> >
> IIRC, using the utility 'screen' might work for you?

Indeed.  VICIdial runs asterisk behind screen, and I see colors in the
console.

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Joseph Stalin)

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Re: [asterisk-users] Half-duplex call on TDM2400p with VPMADT032

[Ruben Zamora]

Today I Install Zaptel 1.4.10 and compiled.No good result.

Then Digium Support send me the last firmware of VPMADT032, and 
installed, at the first sight there was no good news.

But then i move in the driver wcte12xp in the file base.c  and i have 
better results.



Matthew Fredrickson escribió:
> Faraz R. Khan wrote:
>   
>> The newer zaptel (1.4.10) says it includes firmware 1.16 from the
>> CHANGELOG:
>>
>>
>> firmware/Makefile, kernel/wctdm24xxp/base.c,
>>kernel/wctdm24xxp/GpakApi.c, kernel/wctdm24xxp/GpakApi.h: Update
>>wctdm24xxp's VPMADT032 firmware to version 1.16
>>
>>
>> However there seems to be no way to get this firmware and it does not seem 
>> to be included. It checks my firmware and says 1.07 is okay. 
>>
>> 
>
> We had to back that version of the firmware out due to release related 
> problems.  As for all problems related to the VPMADT032, if you have any 
> issues, please contact technical support.  They will be able to help you 
> with whatever issue you may have.
>
> Matthew Fredrickson
>
>   
>> The URL provided does not contain firmware for the VPMADT032
>>
>> I* have logged a query with digum. Is there a URL to get this firmware from?
>>
>> On Tue, 2008-04-08 at 11:12 -0500, Ruben Zamora wrote:
>> 
>>> Lex
>>>
>>> Thanks, I all ready download the last svn branches from zaptel And i 
>>> am going to test these afternoon.
>>>
>>> My phone number es 81-83481611.
>>>
>>> Thanks
>>>
>>> Ruben
>>>
>>> Lex Lethol escribió:
>>>   
 Ruben,

 I am also in Monterrey and have used digium hardware on R2 and PRI.
 MFC/R2 is not supported by digium but the zaptel driver requirement is
 the same.. what changes is using libpri vs unicall.

 Just go ahead and ask them for the firmware update or as Tzafir says
 use a newer zaptel that should include the updated firmware.

 If in trouble add me to gtalk I'll try to help out any way possible,

 Lex

 On Tue, Apr 8, 2008 at 1:52 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
   
 
> On Mon, Apr 07, 2008 at 09:22:30PM -0500, Ruben Zamora wrote:
>  > Lex
>  >
>  > Thanks a lot.   These morning i call Digium Support.   One issue that i
>  > miss in my before e-mail is that i have
>  > my Asterisk installed in Monterrey,Mexico and my TELCO provider gave my
>  > MFC/R2.
>  > Asterisk 1.4.18,Zaptel 1.4.9.2 and Unicall.
>  >
>  > They told me they can help me because they dont have UNICALL support.
>  >
>  > So... I need to investigate more or wait for a new zaptel or anything 
> else.
>
>  Generally you can always use a newer zaptel.
>
>  --
>Tzafrir Cohen
>  icq#16849755  jabber:[EMAIL PROTECTED]
>  +972-50-7952406   mailto:[EMAIL PROTECTED]
>  http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
>
>
>
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>
>
>   

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Re: [asterisk-users] Jumped from 1.2.7 to 1.4.19, missing CLI colors

[Tzafrir Cohen]

On Wed, Apr 09, 2008 at 08:00:38PM -0400, Mike wrote:
> Ah, not bad.   When I start asterisk with "/usr/sbin/asterisk -c" I get the
> colors, but if I start it without -c and then connect to the console using
> "/usr/sbin/asterisk -r" I get no color.
> 
> Since I want this to be running in the background, how do I fix this so I
> get to have my cake and eat it too?

The patch is rather trivial. Just make Asterisk pretend that it is
"vt100" (or whatever) if it is running as a service.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Jumped from 1.2.7 to 1.4.19, missing CLI colors

[Mojo with Horan & Company, LLC]

IIRC, using the utility 'screen' might work for you?

Moj


Mike wrote:
> Ah, not bad.   When I start asterisk with "/usr/sbin/asterisk -c" I get the
> colors, but if I start it without -c and then connect to the console using
> "/usr/sbin/asterisk -r" I get no color.
>
> Since I want this to be running in the background, how do I fix this so I
> get to have my cake and eat it too?
>
> Mike
>
>   
>> -Original Message-
>> From: [EMAIL PROTECTED] 
>> [mailto:[EMAIL PROTECTED] On Behalf Of 
>> Mik Cheez
>> Sent: Wednesday, April 09, 2008 19:06
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] Jumped from 1.2.7 to 1.4.19, 
>> missing CLI colors
>>
>> Correct me if I'm wrong, but if you run asterisk as a service 
>> this happens.  There is/was some dispute as to the fallacy of 
>> using 'safe_asterisk' anyway.
>>
>> Start it at the command line to see the pretty colors.
>>
>> Mike wrote:
>> 
>>> Hi,
>>>  
>>> I`ve just made a leap from * 1.2.7 to 1.4.19.  It took a 
>>>   
>> while to fix 
>> 
>>> all the deprecated stuff, but everything seems to be 
>>>   
>> working fine now, 
>> 
>>> except for a little tiny thing.  I lost all color in my CLI, which 
>>> makes it harder to debug.  Is there something that needs doing? I 
>>> didn't explicitely disable colorization from the command 
>>>   
>> line, and I 
>> 
>>> did try using nocolor=no in the config files. No luck.
>>>  
>>> Regards,
>>>  
>>> Mike
>>>
>>>
>>>
>>>   
>> --
>> 
>>> --
>>>
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>> http://www.api-digital.com --
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>> 
>
>
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>   


-- 

*Mojo Wentworth*
HORAN & COMPANY, LLC
403 Lincoln Street, Suite 210
Sitka, AK 99835
(907) 747-
(907) 747-7417 - Fax
[EMAIL PROTECTED]

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Re: [asterisk-users] Jumped from 1.2.7 to 1.4.19, missing CLI colors

[Mike]

Ah, not bad.   When I start asterisk with "/usr/sbin/asterisk -c" I get the
colors, but if I start it without -c and then connect to the console using
"/usr/sbin/asterisk -r" I get no color.

Since I want this to be running in the background, how do I fix this so I
get to have my cake and eat it too?

Mike

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Mik Cheez
> Sent: Wednesday, April 09, 2008 19:06
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Jumped from 1.2.7 to 1.4.19, 
> missing CLI colors
> 
> Correct me if I'm wrong, but if you run asterisk as a service 
> this happens.  There is/was some dispute as to the fallacy of 
> using 'safe_asterisk' anyway.
> 
> Start it at the command line to see the pretty colors.
> 
> Mike wrote:
> > Hi,
> >  
> > I`ve just made a leap from * 1.2.7 to 1.4.19.  It took a 
> while to fix 
> > all the deprecated stuff, but everything seems to be 
> working fine now, 
> > except for a little tiny thing.  I lost all color in my CLI, which 
> > makes it harder to debug.  Is there something that needs doing? I 
> > didn't explicitely disable colorization from the command 
> line, and I 
> > did try using nocolor=no in the config files. No luck.
> >  
> > Regards,
> >  
> > Mike
> > 
> > 
> > 
> --
> > --
> > 
> > ___
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
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> 


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Re: [asterisk-users] Jumped from 1.2.7 to 1.4.19, missing CLI colors

[Mik Cheez]

Correct me if I'm wrong, but if you run asterisk as a service this 
happens.  There is/was some dispute as to the fallacy of using 
'safe_asterisk' anyway.

Start it at the command line to see the pretty colors.

Mike wrote:
> Hi,
>  
> I`ve just made a leap from * 1.2.7 to 1.4.19.  It took a while to fix 
> all the deprecated stuff, but everything seems to be working fine now, 
> except for a little tiny thing.  I lost all color in my CLI, which makes 
> it harder to debug.  Is there something that needs doing? I didn't 
> explicitely disable colorization from the command line, and I did try 
> using nocolor=no in the config files. No luck.
>  
> Regards,
>  
> Mike
> 
> 
> 
> 
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Re: [asterisk-users] setting dtmf mode for a particular peer

[Brent Davidson]

Have you tried the using the "SIPDtmfMode" function in your dial plan?  
It can be used to change the DTMF mode between two points in a call.  
The problem, I would think, would be if your phones are set up to ONLY 
send inband audio then you have to find someway to get audio to 
transcode the DTMF from inband to info.  I'm not familiar enough with 
the specifics of Asterisk's behavior to know whether that "just works" 
or if it needs some special setup.  Try putting SipDtmfMode(info) just 
before the dial command and see what happens.


Good Luck,
Brent


Brian J. Murrell wrote:

On Wed, 2008-04-09 at 16:18 -0500, Tilghman Lesher wrote:
  

No, that's correct.  The problem is that you aren't using the peer definition
when you dial (as you said, you've never needed it before).

Use
Dial(SIP/[EMAIL PROTECTED])
NOT
Dial(SIP/[EMAIL PROTECTED])



OK.  Trying exactly as you describe above, it does dial:

-- Executing [EMAIL PROTECTED]:2] Dial("SIP/1011002206-b631f650", "SIP/[EMAIL 
PROTECTED]") in new stack

With "sip set debug peer voipmich" I'd expect to see SIP packets for
every digit I press on my phone, right?  I don't.  I don't see anything
beyond the initial call establishment:

Audio is at 67.193.45.68 port 11724
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Reliably Transmitting (NAT) to 69.41.0.50:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 67.193.45.68:5060;branch=z9hG4bK2a84f89b;rport
From: "2003" ;tag=as5c70ce0e
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX  
Max-Forwards: 70

Date: Wed, 09 Apr 2008 21:55:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 207

v=0
o=root 11375 11375 IN IP4 67.193.45.68
s=session
c=IN IP4 67.193.45.68
t=0 0
m=audio 11724 RTP/AVP 0 3 
a=rtpmap:0 PCMU/8000

a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - - 
a=ptime:20

a=sendrecv

---
-- Called [EMAIL PROTECTED]
-- SIP/voipmich-084a5500 is making progress passing it to 
SIP/1011002206-b631f650
  == Spawn extension (macro-ringingdial, s, 2) exited non-zero on 
'SIP/1011002206-b631f650' in macro 'ringingdial'
  == Spawn extension (macro-ringingdial, s, 2) exited non-zero on 
'SIP/1011002206-b631f650'

Of course, in there between the call being established and torn down, I
did hit lots of digits on my phone.

b.

  



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[asterisk-users] multiple simultaneous access to single voice mail box

[Bob Pierce]

We are using Asterisk 1.2.18 at this site. One of the users brought this
to my attention today.

"We have a problem when we take the message off the voice mail. If I am
taking off the messages it used to be [on the old phone system] that no
one else was able to go in & take off the message. Now I can be taking
off the messages & some one else can also be taking off the same
messages. We should not be able to do this!!"

Has anyone else seen this? Is there a way to setup the voice mail so
that each box can only be accessed by one person at a time?

Bob


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Re: [asterisk-users] setting dtmf mode for a particular peer

[Brian J. Murrell]

On Wed, 2008-04-09 at 16:18 -0500, Tilghman Lesher wrote:
> No, that's correct.  The problem is that you aren't using the peer definition
> when you dial (as you said, you've never needed it before).
> 
> Use
> Dial(SIP/[EMAIL PROTECTED])
> NOT
> Dial(SIP/[EMAIL PROTECTED])

OK.  Trying exactly as you describe above, it does dial:

-- Executing [EMAIL PROTECTED]:2] Dial("SIP/1011002206-b631f650", 
"SIP/[EMAIL PROTECTED]") in new stack

With "sip set debug peer voipmich" I'd expect to see SIP packets for
every digit I press on my phone, right?  I don't.  I don't see anything
beyond the initial call establishment:

Audio is at 67.193.45.68 port 11724
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Reliably Transmitting (NAT) to 69.41.0.50:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 67.193.45.68:5060;branch=z9hG4bK2a84f89b;rport
From: "2003" ;tag=as5c70ce0e
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX  
Max-Forwards: 70
Date: Wed, 09 Apr 2008 21:55:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 207

v=0
o=root 11375 11375 IN IP4 67.193.45.68
s=session
c=IN IP4 67.193.45.68
t=0 0
m=audio 11724 RTP/AVP 0 3 
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - - 
a=ptime:20
a=sendrecv

---
-- Called [EMAIL PROTECTED]
-- SIP/voipmich-084a5500 is making progress passing it to 
SIP/1011002206-b631f650
  == Spawn extension (macro-ringingdial, s, 2) exited non-zero on 
'SIP/1011002206-b631f650' in macro 'ringingdial'
  == Spawn extension (macro-ringingdial, s, 2) exited non-zero on 
'SIP/1011002206-b631f650'

Of course, in there between the call being established and torn down, I
did hit lots of digits on my phone.

b.



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Re: [asterisk-users] setting dtmf mode for a particular peer

[Brian J. Murrell]

On Wed, 2008-04-09 at 16:18 -0500, Tilghman Lesher wrote:
> 
> No, that's correct.  The problem is that you aren't using the peer definition
> when you dial (as you said, you've never needed it before).
> 
> Use
> Dial(SIP/[EMAIL PROTECTED])
> NOT
> Dial(SIP/[EMAIL PROTECTED])

Ugh.  That's a problem.  This is ENUM dialing.  So the only reason I my
dialplan will even use tf.voipmich.com is because:

$ enum_lookup 18881234567
7.6.5.4.3.2.1.8.8.8.1.e164.org has NAPTR record 10 100 "u" "E2U+SIP" 
"!^\\+1888(.*)$!sip:[EMAIL PROTECTED]" .
7.6.5.4.3.2.1.8.8.8.1.e164.org has NAPTR record 10 100 "u" "E2U+SIP" 
"!^\\+1888(.*)$!sip:[EMAIL PROTECTED]" .

So the ENUM macro takes the destination from the above NAPTR DNS result.

Any way to deal with that?  I've tried replacing the "[voipmich]" with
"[tf.voipmich.com]" but that doesn't seem to have done the trick either.

If there is a more general way to work around this problem, like by
autodetecting what a SIP server will use for DTMF, I'm all into that.
Having to make sip.conf entries for random SIP servers is a PITA and
doesn't scale.

b.



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Re: [asterisk-users] setting dtmf mode for a particular peer

[Tilghman Lesher]

On Wednesday 09 April 2008 14:12:08 Brian J. Murrell wrote:
> When I make a toll-free call using tf.voipmich.com DTMF doesn't work.
> According to this post:
> http://www.trixbox.org/forums/trixbox-forums/help/enum-strangeness it's
> because voipmich needs dtmfmode set to "info".
>
> How do I specify this for a single SIP peer (tf.voipmich.com) given that
> I normally don't register to them.
>
> I have tried creating a sip.conf entry:
>
> [voipmich]
> type=peer
> fromuser=nobody
> fromdomain=nodomain
> host=tf.voipmich.com
> dtmfmode=info
>
> But that does not appear to be working.  Maybe my approach is all wrong.
> Any ideas?

No, that's correct.  The problem is that you aren't using the peer definition
when you dial (as you said, you've never needed it before).

Use
Dial(SIP/[EMAIL PROTECTED])
NOT
Dial(SIP/[EMAIL PROTECTED])

-- 
Tilghman

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Re: [asterisk-users] RTCP not being sent when on hold

[Adrian A]

The RTP codec 126 is a bogus RTP packet sent by Bria to maintain the NAT
binding.

I've identified the issue as this:

Bria has an inactivity timer that is based on RTCP. Basically, if during the
call there is RTCP, Bria uses it to make sure the call is still alive.
Asterisk does send RTCP when call is active, but it stops when call is put
on hold by Bria. The default timeout for Bria is 30 seconds, thus it
disconnects the call because it has not received any RTP or RTCP during this
time.

I am not sure at this point which is correct implementation. Should the
client not rely on RTP/RTCP when it's on hold or should Asterisk send some
sort of keep alive RTP/RTCP when it knows one of the clients is on hold?


On Wed, Apr 9, 2008 at 7:15 AM, Steve Langstaff <[EMAIL PROTECTED]>
wrote:

>  It would be interesting to see a wireshark trace of the SIP and RTP
> traffic during call setup and hold, to see:
> a) what codec 126 has been negotiated as and
> b) who is sourcing the unknown RTP datagram.
>
>  --
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *Adrian A
> *Sent:* 09 April 2008 00:55
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] RTCP not being sent when on hold
>
> Hello,
>
> When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1 and I
> place the call on hold, the call is dropped after 30 seconds.
> It looks like there is no RTCP/RTP sent to the client from Asterisk while
> on hold (music on hold playing to caller) thus client disconnects the call.
> During this time, I get the following messages in the CLI:
>
> NOTICE[24194] rtp.c: Unknown RTP codec 126 received from '0.0.0.0'
>
> In sip.conf I have rtpkeepalive=15 but that does not seem to help.
>
> Does anyone know what I can do to fix this, other than increase the
> timeout on Bria?
>
> Thanks,
> Adrian
>
>
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[asterisk-users] Jumped from 1.2.7 to 1.4.19, missing CLI colors

[Mike]

Hi,
 
I`ve just made a leap from * 1.2.7 to 1.4.19.  It took a while to fix all
the deprecated stuff, but everything seems to be working fine now, except
for a little tiny thing.  I lost all color in my CLI, which makes it harder
to debug.  Is there something that needs doing? I didn't explicitely disable
colorization from the command line, and I did try using nocolor=no in the
config files. No luck.
 
Regards,
 
Mike
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Re: [asterisk-users] Queues +Exiting

[Rob Schall]

That fixed it. I always thought the "s" would be the fall back from all
extensions that didn't match. I guess that doesn't work in this case.

Thanks!
Rob


Guido Hecken wrote:
>> -Ursprüngliche Nachricht-
>> Von: Rob Schall [mailto:[EMAIL PROTECTED] 
>> Gesendet: Mittwoch, 9. April 2008 15:50
>> An: Asterisk Users Mailing List - Non-Commercial Discussion
>> Betreff: [asterisk-users] Queues +Exiting
>>
>> I'm having a problem getting my queue to function as it should.
>>
>> After 20 seconds or so, it should prompt the user with a 
>> message "thanks
>> for holding. press # to leave a message or stay on the line to
>> continue holding". I set up the "context" in the queues.conf 
>> file, so if
>> a user presses a digit, they should be able to leave. But I get a SIP
>> BUSY message.
>>
>> Here are my confs:
>>
>> queues.conf
>> [custserv]
>> music=default
>> strategy=ringall
>> ;timeout=10
>> retry=20
>> wrapuptime=0
>> maxlen=0
>> context = queue-out
>> periodic-announce=cont_holding
>> periodic-announce-frequency=15
>> ;announce-frequency=15
>> ;announce-holdtime=yes
>> member => SIP/2001
>> member => SIP/2002
>> member => SIP/1004
>> 
>
>
>   
>> extensions.conf
>> [queue-out]
>> exten => s,1,Voicemail(u${vmbox})
>> exten => s,2,Hangup
>> 
>
> Perhaps it's the "s" extension, did you try with
>
> exten => 1,1,Voicemail(u${vmbox})
> exten => 1,2,Hangup
>
>
> Regards,
>
> Guido
>  
> gwsNetTech
> Guido Hecken
>
> Quirrenbacher Str. 36
> 53639 Königswinter
> Germany
>
> fon +49(2244) 870663
> fax +49(2244) 870664
> mobil  +49(179) 1267353
> web http://www.gwsnettech.de
> mailto:[EMAIL PROTECTED]
>
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>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] dial tree crawler?

[Steve Totaro]

On Wed, Apr 9, 2008 at 4:02 PM, Curt Shaffer <[EMAIL PROTECTED]> wrote:
>
>
>
>
> For lack of a better term, I have been tasked with creating a "dial tree
> crawler". The reason is that we have a soon to fail Octel system. The major
> issue is that there is no way to port the dial tree recordings from the
> Octel. So what I envision is creating a script that can somehow dial down
> the trees spawning a recorded call for each tree. I realize that it will
> take time to cut these down and tidy them up but it will be a lot less work
> than recreating thousands of trees for sure.
>
>
>
> So I would need to figure out some way to realize what the options are and
> dial pass the DTMF to get into the subtree. Anyone out there ever had to do
> something similar or have any suggestions on how to accomplish this?
>
>
>
> Thanks.
>

I have not used an Octel system but a quick Google search leads me to
believe that has a harddrive, either 2.5 or 3.5.  Depending on the
filesystem, you could simply remove the drive, attach it to another
machine and copy everything off of it.

But before trying that, I would be surprised if there were no "backup" facility.

Anyways, if you want to use Asterisk, you may be able to use senddtmf
and monitor to get what you want.

Thanks,
Steve Totaro

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[asterisk-users] dial tree crawler?

[Curt Shaffer]

For lack of a better term, I have been tasked with creating a "dial tree
crawler". The reason is that we have a soon to fail Octel system. The major
issue is that there is no way to port the dial tree recordings from the
Octel. So what I envision is creating a script that can somehow dial down
the trees spawning a recorded call for each tree. I realize that it will
take time to cut these down and tidy them up but it will be a lot less work
than recreating thousands of trees for sure. 

 

So I would need to figure out some way to realize what the options are and
dial pass the DTMF to get into the subtree. Anyone out there ever had to do
something similar or have any suggestions on how to accomplish this?

 

Thanks.

 

 

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[asterisk-users] setting dtmf mode for a particular peer

[Brian J. Murrell]

When I make a toll-free call using tf.voipmich.com DTMF doesn't work.
According to this post:
http://www.trixbox.org/forums/trixbox-forums/help/enum-strangeness it's
because voipmich needs dtmfmode set to "info".

How do I specify this for a single SIP peer (tf.voipmich.com) given that
I normally don't register to them.

I have tried creating a sip.conf entry:

[voipmich]
type=peer
fromuser=nobody
fromdomain=nodomain
host=tf.voipmich.com
dtmfmode=info

But that does not appear to be working.  Maybe my approach is all wrong.
Any ideas?

b.



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Re: [asterisk-users] Attrafax

[Matt Watson]

I have a single channel license of Attrafax right now...


It seems to work well from the testing I have done with it so far, which 
admittedly isn't as much as I was hoping to have done at this time.

I;m using Linksys SPA2102 ATA's with it... basically what I;m doing is...


FAX Machine -> Linksys SPA2102 -> SIP/T.38 -> Asterisk -> TDM card (currently 
TDM800P + TDM400P, but moving to TE220B soon) -> PSTN


I had some trouble with Attrafax at first, but updating the firmware on my 
SPA2102 fixed the problem.  I've also tried interfacing a couple of Ricoh 
multifunction copiers with it (we have Ricoh MP 2500 & MP 5000 which can both 
talk SIP if you have the fax option)... I haven't had any luck at all getting 
T.38 negotation to happen between attrafax and the Ricoh's though... I kinda 
decided it wasn;t worth spending the time fiddling with it when I could just 
attach a SPA2102 to it for 80$

Attractel will gives you a 2-week demo license of Attrafax if you request it 
from them, if you want, I can send you the email address of the contact I have 
there, just shoot me an email off list if you want his contact info.

--
Matt

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Hammett
Sent: Wednesday, April 09, 2008 10:18 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Attrafax

Has anyone had any luck with Attrafax?  I'm looking to use it as the T.38 
gateway (PRI in, T.38 out).


--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com


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Re: [asterisk-users] zaptel 1.2.25 compilation error

[Vieri]

--- Tzafrir Cohen <[EMAIL PROTECTED]> wrote:

> Yeah, right. Fixed it a few hours after the release
> :-(
> 
> The fix (run from the top-level directory)
> 
>   svn diff -c 4157
> http://svn.digium.com/svn/zaptel/branches/1.2 |
> patch -p0
> 
> See
> http://svn.digium.com/view/zaptel?view=rev&rev=4157

Thanks.
Compiles ok.

Vieri


__
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Re: [asterisk-users] Half-duplex call on TDM2400p with VPMADT032

[Matthew Fredrickson]

Faraz R. Khan wrote:
> The newer zaptel (1.4.10) says it includes firmware 1.16 from the
> CHANGELOG:
> 
> 
> firmware/Makefile, kernel/wctdm24xxp/base.c,
> kernel/wctdm24xxp/GpakApi.c, kernel/wctdm24xxp/GpakApi.h: Update
> wctdm24xxp's VPMADT032 firmware to version 1.16
> 
> 
> However there seems to be no way to get this firmware and it does not seem to 
> be included. It checks my firmware and says 1.07 is okay. 
> 

We had to back that version of the firmware out due to release related 
problems.  As for all problems related to the VPMADT032, if you have any 
issues, please contact technical support.  They will be able to help you 
with whatever issue you may have.

Matthew Fredrickson

> 
> The URL provided does not contain firmware for the VPMADT032
> 
> I* have logged a query with digum. Is there a URL to get this firmware from?
> 
> On Tue, 2008-04-08 at 11:12 -0500, Ruben Zamora wrote:
>> Lex
>>
>> Thanks, I all ready download the last svn branches from zaptel And i 
>> am going to test these afternoon.
>>
>> My phone number es 81-83481611.
>>
>> Thanks
>>
>> Ruben
>>
>> Lex Lethol escribió:
>>> Ruben,
>>>
>>> I am also in Monterrey and have used digium hardware on R2 and PRI.
>>> MFC/R2 is not supported by digium but the zaptel driver requirement is
>>> the same.. what changes is using libpri vs unicall.
>>>
>>> Just go ahead and ask them for the firmware update or as Tzafir says
>>> use a newer zaptel that should include the updated firmware.
>>>
>>> If in trouble add me to gtalk I'll try to help out any way possible,
>>>
>>> Lex
>>>
>>> On Tue, Apr 8, 2008 at 1:52 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
>>>   
 On Mon, Apr 07, 2008 at 09:22:30PM -0500, Ruben Zamora wrote:
  > Lex
  >
  > Thanks a lot.   These morning i call Digium Support.   One issue that i
  > miss in my before e-mail is that i have
  > my Asterisk installed in Monterrey,Mexico and my TELCO provider gave my
  > MFC/R2.
  > Asterisk 1.4.18,Zaptel 1.4.9.2 and Unicall.
  >
  > They told me they can help me because they dont have UNICALL support.
  >
  > So... I need to investigate more or wait for a new zaptel or anything 
 else.

  Generally you can always use a newer zaptel.

  --
Tzafrir Cohen
  icq#16849755  jabber:[EMAIL PROTECTED]
  +972-50-7952406   mailto:[EMAIL PROTECTED]
  http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir



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-- 
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.

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Re: [asterisk-users] Asterisk 1.4.19 crash with Realtime using SIP peers

[Mark Michelson]

Mindaugas Kezys wrote:
> Hello,
> 
> Asterisk 1.4.19 crashes everytime using Realtime and SIP peers
> 
> gdb asterisk /tmp/coreXXX shows:
> 
> Program terminated with signal 11, Segmentation fault.
> #0  0xb6148968 in find_peer (peer=0xb6042768 "test", sin=0x0, realtime=1) at
> chan_sip.c:2547
> 2547if (!(hp =
> ast_gethostbyname(tmp->value, &ahp)) || (memcmp(&hp->h_addr, &sin->sin_addr,
> sizeof(hp->h_addr {
> 
> 
> Sorry, I have no time to read manual how to correctly put this into bug
> tracker.
> 
> 
> Back to 1.4.18.1
> 
> 
> Regards,
> Mindaugas Kezys
> http://www.kolmisoft.com
> 

For those following this issue, there was a bug filed for this (issue #12362: 
http://bugs.digium.com/view.php?id=12362) and it has been fixed, too (Asterisk 
1.4 svn revision 113240).

Mark Michelson

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Re: [asterisk-users] Asterisk 1.4.19 crash with Realtime using SIP peers

[Atis Lezdins]

On Wed, Apr 9, 2008 at 5:29 PM, Trevor Peirce <[EMAIL PROTECTED]> wrote:
> Mindaugas Kezys wrote:
>  > Hello,
>  >
>  > Asterisk 1.4.19 crashes everytime using Realtime and SIP peers
>  >
>  Yes I also saw this and had to revert. Calls to the IVR seemed to be
>  fine, but as soon as two peers call each other it crashes as the call
>  progresses (never connects). I haven't had a chance to explore any
>  further and therefore haven't posted a bug either. Perhaps this weekend
>  if nobody does first.


So far works fine for me. Sample peer setup below. Had one issue with
peers where ipaddr was 0 (and hostname used instead), but adding this
patch ( 
http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?r1=113012&r2=113240
) seems to solve everything.

Regards,
Atis

*** 1. row ***
id: 2
  name: 21168
   accountcode: NULL
  amaflags: NULL
 callgroup: NULL
  callerid: Atis <21168>
   canreinvite: no
   context: default-sip
 defaultip: NULL
  dtmfmode: rfc2833
  fromuser: NULL
fromdomain: NULL
   fullcontact: sip:[EMAIL PROTECTED]:5061
  host: dynamic
  insecure: NULL
  language: NULL
   mailbox: [EMAIL PROTECTED]
 md5secret: NULL
   nat: yes
  deny: NULL
permit: NULL
  mask: NULL
   pickupgroup: NULL
  port: 5061
   qualify: no
   restrictcid: NULL
rtptimeout: NULL
rtpholdtimeout: NULL
secret: 21168
  type: friend
  username: 21168
  disallow:
 allow: all
   musiconhold: NULL
regseconds: 1207763735
ipaddr: 192.168.1.123
  regexten:
cancallforward: yes
setvar:
call-limit: 4

-- 
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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Re: [asterisk-users] Queues +Exiting

[Guido Hecken]

> -Ursprüngliche Nachricht-
> Von: Rob Schall [mailto:[EMAIL PROTECTED] 
> Gesendet: Mittwoch, 9. April 2008 15:50
> An: Asterisk Users Mailing List - Non-Commercial Discussion
> Betreff: [asterisk-users] Queues +Exiting
> 
> I'm having a problem getting my queue to function as it should.
> 
> After 20 seconds or so, it should prompt the user with a 
> message "thanks
> for holding. press # to leave a message or stay on the line to
> continue holding". I set up the "context" in the queues.conf 
> file, so if
> a user presses a digit, they should be able to leave. But I get a SIP
> BUSY message.
> 
> Here are my confs:
> 
> queues.conf
> [custserv]
> music=default
> strategy=ringall
> ;timeout=10
> retry=20
> wrapuptime=0
> maxlen=0
> context = queue-out
> periodic-announce=cont_holding
> periodic-announce-frequency=15
> ;announce-frequency=15
> ;announce-holdtime=yes
> member => SIP/2001
> member => SIP/2002
> member => SIP/1004


> extensions.conf
> [queue-out]
> exten => s,1,Voicemail(u${vmbox})
> exten => s,2,Hangup

Perhaps it's the "s" extension, did you try with

exten => 1,1,Voicemail(u${vmbox})
exten => 1,2,Hangup


Regards,

Guido
 
gwsNetTech
Guido Hecken

Quirrenbacher Str. 36
53639 Königswinter
Germany

fon +49(2244) 870663
fax +49(2244) 870664
mobil  +49(179) 1267353
web http://www.gwsnettech.de
mailto:[EMAIL PROTECTED]

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[asterisk-users] Magic Jack>>>> usable in Asterisk ?

[nigel.dennis]

Hi Everyone,
    Received my Magic Jack over a week now. Putting it to the test. I have set up my own version of it where I call it my " Travel Jack". I have added my free U.S # given to me by Magic Jack to my asterisk server. I now have mine where I have a number In Canada, Jamaica, U.S so I can have friends call me on it when I am in any of these countries or simply call me locally from these countries as well. With the magic of asterisk you can have a phone # from any country if the DID is available.I have also used it with a softphone from my laptop in a free hotspot. Awesome sound quality. Unfortunately you can only use the call forwarding feature, which they do not openly mention available, from U.S and Canada only. Will be teaching my Magic Jack new 
tricks as time passes...like using it remotely for free unlimited calls from my cell phone as a callback. please also share some positive experiences with this device if anyone have any.
 
 
Nigel Dennis 
425 906 4748


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Re: [asterisk-users] [VOIP-Users-Conference] Potential subject for Friday - Does the Asterisk community need a 3rd party commercial software ecosystem?

[Dean Collins]

Zoa,
Private ping about SF app exchange.


Are you interested in coming on to the call to discuss with other
potential developers how this registration works and how you can bind a
license to a particular server?

I'll also reach out to Gerd at Lumenvox to see if he can come on to
discuss as well.



Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Zoa
> Sent: Wednesday, 9 April 2008 12:39 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Cc: [EMAIL PROTECTED];
[EMAIL PROTECTED]
> Subject: Re: [asterisk-users] [VOIP-Users-Conference] Potential
subject for Friday -
> Does the Asterisk community need a 3rd party commercial software
ecosystem?
> 
> 
> You need to talk about this with digium sales, i suggest you give  Jim
> Webster a call.
> Without going into details, other license agreements are possible.
> (Several companies including mine can distribute software as a module
> similar to how g729 is being sold - we do that for our t.30/t.38 fax
> modules, lumenvox does it for voice recognition,... ).
> 
> OT, where can i find the best info on this salesforce API ? Do you see
> any possibilities to integrate our zoiper softphone with salesforce ?
> (contact me off list for that)
> 
> Cheers,
> 
> Zoa
> 
> Dean Collins wrote:
> > Hi BJ,
> > Further explanation about the 3rd party ecosystem question this
morning
> >
> > Cory Andrews from VoIP supply was on the Voip-Users conference call
last
> > week.
> >
> > I asked the question - how much of VoIP Supply revenue is product
versus
> > applications - he said we don't sell any services such as ITSP
hosted
> > Asterisk so I replied that wasn't what I was thinking of and gave
the
> > example of Snap Dialer which is a low cost (I think I paid $20 for
it)
> > application which allows me to dial names from outlook.
> >
> > I then talked about some of the consulting I did for Salesforce.com
and
> > how they have built an entire ecosystem of third party applications
all
> > built by other people but utilizing the documented API's and
application
> > security etc.
> >
> > My comments were that although Asterisk should always remain a free
open
> > source application that developers need to eat and pay rent as well.
> >
> > If there was some common marketplace that developers could sell
small -
> > low cost third party applications to the Asterisk community that
Digium
> > had some type of overview/management control over who listed etc
that
> > this would deliver a stream of revenue that would encourage further
> > application development.
> >
> > The question I then posed to the group was if anyone knew how Digium
> > managed the sale and licensing of the G729 codes.
> >
> > And if this was an open published standard that it could be used as
the
> > basis for the Asterisk ecosystem license model.
> >
> > Now I know it's not perfect and can be hacked but everything can be
> > hacked. The idea is to build apps cheap enough that it's not worth
the
> > effort of hacking.
> >
> > I know there were discussions in Mexuar about how we could sell
(read
> > license) a single channel of the Mexuar Corraleta application rather
> > than the entire server license for $2000.
> >
> > Earlier this week I sent an original email to Digium and told Kevin
was
> > responsible for the G729 licenses so I was hoping that this Friday
we
> > could get Kevin and possibly the developers of Snap Dialer to talk
about
> > their current license models and how they implemented payment
systems
> > and also maybe the developer of FOP to discuss if this was available
to
> > him and he was able to sell a 100 licenses or something like that a
> > month would this provide an income stream to support further
development
> > etc.
> >
> > Does this make sense?
> > Does anyone have any comments or would you like to be involved with
> > Fridays call?
> >
> >
> >
> >
> > Regards,
> >
> > Dean Collins
> > Cognation Pty Ltd
> > [EMAIL PROTECTED]
> > +1-212-203-4357 Ph
> > +61-2-9016-5642 (Sydney in-dial).
> >
> >
> >> -Original Message-
> >> From: Dean Collins
> >> Sent: Wednesday, 9 April 2008 7:53 AM
> >> To: '[EMAIL PROTECTED]'
> >> Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion';
Ast-biz
> >>
> > (asterisk-
> >
> >> [EMAIL PROTECTED])
> >> Subject: RE: [VOIP-Users-Conference] Subject or Guest for Friday?
> >>
> >> We can talk about the third party application idea Cory and I
> >>
> > discussed last week -
> >
> >> sorry still haven't posted to the list yet as I've had a client in
> >>
> > town for the last few day
> >
> >> but should be able to do this today.
> >> If Fop, Snap Dialer, Mexuar and or any other third party developer
who
> >>
> > has written
> >
> >> commercial asterisk applications (preferably small low value eg $20
> >>
> > etc) is interested
> >
> >> in jumping on a call with me this 

Re: [asterisk-users] zaptel 1.2.25 compilation error

[Tzafrir Cohen]

On Wed, Apr 09, 2008 at 09:39:00AM -0700, Vieri wrote:
> Zaptel seems to compile fine until I enter xpp/utils
> and make there.
> I get:
> 
> xpp/utils # make
> cc -o print_modes -g -Wall  print_modes.c
> print_modes.c: In function `main':
> print_modes.c:9: error: `fxo_modes' undeclared (first
> use in this function)
> print_modes.c:9: error: (Each undeclared identifier is
> reported only once
> print_modes.c:9: error: for each function it appears
> in.)
> print_modes.c:9: error: invalid application of
> `sizeof' to incomplete type `fxo_mode'
> print_modes.c:9: warning: division by zero
> make: *** [print_modes] Error 1

Yeah, right. Fixed it a few hours after the release :-(

The fix (run from the top-level directory)

  svn diff -c 4157 http://svn.digium.com/svn/zaptel/branches/1.2 | patch -p0

See http://svn.digium.com/view/zaptel?view=rev&rev=4157

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] [VOIP-Users-Conference] Potential subject for Friday - Does the Asterisk community need a 3rd party commercial software ecosystem?

[Zoa]

You need to talk about this with digium sales, i suggest you give  Jim 
Webster a call.
Without going into details, other license agreements are possible. 
(Several companies including mine can distribute software as a module 
similar to how g729 is being sold - we do that for our t.30/t.38 fax 
modules, lumenvox does it for voice recognition,... ).

OT, where can i find the best info on this salesforce API ? Do you see 
any possibilities to integrate our zoiper softphone with salesforce ? 
(contact me off list for that)

Cheers,

Zoa

Dean Collins wrote:
> Hi BJ,
> Further explanation about the 3rd party ecosystem question this morning
>
> Cory Andrews from VoIP supply was on the Voip-Users conference call last
> week.
>
> I asked the question - how much of VoIP Supply revenue is product versus
> applications - he said we don't sell any services such as ITSP hosted
> Asterisk so I replied that wasn't what I was thinking of and gave the
> example of Snap Dialer which is a low cost (I think I paid $20 for it)
> application which allows me to dial names from outlook.
>
> I then talked about some of the consulting I did for Salesforce.com and
> how they have built an entire ecosystem of third party applications all
> built by other people but utilizing the documented API's and application
> security etc.
>
> My comments were that although Asterisk should always remain a free open
> source application that developers need to eat and pay rent as well.
>
> If there was some common marketplace that developers could sell small -
> low cost third party applications to the Asterisk community that Digium
> had some type of overview/management control over who listed etc that
> this would deliver a stream of revenue that would encourage further
> application development.
>
> The question I then posed to the group was if anyone knew how Digium
> managed the sale and licensing of the G729 codes.
>
> And if this was an open published standard that it could be used as the
> basis for the Asterisk ecosystem license model.
>
> Now I know it's not perfect and can be hacked but everything can be
> hacked. The idea is to build apps cheap enough that it's not worth the
> effort of hacking.
>
> I know there were discussions in Mexuar about how we could sell (read
> license) a single channel of the Mexuar Corraleta application rather
> than the entire server license for $2000.
>
> Earlier this week I sent an original email to Digium and told Kevin was
> responsible for the G729 licenses so I was hoping that this Friday we
> could get Kevin and possibly the developers of Snap Dialer to talk about
> their current license models and how they implemented payment systems
> and also maybe the developer of FOP to discuss if this was available to
> him and he was able to sell a 100 licenses or something like that a
> month would this provide an income stream to support further development
> etc.
>
> Does this make sense?
> Does anyone have any comments or would you like to be involved with
> Fridays call?
>
>
>
>
> Regards,
>
> Dean Collins
> Cognation Pty Ltd
> [EMAIL PROTECTED]
> +1-212-203-4357 Ph
> +61-2-9016-5642 (Sydney in-dial).
>
>   
>> -Original Message-
>> From: Dean Collins
>> Sent: Wednesday, 9 April 2008 7:53 AM
>> To: '[EMAIL PROTECTED]'
>> Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'; Ast-biz
>> 
> (asterisk-
>   
>> [EMAIL PROTECTED])
>> Subject: RE: [VOIP-Users-Conference] Subject or Guest for Friday?
>>
>> We can talk about the third party application idea Cory and I
>> 
> discussed last week -
>   
>> sorry still haven't posted to the list yet as I've had a client in
>> 
> town for the last few day
>   
>> but should be able to do this today.
>> If Fop, Snap Dialer, Mexuar and or any other third party developer who
>> 
> has written
>   
>> commercial asterisk applications (preferably small low value eg $20
>> 
> etc) is interested
>   
>> in jumping on a call with me this Friday I can put together a series
>> 
> of questions to run
>   
>> the call with.
>> Maybe we can get Kevin from Digium to explain on the call how the g729
>> 
> license
>   
>> registration process works and turn the call into a working
>> 
> discussion.
>   
>> Regards,
>>
>> Dean Collins
>> Cognation Pty Ltd
>> [EMAIL PROTECTED]
>> +1-212-203-4357 Ph
>> +61-2-9016-5642 (Sydney in-dial).
>>
>>
>> 
>>> -Original Message-
>>> From: [EMAIL PROTECTED] [mailto:VOIP-Users-
>>> [EMAIL PROTECTED] On Behalf Of randulo
>>> Sent: Wednesday, 9 April 2008 7:27 AM
>>> To: VOIP Users Conference
>>> Subject: [VOIP-Users-Conference] Subject or Guest for Friday?
>>>
>>>
>>> Hi,
>>>
>>> Does anyone have a guest or subject up their sleeve for this week? I
>>> have neither and I'm getting ready to move house so I don't have a
>>>   
> lot
>   
>>> of time to pursue. Any chance of getting a phone mfr or service
>>> provider on? Anyone out there want to take a crack at this? Present
>>> your pro

[asterisk-users] zaptel 1.2.25 compilation error

[Vieri]

Zaptel seems to compile fine until I enter xpp/utils
and make there.
I get:

xpp/utils # make
cc -o print_modes -g -Wall  print_modes.c
print_modes.c: In function `main':
print_modes.c:9: error: `fxo_modes' undeclared (first
use in this function)
print_modes.c:9: error: (Each undeclared identifier is
reported only once
print_modes.c:9: error: for each function it appears
in.)
print_modes.c:9: error: invalid application of
`sizeof' to incomplete type `fxo_mode'
print_modes.c:9: warning: division by zero
make: *** [print_modes] Error 1



__
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Re: [asterisk-users] Zaptel 1.2.25 and 1.4.10 released

[Brent Davidson]

Jason Parker wrote:

Brent Davidson wrote:
 > Do they mean 1.4.20 instead of 1.4.10?  If not, then this message was
  

seriously delayed  :-D

-Brent



Zaptel, not Asterisk. :)

1.4.10 is correct.


Doh!  My bad.Was looking at the wrong version numbers.  As many 
times as I've recompiled both Asterisk and Zaptel in the past 3 weeks 
you'd think I'd know the versions by hear by now.  LOL.


-Brent
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Re: [asterisk-users] DTMF between Asterisk servers.

[Mark Hamilton]

No, I tried calling the inbound DID to see if DTMF passes through. And most
times it does, however, it's not being relayed to the Asterisk server 2, and
then to the direct external phoneline.

I tried changing all dtmfmodes for the sip peer, for the inbound DID
provider, and it didn't work, even tried playing with canreinvite, etc.

Hence why my desperate plea for help.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies
Sent: April 8, 2008 11:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DTMF between Asterisk servers.

I believe that what you described should "just work" with the caveat
that "dtmf=inband" is rarely the right thing to do over SIP, and is
prone to all sorts of DTMF detection and debounce issues.

I assume you've tried calling a POTS endpoint and listening to see if
you get DTMF passed through?

1) You did not give a great deal of information about what the current
situation was, or what investigations you've already tried, which is
probably why no-one felt they could reply.
2) It may also have been because less than 23 hours had elapsed...

Regards,
Steve

On 08/04/2008, Mark Hamilton <[EMAIL PROTECTED]> wrote:
>
> I find it  hard to believe no one knows, so is it just plain no helping? J
>
> If someone would like to atleast point me in the right direction that will
> deal specifically with what I'm asking, that would be appreciated too.
>
> Much thanks.
>
> From: Mark Hamilton [mailto:[EMAIL PROTECTED]
>  Sent: April 7, 2008 11:48 AM
>  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>  Subject: DTMF between Asterisk servers.
>
> Hello,
>
> I'm a little confused on DTMF.
>
> A sip peer is registered on two Asterisk servers. No dtmfmode is set for
> them, the sip peer is 999 on Asterisk 1 and 999 on Asterisk 2. They both
> register on each other.
>
>
>
> A call comes in on Asterisk server 1, provider 1, dtmf=inband. Then the
call
> is transferred to Asterisk 2:
>
>
RetryDial(/var/lib/asterisk/sounds/connecting,15,10,SIP/[EMAIL PROTECTED],,t
T,)
>
> Where 12351 accepts the call on Asterisk 2, and in some cases, that call
is
> transferred out to a PSTN number, or wherever, but not within Asterisk
> anymore via provider2, dtmf=rfc2833.
>
> When the call comes in, I'd like it to relay DTMF just dandy. How can I do
> so?
>
> There is no NAT between the Asterisk servers or in front of them. However,
> Asterisk2 has iptables which allows all UDP traffic  to/fro Asterisk1.
When
> Asterisk2 transfers the call to external endpoints, there might be a LAN,
> but relative ports are open on those LANs.
>
> Please help.
>
> Thanks in advance,
>
> Mark.

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Re: [asterisk-users] Zaptel 1.2.25 and 1.4.10 released

[Dave Cotton]

Brent Davidson wrote:
> Asterisk Development Team wrote:
>> The Asterisk.org development team has announced the release of Zaptel 
>> versions 1.2.25 and 1.4.10. These releases contain many bug fixes as 
>> well as performance enhancements.
>>
>> A couple of the more major changes include: modifications to the 
>> wctdm24xxp and wcte12xp drivers to increase interrupt latency 
>> resilience, numerous bug fixes and updates to the xpp drivers, as well 
>> as some Makefile updates.  For further details and a more complete list 
>> see the respective Changelog files.
>>
>> Both releases are available as a tarball as well as a patch against the 
>> previous release. They are available for download from downloads.digium.com.
>>
>> Thank you for your support!
>>
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>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>   
> 
> Do they mean 1.4.20 instead of 1.4.10?  If not, then this message was 
> seriously delayed  :-D
> 
> -Brent
> 
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Pity you didn't _read_ the message and then _read_ it again before 
hitting the keys.

D Cotton


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Re: [asterisk-users] Zaptel 1.2.25 and 1.4.10 released

[Tzafrir Cohen]

On Wed, Apr 09, 2008 at 10:32:54AM -0500, Brent Davidson wrote:
> Asterisk Development Team wrote:
> > The Asterisk.org development team has announced the release of Zaptel 
> > versions 1.2.25 and 1.4.10. These releases contain many bug fixes as 
> > well as performance enhancements.
>
> Do they mean 1.4.20 instead of 1.4.10?  If not, then this message was 
> seriously delayed  :-D

They mean Zaptel, rather than Asterisk.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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[asterisk-users] Connecting Asterisk to Nortel Succession 4.0 sip...

[Eugen Soare]

Well I am entering into a realm that I don't know.


3 sites with Asterisk
1 site with Nortel


Asterisk/Sip calls working fine between the 3 sites.

Asterisk to Nortel set calls working fine.  (call comes from asterisk to 
nortel and rings telephone, people answer and talk happens, hangup call 
clears)

Nortel to Asterisk. Set on Nortel gets a busy signal.

Any suggestions on what to look for?

Much appreciated!

Eugen

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Re: [asterisk-users] [VOIP-Users-Conference] Potential subject for Friday - Does the Asterisk community need a 3rd party commercial software ecosystem?

[Dean Collins]

Hi BJ,
Further explanation about the 3rd party ecosystem question this morning

Cory Andrews from VoIP supply was on the Voip-Users conference call last
week.

I asked the question - how much of VoIP Supply revenue is product versus
applications - he said we don't sell any services such as ITSP hosted
Asterisk so I replied that wasn't what I was thinking of and gave the
example of Snap Dialer which is a low cost (I think I paid $20 for it)
application which allows me to dial names from outlook.

I then talked about some of the consulting I did for Salesforce.com and
how they have built an entire ecosystem of third party applications all
built by other people but utilizing the documented API's and application
security etc.

My comments were that although Asterisk should always remain a free open
source application that developers need to eat and pay rent as well.

If there was some common marketplace that developers could sell small -
low cost third party applications to the Asterisk community that Digium
had some type of overview/management control over who listed etc that
this would deliver a stream of revenue that would encourage further
application development.

The question I then posed to the group was if anyone knew how Digium
managed the sale and licensing of the G729 codes.

And if this was an open published standard that it could be used as the
basis for the Asterisk ecosystem license model.

Now I know it's not perfect and can be hacked but everything can be
hacked. The idea is to build apps cheap enough that it's not worth the
effort of hacking.

I know there were discussions in Mexuar about how we could sell (read
license) a single channel of the Mexuar Corraleta application rather
than the entire server license for $2000.

Earlier this week I sent an original email to Digium and told Kevin was
responsible for the G729 licenses so I was hoping that this Friday we
could get Kevin and possibly the developers of Snap Dialer to talk about
their current license models and how they implemented payment systems
and also maybe the developer of FOP to discuss if this was available to
him and he was able to sell a 100 licenses or something like that a
month would this provide an income stream to support further development
etc.

Does this make sense?
Does anyone have any comments or would you like to be involved with
Fridays call?




Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).

> -Original Message-
> From: Dean Collins
> Sent: Wednesday, 9 April 2008 7:53 AM
> To: '[EMAIL PROTECTED]'
> Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'; Ast-biz
(asterisk-
> [EMAIL PROTECTED])
> Subject: RE: [VOIP-Users-Conference] Subject or Guest for Friday?
> 
> We can talk about the third party application idea Cory and I
discussed last week -
> sorry still haven't posted to the list yet as I've had a client in
town for the last few day
> but should be able to do this today.
> If Fop, Snap Dialer, Mexuar and or any other third party developer who
has written
> commercial asterisk applications (preferably small low value eg $20
etc) is interested
> in jumping on a call with me this Friday I can put together a series
of questions to run
> the call with.
> Maybe we can get Kevin from Digium to explain on the call how the g729
license
> registration process works and turn the call into a working
discussion.
> 
> Regards,
> 
> Dean Collins
> Cognation Pty Ltd
> [EMAIL PROTECTED]
> +1-212-203-4357 Ph
> +61-2-9016-5642 (Sydney in-dial).
> 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:VOIP-Users-
> > [EMAIL PROTECTED] On Behalf Of randulo
> > Sent: Wednesday, 9 April 2008 7:27 AM
> > To: VOIP Users Conference
> > Subject: [VOIP-Users-Conference] Subject or Guest for Friday?
> >
> >
> > Hi,
> >
> > Does anyone have a guest or subject up their sleeve for this week? I
> > have neither and I'm getting ready to move house so I don't have a
lot
> > of time to pursue. Any chance of getting a phone mfr or service
> > provider on? Anyone out there want to take a crack at this? Present
> > your product or service or open source contributions.
> >
> > Incidentally, TringMe apparently has an API available and are making
> > it very easy to build AIR and Flash (or Flex) apps. This sounds very
> > interesting for creating branded clients of various kinds. If you're
a
> > programmer in that world, I'd check it out. AIR looks to me like it
> > has legs because of its cross-platform nature. I don't know how it
> > works with linux or FreeBSD if at all. Does anyone here know?
> >
> > If no ideas are forthcoming, we can talk about my personal (VoIP)
> > issues: we're moving and will have two offices, as now. The
difference
> > is, we can not leave the power on at the old office, so we'll be
using
> > a combination of hosted pbx and asterisk, probably with the AA50 at
> > our new home.
> >
> > Thanks in advance for any suggestions on a

Re: [asterisk-users] Zaptel 1.2.25 and 1.4.10 released

[Jason Parker]

Brent Davidson wrote:
 > Do they mean 1.4.20 instead of 1.4.10?  If not, then this message was
> seriously delayed  :-D
> 
> -Brent

Zaptel, not Asterisk. :)

1.4.10 is correct.

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[asterisk-users] PrivacyManager not working

[Jaap Winius]

Hi list,

On my system, PrivacyManager is not reacting to anonymous calls.  
Whenever I dial into my system with my mobile phone's number hidden,  
the CLI message "CallerID Present: Skipping" shows up and and my SIP  
phone rings anyway.

Perhaps the cause is due to the fact that when there is no CID, the  
results are not always the same. For example, I see in my CDR database  
that when anonymous calls come in via the SIP channel, the clid field  
shows:

  "Anonymous" 

However, when anonymous calls came in though my old ISDN line (which I  
can't test anymore because it no longer exists), the clid field would  
show:

  CID withheld

Although I'm not sure, I suspect that PrivacyManager recognizes the  
latter format, but not the former. Has anyone else experienced this  
problem, or know of a fix or workaround?

FYI: I'm using Asterisk 1.4.19 and anonymous calls only come in via SIP.

Thanks!

Jaap


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Re: [asterisk-users] Zaptel 1.2.25 and 1.4.10 released

[Brent Davidson]

Asterisk Development Team wrote:
> The Asterisk.org development team has announced the release of Zaptel 
> versions 1.2.25 and 1.4.10. These releases contain many bug fixes as 
> well as performance enhancements.
>
> A couple of the more major changes include: modifications to the 
> wctdm24xxp and wcte12xp drivers to increase interrupt latency 
> resilience, numerous bug fixes and updates to the xpp drivers, as well 
> as some Makefile updates.  For further details and a more complete list 
> see the respective Changelog files.
>
> Both releases are available as a tarball as well as a patch against the 
> previous release. They are available for download from downloads.digium.com.
>
> Thank you for your support!
>
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>   

Do they mean 1.4.20 instead of 1.4.10?  If not, then this message was 
seriously delayed  :-D

-Brent

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Re: [asterisk-users] queue logging

[Scott Wolfe]

You could ASTassistant to see this. Its Freeware.
www.astassistant.com

  - Original Message - 
  From: Arjan Kroon | Mobillion 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Wednesday, April 09, 2008 1:01 AM
  Subject: [asterisk-users] queue logging


  Hi,

   

  I' using with asterisk a queue with tree members and round robin.

  When a caller enters this queue and it is connecting to one of the members, 
is there a possibility to see which member the caller is connected to?

   

  And is there a way to see in de application to see if the connection from the 
caller to one of the members was successful of not successful?

   

  I know you can see it in de queue. log.

  But I want to know if I can see it also in the hangup (h) in de application?

   

  Kind Regards



   



--


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Re: [asterisk-users] RTCP not being sent when on hold

[Steve Langstaff]

It would be interesting to see a wireshark trace of the SIP and RTP
traffic during call setup and hold, to see:
a) what codec 126 has been negotiated as and
b) who is sourcing the unknown RTP datagram.




From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adrian A
Sent: 09 April 2008 00:55
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] RTCP not being sent when on hold


Hello,

When I receive a call to my CounterPath Bria from Asterisk
1.4.18.1 and I place the call on hold, the call is dropped after 30
seconds.
It looks like there is no RTCP/RTP sent to the client from
Asterisk while on hold (music on hold playing to caller) thus client
disconnects the call. During this time, I get the following messages in
the CLI:

NOTICE[24194] rtp.c: Unknown RTP codec 126 received from
'0.0.0.0'

In sip.conf I have rtpkeepalive=15 but that does not seem to
help.

Does anyone know what I can do to fix this, other than increase
the timeout on Bria?

Thanks,
Adrian


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Re: [asterisk-users] Asterisk 1.4.19 crash with Realtime using SIP peers

[Trevor Peirce]

Mindaugas Kezys wrote:
> Hello,
>
> Asterisk 1.4.19 crashes everytime using Realtime and SIP peers
>   
Yes I also saw this and had to revert. Calls to the IVR seemed to be 
fine, but as soon as two peers call each other it crashes as the call 
progresses (never connects). I haven't had a chance to explore any 
further and therefore haven't posted a bug either. Perhaps this weekend 
if nobody does first.



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Re: [asterisk-users] queue logging

[Matt King]

Hello Arjan,

You can see who is in the queue, which agent is ringing, whether the 
agent is Paused, and which agent is connected to which caller, using 
OrderlyStats (FREE sign up at http://www.orderlyq.com/orderlystats.html 
). This is shown in Real Time, and also in the call history logs.

This will also show you whether the connection succeeded or failed 
as requested.

Kind regards,

   Matt -- OrderlyQ Support.

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[asterisk-users] Attrafax

[Mike Hammett]

Has anyone had any luck with Attrafax?  I'm looking to use it as the T.38 
gateway (PRI in, T.38 out).


--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com

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Re: [asterisk-users] PINCH: Reply-to-munging

[Jay R. Ashworth]

On Wed, Apr 09, 2008 at 03:38:31PM +0200, Benny Amorsen wrote:
> Tilghman Lesher <[EMAIL PROTECTED]> writes:
> > Which is why 'rm' ALWAYS prompts for confirmation, right?  Oh, wait, it
> > doesn't.  ;-)
> 
> Confirmation is useless, it just teaches everyone to blindly type y.

"Are you sure you want to delete $REALLY_IMPORTANT_FILE?"  y

"Are you really sure??" y

"You won't be able to get it back, you know..." yes, yes, just delete it

"Deleted $REALLY_IMPORTANT FILE.  Thanks for playing."  Oh, pants.

> Analogies suck, of course.

Yep, but posited dialogues are fun.  :-)

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Joseph Stalin)

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Re: [asterisk-users] *21* # diverting

[Mindaugas Kezys]

Google is your friend:
http://www.voip-info.org/wiki/view/Asterisk+call+forwarding

 

 

Regards,

Mindaugas Kezys

http://www.kolmisoft.com

MOR PRO - Advanced VoIP Billing Solution

 

 

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of gilbert
saunders
Sent: Wednesday, April 09, 2008 4:50 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] *21* # diverting

 

hi me again im new at asterisk and really need some good tutoring on
asterisk and call forwarding i dont understand it at all pls help i have
attached my extensions.conf file if someone would be so kind to look at it
and tell me what code i must enter to make *21* diverting and #21#
undiverting possible

 __
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Re: [asterisk-users] Catch end of Eagi script when caller hung up...HELP ME PLEASE!!

[equis software]

Excuse me, but I thik this function is ok because I did this...

def run()
   signal.signal(signal.SIGALRM, self.logsignal)
   signal.alarm(10)

def logsignal(self,signum, frame):
self.putCDR()

And work very well, offcourse I need to putCDR() only with SIGHUP not with
the SIGALRM.





On Wed, Apr 9, 2008 at 10:37 AM, Tilghman Lesher <
[EMAIL PROTECTED]> wrote:

> On Wednesday 09 April 2008 07:41:17 equis software wrote:
> > Hi, I need to catch then end of an eagi script (python) when caller
> hungup
> > because I want to generate my own CDR.
> > I try this
> >
> > def run()
> > signal.signal(signal.SIGHUP, self.logsignal)
> >
> > def logsignal(self,signum, frame):
> > self.putCDR()
> >
> > but didn't work. Then try with several signals like:
> > signal.signal(signal.SIGTERM, self.logsignal)
> > signal.signal(signal.SIGTSTP, self.logsignal)
> > signal.signal(signal.SIGPIPE, self.logsignal)
>
> If you read the Python documentation, you'll see that your signal handler
> must
> be a routine that takes 2 arguments, not the 3 that you're providing here.
>
> --
> Tilghman
>
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Re: [asterisk-users] Half-duplex call on TDM2400p with VPMADT032

[Faraz R. Khan]

The newer zaptel (1.4.10) says it includes firmware 1.16 from the
CHANGELOG:


firmware/Makefile, kernel/wctdm24xxp/base.c,
  kernel/wctdm24xxp/GpakApi.c, kernel/wctdm24xxp/GpakApi.h: Update
  wctdm24xxp's VPMADT032 firmware to version 1.16


However there seems to be no way to get this firmware and it does not seem to 
be included. It checks my firmware and says 1.07 is okay. 


The URL provided does not contain firmware for the VPMADT032

I* have logged a query with digum. Is there a URL to get this firmware from?

On Tue, 2008-04-08 at 11:12 -0500, Ruben Zamora wrote:
> Lex
> 
> Thanks, I all ready download the last svn branches from zaptel And i 
> am going to test these afternoon.
> 
> My phone number es 81-83481611.
> 
> Thanks
> 
> Ruben
> 
> Lex Lethol escribió:
> > Ruben,
> >
> > I am also in Monterrey and have used digium hardware on R2 and PRI.
> > MFC/R2 is not supported by digium but the zaptel driver requirement is
> > the same.. what changes is using libpri vs unicall.
> >
> > Just go ahead and ask them for the firmware update or as Tzafir says
> > use a newer zaptel that should include the updated firmware.
> >
> > If in trouble add me to gtalk I'll try to help out any way possible,
> >
> > Lex
> >
> > On Tue, Apr 8, 2008 at 1:52 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> >   
> >> On Mon, Apr 07, 2008 at 09:22:30PM -0500, Ruben Zamora wrote:
> >>  > Lex
> >>  >
> >>  > Thanks a lot.   These morning i call Digium Support.   One issue that i
> >>  > miss in my before e-mail is that i have
> >>  > my Asterisk installed in Monterrey,Mexico and my TELCO provider gave my
> >>  > MFC/R2.
> >>  > Asterisk 1.4.18,Zaptel 1.4.9.2 and Unicall.
> >>  >
> >>  > They told me they can help me because they dont have UNICALL support.
> >>  >
> >>  > So... I need to investigate more or wait for a new zaptel or anything 
> >> else.
> >>
> >>  Generally you can always use a newer zaptel.
> >>
> >>  --
> >>Tzafrir Cohen
> >>  icq#16849755  jabber:[EMAIL PROTECTED]
> >>  +972-50-7952406   mailto:[EMAIL PROTECTED]
> >>  http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
> >>
> >>
> >>
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> >>  asterisk-users mailing list
> >>  To UNSUBSCRIBE or update options visit:
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >> 
> >
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> >
> >   
> 
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> 
-- 
Faraz R Khan
Chief Architect
Emergen Consulting Pvt Ltd
+92.21.111.111.320 x200
www.emergen.biz


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[asterisk-users] *21* # diverting

[gilbert saunders]

hi me again im new at asterisk and really need some good tutoring on asterisk 
and call forwarding i dont understand it at all pls help i have attached my 
extensions.conf file if someone would be so kind to look at it and tell me what 
code i must enter to make *21* diverting and #21# undiverting possible
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[asterisk-users] Queues +Exiting

[Rob Schall]

I'm having a problem getting my queue to function as it should.

After 20 seconds or so, it should prompt the user with a message "thanks
for holding. press # to leave a message or stay on the line to
continue holding". I set up the "context" in the queues.conf file, so if
a user presses a digit, they should be able to leave. But I get a SIP
BUSY message.

Here are my confs:

queues.conf
[custserv]
music=default
strategy=ringall
;timeout=10
retry=20
wrapuptime=0
maxlen=0
context = queue-out
periodic-announce=cont_holding
periodic-announce-frequency=15
;announce-frequency=15
;announce-holdtime=yes
member => SIP/2001
member => SIP/2002
member => SIP/1004

extensions.conf
[queue-out]
exten => s,1,Voicemail(u${vmbox})
exten => s,2,Hangup

[macro-coqueue-vm]
; Call One Queue - Goto to voicemail after 30 secs
; ${ARG1} - Queue Name
; ${ARG2} - Voicemail
exten => s,1,Set(CALLERID(name)=${ARG1}-${CALLERID(name)}) ;Set Caller ID
exten => s,n,Set(vmbox=${ARG2})
exten => s,n,Queue(custserv|tT|||120)
exten => s,n,Voicemail(u${ARG2})
exten => s,n,Hangup()

Any thoughts on how to get the queue to recognize that I pressed a digit
and send it to the queue-out context where they can leave a message?

Thanks.

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Re: [asterisk-users] Advice on best operator phone (with attendant console)

[Faraz R. Khan]

We are very close to taking out that policy ourselves. Problem is that
the other vendors (linksys/Polycom/Aastra/SNom) are awful with
supporting/marketing their phones outside US/Europe. For example a
Polycom 301 imported into Pakistan/India would cost me roughly $280
(thats the reseller price). Aastra is similar. 

On the contrary Grandstream would cost me such that I can STILL sell it
at the MSRP ($109).

Basically Selling Polycom/Aastra an Asterisk based solution becomes as
expensive as a Cisco/Nortel. At which point its really not worth it for
the client.

On Wed, 2008-04-09 at 20:24 +1000, Rob Hillis wrote:
> I've now upgraded one of my phones to the 1.1.6.16 firmware in the
> hope that the firmware is as good as people seem to think it is.
> Certainly the looong list of fixes looks promising.
> 
> Regardless, I would still never recommend Grandstream phones.  The
> place I work for has actually taken on a policy of not supporting
> Grandstream phones.  The damage to their reputation has well and truly
> been done and will take a long time to repair - if it can be.
> 
> 
> Faraz R. Khan wrote: 
> > To be fair to the engineers at grandstream - an update to the latest
> > 1.1.6.16 firmware seems to make the phones very very stable. I now have
> > a couple GXP2000s running at high call volume for the past 3 days
> > without any issue (usually it would happen within an hour).
> > 
> > Problem is that Polycom/Aastra seem to not be interested in sales
> > outside US and Europe. Their channel management seems quite weak and
> > their sales people simply seem uninterested in third world country
> > sales. After being on the phone with Polycom US for 30 minutes I still
> > could not get hold of a person responsible for APAC/EMEA sales (my call
> > got transferred 6 times). Maybe its just my bad luck but it has happened
> > twice now :)
> > 
> > Grandstream on the other hand is extremely helpful in negotiating good
> > deals, giving heavy discount, arranging for shipping from nearby
> > warehouses etc..
> > 
> > I think the problem may be that they release their firmwares WAY too
> > quickly, earning them a bad reputation.
> > 
> > On Sun, 2008-04-06 at 09:41 +0500, faraz wrote:
> >   
> > > Guys thanks a lot. I should be going with a Polycom 650 for all such
> > > jobs.
> > > 
> > > If grandstream receives such bad reviews- how are they selling anything?
> > > Phones hanging or voice cut-outs are simply unacceptable!!
> > > 
> > > On Sun, 2008-04-06 at 14:12 +1000, Rob Hillis wrote:
> > > 
> > > > I'd find that very strange considering that the 57i itself has
> > > > facility for at least 20 BLF buttons and each attendant console has
> > > > facility for another 60!
> > > > 
> > > > 
> > > > Matt Watson wrote: 
> > > >   
> > > > > We are using 57i + 560M combination as well... though we are not 
> > > > > using the 57i ct... but the idea of giving them a cordless is a good 
> > > > > idea.
> > > > > 
> > > > > The only downside to the Aastra 57i + 560M is that it can only 
> > > > > subscribe to 50 extensions for BLF... i haven;t run into this cap yet 
> > > > > myself, but I have heard others talk about it... I think it was a cap 
> > > > > introduced in one of the newer versions of firmware... not sure 
> > > > > though, and not sure why.
> > > > > 
> > > > > I'm running the latest 2.2 firmware on it... the addition of 
> > > > > one-touch transfers in the last firmware was very nice so operator 
> > > > > can transfer very fast, instead of having to do xfer->BLF key->xfer 
> > > > > (for attended transfer), now they can just hit the BLF key for a 
> > > > > blind transfer.
> > > > > 
> > > > > 
> > > > > --
> > > > > Matt
> > > > > 
> > > > > 
> > > > > From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Sigma Networks 
> > > > > [EMAIL PROTECTED]
> > > > > Sent: Saturday, April 05, 2008 12:52 PM
> > > > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > > > Subject: Re: [asterisk-users] Advice on best operator phone (with 
> > > > > attendant console)
> > > > > 
> > > > > We have been marketing ipPBX systems based on asterisk for 3+ years.
> > > > > For the last year we've been placing Aastra 57iCT with 560M sidecars.
> > > > > Our attendants like the idea of a cordless handset so the attendant 
> > > > > can
> > > > > go to the copy room, etc.  The LCD based sidecar means you can keep it
> > > > > up to date without marking up paper strips.   We deploy Thirdlane PBX
> > > > > Manager which allows us to setup the BLF (busy lamp field) via a web
> > > > > interface.
> > > > > 
> > > > > Aastra 57iCT:
> > > > > http://neobits.com/aastra_-_a1758-0131-10-05_-_57i_ct_p11471.html
> > > > > Aastra 560m: 
> > > > > http://neobits.com/aastra_-_a1760--10-55_-_560m_p11472.html
> > > > > Thirdlane PBX Manager: http://www.thirdlane.com/products/pbxmanager
> > > > > 
> > > > > Feel free to contact me off list if I can be of any assistance.
> > > > 

Re: [asterisk-users] Catch end of Eagi script when caller hung up...HELP ME PLEASE!!

[Tilghman Lesher]

On Wednesday 09 April 2008 07:41:17 equis software wrote:
> Hi, I need to catch then end of an eagi script (python) when caller hungup
> because I want to generate my own CDR.
> I try this
>
> def run()
> signal.signal(signal.SIGHUP, self.logsignal)
>
> def logsignal(self,signum, frame):
> self.putCDR()
>
> but didn't work. Then try with several signals like:
> signal.signal(signal.SIGTERM, self.logsignal)
> signal.signal(signal.SIGTSTP, self.logsignal)
> signal.signal(signal.SIGPIPE, self.logsignal)

If you read the Python documentation, you'll see that your signal handler must
be a routine that takes 2 arguments, not the 3 that you're providing here.

-- 
Tilghman

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Re: [asterisk-users] PINCH: Reply-to-munging

[Benny Amorsen]

Tilghman Lesher <[EMAIL PROTECTED]> writes:

> Which is why 'rm' ALWAYS prompts for confirmation, right?  Oh, wait, it
> doesn't.  ;-)

Confirmation is useless, it just teaches everyone to blindly type y.

Analogies suck, of course.


/Benny



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Re: [asterisk-users] RTCP not being sent when on hold

[Drew Gibson]

Adrian A wrote:
> Hello,
>
> When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1 
>  and I place the call on hold, the call is dropped 
> after 30 seconds.
> It looks like there is no RTCP/RTP sent to the client from Asterisk 
> while on hold (music on hold playing to caller) thus client 
> disconnects the call. During this time, I get the following messages 
> in the CLI:
>
> NOTICE[24194] rtp.c: Unknown RTP codec 126 received from '0.0.0.0 
> '
>
> In sip.conf I have rtpkeepalive=15 but that does not seem to help.
>
> Does anyone know what I can do to fix this, other than increase the 
> timeout on Bria?
>
> Thanks,
> Adrian


Is it not up to the phone to send the keep-alive packets?

Sounds like Asterisk does not understand the keep-alive packets coming 
from the phone. Try setting "rtptimeout=300" in sip.conf to test this. 
It should now hangup after 5 minutes.

regards,

Drew

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


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[asterisk-users] Catch end of Eagi script when caller hung up...HELP ME PLEASE!!

[equis software]

Hi, I need to catch then end of an eagi script (python) when caller hungup
because I want to generate my own CDR.
I try this

def run()
signal.signal(signal.SIGHUP, self.logsignal)

def logsignal(self,signum, frame):
self.putCDR()

but didn't work. Then try with several signals like:
signal.signal(signal.SIGTERM, self.logsignal)
signal.signal(signal.SIGTSTP, self.logsignal)
signal.signal(signal.SIGPIPE, self.logsignal)
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Re: [asterisk-users] queue logging

[equis software]

You can´t see the ring event in the queue.log.
You can get this event using the Manager connection.


On Wed, Apr 9, 2008 at 5:01 AM, Arjan Kroon | Mobillion <
[EMAIL PROTECTED]> wrote:

>  Hi,
>
>
>
> I' using with asterisk a queue with tree members and round robin.
>
> When a caller enters this queue and it is connecting to one of the
> members, is there a possibility to see which member the caller is connected
> to?
>
>
>
> And is there a way to see in de application to see if the connection from
> the caller to one of the members was successful of not successful?
>
>
>
> I know you can see it in de queue. log.
>
> But I want to know if I can see it also in the hangup (h) in de
> application?
>
>
>
> Kind Regards
>
>
>
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[asterisk-users] MixMonitor fdiles

[robert boardman]

Hi,

I have a load of files recorded with MixMonitor that are out of sync ie 
one leg of the call is 2-3 seconds behind the other,

is this a bug in Asterisk 1.4.18, or am I possibly doing something wrong


Is it possible to edit the file and re sync the a/b leg?

Thanks for your help

Robb

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Re: [asterisk-users] Message waiting indication(MWI) for voicemail - to H323 endpoints

[Anisha Kumar]

Hi all, 

 I feel that mailing to the point and focussing on the requirement was
more needed than anything else. That's how I have been involved in other
lists too so far. Anyways thank you for pointing that out. I shall try
to change my tone. But I am not sure what created the impression that I
did not do any research. I have been looking into all possible forums
and sites, available tutorials for the same. I posted my query only
after my detailed analysis and after failing to find any suitable
solution for h323, though I  got the solutions for SIP.

 With my analysis so far, I find that Asterisk is used normaly with
ooh323c for H323 support. However it does not provide support to
H.450.7, Message Waiting indication supplementary service. Hence, one
way is to use H323Plus with Asterisk. But then Asterisk code may also
need implementations to support that.  I got this solution from a very
experienced member of H323Plus/OpenH323.

As I am completely new to Asterisk, I do not know , how much of
implementation will be needed in Asterisk. Further more, as I have very
little time to finalize my report, I am more interested in knowing the
existing solutions/methods that is being adopted.

So kindly provide your valuable feedbacks and suggestions about the
existing solutions for MWI for voicemail , that any of you might have
used with the available open source solutions,for an H323 endpoint. Any
input or suggestions will be highly useful. 

Hoping to get a positive response.

Thanks in advance,
Anisha


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Balashov
Sent: Wednesday, April 09, 2008 11:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Message waiting indication(MWI) for
voicemail - to H323 endpoints

Anisha Kumar wrote:

>  How does the Asterisk provide Voicemail Message waiting indication to

> an h323 endpoint configured with Asterisk.

I could be wrong, but a source code search of the H.323 channel drivers
turns up nothing [relevant] for the keywords "mwi," "voicemail," 
"light," "indicator," or anything else that seems pertinent.

>  Please provide the required Setup / comfiguration details or redirect

> to appropriate to resource.

You are decreasing your chances of getting a favourable response with an
imperative tone like that, which also suggests that you are
categorically unwilling to do your own research.

--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] [VOIP-Users-Conference] Subject or Guest for Friday?

[Dean Collins]

We can talk about the third party application idea Cory and I discussed
last week - sorry still haven't posted to the list yet as I've had a
client in town for the last few day but should be able to do this today.
If Fop, Snap Dialer, Mexuar and or any other third party developer who
has written commercial asterisk applications (preferably small low value
eg $20 etc) is interested in jumping on a call with me this Friday I can
put together a series of questions to run the call with.
Maybe we can get Kevin from Digium to explain on the call how the g729
license registration process works and turn the call into a working
discussion.

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:VOIP-Users-
> [EMAIL PROTECTED] On Behalf Of randulo
> Sent: Wednesday, 9 April 2008 7:27 AM
> To: VOIP Users Conference
> Subject: [VOIP-Users-Conference] Subject or Guest for Friday?
> 
> 
> Hi,
> 
> Does anyone have a guest or subject up their sleeve for this week? I
> have neither and I'm getting ready to move house so I don't have a lot
> of time to pursue. Any chance of getting a phone mfr or service
> provider on? Anyone out there want to take a crack at this? Present
> your product or service or open source contributions.
> 
> Incidentally, TringMe apparently has an API available and are making
> it very easy to build AIR and Flash (or Flex) apps. This sounds very
> interesting for creating branded clients of various kinds. If you're a
> programmer in that world, I'd check it out. AIR looks to me like it
> has legs because of its cross-platform nature. I don't know how it
> works with linux or FreeBSD if at all. Does anyone here know?
> 
> If no ideas are forthcoming, we can talk about my personal (VoIP)
> issues: we're moving and will have two offices, as now. The difference
> is, we can not leave the power on at the old office, so we'll be using
> a combination of hosted pbx and asterisk, probably with the AA50 at
> our new home.
> 
> Thanks in advance for any suggestions on any of this :)
> 
> /r
> 
> --~--~-~--~~~---~--~~
> Your participation in the conference is always appreciated! Please try
to be there live
> when it happens.
> 
> You received this message because you are subscribed to the Google
Groups "Asterisk
> Users Conference" group.
> To post to this group, send email to
[EMAIL PROTECTED]
> To unsubscribe from this group, send email to VOIP-Users-Conference-
> [EMAIL PROTECTED]
> For more options, visit this group at
http://groups.google.com/group/VOIP-Users-
> Conference?hl=en
> -~--~~~~--~~--~--~---


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Re: [asterisk-users] For Your Information - Our Experience with ATCom Phones...

[Steve Totaro]

Ship them back and ask for a full refund, if they don't, charge it
back on your credit card.

Thanks,
Steve Totaro

On Wed, Apr 9, 2008 at 4:07 AM, Kashif Naeem <[EMAIL PROTECTED]> wrote:
> Hello All
>
> We purchased 25 new AtCom AT- 530 phones. Four of them did not work for even
> once and some of them lost configuration after some days. I talked to AT Com
> people over chat for support. They have just 1-2 people for support who are
> also busy in some other activities due to which they are unable to
> communicate in proper way. Often support guy left conversation suddenly and
> is unavailable for days.
>
> Regards,
>
>
> --
> Kashif Naeem
>
> MSN: [EMAIL PROTECTED]
> Gmail: [EMAIL PROTECTED]
>
>
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Re: [asterisk-users] Wait for dialtone feature on FXO device

[Steve Davies]

On 08/04/2008, Steve Davies <[EMAIL PROTECTED]> wrote:
>
> http://bugs.digium.com/view.php?id=12382
>
>  Patch has been attached. Currently only for asterisk 1.2.25, but if
>  no-one else provides a 1.4.x patch soon, I will probably need to do
>  that for myself anyway.
>

As a courtesy I have uploaded 1.4.19 and trunk versions of the patch.
They should apply cleanly, but I have not even been able to compile
test them as my 1.4 build environment is out of action right now.

Regards,
Steve

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Re: [asterisk-users] Message waiting indication(MWI) for voicemail - to H323 endpoints

[Doug Lytle]

Rob Hillis wrote:
>>   
>
> Posting the same question to the list twice doesn't help either.

Nor does cross posting it to the Dev list twice.

Doug


-- 
 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."


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[asterisk-users] m switch in dialplan

[Ian]

Hi all

I have a minor inconvenience here.

I want to use the "m" switch in the "dial" command on our outgoing lines 
to play music to the caller whilst asterisk and our telecoms provider 
connects the call. It works, as long as the called person is available. 
When the called  phone (my mobile in this case) is off and it redirects 
to voicemail, asterisk does not detect that the call is answered and 
continues to play the music.

The reason I want to use the switch is that sometimes there is a silence 
period that has been known to last up to 1 minute, before the called 
party's phone begins to ring, and I want to fill that gap with something 
else, I tried the "r" switch as well but for some reason the Grandstream 
I tested on did not like the "r" switch, my x-lite did however indicate 
ringing.

If any of you can direct me to the right site I would be realy greatfull.

If you need any more info, I will be only to happy to provide it, the 
piece of my dialplan is below.

Thanks in advance

Regards
Ian

>
> exten => 9876,1,Progress()
> exten => 9876,n,dial(ZAP/1/0720311294,,m(default))
> exten => 9876,n,Hangup(
>


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Re: [asterisk-users] Message waiting indication(MWI) for voicemail - to H323 endpoints

[Rob Hillis]

Alex Balashov wrote:

Anisha Kumar wrote:

  

 Please provide the required Setup / comfiguration details or redirect
to appropriate to resource.



You are decreasing your chances of getting a favourable response with an 
imperative tone like that, which also suggests that you are 
categorically unwilling to do your own research.
  


Posting the same question to the list twice doesn't help either.

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Re: [asterisk-users] Advice on best operator phone (with attendant console)

[Rob Hillis]

I've now upgraded one of my phones to the 1.1.6.16 firmware in the hope 
that the firmware is as good as people seem to think it is.  Certainly 
the /looong/ list of fixes looks promising.


Regardless, I would still /never/ recommend Grandstream phones.  The 
place I work for has actually taken on a policy of not supporting 
Grandstream phones.  The damage to their reputation has well and truly 
been done and will take a long time to repair - if it can be.



Faraz R. Khan wrote:

To be fair to the engineers at grandstream - an update to the latest
1.1.6.16 firmware seems to make the phones very very stable. I now have
a couple GXP2000s running at high call volume for the past 3 days
without any issue (usually it would happen within an hour).

Problem is that Polycom/Aastra seem to not be interested in sales
outside US and Europe. Their channel management seems quite weak and
their sales people simply seem uninterested in third world country
sales. After being on the phone with Polycom US for 30 minutes I still
could not get hold of a person responsible for APAC/EMEA sales (my call
got transferred 6 times). Maybe its just my bad luck but it has happened
twice now :)

Grandstream on the other hand is extremely helpful in negotiating good
deals, giving heavy discount, arranging for shipping from nearby
warehouses etc..

I think the problem may be that they release their firmwares WAY too
quickly, earning them a bad reputation.

On Sun, 2008-04-06 at 09:41 +0500, faraz wrote:
  

Guys thanks a lot. I should be going with a Polycom 650 for all such
jobs.

If grandstream receives such bad reviews- how are they selling anything?
Phones hanging or voice cut-outs are simply unacceptable!!

On Sun, 2008-04-06 at 14:12 +1000, Rob Hillis wrote:


I'd find that very strange considering that the 57i itself has
facility for at least 20 BLF buttons and each attendant console has
facility for another 60!


Matt Watson wrote: 
  

We are using 57i + 560M combination as well... though we are not using the 57i 
ct... but the idea of giving them a cordless is a good idea.

The only downside to the Aastra 57i + 560M is that it can only subscribe to 50 
extensions for BLF... i haven;t run into this cap yet myself, but I have heard 
others talk about it... I think it was a cap introduced in one of the newer 
versions of firmware... not sure though, and not sure why.

I'm running the latest 2.2 firmware on it... the addition of one-touch transfers in 
the last firmware was very nice so operator can transfer very fast, instead of having 
to do xfer->BLF key->xfer (for attended transfer), now they can just hit the 
BLF key for a blind transfer.


--
Matt


From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Sigma Networks [EMAIL 
PROTECTED]
Sent: Saturday, April 05, 2008 12:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Advice on best operator phone (with attendant 
console)

We have been marketing ipPBX systems based on asterisk for 3+ years.
For the last year we've been placing Aastra 57iCT with 560M sidecars.
Our attendants like the idea of a cordless handset so the attendant can
go to the copy room, etc.  The LCD based sidecar means you can keep it
up to date without marking up paper strips.   We deploy Thirdlane PBX
Manager which allows us to setup the BLF (busy lamp field) via a web
interface.

Aastra 57iCT:
http://neobits.com/aastra_-_a1758-0131-10-05_-_57i_ct_p11471.html
Aastra 560m: http://neobits.com/aastra_-_a1760--10-55_-_560m_p11472.html
Thirdlane PBX Manager: http://www.thirdlane.com/products/pbxmanager

Feel free to contact me off list if I can be of any assistance.

Regards,
Jim
ph: 408-701-9929



Faraz R. Khan wrote:
  


One of our clients is using a Grandstream GXP2000 with an attendant
console. We have used the same phone with past clients successfully
however this particular operator processes around 200 calls a hours and
the GXP2000 for sure does not like the quick line shuffling and call
volume. We get the following problems randomly:

1. menu stops working
2. transfer key stops working
3. Line 1 LED gets stuck
4. Voice 'gaps' (blackouts) for 4-5 seconds
5. The phone also completely locks up regularly
6. ping response goes from 8ms to 3000ms (after which the phone locks
up)

Wondering which operator phone would work best. I have the following
choices:

1. Linksys SPA 932/962 with attendant console
2. Polycom 601/650 with attendant console

I cant confirm online whether the BLF functionality will work with
Asterisk 1.2.26. Is somebody using either of these phones in a high
volume environment successfully?

Thank you.



  

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[asterisk-users] Question about custom Asterisk billing engine

[Pete Kay]

Dear all,

I am working on a customized billing engine.  The reason why I am writing up
one instead of using the existing package is because I have my own UI that I
want to use for displaying data and registration.   Also, I would like to
keep a record of the among of times user spend in the different
functionalities when an incoming call occurs.

I am planning to record an audit trail of any incoming/outcoming call
through the use of AGI which simply puts status info and a session unique
id into the DB before and after an outgoing /incoming call is made.  For the
incoming call, I would like to also record any forwarding call in the audit
trail so users can see how many seconds they spend on listening to voice
mail and how many second they spend in forwarding to an external phone.

In order to fulfill this function, is using AGI call to record the status
info frequently inside the dialplan a feasible solution?  Is this the right
approach in developing a billing solution for Asterisk?

Thanks alot for you inputs.

Regards,
Pete
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Re: [asterisk-users] RTCP not being sent when on hold

[Gordon Henderson]

On Tue, 8 Apr 2008, Adrian A wrote:

> Hello,
>
> When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1 and I
> place the call on hold, the call is dropped after 30 seconds.
> It looks like there is no RTCP/RTP sent to the client from Asterisk while on
> hold (music on hold playing to caller) thus client disconnects the call.
> During this time, I get the following messages in the CLI:
>
> NOTICE[24194] rtp.c: Unknown RTP codec 126 received from '0.0.0.0'
>
> In sip.conf I have rtpkeepalive=15 but that does not seem to help.
>
> Does anyone know what I can do to fix this, other than increase the timeout
> on Bria?

Are you also recording the call?

I had to put this:

   [options]
   transmit_silence_during_record = yes

into asterisk.conf to stop hangups after 30 seconds ...

Gordon


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[asterisk-users] For Your Information - Our Experience with ATCom Phones...

[Kashif Naeem]

Hello All

We purchased *25* new AtCom AT- 530 phones. Four of them did not work for
even once and some of them lost configuration after some days. I talked to
AT Com people over chat for support. They have just 1-2 people for support
who are also busy in some other activities due to which they are unable to
communicate in proper way. Often support guy left conversation suddenly and
is unavailable for days.

Regards,


-- 
Kashif Naeem

MSN: [EMAIL PROTECTED]
Gmail: [EMAIL PROTECTED]
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[asterisk-users] queue logging

[Arjan Kroon | Mobillion]

Hi,

 

I' using with asterisk a queue with tree members and round robin.

When a caller enters this queue and it is connecting to one of the
members, is there a possibility to see which member the caller is
connected to?

 

And is there a way to see in de application to see if the connection
from the caller to one of the members was successful of not successful?

 

I know you can see it in de queue. log.

But I want to know if I can see it also in the hangup (h) in de
application?

 

Kind Regards

 

 

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