Re: [asterisk-users] Asterisk and Skype

[Alex Balashov]

This is not currently possible. Work in progress.

--
Sent from mobile device

On Jul 8, 2009, at 1:31 AM, DHAVAL INDRODIYA  
 wrote:

> Hello All,
>
> can anybody tell me how can i integrate asterisk and skype users
>
> so that skype users can dial my asterisk number or dial internal  
> dialplan form skype
>
> regars
> Dhaval
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Re: [asterisk-users] Small site survivability

[Olivier]

2009/7/6 Jonathan Thurman 

> We are currently moving away from a wide-spread Cisco CallManager
> deployment to Asterisk.  For many of our small sites we have the routers
> configured for what Cisco calls SRST so if we have a WAN failure, the router
> acts as a SCCP registrar.  We are converting to SIP, and from what I can
> tell Cisco wants a license for each router to run SRST over SIP...
>
> So my question to the group is: What are you doing for survivability in
> these small (6-30 phone) sites?  I would like to avoid deploying a lot of
> servers if at all possible.  The requirements would be a simple, easy to
> manage device for the phones to register to in case of WAN failure with 1 or
> 2 POTS lines attached (also used for 911 calls from that site).


What happens for IT when WAN fails ?
Are people still able to work or not ?

If they are, then it should be possible to use current routers (if they have
such POTS interfaces) as Media gateways and have a local resource to act as
a backup Asterisk server.

If they are not, having IT and Telephony to share the same backup WAN is
advisable.


>   Thanks for any suggestions!
>
> -Jonathan
>
>
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Re: [asterisk-users] documentation of DAHDI dial options

[Olivier]

2009/7/7 Jared Smith 

> On Tue, 2009-07-07 at 15:42 +0200, Klaus Darilion wrote:
> > I am searching for the description of the available dialstrin options
> > for the DAHDI channel (and also other channel types).
> >
> > I am not looking for outdated voip-info links, but for the authoritative
> > source, e.g. something like "core show application Dial"
> >
> > Does such thing exists?
>
> I don't think that such a thing exists.  The only ones I'm aware of are:
>
> 1) Channel Groups.
>
> DAHDI/g1/5551212 dials 5551212 on the first available channel in group
> one, searching from lowest to highest
>
> DAHDI/G1/5551212 dials 5551212 on the first available channel in group
> one, searching from highest to lowest
>
> DAHDI/r1/5551212 dials 5551212 on the first available channel in group
> one, going in round-robin fashion (and remembering where it last left
> off), searching from lowest to highest
>
> DAHDI/R1/5551212 dials 5551212 on the first available channel in group
> one, searching in round-robin fashion from highest to lowest.
>
> 2) Distinctive ring
>
> DAHDI/4r1 dials channel 4 (presumably an FXS channel), and uses
> distinctive ring style one.  If I recall, there are four different
> distinctive ring styles... so you could replace r1 with r2, r3, or r4.
>
> 3) Answer confirmation
>
> DAHDI/1c/5551212 tells Asterisk to dial 5551212 on DAHDI channel 1, and
> not consider the call answered until the called party presses #.  This
> is useful because of the way analog signaling works.  Without this
> setting, Asterisk considers any outbound analog call on an FXO port
> answered just as soon as it has been dialed.
>
> 4) Digital calls
>
> DAHDI/1d/5551212 tells Asterisk to dial 5551212 on DAHDI channel 1, and
> that it's a digital call.  If I remember correctly, this is used for
> ISDN calls to set the bearer capability.
>
> I've taken a quick look in channels/chan_dahdi.c in TRUNK, and it seems
> to match up with my understanding, as I didn't see any other options
> stand out.  While poking around in there, I found the following comment:
>
>/*
> * data is ---v
> * Dial(DAHDI/pseudo[/extension])
> * Dial(DAHDI/[c|r|d][/extension])
> * Dial(DAHDI/(g|G|r|R)[c|r|d][/extension])
> *
> * g - channel group allocation search forward
> * G - channel group allocation search backward
> * r - channel group allocation round robin search forward
> * R - channel group allocation round robin search backward
> *
> * c - Wait for DTMF digit to confirm answer
> * r - Set distintive ring cadance number
> * d - Force bearer capability for ISDN/SS7 call to digital.
> */
>
> That's probably as definitive an answer as you're going to get.


What is this was commented such as it could be added to a "core show
application Dial" ?

>
>
>
> --
> Jared Smith
> Training Manager
> Digium, Inc.
>
>
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Re: [asterisk-users] Automatic Gain Control

[Olivier]

2009/7/7 Brent Davidson 

> Is there any possibility of DAHDI supporting Automatic gain control on
> TDM ports?  I'm having issues at a couple of offices where calls made to
> local numbers are fine but a when a calls from or goes to a large
> percentage of long-distance or 1-800 numbers the person at the remote
> end cannot hear the person in my office.  Boosting the gains in
> zapata.conf (I'm still using 1.4.21) to 8 solves the problem with
> long-distance lines, but then local calls say the person in my office is
> too loud.
>

>
> I understand that it is going to be difficult to reliably detect a major
> drop in the volume at the far end of the call, but I'm just wondering if
> there is a good solution for this.  We're using Rhino WC4-FXO-ec cards
> and the OSlec echo canceler (since the on-board echo canceler didn't
> seem to help our echo issues)


hello,

I'm afraid I can't be of any help but as a first step, I'm wondering if
there's a way to translate
user experience such as "too loud" into figures such "volume is 8" ?


>
> Thanks,
> Brent
>
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Re: [asterisk-users] Caller ID (name) - where does it come from?

[Olivier]

Hi,

Reading this thread, is this correct to say CallerName is widely used in the
US ?

Here in France, this service is optional but I don't think many companies
are subscribing to it and I'm not aware of any non-Telco CNAM providers.
I would curious to know how the situation is elsewhere.

Regards
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Re: [asterisk-users] Fax for Asterisk - Fax routing based on TSID

[Olivier]

2009/7/7 Doug Lytle 

> Olivier wrote:
> >
> >
> >
> > Please, allow me to ask what is this Transmitting Station ID ?
> >
>
> Google is you friend:
>
>
> http://encyclopedia.thefreedictionary.com/Transmitting+Subscriber+Identification


Thanks !
I still have to improve my googling !

>
>
> 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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Re: [asterisk-users] Caller ID (name) - where does it come from?

[Trevor Peirce]

Barry D. Hassler wrote:
> So how does Teliax (for instance) go about getting their client's 
> information into these directories? Do they establish a relationship 
> with someone like TargusInfo (described above)?
>
> How do other ITSP's provide this service, or do they ignore it as well?

Yes, either they will work with Targus, Versign, etc who do commercial 
CNAM hosting, or they work with their CLEC partner who would presumably 
already have such an agreement in place.

It's my understanding that with an LOA from your CLEC or ILEC (kind of 
the opposite of the LOA you need to port a number), you can have your 
own CNAM records hosting with one of the companies listed above and make 
money.  We're talking a fraction of a cent per call so unless you have 
many DIDs and many more calls, it's not usually worthwhile.

Most smaller ITSPs either don't know how this or don't have the volume 
to make it feasible. 

In Canada, we just include the name in the SS7 signaling on a per-call 
basis and bypass this whole mess :)

Best regards,

-- 
Trevor Peirce
Digital Conceptions Canada

http://www.digitalcon.ca
1-888-606-3030 / 250 483-0386


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Re: [asterisk-users] Caller ID (name) - where does it come from?

[Barry D. Hassler]

So how does Teliax (for instance) go about getting their client's
information into these directories? Do they establish a relationship with
someone like TargusInfo (described above)?

How do other ITSP's provide this service, or do they ignore it as well?



On Tue, Jul 7, 2009 at 9:49 PM, Frank Bulk  wrote:

> How does that work?
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro
> Sent: Tuesday, July 07, 2009 8:18 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Caller ID (name) - where does it come from?
>
> 
>
> I get paid every time I call someone that subscribes to caller ID.
>
> --
> Thanks,
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
>
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-- 
Barry D. Hassler
President, HCST

http://www.hcst.net/
937-427-9000
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[asterisk-users] Asterisk and Skype

[DHAVAL INDRODIYA]

Hello All,

can anybody tell me how can i integrate asterisk and skype users

so that skype users can dial my asterisk number or dial internal dialplan
form skype

regars
Dhaval
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Re: [asterisk-users] One Way Audio from External Sip Soft & Hard Phone

[Steve Totaro]

On Tue, Jul 7, 2009 at 9:55 PM, Paul Edgar wrote:
> I have a problem with one way audio on Sip and I guess it may be a NAT
> issue, in the example below 204 is rung by 208 (xlite external)
>
>
>
> I dial perfectly but when I get to the answering of the Asterisk, I can hear
> audio from the Asterisk but cannot get audio to the Asterisk, ie If I ring
> the voice mail , Asterisk answers and then cannot hear my password…
>
>
>
> I have put the Ports Forward etc…5004-5080 & 1-2
>
>
>
> Any ideas – even what to test next would be good…
>
>
>
>
>
> -- Executing [...@macro-stdexten:13] Dial("SIP/208-00a10004", "SIP/204") in
> new stack
>
>
>
>     -- Called 204
>
>
>
>     -- SIP/204-00a11584 is ringing
>
>
>
>     -- SIP/204-00a11584 answered SIP/208-00a10004
>
>
>
> [Jul  7 16:15:08] WARNING[834]: chan_sip.c:1993 retrans_pkt: Maximum retries
> exceeded on transmission NjQ1NTA3Mzg3YzI5ZTNlYzI3MWU2NzE4YWE3ZTI2MDE. for
> seqno 2 (Critical Response)
>
> [Jul  7 16:15:08] WARNING[834]: chan_sip.c:2017 retrans_pkt: Hanging up call
> NjQ1NTA3Mzg3YzI5ZTNlYzI3MWU2NzE4YWE3ZTI2MDE. - no reply to our critical
> packet.
>
>
>
>   == Spawn extension (macro-stdexten, s, 13) exited non-zero on
> 'SIP/208-00a10004' in macro 'stdexten'
>
>   == Spawn extension (macro-stdexten, s, 13) exited non-zero on
> 'SIP/208-00a10004'
>
>

Where is the NAT or is it on both sides?

Answer that and turn on SIP debugging and post the output and I am
sure someone can help you.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

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[asterisk-users] g.722 + loudness

[Hose]

Hi, 

  


  
We've been running g.722 in asterisk 1.6.09 for awhile now, with a PRI  

  
and numerous sip clients.  Internal sip to sip and sip to pri (and
vice versa) work fine between g.722 and ulaw - the transcoding is   

acceptable.  
   
The only time it fails is when we utilize a meetme conference bridge.
With a Polycom IP 6000 + a call over the PRI, the person calling in over
  
the PRI sounds distorted when they're barely talking at a normal volume.   
Anything over a normal volume results in terrible clipping.  Bringing   

  
the volume down on the Polycom either via software settings or the  

  
actual volume keys doesn't stop the distortion, so that points to a 

  
problem with asterisk (the volume can be very loud, barely audible, but
you can still hear the clipping occuring).  By clipping, I mean the
static that happens when you have a signal that's too loud.

The thing is, when you call directly into the Polycom over the PRI, it's
fine.  This ONLY happens during a conference call with g.722, though
this might be because asterisk is negotiating a ulaw connection when
called direct from the PRI - is there a way to check what codec it's
negotiated during the call?

I have a feeling that the issue is between transcoding of ulaw to g.722
and it's too loud during the transcoding - anyway to adjust the levels?

Thanks!
hose

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[asterisk-users] One Way Audio from External Sip Soft & Hard Phone

[Paul Edgar]

I have a problem with one way audio on Sip and I guess it may be a NAT
issue, in the example below 204 is rung by 208 (xlite external) 

 

I dial perfectly but when I get to the answering of the Asterisk, I can
hear audio from the Asterisk but cannot get audio to the Asterisk, ie If
I ring the voice mail , Asterisk answers and then cannot hear my
password...

 

I have put the Ports Forward etc...5004-5080 & 1-2 

 

Any ideas - even what to test next would be good...

 

 

-- Executing [...@macro-stdexten:13] Dial("SIP/208-00a10004", "SIP/204")
in new stack

 

-- Called 204

 

-- SIP/204-00a11584 is ringing

 

-- SIP/204-00a11584 answered SIP/208-00a10004

 

[Jul  7 16:15:08] WARNING[834]: chan_sip.c:1993 retrans_pkt: Maximum
retries exceeded on transmission
NjQ1NTA3Mzg3YzI5ZTNlYzI3MWU2NzE4YWE3ZTI2MDE. for seqno 2 (Critical
Response)

[Jul  7 16:15:08] WARNING[834]: chan_sip.c:2017 retrans_pkt: Hanging up
call NjQ1NTA3Mzg3YzI5ZTNlYzI3MWU2NzE4YWE3ZTI2MDE. - no reply to our
critical packet.

 

  == Spawn extension (macro-stdexten, s, 13) exited non-zero on
'SIP/208-00a10004' in macro 'stdexten'

  == Spawn extension (macro-stdexten, s, 13) exited non-zero on
'SIP/208-00a10004'

 

 

 

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Re: [asterisk-users] Caller ID (name) - where does it come from?

[Frank Bulk]

How does that work?  

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro
Sent: Tuesday, July 07, 2009 8:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID (name) - where does it come from?



I get paid every time I call someone that subscribes to caller ID.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

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Re: [asterisk-users] Caller ID (name) - where does it come from?

[Frank Bulk]

For the CNAM vendors who pride themselves on completeness/coverage, don't
you think that they have some interest in getting data from the likes of
Teliax?  Maybe they wouldn't pay for it, but ITSPs have to realize that to
retain certain customers that they have to their customers numbers
disseminated.  But I guess if they can charge extra for what used to be
table stakes, so be it.

Frank

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife
Sent: Tuesday, July 07, 2009 8:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID (name) - where does it come from?

>>My primary issue is for calls that are placed FROM my client's PBX, via 
>>VOIP provider (Teliax). The recipients of those calls are the ones that 
>>are not getting the proper CNAM information as the call comes in.

I neglected to go into detail on this point at the end of my last post 
because I thought it was out of scope.  But now that you ask...

While I am not an expert in the specific architecture of CNAM database, I do

know that (to Frank's point) it is not at all a database in the 'MySQL' or 
'Oracle' sense of the word.  It's a database more analagous to the DNS where

data can be located in many places (cached) but there is a single source 
considered authoritative that ultimately propogates out to cache.  This 
authoritative source is the Telco that provides your DID number--after all, 
they the only ones with a billing relationship to validate the name 
information.

So historically, *normally* your Telco is the authoritative source of the 
CNAM data that populates the 'screens' of the people you call, and 
*normally* the Telco of the calling party is ultimately compensated by the 
Telco of the called party for providing the CNAM data, but this model has 
broken down in the world if IP telephohy.  Your ITSP (Teliax) is one of 
"them-thar new-fangled ITSPs" and the big boys have exactly ZERO interest in

compensating them for CNAM dips.  Meanwhile they are excluded from the holy 
brotherhood of 'real' CNAM.

This is why your name is not populated in the CNAM database.  Teliax is not 
one of the CNAM insiders who exchange name data and compensate each other 
for said data.

That's also why it would never make sense to ask your CNAM lookup serive 
provider to make corrections to errant CNAM data.  It just doesn't work that

way.

It used to be that you could work around this problem by using LNP to port 
your number temporarily to an ILEC .  Your TN would get a CNAM record which 
would persist as an orphan for years.   Recently this has changed, and NOW 
when you port your TN away from the losing LEC, they purge your CNAM record.

:-(

Recently there are some good solutions to this problem.  One is to ask your 
ITSP if they can put your number in the LIDB for a fee or alternatively you 
can just buy a "white pages" entry (also from your ITSP) which accomplishes 
the same thing.  I've seen this for $5 per month, and the BONUS you get a 
white pages entry (which you may or may not want).

I hope this helps.
-Karl

http://www.hcst.net/
937-427-9000




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Re: [asterisk-users] Caller ID (name) - where does it come from?

[Steve Totaro]

On Tue, Jul 7, 2009 at 9:06 PM, Frank Bulk wrote:
> Intersting.  Vitelity is charging for something that they might already be
> getting paid for.  Of course, updating a name for a number takes time, and
> so that's probably why they can justify charging the customer something.
> Most times when you sign up you specify how you want the directory listing
> to look, and that's what is sold/delivered to CNAM vendors and aggregators.
>
> I'm not sure what Vitelity means by "a national database" and "this
> database".  As I discussed before, a telephony provider can choose pretty
> well any CNAM vendor they want.  Beyond the ones that were mentioned by
> someone else in an e-mail, there's also VeriSign, Neustar, and Syniverse.
> It's an oversimplification to tell the customer that it's *a* database --
> Vitelity may sell their data to just one CNAM vendor/aggregator, but that
> doesn't mean every CNAM vendor's database has now been updated.
>
> Frank
>
> -Original Message-
> From: John A. Sullivan III [mailto:jsulli...@opensourcedevel.com]
> Sent: Tuesday, July 07, 2009 7:23 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Cc: frnk...@iname.com
> Subject: Re: [asterisk-users] Caller ID (name) - where does it come from?
>
> On Tue, 2009-07-07 at 16:54 -0400, Barry D. Hassler wrote:
>> This is all excellent information. My primary issue is for calls that
>> are placed FROM my client's PBX, via VOIP provider (Teliax). The
>> recipients of those calls are the ones that are not getting the proper
>> CNAM information as the call comes in.
>>
>> We just recently ported the client's POTS lines to VOIP, and with the
>> exception of this issue, all is working well. But, my client is really
>> unhappy that their callerID NAME isn't showing up.
>>
> 
> I was very curious about this myself.  We successfully set the CallerID
> number by creating different contexts for our various offices and using
> a Set(CALLERID(num)=x) call.  But we could not set the name so I
> asked our new carrier (Vitelity - with whom we have been quite pleased
> thus far).  This is their response to us:
>
> We can have the name set for this number, however there is a one time
> passthrough charge of $xx per number for the update. Outbound caller ID
> is updated into a national database called LIDB (line information
> database), it is the final terminating provider that is responsible for
> querying this database and delivering it to their customers.
>>
> --
> John A. Sullivan III
> Open Source Development Corporation
> +1 207-985-7880
> jsulli...@opensourcedevel.com


I get paid every time I call someone that subscribes to caller ID.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

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Re: [asterisk-users] [asterisk-user] AGI control stream file

[Steve Edwards]

Trying to redirect to -user...

On Tue, 7 Jul 2009, Bryant Zimmerman wrote:

> Hey guys I posted this earlier and did not get any responses.

You posted what appear[s|ed] to be a user question to the dev list.

I did reply (on June 3), but I may have mis-understood.

> I am working on some AGI development that requires control of audio file 
> playback. The control stream file is working for the most part, but to 
> meet one of the project requirements I need to be able to let the user 
> enter stream playback in either, rewind, fastforward or pause states 
> with a single key press. To this point I have not figured out a way to 
> enter into the control stream file command in anything other than a play 
> state. This requires that the user must hit the, rw, ffwd or pause key..

You want to add code to "control stream file" so that you can specify an 
option so that playback is already rewinding, fastforwarding or paused 
when the application is entered?

I can see how paused would work, but how do you see rewinding or 
fastforwarding working? These are not "states" where the user hears the 
audio at double speed (for fastforwarding). I think these would better be 
described as "skip-forward skipms (the third parameter to "control stream 
file") ms" or "skip-backward skipms ms" -- kind of like a properly 
programmed Tivo remote.

Or, are you thinking if your AGI specified "skip-forward," "control stream 
file" would skip forward skipms ms before starting playback at normal 
speed?

Can you explain the "experience" from the user's perspective?

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Caller ID (name) - where does it come from?

[Karl Fife]

>>My primary issue is for calls that are placed FROM my client's PBX, via 
>>VOIP provider (Teliax). The recipients of those calls are the ones that 
>>are not getting the proper CNAM information as the call comes in.

I neglected to go into detail on this point at the end of my last post 
because I thought it was out of scope.  But now that you ask...

While I am not an expert in the specific architecture of CNAM database, I do 
know that (to Frank's point) it is not at all a database in the 'MySQL' or 
'Oracle' sense of the word.  It's a database more analagous to the DNS where 
data can be located in many places (cached) but there is a single source 
considered authoritative that ultimately propogates out to cache.  This 
authoritative source is the Telco that provides your DID number--after all, 
they the only ones with a billing relationship to validate the name 
information.

So historically, *normally* your Telco is the authoritative source of the 
CNAM data that populates the 'screens' of the people you call, and 
*normally* the Telco of the calling party is ultimately compensated by the 
Telco of the called party for providing the CNAM data, but this model has 
broken down in the world if IP telephohy.  Your ITSP (Teliax) is one of 
"them-thar new-fangled ITSPs" and the big boys have exactly ZERO interest in 
compensating them for CNAM dips.  Meanwhile they are excluded from the holy 
brotherhood of 'real' CNAM.

This is why your name is not populated in the CNAM database.  Teliax is not 
one of the CNAM insiders who exchange name data and compensate each other 
for said data.

That's also why it would never make sense to ask your CNAM lookup serive 
provider to make corrections to errant CNAM data.  It just doesn't work that 
way.

It used to be that you could work around this problem by using LNP to port 
your number temporarily to an ILEC .  Your TN would get a CNAM record which 
would persist as an orphan for years.   Recently this has changed, and NOW 
when you port your TN away from the losing LEC, they purge your CNAM record. 
:-(

Recently there are some good solutions to this problem.  One is to ask your 
ITSP if they can put your number in the LIDB for a fee or alternatively you 
can just buy a "white pages" entry (also from your ITSP) which accomplishes 
the same thing.  I've seen this for $5 per month, and the BONUS you get a 
white pages entry (which you may or may not want).

I hope this helps.
-Karl

http://www.hcst.net/
937-427-9000




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Re: [asterisk-users] Caller ID (name) - where does it come from?

[Frank Bulk]

Intersting.  Vitelity is charging for something that they might already be
getting paid for.  Of course, updating a name for a number takes time, and
so that's probably why they can justify charging the customer something.
Most times when you sign up you specify how you want the directory listing
to look, and that's what is sold/delivered to CNAM vendors and aggregators.

I'm not sure what Vitelity means by "a national database" and "this
database".  As I discussed before, a telephony provider can choose pretty
well any CNAM vendor they want.  Beyond the ones that were mentioned by
someone else in an e-mail, there's also VeriSign, Neustar, and Syniverse.
It's an oversimplification to tell the customer that it's *a* database --
Vitelity may sell their data to just one CNAM vendor/aggregator, but that
doesn't mean every CNAM vendor's database has now been updated.

Frank

-Original Message-
From: John A. Sullivan III [mailto:jsulli...@opensourcedevel.com] 
Sent: Tuesday, July 07, 2009 7:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: frnk...@iname.com
Subject: Re: [asterisk-users] Caller ID (name) - where does it come from?

On Tue, 2009-07-07 at 16:54 -0400, Barry D. Hassler wrote:
> This is all excellent information. My primary issue is for calls that
> are placed FROM my client's PBX, via VOIP provider (Teliax). The
> recipients of those calls are the ones that are not getting the proper
> CNAM information as the call comes in. 
> 
> We just recently ported the client's POTS lines to VOIP, and with the
> exception of this issue, all is working well. But, my client is really
> unhappy that their callerID NAME isn't showing up.
> 

I was very curious about this myself.  We successfully set the CallerID
number by creating different contexts for our various offices and using
a Set(CALLERID(num)=x) call.  But we could not set the name so I
asked our new carrier (Vitelity - with whom we have been quite pleased
thus far).  This is their response to us:

We can have the name set for this number, however there is a one time
passthrough charge of $xx per number for the update. Outbound caller ID
is updated into a national database called LIDB (line information
database), it is the final terminating provider that is responsible for
querying this database and delivering it to their customers. 
> 
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society



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Re: [asterisk-users] Caller ID (name) - where does it come from?

[Frank Bulk]

If the calling number shows up correctly for the called party (an obvious
first step), the called party will need to get in contact with their
telephony provider/CNAM vendor to get the calling name fixed.  It’s possible
that because your client (the calling party) ported their number, the called
party’s CNAM source reflects no information because the line was
disconnected.  The called party’s CNAM source is obviously not getting
directory listing directly or indirectly from Teliax.

 

It may be helpful to speak to Teliax and find out where they sell/provide
their directory listings.  Somehow the called party’s CNAM source needs to
get that information from Teliax, either directly, or more likely, via one
or more intermediate parties that aggregates the data.   TARGUSinfo
(http://targusinfo.com/solutions/identification/caller_name/default.aspx),
for example, collects from over 90 sources
(http://targusinfo.com/solutions/identification/caller_name/faq/).

 

I’ve heard that Vonage Canada does not sell/provide their directory
listings, so you’ll never obtain a name-like calling number unless your CNAM
provider collects the data from other sources, and they do (e.g. department
store credit cards applications).   

 

Frank

 

From: Barry D. Hassler [mailto:barry.hass...@gmail.com] 
Sent: Tuesday, July 07, 2009 3:54 PM
To: frnk...@iname.com; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Caller ID (name) - where does it come from?

 

This is all excellent information. My primary issue is for calls that are
placed FROM my client's PBX, via VOIP provider (Teliax). The recipients of
those calls are the ones that are not getting the proper CNAM information as
the call comes in. 

We just recently ported the client's POTS lines to VOIP, and with the
exception of this issue, all is working well. But, my client is really
unhappy that their callerID NAME isn't showing up.

On Tue, Jul 7, 2009 at 3:42 PM, Frank Bulk  wrote:

There's a bit of oversimplification going on here -- it's not "a ...
database".  Different CNAM providers have different databases which are
populated from many sources.  Most of the data probably matches, but not all
of it.

If the Calling Name is incorrect, the person who received the call will have
to check with their telephony provider (or, if they do their own CNAM
lookups, with their CNAM provider) to get the name for the calling party
fixed up (this presumes that the calling party has already verified with
their own telephony provider that their name is correctly listed).  But
that's not all of it, either, because the next time the CNAM provider
refreshes their records, the local fix could be overridden (I'm not sure if
any CNAM providers have the capability to ignore old/bad data for a record,
but perhaps so).  Ideally the CNAM provider shares with the calling party
which database the CNAM provider is using for the calling party, so that the
calling party can try to get it fixed directly with the database provider
(if that's even possible).

In short, it's a mess.

But because accuracy rates are one of the elements that CNAM providers
compete on, these usually do get cleaned up.

Frank


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife
Sent: Tuesday, July 07, 2009 1:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] Caller ID (name) - where does it come from?

The Caller ID name, "CNAM" is a separate database owned and maintained
"cooperatively" by the bell operating companies.

Your ITSP is not doing these CNAM lookups for you because they would have to

pay the BOC's for the 'dips' into the CNAM database.  CNAM is a little cash
cow that the BOC's are quick to protect.  As such CNAM dips may not be
cached or re-sold as a term service that you must agree to with your CNAM
provider.

As far as solving your CNAM problem, you would need to either choose an ITSP

that will provide you with CNAM data on a per-call basis, OR you need to do
CNAM dips yourself as I (and many others) do.  Beware that some ITSP's
provide "best-effort" name data culled from various sources.  It's not
always terrible but it's not 'coke' it's more like 'dollar store' cola. :-)

As a call comes in to your dial plan you can populate the CALLERID(name)
channel variable using the CURL function in your dialplan as so:
exten =>
s,n,Set(CALLERID(name)=${CURL(http://cnam1.edicentral.net/getcnam?q=C
 &f=S&dn
=${CALLERID(num)})})

AND let's not forget the completely separate issue with getting your
ITSP-provisioned number ENTERED INTO the CNAM database in the first place,
so people see "Karl Fife" rather than the "city, state" or worse, some
string of arcane LATA information.  There's a solution to this problem too
but I digress...

I've posted my personal notes below from about 18 months ago when I was

Re: [asterisk-users] Caller ID (name) - where does it come from?

[John A. Sullivan III]

On Tue, 2009-07-07 at 16:54 -0400, Barry D. Hassler wrote:
> This is all excellent information. My primary issue is for calls that
> are placed FROM my client's PBX, via VOIP provider (Teliax). The
> recipients of those calls are the ones that are not getting the proper
> CNAM information as the call comes in. 
> 
> We just recently ported the client's POTS lines to VOIP, and with the
> exception of this issue, all is working well. But, my client is really
> unhappy that their callerID NAME isn't showing up.
> 

I was very curious about this myself.  We successfully set the CallerID
number by creating different contexts for our various offices and using
a Set(CALLERID(num)=x) call.  But we could not set the name so I
asked our new carrier (Vitelity - with whom we have been quite pleased
thus far).  This is their response to us:

We can have the name set for this number, however there is a one time
passthrough charge of $xx per number for the update. Outbound caller ID
is updated into a national database called LIDB (line information
database), it is the final terminating provider that is responsible for
querying this database and delivering it to their customers. 
> 
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] asterisk addon mysql - is mysql connection persistent

[Miguel Molina]

Shahid Tel escribió:

Hi Guys,

As it looks like from CLI command " show cdr mysql " , can somebody 
confirms that cdr-mysql creates persistent connection with in asterisk?


" show cdr mysql " shows " connected to u...@dbhost 
 from 18 hours ..."
 
Yes, the MySQL CDR addon creates a persistent connection to the 
database. If the database server goes down, the addon tries to reconnect 
so if it succeeds no records are lost or only a few, I'm not sure. The 
addon won't die, neither asterisk. For example:


"cdr mysql status" CLI command shows me this:

Connected to @, port 3306 using table  for 19 days, 19 
hours, 32 minutes, 2 seconds.
 Wrote 4256045 records since last restart and 294847 records since last 
reconnect.


Restart is the last asterisk restart. Last reconnect, is the last time 
the connection went down and reconnected because the server went down or 
you killed the connection from the MySQL monitor. So if for any reason 
you need to do a quick restart of MySQL, you won't lose CDR records if 
no calls are hungup during the MySQL restart cycle.


Cheers,

--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center

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Re: [asterisk-users] Call parking with ISDN

[Julien Claassen]

Hello Wilton!
   Thanks for your looking after my problems.
   No I meant the usual asterisk call park. Yes, it should be independet of the 
trunk. But I wondered how to activate it from the asterisk CLI? Should I send 
some special DTMFs (dialing digits) and be done with it? Or should I use some 
special parking command or use a complete dial command? How to do it? and how 
to get my caller back, when I want to continue the call?
   Thanks for caring about this!
   Friendly regards
  Julien


Music was my first love and it will be my last (John Miles)

 FIND MY WEB-PROJECT AT: 
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
=== AND MY PERSONAL PAGES AT: ===
http://www.juliencoder.de

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[asterisk-users] asterisk addon mysql - is mysql connection persistent

[Shahid Tel]

Hi Guys,

As it looks like from CLI command " show cdr mysql " , can somebody confirms
that cdr-mysql creates persistent connection with in asterisk?

" show cdr mysql " shows " connected to u...@dbhost from 18 hours ..."
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Re: [asterisk-users] MixMonitor/Queue and Tranfers

[Miguel Molina]

Darrin Henshaw escribió:


2.   The issue does seem to be limited to MixMonitor and the Queue 
application, as in testing I setup mixmonitor on my extension dialed 
it from outside the company(my cell phone) and transferred the call 
without stopping the recording.



I have a couple of questions on this:

1. Are you using SIP/IAX2/whatever extensions as queue members or Agent 
type members?
2. If you are using Agent members, on the queued calls (though is the 
same call) are you recording from the Agent channel (callee) or from the 
client channel (caller)? That would make a difference in case of a 
transfer, because the callee leg changes but the caller leg is the same.


Cheers,

--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center

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Re: [asterisk-users] Caller ID (name) - where does it come from?

[Barry D. Hassler]

This is all excellent information. My primary issue is for calls that are
placed FROM my client's PBX, via VOIP provider (Teliax). The recipients of
those calls are the ones that are not getting the proper CNAM information as
the call comes in.

We just recently ported the client's POTS lines to VOIP, and with the
exception of this issue, all is working well. But, my client is really
unhappy that their callerID NAME isn't showing up.

On Tue, Jul 7, 2009 at 3:42 PM, Frank Bulk  wrote:

> There's a bit of oversimplification going on here -- it's not "a ...
> database".  Different CNAM providers have different databases which are
> populated from many sources.  Most of the data probably matches, but not
> all
> of it.
>
> If the Calling Name is incorrect, the person who received the call will
> have
> to check with their telephony provider (or, if they do their own CNAM
> lookups, with their CNAM provider) to get the name for the calling party
> fixed up (this presumes that the calling party has already verified with
> their own telephony provider that their name is correctly listed).  But
> that's not all of it, either, because the next time the CNAM provider
> refreshes their records, the local fix could be overridden (I'm not sure if
> any CNAM providers have the capability to ignore old/bad data for a record,
> but perhaps so).  Ideally the CNAM provider shares with the calling party
> which database the CNAM provider is using for the calling party, so that
> the
> calling party can try to get it fixed directly with the database provider
> (if that's even possible).
>
> In short, it's a mess.
>
> But because accuracy rates are one of the elements that CNAM providers
> compete on, these usually do get cleaned up.
>
> Frank
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife
> Sent: Tuesday, July 07, 2009 1:27 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Caller ID (name) - where does it come from?
>
> The Caller ID name, "CNAM" is a separate database owned and maintained
> "cooperatively" by the bell operating companies.
>
> Your ITSP is not doing these CNAM lookups for you because they would have
> to
>
> pay the BOC's for the 'dips' into the CNAM database.  CNAM is a little cash
> cow that the BOC's are quick to protect.  As such CNAM dips may not be
> cached or re-sold as a term service that you must agree to with your CNAM
> provider.
>
> As far as solving your CNAM problem, you would need to either choose an
> ITSP
>
> that will provide you with CNAM data on a per-call basis, OR you need to do
> CNAM dips yourself as I (and many others) do.  Beware that some ITSP's
> provide "best-effort" name data culled from various sources.  It's not
> always terrible but it's not 'coke' it's more like 'dollar store' cola. :-)
>
> As a call comes in to your dial plan you can populate the CALLERID(name)
> channel variable using the CURL function in your dialplan as so:
> exten =>
> s,n,Set(CALLERID(name)=${CURL(
> http://cnam1.edicentral.net/getcnam?q=C&f=S&dn
> =${CALLERID(num)})})
>
> AND let's not forget the completely separate issue with getting your
> ITSP-provisioned number ENTERED INTO the CNAM database in the first place,
> so people see "Karl Fife" rather than the "city, state" or worse, some
> string of arcane LATA information.  There's a solution to this problem too
> but I digress...
>
> I've posted my personal notes below from about 18 months ago when I was
> searchign for CNAM providers:
>
> -Karl
>
> CNAM  PROVIDRES:
>
> Metrostat.com
> about 1.5¢ per dip,
> $30 minimum deposit, refundable
> CNAM service not well documented on web site
> A registerd CLEC
>
> Got Name - Out of business?
> 1.5¢ per dip. no minimums, no setup
>
> ClearReach Networks
> .67¢ per dip $200 monthly minimum, resell ok, significant setup fees
>
> 411xml.com
> more expensive than ClearReach.
>
> - Original Message -
> From: Barry D. Hassler
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Sent: Tuesday, July 07, 2009 12:40 PM
> Subject: [asterisk-users] Caller ID (name) - where does it come from?
>
> Hi Folks, having an issue with outbound calls through a VOIP provider.
> Calls
>
> get sent out with the CallerID(number), but where does callerID(name) come
> from? Apparently not from provider, as we are seeing different (sometime
> missing) names on inbound calls, different than what we have configured.
> Apparently this comes from some telco database somewhere? Numbers were
> ported from a wired-telco.
>
> --
> Barry D. Hassler
> President, HCST
>
> http://www.hcst.net/
> 937-427-9000 
>
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Re: [asterisk-users] Resetting Day/Night setting

[Benny Amorsen]

Ira  writes:

> Well, I'm neither but if it was me I'd just add a test that checks to 
> see if the 280 flag is set and it's after midnight and just turn the 
> flag off if that's true. That  makes it so the first call of the day 
> fixes it automatically. Don't know if that's better or not, but it 
> would me I didn't need to learn how to use cron.

How do you know if it's after midnight?


/Benny


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Re: [asterisk-users] Resetting Day/Night setting

[Benny Amorsen]

Jared Smith  writes:

> Sounds like overkill to me... why not just use a GotoIfTime clause in
> your dialplan?

Because he already has one, but also an override. The override should go
away at midnight, which normally requires a cron job.

A stateless way of doing it is something like:

 exten => _X.,1,GotoIf($["${DB(number1/forceopen)}"="STRFTIME(,,%F)"]?open)
 exten => _X.,n,GotoIf($["${DB(number1/forceclose)}"="STRFTIME(,,%F)"]?close)
 exten => _X.,n,GotoIfTime(...?open)
 exten => _X.,n(close),...
 exten => _X.,n,Hangup
 exten => _X.,n(open),...
 exten => _X.,n,Hangup

 exten => _198,1,Set(DB(number1/forceopen)=STRFTIME(,,%F))
 exten => _199,1,Set(DB(number1/forceclose)=STRFTIME(,,%F))

This should make any overrideopen/overrideclose settings only apply for
the day when they were set. Beware, not tested, not even loaded into an
Asterisk.

I actually feel quite clever for coming up with this :) Of course now
everyone will tell me that they have something twice as clever running
in production since '98...


/Benny


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Re: [asterisk-users] Queues recording & CDR

[Kurian Thayil]

Hi,

My apologies Nicolas, a mistake from my part. And I appreciate for
correcting me. Asternic is a  good piece of work.

Regards,

Kurian Thayil.

On Mon, 2009-07-06 at 09:41 -0300, Nicolás Gudiño wrote:
> Hello,
> 
> Just a correction, Asternic Call Center Stats is not from
> asteriskguru. Asteriskguru has its own statistic program that is not
> open source, but free to use. Asternic was written by me (not
> asteriskguru) and has an open source version and a commercial one.
> 
> Best regards,
> 
> --
> Nicolás Gudiño
> Buenos Aires - Argentina
> 
> 
> 
> On Mon, Jul 6, 2009 at 12:08 AM, Kurian Thayil wrote:
> > Hi Sriram,
> >
> > 1. Set the channel variable MonitorFilename before Queue() in dialplan
> > and you can give some meaningful filename for record.
> > 2. I guess you can use an AGI to capture events and then integrate this
> > with a DB in the Backend. This should help you to track the activity.
> > 3. asternic from asteriskguru is kind of OK. Gives you a live and
> > detailed report. Parses the queue_log to the MySQL DB and works. This
> > parse program could be used in your AGI which I mentioned in point 2.
> >
> > Hope this helps.
> >
> > Regards,
> >
> > Kurian Thayil.
> >
> > On Sun, 2009-07-05 at 22:41 +0530, Sriram wrote:
> >> Hi
> >>
> >> 1. I want to record all calls that land to an agent via a queue using
> >> a meaningful name - as of now i name the recorded file on the fly
> >> using {CALLERID} variable so that the file gets stored using the
> >> caller id iunder /var/spool/asterisk/monitor , now if i want to store
> >> it as  how can i do
> >> this ?
> >> 2. I have a CDR issue - when A calls he is put in Queue and say he is
> >> answered by Agent B ..Agent B transfers the Call to agent C as it is
> >> to Agent C whom A wants to talk..when the call gets d/c the CDR for
> >> that call shows the destination field as B whereas it shd be C...how
> >> do i take care of this ...in my call center agents are paid on the
> >> basis of talk time on inbound calls - this way an agent who just
> >> transfers calls is at merry !!
> >> 3. Are their any GPL based queue reporting software - hows the
> >> asterisk queue statistics program from asteriskguru.com has anyone
> >> tried it ?
> >>
> >> Thanks
> >> Sriram
> >> ___
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> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> > --
> > Kurian Mathew Thayil.
> > (GPG KeyID: E232394F)
> >
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> >
> 
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-- 
Kurian Mathew Thayil.
(GPG KeyID: E232394F)


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Re: [asterisk-users] Caller ID (name) - where does it come from?

[Frank Bulk]

There's a bit of oversimplification going on here -- it's not "a ...
database".  Different CNAM providers have different databases which are
populated from many sources.  Most of the data probably matches, but not all
of it. 

If the Calling Name is incorrect, the person who received the call will have
to check with their telephony provider (or, if they do their own CNAM
lookups, with their CNAM provider) to get the name for the calling party
fixed up (this presumes that the calling party has already verified with
their own telephony provider that their name is correctly listed).  But
that's not all of it, either, because the next time the CNAM provider
refreshes their records, the local fix could be overridden (I'm not sure if
any CNAM providers have the capability to ignore old/bad data for a record,
but perhaps so).  Ideally the CNAM provider shares with the calling party
which database the CNAM provider is using for the calling party, so that the
calling party can try to get it fixed directly with the database provider
(if that's even possible).  

In short, it's a mess.  

But because accuracy rates are one of the elements that CNAM providers
compete on, these usually do get cleaned up.

Frank

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife
Sent: Tuesday, July 07, 2009 1:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID (name) - where does it come from?

The Caller ID name, "CNAM" is a separate database owned and maintained 
"cooperatively" by the bell operating companies.

Your ITSP is not doing these CNAM lookups for you because they would have to

pay the BOC's for the 'dips' into the CNAM database.  CNAM is a little cash 
cow that the BOC's are quick to protect.  As such CNAM dips may not be 
cached or re-sold as a term service that you must agree to with your CNAM 
provider.

As far as solving your CNAM problem, you would need to either choose an ITSP

that will provide you with CNAM data on a per-call basis, OR you need to do 
CNAM dips yourself as I (and many others) do.  Beware that some ITSP's 
provide "best-effort" name data culled from various sources.  It's not 
always terrible but it's not 'coke' it's more like 'dollar store' cola. :-)

As a call comes in to your dial plan you can populate the CALLERID(name) 
channel variable using the CURL function in your dialplan as so:
exten => 
s,n,Set(CALLERID(name)=${CURL(http://cnam1.edicentral.net/getcnam?q=C&f=S&dn
=${CALLERID(num)})})

AND let's not forget the completely separate issue with getting your 
ITSP-provisioned number ENTERED INTO the CNAM database in the first place, 
so people see "Karl Fife" rather than the "city, state" or worse, some 
string of arcane LATA information.  There's a solution to this problem too 
but I digress...

I've posted my personal notes below from about 18 months ago when I was 
searchign for CNAM providers:

-Karl

CNAM  PROVIDRES:

Metrostat.com
about 1.5¢ per dip,
$30 minimum deposit, refundable
CNAM service not well documented on web site
A registerd CLEC

Got Name - Out of business?
1.5¢ per dip. no minimums, no setup

ClearReach Networks
.67¢ per dip $200 monthly minimum, resell ok, significant setup fees

411xml.com
more expensive than ClearReach.

- Original Message - 
From: Barry D. Hassler
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tuesday, July 07, 2009 12:40 PM
Subject: [asterisk-users] Caller ID (name) - where does it come from?

Hi Folks, having an issue with outbound calls through a VOIP provider. Calls

get sent out with the CallerID(number), but where does callerID(name) come 
from? Apparently not from provider, as we are seeing different (sometime 
missing) names on inbound calls, different than what we have configured. 
Apparently this comes from some telco database somewhere? Numbers were 
ported from a wired-telco.

-- 
Barry D. Hassler
President, HCST

http://www.hcst.net/
937-427-9000

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Re: [asterisk-users] Call parking with ISDN

[Wilton Helm]

Since no one has responded to this, I am wondering if there are two kinds of
call park.  I haven't worked with European ISDN, but if it has a call park
feature, that would be distinctly different from the Asterisk PABX call park
feature.

The Asterisk feature should not matter what sort of trunk was involved,
which is why I am wondering.  On the other hand, if there is an ISDN park,
I'm not sure Asterisk would support it.

Wilton



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Re: [asterisk-users] Caller ID (name) - where does it come from?

[Karl Fife]

p.s.
Once you've got a reliable CNAM source, you can save a few bucks per month on 
all of your POTS lines & PRI spans by opting out of the carrier-provided CNAM. 
IIRC, We save something like $40 per month per span on our PRI's & $3 per month 
per line by opting out of CNAM.  When a call comes in we populate it ourselves 
using a quick HTTP GET. 

-Karl 







- Original Message - 
  From: Barry D. Hassler 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Tuesday, July 07, 2009 12:40 PM
  Subject: [asterisk-users] Caller ID (name) - where does it come from?


  Hi Folks, having an issue with outbound calls through a VOIP provider. Calls 
get sent out with the CallerID(number), but where does callerID(name) come 
from? Apparently not from provider, as we are seeing different (sometime 
missing) names on inbound calls, different than what we have configured. 
Apparently this comes from some telco database somewhere? Numbers were ported 
from a wired-telco.



  -- 
  Barry D. Hassler
  President, HCST

  http://www.hcst.net/
  937-427-9000



--


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Re: [asterisk-users] chan_mobile help.

[Sasa Bobek]

Could not agree more.  I had chan_mobile up and running with an older
version of Trix, but never managed to recreate it with the latest versions.
 Other people I talked to even suggested that it was made on purpose.  With
elastix the only problem I had was the missing mobile.conf.example, but you
can create one from the Trix instructions from scratch or download it from
the SVN.

On Tue, Jul 7, 2009 at 7:56 PM, Razza  wrote:

> Seems the only option is to give Elastix a go.
>
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[asterisk-users] Play a recorded message when a fax is detected ?

[Olivier]

Hi,

I'm configuring a system so that end user can receive phone and calls using
the same extension and DID.
At the moment, fax are correctly detected but I'm trying to improve end user
experience.

Relevant dialplan (from extensions.ael) is :
fax => {
Verbose(0,Incoming fax from ${CALLERID(num)});
FAXFILE="/var/spool/asterisk/fax/${UNIQUEID}.tif";
ReceiveFAX(${FAXFILE});
HangUp();
};


What I would to improve is when a fax is detected, instead of hanging up the
receiving extension, play a recorded message like "you're receiving a fax"
(if receiving end is human, or nothing at all if it's a voicemail).

What would you advise me to try ?

Regards
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Re: [asterisk-users] Caller ID (name) - where does it come from?

[Karl Fife]

The Caller ID name, "CNAM" is a separate database owned and maintained 
"cooperatively" by the bell operating companies.

Your ITSP is not doing these CNAM lookups for you because they would have to 
pay the BOC's for the 'dips' into the CNAM database.  CNAM is a little cash 
cow that the BOC's are quick to protect.  As such CNAM dips may not be 
cached or re-sold as a term service that you must agree to with your CNAM 
provider.

As far as solving your CNAM problem, you would need to either choose an ITSP 
that will provide you with CNAM data on a per-call basis, OR you need to do 
CNAM dips yourself as I (and many others) do.  Beware that some ITSP's 
provide "best-effort" name data culled from various sources.  It's not 
always terrible but it's not 'coke' it's more like 'dollar store' cola. :-)

As a call comes in to your dial plan you can populate the CALLERID(name) 
channel variable using the CURL function in your dialplan as so:
exten => 
s,n,Set(CALLERID(name)=${CURL(http://cnam1.edicentral.net/getcnam?q=C&f=S&dn=${CALLERID(num)})})

AND let's not forget the completely separate issue with getting your 
ITSP-provisioned number ENTERED INTO the CNAM database in the first place, 
so people see "Karl Fife" rather than the "city, state" or worse, some 
string of arcane LATA information.  There's a solution to this problem too 
but I digress...

I've posted my personal notes below from about 18 months ago when I was 
searchign for CNAM providers:

-Karl

CNAM  PROVIDRES:

Metrostat.com
about 1.5¢ per dip,
$30 minimum deposit, refundable
CNAM service not well documented on web site
A registerd CLEC

Got Name - Out of business?
1.5¢ per dip. no minimums, no setup

ClearReach Networks
.67¢ per dip $200 monthly minimum, resell ok, significant setup fees

411xml.com
more expensive than ClearReach.

- Original Message - 
From: Barry D. Hassler
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tuesday, July 07, 2009 12:40 PM
Subject: [asterisk-users] Caller ID (name) - where does it come from?


Hi Folks, having an issue with outbound calls through a VOIP provider. Calls 
get sent out with the CallerID(number), but where does callerID(name) come 
from? Apparently not from provider, as we are seeing different (sometime 
missing) names on inbound calls, different than what we have configured. 
Apparently this comes from some telco database somewhere? Numbers were 
ported from a wired-telco.



-- 
Barry D. Hassler
President, HCST

http://www.hcst.net/
937-427-9000




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Re: [asterisk-users] Fax for Asterisk - Fax routing based on TSID

[Doug Lytle]

Olivier wrote:
>
>
>
> Please, allow me to ask what is this Transmitting Station ID ?
>

Google is you friend:

http://encyclopedia.thefreedictionary.com/Transmitting+Subscriber+Identification

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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Re: [asterisk-users] Resetting Day/Night setting

[Tilghman Lesher]

On Tuesday 07 July 2009 11:54:54 Jeremy Winder wrote:
> The problem is in the case of say a holiday on a Friday, when the tech
> support manager decides at the last minute to let his people go early
> and dials *280 and then leaves. Come Monday the override will still be
> in place and since call volumes are usually low in the morning, it could
> be noon or later before someone realizes something is wrong.
>
> My thought was to use cron to run a script that will check the status of
> the Day/Night Control and compare it with the Time Condition and if they
> match, set the Day/Night Control back to default (day). So in our
> holiday scenario, come 5:00pm that Friday night when the Time Condition
> would switch to "after hours" the Day/Night Control switches back to its
> default setting.
>
> Being new to voice systems, I knew how I would handle it as a Linux
> Sysadmin, I was curious how "telecom guys" would go about it.
>
> To answer your question, I believe FreePBX is using a GotoIfTime clause
> for the Time Condition but I'm not exactly sure. I'm more worried about
> giving our tech support manager the ability to override the "normal"
> dial plan without having to call me.

The way that I've implemented it when customers needed it this way was to use
a common flag either in astdb or in a database, which reflected the 1/0 state
of the day/night interface, then used a cronjob to switch the states back and
forth, knowing that a state switch could always occur prior to the automatic
switch (that is, someone could arrive early in the morning and switch the
system to day mode early or leave early and switch the system to night mode
early).

-- 
Tilghman & Teryl
with Peter, Cottontail, Midnight, Thumper, & Johnny (bunnies)
and Harry, BB, & George (dogs)

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Re: [asterisk-users] Fax for Asterisk - Fax routing based on TSID

[Olivier]

2009/7/7 Karl Fife 

> I'm using Fax For Asterisk, and trying to optimize the user experience
> while
> ROUTING faxes based on the Transmitting Station ID [sic] (NOT the
> CALLERID).
>

Please, allow me to ask what is this Transmitting Station ID ?



>
>
> Thanks
> -Karl
>
>
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[asterisk-users] MixMonitor/Queue and Tranfers

[Darrin Henshaw]

Hello,



First off to lay the ground work, I’m running Asterisk 1.4.25, which was
recently upgraded from 1.2 about one month ago. We are running it on CentOS
4.7, on Dell PoweEdge 1950’s.



We are a small MSP(Managed Service Provider) providing
Network/Server/Desktop support for companies based out of the Carribean. The
problem I’m having is as the subject states deals with MixMonitor and
transferring.



When a call comes into our system, it is trunked through IAX to another
office, and then we do some setting of the callerid name based on the
callerid number(so the tech knows what client they are talking to), start
MixMonitor to record the call(helps tremendously in a he said she said
scenario) and also depending on the time drop the call into the right queue.
Then obviously the call is picked up and Bob’s your uncle hopefully the tech
can fix the issue.



The problem we are running into is when the initial tech cannot fix the
issue, or the person needs to speak to someone else. I can see in my CDR
records and queue logs where the call is transferred, but the second leg of
the conversation is not recorded. I can see on the console and through the
logs where MixMonitor stops recording and nothing else is recorded.



I’ve posted a bug here, https://issues.asterisk.org/view.php?id=15426, but
haven’t heard feedback so I thought to post here. If you want configs I
should be able to provide them.



Now for some things we have tried:



1.   We’ve set the AUDIOHOOK_INHERIT variable however, that does not
work.

2.   The issue does seem to be limited to MixMonitor and the Queue
application, as in testing I setup mixmonitor on my extension dialed it from
outside the company(my cell phone) and transferred the call without stopping
the recording.



This did work fine in 1.2, however, switching to 1.4 seems to have
introduced this into our environment. Thank you for any assistance you can
provide.



Cheers,



Darrin Henshaw
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Re: [asterisk-users] How to debug "Nothing to pick up" ?

[Olivier]

2009/7/7 Peder 

>  More info is needed.  Can you send relevant portions of config, version,
> etc?
>
I'm using 1.6.1.1 and (lots of) macros


> Also, are you using Macro’s?  I know there was an issue with call pickup
> when the calls were using macros, but I don’t know when/if that was fixed.
>
Thanks for this information : I had a quick look at mantis before writing
here but couldn't anything obvious ...
I'm about to re-install latest trunk version to see if I can reproduce this
...

Regards

>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier
> *Sent:* Tuesday, July 07, 2009 1:29 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] How to debug "Nothing to pick up" ?
>
>
>
> Hi,
>
> General pickup doesn't seem to work here while directed pickup do.
>
> -- SIP/7530-08338f80 is ringing
>   == Using SIP RTP CoS mark 5
> [Jul  7 08:20:03] NOTICE[2299]: chan_sip.c:18383 handle_request_invite:
> Nothing to pick up for d61a727f746a9304
>
>
> Upgrading debug level to 5, doesn't improve console output.
> Which is the best way to debug this ?
> I'm not hoping to find root cause I dare not file a bug report while I'm
> not certain it's not a configuration issue.
>
> Regards
>
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Re: [asterisk-users] chan_mobile help.

[Razza]

Seems the only option is to give Elastix a go.
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Re: [asterisk-users] Caller ID (name) - where does it come from?

[Steve Totaro]

On Tue, Jul 7, 2009 at 1:40 PM, Barry D. Hassler wrote:
> Hi Folks, having an issue with outbound calls through a VOIP provider. Calls
> get sent out with the CallerID(number), but where does callerID(name) come
> from? Apparently not from provider, as we are seeing different (sometime
> missing) names on inbound calls, different than what we have configured.
> Apparently this comes from some telco database somewhere? Numbers were
> ported from a wired-telco.
>
>
>
> --
> Barry D. Hassler
> President, HCST
>
> http://www.hcst.net/
> 937-427-9000
>

It is in a database or CNAM dip.

You just need to contact your provider and tell them to have it
changed.  What you send is moot on the PSTN.

Also call 911 and make sure they have the correct address and
information on file.  I do it all the time for liability reasons, just
make sure you tell them right off that bat that there is no emergency
and you want to verify what they have in their database is correct.

I have never had a problem doing this and try to do it in front of the
big boss to show them it is correct and that I am thorough and looking
out for them.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

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[asterisk-users] Caller ID (name) - where does it come from?

[Barry D. Hassler]

Hi Folks, having an issue with outbound calls through a VOIP provider. Calls
get sent out with the CallerID(number), but where does callerID(name) come
from? Apparently not from provider, as we are seeing different (sometime
missing) names on inbound calls, different than what we have configured.
Apparently this comes from some telco database somewhere? Numbers were
ported from a wired-telco.



-- 
Barry D. Hassler
President, HCST

http://www.hcst.net/
937-427-9000
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Re: [asterisk-users] Resetting Day/Night setting

[Ira]

At 09:54 AM 7/7/2009, you wrote:
>The problem is in the case of say a holiday on a Friday, when the tech
>support manager decides at the last minute to let his people go early
>and dials *280 and then leaves. Come Monday the override will still be
>in place and since call volumes are usually low in the morning, it could
>be noon or later before someone realizes something is wrong.
>
>Being new to voice systems, I knew how I would handle it as a Linux
>Sysadmin, I was curious how "telecom guys" would go about it.

Well, I'm neither but if it was me I'd just add a test that checks to 
see if the 280 flag is set and it's after midnight and just turn the 
flag off if that's true. That  makes it so the first call of the day 
fixes it automatically. Don't know if that's better or not, but it 
would me I didn't need to learn how to use cron.

Ira 


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Re: [asterisk-users] Resetting Day/Night setting

[Jeremy Winder]

All of this is setup using FreePBX and to be honest, with all of the
macros that FreePBX adds to the *.conf files I'm not really sure what
Asterisk is doing. I've only been playing with all of this for about two
weeks now and most of that was waisted trying to figure out the ZAPTEL
vs DAHDI stuff.

My planned incoming routes will look something like this:

  
 (Incoming)
  
  |
 /\   /\
/  \ /  \
   / if \_After_/Day\\__Day
   \time/ Hours \Nite/ |
\  / \  /  |
 \/   \/   |
  |Work|Night  |
 /\ Hours \/   \/
/  \ -  -
   /Day\\_Day__>(Queue)(VoiceMail)
   \Nite/-  -
\  /   /\
 \/|
  |Night   |
  --

I left out all of the announcements and IVR because I "love" drawing in
ASCII so much, but you get the general idea.

With the Time Condition (if time) we will have automated routing of the
calls to the queue during work hours and voicemail after hours. However,
there are a few times a year when we need to overrule this time
condition for situations where tech support is offered on a weekend or
we are closed on a holiday. My thought was instead of constantly having
to update the Time Condition, we could use the Day/Night Control which
gives a '*280' dial code.

The problem is in the case of say a holiday on a Friday, when the tech
support manager decides at the last minute to let his people go early
and dials *280 and then leaves. Come Monday the override will still be
in place and since call volumes are usually low in the morning, it could
be noon or later before someone realizes something is wrong.

My thought was to use cron to run a script that will check the status of
the Day/Night Control and compare it with the Time Condition and if they
match, set the Day/Night Control back to default (day). So in our
holiday scenario, come 5:00pm that Friday night when the Time Condition
would switch to "after hours" the Day/Night Control switches back to its
default setting.

Being new to voice systems, I knew how I would handle it as a Linux
Sysadmin, I was curious how "telecom guys" would go about it.

To answer your question, I believe FreePBX is using a GotoIfTime clause
for the Time Condition but I'm not exactly sure. I'm more worried about
giving our tech support manager the ability to override the "normal"
dial plan without having to call me.

Thanks again,

Jeremy

On Tue, 2009-07-07 at 11:32 -0400, Jared Smith wrote:
> On Tue, 2009-07-07 at 10:47 -0400, Jeremy Winder wrote:
> > It seemed to me cron was going to be the best solution.
> 
> Sounds like overkill to me... why not just use a GotoIfTime clause in
> your dialplan?
> 
> 


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Re: [asterisk-users] Resetting Day/Night setting

[Jared Smith]

On Tue, 2009-07-07 at 10:47 -0400, Jeremy Winder wrote:
> It seemed to me cron was going to be the best solution.

Sounds like overkill to me... why not just use a GotoIfTime clause in
your dialplan?


-- 
Jared Smith
Training Manager
Digium, Inc.


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Re: [asterisk-users] What is the best way to share extension state

[Benny Amorsen]

Jim Dickenson  writes:

> http://bugs.digium.com/view.php?id=14595 has a patch to add a new class,
> bridge, so you get less events in AMI. This is for 1.6.0.x. It will give you
> an idea of what needs to be changed in order to make the call class of
> messages more granular.

It's nice to see that work is done to make it more granular. However,
doesn't that break backwards compatibility, in the people who request
call now don't get bridge events?

The challenge is that eventually every single manager_event will have
its own type...

Anyway, that's a worry for another time. It's really neat!


/Benny


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[asterisk-users] Automatic Gain Control

[Brent Davidson]

Is there any possibility of DAHDI supporting Automatic gain control on 
TDM ports?  I'm having issues at a couple of offices where calls made to 
local numbers are fine but a when a calls from or goes to a large 
percentage of long-distance or 1-800 numbers the person at the remote 
end cannot hear the person in my office.  Boosting the gains in 
zapata.conf (I'm still using 1.4.21) to 8 solves the problem with 
long-distance lines, but then local calls say the person in my office is 
too loud.

I understand that it is going to be difficult to reliably detect a major 
drop in the volume at the far end of the call, but I'm just wondering if 
there is a good solution for this.  We're using Rhino WC4-FXO-ec cards 
and the OSlec echo canceler (since the on-board echo canceler didn't 
seem to help our echo issues)

Thanks,
Brent

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Re: [asterisk-users] Fax for Asterisk - Fax routing based on TSID

[Kevin P. Fleming]

Karl Fife wrote:

> However if I were to drive the fax process from manager rather than from the 
> dialplan, would the TSID be available for read before RX completion, AND 
> (equally important) would sendfax even tolerate sending a fax from an 
> still-spooling RX file?  In other words, does the  application require that 
> the spool file be complete and closed before it would allow you to begin 
> paying it out 'from the top'?

I don't think that the current FFA applications will do what you want.
Even if you could somehow manage this, any delay in the receiving
process could cause it to get 'behind' the sending process, which would
then abort due to an incomplete file. That's even assuming that the
SendFAX application will accept an incomplete TIFF/F file in the first
place :-)

Even with FAX gateway support you wouldn't be able to achieve this,
because the gateway outbound link would be setup before the TSID is
received.

What might be possible is to have a page-per-file mode, so that as each
page is received it is stored in a separate file; you could then watch
for 'received page' events and transmit the pages that had been received
up to that point. I'll do a bit of research to see if this is possible.

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

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Re: [asterisk-users] Fax for Asterisk - Fax routing based on TSID

[Danny Nicholas]

Receipt of the fax could be handled in the manner you describe using
something like Ghostscript to parse the partial file into pages as it came
in.  Sending the file is pretty much an "all or none" proposition unless you
did something funky like splitting the file into chunks and sending in
consecutive sessions.  I'm pretty sure FFA doesn't let you say send 1&2&3
etc.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife
Sent: Tuesday, July 07, 2009 10:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Fax for Asterisk - Fax routing based on TSID

I'm using Fax For Asterisk, and trying to optimize the user experience while

ROUTING faxes based on the Transmitting Station ID [sic] (NOT the CALLERID).

Specifically I'm trying to eliminate end-user confusion as "that fifty page 
fax" spools to a file, creating a significant delay, leaving the user to 
wonder "why am I not getting my fax--they said it's currently sending".  A 
better user-experience would be to begin TX'ing the fax (at the same or 
slower speed than RX) after a page or two has  been spooled as a buffer.

If I'm using SendFax from the dialplan it appears that I would need to spool

an entire incoming fax to file (to completion) before I could route said fax

based on the Transmitting Station ID

However if I were to drive the fax process from manager rather than from the

dialplan, would the TSID be available for read before RX completion, AND 
(equally important) would sendfax even tolerate sending a fax from an 
still-spooling RX file?  In other words, does the  application require that 
the spool file be complete and closed before it would allow you to begin 
paying it out 'from the top'?


Thanks
-Karl


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[asterisk-users] Fax for Asterisk - Fax routing based on TSID

[Karl Fife]

I'm using Fax For Asterisk, and trying to optimize the user experience while 
ROUTING faxes based on the Transmitting Station ID [sic] (NOT the CALLERID).

Specifically I'm trying to eliminate end-user confusion as "that fifty page 
fax" spools to a file, creating a significant delay, leaving the user to 
wonder "why am I not getting my fax--they said it's currently sending".  A 
better user-experience would be to begin TX'ing the fax (at the same or 
slower speed than RX) after a page or two has  been spooled as a buffer.

If I'm using SendFax from the dialplan it appears that I would need to spool 
an entire incoming fax to file (to completion) before I could route said fax 
based on the Transmitting Station ID

However if I were to drive the fax process from manager rather than from the 
dialplan, would the TSID be available for read before RX completion, AND 
(equally important) would sendfax even tolerate sending a fax from an 
still-spooling RX file?  In other words, does the  application require that 
the spool file be complete and closed before it would allow you to begin 
paying it out 'from the top'?


Thanks
-Karl


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Re: [asterisk-users] Resetting Day/Night setting

[Danny Nicholas]

Every minute seems like overkill.  Why not every hour/half hour/15 minutes?
Also, you could cue a call file to do this a few times a day and let
asterisk do the cron for you.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeremy Winder
Sent: Tuesday, July 07, 2009 9:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Resetting Day/Night setting

It seemed to me cron was going to be the best solution. I'll create a
script that will run every minute or so that compares the Day/Night
Control with the Time Condition and when their outcomes match, reset the
Day/Night Control back to default.

I was curious as to how others would approach the same problem.

Thanks again,

Jeremy

On Tue, 2009-07-07 at 09:14 -0500, Danny Nicholas wrote:
> I concur that this is probably the best solution;  We could provide more
> helpful answers with more question details.  There are usually at least
two
> "correct" solutions to any query you can post; the more detail you can
> provide, the more likely you are to get a correct and efficient answer.
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jimmy
Godbout
> Sent: Tuesday, July 07, 2009 9:08 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Resetting Day/Night setting
> 
> Use a cron entry to reset the setting using a cli command, as long as the
> setting is in the internal database of asterisk.
> 
> > -Original Message-
> > From: jwin...@logicalsi.com
> > Sent: Tue, 07 Jul 2009 09:44:58 -0400
> > To: asterisk-users@lists.digium.com
> > Subject: [asterisk-users] Resetting Day/Night setting
> > 
> > I'm not sure if this is part of Asterisk or FreePBX so I apologize if
> > this is the wrong list to ask my question.
> > 
> > As part of my companies call routes, I have a Time Condition for our
> > tech support queue. I would like to add a Day/Night Control so the Time
> > Condition can be overruled. However, I'm afraid someone will forget to
> > turn it off again. What is the best way of resetting this control back
> > to its default setting at say midnight?
> > 
> > Thanks in advance,
> > 
> > Jeremy
> > 
> > 
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] documentation of DAHDI dial options

[Jared Smith]

On Tue, 2009-07-07 at 15:42 +0200, Klaus Darilion wrote:
> I am searching for the description of the available dialstrin options 
> for the DAHDI channel (and also other channel types).
> 
> I am not looking for outdated voip-info links, but for the authoritative 
> source, e.g. something like "core show application Dial"
> 
> Does such thing exists?

I don't think that such a thing exists.  The only ones I'm aware of are:

1) Channel Groups.  

DAHDI/g1/5551212 dials 5551212 on the first available channel in group
one, searching from lowest to highest

DAHDI/G1/5551212 dials 5551212 on the first available channel in group
one, searching from highest to lowest

DAHDI/r1/5551212 dials 5551212 on the first available channel in group
one, going in round-robin fashion (and remembering where it last left
off), searching from lowest to highest

DAHDI/R1/5551212 dials 5551212 on the first available channel in group
one, searching in round-robin fashion from highest to lowest.

2) Distinctive ring

DAHDI/4r1 dials channel 4 (presumably an FXS channel), and uses
distinctive ring style one.  If I recall, there are four different
distinctive ring styles... so you could replace r1 with r2, r3, or r4.

3) Answer confirmation

DAHDI/1c/5551212 tells Asterisk to dial 5551212 on DAHDI channel 1, and
not consider the call answered until the called party presses #.  This
is useful because of the way analog signaling works.  Without this
setting, Asterisk considers any outbound analog call on an FXO port
answered just as soon as it has been dialed.

4) Digital calls

DAHDI/1d/5551212 tells Asterisk to dial 5551212 on DAHDI channel 1, and
that it's a digital call.  If I remember correctly, this is used for
ISDN calls to set the bearer capability.

I've taken a quick look in channels/chan_dahdi.c in TRUNK, and it seems
to match up with my understanding, as I didn't see any other options
stand out.  While poking around in there, I found the following comment:

/*
 * data is ---v
 * Dial(DAHDI/pseudo[/extension])
 * Dial(DAHDI/[c|r|d][/extension])
 * Dial(DAHDI/(g|G|r|R)[c|r|d][/extension])
 *
 * g - channel group allocation search forward
 * G - channel group allocation search backward
 * r - channel group allocation round robin search forward
 * R - channel group allocation round robin search backward
 *
 * c - Wait for DTMF digit to confirm answer
 * r - Set distintive ring cadance number
 * d - Force bearer capability for ISDN/SS7 call to digital.
 */

That's probably as definitive an answer as you're going to get.


-- 
Jared Smith
Training Manager
Digium, Inc.


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Re: [asterisk-users] Resetting Day/Night setting

[Jeremy Winder]

It seemed to me cron was going to be the best solution. I'll create a
script that will run every minute or so that compares the Day/Night
Control with the Time Condition and when their outcomes match, reset the
Day/Night Control back to default.

I was curious as to how others would approach the same problem.

Thanks again,

Jeremy

On Tue, 2009-07-07 at 09:14 -0500, Danny Nicholas wrote:
> I concur that this is probably the best solution;  We could provide more
> helpful answers with more question details.  There are usually at least two
> "correct" solutions to any query you can post; the more detail you can
> provide, the more likely you are to get a correct and efficient answer.
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jimmy Godbout
> Sent: Tuesday, July 07, 2009 9:08 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Resetting Day/Night setting
> 
> Use a cron entry to reset the setting using a cli command, as long as the
> setting is in the internal database of asterisk.
> 
> > -Original Message-
> > From: jwin...@logicalsi.com
> > Sent: Tue, 07 Jul 2009 09:44:58 -0400
> > To: asterisk-users@lists.digium.com
> > Subject: [asterisk-users] Resetting Day/Night setting
> > 
> > I'm not sure if this is part of Asterisk or FreePBX so I apologize if
> > this is the wrong list to ask my question.
> > 
> > As part of my companies call routes, I have a Time Condition for our
> > tech support queue. I would like to add a Day/Night Control so the Time
> > Condition can be overruled. However, I'm afraid someone will forget to
> > turn it off again. What is the best way of resetting this control back
> > to its default setting at say midnight?
> > 
> > Thanks in advance,
> > 
> > Jeremy
> > 
> > 
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> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
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Re: [asterisk-users] Resetting Day/Night setting

[Danny Nicholas]

I concur that this is probably the best solution;  We could provide more
helpful answers with more question details.  There are usually at least two
"correct" solutions to any query you can post; the more detail you can
provide, the more likely you are to get a correct and efficient answer.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jimmy Godbout
Sent: Tuesday, July 07, 2009 9:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Resetting Day/Night setting

Use a cron entry to reset the setting using a cli command, as long as the
setting is in the internal database of asterisk.

> -Original Message-
> From: jwin...@logicalsi.com
> Sent: Tue, 07 Jul 2009 09:44:58 -0400
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Resetting Day/Night setting
> 
> I'm not sure if this is part of Asterisk or FreePBX so I apologize if
> this is the wrong list to ask my question.
> 
> As part of my companies call routes, I have a Time Condition for our
> tech support queue. I would like to add a Day/Night Control so the Time
> Condition can be overruled. However, I'm afraid someone will forget to
> turn it off again. What is the best way of resetting this control back
> to its default setting at say midnight?
> 
> Thanks in advance,
> 
> Jeremy
> 
> 
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Re: [asterisk-users] Resetting Day/Night setting

[Jimmy Godbout]

Use a cron entry to reset the setting using a cli command, as long as the 
setting is in the internal database of asterisk.

> -Original Message-
> From: jwin...@logicalsi.com
> Sent: Tue, 07 Jul 2009 09:44:58 -0400
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Resetting Day/Night setting
> 
> I'm not sure if this is part of Asterisk or FreePBX so I apologize if
> this is the wrong list to ask my question.
> 
> As part of my companies call routes, I have a Time Condition for our
> tech support queue. I would like to add a Day/Night Control so the Time
> Condition can be overruled. However, I'm afraid someone will forget to
> turn it off again. What is the best way of resetting this control back
> to its default setting at say midnight?
> 
> Thanks in advance,
> 
> Jeremy
> 
> 
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[asterisk-users] Resetting Day/Night setting

[Jeremy Winder]

I'm not sure if this is part of Asterisk or FreePBX so I apologize if
this is the wrong list to ask my question.

As part of my companies call routes, I have a Time Condition for our
tech support queue. I would like to add a Day/Night Control so the Time
Condition can be overruled. However, I'm afraid someone will forget to
turn it off again. What is the best way of resetting this control back
to its default setting at say midnight?

Thanks in advance,

Jeremy


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[asterisk-users] documentation of DAHDI dial options

[Klaus Darilion]

Hi!

I am searching for the description of the available dialstrin options 
for the DAHDI channel (and also other channel types).

I am not looking for outdated voip-info links, but for the authoritative 
source, e.g. something like "core show application Dial"

Does such thing exists?

thanks
Klaus

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Re: [asterisk-users] How to debug "Nothing to pick up" ?

[Peder]

More info is needed.  Can you send relevant portions of config, version,
etc?  Also, are you using Macro's?  I know there was an issue with call
pickup when the calls were using macros, but I don't know when/if that was
fixed.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Tuesday, July 07, 2009 1:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to debug "Nothing to pick up" ?

 

Hi,

General pickup doesn't seem to work here while directed pickup do.

-- SIP/7530-08338f80 is ringing
  == Using SIP RTP CoS mark 5
[Jul  7 08:20:03] NOTICE[2299]: chan_sip.c:18383 handle_request_invite:
Nothing to pick up for d61a727f746a9304


Upgrading debug level to 5, doesn't improve console output.
Which is the best way to debug this ?
I'm not hoping to find root cause I dare not file a bug report while I'm not
certain it's not a configuration issue.

Regards

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Re: [asterisk-users] Answering the nTh call ...

[Jimmy Godbout]

You can take a look at 

-> app_contest allows you to easily run a 'radio station contest line' where 
you can specify a certain caller number and they will be connected, but all 
other callers will be rejected with some message 
http://www.freeswitch.org/asterisk_stuff/app_contest.c

it was made for 1.2 but you could probably port it to 1.4

Jimmy

> -Original Message-
> From: s...@infiltrated.net
> Sent: Tue, 07 Jul 2009 09:05:48 -0400
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Answering the nTh call ...
> 
> 
> Curious to know if anyone's created something similar to the following,
> if so and you'd care to share an AGI or dialplan, much appreciated.
> 
> I will be eventually write a script to answer the nTH call. (if I can't
> find it (why reinvent wheels).
> 
> Looking to do some testing sending anywhere between 50-200 calls to a
> machine. I'd like a Snom/Polycom/whatever to pick up after the nTh call
> where nTh is whatever I set it to.
> 
> exten => _X.,1,{at_N_amount_of_rings}
> exten => _X.,2,Dial(SIP/${ext...@somwhere,45)
> 
> So as a script it would be something like: if call != 25th ; then go
> elsewhere ; fi (make sense?) Where every 25th call or so would go
> through, the others would go wherever, not important.
> 
> --
> 
> =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
> J. Oquendo
> SGFA, SGFE, C|EH, CNDA, CHFI, OSCP
> 
> "It takes 20 years to build a reputation and five minutes to
> ruin it. If you think about that, you'll do things
> differently." - Warren Buffett
> 
> 227C 5D35 7DCB 0893 95AA  4771 1DCE 1FD1 5CCD 6B5E
> http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x5CCD6B5E
> 
> 
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Re: [asterisk-users] Answering the nTh call ...

[Danny Nicholas]

Here is a crack at it
[globals]
CALLCOUNT=0
MAXCOUNT=25
[default]
- exten => s,1,answer()
- exten => s,n,Set(GLOBAL(CALLCOUNT)=${CALLCOUNT}+1)
- exten => s,n,Gotoif($["${CALLCOUNT}" = "${MAXCOUNT}"]?callout)
- exten => s,n,Dial(SIP/100,20,m)
- exten => s,n,Hangup
- exten => s,n(callout),Set(GLOBAL(CALLCOUNT)=0)
- exten => s,n,Dial(SIP/200,20,m)
- exten => s,n,Hangup
 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of J. Oquendo
Sent: Tuesday, July 07, 2009 8:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Answering the nTh call ...


Curious to know if anyone's created something similar to the following,
if so and you'd care to share an AGI or dialplan, much appreciated.

I will be eventually write a script to answer the nTH call. (if I can't
find it (why reinvent wheels).

Looking to do some testing sending anywhere between 50-200 calls to a
machine. I'd like a Snom/Polycom/whatever to pick up after the nTh call
where nTh is whatever I set it to.

exten => _X.,1,{at_N_amount_of_rings}
exten => _X.,2,Dial(SIP/${ext...@somwhere,45)

So as a script it would be something like: if call != 25th ; then go
elsewhere ; fi (make sense?) Where every 25th call or so would go
through, the others would go wherever, not important.

-- 

=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
J. Oquendo
SGFA, SGFE, C|EH, CNDA, CHFI, OSCP

"It takes 20 years to build a reputation and five minutes to
ruin it. If you think about that, you'll do things
differently." - Warren Buffett

227C 5D35 7DCB 0893 95AA  4771 1DCE 1FD1 5CCD 6B5E
http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x5CCD6B5E


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Re: [asterisk-users] Some IAX calls do not disconnect.

[John Novack]

Steve Totaro wrote:
> On Tue, Jul 7, 2009 at 7:43 AM, Tim Panton wrote:
>   
>> On 7 Jul 2009, at 05:05, Steve Totaro wrote:
>>
>> 
>>> On Mon, Jul 6, 2009 at 9:41 PM, Tim Nelson wrote:
>>>   
 - "Steve Totaro"  wrote:
 
> Just use SIP and solve all your problems.
>   
 I seem to be noticing a common element to your posts about IAX. :-)

 I've been successfully using IAX in a large scale environment with no
 problems... yet. Can you shed some light on the reasoning behind your
 obvious dislike of IAX2? It is supposed to be the 'killer' of SIP from a
 usability standpoint (NAT traversal is quick to my mind...). BUT, is it 
 just
 not robust enough in your experience? Are there inherent problems with the
 protocol itself? Is this changing now that IAX2 has it's own RFC? Is it the
 implementation within Asterisk that is the problem? I'm very interested to
 to know where your disdain comes from. :-)

 Thanks Steve!

 --Tim

 
>>> First define large scale.  It certainly means different things to
>>> different people.
>>>
>>> Second, It comes from huge amounts of audio problems over many, many
>>> years, and many, many implementations.
>>>
>>> I actually don't have a disdain for it, it has made me a good deal of
>>> money by fixing ITSPs/carrier's audio issues by switching them to SIP
>>> and still does so I have a fondness for it.  Keep up the sub par
>>> protocol, it helps with the balance sheet!
>>>
>>> Third, it will never kill SIP.
>>>
>>> First of all, Digium owns the name and we have seen what they are
>>> willing to do to attack people for trademark or copyright infringement
>>> (think about the Google Adwords debacle and the the Open letter to
>>> Digium drafted by Trixter that I am not sure was ever fully addressed
>>> by Digium.)
>>>
>>> It would have to be renamed or something.  I think the same thing of
>>> DAHDI.  They want control over the the names Inter Asterisk Exchange
>>> and Digium (whatever the heck the rest of it means.)
>>>
>>> Second, SIP is the industry standard.  Only a couple of goofy phones
>>> do IAX2 as far as I know, some crappy handsets I wouldn't even bother
>>> testing if offered as a free demo unit.  SNOM might now, I am not sure
>>> but I think I read interest in it or it was actually accomplished.
>>> SNOM is OK but I was never a big fan.
>>>
>>> When I see it on a Polycom, Cisco, NEC, 3Com, or any other major
>>> vendor's phones or platforms, then I may rethink my ideas.
>>>
>>> If 3Com and Digium are partnered up now, how come the NBX for V3000
>>> doesn't support IAX2?  They do have SIP.
>>>
>>> Second, there are work arounds for just about every downfall of SIP,
>>> like NAT traversal and the like.
>>>
>>> Third, ALL REAL TIME VOICE traffic is on a single port.  There is a
>>> big issue there, I won't elaborate, but just think about it.
>>>
>>> SIP is here to stay until some other protocol comes about, but
>>> certainly not IAX2.  It will be along the evolution of H323 to SIP to
>>> X., but not IAX,lol.
>>>
>>> Do you realize that most providers are dropping IAX2 support, even
>>> IAX.cc recommends SIP, gotta wonder why?
>>>
>>> Maybe it is all good now, but I won't bank my reputation on it.  I use
>>> what I know works well, period.
>>>
>>> Even unnamed Digium Employees have poo pooed IAX2, albeit a year or two
>>> ago.
>>>
>>> It looks good on paper, didn't perform well historically, and now just
>>> like anything that I have lost trust in, it has to earn my trust back
>>> and that is not easy.
>>>
>>> --
>>>   
>> Obviously Steve and I don't agree about this.
>>
>> There are places where IAX can go that SIP just can't.
>>
>> When Steve says just use SIP, what he is actually recommending is
>> to use SIP/STUN/SDP/RTP/IPSEC to get the same result.
>> (at a 50% bandwidth overhead)
>>
>> i.e. replace a single 100 page RFC with something like 100 RFCs :-)
>>
>> In a big organization where you control the network infrastructure, that is
>> an entirely viable solution, but when you want to get calls through a messy
>> network without having to fill out an infinite number of change requests to
>> the firewall team you should consider IAX.
>>
>> The mess that SIP makes is reflected in the number of bugs and the code
>> size.
>> I'm currently working with a SIP stack that is about 10x the size of the
>> comparable IAX
>> codebase, which matters in some environments.
>>
>> As to the 'everything over a single port' issue, this is no longer such a
>> big deal.
>> (And it is exactly this feature which provides IAX's firewall penetration)
>>
>> Most modern Linuxes support multiple threads reading datagrams from a single
>> datagram socket. The current IAX implementation in Asterisk doesn't support
>> it,
>> but that's an implementation issue, not the protocol itself.
>>
>> Also IAX now supports redirecting the media - which could be used to send
>> it to a se

Re: [asterisk-users] Answering the nTh call ...

[Steve Totaro]

On Tue, Jul 7, 2009 at 9:05 AM, J. Oquendo wrote:
>
> Curious to know if anyone's created something similar to the following,
> if so and you'd care to share an AGI or dialplan, much appreciated.
>
> I will be eventually write a script to answer the nTH call. (if I can't
> find it (why reinvent wheels).
>
> Looking to do some testing sending anywhere between 50-200 calls to a
> machine. I'd like a Snom/Polycom/whatever to pick up after the nTh call
> where nTh is whatever I set it to.
>
> exten => _X.,1,{at_N_amount_of_rings}
> exten => _X.,2,Dial(SIP/${ext...@somwhere,45)
>
> So as a script it would be something like: if call != 25th ; then go
> elsewhere ; fi (make sense?) Where every 25th call or so would go
> through, the others would go wherever, not important.
>
> --
>
> =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
> J. Oquendo
> SGFA, SGFE, C|EH, CNDA, CHFI, OSCP
>
> "It takes 20 years to build a reputation and five minutes to
> ruin it. If you think about that, you'll do things
> differently." - Warren Buffett
>
> 227C 5D35 7DCB 0893 95AA  4771 1DCE 1FD1 5CCD 6B5E
> http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x5CCD6B5E
>
>

To be clear you have defined N two different ways.

1.  Amount of rings
2.  A certain specific in a string of calls.

Which is it?

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

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[asterisk-users] Answering the nTh call ...

[J. Oquendo]

Curious to know if anyone's created something similar to the following,
if so and you'd care to share an AGI or dialplan, much appreciated.

I will be eventually write a script to answer the nTH call. (if I can't
find it (why reinvent wheels).

Looking to do some testing sending anywhere between 50-200 calls to a
machine. I'd like a Snom/Polycom/whatever to pick up after the nTh call
where nTh is whatever I set it to.

exten => _X.,1,{at_N_amount_of_rings}
exten => _X.,2,Dial(SIP/${ext...@somwhere,45)

So as a script it would be something like: if call != 25th ; then go
elsewhere ; fi (make sense?) Where every 25th call or so would go
through, the others would go wherever, not important.

-- 

=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
J. Oquendo
SGFA, SGFE, C|EH, CNDA, CHFI, OSCP

"It takes 20 years to build a reputation and five minutes to
ruin it. If you think about that, you'll do things
differently." - Warren Buffett

227C 5D35 7DCB 0893 95AA  4771 1DCE 1FD1 5CCD 6B5E
http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x5CCD6B5E


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Re: [asterisk-users] Echo and static on PRI with errors

[Steve Totaro]

On Thu, Jul 2, 2009 at 11:15 AM, Tom O'Connor wrote:
>
>
> On Wed, Jul 1, 2009 at 6:58 PM, Steve Totaro 
> wrote:
>>
>> On Wed, Jul 1, 2009 at 6:35 AM, Tom O'Connor  wrote:
>> >
>> >
>> > On Wed, Jul 1, 2009 at 11:08 AM, Steve Totaro
>> >  wrote:
>> >>
>> >>
>> >> On Wed, Jul 1, 2009 at 5:58 AM, Tom O'Connor  wrote:
>> >>>
>> >>>
>> >>> On Wed, Jul 1, 2009 at 10:49 AM, Marco Signorini
>> >>>  wrote:
>> 
>>  Tom O'Connor wrote:
>> 
>>  On Wed, Jul 1, 2009 at 7:37 AM, Francesco Peeters
>>   wrote:
>> >
>> > John F. Ervin wrote:
>> > > What do you do if you find things sharing interrupts (IRQ 11) in
>> > > my
>> > > case with my X100P card.  I believe there is some sort of internal
>> > > audio card in my cheap slow PC.
>> > >
>> > Check the BIOS whether you can:
>> > Change the IRQ assignments
>> > Disable the extra hardware using the same IRQ
>> >
>> > Or otherwise try changing the slot it is in... I had very good
>> > results
>> > in the past swapping card around
>> >
>> > Good luck!
>> >
>> 
>>  I did a bit of investigation WRT the IRQ settings on this box.
>> 
>>  00:02.1 USB Controller: nVidia Corporation CK804 USB Controller (rev
>>  a3) (prog-if 20)
>>      Subsystem: Hewlett-Packard Company Device 3207
>>      Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 11
>>  --
>>  01:05.0 VGA compatible controller: nVidia Corporation NV11 [GeForce2
>>  MX/MX 400] (rev b2)
>>      Subsystem: Hewlett-Packard Company Device 3207
>>      Flags: bus master, 66MHz, medium devsel, latency 64, IRQ 11
>>  --
>>  02:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5721
>>  Gigabit Ethernet PCI Express (rev 11)
>>      Subsystem: Hewlett-Packard Company Device 3209
>>      Flags: bus master, fast devsel, latency 0, IRQ 11
>>  --
>>  81:01.0 Network controller: Tiger Jet Network Inc. Tiger3XX
>>  Modem/ISDN interface
>>      Subsystem: Device 79fe:0001
>>      Flags: bus master, medium devsel, latency 64, IRQ 11
>> 
>>  So basically there's 2 network cards and a USB controller sharing IRQ
>>  11 with the Openvox card.
>> 
>>  I wasn't able to find any settings in the bios to manually configure
>>  IRQ assignments :(
>> 
>>  Could someone tell me how to set which IRQ the ISDN card picks up?
>> 
>>  --
>>  Tom O'Connor
>> 
>>  http://www.twinhelix.org
>>  t...@twinhelix.org
>> 
>>  Hi,
>>  Unfortunately is not always possible and it depends on how the
>>  mainboard was realized. For what I can understand a lot of producers 
>>  decide
>>  to route only a subset of physical IRQ lines to the PCI slots (I think 
>>  is
>>  something related to cost reduction) and to share it with other onboard
>>  peripherals.
>>  This lets impossible to change the IRQ assignment for expansion
>>  cards.
>> 
>>  This is not always true and sometimes swapping add-on cards solves
>>  the problem.
>> 
>>  We had better results with cards based on new Digium technology or
>>  with Sangoma cards.
>> 
>> >>> There is almost no room for manouvering in the HP bios.  There's no
>> >>> ability to disable stuff like parallel ports, or anything else really.
>> >>>
>> >>> I don't think i'd buy digium hardware again.  I'm already considering
>> >>> RMAing these cards and getting Sangoma ones.
>> >>>
>> >>>
>> >>> --
>> >>> Tom O'Connor
>> >>>
>> >>> http://www.twinhelix.org
>> >>> t...@twinhelix.org
>> >>>
>> >>
>> >> That is one option.  The new line Digium cards are on par with Sangoma
>> >> as far as IRQ issues.
>> >>
>> >> I really like Sangoma's lifetime warranty though.  I don't think Digium
>> >> has countered that bold move.
>> >>
>> >> I would try the RMA and if that doesn't work, you can always pickup a
>> >> decent last year's model server at
>> >> http://www.surpluscomputers.com/featured-hardware/cg-69/servers.html
>> >>
>> >> For a basic asterisk server or PBX with nothing special going on, any
>> >> of these servers are more than enough, even overkill.
>> >>
>> >> No affiliation, I have to say the shipping is high and they are slow to
>> >> ship but the prices are great, never had an issue with any of their boxen
>> >> (dozens, knock on wood)
>> >
>> > I don't really know what you mean about the new line Digium cards..
>> > which models are in this "new line"?
>> >
>> > the server i'm using is hardly new, it's one of the older DL145s; so i
>> > don't think this would help much!
>> >
>> > I've tried swapping the card in the slots.  no help :(
>> >
>> >
>> >
>> >
>> >
>> > --
>> > Tom O'Connor
>> >
>> > http://www.twinhelix.org
>> > t...@twinhelix.org
>> >
>>
>> Well I guess if I were you, I would stop posting "woe is me" to the
>> list and call Digium.
>>
>> They do have support people just waiting for your call, you know?
>>

Re: [asterisk-users] Some IAX calls do not disconnect.

[Steve Totaro]

On Tue, Jul 7, 2009 at 7:43 AM, Tim Panton wrote:
>
> On 7 Jul 2009, at 05:05, Steve Totaro wrote:
>
>> On Mon, Jul 6, 2009 at 9:41 PM, Tim Nelson wrote:
>>>
>>> - "Steve Totaro"  wrote:

 Just use SIP and solve all your problems.
>>>
>>> I seem to be noticing a common element to your posts about IAX. :-)
>>>
>>> I've been successfully using IAX in a large scale environment with no
>>> problems... yet. Can you shed some light on the reasoning behind your
>>> obvious dislike of IAX2? It is supposed to be the 'killer' of SIP from a
>>> usability standpoint (NAT traversal is quick to my mind...). BUT, is it just
>>> not robust enough in your experience? Are there inherent problems with the
>>> protocol itself? Is this changing now that IAX2 has it's own RFC? Is it the
>>> implementation within Asterisk that is the problem? I'm very interested to
>>> to know where your disdain comes from. :-)
>>>
>>> Thanks Steve!
>>>
>>> --Tim
>>>
>>
>> First define large scale.  It certainly means different things to
>> different people.
>>
>> Second, It comes from huge amounts of audio problems over many, many
>> years, and many, many implementations.
>>
>> I actually don't have a disdain for it, it has made me a good deal of
>> money by fixing ITSPs/carrier's audio issues by switching them to SIP
>> and still does so I have a fondness for it.  Keep up the sub par
>> protocol, it helps with the balance sheet!
>>
>> Third, it will never kill SIP.
>>
>> First of all, Digium owns the name and we have seen what they are
>> willing to do to attack people for trademark or copyright infringement
>> (think about the Google Adwords debacle and the the Open letter to
>> Digium drafted by Trixter that I am not sure was ever fully addressed
>> by Digium.)
>>
>> It would have to be renamed or something.  I think the same thing of
>> DAHDI.  They want control over the the names Inter Asterisk Exchange
>> and Digium (whatever the heck the rest of it means.)
>>
>> Second, SIP is the industry standard.  Only a couple of goofy phones
>> do IAX2 as far as I know, some crappy handsets I wouldn't even bother
>> testing if offered as a free demo unit.  SNOM might now, I am not sure
>> but I think I read interest in it or it was actually accomplished.
>> SNOM is OK but I was never a big fan.
>>
>> When I see it on a Polycom, Cisco, NEC, 3Com, or any other major
>> vendor's phones or platforms, then I may rethink my ideas.
>>
>> If 3Com and Digium are partnered up now, how come the NBX for V3000
>> doesn't support IAX2?  They do have SIP.
>>
>> Second, there are work arounds for just about every downfall of SIP,
>> like NAT traversal and the like.
>>
>> Third, ALL REAL TIME VOICE traffic is on a single port.  There is a
>> big issue there, I won't elaborate, but just think about it.
>>
>> SIP is here to stay until some other protocol comes about, but
>> certainly not IAX2.  It will be along the evolution of H323 to SIP to
>> X., but not IAX,lol.
>>
>> Do you realize that most providers are dropping IAX2 support, even
>> IAX.cc recommends SIP, gotta wonder why?
>>
>> Maybe it is all good now, but I won't bank my reputation on it.  I use
>> what I know works well, period.
>>
>> Even unnamed Digium Employees have poo pooed IAX2, albeit a year or two
>> ago.
>>
>> It looks good on paper, didn't perform well historically, and now just
>> like anything that I have lost trust in, it has to earn my trust back
>> and that is not easy.
>>
>> --
>
> Obviously Steve and I don't agree about this.
>
> There are places where IAX can go that SIP just can't.
>
> When Steve says just use SIP, what he is actually recommending is
> to use SIP/STUN/SDP/RTP/IPSEC to get the same result.
> (at a 50% bandwidth overhead)
>
> i.e. replace a single 100 page RFC with something like 100 RFCs :-)
>
> In a big organization where you control the network infrastructure, that is
> an entirely viable solution, but when you want to get calls through a messy
> network without having to fill out an infinite number of change requests to
> the firewall team you should consider IAX.
>
> The mess that SIP makes is reflected in the number of bugs and the code
> size.
> I'm currently working with a SIP stack that is about 10x the size of the
> comparable IAX
> codebase, which matters in some environments.
>
> As to the 'everything over a single port' issue, this is no longer such a
> big deal.
> (And it is exactly this feature which provides IAX's firewall penetration)
>
> Most modern Linuxes support multiple threads reading datagrams from a single
> datagram socket. The current IAX implementation in Asterisk doesn't support
> it,
> but that's an implementation issue, not the protocol itself.
>
> Also IAX now supports redirecting the media - which could be used to send
> it to a separate port on the same box.
>
>
> Various Digium employees have also badmouthed SIP (I think we all have
> after a bad day at the SDP coalface), so you can't take such remarks too
> seriously.
>
> I o

Re: [asterisk-users] Some IAX calls do not disconnect.

[Tim Panton]

On 7 Jul 2009, at 05:05, Steve Totaro wrote:

On Mon, Jul 6, 2009 at 9:41 PM, Tim Nelson  
wrote:

- "Steve Totaro"  wrote:

Just use SIP and solve all your problems.


I seem to be noticing a common element to your posts about IAX. :-)

I've been successfully using IAX in a large scale environment with  
no problems... yet. Can you shed some light on the reasoning behind  
your obvious dislike of IAX2? It is supposed to be the 'killer' of  
SIP from a usability standpoint (NAT traversal is quick to my  
mind...). BUT, is it just not robust enough in your experience? Are  
there inherent problems with the protocol itself? Is this changing  
now that IAX2 has it's own RFC? Is it the implementation within  
Asterisk that is the problem? I'm very interested to to know where  
your disdain comes from. :-)


Thanks Steve!

--Tim



First define large scale.  It certainly means different things to
different people.

Second, It comes from huge amounts of audio problems over many, many
years, and many, many implementations.

I actually don't have a disdain for it, it has made me a good deal of
money by fixing ITSPs/carrier's audio issues by switching them to SIP
and still does so I have a fondness for it.  Keep up the sub par
protocol, it helps with the balance sheet!

Third, it will never kill SIP.

First of all, Digium owns the name and we have seen what they are
willing to do to attack people for trademark or copyright infringement
(think about the Google Adwords debacle and the the Open letter to
Digium drafted by Trixter that I am not sure was ever fully addressed
by Digium.)

It would have to be renamed or something.  I think the same thing of
DAHDI.  They want control over the the names Inter Asterisk Exchange
and Digium (whatever the heck the rest of it means.)

Second, SIP is the industry standard.  Only a couple of goofy phones
do IAX2 as far as I know, some crappy handsets I wouldn't even bother
testing if offered as a free demo unit.  SNOM might now, I am not sure
but I think I read interest in it or it was actually accomplished.
SNOM is OK but I was never a big fan.

When I see it on a Polycom, Cisco, NEC, 3Com, or any other major
vendor's phones or platforms, then I may rethink my ideas.

If 3Com and Digium are partnered up now, how come the NBX for V3000
doesn't support IAX2?  They do have SIP.

Second, there are work arounds for just about every downfall of SIP,
like NAT traversal and the like.

Third, ALL REAL TIME VOICE traffic is on a single port.  There is a
big issue there, I won't elaborate, but just think about it.

SIP is here to stay until some other protocol comes about, but
certainly not IAX2.  It will be along the evolution of H323 to SIP to
X., but not IAX,lol.

Do you realize that most providers are dropping IAX2 support, even
IAX.cc recommends SIP, gotta wonder why?

Maybe it is all good now, but I won't bank my reputation on it.  I use
what I know works well, period.

Even unnamed Digium Employees have poo pooed IAX2, albeit a year or  
two ago.


It looks good on paper, didn't perform well historically, and now just
like anything that I have lost trust in, it has to earn my trust back
and that is not easy.

--


Obviously Steve and I don't agree about this.

There are places where IAX can go that SIP just can't.

When Steve says just use SIP, what he is actually recommending is
to use SIP/STUN/SDP/RTP/IPSEC to get the same result.
(at a 50% bandwidth overhead)

i.e. replace a single 100 page RFC with something like 100 RFCs :-)

In a big organization where you control the network infrastructure,  
that is
an entirely viable solution, but when you want to get calls through a  
messy
network without having to fill out an infinite number of change  
requests to

the firewall team you should consider IAX.

The mess that SIP makes is reflected in the number of bugs and the  
code size.
I'm currently working with a SIP stack that is about 10x the size of  
the comparable IAX

codebase, which matters in some environments.

As to the 'everything over a single port' issue, this is no longer  
such a big deal.
(And it is exactly this feature which provides IAX's firewall  
penetration)


Most modern Linuxes support multiple threads reading datagrams from a  
single
datagram socket. The current IAX implementation in Asterisk doesn't  
support it,

but that's an implementation issue, not the protocol itself.

Also IAX now supports redirecting the media - which could be used to  
send

it to a separate port on the same box.


Various Digium employees have also badmouthed SIP (I think we all have
after a bad day at the SDP coalface), so you can't take such remarks  
too seriously.


I overheard a senior Cisco employee saying "So you were right all  
along about IAX "

to a very senior Digium employee, which also proves nothing much :-)

Competition is a good thing - even amongst protocols.

T.

Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk





smime.p7s
Description: 

Re: [asterisk-users] Asterisk & Jabber : WARNING: res_jabber.c aji_recv_loop: JABBER: socket read error

[jonas kellens]

[Jul  7 11:54:38] 
JABBER: asterisk INCOMING: asteriskasterisk90a141d72ee469dc30bc95c661d0c299ead11061

from=192.168.2.5
to=openfire.jocan.local

The 'from'- and 'to'-tag are the same as 192.168.2.5 is the IP-address
of openfire.jocan.local... Is this normal ?

Jonas.


On Tue, 2009-07-07 at 10:19 +0200, Philippe Sultan wrote:

> Or you can disable the digest-md5 authentication mechanism on
> OpenFire. I remember an old related bug :
> https://issues.asterisk.org/view.php?id=11644
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Re: [asterisk-users] Source for OpenVox cards?

[Tom O'Connor]

On Mon, Jul 6, 2009 at 5:03 PM, Tony Mountifield
wrote:

> In article ,
> Timothy Legge  wrote:
> >
> > I am looking for a source for an OpenVox card.  Has anyone purchased
> through
> > http://www.voiplink.com or do you normally use another vendor or
> OpenVox.cn
> > directly?
> >
> > thanks
> >
> > Tim
>
> I have used voipon.co.uk, but I don't know whether that's useful to you,
> as you didn't say which country you are in.
>
> Cheers
> Tony


Yes, I've used voipon.co.uk also, their customer service is very efficient.
They'd dispatched the cards within 2 hours of me ordering them.


-- 
Tom O'Connor

http://www.twinhelix.org
t...@twinhelix.org
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Re: [asterisk-users] Echo and static on PRI with errors

[Tom O'Connor]

On Thu, Jul 2, 2009 at 4:52 PM, Tom O'Connor  wrote:

>
>
> On Thu, Jul 2, 2009 at 4:41 PM, Tzafrir Cohen wrote:
>
>> On Thu, Jul 02, 2009 at 04:15:13PM +0100, Tom O'Connor wrote:
>>
>> > I have tried all suggestions given.  It just happens that none of them
>> have
>> > been much use.  I'm very constrained by time on this project, less than
>> 11
>> > days before i leave the company, so i'd like to have it in a workable
>> > state.  Buying a better server, although it would probably work, would
>> > inevitably cost more money than they're willing to spend.
>> > You might notice that the OpenVox cards they bought are the cheapest on
>> the
>> > market? Coincidence.. no.
>>
>> Cheap? yes. Cheapest? Almost. Have you contacted their support?
>>
>>
>> (I'm completely unafiliated with them etc. etc.)
>
> I tried, I got passed around from person to person until i'd heard the
> script at least 4 times, so I gave up.  Might try again tomorrow.
> I'm currently testing the same card in a different server, with a Tyan
> Intel chipset board, as opposed to a HP / AMD chipset.   I'm somewhat more
> hopeful about this.  I was told that newer servers have some proprietary PCI
> bus chips, which are optimised to deal with RAID cards and suchlike, which
> can be problematic when handing IRQ-heavy cards such as these.
>
> I think this is actually interesting, because the current Asterisk server
> is actually a Dell Optiplex desktop with a Pentium 150 in it.  Can't get
> much more basic, but it works pretty well.  (the only reason we're
> attempting an upgrade, is because the old asterisk 1.0 can't support
> features that the management want to implement, and the physical box is
> taking up too much space in the already cramped server room!)
>
> *Resolved*
I ditched the HP server in favour of the Tyan mobo'd server.  Everything
works perfectly.

Thanks to anyone who offered suggestions, they were most useful!

Tom

-- 
Tom O'Connor

http://www.twinhelix.org
t...@twinhelix.org
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Re: [asterisk-users] chan_mobile help.

[Thomas Kenyon]

Razza wrote:
> I'm running centos, so tried a yum upgrade but nothing was marked for 
> upgrade. I've reinstalled bluez-libs.i386 0:3.7-1.1.
> I've tried a different dongle, but still get the same message.
> 
I tried setting up chan_mobile again about a week ago. (admittedly last 
time I'd tried it, it had a different name and was using callweaver).

I had exactly the same problems as the first time, No audio (although on 
one attempt I had 1 way audio, but strangely the phone and the deskphone 
and other mobile appeared to be conferenced together), answering desk 
phone didn't answer call on mobile, hanging up didn't hang up etc.

Although I think I'm using bluez-4.40, the USB dongle is also a CSR one 
like the OP, asterisk 1.6.1.1, with a nokia e61.

Oh a piece of advice, when the computer is automatically connecting to 
the phone, you can't find it by calling it :-)

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Re: [asterisk-users] Asterisk & Jabber : WARNING: res_jabber.c aji_recv_loop: JABBER: socket read error

[Philippe Sultan]

Or you can disable the digest-md5 authentication mechanism on
OpenFire. I remember an old related bug :
https://issues.asterisk.org/view.php?id=11644

On Mon, Jul 6, 2009 at 8:55 PM, Julian Lyndon-Smith wrote:
> usetls=no
>
> Julian
>
> jonas kellens wrote:
>> On Mon, 2009-07-06 at 16:18 +0100, Julian Lyndon-Smith wrote:
>>> I can assure you that it works, and that it works well. We use it ;)
>>
>> My jabber.conf :
>>
>> [general]
>> debug=yes                               ;;Turn on debugging by default.
>> autoprune=no                            ;;Auto remove users from buddy list.
>> autoregister=yes                        ;;Auto register users from buddy
>> list.
>>
>> [asterisk]                              ;;label
>> type=client                             ;;Client or Component connection
>> serverhost=192.168.2.5                  ;;Route to server for example
>> talk.google.com
>> username=aster...@192.168.2.5           ;;Username with optional roster.
>> secret=XX                     ;;Password
>> port=5222                               ;;Port to use defaults to 5222
>> usetls=yes                              ;;Use tls or not
>> usesasl=yes                             ;;Use sasl or not
>> statusmessage="I am Asterisk"           ;;Have custom status message for
>> Asterisk.
>> ;timeout=100                            ;;Timeout on the message stack.
>>
>> Then I get the following :
>>
>> [Jul  6 20:07:57]
>> JABBER: asterisk INCOMING: > encoding='UTF-8'?>> xmlns:stream="http://etherx.jabber.org/streams"; xmlns="jabber:client"
>> from="openfire.jocan.local" id="56ff9859" xml:lang="en"
>> version="1.0">> xmlns="urn:ietf:params:xml:ns:xmpp-sasl">DIGEST-MD5PLAINANONYMOUSCRAM-MD5> xmlns="http://jabber.org/features/compress";>zlib> xmlns="http://jabber.org/features/iq-auth"/>> xmlns="http://jabber.org/features/iq-register"/>
>> [Jul  6 20:07:57]
>> JABBER: asterisk OUTGOING: > xmlns='urn:ietf:params:xml:ns:xmpp-sasl' mechanism='DIGEST-MD5'/>
>> [Jul  6 20:07:57]
>> JABBER: asterisk INCOMING: > xmlns="urn:ietf:params:xml:ns:xmpp-sasl">cmVhbG09Im9wZW5maXJlLmpvY2FuLmxvY2FsIixub25jZT0iSngyRVZCRmlDNlI4K1hlMU5rbm9PUUNWT1VEN1pGMEpXcnRydUxjdiIscW9wPSJhdXRoIixjaGFyc2V0PXV0Zi04LGFsZ29yaXRobT1tZDUtc2Vzcw==
>> [Jul  6 20:07:57]
>> JABBER: asterisk OUTGOING: > xmlns='urn:ietf:params:xml:ns:xmpp-sasl'>dXNlcm5hbWU9ImFzdGVyaXNrIixyZWFsbT0ib3BlbmZpcmUuam9jYW4ubG9jYWwiLG5vbmNlPSJKeDJFVkJGaUM2UjgrWGUxTmtub09RQ1ZPVUQ3WkYwSldydHJ1TGN2Iixjbm9uY2U9IjQzZTVmYjFkNjZiMTU2OGI1MDFjNzk0ZDQ0MzMyYzFiIixuYz0wMDAwMDAwMSxxb3A9YXV0aCxkaWdlc3QtdXJpPSJ4bXBwLzE5Mi4xNjguMi41IixyZXNwb25zZT1kNGUxYzQ0ZDM0OGNjNWJkN2E2MzJiNzdmZjRjZTQ0OCxjaGFyc2V0PXV0Zi04
>> [Jul  6 20:07:57]
>> JABBER: asterisk INCOMING: > xmlns="urn:ietf:params:xml:ns:xmpp-sasl">
>> [Jul  6 20:07:57] ERROR[24565]: res_jabber.c:606 aji_act_hook: JABBER:
>> encryption failure. possible bad password.
>>
>> I am 100% sure I have the correct password !
>>
>> I even took a very simple password without any special characters...
>>
>> Can you advise ??
>>
>> Jonas.
>
>
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-- 
Philippe Sultan

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Re: [asterisk-users] SIP registry fails during night

[jonas kellens]

Yet another night has passed...

This morning :

Verbosity is at least 25
asterisk*CLI> sip show peers
Name/username  HostDyn Nat ACL Port Status
Realtime  
twinkle-candy/twinkle-can  (Unspecified)D  0
UNKNOWN  
twinkle-jonas/twinkle-jon  (Unspecified)D  0
UNKNOWN  
grandstream/grandstream192.168.1.13 D  5060 OK (27
ms)   
3starsnet/09277907785.119.188.3 N  5060 OK (17
ms)   
4 sip peers [Monitored: 2 online, 2 offline Unmonitored: 0 online, 0
offline]
asterisk*CLI> sip show registry 
HostUsername   Refresh State
Reg.Time 
85.119.188.3:5060   092779077  105 Failed  Mon,
06 Jul 2009 23:37:05

Then a 'sip reload' :

asterisk*CLI> sip reload
[Jul  7 09:28:15]  Reloading SIP
[Jul  7 09:28:15]   == Parsing '/etc/asterisk/sip.conf': [Jul  7
09:28:15] Found
[Jul  7 09:28:15]   == Parsing '/etc/asterisk/users.conf': [Jul  7
09:28:15] Found
[Jul  7 09:28:15]   == Parsing '/etc/asterisk/sip_notify.conf': [Jul  7
09:28:15] Found
asterisk*CLI> sip show peers
Name/username  HostDyn Nat ACL Port Status
Realtime  
twinkle-candy/twinkle-can  (Unspecified)D  0
UNKNOWN  
twinkle-jonas/twinkle-jon  (Unspecified)D  0
UNKNOWN  
grandstream/grandstream192.168.1.13 D  5060 OK (9
ms)
3starsnet/09277907785.119.188.3 N  5060 OK (18
ms)   
4 sip peers [Monitored: 2 online, 2 offline Unmonitored: 0 online, 0
offline]
asterisk*CLI> sip show registry
HostUsername   Refresh State
Reg.Time 
85.119.188.3:5060   092779077  105 Registered   Tue, 07 Jul
2009 09:28:16


So a 'sip reload' got me registered again... Did not need to restart
Asterisk...

/var/log/asterisk/messages :

[Jul  6 23:37:46] NOTICE[24975] chan_sip.c: Peer 'grandstream' is now
UNREACHABLE!  Last qualify: 33
[Jul  6 23:37:46] WARNING[24975] res_config_mysql.c: MySQL RealTime:
Failed to query database. Check debug for more info.
[Jul  6 23:38:03] NOTICE[24975] chan_sip.c: Peer '3starsnet' is now
UNREACHABLE!  Last qualify: 18
[Jul  6 23:38:03] WARNING[24975] res_config_mysql.c: MySQL RealTime:
Failed to query database. Check debug for more info.
[Jul  6 23:39:10] WARNING[24975] chan_sip.c: Maximum retries exceeded on
transmission 399240fc5f0fe8fb0e548a152c2c3...@127.0.0.1 for seqno 248
(Critical Req
uest) -- See doc/sip-retransmit.txt.
[Jul  6 23:39:50] NOTICE[24975] chan_sip.c:-- Registration for
'092779...@85.119.188.3' timed out, trying again (Attempt #1)
[Jul  6 23:40:30] WARNING[24975] chan_sip.c: Maximum retries exceeded on
transmission 399240fc5f0fe8fb0e548a152c2c3...@127.0.0.1 for seqno 249
(Critical Req
uest) -- See doc/sip-retransmit.txt.
[Jul  6 23:41:10] NOTICE[24975] chan_sip.c:-- Registration for
'092779...@85.119.188.3' timed out, trying again (Attempt #2)
[Jul  6 23:41:50] WARNING[24975] chan_sip.c: Maximum retries exceeded on
transmission 399240fc5f0fe8fb0e548a152c2c3...@127.0.0.1 for seqno 250
(Critical Req
uest) -- See doc/sip-retransmit.txt.
[Jul  6 23:42:30] NOTICE[24975] chan_sip.c:-- Registration for
'092779...@85.119.188.3' timed out, trying again (Attempt #3)
...
[Jul  7 00:55:51] NOTICE[24975] chan_sip.c:-- Registration for
'092779...@85.119.188.3' timed out, trying again (Attempt #58)
[Jul  7 00:56:31] WARNING[24975] chan_sip.c: Maximum retries exceeded on
transmission 399240fc5f0fe8fb0e548a152c2c3...@127.0.0.1 for seqno 306
(Critical Req
uest) -- See doc/sip-retransmit.txt.
[Jul  7 00:57:11] NOTICE[24975] chan_sip.c:-- Registration for
'092779...@85.119.188.3' timed out, trying again (Attempt #59)
[Jul  7 00:57:51] WARNING[24975] chan_sip.c: Maximum retries exceeded on
transmission 399240fc5f0fe8fb0e548a152c2c3...@127.0.0.1 for seqno 307
(Critical Req
uest) -- See doc/sip-retransmit.txt.
[Jul  7 00:58:31] NOTICE[24975] chan_sip.c:-- Registration for
'092779...@85.119.188.3' timed out, trying again (Attempt #60)
[Jul  7 00:59:11] WARNING[24975] chan_sip.c: Maximum retries exceeded on
transmission 399240fc5f0fe8fb0e548a152c2c3...@127.0.0.1 for seqno 308
(Critical Req
uest) -- See doc/sip-retransmit.txt.
[Jul  7 00:59:51] NOTICE[24975] chan_sip.c:-- Registration for
'092779...@85.119.188.3' timed out, trying again (Attempt #61)
[Jul  7 00:59:51] NOTICE[24975] chan_sip.c:-- Giving up forever
trying to register '092779...@85.119.188.3'

Trying to register goes on for an hour, every minute...
I don't know why @ 23:39 registration suddenly fails...

Philipp, in my register-statement I use the IP-address of my
SIP-provider, not the hostname.tld.

Jonas.


On Mon, 2009-07-06 at 15:14 +0200, Philipp von Klitzing wrote:

> Hi!
> 
> > Every morning I check my SIP registry to the SIP-provider. And I must
> > conclude that duri