[asterisk-users] You are not the next caller

[Sean Cook]

Ok... I have heard this on Digium's PBX in the past, but can't seem to find
it anymore. 

There was an IVR that you could dial into and Allison had recorded one of
the funniest messages I have ever heard... "you are not the next caller,
hang up not, spend time with your children..."  it was hilareous... 

Does anyone have that recording?  or know where I can find it?

Sean
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[asterisk-users] Modems dialing over sangoma a104d

[Sean Cook]

I have a sangoma 104d that is our main pbx now( legacy system died ).  I
have replaced every phone in the building and things are going very well.
We have fax working well and calls are routing properly...  All is well...

Except for our support modems... we have support people that dial out with
modems across our PRI's.  These modems are attached to an Adtran 750 with 24
FXS's.  I have disabled echo cancelation on the T1 that is connected to the
Adtran but negotiation is still really rough.  I am bridging across the same
card and it isn't doing very well... has anyone done this with reasonably
successful results?  I am not looking for 56K I am looking for around 9600
to 14.4..

Thanks,

Sean
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Re: [asterisk-users] e&m wink, TE110P, * answers too soon

[Sean Cook]

Steve,

I have the exact same problem with a sangoma a104d so I don't think it is
related to the card.  I am trying to figure that one out as well,
fortunately I only have one DID on that trunk so I used _.* to route
everything...

Sorry this doesn't help other than to let you know that it probably isn't
the card...

Sean

On 09-Aug-2006, Steve Linabery wrote:
> Hi,
> 
> I've been googling all over the place and have read the relevant articles in 
> the Digium knowledge base. I have tried all the suggestions I found in the 
> K.B. Spent some time on the asterisk irc, tweaking some parameters as people 
> thereon thought would be helpful, but to no avail.
> 
> I am trying to set up * on an e&m wink trunk currently attached to an Avaya 
> Merlin Magix system. The provider of the T1 is McLeodUSA; our location is St 
> Paul MN USA. I am in the process of getting more specific timing information 
> from their tech support, but it takes days.
> 
> I can call into the * PBX from my cell phone just fine. I can call between 
> the two grandstream phones I bought for testing just fine.
> 
> Here's the problem. When a call comes into *, * attempts to route it to an 
> extension prematurely. For example, if the DTMF digits coming from upstream 
> are '538', * tries to send the call to extn '53'. I still receive the '8', 
> but too late.
> 
> Here's a snip from /var/log/asterisk/messages where the incoming DID digits 
> are '535':
> Aug  7 22:30:00 DEBUG[31492] chan_zap.c: Monitor doohicky got event 
> Ring/Answered on channel 1
> Aug  7 22:30:00 DEBUG[31478] devicestate.c: Changing state for Zap/1 - state 
> 2 (In use)
> Aug  7 22:30:00 VERBOSE[31493] logger.c: Asterisk Ready.
> -- Starting simple switch on 'Zap/1-1'
> Aug  7 22:30:00 DEBUG[31494] app_queue.c: Device 'Zap/1' changed to state '2' 
> (In use) but we don't care because they're not a member of any queue.
> Aug  7 22:30:01 DEBUG[31493] chan_zap.c: DTMF digit: 5 on Zap/1-1
> Aug  7 22:30:01 DEBUG[31493] chan_zap.c: DTMF digit: 3 on Zap/1-1
> Aug  7 22:30:01 DEBUG[31493] chan_zap.c: Enabled echo cancellation on channel 
> 1
> Aug  7 22:30:01 VERBOSE[31493] logger.c:   == Unknown extension '53' in 
> context 'demo' requested
> Aug  7 22:30:04 DEBUG[31493] channel.c: Set channel Zap/1-1 to write format 
> gsm
> Aug  7 22:30:04 DEBUG[31493] channel.c: Scheduling timer at 160 sample 
> intervals
> Aug  7 22:30:04 VERBOSE[31493] logger.c: -- Playing 'ss-noservice' 
> (language 'en')
> Aug  7 22:30:04 DEBUG[31493] chan_zap.c: DTMF digit: 5 on Zap/1-1
> Aug  7 22:30:07 DEBUG[31493] chan_zap.c: Exception on 20, channel 1
> Aug  7 22:30:07 DEBUG[31493] chan_zap.c: Got event On hook(1) on channel 1 
> (index 0)
> Aug  7 22:30:07 DEBUG[31493] chan_zap.c: disabled echo cancellation on 
> channel 1
> Aug  7 22:30:07 DEBUG[31493] channel.c: Scheduling timer at 0 sample intervals
> Aug  7 22:30:07 DEBUG[31493] channel.c: Hanging up channel 'Zap/1-1'
> Aug  7 22:30:07 DEBUG[31493] chan_zap.c: zt_hangup(Zap/1-1)
> Aug  7 22:30:07 DEBUG[31493] chan_zap.c: Hangup: channel: 1 index = 0, normal 
> = 20, callwait = -1, thirdcall = -1
> Aug  7 22:30:07 DEBUG[31493] chan_zap.c: disabled echo cancellation on 
> channel 1
> Aug  7 22:30:07 DEBUG[31493] chan_zap.c: Set option TDD MODE, value: OFF(0) 
> on Zap/1-1
> Aug  7 22:30:07 DEBUG[31493] chan_zap.c: Updated conferencing on 1, with 0 
> conference users
> Aug  7 22:30:07 VERBOSE[31493] logger.c: -- Hungup 'Zap/1-1'
> Aug  7 22:30:07 DEBUG[31478] devicestate.c: Changing state for Zap/1 - state 
> 0 (Unknown)
> Aug  7 22:30:07 DEBUG[31495] app_queue.c: Device 'Zap/1' changed to state '0' 
> (Unknown) but we don't care because they're not a member of any queue.
> 
> 
> Here are some settings from /etc/asterisk/zapata.conf:
> [trunkgroups]
> [channels]
> wink=300
> rxwink=300
> start=3000
> context=default
> switchtype=national
> toneduration=100
> usecallerid=no
> cidsignalling=dtmf
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> relaxdtmf=no
> rxgain=0.0
> txgain=0.0
> group=1
> callgroup=1
> pickupgroup=1
> immediate=no
> callprogress=no
> switchtype = national
> context = demo
> signalling = em_w
> group = 1
> channel => 1-20
> 
> 
> It has occurred to me that I could just set immediate=yes, read the incoming 
> DTMF digits into a variable, and route to the appropriate extension. That 
> seems more fragile to me since we could someday (when I'm not here) start 
> getting more than 3 digits (caller id, for example). Plus I'd like to make it 
> work the way it's *supposed* to.
> 
> Any help/suggestions are appreciated!
> 
> Cheers,
> -- 
> Steve Linabery
> B94B C3C7 8A27 FF09 3C9D  E992 5A20 2492 D5F5 EE51
> 
> 
> This electronic message transmission contains information from the sender's 
> organization that may be proprietary, confidential and/or pri

[asterisk-users] hints causing hang in reload

[Sean Cook]

I have a system right now that has 32 extensions that I am setting up 
hints for



exten => 4521,hint,SIP/4521
exten => 4522,hint,SIP/4522
exten => 4523,hint,SIP/4523
exten => 4524,hint,SIP/4524
exten => 4525,hint,SIP/4525


The problem that I am running into is when I issue a reload, it hangs 
for about 30-40 seconds before completing the reload.  I have found that 
by taking the hints out of the config it reloads immediately as expected.


Has anyone else encountered this?   Is there a decent explanation?

Sean
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Re: [asterisk-users] Looking for an asterisk guru

[Sean Cook]

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Aaron,

Are you interested ;)

Sean

Aaron Daniel wrote:
> LOL, that's descriptive :)
>
> On Wed, 2006-07-05 at 16:03 -0400, Sean Cook wrote: We are looking
> for an asterisk guru / linux geek for full time employment in the
> South West Virigina area.  Please email me off list for more
> details.
>
> Sean
>>>
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[asterisk-users] Looking for an asterisk guru

[Sean Cook]

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We are looking for an asterisk guru / linux geek for full time
employment in the South West Virigina area.  Please email me off list
for more details.

Sean
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Re: [Asterisk-Users] Auto answer an IAXY how

[Sean Cook]

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It would not be the iaxy... it would be the phone that is attached to
it... there are plenty of phones/answering machines /other FXS
signalling devices that can do auto answer... the iaxy is not capable
of doing that...

Sean

Jerry Geis wrote:
> Can an IAXY be setup to auto answer? If so how?
> I mean any call coming into it automatically connect it to the phone
> and send voice traffic.
>
> Thanks,
>
> Jerry
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Re: [Asterisk-Users] Digium Hardware Reliability

[Sean Cook]

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Philippe Lindheimer wrote:
> I would love to see some feedback on this as well. I've lost exact
> count now, but think I've seen about 5-6 failures on their cards
> TDM400P and TDM2400P cards, mostly FXO but in once case FXS. And I
> don't deal with that many systems, which makes this really
> concerning. I've started a thread on the Asterisk Forum to get more
> feedback on the Sangoma cards as an alternative. I'm finding it hard
> to think this experience is a total fluke - it would be great to
> hear other people's experience though - good or bad.
>

H... I have around 10 TDM400 in the field with out a single
failure.  I also have 6-8 sangomas A200's in the field with no
problems... Sangomas are not twice the price at all...

Sangoma a200 with 2xFXO =  $249.95
Digium TDM402b=  $225.90


The only time the cost really goes up with the sangoma is when you add
the echo cancellor. (1 time cost of $300 roughly) at that point you
can only compare the card to the TDM2400 with echo cancellation and
then it too is an even cost.

Both cards in my experience are very reliable, the sangoma IMHO gives
you a little more flexibility in terms of a smaller system and
experimenting with echo cancellation or no echo cancellation.

> philippe
>
>
> From: "M.Hockings" <[EMAIL PROTECTED]>
> To: asterisk-users@lists.digium.com
> Date: Thu, 29 Jun 2006 21:38:20 -0400
> Subject: [Asterisk-Users] Digium Hardware Reliability
>
> How reliable is Digium hardware in general.? My new TDM400P just
> died.
>
> I am trying to determine if I have a lemon. This a new PC with a
> Digium
> TDM400P in it with a single FXO and single FXS card just stopped
> working
> today. It has been running less than three weeks with the the
> FXS card
> and has the FXO card in it only for about a week. Today the
> power went
> out due to a mis-configuration on my part the UPS shut down
> before the
> machine shut down. Now, I would not think this should be a
> problem but
> the Digium card no longer responds. lspci does not show it
> either so I
> presume it dead
>
> So, at over 2x the cost is Sangoma hardware more sturdy than the
> Digium
> stuff?
>
> Right now we are back using the POTS phones with the nice new
> SPA-922's
> looking like cute paperweights.
>
> Mike (totally UNimpressed with Digium)
>
>
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>
> --
> Do you Yahoo!?
> Everyone is raving about the all-new Yahoo! Mail Beta.
> 
>
>
> --
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Re: [Asterisk-Users] (no subject)

[Sean Cook]

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The only issues you could potentially run into is if all the modules
are FXS and they all needed to ring simultaneously... your power
supply may not be suited to handle to voltage requirements.

Sean

Ninneman, Tj wrote:
> 
>
> Hey everybody,
>
>
>
> Is it alright to run two TDM400s on the same machine?  If it is,
> how would one differentiate between the channels on each card?  So,
> if I?m running strait FXS and my first card is fxsks 1-4, would the
>  second be fxsks 5-8?  Would there be any interrupt problems?
>
>
>
> Any help would be great!
>
>
>
> Thanks!
>
>
>
> Tj
>
>
>
>
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Re: [Asterisk-Users] point to point T hookup?

[Sean Cook]

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It always helps to read the original post... so I apologize.  I think
what you are looking to do is route the calls over the existing data
t1 in which case all you need to do is create an IAX trunk between the
two asterisk servers addressing their internal ip addresses ( such
that the route would be over the data T1).

Then you would want to make sure that you are running QoS on all voip
traffic that goes from A to B accross that link giving it the highest
level of priority.

That should in essence do the job...

Sean
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Re: [Asterisk-Users] point to point T hookup?

[Sean Cook]

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Typically with a data t1 you are running either HDLC or PPP on either
end.  I assume you have a cisco router on either end?   Or are you
planning to plug asterisk with a Digium/Sangoma/Other T1 card?

Personally if it is a data t1 I would use a cisco router then do QoS
on both routers and do everything VoIP on the asterisk side... Then
you have no hardware necessary for your trunks (other than the routers
of course)


Jonathan Miller wrote:
> Your response leads me to further question this setup...
>
> It's a full data T that is not provisioned. Being that I control
> the termination at each end, do I get to specify the encoding?
>
>
> On Wednesday 28 June 2006 10:17, Sean Cook wrote:
>>> What kind of T1?  TDM?  Data?  What type of signaling are you
> planning
>>> to use e&m?  There is a lot of information that that question
>>> is lacking for anyone to advise you ...
>>>
>>> Jonathan Miller wrote:
>>>> I have a true leased line (a T1) between the two sites.
>>>>
>>>> What parts do I configure for Asterisk to utilized the link
>>>> bi-directional?
>>>>
>>>> On Wednesday 28 June 2006 09:09, Andrew Kohlsmith wrote:
>>>>>> On Wednesday 28 June 2006 08:48, Jonathan Miller wrote:
>>>>>>> An alternative is to put a router and switch at each
>>>>>>> end and
>>>> extend a
>>>>
>>>>>>> data network to the other site for SIP traffic. Would
>>>>>>> that result in better quality calls?
>>>>>> If you can ensure that voice traffic has top priority in
>>>>>> all the
>>>> routers
>>>>
>>>>>> between the two sites, there should be no difference in
>>>>>> voice
>>>> quality.  For
>>>>
>>>>>> a true point-to-point system this is trivial to achieve,
>>>>>> and
>>>> maximizes the
>>>>
>>>>>> bang-for-buck ratio of your interoffice connection.
>>>>>>
>>>>>> Obviously having two ADSL connections is not true "point
>>>>>> to
>>>> point" -- you
>>>>
>>>>>> will want a leased line, or a dedicated connection to a
>>>>>> common
>>>> provider who
>>>>
>>>>>> has the prioritization of voice traffic in your SLA.
>>>>>>
>>>>>> You could, in theory, have higher than telco quality
>>>>>> voice calls
>>>> with a
>>>>
>>>>>> VOIP system, as you are no longer restricted to
>>>>>> 8kHz-sampled,
>>>> 16-bit audio.
>>>>
>>>>>> Naturally the phones must support this for this to work.
>>>>>>
>>>>>>> What configuration areas are there to be set and how
>>>>>>> are they
>>>> diffent
>>>>
>>>>>>> than just a standard PRI, which I have working now?
>>>>>> If you put a point-to-point DS1 between sites, it's easy.
>>>>>>
>>>> Asterisk can act
>>>>
>>>>>> as a PRI CPE or CO endpoint.
>>>>>>
>>>>>> -A. ___
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>>>>>> options visit:
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Re: [Asterisk-Users] point to point T hookup?

[Sean Cook]

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What kind of T1?  TDM?  Data?  What type of signaling are you planning
to use e&m?  There is a lot of information that that question is
lacking for anyone to advise you ...

Jonathan Miller wrote:
> I have a true leased line (a T1) between the two sites.
>
> What parts do I configure for Asterisk to utilized the link
> bi-directional?
>
>
>
> On Wednesday 28 June 2006 09:09, Andrew Kohlsmith wrote:
>>> On Wednesday 28 June 2006 08:48, Jonathan Miller wrote:
 An alternative is to put a router and switch at each end and
> extend a
 data network to the other site for SIP traffic. Would that
 result in better quality calls?
>>> If you can ensure that voice traffic has top priority in all
>>> the
> routers
>>> between the two sites, there should be no difference in voice
> quality.  For
>>> a true point-to-point system this is trivial to achieve, and
> maximizes the
>>> bang-for-buck ratio of your interoffice connection.
>>>
>>> Obviously having two ADSL connections is not true "point to
> point" -- you
>>> will want a leased line, or a dedicated connection to a common
> provider who
>>> has the prioritization of voice traffic in your SLA.
>>>
>>> You could, in theory, have higher than telco quality voice
>>> calls
> with a
>>> VOIP system, as you are no longer restricted to 8kHz-sampled,
> 16-bit audio.
>>> Naturally the phones must support this for this to work.
>>>
 What configuration areas are there to be set and how are they

> diffent
 than just a standard PRI, which I have working now?
>>> If you put a point-to-point DS1 between sites, it's easy.
> Asterisk can act
>>> as a PRI CPE or CO endpoint.
>>>
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Re: [Asterisk-Users] FXO for PSTN

[Sean Cook]

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Or a TDM2400 with 4 FXO modules... (4x4=16) :)

Lito Lampitoc wrote:
> oh sorry, 2 TDM400P with 4 FXO modules each :=)
>
> On 6/28/06, *Lito Lampitoc* <[EMAIL PROTECTED]
> > wrote:
>
> or TDM400P with four FXO modules perhaps?
>
>
> On 28 Jun 2006 01:47:05 -, [EMAIL PROTECTED]
> *
> <[EMAIL PROTECTED]
> > wrote:
>
> 1 FXO per PSTN, so you would need 16 FXO ports.  That would be
> accomplished by 4 TDM100P with 4 FXO modules on each.
>
> Undrhil
>
> --- Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com
>  wrote: If I have 16 PSTN
> for my trunklines, how many FXO do I need?
>>
>> Thanks.
>>
>> Lito
>>
>>
>>
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[Asterisk-Users] Dell PowerEdge 1650

[Sean Cook]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
 
Anyone have a 1650 running successfully in production mode with 2-4
PRI's?  I want to make sure I don't have a motherboard compatibility
problem before I buy one of these.  We are going to be using a Digium
TE210P to start off with and probably moving to the TE411P down the
road aways?

thanks,

Sean
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Re: [Asterisk-Users] Polycom Buddies in 1.6.6

[Sean Cook]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
 
Double check to make sure you are actually running 1.6.6.  I have it
working with 14 extensions right now with no problems...

Sean

Douglas Garstang wrote:
> All,
>
> Slightly off topic.
>
> Polycom released their SIP software version 1.6.6 for their phones
> recently. I was under the impression that this release fixed a
> previous limitation where the phones would only watch 7 buddies, ie
> send 7 sip subscriptions to Asterisk. I have configured a phone
> directory to watch 30 or so appearances, and it still seems to only
> be sending 7 subscriptions to Asterisk.
>
> Has anyone else got this to work?
>
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Re: [Asterisk-Users] FAX + Digium + SpanDSP

[Sean Cook]

>> Hi,
>>
>>  
>>
>> Anyone using SpanDSP with Digium TDM o TE cards to receive and email
>> Faxes?
>>
>>  
>>
> No. Nobody ever uses this stuff. I just write it to waste my spare time.
>
> Steve

Finish the coffee BEFORE checking the email...
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Re: [Asterisk-Users] Single T1 card with Echo CancellationtoworkwithDell?

[Sean Cook]

Off course for the price, you could by a four port sangoma with echo cancel

David Waugh wrote:
> Hello
>
> Eicon Networks produce a Single T1 card with hardware echo cancellation.
> http://www.eicon.com/worldwide/products/MediaGateways/diva-server-vt1-pr
> i.htm
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Steve
> Davies
> Sent: 15 June 2006 11:43
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
> Discussion
> Subject: Re: [Asterisk-Users] Single T1 card with Echo
> CancellationtoworkwithDell?
>
> On 6/14/06, Cory Andrews <[EMAIL PROTECTED]> wrote:
>   
>> Sangoma is NOT releasing a single T1 with echo cancellation.
>>
>> 
>
> But AFAIK they ARE releasing (have released?) a dual T1/E1 card with
> hardware EC, which may be sufficient compromise.
>
> Cheers,
> Steve
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Re: [Asterisk-Users] Polycom Configuration

[Sean Cook]

Damon Estep wrote:
> What you are proposing is quiet simple, and is done regularly. We
> provision Linksys Sipura ATAs via a perl script with SSL and client
> certificate authentication, as well as Polycom phones via XML file
> drops. The newest Polycom firmware also states that ssl is supported,
> but we have not made the conversion yet.
>
>   
I don't want to go to each phone and provision it... I am already using
ftp provisioning for the polycoms, I would just like a simple
configuration tool that will build the xml files and drop them in the
ftp/tftp/http directory so that I can manage them from a web interface. 
It is a simple task but one that requires time.

What I liked about the sipx tools is that it allowed for almost an asset
management tool.  Again... I have not found anything out there that is
very configurable.   What I am suggestion is a centralized tool that
will support multiple phone types and configurations...

sean
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[Asterisk-Users] Polycom Configuration

[Sean Cook]

I have been playing around with sipX for a couple of days now, and while
I don't really like it (just feels wierd), I do really like the
management interface for provisioning phones.  I was wondering if anyone
had considered ripping this out of sipX or porting it to a simple php
interface or something?

If not I think it would be of interest to create a phone provisioning
tool for asterisk (although not directly in asterisk...).


Takers?  Thoughts?  proverbial bugger off?

Regards,

Sean
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Re: [Asterisk-Users] Quad T1 Card

[Sean Cook]

> One of the primary differences between the two cards is the Sangoma
> h/w echo canceler handles more cases of echo then do the Digium cards.
> Whether you need that additional coverage is 100% dependent on your
> specific implementation (eg, your T1/PRI provider), and not on what
> the list thinks about the two products.
>
> Since there are no affordable tools to truly quantify echo for each
> specific implementation, as a pbx engineer your toolkit should
> probably include both cards. Sort of like try the less expensive card
> and if it doesn't address your echo issues, then try the more
> expensive one.
>

No offense but isn't that like saying  "Don't take what the list has
to say about your purchase... instead you should guess and hope you get
the right answer... but if you don't, gamble again and buy two cards?"
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[Asterisk-Users] Quad T1 Card

[Sean Cook]

Ok... I am reluctant to ask this question as I believe that it may be
like asking what someones favorite linux distribution is... but I need
to make an informed decision.

We are getting ready to upgrade from a TE210P to a quad T1 card with
echo cancellation.  I am trying to decide between the Sangoma card and
the Digium card.  I need this to have great quality and I need it to
work well.

I would like to hear about personal experiences and any other technical
differences between the card.  Again this is not intended to start a
pissing contest or flame war
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Re: [Asterisk-Users] Re: syslog server

[Sean Cook]

syslog is limited to the amount of space you have on disk...  I have
everything centralized by host/date on my syslog server.

In asterisk you can enable syslog by editing the logger.conf

;syslog keyword : This special keyword logs to syslog facility
;syslog.local0 => notice,warning,error

On the phone it depends on the phone... since I don't really care about
the phones... I don't worry about it ;)

A great reference for syslog-ng...
http://www.campin.net/syslog-ng/faq.html
http://www.campin.net/newlogcheck.html


On Tue, 2006-06-06 at 15:48 -0400, Matthew Warren wrote:
> lines

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Re: [Asterisk-Users] syslog server

[Sean Cook]

hmmm... I am a huge fan of syslog-ng, but the stock syslog on your *
system should work well...


Matthew Warren wrote:
> Does anyone know a good syslog server to use for grandstream phones?  I want
> to set this up to see what is happening with the grandstreams.  Easy and
> Free preferably.
>
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Re: [Asterisk-Users] Campusing two Asterisk boxes?

[Sean Cook]

Yes... it is very easy to do...

; on box a
exten => _NXXNXX,1,DIal(IAX2/boxb/${EXTEN})


;on box b
exten => _NXXNXX,1,Dial(IAX2/boxa/${EXTEN})


you just need to make sure that the context on the each side will have a
match for passing in ${EXTEN} to the other side

[from-boxa]
exten => _NXXNXX,1,Dial(ZAP/g0/${EXTEN})


[from-boxb]
exten => _NXXNXX,1,Dial(ZAP/g0/${EXTEN})


> I have been looking around some and I can't seem to find anything which
> will
> answer my question.  If I have two Asterisk boxes in different locations
> which
> are linked to each other over the internet, can I configure the boxes to
> use
> each other's lines as local?
>
> In other words, let's say Site A has Phone1
> for a 1FB line going into it on an FXO port.  Site B has Phone2 for a 1FB
> line going into it on an FXO port.  Is there a way to configure Site A to
> use Phone2 from Site B and vice versa?
>
> Undrhil
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Re: [Asterisk-Users] How to make this into a Macro?

[Sean Cook]

> 
> Just make something like this:
> 
> exten => 8863959,1,Macro(dial,8863959)
> 
> [macro-dial]
> exten => s,1,Dial(SIP/${ARG1},60,r)
> exten => s,2,NoOp(${DIALSTATUS})
> exten => s,3,Voicemail,[EMAIL PROTECTED]
> exten => s,104,Voicemail,b$([EMAIL PROTECTED]
> exten => s,105,hangup
> 
> 

Just to make a bit more modular...

; this will just allow you to pass in the voicemail box
exten => _NXX,1,Macro(dial,SIP/${EXTEN},[EMAIL PROTECTED])

; you can use the same macro...  (note you can also ring two extensions.
; we do this and each extensions softphone is ${EXTEN}h for convenience.
exten => _4XXX,1,Macro(dial,SIP/${EXTEN}&SIP/${EXTEN}h,[EMAIL PROTECTED])


[macro-dial]
exten => s,1,Dial(${ARG1},60,r)
exten => s,2,NoOp(${DIALSTATUS})
exten => s,3,Voicemail,u${ARG2}
exten => s,104,Voicemail,b$(ARG2}
exten => s,105,hangup


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Re: [Asterisk-Users] Polycom-Asterisk hints/presence

[Sean Cook]

>
> Sean,
>
> Where did you find that quote, I would like to see the rest of the
> thread if there was relevant discussions.
>
> Thanks.
>   
It was really a two email thread... I had sent an email asking what the
status of BLA/SCA:  Here is the entire thread:

Sean Cook wrote:

> > I take it SCA/BLA isn't going to make it into 1.4.  Anyone have any idea
> > when support will be added to asterisk for this?
>   

There has been no BLA support written at this point, and it does not
appear that when we do it we will even use SIP-B to get there. SIP-B is
very complex (overly so) and doesn't seem like a practical solution for
implementing basic key system type functionality.

However... I can say that an implementation of this functionality is
being worked on at this time, and we intend to make it available in
Asterisk as soon as we can. It will most definitely not be in 1.4, but I
would expect it to appear some time early in the next development cycle
and be part of Asterisk 1.6.


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Re: [Asterisk-Users] Polycom-Asterisk hints/presence

[Sean Cook]

I would think the biggest issue with this at this point is because
Asterisk is not a SIP only platform, if one were to implement shared
line appearance, it would need to be designed in such a way that
channels other than SIP channels could participate in the "sharing" of
lines.

It has been talked about a great deal, and at some point, it will
probably be done, but just being compliant with the SIP supported
methods of shared line appearance is not all that has to be done in
order to get this feature in.



From Kevin Flemming at Digium:

It will most definitely not be in 1.4, but I
would expect it to appear some time early in the next development cycle
and be part of Asterisk 1.6.



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[Asterisk-Users] Hold Status

[Sean Cook]

Is there a way through AMI or AGI to determine whether a channel is on
hold?  Or if a channel has a call on hold?

Sean
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[Asterisk-Users] Shared Call / Bridged Line Appearances (SIP-B)

[Sean Cook]

I take it SCA/BLA isn't going to make it into 1.4.  Anyone have any idea 
when support will be added to asterisk for this?


Sean
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Re: [Asterisk-Users] AEL #include

[Sean Cook]

No, SVN trunk is not a 'release'. It's a development area. A 'release'
involves packaging it, documenting the changes, and handling bug reports
against it in a different way.
  
I guess I see release as a verb and not a noun.  Probably is a good idea 
to use the appropriate terminology... my bad :P

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Re: [Asterisk-Users] AEL #include

[Sean Cook]

Actually... it means not in the production release.  the subversion
"trunk" is a release but it is not for the faint at heart.  While
generally everything works pretty well, it is expected that you will
find bugs and have issues :)

Sean

Douglas Garstang wrote:
> In non-developer-speak, that means, 'not in current release', correct?
>
>   
>> -Original Message-
>> From: Aaron Daniel [mailto:[EMAIL PROTECTED]
>> Sent: Tuesday, May 30, 2006 2:00 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [Asterisk-Users] AEL #include
>>
>>
>> No, only works in the old language, or in AEL2 which is 
>> released in trunk.
>>
>> On Tue, 30 May 2006, Douglas Garstang wrote:
>>
>> 
>>> Anyone know if #include works in ael yet?
>>>
>>> extensions.ael:
>>> #include "inc/pbx/global.conf"
>>> context test_context {
>>> };
>>>
>>> *CLI> ael reload
>>> May 30 13:56:45 NOTICE[8516]: pbx_ael.c:1120 
>>>   
>> handle_root_token: Unknown root token '#include'
>> 
>>> May 30 13:56:45 WARNING[8516]: pbx.c:3758 
>>>   
>> ast_merge_contexts_and_delete: Requested contexts didn't get merged
>> 
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>>>   
>> -- 
>> Aaron Daniel
>> Computer Systems Technician
>> Sam Houston State University
>> [EMAIL PROTECTED]
>> (936) 294-4198
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Re: [Asterisk-Users] PCI Problems

[Sean Cook]

Kevin P. Fleming wrote:
> Sean Cook wrote:
>
>   
>> lspci -vb # shows IRQ 9 being shared
>> 
>
> That is not a valid piece of information, and wherever you learned it
> you should unlearn it :-)
>
> 'lspci -vb' shows the interrupt that the BIOS assigned to the PCI
> device, but not where the kernel assigned it, which can be radically
> different, especially when an APIC is being used. We have seen cases
> where there were audio issues anyway, but it's not clear that interrupt
> sharing was actually the case in those cases.
>   
What could be the other causes?  I have exhausted everything I know how
to do.  PCI sharing explains it (whether or not it is infact the
problem).  This card shares the BIOS assigned interrupt with the network
card...


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Re: [Asterisk-Users] PCI Problems

[Sean Cook]

Kevin P. Fleming wrote:
> Sean Cook wrote:
>
>   
>> lspci -vb # shows IRQ 9 being shared
>> 
Interesting... I learned it from kenny in training in Huntsville ;)


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Re: [Asterisk-Users] PCI Problems

[Sean Cook]

Andrew Kohlsmith wrote:
> On Thursday 25 May 2006 13:06, Sean Cook wrote:
>   
>> There are no known compatibility issues (IRQ, IO etc) with ANY Sangoma
>> hardware and ANY make/brand of PC/server- NONE
>> 
>
>   

Andrew,

Thank you for all of the information... I will clarify the "any", "none"
verbiage is not mine... it was take from voip-info.  I will admit, I
should have done my homework on the motherboard I purchased.  I assumed
that if you buy quality hardware it would just work.  I am not in that
position right now (again my fault).

Would you say, in your estimation, that the sangoma cards deal with bus
level interrupt sharing better than the digium cards?
>
> There have been some hardware PCI interop issues with Digium's stuff, but I 
> know for a fact that these have been fixed in their rev2 hardware.
>   
Then I am confused: 
lspci -vb
Communication controller: Digium, Inc. Wildcard TE210P (rev 02)

Yet I still have sound quality issues due to PCI interrupt sharing. 

zttest -v
Best: 99.987793 -- Worst: 99.975586 -- Average: 99.976120

cat /proc/interupts
  0:1162398   60615544IO-APIC-edge  timer
  1:  1149IO-APIC-edge  i8042
  4:   1974  92613IO-APIC-edge  serial
  8:  9402IO-APIC-edge  rtc
  9:  0  0   IO-APIC-level  acpi
 12: 16658IO-APIC-edge  i8042
169:  176921220609   IO-APIC-level  eth0
177:   2006 123888   IO-APIC-level  eth1
185:  10753 477340   IO-APIC-level  megaraid
193:  43424   61728058   IO-APIC-level  wct2xxp

lspci -vb # shows IRQ 9 being shared
02:09.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5704
Gigabit Ethernet (rev 03)
Subsystem: Broadcom Corporation Unknown device 1644
Flags: bus master, 66MHz, medium devsel, latency 64, IRQ 9
Memory at fc9c (64-bit, non-prefetchable)
Memory at fc9b (64-bit, non-prefetchable)
Expansion ROM at fc9a [disabled]
Capabilities: [40] PCI-X non-bridge device
Capabilities: [48] Power Management version 2
Capabilities: [50] Vital Product Data
Capabilities: [58] Message Signalled Interrupts: 64bit+
Queue=0/3 Enable-

01:03.0 Communication controller: Digium, Inc. Wildcard TE210P (rev 02)
Flags: bus master, medium devsel, latency 64, IRQ 9
Memory at fc8efc00 (32-bit, non-prefetchable)

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Re: [Asterisk-Users] PCI Problems

[Sean Cook]

Rob Lith wrote:
> Does the sangoma handle sharing interrupts in some other way?
from:  http://www.voip-info.org/wiki/view/Sangoma

There are no known compatibility issues (IRQ, IO etc) with ANY Sangoma
hardware and ANY make/brand of PC/server- NONE

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[Asterisk-Users] PCI Problems

[Sean Cook]

OK... maybe I got a little anxious and ran out and bought a Tyan GX28
with dual Opteron (dual core) processors.  (It is a nice server ;) )  I
did neglect to find out that you can not manually set the IRQ's on this
motherboard.   I am now stuck sharing an IRQ with the ethernet
controller and no foreseeable end to my dilemma. 

I have a Digium TE210P and zttest consistently runs at 99.97% which as
you guessed, is giving rather unpleasing sound quality.  My options as I
see it are:

1.  Buy a new server
2.  Buy a sangoma A102U

I am looking for practical suggestions from those of you out there who
have had a similar experience that may aid me in making this decision. 

Thank you,

Sean
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Re: [Asterisk-Users] Realtime Asterisk Problem

[Sean Cook]

try running realtime load sip_buddies exten  and see what you get...

SEan

On Thu, 2006-05-25 at 05:10 +0530, Chandan Mishra wrote:
> Hi
> i am using the asterisk server on one machine and mysql on another
> machine.I have my mysql running on 192.168.77.75 and asterisk running
> on the 192.168.77.77.
> 
> when executing following cli command  on asterisk server on 192.168.77.77
> 
> *CLI> realtime mysql status
> Connected to [EMAIL PROTECTED], port 3306 with username root for
> 8 minutes, 45 seconds.
> 
> But my phones are not getting registered with the 192.168.77.77. The
> phones have entry in the mysql database.
> 
> my res_msql.conf on machine 192.168.77.77 have the following entries
> [general]
> dbhost=192.168.77.75
> dbname=asterisk
> dbuser=root
> dbpass=root
> dbport=3306
> dbsock=/tmp/mysql.sock
> 
> When the cli shows the asterisk connected then why my phones are not
> able to register the 77 asterisk server. If i am running asterisk on
> 75 itself then the phones are able to register with the 75 server
> (where mysql is installed).
> 
> Please suggest the solution.
> 
> Thanks
> 
> Chandan
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RE: [Asterisk-Users] What and When is the next version of Asterisk?

[Sean Cook]

Not necessarily... my understanding is that the feature freeze was done
about 2 months ago for 1.4 and the release cycle is 6 months putting 1.4
due for release here pretty soon...

As to what is in the new release... probably most of the sip changes
that Olle has been working on as well as the core module
loader/unloader... beyond that... I don't know.

On Wed, 2006-05-24 at 16:29 -0700, Kerry Garrison wrote:
> 1.2.8 would be the logical next version.
> -Kerry
>  
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] 
> > [mailto:[EMAIL PROTECTED] On Behalf Of Obelix
> > Sent: Wednesday, May 24, 2006 1:11 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [Asterisk-Users] What and When is the next version 
> > of Asterisk?
> > 
> > 
> > What and When is the next version of Asterisk?
> > 
> > /Obelix
> > 
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> 
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Re: [Asterisk-Users] database lookup

[Sean Cook]

a very elegant solution would be to do a Realtime lookup and match the
variable that is set.

exten => s,1,Answer()
exten => s,2,Realtime(sometag,cid,${CALLERID(num)},check_)
exten => s,3,gotoif($["${check_daughters}foo" = "foo"]?...)

Then you can just add a record to the database...

On Wed, 2006-05-24 at 15:09 -0600, Bromont - wrote:
> Hi all,
>  
>  I'm looking for an easy way to lookup numbers from the database so I can 
> fork calls from my daughters friends onto her IP phone/answering system. I'm 
> looking for something very similar to LookupBlacklist, but I'm already using 
> LookupBlackist to filter out telemarketers. What I'm doing now is adding 
> multiple exten=> lines to my extensions.conf file to match those numbers. I'd 
> like to just add her friends numbers to the database, then do a "lookup" to 
> match and branch off. Any ideas? Thanks, Marc.
> 
> 
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Re: [Asterisk-Users] Option to reach someone in voicemail?

[Sean Cook]

Matt wrote:
> That would be fine... and I know you can do alot of stuff with
> Asterisk, you just have to think "outside the box"(tm) sometimes.
> but my question with doing that is, then how do I make it go to this
> message only when the person is not available, and not everytime
> someone gets transfered to this extension?
>
exten => _4XXX,1,Dial(SIP/${EXTEN},20,tT)
exten => _4XXX,2,Goto(s-${DIALSTATUS},1)

exten => s-UNAVAILABLE,1,Background(some-ivr-recorded-message)
...

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Re: [Asterisk-Users] Option to reach someone in voicemail?

[Sean Cook]

Why not use "exten => a,..." and do a for more options press *, then
have it drop into an IVR...

Sean


Matt wrote:
> Hi,
> Is there anyway to add an option to dial someone from voicemail?
>
> I know I can make 0 go to operator... however, I want to do something
> like our Nortel did which was "Press 7 to reach XYZ" and 7 could be
> programmed to point to a specific person/extension/number.
>
> Can I do this?
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Re: [Asterisk-Users] asterisk and ODBC

[Sean Cook]

have you tested to make sure that you can connect to the odbc resource
outside of asterisk via perl/php/(insert random language here)?  Make
sure odbc is setup correctly and working before proceeding with the
asterisk part.


Sean

Dumpolid Exeplish wrote:
> Hi,
> I have duetifully followed the instructions in the cdr.txt but
> asterisk still cannot connect to the MS SQL server. I can connect
> using the ODBC native connector bu asterisk still cannot
>
> On 5/16/06, *Bruce Reeves* <[EMAIL PROTECTED]
> > wrote:
>
> Yes you can use MSSQL with the ODBC driver, I have it working for
> CDR logs, I had to install unixODBC and configure it then use the
> cdr_odbc.conf file to specify. Check in your source files or svn
> checkout for a docs folder and the cdt.txt file.
>
>
> On 5/16/06, *Dumpolid Exeplish* < [EMAIL PROTECTED]
> > wrote:
> Hi All,
> How can i use Microsoft SQL server with asterisk, Can the unix
> ODBC diriver interface MSQL?? and what module would i be using to
> access ODBC from asterisk??
>
>
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>
>
>
> -- 
> Bruce
> Nortex Networks
>
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> 
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Re: [Asterisk-Users] Development news :: Smarter medialess calls!

[Sean Cook]

Olle,

Is there a poster of you that I can put up on my wall ;)

Regards,

Sean

Olle E Johansson wrote:
> Friends,
> To update you on recent changes in svn trunk, I can inform you that we
> now have ever smarter
> ways to handle media streams in Asterisk than we do in 1.2 for the
> IAX2 and SIP protocols.
>
> * IAX2: Splitting signalling and media apart
>
> Starting with the IAX2 protocol, we now have the ability to transfer
> media streams to go directly
> between IAX2 servers and keep the signalling path. Before, when
> Asterisk did a native transfer
> to optimize the IAX2 call path, we lost all tracks of the call and
> could not get a CDR. With this
> patch, by Mark, we now have a hybrid solution that releases the media
> but keeps IAX2 signalling.
> This is a very new feature, so I don't expect the various non-asterisk
> IAX2 clients out there to
> support it yet. When they do, it will mean a huge change in the number
> of calls your server can
> handle. For now, this optimizes calls in Asterisk IAX2 "clusters".
>
> * SIP: Removing the media immediately, not as an afterthought
>
> Mark and Kevin have been working on various ways to optimize the setup
> of a SIP call
> where Asterisk has no reason to stay in the media stream. Asterisk
> will now setup the
> call directly between the two devices instead of accepting the call,
> staying in the stream and
> then, as a sudden afterthought, send re-invites to release the media
> stream.
>
> An additional new feature, inspired by a community patch on the bug
> tracker, is that
> we now also release calls if SIP INFO dtmf is used. Since the DTMF is
> not handled in
> the RTP media stream, we can release the call (unless there is another
> reason to stay
> in the media path, like NAT support).
>
> These changes optimize your Asterisk a great deal and will hopefully
> make Asterisk
> scale a bit more. Your development team is as always focused on
> scaling issues, trying
> to go where no software PBX has gone before, explore new telephony
> territories...
> VoiP trekking... Well, enough of that. Sorry, got sidetracked.
>
> * Asterisk 1.4 - I see a shape, an outline
>
> The work with Asterisk 1.4 is going into the final stages. We are
> working hard to commit
> the changes that are ready and finalize the 1.4 release. If you visit
> the bug tracker, you already
> see patches that we've marked "post 1.4" since we feel they're not
> ready. The next release is
> not that far away, so it's not a big thing. We won't wait over 1 year
> like we did between 1.0 and
> 1.2.
>
> This weekend, I'm leaving for my Training in New York. Next training
> is in Stockholm,
> Sweden in June, after that we're launching the Asterisk SIP
> Masterclass in Chicago in
> July - with a gold team teaching: Ed Guy, Terry Wilson and myself.
>
> While I'm travelling around, you can spend all your free time testing
> Asterisk 1.4 for us.
> We need your help, now. Download svn trunk and test in your environment!
>
> On behalf of the community - thank you for testing!
>
> SIP greetings!
> /Olle
>
> ---
> * Olle E. Johansson - [EMAIL PROTECTED]
> * Asterisk Training http://edvina.net/training/
>
>
>
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[Asterisk-Users] Realtime Postgres via ODBC

[Sean Cook]

I am running  unixODBC to connect to postgres for your realtime data for 
things like call forwarding, dnd and have noticed a significant delay 
when running the realtime application.  Has anyone else encountered this?


Even from the CLI if I do "realtime load cf_data exten 4501" it lags for 
almost 2 full seconds before I get the result set back.


Queries directly to the database from anything else are almost 
instantaneous...


In the dial plan...
exten => s,1,RealTime(cf_data,exten,4501,cf_)  <-- this is the point 
where it lags...

exten => s,2,GoToIf...

Any tips or suggestions?
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Re: [Asterisk-Users] monitoring sangoma cards via snmp

[Sean Cook]

[EMAIL PROTECTED] wrote:

Hello,

Digium does not provide snmp support to monitor their
cards !

  
That's like saying Toyota doesn't provide gas with their cars.  You can 
setup snmp with in linux and have it execute commands that you want to 
determine whether or not the hardware is functioning as you wish.  
Hardware is hardware...  Intel doesn't provide snmp for my motherboard 
either...



Anybody has tried Sangoma product A104 Quad T1/E1 or
others ?
  
I don't think you are going to find exactly what you are looking for 
with out purchasing an appliance...


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Re: [Asterisk-Users] TDM4xxP

[Sean Cook]

(never tried it).  Personally I 
would spend $100 bucks and get a FXS or FXO module for it. 

As for the X100P clone, the timing on them IMO is not anywhere near what 
the tdm400.  There are a few documented instance of timing just flat not 
working.



Not working or irregular?
  


irregular.  I have had better luck with the ztdummy drivers (2.6 kernel) 
than I have with the x100p... YMMV


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Re: [Asterisk-Users] TDM4xxP

[Sean Cook]

See here is the interesting thing... zaptel alone is not sufficient to 
provide timing... you need wctdm or something else loaded.  But with out 
modules, I don't think wctdm will load.  (never tried it).  Personally I 
would spend $100 bucks and get a FXS or FXO module for it. 

As for the X100P clone, the timing on them IMO is not anywhere near what 
the tdm400.  There are a few documented instance of timing just flat not 
working.


Sean

Time Bandit wrote:
At that cost of downtime I would grab a module just to be sure and 
stick it
in, no need to actually configure that zapata channel but I would 
think the

card needs some module.


And what about an X100P clone, would that give an accurate timing 
source ?

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Re: [Asterisk-Users] TDM4xxP

[Sean Cook]

hm... why not just use ztdummy and save the $150 for the card?

Sean


Steve Totaro wrote:
I have a TDM4xxp card with no modules.  My question is, will this card 
be sufficient to provide timing or does it need to have modules?


Thanks,
Steve
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Re: [Asterisk-Users] Call Center Phone with Auto Answer

[Sean Cook]

Kevin Savoy wrote:


Can anyone recommend a phone to use in an inbound call center 
environment that has an auto answer feature? We don’t want the agents 
having to acknowledge the call. The call should just activate on the 
headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601. 
None of these support it.


Thanks

May be I am not understanding... Why not use agentlogin and have the 
agents always logged in with MOH... they get a beep and they are 
connected.. Change ackcall=no in agents.conf


Then you don't need auto-answer
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[Asterisk-Users] Call Hold and Retrieve

[Sean Cook]

Our current PBX allows us to put a call on hold and then anyone in the 
building can dial #9XXX and pick up that call.  I know that I can 
replicate a similar function by parking.  But I would really like to 
replicate the existing setup.  Something about having to train people to 
hit more than one button.  Any suggestions?


Sean
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Re: [Asterisk-Users] Unable to get TDM400p working

[Sean Cook]

couple of things... was asterisk compiled after zaptel?  from the cli 
try "load chan_zap.so" and see what you get


Ben Gore wrote:

This has got to be a stupid error I'm making...

I have been experimenting with different hardware and software
configurations before I decide what to use as a production platform. Up
until just recently things were going well.

But now it appears I'm unable to get access to my TDM400p from Asterisk. I
know the TDM card works fine, used it in another machine where it
performed flawlessly. I have been using the same set of conf files, just
copying them over from machine to machine. The hardware is Pentium 4
all-Intel chipset mainboard.

The one difference here is I'm trying CentOS. I've been through the
problems getting Zaptel to compile with the error in spinlock.h. I got
to...

Zaptel Configuration
==


Channel map:

Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)

4 channels configured.

and thought I was home free. Wrong!

zttool recognizes the card properly and reports status "OK"

Asterisk runs and I can make calls on SIP phones with no problems. However
I get no dial tone on the analog phone and outgoing calls through the TDM
(from the SIP phones of course!) produce this on the console:

NOTICE[5934]: app_dial.c:1010 dial_exec_full: Unable to create channel of
type 'Zap' (cause 66 - Channel not implemented)
  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing Congestion("SIP/100-24d1", "") in new stack
  == Spawn extension (internal, 91234567, 102) exited non-zero on
'SIP/100-24d1'

The conf files are the same as they were on another working machine, I
just copied them over. I'll be going over them /again/ next.

What am I missing?

Thanks.

-Ben





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Re: [Asterisk-Users] Volume configuration on Polycom Soundpoint 501 phone

[Sean Cook]

sip.cfg

voice.volume.persist.headset="1" voice.volume.persist.handsfree="1"/>


Jim Freeze wrote:

We are using the polycom 501 phones, and are having some challenges
with the volume setting. When a phone call comes in, the user ups the
volume for the handset, but they have to repeat that for every call.

Currently, the volume level seems to reset itself at about 60%.
Is there a way for the user to change their default volume level?

Thanks

--
Jim Freeze


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Re: [Asterisk-Users] Tool for Polycom configurations

[Sean Cook]

Try this one:

http://www.freedomphones.net/polycom/files/polycom.phone1cfg.pl-script

Sean

Andrew Kohlsmith wrote:

On Thursday 04 May 2006 14:45, Bruce Reeves wrote:
  

I am getting read to roll out close to 100 polycom phones and wondered if
any one knows of a program to take a list of MAC addresses, extensions, and
names and generate the configuration files?



You can do this relatively easily with Perl.  There is a script somewhere that 
will take your sip.conf and generate phone[exten].cfg files, but it knows 
nothing about MAC addresses and as such will not generate the 
[MACADDRESS].cfg files.


Again though, this isn't too tricky to do.  A few hours' worth of work.  The 
tricky part would be making sure you got the right phone to the right desk if 
the extension #s are physically important.  :-)


If you need some help with the script I am available for consulting.  Contact 
me offlist if this is something you'd like to discuss.


-A.
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Re: [Asterisk-Users] IAX Configuration

[Sean Cook]

my guess is that you are trying to dial a sip channel to reach an iax peer.

Dial(SIP/19)

should be

Dial(IAX2/19)

Olivier Saulnier wrote:

Hello,

I have some problems with a new configuration:
I always have on my asterisk console the message:
chan_iax2.c:5886 update registry: restricting registration for peer 
'19' to 60 secondes

I connect only two ip phone with iax protocol.

And when i want to call 19 phone, it's hangup. No information in 
console view, or in file /var/log/asterisk/messages.

Do you have any idea?


My files a there:
extensions.conf:

[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[globals]
CONSOLE=Console/dsp; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest; IAXtel username/password
TRUNK=Zap/g2   TRUNKMSD=1  
[INTERNAL]

exten => 19,1,Dial(SIP/19,20,tr)
exten => 19,2,Voicemail(u19)
exten => 19,hangup
exten => 19,102, Voicemail (b19)
exten => 19,103,Hangup

exten => 20,1,Dial(SIP/20,20,tr)
exten => 20,2,Voicemail(u20)
exten => 20,hangup
exten => 20,102, Voicemail (b20)
exten => 20,103,Hangup


iax.conf:
[general]
bandwidth=low
disallow=lpc10   jitterbuffer=no
forcejitterbuffer=no
[19]
type = friend
username = 19
secret = 19
host=dynamic
context = INTERNAL
mailbox=19

[20]
type = friend
username = 20
secret = 20
host=dynamic
context = INTERNAL
mailbox=20


Best regards,



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Re: [Asterisk-Users] Polycom NTP issue

[Sean Cook]

Haven't see this posted yet but keep in mind the polycom does offsets in 
seconds not in hours... I spent three days figuring that out...


tcpIpApp.sntp.gmtOffset="-18000" is the same as GMT -5

Sean




Aaron Daniel wrote:
Whoops... meant dhcp... Keep in mind that if you're using windows' dns 
server, it doesn't allow negative offsets, but the linux one does.  
That was a pain for us as well.


On Thu, 27 Apr 2006, Aaron Daniel wrote:


What dns server are you running?

On Thu, 27 Apr 2006, Kerry Garrison wrote:

I am ready to pull my hair out. I cannot seem to get the Polycoms to 
read

the time properly. Regardless of the server they are pointed to our the
offset, i am getting the correct time, but 24 hours ahead. So for 
today it

is showing Friday April 28 but with the correct time. Any clues?

Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 -  
[EMAIL PROTECTED]
 http://www.techdatapros.com









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Re: [Asterisk-Users] Really Old Rotary Phone

[Sean Cook]

Well it works!  The pulse detection is a little squirrelly, even with 
the debounce changes to wctdm.c.  I can't get an audible ring but it 
does work.


Sean
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Re: [Asterisk-Users] Really Old Rotary Phone

[Sean Cook]

I do have a TDM400 and the Sangoma A200.  I have done pulse with the 
TDM400, but have not with the A200.  I have just never seen a phone like 
this... ;)


Rusty Dekema wrote:

On 4/25/06, Sean Cook <[EMAIL PROTECTED]> wrote:
  

This worked perfectly! Thank you!

Sean



Now, I think the question is, does your ATA actually support
rotary/pulse dialing? Mine (SPA-2000) did not. I bought a (very cheap)
MITEL-1 "Smart Dialer" and went through a RIDICULOUS amount of pain
trying to configure it to convert from pulse to DTMF dialing, and it
did sort of work although I never seemed to be able to get it
configured exactly the way I wanted it.

I ended up getting a TDM400 with a couple of FXS modules (which I had
been needing to get anyway), and that worked perfectly after patching
the pulse-dial debounce code in Zaptel (although I believe the newest
version of Zaptel already comes with the needed changes).

If there are ATAs that support pulse dialing, I'd like to know about
it, because now I would like to be able to use my pulse phone in
locations other than the physical location of my Asterisk machine with
the TDM400 card in it. So if you find that yours works, could you
please let me know?

Thanks,
Rusty
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Re: [Asterisk-Users] Really Old Rotary Phone

[Sean Cook]

Jerry Jones wrote:

Yellow=ground - not used
Green = tip
Red = ring

connect green/red to rj pins 4/5

You could pick up a quarter mod line cord (mod to spade) and replace 
the cord, or use a screw terminal block to connect to line.


Enjoy



This worked perfectly! Thank you!

Sean
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[Asterisk-Users] Really Old Rotary Phone

[Sean Cook]

Ok... I am not a telephone guy... I was born after rotary phones, so 
forgive my ignorance in this matter.  I am trying to get a really old 
rotary phone up and running with an ATA.  Why?  Who knows... just 
thought it would be cool. The problem is that it does not have an RJ11 
connector, instead it has three wires (green,yellow,red).  Does anyone 
know what that type of connector is called?  Or know of a reference to 
build an adapter to 2 line?


Thanks,

Sean
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Re: [Asterisk-Users] Sangoma A200 preventing Zap channels from disconnecting immediately after PSTN line hangs up (getting empty voicemails)

[Sean Cook]

On Mon, 2006-04-24 at 17:20 -0400, Mike Garey wrote:
> As far as I can tell, after discussing this matter with other asterisk
> users in my area, my telco _does_ provide disconnect supervision..  It
> seems that the problem is actually related to the Sangoma A200 card
> I'm using, as two other people both using this same card have
> expressed the same problem..  Are there any other users on this list
> using the Sangoma A200 FXO port card, and experiencing problems with
> asterisk not detecting when a channel has been disconnected?  Thanks,

Are you using kewlstart?


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Re: [Asterisk-Users] Polycom MWI

[Sean Cook]

Try specifing [EMAIL PROTECTED]  I know their have been some changes
with the implicit defining of the voicemail groupsthat may have
something to do with it... I didn't have to do anything special for my
polycoms.

Sean

On Fri, 2006-04-21 at 06:17 -0400, Andrew Kohlsmith wrote:
> On Friday 21 April 2006 00:28, Kerry Garrison wrote:
> > Didn't help. Could I be missing something else?
> 
> In Avi's footsteps, here is my phone.cfg and sip.conf entry.  This works for 
> 12 phones.  Note that I'm not subscribing to anything on the Polycom; 
> Asterisk sends MWI for the mailbox to the phone when there are messages 
> waiting, and the user can dial '999' to access voicemail.
> 
> 
>  msg.mwi.1.subscribe=""
> msg.mwi.1.callBackMode="contact"
> msg.mwi.1.callBack="999"
> />
> 
> 
> [211]
> context=polycom_outgoing
> type=friend
> host=dynamic
> secret=*
> disallow=all
> allow=ulaw
> mailbox=211
> vmexten=voicemail
> dtmfmode=rfc2833
> callgroup=1
> pickupgroup=1
> callerid=Jack <211>
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Re: [Asterisk-Users] asterisk credit card processing

[Sean Cook]

Shouldn't  be too difficult... perl has some great payment modules: 
check out Business::OnlinePayment 
 
modules on CPAN


Joseph wrote:

Is there a way somehow to implement Asterisk with Credit Card Processing
(IVR system)?

  


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Re: [Asterisk-Users] SIP Asterisk Polycom Reinvite

[Sean Cook]

hahah... I have run into that dozens of times... you can even pull on 
the cord a bit and it seems tight... then you give it a little more push 
and ... you spend about 5 minutes thinking geez I am an idiot...


Sean

Matthew T. O'Connor wrote:
I had a one way audio problem with my Polycom 501's and it turned out 
that the cord wasn't plugged in to the handset all the way.  It looked 
like it was in, but it wasn't in all the way till it clicked.


Matt



Damon Estep wrote:
Wondering if anyone has experienced an intermittent one way audio 
(called party can not hear) problem in these conditions;


 


Several IP501 phones local, same subnet.

Remote asterisk

No NAT anywhere

 


Polycom IP501 ulaw only, canreinvite=yes

Asterisk

Call termination path is to a sonus GSX operated by the upstream 
carrier, ulaw only, canreinvite=no


 

The idea is that if the Polycoms are canreinvite=yes and the PSTN 
termination path is canreinvite=no then calls between polycoms should 
not have asterisk in the media stream and wan link utilization is 
reduced.


 

The problem looks like the Polycom keeps trying to reinvite the sonus 
and the call never sets up right, and not with all calls…


 

Any experience with this? Maybe there is a totally different issue I 
am overlooking?


 

About 3 to 5% of all Polycom to PSTN via asterisk>SIP peer calls are 
impacted.


 

I have not set the Polycom canreinvite=no yet, hoping to not have to 
do that as the wan link is a t1 that is also used for data.


 


Thanks for any help!

 


Damon

 

 





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Re: [Asterisk-Users] Cisco 7960 - hints

[Sean Cook]

Are you using chan_sccp for you cisco implementation? 


Aaron Daniel wrote:
Sadly, no.  The SIP firmware on the Cisco phones doesn't support 
subscribing to other lines.  I heard chan_sccp does though.. now 
to figure out how.


Aaron

On Thu, 6 Apr 2006, Sean Cook wrote:

Is the Cisco 7960 capable of monitoring other extensions (hint 
status) with a SIP implementation?  Seems like it could, just can't 
find any info on it...


Sean
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[Asterisk-Users] Cisco 7960 - hints

[Sean Cook]

Is the Cisco 7960 capable of monitoring other extensions (hint status) 
with a SIP implementation?  Seems like it could, just can't find any 
info on it...


Sean
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Re: [Asterisk-Users] TDM2400P problems

[Sean Cook]

We have had this problem with the TDM400 and just about every thing we 
have ever had... it isn't the card that is chopping off the first 
digit.  It is the fact that it picks up too quickly and starts to dial.  
Change your dial to be Zap/g0/w${EXTEN} and see if that takes care  of 
the problem


Tim Jackson wrote:

I am having issues with a TDM2400P.  It appears when the ZAP channel dials
out, it randomly chops the first digit off of the number.  I have tried
relaxdtmf=yes, turning up and down the txgain, turned off and on the echo
cancellation, generated new zaptel (with updated spinlock.h)...

I am at a loss.  Can someone please offer some help?

Thanks.

TJ
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Re: [Asterisk-Users] Hinting

[Sean Cook]

I started with the polycoms to me its "man those cisco phones boot 
fast" :)


Aaron Daniel wrote:
I think I'm getting there slowly... I notice in your extension, you're 
hinting SIP/2348.  I'll see if that helps me a bit, this damn phone 
takes freaking forever to reboot.


Aaron

On Mon, 3 Apr 2006, Anthony Rodgers wrote:


Hi Aaron,

You need to create an entry in the directory of the _watching_ phone 
with the extension of the _watched_ phone as its contact. Set the 
'Buddy watch' of this entry to 'Yes', so it appears in the list of 
'Buddies' (couldn't they come up with another term for this? :-)


Then, in extensions.conf, set a hint for the _watched_ extension like 
this:


exten => 2348,hint,SIP/2348

Let me know if you have any more questions.

Regards,





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Re: [Asterisk-Users] Hinting

[Sean Cook]

Aaron,

Here is really all you need:

exten => 401,Hint,SIP/401

in the context that the watching phone is in...


Aaron Daniel wrote:
The polycoms have a buddy feature where you can "watch" a buddy.  From 
what I can tell, it sends a subscribe to the server, and only works if 
you're hinting the phone.  That's what was suggested I do since I want 
to be able to tell if someone's on the phone, and I've watched the sip 
debug as it boots up and it does in fact send a subscribe to the 
server for the extensions I want to watch.  The server's not really 
doing anything with it though, so I'm kinda lost on how this is going 
to work.  Sip debug doesn't show asterisk sending any information to 
the phone after it subscribes.


Aaron

On Mon, 3 Apr 2006, Kevin P. Fleming wrote:


Aaron Daniel wrote:

Ok, with the buddies, what "device" do you hint to?  The last line of
the phone?


I don't understand the question... the 'buddy' is effectively a
speed-dial, the same thing you would dial to call that person/extension.
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Re: [Asterisk-Users] definity g3 voicemail

[Sean Cook]

Yeah... I am doing that one now with a merlin system...

Sean

C F wrote:

Well, I did it using DTMF tones on analog channels, it's on the wiki.

On 3/27/06, Sean Cook <[EMAIL PROTECTED]> wrote:
  

My understanding is that the SMDI is a serial interface that passes data
about the call to the system for voicemail and pass MWI info back to the
avaya.  It is the definity side that I am clueless on...

C F wrote:


On 3/27/06, Sean Cook <[EMAIL PROTECTED]> wrote:

  

Is anyone using * to provide voicemail to a definity system?  I
understand with the new SMDI functionality in trunk that this will be
easier to provide some of the integration features.



I have done this on other Avaya systems, you got any idea what that
SMDI does? how it works?



  

Looking for some hints on the definity setup and anything on the SMDI
side.  Anyone with a working solutions would be great if I could pick a
brain.

Sean
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Re: [Asterisk-Users] definity g3 voicemail

[Sean Cook]

My understanding is that the SMDI is a serial interface that passes data 
about the call to the system for voicemail and pass MWI info back to the 
avaya.  It is the definity side that I am clueless on...


C F wrote:

On 3/27/06, Sean Cook <[EMAIL PROTECTED]> wrote:
  

Is anyone using * to provide voicemail to a definity system?  I
understand with the new SMDI functionality in trunk that this will be
easier to provide some of the integration features.



I have done this on other Avaya systems, you got any idea what that
SMDI does? how it works?


  

Looking for some hints on the definity setup and anything on the SMDI
side.  Anyone with a working solutions would be great if I could pick a
brain.

Sean
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[Asterisk-Users] definity g3 voicemail

[Sean Cook]

Is anyone using * to provide voicemail to a definity system?  I 
understand with the new SMDI functionality in trunk that this will be 
easier to provide some of the integration features. 

Looking for some hints on the definity setup and anything on the SMDI 
side.  Anyone with a working solutions would be great if I could pick a 
brain.


Sean
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[Asterisk-Users] Hints in Realtime

[Sean Cook]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
 
Do hints work in Realtime asterisk?  not finding much on the list
archives or anywhere else for that matter... I have tried using -1
priority as mentioned once or twice but no joy

Thought?
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=6gsq
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Re: [Asterisk-Users] Asterisk & Avaya Legend

[Sean Cook]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
 
Yeah... we went through the same thing... our problem was working with
asterisk it was "Oh cool!" then trying to make the legend work was "WTF!".

Our setup is on the back side... legend still connects via PRI to the
PSTN, but we have asterisk running as a tandem off a third PRI.  Many
of the challenges are the same.  I wish we had put the money up and
gotten a 4-port T1 card and done things the way you are.  We would
have caller id and our voicemail would be much more transparent.  We
had to put a channel bank with FXO to replace the Audix voicemail
system.  While it works better than the audix system, there are some
things that I wish it did that asterisk does natively.

Sean

Lacy Moore - Aspendora wrote:

> Thanks Sean!
>
> Fortunately, I don't think I will have to worry about passing
> extensions back and forth between Asterisk and the Legend. But,
> I'm glad to know that it is possible, if the situation arises. We
> don't have that many users. My concern is just making sure that
> the two can coexist for a while. We're going to use the savings
> from switching to the PRI to purchase the rest of the equipment,
> and I'll be keeping everyone connected to the Legend. Once we are
> able to start purchasing the IP phones, I'll move one company at a
> time over. During this transition, we shouldn't have to worry
> about extension to extension going through both.
>
> Light bulb just went off. Looking at this:
>
> [from-pstn] exten => 482,1,Dial(Zap/g0/482)
>
> So if I have all 10 digits being passed and someone placed a call
> to one of our DIDs, for example, 281-604-0532, the dial plan would
> look like:
>
> [from-pstn] exten => 2816040532,1,Dial(Zap/g0/2816040532)
>
> If so, I sure wish the Legend was that easy to setup! I'll
> probably have to get someone to set it up. The card came in
> yesterday for the Asterisk server, so I'll be able to start playing
> with things. I'm guessing I can send my own DID information to the
> Legend so that before the PRI even gets installed I can have
> everything set up and waiting.
>
> Honestly, working with Asterisk is one of the "Oh cool!" moments.
>
> On 3/23/06, *Sean Cook* <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>>
> wrote:
>

>
>> Now, here is what I'm not sure of at this moment. For the time
>> being, is it possible to just pass the PRI through the Asterisk
>> to the Legend? Will there by any type of dialplans or anything
>> that need to be created? Will it pass the DID information
>> through? I was trying to look at this from the perspective that
>> the Legend will be a channel bank, but I don't think that's a
>> correct assumption. I think this more closely fits tying two
>> Asterisk servers together with a T1, but I haven't been able to
>> find any info on this.
>
> basically you will receive the DID information and pass it straight
> through to the DS1
>
> [from-pstn] exten => 482,1,Dial(Zap/g0/482)
>
> It should be pretty straight forward... our setup is similar but
> the asterisk system is configured internally not between the telco
> and legend.
>
>> Can anyone offer any pointers, or maybe point out anything
>> obvious that I am missing? Or, even confirm that what I'm trying
>> to do is possible. I have the two port card on order and would
>> like to play with it before the PRI gets installed. I don't like
>> working in theory on this, I'd feel much better with the
>> equipment in hand, even without the PRI, I can still setup a VoIP
>> account and make sure that I can pass the call through the
>> Asterisk into the
> Legend.
>
> The only thing that I am not sure of is how the legend will be
> passing additional extension across the PRI so that you can dial a
> local extension that is on the Asterisk side... I suppose if you
> set up 4000-4999 in UDP and pass it on the PRI you should be pretty
> good to go, provided that you have did's for every extension set up
> within the legend. (even if they are fake did's not necessarily
> from the telco)
>
>
>> Assuming all this works, I think just having Asterisk in there
>> would solve one problem. It seems that I could set up a dial plan
>> so that if I dial 9 it would use the caller ID of one company,
> dial
>> 8 to use the caller ID of another, etc. On the user side, they
>> would have to dial 9 twice to place an outgoing call (since the
>> Legend requires a 9 for an outside line), but I think I could
>> also set the dialplan up so that if the number dialed is a
>> standard number, it would just us

Re: [Asterisk-Users] Problem with MeetMe Conference!!!

[Sean Cook]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
 
if you have an zaptel card installed and working... try to do a load
app_meetme.so and see what happens... if it loads successfully... you
should be able to conference also check your modules.conf and make
sure you don't have noload=>app_meetme.so

BJ Weschke wrote:

> On 3/24/06, serge messa <[EMAIL PROTECTED]> wrote:
>
>> Hi all
>>
>> I want to use conference in Asterisk. I configure a conference
>> room in meetme.conf (as conf => 600,1234) and extensions.conf as
>> (exten => 600,1,MeetMe(600,i,1234)) . When i call the extension
>> 600, i have the following message in the asterisk logs:
>>
>> WARNING[7758]: pbx.c:1688 pbx_extension_helper: No application
>> 'MeetMe' for extension (conference, 600, 1) == Spawn extension
>> (conference, 600, 1) exited non-zero on 'IAX2/1000-2' -- Hungup
>> 'IAX2/1000-2'
>>
>> I install the zaptel module with the ztdummy timer but the
>> problem still exist.
>>
>> How can i do to fix this problem?
>>
>
> Zaptel and ztdummy must be installed prior to building Asterisk so
> that Asterisk will build and install app_meetme. If you've done
> this after building Asterisk, try budiling it again now that you've
> installed Zaptel and app_meetme.so should now build and install.
>
> -- Bird's The Word Technologies, Inc. http://www.btwtech.com/
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Re: [Asterisk-Users] Stability of Asterisk with 2 x TDM400P cards (6 analogue lines)

[Sean Cook]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
 
I am currently running asterisk 1.0.9 on a system with 2 TDM400P...  I
have had fairly good success with it across the board... my only issue
is that I have monkeys who move stuff around and things get unplugged ;)

Jared Davison wrote:

> I would like to hear from anyone good or bad as what their
> experience has been in recent times with STABILITY of current
> builds of Asterisk and drivers for TDM400P.
>
> The sort of configuration is: 6 incoming POTS lines. ie. 2 TDM400P
> cards.
>
> I am not concerned with: price points, or the advantages or
> disadvantages of using POTS vs ISDN technology, but simply
> RELIABILITY & stability of the Asterisk system & associated
> interface hardware and drivers.
>
> Do people need to reboot their systems regularly?
>
> Thanks in advance.
>
>
> Jared
>
>
>
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Re: [Asterisk-Users] [OT] Polycom provisioning

[Sean Cook]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
 
Aaron,

I have this working quite well.  Are you using FTP? or TFTP...

We are using FTP for about 40 phones and it works like a champ.   For
each phone I have...

0004f2030925.cfg


then I have phone4701.cfg that contains all of the line information
and phone specific data
then the stock sip.cfg with the digitmap and global options


Sean



Aaron Daniel wrote:

> Does anyone have the polycom soundpoint ip's successfully remotely
> provisioning? I've got the phone pulling default configs, and it's
> downloading phone specific information, but it's not actually
> using that information. Any help would be appreciated :)
>
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Re: [Asterisk-Users] Asterisk & Avaya Legend

[Sean Cook]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
 

> Now, here is what I'm not sure of at this moment. For the time
> being, is it possible to just pass the PRI through the Asterisk to
> the Legend? Will there by any type of dialplans or anything that
> need to be created? Will it pass the DID information through? I
> was trying to look at this from the perspective that the Legend
> will be a channel bank, but I don't think that's a correct
> assumption. I think this more closely fits tying two Asterisk
> servers together with a T1, but I haven't been able to find any
> info on this.

basically you will receive the DID information and pass it straight
through to the DS1

[from-pstn]
exten => 482,1,Dial(Zap/g0/482)

It should be pretty straight forward... our setup is similar but the
asterisk system is configured internally not between the telco and legend.

> Can anyone offer any pointers, or maybe point out anything obvious
> that I am missing? Or, even confirm that what I'm trying to do is
> possible. I have the two port card on order and would like to play
> with it before the PRI gets installed. I don't like working in
> theory on this, I'd feel much better with the equipment in hand,
> even without the PRI, I can still setup a VoIP account and make
> sure that I can pass the call through the Asterisk into the Legend.
>
The only thing that I am not sure of is how the legend will be passing
additional extension across the PRI so that you can dial a local
extension that is on the Asterisk side...  I suppose if you set up
4000-4999 in UDP and pass it on the PRI you should be pretty good to
go, provided that you have did's for every extension set up within the
legend.  (even if they are fake did's not necessarily from the telco)

>
> Assuming all this works, I think just having Asterisk in there
> would solve one problem. It seems that I could set up a dial plan
> so that if I dial 9 it would use the caller ID of one company, dial
> 8 to use the caller ID of another, etc. On the user side, they
> would have to dial 9 twice to place an outgoing call (since the
> Legend requires a 9 for an outside line), but I think I could also
> set the dialplan up so that if the number dialed is a standard
> number, it would just use a generic caller ID number.

Right but the legend also provides UDP that will allow you to specify
a range for extensions to be passed back and forth... users that dial
9 and get an outside trunk will be able to match with your dialplan

[from-merlin]
exten => 9NXX,1,Dial(ZAP/g1/${EXTEN:-1})  ; where group=1 is your
telco trunk
exten => 4001,1,Dial(SIP/4001)



>
> I guess I'm still confused as whether Asterisk will pass the DID
> information onto the Legend. If someone could point me in a
> direction regarding this, I'd appreciate it. I haven't found
> anything, which makes me worried this isn't possible (whether
> because of technology or Asterisk).


All you need to know is the digits that are being passed and how they
are mapped in the legend... asterisk can mimic this however you want...


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[Asterisk-Users] RealTime Extensions

[Sean Cook]

-BEGIN PGP SIGNED MESSAGE-
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Ok... so I spent today getting realtime extensions working, which they
are (for the most part) and apart from forgetting to commit
transactions in postgres and trying to figure out why an extension
won't work, all is well.

The only problem that I am running into is "hint".  How the heck do I
put this in the dialplan?  I have tried assigning priority,app,appdata
as  -1,Hint,SIP/ respectively and no joy.

Does anyone have realtime with Hint working?

Sean
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Re: [Asterisk-Users] I'm FED UP with BroadVoice

[Sean Cook]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
 
I would venture to say that all ITSP suck... some just suck less...
It generally speaking boils down to that fact that internet
connectivity is never full reliable (from a consumer standpoint).
Sure if you want to cough up the money for a T1, you probably get
better phone service for $19.95/month... but that really doesn't make
a whole lot of sence.

Even connections from home back to an office that doesn't have much
load but is out on the internet is somewhat flakey because there is no
QOS.

Sean

Ronald Lewis wrote:

> After months of BroadVoice ignoring my trouble tickets for dropped
> calls, delayed termination, etc., I'm throwing in the towel. While
> they have credited $19.95 to my account, they refuse to credit
> anything more, despite ALL of the problems I've had. I feel the
> least they could do is credit the remaining $8.61 to my account,
> yet they won't.
>
> I haven't really been following up on porting between VoIP
> providers, but is there a remote chance I can save my phone number?
> I'd sure hate to change numbers again -- this has been a
> NIGHTMARE. Everyday, calls are dropping, and I'm calling people
> back 2 to 3 times to establish a decent connection.
>
> And their response (paraphrasing): "We've made the best effort to
> ensure your service is functional ... but there are some things
> beyond our control with VoIP." Not good enough! I had great service
> with Vonage, and the times I use VoipJet, it works perfectly!
>
> Thanks in advance for any pointers.
>
> Ronald Lewis Denver, Colorado http://www.ronaldlewis.com/interviews
>
>
>
> --
>
>
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[Asterisk-Users] High Density Analog

[Sean Cook]

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Is anyone using the Adit 600 with CMG g729 gateway?  We are trying to
come up with a solution for 600+ FXS campus and it appears to have the
highest port density of anything out there...

Any other thoughts?
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Re: [Asterisk-Users] Disappearing voicemail

[Sean Cook]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
 
Check to make sure your minimum message length is very short.  You
should be able to view this in the full log.

Sean

Phil Freed wrote:

> Asterisk 1.2, Fedora Core 4:
>
> When I leave a voicemail message, it writes the necessary files to
> the INBOX:
>
> msg.gsm msg.txt msg.wav msg.WAV
>
> When I hang up, the files are erased. There is no indication of
> anything untoward in the logs:
>
> -- x=0, open writing: [...]/INBOX/msg format: wav49, 0x99ce778
> -- x=1, open writing: [...]/INBOX/msg format: gsm, 0x99bed98
> -- x=2, open writing: [...]/INBOX/msg format: wav, 0x99cf5f0
> -- User hung up -- Hungup 'IAX2/phil-4'
>
> Is it possible that Asterisk is attempting to move them someplace
> that doesn't exist?
>
> Thanks as always for any insights you might provide.
>
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Re: [Asterisk-Users] Re: transfers/parked calls + polycom 501

[Sean Cook]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

No... did you get this from the sip.cfg or did you assume that the
default is there?  Asterisk will send a 404 back to the phone if the
entry does not exist but if it is just sending before you are
finished then there is a problem... what do you have the TimeOut set for?

Sean

sdgesa gaeharth wrote:
> [2-9]11|0T|011xxx.T|[0-1][2-9]x|[2-9]x|[2-9]xxxT
> 
> I have never had this changed on any phones. This should be the default.
> 
> Does this value change based on what extensions are available to the
> phone via asterisk extensions file? In other words, does asterisk tell
> the phone what extensions are available and then the polycoms change the
> map themselves?
> 
> thanks
> 
> 
> 
> 
> */Sean Cook <[EMAIL PROTECTED]>/* wrote:
> 
> This sounds like a digitmap issue... from your sip.cfg what is your
> digitmap set to?
> 
> Sean
> 
> sdgesa gaeharth wrote:
>> I am using the latest firmware and bootrom and this is a problem with
>> all 12 polycom 501s that we have in the office. If I want to transfer
>> to 1005 for example while on the p hone with the original caller,
> I press
>> transfer -> blind -> type "1", "0" then the phone clears the display
>> and the transfer fails. It only allows me to dial the first two digits
>> of the extension I want to transfer to. It even happens when I dial
>> local sip to local sip, not just sip to pstn. This seems like a config
>> mistake I made.
> 
>> thanks
> 
> 
>> */Noah Miller /* wrote:
> 
>> Hi -
> 
>> > I am not sure what I did but blind transfers do not work. The
>> Polycom does
>> > not allow me to dial the extension of the person I want to
>> transfer to after
>> > I hit:
>> >
>> > transfer -> blind
> 
>> I would strongly suggest getting the latest firmware, and using the
>> sample
>> configuration files with that firmware to set up your phone. Th is
> SHOULD
>> work. If it still does not work after doing this, there may be a
>> hardware
>> issue with your phone.
> 
>> - Noah
> 
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>> http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> 
> 
>> Brings words and photos together (easily) with
>> PhotoMail
> 
>> - it's free and works with Yahoo! Mail.
> 
> 
> 
> 
> 
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> 
> Yahoo! Mail
> Use Photomail
> <http://pa.yahoo.com/*http://us.rd.yahoo.com/evt=38867/*http://photomail.mail.yahoo.com>
> to share photos without annoying attachments.


> 

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Re: [Asterisk-Users] Re: transfers/parked calls + polycom 501

[Sean Cook]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

This sounds like a digitmap issue...  from your sip.cfg what is your
digitmap set to?

Sean

sdgesa gaeharth wrote:
> I am using the latest firmware and bootrom and this is a problem with
> all 12 polycom  501s that we have in the office. If I want to transfer
> to 1005 for example while on the phone with the original caller, I press
> transfer -> blind -> type "1",  "0" then the phone clears the display
> and the transfer fails. It only allows me to dial the first two digits
> of the extension I want to transfer to. It even happens when I dial
> local sip to local sip, not just sip to pstn. This seems like a config
> mistake I made.
> 
> thanks
> 
> 
> */Noah Miller <[EMAIL PROTECTED]>/* wrote:
> 
> Hi -
> 
> > I am not sure what I did but blind transfers do not work. The
> Polycom does
> > not allow me to dial the extension of the person I want to
> transfer to after
> > I hit:
> >
> > transfer -> blind
> 
> I would strongly suggest getting the latest firmware, and using the
> sample
> configuration files with that firmware to set up your phone. This SHOULD
> work. If it still does not work after doing this, there may be a
> hardware
> issue with your phone.
> 
> - Noah
> 
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> 
> 
> 
> Brings words and photos together (easily) with
> PhotoMail
> 
> - it's free and works with Yahoo! Mail.
> 
> 
> 
> 
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Re: [Asterisk-Users] voicetronix and [EMAIL PROTECTED]

[sean cook]

The channels are VPB/X 

On Fri, 2006-03-10 at 17:35 -0500, Chuck Fletcher wrote:
> Any guidance on how to get my openline4 to get recognized by [EMAIL PROTECTED]
> 
> I've got my vpb drivers running, but not sure how to add it as a trunk, 
> should it be via zap? or is there another way?
> 
> Thanks,
> 
> Chuck
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[Asterisk-Users] Voicetronix OpenSwitch / Sangoma Analog Card

[Sean Cook]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
 
I am looking to trade for a new or used  Sangoma Analog A200 card with
echo cancellation.  I have finished my testing with the OpenSwitch
card and want to test with the sangoma.  Anyone out there looking to
do the same?

Sean
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Re: [Asterisk-Users] IVR woes

[Sean Cook]

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Hash: SHA1
 
If memory servers me correctly DigitTimeout and ResponseTimeout are
depricated...

try:

exten => s,13,Set(TIMEOUT(digit)=5)
exten => s,14,Set(TIMEOUT(response)=30)


Sean

Robert P. McKenzie wrote:

> Hello all. I'm having a problem debugging an IVR I'm building. I
> can't see any reason this shouldn't be working. Firstly the
> asterisk version is:
>
> Asterisk SVN-trunk-r7230 built by root @ localhost.localdomain on a
> i686 running Linux on 2006-02-17 22:44:48 UTC
>
> Basically the problem is this. While the playbacks are happening
> you can push any one of the options and to happily goes off and
> does it. However, if you wait until the messages stop playing back
> it just hangs up with the error at the bottome of this message.
>
> Any help in finding a solution to this werid problem would be
> greatly appreciated.
>
> The IVR context and console logs are:
>
> [lcl-ivr-main]
> ;;
> ; ; This is the main number IVR menu system ;
> ;;
>
> exten => s,1,Answer exten => s,2,NoOp exten => s,3,NoOp exten =>
> s,4,NoOp exten => s,5,Wait(1) exten =>
> s,6,Background(LCL/prompt-00) exten =>
> s,7,Background(LCL/prompt-01) exten =>
> s,8,Background(LCL/prompt-02) exten =>
> s,9,Background(LCL/prompt-03) exten =>
> s,10,Background(LCL/prompt-04) exten =>
> s,11,Background(LCL/prompt-05) exten =>
> s,12,Background(LCL/prompt-09) exten => s,13,DigitTimeout,5 exten
> => s,14,ResponseTimeout,30
>
> ; exten => _1,1,Background(LCL/prompt-20) ; Sales exten =>
> _1,2,Dial(${SALES}|40|trwo) exten => _1,3,Voicemail([EMAIL PROTECTED])
> exten => _1,103,Voicemail([EMAIL PROTECTED]) exten => _1,4,Hangup
>
> ; exten => _2,1,Background(LCL/prompt-30) ; Support exten
> => _2,2,Dial(${SUPPORT}|40|trwo) exten =>
> _2,3,Voicemail([EMAIL PROTECTED]) exten =>
> _2,103,Voicemail([EMAIL PROTECTED]) exten => _2,4,Hangup
>
> ; exten => _3,1,Background(LCL/prompt-40) ; Accounts exten
> => _3,2,Dial(${ACCOUNTS}|40|trwo) exten =>
> _3,3,Voicemail([EMAIL PROTECTED]) exten =>
> _3,103,Voicemail([EMAIL PROTECTED]) exten => _3,4,Hangup
>
> ; exten => _4,1,Background(LCL/prompt-50) ; Reception exten
> => _4,2,Dial(${RECEPTION}|40|trwo) exten =>
> _4,3,Voicemail([EMAIL PROTECTED]) exten =>
> _4,103,Voicemail([EMAIL PROTECTED]) exten => _4,4,Hangup
>
> ; exten => _5,1,NoOp ; Dial Extension
> ; exten => _6,1,Goto(lcl-ivr-menu,s,7) ; Play menu again
> ; exten => i,1,Goto(lcl-ivr-menu,s,7) ; Return to menu
> after a time out exten => t,1,Goto(lcl-ivr-menu,s,7) ;
> Return to menu after a time out
>
>
> Here is he asterisk console output:
>
> -- Accepting AUTHENTICATED call from xx.xx.xx.xx:
>> requested format = unknown, requested prefs = (), actual format =
>> ulaw, host prefs = (ulaw|alaw|gsm), priority = mine
> -- Executing Goto("IAX2/rob-5", "lcl-ivr-main|s|1") in new stack --
> Goto (lcl-ivr-main,s,1) -- Executing Answer("IAX2/rob-5", "") in
> new stack -- Executing NoOp("IAX2/rob-5", "") in new stack --
> Executing NoOp("IAX2/rob-5", "") in new stack -- Executing
> NoOp("IAX2/rob-5", "") in new stack -- Executing Wait("IAX2/rob-5",
> "1") in new stack -- Executing BackGround("IAX2/rob-5",
> "LCL/prompt-00") in new stack -- Playing 'LCL/prompt-00' (language
> 'en') -- Executing BackGround("IAX2/rob-5", "LCL/prompt-01") in new
> stack -- Playing 'LCL/prompt-01' (language 'en') -- Executing
> BackGround("IAX2/rob-5", "LCL/prompt-02") in new stack -- Playing
> 'LCL/prompt-02' (language 'en') -- Executing
> BackGround("IAX2/rob-5", "LCL/prompt-03") in new stack -- Playing
> 'LCL/prompt-03' (language 'en') -- Executing
> BackGround("IAX2/rob-5", "LCL/prompt-04") in new stack -- Playing
> 'LCL/prompt-04' (language 'en') -- Executing
> BackGround("IAX2/rob-5", "LCL/prompt-05") in new stack -- Playing
> 'LCL/prompt-05' (language 'en') -- Executing
> BackGround("IAX2/rob-5", "LCL/prompt-09") in new stack -- Playing
> 'LCL/prompt-09' (language 'en') -- Executing
> DigitTimeout("IAX2/rob-5", "5") in new stack -- Set Digit Timeout
> to 5 -- Executing ResponseTimeout("IAX2/rob-5", "30") in new stack
> -- Set Response Timeout to 30 == Auto fallthrough, channel
> 'IAX2/rob-5' status is 'UNKNOWN' -- Hungup 'IAX2/rob-5'
>
> That hangup is Asterisk just dumping out..
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Re: [Asterisk-Users] Merlin Magix Integration

[Sean Cook]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
 
I actually have this working... on a merlin legend R7

zapata.conf
; turn off caller id otherwise it hangs...
usecallerid=no
usecallingpres=no
callwaitingcallerid=no

; drop into the vm context
relaxdtmf=yes
context=from-vm
group = 4
signalling = fxs_ks
channel=>29-32


extensions.conf
[from-vm]
exten => _#00#XXX##,1,NoOP(Directly dialed 770 from extension ${EXTEN})
exten => _#00#XXX##,2,VoicemailMain()
exten => _#00#XXX##,3,Hangup()

; auto attendant coverage
exten => _#01#XXX##,1,NoOP(outside line ${EXTEN} ${CALLERIDNUM})
exten => _#01#XXX##,2,GoTo(auto-attendant,s,1)

; voicemail coverage for extension

exten => _#02#XXX#XXX#,1,NoOP(extention to extension for coverage
${EXTEN})
;exten => _#02#XXX#XXX#,2,Set(CALLERID(number)=${EXTEN:4:3})
exten => _#02#XXX#XXX#,2,AGI(setcallerid.agi,${EXTEN:4:3})
exten => _#02#XXX#XXX#,3,Voicemail(u${EXTEN:8:3})
exten => _#02#XXX#XXX#,4,Hangup()

; did/outside line coverage for voicemail calls
exten => _#03##XXX#,1,NoOP(did coverage for extension ${EXTEN})
exten => _#03##XXX#,2,NoOp(might set callerid here)
exten => _#03##XXX#,3,Voicemail(u${EXTEN:5:3})
exten => _#03##XXX#,4,Hangup()

exten => s,1,Answer
exten => s,2,WaitExten(1)
exten => t,1,GoTo(auto-attendant,s,1)
exten => o,1,Playback(pbx-transfer)
exten => o,2,Flash()
exten => o,3,SendDTMF(${OPERATOR})
exten => o,4,Hangup();



Darren Ellis wrote:

> Hi List,
>
> Merlin Magix hardware v02
>
> I'm trying to get asterisk to act as a voicemail server for a
> lucent merlin magix PBX that we purchased used. We have 4 FXO
> channels between the two PBXs on a Sangoma A200 card. The 770
> dialgroup is working properly, in that calls to 770 are answered by
> Asterisk. The magix is sending mode codes in the format #XX#XXX#,
> where the 2nd block of digits is the calling extension. I'm
> stripping off the unneeded pound signs and digits, and calling
> voicemailmain. The problem I'm having is that the asterisk is
> starting to play vm-password and then interrupts immediately and
> errors with an incorrect password. Then it works normally. Below
> is the relevant asterisk config and the asterisk log. Zaptel is
> configured to start inbound calls in the inbound context. The
> voicemail accounts and sip accounts are all in the default context.
>
>
> Asterisk log -- Starting simple switch on 'Zap/3-1' Mar 9 10:26:35
> NOTICE[4211]: chan_zap.c:6063 ss_thread: Got event 18 (Ring
> Begin)... -- Executing Answer("Zap/3-1", "") in new stack --
> Executing WaitExten("Zap/3-1", "1") in new stack == CDR updated on
> Zap/3-1 -- Executing NoOp("Zap/3-1", "#00#219#") in new stack --
> Executing Set("Zap/3-1", "[EMAIL PROTECTED]") in new stack --
> Executing NoOp("Zap/3-1", "[EMAIL PROTECTED]") in new stack -- Executing
> VoiceMailMain("Zap/3-1", "[EMAIL PROTECTED]") in new stack -- Playing
> 'vm-password' (language 'en') -- Incorrect password '' for user
> '219' (context = default) -- Playing 'vm-incorrect' (language 'en')
> -- Playing 'vm-password' (language 'en') ||| Caller hangs up here
> ||| Mar 9 10:26:41 WARNING[4211]: app_voicemail.c:4998
> vm_authenticate: Unable to read password -- Hungup 'Zap/3-1'
>
> extensions.conf [inbound] exten => s,1,Answer() exten =>
> s,2,WaitExten(1) ; Allow time for mode code digits to come
> across
>
> ; The following extensions grab the mode code ; coming from the
> Avaya PBX and route the ; call appropriately via the Voicemail() ;
> and VoiceMailMain() apps. ; ; someone pressed vmail check ;
> #00#243# exten => _#XX#XXX#,1,noop(${EXTEN}) exten =>
> _#XX#XXX#,2,Set(CVAR=${EXTEN:4:[EMAIL PROTECTED]) exten =>
> _#XX#XXX#,3,NoOp(${CVAR}) exten =>
> _#XX#XXX#,4,VoicemailMain(${CVAR}) ;exten =>
> _#XX#XXX#,2,VoicemailMain(${EXTEN:4:[EMAIL PROTECTED]) exten =>
> _#XX#XXX#,5,Hangup()
>
>  As can
> be seen, I've tried calling voicemailmain with the ${EXTEN:4:3}
> digit stripping as part of the command, and also I've tried moving
> the digit stripping to a variable. I'd very much appreciate any
> help you folks can offer.
>
> Thanks much.
>
> Darren Ellis
>
>
>
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Re: [Asterisk-Users] Real Time Asterisk

[Sean Cook]

Yes you do need unixODBC before you compile asterisk.  Once you have
installed unixODBC , asterisk will compile and offer you the following
modules:

cdr_odbc.so  
res_config_odbc.so  
res_odbc.so

res_odbc.conf and cdr_odbc.conf are the related config files...

Sean

On Thu, 2006-03-09 at 11:57 -0300, Fernando Lujan wrote:
> using

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Re: [Asterisk-Users] Real Time Asterisk

[Sean Cook]

I am using the odbc set up with postgres right now and it works fine.  

http://www.voip-info.org/wiki/view/Asterisk+RealTime+PostgreSQL

has most of the info to get you running.  As for meetme, I took the
app_cbmysql stuff for webmeetme and rewrote it for postgres.  I am still
testing it, but it seems to work great right now...

if dan is out there anywhere... I would like to help move the webmeetme
part of this to db independant and make it so it can run
register_globals off :)

Sean

On Thu, 2006-03-09 at 09:09 -0300, Fernando Lujan wrote:
> production.

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Re: [Asterisk-Users] Receiving Multiple calls on asterisk at home

[Sean Cook]

To add to the other post... aah or amp actually has a DB that contains
call waiting information.  It may have the default setup such that call
waiting is disabled.  You should be able to dial *70 and enable it.

Sean

On Tue, 2006-03-07 at 11:33 -0700, Rolf Brusletto wrote:
> All - I've been muddling around with this for a few days now.. and I'm 
> trying to figure out why I am not receiving more than one phone call on 
> each polycom 501 phone. I can make more than one phone call out, but not 
> receive another one in, while on a call. Has anybody seen this behaivior 
> before, or is there something simple in the config i'm missing, like.. 
> maxcalls.. or something.
> 
> Thanks!
> 
> Rolf Brusletto
> 
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Re: [Asterisk-Users] Echo Cancelation on TE110P

[Sean Cook]

In theory I would say I agree how ever in practice... I have a PBX
(Merlin Legend) that I am connected to via PRI (10 foot pre-fab'ed
cable) and I get intermittent echo on the voip side.   There is nothing
in between * and the PBX...

sean

On Fri, 2006-03-03 at 13:42 -0600, Michael Sampson wrote:
> It is my understanding that when you hear echo the problem is on the
> other end. So if a caller complains they hear echo that is something
> you should be dealing with, but if you hear echo that is the phone
> companies fault. Now with a normal phone, the phone company will only
> echo cancel long distance calls. For local calls the latency is not
> high enough to matter. But with VOIP the added latency creates echo
> even for local calls. I think the reason you hear it on some numbers
> and not others is that the phone companies are doing echo cancel on
> some of those calls and not on others.
> Michael Sampson
> Information Systems Manager
> Customer Contact Services
> [EMAIL PROTECTED]
> 952-936-4000
> 
> 
> Kerry Garrison wrote: 
> > On a 55 station install onto a Cox PRI with a TE110P (Polycom 501
> > phones) a few users are complaiining about echo. According to the
> > users, the echo seems to be phone number dependant. They claim that
> > certain phone numbers have echo while others dont. Are there any
> > tuning parametes like there is for a TDM400 card? 
> >  
> > Kerry Garrison
> > Director of Technical Services
> > Tech Data Pros - Orange County's Mobile IT Service Provider
> > (949) 502-7819 x200 - [EMAIL PROTECTED]
> > http://www.techdatapros.com 
> >  
> > 
> > 
> > 
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Re: [Asterisk-Users] Sip Realtime Configs Samples with MySQL

[Sean Cook]

I haven't tried sip yet... been finishing voicemail, but the principal
is the same.


res_mysql.conf

[general]
dbhost = localhost
dbname = asterisk
dbuser = someuser
dbpass = somepass
dbport = 3306
dbsock = /var/run/mysqld/mysqld.sock

extconfig.conf
voicemail => mysql,asterisk,voicemail
; i would assume that sip would be 
sippeers => mysql,asterisk,sip
sipusers => mysql,asterisk,sip

Table

CREATE TABLE `sip` ( 
 `id` int(11) NOT NULL auto_increment,
 `name` varchar(80) NOT NULL default '',
 `accountcode` varchar(20) default NULL,
 `amaflags` varchar(13) default NULL,
 `callgroup` varchar(10) default NULL,
 `callerid` varchar(80) default NULL,
 `canreinvite` char(3) default 'yes',
 `context` varchar(80) default NULL,
 `defaultip` varchar(15) default NULL,
 `dtmfmode` varchar(7) default NULL,
 `fromuser` varchar(80) default NULL,
 `fromdomain` varchar(80) default NULL,
 `fullcontact` varchar(80) default NULL,
 `host` varchar(31) NOT NULL default '',
 `insecure` varchar(4) default NULL,
 `language` char(2) default NULL,
 `mailbox` varchar(50) default NULL,
 `md5secret` varchar(80) default NULL,
 `nat` varchar(5) NOT NULL default 'no',
 `deny` varchar(95) default NULL,
 `permit` varchar(95) default NULL,
 `mask` varchar(95) default NULL,
 `pickupgroup` varchar(10) default NULL,
 `port` varchar(5) NOT NULL default '',
 `qualify` char(3) default NULL,
 `restrictcid` char(1) default NULL,
 `rtptimeout` char(3) default NULL,
 `rtpholdtimeout` char(3) default NULL,
 `secret` varchar(80) default NULL,
 `type` varchar(6) NOT NULL default 'friend',
 `username` varchar(80) NOT NULL default '',
 `disallow` varchar(100) default 'all',
 `allow` varchar(100) default 'g729;ilbc;gsm;ulaw;alaw',
 `musiconhold` varchar(100) default NULL,
 `regseconds` int(11) NOT NULL default '0',
 `ipaddr` varchar(15) NOT NULL default '',
 `regexten` varchar(80) NOT NULL default '',
 `cancallforward` char(3) default 'yes',
 `setvar` varchar(100) NOT NULL default '',
 PRIMARY KEY  (`id`),
 UNIQUE KEY `name` (`name`),
 KEY `name_2` (`name`)
) TYPE=MyISAM ROW_FORMAT=DYNAMIC; 



On Thu, 2006-03-02 at 16:10 -0500, [EMAIL PROTECTED] wrote:
> Guys,
> 
> I'm having a hellava time getting realtime to work, focused on sipusers right 
> now, followed the wiki and other examples but still no luck.  Using mysql on 
> a seperate server, asterisk actually sees the database and can poll the table 
> "realtime load sipusers" at the cli but asterisk realtime engine is no 
> pulling the user info.  I'm using 1.2.4 stable and have the database info in 
> sip.conf, extconfig.conf and res_mysql.conf.  Can anyone using mysql send me 
> sample configs and some insight to getting this going?
> 
> Thanks.
> 
> JR
> 
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Re: [Asterisk-Users] web meetme instructions

[Sean Cook]

First things first... use the latest version... (that I know of)

http://www.fitawi.com/Asterisk/


second... which part are you having problems with?  The web piece? or
the app_cbmysql?

For the app_cbmysql, I have found that the easiest way to work with it
is to incorperate it into asterisk-addons...

copy app_cbmysql.c into /usr/src/asterisk-addons-1.2.x

add MODS+=app_cbmysql.so after MODS=format_mp3...
make && make clean

The web portion is pretty straight forward... just make sure you have
register_global=On in php.ini.

That being said... I am personally not a big fan of the web portion of
the interface ;)  I have written my own that allows users to create
there own based on their voicemail login.

Sean


On Fri, 2006-03-03 at 07:31 -0600, Jordan Novak wrote:

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Re: [Asterisk-Users] Zoom 5801 problems with *

[Sean Cook]

But even the FXO -> voip bridging is lacking... you basically dial in
and it answers and provides dial tone for you to dial out your VoIP
service.  

It doesn't provide incoming pots termination except to the FXS port.

Sean

On Wed, 2006-03-01 at 01:46 -0800, [EMAIL PROTECTED] wrote:
> On Tue, 28 Feb 2006, Martin Joseph wrote:
> > On Feb 28, 2006, at 2:35 PM, [EMAIL PROTECTED] wrote:
> >> On Tue, 28 Feb 2006, Cory Andrews wrote:
> >>> Here is a link to some additional resources which may be helpful in 
> >>> configuring the 5801 and other Zoom products
> >>> http://www.zoom.com/salessupport/index.php?pd=ATA_Documentation/
> >> I just found out, the 5801 does not support voip to FXO bridging. Only FXS 
> >> to FXO bridging is supported.
> > That's not an FXO then.
> 
> It is an FXO, it can do FXO->voip bridging. Just not the other way round 
> yet. I guess it's planned for a future firmware revision, like t.38.
> 
> -Dan
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Re: [Asterisk-Users] Asterisk Web-Based Voicemail?

[Sean Cook]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

what about ARI, it gives web based access to the voicemail and is pretty
good at it... the default vmail.cgi is probably not the best as it has a
gaping security hole that allows anyone to listen to anyone elses
messages :)

Sean

Martin Joseph wrote:
> 
> On Feb 26, 2006, at 12:57 AM, Alexander Burke wrote:
> 
>> Hello, list!
>>
>> After Googling and checking out the voip-info wiki, I haven't had much
>> luck in locating a decent web-based voicemail system for Asterisk to
>> check your VM while you're away from the office without using a phone.
>>
>> Can anyone make any recommendations for such packages/applications?
> 
> I like the emailing of messages.  I email them directly to an
> account(IMAP) that has a webmail access, then I can view and listen from
> anywhere.
> 
> This also creates a "backup" of the voicemail messages on a separate
> drive, which I also see as a positive.
> 
> my 2c (us)
> Marty
> 
> 
> 
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Re: [Asterisk-Users] mpg123 alternative?

[sean cook]

Just to through another hat in the ring... I use madplay for mp3s...

[default]
mode=custom
directory=/var/lib/asterisk/mohmp3
application=/usr/bin/madplay -Q -o raw:- --mono -R 8000 -a -12


On Thu, 2006-02-23 at 15:23 -0600, Aaron Daniel wrote:
> I'd suggest using the format_mp3 program that's included in 
> asterisk-addons.  We switched to that after mpg123 wouldn't compile on 
> our newer 64bit machines, and it works like a charm, and you don't have 
> to change anything.
> 
> Aaron
> 
> Rich Adamson wrote:
> > Been using mpg123 for moh for the last two years or so. However, when
> > I have * config errors, often times get a endless stream of console
> > messages and need to kill the two mpg123 processes.
> > 
> > Is there an alternative to mpg123 that eliminates that issue?
> > 
> > I see references in musiconhold.conf relative to madplay, native file
> > format, asterisk-addons, etc. Not sure why the asterisk-addon approach
> > hasn't been moved into trunk, or if madplay is a better choice on this
> > fc3 trunk box.
> > 
> > Any suggestions?
> > 
> > 
> > 
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[Asterisk-Users] Pickup call on Hold

[Sean Cook]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Is it possible to pickup a call that is on hold on another extension?
Does anyone have any magic they can share on this topic?

I am struggling to teach call parking at a local shop where we installed
*.  It would simplify my life so much if they could just put the call on
hold and pick it up on another line.

Sean
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Re: [Asterisk-Users] auto provision of IP501 polycom

[Sean Cook]

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I have actually modified AMP to store the mac address and auto build the
 phone.cfg and 0004XXX.cfg files for ftp.  I use the default
username and password for the phones, so litterally all you do is plug
them in...

I will put together a patch and documentation later tonight...

Sean

Damon Estep wrote:
> Would you mind sending a sample config file?
> 
>  
> 
> Are you able to set the passwords (user and admin)?
> 
>  
> 
> Any ideas on ftp or https vs. tftp for better security?
> 
>  
> 
>  
> 
>  
> 
> 
> 
> *From:* [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] *On Behalf Of *Wojciech
> Tryc
> *Sent:* Thursday, February 23, 2006 6:15 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* RE: [Asterisk-Users] auto provision of IP501 polycom
> 
>  
> 
> Damon,
> 
> I have no problem provisioning 501s through tftp. The tftp address is
> distributed via dhcp.
> 
> Thx,
> 
> Wojtek
> 
>  
> 
> 
> 
> *From:* Damon Estep [mailto:[EMAIL PROTECTED]
> *Sent:* Thursday, February 23, 2006 8:09 AM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [Asterisk-Users] auto provision of IP501 polycom
> 
>  
> 
> Has anyone been able to get the IP501 to discover the FTP server IP
> address (via dhcp or dns) and download 100% of the config from a
> provisioning server?
> 
>  
> 
> We are still having to touch each unit to enter the ftp server address
> and password, as well as set many of the options that will not take from
> the config file.
> 
>  
> 
> Have a sample config file you are willing to share?
> 
>  
> 
> What is required in the way of dhcp options or dns entries to get the
> polycom to discover the ftp boot server?
> 
>  
> 
> What about changing default passwords via ftp?
> 
> 
> 
> 
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Re: [Asterisk-Users] Asterisk behind Centrex

[Sean Cook]

I believe that Centrex is ISDN correct?

Sean

On Tue, 2006-02-21 at 04:55 -0800, Dovid Bender wrote:
> I do not know a lot about centrex but I know that most
> PBX's support POTS lines (usually for faxing). You can
> have them switch over the lines that they send you to
> pots and then you can plug the lines in to a TDM400P.
> 
> Regards,
> Dovid
> 
> --- Devin Heckman <[EMAIL PROTECTED]> wrote:
> 
> > Hi,
> > 
> > I'm looking at setting up an Asterisk PBX in our
> > office, which gets its
> > phone lines (digital signaling, analog voice) from
> > the main campus,
> > which uses Centrex.
> > 
> > Does anyone know if this falls under analog or
> > digital for hardware
> > buying? I was looking at getting a Digium
> > TDM-series, but apparently our
> > lines aren't pots (due to the digital signaling).
> > 
> > Could someone enlighten me a bit?
> > 
> > Thanks a bunch.
> > 
> > 
> > Devin Heckman
> > University of California, Berkeley
> > RSSP-IT Residential Computing
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Re: [Asterisk-Users] Multiple TDM400P's in a single machine

[Sean Cook]

Same setup with two TDM400 (8FXO) running for over a year.

On Tue, 2006-02-21 at 01:37 +0100, Thomas Artner wrote:
> Am Tuesday 21 February 2006 00:24 schrieb Marc Archer:
> > Hi All,
> >
> >
> >
> > Can someone give me a definite answer as to wether or not you can
> > reliably run multiple TDM400P's in the same machine?
> >
> > I need 4 x FXO and 4 x FXS to connect to both the PSTN and existing key
> > system, but I have seen several threads suggesting that this is not a
> > supported configuration
> >
> >
> 
> i have two tdm400p's  (2xFXO, 6xFXS) in one desktop machine used as asterisk 
> server for a small office (so the pc hardware is nothing special).
> This configuration is running since two weeks without any problems!
> 
> 
> 
> >
> > Thanks,
> >
> >
> >
> > Marc.
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