date:20060201

I think they have a 1-800 number so you might be right.

But the important question is still - will Asterisk support this?

PaulH

> Alexander Lopez <[EMAIL PROTECTED]> wrote:
> 
They are using ANI instead of CallerID. If they have an 800 number thya
have the right to know who is calling them because they are paying for
the call.

the *ANI*DNIS* format is known as Feature Grooup D.

Alex

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe
Pukepail
Sent: Wednesday, February 01, 2006 3:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Blocked Callerid

Do they have an 800 number?  If so perhaps their 800 number
provider is doing it via DTMF.  Search around on the internet, I believe
the standard format for the DTMF is *CALLERID*CALLEDNUMBER* (or perhaps
reversed). 

On 2/1/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]>
wrote: 

I have been discussing an asterisk solution with a
company that has a custom written dialogic based solution.

The issue is that their dialogic solution can read
callerid from incoming calls, even if the callerid is blocked.
I have read before that Asterisk can do this, and they
want me to make sure that their new system will be able to do this.

A quick poke around inside the zaptel source code was
unproductive...

Any ideas?

PaulH

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[Asterisk-Users] RE: 5, 000 concurrent calls system rollout question

2006-02-01 Thread John Todd




Signate sells a single server that can get you to the call volumes you need.

Paul Mahler
[EMAIL PROTECTED]
www.signate.com


[snip]

Past conversations on this topic have generated quite a bit of 
controversy within the Asterisk development community, both publicly 
here on the list forums as well as in quite a few more quiet 
discussions with people who often do not post but have extensive 
operational experiences with Asterisk (most of whom monitor the -dev 
list and whose replies will be suited to that audience.)


The subject of load on a single chassis is still the most contentious 
issue to date.  The Signate numbers of >5000 calls per chassis with 
RTP are impressive, and there are others who claim more vaguely of 
1000, 2000, or more calls into a single P4 server (with or without 
media.)  Others say that there are inherent limits in the Asterisk 
code which prevent more than ~500 calls from being processed with RTP 
at any one time.  Opterons, FreeBSD, custom Linux loads, Solaris, and 
other operating systems or hardware have been offered as the magic 
bullets to increase call volumes.  Who knows? (1)  I will say that 
extraordinary claims demand extraordinary evidence, which has been 
pretty thin.  I believe that most large call processing facilities 
still run on distributed systems of some type, as was described in 
the primary thread of this discussion on -users. (2)


I know that there are some projects towards testing Asterisk more 
rigorously to determine these numbers.  However, I would suggest that 
the community at large could benefit from a more open examination of 
high-end system claims immediately than these (better) long-term 
tests which are progressing slowly (if at all.)  Let's just look at 
the "maximum" numbers.  Running a big system? Selling a big system? 
Tell us about it, in detail.  What are the limits that have been hit? 
Be specific.  I keep seeing hand-waving, but no programmers have come 
forward to say "It won't work because of the way X is implemented in 
the file blah.c or libFOO."


To make a bad analogy:  I don't want to see the street rods; I just 
want to see the top-fuel, rocket-powered dragsters on the line.  Any 
takers?  It sounds like Signate has a contender, but quite a few 
people have said that it's impossible without serious modifications 
to the code.  Others have claimed (publicly or privately) that they 
can match those numbers on different hardware.


Here are the criteria:
  - Any O/S
  - An unmodified version of Asterisk from SVN (or CVS)
  OR patches must be available for inspection, as per the GPL
  OR you must be a Digium license-holder (patches can be secret)
  - All calls are IAX2 or SIP (both in and out)
  - No transcoding of any type is required
  - All calls are G.711, 20ms OR 30ms packet size

Documentation:
  - All O/S documentation, kernel tricks, modules, hacks, patches, or 
configuration elements should be documented, but proprietary 
information need not be divulged if that is deemed "secret"

  - Testing method must be reasonably documented
  - Dialplans must be included
  - SIP.conf files must be included
  - All hardware must be fully described (part numbers required)

TEST #1:
   All media must be handled by the server.  This is for both legs of 
the call.  The "canreinvite=no" for SIP and "notransfer=yes" in IAX2 
must be set for all calls.


TEST #2:
   Media may or may not be handled by the server.  Native transfers 
should be allowed in both IAX2 and/or SIP.



(1) I have heard various people saying that it is "impossible" for 
Asterisk to handle a large number of calls due to architectural 
issues (no, it's not just from the people that you'd "expect" to hear 
this from.)  I've not been able to validate this one way or the other 
recently.  I am interested to hear what the developer community has 
as a comment on this topic.  I have an Empirix Hammer system at my 
company, but honestly I just don't have the time to set it up to do 
testing due to day job time constraints...


(2) There are so many ways to spread calls across an Asterisk array 
it makes my head spin, but the question STILL comes down to "how many 
calls can a single chassis handle?"  Even in a farm of servers, there 
has to be a numerator in that ratio.


JT

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Re: [Asterisk-Users] No audio? Update your Asterisk

2006-02-01 Thread Steve Gladden

FIXED!!!

Finally got some free time to really look into it while not rusing around
with work.

The problem boiled down to this:

IN sip.conf:

 externip = X.X.X.X ;Outside address
  localnet = 10.73.73.133 ;Inside address
   localmask = 255.255.255.0 ;Inside subnet

This worked for months & months & months  (since mid 2005) with no
problems for me.

I do remember seeing the warnings about changing it to the new format
quite awhile back but forgot about it.

What I had to do was simple:

Changed it to the new format --

externip = X.X.X.X ;Outside address
localnet = 10.73.73.0/255.255.255.0 ;Inside address

Now everything works! and I have working sip to sip audio once again!
The world smiles on my RTP packets once again.

Was strange how it all failed on Jan, 25 along with the 'time bomb'
failure/problem that everyone else had.

Seems that this had something to do with it!

Simple fix but a major hair pulling experience to track it down in spare
time. :-)

Thanks for the help and I hope I didn't waste your time too badly here.


Steve


































































> Good question,
> But the answer is no.
>
> I have went through the trouble to make sure that all traces of
> other asterisk libraries/modules, config files & excutables
> are removed from the system before compiling running & testing
> anything.
>
> I am also being sure to unload ztdummy & zaptel modules before
> removing the files and recompiling.
>
> For good/bad measure I am also completely powering off the system
> between attemps to ensure the USB hardware being used for timing
> with our 2.4 kernel is getting reset properly before trying again.
> Probably not really needed but I'm at the point of desperation and
> am trying not to leave anything out.
>
> Thanks for your time!
>
> Steve
>
>
>
>
>
>> Steve:
>>
>> I'm picking up the tail end of a thread, so apologies if this is
>> offtrack...
>>
>> Have you perhaps got an old set of EXECUTABLES in your path, that are
>> being picked up before your newly compiled ones?
>>
>> Roger
>>
>> Steve Gladden wrote:
>>
>>>Yes I have.
>>>I have been battling this issue since wednesday 1-25
>>>And so far have tried many things.
>>>
>>>Have also tried RTP debug and do not see ANY RTP when the call is made.
>>>
>>>I will keep working at this until I figure it out but right now am very
>>>stumped and frusterated.
>>>
>>>The software update SHOULD have fixed it as it has for many others.
>>>
>>>Steve
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
Have you tried increasing the debug level and watching the cli?





>No Firewalls involved, the test has been simplified down to two sip
>phones
>on a LAN and still no audio.
>
>For waht it's worth IAX2 still works fine.
>
>Steve
>
>-
>
>
>
>
>
>>>Yep, tried that.
>>>
>>>blew away all my source code, re-downloaded re compiled and re
>>>installed.
>>>it's behaving exactly the same, calls go through but no audio in
>>>
>>>
>either
>
>
>>>direction for sip-sip calls on the LAN or to-from the Internet SIP
>>>providers tested.
>>>
>>>I'm at a loss I feel like I have tried everything.
>>>
>>>even stripped down my configs and tried to make them as simple as
>>>possible
>>>with nothing more than two SIP phones and a default context.
>>>
>>>I'm running a 2.4 kernel with USB timimg for ztdummy
>>>
>>>Another interesting note is that I am getting no DTMF decode
>>>with PAP2 devices set to AVT.
>>>
>>>It was working before Jan 25th along with audio before all suddenly
>>>quite
>>>working.
>>>
>>>I set my system and hardware clock back to 00:00 Jan, 01 2006
>>>and rebooted the system
>>>
>>>
>>>
>>>Anything else I should be checking for?
>>>
>>>
>>Sounds like maybe a firewall is involved somewhere. Are you sure
>> there
>>are none in the path (including on your asterisk box)?
>>
>>
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>>
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>
>
---End of Original Message-


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>>

RE: [Asterisk-Users] Blocked Callerid

2006-02-01 Thread Alexander Lopez




They are using ANI instead of CallerID. If they have an 
800 number thya have the right to know who is calling them because they are 
paying for the call.
 
the *ANI*DNIS* format is known as Feature Grooup 
D.
 
Alex
 

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Joe 
  PukepailSent: Wednesday, February 01, 2006 3:47 PMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: 
  [Asterisk-Users] Blocked Callerid
  Do they have an 800 number?  If so perhaps their 800 number 
  provider is doing it via DTMF.  Search around on the internet, I believe 
  the standard format for the DTMF is *CALLERID*CALLEDNUMBER* (or perhaps 
  reversed). 
  On 2/1/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: 
  
  
I have been discussing an asterisk solution 
with a company that has a custom written dialogic based 
solution.
 
The issue is that their dialogic solution can 
read callerid from incoming calls, even if the callerid is 
blocked.
I have read before that Asterisk can do this, 
and they want me to make sure that their new system will be able to do 
this.
 
A quick poke around inside the zaptel source 
code was unproductive...
 
Any ideas?
 
PaulH
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Re: [Asterisk-Users] TE411P or TE406P

2006-02-01 Thread Matt




 
You will need a minimum 3.4Ghz Dual xeon with 1G 
ECC DDR, and hardware voice processing capable E1/T1 card, such 
as the sangoma 104d quad pci card, in order to run 120 PSTN calls, 1000 
calls is impossible for 1 server. Centos Linux should be fine.
 
We sell supermicro based * solutions you can 
contact me off list.
 
Best Regards
 
Matt

  - Original Message - 
  From: 
  Eduard B. 
  Cleofe 
  To: asterisk-users@lists.digium.com 
  
  Sent: Wednesday, February 01, 2006 8:15 
  PM
  Subject: [Asterisk-Users] TE411P or 
  TE406P
  
  Hi 
  Guys,
     
  I need your recommendation which card to buy?The TE411P or TE406P do they have 
  any difference?I check their brochures they are differ only when it comes to 
  PCI slot voltages.Which card best to use?Btw,Il be using supermicro board with 
  3.3V and 5.5V?Will il be experiencing problem during the implementation later 
  on if i choose the wrong PCI voltage options?Will it depends on what 
  bit(64 or 32 bit) of Linux kernel wil ill be 
  using?
  Thanks 
  so much 
  guys
  eng'r.eduard
  
  
  Yahoo! 
  Autos. Looking for a sweet ride? Get pricing, reviews, & more on new 
  and used cars.
  
  

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[Asterisk-Users] Anyone in or around Redmond, WA?

2006-02-01 Thread Nilesh Londhe


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RE: [Asterisk-Users] winnipeg canada

2006-02-01 Thread Nabeel Jafferali

> Anyone in Winnipeg Canada?

Winnipeg seems to have an active Asterisk group. Their mailing list is at
http://www.muug.mb.ca/mailman/listinfo/asterisk and I believe they are
having some kind of event soon.

Nabeel

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[Asterisk-Users] TE411P or TE406P

2006-02-01 Thread Eduard B. Cleofe

Hi Guys,     I need your recommendation which card to buy?The TE411P or TE406P do they have any difference?I check their brochures they are differ only when it comes to PCI slot voltages.Which card best to use?Btw,Il be using supermicro board with 3.3V and 5.5V?Will il be experiencing problem during the implementation later on if i choose the wrong PCI voltage options?Will it depends on what bit(64 or 32 bit) of Linux kernel wil ill be
 using?  Thanks so much guyseng'r.eduard
	
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Re: [Asterisk-Users] Default value for ASTERISK_VERSION_NUM

2006-02-01 Thread Leo Ann Boon


Kevin P. Fleming wrote:


Leo Ann Boon wrote:

I'm looking at version.h installed by Asterisk 1.2.3/4 - and the 
default value is 00. I thought the value should be 010200. I know 
many people have problems compiling chan_bluetooth because of this 
inconsistency. Anyone has the last word on this?



What is ASTERISK_VERSION in that same file?


Cut n paste from my asterisk-1.2.4/include/asterisk/version.h
/*
* version.h
* Automatically generated
*/
#define ASTERISK_VERSION "1.2.4"
#define ASTERISK_VERSION_NUM 00

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Re: [Asterisk-Users] supermicro server model

2006-02-01 Thread Cory Andrews




Eduard - I would recommend the SuperMicro 6014P-TR, 
which is a 1U rackmount server, with the Intel E7520 Chipset, Dual Redundant HOT Swap 560W Power Supplies, 
Marvell 88SX6081 4-port SATA Controllersupports (4) Hot swap SATA Drives, 
Channel RAID Support, Dual Onboard Gig-Esupports 12GB of DDR 333 RAM or 24GB 
of DDR 266  on (6) DIMM slots.  1.44 Floppy and DVD-ROM onboard, 
(5) Fans.  Only thing missing is the kitchen sink.
 
The followin 
Linux distros are supported:
 


  
  

  Redhat Linux 8.0
   





 

 
 
 
 
  

  Redhat Linux 9.0
   











  

  Redhat Linux AS 2.1
   

   
   










  

  Redhat Linux AS 3 +
   U3
 
 
 
 
 

 
 
 
 
 
  

  Redhat Linux ES 3.0
   

   
   




 





  

  Redhat Linux ES 3.0 Update-3 
   
 
 
 
 
 
 
 
 
 
 

  

  SuSE 8.0 
   
 
 
 
 
 
 
 
 
 
 
 
  

  SuSE 8.1 
   
 
 
 
 
 
 
 
 
 
 
 
  

  SuSE 8.2 
   
 
 
 
 
 
 
 


 

  

  SuSE 9.0 
   











  

  SuSE 9.1 x32
   










 
  

  SuSE 9.1 x64 
   
 
 
 
 
 
 
 
 
 
 
 
  

  SuSE 9.1 Pro 
   
 
 
 
 
 
 
 
 
 
 

  

  FreeBSD 5.2.1
   
 
 
 
 



 

 
 
  

  Fedora Core I x32
   

 
 




 
 
 

  

  Fedora Core II x32
   


 





 
 
 
  

  Fedora Core II x64
   


 







 
  

  SCO OpenServer 5.06
   
 
 
 
 
 
 
 
 
 
 
 
  
Caldera OpenUnix 
8.0
 
 Cory J 
AndrewsVOIPSupply.com454 Sonwil DriveBuffalo, NY 
14225++voice - 716.630.1555 X22email - [EMAIL PROTECTED]AIM - 
B2CORY

  - Original Message - 
  From: 
  Eduard B. 
  Cleofe 
  To: asterisk-users@lists.digium.com 
  
  Sent: Wednesday, February 01, 2006 10:19 
  PM
  Subject: [Asterisk-Users] supermicro 
  server model
  
  Hi Guys,
   
      Im 
  planning to purchase Supermicro server but I dont have idea what 
  Motherboard and chipset model will work with TE406P.I read one post here 
  that the chipset should be Intel E7520.Is it true?Is there other 
  recommendation?My requirement by the way is to handle 1000 simultaneous 
  calls.
      
  Does dualcore xeon (single cpu) and 2gig memory can handle this w/o 
  transcoding using sip?Any recommendation?
   
  Thank you very much guys for your help.
  
  
  Yahoo! 
  Autos. Looking for a sweet ride? Get pricing, reviews, & more on new 
  and used cars.
  
  

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RE: [Asterisk-Users] Skype-to-Asterisk(SIP): progress

2006-02-01 Thread John Todd


 > The developer has indicated that a revised version of the

 PSGW (http://www.rsdevs.com/) code will be available for sale
 shortly with the changes.


Has the developer indicated to you whether this would be a free upgrade for
existing clients or whether additional payments would be expected?

Regards,

Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


No, there has been no indication of this but it hasn't been 
discussed, either.  If you are a registered subscriber, perhaps it 
may be best to ask.  It sounds reasonable to _me_ at least...


JT

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Re: [Asterisk-Users] Re: Asterisk hangs on 1.2.1

2006-02-01 Thread Mark Johnson


Peter Fern wrote:
I'm pretty sure I've seen some commits dealing with channel locking 
since 1.2.1


Brent Torrenga wrote:

Might it be related to the memory leak bug? Upgrade to 1.2.4? (shot 
in the

dark, a brainstorm on my part is all)

 

Here's what the logfile shows.  Any ideas?  And is there a way to 
fix the deadlock without restarting Asterisk?


Feb  1 09:10:33 WARNING[5327]: channel.c:784 channel_find_locked: 
Avoided deadlock for '0xbf002d10', 10 retries!


Feb  1 09:17:08 DEBUG[6606] channel.c: Avoiding deadlock for 
'SCCP/204-0205'
  
Thanks for the suggestions!   I'll try the production box this weekend.  
I just installed the latest in lab and it looks OK.


Mark
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RE: [Asterisk-Users] winnipeg canada

2006-02-01 Thread Technical Support

We're Ontario based, but gladly work across Canada!

Michelle Dupuis
Technical Support Specialist
Oxford Consulting Group Ltd.
Making IT work for your business...
 
T: (519) 672-8238
E: [EMAIL PROTECTED]
W: www.ocg.ca 
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan k.
Creasy
Sent: Wednesday, February 01, 2006 8:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] winnipeg canada

Anyone in Winnipeg Canada?
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Re: RE: [Asterisk-Users] Blocked Callerid


I think the customer is more interested in whether Asterisk can do what their 
current system does, rather than discuss the legalites of this.

PaulH

> Chris Bagnall <[EMAIL PROTECTED]> wrote:
> 
> > The issue is that their dialogic solution can read callerid 
> > from incoming calls, even if the callerid is blocked.
> 
> I don't know what the laws on such things are where you're located, but 
> you
> might want to check into the legality of actually doing that.
> 
> Regards,
> 
> Chris
> -- 
> C.M. Bagnall, Director, Minotaur I.T. Limited
> This email is made from 100% recycled electrons
> 
> 
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Re: [Asterisk-Users] Anyway to do this?


If callerid is received, it will be displayed on the sip phones.

My guess would be that it's not coming in on the analog line in the first place.

PaulH

> Scott Geist <[EMAIL PROTECTED]> wrote:
> 
> How do you retreive the caller id on incoming analog lines and display 
> the
> id on the sip phones on the network?
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[Asterisk-Users] supermicro server model

2006-02-01 Thread Eduard B. Cleofe

Hi Guys,         Im planning to purchase Supermicro server but I dont have idea what Motherboard and chipset model will work with TE406P.I read one post here that the chipset should be Intel E7520.Is it true?Is there other recommendation?My requirement by the way is to handle 1000 simultaneous calls.      Does dualcore xeon (single cpu) and 2gig memory can handle this w/o transcoding using sip?Any recommendation?     Thank you very much guys for your help.
	
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Re: [Asterisk-Users] Re: Asterisk hangs on 1.2.1

I'm pretty sure I've seen some commits dealing with channel locking 
since 1.2.1


Brent Torrenga wrote:


Might it be related to the memory leak bug? Upgrade to 1.2.4? (shot in the
dark, a brainstorm on my part is all)

 

Here's what the logfile shows.  Any ideas?  And is 
there a way to fix the deadlock without restarting Asterisk?


Feb  1 09:10:33 WARNING[5327]: channel.c:784 channel_find_locked: 
Avoided deadlock for '0xbf002d10', 10 retries!


Feb  1 09:17:08 DEBUG[6606] channel.c: Avoiding deadlock for 
'SCCP/204-0205'
   




Sincerely,

Brent A. Torrenga
[EMAIL PROTECTED]

Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771

219.836.8918x325 Voice
219.836.1138 Facsimile
www.torrenga.com

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RE: [Asterisk-Users] determining if a call to a SIP extensions isfrom a queue

Will a variable set when a caller enters a queue pass to all
agents/channels called by the queue?

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Peter Fern
> Sent: Wednesday, February 01, 2006 5:32 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] determining if a call to a SIP
extensions
> isfrom a queue
> 
> Just set one in the dialplan as you enter the queue?
> 
> Damon Estep wrote:
> 
> >I am using agentcallbacklogin for queues
> >
> >I have a desire to modify the call behavior to the agents extension
if
> >the call is from a queue (opposed to from a PRI or another
extension).
> >
> >Is there a channel variable that can be read that would indicate the
> >source channel waiting to be bridged is an agent/queue call?
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Re: [Asterisk-Users] Anyway to do this?

2006-02-01 Thread Paul

Scott Geist wrote:

> How do you retreive the caller id on incoming analog lines and display
> the id on the sip phones on the network?

Assuming the analog lines have caller ID feature:

Do you have an incoming SIP account to test with? Reason I ask is that
my IP phones and ATA do that by default. If I dial a SIP DID from my
pots line and then dial an extension attached to my SPA-2000 I get my
pots line name and number on the extension. I also get that info in my
call logs. Do you get it in your log files for the analog lines? If not
the problems is in your zaptel config.

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Re: [Asterisk-Users] Winnipeg Canada

2006-02-01 Thread Richard Houston

I use Asterisk and I live in Winnipeg. I use it at home and I have a
client install coming up this year.




++
Best regards,
-Richard Houston
-R.L.H.  Consulting
-E-Mail  [EMAIL PROTECTED]
-WWW http://www.rlhc.net
-Bloghttp://www.rlhc.net/blog/


> Anyone in Winnipeg Canada?
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>


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[Asterisk-Users] Anyway to do this?

2006-02-01 Thread Scott Geist

How do you retreive the caller id on incoming analog lines and display the id on the sip phones on the network?
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Re: [Asterisk-Users] winnipeg canada

2006-02-01 Thread Rich Adamson


> Anyone in Winnipeg Canada?

Nope, they all moved to Mexico where its warmer.


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RE: [Asterisk-Users] Skype-to-Asterisk(SIP): progress

2006-02-01 Thread Chris Bagnall

> The developer has indicated that a revised version of the 
> PSGW (http://www.rsdevs.com/) code will be available for sale 
> shortly with the changes.

Has the developer indicated to you whether this would be a free upgrade for
existing clients or whether additional payments would be expected?

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


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RE: [Asterisk-Users] Blocked Callerid

2006-02-01 Thread Chris Bagnall

> The issue is that their dialogic solution can read callerid 
> from incoming calls, even if the callerid is blocked.

I don't know what the laws on such things are where you're located, but you
might want to check into the legality of actually doing that.

Regards,

Chris
-- 
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This email is made from 100% recycled electrons


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RE: [Asterisk-Users] XLite dtmf issue?

2006-02-01 Thread kevin ling




set dtmfmode=rfc2833 in sip.conf and try 
again.


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
AislingSent: Wednesday, February 01, 2006 11:03 PMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] XLite dtmf 
issue?


Hi,
 
I’m wondering if anyone has 
experienced an issue with the XLite softphone and asterisk accepting dtmf? I 
can listen to my voicemail perfectly from my hardphone. However when I dial the 
voicemail number from my XLite softphone and enter the password at the voicemail 
prompt, an error appears vm-incorrect and I get an 
“Unable to read password” message on the asterisk console. Has anyone 
experienced issues with XLite dtmf?
 
Many 
thanks,
Aisling.
 
 ---Legal 
Disclaimer--- The above electronic mail 
transmission is confidential and intended only for the person to whom it is 
addressed. Its contents may be protected by legal and/or professional privilege. 
Should it be received by you in error please contact the sender at the above 
quoted email address. Any unauthorised form of reproduction of this message is 
strictly prohibited. The Institute does not guarantee the security of any 
information electronically transmitted and is not liable if the information 
contained in this communication is not a proper and complete record of the 
message as transmitted by the sender nor for any delay in its receipt. 

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[Asterisk-Users] winnipeg canada

2006-02-01 Thread Jonathan k. Creasy

Anyone in Winnipeg Canada?
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Re: [Asterisk-Users] Dundi key Problem

2006-02-01 Thread Jesus Mogollon

Don't use the file extension (get rid of .pub when defining the key in dundi.conf). It will work.Jesus MogollonGlobal IP Systems, LLChttp://www.globalipsystems.com
On 2/1/06, Jonathan k. Creasy <[EMAIL PROTECTED]> wrote:
I am getting the following message when trying to lookup up a number viaDundi:Feb  1 13:39:24 NOTICE[20146]: pbx_dundi.c:1309 update_key: No such key'office.pbx.bluegrass.net.pub' for creating RSA encrypted shared key for
'00:a0:c9:55:91:89'!I have created keys on each box with "astgenkey -noffice.pbx.bluegrass.net" using the host name for each box of course.I then copied the .pub files to the /var/lib/asterisk/keys folder from
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Re: [Asterisk-Users] Digit timeouts vs includes in diaplan

Don't know why you would have been experiencing pauses beforehand, but 
you can specify a digittimeout with:


Set(TIMEOUT(digit)=<#>)

Michaël Gaudette wrote:


Hi,
 
I have a little situation with my dialplan, and I am wondering if what 
I want is even possible.
 
Here it is: I have three contexts, context1 includes contexts2, and 
context2 includes context3.In other words, in context1 all 
extensions of context2 and context3 are valid (and actually working, 
so that's good).  I am using those context for the sake of code 
clarity and reuse, and for this reason they are absolutely needed.
 
Most extensions work allright, EXCEPT in the cases where there are 
"overlapping" extensions, for exemple 2, 23 and 235.  In a normal 
dialplan, I would expect when dialing "2" that there would be a 
timeout of 5 seconds before that extension is dialed.  When dialing 
23, another 5 second delay and when dialing 235 it would dial 
immediately.  In other words, when I pressed "2" that extension would 
not immediately be dialed but asterisk would wait for other digits.
 
In my case, the extensions are split as follows:

[context1]
include => context2
exten => 2, 1,noop(2)
 
[context2]

include => context3
exten => 23,1,noop(23")
 
[context3]

exten => 235,1,noop(235)
 
And the RESULT is that when I press "2" in context1, it doesnt even 
give me a chance to dial the other digits, it simply connects me to 
extension 2.  What if I wanted to put in 235???
 
Is this:

1) A bug?
2) WAD?
3) I missed something?



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Re: [Asterisk-Users] Grandstream Budgetone BT-101 audio problems

2006-02-01 Thread jurgen

I didn't think SIP had a jitter buffer (yet). There's nothing in the
wiki about it, as it pertains to sip.conf.

In the meantime, I'll try a different NTP server, to see if that
changes anything.

On 01/02/06, Cristian Draghici <[EMAIL PROTECTED]> wrote:
> No. If the same NTP server is used and both phones are in the same
> switch, I don't see how disabling NTP would help.
>
> I'd rather disable jitterbuffer if it was on when the problem is
> manifesting itself.
>
> --
> c
>
>
> On 2/1/06, jurgen <[EMAIL PROTECTED]> wrote:
> > NTP? Really? I would never have thought of that one. All the phones
> > are configured to use pool.ntp.org, and they keep accurate time. Do
> > you think disabling NTP altogether would be a good idea?
> >
> > Thanks!
> >
> >
> > On 31/01/06, Cristian Draghici <[EMAIL PROTECTED]> wrote:
> > > Are you using the same NTP server for both phones?
> > > Are you using NTP at all?
> > >
> > > Is jitterbuffer enabled on the asterisk server?
> > >
> > > Not sure about SIP, but on IAX if the timestamps go haywire,  you can
> > > loose audio from one side.
> > >
> > > hth,
> > > c
> > >
> > >
> > > On 1/31/06, jurgen <[EMAIL PROTECTED]> wrote:
> > > > Hi all,
> > > >
> > > > I'm having a really frustrating time with a bunch of BT-101 phones.
> > > > They've been trouble-free and working very well for the past several
> > > > months. A couple of days ago, some of the phones (but not all of them,
> > > > yet) have started acting very strangely. All phones are running
> > > > firmware 1.0.6.7, and are identically configured (except for the
> > > > user/authenticate/password things) on both the phone side and in
> > > > sip.conf.
> > > >
> > > > After a few hours of relatively heavy use, the phone stops sending the
> > > > remote party's voice to the BT-101 person. Someone else (me, heh heh)
> > > > listening in to the call via ZapScan can hear both sides just fine, so
> > > > it doesn't seem to be Asterisk's problem, at least directly.
> > > >
> > > > I've tried simply power cycling the phones. Doing that buys me a bit
> > > > of time, sometimes a minute, sometimes ten. But the problem always
> > > > comes back relatively quickly. Moving the phones to another physical
> > > > Ethernet connection does nothing either.
> > > >
> > > > The way to make the problem go away for about 24 hours is to swap them
> > > > around. I move a spare from my desk to the person with the bad phone,
> > > > simply by changing the user/auth/pass strings. I set the broken phone
> > > > up with my testing user/auth/pass stuff, and they both start working
> > > > again. Now get this: Simply changing the user/auth/pass strings on the
> > > > bad phone to something else, then setting them back to what they were
> > > > before *doesn't work*. The actual phones have to be swapped around.
> > > > The phones are plugged into the same switch as the Asterisk server, no
> > > > funny stuff there.
> > > >
> > > > I've tried resetting the phones back to the factory settings, and
> > > > reconfiguring them from scratch. That doesn't eliminate the problem
> > > > either, just delays it another 24 hours.
> > > >
> > > > So. That pretty much covers it. Only three of the phones are doing
> > > > this, and they only started a few days ago. They were working *fine*
> > > > for months! Does anyone have any ideas here? I'm about ready to throw
> > > > these phones into a tree shredder.
> > > >
> > > > Thanks!
> > > >
> > > > ..jurgen
> > > >
> > > > --
> > > > [EMAIL PROTECTED] is jurgen's gmail address.
> > > > Visit http://jurgen.ca/ for more yummy goodness.
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> > >
> >
> >
> > --
> > [EMAIL PROTECTED] is jurgen's gmail address.
> > Visit http://jurgen.ca/ for more yummy goodness.
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--
[EMAIL PROTECTED] is jurgen's gmail address.
Visit http://jurgen.ca/ for more yummy goodness.
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Re: [Asterisk-Users] MOH sourced from a sound card?

You don't capture the audio - you set sox to send output to stdout, and 
use it as the custom command in musiconhold.conf, so it works like a 
pipe into asterisk.  If you read the voip-wiki page I listed it should 
make sense.  And with sox you could use either alsa or ossdsp depending 
on support in your kernel.


Mark Phillips wrote:

How does the customer maintain the message if I have to capture it 
every time he changes it?


This is not the solution.

Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com


Peter Fern wrote:

Using the classic MoH, use a custom moh player (see 
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+musiconhold.conf) 
and sox with the alsa pseudo-filetype, and output to stdout with the 
correct bitrate and samples... see the sox manpage for instructions.


Untested, but I think that should do the job for you...

Mark Phillips wrote:

I thought this had been around before but I can't seem to find 
anything about it.


I have a customer whom prior to upgrading to Asterisk invested in 
one of those boxes that plays your company sales campaign into the 
MOH port on your key system.


For reasons of message maintenance he wants to keep the box as part 
of the new system.


Can I couple this to the sound card in the Asterisk server and then 
have it play into the MOH? If so how?


Thanks


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Re: [Asterisk-Users] (newby) Is PING a good indicator of latency?


http://www.voip-info.org/wiki/view/Iperf

Not just latency, but jitter, etc - basically you can simulate various 
types of traffic and generate statistics.  You will need two ends to 
test with.


Cosmin Prund wrote:


As the subject line says: Is PING a good indicator of network latency? If
not, how can I measure latency?

Thanks,
Cosmin Prund



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RE: Using *RT for HA purposes was: [Asterisk-Users]Realtime MultipleAsterisk boxes, iaxusers

2006-02-01 Thread Rusty Shackleford

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Alistair Cunningham
> Sent: Wednesday, January 04, 2006 4:25 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: Using *RT for HA purposes was: 
> [Asterisk-Users]Realtime MultipleAsterisk boxes, iaxusers

> load balacing isn't perfect, and it can give uneven loads at low 
> capacity, but it gets better as load increases which is where 
> it matters.

What kind of loads are we talking about here, please?

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Re: [Asterisk-Users] changing cisco 7940/7960 standard menus ?


Only using SCCP, SIP firmware is set in stone.

Alex Ongena wrote:


Hi,

We are using Asterisk 1.2.1 with Cisco 7940 and 7960 phones.
Most things are running fine ;-)

But, when you are calling and you want to Transfer, you need
to press first on the 'more' button (4th), then you have the
label 'Trnsfr' to Transfer.

these are the lables on the softkeys when having a phone call:
"Holt / EndCall / Confrn / more"

press more and you get

"Transfer /  / BlndXfr / more"

We do more 'Transfers' than 'Confrn', so I which to siwtch the
2 softkeys on the phone.

Can you do that ?
How ?

Thanks
alex


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RE: [Asterisk-Users] 5,000 concurrent calls system rollout question

2006-02-01 Thread Michael Loftis




--On February 2, 2006 9:55:15 AM +1100 Boris Bakchiev 
<[EMAIL PROTECTED]> wrote:





The main sever is still connected via IP, correct?



Does not matter if you use * for media gateways or an APX8000 - the

only

trunking options to get to the main box are IP based.


Are seriously going to tell me that a quad xeon/opteron would not handle
traffic from 4xGIG cards?? :)


Probably not no.  Especially not if you're not using special IRQ load 
balancing software.  Xeon especially not since all teh CPUs share a FSB. 
The opteron *might* if the motherboard were designed such that the multiple 
pci busses terminated at multiple procs, and each proc had local memory 
attached to it.  Then you might be able to approach that rate.

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Re: [Asterisk-Users] determining if a call to a SIP extensions is from a queue


Just set one in the dialplan as you enter the queue?

Damon Estep wrote:


I am using agentcallbacklogin for queues

I have a desire to modify the call behavior to the agents extension if
the call is from a queue (opposed to from a PRI or another extension).

Is there a channel variable that can be read that would indicate the
source channel waiting to be bridged is an agent/queue call?
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Re: [Asterisk-Users] Networking voicemail

You could give each of your users a 'home' asterisk machine by an 
extension numbering convention and route voicemail based on their 
number.  It would mean that voicemail messages won't be able to be 
forwarded to a user homed on another machine though I would imagine.  We 
went for a centralised system, you can setup redundancy still, then 
script a reconciliation when the primary machine comes back.


Joe Pukepail wrote:

Is there a way to network the asterisk voicemail system between 
offices?  We would like the ability to forward a voicemail to another 
user at a branch office (each office would have their own asterisk 
server connected via iax), I guess I would prefer not to use one 
central server for voicemail for redudancy and disaster recovery, but 
I guess I'll have to if others have gone this way and I don't have any 
other choice.
 
 




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Re: Need to terminate 7 lines (was: [Asterisk-Users] RE: Asterisk-Users Digest, Vol 19, Issue 10)

2006-02-01 Thread Jean-Michel Hiver


Kevin Steil a écrit :


Need help...I need to install a card to terminate 7 lines...I have not
order the phone lines yet...I can either do analog lines 1FBs or order a
fractional T1...

If I was you, I would go for the T1 since you can expand it when it's 
needed.



any suggestions on what hardware would be easier to
install and configure...

If you plan on having only one T1 I think you should get a single T1/E1 
Digium card. That being said I'm in a bad position to recommend it since 
I went with a proprietary gateway myself (boo!).



also if I went with a T1...do I need an external
CSU/DSU or anything or does it just plug into the T1 card...thanks..
 


As far as I'm aware you don't.

Cheers,
Jean-Michel.

--
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP & Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE


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RE: [Asterisk-Users] Asterisk SIP phones to Cisco Unity via CCM4.0SIPTrunk

2006-02-01 Thread Dan Austin

>   thanks, using your example, and this url: 
>
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note0
9186a00800dea82.shtml
 

>   I got it to work... then I realized that there's no way the SIP
phone > on asterisk is going to get the MWI ( message waiting indicator
)...  how > are my users going to know they got a unity voice
message?... doh. 
This would be the trickiest, but not unsolvable.
For each Asterisk connect Unity user, modify their MWI extension 
in their Unity sunscriber profile to dial Asterisk.  On Asterisk 
add an extension with a small AGI script to 'touch' the local VM 
status.

>   we might as well buy the 7940s ( I wanted to get a bunch of 941s
), 
> or ditch unity... hrm.

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Re: [Asterisk-Users] (newby) Is PING a good indicator of latency?

2006-02-01 Thread JP Carballo


Cosmin Prund wrote:


Being an ISP I have to disagree - ICMP traffic is rarely prioritized to
"look better"

Most large ISPs actually LIMIT ICMP traffic to counter ICMP flood DOS
attacks.

That does not make it a good indicator of a networks ability to support
voip. Any cheap ATA will have jitter, delay, and packet loss counters -
hook one up and get the real picture.

If you do use ping to pre-qualify a link, grab a copy of pingplotter so
you can tune the icmp packet parameters and packet rate, run it for a
long time or several times during different timeframes to see if there
are periods of congestion.
   



Unfortunatelly good-old Windows based ping is all I can work with before I
actually sign the contract with the ISP. I need to assess rather the service
might be possible before I start throwing money at it (there's no such thing
as an "test account": I'll need to set up an account, pay the instalation
fees, pay one month in advance for the service then pay for the
disconnect!).

Sooo... what might I find using ping? Is there an "good" ping that shows I
can use the link for VoIP? I gues there's no such thing as "bad" ping
showing I can't use the link because ISP's are limiting ICMP traffic?
 


You might want to try the tools here:
http://www.dnsstuff.com/
and here
http://centralops.net/co/

--
JP Carballo

http://www.netfone2x.com
Bringing the world closer.

It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. 


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RE: [Asterisk-Users] Linking Asterisk Boxes with Sip

I assume you change the username/ip address/and passwords? This is way
more info than should be posted in public!

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Tom Vile
> Sent: Tuesday, January 31, 2006 1:41 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Linking Asterisk Boxes with Sip
> 
> It came through fine the first time.  Where is +15703232166 suppose to
go
> to.
> 
> On 1/31/06, Matt <[EMAIL PROTECTED]> wrote:
> > Not sure what's up with the mailing list here.   For some reason
mails
> > are not coming through.
> >
> > Try again...
> >
> > I am trying to link an asterisk box to my provider's asterisk server
> > via SIP. (I know I could use IAX, but the provider does not allow
> > that, so I can't).  When an inbound call happens I get this:
> >
> > Jan 31 13:09:14 NOTICE[3716] chan_sip.c: Failed to authenticate user
> > "+15703232166" ;tag=as3e2d0c2d
> >
> >
> > The config looks like this:
> > [general]
> >
> > port = 5060   ; Port to bind to (SIP is 5060)
> > bindaddr = 0.0.0.0; Address to bind to (all addresses on
machine)
> > disallow=all
> > allow=ulaw
> > allow=alaw
> > context = from-sip-external ; Send unknown SIP callers to this
context
> > callerid = Unknown
> > pendantic=no
> > register=123456789:[EMAIL PROTECTED]
> > [123456789]
> > type=user
> > secret=aaa607ac5aabb25ac037
> > insecure=very
> > context=from-pstn
> > []
> > username=123456789
> > type=peer
> > secret=aaa607ac5aabb25ac037
> > host=voipswitch1.voip..net
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> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> 
> --
> Tom Vile
> Baldwin Technology Solutions, Inc
> Consulting - Web Design - VoIP Telephony
> www.baldwintechsolutions.com
> Phone: 518-631-2855 x205
> Fax: 518-631-2856
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RE: [Asterisk-Users] Canadian Termination $0.0039 / Minute









Aah-ha, the old “do it now and ask forgiveness
latter” trick…

 











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of list
Sent: Tuesday, January 31, 2006
11:24 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Canadian Termination $0.0039 / Minute



 



Sorry I accidentally posted it to the users list instead of
biz. my ba







- Original Message - 





From: list 





To: asterisk-users@lists.digium.com 





Sent: Tuesday, January
31, 2006 12:56 PM





Subject: [Asterisk-Users]
Canadian Termination $0.0039 / Minute





 





All we have a deal on Canadian termination. 





 





Rate: $0.0039 US Dollars





Billing: 1/1





Protocol: SIP or H323





Codec: G729





Terms: Prepaid Only.





 





We have a real-time web interface where you can monitor or
download your CDR's.





 





Please e-mail me offlist if you are interested: [EMAIL PROTECTED]





 





Thanks,





Jon









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Re: [Asterisk-Users] RE: [Asterisk-Announce] Asterisk 1.2.4 and Zaptel 1.2.3

2006-02-01 Thread Kevin P. Fleming


Damon Estep wrote:

Does anyone know what date this memory leak was introduced and/or how to
check source code for it?

I am running a pre-1.2 CVS head version and would like to know if the
potential problem exists.


It has been present since we switched to the 'new' expression parser a 
few months ago. If your system is using ast_expr2.c instead of 
ast_expr.c, you have the problem.

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Re: [Asterisk-Users] Gain adjustment

2006-02-01 Thread Morten Isaksen


On 1/31/06, Rich Adamson <[EMAIL PROTECTED]> wrote:
> > > When adjusting the rxgain and txgain in Asterisk 1.2.1 do I need to restart Asterisk
or> > is it enough to just reload> > > Asterisk in order to apply the new setting?Unless someone just change this recently, the gain settings are notchanged on a reload. Since I've been watching the svn/cvs updates rather
closely, I don't believe any changes have actually been made (but I couldhave missed them as well).
 
I did some tests.
 
The output from ztmonitor is changing when you change the gain settings and then do a reload, so I think the gain values is changed when a reload is forced.
 -- Morten Isaksenhttp://www.misak.dk/blog/ 
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Re: [Asterisk-Users] 5,000 concurrent calls system rollout question

2006-02-01 Thread C F

I don't know how much 1+1 by you is, but lets recalculate this for a moment:
First the bandwidth per channel:
http://www.airewaves.com/aire/support/bandwidth_explain.php
1.5mbps (mega *BITS* not BYTES per second) to a full T1, which equals
1536 Kbits, each channel then takes 64kbps.
64*5,000=320,000kbps.
32,000/1,024=312.5 Mbps (round off to Mbps), no where close to a Gb.
Every single PC made in the last 4 years I came across, can handle
this type of bandwidth.
BTW, this all amounts to just over 39 MBYTES per second. 312.5/8=39.0625

On 1/29/06, Wai Wu <[EMAIL PROTECTED]> wrote:
>
>  To handle 5000 calls coming in over a PRI, you'd need 210 or so T1s or 170
> E1's.All of those would generate 320Mega BYTES of data per second (eg,
> 32Gigabit/sec)
> [Wai Wu]  He not talking about PRI here, but rather SIP to SIP
>
>
>
>
>
>
>
> There is no way possible that you're going to pump that amount of data
> through a PC. Don't care about codecs and dialplans, PC's just don't have
> that sort of internal bandwidth from peripherals.
>
>
>
> If all the endpints support reinvite and he is not doing any voice
> processing at all, there is hardly any data going through the PC
>
>
>
>
>
>
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[Asterisk-Users] Re:Linking Asterisk Boxes with Sip

2006-02-01 Thread steve

I think maybe your register syntax is wrong.  You list it as 
"register=foo-bar" when I believe it should be "register => foo-bar".  
Try adding the right arrow key after the equal sign.

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[Asterisk-Users] Help with sip setup because can't receive calls!!!!!!

2006-02-01 Thread steve

It looks like you have the first extry of the [incoming] context in 
extensions.conf commented out


Hi all, I am resending this message, so far no one has helped me with this incoming call 
issue. there is no problem with outbound call but there is no inbound call to my sip 
phone. the only message I get when I call from pstn is "unable to create local 
channel for call forward to 'Local/[EMAIL PROTECTED]' (case =0)". my configuration 
files are attached below. any help would be greatly appreciated. many thanks in advance.
ABC
 
abc def <[EMAIL PROTECTED]> wrote:

   there is no error message coming up on the pbx for in-bound calls (there is 
only debugging messages for outbound calls).
  
 thanks in advance for any hint or suggestion.

 Ama
  
 I just post my configuration file here for sip phone:

 extensions.conf
-
[globals]
 [default]
include => incoming
include => outgoing
include => iax
inculde => sip
include => sccp
[sip]
exten => 2171,1,Dial(SIP/stargate1,20)
;exten => 2171,1,Dial(SIP/2171,20)
exten => 2171,2,Hangup
exten => 2172,1,Dial(SIP/stargate2,20)
;exten => 2172,1,Dial(SIP/2172,20)
exten => 2172,2,Hangup
exten => 2173,1,Dial(SIP/stargate3,20)
;exten => 2173,1,Dial(SIP/2173,20)
exten => 2173,2,Hangup
 [sccp]
 [skinny]
 [incoming]
_*exten => ; _214943[5-9]6,1,Dial(SIP/stargate3*_) ; <-- this line looks 
like it was commented out on accident
exten => _214943[5-9]6,2,Hangup
 [outgoing]
exten => _,1,Dial(Zap/g1/${EXTEN})
exten => _,2,Hangup


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RE: [Asterisk-Users] changing displayed call info on snom 360

2006-02-01 Thread Phil Blundell

Thanks for the suggestion.  After exchanging some emails with snom
support, it turns out that the 360 firmware doesn't quite support the
facility I was looking for.

My actual problem was that the To: field in the body of the INFO needs
to match the identity of the line in question.  If I make that so, I can
indeed use the From: field in the INFO body to change the calling name
and number that the phone reports.  But this doesn't really help me
much, since I could have changed that anyway by just tweaking Asterisk's
caller ID variables before sending the original INVITE.  I was hoping
that the INFO message would cause the 360 to display the name and number
for both the calling and the (originally) called party.

p.

On Mon, 2006-01-30 at 11:02 +0100, Christian Stredicke wrote:
> That INFO must be inside the extsting dialog, maybe that was the
> problem.
> 
> CS
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] 
> > [mailto:[EMAIL PROTECTED] On Behalf Of 
> > Phil Blundell
> > Sent: Monday, January 30, 2006 10:16 AM
> > To: asterisk-users@lists.digium.com
> > Subject: [Asterisk-Users] changing displayed call info on snom 360
> > 
> > Several of my SIP users are in the habit of diverting all 
> > their calls to an assistant when they're out of the office.  
> > When these calls ring on the assistant's phone, she wants to 
> > be able to tell which number they've been forwarded from so 
> > that she can say "Joe Blow's phone" or whatever when she 
> > picks up the call.  The assistant's phone is a snom 360, 
> > which normally just displays the number of the calling party 
> > while it's ringing.
> > 
> > Snom's FAQ page at http://www.snom.com/wiki/index.php/FAQs 
> > suggests that I can send a SIP INFO message to the phone to 
> > change the displayed call information.  I did a few 
> > experiments with a hacked chan_sip.c, but wasn't able to 
> > produce any visible effect on the phone.
> > 
> > Does anybody have any experience making this snom feature 
> > work with Asterisk, or know of any other way to influence the 
> > information that's displayed on the phone?
> > 
> > Thanks
> > Phil
> > 
> > 
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> > 
> > 
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Re: [Asterisk-Users] Gain adjustment

2006-02-01 Thread Rich Adamson

Think you better check the source code to see how "%" is treated.
According to all documentation that I have, the only valid numbers that
can be used for gain settings are real numbers expressed in terms of db
of gain. (If a percentage was actually supported, what the hell would
it mean? 60% of what?)

In all liklihood, your 0.1% example is treated as 0.1 db.



> This isn't correct in all situations.  The gain adjustments are in 
> percentages, and if you need a LOT of gain to compensate for the signal 
> loss from the CO, then 0.1% really _does_ make a difference.
> 
> For example, when I'm watching the output of the CO's milliwatt 
> generator, an addition of 0.05 on the rxgain brings my 148xx to over 
> 15000.  That's _half_ a /tenth/ of a percent.
> 
> 
> Moj
> 
> Rich Adamson wrote:
> >>Need to stop asterisk and restart it. A reload will not take the new 
> >> setting
> >>into consideration. There is no need to stop/start the zaptel drivers, 
> >> just
> >>asterisk itself.
> >>
> >> 
> >>OK.
> >> 
> >>If I set the gain to a negative number then i decrease the volume? And a 
> >>positive 
number 
> > 
> > increases the volume?
> > 
> >> 
> > 
> > 
> > That's correct. Don't bother with 1/10ths. Just use -2 or 4 or 0 or 
> > whatever.
> > You can enter tenths, but its really not going to get you anywhere.
> > 
> > 
> > ___
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> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> 
> -- 
> Mojo <[EMAIL PROTECTED]>
> Office Manger, Horan & Company, LLC
> (907) 747- x112
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---End of Original Message-


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[Asterisk-Users] RE: Asterisk-Users Digest, Vol 19, Issue 10

2006-02-01 Thread Kevin Steil

Need help...I need to install a card to terminate 7 lines...I have not
order the phone lines yet...I can either do analog lines 1FBs or order a
fractional T1...any suggestions on what hardware would be easier to
install and configure...also if I went with a T1...do I need an external
CSU/DSU or anything or does it just plug into the T1 card...thanks..

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, February 01, 2006 5:41 PM
To: Asterisk User List
Subject: Asterisk-Users Digest, Vol 19, Issue 10

Send Asterisk-Users mailing list submissions to
asterisk-users@lists.digium.com

To subscribe or unsubscribe via the World Wide Web, visit
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or, via email, send a message with subject or body 'help' to
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When replying, please edit your Subject line so it is more specific
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Re: [Asterisk-Users] broadvoice??

2006-02-01 Thread Jerry Glomph Black

I'd suggest using a reliable company with actual humans that answer questions: 
voipjet (NO connection, just a happy customer).  There are other choices which 
are in the same category.


Argentina, Mexico, and Israel are in the 1.5/3/1.5 c/min category.   If you talk 
enough to hit your Broadvoice cost, they would probably be shutting you down for 
breaking the limit on your so-called-unlimited allotment.


I quit BV a long time ago, they do not have their act together, in my 
widely-shared opinion.



On Tue, 31 Jan 2006, Ilan Rabinovitch wrote:


Does anyone have any useful contact information for Broadvoice?  I've
been unable to call Argentina from my Broadvoice line for almost a
month now.  Their techs keep telling me they're "looking into it" but
I get no real information.  I've had several other tickets open for
almost 8 months now.

I called their PR company this morning and apparently they dropped
Broadvoice recently.  They indicated they get a lot of complaint calls
about Broadvoice being unresponsive.

Is there anyone out there?  Any recomendations on a service with
similar pricing on international calls?  I call Argentina, Mexico and
Israel most frequently.

Thanks,
Ilan
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Re: [Asterisk-Users] broadvoice??

2006-02-01 Thread Dovid Bender

>From simulas posts as the one you have written it
seems they are heading in the path of live voip. They
grew too big too fast and cant handle customer
service. I would recomend using myPhoneCompany. One
Negative note, at this time for unlimited plans I know
they only offer ATA's. If you want you can get a DID
from them for $5.00 a month (its caled a
myDevicePlan). Thier rates are pretty good. If you
intend on paying per minute and not using an unlimited
plan I would reccomend using multiple providers with
LCR (Low Cost Routing). This is what I use.

Regards,

Dovid
--- Ilan Rabinovitch <[EMAIL PROTECTED]> wrote:

> Does anyone have any useful contact information for
> Broadvoice?  I've
> been unable to call Argentina from my Broadvoice
> line for almost a
> month now.  Their techs keep telling me they're
> "looking into it" but
> I get no real information.  I've had several other
> tickets open for
> almost 8 months now.
> 
> I called their PR company this morning and
> apparently they dropped
> Broadvoice recently.  They indicated they get a lot
> of complaint calls
> about Broadvoice being unresponsive.
> 
> Is there anyone out there?  Any recomendations on a
> service with
> similar pricing on international calls?  I call
> Argentina, Mexico and
> Israel most frequently.
> 
> Thanks,
> Ilan
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>
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> 


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Re: [Asterisk-Users] polycom ip601 attendant console

2006-02-01 Thread Dovid Bender

I would but my partner isnt eccited about throwing out
there multiple hours of hard work. Also it isnt
complete yet. As of now it's a project. When its
complete I am sure we wil thro something out there.

Dovid

--- Sean Cook <[EMAIL PROTECTED]> wrote:

> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
> 
> care to share with the rest of the class?
> 
> Dovid Bender wrote:
> > To monitor who is doing what we writing a program
> that
> > every user can have on thier windows desktop to
> see
> > the status of all phones on the system. It's AIM
> > style. Has several groups. On the phone, off,
> > Available, Away etc. 
> > Managers can scroll the mouse over the user and
> see
> > what call they are on etc. This is very helpfull
> > because you dont have to program the phone. You
> have
> > all the info right on the desktop. (as of now it
> is
> > just a monitoring program. When we have time we
> will
> > make it so the user can dial from outlook or by
> typing
> > in the number on this program).
> > 
> > Regards,
> > Dovid
> > --- Rob McKrill <[EMAIL PROTECTED]> wrote:
> > 
> > 
> >>Using the "Buddy Watch" functionality on the IP601
> >>you can "watch" up to 
> >>6 people.  The expansion modules are not good for
> >>much more than speed 
> >>dials due to this limitation.
> >>
> >>After talking to our vendor, the reason it is
> >>limited to 6 is due to the 
> >>current version of Asterisk Business Edition's
> lack
> >>of 
> >>[documented/advertised] support for
> >>SUBSCRIBE/NOTIFY.  Since Polycom is 
> >>"certified" against ABE and the most recent
> release
> >>of ABE doesn't 
> >>support this functionality, Polycom will not open
> >>their firmware up to 
> >>allow more than six.
> >>
> >>I am hoping someone from Digium is monitoring this
> >>thread and that they 
> >>might comment on when the new edition of ABE will
> be
> >>released so that we 
> >>can actually utilize the full capabilities of the
> >>IP601's attendant 
> >>consoles.  Right now they (the attendant consoles)
> >>are pretty useless to 
> >>  me.  Has anyone else had any success with them?
> >>
> >>Saul Diaz wrote:
> >>
> >>>Damon Estep wrote:
> >>>
> >>>
> Anyone successfully set up one of the polycom
> >>
> >>soundpoint ip sidecars
> >>
> with asterisk to monitor and allow transfer to
> >>
> >>monitored extensions?
> >>
> How does it work? Any issues?
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> >>
> >
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> > 
>  
> 
> >>>
> >>>It works beautifull and not issues.
> >>>
> >>>regards
> >>>Saul
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[Asterisk-Users] Networking voicemail

2006-02-01 Thread Joe Pukepail

Is there a way to network the asterisk voicemail system between offices?  We would like the ability to forward a voicemail to another user at a branch office (each office would have their own asterisk server connected via iax), I guess I would prefer not to use one central server for voicemail for redudancy and disaster recovery, but I guess I'll have to if others have gone this way and I don't have any other choice. 

 
 
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Re: [Asterisk-Users] ZAP <--> sip(polycom301) can not hear each other

2006-02-01 Thread Ken D'Ambrosio

>From your description, it sounds as if the SIP phones are local to the
Asterisk box.  If this is so, having "nat=yes" might be a problem.

-Ken

sdgesa gaeharth wrote:

> please help!!!
>
> I am dialing into our asterisk server(TDM400p) from the psnt. I hear
> our voicemail message and I press the extention 1000. The Polycom ip
> phone in the office rings. I pickup but neither side can hear one
> another. What have I done wrong?
>
> thanks
>
> sip.conf:
> [general]
> context=local-access ; Default context for incoming calls
> bindport=5060   ; UDP Port to bind to (SIP standard
> port is 5060)
> bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds
> to all)
> srvlookup=yes   ; Enable DNS SRV lookup s on outbound
> calls
> musicclass=default
>
> [authentication]
>
> [1000]
> username=1000
> regexten=1000
> [EMAIL PROTECTED]
> callerid="jon Smith" <1000>
> context=local-access
> nat=yes
> secret=password
> type=friend
> host=dynamic
> canreinvite=yes
> disallow=all
> allow=all
>
> [1001]
> username=1001
> regexten=1001
> [EMAIL PROTECTED]
> callerid="jane doe" <1001>
> context=local-access
> nat=yes
> secret=password
> type=friend
> host=dynamic
> canreinvite=yes
> disallow=all
> allow=all
>
> extensions.conf:
> [general]
> static=yes
> writeprotect=no
> autofallthrough=yes
> clearglobalvars=no
> priorityjumping=no
>
> [globals]
> ATTENDANT=1001
> OUTBOUNDTRUNK=ZAP/g1
>
> [extentions]
> exten => _10XX,1,Ringing
> exten => _10XX,2,Dial(SIP/${EXTEN},20)
> exten => _10XX,3,Answer
> exten => _10XX,4,VoiceMail([EMAIL PROTECTED])
> exten => _10XX,5,Hangup
>
> [voicemail]
> exten => _910XX,1,Wait(1)
> exten => _910XX,2,VoiceMailMain(${EXTEN:[EMAIL PROTECTED])
>
> [local]
> include => extentions
> include => voicemail
>
> [incoming]
> exten => s,1,Answer
> exten => s,2,Background(our-voicemail-sound)
> exten => t,1,Playback(vm-goodbye)
> exten => t,2,Hangup( )
> exten => 0,1,Dial(SIP/${ATTENDANT},20)
> exten => 1,1,Directory(voicemail,internal,f)
> exten => 2,1,Directory(voicemail,internal)
> include => extentions
>
> [local-trunks]
> exten => _9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
> exten => _9XX,2,Congestion( )
> exten => _9XX,102,Congestion( )
> exten => 911,1,Dial(${OUTBOUNDTRUNK}/911)
> exten => 9911,1,Dial(${OUTBOUNDTRUNK}/911)
>
> [local-access]
> ignorepat => 9
> include => local
> include => local-trunks
>
>
> zapata.conf:
>
> [trunkgroups]
> [channels]
> context=default
> switchtype=national
> signalling=fxo_ls
> rxwink=300  ; Atlas seems to use long (250ms) winks
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> rxgain=0.0
> txgain=0.0
> group=1
> callgroup=1
> pickupgroup=1
> immediate=no
> group=1
> echocancel=yes
> switchtype=national
> signalling=fxs_ks
> context=incoming
> echocancelwhenbridged=yes
> channel => 1-4
>
>
> /etc/zaptel.conf:
> fxsks=1,2,3,4
> loadzone = us
> defaultzone=us
>
> log:
> Asterisk Ready.
> -- Star ting simple switch on 'Zap/1-1'
> Jan 31 15:55:28 NOTICE[2525]: chan_zap.c:6040 ss_thread: Got event 18
> (Ring Begin)...
> Jan 31 15:55:29 ERROR[2525]: callerid.c:276 callerid_feed: fsk_serie
> made mylen < 0 (-155)
> Jan 31 15:55:29 WARNING[2525]: chan_zap.c:6070 ss_thread: CallerID
> feed failed: Success
> Jan 31 15:55:29 WARNING[2525]: chan_zap.c:6114 ss_thread: CallerID
> returned with error on channel 'Zap/1-1'
> -- Executing Answer("Zap/1-1", "") in new stack
> -- Executing BackGround("Zap/1-1", "our-voicemail-sound") in new stack
> -- Playing 'our-voicemail-sound' (language 'en')
>   == CDR updated on Zap/1-1
> -- Executing Ringing("Zap/1-1", "") in new stack
> -- Executing Dial("Zap/1-1", "SIP/1000|20") in new stack
> -- Called 1000
> -- SIP/1000-54e4 is ringing
> -- SIP/1000-54e4 an swered Zap/1-1
>   == Spawn extension (incoming, 1000, 2) exited non-zero on 'Zap/1-1'
> -- Hungup 'Zap/1-1'
>
> 
> Bring words and photos together (easily) with
> PhotoMail
> 
> - it's free and works with Yahoo! Mail.
>
>
>
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RE: [Asterisk-Users] Polycom IP501 Endless Loop

Windows 2000 IIS ftp works fine with bootrom 2.6.2 and sip 1.5.2 and
1.5.3

If the ftp server is not on the same subnet as the phone verify that you
are not using nat or a firewall that is preventing or requiring passive
ftp mode.

If there is a firewall try a two way rule between the phone ip and the
ftp ip that allows all ip traffic in both directions (just as a test).

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Jerry Jones
> Sent: Tuesday, January 31, 2006 12:54 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Polycom IP501 Endless Loop
> 
> Also works fine for me, even with the default user/pw combo
> 
> On Jan 31, 2006, at 12:25 PM, Mojo with Horan & Company, LLC wrote:
> 
> > vsftpd has always worked fine for me, but I did change the password
> > the polycom was expecting to send from the default one with capital
> > letters.
> >
> > Moj
> >
> >
> > Ken D'Ambrosio wrote:
> >> I had the /exact/ same problem. Turns out it's the FTP server; in
the
> >> docs, there are several FTP servers specified as being compatible;
> >> proftp is the one I went with, and it fixed it right up. (Note that
I
> >> was using the default Debian FTP server when it was rebooting, so
> >> it's
> >> not just a 'doze issue.)
> >> -Ken
> >> Walt Reed wrote:
> >>> On Tue, Jan 31, 2006 at 08:18:34AM -0700, [EMAIL PROTECTED]
> >>> said:
> >>>
> >>>
>  I have a Polycom IP501 phone and have set it up to download the
>  config from an FTP server, it did this once and now is in an
>  endless loop of trying to contact the FTP server, failing, then
>  rebooting.
> 
>  When I watch the FTP server logs it looks like the phone starts
>  a session, ends it, starts it, ends it until the phone reboots.
>  It is annoying like nothing I can describe!
> 
>  I have tried Windows 2003 FTP service, WSFTP server and a few
>  other Windows based FTP servers.  Anybody have an idea as to how
>  to get around this?  I cannot get support on this phone (Polycom
>  tells me to call the reseller and the reseller won't touch it
>  for less than $95/hour).
> 
> >>>
> >>> Since you are running Asterisk, it would make sense to use a
> >>> Linux based
> >>> FTP server. At least then you would have decent logging (turn on
> >>> verbose
> >>> logging) which you can post the output of. I would also suggest
> >>> sniffing
> >>> the FTP attempt with ethereal or tcpdump to get more info on it.
> >>>
> >>> In any case, you are going to have to get more details:
> >>> When  you say "session", is it actually logging in correctly?
> >>> Finding
> >>> the files it is looking for? Or is it just a connection attempt?
> >>>
> >>> My guess is that it either is not logging in correctly or is not
> >>> finding
> >>> the files it wants, or it IS finding a file but doesn't like it.
> >>> Possibly one or more of the files is corrupt.
> >>>
> >>> ___
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> >>>  http://lists.digium.com/mailman/listinfo/asterisk-users
> >>>
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> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > --
> > Mojo <[EMAIL PROTECTED]>
> > Office Manger, Horan & Company, LLC
> > (907) 747- x112
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Re: [Asterisk-Users] Re: CallerID Problem

2006-02-01 Thread Gary Richardson

Hmm, that's annoying.

If I Set(CALLERID(num)=) (ie, I unset it), the callerid is set to the
default on the router and everything works as expected..

Thanks guys :)

On 2/1/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> This is what i found on Cisco's site:
>
> "Symptoms: Media negotiation fails for SIP calls and the terminating gateway 
> replies with a "488" message to an Invite message.
>
> Conditions: This symptom is observed on a Cisco platform when the terminating 
> gateway is configured with the G279B (annex B) codec and when the Session 
> Description Protocol (SDP) for the incoming Invite message does not have any 
> FMTP attribute line, which means that the default value, that is, the G279B 
> (annex B) codec, is used.
>
> Workaround: There is no workaround."
>
> Regards,
> Jan
>
> -Ursprungligt meddelande-
> Från: [EMAIL PROTECTED] genom Gary Richardson
> Skickat: on 2006-02-01 21:45
> Till: Asterisk Users Mailing List - Non-Commercial Discussion
> Ämne: Re: [Asterisk-Users] Re: CallerID Problem
>
> No, I'm not including the <> -- I was trying to show that it was
> something that I removed from my example..
>
> Thanks.
>
> On 2/1/06, Bromont Quebec <[EMAIL PROTECTED]> wrote:
> > Are you actually putting the < > in there?
> >
> > try:
> >
> > exten => _9.,1,Set(CALLERID(number)=MAINNUMBER)
> >
> > Hey,
> >
> > I'm using a Cisco 2811 to make calls out to a PRI. My asterisk box
> > connects to it using SIP. The asterisk version is 1.2.0.
> >
> > In my sip.conf, I set callerid="First Last" 
> >
> > When I make a an outbound call with the following macro:
> >
> > exten => _9.,1,Dial(SIP/${EXTEN}@,,w)
> > exten => _9.,2,Congestion()
> >
> > The caller id is set to the extension that's defined in sip.conf.
> >
> > If I try something like:
> >
> > exten => _9.,1,Set(CALLERID(number)=)
> > exten => _9.,2,Dial(SIP/${EXTEN}@,,w)
> > exten => _9.,3,Congestion()
> >
> > I get the following error:
> >
> > -- Got SIP response 488 "Not Acceptable Media" back from 
> >
> > It all works fine if I don't set the caller id.. Any ideas on why this
> > may be happening?
> >
> > Thanks.
> >
> >
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> >
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[Asterisk-Users] fax possibilities

2006-02-01 Thread James Harper

I am trying to set up a linux based faxing solution for a client, and
have found that the modem they have (ancient dataplex external unit)
just isn't up to the job. It talks to some remote fax machines but not
others.

A new external modem ranges from AUD$75 to AUD$400, which got me
thinking of other possibilities...

#1 FXO PCI card (more expensive for 1 port, probably cheaper for 2+)
#2 Sipura SPA3000
#3 Grandstream ATA488

I assume there will be no problem getting #1 working as a fax modem, but
what about #2 and #3? Has anyone done this before? Some brief googling
shows that it is possible, but not that it has been done...

James
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[Asterisk-Users] Work in Ukraine

2006-02-01 Thread Wojciech Tryc




 
Hi All,
My friend is looking to for a fulltimer to work on 
various VoIP gateways (mostly Cisco) and few Asterisk servers. Some development 
skills and knowledge of Asterisk's API would be an asset. At least familiarity 
with Asterisk's AGI and Perl/C/PHP would help. He is located in Kiev, so 
obviously Kiev residents are preffered. Remote work to be 
discussed.
Again, this is not a contract, this is Full-Time 
job.
Please send your resumes to [EMAIL PROTECTED]
Regards,
Wojtek
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RE: [Asterisk-Users] 5,000 concurrent calls system rollout question

2006-02-01 Thread Boris Bakchiev


>The main sever is still connected via IP, correct?

>Does not matter if you use * for media gateways or an APX8000 - the
only
>trunking options to get to the main box are IP based.

Are seriously going to tell me that a quad xeon/opteron would not handle
traffic from 4xGIG cards?? :)



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[Asterisk-Users] Re: Polycomm IP600 continues to ring

2006-02-01 Thread Noah Miller

Hi Matt - 

> I have a polycomm IP600 that about 30-50% of the time continues to
> ring after asterisk has signaled an answer.  Has anyone else experienced
> this?

Not I.  Weird!  What versions of things are you running?  Anything on the
CLI or in the logs?

- Noah


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Re: [Asterisk-Users] Asterisk SIP phones to Cisco Unity via CCM4.0SIP Trunk

2006-02-01 Thread sys read

thanks, using your example, and this url: http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a00800dea82.shtml
I got it to work... then I realized that there's no way the SIP phone on asterisk is going to get the MWI ( message waiting indicator )...  how are my users going to know they got a unity voice message?... doh.
we might as well buy the 7940s ( I wanted to get a bunch of 941s ), or ditch unity... hrm.thanks again everyone.On 1/30/06, Dan Austin
 <[EMAIL PROTECTED]> wrote:




It can be done.
 
1.  Setup a new Vm profile on CCM with a mask of  

2.  Setup a CTI route point:
    a. Set the directory number to a 
pattern.  I use *27XX
    but any pattern that you can send from * is 
good, ie. 88XXX
    b.  Set the VM profile to the newly created 
profile
    c.  Set the line to forward all calls to 
VM
3.  Change the dialplan in * to append the extension 
called to
    the prefix pattern.
 
exten 
=> 123,1,Dial(SIP/sipphone,20)exten => 
123,2,Dial(SIP/ccm/88123)
 

I use this setup to allow my CCM users to transfer calls 
directly to VM,
but it should work for this purpose as 
well.

Dan

  
  
  From: [EMAIL PROTECTED]
 
  [mailto:[EMAIL PROTECTED]] On Behalf Of sys 
  readSent: Tuesday, January 24, 2006 7:57 AMTo: Asterisk 
  Users Mailing List - Non-Commercial DiscussionSubject: Re: 
  [Asterisk-Users] Asterisk SIP phones to Cisco Unity via CCM4.0SIP 
  Trunk
  I have my eyes on the Linksys/Sipura 941, ( SIP ), but the core 
  problem is that you can't use SIP phones with CCM.  I have a SIP trunk 
  between asterisk and ccm.  I can route calls back and forth, I just can't 
  get the call to send to vm if no answer on the asterisk side. 
  On 1/24/06, kevin 
  ling <[EMAIL PROTECTED]> 
  wrote:
  
Hi,
 
Maybe 
buy 7912 phone and register to CCM is another choice. or integrated CCM with 
asterisk voicemail system.
 
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED]] On Behalf Of 
sys readSent: Tuesday, January 24, 2006 11:28 
PMTo: Asterisk Users Mailing List - Non-Commercial 
DiscussionSubject: Re: [Asterisk-Users] Asterisk SIP phones to 
Cisco Unity via CCM 4.0SIP Trunk

Hi guys,I want to leave messages on our unity 
box.   I have already converted a couple 7940s to SIP, but I can't 
give them out to our users because I don't want to have to deal with two 
voicemail systems.we have licenses for all our users on unity as 
is.    we're about to buy a bunch more 7940s, but I don't 
want to cause they're expensive. I'd rather buy a 
cheaper SIP phone and have it rollover to the unity vm.
On 1/23/06, Gary 
Richardson <[EMAIL PROTECTED]> wrote: 
You 
  can run a SIP image on a 7940. [EMAIL PROTECTED] has pretty goodsupport 
  for it. Check the voip-info.org wiki for 
  instructions onswitching the firmware.Hopefully that will take 
  a step out of the plan -- you could completely ditch your Cisco system 
  :)On 1/23/06, sys read <[EMAIL PROTECTED]> 
  wrote:>> Hi,>> I've got a CCM ( Cisco Call 
  Manager ), with a Cisco Unity VM server and > about 45 SCCP phones 
  on the ccm, and 200 users on unity.   we want to> migrate 
  all users to IP Phones to ditch our ancient phone system.   I 
  would> love to get Linksys-Sipura SPA-941s for the 150 users not on 
  IP phones yet > and run sip to an asterisk server, but have their 
  voicemail on Unity.>> these phones are $150 each, the 
  alternative is cisco 7940s ( around $250 )> running SCCP through 
  the CCM.  at the quantities I'm talking about, $100 is > 
  significant.>> Does anyone have any idea how to get this 
  done?>> I've tried this:>> exten => 
  123,1,Dial(SIP/sipphone,20)> exten => 
  123,2,Dial(SIP/ccm/3040)>> where 3040 is our VM pilot for 
  ccm.  but all it does is take us to the main> 
  greeting.>> we have smartnet, but they haven't been helpful 
  at all>> I called digium to see if they could help if we 
  paid, but they said they've > never heard of cisco 
  unity>> help?>> thanks.>> 
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  Colocation provided by Easynews.com 
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  update options visit:>> http://lists.digium.com/mailman/listinfo/asterisk-users 
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Re: [Asterisk-Users] Polycom IP301: Pass-through ethernet portunusable?

2006-02-01 Thread Anthony Rodgers

Are you using the ferrite chokes provided with the phone? They should  
be clamped onto BOTH cables, as close to the phone as possible.


Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp

On 31-Jan-06, at 4:52 PM, Nathan Bowyer wrote:


On 1/31/06, Damon Estep <[EMAIL PROTECTED]> wrote:
>
>
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk- 
users-

> > [EMAIL PROTECTED] On Behalf Of Jerry Glomph Black
> > Sent: Monday, January 30, 2006 11:59 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [Asterisk-Users] Polycom IP301: Pass-through ethernet  
port

> > unusable?
> >
> > Have just done a deployment of 45 of these puppies.
> >
> > They are doing their main job quite well, but of course there are
> minor
> > kinks.
> >
> > A not-so-minor one is that if one attempts to plug a PC into  
the 2nd

> RJ-45
> > jack,
> > as soon as you send any reasonable amount of traffic (even  
casual web

> > surfing)
> > the phone seizes.  We had to run a bunch of cables in a big  
rush to

> users'
> > PCs,
> > having (erroneously) believed that the passthru RJ45 would be a  
usable

> > port!
> >
> > Has anyone out there experienced this?
> >
> No issues on the IP501 with 2.6.2 bootrom and 1.5.3 SIP. Ethernet  
port

> works fine for the PC.

Works fine here, with an IP501, 2.6.2 bootrom and 1.6.3 SIP.  Was
using ethernet port for a while, even downloading large files through
it without any hiccups.
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Re: [Asterisk-Users] 5,000 concurrent calls system rollout question

2006-02-01 Thread asterisk


On Wed, 1 Feb 2006, Kristian Larsson wrote:

Indeed, a FreeBSD machine doing just routing
lookups can handle somewhere around 600Kpps.


Not to nitpick, but freebsd has routed 1M+pps using commodity hardware.

Its tricky but can be done.

-Dan
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Re: [Asterisk-Users] Strange echo phenomenon (double tandem)

2006-02-01 Thread asterisk


On Wed, 1 Feb 2006, Rich Adamson wrote:

The echo canceller (EC) on the 488 is rather limited and the call paths
for incoming vs outgoing calls are on the edge of the canceller. The
EC training algorithum might also be a little different for incoming
vs outgoing calls.


If its anything like the SPA-3000 then the EC is extremely limited.
SPA is G.165 EC with 8ms tail.


The only basis for that guess is my playing around with a new 488 running
the latest available firmware on a pstn line that is known to have a
long echo tail. It fails miserably and can't be used on this pstn line
at all.


Still looking for an FXO which doesnt have sucky EC...

-Dan
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Re: [Asterisk-Users] (newby) Is PING a good indicator of latency?

2006-02-01 Thread Chris Mason (Lists)


Cosmin Prund wrote:

As the subject line says: Is PING a good indicator of network latency? If
not, how can I measure latency?

  
The most effective visual indication of network reliability in as far as 
it relates to VOIP is an application called smokeping. The graphs show 
latency as shadows over the ping response which is shown as colours. You 
can see latency, variation in latency, and packet loss in one easy 
display, over 3 hours, 1 day, 1 week.

I can show you this in action is you email me privately.

--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 



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Re: [Asterisk-Users] meetme and dtmf

2006-02-01 Thread Imran Ahmed

On 2/1/06, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
> Imran Ahmed wrote:
>
> > Even though no IAX client supports inband dtmf, An IAX client can send
> > inband dtmf which would have corrected your problem.
>
> No, it won't. No IAX2 client will start a DSP to listen for inband DTMF,
> because IAX2 is defined to always send out-of-band DTMF.
>
> At best, if the receiving IAX2 system is just passing the audio along to
> another protocol that does support inband DTMF, then sending it in the
> audio stream would work. If the application receiving the DTMF is on the
> other IAX2 end, though (like MeetMe in this case), then it will never
> 'see' the DTMF, because Asterisk will not look in the audio stream for DTMF.

I agree, but the other ends of the conference were zap channels in
this case, at least that is what I figured by the first email.

Imran.
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Re: [Asterisk-Users] No Audio on Local Machine, Remote works fine

2006-02-01 Thread Hadar Pedhazur


[EMAIL PROTECTED] wrote:

Is ztdummy loaded properly?

I had a similar problem with a system recently.


The machine has a real Digium T1 card in it, so I didn't think to check 
for a timing source. Since it's a "backup" machine, the actual T1 line 
isn't plugged in at the moment, but chan_zap.so definitely starts up 
correctly.


I'll look into this in the morning (running out of the office now :-).

Thanks for the suggestion!


PaulH

- Original Message - 
From: "Hadar Pedhazur" <[EMAIL PROTECTED]>

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

Sent: Thursday, February 02, 2006 8:16 AM
Subject: [Asterisk-Users] No Audio on Local Machine, Remote works fine



I don't even know where to begin.

I run a lot of production Asterisk servers, for a couple of years now,
with no real problems.

We built a brand new box, CentOS 4.2, and installed Asterisk 1.2.4 from
source tarball(s). Built fine, and started up fine.

Any attempts to do local audio (e.g. a "Playback(welcome)") results in
complete silence. Worse, the Playback command will hang forever (even if
the file is tiny), so it's not just "not being heard", it's like the
command is waiting to do something.

In one specific case (and only in case), I'll hear a 1/2 second burst of
audio, like it's about to start, and then dead air.

The "Record" command creates a zero length file if the format is ulaw,
and hangs forever after that, and a "wav" format is always 44 bytes
before the hang.

If I run the "demo-echo-test", I don't hear the prompt, and it hangs on
the Playback.

OK, now for the weirdness ;-). If I connect this Asterisk to one of our
other servers, and dial the echo test on the remote server through this
same server, I hear the prompts, and can hear my voice echoed correctly,
so this same Asterisk server will happily forward the audio in both
directions, it just won't "generate" it. This is with "notransfer=yes",
so this Asterisk is staying in the audio stream.

I'm stumped, and any help or pointers in the right direction will be
greatly appreciated.
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Re: [Asterisk-Users] RE: Teliax - Codec Preference effective?

2006-02-01 Thread tim panton


Brent Torrenga wrote:

Thanks for your input, everyone, but I still think it is on Teliax's end...
I will present our collective thoughts to their tech.

Kevin,
I am using IAX. When I turn on IAX debug, I get:

--SNIP CLI OUTPUT--

 -- Executing Dial("SIP/Brent_ring-bcf7", "IAX2/teliax/18005558355") in
new
stack
 -- Called teliax/18005558355
 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
Timestamp: 7ms  SCall: 16384  DCall: 0 [208.139.204.228:4569]
VERSION : 2
CALLED NUMBER   : 18005558355
CODEC_PREFS : (g726)
CALLING NUMBER  : 2198368918
CALLING PRESNTN : 0
CALLING TYPEOFN : 0
CALLING TRANSIT : 0
CALLING NAME: Torrenga Engineering
LANGUAGE: en
USERNAME: (REDACTED-BY-THE-EDITOR)
FORMAT  : 16
CAPABILITY  : 63504
ADSICPE : 2
DATE TIME   : 2006-02-01  08:20:42


Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
AUTHREQ
Timestamp: 2ms  SCall: 00237  DCall: 16384 [208.139.204.228:4569]
AUTHMETHODS : (REDACTED-BY-THE-EDITOR)
CHALLENGE   : (REDACTED-BY-THE-EDITOR)
USERNAME: (REDACTED-BY-THE-EDITOR)

 Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
AUTHREP
Timestamp: 00255ms  SCall: 16384  DCall: 00237 [208.139.204.228:4569]
MD5 RESULT  : (REDACTED-BY-THE-EDITOR)


Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
REJECT
Timestamp: 00133ms  SCall: 00237  DCall: 16384 [208.139.204.228:4569]
CAUSE   : Unable to negotiate codec
CAUSE CODE  : 58

 Feb  1 08:20:43 WARNING[4710]: chan_iax2.c:6973 socket_read:  Call rejected
by
208.139.204.228: Unable to negotiate codec
 Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK

a   Timestamp: 00133ms  SCall: 16384  DCall: 00237 [208.139.204.228:4569]
 -- Hungup 'IAX2/teliax-16384'
   == Everyone is busy/congested at this time (1:0/0/1)
 -- Executing Playback("SIP/Brent_ring-bcf7",
"all-outgoing-lines-unavailabl
e") in new stack
 -- Playing 'all-outgoing-lines-unavailable' (language 'en')

  == Spawn extension (internal, *83518005558355, 3) exited non-zero on
'SIP/Bren
t_ring-bcf7'

--SNIP CLI OUTPUT--

I don't see any codec negotiation stuff other than my end requesting g726...
The only "effective" codec I get is GSM, which is my old setting on the
Teliax site.




IAX doesn't exactly do negotiation per se.
In the first packet you send you say CODEC_PREFS = g726
and FORMAT = 16
This means that you are _only_ prepared to do g726
(Thats a Six not a Nine by the way)

They reject the call because they don't want to do 726 and
you didn't offer anything else.

Tim.

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Re: [Asterisk-Users] Analog with channel bank - Inbound works, outbound doesn't

2006-02-01 Thread james.texter

I appreciate all of your help.  I'm starting to think this is a setup/hardware 
issue with the channel bank myself.  I bought the FXO card off of EBay, so who 
knows what kind of shape it's in.

Does anyone else out here on the forums know how to configure a CAC Access Bank 
II SNMP channel bank to work with Asterisk?

Thanks,

James

C F wrote:
> Then I got no clue how to configure it. But it looks like something is
> wrong in that setup there.
>
> On 2/1/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
>> No, it's an Access Bank II SNMP.
>>
>> Thanks,
>>
>> James
>>
>> C F wrote:
>>> Is this an Adit 600?
>>>
>>> On 2/1/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
 The output from the CLI when I put in an inbound call is the following:

-- Starting simple switch on 'Zap/25-1'
-- Executing GotoIf("Zap/25-1", "1?from-pstn-reghours|s|1:") in new 
 stack
-- Goto (from-pstn-reghours,s,1)
-- Executing GotoIf("Zap/25-1", "0?from-pstn-reghours-nofax|s|1:2") in 
 new stack
-- Goto (from-pstn-reghours,s,2)
-- Executing Answer("Zap/25-1", "") in new stack
-- Executing Wait("Zap/25-1", "1") in new stack
-- Executing SetVar("Zap/25-1", "intype=EXT-412") in new stack
-- Executing Cut("Zap/25-1", "intype=intype|-|1") in new stack

 It then goes on to call the extension I have setup.  I think it's coming 
 in on Channel 25, but I'm not sure what the -1 is for in Zap/25-1.

 Not sure if this is relevant or not, but I'm using a Carrier Access 
 Corporation (CAC) channel bank, with 1 FXO card and 1 FXS card.  The 
 analog line is definitely hooked to the FXO card, and I definitely have 
 the T1 plugged in to the FXO card.

 Thanks,

 James


 C F wrote:
> Looks like  channel 25 is not the one hooked up to your POTS, when an
> incoming call arrives, what channel does the CLI report?
>
>
> On 2/1/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
>> Thanks for the reply.  I have tried adding anywhere between 1 and 6 w's 
>> to the dial string, but still no luck.  I hooked up and listened on the 
>> line when the call went out, and never heard any DTMF's.  I'm sure this 
>> must be something simple, I just can't seem to figure out for the life 
>> of me what it is.  What other information can I provide to help sort 
>> this out?
>>
>> Thanks again,
>> James
>>
>> --
>> You could insert a pause by adding a w before the number to be dialed,
>> like this:
>> Dial(zap/25/w1234567890) iirc each w puts a 500ms pause.
>>
>>
>> On 1/30/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
 I am experimenting with an asterisk setup in my office.  The last bit 
 I have to test is working with analog lines.  I have TE411p digium 
 card, with an ISDN line plugged into the first, a channel bank plugged 
 into the second port, and the last two ports empty.  I have the 
 following setup in my zaptel.conf:

 span=1,1,0,esf,b8zs
 bchan=1-23
 dchan=24

 span=2,0,0,d4,ami
 fxsks=25

 And in zapata.conf, I have:
 group=2
 language=en
 context=from-pstn
 signalling=fxs_ks
 channel=>25

 I have one analog line plugged in for testing.  If I dial that analog 
 number, the inbound call arrives, and it works great.  However, when I 
 place an outbound call, I get the following output:
 -- Called g2/5148346
 -- Zap/25-1 answered SIP/412-9b72

 However, my number never rings.  After about 30 seconds, I get a 
 message saying my call could not be completed as dialed.  Almost like 
 it didn't get all of the digits.  Is there a way to inject a pause 
 before dialing?  Any other thoughts?  Any help is greatly appreciated.

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[Asterisk-Users] Polycomm IP600 continues to ring

2006-02-01 Thread Matt TPI


Hey All,
   I have a polycomm IP600 that about 30-50% of the time continues to 
ring after asterisk has signaled an answer.  Has anyone else experienced 
this?

Thanks
-Matt
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SV: [Asterisk-Users] Re: CallerID Problem

2006-02-01 Thread jan.sarin

This is what i found on Cisco's site:

"Symptoms: Media negotiation fails for SIP calls and the terminating gateway 
replies with a "488" message to an Invite message.

Conditions: This symptom is observed on a Cisco platform when the terminating 
gateway is configured with the G279B (annex B) codec and when the Session 
Description Protocol (SDP) for the incoming Invite message does not have any 
FMTP attribute line, which means that the default value, that is, the G279B 
(annex B) codec, is used.

Workaround: There is no workaround."

Regards,
Jan

-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] genom Gary Richardson
Skickat: on 2006-02-01 21:45
Till: Asterisk Users Mailing List - Non-Commercial Discussion
Ämne: Re: [Asterisk-Users] Re: CallerID Problem
 
No, I'm not including the <> -- I was trying to show that it was
something that I removed from my example..

Thanks.

On 2/1/06, Bromont Quebec <[EMAIL PROTECTED]> wrote:
> Are you actually putting the < > in there?
>
> try:
>
> exten => _9.,1,Set(CALLERID(number)=MAINNUMBER)
>
> Hey,
>
> I'm using a Cisco 2811 to make calls out to a PRI. My asterisk box
> connects to it using SIP. The asterisk version is 1.2.0.
>
> In my sip.conf, I set callerid="First Last" 
>
> When I make a an outbound call with the following macro:
>
> exten => _9.,1,Dial(SIP/${EXTEN}@,,w)
> exten => _9.,2,Congestion()
>
> The caller id is set to the extension that's defined in sip.conf.
>
> If I try something like:
>
> exten => _9.,1,Set(CALLERID(number)=)
> exten => _9.,2,Dial(SIP/${EXTEN}@,,w)
> exten => _9.,3,Congestion()
>
> I get the following error:
>
> -- Got SIP response 488 "Not Acceptable Media" back from 
>
> It all works fine if I don't set the caller id.. Any ideas on why this
> may be happening?
>
> Thanks.
>
>
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Re: [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection

2006-02-01 Thread Rob Lith

1500 to 3000 is poor -  Satellite delay is approximately 270 milliseconds (the time 
required for the signal to travel 35,800 km into space and return). If 
associated signal processing time through baseband equipment is included, 
total path delay is closer to 320 milliseconds. In voice 
communications, the most noticeable effect of this delay is echo, anything from 600ms upwards will not be nice.RegardsRobOn 2/1/06, 
Greg Oliver <[EMAIL PROTECTED]> wrote:
Depends on the type of satellite, but generally 1500 - 3000ms.On Wed, 2006-02-01 at 18:28 +0100, Master_PE wrote:> What is a normal dealy on a satelite installation?>> Regards,>> Master_PE
>> Op 1-feb-2006, om 13:26 heeft Garth van Sittert het volgende geschreven:>> > Hi Cosmin> >> > You should be able to get about 12 simultaneous calls on a 128k> > line and about 28 on a 256k line according to asteriskguru's
> > bandwidth calculator http://www.asteriskguru.com/tools/> > bandwidth_calculator.php.> >> > Kind Regards> > Garth> >
> > BitCo Data Communications> > http://www.bitco.co.za> >> > Cosmin Prund wrote:> >> Hello everyone, this is my first post to the list, so hello again.
> >>> >> We're a small company in Romania and we're trying to set up a> >> really small> >> version of "call center". That is, we want to get a few land-lines
> >> from our> >> telco in different countys and "bridge" all calls to our HQ, in> >> order to> >> make it cheeper for our clients to call us.> >>
> >> Unfortunatelly there's no ISP in our area that can deliver a> >> broadband> >> connection for anything less then an arm and a leg, so we're> >> considering> >> runing an * <-> * connection using VoIP over a low bandwidth
> >> connection> >> (we're considering 128kbit but we might be able to go to 256kbit).> >>> >> The bandwidth price is not a problem for our "satelite"> >> installations, we
> >> cand get acceptably priced broadband (~256kbit) so the distant *'s> >> will have> >> propper connections.> >> My question:> >>> >> Is 128kbit a wide enough connection for 1 simultaneous
> >> conversation, using> >> IAX protocol with the comercial version of the g729 codec?> >>> >> I'm expecting this to be engough for more then 1 conversation> >> (after all a
> >> single line analog connection is rated at 64kbit and I'm getting> >> double that> >> bandwidth)> >> Cosmin Prund> >>> >>> >> ___
> >> --Bandwidth and Colocation provided by Easynews.com --> >>> >> Asterisk-Users mailing list> >> To UNSUBSCRIBE or update options visit:
> >>http://lists.digium.com/mailman/listinfo/asterisk-users> >> From - Wed> >>> >> > --
> > Garth van Sittert> > BSc (Physics & Computer Science)> > -> > Mobile: +27 (0)83 791 6662> > Email:  [EMAIL PROTECTED]
> > Phone:  08600 BITCO> > Web:www.bitco.co.za> > ___> > --Bandwidth and Colocation provided by 
Easynews.com --> >> > Asterisk-Users mailing list> > To UNSUBSCRIBE or update options visit:> >   
http://lists.digium.com/mailman/listinfo/asterisk-users>> ___> --Bandwidth and Colocation provided by Easynews.com --
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Re: [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection

2006-02-01 Thread Rob Lith

What codec is that using. G.729 will give you 10 calls at best over 256k unless you're trunking with IAX2? I don't know anyone using lpc10...Remember a G.729 8k codec turns into 23.63 Kbps with all the overheads...
RegardsRobOn 2/1/06, Garth van Sittert <[EMAIL PROTECTED]> wrote:
Hi CosminYou should be able to get about 12 simultaneous calls on a 128k line andabout 28 on a 256k line according to asteriskguru's bandwidth calculator
http://www.asteriskguru.com/tools/bandwidth_calculator.php.Kind RegardsGarthBitCo Data Communicationshttp://www.bitco.co.zaCosmin Prund wrote:> Hello everyone, this is my first post to the list, so hello again.
>> We're a small company in Romania and we're trying to set up a really small> version of "call center". That is, we want to get a few land-lines from our> telco in different countys and "bridge" all calls to our HQ, in order to
> make it cheeper for our clients to call us.>> Unfortunatelly there's no ISP in our area that can deliver a broadband> connection for anything less then an arm and a leg, so we're considering
> runing an * <-> * connection using VoIP over a low bandwidth connection> (we're considering 128kbit but we might be able to go to 256kbit).>> The bandwidth price is not a problem for our "satelite" installations, we
> cand get acceptably priced broadband (~256kbit) so the distant *'s will have> propper connections.>> My question:>> Is 128kbit a wide enough connection for 1 simultaneous conversation, using
> IAX protocol with the comercial version of the g729 codec?>> I'm expecting this to be engough for more then 1 conversation (after all a> single line analog connection is rated at 64kbit and I'm getting double that
> bandwidth)>> Cosmin Prund>>> ___> --Bandwidth and Colocation provided by Easynews.com --
>> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:>http://lists.digium.com/mailman/listinfo/asterisk-users
> From - Wed>--Garth van SittertBSc (Physics & Computer Science)-Mobile: +27 (0)83 791 6662Email:  [EMAIL PROTECTED]
Phone:  08600 BITCOWeb:www.bitco.co.za___--Bandwidth and Colocation provided by Easynews.com
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SV: [Asterisk-Users] Re: CallerID Problem

2006-02-01 Thread jan.sarin

Seems to me like the negotiation fails for some reason. Maybe you are trying to 
use a callerid that isn't allowed?

Regards,
Jan


-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] genom Gary Richardson
Skickat: on 2006-02-01 21:45
Till: Asterisk Users Mailing List - Non-Commercial Discussion
Ämne: Re: [Asterisk-Users] Re: CallerID Problem
 
No, I'm not including the <> -- I was trying to show that it was
something that I removed from my example..

Thanks.

On 2/1/06, Bromont Quebec <[EMAIL PROTECTED]> wrote:
> Are you actually putting the < > in there?
>
> try:
>
> exten => _9.,1,Set(CALLERID(number)=MAINNUMBER)
>
> Hey,
>
> I'm using a Cisco 2811 to make calls out to a PRI. My asterisk box
> connects to it using SIP. The asterisk version is 1.2.0.
>
> In my sip.conf, I set callerid="First Last" 
>
> When I make a an outbound call with the following macro:
>
> exten => _9.,1,Dial(SIP/${EXTEN}@,,w)
> exten => _9.,2,Congestion()
>
> The caller id is set to the extension that's defined in sip.conf.
>
> If I try something like:
>
> exten => _9.,1,Set(CALLERID(number)=)
> exten => _9.,2,Dial(SIP/${EXTEN}@,,w)
> exten => _9.,3,Congestion()
>
> I get the following error:
>
> -- Got SIP response 488 "Not Acceptable Media" back from 
>
> It all works fine if I don't set the caller id.. Any ideas on why this
> may be happening?
>
> Thanks.
>
>
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Re: [Asterisk-Users] No Audio on Local Machine, Remote works fine

Is ztdummy loaded properly?

I had a similar problem with a system recently.

PaulH

- Original Message - 
From: "Hadar Pedhazur" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Thursday, February 02, 2006 8:16 AM
Subject: [Asterisk-Users] No Audio on Local Machine, Remote works fine


> I don't even know where to begin.
>
> I run a lot of production Asterisk servers, for a couple of years now,
> with no real problems.
>
> We built a brand new box, CentOS 4.2, and installed Asterisk 1.2.4 from
> source tarball(s). Built fine, and started up fine.
>
> Any attempts to do local audio (e.g. a "Playback(welcome)") results in
> complete silence. Worse, the Playback command will hang forever (even if
> the file is tiny), so it's not just "not being heard", it's like the
> command is waiting to do something.
>
> In one specific case (and only in case), I'll hear a 1/2 second burst of
> audio, like it's about to start, and then dead air.
>
> The "Record" command creates a zero length file if the format is ulaw,
> and hangs forever after that, and a "wav" format is always 44 bytes
> before the hang.
>
> If I run the "demo-echo-test", I don't hear the prompt, and it hangs on
> the Playback.
>
> OK, now for the weirdness ;-). If I connect this Asterisk to one of our
> other servers, and dial the echo test on the remote server through this
> same server, I hear the prompts, and can hear my voice echoed correctly,
> so this same Asterisk server will happily forward the audio in both
> directions, it just won't "generate" it. This is with "notransfer=yes",
> so this Asterisk is staying in the audio stream.
>
> I'm stumped, and any help or pointers in the right direction will be
> greatly appreciated.
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[Asterisk-Users] No Audio on Local Machine, Remote works fine

2006-02-01 Thread Hadar Pedhazur


I don't even know where to begin.

I run a lot of production Asterisk servers, for a couple of years now, 
with no real problems.


We built a brand new box, CentOS 4.2, and installed Asterisk 1.2.4 from 
source tarball(s). Built fine, and started up fine.


Any attempts to do local audio (e.g. a "Playback(welcome)") results in 
complete silence. Worse, the Playback command will hang forever (even if 
the file is tiny), so it's not just "not being heard", it's like the 
command is waiting to do something.


In one specific case (and only in case), I'll hear a 1/2 second burst of 
audio, like it's about to start, and then dead air.


The "Record" command creates a zero length file if the format is ulaw, 
and hangs forever after that, and a "wav" format is always 44 bytes 
before the hang.


If I run the "demo-echo-test", I don't hear the prompt, and it hangs on 
the Playback.


OK, now for the weirdness ;-). If I connect this Asterisk to one of our 
other servers, and dial the echo test on the remote server through this 
same server, I hear the prompts, and can hear my voice echoed correctly, 
so this same Asterisk server will happily forward the audio in both 
directions, it just won't "generate" it. This is with "notransfer=yes", 
so this Asterisk is staying in the audio stream.


I'm stumped, and any help or pointers in the right direction will be 
greatly appreciated.

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Re: [Asterisk-Users] RE: Euro-ISDN

2006-02-01 Thread Armin Schindler

On Wed, 1 Feb 2006, Aldo Bergamini wrote:
> [EMAIL PROTECTED] is believed to have said: 
> 
> >chan_capi does not set the NT-mode. Your cards driver need to do that.
> >E.g. for Eicon DIVA Server cards, you just set the '-x' option with divactrl
> >or set NT-mode in the config wizard.
> >chan_capi does not (need) to know anything about what protocol the card is 
> >doing. CAPI is independent here.
> 
> Ok.
> 
> >Anyway, if the card is set to NT mode, you should specify ntmode=yes
> >in the capi.conf to tell chan_capi to handle the progress better
> >(get progress tones).
> 
> Fine!
> 
> One last related subpoint: while Eicon Diva cards have their own setup
> application, is there anything standard to control the basic setup of
> generic HFC-S cards? (something similar to the ztconfig tool for analog cards)

Sorry, I cannot answer that one. I don't know enough about these cards and 
their drivers.
 
> >capi driver? I think you mean chan_capi? Yes it does support
> >TE and NT mode, as well as DSS1, 1TR6, JAPAN, NI1, QSIG, 5ESS, and a lot 
> >more protocols.
> 
> Yes, I meant actually the chan_capi module... Nice to know.
> 
> >The card and the cards driver need to support/provide this.
> > 
> >Armin
> 
> So in the end there a lot of reasons to go for a 'better' card.

Yes, a lot reasons. But actually, it depends on what you need and what you
want to do.

Armin

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[Asterisk-Users] how to log agents into a queue

2006-02-01 Thread nik600

hi, i'm testing asterisk

 i would like to log some agents in a queue and see how works the system.

 i've added to queue.conf

  Code: 

 [700]
 wrapuptime=0
 timeout=15
 strategy=ringall
 retry=5
 queue-youarenext=
 queue-thereare=
 queue-thankyou=queue-thankyou
 queue-callswaiting=
 music=default
 monitor-join=yes
 monitor-format=
 maxlen=0
 leavewhenempty=no
 joinempty=yes
 context=
 announce-holdtime=no
 announce-frequency=0
 member => Agent/101


 and in

 agests.conf


  Code: 

 agent => 101,Nicola Mosca


 and then in extension.conf


  Code: 

 exten => 201,1,AgentLogin(101)

 [ext-local]
 include => ext-local-custom
 exten => 100,1,Macro(exten-vm,novm,100)
 exten => 100,hint,SIP/100
 exten => 101,1,Macro(exten-vm,novm,101)
 exten => 101,hint,SIP/101
 exten => 102,1,Macro(exten-vm,novm,102)
 exten => 102,hint,SIP/102


 i've use
[EMAIL PROTECTED] 2.1to manage groups and extensions

 now, for example if log extension 100 and 102 in two different PC i
can call from 100 to 102 and from 102 to 100.

 If i call 700 (the queue) i can hear the music because no-one agents
are logged in the queue

 now, if i have understand what is explained in the wiki, calling the
extension 201 the agest using extension 101 will be logged, is it
true?

 But if i try to call 201 i get a 404 Not Found error, where am i wrong?

 thanks in advance
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Re: [Asterisk-Users] New version of snom soft phone

2006-02-01 Thread Joe Pukepail

I see the announcement for the snom 300 on the website, any idea of the street price for that phone?
On 2/1/06, Christian Stredicke <[EMAIL PROTECTED]> wrote:
Hey we have made a new version of our soft phone which fixes animportant bug in the SRTP SSRC part... It is compatible with our latest
version 5.3 of the hard phones.http://www.snom.com/download/snom360-5.3.exeEnjoy, Christian___
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Re: [Asterisk-Users] Analog with channel bank - Inbound works, outbound doesn't

2006-02-01 Thread C F

Then I got no clue how to configure it. But it looks like something is
wrong in that setup there.

On 2/1/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> No, it's an Access Bank II SNMP.
>
> Thanks,
>
> James
>
> C F wrote:
> > Is this an Adit 600?
> >
> > On 2/1/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> >> The output from the CLI when I put in an inbound call is the following:
> >>
> >>-- Starting simple switch on 'Zap/25-1'
> >>-- Executing GotoIf("Zap/25-1", "1?from-pstn-reghours|s|1:") in new 
> >> stack
> >>-- Goto (from-pstn-reghours,s,1)
> >>-- Executing GotoIf("Zap/25-1", "0?from-pstn-reghours-nofax|s|1:2") in 
> >> new stack
> >>-- Goto (from-pstn-reghours,s,2)
> >>-- Executing Answer("Zap/25-1", "") in new stack
> >>-- Executing Wait("Zap/25-1", "1") in new stack
> >>-- Executing SetVar("Zap/25-1", "intype=EXT-412") in new stack
> >>-- Executing Cut("Zap/25-1", "intype=intype|-|1") in new stack
> >>
> >> It then goes on to call the extension I have setup.  I think it's coming 
> >> in on Channel 25, but I'm not sure what the -1 is for in Zap/25-1.
> >>
> >> Not sure if this is relevant or not, but I'm using a Carrier Access 
> >> Corporation (CAC) channel bank, with 1 FXO card and 1 FXS card.  The 
> >> analog line is definitely hooked to the FXO card, and I definitely have 
> >> the T1 plugged in to the FXO card.
> >>
> >> Thanks,
> >>
> >> James
> >>
> >>
> >> C F wrote:
> >>> Looks like  channel 25 is not the one hooked up to your POTS, when an
> >>> incoming call arrives, what channel does the CLI report?
> >>>
> >>>
> >>> On 2/1/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
>  Thanks for the reply.  I have tried adding anywhere between 1 and 6 w's 
>  to the dial string, but still no luck.  I hooked up and listened on the 
>  line when the call went out, and never heard any DTMF's.  I'm sure this 
>  must be something simple, I just can't seem to figure out for the life 
>  of me what it is.  What other information can I provide to help sort 
>  this out?
> 
>  Thanks again,
>  James
> 
>  --
>  You could insert a pause by adding a w before the number to be dialed,
>  like this:
>  Dial(zap/25/w1234567890) iirc each w puts a 500ms pause.
> 
> 
>  On 1/30/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> >> I am experimenting with an asterisk setup in my office.  The last bit 
> >> I have to test is working with analog lines.  I have TE411p digium 
> >> card, with an ISDN line plugged into the first, a channel bank plugged 
> >> into the second port, and the last two ports empty.  I have the 
> >> following setup in my zaptel.conf:
> >>
> >> span=1,1,0,esf,b8zs
> >> bchan=1-23
> >> dchan=24
> >>
> >> span=2,0,0,d4,ami
> >> fxsks=25
> >>
> >> And in zapata.conf, I have:
> >> group=2
> >> language=en
> >> context=from-pstn
> >> signalling=fxs_ks
> >> channel=>25
> >>
> >> I have one analog line plugged in for testing.  If I dial that analog 
> >> number, the inbound call arrives, and it works great.  However, when I 
> >> place an outbound call, I get the following output:
> >> -- Called g2/5148346
> >> -- Zap/25-1 answered SIP/412-9b72
> >>
> >> However, my number never rings.  After about 30 seconds, I get a 
> >> message saying my call could not be completed as dialed.  Almost like 
> >> it didn't get all of the digits.  Is there a way to inject a pause 
> >> before dialing?  Any other thoughts?  Any help is greatly appreciated.
> >>
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Re: [Asterisk-Users] Blocked Callerid

2006-02-01 Thread Joe Pukepail

Do they have an 800 number?  If so perhaps their 800 number provider is doing it via DTMF.  Search around on the internet, I believe the standard format for the DTMF is *CALLERID*CALLEDNUMBER* (or perhaps reversed). 

On 2/1/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:


I have been discussing an asterisk solution with a company that has a custom written dialogic based solution.
 
The issue is that their dialogic solution can read callerid from incoming calls, even if the callerid is blocked.
I have read before that Asterisk can do this, and they want me to make sure that their new system will be able to do this.
 
A quick poke around inside the zaptel source code was unproductive...
 
Any ideas?
 
PaulH
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Re: [Asterisk-Users] Re: CallerID Problem

2006-02-01 Thread Gary Richardson

No, I'm not including the <> -- I was trying to show that it was
something that I removed from my example..

Thanks.

On 2/1/06, Bromont Quebec <[EMAIL PROTECTED]> wrote:
> Are you actually putting the < > in there?
>
> try:
>
> exten => _9.,1,Set(CALLERID(number)=MAINNUMBER)
>
> Hey,
>
> I'm using a Cisco 2811 to make calls out to a PRI. My asterisk box
> connects to it using SIP. The asterisk version is 1.2.0.
>
> In my sip.conf, I set callerid="First Last" 
>
> When I make a an outbound call with the following macro:
>
> exten => _9.,1,Dial(SIP/${EXTEN}@,,w)
> exten => _9.,2,Congestion()
>
> The caller id is set to the extension that's defined in sip.conf.
>
> If I try something like:
>
> exten => _9.,1,Set(CALLERID(number)=)
> exten => _9.,2,Dial(SIP/${EXTEN}@,,w)
> exten => _9.,3,Congestion()
>
> I get the following error:
>
> -- Got SIP response 488 "Not Acceptable Media" back from 
>
> It all works fine if I don't set the caller id.. Any ideas on why this
> may be happening?
>
> Thanks.
>
>
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Re: [Asterisk-Users] Dumb Dialout Question

2006-02-01 Thread Ronald Wiplinger


[EMAIL PROTECTED] wrote:

I'm still trying to learn some parts of Asterisk, so sorry in advance for the 
dumb question!

How do I set up an extension to dial out to the PSTN through my ZAP interfaces? 
 I want the ability to have a ring group that will ring all of the phones in an 
office and then ring cell phones if nobody answers.  I'm sure this is simple to 
do but I'm at a loss.

I have tried the following configs in extensions.conf to no avail:

exten => 190,1,Dial(ZAP/[EMAIL PROTECTED]) ; Cell Phone

exten => 190,1,Dial(ZAP/800111) ; Cell Phone

exten => 190,1,Dial(SIP/[EMAIL PROTECTED]) ; Cell Phone

exten => 190,1,Dial(ZAP/800111) ; Cell Phone

Thank you in advance!

  

I have something like that:

zapata.conf
group=2for this channel



extensions.conf

in [globals]
PSTN=ZAP/g2

for dialing out:
exten => _9N.,103,Macro(dial-pstn,${EXTEN:1},${LONGTIMEOUT})

and for dialing out I use a macro:

[macro-dial-pstn];
;***
; BEGIN - Outbound Dialing macro
;***
;
;This macro will dial out on PSTN line 1 first
;will use PSTN line 2 if line 1 is in use
;
;Enter with these
;ARG1 = 
;ARG2 = 
;
;Returns with FOUNDME = DIALSTATUS
;
;   the 9w dials 9 then waits 0.5 seconds for outside dialtone, needed 
for dial 9 system only

;
;
exten => s,1,SetGlobalVar(FOUNDME=ANSWER)
exten => s,2,Dial(${PSTN}/w${ARG1},${ARG2})
exten => s,3,NoOp(${DIALSTATUS})
exten => s,4,Goto(s-${DIALSTATUS},1)
;
;Return here if busy
;
exten => s,103,NoOp(${DIALSTATUS})
exten => s,104,Goto(s-${DIALSTATUS},1)
;
;
exten => s-BUSY,1,BackGround(the-party-you-are-calling)
exten => s-BUSY,2,BackGround(is-curntly-busy)
exten => s-BUSY,3,SetGlobalVar(FOUNDME=BUSY)
exten => s-BUSY,4,Goto(s-CLEANEXIT,1)
;
;
exten => s-CANCEL,1,BackGround(canceled)
exten => s-CANCEL,2,SetGlobalVar(FOUNDME=CANCEL)
exten => s-CANCEL,3,Goto(s-CLEANEXIT,1)
;
;
exten => s-CHANUNAVAIL,1,BackGround(channel)
exten => s-CHANUNAVAIL,2,BackGround(is-curntly-unavail)
exten => s-CHANUNAVAIL,3,SetGlobalVar(FOUNDME=CHANUNAVAIL)
exten => s-CHANUNAVAIL,4,Goto(s-CLEANEXIT,1)
;
;
exten => s-CONGESTION,1,BackGround(channel)
exten => s-CONGESTION,2,BackGround(is-curntly-unavail)
exten => s-CONGESTION,3,SetGlobalVar(FOUNDME=CHANUNAVAIL)
exten => s-CONGESTION,4,Goto(s-CLEANEXIT,1)
;
;
exten => s-NOANSWER,1,BackGround(nbdy-avail-to-take-call)
exten => s-NOANSWER,2,SetGlobalVar(FOUNDME=NOANSWER)
exten => s-NOANSWER,3,Goto(s-CLEANEXIT,1)
;
;
exten => s-ANSWER,1,SetGloabalVar(FOUNDME=ANSWER)
exten => s-ANSWER,2,Goto(s-CLEANEXIT,3)
;
;
exten => s-.,1,BackGround(something-terrible-wrong)
exten => s-.,2,NoOp(${DIALSTATUS)
exten => s-.,3,SetGlobalVar(FOUNDME=ERROR)
exten => s-.,4,Goto(s-CLEANEXIT,1)
;
;
exten => s-CLEANEXIT,1,NoOp
exten => s-CLEANEXIT,2,Hangup
exten => s-CLEANEXIT,3,NoOp




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Re: [Asterisk-Users] Is it possible ?

2006-02-01 Thread Sohail Arham

thanks amna i have done it ..
On 2/1/06, amna saleem <[EMAIL PROTECTED]> wrote:

Hi ,
I think i understand what you mean by your mail.I have done the same thing.
You must download following modules from and asterisk site
e.g.
www.digium.com
1.asterisk-1.0.3.tar
2.libpri-1.0.3.tar
3.zaptel-1.0.3.tar
Then there is a process you need to follow which you will find in th read me file of asterisk after you untar these.
If you are unable to figure it out then mail me.
Allah hafiz
Amna

On 1/24/06, Sohail Arham <
[EMAIL PROTECTED]> wrote:

Hi everyone,
I am a new one for that listsactually i have final year project on VOIP & IMS ...so i want to install asterisk on my pc ...IS it possble that ...we can call on small LAN network without buying any card...i will clear my point as that...suppose i have a linux machine on which i want to install asterisk and HOW it will install...and second point is that ..i have 2 windows clients machine i want these two machice can calls each other thorough asterisk .rest of all will be think latter

thanksss -- Muhammad Sohail ArhamU.E.T. LahorePhone No. 0321-4422406 ___--Bandwidth and Colocation provided by
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RE: [Asterisk-Users] Dumb Dialout Question

2006-02-01 Thread Jason Adams

When you dial a zap interface you have to reference the channel.   

So:
exten => 190,1,Dial(ZAP/g1/800111) ; Cell Phone

Using the above, would dial out on GROUP 1.  When you setup your zaptel
hardware you assign the channels to a group.  You can then reference
that group and * will dial out using an available channel.  Using the
lowercase g will tell * to dial out using channels with a lower number.
Using an uppercaes G will tell * to dial out using the highest channel
number available.

You could also create a variable called TRUNK=ZAP/G1 in your [globals]
section.

Then you could do this:
exten => 190,1,Dial(${TRUNK}/800111) ; Cell Phone


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, February 01, 2006 3:11 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Dumb Dialout Question

I'm still trying to learn some parts of Asterisk, so sorry in advance
for the dumb question!

How do I set up an extension to dial out to the PSTN through my ZAP
interfaces?  I want the ability to have a ring group that will ring all
of the phones in an office and then ring cell phones if nobody answers.
I'm sure this is simple to do but I'm at a loss.

I have tried the following configs in extensions.conf to no avail:

exten => 190,1,Dial(ZAP/[EMAIL PROTECTED]) ; Cell Phone

exten => 190,1,Dial(ZAP/800111) ; Cell Phone

exten => 190,1,Dial(SIP/[EMAIL PROTECTED]) ; Cell Phone

exten => 190,1,Dial(ZAP/800111) ; Cell Phone

Thank you in advance!





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Re: [Asterisk-Users] Dumb Dialout Question

You are missing the G

exten => 190,1,Dial(ZAP/g0/800111)

Assumes that you have set up your zap card as group 0.
(zap/g1 is probably more realistic)

later,

PaulH

- Original Message - 
From: <[EMAIL PROTECTED]>
To: 
Sent: Thursday, February 02, 2006 7:11 AM
Subject: [Asterisk-Users] Dumb Dialout Question


> I'm still trying to learn some parts of Asterisk, so sorry in advance for
the dumb question!
>
> How do I set up an extension to dial out to the PSTN through my ZAP
interfaces?  I want the ability to have a ring group that will ring all of
the phones in an office and then ring cell phones if nobody answers.  I'm
sure this is simple to do but I'm at a loss.
>
> I have tried the following configs in extensions.conf to no avail:
>
> exten => 190,1,Dial(ZAP/[EMAIL PROTECTED]) ; Cell Phone
>
> exten => 190,1,Dial(ZAP/800111) ; Cell Phone
>
> exten => 190,1,Dial(SIP/[EMAIL PROTECTED]) ; Cell Phone
>
> exten => 190,1,Dial(ZAP/800111) ; Cell Phone
>
> Thank you in advance!
>
>
>
>
>
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