date:20060414

Interesting... I don't have those problems with the A200D. In fact, 
haven't found any problems at all in about two months of production use. 
I have noticed that mixing a TDM400 with a A200D in the same box has an 
issue with sangoma code disabling echotraining=800 (needed for the TDM). 
But that's it.



Matt wrote:
Well and that is where my love hate relationship with Sangoma is.  
While their technical support seems to be very knowledgeable it

takes forever to get ahold of them!   Additionally I've found 2 bugs
now in their drivers   At first setup wouldn't install... it was a
bug with their setup not creating a file if it was missing.   And now
there is some caller-id bug in their software.   When it patched the
zaptel source... if I have usecallerid=yes on then it crashes... if I
turn usecallerid=no then it is fine.

On 4/13/06, John Novack [EMAIL PROTECTED] wrote:

Have you asked Sangoma support?

Though they can be slow to respond to an E-mail, they can quickly tell
you if it is their problem or not
I am using a 2 FXO 2FXS with Asterisk  SVN-branch-1.2-r13026 with no
problems on CentOS 3.6

John Novack


Matt wrote:


I'm running asterisk 1.2.6 with Sangoma A200 4 port FXO card.
Outbound calls are working fine.  However, when I have an inbound
call.. asterisk will segfault.. and then start again ... then it will
take  1 call fine

I'm running asterisk with a -U and -G of asterisk.  Any thoughts?
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Re: [Asterisk-Users] Ztmonitor shows RX is always on.

2006-04-14 Thread Kyle Sexton

Have you tried putting a Hangup in your extensions.conf?On 4/13/06, Min Hwan Chang [EMAIL PROTECTED]
 wrote:Details:Asterisk 1.0.9 Zaptel 1.0 
Dell P3 1ghz with X100P Clone Location: IndiaThis is an interesting issue where when I open up ZTMonitor, it shows the RX as being on. It seems that Zaptel doesn't know to hang up the line so after a couple of hours when the telecom cuts the line, everythign stops working. Things I've tried include playing with the 
zaptel.conf, trying zaptel v1.2 (with Asterisk 1.0.9), and trying loopstart and kewlstart. Could I have a setting wrong in my extensions.conf or is this just a problem with the Indian phone lines? Regards,
Min *Zaptel.conf*fxsks=1loadzone = usdefaultzone = uszapata.conf[channels]context=incomingsignalling=fxs_ks

language=enrxwink=300  ; Atlas seems to use long (250ms) winksusedistinctiveringdetection=nobusydetect=yesbusycount=6callprogress=yesimmediate=nocallwaiting=nocallwaitingcallerid=no

threewaycalling=notransfer=nocancallforward=nocallreturn=nouseincomingcalleridonzaptransfer=nocallerid=asreceivedusecallerid=norelaxdtmf=noechocancel=yesechocancelwhenbridged=yesechotraining=800
rxgain=0.0txgain=0.0group=0callgroup=1pickupgroup=1faxdetect=nochannel = 1 

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RE: [Asterisk-Users] Performance: Xeon or Opteron?

My main concerns would be, can you have multiple cards like this on a
system, for example, I now have a te110p and 2 tdm04b and Im getting
irqmisses on the te110p (according to zttool and zttest) which makes fax
receiving on the te110p almost impossible.. Plus, voice is getting frame
slips.

I was hoping sangoma cards could be more enterprise friendly. 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|John Novack
|Sent: Thursday, April 13, 2006 4:57 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Performance: Xeon or Opteron?
|
|Have a 2FXO 2FXS card working now.
|More forgiving of the PCI bus. The software installation is a 
|little mean, with the outlined procedure making assumptions 
|about the installers knowledge and familiarity with Sangoma 
|products, and in some places it doesn't really discriminate 
|between their T1 and A200 cards.
|I found one defect in their FXS driver, which they have now fixed.
|
|Overall seems to be a good product, slightly more affordable 
|and less of a problem child than the Digium/TigerJet TDM400
|
|John Novack
|
|Anton Krall wrote:
|
|Has anybody used the sangoma fxo cards with asterisk? Anybody using 
|multiple cards? Problems with irq and such (same as with digium ones)?
|
| 
|
||-Original Message-
||From: [EMAIL PROTECTED]
||[mailto:[EMAIL PROTECTED] On Behalf Of John 
||Novack
||Sent: Wednesday, April 12, 2006 10:29 AM
||To: [EMAIL PROTECTED]
||Cc: Asterisk Users Mailing List - Non-Commercial Discussion
||Subject: Re: [Asterisk-Users] Performance: Xeon or Opteron?
||
||
||
||Rich Adamson wrote:
||
||
|| While talking with one of the sangoma folks very recently, he was 
|| rather emphatic the pci bus was designed to share
||interrupts. I was
|| a little concerned as a test server had the wanpipe driver
||sharing an
|| interrupt with libata and uhc1_hcd. His comment was 
|that's the way 
|| its suppose to work, sharing interrupts as needed. I've 
|not had any 
|| recognizable issues with the A200D card at all, and faxing
||via a A200D
|| fxs port to a A200D fxo (pstn) port functions 100% reliably.
||
|| What that would suggest is the TDM400 pci firmware 
|(whether on card 
|| logic or whatever) is the source of at least part of the
||TDM400 shared
|| interrupt issue. I don't have any digium T1/E1 cards 
|laying around, 
|| but if memory serves correctly, the T1/E1 cards do not use 
|the same 
|| pci controller chip. That would suggest the T1/E1 cards are
||less of an
|| issue then with the TDM400 card.
||
||That's good to know, but considering the response from Digium on the 
||TDM400 ( try another motherboard) when there didn't seem to 
|even be an 
||int. sharing issue, the card just couldn't be seen at all , and the 
||support I received from Sangoma on a recent FXS issue that was 
||resolved within a few days, I would tend to go with Sangoma 
|for the T1 
||card, if and when I have the need.
||
||John Novack
||
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||
|
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RE: [Asterisk-Users] Digium cards, so disappointing !

I must agree with you. I too buy Digium cards because I want to support the
development of asterisk. Asterisk is a great product but digum cards are a
pain, they say they don't support faxing but a lot of people that are
implementing asterisk demand or need faxin as a day to day service on
their PBX's.

Sad to see that faxing is nearly impossible on digium cards. To me is like
saying here you have a great car but.. It cannot handle a car stereo :(
 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Tony ROBIN
|Sent: Thursday, April 13, 2006 7:01 PM
|To: asterisk-users@lists.digium.com
|Subject: [Asterisk-Users] Digium cards, so disappointing !
|
|
|I am so fed up with Digium cards. My company first owned a 
|TE410P, I installed it in a Dell server and enjoyed its 
|instability (we bought it months before Digium warned about 
|the incompatibility issues). Then we switched to a TE411P for 
|the hardware echo cancellation. Now we want to receive fax ( 
|20/day) on it and guess what ? Since April 2006 (again a few 
|months after we bought our brand new card), officially, fax 
|communications is not supported with Digium cards ( 
|http://www.voip-info.org/wiki-Asterisk+fax ).
|Of course, I should have guessed that it is far too much to 
|ask to a $2495 card ! Is the fax extension in Asterisk just 
|there to push us to the competing products ?
|
|We hesitated to buy another Digium card after the problems 
|with TE410P, but I told myself it was nice to support Asterisk 
|by buying some Digium cards. Now Digium make us regret our 
|buys and a disappointed customer is a lost customer forever... 
|Too sad...
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RE: [Asterisk-Users] Digium cards, so disappointing !

Problem is, how to make sure you system WILL have 100% on zttest before
buying the cards.. You need to have stability, compatibility and certainty
that what you buy is going to work :(

Anybody had similar problems or success stories with sangoma cards? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Lee Howard
|Sent: Thursday, April 13, 2006 7:22 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Digium cards, so disappointing !
|
|Tony ROBIN wrote:
|
|Now we want to receive fax ( 20/day) on it and guess what ? Since 
|April 2006 (again a few months after we bought our brand new card), 
|officially, fax communications is not supported with Digium cards ( 
|http://www.voip-info.org/wiki-Asterisk+fax ).
|Of course, I should have guessed that it is far too much to ask to a 
|$2495 card ! Is the fax extension in Asterisk just there to push us 
|to the competing products ?
|
|
|If your zttest has good results (mostly 100%, nothing less 
|than 99.98%) then you should be able to receive faxes (I'd 
|suggest iaxmodem+HylaFAX) despite Digium's disclaimer.
|
|I do not excuse Digium, however, from sidelining fax the way 
|that they have.
|
|Lee.
|
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RE: [Asterisk-Users] Digium cards, so disappointing !

Aaron, have you tried using 1 te110p and 2 tdm04b on the same server? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Aaron Daniel
|Sent: Thursday, April 13, 2006 7:19 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Digium cards, so disappointing !
|
|*shrugs* Ya win some ya lose some.  We've spent about 10 grand 
|plus on Digium cards and have been pretty satisfied with ours 
|:) Faxes have been working great for over 6 months and the 
|cards work wonderfully in our Dell servers.  They just need 
|more documentation on the different configuration options you 
|can pass on load... I think the only problems we've really had 
|are configuration related, or bad hardware on our part, oh, 
|and a server room fry that took out more than just the 
|Asterisk servers :-P
|
|Aaron
|
|On Fri, 14 Apr 2006, Tony ROBIN wrote:
|
|
| I am so fed up with Digium cards. My company first owned a TE410P,
| I installed it in a Dell server and enjoyed its instability (we
| bought it months before Digium warned about the incompatibility
| issues). Then we switched to a TE411P for the hardware echo
| cancellation. Now we want to receive fax ( 20/day) on it and
| guess what ? Since April 2006 (again a few months after we bought
| our brand new card), officially, fax communications is not
| supported with Digium cards ( 
|http://www.voip-info.org/wiki-Asterisk+fax ).
| Of course, I should have guessed that it is far too much to ask
| to a $2495 card ! Is the fax extension in Asterisk just there
| to push us to the competing products ?
|
| We hesitated to buy another Digium card after the problems with
| TE410P, but I told myself it was nice to support Asterisk by
| buying some Digium cards. Now Digium make us regret our buys and
| a disappointed customer is a lost customer forever... Too sad...
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|
|
|-- 
|Aaron Daniel
|Computer Systems Technician
|Sam Houston State University
|[EMAIL PROTECTED]
|(936) 294-4198
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|

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Re: [Asterisk-Users] Digium cards, so disappointing !

2006-04-14 Thread Matt Riddell (IT)

Anton Krall wrote:
 I must agree with you. I too buy Digium cards because I want to support the
 development of asterisk. Asterisk is a great product but digum cards are a
 pain, they say they don't support faxing but a lot of people that are
 implementing asterisk demand or need faxin as a day to day service on
 their PBX's.
 
 Sad to see that faxing is nearly impossible on digium cards. To me is like
 saying here you have a great car but.. It cannot handle a car stereo :(

Is this not possibly also related to the patenting issues on the email
to fax gateways?

-- 
Cheers,

Matt Riddell
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[Asterisk-Users] tdm2400p and asterisk 1.2.1


Hi,
I'd like to change 3 TDM400p with one TDM2400p to avoid echo (my 
intention is to use a TDM2400P echo cancel module). It TDM2400p working 
good with asterisk 1.2.1? Or I need to install a new asterisk version?


TIA

Giorgio Incantalupo
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Re: [Asterisk-Users] Asterisk hardware for new office suggestion


Simone wrote:

Hi list,
I am in the process of setting up Asterisk for a new office and since 
this is going to be my first real installation I'd appreciate some 
advice on the hardware from the real world. We will have 8 channels 
(still not sure if 4xISDN2 or ISDN30 8 channels, but I will definitely 
go for a Digium card with echo hw cancelation) and a DSL 2mbit line (QoS 
on the switch and firewall?), to be configured for both traditional and 
VoIP usage . I was looking at the Xorcom TS-1 server and I was wondering 
if you would recommend it for a 30 employees office or if you'd rather 
build it on a normal server (would a double PIII 1Ghz be enough), and 
also if you could give a suggestion on the phones (we will get an HP 
Gbit switch PoE).

Thanks, any hint really appreciated


Would suggest digging around the wiki as there really is a lot of useful 
info that would help you.


I don't have any isdn stuff, so can't comment on that.

QoS on the switch in a small office is not likely to do anything useful. 
It may help some on the firewall, but it really only impacts outbound 
packets, not inbound. A rather inexpensive way around all that is to 
simply implement a second dsl link used only for voip.


The size of the * box is more oriented around number of simultaneous 
calls and other apps running on the box.  E.g., if the 30 employees 
never place a call, a 600 mhz processor will be just fine. ;)


As I recall (which leaves accuracy questionable), several people have 
implemented a basic * system on old 600 mhz boxes. However, a call 
center with lots of call recording functions (etc) and high call volumes 
may require the largest/fastest processor money can buy.


There is no magic list comparing sip phone quality, features, etc. Lots 
of reasonably good comments on the wiki, but that's about it. Lots of 
opinions, but keep in mind that what one person with soho-only 
experience considers good is highly likely to be rated poor by someone 
that supports a large corporate environment. Interpretations of 
quality varies dramatically based on each person's experience level.


An individual in a less developed country might consider a softphone on 
a PC high quality (compared to their existing telephony infrastructure), 
but another person in a more well developed country would not use a 
softphone in production if their life depended on it.


For sip phones, I'd suggest starting with identifying some basic 
requirements and go from there. For example, if you don't want to do any 
additional cat5 cabling, using a phone with two rj45's (internal 
ethernet switch) might be a requirement. Is speakerphone capability 
needed? How many extensions are truly needed at each desk? Is intercom 
paging required (to individual sip phones)? Is the key system emulation 
of a busy lamp field required? Is PoE truly required? If the answer to 
just those questions are a mandatory yes, you've just eliminated about 
80% of the sip phones on the market.


I'd expect you to find a need for two or three different types of phones 
somewhat oriented around high-quality (exec types), medium (average 
office worker), and lower quality (break rooms or occasional user).


And, don't overlook the differences between key systems and pbx's. If 
your customer is accustomed to an existing key system, they are likely 
to be very disappointed with a pbx unless you spend a fair amount of 
time up front educating the employees and managing expectations (way 
before deployment).


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Re: [Asterisk-Users] tdm2400p and asterisk 1.2.1

Giorgio Incantalupo wrote:

 I'd like to change 3 TDM400p with one TDM2400p to avoid echo (my
 intention is to use a TDM2400P echo cancel module). It TDM2400p working
 good with asterisk 1.2.1? Or I need to install a new asterisk version?

There is no reason not to upgrade to the latest Asterisk version
(barring the little snafu we had yesterday with the 1.2.7 release). The
same is true for upgrading Zaptel; you will get better results using
Zaptel 1.2.5 instead of 1.2.1 or something older.
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Re: [Asterisk-Users] Performance: Xeon or Opteron?

I'd have to guess that combination of cards with almost any mobo would 
be considered an overloaded system. If you replaced the two TDM04b cards 
with an A200D or TDM2400 card, most of those irqmisses (etc) would 
probably go away; but that's a somewhat educated guess on my part.


Factually, the sangoma cards integrate with the pci bus in a much more 
stable/usable way then does the digium TDM card (and I believe the te110 
if it uses the TigerJet pci chip).



Anton Krall wrote:

My main concerns would be, can you have multiple cards like this on a
system, for example, I now have a te110p and 2 tdm04b and Im getting
irqmisses on the te110p (according to zttool and zttest) which makes fax
receiving on the te110p almost impossible.. Plus, voice is getting frame
slips.

I was hoping sangoma cards could be more enterprise friendly. 


|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|John Novack

|Sent: Thursday, April 13, 2006 4:57 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Performance: Xeon or Opteron?
|
|Have a 2FXO 2FXS card working now.
|More forgiving of the PCI bus. The software installation is a 
|little mean, with the outlined procedure making assumptions 
|about the installers knowledge and familiarity with Sangoma 
|products, and in some places it doesn't really discriminate 
|between their T1 and A200 cards.

|I found one defect in their FXS driver, which they have now fixed.
|
|Overall seems to be a good product, slightly more affordable 
|and less of a problem child than the Digium/TigerJet TDM400

|
|John Novack
|
|Anton Krall wrote:
|
|Has anybody used the sangoma fxo cards with asterisk? Anybody using 
|multiple cards? Problems with irq and such (same as with digium ones)?

|
| 
|

||-Original Message-
||
||Rich Adamson wrote:
||
||
|| While talking with one of the sangoma folks very recently, he was 
|| rather emphatic the pci bus was designed to share

||interrupts. I was
|| a little concerned as a test server had the wanpipe driver
||sharing an
|| interrupt with libata and uhc1_hcd. His comment was 
|that's the way 
|| its suppose to work, sharing interrupts as needed. I've 
|not had any 
|| recognizable issues with the A200D card at all, and faxing

||via a A200D
|| fxs port to a A200D fxo (pstn) port functions 100% reliably.
||
|| What that would suggest is the TDM400 pci firmware 
|(whether on card 
|| logic or whatever) is the source of at least part of the

||TDM400 shared
|| interrupt issue. I don't have any digium T1/E1 cards 
|laying around, 
|| but if memory serves correctly, the T1/E1 cards do not use 
|the same 
|| pci controller chip. That would suggest the T1/E1 cards are

||less of an
|| issue then with the TDM400 card.
||
||That's good to know, but considering the response from Digium on the 
||TDM400 ( try another motherboard) when there didn't seem to 
|even be an 
||int. sharing issue, the card just couldn't be seen at all , and the 
||support I received from Sangoma on a recent FXS issue that was 
||resolved within a few days, I would tend to go with Sangoma 
|for the T1 
||card, if and when I have the need.

||
||John Novack


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Re: [Asterisk-Users] Asterisk hardware for new office suggestion

2006-04-14 Thread Tim Panton



On 14 Apr 2006, at 11:29, Simone wrote:


Hi list,
I am in the process of setting up Asterisk for a new office and  
since this is going to be my first real installation I'd  
appreciate some advice on the hardware from the real world. We will  
have 8 channels (still not sure if 4xISDN2 or ISDN30 8 channels,  
but I will definitely go for a Digium card with echo hw  
cancelation) and a DSL 2mbit line (QoS on the switch and  
firewall?), to be configured for both traditional and VoIP usage .  
I was looking at the Xorcom TS-1 server and I was wondering if you  
would recommend it for a 30 employees office or if you'd rather  
build it on a normal server (would a double PIII 1Ghz be enough),  
and also if you could give a suggestion on the phones (we will get  
an HP Gbit switch PoE).

Thanks, any hint really appreciated

Simone


I can only base my advice on what we have done for a smaller office.

If you want 8 lines it is probably as cheap to go for ISDN 30 as for  
4xBRI

at least it is here in the UK.

We have a single span E1 card from digium without echo can in a small  
1U rack mounted server
(spec: 1Ghz Via processor and  512Mb ram). The Via might be a bit  
underpowered for 30 users, but
unless you are transcoding, virtually any modern processor would be  
fine for 8 lines.


You need to look out on the DSL line if it is ADSL, since they have  
low upstream bandwidth.
Heavy outgoing mail messages (eg attachments sent to distribution  
lists) can easily fill the outgoing

(256kbit/s) pipe degrading the voice quality.

I'm very fond of the SNOM phones - elmeg are selling the old SNOM 190  
model
which is a decent office phone. For 30 you should be able to get them  
for less than £70 each.

I've got 6 - 4 SNOMs and 2 elmegs - No problems with any
of them, but they don't support PoE, so you may want to look at other  
models.


Don't underestimate how much training/doc you will need to provide to  
get people going on the new system.
They may have been using the old one for years and written little  
cribsheets about how to transfer etc.




Tim Panton
[EMAIL PROTECTED]



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Re: [Asterisk-Users] Digium cards, so disappointing !

I believe the TDM2400 has the capability of doing on-card fxo-fxs data 
flows (without hitting the pci bus), but that function has not yet been 
implemented. Its basically required to support faxes in an analog 
environment. When it is implemented, that card should work. The TDM400 
card will not work in 99% of the deployments.


Faxing via T1 cards is known to work in a fairly large number of 
deployments, but its likely to be highly dependent on exactly where the 
fax machine is located relative to *. Eg, incoming pstn fax via a T1 
that is expected to be switched to a sip ata adapter has lots of 
technical and specific infrastructure dependencies that have to be 
addressed by the implementor / engineer. The plug-n-play approach will 
have a very high failure rate.



Anton Krall wrote:

I must agree with you. I too buy Digium cards because I want to support the
development of asterisk. Asterisk is a great product but digum cards are a
pain, they say they don't support faxing but a lot of people that are
implementing asterisk demand or need faxin as a day to day service on
their PBX's.

Sad to see that faxing is nearly impossible on digium cards. To me is like
saying here you have a great car but.. It cannot handle a car stereo :(
 


|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Tony ROBIN

|Sent: Thursday, April 13, 2006 7:01 PM
|To: asterisk-users@lists.digium.com
|Subject: [Asterisk-Users] Digium cards, so disappointing !
|
|
|I am so fed up with Digium cards. My company first owned a 
|TE410P, I installed it in a Dell server and enjoyed its 
|instability (we bought it months before Digium warned about 
|the incompatibility issues). Then we switched to a TE411P for 
|the hardware echo cancellation. Now we want to receive fax ( 
|20/day) on it and guess what ? Since April 2006 (again a few 
|months after we bought our brand new card), officially, fax 
|communications is not supported with Digium cards ( 
|http://www.voip-info.org/wiki-Asterisk+fax ).
|Of course, I should have guessed that it is far too much to 
|ask to a $2495 card ! Is the fax extension in Asterisk just 
|there to push us to the competing products ?

|
|We hesitated to buy another Digium card after the problems 
|with TE410P, but I told myself it was nice to support Asterisk 
|by buying some Digium cards. Now Digium make us regret our 
|buys and a disappointed customer is a lost customer forever... 
|Too sad...


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Re: [Asterisk-Users] Segfault on Inbound call?

2006-04-14 Thread Matt

I'm speculating here... but why does Sangoma need to patch zaptel
source?!?  Can't they just use the same interfacing that the digium
cards do?

On 4/14/06, Rich Adamson [EMAIL PROTECTED] wrote:
 Interesting... I don't have those problems with the A200D. In fact,
 haven't found any problems at all in about two months of production use.
 I have noticed that mixing a TDM400 with a A200D in the same box has an
 issue with sangoma code disabling echotraining=800 (needed for the TDM).
 But that's it.


 Matt wrote:
  Well and that is where my love hate relationship with Sangoma is.
  While their technical support seems to be very knowledgeable it
  takes forever to get ahold of them!   Additionally I've found 2 bugs
  now in their drivers   At first setup wouldn't install... it was a
  bug with their setup not creating a file if it was missing.   And now
  there is some caller-id bug in their software.   When it patched the
  zaptel source... if I have usecallerid=yes on then it crashes... if I
  turn usecallerid=no then it is fine.
 
  On 4/13/06, John Novack [EMAIL PROTECTED] wrote:
  Have you asked Sangoma support?
 
  Though they can be slow to respond to an E-mail, they can quickly tell
  you if it is their problem or not
  I am using a 2 FXO 2FXS with Asterisk  SVN-branch-1.2-r13026 with no
  problems on CentOS 3.6
 
  John Novack
 
 
  Matt wrote:
 
  I'm running asterisk 1.2.6 with Sangoma A200 4 port FXO card.
  Outbound calls are working fine.  However, when I have an inbound
  call.. asterisk will segfault.. and then start again ... then it will
  take  1 call fine
 
  I'm running asterisk with a -U and -G of asterisk.  Any thoughts?
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RE: [Asterisk-Users] Will VoIP ITSP's be Next?

2006-04-14 Thread Alexander Lopez

Title: [Asterisk-Users] Will VoIP ITSP's be Next?

Comsumer acceptance of VoIP has caused the TelCo's to
rethink their profitablility, but that so no secrect to those on this
list.

It would appear that the treat is real enough for
others, such as Googleto start taking preventive
measures
http://www.ivt.com.au/latestnews/id/68

VoIP as a technology is the way to go, especially with
convergence. However, as it has been stated before, with the Public Internet you
cannot control QoS or even connectivity. This solidifies that
fact.

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wai
WuSent: Thursday, April 13, 2006 8:49 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: RE:
[Asterisk-Users] Will VoIP ITSP's be Next?

I would say what is going
to prevent content providers like google and yahoo becoming telcos. Now they
too would have their peer arrangement.

From: [EMAIL PROTECTED]
on behalf of Bob's Leaky News ServiceSent: Thu 4/13/2006 8:26
PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: [Asterisk-Users] Will VoIP ITSP's be
Next?

Will VoIP be Next?Telco's that provide Internet
services to their customers are nowtrying to charge select companies for
large volumes of content thatpass over their network to their paying
customers! What part of this"greed fest' makes any sense to you? Telco's
sell DSL tellingcustomers how much faster it is, how much they can do with
HighspeedInternet connections and now they want to charge non-customers
for theright to see their paying customers. CONFUSED? Damn so am I.
Telco'spay nothing for Internet connections as they get their traffic
fromwhat are called peering agreements. Say I have a bunch of
customersand you have a bunch of customers...Then it is in our best
interest topeer so our customers can communicate with each other. This is
done99% of the time at NO COST to company A or company B. Now
companieslike Verizon are demanding companies like Google pay them stating
thatbandwidth intensive content is being passed over their networks
andthey should be compensated. Ok so you flip to the other side and
theVerizon DSL customer has been sold on the fact services like
Google,Yahoo and many many smaller just plain super Internet goodies are
nowgoing to travel into their computers at highspeed. Why the
doublestandard? GREED. That is it just plain old fashion GREED. The past
fewyears have not been good for Telco's. Cable companies are able
tooffer better service and are now entering the phone markets as
well.Hey call me stupid but can we see where this is going? From the How
DoI Get More Profit Department Mr. Telco is at it again. The customer
ispaying for access to a river of sorts. That river is called the
GlobalInternet. The Telco decided to enter that market. The Telco decided
toagree to PEER and accept traffic from other Global Internet
providers.The Telco decided to spend millions telling customers BUY
OURHIGHSPEED DSL service so they could access bandwidth intensive
contentlike music, TV, large files and more. Huge marketing plans are
inaction daily. So what have these customers really bought?
Goodquestion. Maybe it is time for a Telco customer to get an
Internetsavvy lawyer or two and sue the hell out of these companies that
aretrying to restrain competition, block free trade and maybe just
maybeeven eliminate services their customers count on to run their
dailybusiness.To the TELCO I say "Folks pull your heads out of
your collectiveasses" or your going to fail at this worse then you have
failed atDSL. Stick with your core business and stop trying to profit off
theinventions of others. Most of all stop trying to restrain trade.
Ifyou fail to hear the phone ring it is going to cost your
shareholdersa huge pile of cash one day very soon.PS: The Internet
is a place where new ideas come and go by the minute.If this sort of
stupid nonsense is allowed to start it will never end.What has become a
world without borders, the connected will fastbecome the disconnected. Mr.
Telco the Internet happened while all ofyou were asleep at the "switch" so
wipe that evil smile off yourfaces. To me Telcos are nothing more than
OED's ( One Eyed Droids
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Re: [Asterisk-Users] Digium cards, so disappointing !


Anton Krall wrote:

Problem is, how to make sure you system WILL have 100% on zttest before
buying the cards.. You need to have stability, compatibility and certainty
that what you buy is going to work :(

Anybody had similar problems or success stories with sangoma cards? 


Running zttest on my box with both a TDM04b and A200D installed 
indicates and average of 99.96% for both. Not sure how accurate that 
might be as the A200D card appears as a 24 channel interface in terms of 
/dev/zap even though only four ports are equipped.


The TDM04b won't support faxes on this box under any circumstances and 
I've played around with about every possible pci latency, etc, change 
that folks have suggested in the last two years.


Based on my heavily invested testing to date (which includes about two 
years of doing this), the only usable fax support thus far comes from 
using the A200D card with the fax machine directly connected to a fxs 
port on that card, and an fxo (pstn) port on the exact same card. Those 
fax tests have been 100% solid using a cheap/older Brother fax machine.




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Re: [Asterisk-Users] tdm2400p and asterisk 1.2.1


Hi Kevin,
I know upgrading is better, sorry, maybe my question was malformed...the 
exact question is which is the minimum asterisk version supporting 
TDM2400P?
(I have 10 pbx and  I want to change 3 TDM400P with one TDM2400P on 
every pbx without reinstalling a new asterisk version on every machine!!)


TIA

Giorgio Incantalupo



Kevin P. Fleming wrote:

Giorgio Incantalupo wrote:

  

I'd like to change 3 TDM400p with one TDM2400p to avoid echo (my
intention is to use a TDM2400P echo cancel module). It TDM2400p working
good with asterisk 1.2.1? Or I need to install a new asterisk version?



There is no reason not to upgrade to the latest Asterisk version
(barring the little snafu we had yesterday with the 1.2.7 release). The
same is true for upgrading Zaptel; you will get better results using
Zaptel 1.2.5 instead of 1.2.1 or something older.
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[Asterisk-Users] asterisk with bluetooth headset howto

2006-04-14 Thread Andreas Nitsche

Hello!

What do I need to use the asterisk on my notebook with a bluetooth
headset?

Is there anywhere a good howto?

Thanks!

Andi
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[Asterisk-Users] change/toggle flash operator panel components


Hi,
is it possible to remove the no timeout combo box in flash operator panel?
How can I reduce the flash area? I set small buttons and half of the 
area is white and I want to resize it.


TIA

Giorgio Incantalupo
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RE: [Asterisk-Users] asterisk with bluetooth headset howto

2006-04-14 Thread Alexander Lopez

I think that the BT channel in Asteisk is to support a phone connection
via Bluetooth, (ie asteisk mimics your Headset, and a bit more).

If you have a Bluetooth chipset in your laptop or add one via USB. Any
softphone that can you the audio to/form your headset should work, I do
not know if ALSA or OSS will, so using the console may not be an option
if you are running * on your Laptop.

 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Andreas Nitsche
 Sent: Friday, April 14, 2006 9:39 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] asterisk with bluetooth headset howto
 
 Hello!
 
 What do I need to use the asterisk on my notebook with a 
 bluetooth headset?
 
 Is there anywhere a good howto?
 
 Thanks!
 
 Andi
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Re: [Asterisk-Users] Digium cards, so disappointing !

2006-04-14 Thread Rusty Dekema

On 4/14/06, Rich Adamson [EMAIL PROTECTED] wrote:
 I believe the TDM2400 has the capability of doing on-card fxo-fxs data
 flows (without hitting the pci bus), but that function has not yet been
 implemented. Its basically required to support faxes in an analog
 environment. When it is implemented, that card should work. The TDM400
 card will not work in 99% of the deployments.

I apologize if I'm being dense, but I don't understand why the fact
that a call traverses the PCI bus would kill a fax transmission. I
made the following setup, and it consistently gets 31200 and 33600
connects, no disconnects, good throughput:

Modem --- SIP ATA(G.711u) --- (LAN) --- Asterisk --- (TDM400FXS) --- Modem

I get the same thing (although alwyas 31200 connects, never 33600) with:

Modem --- (TDM400FXS) --- Asterisk --- (TDM400FXS_same_card) --- Modem

If this works, I don't see why a fax transmission wouldn't work. Is it
because the fax protocol doesn't have error correction? Is that even
true?

I realize that clearing up my confusion about this isn't probably
going to result in the problem being fixed, but I sure would like to
know...

-Rusty
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Re: [Asterisk-Users] tdm2400p and asterisk 1.2.1

Giorgio Incantalupo wrote:
 Hi Kevin,
 I know upgrading is better, sorry, maybe my question was malformed...the
 exact question is which is the minimum asterisk version supporting
 TDM2400P?
 (I have 10 pbx and  I want to change 3 TDM400P with one TDM2400P on
 every pbx without reinstalling a new asterisk version on every machine!!)

That is not the correct question; Asterisk does not talk to the hardware
directly, it uses Zaptel.

I would not suggest converting to TDM2400Ps without also upgrading to
the latest 1.2.x Zaptel release; the driver for that card has undergone
some modifications since it was originally added.
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Re: [Asterisk-Users] Digium cards, so disappointing !

Rusty Dekema wrote:

 If this works, I don't see why a fax transmission wouldn't work. Is it
 because the fax protocol doesn't have error correction? Is that even
 true?

FAX transmission is massively more complex than modem transmission. At
higher speeds, it involves 3 or 4 different 'carrier' frequencies and
signaling rate shifts, and these are done with very critical timing
requirements.

Yes, error correction is available, but it just means that sending FAXes
over a lousy connection will take a very long time, instead of failing
completely.
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Re: [Asterisk-Users] asterisk with bluetooth headset howto

2006-04-14 Thread Andreas Nitsche

My bluetooth headset works with the snd-bt-sco module. I also compiled
chan_bluetooth into my asterisk but I can't find any howto for using a
bt headset with asterisk.

Does this just work with a smartphone as gateway? Is it not possible
to connect the bt headset directly to my notebook?

Andi

On Fri, Apr 14, 2006 at 09:47:09AM -0400, Alexander Lopez wrote:
 I think that the BT channel in Asteisk is to support a phone connection
 via Bluetooth, (ie asteisk mimics your Headset, and a bit more).
 
 If you have a Bluetooth chipset in your laptop or add one via USB. Any
 softphone that can you the audio to/form your headset should work, I do
 not know if ALSA or OSS will, so using the console may not be an option
 if you are running * on your Laptop.
 
  
 
  -Original Message-
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  Andreas Nitsche
  Sent: Friday, April 14, 2006 9:39 AM
  To: asterisk-users@lists.digium.com
  Subject: [Asterisk-Users] asterisk with bluetooth headset howto
  
  Hello!
  
  What do I need to use the asterisk on my notebook with a 
  bluetooth headset?
  
  Is there anywhere a good howto?
  
  Thanks!
  
  Andi
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Re: [Asterisk-Users] Digium cards, so disappointing !


Rusty Dekema wrote:

On 4/14/06, Rich Adamson [EMAIL PROTECTED] wrote:

I believe the TDM2400 has the capability of doing on-card fxo-fxs data
flows (without hitting the pci bus), but that function has not yet been
implemented. Its basically required to support faxes in an analog
environment. When it is implemented, that card should work. The TDM400
card will not work in 99% of the deployments.


I apologize if I'm being dense, but I don't understand why the fact
that a call traverses the PCI bus would kill a fax transmission. I
made the following setup, and it consistently gets 31200 and 33600
connects, no disconnects, good throughput:

Modem --- SIP ATA(G.711u) --- (LAN) --- Asterisk --- (TDM400FXS) --- Modem

I get the same thing (although alwyas 31200 connects, never 33600) with:

Modem --- (TDM400FXS) --- Asterisk --- (TDM400FXS_same_card) --- Modem

If this works, I don't see why a fax transmission wouldn't work. Is it
because the fax protocol doesn't have error correction? Is that even
true?

I realize that clearing up my confusion about this isn't probably
going to result in the problem being fixed, but I sure would like to
know...


The fax issue revolves around the fact that fax signals are analog audio 
(eg, modem) that have to be accurately reproduced end-to-end (whatever 
that happens to mean in your environment).


If a fax machine is attached to a sip ata device, the network 
infrastructure has to be 100% rock solid (no dropped packets, no 
congestion, relatively low utilization, no contention for resources 
anywhere between asterisk and the adapter). If those items are unknown 
or poor, the analog fax signal will not be accurately reproduced at any 
sip ata device.


Likewise, the transfer of data across the pci bus has to be 100% 
accurate with no dropped/slipped packets, no jitter, etc. That has 
been an issue with an estimated 95% of the TDM implementations to date.


If you go visit some of the sites where developers work with real time 
audio, you'll find lots of comments relative to the inadequacies of the 
pci bus as implemented on many many mobo's. Most of what I've read 
relates to the North/South pci bridge chipsets, and design errors in 
those chipsets including some of the Intel products. Those same issues 
seem to be impacting the TDM card in one form or another, and no one has 
openly put a finger on exactly why.


The majority of implementations that need fax capability is related to 
an internal fax machine (however its connected) to an fxo (pstn) port. 
If you have that working via a TDM400 card, everyone on this list 
would love to know exactly which mobo you are using.


A modem's error detection / error correction capability (which does not 
exist in a lot of fax machines and point of sale devices) can handle 
small amounts of those issues noted above. However, it cannot handle 
anything more then a glitch here and there without significantly 
impacting data throughput.



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Re: [Asterisk-Users] still no solution for me, if one provider fails.

2006-04-14 Thread Ronald Wiplinger


Peter J Dean wrote:
We do it slightly different, rather than multiple macros, we do it 
within a single macro.



Peter,

I have to this some questions:
1. I have not seen n(tryiax01) construction before. Can you explain 
it, please and how you give this to the macro?

I know only exten = s,4,Goto(s-${DIALSTATUS},1)

2. Your macro covers only CHANUNAVAIL and CONGESTION
There are more than that, like BUSY, CANCEL, NOANSWER, ANSWER
What does each one of them exactly mean? When is it CONGESTION and when 
is it BUSY? When is CHANUNAVAIL and when NOANSWER? There is a very fine 
line between.


3. You are using the options tT the called and the calling party can 
transfer the call. When is important that the calling party can transfer 
a call? If we use tT or other options, we cannot use anymore 
canreinvite=yes   - or when and when can we not do that?
I did not add any options to my users, however, if they do not hear the 
ring, they are not happy! On the other side, I cannot route all calls 
through my *


4. If a call cannot be completed, than I would like to know it!!!
I tried to accomplish that once:
;exten = _9011Z.,513,SYSTEM(mail -s 'VPBX all lines in use' 
[EMAIL PROTECTED])

However, this did not work. Do you know what I made wrong here?

;
;
;
[macro-outbound-calling]
exten = s,1,NoOp(Debug:  Outbound Call from ${CALLERID})
;
exten = s,n(tryiax01),NoOP(Debug [${CONTEXT}]: Trying 1st IAX2 
Service)

exten = s,n,Dial(IAX2/${CarrierA}/${ARG1},60,tT)
exten = s,n,GotoIf($[${DIALSTATUS} : 
(CHANUNAVAIL|CONGESTION)]?tryiax02:Hangup)

;
exten = s,n(tryiax02),NoOP(Debug [${CONTEXT}]: Trying 2nd IAX2 
Service)

exten = s,n,Dial(IAX2/${CarrierB}/${ARG1},60,tT)
exten = s,n,GotoIf($[${DIALSTATUS} : 
(CHANUNAVAIL|CONGESTION)]?trypstn:Hangup)

;
exten = s,n(trypstn),NoOP(Debug [${CONTEXT}]: Now trying the PSTN 
Backup Lines)

exten = s,n,etc, etc
;
exten = s,n,Hangup()

Then the extension string becomes;

exten = _9011Z.,103,Macro(outbound-calling,${EXTEN})


On 11/04/2006, at 6:55 PM, Mimmus wrote:




I have now:
exten = _9011Z.,103,Dial(SIP/011${EXTEN:[EMAIL PROTECTED])
;exten = _9011Z.,103,Dial(SIP/011${EXTEN:[EMAIL PROTECTED])
;exten = _9011Z.,103,Dial(SIP/011${EXTEN:[EMAIL PROTECTED])
exten = _9011Z.,104,NoOp(${DIALSTATUS})


I configured two trunks for my outgoing calls:

 [outrt-001-out]
 exten = _0.,1,Macro(dialout-trunk,2,${EXTEN:1},)
 exten = _0.,2,Macro(dialout-trunk,5,${EXTEN:1},)
 exten = _0.,3,Macro(outisbusy) ; No available circuits

If first fails, second is automatically used but I get a CDR with
disposition = 'FAILED'. How can I avoid this?

--
Domenico Viggiani

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--
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Tel. (M) +886.939.775.516  (O) +886.2.2835.7765 (ENUM)   or FWD 511208

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[Asterisk-Users] Work available - India

2006-04-14 Thread Sahil Gupta


Hi there,
If there is anybody on-list looking for VoIP related work in India, please 
contact me off=list with your details.


Positions are of a full-time nature.

Regards,


Sahil Gupta
VoiceValley
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[Asterisk-Users] A weekend of upgrade is coming for me - any hints?

2006-04-14 Thread Ronald Wiplinger


I want to upgrade * this weekend.

What can I prepare? What will I have to change in the settings? Where 
can I read about it?


I use now:
*CLI show version
Asterisk SVN-trunk-r8447M built by root @  on a x86_64 running Linux 
on 2006-01-25 15:33:01 UTC


bye

Ronald


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Re: [Asterisk-Users] Callback Agents and Dial 'g' option

2006-04-14 Thread Johann

Small update, I've been able to sort of work around the problem by making the 
AgentcallbackLogin() direct to a context that in turn does another dial over a 
local channel with the /n that gets around part of the problem.  Still kinda 
nasty seeing 5 channels around for 1 call...



--johann

Johann wrote:
I'm unable to get the Dial option 'g' to work with callback agents.  The 
plan is to use it so that I can redirect a customer to a menu so they 
can rate the call they just had with the agent.  However, when the agent 
hangs up the call does not continue in the dialplan.


I login with the agent.  Call joins the queue.  The agent and call get 
connected.  The agent hangs up and the call should continue to the 
Playback(beep) and the Noop(), however the call is hung up on both sides.


Extensions.conf:
[default]
; Handle login and logout
exten = ,1,Agentcallbacklogin(1,,[EMAIL PROTECTED])
exten = ,1,AgentCallbackLogin(1,s)

; join the queue
exten = ,1,Answer
exten = ,2,Queue(testing)

[queue]
exten = 1,1,Dial(Sip/4000||got)
exten = 1,2,Playback(beep)
exten = 1,3,Noop(Jump to the QA menu now)

Any ideas?


--johann
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[Asterisk-Users] Packet Testing

2006-04-14 Thread Kevin Smith


Hi everyone,

On the Polycom 601 phones we are using, the forward feature works very 
nicely for agents that are out on trips. I was wondering if there is a 
way to test to see if they have the forward option enabled.


When it is enabled the call comes in and gets -- Got SIP response 302 
Moved Temporarily response and then it uses the correct outbound macro 
to forward the call to the number specified. I am wondering if I am able 
to test for that SIP response or something in the SIP packet that I can 
grab to test. From what I read online, I didn't see much that would 
allow me to test for it, but  I may have just missed it.


Thanks,
Kevin
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[Asterisk-Users] Background music in call

2006-04-14 Thread nzrh

Hi all,

I urgently need a solution in a part of a project. 
I appreciate all types of help.

The thing I absolutely need is. To play a background
music in call. 
If I have the opportunity to stop it via entering a
dtmf combination is would be very very nice also.
Does anybody know some application do this.


NZR


__
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
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Re: [Asterisk-Users] A weekend of upgrade is coming for me - any hints?


Ronald Wiplinger wrote:

I want to upgrade * this weekend.

What can I prepare? What will I have to change in the settings? Where 
can I read about it?


I use now:
*CLI show version
Asterisk SVN-trunk-r8447M built by root @  on a x86_64 running Linux 
on 2006-01-25 15:33:01 UTC


As Kevin as already posted, upgrading to current trunk is not 
advisable (other then for testing) as significant changes are in the 
middle of being implemented. Your probability of having a working system 
is likely to be very low. Maybe in another week or two.


Read the CHANGES text file, read the files that reflect recent changes 
in asterisk/conifgs, and read files that have changed in asterisk/doc.


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[Asterisk-Users] Unicall and Fax

2006-04-14 Thread Carlos Chavez

 Has anyone been able to send a fax through a Unicall channel?  I am
unable to send or receive faxes using either rxfax or a fax machine connected
to an ATA.  Can someone point me in the right direction?

--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001

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RE: [Asterisk-Users] Digium cards, so disappointing !

2006-04-14 Thread Ryan Amos

I am using a digium TE110P and a TDM04b (or whatever the one with 4 FXS
ports is called) on a Dell PowerEdge 2850. No problems at all with
faxing with a cheap fax machine, though the asterisk box almost never
goes above 5% CPU usage unless there are some conference calls going on.
I can run modems/faxes just fine (though the modem connections seem to
have a bit more latency than through a POTS line, it is acceptable for
our use.)

Just be sure to set echocancelwhenbridged=no and tweak your txgain and
rxgain on the line (this is not a do it once and you're done thing, I
had to go back probably 5 times over the course of 2 weeks to get the
right numbers.) I am even doing a redirect to eFax (I'd do with asterisk
but we already had an efax account and it works well enough) on one of
my DIDs and it works great.

Quite honestly I found a lot of documentation on how faxing in Asterisk
is hard, and I just never saw that. Maybe I got lucky with a magic
combination of hardware and forgiving fax machine, but it kind of just
worked the first time I tried it. Now, if only setting up a 'page all'
function on Cisco 79XX SIP phones without using a line appearance was so
easy...

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Friday, April 14, 2006 8:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Digium cards, so disappointing !

Anton Krall wrote:
 Problem is, how to make sure you system WILL have 100% on zttest
before
 buying the cards.. You need to have stability, compatibility and
certainty
 that what you buy is going to work :(
 
 Anybody had similar problems or success stories with sangoma cards? 

Running zttest on my box with both a TDM04b and A200D installed 
indicates and average of 99.96% for both. Not sure how accurate that 
might be as the A200D card appears as a 24 channel interface in terms of

/dev/zap even though only four ports are equipped.

The TDM04b won't support faxes on this box under any circumstances and 
I've played around with about every possible pci latency, etc, change 
that folks have suggested in the last two years.

Based on my heavily invested testing to date (which includes about two 
years of doing this), the only usable fax support thus far comes from 
using the A200D card with the fax machine directly connected to a fxs 
port on that card, and an fxo (pstn) port on the exact same card. Those 
fax tests have been 100% solid using a cheap/older Brother fax machine.



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[Asterisk-Users] Polycom 501 resource full problems ...

2006-04-14 Thread phil . dawson


Hi List,

Not sure if this is the place for this so here goes ...

We have a number of Polycom 501's connected to our * box and they work
great.  Some of our users have added a few entries into the directory on
the phone.  The problem is on those particular phones they now sometimes
get resource full on the phone when accessing the directory.  No central
directory was configured.  All phones are flashed with the latest
publically available sip and boot image.

Any help would be greatly appreciated.


Phil.

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Re: [Asterisk-Users] Polycom 501 resource full problems ...

2006-04-14 Thread stoffell

On 4/14/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
 We have a number of Polycom 501's connected to our * box and they work
 great.  Some of our users have added a few entries into the directory on
 the phone.  The problem is on those particular phones they now sometimes
 get resource full on the phone when accessing the directory.  No central
 directory was configured.  All phones are flashed with the latest
 publically available sip and boot image.

How weird, what is the exact SIP firmware you are using? 1.6.5 ? And
how often does the problem occur? I'm willing to test it out on a
bunch of phones also if you can share those details.

cheers
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Re: [Asterisk-Users] Polycom 501 resource full problems ...

2006-04-14 Thread Michael Welter

My customers are reporting that the contact directory can only hold 
about 45+ entries.



--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
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RE: [Asterisk-Users] the best billing tool for Asterisk

2006-04-14 Thread Guillermo Salas M.

On Fri, 2006-04-14 at 13:08 -0700, Mindaugas Kezys wrote:
 Hello,
 
 You can try: http://www.paskambink.lt/mcc
 

Or can try http://www.asterisk2billing.org/ it supports postgresql

 
 Regards/Pagarbiai,
 Mindaugas Kezys
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Joshua Colp
 Sent: Tuesday, April 11, 2006 9:55 AM
 To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
 Discussion
 Subject: Re: [Asterisk-Users] the best billing tool for Asterisk
 
 On 4/11/06 8:14 AM, Joao Pereira [EMAIL PROTECTED] wrote:
 
  Hello to all
  I would like to know some opinions of people that are using billing
  tools for Asterisk.
  Can you please advise me in wich billing tool to I use?
  
  Thanks
  Joao Pereira
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 Lots of people whip together their own solution as there is no billing
 solution out there for Asterisk that fits all. Usually you end up making
 tweaks here and there even if you do use a prebuilt solution.
 

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Re: [Asterisk-Users] tdm2400p and asterisk 1.2.1


Hi Kevin,
I'm sorry..I was wrong again...when I wrote Asterisk I meant Zaptel (I 
always use Asterisk x.y.y + Zaptel x.y.z + Libpri x.y.z, same version 
for all!)

By the way your answer satisfy me. I already use 1.2.x zaptel driver.  ::)

Thanks again !

Giorgio Incantalupo


Kevin P. Fleming wrote:

Giorgio Incantalupo wrote:
  

Hi Kevin,
I know upgrading is better, sorry, maybe my question was malformed...the
exact question is which is the minimum asterisk version supporting
TDM2400P?
(I have 10 pbx and  I want to change 3 TDM400P with one TDM2400P on
every pbx without reinstalling a new asterisk version on every machine!!)



That is not the correct question; Asterisk does not talk to the hardware
directly, it uses Zaptel.

I would not suggest converting to TDM2400Ps without also upgrading to
the latest 1.2.x Zaptel release; the driver for that card has undergone
some modifications since it was originally added.
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Re: [Asterisk-Users] Segfault on Inbound call?

2006-04-14 Thread Dinesh Nair




On 04/14/06 20:05 Matt said the following:

When it patched the
zaptel source... if I have usecallerid=yes on then it crashes... if I
turn usecallerid=no then it is fine.


we've tested the sangoma A101, A102 and A104 cards with usercallerid=yes, 
and it hasn't crashed. this is on FreeBSD though, and the sangoma driver 
installation did not patch the zaptel-bsd drivers at all. ymmv on linux.


--
Regards,   /\_/\   All dogs go to heaven.
[EMAIL PROTECTED](0 0)http://www.alphaque.com/
+==oOO--(_)--OOo==+
| for a in past present future; do|
|   for b in clients employers associates relatives neighbours pets; do   |
|   echo The opinions here in no way reflect the opinions of my $a $b.  |
| done; done  |
+=+
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Re: [Asterisk-Users] tdm2400p and asterisk 1.2.1

Giorgio Incantalupo wrote:

 I'm sorry..I was wrong again...when I wrote Asterisk I meant Zaptel (I
 always use Asterisk x.y.y + Zaptel x.y.z + Libpri x.y.z, same version
 for all!)

FYI... those version numbers are no longer kept in sync. The Zaptel and
libpri version numbers are incremented only when needed, not just
because Asterisk has a new release. So, for example, there is no Zaptel
1.2.7 release yet, even though there is an Asterisk 1.2.7.1 release.

In other words, if you run the latest release of each package, you will
be up to date, even though the version numbers no longer match.
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Re: [Asterisk-Users] Asterisk stops responding when internet is down

2006-04-14 Thread Jay Milk


Michael Strelnikov wrote:

I just never used one. Is BIND good enough?
dnsmasqd is quick and easy.  All the joys of DNS caching without the 
pain of configuring a full-blown bind.  Unless, of course, you do this 
sort of thing every day ;-)

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Re: [Asterisk-Users] Asterisk stops responding when internet is down


Jay Milk wrote:

Michael Strelnikov wrote:

I just never used one. Is BIND good enough?
dnsmasqd is quick and easy.  All the joys of DNS caching without the 
pain of configuring a full-blown bind.  Unless, of course, you do this 
sort of thing every day ;-)

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The software that this fellow is referring to is available at:

http://thekelleys.org.uk/dnsmasq/doc.html

It's widely used on Linux based routers too, so you use your router as 
the DNS server and it proxies to your ISP... and caches information. It 
works VERY well, but I've never personally tested it with helping with 
the DNS issue in Asterisk so if someone does then please post your 
results so others will know!


--
Joshua Colp
Software Developer
Digium
P - 256-428-6066
C - 506-878-0147
[EMAIL PROTECTED]
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Re: [Asterisk-Users] Digium cards, so disappointing !


Ryan Amos wrote:

I am using a digium TE110P and a TDM04b (or whatever the one with 4 FXS
ports is called) on a Dell PowerEdge 2850. No problems at all with
faxing with a cheap fax machine, though the asterisk box almost never
goes above 5% CPU usage unless there are some conference calls going on.
I can run modems/faxes just fine (though the modem connections seem to
have a bit more latency than through a POTS line, it is acceptable for
our use.)

Just be sure to set echocancelwhenbridged=no and tweak your txgain and
rxgain on the line (this is not a do it once and you're done thing, I
had to go back probably 5 times over the course of 2 weeks to get the
right numbers.) I am even doing a redirect to eFax (I'd do with asterisk
but we already had an efax account and it works well enough) on one of
my DIDs and it works great.

Quite honestly I found a lot of documentation on how faxing in Asterisk
is hard, and I just never saw that. Maybe I got lucky with a magic
combination of hardware and forgiving fax machine, but it kind of just
worked the first time I tried it. Now, if only setting up a 'page all'
function on Cisco 79XX SIP phones without using a line appearance was so
easy...

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Friday, April 14, 2006 8:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Digium cards, so disappointing !

Anton Krall wrote:

Problem is, how to make sure you system WILL have 100% on zttest

before

buying the cards.. You need to have stability, compatibility and

certainty

that what you buy is going to work :(

Anybody had similar problems or success stories with sangoma cards? 


Running zttest on my box with both a TDM04b and A200D installed 
indicates and average of 99.96% for both. Not sure how accurate that 
might be as the A200D card appears as a 24 channel interface in terms of


/dev/zap even though only four ports are equipped.

The TDM04b won't support faxes on this box under any circumstances and 
I've played around with about every possible pci latency, etc, change 
that folks have suggested in the last two years.


Based on my heavily invested testing to date (which includes about two 
years of doing this), the only usable fax support thus far comes from 
using the A200D card with the fax machine directly connected to a fxs 
port on that card, and an fxo (pstn) port on the exact same card. Those 
fax tests have been 100% solid using a cheap/older Brother fax machine.




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Some people have problems, some people don't. There is no way you can be 
prepared for every situation out there. We try our best.


--
Joshua Colp
Software Developer
Digium
P - 256-428-6066
C - 506-878-0147
[EMAIL PROTECTED]
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Re: [Asterisk-Users] Packet Testing


Kevin Smith wrote:

Hi everyone,

On the Polycom 601 phones we are using, the forward feature works very 
nicely for agents that are out on trips. I was wondering if there is a 
way to test to see if they have the forward option enabled.


When it is enabled the call comes in and gets -- Got SIP response 302 
Moved Temporarily response and then it uses the correct outbound macro 
to forward the call to the number specified. I am wondering if I am able 
to test for that SIP response or something in the SIP packet that I can 
grab to test. From what I read online, I didn't see much that would 
allow me to test for it, but  I may have just missed it.


Thanks,
Kevin
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You won't know that the phone is going to forward the call to an 
alternate number until you send the call to it. You could theoretically 
cache the information... but then it wouldn't be real time up to date 
with the phone's settings.


--
Joshua Colp
Software Developer
Digium
P - 256-428-6066
C - 506-878-0147
[EMAIL PROTECTED]
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[Asterisk-Users] asterisk 1.2.7.1 and app_rxfax

Hi!

After upgrading to asterisk 1.2.7.1 (from 1.2.4) it seems that app_rxfax
doesnt work any more.

I've installed spandsp-0.0.2pre25 (the same problem with
spandsp-0.0.3pre6.tgz ) app_rxfax.c, app_txfax.c and made the Makefile
patch from the same directory.

When starting asterisk I always get the follwing error message:

[app_rxfax.so]Apr 14 18:54:20 WARNING[7223]: loader.c:325
__load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined
symbol: t30_get_far_ident
Apr 14 18:54:20 WARNING[7223]: loader.c:554 load_modules: Loading module
app_rxfax.so failed!


Does anyone have any idea how to fix that?


cheers,
Tom
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Re: [Asterisk-Users] asterisk 1.2.7.1 and app_rxfax

2006-04-14 Thread Rob Terhaar

did you try to recompile the plugin?On 4/14/06, Thomas Artner [EMAIL PROTECTED] wrote:
Hi!After upgrading to asterisk 1.2.7.1 (from 1.2.4) it seems that app_rxfaxdoesnt work any more.I've installed spandsp-0.0.2pre25 (the same problem withspandsp-0.0.3pre6.tgz
 ) app_rxfax.c, app_txfax.c and made the Makefilepatch from the same directory.When starting asterisk I always get the follwing error message:[app_rxfax.so]Apr 14 18:54:20 WARNING[7223]: loader.c:325
__load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefinedsymbol: t30_get_far_identApr 14 18:54:20 WARNING[7223]: loader.c:554 load_modules: Loading moduleapp_rxfax.so failed!Does anyone have any idea how to fix that?
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Re: [Asterisk-Users] asterisk 1.2.7.1 and app_rxfax

Rob Terhaar wrote:
 did you try to recompile the plugin?
 


yes, of course...


 On 4/14/06, Thomas Artner [EMAIL PROTECTED] wrote:
 Hi!

 After upgrading to asterisk 1.2.7.1 (from 1.2.4) it seems that app_rxfax
 doesnt work any more.

 I've installed spandsp-0.0.2pre25 (the same problem with
 spandsp-0.0.3pre6.tgz ) app_rxfax.c, app_txfax.c and made the Makefile
 patch from the same directory.

 When starting asterisk I always get the follwing error message:

 [app_rxfax.so]Apr 14 18:54:20 WARNING[7223]: loader.c:325
 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined
 symbol: t30_get_far_ident
 Apr 14 18:54:20 WARNING[7223]: loader.c:554 load_modules: Loading module
 app_rxfax.so failed!


 Does anyone have any idea how to fix that?


 cheers,
 Tom
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[Asterisk-Users] Re: Asterisk-Users Digest, Vol 21, Issue 81

2006-04-14 Thread Martin

I too had a server room fry and need to replace h/w.
So what specific Dell servers did/do you deploy?
Where is the link w/Digium/s Dell caveats?
I'm using the Digium  TDM400 card w/*

Date: Thu, 13 Apr 2006 19:19:13 -0500 (CDT)
From: Aaron Daniel [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Digium cards, so disappointing !
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed

*shrugs* Ya win some ya lose some.  We've spent about 10 grand plus on
Digium cards and have been pretty satisfied with ours :) Faxes have been
working great for over 6 months and the cards work wonderfully in our Dell
servers.  They just need more documentation on the different configuration
options you can pass on load... I think the only problems we've really had
are configuration related, or bad hardware on our part, oh, and a server
room fry that took out more than just the Asterisk servers :-P

Aaron

On Fri, 14 Apr 2006, Tony ROBIN wrote:

I am so fed up with Digium cards. My company first owned a TE410P,
I installed it in a Dell server and enjoyed its instability (we
bought it months before Digium warned about the incompatibility
issues). Then we switched to a TE411P for the hardware echo
cancellation. Now we want to receive fax ( 20/day) on it and
guess what ? Since April 2006 (again a few months after we bought
our brand new card), officially, fax communications is not
supported with Digium cards ( http://www.voip-info.org/wiki-Asterisk+fax ).
Of course, I should have guessed that it is far too much to ask
to a $2495 card ! Is the fax extension in Asterisk just there
to push us to the competing products ?

We hesitated to buy another Digium card after the problems with
TE410P, but I told myself it was nice to support Asterisk by
buying some Digium cards. Now Digium make us regret our buys and
a disappointed customer is a lost customer forever... Too sad...

--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198

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Re: [Asterisk-Users] asterisk 1.2.7.1 and app_rxfax


after a few hours of debugging it works now...
I got some version mixes of spandsp on my system...

sorry for the spam


tom

Thomas Artner wrote:
 Hi!
 
 After upgrading to asterisk 1.2.7.1 (from 1.2.4) it seems that app_rxfax
 doesnt work any more.
 
 I've installed spandsp-0.0.2pre25 (the same problem with
 spandsp-0.0.3pre6.tgz ) app_rxfax.c, app_txfax.c and made the Makefile
 patch from the same directory.
 
 When starting asterisk I always get the follwing error message:
 
 [app_rxfax.so]Apr 14 18:54:20 WARNING[7223]: loader.c:325
 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined
 symbol: t30_get_far_ident
 Apr 14 18:54:20 WARNING[7223]: loader.c:554 load_modules: Loading module
 app_rxfax.so failed!
 
 
 Does anyone have any idea how to fix that?
 
 
 cheers,
 Tom
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Re: [Asterisk-Users] call center running Asterisk -soundquality-critical!

2006-04-14 Thread Matt Roth


Wai Wu wrote:


I did not install soxmix in my linux box. If you having issues with
mixmonitor, you can put both legs of the call into a conference and
record the conference 


Wai,

How are you mixing the leg files?  Do you run a process that moves them 
to a remote box with soxmix installed?


You have also mentioned that you are recording to a SCSI drive and I'm 
curious as to the details.  Is this a single drive or a RAID (and if so 
what RAID level)?  What filesystem does the partition the leg files are 
written to use?


Thank you,

Matthew Roth
InterMedia Marketing Systems
Software Engineer and Systems Developer
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Re: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 21, Issue 81

2006-04-14 Thread Aaron Daniel

We're using 1850's for our asterisk system right now (well, all but two, 
but they'll be upgraded soon).  1U boxes that work like a dream, I think 
the biggest problem I've had with them is mpg123 wouldn't compile since 
they're 64bit, and that was a simple fix.


This is what Digium has to say about the 1850's:
TE411P, TE410P, TE406P and TE405P, TE210P, TE205P

The Dell PowerEdge 1850 has an onboard Intel e1000 Ethernet controller 
that can interfere with the operation of the TE411P, TE410P, TE406P and 
TE405P, TE210P, TE205P cards. The recommended action for this server is to 
disable the onboard Ethernet controller and use a PCI-based solution.


Basically the problems are IRQ based if I'm reading right, but we haven't 
had a single problem with IRQ usage on the box, and we're using both of 
the ethernet links on the server.


Aaron

On Fri, 14 Apr 2006, Martin wrote:


I too had a server room fry and need to replace h/w.
So what specific Dell servers did/do you deploy?
Where is the link w/Digium/s Dell caveats?
I'm using the Digium  TDM400 card w/*


Date: Thu, 13 Apr 2006 19:19:13 -0500 (CDT)
From: Aaron Daniel [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Digium cards, so disappointing !
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed

*shrugs* Ya win some ya lose some.  We've spent about 10 grand plus on
Digium cards and have been pretty satisfied with ours :) Faxes have been
working great for over 6 months and the cards work wonderfully in our Dell
servers.  They just need more documentation on the different configuration
options you can pass on load... I think the only problems we've really had
are configuration related, or bad hardware on our part, oh, and a server
room fry that took out more than just the Asterisk servers :-P

Aaron

On Fri, 14 Apr 2006, Tony ROBIN wrote:



I am so fed up with Digium cards. My company first owned a TE410P,
I installed it in a Dell server and enjoyed its instability (we
bought it months before Digium warned about the incompatibility
issues). Then we switched to a TE411P for the hardware echo
cancellation. Now we want to receive fax ( 20/day) on it and
guess what ? Since April 2006 (again a few months after we bought
our brand new card), officially, fax communications is not
supported with Digium cards ( http://www.voip-info.org/wiki-Asterisk+fax 
).

Of course, I should have guessed that it is far too much to ask
to a $2495 card ! Is the fax extension in Asterisk just there
to push us to the competing products ?

We hesitated to buy another Digium card after the problems with
TE410P, but I told myself it was nice to support Asterisk by
buying some Digium cards. Now Digium make us regret our buys and
a disappointed customer is a lost customer forever... Too sad...

--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198



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--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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[Asterisk-Users] How to get 1.2.7 asterisk

2006-04-14 Thread Wai Wu

Hi,

Does cvs checkout asterisk gets the later version of asterisk? I tried
cvs checkout -r v1-2-7 asterisk, and didn't work for me. The only
thing works is cvs checkout -r v1-2 asterisk. What exactly is version
tag for version 1.2.7? Thnx
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RE: [Asterisk-Users] call center running Asterisk-soundquality-critical!

2006-04-14 Thread Wai Wu

The files were never mixed until they are actually listened to (They
have 3 people, 10 hours a day listening to calling recordings), and it
is done on a separate * box. As for the drive, all I know is an separate
external unit out of the main * box. (I didn't setup the linux server
myself).

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Roth
Sent: Friday, April 14, 2006 2:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] call center running
Asterisk-soundquality-critical!

Wai Wu wrote:

I did not install soxmix in my linux box. If you having issues with 
mixmonitor, you can put both legs of the call into a conference and 
record the conference

Wai,

How are you mixing the leg files?  Do you run a process that moves them
to a remote box with soxmix installed?

You have also mentioned that you are recording to a SCSI drive and I'm
curious as to the details.  Is this a single drive or a RAID (and if so
what RAID level)?  What filesystem does the partition the leg files are
written to use?

Thank you,

Matthew Roth
InterMedia Marketing Systems
Software Engineer and Systems Developer
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[Asterisk-Users] attended transfer issue

Hi!


A few months ago I needed some help for the following issue:

.) a call comes in
.) Person A takes the call and does an attended transfer to Person B
.) Person A hangs up the phone without waiting for Person B taking the call
.) the caller get lost at this point !!

At this point the attended transfer should go into a blind transfer. The
phone of Person B should still be ringing and the caller shouldnt get lost.

I think this is the most usual behaviour of a call transfer also on the
cheapest systems on the market.

Why doesnt this work well with asterisk? Will there be a solution for
that in the near future?

I am thankful for any kind of help!


thx,
Tom

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Re: [Asterisk-Users] How to get 1.2.7 asterisk

2006-04-14 Thread Matteo Brancaleoni


Hi,


Does cvs checkout asterisk gets the later version of asterisk? I tried
cvs checkout -r v1-2-7 asterisk, and didn't work for me. The only
thing works is cvs checkout -r v1-2 asterisk. What exactly is version
tag for version 1.2.7? Thnx
 


Why don't you download the package from the asterisk.org website?
Or checkout on the same website how to download the release
with subversion.

matteo
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Re: [Asterisk-Users] How to get 1.2.7 asterisk


Wai Wu wrote:

Hi,

Does cvs checkout asterisk gets the later version of asterisk? I tried
cvs checkout -r v1-2-7 asterisk, and didn't work for me. The only
thing works is cvs checkout -r v1-2 asterisk. What exactly is version
tag for version 1.2.7? Thnx
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We've moved to using subversion so it's highly recommended to use it 
instead of depending on CVS. Information about using SVN to check out 
things is available at http://www.asterisk.org/download


--
Joshua Colp
Software Developer
Digium
P - 256-428-6066
C - 506-878-0147
[EMAIL PROTECTED]
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RE: [Asterisk-Users] attended transfer issue

2006-04-14 Thread Michael Collins

 A few months ago I needed some help for the following issue:
 
 .) a call comes in
 .) Person A takes the call and does an attended transfer to Person B
 .) Person A hangs up the phone without waiting for Person B taking the
 call
 .) the caller get lost at this point !!
 
 At this point the attended transfer should go into a blind transfer.
The
 phone of Person B should still be ringing and the caller shouldnt get
 lost.
 
 I think this is the most usual behaviour of a call transfer also on
the
 cheapest systems on the market.


Could you remind us of what kinds of phones you are using, and whether
you're using SIP, Zap or something else?

Thanks!

-MC
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Re: [Asterisk-Users] attended transfer issue

2006-04-14 Thread John Novack




Michael Collins wrote:


A few months ago I needed some help for the following issue:

.) a call comes in
.) Person A takes the call and does an attended transfer to Person B
.) Person A hangs up the phone without waiting for Person B taking the call
.) the caller get lost at this point !!

At this point the attended transfer should go into a blind transfer.
   


The phone of Person B should still be ringing and the caller shouldnt get lost.

I think this is the most usual behaviour of a call transfer also on the 
cheapest systems on the market.
 




Could you remind us of what kinds of phones you are using, and whether you're 
using SIP, Zap or something else?

Thanks!

-MC

I think the point of this post and other related ones is the fact that 
there are attended and blind transfers, initiated by different actions, 
where phone systems for at least the last 20 years have one action, or 
transfer.
The person initiating the transfer starts the procedure, and if the 
destination extension answers, either through the facilities of 
handsfree intercom or picking up the phone, the initiator and the 
receiver can confer BEFORE the transfer is complete.
If, on the other hand the initiator either chooses to hang up after 
starting the transfer, the transfer is then complete, and the 
destination extension rings until answered or overflows into voice mail.
In NO case should the call get lost. Attended and blind transfer SHOULD 
start with the same action and be considered as ONE function

Irrelevant what phones are being used.

JMO

John Novack

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Re: [Asterisk-Users] attended transfer issue

2006-04-14 Thread Jerry Jones

Yes it should all behave the way we are used to. However SIP IS  
different. The exact behavior will be dependant upon the individual  
hard phone.


This of course is if using SIP which we do not know yet...

On Apr 14, 2006, at 1:43 PM, John Novack wrote:




Michael Collins wrote:


A few months ago I needed some help for the following issue:

.) a call comes in
.) Person A takes the call and does an attended transfer to Person B
.) Person A hangs up the phone without waiting for Person B  
taking the call

.) the caller get lost at this point !!

At this point the attended transfer should go into a blind transfer.

The phone of Person B should still be ringing and the caller  
shouldnt get lost.


I think this is the most usual behaviour of a call transfer also  
on the cheapest systems on the market.




Could you remind us of what kinds of phones you are using, and  
whether you're using SIP, Zap or something else?


Thanks!

-MC

I think the point of this post and other related ones is the fact  
that there are attended and blind transfers, initiated by different  
actions, where phone systems for at least the last 20 years have  
one action, or transfer.
The person initiating the transfer starts the procedure, and if the  
destination extension answers, either through the facilities of  
handsfree intercom or picking up the phone, the initiator and the  
receiver can confer BEFORE the transfer is complete.
If, on the other hand the initiator either chooses to hang up after  
starting the transfer, the transfer is then complete, and the  
destination extension rings until answered or overflows into voice  
mail.
In NO case should the call get lost. Attended and blind transfer  
SHOULD start with the same action and be considered as ONE function

Irrelevant what phones are being used.

JMO

John Novack

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Re[2]: [Asterisk-Users] asterisk 1.2.7.1 and app_rxfax

2006-04-14 Thread Melcon Moraes

So, what version of spandsp are using afterall?

[]'s
MM

 -Original Message-
From:   Thomas Artner [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Cc: 
Sent:  Fri, 14 Apr 2006 19:50:47 +0200
Delivered:  Fri,  14 Apr 2006 11:52:45 
Subject:[Asterisk-Users] asterisk 1.2.7.1 and app_rxfax

after a few hours of debugging it works now...
I got some version mixes of spandsp on my system...

sorry for the spam

tom

Thomas Artner wrote:
 Hi!

 After upgrading to asterisk 1.2.7.1 (from 1.2.4) it seems that app_rxfax
 doesnt work any more.

 I've installed spandsp-0.0.2pre25 (the same problem with
 spandsp-0.0.3pre6.tgz ) app_rxfax.c, app_txfax.c and made the Makefile
 patch from the same directory.

 When starting asterisk I always get the follwing error message:

 [app_rxfax.so]Apr 14 18:54:20 WARNING[7223]: loader.c:325
 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined
 symbol: t30_get_far_ident
 Apr 14 18:54:20 WARNING[7223]: loader.c:554 load_modules: Loading module
 app_rxfax.so failed!

 Does anyone have any idea how to fix that?

 cheers,
 Tom
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E-mail classificado pelo Identificador de Spam Inteligente Terra.
Para alterar a categoria classificada, visite
http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz_l=1,1145037166.47636.6654.arrino.terra.com.br,4381,Des15,Des15

 --Original Message Ends--

-- 
Melcon Moraes [EMAIL PROTECTED]

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Re: [Asterisk-Users] attended transfer issue

2006-04-14 Thread John Novack




Jerry Jones wrote:

Yes it should all behave the way we are used to. However SIP IS  
different. The exact behavior will be dependant upon the individual  
hard phone.



Isn't that true only if it has a preprogrammed transfer key?
an Asterisk feature code should work as discussed.
There SHOULD be a way to make SIP phones work the same.
( easy to say, perhaps not so easy to do )

John Novack


This of course is if using SIP which we do not know yet...

On Apr 14, 2006, at 1:43 PM, John Novack wrote:




Michael Collins wrote:


A few months ago I needed some help for the following issue:

.) a call comes in
.) Person A takes the call and does an attended transfer to Person B
.) Person A hangs up the phone without waiting for Person B  taking 
the call

.) the caller get lost at this point !!

At this point the attended transfer should go into a blind transfer.

The phone of Person B should still be ringing and the caller  
shouldnt get lost.


I think this is the most usual behaviour of a call transfer also  on 
the cheapest systems on the market.




Could you remind us of what kinds of phones you are using, and  
whether you're using SIP, Zap or something else?


Thanks!

-MC

I think the point of this post and other related ones is the fact  
that there are attended and blind transfers, initiated by different  
actions, where phone systems for at least the last 20 years have  one 
action, or transfer.
The person initiating the transfer starts the procedure, and if the  
destination extension answers, either through the facilities of  
handsfree intercom or picking up the phone, the initiator and the  
receiver can confer BEFORE the transfer is complete.
If, on the other hand the initiator either chooses to hang up after  
starting the transfer, the transfer is then complete, and the  
destination extension rings until answered or overflows into voice  
mail.
In NO case should the call get lost. Attended and blind transfer  
SHOULD start with the same action and be considered as ONE function

Irrelevant what phones are being used.

JMO

John Novack

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RE: [Asterisk-Users] attended transfer issue

2006-04-14 Thread Michael Collins

 Jerry Jones wrote:
 
  Yes it should all behave the way we are used to. However SIP IS
  different. The exact behavior will be dependant upon the individual
  hard phone.
 
 Isn't that true only if it has a preprogrammed transfer key?
 an Asterisk feature code should work as discussed.
 There SHOULD be a way to make SIP phones work the same.
 ( easy to say, perhaps not so easy to do )

Agreed!  In regular PBX telephony there is no technological difference
between a blind transfer and an attended/supervised transfer.  As far as
the generic PBX or key system is concerned, a transfer is a transfer.
The only thing the differentiates the blind from the attended is the
BEHAVIOR of the person initiating the transfer.

HOWEVER, with the various types of SIP phones, firmware revs, etc.,
there are some issues.  When the OP gives us the details on his exact
config hopefully we'll be able to point him in the right direction.

-MC
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Re: [Asterisk-Users] Digium cards, so disappointing !

On Fri, 2006-04-14 at 08:19 -0500, Rich Adamson wrote:
 I believe the TDM2400 has the capability of doing on-card fxo-fxs data 
 flows (without hitting the pci bus), but that function has not yet been 
 implemented. Its basically required to support faxes in an analog 
 environment. When it is implemented, that card should work. The TDM400 
 card will not work in 99% of the deployments.
 
 Faxing via T1 cards is known to work in a fairly large number of 
 deployments, but its likely to be highly dependent on exactly where the 
 fax machine is located relative to *. Eg, incoming pstn fax via a T1 
 that is expected to be switched to a sip ata adapter has lots of 
 technical and specific infrastructure dependencies that have to be 
 addressed by the implementor / engineer. The plug-n-play approach will 
 have a very high failure rate.

What about:

T1 card - * - different T1 card - channel bank - fax

or

T1 card - * - FXS card - fax

Is the rule as long as the fax doesn't go over an IP network, then
faxing should work?

...Jeff
 
 
 Anton Krall wrote:
  I must agree with you. I too buy Digium cards because I want to support the
  development of asterisk. Asterisk is a great product but digum cards are a
  pain, they say they don't support faxing but a lot of people that are
  implementing asterisk demand or need faxin as a day to day service on
  their PBX's.
  
  Sad to see that faxing is nearly impossible on digium cards. To me is like
  saying here you have a great car but.. It cannot handle a car stereo :(
   
  
  |-Original Message-
  |From: [EMAIL PROTECTED] 
  |[mailto:[EMAIL PROTECTED] On Behalf Of 
  |Tony ROBIN
  |Sent: Thursday, April 13, 2006 7:01 PM
  |To: asterisk-users@lists.digium.com
  |Subject: [Asterisk-Users] Digium cards, so disappointing !
  |
  |
  |I am so fed up with Digium cards. My company first owned a 
  |TE410P, I installed it in a Dell server and enjoyed its 
  |instability (we bought it months before Digium warned about 
  |the incompatibility issues). Then we switched to a TE411P for 
  |the hardware echo cancellation. Now we want to receive fax ( 
  |20/day) on it and guess what ? Since April 2006 (again a few 
  |months after we bought our brand new card), officially, fax 
  |communications is not supported with Digium cards ( 
  |http://www.voip-info.org/wiki-Asterisk+fax ).
  |Of course, I should have guessed that it is far too much to 
  |ask to a $2495 card ! Is the fax extension in Asterisk just 
  |there to push us to the competing products ?
  |
  |We hesitated to buy another Digium card after the problems 
  |with TE410P, but I told myself it was nice to support Asterisk 
  |by buying some Digium cards. Now Digium make us regret our 
  |buys and a disappointed customer is a lost customer forever... 
  |Too sad...
 
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Re: [Asterisk-Users] attended transfer issue

2006-04-14 Thread Jerry Jones

Keep in mind that with a SIP phone you are not communicating directly  
with asterisk but with the phone which acts on your behalf with  
asterisk. On traditional systems if you performed a hook flash to  
transfer, you were definately signalling directly to the PBX. Now  
when you push a button, hard or soft, on a SIP phone you are telling  
the phone to perform as series of actions to accomplish a goal. It is  
very much up to the phone software on exactly how the set behaves.
As stated previously, yes there should be a standard, but afaik there  
are no standards bodies specifying the ui for voip devices.



On Apr 14, 2006, at 2:16 PM, John Novack wrote:




Jerry Jones wrote:

Yes it should all behave the way we are used to. However SIP IS   
different. The exact behavior will be dependant upon the  
individual  hard phone.



Isn't that true only if it has a preprogrammed transfer key?
an Asterisk feature code should work as discussed.
There SHOULD be a way to make SIP phones work the same.
( easy to say, perhaps not so easy to do )

John Novack


This of course is if using SIP which we do not know yet...

On Apr 14, 2006, at 1:43 PM, John Novack wrote:




Michael Collins wrote:


A few months ago I needed some help for the following issue:

.) a call comes in
.) Person A takes the call and does an attended transfer to  
Person B
.) Person A hangs up the phone without waiting for Person B   
taking the call

.) the caller get lost at this point !!

At this point the attended transfer should go into a blind  
transfer.


The phone of Person B should still be ringing and the caller   
shouldnt get lost.


I think this is the most usual behaviour of a call transfer  
also  on the cheapest systems on the market.




Could you remind us of what kinds of phones you are using, and   
whether you're using SIP, Zap or something else?


Thanks!

-MC

I think the point of this post and other related ones is the  
fact  that there are attended and blind transfers, initiated by  
different  actions, where phone systems for at least the last 20  
years have  one action, or transfer.
The person initiating the transfer starts the procedure, and if  
the  destination extension answers, either through the facilities  
of  handsfree intercom or picking up the phone, the initiator and  
the  receiver can confer BEFORE the transfer is complete.
If, on the other hand the initiator either chooses to hang up  
after  starting the transfer, the transfer is then complete, and  
the  destination extension rings until answered or overflows into  
voice  mail.
In NO case should the call get lost. Attended and blind transfer   
SHOULD start with the same action and be considered as ONE function

Irrelevant what phones are being used.

JMO

John Novack

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Re: [Asterisk-Users] attended transfer issue

2006-04-14 Thread Rob Terhaar

on my system, if i do a blind-xfer, it rings the destination's phone and finally flips to voicemail. Sometimes, If the destination/recipient is an exec or otherwise important, our attendant does a normal xfer to see if they're at their desk, if the destination doesn't respond, then the attendant asks the caller if they would like to go to the destination's voicemail.
On 4/14/06, John Novack [EMAIL PROTECTED] wrote:
Jerry Jones wrote: Yes it should all behave the way we are used to. However SIP IS different. The exact behavior will be dependant upon the individual hard phone.Isn't that true only if it has a preprogrammed transfer key?
an Asterisk feature code should work as discussed.There SHOULD be a way to make SIP phones work the same.( easy to say, perhaps not so easy to do )John Novack This of course is if using SIP which we do not know yet...
 On Apr 14, 2006, at 1:43 PM, John Novack wrote: Michael Collins wrote: A few months ago I needed some help for the following issue:
 .) a call comes in .) Person A takes the call and does an attended transfer to Person B .) Person A hangs up the phone without waiting for Person Btaking
 the call .) the caller get lost at this point !! At this point the attended transfer should go into a blind transfer.
 The phone of Person B should still be ringing and the caller shouldnt get lost. I think this is the most usual behaviour of a call transfer alsoon the cheapest systems on the market.
 Could you remind us of what kinds of phones you are using, and whether you're using SIP, Zap or something else? Thanks!
 -MC I think the point of this post and other related ones is the fact that there are attended and blind transfers, initiated by different actions, where phone systems for at least the last 20 years haveone
 action, or transfer. The person initiating the transfer starts the procedure, and if the destination extension answers, either through the facilities of handsfree intercom or picking up the phone, the initiator and the
 receiver can confer BEFORE the transfer is complete. If, on the other hand the initiator either chooses to hang up after starting the transfer, the transfer is then complete, and the
 destination extension rings until answered or overflows into voice mail. In NO case should the call get lost. Attended and blind transfer SHOULD start with the same action and be considered as ONE function
 Irrelevant what phones are being used. JMO John Novack ___ --Bandwidth and Colocation provided by 
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Re: [Asterisk-Users] asterisk with bluetooth headset howto

2006-04-14 Thread Rob Terhaar

previously, i've found directions on how to get a bluetooth headset to act like a sound card in windows. Can't remember the URL though...On 4/14/06, Andreas Nitsche
 [EMAIL PROTECTED] wrote:
My bluetooth headset works with the snd-bt-sco module. I also compiledchan_bluetooth into my asterisk but I can't find any howto for using abt headset with asterisk.Does this just work with a smartphone as gateway? Is it not possible
to connect the bt headset directly to my notebook?AndiOn Fri, Apr 14, 2006 at 09:47:09AM -0400, Alexander Lopez wrote: I think that the BT channel in Asteisk is to support a phone connection
 via Bluetooth, (ie asteisk mimics your Headset, and a bit more). If you have a Bluetooth chipset in your laptop or add one via USB. Any softphone that can you the audio to/form your headset should work, I do
 not know if ALSA or OSS will, so using the console may not be an option if you are running * on your Laptop.  -Original Message-  From: 
[EMAIL PROTECTED]  [mailto:[EMAIL PROTECTED]] On Behalf Of  Andreas Nitsche  Sent: Friday, April 14, 2006 9:39 AM
  To: asterisk-users@lists.digium.com  Subject: [Asterisk-Users] asterisk with bluetooth headset howto   Hello! 
  What do I need to use the asterisk on my notebook with a  bluetooth headset?   Is there anywhere a good howto?   Thanks!   Andi
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[Asterisk-Users] Re: Cisco 7960 6.3 unlock/reset?

2006-04-14 Thread R.Trommer

Shaun schrieb:
 Anybody know the proceedure to factory reset the a 7960 phone running 6.3 
 SIP software?  I've tried holding # when booting the phone and nothing, i 
 can do that on my 8.2 phone but this phone i just got with 6.3 isnt working. 
 Also **# doesnt work either..
 

Hi Schaun,

i have found this information.

http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a00800941bb.shtml

I hope it helps

Ronny

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RE: [Asterisk-Users] attended transfer issue

2006-04-14 Thread Damon Estep

There is some kind of issue with SIP transfer interaction between some
SIP phones and asterisk, I have personal experience with Polycom phones
not being able to do a blind xfer using the feature key.

We have to use the asterisk # blind xfrer functionality for blind
transfers

The phones will drop the call if you initiate a transfer with the
feature key but do not wait for the remote line to answer before
releasing the call. In other words, if you hit transfer on the phone,
wait for the remote phone to ring, and hang up, you will drop the call.

If you wait for the remote phone to answer (live or voicemail) the
transfer will complete.

It IS confusing to users to have 2 transfers, # for blind and the
feature key for attended.

Damon

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Jerry Jones
 Sent: Friday, April 14, 2006 1:28 PM
 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-
 Commercial Discussion
 Subject: Re: [Asterisk-Users] attended transfer issue
 
 Keep in mind that with a SIP phone you are not communicating directly
 with asterisk but with the phone which acts on your behalf with
 asterisk. On traditional systems if you performed a hook flash to
 transfer, you were definately signalling directly to the PBX. Now
 when you push a button, hard or soft, on a SIP phone you are telling
 the phone to perform as series of actions to accomplish a goal. It is
 very much up to the phone software on exactly how the set behaves.
 As stated previously, yes there should be a standard, but afaik there
 are no standards bodies specifying the ui for voip devices.
 
 
 On Apr 14, 2006, at 2:16 PM, John Novack wrote:
 
 
 
  Jerry Jones wrote:
 
  Yes it should all behave the way we are used to. However SIP IS
  different. The exact behavior will be dependant upon the
  individual  hard phone.
 
  Isn't that true only if it has a preprogrammed transfer key?
  an Asterisk feature code should work as discussed.
  There SHOULD be a way to make SIP phones work the same.
  ( easy to say, perhaps not so easy to do )
 
  John Novack
 
  This of course is if using SIP which we do not know yet...
 
  On Apr 14, 2006, at 1:43 PM, John Novack wrote:
 
 
 
  Michael Collins wrote:
 
  A few months ago I needed some help for the following issue:
 
  .) a call comes in
  .) Person A takes the call and does an attended transfer to
  Person B
  .) Person A hangs up the phone without waiting for Person B
  taking the call
  .) the caller get lost at this point !!
 
  At this point the attended transfer should go into a blind
  transfer.
 
  The phone of Person B should still be ringing and the caller
  shouldnt get lost.
 
  I think this is the most usual behaviour of a call transfer
  also  on the cheapest systems on the market.
 
 
 
  Could you remind us of what kinds of phones you are using, and
  whether you're using SIP, Zap or something else?
 
  Thanks!
 
  -MC
 
  I think the point of this post and other related ones is the
  fact  that there are attended and blind transfers, initiated by
  different  actions, where phone systems for at least the last 20
  years have  one action, or transfer.
  The person initiating the transfer starts the procedure, and if
  the  destination extension answers, either through the facilities
  of  handsfree intercom or picking up the phone, the initiator and
  the  receiver can confer BEFORE the transfer is complete.
  If, on the other hand the initiator either chooses to hang up
  after  starting the transfer, the transfer is then complete, and
  the  destination extension rings until answered or overflows into
  voice  mail.
  In NO case should the call get lost. Attended and blind transfer
  SHOULD start with the same action and be considered as ONE
function
  Irrelevant what phones are being used.
 
  JMO
 
  John Novack
 
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[Asterisk-Users] asterisk or ser

2006-04-14 Thread Xaji Gaid

Hello:I noticed in few references that asterisk and ser and complementary. Meaning asterisk handles connections to PSTN and voicemail but SER is better for routing SIP traffic.Is anyone using just asterisk for production purpose. Meaning serving a high number of callers.
Is it mandatory to use SER behind asterisk? your feedback would appreciated.-Gaid
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Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)

On Fri, 2006-04-14 at 13:49 -0300, Joshua Colp wrote:

 Some people have problems, some people don't. There is no way you can be 
 prepared for every situation out there. We try our best.
 

I was looking at using a Dell server for running Asterisk and noticed
that Dell has started using PCI-X on a lot of their new systems.  Does
this newer bus standard help the situation with faxing?  
Will Digium offer cards that support the new bus? 
What about a new line of Digium cards that have bridge cables that run
between the various cards and bypass the PCI bus?  Since one of the best
aspects of using Asterisk is standards.  This bridge cable should be
standardized and published so that other companies can adopt the
standard.  For example an ISDN card could bridge to a Digium T1 card.
Or a card that supported legacy digital phones could bridge to other
cards.
...Jeff

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Re: Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)

Jeff Gustafson wrote:

   I was looking at using a Dell server for running Asterisk and noticed
 that Dell has started using PCI-X on a lot of their new systems.  Does
 this newer bus standard help the situation with faxing?  

No. PCI-X is just a wider/higher-speed version of PCI, not a new bus.

   Will Digium offer cards that support the new bus? 

All of our cards work in PCI-X slots, but none of them take advantage of
64-bit slots or speeds higher than 33MHz.

   What about a new line of Digium cards that have bridge cables that run
 between the various cards and bypass the PCI bus?  Since one of the best
 aspects of using Asterisk is standards.  This bridge cable should be
 standardized and published so that other companies can adopt the
 standard.  For example an ISDN card could bridge to a Digium T1 card.
 Or a card that supported legacy digital phones could bridge to other
 cards.

That is called H.100, and it has existed for many years. It's also
ludicrously expensive to implement, so you won't see it on Digium cards
any time soon :-)
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Re: Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)

2006-04-14 Thread Aaron Daniel

Well, the TE410P and TE411P work in the PCI-X slots since it's backwards 
compatible.  So I guess in effect, the Digium's cards already do support 
it :)


Aaron

On Fri, 14 Apr 2006, Jeff Gustafson wrote:


On Fri, 2006-04-14 at 13:49 -0300, Joshua Colp wrote:


Some people have problems, some people don't. There is no way you can be
prepared for every situation out there. We try our best.



I was looking at using a Dell server for running Asterisk and noticed
that Dell has started using PCI-X on a lot of their new systems.  Does
this newer bus standard help the situation with faxing?
Will Digium offer cards that support the new bus?
What about a new line of Digium cards that have bridge cables that run
between the various cards and bypass the PCI bus?  Since one of the best
aspects of using Asterisk is standards.  This bridge cable should be
standardized and published so that other companies can adopt the
standard.  For example an ISDN card could bridge to a Digium T1 card.
Or a card that supported legacy digital phones could bridge to other
cards.
...Jeff

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--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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Re: Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)

On Fri, 2006-04-14 at 15:10 -0500, Aaron Daniel wrote:
 Well, the TE410P and TE411P work in the PCI-X slots since it's backwards 
 compatible.  So I guess in effect, the Digium's cards already do support 
 it :)
 

My fault.  I meant to say PCI-e, which is a newer bus that Dell is
shipping on their server class machines.

...Jeff

 Aaron
 
 On Fri, 14 Apr 2006, Jeff Gustafson wrote:
 
  On Fri, 2006-04-14 at 13:49 -0300, Joshua Colp wrote:
 
  Some people have problems, some people don't. There is no way you can be
  prepared for every situation out there. We try our best.
 
 
  I was looking at using a Dell server for running Asterisk and noticed
  that Dell has started using PCI-X on a lot of their new systems.  Does
  this newer bus standard help the situation with faxing?
  Will Digium offer cards that support the new bus?
  What about a new line of Digium cards that have bridge cables that run
  between the various cards and bypass the PCI bus?  Since one of the best
  aspects of using Asterisk is standards.  This bridge cable should be
  standardized and published so that other companies can adopt the
  standard.  For example an ISDN card could bridge to a Digium T1 card.
  Or a card that supported legacy digital phones could bridge to other
  cards.
  ...Jeff
 
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Re: [Asterisk-Users] attended transfer issue

2006-04-14 Thread John Novack




Damon Estep wrote:


There is some kind of issue with SIP transfer interaction between some SIP 
phones and asterisk, I have personal experience with Polycom phones not being 
able to do a blind xfer using the feature key.

 


Is that a Polycom or Asterisk defect?


We have to use the asterisk # blind xfrer functionality for blind
transfers
 




The phones will drop the call if you initiate a transfer with the
feature key but do not wait for the remote line to answer before
releasing the call. In other words, if you hit transfer on the phone,
wait for the remote phone to ring, and hang up, you will drop the call.

 


Not so good.
Asterisk or Polycom doing that?

John Novack


If you wait for the remote phone to answer (live or voicemail) the
transfer will complete.

It IS confusing to users to have 2 transfers, # for blind and the
feature key for attended.

Damon

 

 


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Re: [Asterisk-Users] attended transfer issue

Michael Collins wrote:
 A few months ago I needed some help for the following issue:

 .) a call comes in
 .) Person A takes the call and does an attended transfer to Person B
 .) Person A hangs up the phone without waiting for Person B taking the
 call
 .) the caller get lost at this point !!

 At this point the attended transfer should go into a blind transfer.
 The
 phone of Person B should still be ringing and the caller shouldnt get
 lost.

 I think this is the most usual behaviour of a call transfer also on
 the
 cheapest systems on the market.
 
 
 Could you remind us of what kinds of phones you are using, and whether
 you're using SIP, Zap or something else?


i am using analog phones on digium cards (zaptel).



 
 Thanks!
 
 -MC
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Re: [Asterisk-Users] asterisk 1.2.7.1 and app_rxfax

Melcon Moraes wrote:
 So, what version of spandsp are using afterall?

i am using spandsp-0.0.2pre25 now.
In the 0.0.3 package, there is no app_rxfax.c and no app_txfax.c. No
idea why thats missing there.


tom


 
 []'s
 MM
 
  -Original Message-
 From:   Thomas Artner [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Cc: 
 Sent:  Fri, 14 Apr 2006 19:50:47 +0200
 Delivered:  Fri,  14 Apr 2006 11:52:45 
 Subject:[Asterisk-Users] asterisk 1.2.7.1 and app_rxfax
 
 
 after a few hours of debugging it works now...
 I got some version mixes of spandsp on my system...
 
 sorry for the spam
 
 
 tom
 
 Thomas Artner wrote:
 Hi!

 After upgrading to asterisk 1.2.7.1 (from 1.2.4) it seems that app_rxfax
 doesnt work any more.

 I've installed spandsp-0.0.2pre25 (the same problem with
 spandsp-0.0.3pre6.tgz ) app_rxfax.c, app_txfax.c and made the Makefile
 patch from the same directory.

 When starting asterisk I always get the follwing error message:

 [app_rxfax.so]Apr 14 18:54:20 WARNING[7223]: loader.c:325
 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined
 symbol: t30_get_far_ident
 Apr 14 18:54:20 WARNING[7223]: loader.c:554 load_modules: Loading module
 app_rxfax.so failed!


 Does anyone have any idea how to fix that?


 cheers,
 Tom
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 E-mail classificado pelo Identificador de Spam Inteligente Terra.
 Para alterar a categoria classificada, visite
 http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz_l=1,1145037166.47636.6654.arrino.terra.com.br,4381,Des15,Des15
 
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Re: Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)

On Fri, 2006-04-14 at 15:10 -0500, Kevin P. Fleming wrote:
 Jeff Gustafson wrote:
 
  I was looking at using a Dell server for running Asterisk and noticed
  that Dell has started using PCI-X on a lot of their new systems.  Does
  this newer bus standard help the situation with faxing?  
 
 No. PCI-X is just a wider/higher-speed version of PCI, not a new bus.
 
Sorry, I meant PCI-e.

[...]
  What about a new line of Digium cards that have bridge cables that run
  between the various cards and bypass the PCI bus?  Since one of the best
  aspects of using Asterisk is standards.  This bridge cable should be
  standardized and published so that other companies can adopt the
  standard.  For example an ISDN card could bridge to a Digium T1 card.
  Or a card that supported legacy digital phones could bridge to other
  cards.
 
 That is called H.100, and it has existed for many years. It's also
 ludicrously expensive to implement, so you won't see it on Digium cards
 any time soon :-)

Is there any reason an easier implementation of the same, basic, idea
could be created for the Asterisk generation?  According to a quick
search of H.100 it's just a TDM bus.  It handles 2,048 full duplex
calls.  Would a lightweight version that only supports 512 or 256 calls
be any cheaper?

...Jeff

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RE: Faxing and PCI (was Re: [Asterisk-Users] Digium cards, sodisappointing !)

2006-04-14 Thread Jesus M. Lares

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Jeff
 Gustafson
 Sent: Friday, April 14, 2006 1:00 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Faxing and PCI (was Re: [Asterisk-Users] Digium cards,
 sodisappointing !)

 On Fri, 2006-04-14 at 13:49 -0300, Joshua Colp wrote:

  Some people have problems, some people don't. There is no way
 you can be
  prepared for every situation out there. We try our best.

   I was looking at using a Dell server for running Asterisk
 and noticed
 that Dell has started using PCI-X on a lot of their new systems.  Does
 this newer bus standard help the situation with faxing?
   Will Digium offer cards that support the new bus?
   What about a new line of Digium cards that have bridge
 cables that run
 between the various cards and bypass the PCI bus?  Since one of the best
 aspects of using Asterisk is standards.  This bridge cable should be
 standardized and published so that other companies can adopt the
 standard.  For example an ISDN card could bridge to a Digium T1 card.
 Or a card that supported legacy digital phones could bridge to other
 cards.

This standard already exists, it's called H.100 and it uses a ribbon cable
between cards.

   ...Jeff

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Re: Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)

Jeff Gustafson wrote:

   My fault.  I meant to say PCI-e, which is a newer bus that Dell is
 shipping on their server class machines.

Right. That is not supported by any Digium products yet, but it still
won't help the FAXing issue, since the issue is _not_ PCI bus bandwidth.
In fact, the FAXing issue is really more a problem with specific card
designs and other system issues than it is with the bus at all.
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RE: [Asterisk-Users] attended transfer issue

2006-04-14 Thread Damon Estep

Not sure, but the fact that the # xfer in asterisk releases the call
without the ability to do an attended transfer is an asterisk issue,
maybe not a defect, but a design issue inconsistent with typical PBX
behavior.

To be typical it would act like this;

Press pound to get secondary dial tone
Dial the number for the transfer
Either hang up or stay on the line after progress (ring)
If you stay on the line the transfer completes when you hang up
If you hang up during the ring the call is blind transferred
If you press the same feature access key (#) again you get the call back
and terminate the transfer.

Make sense?

Is this feature already there? If it is it would be easy to code the
feature key on the phone to use the sip feature server function, if it
is not there then it is hard to have a phone ask asterisk to do
something it does not know how to do.

I agree that this feature could be built into the phone firmware, but it
would be different for every phone. Putting it in asterisk provides a
bit of consistency on a feature every business uses every day.

Damon

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of John Novack
 Sent: Friday, April 14, 2006 2:22 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] attended transfer issue
 
 
 
 Damon Estep wrote:
 
 There is some kind of issue with SIP transfer interaction between
some
 SIP phones and asterisk, I have personal experience with Polycom
phones
 not being able to do a blind xfer using the feature key.
 
 
 
 Is that a Polycom or Asterisk defect?
 
 We have to use the asterisk # blind xfrer functionality for blind
 transfers
 
 
 
 The phones will drop the call if you initiate a transfer with the
 feature key but do not wait for the remote line to answer before
 releasing the call. In other words, if you hit transfer on the phone,
 wait for the remote phone to ring, and hang up, you will drop the
call.
 
 
 
 Not so good.
 Asterisk or Polycom doing that?
 
 John Novack
 
 If you wait for the remote phone to answer (live or voicemail) the
 transfer will complete.
 
 It IS confusing to users to have 2 transfers, # for blind and the
 feature key for attended.
 
 Damon
 
 
 
 
 
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Re: Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)

Jeff Gustafson wrote:

   Is there any reason an easier implementation of the same, basic, idea
 could be created for the Asterisk generation?  According to a quick
 search of H.100 it's just a TDM bus.  It handles 2,048 full duplex
 calls.  Would a lightweight version that only supports 512 or 256 calls
 be any cheaper?

It's doubtful. The issues are the cables and connectors are not cheap,
and getting the boards to pass EMI and other certifications would be
more complex. In addition, it means every board now has to have support
for a super-speed TDM bus, even if it's only a 4-port analog interface
card, and it also needs onboard logic to be able to map channels around.
That would increase the card cost quite a bit, even for people that have
no desire to use this method of connection.
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[Asterisk-Users] Bluetooth (chan_btp): dialing external phone number through BTP/Zap when bluetooth device not present?

2006-04-14 Thread Mike Garey

I sent the following message a few days ago, but never received a
reply, so I thought I'd ask again..

Can anyone tell me how me to get asterisk to dial out a phone number using BTP
when a bluetooth device is not detected?  I can get BTP to dial to a
SIP phone, but I can't get it to dial through a POTS phone line using
the Zap interface..

I've tried putting the following under the clients section in
/etc/asterisk/btp.conf:

client =user,00:12:34:56:78:90,Zap/4/1234567890

and in extensions.conf:

exten = 222,1,Playback(pls-hold-while-try)
exten = 222,2,Dial(BTP/user,60,m)
exten = 222,3,Hangup

but asterisk doesn't dial the phone number 1234567890, it simply does:

Zap/4-1 answered SIP/304-fc8a

and then gives me a dial tone..  From btp.conf, it says:

;If a default channel is specified, we
; use that channel if nobody has found the bluetooth device.

so it seems as though I can connect to a channel (in this case Zap/4),
but I can't actually get the channel to dial the given phone number..
If anyone can tell me what I'm doing wrong, I would very much
appreciate it.  Thanks,

Mike
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Re: Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)