RE: [asterisk-users] Fax with T.38

[Jbebeau]

Hi Bill,

I'm in exactly the same boat with T.38 and OpenPBX.  I too think the Cisco-T.38 
Gateway is the most practical at this moment.  Where are you on testing this 
and can you share the 3660 config?  In researching the CIsco/voice, there is a 
TON of hardware options you need, or so it seems.

Jon

-Original message-
From: "Bill Gibbs" [EMAIL PROTECTED]
Date: Thu, 22 Feb 2007 15:02:18 -0500
To: "Asterisk Users Mailing List - Non-Commercial 
Discussion"asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] Fax with T.38

> Ray,
> 
> I have been playing with OpenPBX.  My core servers are Asterisk so I was 
> playing around with their T38Gateway application.  Long story short - I can 
> get the ATA (behind NAT) to talk T38 to the rxfax app on an OpenPBX server 
> but the gateway feature of that product is still under development so I was 
> sending IAX calls to it and it would try to talk T38 to my ATA (behind NAT or 
> public IP) and eventually the call would fail.  Clearly T38 was working 
> though, debug output was full of T38 talk.  However the wiki clearly states 
> it's experimental still.
> 
> I personally have decided to go with a 2nd PRI port to a 3660 I have on hand 
> that will do T38 SIP.  I am going to set that up to talk to * 1.4.0 and do 
> T38 pass through.  I to will be doing NAT for the ATAs so...hopefully it will 
> work.  We shall see.
> 
> So my call flow will be
> 
> PRI -> Asterisk 1.2.x
> Out the 2nd PRI to the 3660
> 3660 dial-peer, with T38 fax settings talk SIP to Asterisk 1.4.x then t38 
> pass through to my ATA.
> 
> I have the 3660 there to take the call via TDM and convert to T38.  I only 
> have a single PRI which is why I don't want to have to purchase other lines 
> dedicated to a T38 faxserver, and this will give me the ability to use my 
> DIDs already assigned.
> 
> That's how I plan to set it up.
> 
> Bill
> 
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ray Jackson
> Sent: Wednesday, February 21, 2007 10:43 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Fax with T.38
> 
> Could anybody give me an authoritative answer on whether Asterisk can 
> support T.38 pass-through when the clients are behind NAT?  We have 
> Asterisk servicing clients behind NAT (with nat=route, canreinvite=no) 
> and would love to get T.38 going but have had no luck so far.  The 
> following case:
> 
> http://bugs.digium.com/view.php?id=7844
> 
> ...suggests that T.38 *does* now work for clients behind NAT but I have 
> the latest SVN trunk but still cannot get it to work?  On the other side 
> I have seen on this list only 2 weeks or so ago:
> 
> http://www.mail-archive.com/asterisk-users@lists.digium.com/msg172556.html
> 
> This suggests that T.38 does *NOT* work behind NAT?  So, can anybody 
> save me the trouble and tell me how it is.  Am I on a hiding to nothing 
> trying to get T.38 going with NAT?  Please put me out of my misery! :)
> 
> Cheers,
> Ray
> 
> PS. Does anybody know whether OpenPBX would support T.38 and NAT 
> configurations?  This was my backup plan if I couldn't get it to go in 
> Asterisk.
> 
> Thomas Deillon wrote:
> > Yes, the canreinvite means Re invite, but there is a consequence in 
> > Asterisk configuration.
> > 
> > For sure, all the signalisation traffic will go through the asterisk … 
> > but for the RTP traffic?
> > 
> > If canreinvite = No, all RTP traffic will go through the Asterisk 
> > (useful for NATed phoned without ALG/STUN/…)
> > 
> > If canreinvite = Yes, the phones will try to exchange RTP packets directly.
> > 
> >  
> > 
> > Do you thing there is a way to allow Re Invite (because you’re right) 
> > without the RTP consequence?
> > 
> >  
> > 
> > Thanks a lot for your help,
> > 
> >  
> > 
> > Thomas
> > 
> >  
> > 
> > 
> > 
> > *De :* [EMAIL PROTECTED] 
> > [mailto:[EMAIL PROTECTED] *De la part de* Rajnish 
> > Jain
> > *Envoyé :* lundi, 19. février 2007 16:25
> > *À :* Asterisk Users Mailing List - Non-Commercial Discussion
> > *Objet :* Re: [asterisk-users] Fax with T.38
> > 
> >  
> > 
> > A T.38 fax call typically begins as a normal voice media call. The 
> > call then dynamically switches over T.38 image media on detection of fax 
> > handshake tones.  The dynamic modification of session from audio to 
> > image is accomplished through SIP RE-INVITE messages. I would imagine 
> > canreinvite= flag controls if an end-point is allowed to send/recv 
> > RE-INVITE to/from Asterisk. If so, you'll need to set it to yes for T.38 
> > to work.
> > 
> >  
> > 
> > 
> >  
> > 
> > On 2/19/07, *Thomas Deillon* <[EMAIL PROTECTED] 
> > > wrote:
> > 
> > Hi all,
> > 
> > I make others tests.
> > Analog Fax 1 -> PATTON M-ATA -> Asterisk -> PATTON M-ATA -> Analog Fax2
> > 
> > It works only if I use canreinvite= yes.
> > But all my clients are behin

Re: [Asterisk-Users] Can someone tell me why I'm gettingthese? (mailing list probe message)

[jbebeau]

OK - I should know this... How does someone call in and pick up there 
messages remotely?

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

Sent: Tuesday, February 08, 2005 3:20 PM
Subject: Re: [Asterisk-Users] Can someone tell me why I'm gettingthese? 
(mailing list probe message)


Andrew Thompson wrote:
Twice in the last week or so, I've received a message similar to the 
attached.

A portion of the attachment that's attached is not in English. Is this my 
mail server failing, or someones who's on the list?
Andrew,
1) - When you signed up you were given the option for a monthly password 
reminder.  That is what you recieved.

2) - Speaking of passwords, you might want to change yours now that we all 
know what it is!

--
Kristian Kielhofner
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Re: [Asterisk-Users] Any experience with Sangoma cards?

[jbebeau]

No, I don't work for Sangoma or any affiliate.  I receive no compensation or 
benefit from Sangoma or any affiliate in any way.  I'm just trying to fine 
the best stuff for my use.

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

Sent: Monday, January 24, 2005 7:20 PM
Subject: Re: [Asterisk-Users] Any experience with Sangoma cards?


On Mon, 24 Jan 2005 11:38:52 -0500, Jon Bebeau  wrote:
I'm exactly in the middle of benchmarking the A104 and T410p.  I'm
developing a matrix of CPU, bandwidth throughput and trying to find high
water marks under several loads; single processor, multi processor, Xeon 
vs.
P4, Hyperthreading vs. not, and mixed voice and data T1s for the Sangoma
boards.  It's turning out to be an ordeal.  I'll post the findings when I
results.

Do you work for sangoma ?
regards
m.
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Re: [Asterisk-Users] RE: Zaptel/Zapata config from T410p to BrooktroutT1

[jbebeau]

I had tried g3 first...the individual address was at guess that g3 didn't 
work.

Thanks.
- Original Message - 
From: "Jason Kawakami" <[EMAIL PROTECTED]>
To: 
Sent: Wednesday, December 22, 2004 9:43 PM
Subject: [Asterisk-Users] RE: Zaptel/Zapata config from T410p to 
BrooktroutT1



-Original Message-
Message: 8
zaptel.conf
span=2,0,0,esf,b8zs
e&m=25-48
zapata.conf
signaling = em_w
context = faxserver
group = 3
channel = 25-28
exten.conf
exten => 1231231234,1,Dial(Zap/2-2/${exten})
I think if you change the last line to :
exten => 1231231234,1,Dial(Zap/G3/${EXTEN})
it should work.  I have had difficulty in the past doing direct selection 
of
individual channels on t-1 spans.


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Re: [Asterisk-Users] CP-7960

[jbebeau]

Yes...please contact me off list.  Condition? 
new / used  what do they come with? in boxes?  Cost?
 
I need about 20.
 
Jon Bebeau

  - Original Message - 
  From: 
  Garrett 
  Smith 
  To: 'Asterisk Users Mailing List - 
  Non-Commercial Discussion' 
  Sent: Tuesday, November 23, 2004 10:53 
  AM
  Subject: [Asterisk-Users] CP-7960
  
  
  Anyone in need of some of 
  these?
   
  Garrett 
  Smith
  Sales 
  Executive
   
  [EMAIL PROTECTED]
   
  B2 
  Technologies
  454 Sonwil 
  Drive
  Buffalo, 
  NY 14225
   
  (716) 250-3408 
  Direct
  (716) 630-1548 
  Fax
  (716) 903-9495 
  Cell
   
  AOL IM: 
  B2sales
   
  Specializing in New and Used equipment from vendors 
  including Cisco Systems, Juniper, Adtran, Dialogic, Lucent, Nortel, Sipura, 
  Granstream, Snom, Mediatrix, Carrier Access, Digium, Zultys, IPDialog and 
  more.
   
   
   
  
  

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Re: [Asterisk-Users] Cisco 7960 version 7.3 SIP not always able to hearcalling person

[jbebeau]

You should check the preferred and allowed 
CODEC in the phone SIP config.  The upgrade might have changed it on 
you.
 
Jon

  - Original Message - 
  From: 
  Jerry 
  Geis 
  To: [EMAIL PROTECTED] 
  
  Sent: Monday, November 22, 2004 1:51 
  PM
  Subject: [Asterisk-Users] Cisco 7960 
  version 7.3 SIP not always able to hearcalling person
  I have the Cisco 7960 SIP version 7.3 
  phone.When someone calls in I cannot always hear that person.They can 
  hear me though. (The ear piece is DEAD quite like it is muted or something 
  - no noise at all).This never happens with the other 4 grandstream SIP 
  phones I have.Is there a problem in my setup?Is there a problem with 
  this version of cisco SIP?Any ideas? or is this happening to other 
  users of this phone also?Thanks,Jerry
  
  

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