[asterisk-users] FAX with SIP

[Israel Gottlieb]

On Fri, Jul 22, 2011 at 12:50 AM, Kevin P. Fleming <kpfleming at digium.com>wrote:

> On 07/21/2011 04:43 PM, Israel Gottlieb wrote:
>
>>
>>
>> On Fri, Jul 22, 2011 at 12:39 AM, Kevin P. Fleming <kpfleming at digium.com
>> <mailto:kpfleming at digium.com>> wrote:
>>
>>    On 07/21/2011 04:34 PM, Joaquin Sosa wrote:
>>
>>        On Mon, Jul 18, 2011 at 07:58, Steve Davies<davies147 at gmail.com
>>        <mailto:davies147 at gmail.com>>  wrote:
>>
>>
>>            The magic sauce that you need is "T.38" - Asterisk 1.6
>>            supports this
>>            to a limited degree, and your ITSP will need to support it.
>>
>>            The sip.conf.sample file and the voip-info wiki has all the
>>            information you need to try it out.
>>
>>
>>        Correct. However it would be helpful to note T.38 support in
>>        Asterisk
>>        is broken. Take a T.38 ITSP, T.38 enabled Asterisk and T.38 ATA and
>>        try to send a fax. It won't work!
>>
>>
>>    We do this in our testing all the time, and it works fine. Since you
>>    didn't specify any particular version of Asterisk, there's no way to
>>    associate your "It won't work" statement with anything in
>>    particular. Given the variations of T.38 implementations that exist
>>    in ATAs, carrier networks and other places, *any* T.38 connection
>>    that involves implementations from more than one vendor is
>>    (unfortunately) likely to have problems, whether any version of
>>    Asterisk is involved or not
>>
>>
>>
>> well I tried  a linksys spa 8000 and 2102 thru
>> asterisk 1.8.3
>> 1.8.4
>> 1.6.2.16-19
>> sonus switch at itsp (012 israel)
>>
>> and no luck
>>
>
> We'd be happy to investigate why it failed, if you can capture the packet
> streams on both sides of Asterisk. Frequently, it's a configuration issue in
> at least one of the devices in the system.
>

NP I'll get that for you I have spent days trying to get it to work with no
luck

>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at www.digium.com & www.asterisk.org
>
> --
> ______________________________**______________________________**_________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>              http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>  http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110722/ec99e873/attachment.htm>