. in the directory /usr/src/asterisk-1.8.18.1/contrib/realtime/mysql
sippeers.sql
There isnot encryption field, if I add it by hand, it can not take effect.
Regards.
Hugo
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Hugo Serrano
hugo.serr...@javali.pt
Javali
/?string=Linksys
For the Linksys Sipura, they should all start with 00:0E:08, because
that was the Vendor ID for Sipura.
Regards!
Hugo Serrano
On 12/07/10 16:41, Frank Church wrote:
Is there a database of MAC address prefixes used the common VoIP
devices. I see the Linksys Sipura devices state
of the week
/sbin/reboot - Comand
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Hugo Serrano
hugo.serr...@javali.pt
Javali
Administração e Desenvolvimento de Sistemas Informáticos, Lda.
Madan ParqueEdifício VI Campus da FCT/UNL
Quinta da Torre 2829-516 Caparica Portugal
Phone: +351 212949666
. But for the record - I have GSM
failover...) :-)
What are the options? Any phones I should avoid?
Maybe anyone of you have done something by yourself? :)
Best wishes, have a nice weekend!
Motiejus
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Hugo Serrano
hugo.serr...@javali.pt
Javali
Administração e Desenvolvimento de Sistemas
Hi,
I'm using a Cisco IP Communicator 7.0 enable with SIP, but I can't to
configure it to add the lines because I can't to Select the options, for
example: the Name, Authentication name...
Somebody help me??
Thank you!!!
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São Paulo, SP here ;)
On 7/29/07, Luis Antonio Prata Barbosa [EMAIL PROTECTED] wrote:
Yep. From Brasilia-DF ! :)
2007/7/29, Jozeph Brasil [EMAIL PROTECTED]:
Some brazilian here on list?
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Hugo da Costa Rebelo
Tel: +55 (11) 7165-4630
www.hugorebelo.com
Hi Kyle,
You need to set up a inbound route from DID=skype1 and tell him where to finish.
Something like:
exten = skype1,1,Set(FROM_DID=skype1)
exten = skype1,n,Goto(ext-local,1000,1)
Hope it helps.
Best Regards,
Hugo Picão
Link Consulting - RedesSegurança
Tel: 213 100 182
Av. Duque de
Hi There,
Good guide on setting up chanskype on trixbox
http://www.geek-pages.com/articles/asterisk/setting_up_trixbox/asterisk_to_use_skype.html
also:
http://www.chanskype.com/
working on my trixbox 2.0 :)
Best Regards,
Com os melhores cumprimentos,
Hugo Picão
Link Consulting
if anyone calls from the outside and tries to reach a PC.
Any ideas?
Regards.
--
IEE José Hugo Pérez Casanova
Profesor Investigador
Departamento de Ingeniería Electrónica
Instituto Tecnológico de Veracruz
M.A. de Quevedo #2779, colonia Formando Hogar
Veracruz, Ver. Mexico
Tel/Fax: (52) 229-938
I'm using Asterisk (1.2.14, RedHat 9) but I've been having trouble with SIP
re-invites.
I have a DiD from an ITSP and when someone calls in, Asterisk plays a menu
recording and transfers the call to the external line the caller selects.
Since both sides of the call are external, I want to use
If a call comes into my Asterisk server on a DiD provided by an ITSP and the
dialplan sends that call to another external number throught the same ITSP's
network, I don't want the RTP packets to pass through my server once the
call is bridged.
I have had great success getting this to work using
SetUp:
- Asterisk behind a NAT,
- Red Hat 9.0
- Asterisk 1.2.14
My Asterisk box is behind a NAT and I have a DiD from an ITSP. I have my
dial plan set up so that when outside callers dial the DiD, the call is
answered by my auto-attendant. The caller can then select who they'd like
to speak to
configuration is differentfrom DYNAMIC).Regards,Hugo
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Anyone could help to use Static RealTime with SIP.CONF. I use Dynamic Realtime successfully. In fact, I want to know how to compos the correct DB(postgres or mysql) fields (I think
STATIC configuration is different from DYNAMIC).
Regards,
Hugo
Hello again everyone,
I'm having this same issue with Asterisk. Any ideas ?
Hugo
Cullin J. Wible wrote:
After all of your feedback and a discussion at Teliax we have fixed this
issues.
It appears that when dialing a PSTN number, using the 'r' option is really
unnecessary.
Furthermore
Thank Brian !
That gave us the last piece we needed.
Regards,
Hugo
Brian Roy wrote:
On 6/7/05, Johann [EMAIL PROTECTED] wrote:
Hugo,
1118098465|1118098465.47|salesq|NONE|ENTERQUEUE||Ray Balbin 25
(716)250-3405
2nd column is not really sure...maybe the duration
Thanks Johann. - that helps out .
Johann wrote:
Hugo,
1118098465|1118098465.47|salesq|NONE|ENTERQUEUE||Ray Balbin 25
(716)250-3405
1st column is unixtime stamp for the current date
2nd column is not really sure...maybe the duration?
3rd column is the queue name
4th column is their agent
(716)250-3405
I found a doc that tells me about everything from ENTERQUEUE and on
but nothing on the 4 fields before it.
Can anyone she some light on this ?
Hugo
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-terminated line.
My question is: what do I need to hook that TDM T1 line into the server
where I'm running Asterisk? Is it simply a Digium card like the Wildcard
TE410P?
Many thanks!
Hugo
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