Hi,
I had no issues connecting/calling to my asterisk
box (on public ip), even my phone is behind a
hotspot. Its just that i need to use G711 codec.
At 03:34 PM 7/5/2006, you wrote:
Hello
Has anyone tried a Nokia E6x phone when it is
natted? Like behind a hotspot or similar?
BR
Amund Ny
Hi,
configuration for E61 is the same as E60.
As for the codec, G729 works between E60/61 phones (G729 passthru).
At 03:44 PM 7/4/2006, you wrote:
Devraj Mukherjee wrote:
> Hello world,
>
> Any success stories of getting a Nokia E61 to work with Asterisk
> server? Interested to hear before
try using g711 ulaw codec.
At 03:51 PM 5/31/2006, you wrote:
Hi
I want to check out from the members , about their
experience with Nokia E60 phone as SIP client , I was
able to register the phone , but my voice gets
broken during the calls . My other Wi-Fi VoIP SIP
phone are working fi
Hi,
Try this on your extensions.conf
[from-mysipprovider]
exten => 1140636249,1,Dial(SIP/29650)
exten => 1140636249,2,Hangup
At 07:34 AM 3/30/2006, you wrote:
Hello All,
I am using incoming DIDs for the first time. I ll very happy if someone
can help me on that serttings ... I need to
f and
sip.conf for my voicemail users. Does it really has a limit?
Thanks,
Ryan
At 08:23 PM 3/23/2006, Antonio Rabena wrote:
How about moving your voicemail users into db?
At 03:50 AM 3/23/2006, you wrote:
Hi,
Is there an account limit for voicemail? I have 80+ users
in the voicemail and
How about moving your voicemail users into db?
At 03:50 AM 3/23/2006, you wrote:
Hi,
Is there an account limit for voicemail? I have 80+ users
in the voicemail and I can only reach the 70-ieth user. If there is
a limit how can I increase it to hundred for example?
Thanks,
Ryan
___
Hi,
I also experienced broken page receiving fax on asterisk + spandsp
with Digium TE410P. I also tried diff. versions of spandsp and
asterisk, still no luck.
I had no issues using the same asterisk + spandsp config with TE110P.
Any ideas?
At 09:21 AM 11/24/2005, you wrote:
Hi, All
HI,
I'm using asterisk + voicetronix openswitch12 (using fxo). I just
noticed when I call a pstn number (mobile number), asterisk will
answer the call first before it actually dials the destination
number. Is this normal?
-- Executing Wait("SIP/192.168.1.130-081671b0", "1") in new sta
Hi,
Is it possible to disable the retries in callback?
Regards,
Antonio
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Yes there is.. the default setting is /init.cfg. Not sure about these
parameters.. I cant find this provisioning setting on the user-guide.
maybe anyone can help?
Regards,
Antonio Rabena
Andres wrote:
Antonio Rabena wrote:
I have 1 PAP2-NA. Configuration is done thru Phone (IVR) and Web
I have 1 PAP2-NA. Configuration is done thru Phone (IVR) and Web. Im
wondering if this ATA supports auto-provisioning.
Matthew Boehm wrote:
Does anyone have one of these models? Can they confirm that it works with
any other SIP server? How is the PAP2-NA configured? Web? Phone?
The pdf I downlo
You need an E1 back-to-back cable.
Regards,
antonio
Francesco Delfino wrote:
Hi,
I would like to realize a voip testbed that should simulate the scenario
in which two companies have an asterisk PBX connected through a PRI-ISDN
to the Telco operator.
I have no experience of T1/E1 connection but I th
Hi,
How can i send dtmf tone upon connection?
On Friday 30 January 2004 01:15, Nick Bachmann wrote:
> > Dear to all
> > someone know how is possible to have a DTMF tone like "C"
AKA Alpha
> > Tone
> > (connect tone) to the caller?
>
> Yes, it's possible.
>
>
Regards,
Antonio
maybe you can if try this..
span=2,1,0,ccs,hdb3,crc4
Claus Futtrup wrote:
Here you go.
loadzone = no
defaultzone = no
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
bchan=17-31
dchan=16
span=2,2,0,ccs,hdb3,crc4
bchan=32-46
bchan=48-62
dchan=47
Both E100P are connected to PSTN.
Kind Regards
Claus Futtrup
_
try to put hidecallerid=no in your zapata.conf
Pedro Vela wrote:
Yes, my phone company has enabled the Caller ID hiden possibility, thats
because with a Panasonic PBX works fine but with Asterisk not. Thanks for
your aproach, what can I do now?
Regards,
Pedro
___
=2101
qualify=yes
nat=yes
canreinvite=no
and my extensions.conf
[sip-users]
exten =>_21XX,1,Dial(SIP/[EMAIL PROTECTED])
[default]
exten s,1,Hangup
Regards,
Antonio Rabena
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x this bug.
I noted
the same problems with CVS from 03/30/2004 when incoming calls
were sent to voicemail. Anyway I had to roll back to 03/05 since
last
Zaptel was giving me yellow alarms con my TE410P on a E1 PRI.
--
Juanjo sin .sig
Regards,
Antonio Rabena
install..
later..
Regards,
Antonio Rabena
Zap/g2/65699062
Regards,
Antonio Rabena
At 05:44 AM 2/18/2004, you wrote:
2. can Two SIP phones login
to * at the same time with the same
username/pwd ? how to prevent
this?
I also want to know if its possible to prevent multiple
logins..
Regards,
Antonio Rabena
Hi all,
Is it possible to have multiple context= for user configuration in sip.conf?
Regards,
Antonio Rabena
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Hi,
Is it possible to have multiple context= for user configuration in sip.conf?
Regards,
Antonio Rabena
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