I've had good experience with Algo's VoIP paging solutions.
--
Nabeel Jafferali
On Thu, Mar 21, 2019 at 3:00 PM Michael Munger wrote:
> Does anyone have an (overhead) paging system that they like that works
> with SIP?
>
>
>
> We’ve got a client with an old pa
.
Nabeel
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Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display
Hi,
When I call my Asterisk server using a SIP client, the ICE server is used
correctly if calling as a 'voice call', but the ICE server is not being
used if calling as a 'video call'.
Please let me know what could be
>
> What happens when you dial "*98" from your own
> phone.
I get password prompt if a password is set, and no password prompt if no
password is set.
On 4 August 2016 at 14:36, D'Arcy J.M. Cain wrote:
> On Thu, 4 Aug 2016 14:03:39 +0100
> Nabeel wrote:
&
> On 4 August 2016 at 13:18, D'Arcy J.M. Cain wrote:
>
>> without asking for a password.
>>
>
>
I should add, a password is *always* asked if a password has been set.
There isn't a way to bypass that.
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uring the message should put you into the callee's
> mailbox and ask for a password. Calling '*98' from your own phone, if
> the extension I originally showed you exists, should put you directly
> into your own mailbox without asking for a password.
The password is only as
this does not necessitate setting
a password.
The problem is that the 'mailbox' prompt allows a way for accessing any
other mailbox, which is not necessary in my case. If anyone knows a way to
remove this 'mailbox' prompt, please let me know.
Nabeel
--
e it is more efficient for
users to skip that step if possible.
Nabeel
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New to Asterisk? Join us for a live introductory webinar every Thurs:
http:
t to test this behaviour in Asterisk during the Unavailable/Busy
message. However, if this is the case, then this seems to be an illogical
security hole in Asterisk's design. Why does Asterisk allow accessing
another person's mailbox by pressing
'unavailable
message' or 'busy message', if they press '*' at that time they will enter
the other person's mailbox?
Nabeel
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If I remove the password, how can anyone access the mailbox if the
'mailbox' prompt is not played?
Nabeel
On 30 Jul 2016 3:19 p.m., "D'Arcy J.M. Cain" wrote:
> On Sat, 30 Jul 2016 06:43:47 +0100
> Nabeel wrote:
> > I am using Asterisk voicemail on a Ce
Hello,
I would like to remove the 'Comedian Mail' name/message played when a user
tries to access their voicemail. Please let me know how to do this.
Nabeel
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x27;mailbox' prompt is always heard. Please let
me know how Asterisk can be configured to remove these prompts.
Nabeel
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BTW I just did some quick experimentation. Example 1 did not work, example 2
did work. So that's a solution to your issue.
Example 1:
exten => 999,1,Goto(nabeel,1)
exten => _nabeel,1,Goto(800,1)
Example 2:
exten => 999,1,Goto(nabeel,1)
exten => _[n]abeel,1,Goto(800,1)
--
Na
Asterisk is looking for hilto[1-9]-2[0-9][0-9], if you know what I mean?
--
Nabeel Jafferali
X2 Networks
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall
Sent: February-11-09 8:18 PM
To
Disconnect and Polarity Reversal settings, to no
avail.
Anyone experienced this issue with Audiocodes or any other gateway in general?
Any tips?
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Nabeel Jafferali
X2 Networks
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Looks like a rebadged Patton 6075 to me:
http://www.patton.com/products/pe_products.asp?category=337
Nabeel
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Dean Collins
> Sent: May 27, 2007 11:53 AM
> To: Asteris
There is some info on page 3 of
http://www.digium.com/en/docs/TE120P/TE120P-user-manual.pdf.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Klaverstyn, David C
> Sent: May 16, 2007 6:38 PM
> To: Asterisk Users Mailing List - Non-Comm
Did you have the IP specified in sip.conf?
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Yaakov Menken
> Sent: May 13, 2007 10:43 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Sudden appearance of SIP/2.0 401
Hello.
I have a customer that needs to downgrade the firmware on their SPA841 to
3.1.1(a). I can't seem to find the firmware file. Google turned up
3.1.2-something and Linksys is taking a while to get back to me.
Anyone happen to have that file lying around?
Thanks,
N
k-users] headsets for linksys/sipura phones?
>
> Nabeel Jafferali wrote:
>
> > You can look for headsets made for Motorola cell phones. Also,
> > Plantronics has some compatible models - I can dig up part
> numbers if
> > you're interested.
> >
>
You can look for headsets made for Motorola cell phones. Also, Plantronics
has some compatible models - I can dig up part numbers if you're interested.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Per Jessen
> Sent: April 27, 2007 8:32 AM
> T
You can purchase the Linksys part PA100-NA and plug it into a WBP54G and
then ignore the power connector hanging off the WBP54G.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Mike
> Sent: April 27, 2007 3:00 PM
> To: 'Asterisk Users Mailing List
Does anyone have any experience using the Huawei Videophones in a
point-to-multipoint configuration using Asterisk?
Thanks,
Nabeel
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~$50, I'd say that's a decent deal,
assuming the phones serve your purpose.
Nabeel
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Tim
> Sent: September 28, 2006 7:54 AM
> To: asterisk-users@lists.digium.com
> Subject: [
I don't belive there is a way to turn it off, but you can prevent the IVR
menu being used to factory reset the device using the provisioning tools.
Nabeel
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Jamin W. Collins
>
In that case, it is likely your reseller is not a "Polycom certified VoIP
reseller". Contact me off-list and I'll help you.
Nabeel Jafferali
www.voipdepot.ca
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Lucas Alva
Up here in Toronto, on a PRI to an Asterisk box with a Sangoma T1 card, I
can send DTMF fine and the call is immediately "connected".
Nabeel
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Steven
> Sent: June 21, 2006 2:12 P
ly outside our company WiFi network, so I can't
say anything about that.
Nabeel
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> Anyone know what the differences are between the spa3000 and
> spa3102 other then packaging?
The SPA3102 has a built-in router (LAN + WAN), T.38 on the FXS port and can
handle dual G.729 sessions.
Nabeel Jafferali
www.voipdepot.ca
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> Is the 94x any better? seems without backlighting, any are
> next to useless.
The SPA-9x2 have backlit displays.
Nabeel
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> I have followed these two for configuration of sangoma
> A101
> http://www.ss7box.com/s01_setup.html
> http://www.ss7box.com/support_wancfg_1.html
Why not go straight to the source and use the instructions at
http://sangoma.editme.
> there some sort of explicit dialplan command that stops
> execution and immediately ends the dialplan? Something like
> MacroExit() in a macro Can't see it in the docs.
Hangup(), maybe?
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That list is obviously not complete ;)
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Eric "ManxPower" Wieling
> Sent: May 11, 2006 2:43 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Asterisk a
be IVR-accessible and is only to record
announcements for different groups by the respective groups.
However, I see your point. They need a sort of "tenant" capability.
Nabeel
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Asterisk-Users ma
r, regardless, does [EMAIL PROTECTED] have built-in functionality to have
a hidden
menu for a user to modify a recorded file (i.e. a file played as an option
on the IVR). And I don't mean from the [EMAIL PROTECTED] web interface, I mean
an option
that can be added
f hours to write and test this, but I thought if someone has
something already written, I could just "borrow" it from you.
Thanks,
Nabeel
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Early audio is played, as long as you do not have a "r" in your Dial
statement.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Mohammed Salim
> Sent: April 13, 2006 2:17 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-U
> Is there anyway I can make one particular extension always dial out on
> one specific pots line(group)
Set the context= for the extension's sip/iax/zapata.conf entry and then in
that context dial out the specific Zap channel/gro
.
BTW I'm running 1.2.0.
> PS. While I'm here, is the A102d worth the money? I assume it works as
> easily as the A104d?
Yes, great card. Worked right away using the simple instructions at
http://sangoma.editme.com/. Contact me off-list if you need more info.
Nabeel
> try SetCallerId or set callerid=name <(xxx)xxx-> in sip.conf or
> iax.conf (depending on what you are using)
I am not using SIP or IAX2 clients. As mentioned in the original email, this
is from PRI to PRI.
I could use SetCallerID, but the zapata.conf method should w
RI. However, the CLID is not set to the
number I set.
In fact, the ${CALLERID} variable is empty (which is what the PBX is
sending, but what I would have expected the callerid= setting to overwrite).
Any ideas?
Nabeel
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I call AmEx at 1-800-297-1000 and my VoIP provider
drops the call after 30 secs thinking nobody is answering.
But if I just hit 0 the call "connects".
Nabeel
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T
and a call comes in on 103, the phone responds "Busy
Here".
Is there any way of assigning groups of Line buttons to different
registrations (like the snom phones)?
Nabeel
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Asterisk-Use
if you try to dial a missed call by simply hitting the dial
> button. Can anyone else verify this problem?
Yeah, that bothered me so I rolled back to SIP 7.4.
Nabeel
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> Anybody using sipuras 2002 knows if there is a way to make
> the phones connected to it to autodial an extension when the
> phone is picked up?
http://www.sipura.com/Documents/faq/Section_2.html#5
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cnf to (192.168.1.60), 6173 bytes
09/03/2006 10:12 :Sending RINGLIST.DAT to (192.168.1.60)
09/03/2006 10:12 :Sent RINGLIST.DAT to (192.168.1.60), 19 bytes
09/03/2006 10:12 :Sending dialplan.xml to (192.168.1.60)
09/03/2006 10:12 :Sent dialplan.xml to (192.16
> exten => _50,1,Dial(...)
> exten => _5!,1,Dial(...)
Remove the "_" from the first line.
Nabeel
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h
> Is there a way to display the time of the 7960 running firmware 7.4? Im
> unable to find any information.
Add the following to SIPDefault.cnf or SIP.cnf:
sntp_server: "time.nrc.ca"
sntp_mode: unicast
time_zone: EST
You should of course change your NTP server and/or ti
y.
Give me a few minutes to write up the procedure.
Nabeel
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bin/tablebuild.pl/sip-ip-phone7960.
I do see a .cop file for the 7941/7961 8.x SIP load, but nothing for the
7960.
Nabeel
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the
VM number. This is incorrect behaviour to the best of my knowledge.
Nabeel
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> I am using a snom 320 running 5.3.6 with Asterisk 1.2.4. In the sip.conf
> entry, I have [EMAIL PROTECTED] and vmexten=*98.
>
> The light on the snom 320 turns on when I have voicemail and the retrieve
> button dials the correct extensions.
>
> However, the light turns off immediately after mak
sys appears to have maintained the same policy for the SPA- devices.
So, contact your reseller.
BTW I always get quick responses from [EMAIL PROTECTED] (which I believe is
forwarded to [EMAIL PROTECTED]).
Nabeel
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the call to voicemail,
even if I do not check the voicemail.
Any idea on how to get this to behave properly?
Nabeel
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> Does anyone have a way to do wake calls?
http://www.voip-info.org/wiki/view/Asterisk+tips+Wake-Up+Call+PHP
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> What is the best way to ignore a DID and not pick up the line?
> I don't want to incur charges on the line (T1 PRI), so would
> Hangup pick up the line, then hang up? Or can I use Hangup?
exten => 416967,1,Busy()
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CTech wrote:
> Nabeel Jafferali wrote:
> > Am I missing something completely obvious? Is there a way to see why
> > Asterisk is sending 488 (i.e. what is not acceptable?).
>
> Did you solve the 488 error? I run into the same problem as you.
>
> Below is my invite
ep the port(s) open. At the same time,
most SIP devices have a NAT Keep Alive option, if that is an issue.
Obviously, this doesn't explain what changed to cause the issue, but this
should at least have you up and running until you do figure it out.
Nabeel
___
> Hi. I'm wondering if it is possible to make asterisk honor the
> Presentation Restricted bit on incoming PRI calls.
I'm guessing you have to make your dialplan remove the CLI based on the
${CALLINGPRES} variable.
Nabeel
___
> How can I send the hook flash to the x100P card to switch to the call
> coming in from the PSTN?
http://www.voip-info.org/wiki-Asterisk+cmd+Flash
Scroll down to "Re: X100P + Call-Waiting how-to"
Enjoy.
Nabeel
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> Anyone in Winnipeg Canada?
Winnipeg seems to have an active Asterisk group. Their mailing list is at
http://www.muug.mb.ca/mailman/listinfo/asterisk and I believe they are
having some kind of event soon.
Nabeel
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rrect user, however it seems to forget
the user's AccountCode (which is set in sip.conf).
Any ideas?
Nabeel
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> Outbound proxy
> Proxy IP stun01.sipphone.com
> Port:: 3478
STUN servers are not outbound SIP proxies.
Nabeel
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> I got some troubles with my wifi phone.
What phone is this?
Nabeel
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uccessful, while pbx-
> >pstn works fine. The tones simply don't make it through. Tiny brief
> fragments are all.
It might help to describe what interfaces your Asterisk PBX with the PSTN.
Is this a VoIP provider DID you are using, or a POTS line with an interface
card, or a PRI with
If you have, in sip.conf, a register => blah:[EMAIL PROTECTED]/12345, you
would also have:
[blah]
host=sip.blah.com
context=from-blah
Then, in extensions.conf, you would have:
[from-blah]
exten => 12345,1,Dial(whatever)
...
Nabeel
From:
> But I am still trying to find out what the "Shared" line appearance is
> on these phones?
The SPA-841 Admin Guide mentions that the "Shared Call Appearances" features
is for Broadsoft server implementations, and that other vendors' server
implementations will supp
> I don't suppose anyone here knows where I can find a list of a whole heap
> of US resellers do you in either VOIP or IP space?
This might help:
http://www.voip-info.org/tiki-index.php?page=Asterisk+system+vendors
--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
1.877
Jorge Mendoza said:
> We are using in production Bodhicom gateways (www.tainet.net).
> They work fine and are not expensive.
Any idea where to buy these in US/Canada?
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Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
1.877.VOIP.X2N
F: 1.866.655.6698
FWD:
f you dial 35, you want Asterisk to run the 33 extensions
instead. If so, you need, for example:
exten => 35,1,Goto(33,1)
exten => 33,1,Voicemail
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Nabeel Jafferali
X2 Networks
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T: 1.647.722.6900
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F: 1.866.655.6698
FWD: 46990
_
> Is it possible to terminate (or forward) lets say 800 DID number to FWD
> number.
Depends on your carrier. Just ask them to forward the DID to
sip:[EMAIL PROTECTED]
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Nabeel Jafferali
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FWD:
with a slightly lower price, I think). The current
version of the IAX2 firmware has some limitations. However, the IAX2
firmware (and maybe the others) has recently been released as open-source,
so you can expect significant community-driven improvements.
--
Nabeel Jafferali
X2 Networks
www.x2n
> Has anyone got recommendations on WIFI Phones that work with Asterisk?
I use the Hitachi Cable Wireless WiFi phone every day with my Asterisk setup
through a home NAT router - works great.
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Nabeel Jafferali
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ll setup even though it might end up using G711). The
> codec is only released once the call is set up.
I know that, which is why the sip.conf entry is set to allow=g729 and
allow=ulaw.
Any other ideas?
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Nabeel Jafferali
X2 Networks
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T: 1.647.722.6900
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F:
entry is:
[3255]
type=friend
username=3255
secret=**
accountcode=3255
callerid="Bode Dar" <>
context=clients-int
host=dynamic
qualify=yes
nat=yes
disallow=all
allow=g729
allow=ulaw
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Nabeel Jafferali
X2 Networks
www.
the SIP packet does
not have an entry for accountcode.
> My solution was to send the accounts into their own contexts and set the
> accountcode and callerid their instead.
That is a workaround, but it is not a scalable solution. Any other ideas?
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Nabeel Jafferali
X2 Networks
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T:
tch in the file is choosen.
No, there is not. I even checked sip debug, it has a line saying 'Found peer
"customer"' indicating the right peer entry is being used.
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Nabeel Jafferali
X2 Networks
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eason, the AccountCode is blank. I have a
NoOp(${ACCOUNTCODE}) and the CLI shows:
-- Executing NoOp("SIP/x.x.x.x-0821e058", "") in new stack
indicating a blank AccountCode.
I tried username-based authentication and have the exact same issue. Anyone
have any ideas on why this
nels, search for a posting by me a week or two ago
regarding this problem.
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Nabeel Jafferali
X2 Networks
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t context in iax.conf, under [general], that
points to the extensions.
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w to have a
guest peer.
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> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Anton Krall
> Sent: May 18, 2005 1:54 PM
ship COD (within Canada).
I have never done business with netVOICE but have heard only good things
about them. At the same time, we carry most of the same products and would
be happy to help any Canadians out :) Check us out at www.voipdepot.ca.
Competition is great, isn't it?
--
Nabeel Jaffe
> My 7960 is configured for two lines, and I can turn the other appearance
> buttons into speed dials from the menus, but is there any way to program
> the speed dials in the SIP.conf file?
You can not: http://tinyurl.com/az4fp
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Nabeel Jafferali
X2 Networks
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T: 1.647
nd calls using your prefix, for example *356
> >200, to, for example, [EMAIL PROTECTED]
> How about sip.conf? or iax2.conf?
As long as [default] is the context in [general] in sip.conf, I believe it
should work.
--
Nabeel Jafferali
X2 Networks
www.x2
With STUN set up correctly on the phones, you really should not need to port
forward. The phone should be able to discover the mapping and work with it.
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Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
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F: 1.866.655.6698
FWD: 46990
> -Original Mess
Make this your dialplan:
(S0<:100>)
where 100 is the number to be dialled.
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Nabeel Jafferali
X2 Networks
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T: 1.647.722.6900
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F: 1.866.655.6698
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> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTE
r, you would Dial(SIP/[EMAIL PROTECTED]).
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Nabeel Jafferali
X2 Networks
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T: 1.647.722.6900
1.877.VOIP.X2N
F: 1.866.655.6698
FWD: 46990
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Anton Krall
> Sent: M
r example *356
200, to, for example, [EMAIL PROTECTED]
--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
1.877.VOIP.X2N
F: 1.866.655.6698
FWD: 46990
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voicemail.
Any ideas on what to do? The only thing I came up with is to also create
local voicemail boxes that email the voicemail to the user when the 'net
connection comes back. I was wondering if there were any more creative
ideas.
Thanks for reading.
--
Nabeel Jafferali
X2 Networ
> I was just wondering, Do sipura 200 and linksys pap2 ATAs send their
> mac address in REGISTER message? Is their any other way to get the MAC
> address of sip peer who is trying to register?
No, the PAP2-NA does not send its MAC address in a REGISTER message.
--
Nabeel Jaf
> Will this fix it or is the \@ change necessary also. If it is, I will get
> a patch in
> tomorrow.
I just checked - Dial(Local/1416967/routes/n) is not valid.
Dial(Local/[EMAIL PROTECTED]/n) is valid. Therefore, you would need the
other change as well.
--
Nabeel Jafferali
X2
be:
$dialstr = "Local/[EMAIL PROTECTED]>{path}/n|30|HL(" . ($maxtime * 60 * 1000) .
":6:3)";
which would result in the channels not being bridged. This fixed the problem
I was having with 0 sec billed calls.
--
Nabeel Jafferali
X2 Networ
> I found records from my provider about 36 minutes phone call, while
> ASTCC says
>
> .. and of course the user is not billed for that call.
Are you by any chance using Local channels to route the call?
--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
1.87
> I have a customer that's located in France and he wants french phones
> if possible. So I'm wondering if there's any one out there that found
> a phone that can be change to french.
I believe snom phones have the option.
--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T:
your site.
Am I the only one who gets ticked off when people advertise their own site
acting as a disinterested third party? If you made the headset adapter and
want to sell it, say "that's my site, I have 'em available."
--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.69
ear 'collate utf8_unicode_ci NOT NULL
default '',
`cid_name` varchar(50) collate u' at line 3
I have no experience with mySQL. Any idea what that means?
--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
1.877.VOIP.X2N
F: 1.866.655.6698
FWD: 46990
the external address?
We did not misunderstand. I answered the question exactly as you asked it.
Someone else veered the topic away from what you intended.
--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
1.877.VOIP.X2N
F: 1.866.655.669
> And if asterisk is behind nat doing prot forwarding? Say you just
> forwarded
> udp 4569 5060 5004 1-2000?
You'd just need to set externip correctly, assuming you have a static public
IP.
--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
1.877.VOIP.X2N
F: 1.86
> Any special settings on * or your nat firewalls?
Nope.
--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
1.877.VOIP.X2N
F: 1.866.655.6698
FWD: 46990
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