I am dialinig the far end fist, because I need to. how ever I tried the
reverse, and the result is the same i.e. info on one end missing. This
time the extension is missing. I think the CDR generator of the Asterisk
needs change to record the complete information.
***
On 30 dec 2006, at 04:59, Vernier Umali wrote:
I do not have
any luck using nokia E61 (doesn't register and keeps on hanging). I
would think it's the same with all wifi enabled nokias.
To balance this out, i'm using a nokia e61 on a daily basis as my
main phone in the office without
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On 30 Dec 2006, at 04:59, Vernier Umali wrote:
I do not have
any luck using nokia E61 (doesn't register and keeps on hanging). I
would think it's the same with all wifi enabled nokias.
Sweeping generalizations never work.
My E60 works fantastic
Hi,
I found a solution to my own problem... well, sort of.
When I put the follwing in my misdn.conf:
[intern]
ports=1
callgroup=1
pickupgroup=1
always_immediate=yes
nodialtone=yes
context=intern
I.e. send misdn_chan directly to the s extension and do not allow it to
create a dial tone, and
On an Asterisk 1.2.12.1 system using a TDM400P with two FXO and two FXS
modules, servicing two POTS lines:
When dialing a number, such as a bank, or pharmacy, where it is required to
enter a long series of numbers via the phone's keypad, an unexpected hangup
occurs.
The hangup does *not*
Hi all,
My problem seems to be solved,
When we have multiple SIP accounts on the same phone with RealTIme
configuration, Asterisk can't authenticate correctly the second account, I
think it's because of the same IP and port number.
My solution is to use insecure=invite on the second SIP
Vernier Umali a écrit :
[...] I do not have
any luck using nokia E61 (doesn't register and keeps on hanging). I
would think it's the same with all wifi enabled nokias.
Nokia E70 with latest firmware works perfectly.
I used an Ipaq
6900 series and Asus P55 and both worked well with SIP
Leo Ann Boon wrote:
Have you tried using the agi unit at
http://home.cogeco.ca/~camstuff/agiunitpas.txt?
Leo
Yes. I have tried that and get the same thing.
--
Warm Regards,
Lee
___
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I guess I misunderstood your issue, Fred.
Have a great New Years.
Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
iaxtel: 700:248:2637 x:1500
http://www.sheltonjohns.com
On Dec 30, 2006, at 8:59 AM, Asterisk
Have you tested the Ipaq or Asus with softphones in a roaming environment?
Jorge Mendoza
Vernier Umali wrote:
The best experience I had in using a wifi handset to connect to
asterisk is a windows mobile based PDA. I had the priviledge of
testing a few phones in our company to connect via VOIP.
Andrew, in your experience, what has changed from version to version. I
work daily with Avaya gear and do regression testing on new releases before
they hit the public. In my experience, I can't recall anything changing
with h.323 trunking other than the maximum number of trunk members managed
Lee Jenkins wrote:
Hi all, after trying a number of different ways to get this to work, I
have found a way.
For some reason, it does not appear to me that using standard Writeln()
to send commands to Asterisk are ignored for some reason, even when
appending #10 or #13#10 or #13 to the
Hello Everyone,
I can see that a few people are interested in SIP WIFI phones. I have
tested several Linksys 300,and it is OK. More of a toy then a business
tool. It a poor built in ear speaker, which makes all calls sound tinny,
and the unit is known to hang. I have two Linksys 300's that are
Always...
Desire that in the New Year that if you really initiate...
It hears the words that always it desired to hear. It pronounces the phrases
that one day it desired to repeat.
It feels the emotion that always waited to feel.
It walks for the tracks that one day it desired to follow.
It
I think the CDR generator of the Asterisk
needs change to record the complete information.
Agreed. However, there are still challenges here. First, you could use
the custom_csv to create your own CDR layout that includes the dialed
number, but you'd still need to come up with a way to get
Matthew Mackes wrote:
Check them out:
www.neobits.c_m/zultys_wip_2__wi-fi_ip_voip_sip_telephone_p9656.html
Got 404 error on that link.
--
Warm Regards,
Lee
___
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asterisk-users mailing list
On 30 Dec 2006, at 23:14, Lee Jenkins wrote:
Matthew Mackes wrote:
Check them out: www.neobits.c_m/zultys_wip_2__wi-
fi_ip_voip_sip_telephone_p9656.html
Got 404 error on that link.
try :
http://www.neobits.com/zultys_wip_2__wi-
fi_ip_voip_sip_telephone_p9656.html
Tim Panton
www.neobits.com/zultys_wip_2__wi-fi_ip_voip_sip_telephone_p9656.html
On 12/30/06, Lee Jenkins [EMAIL PROTECTED] wrote:
Matthew Mackes wrote:
Check them out:
www.neobits.c_m/zultys_wip_2__wi-fi_ip_voip_sip_telephone_p9656.html
Got 404 error on that link.
--
Warm Regards,
Lee
I removed the o in com in case the list serv blocked web addresses. Here
is the whole address:
www.neobits.com/zultys_wip_2__wi-fi_ip_voip_sip_telephone_p9656.html
And here is the Zulty manufacturers link:
*snipped
Second, when using a .call file (or the manager interface's Originate
action) the 'Dial' action is executed BEFORE entry into the dialplan, so
if it fails, nothing in your dialplan is executed and you get a somewhat
*snipped
not *exactly* true.
you need to add
;this extension
Hello everybody,
currently I'm implementing redirection into my dialplan.
What I want to do is: If a call comes in to my extension I want to dial
back out to my cell phone.
So far it works very well, but I've got a problem with the displayed
number on the cell phone.
What I want is that the
Jorge:
Roaming is irrelevant in VOIP. You just need a fairly good wifi connection.
Marcel,
As I've said, I'm using a pre-production unit (for testing purposes).
I'll give your comment a try and upgrade the firmware to 2.x or 3.x.
Thanks for the tip.
We have a PRI line setup on an asterisk box.
I want to provide a couple dial-in lines for internet access from our remote
offices.
The remote offices only have analog lines.
I thought I could do this with the ZapRAS or PPPD functions in Asterisk, but
the more I read this is only for ISDN
Sounds like an EBay ad...
On 12/30/06, Josué Conti [EMAIL PROTECTED] wrote:
Always...
Desire that in the New Year that if you really initiate...
It hears the words that always it desired to hear. It pronounces the
phrases that one day it desired to repeat.
It feels the emotion that always
Thanks guys, looks like that is the problem I am wondering why I
didn't notice it. Will try to change it in the bios and see if that
helps. I'll report back
On 12/29/06, Gordon Henderson [EMAIL PROTECTED] wrote:
On Fri, 29 Dec 2006, Paul wrote:
Doug Lytle wrote:
C F wrote:
Gordon, how did
Hi all,
I have a requirement.
I would like to connect an SIP based VOIP server to my asterisk and receive
multiple calls from it and terminate it to my trunks based on my dialplan. I
know how to do that using IAX2 protocol but wondering is that is possible
using SIP.
Has anyone done a
Hi Carlos,
Im interested in knowing how we can connect 2 server using SIP. Well for me
both are not asterisk servers, 1 is asterisk and 2nd is an SIP based Server.
i need to take multiple calls from the SIP based server and terminate it
using my asterisk peers based on my dialplan. I can use SIP
Roaming is irrelevant in VOIP. You just need a fairly good wifi connection.
I don't think they mean roaming in traditional cell-phone terms. I
think they mean moving between different Access Points on a single
WiFi network. Judging by the reports in this thread, some Wifi phones
do this
HOWEVER- The Zultys WIP 2 is an INCREDIBLE WIFI B/G SIP PHONE- IT IS
EXCELLENT IN ALL RESPECTS.
Thanks for the tip! I hadn't seen these advertised before, and I've
been searching for some time for a Wifi SIP phone that can handle
multiple line appearances.
One Question: Really only 12-13
Hi Muhammad -
i have 8-port e1 controller, i am some confuse about e1 commands
that is
when and why we use cahnnel-group and pri-group e1 controller command
let me konw the above question,
i have further more questions related to this issue.
i shall be very thankfull to you
I believe you're
I agree with regards to the standby time. The Dlink DPH 540 has comparable
talk time, but 30 hours of standby time. I sometimes go for 18 hours or so
before my phone can see a charger again...
Alex
On 12/30/06, Noah Miller [EMAIL PROTECTED] wrote:
HOWEVER- The Zultys WIP 2 is an INCREDIBLE
Well, We have found that it will stay on line for around 6-7 hours with
a full charge and no talk time, and we can get about 2 hours of straight
talk time. We have given these units to our cashiers that are located
out in parking lots, and our location mangers who walk a large location
Tom
I don't mean to say that things change often or ever on some newer
models. I have some experience _cleaning up_ after messes. When
attaching to third party hardware and software configurations get
tight. When unnecessary equipment is installed because the sales
person does not know the
Not sure if anyone experienced the same - or if anyone ever connected a POTS
phone to the Phone jack on an X100P card.
The POTS phone rings normally when the FXO receives a call. The POTS phone
can also make outgoing calls when FXO is not holding the line. This is
desired. But if a call
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