.
Also, there seems to be some problems registering all 3 clients
simultaneously.
I have NAT=yes in all users and nat=no is commented out in sip.conf
Is there any other place to check?
Could anyone help, please?
Thank you very much,
Roman Zhovtulya
in that file because the
default is no.
On 9/8/05, Roman Zhovtulya [EMAIL PROTECTED] wrote:
Hello,
I'm still looking for any ideas on this problem:
I've got 3 sip clients behind the router, and they all register with
Asterisk using the same IP address.
Now, wenn all are registered, all the calls get
client.
Also, there seems to be some problems registering all 3 clients
simultaneously.
Could anyone help, please?
Thank you very much,
Roman Zhovtulya
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Asterisk-Users mailing list
Asterisk
risk Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] Several SIP clients behind router register withthe same IP,
messing up call routing, any ideas?
Do
you have NAT turned on?
On 9/7/05, Roman
Zhovtulya [EMAIL PROTECTED]
wrote:
Dear
all,Has a
http://www.freeworldialup.com/advanced/peering_numbers
But I'm not sure if they would like you to terminate a lot of minutes over
it, just check it out.
Roman
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pedro
Sent: Freitag, 10. Juni 2005
Thanks a lot to all for the input.
I have now switched to the voipjet east coast back-up server and everything
seems to be back to normal now.
Thanks,
Roman
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Freitag, 10. Juni 2005 17:58
Dear all,
I've noticed some significant voice quality deterioration when calling US
landline via VoIPjet.com in the last week or so.
Before that the quality was pretty good.
Has anyone else experienced any voice quality problems with voipjet
recently?
Thanks,
Roman
Hello,
I'm working with one institution that considers deploying Asterisk as their
internal PBX.
To overcome their initial undecisiveness, I would very much like to
present them with some examples of Asterisk deployment in similar
environments.
I'd be happy to have some info on:
Asterisk
www.voipjet.com is much cheaper, by the way (but they charge per-minute)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Kanuri, Seshu (Company IT)
Sent: Montag, 2. Mai 2005 19:24
To: Kumara Jayaweera; Asterisk Users Mailing List -
I think you should use the sip://name syntax.
I've wasted a lot of time before I figured it out myself.
Regards,
Roman
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kib Eki
Sent: Montag, 2. Mai 2005 14:57
To: Asterisk Users Mailing List -
Title: Message
As far
as I know speex is an adaptive codec, i.e. it will automatically adjust to the
conditions and provide the best quality possible.
Therefore, there should be no need to configure that
manually.
Could
anyone correct me if I'm wrong?
On the
other hand, I was thinking
Hello,
I was trying to configure 2 Digium X100P-Card compatible FXO cards.
Everything went OK up to starting Asterisk, the following error occures
while parsing zapata.conf:
Parsing '/etc/asterisk/zapata.conf': found
Mar 27 16:24:20 ERROR[1075656544]: chan_zap.c:5196 mkintf: Unable to get
What should also be possible is to register a FWD number and link it to
Asterisk.
Then ask users to use the FWD Talk
(http://www.freeworlddialup.com/content/view/full/332/)
(IE ActiveX component - softphone)
They would call the FWD number, which will then get forwarded to your
Asterisk.
Roman
This time it did.
You should normally see your own messages as well.
Make sure you post messages using THE SAME email address as you
registered on the list.
Roman
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Kris Edwards
Sent: Freitag, 25.
Hello,
I was in the same situation about half a year ago when I evaluated the
billing systems for our Asterisk setup.
Sice I was already putting the cdrs in MySQL and had all the users and
extensions there as well, my solution was to develop our own rating
engine (jsp - server-side-java- based)
- Billing Solutions with Asterisk?
Roman Zhovtulya wrote:
Hello
We was in the same situation about the same time. we take the
a version
of rate_engine (routecall APP) and modify the implementation now our
solutions is able to find a least cost route and at the same
time show
all
Hello Scott,
Thanks a lot for the info. Ill need to do a bit more testing with 3 to
5 simultaneous calls to see if there is any problem.
I dont really like the hardphones, because you dont have all the
flexibility offered by the sofphone solution (click-to-call,
paste-n-call, hands-free
Im using the mute switch on the Plantronics headset 90 with SJPhone, so
never had this issue :-)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Dana Olson
Sent: Montag, 21. März 2005 17:53
To: Asterisk Users Mailing List - Non-Commercial
Did you check SJPhone?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Scott Bussinger
Sent: Montag, 21. März 2005 22:22
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] why even use SIP
Did you
Thats really strange.
Im testing a softphone-only setup (SJPhone with Plantronics 80 Headsets
plugged into Soundcard) with around 40 users for that are linked over
LAN in an organization of around 300 people and never had any of the
problems you described (the test is going for over a month
Another solution would be the FWD web-based phone, where youd call a
FWD number, that is linked to Asterisk:
http://www.freeworlddialup.com/content/view/full/332/
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Kristof Hardy
Sent: Sonntag, 20.
Did you try enabling sip debug on Asterisk and checking what it tells
you ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tim Pushor
Sent: Montag, 14. März 2005 21:58
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Looking for a free SIP/IAX
softphone with IMandpresence support
Firefly?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Roman Zhovtulya
Sent: Domingo
softphonewith IMandpresence support
Roman Zhovtulya wrote:
Yes, I've seen it already, but it's not really as user-friendly as
sjphone. In firefly, you cannot even paste the phone number in.
Any other ideas?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hello,
I wonder if I would have to sacrifice anything if I set NAT=yes for
all sip clients I have, regardless of whether they are behind the NAT or
not.
The idea is to have the setting that works regardless of whether the
user is behind the NAT or not, since I'm not sure what connection that
Thanks!
Could that mean any security problems?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Nabeel Jafferali
Sent: Montag, 14. März 2005 19:19
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Setting
Who is being impolite???
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
César Davi Ávila do Nascimento
Sent: Montag, 14. März 2005 21:31
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Skype - Bandwidth
Hello,
Could anyone recommend something similar in functionality and
user-friendliness to SJPhone, but that would additionaly have IM and
presence support?
Thanks a lot,
Roman Zhovtulya
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Pulver.communicator (FWD) ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
FCG ZHAO Zigang
Sent: Freitag, 11. Mrz 2005 06:17
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] what is best free softphone.
I use xlite , but it
expensive for a softphone)?
Thanks,
Roman Zhovtulya
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Kevin P. Fleming
Sent: Sonntag, 13. März 2005 00:05
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users
For SIP incoming/outgoing you normally need ports 5060 and the port
range 1-2 open.
At least it works in my setup.
Could anyone correct it if it's not exactly all the truth?
Regards,
Roman
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott
I've had problems with VoipJet this morning, they stoped supporting iLBC
and GSM codecs now.
I had to enable ulaw (both in iax.conf and sip.conf) to get it back to
work.
They promissed, though, to bring the codec support back.
Also, it looks like they've lost all the credit on my account
,BackGround(timeout); Play a timeout message
exten = t,n,Hangup
Any suggestions are highly appreciated.
Thanks a lot,
Roman Zhovtulya
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What do folks have to say about www.voipjet.com?
(IAX, call termination only)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John
Novack
Sent: Montag, 7. März 2005 00:58
To: The Phone Guys; Asterisk Users Mailing List - Non-Commercial
Discussion
Could you also post your extensions.conf where to route the call
further?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Woojin Lee
Sent: Freitag, 4. März 2005 16:38
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Broadvoice + incoming
:[EMAIL PROTECTED]/557110304
Should be the same as a context name:
[31557110304]
type=friend
context=from-budgetphone
host=sip.budgetphone.nl
username=31557110304
secret=my_budgetphone_pass
qualify=yes
nat=yes
canreinvite=no
insecure=very
Hope it helps.
Regards,
Roman Zhovtulya
-Original
I wonder if you could share your configuration (sip.conf and
extensions.conf) on handling incoming calls from VoipLive, since I'm
trying to set it up also.
Thanks a lot,
Roman Zhovtulya
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Fielding
Title: Message
You've
got to check if you have all the required mysql libraries installed (mysql
client and mysql-devel)
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of asterisk
asteriskSent: Donnerstag, 3. März 2005 10:13To:
if the comments I made to the code below are correct.
Thank you very much,
Roman Zhovtulya
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. Can it pose a problem?
Thank you very much,
Roman Zhovtulya
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Goryachev
Sent: Donnerstag, 3. März 2005 15:05
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users
Where did you get it?
I was looking on the internet and couldn't find any link to install this
Mozilla extension.
Is it also possible to install it on Firefox?
Thanks,
Roman Zhovtulya
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
administrator
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