look for 'mytrunk' as thats the trunk its dialing
On Wed, Mar 18, 2020 at 02:41:51PM -0300, Joshua C. Colp wrote:
> On Wed, Mar 18, 2020 at 2:37 PM John Roman wrote:
>
> > ive enabled logging. aside from a realm error i see on my endpoint, im
> > still not s
ax-Forwards: 70
From: "demo-alice"
;tag=3166828162
To:
;tag=f1b212ab-9b55-4d13-9055-f49ce55f214e
Via: SIP/2.0/TCP
[2605:e000:130a:fb:de1:71fc:e257:6f4e]:60629;branch=z9hG4bKd67297ab9853f9168c
greetings asterisk users :)
ive just deployed version 17 and migrated as best I can to pjsip. I can
receive calls, and get to my mailbox prompt, however placing calls seems
impossible with the following error on dial:
Connected to Asterisk GIT-master-0cde95ec89 currently running on dunkel (pid =
Are you sure the packets are not lost and reach their destination?
Do the restored configs have the necessary permissions/ownership?
Roman
> 4 авг. 2017 г., в 22:27, Jerry Geis написал(а):
>
> Audio packets are running...
>
> 961 16.150421076 192.168.5.150 -> 192.168.5.25
Hello, Jerry,
What does it mean exactly «…just no audio»?
Maybe the audio packets don’t come through.
Have you tried Wireshark?
All the best,
Roman
> 4 авг. 2017 г., в 22:04, Jerry Geis написал(а):
>
> Hi all,
>
> I had a box with CentOS 6... I backed up, installed C7. rest
serach the option en sip.conf:
externip = you public ip
localnet=tus direcciones locales (address local)
saludos
Roman
On Tue, Dec 22, 2009 at 4:26 AM, zehra yildiz wrote:
> Hello All,
>
> I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The
> softphone can ca
the
same for all destinations, you can skip using AGI. And if you need to
make the calls just once you might want to take a look at asterisk
call files instead of AMI, probably they will suit your needs,
depending on how much control do you need over
SIP conntrack module or
something like that). The 1.50 firmware version solved the problem and also
gave the impression of working faster overall than 1.20.
Thanks a lot.
Roman.
smime.p7s
Description: S/MIME cryptographic signature
___
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one paid
account between several users). So a SIP server + possibility for dialplans
through 3rd party SIP servers. Maybe something like SER would suffice? Or SER
as a proxy in front of Asterisk is the way to go?
--
TIA
Roman.
smime.p7s
Description: S/MIME crypt
nux) software clients are known to have maximum compatibility
with Asterisk?
--
TIA
Roman.
smime.p7s
Description: S/MIME cryptographic signature
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t of your Dial
> Command.
Doesn’t help, alas. Also, it works the same (disconnect after 20 seconds) both
for Dial and Echo, regardless of presence of Answer.
--
TIA
Roman.
smime.p7s
Description: S/MIME cryptographic signature
___
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IP clients does not help.
Maybe there’s something wrong with the headers of the packet that makes the
client think the packet is misaddressed? Twinkle says, “you have the
following registrations ” while I’d expect
. So how do I make sure the client sends its ACK?
--
TIA
Roman.
smime.p7s
Desc
cified slot.
2. I am using 2.6.9-42.0.3.ELsmp kernel
Roman Marchevsky
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: Sunday, December 10, 2006 3:24 PM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] Kernel crash
On Thursday 30 November 2006 21:49, Tzafrir Cohen wrote::
> On Thu, Nov 30, 2006 at 07:19:14PM +0200, Roman Yeryomin wrote:
> > Hello!
> >
> > I have problems compiling zaptel (tried 1.2.11, 1.2.10 and 1.4.0-beta2 --
> > all give the same error) with 2.6.19 kernel
&g
Hello!
I have problems compiling zaptel (tried 1.2.11, 1.2.10 and 1.4.0-beta2 -- all
give the same error) with 2.6.19 kernel
CC [M] /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.o
In file included
from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xpd.h:26,
from
Hello all!
Did anyone manage to run spandsp with t.38 support on asterisk 1.2.7.1 ?
Roman
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On Saturday 20 May 2006 16:31, Roman Yeryomin wrote::
> Hello all!
>
> I have a problem with ringing indication when calling from h323 (oh323+open
> phone client) to sip users. The phone rings and users can talk to each
> other with no problems but the calling h323 user hear sil
from sip to that open phone client also
no problems.
I tried latest ooh323 and oh323... no difference
Also passing "r" option to dial doesn't help.
Does anyone know where could be the problem?
Roman
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On Wednesday 03 May 2006 13:19, ChaosMedia > WebDev wrote::
> In fact i was expecting asterisk to have some kind of configuration that
> would enable mysql but i can't seem to find anything like that, like it
hope this will help
http://www.voip-info.org/wiki/view/Asterisk+RealTime
Hello!
Can anybody tell me how asterisk handles mysql connection failures? f.e. mysql
database is on another maschine and there was a network failure, does it
buffer something somewhere so it will be able to write cdrs later when mysql
is up?
Roman
erent numbers of digits.
Any questions, let me know.
Roman
Carlos Chavez wrote:
I am having a problem with some Polycom 601 phones. If I dial without
picking up the handset or selecting the speaker I can dial numbers that are
any lenght. But if I pick up the handset or are using the s
XT-PIC ide1
NMI: 0
LOC:3115228
ERR: 0
MIS: 0
Any other information you need to help me figure this out, please let me
know.
- Roman
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29164?site=sr:SEARCH:MAIN_RSLT_PG
--
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
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good things about them:
http://www.vegastream.com/vega400.asp
--
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
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ar/log/asterisk/full"
Any suggestions?
Thanks!
-Ken
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Post your extensi
ds since Jan 1, 1970. May be negative.
defaults to now.
timezone: timezone, see /usr/share/zoneinfo for a list.
defaults to machine default.
format: a format the time is to be said in. See voicemail.conf.
defaults to "ABdY 'digits/at
Krystian Filiks wrote:
What about plain g729?
My main concern is the Hardware, anyone that can tell me if this
Supermicro 6014H-32 is stable and sutible for asterisk?
Supermicro Superservers are traditionally extremely stable and reliable.
--
Roman Volf
Keystreams Internet Solutions
[EMAIL
On Tuesday 15 November 2005 09:30, Dmitry Ivanov wrote:
> On Tuesday 15 November 2005 06:26, [EMAIL PROTECTED] wrote:
> > Hi. I'm setting up an Asterisk hobby box for me to play around with.
> > Is it possible to use a regular 56k modem and a regular home phone
> > for it?
>
> Yes, but forget G.711
On Tuesday 01 November 2005 18:11, Anton Krall wrote:
> Unicall is Steves Underwood E1 R2MFC support using spandsp.
Well, don't know about that...
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Nothing special... yes I patched them manually -- current spandsp patch is
broken a little.
On Tuesday 01 November 2005 18:11, Anton Krall wrote:
> Any special considerations? How did you patch the files? Manually?
> |
> |On Tuesday 01 November 2005 12:19, Anton Krall wrote:
> |> Anybody already
On Tuesday 01 November 2005 15:45, Robert Webb wrote:
> Hi all..
>
> I just setup a test box with Debian running kernel 2.6.
> Went to CVS and did a checkout of the new beta 2 release
> using the command: cvs checkout -r v1-2-0-beta2 zaptel
> libpri asterisk asterisk-addons asterisk-sounds.
>
> I
On Tuesday 01 November 2005 12:19, Anton Krall wrote:
> Anybody already tried compiling spandsp with the new 1.2beta2?
> How about unicall?
I did, and it works fine.
What is unicall?
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t to ftp server.
http://ftp.digium.com/pub/asterisk/asterisk-1.2.0-beta2.tar.gz
gives NOT FOUND
Regards,
Roman
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On Friday 07 October 2005 17:48, Michael Stahl wrote:
> Roman:
>
> I created two bash scripts called Mail2Fax and Fax2Mail for use with the
> asterisk sever.
>
> They leverage the app_txfax and app_rxfax scripts, along with ast_fax.
> They make using these apps a lot easier,
On Friday 07 October 2005 13:52, Bohuslav Coufal wrote:
> Hi all,
>
> does anybody have $subj apps.
>
> Thanks,
>
> Bob.
you can download them from spandsp website
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A
On Thursday 06 October 2005 09:27, Roman wrote:
> On Wednesday 05 October 2005 18:58, Technical Support wrote:
> > I found the problem! Installing spandsp .3 created a symlink that was
> > not removed. Installing spandsp .2 did not replace the link. That cause
> > the wro
On Wednesday 05 October 2005 18:58, Technical Support wrote:
> I found the problem! Installing spandsp .3 created a symlink that was not
> removed. Installing spandsp .2 did not replace the link. That cause the
> wrong library linking in
can you tell where it is?
Thanks!
___
On Wednesday 05 October 2005 12:31, Roman wrote:
> Anyone run app_txfax and app_rxfax successfully on asterisk 1.2 beta1?
> I get an error when trying to run asterisk:
>
> [app_txfax.so]Oct 5 12:05:24 WARNING[14665]: loader.c:314
> __load_resource: /usr/lib/asterisk/modul
Anyone run app_txfax and app_rxfax successfully on asterisk 1.2 beta1?
I get an error when trying to run asterisk:
[app_txfax.so]Oct 5 12:05:24 WARNING[14665]: loader.c:314
__load_resource: /usr/lib/asterisk/modules/app_txfax.so: undefined symbol:
fax_set_header_info
Oct 5 12:05:24 WARNING[146
cative of?
I'd really appreciate it if we could figure it out.
Thanks a lot,
Roman
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Flobi
Sent: Donnerstag, 8. September 2005 20:58
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
ect client.
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 m
Title: Message
Yes,
NAT=yes in all users and "nat=no" is commented
out in sip.conf
Is there any other
place to check?
Thanks,
Roman
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
FlobiSent: Mittwoch, 7. September 2005 22:48To:
rrect 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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Ast
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.
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: Freit
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,
e and
particulary in Germany.
Any ideas, links, contacts?
Thank you very much,
Roman
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htt
loads".
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Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
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Sounds like you are missing the mysql client libraries.
--
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED
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Please stop double posting your questions. This will not help you get
any answers.
--
Roman
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: Asteris
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 -
> Non-
Have you looked here:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SetCallerID
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
Jason Brown wrote:
Here is something I wasn’t quite expecting from a business deployment,
and don’t have an answer for. Maybe one of you do
Have you tried putting both access points on the same channel?
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
Jim Meehan wrote:
I've got a Hitachi WIP-5000 phone. Seems to work well with my Asterisk setup,
except for a few annoyances:
1) If the phone has been sitting unused
Oh and it was just a test to see how it worked. Pretty easy to setup
Asterisk-users
On Apr 8, 2005 8:47 PM, Roman Volf <[EMAIL PROTECTED]> wrote:
> I setup this google group because Google seemed to be good at
> threading the topics from the list. I have noticed that many threads
&
I setup this google group because Google seemed to be good at
threading the topics from the list. I have noticed that many threads
don't go as well as planned and wind up in the wrong place.
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fi
This should work fine.
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
Matt Riddell wrote:
Peter Bowyer wrote:
exten 456,1,Background(Please-set-time-mmddhhmm)
exten _.,1,System (date ${EXTEN})
If I dial 456 I get the message, so I type 04021305 (2nd April, 13:05).
On the cons
y
help will be appreciated.
Thanks a lot,
Roman
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dominic
LuSent: Montag, 28. März 2005 19:43To:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] How to
config speex?
Hello,
I&#
Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
2 channels configured.
Any ideas would be highly appreciated.
Roman
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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
>
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
>
>
>
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
m now using a month-old CVS version).
Roman
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Scott Bussinger
> Sent: Montag, 21. März 2005 22:32
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
&g
ubject: Re: [Asterisk-Users] VOIP - 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
you are interested in taking a look at our solution, just mail me at
[EMAIL PROTECTED] and Ill get you a test account.
Roman
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> Sent: Dienstag, 22. März 2005 17:20
> To: Asteris
?
Regards,
Roman
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Scott Bussinger
> Sent: Montag, 21. März 2005 20:19
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [A
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,
Or if google is too complex, http://asterisk.keystreams.com
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
Robert Webb wrote:
On Tue, 15 Mar 2005 11:56:18 -0500
"Fabian Borot" <[EMAIL PROTECTED]> wrote:
Hello all
I have been learning * from almost 1 month now.
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:
Because SIP works with things other than Asterisk. IAX does not.
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
Joseph wrote:
I'm just curious why Sipura isn't using free IAX protocol with their
devices instead of SIP?
With IAX NAT traversal would have been easier, so wh
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 - Band
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] Set
connection that
particular user has to Internet at the moment.
I've tried and NAT=yes works even for those clients that are not behind
the NAT.
Is there any peformance problems/etc if I set NAT=yes for all clients?
Thanks,
Roman
> -Original Message-
> From: [EMAIL PROTECTED]
er way/any other client?
Thanks a lot,
Roman
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Kavit Munshi
> Sent: Montag, 14. März 2005 15:06
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asteris
Montag, 14. März 2005 02:21
> 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 PROTECTE
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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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
> S
Pulver.communicator (FWD) ?
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> FCG ZHAO Zigang
> Sent: Freitag, 11. MÃrz 2005 06:17
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] what is best free softphone.
>
>
>
> I use xli
my account (there was
only 4 dollars there) after their software update, so I couldnt phone
because I had no funds on the account. They said they'll check it.
So, you'd better check if you still have enough funds on your account.
Now I can phone without problems.
Hope it helps.
Roma
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
; Wait for an extension to be dialed.
exten => t,1,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
Subject
It would be helpful if you pasted the relevant sections of sip.conf and
extensions.conf
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Im new to Astererisk. I compiled the latest CVS and setup the server. It
looks like things are working. I'm running k
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
Fie
ber (557110304) in
register => 31557110304:[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 h
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 cal
e of the extensions based on the last 3 numbers a
caller entered?
By the way, I'm using RealTime (mysql) for sip (just users) and
extensions. Can it pose a problem?
Thank you very much,
Roman Zhovtulya
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
dn't it
read the numbers punched on the phone? The Voicemail works very well.
I use dtmfmode = rfc2833 and iLBC codec.
Also, please check if the comments I made to the code below are correct.
Thank you very much,
Roman Zhovtulya
___
Asterisk-Users ma
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:
aster
eb of 2005. It will *not* be updated in real time (at
least not for now)
Please direct flames/questions/comments to [EMAIL PROTECTED]
--
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
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Asterisk-Users@lists.digiu
In case you didn't get the last 5 responses, you just need to create an
alias for the two email accounts.
But honestly people, do you not read the rest of the thread before
responding? Its already been answered.
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
C F wrote:
yes, c
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
administ
Have you seen the user guide?
http://www.sipura.com/Documents/SPA841UserGuide.pdf
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
Scott Bussinger wrote:
There isn't an Admin Guide for the SPA-841 as far as I know.
However, I have found that the Admin Guides for their other
product
de = rfc2833 and iLBC codec.
Also, please check if the comments I made to the code below are correct.
Thank you very much,
Roman Zhovtulya
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Problem solved. It was NAT. h323 not work behind NATD
-Original Message-
From: Roman Bessyadovskii
Sent: 10 ÓÅÎÔÑÂÒÑ 2004 Ç. 12:59
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Netmeeting i can't hear voice
Hi.
After a small war with "underfined sybol" error and co
Hi, another problem.
I configure TAPI driver for outlook.
(https://sourceforge.net/projects/asttapi/)
Yesterday all work fine.
I configure that "Local Phone" is Zap/g1/772323 and external call is going
to context default.
When I call to sip - all work ok.
When I call to city (via Zap) local phone
Hi.
I see that message in console.
-- Zap/1-1 is ringing
!! Unknown IE 40 (cs6, Unknown Information Element)
-- Zap/1-1 answered SIP/1016-e34b
As see in older messages it some Information send by phone station to my via
PRI.
But what does IE 40 mean? I cann't find information element 40
Hi.
After a small war with "underfined sybol" error and conflicts between h323
and oh323 I successfully install h323 channel.
Now, I can connect from Netmeeting to SIP and ZAP channels, but I can't here
anything.
When I call at phone, and try to speak, on another end of line man said,
that my voi
Thanks for help.
All works now.
Problem was in codecs on different sides
Definity: display ds1 1b14 CRC? n
Interface Companding: mulaw
And when making call via asterisk
Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Speech (0)
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