Displaying 9 results from an estimated 9 matches for "suleymanov".
2006 Mar 27
3
sipura spa2 + asterisk bug ?
...sole
gives me this short error:
Mar 27 07:01:43 NOTICE[29656]: chan_sip.c:3629 process_sdp: No
compatible codecs!
My sipura adapter is using g729a codec.
When using both of sipura lines separately everything works fine,
until someone tries to use both lines simultaneously.
Any advice ?
Tofik Suleymanov
2006 Apr 05
6
transforming g729 files to wav files
Hello list,
is there any open-source software that recodes g729 sound files to wav
sound files ?
The only way (at least) to do such transformation is with interactive
form on: http://www.asteriskguru.com/audio_conversion.php
Tofik Suleymanov
2006 May 07
1
another question about hardware for using with asterisk
Hello folks,
firstly, thank you for your useful and fast answers !
Is there anybody using D-Link SIP phones ?
Are D-Link SIP phones ok to install in production environment ?
Give your comments please.
Tofik Suleymanov
2006 Apr 15
2
asterisk voicemail question
Hello list,
When new voicemail comes and i pick up the phone i hear special tones
indicating that the new voicemail arrived.I've never had any problems
with this feature, but several days ago it begin to behave strangely:
1. new voimcemail arrives, but i dont hear the special indicating tones
when picking up the phone
2. there is no new voicemail (checked mailbox on filesystem), but when i
2006 May 07
0
[Fwd: Re: asterisk hardware]
Tofik Suleymanov wrote:
> Steve Totaro wrote:
>
>> Give idefisk a try. It works very well for me, its free, and does
>> not crash all the time like Cubix (formerly Firefly).
>>
>>
>
> Hello Steve,
>
> As far as i know 'idefisk' is a softphone, but i need a hardwar...
2006 Mar 28
4
ISDN and Analog DIAL UP Connection Through Asterisk and Digium TE405P
Did anybody know,
Is it possible to establish a ISDN DIAL up Connection and Analog Dial up
Connection (V90) trough asterisk with Digium TE405?
Thanks a lot for help.
Nico Giefing
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2004 Sep 30
7
Asterisk hardware
Hi to all,
I already setup asterisk on REDhat 9.0 linux machine.
I will have 4 physical phone lines and 10 IP phones for it to use. I have a
network setup already.
Is getting TDM400P - 4port FXO from digium enough to start? Do I need
anything else?
Thank you
2005 May 16
11
H323 to SIP
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2006 Mar 13
0
Anyone seen this scsi error before ?
Graham Bentley wrote:
> Recently swapped out my Sony for a HP DAT and
> got this first time in dmesg
>
> (sa0:ahc0:0:6:0): PREVENT ALLOW MEDIUM REMOVAL. CDB: 1e 0 0 0 0 0
> (sa0:ahc0:0:6:0): CAM Status: SCSI Status Error
> (sa0:ahc0:0:6:0): SCSI Status: Check Condition
> (sa0:ahc0:0:6:0): UNIT ATTENTION asc:28,0
> (sa0:ahc0:0:6:0): Not ready to ready change, medium may