Displaying 6 results from an estimated 6 matches for "supura".
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sipura
2006 Jun 23
1
Can I get caller id passed to a phone connected to a Supura 2100?
I have a Uniden wireless phone connected into Linksys/Supura 2100. It
works well, except I never see any caller ID information displayed on
the phone. Is that a setting in the 2100 that I'm missing, or is it an
Asterisk setting or isn't it possible?
Thanks,
Jim.
2006 Nov 22
1
Sipura phone does not ring
Problem: SPA3000 phone does not ring for incoming PSTN call although I
can dial out.
I set up my Sipura with the Voxilla Wizard which is pretty good but
leaves out some important details.
The Voxilla Wizard for Supura SPA3000 gave me a setting for PSTN Tab ->
Dial Plans ->
Dial Plan 8 (<S0:66610>)
Should I put extension [66610] in sip.conf with a context in
extensions.conf that will contain dialing instructions?
Can someone please tell me what the entries under [66610] and the
associated context...
2004 May 19
1
voicemail notify problem on sip extension
...: Bruce Komito
To: <asterisk-users@lists.digium.com>
Subject: [Asterisk-Users] voicemail notify problem on sip extension
Sent: 5/19/2004 4:27:51 PM
I'm having a problem with the voicemail notify feature. Although I have
the voicemail box configured for the sip extension, the sip device
(Supura) does not produce a stutter dialtone to indiate voicemail waiting.
sip.conf and voicemail.conf follow below.
Has anyone seen this problem? I am running CVS-HEAD-05/06/04-17:52:43.
TIA
sip.conf:
[7752365815]
type=friend
username=xxxx5815
secret=xxxx
host=dynamic
mailbox=7752365815
context=vpbx-...
2006 Jun 26
1
registering a Motorola vt1005
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2006 Apr 20
1
MeetMe: lots of buffer overruns/underruns when connecting over IAX
...size: 320
Apr 20 17:37:28 NOTICE[20970]: app_meetme.c:1276 conf_run: Audio
bytes: 320 Buffer size: 640
Apr 20 17:37:28 NOTICE[20970]: app_meetme.c:1276 conf_run: Audio
bytes: 640 Buffer size: 320
This always happens, even if there's only one participant in the
conference.
All phones are Supura/Linksys SPA-941 phones. Everything is working
fine (users can talk to each other, voicemail is working, etc), exept
for meetme.
In meetme.conf I've got audiobuffers=32, which doesn't help.
Any clue?
--
Erik Hensema (erik@hensema.net)
2005 Mar 27
6
Sipura 2000 x dual g729 channels x other choices?
I found a thread [1] last month about the poor/crappy g729 quality on
Sipura units. Anyone noticed an improvement or the quality is still poor?
If the Sipura firmware/g729 offers no quality yet, who else is offering
a dual channel g729 ATA? I heard about Uniden, but I have no "reports"
about their ATA...
[1] Sipura g729 call quality to PSTN