search for: supura

Displaying 6 results from an estimated 6 matches for "supura".

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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