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Thread: RE: pbxes going crazy
smi

Replies: 2
Views: 7720

pbxes going crazy 13.07.2012 23:12 Forum: Bugs

phones disconnected, submit and start does not help

Lots of these in logs:

Jul 13 16:42:13 NOTICE[58600] chan_sip.c: -- Registration for '*removed*@callcentric.com' timed out, trying again (Attempt #4)

lots of these in logs are not my users

Jul 13 16:38:17 VERBOSE[58600] chan_sip.c: SIP REGISTER attempt failed for (null) : Bad digest user
Jul 13 16:38:17 NOTICE[58600] chan_sip.c: Registration from '"*removed*-200" <sip:*removed*@pbxes.org>' failed - Username/auth name mismatch

Thread: RE: Submit and Start
smi

Replies: 5
Views: 10486

RE: Submit and Start 28.09.2011 15:12 Forum: Bugs

Phones do not work is the first clue. We are found no log entries as described above. I wonder if it has something to do between Seattle and New York servers?

Thread: RE: Submit and Start
smi

Replies: 5
Views: 10486

traurig Submit and Start 28.08.2011 05:12 Forum: Bugs

Twice recently our phones and voicemail have stopped working and I had to go under personal data and hit Submit and Start.

Most recently there was nothing under System Log since Aug 22, but there was stuff under Call Monitor up until it stopped working then everything was duration 00:00:00. After Submit and Start there are recent log entries again.

Is there something we are doing wrong or need to do better?

Thank you.

Thread: RE: Google Voice Peering
smi

Replies: 206
Views: 1018196

RE: Google Voice Peering 18.01.2011 22:19 Forum: News

same experience here, inbound over google talk the recording sounds fine, but ring tones when it hits the ring group are just static

Thread: Dscp 46 (ef)
smi

Replies: 0
Views: 7099

Dscp 46 (ef) 16.01.2011 22:26 Forum: Feature Requests

My wireless ISP rate limits us to 400k but prioritizes DSCP 46 (EF) tagged packets so they get through. We just upgraded to SOHO but while on the free account I did some wireshark and saw my phones were setting DSCP EF, but pbxes.com was not. Is that something that could be enabled? Thanks!

http://www.voip-info.org/wiki/view/DiffServ
# Asterisk 1.0.9 (ie sip.conf) - EF DiffServ hexadecimal: tos = 0xb8

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