search for: ferrency

...r.c:237 #17 0x0805a90c in main (argc=5, argv=0xbfbfec14) at deliver.c:1093 Does anyone have any clues on how to debug this further? I wonder if the original bug reporter removed both the mailbox and the index files at the same time; the current explanation is less than satisfactory. Thanks, Alan Ferrency pair Networks, Inc.

...ers, but we have a proliferation of pop3-login processes. Currently we have 128 such processes. We have 11 imap-login processes, but only a few actual imap processes running. Is this normal? Can we stop it? Is anyone else successfully running dovecot on FreeBSD 6.2? Thanks for your help. Alan Ferrency pair Networks, Inc. alan at pair.com

...home/test/mailbox does not contain/usr/home/test/mailbox_folders, and this condition should fail. If some other code were to enforce that /usr/home/test/mailbox be terminated with a directory separator, then this is not a problem and my patch should work. I hope this makes sense. Thanks, Alan Ferrency pair Networks, Inc. alan at pair.com

I posted this 1.2.0-beta1 success story to asterisk-dev, and someone recommended that asterisk-users might benefit from it as well. Thanks, Alan Ferrency pair Networks, Inc. alan@pair.com ---------- Forwarded message ---------- Date: Thu, 22 Sep 2005 17:35:08 -0400 (EDT) Subject: [Asterisk-Dev] Re: Ring requested on channel already in use To: asterisk-dev@lists.digium.com > alan wrote: > > A problem was recently posted on the Asterisk-Us...

...l authentication script, but I want to explore more direct configuration options, first. Does anyone have any recommendations? If what I want can't already be done, and I patched dovecot to do one of these things, how likely would it be to have that patch merged into the project? Thanks, Alan Ferrency pair Networks, Inc. alan at pair.com

...master user to log into a username that is in the passdb with a "user=" field, dovecot looks in the userdb for the original username, and not for the "user=" username specified in the passdb passwd-file. Is this a known bug? Maybe I'm doing something wrong? Thanks! Alan Ferrency pair Networks, Inc. alan at pair.com

...REFER. Now, the phone has dropped out of the loop, and Asterisk has connected the two call legs into a loop, as far as I can tell. I tried a "soft hangup" on each of the channels, but nothing happened. Is there any way to recover from this, short of an asterisk restart? Thanks, Alan Ferrency pair Networks, Inc. alan@pair.com

...egister, but to use the outbound proxy for outgoing calls, and produce a dial tone anyway. - tell Asterisk to enforce an IP address restriction for a specific SIP channel, but still accept an incoming registration from that IP. Vitals: - Asterisk 1.0.7 - snom 190 firmware 3.60i Thanks, Alan Ferrency pair Networks, Inc. alan@pair.com

...dio effects such as a periodic "bzzt" sound in the handset. Does anyone have any familiarity with "decode latency," specifically with Sipura devices? Why would it take 200+ms to decode a 20ms RTP packet? G.711u has existed for over 30 years, how hard could it be? Thanks, Alan Ferrency pair Networks, Inc. alan@pair.com

...lbacklogin app, then I will use it from today on. Othervise it is a reinventing of the wheel. Hope there will be a alternate application in newer versions of asterisk. Thanks Martin ----- Original Message ----- From: "Kevin P. Fleming" <kpfleming at digium.com> To: "Alan Ferrency" <alan at pair.com> Cc: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Sent: Wednesday, April 11, 2007 11:45 PM Subject: Re: [asterisk-users] AgentCallBackLogin vs AddQueueMember Alan Ferrency wrote: > However, this is...

...ion deployment, I experienced the same problems with analog phones conected to all four FXS ports. Does this sound familiar to anyone? Where else should I look for signs of the Zap failures? What extra logging should I put in place to make diagnosis easier in case it happens again? Thanks, Alan Ferrency pair Networks, Inc. alan@pair.com

...nal server error is received. I have verified that this is only a problem for PRI->SIP calls. SIP->SIP calls do not generate DTMF INFO packets prior to the call being answered. A complete SIP debug log (with some console verbosity as well) follows. Thank you very much for your help. Alan Ferrency pair Networks, Inc. alan@pair.com ------------------- SIP DEBUG LOG --------------------- -- Executing NoOp("Zap/3-1", "Dialing staff extension sip/sip_alan by XXXXXXXXXX") in new stack -- Executing Dial("Zap/3-1", "sip/sip_alan|20|o") in new stack...

...igital, and the loopback path to call our own internal milliwatt source will almost definitely be all digital, so where would the loss come from? - Assuming gains are tuned correctly, or don't matter for the PRI, is there any other hope I have for solving these echo issues? Thanks, Alan Ferrency pairNIC pair Networks, Inc. alan@pair.com

I finally added beginnings of unit tests to Dovecot v1.1 code tree: http://hg.dovecot.org/dovecot/rev/4e4a5d6bb2cb There are currently just a couple of tests implemented in test-lib (lib/ functions) and test-mail (lib-mail/ functions). I'm probably going to be pretty lazy at adding more of those tests, so if you've ever wanted to get your own code to Dovecot, now would be a great time to

...en "no." Theirs is a proprietary system which from what I've read, seems to use SMS in combination with IMAP. Wacky. (If you aren't talking about iPhone compatibility, I'm less knowledgeable, but dovecot does support IDLE-style "imap push" style notification.) Alan Ferrency

...s to have an agent who is a member of two queues, hear a different ring depending on what queue the call is coming from. (Yes, I'm aware of the queue announcement, and I intend to use it. But this is another requested feature which I'd like to implement if it's feasible.) Thanks, Alan Ferrency pair Networks, Inc. alan@pair.com

...quot;goto context" would be more useful I'd expect. I know this is possible by having the internal extension simply transfer the call to a different extension; however, this is not a suitable solution in this case. I need a "one button press" (dtmf tone) solution. Thanks, Alan Ferrency pair Networks, Inc. alan@pair.com

Anyone have any ideas why DTMF would not work on only one number? Looking through the logs, anytime a button is pressed, this is what shows up: 2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Exception on 9, channel 1 2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Got event Dial Complete(9) on channel 1 (index 0) 2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Echo cancellation already on We

Hi there, I have bought 70 sipura 841 phones for a customer of mine. When following the mass provisioning guide in the admin manual for the sipura, I see it download the spa841.cfg file from my tftp server Sometimes the phone also downloads is phone specific file via tftp, and it works okay then. But, after a reboot of the phone, it is very very likely that it won't startup

...t;voicemail prompt" problem, I see: Jan 9 09:33:20 WARNING[7627] app_voicemail.c: Couldn't read username This makes sense, but isn't very helpful. I'd appreciate it if anyone could shed any light on this situation, though I admit I don't have very high hopes. Thanks, Alan Ferrency pair Networks, Inc. alan@pair.com