similar to: Polycom phones-buggy SIP firmware or am I missingsomething in the XML configs?

> >> Jason Brown wrote: > >> | Anyone have experiece with polycom phones? > >> | > >> | I am experiencing a really weird problem. In an office > where I have > >> | the following extensions: > >> | On the Polycom phones, when I want to dial from extension > >> 100 to any > >> | extension 120 or above, or dial out, it

Anyone have experiece with polycom phones? I am experiencing a really weird problem. In an office where I have the following extensions: 100 101 102 103 104 110 111 120 130 140 141 150 200 On the Polycom phones, when I want to dial from extension 100 to any extension 120 or above, or dial out, it dials just fine. If I want to dial from extension 100 to extension 101,or 102 or 103 or 104,

Hello all, For some years now, I've been using Thunderbird for dovecot. I am not very satisfied with t/b so I thought of using m/s outlook but then I thought that I want to distance my clients from office products. I have a newly created dovecot installation on a very small site. Three nodes, all x86 Windows 7 professional with an ubuntu v14.04 server (x86 again) running dovecot 1.2.17.

Op 16-7-2016 om 18:12 schreef Charles Marcus: > > On July 16, 2016 4:02:33 AM EDT, Spyros Tsiolis <stsiol at yahoo.co.uk> wrote: >> Since I have quite some experiece with thunderbird, I know most of >> its shortcomings > Care to elaborate? Thunderbird is far from perfect, but is by far the best IMAP client available. > > Most times you can work around supposed

I have two T1s and I'd like to split my SIP traffic over the two. I am looking at this: http://lartc.org/howto/lartc.rpdb.multiple-links.html what bothers me about it is the note "Note that balancing will not be perfect, as it is route based, and routes are cached. This means that routes to often-used sites will always be over the same provider.". If all my traffic goes to the same

Does render_partial work within a RXML file? I am attempting the following: xml.users do for user in @users render_partial ''user'', user end end All that is output is: <user> </user> Without any exception or anything to indicate that I''m doing something wrong. Does anyone have any experiece with this? Josh

> > No, this is not the case currently with any of the Cisco SIP software > loads that I am aware of. If you find this to be incorrect, please > let the list know. Cisco has not deployed much of the featureset in > their SCCP phones (such as paging/intercom) into the SIP phones due > to lack of standards/interest/political capital. > > JT Ok, after further

Hi all, I got a couple of Snom 190's through this week and after some initial foolishness I have them both setup no problems. But here comes the except. When there is voicemail waiting the softbutton appears but the phone always dials its own extension. No matter what I put into the "mailbox" parameter on the line settings. (Phone registers correctly with * and all standard

I have these phones working well with Asterisk, thanks to http://www.freedomphones.net/polycom/files The code there is not the most current (and my Polycom Extranet login hasn't gotten updated yet to allow downloads). Been waiting a while for their support folks to update the access. If you know where I can get the 2.5.0 BootROM and 1.3.1 SIP code? Thanks, L.

Hi, I've ordered a few IP501s from PC Connection, basically since we have an account with them. I like the phones for what they do, and now would like establish a relationship with a reseller that can give us maintenance and access to the most current firmware. What are some good resellers out there? Regards, --- Gavin Adams VP of Technology Promisant (USA) Inc. Email: me@gavinadams.org

Stay away from Alliance Systems. We ordered $15k worth of Polycom's over a month ago and we're still waiting. Our account rep's communication with us on what the delay has been, has been terrible. Doug. -----Original Message----- From: Gavin Adams [mailto:me@gavinadams.org] Sent: Friday, January 27, 2006 8:26 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

Hello puppeters; I want to set/test the PuppetShow "Puppet''s Web Interface" with Ruby on Rails (ROR). If somebody has setup such show, please share your experiece with EXAMPLES...I''m not the programmer.so please share your STEPS....I have setup the ROR environment, currently learning Rails with different frameworks....HOBO (http://hobocentral.net/) is great for starter

Does anyone know if the 7912G (which the wiki says can do either sccp or sip) uses the 7940/60 sip firmware? I ask this because the only firmware I can seem to find on TAC for the 7912G is sccp, no sip...if it takes it's own firmware and doesn't use 7940/60 firmware, can someone point me to the right location for it? Thanks, Marty Mastera M3 Resources marty@m3resources.com Phone:

Has anyone been able to get the minibrowser on the Polycom SoundPoint IP 500/600 phones working? If so could you share the relevant sections of your config with me?

Folks, I am in the process of fighting my way through my first foray into ADSI phones. I have a very nice Aastra 480 from Sayson, purchased through Cylogistics. Very nice phone. However, my experience in configuring ADSI has been pure pain so far, and it's not all working yet... 1) The locking mechanism for these ADSI phones is... baroque? Arcane? Perhaps there is a reason why you would

I had the same problem with snom 190 phones. Using the transfer with # instead of "Transfer Button on the phone" worked for me. In my configuration "REFER" was not send, so the transfer with the button on the phone did not work. Guido Hecken -----Urspr?ngliche Nachricht----- Von: steve szmidt [mailto:steve@szmidt.org] Gesendet: Sonntag, 26. Dezember 2004 17:14 An:

Did you try to boot without lan just the power ... I've had this same problem to and rebooted the device without lan connection -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Louis-David Mitterrand Sent: woensdag 26 januari 2005 11:36 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

> I ran into this problem with an early batch of IP650s. Polycom's firmware > version 2.0.3b made this issue go away. Speaking of Polycom firmware, anyone have an up to date source for the stuff? The site I ordered from took down their FTP site that had it. :( -Kenneth

Hi, Trying to use Ajax.Updater to update a screen element. Works well, except in one place where I want to use a 302 Redirect page. This works absolutely fine in the Browser, and takes me to the page I want to see. However, when calling using Ajax Updater, I get a message like this: This document you requested has moved temporarily. It''s now at http://.... The on302 event fires,

Hello, I got 3 Polycom 300 phones, and upgraded to the latest firmware provided by the reseller. This is my first experience with Polycom and I cannot make them register in my Asterisk Box. I followed every advice I found (including separating [user] and [peer] in sip.conf. Using ethereal, I found that it tries to SUBSCRIBE to the asterisk box and it receives a 403 FORBIDDEN message. I