similar to: Which IP Phone is really the best?

Hello to you all. Just got my first iP0020 phone and no matter what I do to it when I try to call I get a busy signal even though Asterisk and the phone web gui shows that the phone is "registered". Has any body had any similar experience with this type of phone (or nay phone)? William Herrera LAN/WAN Technical Consultant LAN Solutions <http://www.lan-solutions.net/>

Hello all: For several years I have been running the following in a Linux server. Dovecot Version: 2.0.9 *IMAP:* Connection Security: SSL/TLS Port: 993 Authentication Method: Normal Password *SMTP:* Connection Security: STARTTLS Port: 587 Authentication Method: Normal Password The E-mail client is Thunderbird on Windows. I am preparing a new server, with Dovecot 2.2.36 and would like to know

I?m trying to migrate mdbox to maildir based on https://wiki.dovecot.org/Migration/MailFormat * Modified dovecot.conf and set mail_location=maildir:/tmp/destination/mailboxes/ * Then I had run /mnt/testmail/bin/dsync -Dv -u user at domain.com<mailto:user at domain.com> mirror mdbox:/source/mailboxes/user at domain.com<mailto:/source/mailboxes/user at domain.com> but that

I see a variety of DECT 6 phones available CHEAP at costco. Is there a way to convert these to SIP? I recall someone talking about a Siemens devices that works with all DECT phones, making them SIP (I think).... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081205/fdd14af9/attachment.htm

On this weeks VUC call we will collectively be our own guests. That is, we'd like to know what was the big issue that impacted YOU in 2010? All opinions welcome. Here are a few things to get you thinking in advance: - Apple's Antenna-gate - Asterisk 1.8 Launches - Amazon EC2 as a DOS platform - Cisco launched UMI video conference device - More HDVoice capable phones - Skype Outage - VoIP

I will have a small shop with ~4 phones using an HP server with Asterisk on it, it has two NICS and so I planned on plugging one into the cable modem, and the other into the switch. I was going to let this box perform NAT for the company but I am concerned about QOS for the VOIP portion. Anyone got a similar setup and care to share what they successfully implemented? Thanks! jlc

For some unfathomable reason, Siemens USA doesn't offer the Gigaset IP range in the U.S. I'm particularly interested in the Gigaset S685 IP. Since it's DECT 6.0, and there's an English (UK) version, I'm thinking it should work just fine, after dealing with the walwart issue (and maybe caller ID signalling). Anyone imported one from the UK and using it in the US? for how

Hello, I have installed oh323 channel driver. Outgoing calls to H.323 world do not include RFC2833 in the outgoing TerminalCapabilitiesSet despite that userInputMode=RFC2833 has already been set. Does anyone know how to make RFC 2833 DTMF relay work over oh323 channel? Kind regards, Fernando Herrera _____ De: Fernando Herrera [mailto:fherrera@iplan.com.ar] Enviado el:

I need to delete the DNS record and I could not have any idea something that is going bad root at zdtoe-02:~# dig axfr dtoe.etecsa.cu |grep zdtoe-01 *_msdcs\010CNF:946b89d3-b07b-4f43-a6b9-0de3bf3f3a10.dtoe.etecsa.cu. 900 IN NS zdtoe-01.dtoe.etecsa.cu.* *_msdcs\010CNF:4b642bd5-765f-4b16-93ce-3921b46ce220.dtoe.etecsa.cu. 900 IN NS zdtoe-01.dtoe.etecsa.cu. *root at zdtoe-02:~#*samba-tool dns

Hi As you may''ve seen from my last posts I''m struggling to get the setup on my subject line to work without flaws. I''ve tried compiling the Kernel from Source, I''ve also tried Fedora 14, which has a flaky pv_ops Dom0 kernel which DOESN''T work at all, but none of these seem to work. I''m a mid-time XEN user, I just love it and want to stay

Hi As you may''ve seen from my last posts I''m struggling to get the setup on my subject line to work without flaws. I''ve tried compiling the Kernel from Source, I''ve also tried Fedora 14, which has a flaky pv_ops Dom0 kernel which DOESN''T work at all, but none of these seem to work. I''m a mid-time XEN user, I just love it and want to stay

Hi there, in case anyone is interested, I've just taken ownership of a small home network (3 handsets) of the brand new Siemens DECT PSTN/VOIP phone. It works great with Asterisk. Here's my overview and review so far... http://www.theopensourcerer.com/2008/04/27/siemens-gigaset-685ip-phones/ Cheers Al -- The way out is open! http://www.theopensourcerer.com

Yeah, I think dovecot was a custom compiled build Version is 2.2.18 Config: # 2.2.18: /etc/dovecot/dovecot.conf # Pigeonhole version 0.4.8 # OS: Linux 2.6.32-754.14.2.el6.x86_64 x86_64 CentOS release 6.10 (Final) xfs auth_master_user_separator = * auth_mechanisms = plain login dict { acl = mysql:/etc/dovecot/dovecot-dict-sql.conf.ext } disable_plaintext_auth = no lda_mailbox_autosubscribe =

Hi people, Now I am using the version of gluster 3.6.2 and I want configure the system for fault tolerance. The point is that I want have two server in replication mode and if one server down the client do not note the fault. How I need import the system in the client for this purpose.

On Tue, 4 Jan 2011, Andy Graybeal wrote: >> The Polycom 321 has not been EOL'd and supports VLAN. It is, however, >> lacking a 2nd ethernet port if you were to go that route. >> >> -M >> > Thanks for the response Mark. I see the 331 has two ports and the same > features as the 321. > > I'm wondering what phone would be best being used as an

I could delete all records of this server that no longer exists no problem even record type NS I only have these two that there has been no way to erase any other ideas thanks samba-tool dns delete zdtoe-02.dtoe.etecsa.cu dtoe.etecsa.cu _msdcs\010CNF:946b89d3-b07b-4f43-a6b9-0de3bf3f3a10.dtoe.etecsa.cu. NS zdtoe-01.dtoe.etecsa.cu -U administrator GENSEC backend 'gssapi_spnego'

Has anybody have any experience using the KX-TGP500 Cordless DEC 6.0 "System" with asterisk ? I run a small asterisk server at home using two SPA3102s, and thinking of upgrading my cordless analog phones to something a little newer. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have quite a number of users complaining that when they are using handsfree to talk over a landline, the other end can't hear them. It's like the person is speaking 5 feet away and can barely hear their voice. However between internal SIP calls, it's fine. What could be the problem? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Using a Polycom 550 and 650 phones on my Asterisk server for testing. I can't figure out why the volume is so low. How can I adjust the volume control on Asterisk? It's at max on the handset phones. Thanks! Hin

While I myself am not a MagicJack user, I'm curious as to whether anyone here has managed to heist their MagicJack account's sip credentials, and use them to terminate calls using asterisk. Apparently it's as simple as sniffing the SIP credentials. If so, said person would enjoy unlimited termination for $20 year while retaining the flexibility of setting their CallerID to a