similar to: Linksys/Sipura SPA-3201 FXO/FSA with Asterisk

Greetings list, Does anyone have any experiences they'd like to share deploying these phones in medium-size asterisk setups, e.g. 40+ users? I have a project coming up to deploy 100 phones over 2 offices and the client rather likes these phones. Are there any obvious pitfalls/configuration difficulties/quality issues etc. using these phones? If so, what alternatives would people suggest with

Hallo! I had ordered a Sipura SPA-3000 in the UK, but the supplier turned out to be unreliable and never shipped. Yesterday I went looking for alternative suppliers and found the Linksys SPA3000 device. It's a different box, but the specs look very similar. Is this the same device? Has anyone used this Linksys SPA3000 successfully with Asterisk? Thanks, Frank

Hello I'm about to order a GrandStream GXP-2000 and a Linksys SPA-921 I'd like to have some user feedback about how those phones perform, and whether their LCD screen displays both the caller ID name and number (The GrandStream BT-100 only displays numbers, which isn't very helpful). Thank you.

All, I just had a power failure that caused me a little drama. This spurred me into trying to get NUT up on my cyberpower 1500 again. For about 6 months I tried to get it to work with the new power panel drive with no luck. Today I removed all the old files for NUT and installed the newest version I could find (2.4.3) and it works perfectly now with my UPS. Hopefully now I

Karl Heyes wrote: > Seann Clark wrote: > >> The biggest being is that the server binds to whatever port it feels >> like instead of the bind port specified. The rest of the issues I have > > a random port bind is a new issue. Can you show us the > netstat -tnlp | grep icecast > for the xml provided? Thu Jul 03-13:54:17-root at haruhi-new:~> netstat -tnlp | grep

Hi All, I have a telephony project for which I need to build a prototype to demo for management. The prototype must work on a GSM phone network. In the demo system, a call from GSM phone comes into the demo box. The demo box runs CallWeaver. Callweaver picks up the GSM call, answers it and plays a sould file, then dials out to a second GSM phone somewhere and connects them so they talk. My

To: Twin Cities Asterisk Users From: asterisk_help at iwishi.nu Subject: TwinCities Asterisk Users Group Meeting this Saturday - Only 1 and 1/2 days away! Meeting Start: 09/08/2007 - 11:30am Hello all Twin Cities Asterisk Users, It's time once again to have another meeting. I've not had much time to prepare, but I'd really like to review and install with the group at our next

All, I am toying with an idea of using an AGI to be able to 'call' my phone, or phones, in case of severe weather warnings. I have been tinkering with a script that reads from weather underground for the forecast, based off a PHP version of a weather AGI I found on the net. It seems rather trivial to have the AGI as a script, that does nothing unless a condition is met, and

n active project, than a dead one. Otherwise who is going to patch vulnerab= ilities? Not me. I'm not a software developer. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . From: Steve Totaro [mailto:stotaro at totarotechnologies.com]=20 Sent: Thursday, March 24, 2011 11:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What

with an active project, than a dead one. Otherwise who is going to patch vulnerabilities? Not me. I'm not a software developer. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . From: Steve Totaro [mailto:stotaro at totarotechnologies.com]=20 Sent: Thursday, March 24, 2011 11:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]

With the increased number of "brute force" login attempts against port 22, I am concerned that an intruder may actually stumble accross a valid user/pass combination. To combat this, I would like to request an sshd_config option that would cause the running sshd parent process to keep track of login failures by IP address. If there are more than X number of login failures for a

I am running into an issue with rsync that I need some help with. When syncing large files (e.g. 1GB), the rsync algorithm creates a temporary 1GB file and then renames it when the transfer is finished. The issue I am running into is if the two large files have very few differences between them, the bottleneck is creating the 1GB temporary file on the target box. This process takes several

Hi list, I'm evaluating a private telephony scenario of about 20 locations - 300 phones, 50 FAX machines. Initial overview points to the installation of asterisk at three locations connected to the PSTN via ISDN PRI. All other locations, small by themselves, would get SIP phones managed by asterisk, since there is good IP connectivity between all sites. Now on to the

All, I just noticed this in my logs, and am rather lost as to what module it pertains to. I would assume pseudo-realtime priority for the process, but I am looking for a little confirmation from the group: [Apr 28 12:28:36] WARNING[20773] asterisk.c: The canary is no more. He has ceased to be! He's expired and gone to meet his maker! He's a stiff! Bereft of life, he rests in

All, I have a Cyberpower UPS that I have been working with for about a year. I have used NUT in the past with good results, but as of late have been seeing issues with nut talking to the UPS. I will start with the information on the server that is running NUT, and has the UPS connected VIA an RS232 cable. O/S : Fedora 9 Kernal (uname -a): Linux haruhi 2.6.26.5-45.fc9.x86_64 #1 SMP Sat

All, I have a Cyberpower UPS that I have been working with for about a year. I have used NUT in the past with good results, but as of late have been seeing issues with nut talking to the UPS. I will start with the information on the server that is running NUT, and has the UPS connected VIA an RS232 cable. O/S : Fedora 9 Kernal (uname -a): Linux haruhi 2.6.26.5-45.fc9.x86_64 #1 SMP Sat

Hi Guys, Just did an Asterisk 1.6.x (repo install) and FreePBX (source install). When trying to dial a number, I get this: tel*CLI> Use of uninitialized value in hash element at /var/www/html/panel/ op_server.pl line 3367. Use of uninitialized value in concatenation (.) or string at /var/www/html/panel/op_server.pl line 3372. Use of uninitialized value in pattern match (m//) at

All, I have been fighting with my dialplan for hours now, and google searches talk lots but offer nothing in terms of explication for this. I have my SIP peer set up and working with Broadvoice: [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=5555551234 secret=password defaultuser=5555551234 insecure=port,invite context=broadvoice

Hello list, Topology diagram first: [PSTN]-----[SPA 400]--------[NETGEAR SWITCH]-------[ASTERISK BOX] | | | |__[SPA 921] | [SPA 901] The problem is the

Hi, I was wondering if anyone stumbled upon the correct event in a sip NOTIFY (from a SUBSCRIBE) to make the BLF lamps on a Linksys/Cisco SPA9xx/5xx go yellow? I'm trying to differentiate between "On the Phone" and "DND" with the BLF. Thanks. -- James