search for: 200's

how do these phones perform? ok for office use? work well with asterisk? any info would be appreciated.

...the other problem I have is that sometimes ( like maybe 4 times a day ) we will be talking to PSTN calls and one side of the call will go silent and then 5 to 10 seconds later come back. we have this setup. PSTN <--> Asterisk /w T100p <-IAX2-> Asterisk2 /w zapdummy <-sip> snom 200's, and locally in our office it looks like this PSTN <-T100p-> Asterisk <-sip-> snom 200's (the echo is on this setup as well) any help would be appreciated.

Look like SNOM have rolled back the firmware version of the 200's from 1.16w to 1.16q.. Anyone know why? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Dear Group, Does any one have experience using SoundPoint(r) IP 300? I have one call center on Snom 200's I'm adding a second and was looking at the SoundPoint, but needed some input. Thanks Shad Mortazavi --------------------------------------------------- Nexus Technical Manager n|m Nexus Management Inc Sydney -------------- next part -------------- An HTML attachment was scrubbed... UR...

Our basic system is as follows: P4 3.0 Ghz w/ HT, 1GB PC3200 RAM, 120 GB HDD, RH 9.0 OS, * from CVS several weeks ago, working OK for routing, VM, and AA, calls in on separate PSTN lines to Adtran TSU 600, into * server through T100P card. The hardware is not taxed at all with little over 20% proc utilization ever, low mem use, etc. All Phones are SNOM 200...

I'm trying to put multiple Linphones and Snom 200's into a Meetme room. With two devices, echo is quite noticeable. With 3 or more it degenerates into white noise. Which part of the software is responsible for echo cancellation in a MeetMe room? Is it a setting on the phones themselves, or within Asterisk? And is this related to echo...

is there a firmware for IAX for the snom 200's. or are there any other hard phones that use iax(2)? Thanks in advance!

We are running * 1.0 and have 6 snom 200's and one snom 220. the line hints work until I issue a reload command and then the snom hints stop working. I can re-create this scenario on 2 systems and with snom 190's also. is there a way to fix this? the only solution I can come up with is to reboot the phones. Thanks in advance,...

My * at home is a P-3 400 256 meg with a TDM 400. 2 cordless phones, 3 snom 200's. Termination is through an IAX provider. All of the standard stuff works, transfer to cell, web voicemail,etc but the interesting thing that I do is a script that polls the Canadian weather service every 10 min looking for severe weather warnings (happens quite often in Alberta) and if it fin...

I had a working system using * 1.0 and then 1.2 and now Asterisk 1.4.13 with addons 1.4.4, zaptel 1.4.6, libpri 1.4.2. I have a mix of Grandstream (GXP2000), Uniden uip-200, Linksys Wireless G, and analog phones via Adtran chan bank. When I went to * 1.4.13, the Uniden phones stopped being able to login to voicemail. All phones are on same lan with Asterisk. I get 'Login incorrect' from Allison. I go to any other phone and I can log in ju...

I would like to use one of my Snom 200's 5 programmable keys to park calls. I am using image SIP 2.04g. I have tried a variety of combinations and have come to the conclusion that: 1) On the Key Mappings administration page, I must select the "Transfer" under the "Break Keys" option box to be able to successfull...

where can I get the 2.05 firmware all i see is the 2.04 firmwares :-) also anyone got a fix for the horrible speaker phone on the 200's

That'd be the 3CNJP24SE, we have one that powers 3COM NJ-200's. Works well. -----Original Message----- From: Martin [mailto:marrandy@chaossolutions.org] Sent: Tuesday, January 20, 2004 9:23 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Power Over Ethernet for *any* ethernet switch (or hub); product idea On Tuesday 20 January 20...

Hi, I'm using * with the latest CVS image as of this afternoon on Fedora1. I am using snom 200's with the latest firmware. When I dial the voicemail extension and try to enter my password, * does not receive the password. The console complains about not receiving the password. Also, when I try to leave voicemail messages there seems to be no talk path and the recording session times out...

...e talking on a call the caller hears nothing for 5 to 10 seconds. The person on the other end of the call hears everything just fine. Then the call returns to normal and both parties can hear. our network: PSTN <--> Asterisk /w T100p <-IAX2-> Asterisk2 /w zapdummy <-sip> snom 200's Any insight would be much appreciated. lach

...bc frame that isn't a multiple of 50 bytes long from RTP (4)? This means that I cannot get access to voicemail from the handsets !!! Any clues??? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040602/b35a44b3/attachment.htm

What are these messages that * constantly sends to the client? And why does it always say 'looking for [server ip] in default' ? Sometimes I see 404's instead of 200's, and I know something is wrong but I don't know what is being signalled here. In this case 206.80.111.117 is the * server, and 4.27.217.38 is my nat box. <-- SIP read from 4.27.217.38:61384: OPTIONS sip:206.80.111.117 SIP/2.0 Content-Length: 0 Call-ID: 5F620712-81EE-4E77-925D-DCF4...

...ve echo cancelled. We are using analog phones connected to asterisk TDM cards. Thanks a lot. Michael ======================================== Hi, Oops, my bad. Turns out it was just mixer settings, feeding back through the soundcard. Sorry for the noise. Message: 14 Date: Wed, 03 Dec 2003 17:43:16 -0500 From: Matt Lawson <matt@1control.com> To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Echo cancel in MeetMe? Reply-To: asterisk-users@lists.digium.com I'm trying to put multiple Linphones and Snom 200's into a Meetme room. With two devices, echo is qui...

...an to allow international dialing from my US location. As I understand it, an international call can have 9 to 15 digits including country code. The problem is that the call always goes through after I've entered the 9th digit. My service provider is BroadVoice, my phones are Grandstream GXP-200's. DTMF is set on the phone to be via SIP INFO and Early Dial is set to "Yes". I have this in my plan: [outbound-long-distance-forio] ; this is how outbound long distance calls are handled ; international numbers are max 15 digits included 2-3 digit country code (E-164 ITU protocol...

Hi all, thanks to the all of you. This list is very interesting also for a newby like me. My problem: I just setup my first full working asterisk installation with this config: 1. n.1 GXP-2000 2. n.4 Budgetone 102 3. n.1 TDM400p (3 FXS, 1 FXO) Everything seems to work fine, but the sidetone... it's really annoying! We can hear the sidetone only when we call to the outside (PSTN), it doesn't matter if we call a local, a mobile or a longdistance call. Only we hear the echo, not t...