similar to: Uniden, Polycom or SwissVoice???

Hello, I am trying to come up with a good way to enforce a limit on the number of simultaneous calls that can occupy an IAX trunk at any given time. I have searched around and so far can't locate a config option that would directly label a IAX trunk with a specific number to obey (is there one?). Based on examples for the SetGroup and CheckGroup commands, I am thinking of using SetGroup

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:

Can anyone help me to understand what the significance of this output is? May 17 10:50:23 DEBUG[2030]: Didn't get a frame from channel: SIP/105-1ae4 May 17 10:50:23 DEBUG[2030]: Bridge stops bridging channels SIP/105-1ae4 and SIP/outbound-7dc3 I searched for these phrases but am coming up short on what they really mean. I'm trying to investigate problems we are having with two

Hello, I'm looking for any comments or user experiences from anyone who is using 7912G phones with SIP. Any installation issues? Usability problems? Do the features seem to work, etc...In short, I'm looking for your opinions on how suitable this phone is for an asterisk implementation for approx. 10 users. Next logical question: what other phones would you recommend for a situation

Since upgrading to 7.2, I've noticed a random problem where I dial a number and hear all the correct tones in the handset, but the display won't show all the numbers I dialed. So you sit there waiting for the dialplan to kick the call off (b/c you heard the proper amount of tones played and think it's all good) but the phone is just sitting there b/c it somehow "missed"

Ok, ok, I know there has been plenty of discussion on asterisk and fax - from this I understand: 1) First and foremost, use g.711 ulaw 2) Packet loss, etc...makes faxing over the internet unreliable My need is for a fax to come in on a X100P and be forwarded to a fax machine on the local lan. I don't currently have any fxs as I'm using all sip phones at this point. I see the

I haven't seen any status on the 7960 backlight project lately...I tried to email the original poster but his mailbox appears to be over quota. Does anyone have an update on this? Thanks, Marty Mastera M3 Resources marty@m3resources.com Phone: 303.680.1283 x200 FAX: 303.680.1283 IAXTel: 700.206.7507 FWD: 484162 -------------- next part -------------- An HTML attachment was

Has anyone got this to work? Under Idle Display Animation, the administrators guide says "For example, a company logo could be displayed".. In the ipmid.cfg file, I enabled 'ind.idleDisplay.enabled' (ie changed it to 1), and under the IP 500 section, I added an entry for the bitmap that I want to display: bitmap.IP_500.66.name ="arf" but from there I'm not sure

I was looking at the sample "follow me" config (http://www.voip-info.org/wiki-Asterisk+Tips+follow+me <http://www.voip-info.org/wiki-Asterisk+Tips+follow+me> ) which uses a dial modifier 'c' to enable Answer confirmation - "If the letter c follows, then "Answer Confirmation" is requested, in which the call is not considered answered until the called user

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'

Hi - I'm wondering if any has experience with the Uniden UIP-200 phones. The product info says that the 8 led buttons at the top are all programmable. Can they be programmed as separate line appearances (ala Snom 200, Cisco 7960, Zultys Zip4x4, etc)? In other words - is the phone capable of multiple SIP registrations? Also, the post about these phones at voip-info.org mentions some

Hi, I'm new to asterisk, and have a uniden UIP 200 that I got off of ebay. I'm having trouble getting the phone to register with asterisk. I've tried a few different settings. I'd be extremely grateful if someone with a similar setting could give me the sip.conf block for the UIP and the settings you're using in uniden.txt. Here's what I have currently: IP of phone

I have several too and I also see this problem on occasion. Like you say, it is fairly rare and I can't pinpoint a cause or tell if it is a symptom of something else. I think I wrote to tech support about it but never heard anything. I'm wondering how long they will continue to support the phone. -Nate > -----Original Message----- > From: Jerry Geis

I have set up a new SwissVoice phone and it can receive calls but I cannot make calls out from it. The setup is simple for now, 2 phones: SwissVoice is ext 7726 and Cisco 7960 (SIP) is ext 7999. I can call from the Cisco phone and it rings on the SwissVoice phone but when I dial from the SwissVoice phone I get a busy tone upon dialing the second digit. The log reads as follows: -- Endpoint

I just spent the last hour or so trying to get this firmware to work across a NAT with no success. I have a GS BT101 working through the same NAT, so I don't think it's the NAT itself. I have a STUN setup in * and pointed the UIP200 to it and I tryed several combinations of nat= in the sip.conf and in the config files for this phone. No luck(yes, I did a reload now with each change in

Has anyone used the Swissvoice IP 10S (www.swissvoice.net) VoIP Phone with *? Adrian -- Adrian Walker adrian@digitaltraffic.co.uk ======================================================================= This email has been scanned for Virus infection by MessageLabs For more information please contact messagelabs@atomwide.com

Hi, Does anybody know the place to download the firmware for swissvoice ip10s I have several phones with application IP10 H3 v1.0.0 (Build 1) I'm looking for newer H.323 and also MGCP firmwares Are the SIP firmware available, according to web its targeted to Q1 2004, but we have week left in Q2 I sent several email to swissvoice support,, no answers Regards Juri

I have 1 IP phone (Swissvoice IP10S) and 1 POTS phone. When I dial the number for the IP phone off the POTS phone, the IP phone rings. But when I pick up the handset on the IP phone, I get a busy signal and this message on *: Aug 23 09:38:57 NOTICE[1142106560]: chan_mgcp.c:2243 handle_response: Terminating on result 502 from svip10@00059002042b-1 Here is the entire session. svip10 is the 1 and

Hi there, here some questions and experiences after playing for one day with 3 Swissvoice ip10s and the latest * CVS: QUESTIONS: - what is the user option "enter voice mail number" good for? It doesn't appear to be of any practical use - does anyone have some Swissvoice info that I cannot find on their web site like the guide to MGCP XML (.svd), guide to configuration file

Hi there, tonight I upgraded one of my ip10 phones to "Appli 1.0.0 build 3" with "Boot 0.3.6" This firmware is dated Dec.19, 2003. For Robert (aka "info lists") this fixed the multiplied digits problem (which I never had). Since there is no website of Swissvoice I guess you'll have to e-mail them if you want upgrade... Unfortunately the upgrade did