similar to: 7912G via SIP, looking for comments

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 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

I need to purchase approx. 10 phones for a small office implementation. Nothing fancy is required besides a full-duplex speakphone, in the sub-$200 range. I am currently looking at the Polycom Soundpoint 500, Uniden UIP-200 and SwissVoice IP-10. I have searched around and found various posts regarding each phone's ability to work with asterisk (SIP, I probably should have mentioned), but

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

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 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

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

Friends I'm trying fit a survival model by maximum likelihood estimation using this function: flver=function(a1,a2,b1,b2) { lver=-(sum(st*log(exp(a1*x1+a2*x2)))+sum(st*log(hheft(exp(b1*x1+b2*x2)*t,f.heft))) -(exp(a1*x1+a2*x2)/exp(b1*x1-b2*x2))*sum(-log(1-pheft(exp(b1*x1+b2*x2)*t,f.heft)))) } emv=mle(flver,start=list(a1=0,a2=0,b1=0,b2=0)) where hheft and pheft are functions defined in

Hello All! I have a strange problem with Cisco 7905. It is forwarding unanswered calls to VoiceMail even thought I have setup it not to. My ring timer on cisco 7905 is 60s, and my ForwardToVMDelay is 3000s. This means that call should never be forwarded to VM! This is true if I call from internal number then this happens on asterisk: -- SIP/104-6073 is ringing -- Nobody picked up in

Mmm...I swear I read somehwere that the 7912G did SIP? Cisco lists it as an SCCP only phone? -- Undocumented Features quote of the moment... "It's not the one bullet with your name on it that you have to worry about; it's the twenty thousand-odd rounds labeled `occupant.'" --Murphy's Laws of Combat

Hi, How well to the Cisco 7905G or Cisco 7912G phone work with Asterisk? Cisco claims both phones do SIP. I was strongly considering Polycom phones. However, it appears to be quite difficult to obtain support or firmware for Polycom phones. On the other hand, I find Cisco is very well supported. Thanks, Adi

I got this figured out much faster than I thought I would. Thanks to all of your help, Peter, Mattias, Thomas and Helmut. And Thomas, that 8 partition live OS sounds right up my alley, and I will definitely check it out. The second partition is working well with `parted' and `fdisk', so I am quite pleased. Below is, again, "what I came up with". I tried to integrate each of your

I recently got a 7905G w/ Sip software preloaded. I got it working w/ asterisk with no problem setting it up through the phone. I am now trying to make it download the config file from the tftp server. I have set all of the options in the file and the file is definately named correctly. But the phone is simply not processing the config file for some reason. Two commands Im trying to get

Friends, I changed one line of a package at the source level and then rebuilt it. When I run R CMD check, I get an error: installing R.css in C:/polsplineRS.Rcheck ---------- Making package polsplineRS ------------ adding build stamp to DESCRIPTION making DLL ... making hareall.d from hareall.c making heftall.d from heftall.c making lsdall.d from lsdall.c making lspecall.d from lspecall.c

Hi, a small question.. I'm using NTP to synch our phones with an ntp server, but it seems the Cisco 7912G (with SIP image) does not handle daylight savings time very well? Am I overlooking something or is this a known feature? I'm using GMT+1 and minutes are correct but it doesn't respect DST. SIP software seems to be: v1.02.00(040406A). Cheers, Kristof.

hi folks well, tinc is a really nice tool and we implemented it on 3 linux servers and 2 mobile clients (XP notebooks) so far. one of the 3 tinc servers is making troubles, when a connection is initiated from this server over a zyxel 642 adsl router out to the other 2 servers in the internet. the logfiles of the other 2 servers shows: > tinc[1398]: Received UDP packet from unknown

-----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of asterisk-users-request at lists.digium.com Sent: Wednesday, August 22, 2007 10:51 PM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 37, Issue 88 Send asterisk-users mailing list submissions to asterisk-users at lists.digium.com

Dear all, I have a question on how to set up the starting model in POLYCLASS and make sure the terms in the starting model retained in the final POLYCLASS model. In the function POLYMARS, this can be done using the STARTMODEL option. See below for example, I started with model y= b0 + b1*X1 + b2*X2 + b3*X4 + b4*X5 + b5*X2*X5 + e > m00 <- matrix(c( 1, NA, 0, NA, 1, 2,