similar to: IP300 soft key configuration

I searched and read all the relevant posts, but I still don't have a solution to my problem.. I've got a small queue for tech support calls using AddQueueMember. The agents are using IP300's from polycom. In my example, only one agent is logged int. When a call comes into the queue, asterisk sends the call to the one agent logged in. The agent answers and is talking to the

hi all, anybody could help me with polycom ip300 bootrom-2.6.1 sip 1.4.1 conf files store on ftp server SUBSCRIBE/NOTIFY and MESSAGE method failed. either chan_sip is bad or i made mistakes but SUBSCRIBE/NOTIFY and MESSAGE method work with many sip servers. REGISTER is ok but asterisk send 407 error for SUBSCRIBE ?! I do hope a user could help me. Harry D?couvrez le nouveau Yahoo!

Hi, Has anyone here reprogrammed their Polycom features keys using sip/ipmid.cfg? If so I would be really grateful if someone could send me an example as I have tried various entries for hours now and don't seem to be getting anywhere. Any help appreciated. Kind regards Jamie Heckford Technical Consultant

Hi, I'm trying to configure a Polycom IP Phone SoundPoint 500 to connect it to my Asterisk PBX but with no success. First of all, I downloaded the SoundPoint IP SIP Administration guide I found on internet and then I tried to make a boot server creating an FTP account on my Mandrake 9.1 Linux box but I needed the following files: 000000000000.cfg sip.cfg phone1.cfg ipmid.cfg sip.ld so I

Hi, Just started using an asterisk-based PBX with Polycom IP501 phones. Am Fairly satisfied and am starting to get into FTP setup of the phones. Have figured out most things except for how button remapping works. In sip.cfg, I have this entry: <keys key.IP_500.31.function.prim="DoNotDisturb"></keys> This works as expected but if I try to change the remapping to any

Hi all, I've just updated to current CVS, and have 2 polycom IP phones, one is a IP600 and the other is a IP300. The IP600 shows the status of the IP300 and a ZAP line quite nicely, but the IP300 won't show the status of the IP600.... Is there any additional debug apart from "show hints" to see why this might not be working ?? -= Registered Asterisk Dial Plan Hints =-

The current code for primitive splitting and emission on pre-nv50 is severely broken. In particular: 1. Quads and lines are totally broken because "&= 3" should be "&= ~3" and similar for lines 2. Triangle fans and polygons are broken because the first vertex must be repeated for each split chunk 3. Line loops are broken because the must be converted to a line strip,

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

hello list, i have problem when i am useing wrapuptime with agents.conf my agents.conf looks like this [agents] autologoff=15 musiconhold => default wrapuptime=50000 group=1 agent => 1001,4321,Mark Spencer recordagentcalls=yes my aim is every call needs have wrapuptime of 5000 ms but when ever a call comes its directly connecting not wating any more. your views will be highly

I need to find a way to have the Polycom phones automatically park calls. Right now my users hit #70# (I know the last # is optional but it speeds it up.) to park a call. Personally I think this is easy, but my users would like one button to do this for them. The Polycom has buttons we don't need (Transfer & Services), it would be nice if I could remap one of those buttons to dial

Hi all, we're buildin an Asterisk box based on an Intel IXP425 board. The board uses a Beronet BN2S0 ISDN card, mISDN 1.1.4 and asterisk 1.4.2. hfc_multi has been patched to compile under big endian cpu, and so also capi kernel files. All the modules seem to load correctly (configuration was made with misdn-init config), but when starting cha_misdn, asterisk outputs the following lines: P[ 1]

Is it possible to rewrite caller id's? I would like to have sip phones appear by their local cid (like Henk <208>) but when they call out using the PRI I would like their full DID (MSN) to appear (like 0031201234567) I could ofcourse set callerid to the main phonenumber but surely there must be a better solution? Thanks!! Remco

i am using asterisk 1.6.2 with mISDN and a passive AVM Fritz!card on CentOS 5.5. The only thing, i want to do is a call-redirection from an isdn-call to my mobile via sip-account. My extension conf is: general] static=yes writeprotect=no [globals] OUT_PORT=1 [ISDN] exten => 12345,1,Dial(SIP/012346737222 at sipprovider.local) If i call to the msn 12345, the SIP-call is going out, but after

Hi. I'm running Asterisk server with 10 sip phones, and 2 grouped T0 lines with 10 DDI numbers. My provider is France Telecom and my setup is : - Debian Lenny - Asterisk 1.4 - Linux kernel 2.6.25.17 - mISDN 1.1.8 driver - Sip phones Thomson ST2030 No problem with the SIP . But when reveiving a call on RNIS line (any of the DDI numbers), the associated SIP phone rings indicating _two_

recently i've upgraded asterisk from 1.2.4 to 1.4 All works fine but i'me experencing some instability on misdn channels. In the last week i've experienced twice some problems with misdn (I am using mISDN-1_0_4) dmesg output: mISDN_rdata: rport queue overflow 256/256 [addr:52020501 prim:120282 dinfo:ffffffff] mISDN_rdata: rport queue overflow 256/256 [addr:52020501 prim:120282

I found on the web that there is way to customize the softkeys for the 7941/7961 phones. In the SEP<mac>.xml there is a section called "softKeyFile" where you can specify an xml file for the softkeys. I couldn't find any examples of this softkey file or the format to this file. Does anyone have a call manager enviroment where they can look at this file and send me an

Hi All, I have some Polycom IP300's and I'm interested in increasing the max volume for the headset (not handset), I'm wondering if anyone has experience adjusting these values: <gains voice.gain.rx.analog.handset="0" voice.gain.rx.analog.headset="0" voice.gain.rx.analog.chassis="3" voice.gain.rx.analog.chassis.obs="-12"

Hi, We are using Asterisk 1.2.1 with Cisco 7940 and 7960 phones. Most things are running fine ;-) But, when you are calling and you want to Transfer, you need to press first on the 'more' button (4th), then you have the label 'Trnsfr' to Transfer. these are the lables on the softkeys when having a phone call: "Holt / EndCall / Confrn / more" press more and you get

I'm trying to set caller ids on outgoing calls. I have a quad BRI B410P card connected to my telephony provider. I know the list of DID numbers the provider assigned to my company. If I don't set the caller id then the callee always sees the same "top-level" number. If I set the caller id to a specific DID number we own, the callee keeps seeing the "top-level" number,

I am getting to my wits end with these phones (and so is my boss). I am getting an random echo on these phones and I have an issue opened with Polycom and its been in their research and development department for almost a month with no results. I have noticed that I get a message "RFC3389 support incomplete. Turn off on client if possible" in asterisk. I have researched this and made