similar to: Last two digits getting cut off?

Problem: Polycom SoundPoint IP phones (running SIP) ceases to send DTMF that Asterisk can detect and use. It worked in 1.0.5, but has not worked since. This has been verified on SoundPoint IP 300's and SoundPoint IP 600's. Workaround: It used to be that for DTMF to work, I had to set the mode in sip.conf to "inband". Without making any configuration changes on the

hi all, I noticed that when I upgraded from Samba 2.0.7 (redhat 7.0) to samba 2.2.1a I lost this nice little script that came with redhat to start and stop the samba server. I would type in /usr/sbin/samba start and get a response like... /usr/sbin/smbd start (OK) /usr/sbin/nmbd start (OK) I liked that...does any one have a substitute that I could use for 2.2.1a? Thanks Carl

I am trying to use if...else loop and have included a code snippet which I might like to expand. Maybe you could steer me in the right direction. library(stats) library(prob) { a <- sample ( 1:4,100, replace=T,prob=c(0.1,0.2,0.5,0.3)) b<-sample(3:6,100,replace=T,prob=c(0.2,0.2,0.2,0.4)) } dd <- data.frame(a,b) if (subset finds a vector) ( print that vector) (else

Dear R help I was hoping you might be able to show me how to write a loop function take would ccomplish this task. library(prob) { a <- sample ( 1:20, 100, replace=T) b<-sample(5:24,100,replace=T) } dd <- data.frame(a,b) dd # code piece I am looking for if(subset(dd,c(1,23,ordered=F))is found))( print subset) else( continue evaluating subsets) subset(dd,isin(dd,c(1,23), ordered =

I just got a Polycom soundpoint and I set it up using the phone and web based admin. I cant seem to figure out the config files and they are confusing me greatly and I dont have time for it :) Some things are odd, like on every reboot it seems the volume I set is reset? is there any way to fix that. And the ringer seems low. - Even all the way up Anyone willing to point out a good asterisk

Hi folks, I started testing the antispoof feature of xen stable (2.0.7). I am stuck with it. I have setup a standard bridged environment. I understood it like this: in domU config I set up the virtual NIC like vif = [ ''mac=ae:00:00:78:78:78, ip=192.168.0.100'' ] Then I configure /etc/network/interface of this domU to show the same IP address for eth0. After restarting

Hi folks, I've got a very bizarre problem recurring when making calls with Polycom SoundPoint IP500 SIP phones and Asterisk. Sometimes when a call comes in to an IP500, one of the sides of the conversation is cut off (i.e. the caller can't hear the callee, or vice-versa). This isn't easily repeated, and rebooting the phone, or restarting Asterisk, doesn't seem to have an

Hi We've setup an Asterisk PBX recently and I encountered the following problem: When [mac address]-registration.cfg file includes the FQDN of the Asterisk PBX for the Polycom SoundPoint 501 phones it will not (even try to) register with the Asterisk PBX unless the DNS (it asks) successfully resolves the name: _sip._udp.[Asterisk FQDN]. Did this happen to anyone else? PS - The

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

I have an extension setup in my extensions.conf for hold music. ext. 600. If I pick up a phone (polycom soundpoint 300 sip) and dial extension 600 I hear the hold music playing. If I call another extension and pick it up and put the call on hold with the hold button on the phone I hear nothing at all. Does anyone have any experience with these phones and getting the hold button to work?

Has anyone been able to get the minibrowser on the Polycom SoundPoint IP 500/600 phones working? If so could you share the relevant sections of your config with me?

Hi, What is out there in terms of SIP enabled handsfree speakerphones? Looking for something that works well and also fits a low budget. I am used to using a Cisco 7940. It is a great phone but a bit expensive. Thought about the Polycom SoundPoint 300 until I realized that it does not include speakerphone functionality. Thanks, Adi

Does anyone have the polycom soundpoint ip's successfully remotely provisioning? I've got the phone pulling default configs, and it's downloading phone specific information, but it's not actually using that information. Any help would be appreciated :) -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

I've got a couple Shoreline IP phones, their Shoreline model number is Shoreline IP 100. I believe this is actually a Polycom SoundPoint IP 300 phone. I believe the phone is using a MGCP stack. I want to use it for testing with Asterisk. 1. I suspect I need to re-image the phone to make it work with *. 2. How can I preserve the current image on the phone? 3. What is preferred image to use

Hi Folks; I just bought a Polycom SoundPoint 500 off of ebay after having spent way too much time trying to get updated sip images for our cisco phones. The phone I bought didn't have an AC power adapter; Could someone please tell me the volts & amps that the dc plug that comes with the phone puts out? Thanks! Mike

Hello, I have finally received the details from Polycom to get into the backend configuration of their SoundPoint IP 600 SIP VOIP phone. The phone is quite nice looking but the configs are very sparse, not even a place for a secret(password) field in their SIP registration section. If anyone else has one of these and needs the passwords to get into the back end configurations, just send me an

We are looking to deploy asterisk at one of our locations that will have about 50 phones. I have been buying different phones to test there quality and feature set. So far we have a Grandstream 2000 Grandstream HandyTone 488 Cisco 7912 Polycom SoundPoint IP And we are looking at getting a Linksys SPA-942 Anyone have a favorite? -------------- next part --------------

Hello all, Has anyone used the SIP version of this phone with Asterisk? I see Polycom has a H.323 and MGCP version also, does anyone know if you flash the phone to swith protocols? Thanks in advance for the info. Ed -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031029/a9377305/attachment.htm

I'm attempting to set up my SoundPoint 501 with my Asterisk server. I've configured DHCP and TFTP and successfully updated both the BootRom and SIP application. I've also created a custom cfg file for this phone's MAC address and the settings seem to be taking just fine. I can see that the phone registers with my Asterisk server but 'sip show peers' reports that the phone

Hi all, I have a Polycom Soundpoint IP 600 that looks like it is fried. It either has a bad or corrupt bootrom (I'm guessing). It never gets to the prompt that asks if I want to enter the setup configuration or continue booting. When plugged into power, it turns on, shows the Polycom logo for 3 seconds and then pretty much goes dead with two red lights on. The one in the top right corner