similar to: hints and polycom IP 300 phones

Hello, I patched asterisk cvs head sources with http://juraj.bednar.sk/work/software/asterisk/messaging/ and presnce patch without success. asterisk send "405 method not allowed" to sender. I use polycom ip300. Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger

Hello, I am looking at the Polycom phones. The ip600 has a very nice screen, is that the only real advantage over the ip500 and ip300.. Is it worth the extra dollars? Kind regards Michael Felder IT Medic Australia Pty. Ltd. P: 03 9557 2213 F: 03 9557 2214 M: 0419 568 217 E: Michael@ITMedic.com.au <blocked::mailto:Michael@ITMedic.com.au> http://www.ITMedic.com.au

Hi All, I'm using Asterisk for my PBX, I have a remote office that is connected by a VPN link. I am using Openswan on my side and a Linksys box on the remote side. I have a Polycom IP300 on the remote side configured with a static IP address. When I call the phone on the remote side, it rings and establishes the call fine. The problem I am having is that the remote side can hear the call

Hello I have a Junghanns ISDN BRI card for incoming calls and use SIP Polycom IP300 phones. My config files look like this: features.conf pickupextn = *8 zapata.conf context=frompstnisdn group=1 callgroup=1 pickupgroup=1 I also edited sip.conf like this: group=1 callgroup=1 pickupgroup=1 But on internal and incoming calls if I dial *8 from any phone I cannot pickup. Do I need to add

Hi All - I'm running version SIP version 1.3.4 on various IP300, IP500, and IP600 Polycom phones. I'm having a tough time with MWI. I thought I remembered somebody on the list saying that they had it working, but I can't find it in the archives now. I have all the phones configured for MWI as specified in the WIKI: ipdmid.cfg: up.oneTouchVoiceMail="1"

I've had an IP300 for a while now and it's been working fine. I just got an IP500 and when it connects to the FTP server it downloads the new bootrom and says error loading. The bootrom is fine and works on the 300... In addition, I downloaded a new copy to be sure and it still doesn't work. Can anyone give me some advice?

At the most recent project I completed I have to post a intranet web page with instructions on using the system and phones. Asterisk is 1.07 stable and the phones are Polycom IP300, IP500, and IP600. Has anyone done an Astersik users guide? Something non-technical but covering most of the features an office worker would use. If nothing exists, should we develop this as a documentation project?

Hi All, I am a remote office that is connected to my office via openvpn on UDP. Voip has always worked well (after discovering g729). Initially I used a softphone, then an analog set on a sipura 2000, then a polycom IP500 (I still LOVE this phone). At that point, I started noticing that the polycom doesn't ring a lot of the time. Since I was desperate for a phone, I didn't upgrade

Hello, We have a problem on a recent asterisk install with Polycom 30x phones; Sometimes (can not reproduce or find the logic of the problem after one week one analysis), the called party (even incoming or outgoing call) can not hear the calling party, as other flow works (caller hears called). This occurs between 5 and 10% of the time. The configuration is the following: - Asterisk 1.2.9.1 -

I'm trying to reconfigure my IP300 softkeys.. Currently when on a call, I have to hit more and then transfer.. I'd like make transfer appear on the first screen. Right now there's hold on there.. and hold is kind of redundant, since the IP300 has a "hard" hold button. I tried doing it in the <keys/> section of ipmid.cfg, but it doesn't seem to work.. anyone done

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"

All, Slightly off topic. Polycom released their SIP software version 1.6.6 for their phones recently. I was under the impression that this release fixed a previous limitation where the phones would only watch 7 buddies, ie send 7 sip subscriptions to Asterisk. I have configured a phone directory to watch 30 or so appearances, and it still seems to only be sending 7 subscriptions to Asterisk.

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!

I'm currently using the Linksys PAP2, and since there's a shortage I'm looking for different devices. I'm mainly looking at the Sipura SPA sets since they are the base of the pap2. Anyone else have experience using them, and which one? Thanks Sherwood McGowan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Have the Sayson IP phon started to deliver yet? I'm thinking about two new phones for my office and considering the Sayson 480i and Zultys 4x4. Would also consider the Virbiage phone if it becomes available. I have Snom 200s and a Pingtel phone at the moment. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist

I have had very bad experiences with IAXYs so far.. I have pulled them and will be attempting a refund shortly. Bad audio, overheating and shutting down until allowed to cool, etc. make it unusable in a business environment. That said, is there a low-mid priced solution for a remote office to connect to a home office runing asterisk? There seems to be a hole in the market for 6-8 person

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

>Message: 9 >Date: Thu, 02 Dec 2004 11:33:16 -0700 >From: "Kevin P. Fleming" <kpfleming@starnetworks.us> >Subject: Re: [Asterisk-Users] Polycom 500, asterisk user opinions? >To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >Message-ID: <41AF5FEC.7050406@starnetworks.us> >Content-Type: text/plain;

When I bought two Polycom 501 SIP phones, I naively thought they were Power-over-Ethernet (IEEE 802.3af) because they were "powered over ethernet." Silly me. Polycom must have some odd voltage or funny way of injecting the power, because the POE switch I bought for them (Netgear F@510P) won't power them, though if I use the Polycom-supplied AC adapter and ethernet power

I have just finished compiling and installing Asterisk on a test Debian system. All is working well. We are now attempting to get remote offices to test the system I have installed both a SIP and an IAX client at a remote office. Then I connect to our office via Microsoft ISA firewall and the Windows XP VPN client. Neither of the softphones will connect. On the IAX softphone I just get a ringtone