I am getting ready to install Asterisk and I was looking into the Polycom IP600 phones. I spoke with Polycom sales to verify the multiple line appearance and they said it would work. More specifically, if lines 1-3 all contain the same SIP registration info, the Polycom will only send out 1 SIP registration to the server and then handle the calls ringing on multiple lines. I was wondering if anyone can confirm that this works with the polycoms. I know the 7960s support this, but I want to make sure the Polycom sales team wasn't just saying Yes to make the sale. Any comments are appreciated. -Eric -----Original Message----- Subject: fwd on busy when calling multiple extensions at once Chris A. Icide wrote:> IMHO, the Polycom IP600 is a superior phone to the cisco 79XX phones. > I base this off of having had both an IP600 and a 7960. The two > advantages the 7960 had over the IP600 was appearance and ease of > configuration. Outside of that, the IP600 (IMHO) beat the cisco hands > down. > > Now, you MAY want to try registering all 6 lines on the polycom to the > same line and see if the phone handles that as well as the cisco. If > it does, then you are set. Otherwise, you will need some complex > configuration work in your extensions.conf to achieve what you are > looking to achieve. > > Some thoughts: > > What do you want to happen when one of the call takers has all 6 lines > in use? > > Have you considered using queues to do what you need? > > -Chris > > On 10:08 AM 5/22/2004, Brian Cuthie wrote: > > > >You might consider using the Cisco SIP phones. They're smart enough > >to accept incoming calls for as many call appearances you have with > >the same SIP registration. > > > >-brian > > > >Tor Roberts wrote: > > > >> Hi, > >> I am setting up a dispatch center where will have 4 call takers, > >> all with Polycom IP 600 Sip phones. Each phone will be setup with 6 > >> extensions each. When a new call comes in, the first extension on > >> all the phones will ring. This works fine, the problem is when one > >> of the dispatchers is already using her first extension and another > >> call comes in. What happens now is that the remaining 3 phones ring > >> on the first extension, but the dispatcher who is on a call, her > >> phone does not ring. I want her second extension ring along with > >> the other 3 phones first extensions. > >> > >> In sip.conf I have all the extensions set to incominglimit=1 and > >> the pertinent part of extensions.conf is: > >> > >> exten => s,1,Dial(SIP/5000&SIP5001&SIP5002&SIP5003,20,tr) > >> exten => s,2,Dial(SIP/5004&SIP5005&SIP5006&SIP5007,20,tr) > >> > >> and so on. > >> > >> If anybody has any insight, or a better solution, that would be great. > >> > >> Thanks, > >> > >> -Tor Roberts
I have a polycom IP600 for testing and will try it out if possible. Not sure at the moment if the phone as MGCP or SIP software on it. Btw, has polycom released the IP600, I was under the impression that it was still in testing. Umar. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Eric Mandel Sent: 14 June 2004 17:24 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Polycom IP 600 I am getting ready to install Asterisk and I was looking into the Polycom IP600 phones. I spoke with Polycom sales to verify the multiple line appearance and they said it would work. More specifically, if lines 1-3 all contain the same SIP registration info, the Polycom will only send out 1 SIP registration to the server and then handle the calls ringing on multiple lines. I was wondering if anyone can confirm that this works with the polycoms. I know the 7960s support this, but I want to make sure the Polycom sales team wasn't just saying Yes to make the sale. Any comments are appreciated. -Eric -----Original Message----- Subject: fwd on busy when calling multiple extensions at once Chris A. Icide wrote:> IMHO, the Polycom IP600 is a superior phone to the cisco 79XX phones. > I base this off of having had both an IP600 and a 7960. The two > advantages the 7960 had over the IP600 was appearance and ease of > configuration. Outside of that, the IP600 (IMHO) beat the cisco hands > down. > > Now, you MAY want to try registering all 6 lines on the polycom to the > same line and see if the phone handles that as well as the cisco. If > it does, then you are set. Otherwise, you will need some complex > configuration work in your extensions.conf to achieve what you are > looking to achieve. > > Some thoughts: > > What do you want to happen when one of the call takers has all 6 lines > in use? > > Have you considered using queues to do what you need? > > -Chris > > On 10:08 AM 5/22/2004, Brian Cuthie wrote: > > > >You might consider using the Cisco SIP phones. They're smart enough > >to accept incoming calls for as many call appearances you have with > >the same SIP registration. > > > >-brian > > > >Tor Roberts wrote: > > > >> Hi, > >> I am setting up a dispatch center where will have 4 call takers, > >> all with Polycom IP 600 Sip phones. Each phone will be setup with 6 > >> extensions each. When a new call comes in, the first extension on > >> all the phones will ring. This works fine, the problem is when one > >> of the dispatchers is already using her first extension and another > >> call comes in. What happens now is that the remaining 3 phones ring > >> on the first extension, but the dispatcher who is on a call, her > >> phone does not ring. I want her second extension ring along with > >> the other 3 phones first extensions. > >> > >> In sip.conf I have all the extensions set to incominglimit=1 and > >> the pertinent part of extensions.conf is: > >> > >> exten => s,1,Dial(SIP/5000&SIP5001&SIP5002&SIP5003,20,tr) > >> exten => s,2,Dial(SIP/5004&SIP5005&SIP5006&SIP5007,20,tr) > >> > >> and so on. > >> > >> If anybody has any insight, or a better solution, that would be great. > >> > >> Thanks, > >> > >> -Tor Roberts_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Eric, I tried this and could not get it to work. What I ended up doing is giving each button a different extension and then set the phone to "divert" to the second extension on busy. This is done with the "busy divert.busy.1.enabled=" and "divert.busy.1.contact=" tags in the phones config file. This has some drawbacks in that you have to make way more extensions, but you also have to use call waiting on each line. If there is a better way to do it, I would like to hear it. But this does get the job done. It would be great if you could just register all the lines to the same extension. -Tor Roberts Eric Mandel wrote:>I am getting ready to install Asterisk and I was looking into the Polycom >IP600 phones. I spoke with Polycom sales to verify the multiple line >appearance and they said it would work. More specifically, if lines 1-3 all >contain the same SIP registration info, the Polycom will only send out 1 SIP >registration to the server and then handle the calls ringing on multiple >lines. > >I was wondering if anyone can confirm that this works with the polycoms. I >know the 7960s support this, but I want to make sure the Polycom sales team >wasn't just saying Yes to make the sale. > >Any comments are appreciated. > >-Eric > >-----Original Message----- >Subject: fwd on busy when calling multiple extensions at once > >Chris A. Icide wrote: > > > >>IMHO, the Polycom IP600 is a superior phone to the cisco 79XX phones. >>I base this off of having had both an IP600 and a 7960. The two >>advantages the 7960 had over the IP600 was appearance and ease of >>configuration. Outside of that, the IP600 (IMHO) beat the cisco hands >>down. >> >>Now, you MAY want to try registering all 6 lines on the polycom to the >>same line and see if the phone handles that as well as the cisco. If >>it does, then you are set. Otherwise, you will need some complex >>configuration work in your extensions.conf to achieve what you are >>looking to achieve. >> >>Some thoughts: >> >>What do you want to happen when one of the call takers has all 6 lines >>in use? >> >>Have you considered using queues to do what you need? >> >>-Chris >> >>On 10:08 AM 5/22/2004, Brian Cuthie wrote: >> >> >>>You might consider using the Cisco SIP phones. They're smart enough >>>to accept incoming calls for as many call appearances you have with >>>the same SIP registration. >>> >>>-brian >>> >>>Tor Roberts wrote: >>> >>> >>> >>>>Hi, >>>>I am setting up a dispatch center where will have 4 call takers, >>>>all with Polycom IP 600 Sip phones. Each phone will be setup with 6 >>>>extensions each. When a new call comes in, the first extension on >>>>all the phones will ring. This works fine, the problem is when one >>>>of the dispatchers is already using her first extension and another >>>>call comes in. What happens now is that the remaining 3 phones ring >>>>on the first extension, but the dispatcher who is on a call, her >>>>phone does not ring. I want her second extension ring along with >>>>the other 3 phones first extensions. >>>> >>>>In sip.conf I have all the extensions set to incominglimit=1 and >>>>the pertinent part of extensions.conf is: >>>> >>>>exten => s,1,Dial(SIP/5000&SIP5001&SIP5002&SIP5003,20,tr) >>>>exten => s,2,Dial(SIP/5004&SIP5005&SIP5006&SIP5007,20,tr) >>>> >>>>and so on. >>>> >>>>If anybody has any insight, or a better solution, that would be great. >>>> >>>>Thanks, >>>> >>>>-Tor Roberts >>>> >>>> > > > > > > > > > > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >
Hello, the new Soundpoint 1.2 SIP release is on the Asterisk Polycom site for download: http://www.freedomphones.net/polycom/files/SoundPoint-IP_SIP_1.2.0.zip included with it is the new Admin Guide. Other older Polycom Soundpoint files are also available for download on the site: http://www.freedomphones.net/polycom/files/ Polycom Support will tell you that they don't send releases to non-certified partners, but the truth is that they are a hardware company and they will send you whatever releases you want if you either buy enough phones, go through a reseller that will support you(like ReviewVideo in Chicago) or get a contact far enough up the chain of command at Polycom to decide to give you the software. I was in talks with the VP of IP phone sales at Polycom for months and he said they were ready to work with the Asterisk community if a large reseller would step up to be the official reseller that they deal with in relation to the Asterisk community(they would even give us a contact in their engineering department). I was never able to get one of their resellers to agree to it without adding extra cost to the price of the phones. If someone is interested in taking up the cause and they have more free time than I do, send me an email and I'll let you know everything I know about it. Enjoy, MATT--- -----Original Message----- From: Brent Franks [mailto:mwless@mindworks.net] Sent: Monday, June 14, 2004 10:15 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Polycom IP 600> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Tor Roberts > Sent: Monday, June 14, 2004 8:22 PM > > John, > No, I have not tried 1.2, I did not know it was even out. Can this be > downloaded from Polycom's site? If so, I will try it out. > > -TorWe had e-mailed Polycom about 2 months ago requesting that the ADA compliance (auto-reset Volume per call) be a feature as per the ADA specs rather than a requirement. Polycom wrote back and said it is in the 1.2 release. Polycom will only release the release to certified partners.> John Baker wrote: > > >Did you try the new sip firmware update? The latest is version 1.2and> >has some fixes for what you're trying to do.John, do you care to share the new firmware? Unfortunately, like Tor, we do not have access to this either. What is the easiest way to get this release? When I call Insight (Where we purchased the phones) they have no clue what we are talking about. Thanks! - Brent _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Hi Tor, Thanks for trying. I opened up a case with Polycom today to look into this. I spoke with a tier 2 engineer. He didn't know the answer but was going to research it and get in contact with the polycom engineering team if necessary. I'll keep you posted on my findings. -Eric --__--__-- Message: 8 Date: Thu, 17 Jun 2004 12:29:39 -0700 From: Tor Roberts <voip@sscsinc.com> To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Polycom IP 600 Reply-To: asterisk-users@lists.digium.com Matt, I tried 1.2 out, but could not get multiple lines to register to the same sip channel. I am by no means an expert, so it is possible that somone else could figure out how to do it. Oh well, it would be nice if it worked. -Tor