similar to: Newbie Polycom: Instant Messaging

Just wondering if anyone has gotten instant messaging working between two polycom phones/and or MSN messenger with asterisk in the middle? I tried a few things but couldn't get it to work... Regards, Adam

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Is it possible to use the instant messaging feature of Polycom phones in Asterisk? At the moment I'm seeing this in the SIP messaging when I try to send one from a Polycom 450. <--- SIP read from UDP:<CENSORED POLYCOM IP>:5060 ---> INVITE sip:0100@<CENSORED>:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP <CENSORED POLYCOM

Can anyone explain the Polycom Text Messaging features built in to the IP 500/600? Can Asterisk (or something else) talk to it? I've seen vague references to MSN Messenger, and somehow that's mentally disturbing. Chris Coulthurst <mailto:chris@shuksan.com> chris@shuksan.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

When I downloaded the sip and bootrom from Polycom website, I noticed a file called SoundPointIPWelcome.wav. However, I have no idea where and when it was used. I played the wav file but I have never heard the phone using this wav file before. Does anyone know what it is used for?

I have a question about DHCP and boot server supporting more than 1 model of Polycom phones. According to Polycom "standards", Polycom phone boots up to get a DHCP address and at the same time getting a boot server string (with username and password) to logon to boot server to download SIP, bootROM and etc. That is okay if there is only one type of phone (that requires a specific SIP

I am using Polycom IP600 phone. If I call a phone which has DND (do not disturb) enabled, the message to the caller will be "The person on extension ... is on the phone, please leave a message ...". Is there a way to pick the "person ... not available" message instead?

I didn't know that instant messaging was a feature with the Polycom's. Do you have any documentation, how-to's, etc. that you can point me to? That would be just way too cool. Thanks; John -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Michael Englehorn Sent: Monday, January 12, 2015 12:19

I'm guessing that I need more info entered into the 'message centre' section. What did you key in? Paul Hales IT Support Adairs -----Original Message----- From: Jeff Pyle [mailto:jpyle490@gmail.com] Sent: Friday, 17 September 2004 11:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 I have two IP 500's on my Asterisk

I am working on Polycom IP601 console with expansion module. I want to put on the BLF (busy lamp field) feature on all the contact/speed dial names I put on the console but I could not get it to work. *CLI> core show version Asterisk 1.4.13 built by root @ hostname on a i686 running Linux on 2007-11-20 05:26:15 UTC *CLI> sip show subscriptions Peer User Call ID

Calling all Polycom gurus: I am using Polycom IP601 phones with Asterisk 1.4.21.2 In all Polycom phones, I set the following in sip.cfg. <dialplan dialplan.impossibleMatchHandling="2"> </dialplan> (I leave the digitmap unchanged because I thought setting impossibleMatchHandling will ignore the bitmap) ...so that I could dial any number by entering a variable-size

I have been testing with Polycom IP600 phones for a month or so. I found out that there are frequent problems with the handset. The problem is I can hear the other end but the other end cannot hear me. I have already downloaded the latest bootROM 3.1.3 and sip 2.1.2 However, there are no problems with the headset or speaker phone. Has anyone encountered such problems before? Thanks.

I was reading a Polycom brochure and it appears that there is really no special receptionist console and the console is basically a IP601. Is this correct? The only difference is to purchase an expansion module in order to have more shortcut keys for the girls. So, apart from the hardware, as far as the dialplan is concerned, do I just treat the receptionist console as any other extension? Are

hello, I try to set up two lines per ip 300 phone, registration is ok but i get Failure to authenticate 407 for subscribe. Anybody could help me to configure Asterisk in order to set instant message and presence ? I've tried with Ondo sip server it's ok ! Regards D?couvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Cr?ez votre Yahoo! Mail

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Any suggestions? Here is the console log:

I did not really spend too much time on looking at call forwarding and wonder if someone could help me. It seems that for setting call forwarding on the Polycom phone itself, only "forward all calls" will work. The other call forward function like "forward if no-answer for n rings" or "forward if busy" does not work at all on the phone. If that is the case, it

Hello, I am looking for documentation support for enabling instant messaging between endpoints using Asterisk 13.1.0 and vanilla VoIP clients such as Zoiper. Where do I enable this support on the server side and does it need anything on the client side? I see plenty of online help for chan_sip, but nothing for chan_pjsip. I imagine there is both pjsip.conf configuration and extensions.conf

Hi, I have a strange problem with all of our Polycom 550 & 650 phones. I am running a TFTP server on my Asterisk server and option 66 Boot Host pointing to Asterisk on my DHCP server. The auto-provisioning is working because the phones are registering correctly with their extension. If I change the MAC.cfg file to another extension and reboot the phone, it will reflect the new ext. The

I setup a number of remote phones on public IPs using the web interface. Now my question is how do I change the default Polycom:456 password via the web interface. Is there a hidden way or does it have to be done via FTP TFTP? Thanks, Steve Totaro

Hi, Is there ANY way for me to see the status of the Polycom DND buttons in the Asterisk hints? I`m using the BLF buttons to see the status of other people`s lines, and DND should logically be somehow reflected (I don`t care as much about Polycom showing the BLF button as DND, but I do care about Asterisk hints showing it in the CLI). Right now, a Polycom phone on DND shows up as being

Everybody, I've had a request from my manager that I figure out how to get our Asterisk 13.x system using chan_sip to be able to display on the Polycom VVX series phone display (firmware 5.9.5), when an extension is called and the person on the other end is on the phone. He said, "Our old Analog phone system could do it, how hard can it be?" I've gone down the path of trying