similar to: polycom overlap dialing?

Where's the setting for overlap dialing with Polycom IP601? -Dan

Hey All, This is more a polycom question, but involves how to make it specifically interact with asterisk... I have setup a special extension in asterisk which toggles whether our "Night Service" is turned on. However, I would like to have 2 things happen on our Polycom 501 (or 601)s. Where there are buttons which say "New Call" and "Forward", I would like to add a

Greetings, Has anyone noticed that attempting to place a call from the "Placed Calls" list on a Polycom IP501 by pressing the 'Dial' softkey sometimes simply returns the phone to the idle screen? It is not related to the number being dialed, as we have observed two entries for the same number, one of which worked and the other didn't. We've experimented with calls

I'm evaluating using Polycom phones for our call center and I've set up my first phone (a SoundPoint 560) to give it a try. The phone is working and can successfully place and receive calls. But every minute, there's an error in the log file: chan_sip.c: Registration from '<sip:6193644850 at jtsd05>' failed for '192.168.200.99' - Username/auth name

I'm having a problem with polycom ip601. If I Dial() directly eg Dial(SIP/4000) it works perfectly. The polycom rings, and stops ringing as soon as I hang up. But if the phone is called via a queue, the polycom continues to ring long after I've hung up. Other phones in the queue (grandstream, cisco) don't have this problem, and stop ringing properly when I hang up. polycom bug

Hi All - I noticed that the Polycom IP501's are now shipping. Has anyone gotten one yet, and if so, what's different about the phone? Any UI improvements, or is it just better hardware? Thanks, Noah

Hi - Has anybody been able to successfully remap SpeedDials on Polycom phones? The manual seems to indicate that you can, and I followed the advice in this list message: http://lists.digium.com/pipermail/asterisk-users/2005-October/129142.html The result I get is that the remapped buttons act as the VolumeUp button, rather than as a speed dial. My configs look like this: <Mac

Hi, I am lost trying to figure out how to enable overlap dialing for calls coming in and coing out on a pri span. DISA looked promising at first, but does not seem to support overlap dialing. Just picking up a call by and trying to dial out does not seem the way to do it either. I tried: [dialincontext] exten => 12341234,1,Goto(dialoutcontext,s,1) [dialoutcontext] exten => s,1,Wait,1

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

Now i'm receiving this and my phone no longer can dial out... ERROR[4391]: chan_sip.c:11169 handle_request: Missing Cseq. Dropping this SIP message, it's incomplete. I'm having to use the configs that came with the zip because apparently my previous configs no longer are valid and lock the phone from dialing with a url disabled message... anyway, these polycom phones are driving

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Noah Miller > Sent: Wednesday, April 25, 2007 9:52 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] My Polycom IP 501 is formatted its file > systemitself > > Hi Chandra - > >

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"

hello all i want to have overlap dialing in asterisk. it works fine if i don't have "!" in my pattern. for example pattern "07." works fine and i can call 07122 by overlap dialing via it. but if i define "07!" i can't call 07122 because it doesn't wait to collect all digits and therefore call 07. if i define "07.!" it works fine again and

I have four Polycom 330 phones connected to an asterisk system. There are other VoIP phones connected too. All of the extensions are four digits beginning with 11. From any of the phones, except the Polycom, picking up the handset to call extension 1103 for example works fine. With the Polycom 330, as I press the second 1 of 1103 it stops taking input and gives me an error. I tried

Hi - Just wondering if anyone has gotten a Polycom VVX phone to successfully do an Auto Answer with asterisk. I have an older generation of Polycom phones that do this just fine, but I can't seem to make the VVX phones work. I tried the guide here: http://community.polycom.com/t5/VoIP/FAQ-How-can-I-change-my-Ringtone-or-Ring-in-a-special-manner-for/td-p/5167 And I have this in my diaplan:

Hi Andres - > I have a small setup with 2 SPA3000 1 SPA2001 and 1 Polycom 301 > > The Polycom misses 1 out of 2 dialout calls, this is the full log > from a > call which didn't go through. > > 303094 Sep 14 10:45:15 DEBUG[15073]: Stopping retransmission on > 'f4e376c3-7531ff39-c86f6812@192.168.1.18' of Response 2: Found > 303095 Sep 14 10:45:15

This is a dialpaln issue. I solved the same problem recently. For 4 digit extensions you need to append the dialplan statement in the sip.cfg configuration file as follows <digitmap dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2 -9]xxxT|1xxxT" dialplan.digitmap.timeOut="3"/> Michael > I am not sure what I did but blind transfers do not

Hello, I just wiped out my old asterisk install and installed Asterisk at Home. I was quickly able to get my Digium TDM422P working, 2 POTS lines, 2 phones. I also got X-Lite working as a SIP extension. I then tried to setup my Polycom IP 500, and this was not so easy... Using AMP I created SIP extension 205 to be used with my Polycom phone. I setup username = 205, secret = 123, context =

via google, I found the reference regarding digit maps for the Polycom phones: http://lists.digium.com/pipermail/asterisk-users/2005-January/082884.html But I don't see any instruction for prepending digits to the number dialed. Does anyone know how to prepend a digit to the number dialed (from the Polycom side, not Asterisk)? I can do this pretty easily on a Sipura. i.e. Say I want to

Hi, all. I want to have a button on my receptionist's 601 that, when pressed, will forward her current call to a given extension. Is there any way to do that? I've tried to RTFM, but I'm coming up empty. Thanks, -Ken D'Ambrosio