similar to: Anybody will to share a sm bus polycom dialplan to share?

Contact me off list, I have a sample extensions.conf file that I can share. It has Paging (one to one and One to Many) Ivr includes, time of da routing and it is geared towards Polycoms. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Thomas Johnson > Sent: Friday, January 20, 2006

Hello- We've got a patched-together extensions.conf that's barely working for us, and we need to get real about using Asterisk. We've got a couple of remote workers with Polycom IP-601 phones, and a single asterisk server, using a couple of incoming DIDs from teliax and sixtel. Does anyone have a good extensions.conf that they'd be willing to share, that provides a

This is not necessarily Asterisk specific but if I have Polycom 301/501 and 601s and want to dial a missed call back, how do I prepend a 9 - can I do this via the polycom config? I can't find anything in the docs. Bill -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have a bunch of Polycom 601's and Asterisk 1.4.13. The problem is that the MWI indicators will never go off (The blinking red light and envelope in the LCD). I have tried to upgrade to 1.4.14 and all different SIP versions on the Polycoms. I am now at 1.6.7 Here is the SIP Message that turns on the lights: Scheduling destruction of SIP dialog '

Does anybody happen to know the input power specs for the Polycom IP 500 and IP 600? We've mixed up our power supplies and we've got a whole box of them and can't figure out which go to the Polycoms. I would rather not kill the phones by trying random ones....

I am having a problem with some Polycom 601 phones. If I dial without picking up the handset or selecting the speaker I can dial numbers that are any lenght. But if I pick up the handset or are using the speaker I can only dial numbers that are 8 digits. When I dial the 8th digit it dials immediately. Obviously this creates problems when I am dialing long distance numbers or anything that

Wondering if anyone has experienced an intermittent one way audio (called party can not hear) problem in these conditions; Several IP501 phones local, same subnet. Remote asterisk No NAT anywhere Polycom IP501 ulaw only, canreinvite=yes Asterisk Call termination path is to a sonus GSX operated by the upstream carrier, ulaw only, canreinvite=no The idea is that if the Polycoms are

Hi everyone, I have been dealing with a certain issue with a particular customer site for months now. The problem occurs when there is an error with caller id as shown in the following: WARNING[16036]: chan_zap.c:6309 ss_thread: CallerID returned with error on channel 'Zap/3-1' When this happens, it appears that the call still goes through as I can see the caller still navigating

I recently had a bear of a time getting a Polycom Soundpoint 500IP up and registered.. Now that its registered I ran into a problem w/ the dialplan. Needing to dial x101 I'd dial 10 - then get a fast buzy.. Also making a local call - dialing 95551212- would give me a fast busy after the 7th digit - so 9555121.. Same w/ LD calls... This dialplan really got me down as I didn't find

Andrei, Do you have X-Windows running on the linux box? I had a similar issue that was eliminated when I stopped this process and samba from running. Now samba is only allowed to come up during non-business hours, for changing BG music. Also, make sure your registration period in either (polycom) ipmd.cfg or sip.cfg is set to be at least the default 3600 time period. I also removed the

I'm replacing a SoundPoint IP 600 with a SoundPoint IP 650. I attempted to simply reuse the existing config files for the old phone on the new phone, but the new phone would lock up on the 4th digit when attempted to dial out certain numbers. So, I downloaded the newest firmware and config templates from Polycom, and attempted to migrate the settings. Seems I'm missing something from

Hi, We are in the process of moving from an Avaya Definity to Asterisk for our institution's phone system. I got one feature that the Avaya had, which I have not been able to reproduce with Polycom phones and Asterisk; since this feature seemed so small and useless to me when testing, I kind of ignored it. Now I am getting more "I miss that" requests than I expected. =) On the

For the last few months we have intermittently been experiencing some very strange registration problems with certain polycom phones. Here is some background information: I have about 150 Polycom Soundpoint IP 600s, 601s, and 650s spread between 8 servers at different locations. Each phone is on the same network (and subnet) as the server it connects to. There is no NAT or anything else strange

Wiley, There are a couple of issues that we saw while not using this option. 1) sip authentication failures as Asterisk is not able to reach Polycom phones. A typical problem description is here: http://lists.digium.com/pipermail/asterisk-users/2004-December/079251.ht ml 2) DTMF issues for Transfers, Hold or simply to dial extensions. This problem is more pronounced when you are using

I am in the process of setting up an * system using Polycom IP 500's. I don't want to spend time setting a ftp server for application and configuration files at the moment, just want to use the web interface to the Polycoms. DCHP works OK and IP is obtained correctly. Polycom fails to load .cfg file and holts. I have read the 143 page admin user guide a couple of times...and I missing

I?ve been tearing my hair out trying to get a Polycom phone (behind a NAT) to register to an * box behind a Cisco SIP ALG. With known good credentials configured on the phone and in *, I get 403 Bad Auth when trying to register. If I put the phone onto the same LAN as * it works fine without changing any authentication parameters whatsoever. If I make the secret blank (null) on the phone and *,

Hello all, We have an Asterisk server that has worked without issue for a while. Before, only Sipura and Polycom 500 series phones were used. Recently, we've added a few POE switches and 20 or so Polycom 330's. The 330's seem to lock up often. One easy way to do this is by hitting the * key. I looked thru the XML files for dialplan and what not, nothing stood out.

I am exploring the contacts directory in Polycom and I am wondering if a big 000000000000-directory.xml on the boot server will eat up the memory and crash the Polycom phone once downloaded onto the phone. The asterisk directory extension is good but because users cannot see the names I thought to find another alternative. Does anyone have any experience?

I`m trying to provision my Polycom 501s using Pure-FTP. Everything has been working fine, until I put a new bootlog.ld file. Sniffing the packets between my phone and the FTP server during transfer, I noticed that the file is not 100% sent (it seems to send about a third of it, and then the phone wisely detects it as being corrupted and reboots). I have suceeded in using other FTP clients to

I'm using Polycom Soundpoint phones and I want to use some extensions beginning with # for features setup. I'm getting the fast busy "can't match it" signal. I want to match #50 for call forwarding, for instance, and #505551212 to set the call forwarding number and turn it on. I have tftp set up and sip.cfg contains the following: <dialplan