similar to: Polycom IP601 - some hints working, not others?

I have Teliax trunk set to ulaw and g729 and I have a modem/fax extension from a sipura forced to ulaw. When the call goes out through Teliax IAX trunk, asterisk transcodes to g729. Is there a way to tell asterisk not to transcode calls from/to a specific extension? I'm running asterisk 1.2.4 and that extension is for my home alarm/dish network and fax calls. Thanks -------------- next part

Hello people, I am having difficulties figuring out the POE power draw in watts from a Polycom IP601. I want to know how many IP601's can be powered from the Cisco Catalyst 3560G-48PS. The IP601 wallwart has: Input 120VAC 60hz 19W, Output 24VDC 500mA. I assume the output is appropriate value to figure out how many phones can be powered. The Cisco 3560 datasheet says "the 48-port PoE

Hi all, I am working on asterisk 1.2.18 zaptel 1.2.17 Polycom 650 polycom 430 SIP version 2.0.3.0131 for IP 650 SIP version for IP430 2.0.3.0127 freepbx 2.2.1 I am trying to configure BLF using asterisk but failed. I would be thankfull if somebody help me. Regards FArooq ********************************************** 1 ********************************************** in my

Hi, I configured Buddy Watch function on my Polycom IP 601. It works well, until I make a reload of Asterisk. After reload, if I give the "show hints" command in Asterisk's CLI, it says that there are no watcher for the extensions that I configured. Before the reload in the CLI appears: -= Registered Asterisk Dial Plan Hints =- 3002 : SIP/3002 State:

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

Make sure you set nat=yes for the sip user. Asterisk will then send replies back to the source IP address, rather than what's in the Via: header. > -----Original Message----- > From: Warren (mailing lists) [mailto:warren-lists@icruise.com] > Sent: Friday, October 27, 2006 5:45 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Taking a

Today is a strange day. My asterisk server is suddenly saying that all extensions are on hold. All my hints are like this: -= Registered Asterisk Dial Plan Hints =- 4101 at hints : SIP/4101 State:Hold Watchers 0 4100 at hints : SIP/4100 State:Hold Watchers 0 4002 at hints

Hey list, Im in this HUGE crisis. Im trying to get a Polycom 601 with two expansion modules to work. I need the XML config files I guess. Does anyone have these I can have? Im trying to get this phone up and running, and haveing MUCHO problems, can someone help me out!! Im sure if I see the configs I can see how it works, just need those XML files!! The ones from the 501 that I have dont seem to

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! Are hints not yet implemented in res_jabber? I have this here: exten => 3000,hint,gtalk/gtalk_account/mari.xxxxxxx at gmail.com But the hint doesn't show any difference. It always shows online on the phone and core show hints always shows that: 6003 at internal : SCCP/6003 State:Unavailable Watchers 0 6002 at internal :

Hi all, I've just updated to current CVS, and have 2 polycom IP phones, one is a IP600 and the other is a IP300. The IP600 shows the status of the IP300 and a ZAP line quite nicely, but the IP300 won't show the status of the IP600.... Is there any additional debug apart from "show hints" to see why this might not be working ?? -= Registered Asterisk Dial Plan Hints =-

Hi all, I am working on asterisk 1.2.18 zaptel 1.2.17 Polycom 650 polycom 430 SIP version 2.0.3.0131 for IP 650 SIP version for IP430 2.0.3.0127 freepbx 2.2.1 I am trying to configure BLF using asterisk but failed. I would be thankfull if somebody help me. Regards FArooq ********************************************** 1 ********************************************** in my

Group I'm having trouble getting hints to work correctly using SVN-branch-1.4-r59289 I have hints working on several other systems but I must be missing something this time around. VoIPGW*CLI> show hints -= Registered Asterisk Dial Plan Hints =- 30@default : State:Unavailable Watchers 3 29@default :

Version - Asterisk SVN-trunk-r12793M (1.2.4) I have 4 Grandstream GXP 2000 phones configured. However at the moment, I have had to disable BLF, Hints, and Call Limiting due to an extremely annoying bug which seems to make the phones channels "lock" in busy after a call has been hungup. If I do a show hints after say extension 200 has hung up I get the following -= Registered

Hi, Returning to my office I find two "missed calls" (from autodialers) that my IP601 displays as originating from 01111111111. However the CDR database recorded the call this way: calldate: 2008-04-04 14:18:16+02 clid: 0172752780 src: 0172752780 dst: 2131 dcontext: default channel: Zap/1-1 dstchannel: SIP/0146472131-007a7e80 lastapp:

One more Polycom IP601 question please (sorry for the long intro here to document) ... In order to closely approximate the behavior of the previous telephone system that many of the users are familiar with, I have set up call parking like this: - features.conf [general] section contains: parkext => ** ; What extension to dial to park parkpos => 10-11 ; What

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

I've got hints setup for my MeetMe conferences like so: exten => _60X,hint,MeetMe:${EXTEN} and they show up in "core show hints" like so 600 at dialtone : MeetMe:600 State:Unavailable Watchers 1 _60X at dialtone : MeetMe:${EXTEN} State:Unavailable Watchers 0 I'm wondering why they're Unavailable instead of

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.

In testing Asterisk 11, I've found that Asterisk doesn't seem to be sending BLF updates to SIP peers that have subscribed to a hint looking at a Custom device if that Custom device state is RINGING or RING_INUSE. All other states seem to be working correctly. The hint section of the dialplan is: [hints] exten => _3XX,hint,Custom:${EXTEN} Console shows the following for core show

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....