similar to: Got SIP response 400 "Bad Request" back from

Hi, I have a 1.6.2.6 fax installation with a FFA license which seems to be installed correctly (in fax show stats, I see that I have 1 Digium G.711 licensed channel, and 1 Digium T.38 licensed channel). When trying to call my business line with a fax machine, it looks like it's ringing to my asterisk box, then transfer the call to my extension. In the logs, I see (after the line where

Just installed a new SAMBA-LDAP-PDC server on my network. I followed the Linux Samba-OpenLDAP Howto, revision 20060710, so I would assume my setup is correct. My LDAP setup is a master/slave setup. The master is on a remote server, and the lave is local. When joining the domain (WinXP), it sometimes takes three-four times to be able to reach the domain(user administrator with uidNumber to

Hello, I'm new to asterisk and this list. The ISO download appears to have 1.6 with the FreePBX GUI but I am looking to use the Asterisk GUI. The only option for the Asterisk GUI is to use 1.4. Is it as simple as installing 1.6 only then using the yum repository to install the Asterisk GUI? If so, what packages are needed? Thanks!

Hello Rhelpers, I am trying to represent the following data (hereafter named donnees) in a barchart with a grouping variable : site traitement date s res 1 NT 17/10/2005 normal 76.2 1 T 17/10/2005 normal 103.2 1 NT 23/11/2005 tardif 81.6 1 T 23/11/2005 tardif 98 2 NT 15/10/2005 normal 72.71 2 T 15/10/2005 normal 94.47 2 NT 15/11/2005 tardif 79.65 2 T 15/11/2005 tardif 94.7

Are all of the sip phones in the same context? > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Robert Jenkins > Sent: Tuesday, January 16, 2007 1:44 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: [asterisk-users] Polycom IP601 - some hints working,

Is it possible to limit the imap/pop connections per user in dovecot? If so i couldnt find the option to do that. Thanks! -- Jordan Tardif DreamHost

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

Is there a way to disable this feature? Seems to be causing more harm then good right now. Without getting into too much details it is fixing the duplcates but it does not remove the old file, and the new file it creates is not marked as read like the old one was. I understand that this problem can be created by multiple rsyncs and files changing in between, but is there any way that I can

So after some problems with NFS and transaction logs getting corrupted. Im tyring out storing them localy. via.. mail_location: maildir:~/Maildir:INDEX=/var/dovecot/indexes/%u the problem is with a large amount of users.. the indexes dir becomes huge with this setup. So a change to make this better is... mail_location: maildir:~/Maildir:INDEX=/var/dovecot/indexes/%d/%n to seperate the users

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

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

So Ive been looking into some problems with users reporting their mail goes missing for a second but then re-apears very soon after. I have tracked it down to happening at the same time these appear in the logs. Seems the transaction logs is getting corrupted somehow and then rebuilding itself (which works fine). The problem is while its re-builing the users think all of their mail is gone.

How do you disable call waiting on Polycom IP601 phones? I've looked through the user and admin guides and can't see anything about disabling it. -Dan

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

What hardphones support xml/html/xhmtl/microbrowser? I need an inexpensive SIP hardphone that can run simple applications (queue status, etc). The phones I know of: Aastra 480i, 9112i, 9133i (though limited by 3 LCD lines on the 91xx seems kind of silly) Cisco 79xx Mitel 5235 Polycom IP601 Any others? -Dan

I use WFW and Samba Version 1.9.17p1, on Sun Solaris 2.4 I want to use remote shell command from my PC Can you help me best regards -------------- next part -------------- HTML attachment scrubbed and removed

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.

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

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