Re: [asterisk-users] PRI Splitter

Adtran Atlas 550. We were bring in a single pri into an atlas 550 and then splitting it up so that 6 channels went to a video system (h.320) and 17 channels to our PBX. You can also convert the signaling or send out on different type of connections like v.35. Pretty cool device and rock solid.

Re: [asterisk-users] Polycom LDAP Corporate Directory

it. I have quite a few polycoms and didnt even know polycom had this feature! :) This is obviously a separate peice of software that must be purchased and installed on the phones. Looks amazing though- any idea on pricing?. On Fri, 2008-04-18 at 14:53 -0400, Anciso, Roy wrote

[asterisk-users] Polycom LDAP Corporate Directory

Anyone use the LDAP feature yet on the polycom phones? If so how well does it work? How are you using it in your environment? http://polycom.com/usa/en/products/voice/desktop/soundpoint_ip/applicati ons/corporate_directory_access.html Roy Anciso Director of Technology Manistee Intermediate

Re: [asterisk-users] Hardphone SIP phone costs

I understand the maximizing pricing and branding aspect of phones but when you look at feature set it just doesn't make sense. And as far as purchasing the phone you can get it without a contract at the same price. When I starting thinking about it, can anyone else see a time when desk phones

[asterisk-users] Hardphone SIP phone costs

I'm trying to understand something that just doesn't seem to compute. How can companies like Cisco justify selling their hard phones for as much as they do? I know there is a matter of recouping RD costs but when you look at the iPhone with all its amazing features for less than $500.00 it just

Re: [asterisk-users] Druid Open Source Edition

Is there a limit on how many phones you can use? I couldn't find anything on the website about this. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua Wilson Sent: Wednesday, March 12, 2008 1:43 PM To: asterisk-users@lists.digium.com

[asterisk-users] Linksys SPA-942 Phones

Hello List, After seeing a few positive responses for the Linksys SPA-942 phones I was hoping to get some answers on the following questions: * How do the phones handling system wide paging? Is it similar to the Polycom phones? * Can a corporate directory be configured with the phones

[asterisk-users] Script for seeding polycom phones with an extension directory

Hello List, Not sure if this will be helpful but I made changes to the original Cisco directory.php.txt script and applied them for use on the Polycom phones. This will create an extension directory and alphabetize it based on the sip registrations you have setup in sip.conf. Note that this

[asterisk-users] Followme

I've been reading up on followme app for asterisk 1.4 and I have it working but I was wondering if the following was possible: Based on followme.conf present the caller with the option to locate the person: Call comes in (external or internal) and rings extension with followme configured.

Re: [asterisk-users] Followme

the followme app. On Jan 22, 2008 10:25 AM, Anciso, Roy [EMAIL PROTECTED] wrote: I've been reading up on followme app for asterisk 1.4 and I have it working but I was wondering if the following was possible: Based on followme.conf present the caller with the option to locate the person: Call

Re: [asterisk-users] Voicemail - is it possible to automatically usethe extension being dialed from?

Heres what I do for this: exten = *85,1,VoicemailMain(${CALLERID(NUM)}) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of arkda Sent: Tuesday, January 22, 2008 5:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject:

Re: [asterisk-users] cisco ip phne 7911G with asterisk

Of Christian Pinedo Zamalloa Sent: Friday, January 18, 2008 8:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cisco ip phne 7911G with asterisk On Wed, Jan 16, 2008 at 10:26:04AM -0500, Anciso, Roy wrote: Now that you have your 7911g phone up running, would

Re: [asterisk-users] cisco ip phne 7911G with asterisk

Now that you have your 7911g phone up running, would you mind checking the audio quality when leaving a voicemail for on another local asterisk user from this phone? I have a 7911g and I hear loud audio taps from the phone. The 7961g phone doesn't have this issue. I'm just trying to rule out the

[asterisk-users] Record calls then send them to users voicemail

Just wondering if this is possible: Make a call from a registered sip extension (Doesn't matter if it's internal or external) during the call press a key sequence let say *90 to start recording call. When the call ends the recording automagically goes to their voicemail. Thanks Roy Anciso

Re: [asterisk-users] Cisco 79xx XML services

Although it's not LDAP I used a script that I found on the voip wiki and changed it so it looked at only sip configuration files. It also alphabetizes the output so it can be displayed that way on the phone. Below are my notes on the subject. If someone is willing to post this to the wiki and

Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee

I've upgraded from SCCP to SIP 8.x.x branch on 7961g and 7911g without any problems. As far as the CTLSEP File (Straight from Cisco): http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/i pp7960/addprot/mgcp/frmwrup.htm#wp1047292 The CTLSEP MAC file is a certificate trust

Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee

NO CTL and CTL processing FAILED** ctl-err 12 (file is too small) NOT 09:29:03.227508 DHCP: Restart - delay = 1 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anciso, Roy Sent: Friday, January 04, 2008 8:43 AM To: Asterisk Users Mailing List - Non

Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee

and CTPSEP odyssee That is not the name the phone requests When uping my 7960, the empty file did the trick I so far am unable to go beyond 7.1 however, as Asterisk rejects anything I dial with 7.3 Anyone have SIMPLE sample config files? John Novack Anciso, Roy wrote: Try naming the empty

Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee

:02.696335 SECD: EROR:updateCTL: ** had NO CTL and CTL processing FAILED** ctl-err 12 (file is too small) NOT 09:29:03.227508 DHCP: Restart - delay = 1 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anciso, Roy Sent: Friday, January 04, 2008 8:43

Re: [asterisk-users] 7970 CTLFile.tlv?

I believe you can create a blank file to keep the phone from complaining. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Rubenstein Sent: Friday, December 21, 2007 10:16 AM To: Asterisk -Users Subject: [asterisk-users] 7970 CTLFile.tlv?

Re: [asterisk-users] Cisco 7961 new firmware stops readingconfiguration files

Chad, You might want to upgrade to the latest firmware. I have 7961g on 8-3-3SR2S and works very well. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chad Osmond Sent: Thursday, December 20, 2007 10:33 AM To: asterisk-users@lists.digium.com Subject:

[asterisk-users] Cisco 7911g Poor Audio Quality w/ Asterisk Voicemail and MOH

For those using Cisco 7911g phones, I am running into an issue with one the Cisco demo phones we have. The 7961 works great with asterisk no problems However, the 7911g gets audio clipping when recording voicemails or the unavailable message. Also when a call is transferred using the 7911g the

[asterisk-users] Play Beep instead of MOH

Is there a way to tell asterisk to beep every few seconds rather than play MOH. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED]

[asterisk-users] Digium TE120P versus Sangoma A101D-X

Hello List, We purchased a TE120P card from Digium and it works great. The only problem is that we are still experiencing echo on some calls. I've tried various echo cancellers (right now we are using OSLEC) and still no luck. My question has anyone gone from the TE120P to a Sangoma

Re: [asterisk-users] Music on Hold Problem w/ Transfers

I'm having this problem. Here is my output with verbosity on 10: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/2524-099012b0, SIP/2523|15) in new stack -- Called 2523 -- SIP/2523-09905220 is ringing -- SIP/2523-09905220 answered SIP/2524-099012b0 -- Packet2Packet bridging

Re: [asterisk-users] Music on Hold Problem w/ Transfers

Asterisk version 1.4.13 Also when I listened in on a transfer it sounds like the moh is trying to start but then immediately stop and tries to start again. Below is my musiconhold.conf: [default] mode=files directory=/var/lib/asterisk/moh random=no -Original Message- From: [EMAIL

[asterisk-users] Cisco phones and 32 directory object limit

Hello List, For those of you with Cisco phones and XML directories and large user bases, how do you handle the 32 directory object limit? I know you can create multiple xml files with 32 objects in each but this just seems really sloppy. I would like to have one large directory. Thanks

[asterisk-users] Page Command

Hello List, I'm looking at the page command. I was wondering if there was a way to set a wild card to dial all registered sip devices. For example page all 1XXX extensions. Thanks in advance Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road

Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files

Sorry forgot the images: http://picasaweb.google.com/ranciso/AsteriskImagesCiscoPhones From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anciso, Roy Sent: Thursday, November 15, 2007 3:47 PM To: Asterisk Users Mailing List - Non-Commercial

Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files

7911/7941/7970/7971 Softkey XML Files 2007/11/15, Greg Oliver [EMAIL PROTECTED]: On Thu, 2007-11-15 at 07:31 +0100, Olivier wrote: 2007/11/14, Greg Oliver [EMAIL PROTECTED]: On Tue, 2007-11-13 at 08:44 -0500, Anciso, Roy wrote: Hello List, Does

Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files

, 2007-11-13 at 08:44 -0500, Anciso, Roy wrote: Hello List, Does anyone have access to the soft key configuration files for the Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and didn't find much up there. Thanks Softkeys running both SCCP and SIP firmware are both sent

[asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files

Hello List, Does anyone have access to the soft key configuration files for the Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and didn't find much up there. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI

Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files

To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files Anciso, Roy wrote: Hello List, Does anyone have access to the soft key configuration files for the Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site

Re: [asterisk-users] Cisco IP Communicator with Asterisk

The version I have is 2.1.2.0. It makes for a really nice software sip phone:) The other thing I should note is that you only need the SEPXXX.cnf.xml file and dialplan.xml file in your tftp directory. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of

[asterisk-users] Cisco IP Communicator with Asterisk

I'm not sure if anyone has done this before or not but, I was able get the Cisco IP Communicator soft phone to work with Asterisk using SIP. Thought I would share my experiences. The key is on the installation. To have the software use the SIP protocol type the following command: msiexec /i

[asterisk-users] Selecting OSLEC for zaptel-1.4.6

Hello list, Can someone outline the steps for selecting OSLEC canceller in 1.4.6? I know there was a bug fix for this but I can't figure out how to select it. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph:

Re: [asterisk-users] Selecting OSLEC for zaptel-1.4.6

] Selecting OSLEC for zaptel-1.4.6 Dave Fullerton wrote: Anciso, Roy wrote: Hello list, Can someone outline the steps for selecting OSLEC canceller in 1.4.6? I know there was a bug fix for this but I can't figure out how to select it. snip / Roy Anciso You shouldn't need to. As long

Re: [asterisk-users] Outgoing PRI CID?

I do this to tie extensions to a particular number: exten = _9X./_2XXX,1,SET(CALLERID(all)=Manistee ISD2317231516) exten = _9X./_1XXX,1,SET(CALLERID(all)=MISD Tecnology2317234264) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Turbo Fredriksson Sent:

Re: [asterisk-users] Cisco Phones

That is correct. Our Cisco rep is sending us a 7911G and 7941G so we can test with asterisk. We plan on converting them over to SIP for testing. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo with Horan Company, LLC Sent: Thursday, October 25,

[asterisk-users] Cisco Phones

For those of you running Cisco phones, did you start out with a Cisco CallManager and move to Asterisk? And if you did switch do you find that you or your users are missing features they once had? How have you handle the issue? Thanks Roy Anciso Director of Technology Manistee

[asterisk-users] Cisco phones with Asterisk

Hello List, For those of you using Cisco phones, did you have to purchase a 'SIP license' for each phone? Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED]

[asterisk-users] What web GUI are people happy with?

Just wondering what web GUI people like for asterisk. I installed asterisk from source and I was looking at possibly installing web GUI for system management. So far freepbx.org looks promising anybody else have any suggestions. Thanks Roy Anciso Director of Technology Manistee

Re: [asterisk-users] What web GUI are people happy with?

: [asterisk-users] What web GUI are people happy with? Anciso, Roy wrote: Just wondering what web GUI people like for asterisk. I installed asterisk from source and I was looking at possibly installing web GUI for system management. So far freepbx.org looks promising anybody else have any