Re: [sipx-users] POLYCOM configuration - display FIRST NAME of user

in the Polycom template where I would put this information. I don't want to update each phone individually to accomplish this. Thanks! -Original Message- From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] Sent: Monday, May 24, 2010 6:23 PM To: dhub...@ezuce.com; IT Services Cc

Re: [sipx-users] PSTN delay

* * * * * -Original Message- From: Jim Canfield [mailto:jcanfi...@emstar.com] Sent: Monday, May 24, 2010 5:22 AM To: IT Services Cc: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] PSTN delay On Fri, May 21, 2010 at 12:49 PM, IT Services itservi...@apiwellness.org wrote: I'm using sipxecs 4.2

[sipx-users] POLYCOM configuration - display FIRST NAME of user

Hi there: Sipxecs ver 4.2 Polycom phones As the default, the polycom phone display shows the extension number. I want to display the first name of the user. I've created a phone group for the polycom 430s. In the Registration settings page, the field LABEL is where I can make the change. But I

[sipx-users] PSTN delay

Hi all: I'm using sipxecs 4.2 with patton PSTN gateways and polycom phones. When I make an outgoing call, there is a delay of 5-8 seconds of silence before I hear the call being made. Is this typical? Does it take this long to connect to the patton gateways? Thanks for your help! Notice of

[sipx-users] Some Polycoms not Picking up Bootrom Upgrades

Hi there: I'm running sipxecs 4.2 with a mixture of polycom models (301, 430, 501). I'm upgrading the bootrom to 4.x but some polycoms pick up the upgrade and others do not. I format the file system to do a clean install but it still doesn't work. Any ideas? Thanks... Notice of

[sipx-users] Ploycom Phones - Time/Date is 12/31 4:00pm

Hi all: I have a problem with some polycom phones not picking up the correct date/time. I have a set of polycom 501s with half displaying the correct time and date, and half not showing the correct date/time (but with flashing 12/31 4:00pm instead). The phones have been reformatted so that means

Re: [sipx-users] Ploycom Phones - Time/Date is 12/31 4:00pm

thoughts? Thanks... -Original Message- From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] Sent: Friday, April 23, 2010 4:49 PM To: IT Services; sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Ploycom Phones - Time/Date is 12/31 4:00pm There is a field in sipxconfig which you can

[sipx-users] How can you send a voicemail to a group

Hi there: On our old PBX system, we were able to send a voicemail to a group extension where each member in that group received the voicemail in their box. How do I set that up in sipx? For example, a user is in voicemail and records a voicemail to be sent to all managers. The extension we use

Re: [sipx-users] Issue with connecting phones to 8-port hub

13, 2009 5:51 AM To: Dale Worley; IT Services Cc: sipx-users@list.sipfoundry.org Subject: RE: [sipx-users] Issue with connecting phones to 8-port hub Dittos... shouldn't be a problem. Although hopefully this is a switch and not a hub :-) -Original Message- From: sipx-users-boun

[sipx-users] VLAN question

Hi all: I want to VLAN our network to put data on VLAN1 and the phone system on VLAN2. we have Cisco 2950 switches. Is it possible to create VLANs if our PCs are connected to the phones which then connect to the switches (i.e., one port for a phone and PC)? Or must I run two network cables and

[sipx-users] Forwarding from Patton PSTN Gateway

Hi there: I am using 4.0.2 with a Patton 4114 PSTN gateway. I used the config as suggested with the gateway forwarding incoming PSTN calls to the auto attendant at extension 100. I want all incoming calls to go to extension 301 (receptionist) but when I change the extension from 100 to 301,

[sipx-users] SNOM phones not registering

Hi there: I am running 4.01 with windows DNS. There are SRV records. The SNOM 320 phones are not registering. The phones pick up the config file, but fail to register. In the logs, there are these entries: [5]24/8/2009 16:41:21: sip::process_auth:Match challenge for user=341, realm=apiwc.local

[sipx-users] 4.01 ISO install - receive 'spixecs-release package not found'

Hi all: When installing on an HP Proliant RAID 1, I get this message: 'Sipxecs-release package not found.' Abort or continue. I can continue and install, but I feel something is not getting installed. Any thoughts? tommy Notice of Confidentiality: **This communication and any of its

[sipx-users] Phones not Restarting

Hi all: I am using 4.01 from a CD install on a single server. When I send a new phone profile, the phones (snom 320 or LG-6812) do not restart automatically and hence, I get a failed message in the job status area. Any suggestion on where to start troubleshooting? Thanks... tommy Notice of

Re: [sipx-users] Low Volume on PSTN calls

) that can increase only the volume to the IP phone? Thanks for your help! tommy -Original Message- From: Picher, Michael [mailto:[EMAIL PROTECTED] Sent: Friday, October 31, 2008 1:44 AM To: IT Services; sipx-users@list.sipfoundry.org Subject: RE: [sipx-users] Low Volume on PSTN calls

Re: [sipx-users] Low Volume on PSTN calls

31, 2008 3:38 PM To: IT Services; [EMAIL PROTECTED]; sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Low Volume on PSTN calls No. That is the function of the phone itself. Once the gateway sends the calls into your network it is digital. Gain is a function of the analog gateway before

[sipx-users] Low Volume on PSTN calls

Hi all: I'm using 3.10.2 with audiocodes gateways. Internal calls are fine in terms of volume. But incoming and outgoing calls via PSTN have very low volume. Is there anything I can do to increase the volume on these calls yet maintain the volume on internal calls? Thanks! Notice of

[sipx-users] Cannot connect to Audicodes MP-11X gateways

Hi all: I'm still trying to connect to audiocodes gateways. I can connect to a Grandstream 4108, but need to use another gateway so I got audiocodes. From the logs, the one difference between a successful connect and the unsuccessful connect is a 404 error: Sipx server: 192.168.1.49 Audiocodes:

Re: [sipx-users] phones/gateways do not restart via sipxecs

[mailto:[EMAIL PROTECTED] Sent: Tuesday, October 21, 2008 12:31 AM To: IT Services; Tony Graziano; sipx-users@list.sipfoundry.org Subject: RE: [sipx-users] phones/gateways do not restart via sipxecs If they are not restarting, I would suggest that either they are on really old firmware or your

[sipx-users] Having problems connecting to AudioCodes MP-118

Hi there: I followed the wiki to set up a MP-118 FXO but it's not working. I created the gateway in sipxecs and assigned a static IP to the audiocodes box. I uploaded the .cpm and .ini files successfully. When I make a call, I eventually get a fast busy signal. In the logs, it does show sipx

[sipx-users] phones/gateways do not restart via sipxecs

Hi there: I notice that when I issue a 'Send Profile' command, the phone units (and gateways) do not receive the 'restart' command from sipxecs. In the 'Job Status' section, it says that the restart command was issued successfully, however. Any help is appreciated. Thanks! tommy Notice of

Re: [sipx-users] phones/gateways do not restart via sipxecs

Thanks for the tip. But even the managed phones (lg Nortel 6812, snom 320) do not restart after I issue the 'send profiles' commands. Any thoughts? tommy -Original Message- From: Tony Graziano [mailto:[EMAIL PROTECTED] Sent: Monday, October 20, 2008 4:26 PM To: IT Services; sipx-users

Re: [sipx-users] Call Forwarding to External Number via PSTN not Working

. Miller [mailto:[EMAIL PROTECTED] Sent: Friday, September 26, 2008 5:25 AM To: IT Services Cc: sipx-users@list.sipfoundry.org Subject: RE: [sipx-users] Call Forwarding to External Number via PSTN not Working I'm the one who posted re Grandstream gateways using the To-URI rather than the Request

[sipx-users] Call Forwarding to External Number via PSTN not Working

Hey there: I am configuring 3.10.2 with PSTN lines. I'm testing two gateways: Linksys 3102 Grandstream 4108 Both gateways were virtually plug-and-play with respect to receiving external calls and placing outbound calls. No problems there (except for low volume, but that's a different issue).

Re: [sipx-users] Callresolver not installed - 3.10.2

by using the 32-bit OS? -Original Message- From: Tony Graziano [mailto:[EMAIL PROTECTED] Sent: Thursday, August 14, 2008 2:41 AM To: IT Services; sipx-users@list.sipfoundry.org Subject: RE: [sipx-users] Callresolver not installed - 3.10.2 From the logs, it seems that a file is missing

Re: [sipx-users] Callresolver not installed - 3.10.2

Hey tony: Thanks for all of your help and suggestions. I will revert back to the ISO as I realize that if it works, it works! tommy From: Tony Graziano [mailto:[EMAIL PROTECTED] Sent: Thursday, August 14, 2008 12:23 PM To: IT Services; sipx-users

Re: [sipx-users] Callresolver not installed - 3.10.2

[mailto:[EMAIL PROTECTED] Sent: Saturday, August 09, 2008 3:58 AM To: IT Services; sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Callresolver not installed - 3.10.2 What is your output from: [EMAIL PROTECTED] ~]# rpm -qa | grep -E sipx You should have, sipxconfig-ftp-3.10.2-013143

[sipx-users] Callresolver not installed - 3.10.2

hi there: for some reason, sipxcallresolver isn't installed. i installed centos 5.2 64-bit and then did a 'yum install sipxecs'. the only thing not installed is callresolver. here is a partial log: Aug 8 15:23:35 sipx sipXpbx: sipXpbx configuration problems found: Aug 8 15:23:35 sipx

Re: [sipx-users] Call Forwarding Not Working

:[EMAIL PROTECTED] Sent: Monday, July 07, 2008 12:20 PM To: IT Services Cc: Tony Graziano; sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Call Forwarding Not Working On Mon, 2008-07-07 at 09:41 -0700, IT Services wrote: *** log files listed below! Remember -- when sending log files, do

Re: [sipx-users] Call Forwarding Not Working

An update: 1. the scenario is this: Ext 343 has calls forwarded to an external number 793-4068. Ext 355 dials ext 343. gets failed call. Gateway and SIPX trace is listed below. Every ext has mobile turned ON. 2. version: sipXconfig (3.10.1-012233 2008-04-08T22:39:19 ecs-centos5) 3.

Re: [sipx-users] Call Forwarding Not Working

I did upgrade the version to current so it doesn't seem to be that. For some reason, it seems that sipx is not relaying the 'forward-to' number to the gateway. The sipx-trace below shows the external number (793-4068) being routed but then it switches back to ext 343. therefore, the gateway

Re: [sipx-users] Call Forwarding Not Working

To: IT Services; [EMAIL PROTECTED] Cc: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Call Forwarding 1. Please use a subject that is consistent and helps indicate your need. I'm sure it's an oversight, but it really does help. 2. The gateway is forwarding calls from 336 to 343 (an internal

Re: [sipx-users] Call Forwarding Not Working

way more than I, so I appreciate this a lot! tommy -Original Message- From: Tony Graziano [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 02, 2008 4:17 PM To: IT Services; [EMAIL PROTECTED] Cc: sipx-users@list.sipfoundry.org Subject: RE: [sipx-users] Call Forwarding Not Working It's most

Re: [sipx-users] sipx-users Digest, Vol 52, Issue 61

Hey there: I'm having a problem with call forwarding. If I list an external (PSTN) number, the caller gets an '...if you want to try your call again, please hang up and dial again.' If I list an internal extension, the caller gets the voicemail of the original extension. I'm stuck right now so

[sipx-users] converting a lg-nortel 6812 phone from MGCP to SIP

Hi there: I have one LG-Nortel 6812 phone that is configured for MGCP. Is there a way to convert to SIP? Thanks! tommy Notice of Confidentiality: **This communication and any of its attachments is intended for the use of the person or entity to whom it is addressed and may contain

[sipx-users] Dial by Name - How can i use First Name ?

Hi all: The dial by name feature works great, but I would prefer if Dial by Name is by FIRST NAME rather than last name. How do I modify this feature? THANKS! ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: