[sipx-users] Cisco MWI - Fixed?

I've had 7960's work fine with MWI, however the 7941 and 7970 don't, I've seen some comments in relation to this dating back a few years, however I have not seen a clear fix to the problem, is it fixed for newer Cisco phones? has anyone got the MWI working properly on a 7941 or 7970? Regards, Be

[sipx-users] SRTP

I have noticed entries in the phone configs for SRTP. Is this supported in sipXecs? If I wanted to geek-out and turn this on somehow, any pointers to what is needed to make it work? Thanks s ___ sipx-users mailing list sipx-users@list.sipfoundry.org

Re: [sipx-users] may i know how to configure this pennyTel to my new sip gateway?

Under permissions, does this caller have permission for this dial plan - local dialing? You have indicated 8 digits for the resulting number. So are you really dialing 00603xxx for your call, or is it 00603 I'd confirm this section of our dial plan. To troubleshoot this, cl

[sipx-users] may i know how to configure this pennyTel to my new sip gateway?

I looking for some sip trunk provider just purely make outbound call, Because for inbound we already have our PSTN E1 PRI line to do the receive call. So i check for this pennyTel actually is cheap and near by our place (i ping just 150++ms) 1) i register one account from PennyTel, 2) create

Re: [sipx-users] G-Tek phone keep "hanging" because is my codec problem?

Thank all of you to reply, Actually i already get some Idea and change it is can work properly now, I check for this G-tek  adapter actually just only 6watt. I have 35 same model phone under deference Vlan., (base on department) FD = Phone - 192.20.102.* , PC - 192.20.2.* MG = Phone - 192.20

Re: [sipx-users] G-Tek phone keep "hanging" because is my codec problem?

Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum In-Reply-To: <4b6164dc.7020...@elabram.com> X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <41086> Message-ID: Power requirements are also affected by run lenght. Limiting your cisco to suppl

Re: [sipx-users] call disconnects when listening to saved messages

Seems to be the case. Maybe another ticket needs to be opened in Jira. -Original Message- From: gabriel [mailto:g...@bayintegrated.net] Sent: Friday, January 29, 2010 5:29 PM To: Todd Hodgen Cc: sipx-users@list.sipfoundry.org Subject: RE: [sipx-users] call disconnects when listening

Re: [sipx-users] call disconnects when listening to saved messages

so this works if the saved messages are from an internal extension. If the messages are from outside, the call drops. these are examples of the 2 XML files associated with the WAV files. 2...@sipx.company.net "Conference Room" 5 Fri, 29-Jan-2010 04:30:28

Re: [sipx-users] New installation registers but can't call

Is it an option to put it on its own subnet and let your sipx server be the DHCP and DNS server? It will make the process easier. If you are failing those tests, you are going to have issues. Sent via BlackBerry from T-Mobile -Original Message- From: Ben Cecka Date: Fri, 29 Jan 2010 15

Re: [sipx-users] New installation registers but can't call

Can you be more specific about what tests failed? Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Cu

[sipx-users] New installation registers but can't call

Hello, I've been working with sipx for a week or so now with various installation configurations. I am finally able to register several softphones to it, but when I try to call between them or to a queue extension I get a 'Destination not found' message. I'm thinking this could be a DNS or DHC

Re: [sipx-users] call disconnects when listening to saved messages

I found that when I pulled two specific messages out of my voicemail, the problem went away. It has worked ever since I did that with no problems. Running the same software as you as well. Open Winscp, pull all the voicemails for that box out and try it. Then put them back in one at a time until

Re: [sipx-users] call disconnects when listening to saved messages

The system is currently running 4.1.0-017444 installed from the ISO (centos) and I think it got updated once about 3 weeks ago. the reason I decided to go with the dev version was that I haven't had any issues with it so far - looks like I do have one now :) I was trying to find a workarou

Re: [sipx-users] call disconnects when listening to saved messages

I found that by deleting the conference recordings it worked for me. I think it had to do with how they were marking something in the database - I read that part but don't try to retain - with my limited size noodle. -Original Message- From: gabriel [mailto:g...@bayintegrated.net] Sent

Re: [sipx-users] call disconnects when listening to saved messages

On Fri, Jan 29, 2010 at 4:53 PM, gabriel wrote: > right, it's the dev version. > in the mean time I searched on sipx-dev and saw the bug :( > > not sure what the conf recordings had to do with it but will try what you > said now. > > -gabriel > > On Fri, 29 Jan 2010, Todd Hodgen wrote: > > > Wha

Re: [sipx-users] call disconnects when listening to saved messages

right, it's the dev version. in the mean time I searched on sipx-dev and saw the bug :( not sure what the conf recordings had to do with it but will try what you said now. -gabriel On Fri, 29 Jan 2010, Todd Hodgen wrote: > What version are you running? > > If you are running the unstable dev

Re: [sipx-users] call disconnects when listening to saved messages

That log snippet is only showing "part" of the call conversation. http://wiki.sipfoundry.org/display/xecsuserV4r0/Display+SIP+message+flow+using+Sipviewer A siptrace would show you more. It's also important to understand how the call to the VM system is arriving (local phone or softphone, remote

Re: [sipx-users] call disconnects when listening to saved messages

What version are you running? If you are running the unstable dev version, delete any recorded conference call you have and try it. There is an open JIRA on an issue with Conference recordings that are transferred if I recall correctly. -Original Message- From: sipx-users-boun...@list.si

[sipx-users] call disconnects when listening to saved messages

so I can listent to the voice mail messages only from the web interface, the call gets disconnected when trying it from the phone. I see this error in the log: --- "2010-01-29T21:15:04.782000Z":18063:sipXivr:INFO:sipx.company.net:Thread-359::sipxivr:"Retrieve::playMessages SAVED Mail

Re: [sipx-users] sipx paging

I do all that with Asterisk and PHPAGI. If you want more info I'll send it to you off list. Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 1/29/2010 2:19 PM, Eric Varsanyi wrote: Speaking of paging: does anyone have a pointer to a good command line based

Re: [sipx-users] sipx paging

Speaking of paging: does anyone have a pointer to a good command line based way to play a page (from a wav file for instance)? I'd like to send timed pages out of cron. I was thinking of trying the pjsip/pjsua SIP client, it looks from a quick scan like it might be able to make a call to a SIP

Re: [sipx-users] sipx paging

25 is the max I would go. I've written an application for Asterisk that can page multiple page groups in series, but that's about the best I think you're going to get. It's unicast Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 1/29/2010 1:32 PM, birchst

Re: [sipx-users] sipx paging

A dozen'ish. Above that you can use an overhead sip based paging system (valcom, etc.). Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fa

[sipx-users] sipx paging

Hi there, Server hardware limitations aside does anyone know of a limitation on the number of end points that can be paged simultaneously through sipxecs? Is it multi-cast or unicast? In this case, I am looking to page a 'large' number of endpoints. B

Re: [sipx-users] Wiki account

This place is as good as any to ask. Scott Lawrence can probably set you up. Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 1/29/2010 10:34 AM, Eric Varsanyi wrote: > I'd like to add a page with a complex (and even working) Patton FXO config I > came up wit

[sipx-users] Wiki account

I'd like to add a page with a complex (and even working) Patton FXO config I came up with but it says account adding requires email to 'mar...@sipfoundry.org'. This email address bounces at Nortel. Who can I ask for write access to the wiki? Thanks, -Eric Varsanyi __