Re: [sipx-users] [sipX-dev] Handling firmware for Polycoms in a mixed legacy/non-legacy environment

From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano Sent: Thursday, January 07, 2010 11:10 AM To: Eric Varsanyi Cc: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] [sipX-dev] Handling firmware for Polycoms in a mixed

Re: [sipx-users] Vitelity trunk

On Tue 05.Jan.10 09:57, dan wrote: >On Tue 05.Jan.10 11:21, M. Ranganathan wrote: >>On Tue, Jan 5, 2010 at 11:12 AM, Dan McDaniel wrote: >>> As a follow-on question, when I look at the trace with sipviewer, which >>> component should be talking to the ISTP? I'm seeing sip-proxy contacting >>> the

Re: [sipx-users] Call Forwarding - No Audio

Check the signaling using the trace viewer. Does the ITSP support p-asserted-identity ( that is the default ). If not you will have to tweak the settings a bit to get things to work. Ranga On Thu, Jan 7, 2010 at 1:08 PM, Ken Fulmer wrote: > No AA or hunt group in this configuration. It’s just a

Re: [sipx-users] Call Setup Delay

Then it is an fxo connection? I was not aware fxo connections needed to register. 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

Re: [sipx-users] Call Setup Delay

Why is on phone registered with the Audiocodes? What purpose does that server? On Thu, Jan 7, 2010 at 6:30 PM, Joseph Ofosu wrote: > Hello All, > > I have installed SIPX with two phones registered. One phone is registered > with Audiocodes MP112 and the other is an eye beam softphone. > Is you

[sipx-users] Call Setup Delay

Hello All, I have installed SIPX with two phones registered. One phone is registered with Audiocodes MP112 and the other is an eye beam softphone. The server, and phones are all on the same network and so I decided to use the IP address of the server as the proxy. I also set the domain alias usi

Re: [sipx-users] [sipX-dev] Handling firmware for Polycoms in a mixed legacy/non-legacy environment

FWIW, this really should be a sipx-dev discussion (IMO). The polycom phones will "discard" any parameter they are given that they do no know how to handle (like EFK). That being said, I think there was a workaround being suggested by Paul to allow sipx to "know" the model of a phone and hence, pu

Re: [sipx-users] [sipX-dev] Handling firmware for Polycoms in a mixed legacy/non-legacy environment

I got the impression from the polycom release notes that the base config files that come with each firmware release have to be used with that firmware release, I looked around in the template stuff but I couldn't find any obvious way to use different templates for different instances of a polyco

Re: [sipx-users] Hardware Timing Source

Thanks Tony. Tony Graziano wrote: > Since all of the hardware gateways for sipxecs are standalone > appliances, the only timing you would care about is timing for the > CSU. Typically your TELCO provides timing (example: for a PRI or > switched T1) and you would configure your appliance to be a

Re: [sipx-users] Hardware Timing Source

Since all of the hardware gateways for sipxecs are standalone appliances, the only timing you would care about is timing for the CSU. Typically your TELCO provides timing (example: for a PRI or switched T1) and you would configure your appliance to be a slave (with the provider being the master). T

Re: [sipx-users] Hardware Timing Source

Thanks Scott. --Original Message-- From: Scott Lawrence To: Thomas McConnell Cc: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Hardware Timing Source Sent: 7 Jan 2010 18:35 On Thu, 2010-01-07 at 18:22 +, Thomas McConnell wrote: > Coming from a Asterisk background, I was alwa

Re: [sipx-users] Hardware Timing Source

On Thu, 2010-01-07 at 18:22 +, Thomas McConnell wrote: > Coming from a Asterisk background, I was always told that a hardware > timing devices was important for a full VoIP environment, we used > Sangoma UT50s(USB timing device) in all of our installs of Asterisk. > Does SipXecs require som

[sipx-users] Hardware Timing Source

Coming from a Asterisk background, I was always told that a hardware timing devices was important for a full VoIP environment, we used Sangoma UT50s(USB timing device) in all of our installs of Asterisk. Does SipXecs require something similar. Regards, Thomas __

Re: [sipx-users] Call Forwarding - No Audio

No AA or hunt group in this configuration. It's just a simple user / phone combination. Normal calls work properly. It's the calls that are forwarded that don't have audio. Ken From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] Sent: Thursday, January 07, 2010 11:58 AM To: Ken Fulme

Re: [sipx-users] FXO+FXS gateways

Dittos From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano Sent: Thursday, January 07, 2010 11:23 AM To: Eric Varsanyi Cc: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] FXO+FXS gateways PATTON. If you like intero

Re: [sipx-users] TAPI with sipx

Sorry, not a programmer... check with the dev list. From: Ola Samuelson [mailto:ola.samuel...@attendit.se] Sent: Thursday, January 07, 2010 6:59 AM To: Picher, Michael Cc: Ola Samuelsson; sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] TAPI with sipx Thanks. Looks promising. I ha

[sipx-users] Auto Attendant forwarding to external number problem

Hi, I am having trouble with getting the Auto Attendant forwarding to external number. 2 Methods tried 1) Press 2 in the auto attendant and it has a UK mobile number to forward to 2) Press 2 in the auto attendant and it forward to an extension which in turns forwards to a UK mobile number Bo

Re: [sipx-users] Call Forwarding - No Audio

No. If I call a DID assigned to a user, and they have call forwarding turned on, the call goes back out and audio works in both directions. If you are hitting an AA or hunt group first, please acknowledge that. On Thu, Jan 7, 2010 at 12:49 PM, Ken Fulmer < kenful...@icstechnologysolutions.com> wro

[sipx-users] Call Forwarding - No Audio

A user in sipX is configured for call forwarding to an external destination. A call that originated in the PSTN gets forwarded back out to a destination in the PSTN. However, there is no audio in either direction. We are using the sipXbridge and a sip trunk. Routine inbound and outbound calls

Re: [sipx-users] FXO+FXS gateways

Do't know about the m-ata. I know someone uses it with mitel and it works. http://forum.sipfoundry.org/index.php?t=msg&th=4618&start=0&S=595485b1b93cb9adff8baedf1a81caad You might post a question on th

Re: [sipx-users] FXO+FXS gateways

You will find example configs in the sipx wiki. On Thu, Jan 7, 2010 at 12:32 PM, Eric Varsanyi wrote: > Thanks for the pointer. I went to the site and immediately found manuals > and firmware. The manuals are readable and configuration seems pretty > straightforward (like old school cisco router

Re: [sipx-users] FXO+FXS gateways

Thanks for the pointer. I went to the site and immediately found manuals and firmware. The manuals are readable and configuration seems pretty straightforward (like old school cisco routers) and logical. I have a 4114 FXO and a 4114 FXS on the way to test with! Thanks, -Eric PS: Is the little

Re: [sipx-users] No audio problem in both sides doing outgoingcalls using voicetrading.com as ITSP

Sorry. the first paragraph answers your question I think. On Thu, Jan 7, 2010 at 12:23 PM, Tony Graziano wrote: > Most firewalls randomize ports (rewrite the source port) of outbound > traffic. This is problematic for some protocols (like PPTP, IPSEC and SIP). > sipXbridge needs static port NA

Re: [sipx-users] No audio problem in both sides doing outgoingcalls using voicetrading.com as ITSP

Most firewalls randomize ports (rewrite the source port) of outbound traffic. This is problematic for some protocols (like PPTP, IPSEC and SIP). sipXbridge needs static port NAT, or symmetric signalling in order to work properly. This means when sipXbridge makes an media connection at port 30001,

Re: [sipx-users] No audio problem in both sides doing outgoingcalls using voicetrading.com as ITSP

I think we found out where is the problem. We disabled Symmetrical RTP on voicetrading profile interface and now we have voice :) How does Sipxecs work with symmetrical and not symmetrical RTP? I suppose that this is related again with Nat Traversal thing. I will be happy if someone explain to s

Re: [sipx-users] FXO+FXS gateways

PATTON. If you like interoperability, and a thorough 500+ page user manual and complete CLI, plus factory direct support anyway. Their product has never disappointed me, we never even touch any other gateways anymore. On Thu, Jan 7, 2010 at 11:17 AM, Eric Varsanyi wrote: > OK, while I'm still su

[sipx-users] FXO+FXS gateways

OK, while I'm still suffering with SPA3102 brokenness (bad NOTIFY message, bad From headers, broken CPC detection) and having some success I do really want a rock solid and known to work with sipXecs (4.0 or 4.2+) small FXO+FXS solution. Price isn't an overriding factor but active vendor mainten

Re: [sipx-users] Anyone ever used VM to text services like Spinvox?

Not that's really a discussion option. Either way the encoding/decoding has to be done. If the end result is an mp3 it still has to be paid a royalty to be legitimate. A voicemail server destination as a google voice account makes more sense to me though. Tony Graziano,

Re: [sipx-users] Anyone ever used VM to text services like Spinvox?

I suppose Ogg Vorbis is out of the question? Mike Burden Lynk Systems, Inc e-mail: m...@lynk.com Phone: 616-532-4985 -Original Message- From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] Sent: Thursday, January 07, 2010 10:09 AM To: Burden, Mike Cc: sipx-users@list.sipfoun

Re: [sipx-users] Anyone ever used VM to text services like Spinvox?

Encoding/decoding consumes cpu on server, plus royalty 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 Contrac

Re: [sipx-users] No G.711 = no Polycom HD voice?

Verizon has done everything properly with codec negotiation. The wireshark captures show me what I was hoping for. On 1/6/2010 8:21 PM, Tony Graziano wrote: > Glad you have that worked out. When an invite goes out, it offers codec > choices, and typically the first agreed match is what will be us

Re: [sipx-users] Anyone ever used VM to text services like Spinvox?

What's the argument against MP3? Mike Burden Lynk Systems, Inc e-mail: m...@lynk.com Phone: 616-532-4985 From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano Sent: Thursday, January

Re: [sipx-users] Anyone ever used VM to text services like Spinvox?

Well, you can always just forward the call to google voice and bypass the internal voicemail. Personally, I like the idea of not having that kind of service on the vm system. I would prefer, though, changing the format of vm to mp3, but that's an argument I won't win. On Thu, Jan 7, 2010 at 9:40 A

Re: [sipx-users] Anyone ever used VM to text services like Spinvox?

On Thu, 2010-01-07 at 08:40 -0600, Robert B wrote: > I can't figure out how it works on their page, but I'd imagine it must > have something to do with forwarding voicemail... > > This is another "gee it would be cool" kind of things to support. I experimented briefly with a couple of services fo

[sipx-users] Anyone ever used VM to text services like Spinvox?

I can't figure out how it works on their page, but I'd imagine it must have something to do with forwarding voicemail... This is another "gee it would be cool" kind of things to support. I recall Nerd Vittles did some Asterisk thing with Google Voice but it was a hell of a contraption when the

Re: [sipx-users] TAPI with sipx

Thanks. Looks promising. I have looked att files/source but fails to se how i can get like "realtime events" upon device ringing/hanging up etc. I would rather not rely on timings/queries but get events. Any clues with regards to that? *Vänliga Hälsningar/Best Regards* /Ola Samuelson / 201

Re: [sipx-users] TAPI with sipx

Research the soap interface... From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ola Samuelson Sent: Thursday, January 07, 2010 4:58 AM To: sipx-users@list.sipfoundry.org Subject: [sipx-users] TAPI with sipx Hi! I have searched the wi

[sipx-users] TAPI with sipx

Hi! I have searched the wiki but it is a bit unclear to me which is the easiest way to do tapi integration with sipxecs. I realize building a full client with sipxtapi could be done but it seems a bit to much work just to trigger an external app at incoming call and to obtain click to call in e

Re: [sipx-users] Brain picking

Yes, as long as the 'internet calling' check box is un-checked on the 'internet calling' page... -Original Message- From: Nathan Nieblas [mailto:nathan.nieb...@sacatech.com] Sent: Thursday, January 07, 2010 3:20 AM To: Picher, Michael; Tony Graziano Cc: sipx-users@list.sipfoundry.org Subj

Re: [sipx-users] Brain picking

Cool, I've disabled it - I am really looking forward to no problems come morning. I'm assuming sipXproxy/sipXbridge behaves nicely with their built in NAT functionality then with remote clients, etc? Nathan Nieblas SACA Technologies, Inc. -Original Message- From: Picher, Michael [mailt

Re: [sipx-users] Brain picking

On the ASA's under the protocol inspection make sure SIP is disabled... works better for me that way. Mike -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of nathan.nieb...@sacatech.com Sent: Thursday, January 07, 2