similar to: Looking for a patch cable for my SPA941 Phones

Hi, I'm experiencing the same problem. We have 2 office locations and the Asterisk server is at one of them. At the other location, all SPA941 access the Asterisk server over an Internet link. All phones are set to "nat=yes" at the remote location. So my problem is that the MWI doesn't work at the remote location. The Sipsak messages are sent properly, but it's sent to the

The prospect of people actually putting B-format audio (via the panner or directly input) into Ogg/Vorbis brings an interesting challenge: What do we do with the audio after decoding it? The following sane options exist: A) Simply output the B-format audio B) Produce a downmix 1) Mono. 2) Stereo blumlein crossed pairs 3) Stereo UHJ 4) binaural C) Produce speaker feeds 1) Fully

Does anyone know how to set the distinctive ring on the Linksys SPA941? I want to be able to dial one extension and have the phone ring with a certain tone and then dial another and have the phone ring with a different tone. I have tried the following ------------------------------------------------------------------- exten => 802,1,SIPAddHeader(call_info=Classic-4) exten =>

Dear Group, I have just purchased two Linksys SPA941 and flashed these to the latest firmware. Everything works well except for the Hold button? Has anyone else experienced the same issue? What was the solution? Kind Regards Shad Mortazavi

Hi all, i have the following setup PSTN -> routing server (asterisk 1.6.2.11) -> IAX -> callcenter asterisk 1.6.2.9 -> SIP -> agent Does work quit fine - then agent does have the abibility to transfer a call to a third party - the agent can initiate the transfer over a web interface - it does generate a asterisk manager atxfer request... So agent does initiate transfer - call

Hi all, I'm new to this list, so forgive me if I'm not supposed to ask this: I currently own a Linksys SPA941 SIP phone with 5.1.8 firmware. Is there any way to use TLS with this phone<--->asterisk (v 1.6.0.9)? It is said that is supports TLS/SRTP but I don't see any of these options in the configuration file or the admin (advanced) SIP conf panel. Am I missing something? Thnx

I have an a bunch of SPA941 Linksys phones for users in and out of the office. When the phones are in the office (and on the same network as the asterisk server) the WMI goes on when it should and off when it should. But when the phone is on another network and natted it fails to do so. The light always stays off. Has anybody had a similar problem (and hopefully a resolve)?

Hello, Yesterday night we have upgraded our Asterisk from 1.2.32 to 1.4.24.1 with lipbri 1.4.10, dahdi-linux-2.2.0-rc4 and dahdi-tools-2.2.0-rc2. Libpri and dahdi is only for dahdi dummy cause of the meetme function. After the upgrade we had the problem that some Linksys spa941 phone at one location could not dial out. incoming calls to the phones works without any problem, but outbound the

Hi all. The SPA941 and friends have pretty multicoloured LEDs, but there doesn't appear to be any support for SUBSCRIBE/NOTIFY as * as implemented for extension hinting. Has anyone managed to get the phone to support this? Thanks! -- David Zanetti <david.zanetti@catalyst.net.nz> Team Leader, Systems Administration Catalyst IT Limited +64-4-8032233 +64-21-402260 -------------- next

Appearently SPA941 is less than a SPA942 without two ports, poe and backlight. There is less features too, it doesn't support BLF. Is it possible to hack 942-software into 941, or is there another workaround? Leif -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091112/0a6cbf82/attachment.htm

Hi I have a big problems on my Asterisk systems : I have one Asterisk Server with static IP (no nat) and 6 Linksys SPA941. All SPA are after a router with NAT: * SPA-1 and SPA-2 are on the same network, we have a pat 5060 => SPA-1 and 5061=> SPA-2 on the internet router * SPA-3, we have a pat 5062 => SPA-3 * SPA-4, we have a pat 5063 =>

hi all, we do have some troubles with zaptel timing source - we have a setup with 3 telco PRI lines, connected to asterisk digium card 0 - asterisk does some handling - calls are leaving on digium card 1 - going to a siemens hipath - there is some call handling - some of the calls then are going from the hipath over a qsig line to a bosch integral PBX - handling the rest of the calls. To be able

Hello list, I got a couple of those "wouldn't it be great questions", as following: 1. Is it possible, with asterisk to hold a central phonebook directory of callers?, so that when this party calls a "textual" caller ID will be displayed on the phone display. 2. How can this be configured with Trixbox, I've looked at the configuration options - I assume it plays no

Hi Opus list. I subscribed because your discussion on the IETF draft ("Ambisonics in an Ogg Opus Container") was mentioned on the sursound list. I tried Opus for ambisonics more than a year ago. It does works with uncoupled channels (I had to patch the encoder). I don't know what else could be done to optimize support for ambisonics, as I'm not a codec expert. So I think that

On Tue, 31 May 2016 09:41:37 -0700 Michael Graczyk <mgraczyk at google.com> wrote: > UHJ is an interesting way to preserve compatibility with non-ambisonic > playback systems. However, I have not seen it generalized to higher > orders. I expect that its popularity will decrease as HOA becomes more > and more common. If UHJ becomes popular in the future, we could > specify

> Many thanks Julian. Are you looking for the pinout for a single plug 2.5mm (cellphone) headset or a dual plug 3.5mm (computer) headset? -- Nabeel Jafferali Tel: +1 (416) 628-9342 Toronto +1 (646) 225-7426 New York FWD: 46990 Email/MSN: nabeel<at>jafferali.net

On Sat, 28 May 2016 16:21:33 -0700, Michael Graczyk <mgraczyk at google.com> wrote : > Hi Marc, Hi Micheal. > On Sat, May 28, 2016 at 10:44 AM, Marc Lavallée <marc at hacklava.net> > wrote: > > I subscribed because your discussion on the IETF draft ("Ambisonics > > in an Ogg Opus Container") was mentioned on the sursound list. > > Thanks for

I was just browsing my local suppliers list of headsets - not a single one with a single 2.5mm jack. Either USB or 1-2 3.5mm jacks. Can anyone recommend a headset that works with e.g. SPA-921 and -941? /Per Jessen, Z?rich

Not exactly sure if this is what you are looking for, but here goes: RC3 and prior maintain channel seperation with Q>=5.0 (cut-off is exact) Vorbis 1.0 maintains channel seperation with Q>=6.0 (not sure of the exact cut-off) with release code, selecting a Q of 5.99 or lower will make assumptions for channel seperation to save bits. earch the archives for "ambisonics" - you may

Hi. We have an odd problem with incoming SIP calls. I have attached a SIP debug log, with some asterisk verbosity as well, demonstrating the problem, below. Is this a known bug? Vital stats: - Asterisk 1.2.3 - Sipura SPA-841, SPA-941 phones - Fedora core 3 The problem manifests itself with these symptoms: - an internal SIP extension receives a call from our PRI - the SIP phone answers the