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Does anyone succeed on upgrading Mediatrix 1204 Frimware? If yes How... thanks

1-You CANNOT register PSTN line as an extension, it MUST be as a trunk 2-You CANNOT dial an extension to get an external line! Unless I am mistaking or not getting your question. You can use SPA3102 to dial true PSTN line without going to PBX at all, but why would you like to do that! If you need to dial to extension to place an external call, you can use CallingCard feature. Mechanism will do what I understand from your question. their is 2 thinkgs on SPA3102 one is FXO port (Your extension) and second FXO (your PSTN line) they need to work together! Maube some will have an answer for you, I never used SPA3102 in a process like you are asking to do!

Very true, it took me a while to get it working fine ... since version 2.0 I have one running for several mounts on a test system, here's my settings: PBX - Trunk should be configured with username and password that match spa3102, Outbound Proxy: xxx.xxx.xxx.xxx:port (for me it is 192.168.1.50:5061) Send call to extension: AA (for me 10) SPA3102: firmware 3.2.6(GWa) newer will never register under PSTN Line Proxy: your PBX IP (for me 192.168.0.1) Outbound Proxy: your SPA3102 IP (192.168.0.50) Use Outbound Proxy: Yes Use OB Proxy In Dialog: Yes SIP Settings SIP Port: 5061 Subscriber Information - this must match your Trunk set-up Display Name: User ID: Password: Use Auth ID: Auth ID: Dial Plans Dial Plan 1: (S0< :10@192.168.0.1:5060>) Dial Plan 2: nothing Dial Plan 3: (xx.) - and all others VoIP-To-PSTN Gateway Setup VoIP-To-PSTN Gateway Enable: Yes VoIP Caller Auth Method: none VoIP PIN Max Retry: 3 One Stage Dialing: yes Line 1 VoIP Caller DP: 1 VoIP Caller Default DP: 2 Line 1 Fallback DP: none PSTN-To-VoIP Gateway Setup PSTN-To-VoIP Gateway Enable: Yes PSTN Caller Auth Method: None PSTN Ring Thru Line 1: Yes PSTN PIN Max Retry: 3 PSTN CID For VoIP CID: Yes PSTN CID Number Prefix: PSTN Caller Default DP: 1 Off Hook While Calling VoIP: no Line 1 Signal Hook Flash To PSTN: Disabled PSTN CID Name Prefix: Line 1 DialPlan: (*xx.|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) xx must match the max number of digets for your system ex if your system is 2 digets you must put xx, if 4 mut put xxxx Hope this helps

I can certenelly do that, what screen do you need? Yes, conference room is a scheduled room

HI, I need to set a new PBXnSIP small office system and looking for proper part before. What gateway would be suitable for 2 FXO only? For 4FXO I would be usign Mediatrix 1204 but for 2 only what would be good and not expensive? many thanks.

Good day, Running 3.3.0.3165 CallCenter edition Loging in as user and going to the conference tab to create a conference. Entering all the info and picking an availible room. It has absolutelly no effect waht so ever. Email is not sent out, conference is not registering, I see no trace what so ever just like if nothing ever happend regards

Settings URL tabs does not exist under snom820-SIP 8.1.3 15856. How can it be called?

I have download and install new 3.3.0.3147 version and so fare I loe the new features but I found that something is not working related to cell phone transfert. Incoming tranfer call to cell phone is not receving audio on the cell phone but cell phone transmit audio no problem. I went back to 3.2 and everything seams to be OK.

"Max. number of concurrent registrations per extension: " is a fantastic idea and I LOVE it, I suggest to impliment it on the extension setting level to control each extensions separatelly instead of a general setting for all domaines. I love it! more control over registration, security, services, ... i love it!

"In order for the PAC to work with the PBX, you need to use later versions of 3.3.x PBX software" PBX v3.3 , noT yet public?

I am looking for that as well, so fare no success!

Many thanks for your reply, it does not seams to like PBXnSIP so fare... I am looking for a simple RTP server that I can run locally and have PBX listenning to it something like iTune would be perfect!

According to: http://wiki.pbxnsip.com/index.php/Music_on_Hold we can use RTP to stream to PBX server "There are several external tools available that are able to generate a compliant RTP stream" What would be a good application- tool if I want to run my own RTP server on Win32 system?

Their is a major problem with PBX/SNOM 8.0.12. V 3.2.0.3144 Win 32 CallCenter edition. several LinkSys Ciaso7960, SPA962, SPA942, SPA525G and 1 new snom 820 that create problems. When accessign VM from SNOM phone system is asking to record personal greedings even if it is present/valide/on You record and it saves it on PBX but ask to record it again. Nerver had a problem like that with LinkSys and Cisco phones. Any ideas on waht to be looking for and where? MWI Notification is set to "silent" MWI Dial tone to "normal" tkx

-UPDATE- Feb 17 After several days, deveice is extremelly stable and still in a working state. I can conclud by recommending SPA525G in a mixte installation and 99% PBXnSIP friendelly. Cisco IS extremelly responsive and efficient with this product line.

I need to set routers which supports DMZ on all user's locations. Not best way. SIP ALG? No just on the PBX side, on the client side it should be working with no change at all, If needed only you just have to route phone IP with 5060-5061 and RDT ports. better to use static IP for the phone! 50/50. It can help in case when static IP used. Most of ISP's changes external IP addresses of home users (DSL) once a day. I need to seek this addresses.... I use no-ip.com dynamic DNS it prevent loosing adresse everyday and I use PBX IP/DNS name

I have faced the same situation, to resolve I didi the following: 1- use a router that support DMZ and set to the adress of your PBX server - work fine OR 2- set your router to allow ports for your PBX - all of them RTP is transporting audio you can also set your PBX to use the following route table : under Admin>Ports set "SIP IP Replacement List" to first IP of your PBX and IP of your router - something like 192.168.0.1/xxx.xxx.xxx.xxx -> PBXIP/RouterIP restart PBX and It should work fine This run fine on my side BTW you should have a very very very good Internet connection if you want to keep audio quality for all external registration on your PBX otherwise it will be more like a cell phone quality, forcing codec might help. hope it help!

Tanks for the update. We'll wait for 3.3 :-)

I have the same experience with CallCenter edition was installed with v 3.1.2 and manually upgraded to 3.2.0 Don't see that email setting

Many thanks, Now that I have documentation, I create a file and put it somewhere??? How do I access it from a specific adress true a web browser??? Documentation is very helping but process would be too. many thanks for you reply!

A complete email has been sent to the requested address.

Status>Calls I now get numbers under Action colomn. Waht does that means? (-29.5) Start From To State Action 2009/02/04 08:19:58 Guylaine (73) 9xxxxxxxxxxxxx [xxxxxxxxxx] connected X (-29.5) values are different between calls and lignes used. Is this audio Gain based on the Audio test from Logs?

Hi I have a an Installation that moved from a private IP to a public IP to allow external user to register phones from their house. My problem is that all phone register but the one outside of the company location do not play any audio at all. I've tried to check if ports are open/close and I can see that 5060-5061 are open but 69-close 49152 to 64512 (that are used for audio if I am not mistaking) are closed event on public IP?? From the Internet service provider ? is that possible? Does that make sense and can it be the reason I get no audio at all?

Fixed! many thanks for the precision

This seams to be extremelly great. But how do you code it, how do you access it... If we are not familiar with SOAP your reference link would be helpfull, unfortunately is is borken. Where can I get the bases of SOAP to start usign it? regards