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YMSL

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Posts posted by YMSL

  2. Linksys SPA3102

    in Gateways

    Posted

    Yes, my problem, was that I needed to set it up as an extension and let it (SPA 3102) register as that extension. For dialing out, I would have to dial that extension number first, and then dial my intended external PSTN number. For incoming from PSTN->VOIP, my dial plan was simply (S0<:10>) where 10 is the extension I wanted the PSTN call sent to.

     

    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!

  3. Linksys SPA3102

    in Gateways

    Posted

    Hello,

     

    I have a Linksys spa3102. I have had difficulty trying to make this work with pbxnsip. Is there anyone out there who has these two working together? I have heard that I should set it up in pbxnsip as an extension. I have tried this, but no joy. There is no PSTN setup. It has been a couple of days. Any help would be appreciated.

     

    Thanks.

     

    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

  5. FXO gateway

    in Gateways

    Posted

    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.

  6. Conference system

    in Web Interface

    Posted

    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

  9. PBX 3.3

    in PBX Lounge

    Posted

    "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!

  11. MoH RTP stream

    in General Setup

    Posted

    I am looking for some assistance with getting this working. What player with best and what port is used. Also would like to provide different RTP streams for different domains. Thanks for the help.

     

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

  14. snom 820 and VoiceMail

    in snom Phones

    Posted

    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

  15. SPA525 G

    in Linksys Phones

    Posted

    Test done,

     

    We have received from Cisco SPA525G unit and put them under testing.

     

    result:

    GRATE PHONE, look, size, feel... very handsome device

    handfree seeker is so fantastic

    so fare interoperability seams to be just fine

    provisionning need be be done partially manually...

    needs lots of manual adjustment for Wi-Fi usability

    USB (mp3) player need standard mp3 files

    need firmware upgrade from 7.1.3 to 7.1.7 (A MOST)

     

    -UPDATE- Feb 2nd

    SPA525G product manager from California is providing support,

    so it look like cisco is getting very very serious with this promessing device.

     

     

     

    -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.

  16. Users behind NAT

    in General Setup

    Posted

    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

  17. Users behind NAT

    in General Setup

    Posted

    Hi,

     

    We trying to configure pbxnsip to work with remote users who would like to work at home. Most of remote users have internet connection behind NAT. It causes problem when users don't hear audio at all. What happens:

     

    This is INVITE from remote user. This INVITE coming from real address.

    1 0.000000 195.146.74.104 192.168.22.43 SIP/SDP Request: INVITE sip:112@tw;user=phone, with session description

    pbxnsip answering to this address. Good.

    2 0.003732 192.168.22.43 195.146.74.104 SIP Status: 100 Trying

    3 0.010382 192.168.22.43 195.146.74.104 SIP/SDP Status: 200 Ok, with session description

    and trying to send RTP to private address. This address taken by pbxnsip from INVITE.

    4 0.021485 192.168.22.43 192.168.1.25 RTP Payload type=ITU-T G.711 PCMU, SSRC=1237099670, Seq=24774, Time=3241200, Mark

    6 0.041015 192.168.22.43 192.168.1.25 RTP Payload type=ITU-T G.711 PCMU, SSRC=1237099670, Seq=24775, Time=3241360

     

    Sure that pbxnsip working fully according SIP RFC - sends RTP to IP address from INVITE/SDP.

     

    Quesion is: maybe possible to configure pbxnsip to send RTP to addres from INVITE received? As part of Mini-SBC function..

     

     

    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!

  20. VoiceMail box reset

    in Mailbox

    Posted

    Wiki link is fixed now. Thanks for pointing it out.

    Also, actual SOAP messages that you can send to the PBX are listed in each section on our wiki page

     

     

    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!

  21. No Audio

    in General Setup

    Posted

    Could be that the firewall on the home routers blocking the traffic. Is there a log file ? also if you have the wireshark trace, please PM to support@pbxnsip.com

     

    A complete email has been sent to the requested address.

  22. Understanding what pbx display

    in General Setup

    Posted

    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?

  23. No Audio

    in General Setup

    Posted

    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?

  24. 3.2.0

    in New Features/Versions

    Posted

    Just verify the "Permission" page for the extension. Old settings were 'true' or 'false'. But in the new version, is is changed to specific accounts or * for everything.

     

     

    Fixed! many thanks for the precision

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