Maribel

I did a test with shoutcast, tried to stream the radio on the snomONE plus and still no success I followed step by step both links and it didn't work Thanks Maribel

But shoutcast only show US radio stations, I need canadian radio stations Thanks

Thanks for the link, do I need to use shoutcast? What if the customer wants another radio station? For example: http://player.rogersradio.ca/ckis/recently_played

This link is broken I am trying to stream a live radio too I tried using VLC but no success I was also checking the wiki.snomone.com and they explain about uploading an MP3 file using Magix, Goldwave, etc I don't want to use MP3 files I just want to stream the live radio from the internet, do I still need to convert the link to something else? Please ideas, suggestions, anything that can help. Thanks

You can check the following link: http://claricomsolutions.com/faq/index.php?action=artikel&cat=1&id=18&artlang=en&highlight=snomONE Thanks Maribel

Hello I have the same problem. I have set an extension with and action URL only to allow it to call one extension and set the NOT ALLOWED plan for the rest of the extensions or any other external phone number. The dial plan doesn't work on calling other extensions. Is this a bug on the software? I am running 2011-4.2.0.3981 (Linux) version Thanks Maribel

If you are running the CS410 locally, you need to put the MAC address of the M9 in the Bind to Mac field of the extension or extensions (if you have more than one handset) you want to register the M9. After that, got the the web interface of your M9 and under Network Settings set the ip address or the domain name of the CS410, save and reboot, once it is rebooted you should be able to see the Identity or Identities with the contact information of the extension or extensions.

Estimado usuario, La versión no tiene nada que ver con el pagmoh. De acuerdo al documento guía para la instalación del Paging and Moh hemos notado que no hemos sido específico en un paso. Al momento de llegar a la opción de GENERAL ----> MoH es necesario crear el pagmoh rtp. En el campo NAME colocar pagmoh rtp, TYPE RTP stream, DOMAIN sería all domains ó el dominio de su preferencia y en PORT NUMBER coloque el número de puerto 6000. Una vez finalizado haga click en el botón CREATE el cual lo salvará y lo colocará disponible en el AVAILABLE RESOURCES. A partir de este paso puede continuar con las indicaciones del documento guía. Saludos y gracias, Maribel

I'm setting the inum accounts as SIP Gateway trunks with inbound and outbound calls. Both are with the same ip address of outbound proxy, but the calls still are ringing on the last extension I set up. Thanks, Maribel

Never mind, I figured out. Now the issue I'm dealing with is that every trunk I set on my pbxnsip server goes to the last extension number I set with and specific INUM account number, I mean if I set my inum account number ended in 000 to ring at my extension 101, and I have my other inum account number ended in 040 to ring at extension 102; all incoming calls from both trunks are ringing on my extension 102. Any ideas of what could be happening here? Thanks, Maribel

Hello, I'm working with two inums account, registered on my pbxnsip but I can't received or make calls. Can you help me how can I set up my dial plan? Thanks, Maribel

Hello Everyone, I have problems with a CS425, it was running the latest version 3.2.0.3143 and working fine until updated to the 3.3.2.3181 version. We can’t make outgoing calls thought Trunk PSTN, I'm attaching the settings. The original settings on Rewrite global numbers is “Check Domain Country Code” I saw a solution in this forum that said changing to NANPA (11 Digits), I changed it to this option and listened a record from our carrier saying that dial number is incorrect, when the option is “Check Domain Country Code” we receive a busy tone immediately and in the log file mark this error: [7] 2009/05/13 20:45:26: Cannot convert number 92603200 into global format And on the domain country code is set to 1 and area code to 956. Any help please!!!

Hello, What OS are you using to access the CS410? Because one thing is the OS the CS410 uses (which is Linux) and the other one is the OS you use to access the CS410. If it's windows, you just donwload the software upgrade on your PC and uploaded via web browser to the CS410 and reboot the appliace. It's that simple. Maribel

Hello, We've been working on replacing every white box from our clients and users. You need to contact your vendor or any sales personel from pbxnsip who can help you with this. Regards

Hi, Can you please make a call and sen the log, if it is possible set the level to 7 to see the details. Thanks Maribel

Hi, I don't know what happened this weekend but once I turned off my computer and restarted today. The new 3.1 beta is working now Thanks

So do I need to set up something in the ATA or the CS410? Thanks

Yes I did.

Hi guys, I tried to update the pbxnsip with the new beta image 3.1 but by the time I restart the service I got the error 1067: the process terminated unexpectedly message. I have two PCs both with windows vista and I got the same error message in both. I did it with an old image and it worked fine. Can anyone help me to solve this? Thanks, Maribel

Hi Juan, The new release for the software has the ANI field option. Did you upgrade your software? Maribel

Hello guys, I've been trying to register a grandstream handy tone 488 with my CS410, but every time I update it and reboot it with the SIP information, the register status says NO. Do I have to necesary plug an analog extension to the FXS to get this registered? This is the SIP message of the CS410 and it is constantly repeating and there is no a SIP/2.0 OK message after that. REGISTER sip:192.168.1.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bK3c54f49ffe0ab8bf From: "Grand Stream" <sip:43@192.168.1.102;user=phone>;tag=3abfd8190bc9d8e9 To: <sip:43@192.168.1.102;user=phone> Contact: * Call-ID: d10af6bf7e3655d8@192.168.1.100 CSeq: 100 REGISTER Expires: 0 User-Agent: Grandstream HT488 1.0.3.96 FXS Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 Can you help me with this? Thanks

Do these gateways support a multiplexed line of 30 channels? Thanks, Maribel

Hi guys, I would like to know which E1/T1 PSTN gateways or cards work with PBXNSIP? Thanks, Maribel

Hi guys, Does anyone know how to configure pbxnsip with multiple interfaces? The application is running under Linus Redhat. Thanks, Maribel

Try the TELMEX ITPS, there are ohters but this has much time in the mexican market thant the rest. Regards,