Vodia support

On call 501 to the conference room, can you turn up the log length as well, looks bit short? Also can you create another conference room and have ext 201 call just to see if you get the same results. Thanks

The software was created for both products to work seamlessly together, once you understand how PNP works on the phone you will realize how easy it's to set up an installation, Lets say you have an installation of 50 phones on the network, are you going to program every phone, every button? The point is that admin will have total control of his installation and phones. Click on this URL "Quick video" on PNP PNP Overview You are more welcome to open a ticket with us by all means. https://support.snom.com/ You will have to register with us to open the ticket.

Can you send us a SIP trace with the following logging events. We have tested your scenario and could not reproduce it.

We will test this in our lab to see if it's reproducible.

Welcome, I don't see why not it looks like your provider likes g711a.

The log looks scattered only thing that caught my was the following. [6] 20120123190302: Call hold from trunk [6] 20120123190302: Different Codecs (local pcmu/8000, remote pcma/8000), callid 5l5um9chha, falling back to trans-coding. To avoid transconding you will have to make use of g711a you can change this in the port section choose only g711a and on the trunk section use g711a Also I also noticed that you have ports 30000 to 40000 ports open in the "port section" there is field called "RTP" we are using ports 49152-64512 to receive RTP can you open these ports on your router, just to be on the safe side or you can add 30000 to 40000 on the RTP section and restart the service as well.

Any complex networking in you facility DMZ, NAT, Firewalls? Can you post the logfile? Thanks

Many users experience this routing issue when using DMZ, You can check this forum post for more detail. http://forum.snomone.com/index.php?/topic/5543-ip-authenticated-sips/page__p__23276#entry23276

Can you verify if the phone actually pnp? you can check on the phones GUI --> Function keys usually you will see a button configuration already populated check the below jpg. You can also check if the pbx created a generated folder that we create when the phones PNP to the system you can find this folder in Windows, Program Files/snom/snomONE/generated or in Linux usr/local/snom/snomONE/generated

If you download version 4.5.0.1016 http://wiki.snomone.com/index.php?title=Alpha_Monocerotids(2011-4.5.0.1016) Custom Header is a new feature on this version. here is an explanation on how to use the custom header feature. http://wiki.snomone.com/index.php?title=Trunk_Custom_Headers

Before you can assigned a button configuration to the phone you have to PNP the phone 1st. You can check the link below for some information on how to do that. Today park orbits are accounts, you will have to create a button configuration in Domains->settings->Buttons usually we have a template you can use, Choose Park Orbit as a type and the park orbit account on the parameter. Once you have finish creating your buttons you will have to assigned them on the extension level. I would create these things 1st and then plug and play the phone afterwards. http://wiki.snomone.com/index.php?title=Before_Doing_Plug_and_Play http://wiki.snomone.com/index.php?title=Plug_and_Play

If you're cs410 is on a public, send me a PM and I will be able to help you.

The ticket has been updated, awhile ago. We have uploaded your domain.tar file on our windows machine and downgraded our phone to 8.4.32 we tested you scenario on 4.2.1.4025 "pbxnsip" and we could not reproduce you scenario. If you can reproduce it, follow the direction on the ticket and update it with the new log file. Thanks

We are working on a new snom console and have discontinued support on the WAC for the moment. You can download our phone app in the meantime. The snom console does not have a release date yet, but when it's available you will find it on the forum. http://wiki.snomone.com/index.php?title=The_snom_phone_application

In the redirection page of the extension you can set "immediately under "Include the cell phone in calls to extension" If the cell phone does not pick up then the system will send the call to the systems VM.

You will have to create a sip gateway on the soho to the Epygi in order to make outbound calls and inbound calls. I think you will have to create an association in the Epygi as well. http://wiki.snomone.com/index.php?title=SIP_Gateway After the association has been made on the soho you will then have to create a dial plan so the user can call outbound. http://wiki.snomone.com/index.php?title=Dial_Plan_Samples

Thanks for the PM, you can use this forum post to open a ticket in the future as well. Good Luck!

Can you post the logfile so we can diagnose it. Thanks

Can you check if the star code has been changed? We have just tested 4.2.1.4025 and was able to transfer the call. 1. 40 calls 41, 2. 41 puts ext 40 on hold 3. 41 press *7742 4. pbx disconnects 41 and transfers 42 and 40.

Some radio station hide the radio link which is difficult to find when looking at their source-code. With shoutcast.com they give you the direct link to the station.

Hola tu! try this link http://plugvoip.com/How%20to%20guides/HOW%20TO%20STREAM%20A%20RADIO%20STATION%20ON%20SNOMONE.pdf ignore step 7 since the new alpha version corrects the codec issue. press stream button on VLC. Then change the neccesary settings on the pbx concering the MOH.

Please review this attachd article on how to configure skype and snomONE after you have configure the trunk follow the steps above concering the skype certificate. SkypeTrunk.pdf

We will open a feature request for this item.

I would open a ticket with sangoma, http://www.sangoma.com/contact/technical_support.html

By default we broadcast the multicast address 224.x.x.x can you restart the pbx service and then log into the pbx and navigate to status-logfile and check if the multicast is being broadcast.