search for: snome

hi We are using snome 200 IP phone with *. It works OK. But after a period of time we can't hear any sounds for any icoming or outgoing calls. I've got two of these phones. Same symptoms occur to both of these( not at the same time ) and the problem remains until the phone is completely rebooted. Don't know...

Hi, unfortunately the pbx control sequences which I have to use within asterisk (chan_h323) aren't the same as predefined in our snome phones. Now I found in an old message in the mailing list archive, that the control sequence for direct transfer is just #. Is there somewhere a list with further pbx control sequences (consultant transfer, ...)? Thanks for any hints! Roger.

...3.org and installed it and the module loads correctly when called. What I am confused about is how Asterisk will interact with an h323 gateway. Will I lose an functionality of the Asterisk PBX because of this. What I am ultimately trying to do is have a heterogeneous group of phones(Cisco, avaya, snome, etc) in my company that can use the Asterisk PBX for all their functionality not matter what protocol or Codec that use. Is this something that is wishful thinking? Also, do I need an h323 gateway for my h323 phone and Asterisk. If so, how does Asterisk use this h323 gateway? I thought Asteris...

...that are answered out of a queue must be done using Asterisk '#' transfers (enabled with the 't' option above). SIP transfers result in the Agent remaining affiliated with the call until its eventual termination, preventing that agent from being offered another call." We have a snome 220 that does consultative transfer..with the buttons on the phone Does this mean I wont be able to do this? Please Help and andvice

Hello list, I'm having some troubles with DTMF tones. When pressing numbers on a Snom phone, the DTMF-signal takes too long. I have the following in sip.conf : dtmfmode = rfc2833 which works well for Grandstream, Yealink and Cisco phones. But not for Snom. Snom support tells me I should use SIP info. Is it possible to have something like this : dtmfmode = rfc2833, info ?? Because

...transfer out of the queue went away. Something must have changed. I tried to implement the call transfer mechanism in features.conf. I selected #1 for blind transfer and #2 for attended transfer. The * on the Snoms has an internal use. This seems to work well with other phones than the Snoms. On a Snome one have to press the "check" button to "send" the #2. This seems to work occasionally, but not reliably. The transfer call doesn't seem to work at all. I have to confess, that I only have user reports on this. Does anyone have a tip how to get this to work? The users would...

Hello! I have finally ordered some Asterisk hardware: the TDM DevKit. However, I want to use VoIP phones (or possibly adapters) for remote users. I would like to get some suggestions on which phones to buy. I'm hoping that some of you with real experience might be able to help me out! Here are the features that are important to me: * While these phones are initially going to be

Hi, I am sorry to be asking this but the wiki is down and has been for a couple of days and I need to get this working before Monday to get my live system setup. Trying to get the Snom 190's and soon to arrive 3com 3102's to use the function keys and for the life of me I can't work it out from the conversations on the archive what I am going exactly wrong here? The snom 190 with