Steve B

I have it working pretty good, dialing out is sometimes quick and sometimes a 10 second wait. Here is my IP registration setup: # Trunk 17 Name: Vitelity Type: proxy To: sip RegPass: ******** Direction: Disabled: false Global: false Display: RegAccount: RegRegistrar: inbound.server.net RegKeep: RegUser: Icid: Require: OutboundProxy: out.bou.nd.ip Ani: DialExtension: 112 Prefix: Trusted: false AcceptRedirect: false RfcRtp: false Analog: false SendEmail: UseUuid: false Ring180: false Failover: never HeaderRequestUri: {request-uri} HeaderFrom: {from} HeaderTo: {to} HeaderPai: {trunk} HeaderPpi: HeaderRpi: HeaderPrivacy: HeaderRpiCharging: BlockCidPrefix: Glob: RequestTimeout: Codecs: CodecLock: true DtmfMode: Expires: 3600 FromUser: Tel: true TranscodeDtmf: true AssociatedAddresses: inb.oun.d.ip InterOffice: false DialPlan: UseEpid: false CidUpdate: Ignore18xSDP: false UserHdr: Diversion: Colines: DialogPermission: I would be willing to make snomone a temp account if they want to work with Vitelity. I have been trying Sotel lately and everything has been running great. some of the trunk options are wrong on the wiki though. I will post the changes I made later in a post on that.

I am thinking of building an m350 also, looks very promising with all the mounting options.

They are now Sumofiber, info@sumofiber.com 801-320-1000. They have been very good to my customers.

First thing that jumped out at me was you have the G729 Codec as number 1, that would compress the audio if I am not mistaken...

This is pretty old news but as I understand it snom spun off snomONE into a separate company called Vodia Networks. This allows them to have third party PNP and make upgrades faster with better features.

Haha, the sooner the better. Nothing like 50+ emails saying i'm blacklisted until I white list my IP.

Could this be related to rtcp? I can try turning it off and seeing if that helps.

We have not had problems on extension to extension calling even with remote extensions involved. We only have problems when using certain outbound providers. I have not run a pcap in this instance but last time I had this problem and had to switch to UDP (ugh) to solve it, I could hear both sides of the conversation on the PBX but only one side at the phone.

So I have an issue on two new trunk providers (One I am testing and One I inherited). If I have my snom 320 phones connected to the PBX in TLS, I get some intermittent one way audio problems when calling to an outbound number. If I change the phones to TCP or UDP, no one way audio. I know from one of the providers that they don't support SRTP, could this be causing the problem? If the provider does not support SRTP, is there a setting somewhere for this? I am running 4.5.1090 and have a 5.0.5 pbx that has seen similar issues. -Steve

Well with a key system you can set the extension to auto pick-up in speaker when dialed from another internal phone on the system. When you transfer a call to the extension, the system knows the call is from an external number to the system and rings the phone instead of the extension auot picking up. Could you make the button on snomone act like this: When the BLF button is pressed (without transfer) it sends a call-info but when it is transferred it sends a ring tone? Or if the call has an attended transfer, to send call-info first to alert the person who will be taking the call, then when the transfer button is pressed a second time or a hangup happens the call is transferred as a ringtone.

I totally agree, we have a small office that just came off a key system and that is exactly what they are wanting. I third this motion...

FYI, all the video links are dead.

G711U is the only codec I used with it. I can't recall getting one way audio with it though.

DTMF with this unit is why I switched to Patton for my pots adapter choice for a production box. It is just not consistent enough for me. Some people swear by it though.

IF DTMF still gives you an issue, then you may want to set the PBX trunk to yes on out of band-DTMF tones and test the different DTMF Methods settings again.

Are the phones lines based in the USA? If so the the FXO page shows the caller ID scheme ok but you will probably need to change the "Number of rings to pickup" to 2 (ch1-4:2;), usually caller ID comes between ring 1 and 2 on pots lines. ON DTMF, click the channels page, and change DTMF Methods to 1 (ch1-4:1;), try it, if that didnt work then 2, try it or 4. I was told by grandstream not to use 3, it doesn't make sense to use because it is sent in one or the other not both. Also, don't change the progress tones, even if the test tells you to unless you know your provider needs something different and you know the info. The channel test gives the wrong settings. I have attached a PDF of my channels page. Grandstream Device Configuration.pdf

These are my updated notes on the grandstream 4104, I just had it working last week with 5.0.4 without a problem. I don't think there are any other changes needed from this but I will check my adapter tonight. I worked through this with my adapter and here is how I set mine up: Grandstream POTS Adapter: Product Model: GXW4104 Software Version: Program--1.3.4.10 Loader--1.1.3.4 Boot--1.1.3.2 (The current firmware that is available today works fine also) If your adapter is exposed to the internet change the password under advanced settings. Settings: FXO Lines: - Channel Dialing To PSTN 1. Wait for Dial-Tone(Y/N): ch1-4:N; 2. Stage Method(1/2): ch1-4:1; 3. Min Delay Before Dialing Out: ch1-4:500; FXO Lines: - Channel Dialing to VoIP User ID: ch1-4:1; ## Without a user ID the adapter would not call the pbx, it seems to work with any userid. I believe you can change the user ID on the individual lines to 888 (or something else you prefer ie an actual extension number) and route that to a hunt group or extension. Sip Server: ch1-4:p1; Sip Destination Port: ch1-4:5060; Channels: Make sure all the channels you are using are set to profile 1. 1. DTMF Methods(1-7): ch1-4:3; 1. DTMF Methods(1-7): ch1-4:2; Must be 1 or 2 depending on, In audio is probably better for analog.Experiment on what works best for you. (EDIT) I noticed the audio in was kind of low so upped the RX to 7. Beware that all POTS lines I have dealt with are different and will have to be tuned to your liking. Sometimes the ringback tone is too low and may need to be increased. Profile 1 SIP Server: xxx.xxx.xxx.xxx (PBX IP) SIP Registration: No snomONE PBX: Type: SIP Gateway Direction: Inbound and Outbound Trunk Destination: Generic Sip Server State: Enabled Display Name: Grandstream Domain: xxx.xxx.xxx.xxx "IP of your gateway" No User Name Or Password Accept Redirect: Yes Interpret SIP URI always as telephone number: Yes Send Call To Extension: "The Extension You Use" (if you set different user ID's you can route them to different parts of the snom PBX) Explicitly list addresses for inbound traffic: xxx.xxx.xxx.xxx "IP of your gateway" Message 180 to yes (A bad clicking sound will be on one side of the call if this is not enabled.) Make sure you remove the area code out of your domain settings so it does not dial 10 or 11 digits Make sure to have a 7 digit dial plan active that points to the pots adapter.

I had this problem once when I was using UDP and a snom m9. When I changed to TCP/TLS it went away. This may not be your solution but I am just throwing out my experience.

Sorry to hijack the thread, but have you been able to keep the Voip.ms registration stable. Mine seems to go 408 about 50 times a day on two different PBXs, good thing they are only back up trunks. I am thinking of doing an IP registration but was just wondering if you have had better luck.

Snomone 5.04 Debian 64 On a 16 phone system with 8 of the phones external, is the Maximum number of SIP connections per second set to 2 still safe or should I bump this up?

Hi Guys, Snomone V 5.0.4 Debian 64 I have a slight issue, a call will come in and the Request URI is sip:2085551234@000.000.000.000 (Number and IP changed for security). The DID 2085551234 is on Huntgroup 371. Intermittently, the system will give a 404 error back to the sip server when that DID is dialed. Does this happen a lot? I have PCAPs of a rejected call if that helps.

Sorry, it is windows 8 as the client. I will check the firewall later to see if that is an issue.

I have a similar problem, I can upload and import csv from Windows 7 just fine but in Windows 8, I cant upload or import csv files in Chrome, Firefox or IE. This is on 4.x.1090 and 5.0.4

So I just experimented with my Yealink T26P, I had to make a transfer button to 500 (Park orbit) to transfer the call to the orbit and a BLF to 500 to pick up and monitor the call. If they are comfortable transferring to 500 manually they will have to press the transfer soft button, press 500 then press the transfer button again. I would program a BLF button to pick it up, also to see if someone is already in the parking spot. I think it is easier to just make the transfer button and the BLF button, especially if the client is like most of mine, they just want it stupid simple. Steve

I do it with Grandstream, I just set the BLF to the parking spot number. For instance I park calls on 500 - 501, I just set up the BLF to 500 and another to 501. I can transfer to 500 or the corresponding BLF without having to enter the * code. On Grandstream, you have to dial the extension manually to alert the phone of the BLF, then the BLF works.