Steve B
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Posts posted by Steve B
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Hi,
I am glad that you brought out this topic. I have been looking for solutions and have already tried a couple that I can share. I already checked the MB-D38 that appears on the following webpage: http://www.ewayco.com/100-low-cost-pc-products-low-cost-systems-embedded-systems-servers-lcd-pc-panel-pc.html I added a 2.5" 32G SSD and 2G of DDR3 RAMto round it up and it does work. Its pros are: It does work, Debian Linux 64 bits runs there with no problems, its small form factor. Its cons, at least to me are: Coming from China to my country, some 45% of cost is shipping and duties. Although it is Atom based, it does have a couple of small fans which in such a reduced form factor makes the solution a bit noisy, which may not be important in some sites, but it may be a problem in others. Something that I do not know if it is a con or a pro, but that grabbed my attention, is that the chassis appears to be made of thick caliber steel. It would not surprise met hat it could stop a bullet. I also tried assembling a pc myself. All the needed parts are available in my town, exception made of a small good looking chassis and corresponding power supply.The closest chassis that I found was a slim mid tower unit. Not so cool looking, but the unit worked and was affordable. That partial success encouraged me to a next project. Import just the chassis and PSU and find everything else at home. The chassis that I will try next is the M350 http://www.mini-box.com/M350-universal-mini-itx-enclosure I will let you guys know. The chassis is still somewhere on the road.
Cheers!
I am thinking of building an m350 also, looks very promising with all the mounting options.
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Do you know if nexsip is still in business? I tried to email sales at nexsip.com and it bounced. Also, many of the links on the web site are broken.
They are now Sumofiber, info@sumofiber.com 801-320-1000.
They have been very good to my customers.
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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...
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SNOM sold there PBX software to them, the whole thing (SNOMONE).
I have just found out and had a call from them
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.
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Yea, this is becoming a useless pain in the neck. We'll change the algorithm so that the user gets only blacklisted if there are several real attempts to get into the system without having the right credentials.
Haha, the sooner the better. Nothing like 50+ emails saying i'm blacklisted until I white list my IP.
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Could this be related to rtcp? I can try turning it off and seeing if that helps.
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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.
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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
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How do you want intercom and speed dial on the same key?!
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.
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This would be a GREAT feature.
snomone is most attractive for people migrating from Key Systems. most key systems by default intercom not ring.
second this motion!
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...
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If you're wondering where the Plug and Play Parameters section is located check this article.
http://wiki.snomone.com/index.php?title=Plug_and_Play#PNP_Firmware
FYI, all the video links are dead.
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So it appears that maybe it is the one way audio and not a true DTMF that I am having? Which codec works best with these units?
G711U is the only codec I used with it. I can't recall getting one way audio with it though.
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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.
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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.
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.
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OK. It is now picking up but it is not hearing the keyed input/tones. It is also not showing caller ID.
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.
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With the analog part I can not help you... However on the SIP side, if you can, use a VPN that makes the setup a lot more easier. Then you can just use a gateway trunk with full control over all headers, so that it should be easy to get the gateway working.
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.
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I generated two test scenarios without TCP connection between pbx and the phones.
192.168.0.220 = Patton
192.168.0.200 = snomONE 5.0.5
192.168.0.206,207 and 214 = snom370-SIP 8.7.3.19 in huntgoup stage one
normal cancel message after 5 seconds ringing(between line 9 and 45):
cancel after 20 milliseconds ringing (between line 9 and 10) with ghost ringing on all stage one extensions :
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.
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After upgrading to 5.0.5, are any tweaks required on the trunks? After I upgraded, I mysteriously can not make outbound calls anymore. Incoming calls work just fine. The two trunks that are having trouble are CallCentric and VoIP.ms. Luckily, I have a backup; but just wondering if there's something obvious that I'm missing?
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.
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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?
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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.
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Windows 8 on the PBX or on the client? Anything with the personal firewall?
Sorry, it is windows 8 as the client. I will check the firewall later to see if that is an issue.
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I checked the snom one technical manual and it makes no reference to uploading audio. I would assume by the snomone webUI that you can upload audio due to the fact you can browse files on the local machine, but im not finding documentation provided by snom or anyone else that outlines the requirements of uploading audio from the webUI.
A heads up from snom one letting me know if this is possible and the requirements would be nice. Good customer service...you know
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
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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.
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
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I've got a question I couldn't find a topic home for. Basically we have a customer that is going to have several Yeahlink phones. On these phones they want the ability to see a call put on hold. This would be indicated by a light of some sorts. Then able to be answered. by presssing a button or two
How can we make this a reality using these phones? There's tons of BLF settings i just don't know where to start.
Thanks
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.
Vitelity Registrations
in Trunk Setup
Posted
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.