[Call drops after a minute]

The only thing that sticks out to me is that you have only one CO line. But that would not explain 66 seconds. I assume the PBX is behind a NAT router? This is the #1 reason for trouble... Do you have more than one IPv4 address on the host? Maybe you can just filter out the SIP traffic and send a link in a private message to me and then we can take a look.

 

Thanks I did send you a PM.

 

I have only 1 Co Line because they have only 3 avail and 2 are in a different trunk to be able to send out different DID.

 

Calls go out in 2 different Trunks for that reason. Incoming trunk will be never used probably.

[Call drops after a minute]

Hello I have a snomONE Plus with Broadvox as SIP trunk provider, and after 66 seconds the PBX asks Broadvox to reset and when broadvox does the PBX never replays back and we get a one sided audio only until the call drops all together.

here is my Trunk Setup and a pcap from broadvox.

 

Capture is too big to post here. I can send if you need it.

[SAngoma Failed to resolve URI]

All the sudden no calls can be made out.

 

The sangoma error message is Yellow with this legend

 

Failed to resolve hostname of "From" URI sip:200@pbx.xxxxx.com

 

HELP

[Hot to divide 2 trunks from 2 Sangoma cards]

For outbound, you can create an "Only outbound" trunk on each domain with 4 co-lines and then use it in the dial plans of the respective domains.

 

For the right caller-id, you can put that in the trunk ANI or the Domain ANI.

 

Everybody is part of the 2 companies, so I guess I'll need co lines, but I can't figure it out how the sangoma will identify this.

[Hot to divide 2 trunks from 2 Sangoma cards]

Hello, I have a multi-company situation, using 2 sangoma cards right now, 4 Lines should go to 1 company and the other 4 to the other.

Inbound is fine I set up the DID's on the Auto atendants,but Outband I can't find a way to make the calls going out wiht the right caller ID.

[DTMF problems & Sangoma Card]

Here is the second Auto Attendant. First one is basically pres 1 press 2 .

 

It will go anywhere in the list but not if I dial an Extension while the recording is going.

 

If you want send me a provate and i can give you IP and access if you want to take a look at the PBX

[DTMF problems & Sangoma Card]

looks like the pbx detected the DTMF from the sangoma card.

[7] 2012/01/28 08:56:00: Received RFC4733 DTMF on codec 101

[6] 2012/01/28 08:56:00: Received DTMF 2

[8] 2012/01/28 08:56:00: call port 30: state code from 200 to 200

[8] 2012/01/28 08:56:00: Play audio_moh/noise.wav

[6] 2012/01/28 08:56:01: Received DTMF 1

[8] 2012/01/28 08:56:01: Packet authenticated by transport layer

[6] 2012/01/28 08:56:01: Received DTMF 2

[8] 2012/01/28 08:56:03: Packet authenticated by transport layer

Can you add 212 to the Direct Destinations options and dial 1.

 

Lets see if the extension can be reached.

 

No it doesn't that is the problem, it seams to get the dtmf but it doesn't go any where.

I added the trunk to my prior post probably after you replied to it

[DTMF problems & Sangoma Card]

OK I'm remote so I can't get to the Office phones right now.

But here is the inbound call log.

 

I'm also attaching the Trunk setup. Note that I already tried with the Out of band DTMF

 

[8] 2012/01/28 08:55:49: Last message repeated 34 times

[9] 2012/01/28 08:55:49: UDP: Opening socket on 0.0.0.0:64064

[9] 2012/01/28 08:55:49: UDP: Opening socket on 0.0.0.0:64065

[9] 2012/01/28 08:55:49: UDP: Opening socket on [::]:64064

[9] 2012/01/28 08:55:49: UDP: Opening socket on [::]:64065

[9] 2012/01/28 08:55:49: Resolve 19925: aaaa udp 127.0.0.1 5066

[9] 2012/01/28 08:55:49: Resolve 19925: a udp 127.0.0.1 5066

[9] 2012/01/28 08:55:49: Resolve 19925: udp 127.0.0.1 5066

[7] 2012/01/28 08:55:49: Set packet length to 20

[6] 2012/01/28 08:55:49: Sending RTP for 2a51371a-1dd2-11b2-8cc6-fbcd0b20cc98@S1-MYPBX to 127.0.0.1:14006, codec not set yet

[8] 2012/01/28 08:55:49: Incoming call: Request URI sip:2148281088@localhost:5060;transport=udp;user=phone, To is "2148281088" <sip:2148281088@localhost:5060>

[8] 2012/01/28 08:55:49: Trunk Netborder Express@pbx.MYPBX.com has country code not set, area code not set

[9] 2012/01/28 08:55:49: Incoming: formatted From is = "GRANDPRARI TX" <sip:12144486474@127.0.0.1:5066;user=phone>

[9] 2012/01/28 08:55:49: Incoming: formatted To is = "2148281088" <sip:2148281088@localhost:5060;user=phone>

[9] 2012/01/28 08:55:49: Incoming: formatted URI is = sip:2148281088@pbx.MYPBX.com:5060;transport=udp;user=phone

[8] 2012/01/28 08:55:49: Set the To domain based on To user 401@pbx.MYPBX.com

[7] 2012/01/28 08:55:49: set_codecs: for 2a51371a-1dd2-11b2-8cc6-fbcd0b20cc98@S1-MYPBX codecs "0", codec_preference count 2

[8] 2012/01/28 08:55:49: Play audio_moh/noise.wav

[8] 2012/01/28 08:55:49: call port 30: state code from 0 to 183

[7] 2012/01/28 08:55:49: Set packet length to 20

[9] 2012/01/28 08:55:49: update_codecs for 2a51371a-1dd2-11b2-8cc6-fbcd0b20cc98@S1-MYPBX: adding codec pcmu/8000 to available list

[9] 2012/01/28 08:55:49: update_codecs for 2a51371a-1dd2-11b2-8cc6-fbcd0b20cc98@S1-MYPBX: codec_preference size 2, available codecs size 2

[6] 2012/01/28 08:55:49: Codec pcmu/8000 is chosen for call id 2a51371a-1dd2-11b2-8cc6-fbcd0b20cc98@S1-MYPBX

[9] 2012/01/28 08:55:49: Resolve 19926: aaaa udp 127.0.0.1 5066

[9] 2012/01/28 08:55:49: Resolve 19926: a udp 127.0.0.1 5066

[9] 2012/01/28 08:55:49: Resolve 19926: udp 127.0.0.1 5066

[9] 2012/01/28 08:55:49: Resolve 19927: aaaa udp 127.0.0.1 5066

[9] 2012/01/28 08:55:49: Resolve 19927: a udp 127.0.0.1 5066

[9] 2012/01/28 08:55:49: Resolve 19927: udp 127.0.0.1 5066

[8] 2012/01/28 08:55:49: Packet authenticated by transport layer

[8] 2012/01/28 08:55:50: Play recordings/att110.wav space20

[8] 2012/01/28 08:55:50: call port 30: state code from 183 to 200

[9] 2012/01/28 08:55:50: Resolve 19929: aaaa udp 127.0.0.1 5066

[9] 2012/01/28 08:55:50: Resolve 19929: a udp 127.0.0.1 5066

[9] 2012/01/28 08:55:50: Resolve 19929: udp 127.0.0.1 5066

[8] 2012/01/28 08:55:55: Packet authenticated by transport layer

[8] 2012/01/28 08:56:00: Last message repeated 4 times

[7] 2012/01/28 08:56:00: Received RFC4733 DTMF on codec 101

[6] 2012/01/28 08:56:00: Received DTMF 2

[8] 2012/01/28 08:56:00: call port 30: state code from 200 to 200

[8] 2012/01/28 08:56:00: Play audio_moh/noise.wav

[6] 2012/01/28 08:56:01: Received DTMF 1

[8] 2012/01/28 08:56:01: Packet authenticated by transport layer

[6] 2012/01/28 08:56:01: Received DTMF 2

[8] 2012/01/28 08:56:03: Packet authenticated by transport layer

[DTMF problems & Sangoma Card]

Hello, recently installed a Snom one plus with 2 sangoma cards on it.

 

I can receive and make calls fine.

 

Some of the DTMF work, like when pressing 1 2 and so, (just one digit. but if they dial an extension it wont go anywhere.

 

Here is the Log file of 2 different calls.

 

Log 1 INFO:

- Called into Attendant, and dialed extension 212. Nothing happened, I hanged up.

- 401 is main Attendant

- 212 is an extension.

-Attached image is the settings on 401 Attendant

-------------------------------- LOG 1 ------------------------------------------

[5] 2012/01/27 16:12:43: Identify trunk (IP address and DID match) 1

[7] 2012/01/27 16:12:43: Set packet length to 20

[6] 2012/01/27 16:12:43: Sending RTP for 08cd5ede-1dd2-11b2-b7ac-97770de55725@S1-XXXXX to 127.0.0.1:14182, codec not set yet

[5] 2012/01/27 16:12:43: Domain trunk Netborder Express@pbx.XXXXX.com sends call to 401 in domain pbx.XXXXX.com

[7] 2012/01/27 16:12:43: set_codecs: for 08cd5ede-1dd2-11b2-b7ac-97770de55725@S1-Sliquid codecs "0", codec_preference count 2

[7] 2012/01/27 16:12:43: Set packet length to 20

[6] 2012/01/27 16:12:43: Codec pcmu/8000 is chosen for call id 08cd5ede-1dd2-11b2-b7ac-97770de55725@S1-XXXXX

[7] 2012/01/27 16:12:54: Received RFC4733 DTMF on codec 101

[6] 2012/01/27 16:12:54: Received DTMF 2

[6] 2012/01/27 16:12:55: Received DTMF 1

[6] 2012/01/27 16:12:55: Received DTMF 2

--------------------------------END Log 1 ---------------------------------------

 

Log 2 INFO: (Just 1 digit option works fine)

- Called into attendant, selected Option 2 (secondary Attendant), then SELECTED option 3 (Agent Group) and it took me to a voicemail since it is the normal when no agents are logged on.

 

------------------------------ LOG 2 -------------------------------------------

[5] 2012/01/27 16:30:32: Identify trunk (IP address and DID match) 1

[7] 2012/01/27 16:30:32: Set packet length to 20

[6] 2012/01/27 16:30:32: Sending RTP for 862e1dd0-1dd2-11b2-af4c-fe9c9b9c9ee8@S1-Sliquid to 127.0.0.1:14188, codec not set yet

[5] 2012/01/27 16:30:32: Domain trunk Netborder Express@pbx.xxxxx.com sends call to 401 in domain pbx.xxxxx.com

[7] 2012/01/27 16:30:32: set_codecs: for 862e1dd0-1dd2-11b2-af4c-fe9c9b9c9ee8@S1-Sliquid codecs "0", codec_preference count 2

[7] 2012/01/27 16:30:32: Set packet length to 20

[6] 2012/01/27 16:30:32: Codec pcmu/8000 is chosen for call id 862e1dd0-1dd2-11b2-af4c-fe9c9b9c9ee8@S1-xxxxx

[7] 2012/01/27 16:30:44: Received RFC4733 DTMF on codec 101

[6] 2012/01/27 16:30:44: Received DTMF 2

[6] 2012/01/27 16:31:07: Received DTMF 3

[6] 2012/01/27 16:31:07: ACD 601: Set priority to 1

[7] 2012/01/27 16:31:07: set_codecs: for 2c8d6786@pbx codecs "", codec_preference count 7

[7] 2012/01/27 16:31:08: Call 2c8d6786@pbx: Clear last request

[7] 2012/01/27 16:31:27: Last message repeated 2 times

[7] 2012/01/27 16:31:27: Call 2c8d6786@pbx: Clear last INVITE

[5] 2012/01/27 16:31:27: INVITE Response 487 Request Terminated: Terminate 2c8d6786@pbx

[6] 2012/01/27 16:31:35: Reg 156, Sent MWI notification to user 200@pbx.xxxxx.com

-----------------------------END Log 2 -------------------------------------------

 

Please HELP, thank you.

[Snom ONE PLUS - Static IP setup]

I can't find the way to setup a Static IP on the snom one plus.

 

Also can't decide if is better to use the Lan port or Wan port. I will have phones outside the office, just 2 but I guess WAN will be correct unless I stablish a VPN and use LAN. correct?

 

still can't setup a static ip on ether of them.

 

HELP

[Licensing]

Hello any body out there?

[Licensing]

Can't it says snom one is a groupor something like that.

 

pbx_support? in here or @snom.com?

 

 

The member snom ONE cannot receive any new messages

 

Can you send me a PM with your User? same thing happens with pbx_support

[Licensing]

You can send me a private message?! Anyway if that does not work send it to pbx_support instead.

 

Can't it says snom one is a groupor something like that.

 

pbx_support? in here or @snom.com?

[Licensing]

Still not working where should I send the key?

Won't let me send to snom one

 

Hello, thank you for your answer.

 

I have a cable on the wan plier to my router (privaye IP) and another for the internal LAN and is still not activating.

[Licensing]

I just started a new snom ONE, I got the License on the back of the book that is 12 characters long xxx-xxx-xxx-xxx (groups of 3) the Help shows groups of 4, it Reads Activation Key, but every time I try to activate it wont do it. Does it need to be connected to the internet to activate? It doesn't say that any where, I'm pre-configuring it, so didn't think it will be necessary.

 

I'm trying to get it ready for monday

 

Ideas?

[Need help Setting multiple DID outgoing calls]

Right, but you have to register only the inbound trunk. The outbound trunks are not registering, they are just sending the (outbound) traffic to the same location where the inbound trunk is registering to; using the gateway mode. So that the carrier does not "see" that you actually have multiple trunks.

 

OK sounds perfect then.

To be clear.

 

I just need to point to same IP as the Main trunk and not even user and password matter? or it does?

[Need help Setting multiple DID outgoing calls]

Well, my point is that you'll have to split hte trunk up. You can keep your inbound trunk as is (just change it to inbound only), and then add two more trunks for outbound. If you are using a registration trunk, those two outbound trunks should be changed to gateway/outbound only with the same outbound proxy like the inbound trunk.

 

Ok that sounds interesting, but I don't know or think my SIP provider will let me login twice with he same trunk.

 

Let's see if I got it right, same trunk I have now, just use one for inbound and separate it into 2 for outbound. in one I can assign Houston in the other just Dallas.

 

Correct?

[Need help Setting multiple DID outgoing calls]

You can make the DID (actually, in this case it is called ANI) dependent on the trunk that is being used. In the ANI field, you can prepend the ANI with the trunk name. Use space to seperate the multiple choices. For example, "Houston:2811234567 Dallas:9721234567 2121234567" would mean if the call goes out on the Houston trunk, use 2811234567, if it goes out on the Dallas trunk, use 9721234567 and otherwise use 2121234567. This implies that trunk names must not have spaces in the name. If you are using the same trunk for everything right now, you should consider splitting it up into one inbound trunk and two outbound trunks (to avoid two overlapping inbound trunks).

 

I thought on making it more explanatory, my bad I guess.

 

I don't have multiple Trunks. Just one Trunk. many DID's

 

That is my problem, IF I had multiple Trunks I could use the Dial Plan and the ANI on the Trunks, I knew as much. Problem is how to do it with different DID's and only 1 Trunk.

[Need help Setting multiple DID outgoing calls]

Hello guys.

 

I just setup a new snomOne and the client wants to be able to choose when to send a DID on the outgoing calls.

 

Here is an example.

They have DID's for Houston area and Dallas area. SO if they dial a Houston Number they want to show the Houston DID they have in the caller ID of the person they are calling.

 

THey are ok with making the selection manually, but If we can do it auto will be great.

 

Any ideas how can I do this?

[In Cloud PBX]

Read my prev post - SPA2102

 

You right, I'm loosing my mind. Sorry.

[In Cloud PBX]

on your 3.0GHZ servers we try to keep the concurrent calls to about 30. I have seen them go a little higher, but that is our magic number.

 

All of our hosted customers are on dedicated blade servers we do not share resources. our blades go from 10 concurrent calls to 35. our 1.5Ghz blades are 10 concurrent calls our 3.8GHZ blades are 35.

 

db

 

It seams a lot of resources for just 30 CCS.

 

I've been checking the snom ONE and Processor is not used that much.

 

What PBX you guys use?

[In Cloud PBX]

OK I'm checking nexsip, now that will fix the fax side, what about the Credit Card machines?

 

Any ideas on that?

[Conference Rooms & Multi PBX]

thanks. Pretty clear. Conference Shut down ;-)

[In Cloud PBX]

How much will you guys run on this server:

Core 2 Duo E8400

1x Core2Duo E8400 Dual Core

2 GB

1x 250 GB SATA

 

OS Debian

 

I mean how many extensions and CCS will you guys be willing to run here?

[Conference Rooms & Multi PBX]

Hi guys, some of you may have seen my other posts, I'm trying to get a big amount of extensions and faxes setup together.

 

I'm pretty sure I'll have to get multiple PBX units probably 3 when all the stores we are trying to connect get switched over.

 

Now can I chain conference rooms? like the Conference room of one PBX dials the conference of the other PBX?

 

Is there any limit on the quantity of participants other than the amount of trunks? and the number of CCS the server hardware can handle?