AG1

Yes PnP Here is something I noticed last night, if I dial a number from the buttons the calls never go through and a message comes up saying the call cannot be completed as dialed BUT if I select the number on the touchscreen dialed numbers screen the calls go through because even though a 9 is dialed (selecting my Nextiva trunks) the calls go through on my Sangoma Netborder express card 2013/08/20 07:25:51A Conference Room (2203@192.168.100.151) 97015720767 00:07 Nextiva 2013/08/20 07:25:11A Conference Room (2203@192.168.100.151) 94062094291 00:02 Nextiva 2013/08/20 07:24:36A Conference Room (2203@192.168.100.151) 94062094291 00:26 Netborder Express 2013/08/20 07:24:12A Conference Room (2203@192.168.100.151) 97015720767 00:10 Netborder Express 2013/08/20 07:23:33A Conference Room (2203@192.168.100.151) 94783057 00:03 Nextiva 2013/08/20 07:22:55A Conference Room (2203@192.168.100.151) 94783057 Nextiva 2013/08/20 07:21:38A Conference Room (2203@192.168.100.151) 94783057 00:10 Netborder Express [5] 2013/08/20 07:24:17: SIP Rx tls:192.168.100.93:3372: ACK sip:2203@192.168.100.151:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-abtasztybeiq;rport From: "Conference Room" <sip:2203@192.168.100.151>;tag=hpp68z45m6 To: <sip:97015720767@192.168.100.151;user=phone>;tag=1aea6c0ddd Call-ID: b86e1352c5a3-ux2gbyham388 CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:2203@192.168.100.93:3372;transport=tls;line=zl1csrsm>;reg-id=1 Proxy-Require: buttons-snom870 Content-Length: 0 [8] 2013/08/20 07:24:17: Packet authenticated by transport layer [8] 2013/08/20 07:24:27: Last message repeated 20 times [5] 2013/08/20 07:24:27: SIP Rx tls:192.168.100.93:3372: BYE sip:2203@192.168.100.151:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-hohlier5joxz;rport From: "Conference Room" <sip:2203@192.168.100.151>;tag=hpp68z45m6 To: <sip:97015720767@192.168.100.151;user=phone>;tag=1aea6c0ddd Call-ID: b86e1352c5a3-ux2gbyham388 CSeq: 3 BYE Max-Forwards: 70 Contact: <sip:2203@192.168.100.93:3372;transport=tls;line=zl1csrsm>;reg-id=1 User-Agent: snom870/8.7.3.19 RTP-RxStat: Total_Rx_Pkts=730,Rx_Pkts=724,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=721,Tx_Pkts=721,Remote_Tx_Pkts=2 Proxy-Require: buttons-snom870 Content-Length: 0 [8] 2013/08/20 07:24:27: Packet authenticated by transport layer [5] 2013/08/20 07:24:27: SIP Tx tls:192.168.100.93:3372: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-hohlier5joxz;rport=3372 From: "Conference Room" <sip:2203@192.168.100.151>;tag=hpp68z45m6 To: <sip:97015720767@192.168.100.151;user=phone>;tag=1aea6c0ddd Call-ID: b86e1352c5a3-ux2gbyham388 CSeq: 3 BYE Contact: <sip:2203@192.168.100.151:5061;transport=tls> User-Agent: snomONE/5.0.10 Content-Length: 0 [7] 2013/08/20 07:24:27: 1aad7a3e@pbx: Media-aware pass-through mode [8] 2013/08/20 07:24:27: Clearing call port 298, SIP call id b86e1352c5a3-ux2gbyham388 [8] 2013/08/20 07:24:27: Call port 299: state code from 200 to 486 [8] 2013/08/20 07:24:27: Remove leg 4390: Call port 298, SIP call id b86e1352c5a3-ux2gbyham388 [5] 2013/08/20 07:24:27: SIP Tx udp:192.168.100.151:5066: BYE sip:NetborderExpressGateway@192.168.100.151:5066;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-71d541664ae945d23b8a7c845ced56ea;rport From: "Conference Room" <sip:4064888066@192.168.100.151;user=phone>;tag=366627966 To: <sip:97015720767@192.168.100.151;user=phone>;tag=ds-2c2e9ec6-e413e3b4 Call-ID: 1aad7a3e@pbx CSeq: 21846 BYE Max-Forwards: 70 Contact: <sip:4064888066@192.168.100.151:5060;transport=udp> P-Asserted-Identity: "Netborder Express" <sip:192.168.100.151:5066> Content-Length: 0 [8] 2013/08/20 07:24:27: Hangup: Call 298 not found [8] 2013/08/20 07:24:27: Last message repeated 2 times [5] 2013/08/20 07:24:27: SIP Rx udp:192.168.100.151:5066: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-71d541664ae945d23b8a7c845ced56ea;rport=5060 From: "Conference Room" <sip:4064888066@192.168.100.151;user=phone>;tag=366627966 To: <sip:97015720767@192.168.100.151;user=phone>;tag=ds-2c2e9ec6-e413e3b4 Call-ID: 1aad7a3e@pbx CSeq: 21846 BYE Content-Length: 0 [5] 2013/08/20 07:26:06: SIP Tx tls:192.168.100.93:3372: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-a6oxayp0t2cs;rport=3372 From: "Conference Room" <sip:2203@192.168.100.151>;tag=uhjcug5i0f To: <sip:97015720767@192.168.100.151;user=phone>;tag=46efdabc08 Call-ID: 246f13529cad-iwcc2s3ayfqn CSeq: 3 BYE Contact: <sip:2203@192.168.100.151:5061;transport=tls> User-Agent: snomONE/5.0.10 Content-Length: 0 [7] 2013/08/20 07:26:06: d6463b23@pbx: Media-aware pass-through mode [8] 2013/08/20 07:26:06: Clearing call port 304, SIP call id 246f13529cad-iwcc2s3ayfqn [8] 2013/08/20 07:26:06: Remove leg 4396: Call port 304, SIP call id 246f13529cad-iwcc2s3ayfqn [8] 2013/08/20 07:26:06: Call port 305: state code from 200 to 486 [8] 2013/08/20 07:26:06: Hangup: Call 304 not found [8] 2013/08/20 07:26:06: Last message repeated 2 times [5] 2013/08/20 07:26:06: SIP Tx udp:208.73.146.95:5060: BYE sip:7015720767@208.73.146.95:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-07bd5d0424065697730836ee33cc851e;rport From: "Nextiva" <sip:14062094291@208.73.146.95>;tag=1878955888 To: <sip:7015720767@208.73.146.95>;tag=3585994152-502418 Call-ID: d6463b23@pbx CSeq: 31552 BYE Max-Forwards: 70 Contact: <sip:14062094291@192.168.100.151:5060;transport=udp> Remote-Party-ID: "Conference Room" <sip:4064888066@192.168.100.151;user=phone> Content-Length: 0 [5] 2013/08/20 07:26:06: SIP Rx udp:208.73.146.95:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.151:5060;received=216.228.51.194;branch=z9hG4bK-07bd5d0424065697730836ee33cc851e;rport=13291 From: "Nextiva" <sip:14062094291@208.73.146.95>;tag=1878955888 To: <sip:7015720767@208.73.146.95>;tag=3585994152-502418 Call-ID: d6463b23@pbx CSeq: 31552 BYE Content-Length: 0 [7] 2013/08/20 07:26:06: Call d6463b23@pbx: Clear last request [5] 2013/08/20 07:26:06: BYE Response: Terminate d6463b23@pbx [8] 2013/08/20 07:26:06: Clearing call port 305, SIP call id d6463b23@pbx [8] 2013/08/20 07:26:06: Remove leg 4397: Call port 305, SIP call id d6463b23@pbx [8] 2013/08/20 07:26:07: Packet authenticated by transport layer [8] 2013/08/20 07:26:23: Last message repeated 19 times [8] 2013/08/20 07:26:23: Trunk 4: Preparing for re-registration [8] 2013/08/20 07:26:23: Trunk Nextiva: Sending registration to 208.73.146.95 [8] 2013/08/20 07:26:23: Trunk 4: setup callback to send re-registration after 37 seconds [8] 2013/08/20 07:26:23: Packet authenticated by transport layer [8] 2013/08/20 07:27:00: Last message repeated 60 times [8] 2013/08/20 07:27:00: Trunk 4: Preparing for re-registration [8] 2013/08/20 07:27:00: Trunk Nextiva: Sending registration to 208.73.146.95 [8] 2013/08/20 07:27:00: Trunk 4: setup callback to send re-registration after 38 seconds [8] 2013/08/20 07:27:01: Packet authenticated by transport layer [8] 2013/08/20 07:27:15: Last message repeated 21 times [5] 2013/08/20 07:27:15: Identify trunk (IP address/port and domain match) 3 [8] 2013/08/20 07:27:16: Packet authenticated by transport layer [8] 2013/08/20 07:27:35: Last message repeated 30 times [8] 2013/08/20 07:27:35: Could not find a trunk (3 trunks)

I cannot get my 870's to dial out on my Nextiva SIP trunks The 820s work fine so it has to be an 870 firmware issue....does it not? If so does the beta version 8.7.4.8 fix this problem? I have tried every possible combination of "1" or "Area code" , "country code"in the general settings and I have even tried some wildcard stuff in the dial plan, but like I said the 820s work fine its only on our touchscreen 870s Is anyone else having this problem with 870's or am I just the lucky one again?

I still cannot get my 870's to dial out on my Nextiva SIP trunks The 820s work fine so it has to be an 870 firmware issue....does it not? If so does the beta version 8.7.4.8 fix this problem? Is anyone else having this problem with 870's or am I just the lucky one again?

I have call waiting enabled Still doesnt go to that extensions voicemail. (Yes their voicemail box is set up)

I have a Snom One Plus and I am having a few problems with voicemail. When a user is on the phone and a call is transferred to them, the calls do not go to voicemail. When somebody in our office dials an extension it rings but if the person at that extension doesnt pick up an error message comes on saying you dont need to dial a 1 when dialing this number instead of going to voicemail. Any suggestions

Cool Thanks

[5] 2013/07/03 08:38:27: Sending IM from "Mike" <sip:2202@192.168.100.151> to "Mike" <sip:2202@192.168.100.151> (1 destinations) This happens on every extension all the time, its not doing anything to make the system not work I just wanted to know what this is and why this is happening. Thanks, AG1

Right now I am not using a country code. Do most people use that or not? Pros...Cons? Thanks for the other info, much appreciated

I am having trouble setting up a dial plan that allows me to dial local numbers with or without the area code or a 1 in front of the area code. This is troubling because the cell phone numbers need to have the area code in front of them to call different city cell numbers. Any ideas?

So if you know this fixes the low audio problem.......why hasnt Snom fixed the problem in their newer firmware updates? It would be cool if they did...........................

Thank you Marc.....I will try that

Says call cannot be completed as dialed or you need to dial a one first even though I am trying that call with a 1

Another problem on my Sangoma card is that as of last Friday we are not able to dial local cell phones from the Snom One plus. I can use my VoIP trunk to dial them and they are available cell phone to cell phone just not using the Sangoma Gateway. Other local numbers are accessible.

I have a Snom One Plus with a Sangoma 4 FXO card that we use with our 4 analog lines. I am still having the same if not more MULTIPLE problems with this equipment and right now I am so frustrated I feel like quitting and getting a new system. Problems are as follows: Low incoming audio levels on my Sangoma A200 card that are very evident on my Snom 821 phones 870 phones that will now not dial out on my ITSP (Nextiva) I can use the 821's just fine its only the 870's they will still receive calls If I adjust the incoming audio on the Sangoma card the unit fails and will not let me dial out it only works on factory defaults. I have been trying to configure a Patton 4114 since last Saturday and I am about ready to send it back. This isnt even a complete list its just all I can think of right now after being up until 2:30am working on this stuff I know this may not sound like much to you guys that have been doing this stuff for years but for me and my first time installing this equipment with no training on any of it its beyond miserable. I just had this project dumped on me..."here do this". I dont mean this with any offense in any way but the responses I have got on here though greatly appreciated have not helped very much other than focusing the area that needs to be looked at. Part of that may be my inability to ask the right questions because in some way I am in the dark on what is even going on. Thanks again for the help and letting me vent.

I uninstalled the program and reinstalled it. Place the same information in it that I had before same result.......then I cleared everything out of the boxes and added the same domain and password etc.............still nothing. I closed the program opened it 10 mins ago and entered the same parameters and it came up and registered. so.......great...one of my problems is not there anymore.

Yeah, I never have been able to get mine to register. Still cant

On my Nextiva VoIP trunk the status doesnt say enabled just the following? 200 OK (Refresh interval 38 seconds) I can still receive incoming calls but cannot call out? Any ideas?

Also I have noticed that since I had some problems a week ago while we upgraded firmware on my Sangoma A200 I cannot call out on my VoIP lines/trunks. Incoming works fine ?????

I am reading through the Snom One book and I see I should be able to make entries in a dial plan in tandem to ensure a call goes through even if I lose my internet connection. I currently have 2 SIP trunks and I also have a Sangoma gateway with 4 FXO ports. In the book they use 1978* in the Pattern field and again in the Replacement field Is this the correct code or pattern to use to accomplish redundancy?

I created 5 agent groups and none of them are actively being used. I was doing this to mess around and get familiar with the setup and I currently only have 2 licenses. I wanted to delete the 3 I wont be using and they wont delete when I check the box and click delete. Is this a bug? I know it doesnt matter if they are there and not being used just wondering what I was doing wrong. Thanks,

I am toying around with nesting agent groups and have 6 AGs. The first 2 I can enable and disable but the other 4 wont. it says they are not licensed. So how do you get them licensed? Is this something extra you have to purchase?

OK, I think I have this working now but I do have one more question. I can now see the buttons turn green when the call ring in, they turn yellow when somebody is on that line etc but is there a way to edit that button to display the extension that in on that line?

I have 4 PSTN lines coming into a Sangoma A200 card and I want to make those lines accessible on my 870 virtual keys. 1. Is this even possible? 2. I have already selected the buttons tab and edited the button plan for the 870 using the following settings "shared line" - "CO1-CO4" for the parameter and labeled them lines 1-4 Doesnt work I was wondering if this is even possible with a Sangoma card?

Ok, I have yet another glitch/question for you guys. When I open up IE and log into my extension the web interface comes up and DOES NOT display the Extensions that are on or off the phone. the page is blank other than calls missed calls etc. This works when I use Chrome or Firefox.........any ideas?

Orange error message - call-id=1367931853-627024-955298136-162 Failed to send packet: Some other RTP error