Bradley_M

Alright, I've figured out what I need to do here is just keep asking the right questions and ignore the people not being beneficial on actual support issues. I have someone from pbxnsip that I'm talking to now and working to see if we can replicate problems. I did discover a new little gem in the system . . . if you are keying touch tones from a phone and the other side of the call disconnects, the channel will hang open. Th call never terminates, and no timeout ever occurs on the line either. I don't know what that problem is . . .but it's been handed off to pbxnsip to see if they can replicate. Another question I have is how the system decides if all channels are full? If the PSTN side never releases, I haven't figured out yet if the system will allow someone to dial out again, or if it will give service unavailable message. If it gives service unavailable message, then why wouldn't the CS410 be able to determine "hmm -- I show no calls online over here on the web side . . . what's up?"

From the latest eBay sale of the pbxnsip system: "Item Description PBXNSIP CS410. Appliance Single-Domain, 10 extensions and 10 accounts (Choose 10 of any of the following to meet your needs: Auto Atendant, Conference, Hunt Group, Agent Group, Calling Card, Paging, Service Flag, and IVR Node). Fully SIP compliant, built-in 4 port FXO gateway for PSTN connectivity. Now with additional WAN port. It is a perfect solution for small to medium sized business’s (SMB) that want to take advantage of VoIP’s benefits and have the option to keep their existing phone lines. The pbxnsip CS 410 contains 4 integrated FXO ports to connect to existing PSTN trunks and the system also supports IP trunks to connect to Internet Telephony Service Providers that support SIP. The system includes all available features from the other pbxnsip PBX editions like voicemail, auto attendant or conferencing, but also advanced features like NAT Transversal, call barge in or cell phone integration. Key features of the CS 410 include: * Built-in 4 port FXO PSTN gateway and SIP trunking * Paging and "Music on Hold" audio jacks * Full PBX features including Auto Attendant, Conference Server, and Voicemail * Unified Messaging integration with Microsoft Exchange 2007 UM * Plug and Play phone support and full security support with TLS and SRTP * Quiet, compact hardware design utilizing a Mindspeed Comcerto VoIP processor * Easily upgradable to 25 extensions The pbxnsip CS 410 is opening new markets for Value Added Resellers and Systems Integrators in the US and abroad. "Our company delivers high end VoIP solutions to Internet Telephony Service Providers in the UK and US", said Jonathan Greenwood, founder and CEO of Solutions11 Limited located in Aspect Court, Leeds, UK. "With the introduction of the CS 410, we now have an IP phone system that we can deliver to offices of 10 to 25 employees. We can integrate the system with IP phones from leading vendors like Polycom, snom, and Cisco and provide a plug and play solution. This opens up a brand new market segment for us." The pbxnsip CS 410 is shipping now and is available through pbxnsip channel partners in the US and throughout the world." Let me re-iterate, this says "plug and play solution" and I could really care less what your years of outdated experience indicate, today more and more solutions are based on very easy to integrate products that should be easy to diagnose and work with. If I fill my car with 91 octane fuel, or 87 octane, the engine nowadays is brilliant enough to listen for knock and retard the timing based on what it knows is going on. Ten+ years ago, that was around too, but didn't work right all the time . . . so nobody relied on that for consumers. This is 2008, the telephone system has been a known quantity for at least a few years. You want questions? Let's ask a few questions: 1) How can a device show FX0 ports as active, yet the web interface shows no calls? How do you resolve such a condition? Please don't tell me "reboot the device" in a production environment. The last thing I need is phones ringing open while the system comes back up, or the calls get routed to a voice mail box that will never be checked because the phones are still registering. 2) How can you find adequate logging information on this system? -- one of the questions (still unanswered Mr. Gunslinger) is what level of logging is required. 3) If this product isn't for general consumption -- why do some resellers have product to push on eBay? Most products with exclusivity to "installers only" have provisions in their sale to say only VAR installations are support in terms of warranty and support. My product(s) came with a couple of sheets of 8x11 pages printed from the wiki for installation . . and that was it. And that's what was provided by pbxnsip!

Most immediate problem is the call drop issue -- for no rhyme/reason. I've got very frustrated users in a staffing situation where they can't afford to have a call drop. It just happens to one person . . . seems to be after about 3 minutes or so, but not consistent on that time. ???

Actually I'm not doing too bad, but I'm really having an issue on finger pointing. I had an Asterisk box that had the occasional Digium card channel hang issue that would require a reset. No biggy. We bought this system to eliminate any problems like that and to get call transfers to work like a key system. Now I'm having calls dropping occasionally and not 100%, but might have channels hanging also. I had a very wacky roll-over line issue where the telco side was ringing channel without rolling line to our corp office. I *think* they got that fixed, but now wondering who to point at with the drop issue. Is it the FXO card in the CS410? Is it the telco? It's slighly odd that the problem also happened on the Asterisk/Digium stuff, so I'm leaning a bit towards that side of things. I'm ABSOLUTELY FRUSTRATED with Digital Communications (one of suppliers) . .. . they are selling units on eBay and then not standing behind the product -- I've called and emailed and received no response back. (Not that I believe they would be very helpful in the first place, but I'd still like a "ok, I tried this channel for escalation". Has anyone else dealt with call drops or channel hanging in an install?

Ok -- on my local network, I've been able to setup a port in DMZ with it's own IP address, assigned CS410 that IP address, and then hooked the whole shebang together . . . so far so good. I can dial between two phones, traffic flows perfectly, audio/etc... I then put one one on network inside my class C (192.168.x.x) and pointed to phone server . . . works great, audio/etc... goes right where it should. I'm working on getting two fied IP addresses for my other office, and then I'll set it up just like I have here, one connection to WAN with public IP and then internal on it's own. All this jumping through hoops just to figure out why calls are dropping and why FX0 ports are hanging occasionally. Grrrrrrr.

comcerto:~# more /proc/net/route Iface Destination Gateway Flags RefCnt Use Metric Mask MTU Window IRTT eth0 0001A8C0 00000000 0001 0 0 0 00FFFFFF 0 0 0 eth1 00010101 00000000 0001 0 0 0 00FFFFFF 0 0 0 eth0 00000000 0201A8C0 0003 0 0 0 00000000 0 0 0

Active Internet connections (w/o servers) Proto Recv-Q Send-Q Local Address Foreign Address State tcp 0 0 192.168.1.95:sip-tls 192.168.1.102:2082 ESTABLISHED tcp 0 0 192.168.1.95:sip-tls adsl-99-163-43-150:2141 ESTABLISHED tcp 0 0 192.168.1.95:sip-tls 192.168.1.101:2082 ESTABLISHED tcp 0 132 192.168.1.95:ssh adsl-99-163-43-150:1172 ESTABLISHED tcp 0 0 192.168.1.95:sip-tls 192.168.1.110:2086 ESTABLISHED This is what my one remote office shows -- the adsl connection is my local office phone connected up to there, and the SSH connection. My phone on my desk (local) is on 192.168.1.122 on this side . . . perhaps an issue since the other office is also on same class C?

I have CS410 at two locations . . . setup everything and I can call one location and phone rings, answers, but no audio either way. I took a phone home and tried it there, I can get "we're sorry" message from telco on the phone, but no audio otherwise. I'm pulling my hair out here . . . neither my simple WRTG54 nor my larger 16 port router are working right . . but I was able to get my asterisk box through the system before. Is there some troubleshooting things to look at -- and yes, I've looked at the WIKI on one way audio and that wasn't very useful at all. The phones are SNOM 360's on the latest firmware 7.1.30. Thanks!!!

Ok -- I turned on PSTN logging and will see what that turns up. What level logging should I be set to? I'm at level 5 right now.

I just upgraded everything about an hour ago . . . DBT is off, DDT is off, DPC is on. Should I change polarity and see what that does? My biggest issue is when facility calls in and is getting busy signal first. I definitely need those lines to be reflected properly. Why do the lines show hot via the LED's but not on the web interface?

Ok, about once a day, or perhaps every other day . . it's totally random, a person is on a call and the system will just drop them. That's issue#1. I don't know if this is related also, but apparently the system will forget FX0 ports and leave them hanging open. I went in and tried to call and got a busy signal (main line with extra line roll-over). I had the person at facility look and all FX ports were lit up . . . so I looked at the onscreen call status . . . it only showed two calls. Hmm. I ended up rebooting the appliance to get the calls to drop. I've attempted to contact my VAR - Digital Communications out of Bakersfield, CA and so far have not received any calls back. I sent an email to Kevin Moroz and his suggestion was contacting abptech and they said they hadn't hear of any thing like I was describing, but offered a $120/hour support contract. I replaced a Asterisk system with this CS410 and I'm really beginning to regret that decision . . . they were having operational issues with the Asterisk system, but at least calls weren't being dropped. What can I go to resolve these issues? I'm in the middle of evaluating this product for another identical office, as well as our corporate system, but unless I have support and a rock solid system I can't proceed. Any assistance would be greatly appreciated. Here's our stats: Snom 360 Phones - All on version 7.1.30 CS-410 (Black) Version: 3.0.0.2905 (Linux) Four POTS lines in with rollover from main DID.

I *think* I found the smoking gun here . . . is that WAN port still accessible despite no connection? I yanked the system and put it on my test network and it started working . . . moved it back and it died again . . . finally looked and decided to yank the WAN off fixed IP and put it on DHCP too temporarily . . . . viola . . . problem seems to be fixed. I suspect that when my users came in on Monday, .99 got handed out on DHCP request, causing issues . . . but I'm curious as to why the unhooked WAN would have had any influence here???? Hmm. My VAR still hasn't replied to my earlier message, and as of yet, the replacement device isn't here yet either. This could coincide with the documentation issues mentioned by other resellers in the forums that I've read about . . . hmm.

That didn't help either . . . still died.

Any hints on what to kill???? comcerto:~# ps aux|more USER PID %CPU %MEM VSZ RSS TTY STAT START TIME COMMAND root 1 13.0 0.4 1512 492 ? S 2007 0:10 init [2] root 2 0.0 0.0 0 0 ? SN 2007 0:00 [ksoftirqd/0] root 3 0.0 0.0 0 0 ? S< 2007 0:00 [events/0] root 4 0.0 0.0 0 0 ? S< 2007 0:00 [khelper] root 5 0.0 0.0 0 0 ? S< 2007 0:00 [kthread] root 6 0.0 0.0 0 0 ? S< 2007 0:00 [kblockd/0] root 7 0.0 0.0 0 0 ? S 2007 0:00 [pdflush] root 8 0.0 0.0 0 0 ? S 2007 0:00 [pdflush] root 10 0.0 0.0 0 0 ? S< 2007 0:00 [aio/0] root 9 0.0 0.0 0 0 ? S 2007 0:00 [kswapd0] root 11 0.0 0.0 0 0 ? S 2007 0:00 [mtdblockd] root 185 0.0 0.9 2220 1044 ? Ss 2007 0:00 dhclient3 -pf /va r/run/dhclient.eth0.pid -lf /var/run/dhclient.eth0.leases eth0 daemon 202 0.0 0.3 1632 440 ? Ss 2007 0:00 /sbin/portmap root 230 0.1 0.6 1724 664 ? Ss 2007 0:00 /sbin/syslogd root 233 0.0 0.3 1512 440 ? Ss 2007 0:00 /sbin/klogd root 241 0.0 0.4 1572 508 ? Ss 2007 0:00 /usr/sbin/inetd root 268 0.9 0.4 5140 444 ? S 00:00 0:00 /pbx/sipfxo --con fig /etc/sipfxo.conf --agc --busy-det --show-version /etc/sipfxo-release --pbx-a dr 127.0.0.1 --sip-adr 127.0.0.1 --csp-mac 00:19:15:68:40:A0 --msp-mac 00:1A:1B: 1C:1D:1E root 269 0.0 1.1 2684 1236 ? S 00:00 0:00 /bin/bash /etc/rc 2.d/S20pbxnsip start root 270 50.3 21.5 41644 23812 ? S 00:00 0:22 /pbx/pbxctrl-debi an3.1 --dir /pbx --default default.xml --no-daemon root 275 0.0 0.4 5140 444 ? S 00:00 0:00 /pbx/sipfxo --con fig /etc/sipfxo.conf --agc --busy-det --show-version /etc/sipfxo-release --pbx-a dr 127.0.0.1 --sip-adr 127.0.0.1 --csp-mac 00:19:15:68:40:A0 --msp-mac 00:1A:1B: 1C:1D:1E root 276 0.0 0.4 5140 444 ? S 00:00 0:00 /pbx/sipfxo --con fig /etc/sipfxo.conf --agc --busy-det --show-version /etc/sipfxo-release --pbx-a dr 127.0.0.1 --sip-adr 127.0.0.1 --csp-mac 00:19:15:68:40:A0 --msp-mac 00:1A:1B: 1C:1D:1E root 277 0.0 0.4 5140 444 ? S 00:00 0:00 /pbx/sipfxo --con fig /etc/sipfxo.conf --agc --busy-det --show-version /etc/sipfxo-release --pbx-a dr 127.0.0.1 --sip-adr 127.0.0.1 --csp-mac 00:19:15:68:40:A0 --msp-mac 00:1A:1B: 1C:1D:1E root 279 0.0 1.3 4040 1508 ? Ss 00:00 0:00 /usr/sbin/sshd root 285 0.0 0.7 1864 780 ? Ss 00:00 0:00 /sbin/rpc.statd root 291 0.0 0.4 1508 480 ttyS0 Ss+ 00:00 0:00 /sbin/getty -L tt yS0 115200 vt100 root 299 0.0 21.5 41644 23812 ? S 00:00 0:00 /pbx/pbxctrl-debi an3.1 --dir /pbx --default default.xml --no-daemon root 300 0.2 21.5 41644 23812 ? S 00:00 0:00 /pbx/pbxctrl-debi an3.1 --dir /pbx --default default.xml --no-daemon root 301 0.0 21.5 41644 23812 ? S 00:00 0:00 /pbx/pbxctrl-debi an3.1 --dir /pbx --default default.xml --no-daemon root 302 0.1 21.5 41644 23812 ? S 00:00 0:00 /pbx/pbxctrl-debi an3.1 --dir /pbx --default default.xml --no-daemon root 303 0.0 21.5 41644 23812 ? S 00:00 0:00 /pbx/pbxctrl-debi an3.1 --dir /pbx --default default.xml --no-daemon root 304 0.0 21.5 41644 23812 ? S 00:00 0:00 /pbx/pbxctrl-debi an3.1 --dir /pbx --default default.xml --no-daemon root 305 0.0 21.5 41644 23812 ? S 00:00 0:00 /pbx/pbxctrl-debi an3.1 --dir /pbx --default default.xml --no-daemon root 306 0.0 21.5 41644 23812 ? S 00:00 0:00 /pbx/pbxctrl-debi an3.1 --dir /pbx --default default.xml --no-daemon root 307 1.7 1.9 7024 2180 ? Ss 00:00 0:00 sshd: root@pts/0 root 309 0.0 1.9 7024 2180 ? S 00:00 0:00 sshd: root@pts/0 root 311 1.0 1.3 2744 1448 pts/0 Ss 00:00 0:00 -bash root 315 0.0 0.7 2664 864 pts/0 R+ 00:00 0:00 ps aux root 316 0.0 0.5 1872 632 pts/0 S+ 00:00 0:00 more comcerto:~#