Worm78

Thanks for the quick reply. I will check the firmware and see what else I can do to eliminate the errors.

Not sure what you mean PBXNSIP. Any other suggestions for troubleshooting? I added the device to a new sip trunk and the issue remains. Still same company. Andy suggested monitoring the WAN port which I will do tomorrow and then post a new pcap file. All users are using X lite. Windows firewall has an exception for the program. Any known issues with this soft phone? I have searched their forum with no luck. Thanks, Brian

Not sure if I explained my setup correctly. CS410 -- Wan port goes to public switch/router. Lan port goes to local 24 port switch. Netgear internet router -- Wan port goes to switch/router. Lan port goes to local 24 port switch. Both devices have their own public IP. All SIP devices on lan have their reg/gateway pointed to the local IP of the cs410. Thanks

I placed my laptop on the local network and installed xlite. I called a 800 number job hotline recording until the issue started. I started a wireshark snapshot and let it run until the audio of the recording came back. I have attached both. It looks like i'm getting a udp checksum error but i just don't know what I'm looking at well enough to know the issue. I also attached a normal portion of a noneffected call. Also to note they are using the phone more often and it seems to be happening alot more then I first thought. I also seem to have the issue at the same time on multiple phones. PC's are on the local side of he network behind (same side) a netgear router. Lan port is direct to the public side with its own ip. Just curious should I have anything in the gateway section of the local ip settings. I seem to recall taking that out to fix a ntp issue. I think this was on purpose to keep the trunk going out the correct port but want to make sure while I'm asking questions. Thanks pcaps.zip

Issue is still there. Any other ideas? Anything specific to look for in the log and if so at what log level?

Thanks for the tip. It disappeard for sure. Still testing but seems to have resolved the issue. I will repost when resolved.

I have a cs410 hooked direct to a public IP for testing. About 2-3 times a day I get complaints of the person on the other end hearing us but us only hearing silence on our end for 10-15 seconds. If you wait audio does go back to normal. I'm using a trunk through Teliax. Any ideas? Happens on several phones. Please use the information on this web page when you address the support. Version: 3.0.0.2899 (Linux) License Status: cs410 License Duration: Permanent Additional license information: Extensions: 8/10 Accounts: 14/20 Working Directory: /pbx IP Addresses: eth2 x.x.x.x 255.255.255.0 eth1 1.1.1.1 1.1.1.0 255.255.255.0 eth0 192.168.10.200 192.168.10.0 255.255.255.0 lo 127.0.0.1 127.0.0.0 255.0.0.0 default 96.11.x.x MAC Addresses: xxxxxxxxxx Calls: 168/35 (CDR: 203) 0/0 Calls SIP packet statistics: Tx: 139657 Rx: 128098 Emails: Successful sent: 39 Unsuccessful attempts: 0 Uptime: 12 19:24:44 (4467 4957440-0) WAV cache: 0 Media CPU Usage: 100% 0% 0 24 Thanks

I have a cs410 newly installed and I'm seeing a weird issue with the trunk registration. First it has the wan and lan port behind a firewall with the ports opened up fully to the wan port which may be the issue. 1 ip office scenario. System is working just fine however if you reboot the box you do not get the ability to call in until you try to call out from the pbx. I have waited up to 5 minutes and nothing. It shows the trunk is live. I have asked teliax and they said they are not seeing anyhting coming from the device prior to my call out. Is there a setting anywhere for the reconnect delay or force reconnect? I also have one other off the subject issue where I can only add two auto attendants. It says no more license however I can add a hunt group or acd no problem. Is there a 2 max on auto attendants on the base 410? Thanks Version: 3.0.0.2899 (Linux) License Status: Appliance Key License Duration: Permanent Additional license information: Extensions: 8/32 Accounts: 15/40 Working Directory: /pbx IP Addresses: eth1 1.1.1.1 1.1.1.0 255.255.255.0 eth0 192.168.10.200 192.168.10.0 255.255.255.0 eth2 192.168.10.202 192.168.10.0 255.255.255.0 lo 127.0.0.1 127.0.0.0 255.0.0.0 default 192.168.10.202

No, this is on 1.5.2.10 server. Issue above was on a cs410

If using call waiting option will this allow for several calls to be put on hold? I was debating just call forwarding busy to the auto attendant. That way if she in on the phonethey don't let it ring for 10 seconds for no reason.

How would I setup my receptionist phone if i don't have co lines? If I remove them I get a busy signal once one call comes in. She uses a snom 360. Scenario is trunks rolls to hunt group 700, rings 101 being the receptionist for 10 seconds then rolls to 701 which is the auto attendant, or 702 at night. Without co lines, I called in and answered the call then took another cell and called in and I get a busy.

The original scenario seemed to work ok for the first 4 hours and then the network craps out. I was not on site but you lose internet and ability to ping the router. It seemsto come and go every 30 seconds or so until you disconnect the PBX wan and lan from the router. No router reset needed at this point and everything is working ok. I also made sure I didn't have a GW in the local side as this was causing issues for the device to hit the time server. I'm working on trying to get a private ip for the wan side but need a temp solution if possible. Any ideas? Thanks

I was wondering what router people have had good luck with in small offices using the cs410. I have a small office using a cs410 and a Linksys BEFSR41. It works great using a DMZ or opening up ports for the wan port The lan is just sitting on the normal side of the network. If I replace this with a new netgear 814 or I think 614 I lose registration and have intermittent audio incoming. I have removed packet imspection, setup on dmz, and all other items I can think of but no luck. They currently only have one static ip from their provider. Thanks

I checked all the xml files and removed the dups and redirections in the other folder with another forum assisted fix. We removed all co lines then went in and removed the refs for the ramining ones listed in accounts. I then addedthe co lines back in. So I can just delete all refrences to co lines in both trunks?

I found the issue. It appears all the co lines are staying active but only 1 person is on a call. We have 12 total lines. I had an issue with several co lines that did not disappear when deleting an old trunk so this may be a remaining issue.

The log from teliax or is this something from my end?

Thanks

I'm running 1.5.2.10 I keep running into an issue where I can't dial out on my sip or pots line trunks. Happens every 3-4 days. It happened twice right after I registered a user on a phone but I thought this was a coincidence. If I reset the pbxnsip service all is good again. I have 9 set for my sip and 8 set for my gateway. If you try to dial it gives you a recording saying you are dialing an invalid extension. Trunks both show good and registered at this point. It normally happens mid day and I need to get us back up so I don't have much time for troubleshooting. Yesterday I upped the log levels and tried to make a call during the down time, then cleared the log and made another call after the service was reset. I don't see any errors jumping out however I'm failry new to voip. anyone see anything? .0.0.0

Is there a feature to allow you to call into the PBX say using your local home number, enter a pin code and then dial long distance to use the sip to reduce long distance fees? Thanks

Figured it out. Provider was not sending the 10 digit number in the caller id To: <sip:Consciencium@192.168.1.90:5060;transport=udp;line=c81e728d>;tag=520db5f9ff Call-ID: 26278635405878f077deb4145429fc53@64.74.188.23 Thanks for your help. I still couldn't get the filter to work so I went with using an alias. Working well

hhh

Figured it out. Provider was not sending the 10 digit number in the caller id To: <sip:Consciencium@192.168.1.90:5060;transport=udp;line=c81e728d>;tag=520db5f9ff Call-ID: 26278635405878f077deb4145429fc53@64.74.188.23 Thanks for your help. I still couldn't get the filter to work so I went with using an alias. Working well

With this they both go to 396. 5] 2007/12/31 19:05:19: Trunk Teliax sends call to 396 [5] 2007/12/31 19:06:05: Identify trunk (line match) 2 [5] 2007/12/31 19:06:05: Trunk Teliax sends call to 396 [5] 2007/12/31 19:06:22: Identify trunk (line match) 2 [5] 2007/12/31 19:06:22: Trunk Teliax sends call to 396 Any other thoughts? Ca I use alias to do the same task?

I have this line in where I did have just 396. With only 396 in the field both phone numbers rolled to the auto attend. Im trying to add a direct line for the fax. !1(3176440272)!\1!t!396 !1(3176440274)!\1!t!310 and this is what I get in the log. [5] 2007/12/31 20:41:28: Trunk Teliax sends call to Consciencium [5] 2007/12/31 20:41:28: Trunk call: Could not identify user [5] 2007/12/31 20:59:37: Identify trunk (line match) 2 [5] 2007/12/31 20:59:37: Trunk Teliax sends call to Consciencium [5] 2007/12/31 20:59:37: Trunk call: Could not identify user [5] 2007/12/31 21:00:05: Dialplan Standard User: Match 99858299@192.168.1.79 to <sip:3179858299@nyc.teliax.net;user=phone> on trunk Teliax [5] 2007/12/31 21:00:12: INVITE Response: Terminate 6ca9e8eb@pbx [5] 2007/12/31 21:00:20: Identify trunk (line match) 2 [5] 2007/12/31 21:00:20: Trunk Teliax sends call to Consciencium [5] 2007/12/31 21:00:20: Trunk call: Could not identify user [3] 2007/12/31 21:03:30: Could not connect to 68.142.234.44:443 [1] 2007/12/31 21:03:30: Could not send via TCP:?;Gy�r�a���:U����%3�h���L ǧ�5l db [5] 2007/12/31 21:04:04: INVITE Response: Terminate 95fe1964@pbx [5] 2007/12/31 21:04:21: BYE Response: Terminate NWRiYTIwOTg3ZmY1NTlkNTJlM2E4MmEyZTBlY2YxZjU. 396 is the auto attend and 310 is my fax Any ideas? This is a cs410 device if that matters. the info is entered under "Send call to extension:" under the trunk screen.

I think I understand the patterns but lost on where this gets placed. Would it go on the trunks page in the extension field? If so does it sort in order and the use a back up at the end? 300 would be the auto attendant. Example exact typing in field being Extension: !1([8128675308]*)!\1!t!301 !1([8128675309]*)!\1!t!302 300