Bradley_M Posted May 27, 2008 Report Share Posted May 27, 2008 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!!! Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted May 27, 2008 Report Share Posted May 27, 2008 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. How is your IP config? Are you mixing DHCP with static IP addresses? Maybe it makes sense to log in through SSH and take a look at the IP config with ping. Quote Link to comment Share on other sites More sharing options...
Bradley_M Posted May 27, 2008 Author Report Share Posted May 27, 2008 How is your IP config? Are you mixing DHCP with static IP addresses? Maybe it makes sense to log in through SSH and take a look at the IP config with ping. 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? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted May 27, 2008 Report Share Posted May 27, 2008 What does /proc/net/route say? Quote Link to comment Share on other sites More sharing options...
Bradley_M Posted May 28, 2008 Author Report Share Posted May 28, 2008 What does /proc/net/route say? 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 Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted May 28, 2008 Report Share Posted May 28, 2008 comcerto:~# more /proc/net/routeIface 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 Well, that means that the PBX has only 192.168.1.x as IP address. That is not a public IP address... If you want it in the public you have to do something about the public IP address (http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses is a good start). Do you have a public IP address? Can you use your router in router mode, no NAT? Otherwise things will get complicated or instable. Quote Link to comment Share on other sites More sharing options...
Bill H Posted May 28, 2008 Report Share Posted May 28, 2008 Well, that means that the PBX has only 192.168.1.x as IP address. That is not a public IP address... If you want it in the public you have to do something about the public IP address (http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses is a good start). Do you have a public IP address? Can you use your router in router mode, no NAT? Otherwise things will get complicated or instable. If PBXNSIP has only 192.168.1.x as an IP Address, how would he connecting to it remotely? I think your problem with voice is that someone or both parties are not getting the correct IP Address to respond to. With as much bad rap that STUN recieves I still use it and it works OK under certain conditions. Here is what I would look at: Set up Logging in PBXNSIP to see all the SIP Messages. Make a few test calls. Look in the SDP (Session Description Protcol) area of the message (its long and at the end) and see if there are any Local IP Addresses like 192.168.x.x. There should not be any. INVITE sip:2227878@Proxy.ac SIP/2.0 Via: SIP/2.0/UDP 172.215.xxx.xxx:5060;branch=z9hG4bKac710131078 From: "5022" <sip:5022@Proxy.ac>;tag=342231859 To: <sip:2227878@Proxy.ac> Call-ID: 515987274-5062-2@71.190.187.23 CSeq: 10 INVITE Contact: <sip:5022@71.190.xxx.xx:5062> <<< This should be a Public IP Address Max-Forwards: 69 Supported: replaces, path, timer User-Agent: Grandstream HT-502 V1.1B 1.0.0.86 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 433 v=0 <<< SDP Starts here o=5022 8002 8000 IN IP4 xxx.xxx.xxx.xx <<< This should be a Public IP Address s=SIP Call c=IN IP4 xxx.xxx.xxx.xxx <<< This should be a Public IP Address t=0 0 m=audio 5050 RTP/AVP 0 8 4 18 2 97 103 102 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G I use STUN. It solves the NAT problem for me. I don't know if Snom has a STUN feature or not. Use a STUN Server like stun.xten.com or stun.softjoys.com Also, different routers can create VOIP problems. Turn off SPI (Stateful Packet Inspection) Some routers (Netgear) have a hightend NAT control. Set it to Standard or Simple. (something like that) Give your Snom phone a Private Static IP Address to use with the router. Put that IP Address in the DMZ for full exposure. Someone wise (Kevin Moroz at PBXNSIP) once told me that all the answers are in the SIP Traces (Messages). ....... He was right. Bill H Quote Link to comment Share on other sites More sharing options...
Bradley_M Posted May 29, 2008 Author Report Share Posted May 29, 2008 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. Quote Link to comment Share on other sites More sharing options...
andrewgroup Posted May 29, 2008 Report Share Posted May 29, 2008 All this jumping through hoops just to figure out why calls are dropping and why FX0 ports are hanging occasionally. Grrrrrrr. Jumping through hoops is the result of not having a plan. This technology isn't easy because it's IP and without a good IP foundation and experience you cannot make and execute a good plan. All you are left to do is GunSling hoping to find the magic bullet. Building and testing a plan prior to rolling out this or any other VoIP solution is always the best plan. If you are coming from the TDM telecom world, Congrates on your successes. Quote Link to comment Share on other sites More sharing options...
Bradley_M Posted May 30, 2008 Author Report Share Posted May 30, 2008 Jumping through hoops is the result of not having a plan. This technology isn't easy because it's IP and without a good IP foundation and experience you cannot make and execute a good plan. All you are left to do is GunSling hoping to find the magic bullet. Building and testing a plan prior to rolling out this or any other VoIP solution is always the best plan. If you are coming from the TDM telecom world, Congrates on your successes. 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? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted May 30, 2008 Report Share Posted May 30, 2008 Has anyone else dealt with call drops or channel hanging in an install? (Trying to find a way back to what the problem was:) So that NAT problem is solved, now the problem is that the FXO lines hang? "Hanging" meaning that the call stays connected, although it should be disconnected? Quote Link to comment Share on other sites More sharing options...
andrewgroup Posted May 30, 2008 Report Share Posted May 30, 2008 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? We previously support many Asterisk Solutions with a full time Asterisk Developer. We were able to plug-n-play PBXnSIP into the Asterisk clients without missing a beat and our support time fell and we exclusively used Digium Hardware. Our largest installatiion was and still is 45,000 call minutes a month across a full PRI. We have numerous sites with Audio Code Analog gateways, CS 410's, SPA2002 gateways and soon to be another brand. Your references to channels hanging, wacky rollover issues, duplicates of troubles experienced with Asterisk, all indicate to me with no personal knowledge of your installation that an external event may be related. Successful interconnect companies for years have had to deal with these issues and adding the complexities of VoIP only add to the possibilities. Line Quality is always an issue - and previously dropped calls that were once tolerated will now be blamed on the new system. Fact O Life. Quote Link to comment Share on other sites More sharing options...
andrewgroup Posted May 30, 2008 Report Share Posted May 30, 2008 Can you monitor a line with a linemans Butt set and listen to the line when the hangup occurs? Until you can validate the LEC is providing call disconnect you are likely spinning your wheels. Call disconnect on lines is a switch setting in the LEC switch and was not always a default settting. The 410 has FXO settings to assistn but they are no substitute for no call disconnect. We have also seen the following trouble. The carrier is providing FXS ports from a CISCO IAD 2431 and when the call is disconnected the CS410 see's these tones and immediately goes off hook and must time out in order to accept another call on that channel. The CS410 FXO gateway needs a timer setting for a wait timer before it answers the next incoming call, this would prevent this problem. I can duplicate this trouble if PBXnSIP might be interested in this issue. I'm sure it can happen with any IAD CISCO 2431 device. Quote Link to comment Share on other sites More sharing options...
Bradley_M Posted May 30, 2008 Author Report Share Posted May 30, 2008 (Trying to find a way back to what the problem was:) So that NAT problem is solved, now the problem is that the FXO lines hang? "Hanging" meaning that the call stays connected, although it should be disconnected? 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. ??? Quote Link to comment Share on other sites More sharing options...
andrewgroup Posted May 30, 2008 Report Share Posted May 30, 2008 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. ??? Quote Link to comment Share on other sites More sharing options...
andrewgroup Posted May 30, 2008 Report Share Posted May 30, 2008 This all points to more line related issues. Having a high empendance line recorder is often the first line of defense. http://www.999.co.jp/us/telecoderan8us/index.html http://www.elyssacorp.com/PDF_Cutsheets/Vo...ogCut_05_04.pdf PLus google searching will find many more - The betters ones can detect / decode DTMF tones while dialing and analyze wav forms Despite the fact that PSTN is over 100 years old, it has troubles and installing new technology onto know crap makes you the goat. Having the most and the best diagnostic evidence allows you to stand clear of the debris' and hopefully to be paid for your expertise.... Add clauses to your purchase agreements that you invoice for all time - including time diagnosing pre-existing conditions with Telecom Providers. Cheers. Quote Link to comment Share on other sites More sharing options...
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