ugrade to new 5.1

Discussion in '3CX Phone System - General' started by kirimis, Feb 15, 2008.

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  1. kirimis

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    evryting ok with new version beta exept forwarding i tried to give an internal call to another internal
    and it could not tranfer and the internal i want to pass the line is stuck. the comes out like this

    "11:23:51.511 MediaServerReporting::RTPReceiver :[MS105000] C:128.2: No RTP packets were received:remoteAddr=0.0.0.0:59000,extAddr=0.0.0.0:0,localAddr=100.100.100.253:7310
    11:23:47.402 Call::Terminate :[CM503008]: Call(126): Call is terminated
    11:23:47.402 Call::Terminate :[CM503008]: Call(126): Call is terminated
    11:23:40.793 MediaServerReporting::RTPReceiver :[MS105000] C:127.1: No RTP packets were received:remoteAddr=0.0.0.0:59000,extAddr=0.0.0.0:0,localAddr=100.100.100.253:7304
    11:23:40.793 MediaServerReporting::RTPReceiver :[MS105000] C:127.2: No RTP packets were received:remoteAddr=100.100.100.162:59000,extAddr=0.0.0.0:0,localAddr=100.100.100.253:7306
    11:23:40.777 Call::Terminate :[CM503008]: Call(127): Call is terminated
    11:23:19.325 Call::Terminate :[CM503008]: Call(128): Call is terminated
    11:23:19.325 Call::Terminate :[CM503008]: Call(128): Call is terminated
    11:23:11.154 MediaServerReporting::RTPSender :[MS004001] C:128.1: Can't send RTP stream to 1 destination unreachable
    11:23:02.404 CallCtrl::eek:nLegConnected :[CM503007]: Call(128): Device joined: sip:145@100.100.100.146:5060
    11:23:02.341 CallCtrl::eek:nLegConnected :[CM503007]: Call(128): Device joined: sip:209@100.100.100.162:5060
    11:22:57.092 CallCtrl::eek:nSelectRouteReq :[CM503004]: Call(128): Calling: Ext:145@[Dev:sip:145@100.100.100.146:5060]
    11:22:56.998 CallCtrl::eek:nIncomingCall :[CM503001]: Call(128): Incoming call from Ext.209 to [sip:145@100.100.100.253]
    11:22:49.201 MediaServerReporting::RTPSender :[MS004001] C:127.2: Can't send RTP stream to 1 destination unreachable
    11:22:28.155 CallCtrl::eek:nLegConnected :[CM503007]: Call(127): Device joined: sip:209@100.100.100.162:5060
    11:22:28.093 CallCtrl::eek:nLegConnected :[CM503007]: Call(127): Device joined: sip:222@100.100.100.154:5060
    11:22:24.140 CallCtrl::eek:nSelectRouteReq :[CM503004]: Call(127): Calling: Ext:209@[Dev:sip:209@100.100.100.162:5060]
    11:22:24.046 CallCtrl::eek:nIncomingCall :[CM503001]: Call(127): Incoming call from Ext.222 to [sip:209@100.100.100.253]
    "

    what does this message means can somebody help?
    11:22:49.201 MediaServerReporting::RTPSender :[MS004001] C:127.2: Can't send RTP stream to 1 destination unreachable
     
  2. Nick Galea

    Nick Galea Site Admin

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    You have a networking issue. Your network looks very strange, ips 100.100.100.162 or whatever are not correct IPs to use for an internal network.
     
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  3. kirimis

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    why it is strange? i am using range from 100.100.100.1 - 100.100.100.254 to all devices. everything else has no problem with those ip from erp to network devices.
     
  4. SY

    SY Well-Known Member
    3CX Support

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    First of all
    Thanks for your attention.
    message "Can't send RTP stream to 1 destination unreachable" is incorrect
    "1" - should be IP of destination. It is mistake in the code and it is fixed right after your post appeared on the forum

    Other problems:
    Please specify phone models or version of softphones you are using. Some phones require specific setting for "PBX deliver audio"/"Supports re-Invite"/"Supports replaces"

    Regards
     
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  5. kirimis

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    i am using sip phones of onevoip a company propably not known to you. i would be pleased if i could send you a device to test it with 3cx since i bought around 40 of them because of their low cost. you can see it at http://www.onevoip.net/index.php?dstCatid=251. if you want me to send it to you for testing with no cost from you side please tell me the adress.

    Thank you once more , for you response.
     
  6. SY

    SY Well-Known Member
    3CX Support

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    It is unsupported device but you can try to force it to work. Please note, you will not have all features of 3CX PBX available on this phone. Some of features will not work.

    Please try to perform following tests:
    Configure 3 devices as "Supports replaces"=off. Replicate scenario using them.
    If it still doesn't work. Configure them as "Supports re-INVITES"=off (it will set "Supports replaces"=off as well).
    and try to replicate scenario again.

    Please try blind, attendant transfers, transfers form DR and then publish your results here.

    Thanks

    P.S. Network configuration you are using doesn't correspond any RFC. http://www.rfc-editor.org/rfc/rfc3330.txt
     
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  7. blackburwood

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    hi,

    i have the same problem, all my extention are green and my voip gateway is ok
    i use the grandstream gxp 2000 with 3cx v5
    evrerything is ok inside my lan but when i want to call outside, i can t have any sound and it s the the same trouble when i want to call from the outside.

    on the 3cx server statut i find this

    : No RTP packets were received:remoteAddr=212.27.x.x:36526,extAddr=x.x.126.243:9012,localAddr=x.x.126.243:9012

    please help me and pardon my french
     
  8. Chris-FR

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    Hi all,

    Same problem, no audio for incomming or outgoing from my french freephonie provider.
    Working perfect between extansions.

    Same report on the logs : no RTP Packet......

    All was ok with dinosorus 3.1 version of 3CX ( only the call group not working ).

    PS: My network is full RFC1918 compliant using Class C Mask
    Machine : W2003 SBS Server
     
  9. blackburwood

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    hi all

    it s seems that i have the same trouble with my voip provider freephonie (free.fr), but i don t understand because i have the same trouble with my fxo (spa3102) no sound!!
     
  10. Chris-FR

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    That's the problem with 3CX !

    When it woks, it's fantastic...but from a version to another you can have something going wrong.
    I remember to have the same problem with oldies version ( loosing voice on my provider ). 3CX team made a patch and all OK.

    So Now i Wanted to know if someone have sound with his provider and this verion of 3CX.

    Can someone answer my question please ?
     
  11. Nick Galea

    Nick Galea Site Admin

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    An error like this ": No RTP packets were received:remoteAddr=212.27.x.x:36526,extAddr=x.x.126.243:9012,localAddr=x.x.126.243:9012" points to a firewall problem. This is not a 3CX problem. If we are not receiving RTP data there will never be audio.

    It means that either your firewall is blocking the RTP data, or the VOIP provider is not sending it. First step:

    1. See if you can complete firewall check. No warnings or errors must be present
    2. If check passes then run wireshark on the PBX machine and see if RTP data is arriving. If not, then you will have to resolve that first.

    Remember we do not support SYMMETRIC NAT and you MUST Have static port mappings. This is frequently the cause of problems.
     
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  12. Chris-FR

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    Hi,

    No firewall problem. All rules ok, and after all it work perfect with the 3.1.2434.0 version of 3CX
    Running on W2003 SBS ( DNS / EXCHANGE / IIS )
    2 HardPhone Cisco 7941G, 1 Nokia N80IE, 2 Software IP Phone.

    The ring group of 3.1.2434.0 version of 3CX does not work....Big problem but we can call and beeing called.

    The new 5.1 is perfect !!!! All is working fantastic, but no audio from my provider.

    For the story, i am an old user of 3CX (before version 2) and i congratulate all the 3CX Team for their fantastic work.
    They are allways small bug in all version, but it's the secound time we have a perfect version, and the secound time i have no sound from my provider " freephonie " :cry: .
     
  13. Nick Galea

    Nick Galea Site Admin

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    Hi,

    Thanks for your feedback :) Actually 3.1 worked differently in that it had a keep alive manager which allowed for connections to be kept alive and this would reduce firewall problems. This is what soft phones do. This works fine for 1 or maybe 2 lines but beyond that other problems start to occur that is why we stopped this. 3CX Phone System has to work as a PBX, and has much less flexiblity then for example a soft phone in establishing and receiving connections.

    However, what i suggest is, can you clean the logs, make a call and then send me the wireshark capture so we can check it out? I would like to double check.

    Also can you confirm that the firewall check test runs OK? Can you send me output?

    I look forward to hearing from you.
     
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  14. Chris-FR

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    I am back with the log...


    18:35:25.234 MediaServerReporting::RTPReceiver :[MS105000] C:8.2: No RTP packets were received:remoteAddr=212.27.**.**:30656,extAddr=82.229.**.**:9010,localAddr=82.229.**.**:9010
    18:35:25.234 Call::Terminate :[CM503008]: Call(8): Call is terminated
    18:35:25.234 Call::Terminate :[CM503008]: Call(8): Call is terminated
    18:35:02.203 CallCtrl::eek:nLegConnected :[CM503007]: Call(8): Device joined: sip:095*********@freephonie.net:5060
    18:35:02.187 CallCtrl::eek:nLegConnected :[CM503007]: Call(8): Device joined: sip:300@192.168.0.60:5060;transport=udp
    18:34:57.500 StunClient::eek:nInitTests :[CM506001]: STUN request to resolve SIP external IP:port mapping is sent to STUN server 64.69.76.23:3478 over Transport 192.168.0.150:5060
    18:34:44.578 Line::printEndpointInfo :[CM505003]: Provider:[FreePhonie] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] Transport: [sip:192.168.0.150:5060]
    18:34:41.968 CallCtrl::eek:nSelectRouteReq :[CM503004]: Call(8): Calling: VoIPline:10000@[Dev:sip:095*******@freephonie.net:5060, Dev:sip:095*******@freephonie.net:5060]
    18:34:41.921 CallCtrl::eek:nIncomingCall :[CM503001]: Call(8): Incoming call from Ext.300 to [sip:0148******@192.168.0.150]

    Log Firewall

    1 9000 Information (9) Le port est ouvert et peut être utilisé pour les appels. externalAddress = 82.229.**.**:9000
    2 9001 Information (9) Le port est ouvert et peut être utilisé pour les appels. externalAddress = 82.229.**.**:9001
    3 9002 Information (9) Le port est ouvert et peut être utilisé pour les appels. externalAddress = 82.229.**.**:9002
    4 9003 Information (9) Le port est ouvert et peut être utilisé pour les appels. externalAddress = 82.229.**.**:9003
    5 9004 Information (9) Le port est ouvert et peut être utilisé pour les appels. externalAddress = 82.229.**.**:9004
    6 9005 Information (9) Le port est ouvert et peut être utilisé pour les appels. externalAddress = 82.229.**.**:9005
    7 9006 Information (9) Le port est ouvert et peut être utilisé pour les appels. externalAddress = 82.229.**.**:9006
    8 9007 Information (9) Le port est ouvert et peut être utilisé pour les appels. externalAddress = 82.229.**.**:9007
    9 9008 Information (9) Le port est ouvert et peut être utilisé pour les appels. externalAddress = 82.229.**.**:9008
    10 9009 Information (9) Le port est ouvert et peut être utilisé pour les appels. externalAddress = 82.229.**.**:9009
    11 9010 Information (9) Le port est ouvert et peut être utilisé pour les appels. externalAddress = 82.229.**.**:9010
    12 9011 Information (9) Le port est ouvert et peut être utilisé pour les appels. externalAddress = 82.229.**.**:9011
    13 9012 Information (9) Le port est ouvert et peut être utilisé pour les appels. externalAddress = 82.229.**.**:9012
    14 9013 Information (9) Le port est ouvert et peut être utilisé pour les appels. externalAddress = 82.229.**.**:9013
    15 9014 Information (9) Le port est ouvert et peut être utilisé pour les appels. externalAddress = 82.229.**.**:9014
    16 9015 Information (9) Le port est ouvert et peut être utilisé pour les appels. externalAddress = 82.229.**.**:9015

    Trying everything, this version doesn't work with external (WAN) IP Provider.
    So come back with 3.1 Version :cry:
     
  15. SY

    SY Well-Known Member
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    Chris,

    For some reason, there is the missed information in PBX log. I can see "No RTP..." but there are no any other reports from media server.
    Please repeat your scenario with "Verbose" level of log. (please don't forget to restart PBX after you change it)
    By the way, using WireShark you can capture traffic on PBX host and see are there any incoming RTP packets as well as analyze negotiations. It is objective source of information regarding "what was sent and what was received"

    Thanks
     
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  16. Chris-FR

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    Hi,

    Here the MAX log :

    19:05:37.046 MediaServerReporting::RTPReceiver [MS105000] C:3.2: No RTP packets were received:remoteAddr=212.**.**.**:35740,extAddr=82.22*.**.**:9014,localAddr=82.22*.**.**:9014
    19:05:36.921 Call::Terminate [CM503008]: Call(3): Call is terminated
    19:05:36.921 Call::Terminate [CM503008]: Call(3): Call is terminated
    19:05:26.828 CallLeg::eek:nConfirmed Session 18 of leg C:3.1 is confirmed
    19:05:26.656 CallCtrl::eek:nLegConnected [CM503007]: Call(3): Device joined: sip:095*********@freephonie.net:5060
    19:05:26.640 CallCtrl::eek:nLegConnected [CM503007]: Call(3): Device joined: sip:300@192.168.0.60:5060;transport=udp
    19:05:26.640 MediaServerReporting::SetRemoteParty [MS210003] C:3.1:Answer provided. Connection(transcoding mode):192.168.0.150:7032(7033)
    19:05:26.640 MediaServerReporting::SetRemoteParty [MS210001] C:3.2:Answer received. RTP connection: 212.**.**.**:35740(35741)
    19:05:26.640 CallLeg::setRemoteSdp Remote SDP is set for legC:3.2
    19:05:10.953 Line::printEndpointInfo [CM505003]: Provider:[FreePhonie] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] Transport: [sip:192.168.0.150:5060]
    19:05:10.953 CallCtrl::eek:nAnsweredCall [CM503002]: Call(3): Alerting sip:095*********@freephonie.net:5060
    19:05:08.484 MediaServerReporting::SetRemoteParty [MS210002] C:3.2:Offer provided. Connection(transcoding mode): 82.22*.**.**:9014(9015)
    19:05:08.031 CallCtrl::eek:nSelectRouteReq [CM503004]: Call(3): Calling: VoIPline:10000@[Dev:sip:095*********@freephonie.net:5060]
    19:05:08.015 CallCtrl::eek:nSelectRouteReq [CM503010]: Making route(s) to [sip:014*******@192.168.0.150]
    19:05:08.015 MediaServerReporting::SetRemoteParty [MS210000] C:3.1:Offer received. RTP connection: 192.168.0.60:30214(30215)
    19:05:08.015 CallLeg::setRemoteSdp Remote SDP is set for legC:3.1
    19:05:08.015 Extension::printEndpointInfo [CM505001]: Ext.300: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Cisco-CP7941G/8.0] Transport: [sip:192.168.0.150:5060]
    19:05:08.000 CallCtrl::eek:nIncomingCall [CM503001]: Call(3): Incoming call from Ext.300 to [sip:014*******3@192.168.0.150]

    I dont have firewall on local machine, so all port are open on LAN.
    All port also open from LAN to WAN.
    So I doesn't understand why i doesn't been at least eard by someone i call ( via VoIP Provider ).

    I have not made any test with ethereal to see what happend to my local IP.
     
  17. SY

    SY Well-Known Member
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    I asked you to check it using ethereal(wireshark) just to be sure that traffic is coming to your host (sent from it). You don't want to do it, but you can ask ISP and VoIP provider support about reasons of "no traffic". They should provide information for user who cannot receive packets on specified destination (82.22*.**.**:9014(9015)). from the source 212.**.**.**:35740(35741)

    Did you try it?

    Thanks
     
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  18. blackburwood

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    hi
    i have try wireshark but i don t know this soft so if you could help me to understand what i need to looking for on my log from my server 3cx activities( ethernet)

    i can now have sound on my gateway pstn spa3201 but freephonie provider voip stll have troubles of sounds

    thank
     
  19. SY

    SY Well-Known Member
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    Your VoIP provider provides service and support, isn't it?
    Sorry for inconveniences, but 3CX PBX just use service of freephonie, it doesn't "wrap" it.
    Could you please ask freephonie for support?

    Thanks
     
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  20. Chris-FR

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    Hi,

    I don't have made the test from what appening on ethernet local, but my hardware Cisco PIX firewall is quite clear : there is no request on RTP port.

    When I uses 3CX 3.1 All are OK
     
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