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bskeddle

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Everything posted by bskeddle

  1. I find the same problem. It ONLY works at registration/initialization of an extension.. then does not update. (I don’t know if it related.. but when a chat session is initiated a chat window opens on the initiators computer; however when the other party replies, a 2nd chat window opens over the original chat window with the reply and then supersedes the original window.) Seems to me this could probably be fixed by having a PBXnSIP engineer call the Counterpath people.
  2. Any update on this??!? We really need to get it working with x-lite! I am willing to do any testing, logging, etc. that would help PBXnSIP either fix this problem or to document what needs to be fixed on the CounterPath side. Thanks!
  3. I have read: http://wiki.pbxnsip.com/index.php/Counterpath It states; “If you want to use the presence feature of X-lite, you need to change the presence mode to "Presence Agent". The other settings, poll time and refresh interval are fine; you don't have to change them.” I find that the Presence / Availability indicators are not updated regularly (or at all) on some of my extensions. What does “Poll time” do? What does “Refresh interval” do? (Also, the http://wiki.pbxnsip.com/index.php/Counterpath page needs to be updated as the picture shows a checkmark for ICE but later the page states: “You can also turn ICE off because the PBX does not support ICE.” Furthermore the page also states “Presence is also not supported yet by the PBX”)
  4. This solved the problem. Thank You!
  5. Testing it now. Looks like it will work! Will Report results tommorow.
  6. I DO NOT know!!! but Time Warner did a call trap for me and sent me this info: After carefully reviewing the call trap I have come to this conclusion. This call was terminated by the NON-TWC customer. Date/Time: 3/24/09 10:32 AM Originating TN: 760xxx8621 Terminating TN: 8588057174 Start with the RTP, which is the voice audio. The BYE response in BLUE is the off-net caller which was me hanging up. 481.600 seconds into the call. The 200 OK response in BLUE back from the soft switch to the Signaling Interface at 481.603 seconds into the call. The DLCX response in PURPULE from the soft switch to the MTA came at 481.604 seconds into the call. The 250 response in PURPULE from the MTA to the soft switch came at 481.692 seconds into the call. This was followed immediately by a NTFY on-hook from the MTA to the soft switch at 489.678 seconds into the call. This gave us a delay of 8 to 9 seconds from the DLCX to the on-hook response back to the soft switch. I have been advised by our Voice Analyst that this is normal and there should be delay of no more than 10 seconds. They stated that the was setup for 911 situations where if someone was calling the call would not immediately hang up by pressing the hook. I hope this information clarifies the situation. Please let me know if there is anything else that I can do to assist. Agilent NgN Monitor and Analysis Call Trace: Call Sequence:
  7. I have tried 400 down to 1 (and upto 1500). No joy. According to my research "Detect Polarity Change" is a holdover from the days that batteries were used by the phone co. (they would reverse polarity to signal disconnect) and is no longer used in the USA. What does "FX Impedance" do? I think this point in the log is where the call (620e2245@pbx) is terminated by the user and the system creates the "ghost" call (6931fac9@fxo) back: [7] 2009/03/23 19:23:22: Call 620e2245@pbx#1847375296: Clear last request [5] 2009/03/23 19:23:22: BYE Response: Terminate 620e2245@pbx [3] 2009/03/23 19:23:22: PSTN: Channel 0: Hangup [5] 2009/03/23 19:23:22: PSTN: Channel 0 goes onhook [5] 2009/03/23 19:23:22: PSTN: enable_callerid 0 [3] 2009/03/23 19:23:22: PSTN: Channel 0 going to GO_ONHOOK [3] 2009/03/23 19:23:23: PSTN: Channel 0 going to IDLE [5] 2009/03/23 19:23:24: PSTN: eVAPI_CALLER_ID_DETECTED_EVENT [7] 2009/03/23 19:23:24: SIP Rx udp:127.0.0.1:5062: INVITE sip:760xxx8620@localhost;user=phone SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5062 From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1804289383 To: <sip:760xxx8620@localhost;user=phone> Call-ID: 6931fac9@fxo Contact: <sip:127.0.0.1:5062;line=1> CSeq: 1 INVITE Content-Type: application/sdp Content-Length: 137 What does the line: [5] 2009/03/23 19:23:24: PSTN: eVAPI_CALLER_ID_DETECTED_EVENT indicate is happening?
  8. According to my research, CPC Duration (or Loop Interruption) is the length of time that the phone co. interrupts the line voltage in order to signal the end of a call. If the CPC Duration is set longer than the actual length of the CPC "issued" by the phone co. the PBXnSIP will not interpret the voltage interruption as a CPC and therefore never disconnect. i.e. If I set it to 1000ms, the system will only disconnect after the line has 0 voltage for Greater than 1000ms. If the actual CPC Duration is only 600ms the PBXnSIP will not recognize it as a CPC and therefore not disconnect.. However if the CPC Duration is set to 400ms and the actual CPC is 600ms then the PBXnSIP will interpret any voltage interruptions Greater than 400ms (including real 600ms CPC's) as a CPC and will therefore disconnect. It seems that the purpose of setting a longer CPC Duration is to prevent unintended disconnects caused by random voltage interruptions. As the problem appears to be that the PBXnSIP is NOT disconnecting, how would setting the CPC higher help? Your help is very appreciated, as this phone stuff is new territory to me!
  9. Thanks for getting back to me! "That of course raises the question: Has anything changed in the setup?" Nothing has changed on our side, and TimeWarner claims nothing has changed on their side. "It sounds a little bit like the PBX has problems with the FXO. Maybe it was on the edge already and now it is over the edge. Is the signal quality okay on the FXO (I don't know the Time Warner - is it connected through a short cable?). " TimeWarner is a cable co. they offer phone service though a Motorola cable modem (voip->pots). The pots cable from the modem is about 1 foot long. "Maybe you can attache a screenshot of the settings for the FXO gateway on the CS410. It could be there is a parameter wrong." OK.. here it is...
  10. The CS410 is connected to a Time Warner SBV5322 that provides 3 “pots” lines connected to FXO1 – FXO3. (the 4th line runs directly to a fax.) CS410 Version 3.3.0.3160 X-Lite soft phones 3.0 build 47546. The system worked perfectly for 6 months. 1) For the last week we have been "receiving" calls from Anonymous (anonymous@localhost) [PSTN] after any call has been ended (from either side). a. None of the FXOx port indicators are lit, but the system does list the “call” on the “Currently Active Calls” page. b. The line is silent when answered and when unanswered the Ghost calls connect to VM boxes and fill them with blank messages. c. An email with a RE line similar to:"CS410: RTP Timeout on 81304908@fxo#d1caa153cb" is sent stating that "The call between sip:7608328620@localhost;user=phone and sip:anonymous@localhost;user=phone has been disconnected because of media timeout (120 seconds), 0/5994 packets have been received/sent" 2) For the last month FXOx ports will randomly “lock up” making that line unavailable. a. The FXOx port indicator will light, but the system does not indicate a call in the “Currently Active Calls” page. b. If two internal users are on lines, when the third user dials out the system will drop one of the existing calls. c. Sometimes unplugging the line from the offending FXO port will clear the problem, other times the CS410 will have to be restarted. We purchased our CS410 directly from PBXnSIP (and therefore do not have a reseller to turn to for support).
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