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I need to find out if Cox has any published standards regarding their Company Provided TA devices, and how they are supposed to behave, with regard to basic D.C. Telephony services. Does Cox comply with the Bellcore G.R.s regarding things like on-hook timing for flash, leakage across T-R for determining on-hook vs off conditions.
Here are my reasons why I need this information:
1. Off-the shelf Uniden wireless phones are constantly keeping line siezed on hangup, causing High & Wet conditions, or failure to provide Dial tone on next off hook. The only remedy to date is to break the D.C. Path between the TA and the base unit for the phones. This NEVER occurred on a 5E-hosted Dialtone line at 18KF from C.O.
2. Off hook duration for 3W-calling or SW-Flash for call waiting, more often than not causes the TA to lock-up, sending 120 IPM Dial Tone to me, or tears down the call in progress, forcing us to try and start all over. I've lost count of the times I've bid for Dialtone, only to hear Silent Batt, or 120 IPM, then gone thru the open-pair drill before it will clear.
3. Calls to National 800 number companies with IVRs will fail, with no ability to get into the company's directory or menu choices, because the IVR is receiving some sort of voice primitive, or tone or impluse noise that prevents me from instructing it, when I am NOT SENDING ANY TONE OR VOICE through the call session. The Same IVR will work as expected from any Cell phone or C.O. based origination in the same area.
For this trouble, I need tier 2 or 3 support to prove the trouble either towards the IVR sponsor ( NOT likely since this symptom is occurring on Companies all over the U.S., therefore at different distances [echo return variances, and possibly +|- .5 DB level variances]) or towards your network or the default level settings in the TA or the head end with regard to how your network converts Analog voice and background noise before sending it towards the called IVR.
Ideally I would like to work with the tech who has No-Test capability and a "GOD's eye" view of the network and stable path through it so he or she can hear for themselves what I am putting up with. The most recent & frustrating site for me has been the national U.S.P.S. call center at 1-800-275-8777. Before the first choice offers are even completed, the IVR halts and returns "I didn't understand that input" the re-starts the menu choices over again, never to be satisfied. I eventually have to bail out from the call.
Wow, this is the first thread I have read here that is WAY above my head. See this thread. Who you want to talk to is dward5665 and Michael2062,
Over my head. Just had two new modems installed and 12-year-old Panasonic and alarm system working just fine. According to the expert who did the install they’re having issues with phone systems because most contractors doing the work don’t understand the “polarity” requirements on the connections. I’ll translate to ... there are two wires for a CO line and these are sensitive to how you connect them. But everything worked fine after conversion and we only had to boost the RF in the house because we used two modems for 3 lines.
First time every asking this on this forum, but could you dumb it down a bit? I feel I have good input to give, but I don't quite understand what you are asking. Are you looking for some kind of standard agreement from Cox to make the signal over Packet Switch the same as that over Circuit switch? Also, are you aware that Cox is transitioning to Cox Voice, which is still SIP, but SIP that is ran over a now virtual PBX instead of standard switches.
Also, have you checked out the DPQ3212 datasheet? I seem some standards in there I think Cox follows. Is that the type of thing you are looking for, but more specific?
● MGCP/NCS including configurable IPsec encryption● Configurable to support RFC2833 event signaling● Supports Bell103 detection: Improves alarm panel and Point of Sale (POS) interoperabilityby optimizing DSP for Bell 103 protocol● Software upgradeable to support Session Initiation Protocol (SIP)● The following SIP standards are supported◦ RFC 2617 HTTP Authentication: Basic and Digest Access Authentication◦ RFC 2976 The SIP INFO Method◦ RFC 3261 SIP: Session Initiation Protocol◦ RFC 3262 Reliability of Provisional Responses in Session Initiation Protocol (SIP)◦ RFC 3263 Session Initiation Protocol (SIP): Locating SIP Servers◦ RFC 3264 An Offer/Answer Model with Session Description Protocol (SDP)◦ RFC 3265 Session Initiation Protocol (SIP)-Specific Event Notification◦ RFC 3420 Internet Media Type message/sipfrag◦ RFC 3428 Session Initiation Protocol (SIP) Extension for Instant Messaging◦ RFC 3515 The Session Initiation Protocol (SIP) Refer Method◦ RFC 3842 A Message Summary and Message Waiting Indication Event Package for theSession Initiation Protocol (SIP)◦ RFC 3892 The Session Initiation Protocol (SIP) Referred-By Mechanism◦ RFC 3903 Session Initiation Protocol (SIP) Extension for Event State Publication◦ Draft-ietf-mmusic-sdp-new-24 SDP: Session Description Protocol (Replacement forRFC 2327)◦ Draft-ietf-sipping-cc-transfer-01 Session Initiation Protocol Call Control – Transfer◦ Draft-ietf-sip-session-timer-08 The SIP Session Timer◦ Draft-ietf-sipping-realtimefax-01 SIP Support for Real-time Fax: Call Flow Examples andBest Current Practices◦ Draft-ietf-mmusic-sdescription-09 Session Description Protocol Security◦ Descriptions for Media Streams◦ Draft-ietf-sip-replaces-02 The Session Initiation Protocol (SIP) "Replaces" Header
I've moved discussion over to this thread because I think the problem definition is better.
How many devices are physically plugged into your line? Is your Uniden a base station with wireless handsets, or several separate devices? Have you checked your hookflash timers? They should be about 500ms. Is the audio quality good? Have you tried plugging straight into the modem and testing?