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HighCtenor

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Posts posted by HighCtenor

  1. Certainly glad to supply it to you. I will supply what I have been told and what I know. If you can help me to fill in the blanks that should let me get it to work today. How shall I send the info to you privately? Please advise

  2. Here is the info from their help. It refers to the Router but not to the entries we need to make. When I asked about the registration I was told to not use a password but to use

    tel.t-onlne.de for the domain and proxy and to use anonymous@t-online.de as the username. Below you will find the router entries:

     

    2/25/2015 Welche Einstellungen sind für die IP­Telefonie mit anderen Clients nötig?
    Welche Einstellungen sind für die IP­Telefonie mit anderen Clients nötig?
    Die folgenden Einstellungen benötigen Sie, wenn Sie einen SIP­fähigen Software­Client oder Ihr Handy oder
    den Router eines anderen Anbieters für die Internettelefonie nutzen möchten. Mit den Routern der Telekom, die für
    den IP­basierten Anschluss optimiert sind, benötigen Sie diese Einstellungen nicht.
    Wir bieten Ihnen dazu zwar keine detaillierten Konfigurationsanleitungen, da es eine sehr große Anzahl an Angeboten
    für die Software­Clients und Hardware­Clients gibt. Dafür erhalten Sie in der folgenden Übersicht aber alle erforderlichen
    Angaben, um jeden üblichen Client einzurichten.
    Allgemeine Einstellungen:
    SIP­ID/Benutzer: Ihre Telefonnummer
    Bildschirmname (falls vorhanden): Ihre Telefonnummer
    Authentifizierungsname / Benutzername: Ihre E­Mail­Adresse, z. B. ihr­name@t­online.de
    Passwort: Ihr Passwort
    SIP­Proxy: tel.t­online.de
    Registrar: tel.t­online.de
    Realm: tel.t­online.de
    STUN­Server: stun.t­online.de
    Outbound­Proxy: leer lassen oder ebenfalls tel.t­online.de
    Bei Einsatz einer Firewall oder eines Routers sind für die IP­Telefonie folgende Portfreischaltungen erforderlich:
    UDP (out): Ports 5060, 30000­31000, 40000­41000, 3478, 3479
    UDP (in): Ports 5070, 5080, 30000­31000, 40000­41000
    TCP (out): Port 80, 443
    Hinweise:
    Geben Sie die SIP­ID ohne Leerzeichen und Sonderzeichen ein (entspricht Vorwahl und Rufnummer)
    Die Eingabe der SIP­ID und des Bildschirmnamens müssen übereinstimmen.
    Den Benutzernamen bitte vollständig klein schreiben.
    Haben Sie Ihr Passwort vergessen? In diesem Fall halten wir für Sie die Anleitung für
    denWiederherstellungsprozess bereit.
    Falls Sie E­Mail­Adresse und Passwort noch nicht vergeben haben, können Sie diese im Kundencentereinrichten.
  3. Hi

     

    The Deutsche Telekom has offered us an upgrade to a VDSL IP-Based Anschluss. That means I have to change to a Sip Gateway or Registration. I have previously used ISDN but that has been turned off.

     

    Does anyone have a template of how the Truck is set up for the Deutsche Telekom in Germany? It would help to have an example as a template for making the proper adjustments to the details.

     

     

  4. Serveral issues are a problem. The provisioning of the Snom 370 has locked me out of the admin mode. I need a fix for that or that phone out.

     

    The analog phone which has worked for 5 years on pbxnsip going through a Linksys PAP2 now is not registering but fails to pickup calls and when it has clicked like the devil. I have disconnected everything and doce a sequential restart. Will restart the analog device after letting it go to a cold start 15 minutes. I need to get that fixed

     

    Without the ability to fix those 2 issues I have one phone and one fax that work and my girlfirend is getting touchy. Flowers do not work if she can not call her girl friends, doctors and work. So I need to get that fixed. I am trying to put the pbxnsip server back into operation so I have a working system.

     

    The problems with the analog phone happen with either Vodia or pbxnsip so I think that must be hardware related? Could it be voltage, codex alignments or what...clicking is something I have never had.

     

    If I could use my NFR rather than the Free config perhaps some of these birthing problems would not be happening.

     

    I am able on either system to dial out and in on the Siemens phone and the Snom when it registers. The fax on the ATA also seems to function within the network and is reachable.

     

    The last couple of days the system would start out allowing me to phone with the siemens and Snom only to fail later on. Very strange behavior. Nothing like it before on pbxnsip.

  5. I am going to attach an additonal config for another smartnode for you. This config does not work for me because my SN has only 1 00/01 port. The attachment is for another in their line with support for the 00/01, but also 4 additional ISDN ports. I would imagine one could use this for companies that need 8 outbound lines at any one time with a fallback. Does not fit me, but it may help you config for larger SMU Patton units and it is ISDN. Hope the attaching works.

    Sample configuration SmartNode connected as trunk gateway between a SIP network and PBX with 4 S0 port.txt

  6. I went to their website, which is not as good as speaking with them directly. I will send a copy of my working config to them. That configuration is what the Vodia has modified. My gateway has just one BRI it would be less detailed than later gateways but would work in a similar way. Attached is a config for your review. I believe it was prepared for a system which is not connecting outbound to a sip provider but only to an ISDN trunk. If I am correct, the attachment may help you see how they conceived it.

    My gateway is older than the one this example is for, I will followup to make sure what they suggest is in compliance with the gateway I am using now.

     

    What is attached is for a 46xx Gateway which came out after mine so it is actually newer and more modern than mine. It is supplied for you to use for comparison. This configuration description reads: "SN46xx to SIP to ISDN Trunk Gateway configuration without SIP Registration. The ISDN PSTN may be connected to Port BRI 0/0 to 0/x."

     

    I will make a suggestion to them to cover my older unit and make the config as forward compatable as possible. What is interesting is what you previously programmed is capturing non-mobile exchanges accurately.

     

    One open concern I have is that if I eventually bring on a new SIP Provider Trunk a similar situation might arrise. Perhaps apples and oranges...

     

    Expect a response next week. More to come.

    isdntrunktosip.txt

  7. I am a little unclear on how to formulate my questions to Patton support. Should I simply ask them to review the template you are using? If they were kind enough to do so, perhaps the resulting edit would be something I can send you.

     

    As far as you presently know, this is the only incident of this happening that you have? or has this appeared under partner configs in Germany. Perhaps one of them had a similar experience. Do I have to enter this incident in the German part of the Forum or is it already viewable there now?

  8. My Gateway is a Patton 4562 and is provisioned by the Vodia as far I understand the currently supplied Vodia software. Is there a specific report on the Patton that would list or indicate such a source of the problem...The same gateway delivers and no coding mismatch exists on landed lines. If your suggestion is valid, then I should be able to trace that back but a Patton 4562 is no longer new.

     

    Does anyone know how to interegate a Patton 4562 to obtain the data you are suggesting?

  9. Following problem is repeating and is not resolved. S/W ver. 5.2.4, Build 22.8.2014

     

    The pbx inserts into inbound call logs the field content "default location - local" rather than accepting the Caller-ID that is presented by the inbound call. The result of this false transcription is that the system goes to the field mentioned and pulls the country code and the local code and inserts them into the record of the incoming (now received) call. The system does not exhibit this behavior on inbound calls received from "landed" phones...only from mobile devices.

     

    The inbound calls are European (German mobile carriers) the typical structure is in this case 10 places deep and in the case of mobile phones defined as +49 1xx xxxxxxx. What the system produces is +4921xx171645xxxx. To make sense of this one has to strip out the 21xx otherwise one has two exchanges listed one after another in sequencefollowed by the proper inbound caller Id.

     

    The system inserts the "domain default location- land code" and "default location - local" on EVERY mobile call regardless of mobile carrier. This creates extra soup in both the calllog and affects the ability to create clean callbacks. The soup also carries into the E-mail strings, but there the system seems able to reference a recall of the string in the Email despite it being rendered improperly. The system seems to be having real problems with these mobile exchange entries.

     

    Is this a bug? Or are there a number of fields or settings that are causing these errors?

     

    Looking forward to your review and imput.

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