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SIP 403 Forbidden auth ID

SIP phone Registration error: 403 - Forbidden (Bad auth) Registration error: 403 - Forbidden (Bad auth) Registration error: 408 - Forbidden (Bad auth) These errors are caused by the firewall, the f5 tmos commands. Log in to the Traffic Management Shell (tmsh) by entering the following command: Tmsh delete a partition To delete the VLAN named.. CM Route pattern --> SIP_Outgoing-RL (Three sip trunks toward VGW with 172.18..10 ; 10.14.22.1 ; 10.14.22.5 destination IPs respectively.) Route Pattern, VGW configu, Numbers and Traces are attached for working and non working calls to Local and National number. In the traces i can see there is SIP/2.0 403 Forbidden

403 Forbidden, weil die Daten natürlich falsch waren ich wollte ja nur sehen, in welchen Feldern was auftaucht. Eine SIP-Anmeldung funktioniert eben ähnlich einer HTTP-Authentifizierung, bei der auch erst mal anonym gearbeitet wird um mit der ersten 401 unautorized zu wissen, welche Anmeldeverfahren der Server denn unterstützt Try to disable the Remote Party ID parameter for outgoing call on the SIP trunk page setting page. Comment actions Permalink. 0. Chad November 27, 2019 13:39. Thanks Riddhe for the reply. Did what you wrote above but the Forbidden message issue still there. See the Trunk -> Advanced Tab . And General -> SIP -> Advanced Tab. I have an S20 in the office and I will replicate the problem and see. SIP/2.0 403 Forbidden 0102301005001 Die Ziffernfolge wird ja wohl vom Telekom-Server übermittlelt. Oft gibt es die lt. google aber nicht. Und da google noch nicht mal meine eigenen Beiträge dazu..

Plötzlich und ohne Vorwarnung, zap aus vorbei die VoIP Nummer geht nicht mehr keine Registrierung möglich. SIP-Fehler 403 (forbidden). Einstellungen prüfen, alles OK. Einstellungen nochmal prüfen mit einem zweiten paar Augen. Alles Ok. Anruf bei der Hotline, Sie verwenden ein Fremdgerät, wir leisten keinen Support. Die Leitung ist O 49 3.504580000 217..18.16 192.168.1.100 SIP 419 Status: 403 Forbidden 0102401005001 (0 bindings) | Es wird Forbidden 0102401005001 zurück gemeldet. Aber: Warum auf einmal Wie es der Zufall will, gibt es zwei SIP-User mit dem gleichen Auth-Name, was ja keinesfalls falsch ist, schließlich sind diese Authentifizierungsdaten ja nur für die jeweilige SIP-ID gültig. Ein REGISTER funktioniert auch anstandslos (was dazu geführt hat, dass das Problem zunächst nicht bemerkt wurde). Kommt jedoch ein INVITE rein, kommt es dazu, dass der LANCOM sagt 403 Forbidden. Im.

SIP-Status-Codes, ungenau auch SIP-Fehler-Codes oder SIP-Responses genannt, bezeichnen die möglichen Antworten auf eine SIP-Anfrage.Das Session Initiation Protocol (SIP) für Aufbau, Steuerung und Abbau einer Kommunikationssitzung (zumeist IP-Telefonie) ist an das Hypertext Transfer Protocol angelehnt. Die Teilnehmergeräte senden sich Anfragen (englisch requests) und beantworten diese. I am receiving a reorder tone when trying to dial outbound and getting SIP/2.0 403 Forbidden-Source Endpoint Lookup Failed with Cause Code 57 in response. The one thing I did notice is that the response is being show VIA the outside CUBE interface (10.0.1.1) and not the provider IP. Neither inbound or outbound work. I do not see a registration unde Mit dem Eintrag des Telekom DNS Servers wird das Problem 403 Forbidden umgangen. Einfach mal 403 forbidden und DNS googeln. ISt seit Wochen nicht abgestellt und betrifft einige PBX System Comparing to one of my Asterisk-to-Asterisk SIP trunks... It looks like what I use is the defaultuser= parameter in my sip.conf as opposed to fromuser=. From the original sip.conf that comes with a make samples-- defaultuser is described as Authentication user for outbound proxies. While it's not a proxy in this case, I believe this is the parameter that will be used when making this SIP.

I get from the PBX a sip packet SIP 403 forbidden (bad auth). To compare, I looked at another register packet from a different sip client that registered successfully. (the PBX sends back OPTIONS Packet). Here are the packets: This packet gets FORBIDEN (bad auth) response (failed I went through the whole setup guide and am trying to setup xlite to start testing. I ran the mysql query to get the password for the 701 extension, entered it into xlite, and am getting an error 403 bad auth message. I proceeded to try configuring my PAP2 and iphone softphone and they, too, won't register <- 407 Proxy Authentication Required-> ACK-> INVITE (now with Auth info) <- 403 Forbidden Not sure why it is forbidden, when the phone talks direct it's OK and works perfectly. The final invite (the one that results in the 403 error) is: INVITE sip:31@10.1.1.3 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.46:5060;branch=z9hG 4bK36B1D64 BA09A41087 FCC96BEF8F AF61 I'm trying to setup a client PBX to register to our SIP server, and i keep getting 403 forbidden (bad auth) back. We are running A2Billing on top of Freepbx. I have posted this on the A2Billing forums but thought i'd try here too. I have checked the username & secret and these are correct. I've recreated the VOIP account as well; still no joy. The PBX is IPFX director; i know i can register this PBX type to this system because i've done it before, and i'm certain i have.

SIP 403 is shown when the server understands your request, but is refusing to fulfill it. You will need to contact your VoIP service provider or PBX administrator for assistance. The most common reasons for this error: - The username and / or password are wrong User-Agent: PolycomSoundStationIP-SPIP_6000-UA/3..4.0061. Max-Forwards: 70. Expires: 3600. Content-Length: 0. 16968981mS SIP Tx: UDP 10.1.40.2:5060 -> 10.1.40.49:5060. SIP/2.0 403 Forbidden. Via: SIP/2.0/UDP 10.1.40.49;branch=z9hG4bKe7fcb180CD916207. From: 259 <sip:259@10.1.40.2>;tag=26869EF3-72D980A2. To: <sip:259@10.1.40

RE: SIP/2.0 403 Forbidden teletechman (TechnicalUser) 22 Feb 19 13:17 Check SSA extension and see if you can see the B179, also in Monitor go to status and SIP phone status it will tell you if it is registered or not gibt es eine Liste der Gründe, bei denen der SIP-Registrar phone.mnet-voip.de eine REGISTER-Anfrage des VoIP-Telefons mit Status SIP/2.0 403 Forbidden beantwortet? Der folgende JSON-Dump wurde mit tcpdump auf dem Router erzeugt und ich sehe auf Anhieb nicht, was hier falsch ist (alle beteiligten Adressen sind IPv6 und meine Adressen kommen aus dem mir zugeteilten /56 Präfix) D29 - 403 Invalid X-Telnyx-Token #{telnyx_auth_token} or IP #{caller_ip} D29 - The SIP INVITE we received contains an Invalid X-Telnyx-Token with IP. Please ensure the token matches what was set on the expert settings of the SIP Connection. Tokens are used to segregate traffic that come where different SIP Connections share the same IP address. D3X. D3 - 403 Connection channel limit exceeded.

Why your OpenSIPS rejects the call with 403 depends on your configuration script....Go and check your opensips cfg and see when the 403 relay forbidden is generated . Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 08/23/2012 03:45 PM, yassine.ab wrote SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 196.17.242.198:5060;branch=z9hG4bK7f227b73;received=196.17.242.198;rport=5060 From: 06649662860 sip:06649662860@196.17.242.198;tag=as090ae269 To: sip:42335122550@196.17.247.227;tag=as5b68b468 Call-ID: 0f81b2d63d525b6d5940ba2d77219966@196.17.242.198 CSeq: 103 INVITE User-Agent: Asterisk PB

Cause: Your SIP infrastructure is replacing a Twilio-specific private IP address in a stacked Via header with a different IP address in a 200 OK. This is likely due to a Global replacement of certain private IP ranges. This will cause the 200 OK to be dropped inside Twilio's infrastructure, preventing an ACK from being sent, and causing your infrastructure to terminate the call IP Office SIP Registration Small and Medium Business Communications Iskratel is sending Unauthorized messaging, then IP Office is sending another Register message with authentication header, and then Iskratel responds with 403 Forbidden. - In the Unauthorized message the Iskratel is giving the parameters for the authentication WWW-Authenticate: Digest realm=172.16.17.10,nonce.

Lync Front-end Service Error: User Authentication: 403, Forbidden When scanning a Lync Front-end service, it may fail to run a User Authentication test. This can be due to bad credentials being used for the Test account configured. This article describes how to validate user credentials outside of GSX Monitor using PowerShell commands or cmdlets. GSX Monitor 10.9+ | Lync 2013 | Powershell. @Klaus_VoIP Danke für deine Einschätzung.. @schluej Klar sollte das mit einer Kombination aus STUN & portweiterleitung in deinem Router irgendwie zu machen sein. Was ich machen möchte ist einen SIP proxy verwenden, wie SIP das vorsieht. Der einzige Grund warum das bei O2 nicht geht ist, dass sie Anfragen ablehnen nach scheinbar willkürlichen Regeln Is it possible that you can use SIP with a non TCP/TLS transport, going to the standard default UDP at 5060? In the the 403 response, I am only seeing the IP of the provider's interface to you and their B2BUA (back-to-back user agent) in the VIA. Are you also using TLS with the other provider? I went back and looked at the syslog

SIP phone Registration error: 403 - Forbidden (Bad auth

  1. SIP Error: 403 Forbidden : make automatic phone calls with auto dialer software with dialing features. Products Download Contact There is normally no such a thing as a 'local call' for your VOIP account. You can do so also by setting a dialing rule on Voicent gateway. If all of the above does not fix the problem, try to use a softphone (like x-lite) or Voicent AgentDialer semi.
  2. And have you #included the extra SIP A2Billing config files? Joe I am sorry i did not understand, which config files and what i need add #included can you tell me
  3. Nach diesen müsste ich auf meine REGISTER-Anfrage vom SIP-Client einen 401 Unauthorized vom Server erhalten und dadurch eine Anmeldung durch dein SIP-Client initieren. Ich erhalte aber ein 403 Forbidden zurück, welches ich jetzt als REGISTER ist von dir nicht erlaubt betrachte. Mein IP-Adresse im SIP-Paket ist meine private IPv4-Adresse.
  4. 403 Forbidden Der Server hat die Anfrage verstanden, aber leider wird dieser Service nicht unterstützt. 404 Not Found Der Server hat die Information übermittelt, dass der angerufene Teilnehmer in dem Netzwerk nicht existiert. 405 Method Not Allowe
  5. The log could perhaps reveal the problem. I see some SIP/2.0 403 Forbidden (Bad auth) and SIP/2.0 401 Unauthorized and SIP/2.0 404 Not Found and Forbidden - wrong password on authentication for REGISTER for '044123456' to '80.91.XXX.XXX' but I do not understand the problem. Could it have something to do with MD5
  6. SIP-Status-Codes, ungenau auch SIP-Fehler-Codes oder SIP-Responses genannt, bezeichnen die möglichen Antworten auf eine SIP-Anfrage.Das Session Initiation Protocol (SIP) für Aufbau, Steuerung und Abbau einer Kommunikationssitzung (zumeist IP-Telefonie) ist an das Hypertext Transfer Protocol angelehnt. Die Teilnehmergeräte senden sich Anfragen (englisch requests) und beantworten diese.

Solved: SIP/2.0 403 Forbidden - Outgoing SIP trunk call ..

Der SIP Trunk funktioniert fehlerfrei mit einem angeschlossennen IP-Telefon. Mit freundlichem Gruß booker solved: Nach Wireshark-Analyse gab der SIP 403-Fehlercode 'Forbidden' den entscheidenen Hinweis. Die Anmeldedaten während des Fax Versand waren falsch. Der SIP Anmeldename stand auf 'Anonym' The identify section makes it so that incoming SIP traffic from the IP address in the The forbidden_retry_interval can be set such that if Asterisk receives a 403 Forbidden response from a registrar, Asterisk can wait the number of seconds indicated and re-attempt registration. Retries that are attempted in this manner count towards the same max_retries value as temporary failure retries. Author Topic: Can't register to my SIP provider, get 403 forbidden (Read 24471 times

SIP Authentication - MSXFA

  1. <--- SIP read from UDP:remote-server-ip:5060 ---> INVITE sip:492212XXXXXXXX6@our-server-ip:5060 SIP/2.0 Via: SIP/2.0/UDP remote-server-ip:5060;rport;branch=z9hG4bK-6d70-1406058084-1272-462 Call-ID: 64d6-439-6222014194124-inCGN2-2-remote-server-ip CSeq: 2 INVITE Max-Forwards: 70 To: <sip:492212XXXXXXXX6@our-server-ip:5060> From: 43650XXXXXXX<sip:43650XXXXXXX@remote-server-ip>;tag.
  2. Hi, we have Lync 2010 Enterprise Enviroment with Server1 (FE, Med, A/V, Conf), Server2 (Edge) and Server3 (Monitoring). Lync 2010 CU1 is installed. Between Sever1 and Server2 is Forefront TMG 2010 SP1 RU4 installed. Server1 (Mediation) connects to an Cisco Callmanager from a Service Provider · Please have a try the following solution: In.
  3. The meaning of SIP-403 Forbidden. The answer that is received when a forbidden request sent to the other side. The consequences of Sip 403. This code causes to lower the VoIP traffic's ASR rates. Low ASR rates are not desired values for healthy traffic. The reasons fo Sip 403 Forbidden. Maybe there is a problem with IP; Problem with number
  4. 2013-03-12 13:17:21.500287 [WARNING] sofia_reg.c:1447 SIP auth failure (REGISTER) on sofia profile 'internal' for [[hidden email]] from ip user_agent_ip send 573 bytes to udp/[user_agent_ip]:5060 at 12:17:21.509394: ----- SIP/2.0 403 Forbidden Via: SIP/2.0/UDP user_agent_ip;branch=z9hG4bK0f9e3669f4d198d3
  5. Interval used when receiving a 403 Forbidden response. fatal_retry_interval . Unsigned Integer. 0. false. Interval used when receiving a Fatal response. server_uri. String false. SIP URI of the server to register against . transport. String false. Transport used for outbound authentication . line. Boolean. no. false. Whether to add a 'line' parameter to the Contact for inbound call matching.
  6. I am doing ip authentication on my voip setup on a elastix box each time I make a call i get a all circuits are busy message and when I do I wireshark trace my service provider is giving me a 403 forbidden message on on the trunk link. I have done all I have know from my limited knowledge so I am looking for some help on this. User #346099 625 posts. IT1rB. Whirlpool Enthusiast reference: whrl.
  7. I setup my PBX connect to sip.skype.com and registration working fine. When I tried to make call out by using E.164 international number, i always get the 403 Forbidden code. The authentication is correct and i have sufficient balance in my account. What can be the reason and how to solve the issue

Response Group call transfer to agent fails with 403 Forbidden Referred-by header parameters are not vali Re: [Sip] About 403 Forbidden. Dean Willis <dean.willis@softarmor.com> Wed, 22 December 2004 17:22 UT Step 2 Log in to the SoftCo and run the show sipue eid command to view the IP phone configurations. This command output also contains the information about whether the IP phone has been registered. The information contains the IP address that the phone is using. Verify that the IP address is the same as the actual IP address of the IP phone. If.

SIP status 403 Foridden when calling an international

mögliche Ursachen für SIP-Registrierungsfehler 403 bei

Gelöst: SIP Fehler 403 in verbindung mit Fremdroutern (Fr

SIP-Status 403 FORBIDDEN 0102401005001 - warum? Telekom

  1. What does registration failed (Forbidden (403)) mean? 0 votes. My Zoiper won't register. It was working fine till a few days ago when that message popped up. -registration-android. asked Aug 28, 2015 in Android by Lilford (120 points) flag answer comment. share. share . share 1 Answer. 0 votes. Please contact your provider, if it worked before and you didnt change the settings, it should still.
  2. Seit Sonntagmorgen gegen 11 Uhr können wir über o2 VoIP keine ausgehenden Telefonate mehr führen. Die Logs der Fritz!Box 7590 melden den Fehler 403 - Forbidden. Eingehend ist möglich, also ist der Account wahrscheinlich nicht der Fehler. Den Router habe ich bereits mehrfach neu gestartet. Das letzte..
  3. SIP 403 - Forbidden. SIP 403 is shown when the server understands your request, but is refusing to fulfill it. You will need to contact your VoIP service provider or PBX a... Last Modified: 2017-02-13 18:31:03 . calls get disconnected after 30 seconds. This issue appears mostly because the session was not actually established and is in most cases network(NAT) related. For example when you.

All-IP/VoIP: SIP-Authentifizierung schlägt fehl (9

Subject: Re: [sipx-users] Voip.ms 403 Forbidden Errors after registered for several hours Just yesterday we figured out, you don't have to use registration if you have a static IP. Everyone says you cannot use IP auth with voip.ms because they cannot send inbound calls via port 5080. The answer is to setup you DID's to use SIP URI for inbound. SIP: Status code : Q.850: Cause value in decimal representation [ITU-T Q.850] Preemption: Cause value in decimal : EMM: Cause value in decimal representation [3GPP TS 24.301 subclause 9.9.3.9, Table 9.9.3.9.1] ESM: Cause value in decimal representation [3GPP TS 24.301 subclause 9.9.4.4, Table 9.9.4.4.1] S1AP-RNL: Radio network layer cause value in decimal representation [3GPP TS 36.413] S1AP.

SIP-Status-Codes - Wikipedi

  1. READ NEXT › How to Search for Tweets from a Specific Date or Time Period › How to Restart or Reset Your Amazon Kindle › How to Manually Approve Facebook Posts You're Tagged in › How to Get the iMac's Hello Screen Saver on Your Mac › How to Hide People from Memories in Google Photo
  2. SIP is a signalling protocol used for creating, modifying, 403 Forbidden 3261 404 Not Found 3261 405 Method not allowed 3261 406 Not acceptable 3261 407 Proxy authentication required 3261 408 Request timeout 3261 410 Gone 3261 412 Conditional request failed 3903 413 Request entity too large 3261 414 Request-URI too long 3261 415 Unsupported media type 3261 416 Unsupported URI scheme 3261.
  3. The Session Initiation Protocol (SIP) is a signalling protocol used for controlling communication sessions such as Voice over IP telephone calls.SIP is based on request/response transactions, in a similar manner to the Hypertext Transfer Protocol (HTTP). Each transaction consists of a SIP request (which will be one of several request methods), and at least one response
  4. SIP bedeutet Session Initiation Protocol und ist in der IP-Telefonie ein Netzprotokoll zum Aufbau, Abbau und Steuerung eines Gespräches. SIP-Fehler-Meldungen helfen Ihnen bei der Identifikation von Fehlerursachen. Sie können diese Fehlermeldungen im Ereignisprotokoll Ihres 1&1 DSL-Routers nachlesen

Hello Rajaram, The first Request-URI is right. You are REGISTERing to the sip:open-ims.test domain. People often make this mistake of assuming that the registration is between a UE and a specific SIP Proxy, when it is between the UE and the abstract network from the UE's point of view SIP认证的框架和HTTP非常接近(RFC2617[17])。特别式,auth-scheme的BNF范式,auth-param, challenge,realm,realm-value,以及信任书都是一样的(虽然对Basic认证方案是不允许的)。在SIP,UAS使用 401(Unauthorized)应答来拒绝UAC的身份(或者讲是考验UAC的身份,如果不通过,就是401)。另外,注册服务器,转发服务器. nachfolgend noch ergänzende (teils anonymisierten) Informationen und der relevante Ausschnitt aus dem SIP-Packet-Trace (in dem nachvollziehbar ist, dass die Telekom den Ruf mit 403 Forbidden ablehnt): Telekom Business-Anschluss Feste IP-Adresse: 12.345.67.89 Quell-Rufnummer (extern): +491721234567 Ziel-Rufnummer (extern): +49691234562 Possible reasons for 403 Forbidden: You set the IP address in the extension settings (Trusted IP Addresses in the extension/registration settings) The From- and To-domain are not the same (should not be the problem here) If you try to subscribe to dialog, the endpoint must have the permission for hat (not the case here because you try to.

SIP Trunking to PSTN - 403 Forbidden / Cause Code 57

403 Forbidden mit Telekom (nicht SIP-Trunk) 3CX

Route: sip:[P-CSCF-IP], // Route specifies the IP of next node for this REGISTER to reach. In this case, 'Next Node' is P-CSCF . Max-Forwards: 20, From: <sip:name@hims.net>;tag=abbb, To: <sip:name@hims.net>, Contact: <sip:[UE-IP]>;expires=90000, Call-ID: ababab, CSeq: 25 REGISTER, Security-Client: port-s, port-c, Authorization: Digest username = name.private@hims.net, Content-Length: 0 . Note. SIP responses, class 4. The 4xx response codes indicate request failures . 400 Bad Request; 401 Unauthorized: Used only by registrars. Proxies should use proxy authorization 407; 402 Payment Required (Reserved for future use) 403 Forbidden 403 Subcode - Forbidden. User Permission Access Issue: 403.1 - Execute access forbidden; 403.2 - Read access forbidden; 403.3 - Write access forbidden . If the user sees the above Error(s), Group Policy has blocked the user from accessing the System. IIS leverages the same Authenticate access as if a user was logging into the system Jul 5 15:27:12 sua [570]: DLG <6+info > [000] SIP/2.0 403 Forbidden^M I verified with the administrator that the password I am entering is correct. We reduced the password length to 8 characters in case the T46G doens't support longer passwords, but that was apparently not the problem Check the codecs allowed in the SIP trunk configuration above, VoiceHost only supports: alaw, ulaw, gsm If a codec is defined in Asterisk that is not one of the above, or is offering a differing sample rate or interval rate (e.g. 8000/20i - 8000Hz at 20ms) cannot interwork with 16000/30i - 16000Hz at 30ms) the call will fail and the codecs in the SIP trunk configuration need to be aligned to.

debugging - Asterisk to asterisk call: 403 Forbidden

How to configure SIP Trunking for Asterisk IP PBX based systems. Our service is 100% compatible with Asterisk using either standard SIP registration or IP authentication where SIP trunks are configured as such. Asterisk is the base software behind many open-source PBX distributions, including FreePBX, Trixbox and Elastix, and is also the enabler behind many other ITSPs and commercial PABX. Error 403: Forbidden! {{error_403_text}} Quick Links: Home Page Web hosting Domain Names 1 Click Scripts Installer Orde

python - pjsua - getting 403 forbidden (bad auth) - Stack

For example, if the OCMS SIP servlet container receives a request that includes the appId parameter, the container searches for the value set for the parameter, such as presence in Example 3-1. If the container cannot find an application named presence, it returns a 403 Response (Forbidden) The example below shows the SIP 1 account setting, Section Services - Phone - SIP 1, The HIP device is registered to the SIP Proxy (2N Netstar SW) 192.168.50.150, The phone number ID is 100, SIP Proxy and SIP proxy registrar Server is also 192.168.50.150 and there is used a default port 5060. The Authentication ID and password is in case also required from the Pbx. It is possible to check the.

Error 403: forbidden (bad auth) The VoIP-info Foru

HTTP/1.1 403 Forbidden Cache-Control: no-cache Content-Type: text/html Server: Microsoft-IIS/8. X-Ms-diagnostics: 28070;source=METFE02.metio.ms;reason=Service does not allow a cross domain request from this origin. X-MS-Server-Fqdn: METFE02.metio.ms X-Powered-By: ASP.NET X-Content-Type-Options: nosniff Date: Mon, 11 May 2015 16:26:08 GMT Content-Length: 123 Wow we can see that the server told us 403 (SIP Forbidden). But, at least the server is nice enough to tell us we were forbidden a location profile because only users that are UC enabled have a location profile. So our user is not UC enabled and therefore gets no location profile. This is part of the in-band provisioning process that all Lync clients go through when they log into the system <synopsis>Interval used when receiving a 403 Forbidden response.</synopsis> <description><para> If a 403 Forbidden is received, chan_pjsip will wait <replaceable>forbidden_retry_interval</replaceable> seconds before: attempting registration again. If 0 is specified, chan_pjsip will not: retry after receiving a 403 Forbidden response. Setting. The tag AUTHTAG_METHOD() specifies the authentication module and scheme to be used by the auth_module. The name can specify a basic authentication module, like Digest or Basic, or an plugin module, like SGMF+Digest. See also: See <sofia-sip/auth_plugin.h> for plugin interface

403 Forbidden 0102301005001; from IP:217..23.100:5060 - Forum Telekom fsobxmeyu.com (hosted on telekom.me) details, including IP, backlinks, redirect information, and reverse IP shared hosting dat was sagt denn die Fehlemeldung 403 (VoiP) bei Euch aus? Laut AVM muss ich diese Information bei Euch einholen. Können Sie evtl. überprüfen (Kundennummer ist hinterlegt), was für ein Fehler bei. Mir blieb also nichts anderes übrig, als die PI-Adressen von sip.1und1.de direkt als Server 1 und Server 2 im Yealink einzutragen. Das ist eine ziemlich dämliche Lösung, vor allem wenn 1und1 ihren SIP-Servern andere (oder noch weitere) PI-Adressen zuweisen sollte. Um dem Ziel (schnell und einfach) zu dienen hier die Zusammenfassung für den interessierten Forumsbesucher: Yealink W60B mit. Also an sip.sipdiscount.com kann ich mich nicht anmelden, da kommt auch immer 403 im Systemlg der Fritzbox. Mit connectionserver- oder sip1.sipdiscount.com (exakt die gleichen Server-IPs) kann ich mich registrieren, aber anstelle dem hier schon mehrfach beschrieben Problem vonwegen es klingelt durch und Angerufener hoert nix, kommt nun auch Fehler 403 und es klingelt nicht mal Polycom has recently announced native Lync support for a wide variety of standard SIP phone devices which all run on the same Polycom Unified Communications Software (UCS) software release. This means that the large variety of SoundPoint IP, SoundStation IP, SpectraLink Wireless, and VVX Business Media Phones can all now natively register directly to Lync Server, adding a variety of choices.

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