Zoiper request timeout code 40812/8/2023 ![]() ![]() As stated, we will review softphones in a later in the tutorial. The users and extensions are now registered on Asterisk but the users must also be registered on a SIP or IAX client softphone. When any change is made in conf files from /etc/asterisk/ or changes that relate with some of these files, you must type 'reload' in the Asterisk Command Line Interface (CLI) to make the changes effective. You have completed the registration of 4 users(2 SIP/2 IAX) and 4 extensions. Note that the Dial command when using IAX2 protocol is : call the second registered user, create extension 2222. To call the first user 'ivan_iax' dial 1111. Now register a second IAX user following the same steps. Secret equals your chosen password, host equals 'dynamic IP' and context is 'tutorial'. The user is 'ivan_iax' and type is 'friend' again (Inbound/Oubound calls allowed). Set the host IP to dynamic and create a password as described previously. More detailed configuration information for a series of phones can be found here:Ĭreate user 'ivan_iax' with the same username and join it to the tutorial context. For now, just make sure you have registered the users and extensions. However, softphones will be reviewed later. The final step is to register the user to a compatible softphone. The client MAY repeat the request without modifications at any later time.: §21.4.9 409 Conflict User already registered. The server could not produce a response within a suitable amount of time, for example, if it could not determine the location of the user in time. ![]() Now when user 'ivan' or any other user from the tutorial context dials 4321, the user 'test' will be called.ģ. 408 Request Timeout Couldnt find the user in time. Register the extension(4321) in /etc/asterisk/nf in the same context = tutorial. Start by registering the second user in the same way in /etc/asterisk/sip.conf Follow this same process to register another SIP user and extension in order to place test calls. We now have a registered SIP user and extension on Asterisk. The priority determines the sequence in which the extensions will be executed. The command is : exten => number, priority, Dial(protocol/user). 1 Answer 0 votes Please make sure to disable all features that are not explicitly required by your VoIP provider. Sometimes the error is passed back from IP Exchange through VoiceHost to the customer's system, at which point the call will usually hunt to another route to try and place the call.When dialing number 1234, Asterisk will first Dial the user xlite through SIP protocol. ![]() If our platform replies back with 503 it usually means the gateway trying to process the call can't due to "issues", or the customer has hit their Calls-Per-Second (CPS) limit and is sending too many calls at once. Calls fail with SIP error 503, I-SUP errors 34 or 38: Ascertain how long the 408 error took to come back if it was immediate the trunk is usually unregistered if it took a few seconds the number is usually not mapped correctly. if Asterisk is configured to use plus somewhere else. Inbound calls fail with SIP error 408 (Request Timeout):Ĭheck the inbound number is mapped in the system correctly, if necessary the SIP trunk on the portal can be configured to strip the plus, 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 use one of the above codecs. 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. Outbound calls fail with SIP error 488 (Not Accepted Here) or I-SUP errors 58 (bearer capability not available) or 88 (incompatible destination):Ĭheck the codecs allowed in the SIP trunk configuration above, VoiceHost only supports: alaw, ulaw, gsm Just because a trunk is showing as registered does not mean it will authenticate correctly. This error most commonly occurs when the call is not authenticating properly, at which point check the above in the SIP trunk configuration (If Asterisk, swap username= for defaultuser= to see if this solves the issue. This is the default configuration of Asterisk regardless of the actual error generated (which is infuriating when you are trying to diagnose the real problem) unless PBX is updated to send back the real error rather than the changed error. Outbound calls error with "all circuits busy" or "congestion": ![]()
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