Freepbx hangup cause 16

These codes are provided by your phone carrier to your PBXact system. They are here mainly for reference so when a problem arises on your PRI, you can get an understanding of what the cause code means. Cause No. This cause usually occurs in the same type of situations as cause 1, cause 88, and cause This cause indicates that the destination requested by the calling user cannot be reached because, although the number is in a valid format, it is not currently assigned allocated.

This cause indicates that the equipment sending this cause has received a request to route the call through a particular transit network which it does not recognize.

The equipment sending this cause does not recognize the transit network, either because the transit network does not exist, or because that particular transit network does not serve the equipment which is sending this cause. This cause indicates that the called party cannot be reached because the network through which the call has been routed does not serve the destination desired.

This cause is supported on a network dependent basis. This cause indicates that the called party cannot be reached for reasons that are of a long term nature and that the special information tone should be returned to the calling party. This cause indicates the erroneous inclusion of a trunk prefix in the called party number. This number is snipped from the dialed number being sent to the network by the customer premises equipment.

This cause indicates that the channel most recently identified is not acceptable to the sending entity for use in this call. This cause indicates that the user has been awarded the incoming call and that the incoming call is being connected to a channel already established to that user for similar calls e. This cause indicates that the call is being preempted and the circuit is reserved for reuse by the preempting exchange.

This cause indicates that your provider has sent us a hangup request because the remote party or person you called has hung up their phone.

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This cause is used to indicate that the called party is unable to accept another call because the user busy condition has been encountered. This cause value may be generated by the called user or by the network. In the case of user determined user busy it is noted that the user equipment is compatible with the call.

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What this means is the person you are trying to call is busy. This cause is used when a called party does not respond to a call establishment message with either an alerting or connect indication within the prescribed period of time allocated. What it means is the equipment on the other end does not answer the call. Usually this is a misconfiguration on the equipment being called.

This cause is used when the called party has been alerted but does not respond with a connect indication within a prescribed period of time. Note - This cause is not necessarily generated by Q.

This cause value is used when a mobile station has logged off. Radio contact is not obtained with a mobile station or if a personal telecommunication user is temporarily not addressable at any user-network interface. This cause indicates that the equipment sending this cause does not wish to accept this call, although it could have accepted the call because it is neither busy nor incompatible.

This cause may also be generated by the network, indicating that the call was cleared due to a supplementary service constraint. The diagnostic field may contain additional information about the supplementary service and reason for rejection. What it means is usually a telco issue. The call never reaches the final destination, which can be caused by a bad switch translation, or a misconfiguration on the equipment being called. This cause is returned to a calling party when the called party number indicated by the calling party is no longer assigned.

The new called party number may optionally be included in the diagnostic field. If a network does not support this cause, cause no. This cause indicates that the user has not been awarded the incoming call. This cause indicates that the destination indicated by the user cannot be reached because the interface to the destination is not functioning correctly. The term "not functioning correctly" indicates that a signal message was unable to be delivered to the remote party; e.

This cause indicates that the called party cannot be reached because the called party number is not in a valid format or is not complete.Hope I am not posing a common question…I did have a good hunt around for a solution. I am running FreePBX 14 behind a pfsense firewall. I have followed a couple of guides which improved the time from 15mins to 60mins but it still drops …I changed the firewall optimisation setting to conservative and also Disabled the Firewall Scrub.

This is not needed for SIP trunk providers that use outbound registration. Also, he already has calls working with 2-way audio. So this kind of change would do nothing. This is not accurate. Even in cases where media and signalling come from the same IP, there are certain call scenarios where audio will fail if port forwarding is not in place as I described once in this post. If there is two way audio, then the port mapping is in place and working. If it then stops, something is dropping it.

In this case something in pfSense. The OP changed some settings and things went from 15 minutes seconds to 60 minutes seconds. This clearly tells you that something in the settings is killing things. The issues where happening with calls dropping after 15mins and now they are about an hour or so before dropping. The first re-INVITE generally happens around the minute marker of the call and depending on the provider could happen every 15 or even more minutes to make sure everything is where it still is supposed to be.

If either of those fail to respond the call can be dropped. The server is registering outbound to the sip provider. I have not seen anything odd to suggest that the FW is blocking traffic on the basis of the ports so far. That said, there is a lot of chatter in the logs regarding RTP which I do not understand. I am guessing this could be exactly related to the point made regarding not all required ports being opened.IE stands for Information Element.

Unspecified causes codes no value in the "SIP Equiv. These codes are used internally to FreeSwitch to indicate other states. These codes do not map directly to SIP error codes either. Another set of mappings are the Q. SIG is one of many extensions to Q. Evaluate Confluence today. Page tree.

freepbx hangup cause 16

Browse pages. A t tachments 0 Page History. Jira links. Created by Belaid Areskilast modified by livem Chan on In practice it appears that FreeSwitch implements neither Q. ITU-T Q. No other cause codes applicable. This is usually given by the router when none of the other codes apply.

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This cause usually occurs in the same type of situations as cause 1, cause 88, and cause The equipment sending this cause does not recognize the transit network either because the transit network does not exist or because that particular transit network, while it does exist, does not serve the equipment which is sending this cause.

This cause is supported on a network dependent basis. Under normal situations, the source of this cause is not the network. This cause value may be generated by the called user or by the network. In the case of user determined user busy it is noted that the user equipment is compatible with the call.

Note - This cause is not necessarily generated by Q. The network may also generate this cause, indicating that the call was cleared due to a supplementary service constraint. The diagnostic field may contain additional information about the supplementary service and reason for rejection. If a network does not support this cause, cause no: 1, unallocated unassigned number shall be used. Such an exchange can invoke a redirection mechanism, by use of this cause value, to request a preceding exchange involved in the call to route the call to the new number.

This cause is generated by an intermediate node, which when decrementing the hop counter value, gives the result 0. The term "not functioning correctly" indicates that a signal message was unable to be delivered to the remote party; e.

It is noted that the particular type of access information discarded is optionally included in the diagnostic. However, the information element is not required to be present in the message in order for the equipment sending the cause to process the message.

This is often associated with NAT problems. If it is not NAT related it can sometimes be provider related, make sure to ensure another outbound provider does not solve the problem. The cause indicates that the parameter s were ignored.

In addition, if the equipment sending this cause is an intermediate point, then this cause indicates that the parameter s were passed unchanged. Content Tools. Powered by Atlassian Confluence 6.This section describes terminology, tips and settings that might aid in troubleshooting hangup detection issues most notably within applications using analogue phones. Note in the opposite case the FXO would indicate a hangup by bringing the line to the on-hook voltage level this section is to help with configuring the former.

This enables listening for the beep-beep busy pattern. If busydetect is enabled, it is also possible to specify the cadence of your busy signal.

In many countries, it is msec on, msec off. If you specify busypattern, then we'll further check the length of the sound tone and silence, which will further reduce the chance of a false positive. Use a polarity reversal to mark when a outgoing call is answered by the remote party. In some countries, a polarity reversal is used to signal the disconnect of a phone line.

Few zones are supported at the time of this writing, but may be selected with "progzone". Progzone also affects the pattern used for buzydetect unless busypattern is set explicitly. The symptoms of this is being disconnected in the middle of a call for no reason.

Set the tonezone. This sets the tone zone by number.

freepbx hangup cause 16

Note that you'd still need to load tonezones loadzone in dahdi. The default is not to set anything. FXO FXS signalled devices must have a timeout to determine if there was a hangup before the line was answered. This value can be tweaked to shorten how long it takes before DAHDI considers a non-ringing line to have hungup. This value determines the level which the line voltage must be strictly less then in order for the line to be in a "battery removed" state.

This value defines the time period the wanpipe driver will wait for voltage level changes to settle on the line. So if the far end sends a battery debounce e. The value is decremented once every 4 interrupt periods 1 ms interrupts and therefore determines the settling time with the simple relation:. Evaluate Confluence today.

Telephony Cards. Pages Blog. Page tree. Browse pages. A t tachments 0 Page History. Jira links. Created by Leo D'Alessandro on 14 Jan No labels. Powered by Atlassian Confluence 6.Please could someone help me with this problem. We are using Freepbx for our phone system in school.

We need this to be extended because quite often Government departments, Social Services etc mean that calls being on hold for longer than this is quite frequent.

Can you provide the logs and SIP debug for an example call? This could just as easily be a network issues as opposed to a FreePBX setting.

freepbx hangup cause 16

Sorry for my ignorance - is what I copied and pasted above not the SIPs log? I have no idea. Where should I look to find what you need? I have the administrator password but not sure which log I should look at?

Can you explain how these calls are placed on hold? What it seems to me, that the extension disconnected the call. You might want to check the phone as well if it has any hold limitations. We just spent 30 minutes on the phone call - actively talking and at exactly 30 minutes duration seconds to be exact! I will try to get the logs you have requested - I have the call identifications for the call we tested with this morning so will come back to you.

Thank you for your help! For the purpose of keeping everything together - I contacted Gamma and they have come back with the following. I had a look at the call at from to and I believe I have found the issue.

When the call is setup, the invite you send specifies that the Timer function is supported which is fine. Can you contact your PBX maintainer and ask them to look over the settings and see if they can get the signalling to include refreshers? There are two approaches and the choice is yours. Both of these are common misconfigurations, so there are plenty of people in the forum that have solved this problem.

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Either solution, though, is workable. Thank you so much! The problem is now resolved using Option 1 and Option 2 will be implemented once I can get my head around what I need to do!!

This topic was automatically closed 31 days after the last reply. New replies are no longer allowed. Hi, Please could someone help me with this problem. Can someone please help me fix this as it is becoming quite an issue with the Office staff.

Many Thanks, Shelley. And in case you are not aware of, parking works wonderfully. For the purpose of keeping everything together - I contacted Gamma and they have come back with the following Hi Shelley, I had a look at the call at from to and I believe I have found the issue.

I have attached a pcap for review. Kind regards.Testing the connection by placing an inbound call. When I answer the call just hangs up. Log entry below. Can anyone please assist? The new section of the log is below. Your assistance would be greatly appreciated. Also, in the future, please post pastebin links, or select preformatted text. Still get the messages despite changing my phone to use PCMU as the prefered codec. I believe that is the same as ulaw. Changed the sip settings to only have ulaw as well as the trunk but still the above.

Am I missing a setting? The elephant in the room is why are you allowing anonymous calls into your PBX? That should be stopping your calls way before the phones get involved. This is a trunk issue and with the recent spate of PBX Sangoma is seeing from Iran, Turkey, and Russia, it would really be a good idea to get your configuration under control.

Preferred Vocoder Configures vocoders in a preference list up to 8 preferred vocoders that will be included with same order in SDP message. Vocoder types are G. At the Asterisk command prompt, type pjsip set logger on make a failing incoming call, paste the Asterisk log for the call which will now include a SIP trace at pastebin.

Thank you. You should also fix your trunk settings, because allowing anonymous calls is a security risk. Set the Match Permit field to include all IP addresses from which the provider can send you calls. The trunk should be Diamondcard which is a pjsip trunk. My sipgate trunk is disabled at the moment. I already only had ulaw enabled in the main general area but I have enabled u and a and moved alaw to the top.

With luck, it will at least recognize the trunk. If you post a new log, you must reissue pjsip set logger on because a reload cancels that setting. But your call came from Inbound call - phone rings but hangup on answer FreePBX. Still a codec issue. From the manual Preferred Vocoder Configures vocoders in a preference list up to 8 preferred vocoders that will be included with same order in SDP message. Stewart1 UTC Kind regards. In your DiamondCard pjsip trunk, set Match Permit to Thanks again Stewart.

That fixed the issue!Friends, I have been beating my head against the wall all day on this and just cannot figure it out. They claim there is no ID or secret to use, they use the external IP address for security. Firewall is turned off. This is the same type of config I have done for years on many different providers. I have been playing with trunk settings, adding and removing a myriad of settings and cannot get calls to move in or out.

Retransmitting 1 NAT to If I try to make a call in our out - the call just sits and never goes anywhere, just silence.

Ive never had to do this before with any other provider… WHat can I send? The current trunk settings are:. Currently inbound is blank - but I have tried many of the above settings in there too…If I ping the SIP trunk address from the console, it responds fine. Hopefully this makes sense - let me know what else I can give. This is the latest FreePBX distro with all updates as of Does eth1 has a public static IP? If so, you need to set it on the SIP settings on the settings menu.

Made those changes. Are you sure that the static route is actually working?

Have you tried a traceroute to the IP of the provider? Looks like the provider is not proxying RTP. Routing the SIP server address alone to eth1 is not enough. If the default gateway is eth0, then all the audio traffic is going out the wrong interface.

Is any of that familiar, perhaps the location of a remote extension? Are they your ISP as well as your trunking provider? If not, please explain the setup. If yes: Why do you have the default gateway on eth0? Is there a different ISP providing that? Thanks for all the help everyone. I have narrowed this down to an actual networking problem… Your guidance certainly helped.

The traceroutes and pings were being rerouted over the internal LAN connection… I missed that part. Thanks all - ill post back when I finally get it. I see this written all the time and it makes ZERO sense. Gasoline is manufactured to specific standards to which the engine specifications conform. The same is largely true of SIP.


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