Phone still ringing after answer

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If the DIDs are over the same provider, then compare an incoming log of each. What is different? Routing, number format? At what point to they fail? If you send the failing number to the same destination as the working number, does it then work? If so, it's a destination issue. Can you call that destination internally?

Outgoing, even with DIDs simply uses a trunk. the only difference is the caller ID you attach to each, you can't think of each DID call using dedicated routes/channels unless you have more than one provider. The only difference is the number information attached to each call.

All the trunk options, at all locations are the same?
 
Hello,
1. I can call the destination internally. Internal calls do not have the problem.

2. I have setup both DIDs to call the same destination. all settings are exactly same, but one works and other doesnot. Basically i have a phone with Extention number 05. And it is setup to receive calls from both DID-1 and DID-2. When I call DID-1 - i can answer the call and speak. When I call DID-2 the phone rings - but when i answer - i can hear nothing, while my mobile continues to give ringign signals as if the call was not answered.

3. I have checked and all the trunk options at all locations are same. I tired switcheing PBX delivers audio back and forth - no success. yes i understand that outgoing calls all go through the same SIP trunk - its just different Caller id info that is attached. and its great that all outgoing calls work fine now.

4. I will make a verbose log for working and not working call. Can i send it in Private Message?

thank you
 
I have found that my SIP truunk provider was blcoking outgoing calls, if caller ID was set to secondary DID. Now they have changed this setting back and outgoing calls are going out fine.

Is it possible that they have similar restrictions for incoming calls?
 
Anything is possible. For specifics you'd want to talk to them.
 
Yeah, i talk to them - but they say - its not them - because its not possible technically....

Thanks for letting me know though...
 
I have captured a PCAP file. Is it ok to upload it here or should I modify it first?

thank you
 
If it contains sensitive information, such as your public IP or FQDN, then best to edit it.
As far as the sending of calls, to you. Your provider will send them, or not. It is the information that is sent, that will make the difference in how 3CX is able to deal with (act on) the call. When DID information is involved, and you have created rules to use those numbers, then you must be certain that the rules match exactly (or use wildcards), or 3CX will not be able to route correctly. The Activity Log will show you what the number format is.
 
Thank you.

1) I am confident, that my provider send the call info and it is received by 3CX and then 3CX can read the call info and forwards it to the correct destination based on DID info attached. I can see the actual call, just - when I answer it - nothing happens - the call comes in but I am unable to answer it.

2) How can I edit sensitive info from a PCAP file? I can open it with FireShark, but if I edit PCAP file with notepad editor - then when I open it with Wireshark it says it is corrupted..
 
You may want to use Wireshark on the network between the set and the SBC first. See what is happening when the set gets an Invite, then goes off-hook. the messaging should be passed onto 3CX if you look again on the WAN side of the SBC, then again on the WAN side of the Modem/Router. You can then check at the 3CX end. What is missing? If you are monitoring internally, then the FQDN may show in the logs. Not certain about editing those, then reading again using Wireshark
 
I have checked wireshark - it seems like the difference between working and not working instances is that working system connects with G711 Ulaw codec, with session timers disabled, while not working instance connects with PCMA and session timers enabled.

therefore i have 2 questions
  1. Can i disable session timers in 3CX?
  2. I have setup the codecs for SIP trunk i use to g711 Ulaw only, but it still connects via PCMA. Is there anyway to fix this?
 

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Ok we are on page 2... time for the 'This is why you use a supported provider comment' :)

Now that being said, there really is no way it should be using a codec not listed on the trunk. Are you sure you're reading the wireshark correctly? PCMA might show on the extension to 3CX leg (assuming it is one of the configured options there) but it should only be one of the configured codecs on the 3CX to provider side. Something is definitely amiss here.
 
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Dear Cobaltit

Yes i would be happy to use supported provider, Unfortunately there are no supported providers in Japan (international providers do not work because of latency issues and because they can not show correct CID due to Japanese law restrictions. If I use any SIP trunk except the one I am using now - everyone i Call will see UNKNOWN NUMBER instead of my phone number CID).

I really appreciate your help in dealing with me - as I have no alternative. ;)

I will check wireshark again.

How about session timers? any chance of disabling it in 3CX or can it only be done at SIP trunk provider side?

thank you
 
The initiating device will decide the Codec, if it is one supported at the other end. So...if the call you are looking at is between a set and 3CX, then it when the set places a call, the Codec list in the set will determine which is used (first choice if possible). If the call continues to an outside trunk, then 3CX will negotiate with the provider. If the set isn't using the same Codec as 3CX is using with the provider, then transcoding will take place.
You may need to review, the configuration file used by the sets involved, and perhaps compare with the working sets. The you have to figure out why they are different, which is wha it sounds like.
 
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