Hello ,Thanks in advance for any and all assistance on this issue.
Recently i began using Bluetooth voip gateway -Gempro GP-708 GSM/3G/WCDMA VoIP PBX for call termination based on Asterisk 14.1 .
To controll each channel from voip switch i created multiple SIP trunks to the same ip with different user name and password ,where each sip trunks connects to 1 specified bluetooth devices for example Sip Trunk 1 sends call to BT1 only,Sip Trunk 2 sends calls to BT2 only,etc…
Sip trunks succefully registered to the voip switch but impossible to get calls. getting promt 503 after number is dialed.
Can anyone advice what can be issue or how to solve it?
Please not in case i removed all trunks ,remained 1 trunk with the same settings work as well.
Regards
You need to be more specific. What channel driver are you using? Also, Asterisk 14.1 looks a bit old. All I can say is that Asterisk 19.2 using PJSIP works fine with multiple “trunks” to the same IP. In case you are using chan_sip, there might not be the necessary options to filter the different SIP accounts.
Dear EkFudrek Thanks for your reply
Unfortunately I can not understand asterisk well, I am still studying. I will be glad if u find a way to log in to my computer with Any Desk and help me solve this problem, I think only a professional can help me. I hope you understand me correctly, it is important to me and I am ready to pay for it on demand.
regards
nikol
Telegram/whatsapp/viber :+995555577100
Sorry, but this is not possible. You probably need to ask for paid support.
that`s ok. Thanks for your attention and support
Dear Friends
CDR analysis revealed that the call from all sip trunks arrives the GP-708 PBX, but despite of there are connected BT-1(first) and BT-2(second) bluetooth devices it tries to connect only BT-8 (eighth channel).
May be it gives a tittle clarification to solve this issue
Regards
“”,“001001”,“5689776643”,“DID_1BT008”,“001001”,“SIP/1BT008-00d58198”,“Zap/8-1”,“NoOp”,"",“2007-01-01 04:21:32”,“2007-01-01 04:21:32”,0,0,"NO
That would suggest Asterisk 1.4, rather than Asterisk 14. or at least that you used documentation for Asterisk 1.4 or earlier. I guess the code might still recognize “Zap” as a synonym of “dahdi”. Asterisk 1.4 is less than two months short of being a decade past full end of life.
Also the CDR is of limited use without full details of the configuration.
You also seem to be using some sort of GUI that I’ve never seen before. Generally Asterisk GUIs include a lot of custom dialplan code and can only be supported by their creators. (Currently there is only one GUI that has any real support, and that is FreePBX.)
I am Using the Gempro GP-708 to terminate VOIP calls via 8 Bluetooth trunks.
I created 8 Sip Truks for incoming calls, where each trunk must have a corresponding outgoing Bluetooth trunk for termination (SipTrunk1 to BluetoothTrunk1, SipTrunk2 to BluetoothTrunk2, etc.)
created an inbound routing rule Destination [run Macro Trunkdial]
exten=_X!,1,Macro(trunkdial,${trunk_1}/${EXTEN:0},trunk_2,)
exten=_X!,1,Macro(trunkdial,${trunk_2}/${EXTEN:0},trunk_3,) etc.
But no matter which Sip trunk I call from, it only rings in the eighth Bluetooth trunk
Has anyone had a similar situation?
Could you recommend something?
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