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SIP Simple massage error

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Dear all

I’m not expert and I need some helps:

I configured a working freePBX instance using SANGOMA distribution for my home (NOT OFFICE… home :slight_smile: ).
Inbound, outbound and IVR is working fine, but I need your help for SIP messages.

I explain my situation
DNS CNAME record registered sip.mydomain.it that redirect to a dynamic DNS mydynamycdns.ddns.net ( I have no fixed IP)

I enabled sip messages for my 4 digit extension using the following script in extensions_custom.conf


[myMessages] ; myMessages is the name of the dialplan and can be changed to whatever you want ( also change in extensions )
exten => _.,1,NoOp(mymessage receiving dialplan invoked)
exten => _.,n,NoOp(To ${MESSAGE(to)})
exten => _.,n,NoOp(From ${MESSAGE(from)})
exten => _.,n,NoOp(Body ${MESSAGE(body)})
exten => _.,n,Set(ACTUALTO=${CUT(MESSAGE(to),@,1)})
exten => _.,n,MessageSend(${ACTUALTO},${MESSAGE(from)})
exten => _.,n,NoOp(mymessage Send status is ${MESSAGE_SEND_STATUS})
exten => _.,n,GotoIf($["${MESSAGE_SEND_STATUS}" != "SUCCESS"]?sendfailedmsg)
exten => _.,n,Hangup()
;
; Handle failed messaging
exten => _.,n(sendfailedmsg),Set(MESSAGE(body)="[${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)}] Your message to ${EXTEN} has failed. Retry later.")
exten => _.,n,Set(ME_1=${CUT(MESSAGE(from),<,2)})
exten => _.,n,Set(ACTUALFROM=${CUT(ME_1,@,1)})
exten => _.,n,MessageSend(${ACTUALFROM},ServiceCenter) ; ServiceCenter is just a name for the sender of the message
exten => _.,n,Hangup()
exten => _.,n,Hangup() 

in Extension–>Advanced I set message context to myMessages

The result is that when I send a message from extension 1001 to extension 1002 it is delivered correctly, when extension 1002 reply, often it doesn’t arrive (rarely yes and I don’t know why)

this is the log, in this case 1001 sent a message to 1002, it was received by 1002;
1002 reply to 1001 but 1001 doesn’t receive it:

10254	[2024-03-12 08:29:44] VERBOSE[2404][C-00000001] pbx.c: Executing [1002@myMessages:1] NoOp("Message/ast_msg_queue", "mymessage receiving dialplan invoked") in new stack	
10255	[2024-03-12 08:29:44] VERBOSE[2404][C-00000001] pbx.c: Executing [1002@myMessages:2] NoOp("Message/ast_msg_queue", "To pjsip:1002@sip.mydomain.it") in new stack	
10256	[2024-03-12 08:29:44] VERBOSE[2404][C-00000001] pbx.c: Executing [1002@myMessages:3] NoOp("Message/ast_msg_queue", "From <sip:1001@sip.mydomain.it>") in new stack	
10257	[2024-03-12 08:29:44] VERBOSE[2404][C-00000001] pbx.c: Executing [1002@myMessages:4] NoOp("Message/ast_msg_queue", "Body tyu") in new stack	
10258	[2024-03-12 08:29:44] VERBOSE[2404][C-00000001] pbx.c: Executing [1002@myMessages:5] Set("Message/ast_msg_queue", "ACTUALTO=pjsip:1002") in new stack	
10259	[2024-03-12 08:29:44] VERBOSE[2404][C-00000001] pbx.c: Executing [1002@myMessages:6] MessageSend("Message/ast_msg_queue", "pjsip:1002,<sip:1001@sip.mydomain.it>") in new stack	
10260	[2024-03-12 08:29:44] WARNING[13307] res_pjsip_messaging.c: To address '1002@sip.mydomain.it' is not a valid SIP/SIPS URI	
10261	[2024-03-12 08:29:44] VERBOSE[2404][C-00000001] pbx.c: Executing [1002@myMessages:7] NoOp("Message/ast_msg_queue", "mymessage Send status is SUCCESS") in new stack	
10262	[2024-03-12 08:29:44] VERBOSE[2404][C-00000001] pbx.c: Executing [1002@myMessages:8] GotoIf("Message/ast_msg_queue", "0?sendfailedmsg") in new stack	
10263	[2024-03-12 08:29:44] VERBOSE[2404][C-00000001] pbx.c: Executing [1002@myMessages:9] Hangup("Message/ast_msg_queue", "") in new stack	
10264	[2024-03-12 08:29:44] VERBOSE[2404][C-00000001] pbx.c: Spawn extension (myMessages, 1002, 9) exited non-zero on 'Message/ast_msg_queue'	
10265	[2024-03-12 08:29:50] VERBOSE[16476] res_pjsip_registrar.c: Added contact 'sip:1002@5.90.140.64:55099;rinstance=1FB1CA5F;x-ast-orig-host=10.51.182.176:2839' to AOR '1002' with expiration of 600 seconds	
10266	[2024-03-12 08:29:50] VERBOSE[8223] res_pjsip/pjsip_options.c: Contact 1002/sip:1002@5.90.140.64:55099;rinstance=1FB1CA5F;x-ast-orig-host=10.51.182.176:2839 is now Reachable. RTT: 38.994 msec	
10267	[2024-03-12 08:29:59] VERBOSE[2404][C-00000001] pbx.c: Executing [1001@myMessages:1] NoOp("Message/ast_msg_queue", "mymessage receiving dialplan invoked") in new stack	
10268	[2024-03-12 08:29:59] VERBOSE[2404][C-00000001] pbx.c: Executing [1001@myMessages:2] NoOp("Message/ast_msg_queue", "To pjsip:1001@sip.mydomain.it") in new stack	
10269	[2024-03-12 08:29:59] VERBOSE[2404][C-00000001] pbx.c: Executing [1001@myMessages:3] NoOp("Message/ast_msg_queue", "From <sip:1002@sip.mydomain.it>") in new stack	
10270	[2024-03-12 08:29:59] VERBOSE[2404][C-00000001] pbx.c: Executing [1001@myMessages:4] NoOp("Message/ast_msg_queue", "Body uyt") in new stack	
10271	[2024-03-12 08:29:59] VERBOSE[2404][C-00000001] pbx.c: Executing [1001@myMessages:5] Set("Message/ast_msg_queue", "ACTUALTO=pjsip:1001") in new stack	
10272	[2024-03-12 08:29:59] VERBOSE[2404][C-00000001] pbx.c: Executing [1001@myMessages:6] MessageSend("Message/ast_msg_queue", "pjsip:1001,<sip:1002@sip.mydomain.it>") in new stack	
10273	[2024-03-12 08:29:59] WARNING[13307] res_pjsip_messaging.c: To address '1001@sip.mydomain.it' is not a valid SIP/SIPS URI	
10274	[2024-03-12 08:29:59] VERBOSE[2404][C-00000001] pbx.c: Executing [1001@myMessages:7] NoOp("Message/ast_msg_queue", "mymessage Send status is SUCCESS") in new stack	
10275	[2024-03-12 08:29:59] VERBOSE[2404][C-00000001] pbx.c: Executing [1001@myMessages:8] GotoIf("Message/ast_msg_queue", "0?sendfailedmsg") in new stack	
10276	[2024-03-12 08:29:59] VERBOSE[2404][C-00000001] pbx.c: Executing [1001@myMessages:9] Hangup("Message/ast_msg_queue", "") in new stack	
10277	[2024-03-12 08:29:59] VERBOSE[2404][C-00000001] pbx.c: Spawn extension (myMessages, 1001, 9) exited non-zero on 'Message/ast_msg_queue'	

furthermore I have two clients installed on my phone (to test them and choose the best one for my purposes) Zopier pro and GroundWire, but when messages were recieved, sometimes go to GroundWire, sometimes to Zopier…there is the possibility to set that messages are delivered to all clients configured for the extensions?

thanks in advance for your help

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Upgrading from v14 to 15 using the upgrader tool

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Hello,
I want to upgrade my FreePBX v14 system to 15 using the upgrader tool. This is my first time upgrading the system. We currently do not have a test environment. I wanted to know approximately how long the upgrade process takes?

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Extensions/user management platform

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I’ve never been a 3CX fan, however I played a little with its web centralized user management platform, where one can manage user status, presence, call progress, etc.
Single user can eventually even display all collegues status in a web panel.
Does something like exist for freepbx, natively or maintained third party suite ?

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Ivr location

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How can I find the ivr file on MobaXtern. please answer me

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Copy ivr file

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how can i copy ivr file
somebody answer me please
thanks

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Sometimes ring time is short

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On one installation my customer is reporting short ring times of about 15 to 16 sec… After this time app-missedcall-hangup is executed. In most cases this doesn´t matter, because the call is answered within this time. The effect seems to appear only on internal calls. There are times the effect doesn´t occure. Probably not all extensions are affected in this case. Any suggestions or similar experiances?

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Astricon: FreePBX Talk about changes over the last year and what is coming

Wrong "outisbusy" message on some phone numbers

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I have a working FreePBX install. We are calling about 200 phone numbers each day without any problem. However I encounter a problem with some numbers that are reporting “outisbusy”. I can NEVER call those specific numbers. That call does not even reach our SIP provider. It is blocked in the PBX. I can normally call that number with my cellphone. So it is NOT busy. No numbers are blacklisted. Even if I specify this specific number in my DIALPATTERN list I am unable to call it. Again, the call attempt never shows up in my SIP providers log.

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Specific outside number dialed from any extension routes to main IVR instead of Outbound Route

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Whenever any user dials a specific legitimate outside phone number from any extension, FreePBX ignores the Outbound Route pattern rules and directs the call to our main IVR. I even tried adding an outbound route pattern specifically for the number, and it still routes to the main IVR. Thinking our SIP trunk provider might be routing the calI back to us I contacted them and they have no idea.
Thoughts!!! Be forewarned I am not a newbie, but I am a LONG way from an expert…

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Cannot determine best translation path since one capability supports no formats

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– Executing [12039874400@from-internal:1] Macro(“Message/ast_msg_queue”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“Message/ast_msg_queue”, “TOUCH_MONITOR=1710518940.0”) in new stack
– Executing [s@macro-user-callerid:2] Set(“Message/ast_msg_queue”, “CHANCONTEXT=”) in new stack
– Executing [s@macro-user-callerid:3] Progress(“Message/ast_msg_queue”, “”) in new stack
– Executing [s@macro-user-callerid:4] Set(“Message/ast_msg_queue”, “CHANCONTEXT=”) in new stack
– Executing [s@macro-user-callerid:5] Set(“Message/ast_msg_queue”, “CHANEXTENCONTEXT=ast_msg_queue”) in new stack
– Executing [s@macro-user-callerid:6] Set(“Message/ast_msg_queue”, “CHANEXTEN=ast_msg_queue”) in new stack
– Executing [s@macro-user-callerid:7] Set(“Message/ast_msg_queue”, “CALLERID(number)=”) in new stack
– Executing [s@macro-user-callerid:8] Set(“Message/ast_msg_queue”, “AMPUSER=”) in new stack
– Executing [s@macro-user-callerid:9] Set(“Message/ast_msg_queue”, “HOTDESCKCHAN=ast_msg_queue”) in new stack
– Executing [s@macro-user-callerid:10] Set(“Message/ast_msg_queue”, “HOTDESKEXTEN=ast_msg_queue”) in new stack
– Executing [s@macro-user-callerid:11] Set(“Message/ast_msg_queue”, “HOTDESKCALL=0”) in new stack
– Executing [s@macro-user-callerid:12] ExecIf(“Message/ast_msg_queue”, “0?Set(HOTDESKCALL=1)”) in new stack
– Executing [s@macro-user-callerid:13] ExecIf(“Message/ast_msg_queue”, “0?Set(CALLERID(name)=)”) in new stack
– Executing [s@macro-user-callerid:14] GotoIf(“Message/ast_msg_queue”, “0?report”) in new stack
– Executing [s@macro-user-callerid:15] ExecIf(“Message/ast_msg_queue”, “1?Set(REALCALLERIDNUM=)”) in new stack
– Executing [s@macro-user-callerid:16] Set(“Message/ast_msg_queue”, “AMPUSER=”) in new stack
– Executing [s@macro-user-callerid:17] GotoIf(“Message/ast_msg_queue”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:18] Set(“Message/ast_msg_queue”, “AMPUSERCIDNAME=”) in new stack
– Executing [s@macro-user-callerid:19] ExecIf(“Message/ast_msg_queue”, “0?Set(__CIDMASQUERADING=TRUE)”) in new stack
– Executing [s@macro-user-callerid:20] GotoIf(“Message/ast_msg_queue”, “1?report”) in new stack
– Goto (macro-user-callerid,s,29)
– Executing [s@macro-user-callerid:29] NoOp(“Message/ast_msg_queue”, “Macro Depth is 1”) in new stack
– Executing [s@macro-user-callerid:30] GotoIf(“Message/ast_msg_queue”, “1?report2:macroerror”) in new stack
– Goto (macro-user-callerid,s,31)
– Executing [s@macro-user-callerid:31] GotoIf(“Message/ast_msg_queue”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,50)
– Executing [s@macro-user-callerid:50] Set(“Message/ast_msg_queue”, “CALLERID(number)=”) in new stack
– Executing [s@macro-user-callerid:51] Set(“Message/ast_msg_queue”, “CALLERID(name)=”) in new stack
– Executing [s@macro-user-callerid:52] GotoIf(“Message/ast_msg_queue”, “1?cnum”) in new stack
– Goto (macro-user-callerid,s,54)
– Executing [s@macro-user-callerid:54] Set(“Message/ast_msg_queue”, “CDR(cnum)=”) in new stack
– Executing [s@macro-user-callerid:55] Set(“Message/ast_msg_queue”, “CHANNEL(language)=en”) in new stack
– Executing [12039874400@from-internal:2] Gosub(“Message/ast_msg_queue”, “sub-record-check,s,1(out,12039874400,dontcare)”) in new stack
– Executing [s@sub-record-check:1] GotoIf(“Message/ast_msg_queue”, “0?initialized”) in new stack
– Executing [s@sub-record-check:2] Set(“Message/ast_msg_queue”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [s@sub-record-check:3] Set(“Message/ast_msg_queue”, “NOW=1710518940”) in new stack
– Executing [s@sub-record-check:4] Set(“Message/ast_msg_queue”, “__DAY=15”) in new stack
– Executing [s@sub-record-check:5] Set(“Message/ast_msg_queue”, “__MONTH=03”) in new stack
– Executing [s@sub-record-check:6] Set(“Message/ast_msg_queue”, “__YEAR=2024”) in new stack
– Executing [s@sub-record-check:7] Set(“Message/ast_msg_queue”, “__TIMESTR=20240315-120900”) in new stack
– Executing [s@sub-record-check:8] Set(“Message/ast_msg_queue”, “__FROMEXTEN=unknown”) in new stack
– Executing [s@sub-record-check:9] Set(“Message/ast_msg_queue”, “__MON_FMT=wav”) in new stack
– Executing [s@sub-record-check:10] NoOp(“Message/ast_msg_queue”, “Recordings initialized”) in new stack
– Executing [s@sub-record-check:11] ExecIf(“Message/ast_msg_queue”, “0?Set(ARG3=dontcare)”) in new stack
– Executing [s@sub-record-check:12] Set(“Message/ast_msg_queue”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [s@sub-record-check:13] ExecIf(“Message/ast_msg_queue”, “0?Set(REC_STATUS=NO)”) in new stack
– Executing [s@sub-record-check:14] GotoIf(“Message/ast_msg_queue”, “3?checkaction”) in new stack
– Goto (sub-record-check,s,17)
– Executing [s@sub-record-check:17] GotoIf(“Message/ast_msg_queue”, “1?sub-record-check,out,1”) in new stack
– Goto (sub-record-check,out,1)
– Executing [out@sub-record-check:1] NoOp(“Message/ast_msg_queue”, “Outbound Recording Check from unknown to 12039874400”) in new stack
– Executing [out@sub-record-check:2] Set(“Message/ast_msg_queue”, “RECMODE=”) in new stack
– Executing [out@sub-record-check:3] ExecIf(“Message/ast_msg_queue”, “1?Goto(routewins)”) in new stack
– Goto (sub-record-check,out,7)
– Executing [out@sub-record-check:7] Gosub(“Message/ast_msg_queue”, “recordcheck,1(dontcare,out,12039874400)”) in new stack
– Executing [recordcheck@sub-record-check:1] NoOp(“Message/ast_msg_queue”, “Starting recording check against dontcare”) in new stack
– Executing [recordcheck@sub-record-check:2] Goto(“Message/ast_msg_queue”, “dontcare”) in new stack
– Goto (sub-record-check,recordcheck,3)
– Executing [recordcheck@sub-record-check:3] Return(“Message/ast_msg_queue”, “”) in new stack
– Executing [out@sub-record-check:8] Return(“Message/ast_msg_queue”, “”) in new stack
– Executing [12039874400@from-internal:3] ExecIf(“Message/ast_msg_queue”, “0 ?Set(CDR(accountcode)=)”) in new stack
– Executing [12039874400@from-internal:4] Set(“Message/ast_msg_queue”, “_ROUTEID=3”) in new stack
– Executing [12039874400@from-internal:5] Set(“Message/ast_msg_queue”, “_ROUTENAME=VoIP_Innovations-Out”) in new stack
– Executing [12039874400@from-internal:6] Set(“Message/ast_msg_queue”, “MOHCLASS=default”) in new stack
– Executing [12039874400@from-internal:7] Set(“Message/ast_msg_queue”, “_CALLERIDNAMEINTERNAL=”) in new stack
– Executing [12039874400@from-internal:8] Set(“Message/ast_msg_queue”, “_CALLERIDNUMINTERNAL=”) in new stack
– Executing [12039874400@from-internal:9] Set(“Message/ast_msg_queue”, “_EMAILNOTIFICATION=FALSE”) in new stack
– Executing [12039874400@from-internal:10] Set(“Message/ast_msg_queue”, “_NODEST=”) in new stack
– Executing [12039874400@from-internal:11] Macro(“Message/ast_msg_queue”, “dialout-trunk,1,12039874400,”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“Message/ast_msg_queue”, “DIAL_TRUNK=1”) in new stack
– Executing [s@macro-dialout-trunk:2] ExecIf(“Message/ast_msg_queue”, “0?Set(DIAL_OPTIONS=Hhtr)”) in new stack
– Executing [s@macro-dialout-trunk:3] GosubIf(“Message/ast_msg_queue”, “0?sub-pincheck,s,1()”) in new stack
– Executing [s@macro-dialout-trunk:4] ExecIf(“Message/ast_msg_queue”, “0?Set(CALLERID(num)=)”) in new stack
– Executing [s@macro-dialout-trunk:5] GotoIf(“Message/ast_msg_queue”, “0?disabletrunk,1”) in new stack
– Executing [s@macro-dialout-trunk:6] Set(“Message/ast_msg_queue”, “DIAL_NUMBER=12039874400”) in new stack
– Executing [s@macro-dialout-trunk:7] Set(“Message/ast_msg_queue”, “DIAL_TRUNK_OPTIONS=HhTtr”) in new stack
– Executing [s@macro-dialout-trunk:8] Set(“Message/ast_msg_queue”, “OUTBOUND_GROUP=OUT_1”) in new stack
– Executing [s@macro-dialout-trunk:9] Set(“Message/ast_msg_queue”, “DIAL_TRUNK_OPTIONS=”) in new stack
– Executing [s@macro-dialout-trunk:10] GotoIf(“Message/ast_msg_queue”, “1?nomax”) in new stack
– Goto (macro-dialout-trunk,s,12)
– Executing [s@macro-dialout-trunk:12] GotoIf(“Message/ast_msg_queue”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:13] Macro(“Message/ast_msg_queue”, “outbound-callerid,1”) in new stack
– Executing [s@macro-outbound-callerid:1] NoOp(“Message/ast_msg_queue”, “”) in new stack
– Executing [s@macro-outbound-callerid:2] NoOp(“Message/ast_msg_queue”, “”) in new stack
– Executing [s@macro-outbound-callerid:3] NoOp(“Message/ast_msg_queue”, “”) in new stack
– Executing [s@macro-outbound-callerid:4] ExecIf(“Message/ast_msg_queue”, “0?Set(CALLERID(name-pres)=)”) in new stack
– Executing [s@macro-outbound-callerid:5] ExecIf(“Message/ast_msg_queue”, “0?Set(CALLERID(num-pres)=)”) in new stack
– Executing [s@macro-outbound-callerid:6] Set(“Message/ast_msg_queue”, “HOTDESCKCHAN=ast_msg_queue”) in new stack
– Executing [s@macro-outbound-callerid:7] Set(“Message/ast_msg_queue”, “HOTDESKEXTEN=ast_msg_queue”) in new stack
– Executing [s@macro-outbound-callerid:8] Set(“Message/ast_msg_queue”, “HOTDESKCALL=0”) in new stack
– Executing [s@macro-outbound-callerid:9] ExecIf(“Message/ast_msg_queue”, “0?Set(HOTDESKCALL=1)”) in new stack
– Executing [s@macro-outbound-callerid:10] ExecIf(“Message/ast_msg_queue”, “0?Set(CALLERID(name)=)”) in new stack
– Executing [s@macro-outbound-callerid:11] Set(“Message/ast_msg_queue”, “ALLOWTHISROUTE=NO”) in new stack
– Executing [s@macro-outbound-callerid:12] ExecIf(“Message/ast_msg_queue”, “0?Set(ALLOWTHISROUTE=YES)”) in new stack
– Executing [s@macro-outbound-callerid:13] ExecIf(“Message/ast_msg_queue”, “0?Hangup()”) in new stack
– Executing [s@macro-outbound-callerid:14] ExecIf(“Message/ast_msg_queue”, “1?Set(REALCALLERIDNUM=)”) in new stack
– Executing [s@macro-outbound-callerid:15] ExecIf(“Message/ast_msg_queue”, “0?Set(AMPUSER=unknown)”) in new stack
– Executing [s@macro-outbound-callerid:16] GotoIf(“Message/ast_msg_queue”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,20)
– Executing [s@macro-outbound-callerid:20] Set(“Message/ast_msg_queue”, “USEROUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:21] Set(“Message/ast_msg_queue”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:22] ExecIf(“Message/ast_msg_queue”, “0?Set(SCDEXT=YES)”) in new stack
– Executing [s@macro-outbound-callerid:23] ExecIf(“Message/ast_msg_queue”, “0?Set(EMERGENCYCID=)”) in new stack
– Executing [s@macro-outbound-callerid:24] ExecIf(“Message/ast_msg_queue”, “0?Set(CALLERID(name)=)”) in new stack
– Executing [s@macro-outbound-callerid:25] ExecIf(“Message/ast_msg_queue”, “0?Set(EMERGENCYCID=)”) in new stack
– Executing [s@macro-outbound-callerid:26] Set(“Message/ast_msg_queue”, “TRUNKOUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:27] GotoIf(“Message/ast_msg_queue”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,33)
– Executing [s@macro-outbound-callerid:33] ExecIf(“Message/ast_msg_queue”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:34] ExecIf(“Message/ast_msg_queue”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:35] ExecIf(“Message/ast_msg_queue”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:36] ExecIf(“Message/ast_msg_queue”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:37] ExecIf(“Message/ast_msg_queue”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:38] Set(“Message/ast_msg_queue”, “TIOHIDE=no”) in new stack
– Executing [s@macro-outbound-callerid:39] ExecIf(“Message/ast_msg_queue”, “0?Set(CALLERID(name-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:40] ExecIf(“Message/ast_msg_queue”, “0?Set(CALLERID(num-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:41] ExecIf(“Message/ast_msg_queue”, “0?Set(CALLERID(name-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:42] ExecIf(“Message/ast_msg_queue”, “0?Set(CALLERID(num-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:43] Set(“Message/ast_msg_queue”, “CDR(outbound_cnum)=”) in new stack
– Executing [s@macro-outbound-callerid:44] Set(“Message/ast_msg_queue”, “CDR(outbound_cnam)=”) in new stack
– Executing [s@macro-dialout-trunk:14] GosubIf(“Message/ast_msg_queue”, “0?sub-flp-1,s,1()”) in new stack
– Executing [s@macro-dialout-trunk:15] Set(“Message/ast_msg_queue”, “OUTNUM=12039874400”) in new stack
– Executing [s@macro-dialout-trunk:16] Set(“Message/ast_msg_queue”, “custom=PJSIP”) in new stack
– Executing [s@macro-dialout-trunk:17] ExecIf(“Message/ast_msg_queue”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))”) in new stack
– Executing [s@macro-dialout-trunk:18] ExecIf(“Message/ast_msg_queue”, “0?Set(DIAL_TRUNK_OPTIONS=M(confirm))”) in new stack
– Executing [s@macro-dialout-trunk:19] ExecIf(“Message/ast_msg_queue”, “0?AGI(allowlist-autoadd.agi,)”) in new stack
– Executing [s@macro-dialout-trunk:20] Macro(“Message/ast_msg_queue”, “dialout-trunk-predial-hook,”) in new stack
– Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“Message/ast_msg_queue”, “”) in new stack
– Executing [s@macro-dialout-trunk:21] GotoIf(“Message/ast_msg_queue”, “0?bypass,1”) in new stack
– Executing [s@macro-dialout-trunk:22] ExecIf(“Message/ast_msg_queue”, “0?Set(CONNECTEDLINE(num,i)=12039874400)”) in new stack
– Executing [s@macro-dialout-trunk:23] ExecIf(“Message/ast_msg_queue”, “0?Set(CONNECTEDLINE(name,i)=CID:)”) in new stack
– Executing [s@macro-dialout-trunk:24] ExecIf(“Message/ast_msg_queue”, “0?Set(CONNECTEDLINE(name,i)=CID:(Hidden))”) in new stack
– Executing [s@macro-dialout-trunk:25] GotoIf(“Message/ast_msg_queue”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:26] ExecIf(“Message/ast_msg_queue”, “0?Set(DIAL_TRUNK_OPTIONS=)”) in new stack
– Executing [s@macro-dialout-trunk:27] Set(“Message/ast_msg_queue”, “HASH(__SIPHEADERS,Alert-Info)=unset”) in new stack
– Executing [s@macro-dialout-trunk:28] Dial(“Message/ast_msg_queue”, “PJSIP/12039874400@voipInn-1-Qr7hQAtoYtYpfL8__B1E,300,b(func-apply-sipheaders^s^1,(1))”) in new stack
– PJSIP/voipInn-1-Qr7hQAtoYtYpfL8__B1E-00000000 Internal Gosub(func-apply-sipheaders,s,1(1)) start
– Executing [s@func-apply-sipheaders:1] NoOp(“PJSIP/voipInn-1-Qr7hQAtoYtYpfL8__B1E-00000000”, “Applying SIP Headers to channel PJSIP/voipInn-1-Qr7hQAtoYtYpfL8__B1E-00000000”) in new stack
– Executing [s@func-apply-sipheaders:2] Set(“PJSIP/voipInn-1-Qr7hQAtoYtYpfL8__B1E-00000000”, “TECH=PJSIP”) in new stack
– Executing [s@func-apply-sipheaders:3] Set(“PJSIP/voipInn-1-Qr7hQAtoYtYpfL8__B1E-00000000”, “SIPHEADERKEYS=Alert-Info”) in new stack
– Executing [s@func-apply-sipheaders:4] While(“PJSIP/voipInn-1-Qr7hQAtoYtYpfL8__B1E-00000000”, “1”) in new stack
– Executing [s@func-apply-sipheaders:5] Set(“PJSIP/voipInn-1-Qr7hQAtoYtYpfL8__B1E-00000000”, “sipheader=unset”) in new stack
– Executing [s@func-apply-sipheaders:6] ExecIf(“PJSIP/voipInn-1-Qr7hQAtoYtYpfL8__B1E-00000000”, “0?SIPRemoveHeader(Alert-Info:)”) in new stack
– Executing [s@func-apply-sipheaders:7] ExecIf(“PJSIP/voipInn-1-Qr7hQAtoYtYpfL8__B1E-00000000”, “1?Set(PJSIP_HEADER(remove,Alert-Info)=)”) in new stack
[2024-03-15 12:09:00] ERROR[22281]: res_pjsip_header_funcs.c:723 remove_header: No headers had been previously added to this session.
– Executing [s@func-apply-sipheaders:8] ExecIf(“PJSIP/voipInn-1-Qr7hQAtoYtYpfL8__B1E-00000000”, “0?Set(sipheader=http://127.0.0.1;info=unset)”) in new stack
– Executing [s@func-apply-sipheaders:9] ExecIf(“PJSIP/voipInn-1-Qr7hQAtoYtYpfL8__B1E-00000000”, “0?Set(sipheader=http://127.0.0.1unset)”) in new stack
– Executing [s@func-apply-sipheaders:10] ExecIf(“PJSIP/voipInn-1-Qr7hQAtoYtYpfL8__B1E-00000000”, “0?SIPAddHeader(Alert-Info:unset)”) in new stack
– Executing [s@func-apply-sipheaders:11] ExecIf(“PJSIP/voipInn-1-Qr7hQAtoYtYpfL8__B1E-00000000”, “0?Set(PJSIP_HEADER(add,Alert-Info)=unset)”) in new stack
– Executing [s@func-apply-sipheaders:12] EndWhile(“PJSIP/voipInn-1-Qr7hQAtoYtYpfL8__B1E-00000000”, “”) in new stack
– Executing [s@func-apply-sipheaders:4] While(“PJSIP/voipInn-1-Qr7hQAtoYtYpfL8__B1E-00000000”, “0”) in new stack
– Executing [s@func-apply-sipheaders:13] Return(“PJSIP/voipInn-1-Qr7hQAtoYtYpfL8__B1E-00000000”, “”) in new stack
== Spawn extension (from-pstn, 12039874400, 1) exited non-zero on ‘PJSIP/voipInn-1-Qr7hQAtoYtYpfL8__B1E-00000000’
– PJSIP/voipInn-1-Qr7hQAtoYtYpfL8__B1E-00000000 Internal Gosub(func-apply-sipheaders,s,1(1)) complete GOSUB_RETVAL=
– Called PJSIP/12039874400@voipInn-1-Qr7hQAtoYtYpfL8__B1E
[2024-03-15 12:09:00] ERROR[22212][C-00000001]: translate.c:1402 ast_translator_best_choice: Cannot determine best translation path since one capability supports no formats
== Spawn extension (macro-dialout-trunk, s, 28) exited non-zero on ‘Message/ast_msg_queue’ in macro ‘dialout-trunk’
== Spawn extension (from-internal, 12039874400, 11) exited non-zero on 'Message/ast_msg_queue

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Admin GUI times out after about 30 minutes

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Hello all, I’m new to FreePBX, so please bear with me!

I have a new install of FreePBX on Oracle VM VirtualBox. When logged into the admin portal, it’ll eventually timeout after about 30 minutes and won’t work again until I reboot the virtual machine. I’ve read that this might have to do with the firewall.

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Problem with appsthey are entering

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Why I get this error when making an incoming call and how to resolve it

2 posts - 2 participants

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Testing multiple DID Numbers

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I have about large amount of DID’s that terminate to our system. I need to test these numbers to make sure that they all work properly. Without calling every DID by hand and get the results.please help me out

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How to record a Call originated by AMI that goes to a context

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Friends, good morning!
I have a client who is a medical clinic, and they needed an automatic confirmation solution (where the PBX pulls the doctor’s schedule, saves it in a database, calls the patients, informs them of the appointment via Azure Voice AI and gives options to confirm, cancel, reschedule, among others).

We managed to achieve this goal with the help of some Python scripts and the creation of some ODBC functions, the scheme is as follows:

Scripts location: /opt/medical

agenda.py (script that pulls the agenda from their system endpoint and saves it in a database, runs every day at a specific time getting the next day’s agenda)

dialer.py - calls patients (originating a call via AMI), and when they answer, forwards to the ivr-medical context

confirmation.py - script that runs in a loop, whenever a new entry is detected in the confirmations table (when a client confirms a query via ivr), sends this to their system endpoint

see the context here: Context - AMI AI Calls - Pastebin.com
and the python script who initiate the calls: Context AI Calls - Pastebin.com

In general, this scheme has been working well, and there are a few more scripts (like the one that calls the Microsoft Azure API and generates the audio that the context plays) but one difficulty that I haven’t been able to solve yet is getting these calls to be recorded correctly .

I even managed to get the audio to be generated through context, but it remains silent and only lasts for a second. Is it something I got wrong in the context? or in the script? Is there an easier way to do what these scripts are doing? If someone can help me with this I would be very happy.

ps: some things were translated from pt-br (original language where everything was written, like the comments) to English, so if something was confusing I can clarify… also, if anyone is interested in some of the scripts (start call, pull agenda to save in the database, script that generates audio with azure) I can send or publish on pastebin.

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Asterisk errors

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I’ve just noticed that my FreePBX system is giving quite a few Asterisk errors. Here are a few examples:

[2024-03-18 18:01:02] ERROR[9544][C-000000e0]: 
res_pjsip_header_funcs.c:622 func_read_header: This function 
requires a PJSIP channel.

[2024-03-19 09:13:41] ERROR[24021]: res_sorcery_config.c:328 
sorcery_config_internal_load: Unable to load config file 'pjproject.conf'

[2024-03-19 09:13:41] ERROR[20722]: res_pjsip_config_wizard.c:1080 
object_type_loaded_observer: Unable to load config file 'pjsip_wizard.conf'

[2024-03-19 09:13:41] ERROR[24021]: config_options.c:710 
aco_process_config: Unable to load config file 'resolver_unbound.conf'

[2024-03-19 09:13:42] ERROR[24021]: res_sorcery_config.c:328 
sorcery_config_internal_load: Unable to load config file 'stir_shaken.conf'

I have been unable to find a list of these codes to give me a better idea of how to fix the errors. Could someone give me a steer in the right direction, please?

Bill.

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E1 link down

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When I try to make an outbound call, an error occurs on the Sangoma VEGA,
Fail to send SETUP as E1 link is down but inbound is working fine.

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Campon alternative

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Hi everyone,
I know Call Completion Supplemental Services (CCSS) has been deprecated. I was wondering if there is any alternative to this. Is there a commercial module for this feature?
If neither options are available, any possibility to getting this module made?

Thanks in advance.

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Почему у freepbx не проходят локальные звонки?

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День добрый. подскажите куда смотреть, поднял freepbx (16.0.40.7), все в дефолте. Добавил несколько локальных номеров. Аппараты успешно зарегистрировались на сервере. Но звонки не проходят. В логах:

[2024-03-21 18:45:33] WARNING[280794] res_pjsip_pubsub.c: No registered publish handler for event presence from 111
[2024-03-21 18:42:11] WARNING[281887][C-0000000a] pbx.c: No application ‘Macro’ for extension (from-internal, 101, 3)
[2024-03-21 18:42:11] WARNING[281887][C-0000000a] pbx.c: No application ‘Macro’ for extension (from-internal, h, 1)
телефоны в локальной сети, nat не используется.

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Transfer a call back to the PBX from a mobile phone

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Is it possible at all, very likely using DTMF signals, to transfer a call back to the PBX from a mobile phone?
We have an extension setup to call out if the internal phones are not answered.
However, it could be handy to transfer the call back to an internal extension when it has been answered on the mobile device (over GSM).
Is there a way to achieve this?

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Best way to play audio files to callers?

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Hello all!

I am relatively new to FreePBX and am wondering if there is a good way to play audio files to callers? I realize that you can manually upload audio files one by one, but I am looking for some sort of solution that will play audio files out of a folder on the server and have playback controls like pause, fast forward etc. Is there a module that adds this functionality?

I would like to setup a call-in number where folks could call in and listen to audio files. Can somebody point me in the right direction?

Thanks in advance!

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Inbound calls only work after placing an outbound call first

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One of our SIP providers is a low traffic backup that’s rarely used. We got reports from users that inbound calls to DIDs on that alternate ITSP were failing. I did a few test calls to those DIDs and they all returned fast-busy and nothing showing up in our logs. I then placed an outbound call on that same ITSP trunk and that went through fine. Afterwards calls to those same DIDs were now passing normally.
Before I reach out to the provider, could there be something I’m missing on my end?
The only two settings I have setup under Connectivity > Trunks > “ITSP” > sip Settings > Outgoing> PEER Details is
type=peer
host=x.x.x.x

Incoming is empty
There is no firewall between our system and the ITSPs router.

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Fwconsole reload error message

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I get this error message on fwconsole reload. Tried reinstall the respapps , yet have same error message. Any fix for that? Apply config from webgui works.

In Restapps.class.php line 1429:

[Whoops\Exception\ErrorException (2)]
Invalid argument supplied for foreach()

Exception trace:
() at /var/www/html/admin/modules/restapps/Restapps.class.php:1429
Whoops\Run->handleError() at /var/www/html/admin/modules/restapps/Restapps.class.php:1429
FreePBX\modules\Restapps->parseQueues() at /var/www/html/admin/modules/restapps/Restapps.class.php:1704
FreePBX\modules\Restapps->doDialplanHook() at /var/www/html/admin/libraries/BMO/DialplanHooks.class.php:107
FreePBX\DialplanHooks->processHooks() at /var/www/html/admin/libraries/Console/Reload.class.php:318
FreePBX\Console\Command\Reload->reload() at /var/www/html/admin/libraries/Console/Reload.class.php:91
FreePBX\Console\Command\Reload->execute() at /var/www/html/admin/libraries/Composer/vendor/symfony/console/Command/Command.php:255
Symfony\Component\Console\Command\Command->run() at /var/www/html/admin/libraries/Composer/vendor/symfony/console/Application.php:960
Symfony\Component\Console\Application->doRunCommand() at /var/www/html/admin/libraries/Composer/vendor/symfony/console/Application.php:255
Symfony\Component\Console\Application->doRun() at /var/www/html/admin/libraries/Composer/vendor/symfony/console/Application.php:148
Symfony\Component\Console\Application->run() at /var/lib/asterisk/bin/fwconsole:163

reload [–json] [–dry-run] [–skip-registry-checks] [–dont-reload-asterisk]

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Medical call - to two participants number with announcements and transfer

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Hello community, we need a medical call solution for our system. > see flow_chart.pdf

All works fine, but we cannot play announcement 2 if answer extern number 4000 (or 471 for testing)

We tested with various methods, it is certainly not difficult - but we are standing on the hose. :upside_down_face:

I hope it is understandable
Thank YOu
MfG Steffen









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When uploading the CSV file for the dialplan the "." is not imported!

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When I import the dialplan CVS with the “.” wildcard set after the codes, freepbx does not import the “.”. Does anyone know how to fix this please?

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Internal trunk to trunk call failure

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Hello,

I am having very easy to replicate trunk to trunk failures between two internal trunks to our FreePBX box. I have spent hours trying to find why one SIP trunk to SIP trunk call is working and why the other SIP trunk to SIP call is failing. I did manage to find a difference in the packet capture that I do not understand. I have obfuscated the names/numbers for privacy. Please see below:

Call #1, Rejected

“8.528233”,“8.714564”,“100.101.58.3”,““Orig Cust"sip:4062573333@10.100.0.12:5060”,“sip:4067523333@10.100.0.12:5060”,“SIP”,“00:00:00”,“4”,“CALL SETUP”,“INVITE”
“8.696185”,“8.713447”,“10.100.0.12”,”“Term Cust #1sip:14067523333@10.100.0.12",“sip:4067523333%404067523333@100.103.255.3”,“SIP”,“00:00:00”,“3”,“REJECTED”,“INVITE 404”

Call #2, Successful:

“7.597821”,“13.162922”,“100.101.58.3”,““Orig Cust"sip:4062573333@10.100.0.12:5060”,“sip:4067553333@10.100.0.12:5060”,“SIP”,“00:00:05”,“6”,“IN CALL”,“INVITE 200”
“7.750494”,“13.089710”,“10.100.0.12”,”“Term Cust #2sip:14067553333@10.100.0.12",“sip:4067553333@192.168.105.242”,“SIP”,“00:00:05”,“8”,“IN CALL”,“INVITE 401 200”

Notice in the first failing call the string from FreePBX to Term Cust #1 (10.100.0.12 to 100103.255.3) has a much longer number string and includes a % character than the successful call #2.

I believe the rejection of the first call is coming from the term PBX not liking the extra garbage in the number spill.

Does anyone have thoughts as to what might cause the number spills to be so different as they should both be sending 10 digits?

Thanks for your time!

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