SIP Station and VoIP innovations are both supposed to support SMS.
But do they both support MMS?
Is it supported in Zulu and SangomaConnect for both trunking providers?
Mobile as well as desktop?
I don’t want to be restricted to UCP.
1 post - 1 participant
I have setup a freePBX system with trial SIPSTATION. I am able to call in, but the caller never hears any audio. I can see a configured phone the incoming call, and can see the call in the logs. The caller just never gets any response. Also, when calling out the system response is “all circuits are busy”. Fairly sure these items are linked, but I am not sure where to start looking.
TIA.
3 posts - 3 participants
Installed FreePBX 16 with SIPSTATION. I have confirmed that one of my extensions can send and received SMS messages through the UCP, however I am looking to automate the SMS process.
Does anyone know if asterisk, or FreePBX has a command that will do the following from either the asterisk CLI or a FreePBX program / script?
Thank you in advance,
3 posts - 3 participants
We have a landline at our business that we would like to use as a failover number to our sipstation trunking. We have a physical pbxacct device here at our facility.
We understand that calls will be forwarded to this number should our internet connectivity or hardware would fail, perhaps because of a power failure or internet problem at our end.
I’ve read about ddos attacks on voip services. In that case our calls would typically fail, because it would be the trunking company whose servers are down, and there would be no forwarding. Is this correct?
Further to this, how is sipstation fairing with with these attacks? Have they been hit? Is there any way to mitigate or immunize a service like this? Is there anything I should be doing at our level to mitigate the risk of sipstation having a problem that degrades service for more than a short while?
Thanks for your input
1 post - 1 participant
Just started happening for no reason:
PBX Version:14.0.16.11
PBX Distro: 12.7.8-2107-3.sng7
Asterisk Version: 13.38.3
PHP Version 5.6.40
Is this something I can fix in php.ini?
1 post - 1 participant
/var/log/asterisk/full output below:
[2022-02-14 05:50:27] VERBOSE[17611] netsock2.c: Using SIP RTP Audio TOS bits 184
[2022-02-14 05:50:27] VERBOSE[17611] netsock2.c: Using SIP RTP Audio TOS bits 184 in TCLASS field.
[2022-02-14 05:50:27] VERBOSE[17611] netsock2.c: Using SIP RTP Audio CoS mark 5
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [5194662486@from-internal:1] Macro(“PJSIP/121-00000000”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
[2022-02-14 05:50:27] WARNING[26537][C-00000001] app_macro.c: Macro() is deprecated and will be removed from a future version of Asterisk.
[2022-02-14 05:50:27] WARNING[26537][C-00000001] app_macro.c: Dialplan should be updated to use Gosub instead.
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:1] Set(“PJSIP/121-00000000”, “TOUCH_MONITOR=1644835827.0”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:2] Set(“PJSIP/121-00000000”, “CHANCONTEXT=”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:3] Set(“PJSIP/121-00000000”, “CHANCONTEXT=”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:4] Set(“PJSIP/121-00000000”, “CHANEXTENCONTEXT=121-00000000”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:5] Set(“PJSIP/121-00000000”, “CHANEXTEN=121-00000000”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:6] Set(“PJSIP/121-00000000”, “CALLERID(number)=121”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:7] Set(“PJSIP/121-00000000”, “AMPUSER=121”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:8] Set(“PJSIP/121-00000000”, “HOTDESCKCHAN=121-00000000”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:9] Set(“PJSIP/121-00000000”, “HOTDESKEXTEN=121”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:10] Set(“PJSIP/121-00000000”, “HOTDESKCALL=0”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:11] ExecIf(“PJSIP/121-00000000”, “0?Set(HOTDESKCALL=1)”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:12] ExecIf(“PJSIP/121-00000000”, “0?Set(CALLERID(name)=)”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:13] GotoIf(“PJSIP/121-00000000”, “0?report”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:14] ExecIf(“PJSIP/121-00000000”, “1?Set(REALCALLERIDNUM=121)”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:15] Set(“PJSIP/121-00000000”, “AMPUSER=121”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:16] GotoIf(“PJSIP/121-00000000”, “0?limit”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:17] Set(“PJSIP/121-00000000”, “AMPUSERCIDNAME=Dave Bowles”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:18] ExecIf(“PJSIP/121-00000000”, “0?Set(__CIDMASQUERADING=TRUE)”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:19] GotoIf(“PJSIP/121-00000000”, “0?report”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:20] Set(“PJSIP/121-00000000”, “AMPUSERCID=121”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:21] Set(“PJSIP/121-00000000”, “__DIAL_OPTIONS=HhTtr”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:22] Set(“PJSIP/121-00000000”, “CALLERID(all)=“Dave Bowles” <121>”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:23] ExecIf(“PJSIP/121-00000000”, “0?Set(CUSDIAL=)”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:24] ExecIf(“PJSIP/121-00000000”, “0?Set(CALLERID(all)=“Dave Bowles” <121>)”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:25] GotoIf(“PJSIP/121-00000000”, “0?limit”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:26] ExecIf(“PJSIP/121-00000000”, “1?Set(GROUP(concurrency_limit)=121)”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:27] ExecIf(“PJSIP/121-00000000”, “0?Set(CHANNEL(language)=)”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:28] NoOp(“PJSIP/121-00000000”, “Macro Depth is 1”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:29] GotoIf(“PJSIP/121-00000000”, “1?report2:macroerror”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx_builtins.c: Goto (macro-user-callerid,s,30)
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:30] GotoIf(“PJSIP/121-00000000”, “1?continue”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx_builtins.c: Goto (macro-user-callerid,s,49)
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:49] Set(“PJSIP/121-00000000”, “CALLERID(number)=121”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:50] Set(“PJSIP/121-00000000”, “CALLERID(name)=Dave Bowles”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:51] GotoIf(“PJSIP/121-00000000”, “0?cnum”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:52] Set(“PJSIP/121-00000000”, “CDR(cnam)=Dave Bowles”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:53] Set(“PJSIP/121-00000000”, “CDR(cnum)=121”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-user-callerid:54] Set(“PJSIP/121-00000000”, “CHANNEL(language)=en”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [5194662486@from-internal:2] Gosub(“PJSIP/121-00000000”, “sub-record-check,s,1(out,5194662486,dontcare)”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@sub-record-check:1] GotoIf(“PJSIP/121-00000000”, “0?initialized”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@sub-record-check:2] Set(“PJSIP/121-00000000”, “__REC_STATUS=INITIALIZED”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@sub-record-check:3] Set(“PJSIP/121-00000000”, “NOW=1644835827”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@sub-record-check:4] Set(“PJSIP/121-00000000”, “__DAY=14”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@sub-record-check:5] Set(“PJSIP/121-00000000”, “__MONTH=02”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@sub-record-check:6] Set(“PJSIP/121-00000000”, “__YEAR=2022”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@sub-record-check:7] Set(“PJSIP/121-00000000”, “__TIMESTR=20220214-055027”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@sub-record-check:8] Set(“PJSIP/121-00000000”, “__FROMEXTEN=121”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@sub-record-check:9] Set(“PJSIP/121-00000000”, “__MON_FMT=wav”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@sub-record-check:10] NoOp(“PJSIP/121-00000000”, “Recordings initialized”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@sub-record-check:11] ExecIf(“PJSIP/121-00000000”, “0?Set(ARG3=dontcare)”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@sub-record-check:12] Set(“PJSIP/121-00000000”, “REC_POLICY_MODE_SAVE=”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@sub-record-check:13] ExecIf(“PJSIP/121-00000000”, “0?Set(REC_STATUS=NO)”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@sub-record-check:14] GotoIf(“PJSIP/121-00000000”, “3?checkaction”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx_builtins.c: Goto (sub-record-check,s,17)
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@sub-record-check:17] GotoIf(“PJSIP/121-00000000”, “1?sub-record-check,out,1”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx_builtins.c: Goto (sub-record-check,out,1)
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [out@sub-record-check:1] NoOp(“PJSIP/121-00000000”, “Outbound Recording Check from 121 to 5194662486”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [out@sub-record-check:2] Set(“PJSIP/121-00000000”, “RECMODE=force”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [out@sub-record-check:3] ExecIf(“PJSIP/121-00000000”, “0?Goto(routewins)”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [out@sub-record-check:4] ExecIf(“PJSIP/121-00000000”, “0?Goto(routewins)”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [out@sub-record-check:5] Gosub(“PJSIP/121-00000000”, “recordcheck,1(force,out,5194662486)”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [recordcheck@sub-record-check:1] NoOp(“PJSIP/121-00000000”, “Starting recording check against force”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [recordcheck@sub-record-check:2] Goto(“PJSIP/121-00000000”, “force”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx_builtins.c: Goto (sub-record-check,recordcheck,5)
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [recordcheck@sub-record-check:5] Set(“PJSIP/121-00000000”, “__REC_POLICY_MODE=FORCE”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [recordcheck@sub-record-check:6] GotoIf(“PJSIP/121-00000000”, “1?startrec”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx_builtins.c: Goto (sub-record-check,recordcheck,16)
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [recordcheck@sub-record-check:16] NoOp(“PJSIP/121-00000000”, “Starting recording: out, 5194662486”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [recordcheck@sub-record-check:17] Set(“PJSIP/121-00000000”, “__CALLFILENAME=out-5194662486-121-20220214-055027-1644835827.0”) in new stack
[2022-02-14 05:50:27] WARNING[26537][C-00000001] func_strings.c: EVAL requires an argument: EVAL()
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [recordcheck@sub-record-check:18] MixMonitor(“PJSIP/121-00000000”, “2022/02/14/out-5194662486-121-20220214-055027-1644835827.0.wav,abi(LOCAL_MIXMON_ID),”) in new stack
[2022-02-14 05:50:27] VERBOSE[26539][C-00000001] app_mixmonitor.c: Begin MixMonitor Recording PJSIP/121-00000000
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [recordcheck@sub-record-check:19] Set(“PJSIP/121-00000000”, “__MIXMON_ID=0x7f38d80315d0”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [recordcheck@sub-record-check:20] Set(“PJSIP/121-00000000”, “__RECORD_ID=PJSIP/121-00000000”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [recordcheck@sub-record-check:21] Set(“PJSIP/121-00000000”, “__REC_STATUS=RECORDING”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [recordcheck@sub-record-check:22] Set(“PJSIP/121-00000000”, “CDR(recordingfile)=out-5194662486-121-20220214-055027-1644835827.0.wav”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [recordcheck@sub-record-check:23] Return(“PJSIP/121-00000000”, “”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [out@sub-record-check:6] Return(“PJSIP/121-00000000”, “”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [5194662486@from-internal:3] ExecIf(“PJSIP/121-00000000”, “0 ?Set(CDR(accountcode)=)”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [5194662486@from-internal:4] Set(“PJSIP/121-00000000”, “_ROUTEID=3”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [5194662486@from-internal:5] Set(“PJSIP/121-00000000”, “_ROUTENAME=SIPStation-Out”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [5194662486@from-internal:6] Set(“PJSIP/121-00000000”, “MOHCLASS=default”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [5194662486@from-internal:7] Set(“PJSIP/121-00000000”, “_CALLERIDNAMEINTERNAL=Dave Bowles”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [5194662486@from-internal:8] Set(“PJSIP/121-00000000”, “_CALLERIDNUMINTERNAL=121”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [5194662486@from-internal:9] Set(“PJSIP/121-00000000”, “_EMAILNOTIFICATION=FALSE”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [5194662486@from-internal:10] Set(“PJSIP/121-00000000”, “_NODEST=”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [5194662486@from-internal:11] Macro(“PJSIP/121-00000000”, “outisbusy,”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-outisbusy:1] Progress(“PJSIP/121-00000000”, “”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-outisbusy:2] GotoIf(“PJSIP/121-00000000”, “0?emergency,1”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-outisbusy:3] GotoIf(“PJSIP/121-00000000”, “0?intracompany,1”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-outisbusy:4] Playback(“PJSIP/121-00000000”, “all-circuits-busy-now&please-try-call-later, noanswer”) in new stack
[2022-02-14 05:50:27] VERBOSE[26537][C-00000001] file.c: <PJSIP/121-00000000> Playing ‘all-circuits-busy-now.ulaw’ (language ‘en’)
[2022-02-14 05:50:28] VERBOSE[26537][C-00000001] file.c: <PJSIP/121-00000000> Playing ‘please-try-call-later.ulaw’ (language ‘en’)
[2022-02-14 05:50:29] VERBOSE[26537][C-00000001] pbx.c: Executing [h@from-internal:1] Macro(“PJSIP/121-00000000”, “hangupcall”) in new stack
[2022-02-14 05:50:29] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/121-00000000”, “1?theend”) in new stack
[2022-02-14 05:50:29] VERBOSE[26537][C-00000001] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2022-02-14 05:50:29] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/121-00000000”, “0?Set(CDR(recordingfile)=)”) in new stack
[2022-02-14 05:50:29] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-hangupcall:4] NoOp(“PJSIP/121-00000000”, " montior file= /var/spool/asterisk/monitor/2022/02/14/out-5194662486-121-20220214-055027-1644835827.0.wav") in new stack
[2022-02-14 05:50:29] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-hangupcall:5] GotoIf(“PJSIP/121-00000000”, “1?skipagi”) in new stack
[2022-02-14 05:50:29] VERBOSE[26537][C-00000001] pbx_builtins.c: Goto (macro-hangupcall,s,7)
[2022-02-14 05:50:29] VERBOSE[26537][C-00000001] pbx.c: Executing [s@macro-hangupcall:7] Hangup(“PJSIP/121-00000000”, “”) in new stack
[2022-02-14 05:50:29] VERBOSE[26537][C-00000001] app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘PJSIP/121-00000000’ in macro ‘hangupcall’
[2022-02-14 05:50:29] VERBOSE[26537][C-00000001] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/121-00000000’
[2022-02-14 05:50:29] VERBOSE[26539][C-00000001] app_mixmonitor.c: MixMonitor close filestream (mixed)
[2022-02-14 05:50:29] VERBOSE[26539][C-00000001] app_mixmonitor.c: End MixMonitor Recording PJSIP/121-00000000
16 posts - 6 participants
I have SMS/MMS setup with the Sangoma Connect Module and using Voip Innovations trunks…and it is working. Does anyone know if there is a way to send a broadcast or have multiple recipients for an SMS or MMS message? I’m getting an invalid number error in UCP when I try and send with multiple numbers.
Thanks!
3 posts - 3 participants
I have some calls coming in showing the [V] on the caller Id name but it is not showing the actual name of the person calling in, but only [V] and the calling number as the name. I know this is cause of the Verified Caller id that is starting to show up. my incoming carrier is SIPStation, I wasn’t sure if I needed to start with them as I have always been told Caller Id lookup originates with the receiving carrier, or is it a config issue on the pbx setup end. I know that Caller Id is working as the calls that does not have the [V] is showing names like it always did.
3 posts - 3 participants
I have installed the Voip Innovations Module and the Registration Status are all green. But when I go to the ROUTING tab it will NOT allow me to enable the Primary Gateway. Both Routes I have are greyed out. I’m also unable to make outbound calls but I receive inbound calls. Not sure if this have anything to do with the Primary gateway not being enabled. Please advise
5 posts - 3 participants
I’m new to freepbx user. Our company setup a FreePBX 15.0.23 box with a SIPStation free trial. The registration with the server and ip phone (Grandstream GPX1628) were successful. Calling to the the DID number goes to SIPStation DID validation message. It seems OK. But when I changed the Inbound route destination to the extension, I was always directed to voice messages. Is that normal?
I couldn’t found any doxumentation what sip station free trial can do. Can anybody give me some hints?
3 posts - 2 participants
I posted this in asterisk’s forums but was told this was something I should post here
I’ve been testing some POC code and system architecture I’ve designed to make responding to SMS received from SIPSTATION gateway easier and more mobile than having to log into UCP to do so.
I’ve already written automation to scrape the asterisk sms tables in the database (in the future an event-driven model would be better) and email to me every few seconds.
The easiest way I’ve found to respond would be to reply to said email. I operate my own email relay and freepbx server. The emails sent from scraping are sent as an authenticated user on my relay with proper dkim/spf, etc. I have the relay set up to redirect all emails sent to that specific user (in this case all replies to said scrape emails) to the asterisk system, which runs a basic smtp routine i wrote to process the email and handoff to some sort of Asterisk AMI routine (written in php). The goal is to find a way to query Asterisk AMI (or some other method) to send an SMS through the SIPSTATION gateway i use.
the following snippet works, so I’m able to connect to AMI and issue commands
$asm = new AGI_AsteriskManager();
if($asm->connect('127.0.0.1', 'username', 'password')) {
$peers = $asm->command("sip show peers");
print_r($peers);
}
However, I can’t seem to call the MessageSend method without failure, provided this is what I will need to call.
$asm = new AGI_AsteriskManager();
if($asm->connect('127.0.0.1', 'username', 'password')) {
$result = $asm->MessageSend('sip:19999999999', '"Caller ID Name" <19999999999>', 'hello');
print_r($result);
}
Array
(
[Response] => Error
[Message] => Message failed to send.
)
asterisk cli debug logs show:
[2022-04-23 23:30:43] WARNING[31990]: chan_sip.c:6331 create_addr: Purely numeric hostname (19999999999), and not a peer–rejecting!
Is it possible for me to send an SMS message out to a 10 digit cell number like this?
2 posts - 2 participants
Hello,
I’m setting up a new install.
Using FreePBX 16.0.21.8
And I’m in the trial of SIP STATION (first time I use it).
I went though the setup and everything is fine except that when I receive a call, the other end doesn’t hear my voice.
This is only occurring on inbound calls.
Outbound calls are fine!
My firewall is set to forward the following ports (UDP) to FreePBX : 5060, 5061, 5160, 5161 & 10000 to 20000.
Any idea what could be causing the outgoing voice to not be sent in inbound calls?
Thank you!
9 posts - 4 participants
I am trying to get a SIPSTATION Trial, but when I go to step 2 I get a red stripe at the top.
It keeps loading and doesn’t give me the next step.
1 post - 1 participant
Is SipStation signing their calls? Or are there steps customers of SipStation need to take in order for their calls to be signed?
14 posts - 4 participants
I have Sipstation as my provider and freepbx setup. I am able to send sms from the UCP but I cannot receive. Has anyone ever had this issue?
2 posts - 2 participants
Hello Guys ,
I am trying to get a SIP trunk register on Sangoma FreePBX , in Advanced Setting , under SIP channel driver, i disabled PJSIP and left only SIP channel enable.
Having Elastix with SIP trunk configuration , same parameter and it works well, when issue with the show sip registry cmd , this is displayed the SIP trunk status Register .
I have copy paste the same parameters in my new FreePBX Sangoma, in Peer details field ,and Register String , so now i am looking with the good syntax to get the status of the SIP trunk configuration . Have you guys have an idea ?
pjsip show registrations is the default cmd under the Asterisk CLI but for my case i ve disabled the PJSIP channel , so i used SIP channel instead .
Looking forward to your reply.
Thanks !
3 posts - 3 participants
So just seeing if anyone has run into this…
We put new Resident Call Boxes (Mircom) in a high rise client. These boxes take standard copper phone line only when NOT using their in house VOIP solution… They just directed us to use an FXS if using our own VOIP line to transition to copper to connect to the MIrcom call boxes…
We have a fully licensed FreePBX at this site. We just setup 2 new extensions for a CISCO SPA112 and plugged the 2 call boxes into the 2 FXS ports on the CISCO… Zero issues, works great…
This deployment uses SipStation trunks… Everything has been working great for about a year since we installed it.
Over the last few months, we have been receiving complaints from this client that the residents in the building that have Comcast Digital Voice landlines in their condo/apartments… and that number is programmed into the call box for these specific residents, the call never rings on their house phones and the person at the call box gets the Comcast voicemail… Even though these residents have had techs out, confirmed forwarding and comcast vm is turned off.
If a cell phone or different VOIP service is called like another Sipstation or Nextiva, Verizon PSTN, etc then the call rings through normally and person answers, presses 9 and door unlocks…
Has anyone see this issue ??? Where calls from FreePBX/SipStation to Comcast phone numbers go straight to VM? We did submit a proper LIDB request to SipStation when we setup this line and they applied it correctly. The correct client name comes up on my office CallerID when they call me…
Thoughts/suggestions? Anyone seen this? Doing some googling, it seems other comcast customers are experiencing this issue as well. Just seeing if any of you guys have seen or heard of this…
1 post - 1 participant
Hey All,
I am having trouble getting any SMS sent. I get an error in the Sangoma Talk app that says “The message could not be sent. Please try again.” and in the sangomaconnect_err.log I get a line that says
Error in API request (/mobile/sms/send) null Error code400 {“status”:false,“message”:“”}
I can receive SMS just fine.
VoipInnovations says they do not see any outbound SMS traffic on that DID.
My DID shows SMS is enabled with SIP.
Any advice would be appreciated.
1 post - 1 participant
Hey everyone,
I have everything configured on my FreePBX, and I am just about to push this into production. I was using the SIPStation trial - the trial trunks are different from the production trunks.
Once I activated the SIPStation account, the endpoints connected to FreePBX began adding a +1 on all outbound calls. This is problematic because we can’t redial from our call history (says “No dial plan rules matched.”)
I’m not sure why the change after moving from the trial to activated SIPStation account. I guess I need to either remove the +1 or add the +1 to the dial plan rules.
Any suggestions on how to resolve this? Is there a setting that I am missing after having upgraded to the full SIPStation account?
Thanks in advance!!
7 posts - 2 participants
I was having a discussion about HD voice codecs and it just reminded me that Sipstation once had a trial of hd codec support
I understand hd codecs are extremely hit or miss (mostly miss) when connecting to another carrier, but it seems like there must be some that at least supported it.
Was it dropped because of lack of carrier support, customer interest, technical reasons?
Will it ever make a resurgence?
1 post - 1 participant