› VitalPBX Community Support › General Discussion › DID Number not set and problem on inbound route
- This topic has 22 replies, 2 voices, and was last updated 2 years, 5 months ago by
pdifeo.
- Post
-
- August 9, 2018 at 4:14 pm
Hi,
I have a SIP Trunk with address 50 numbers.
The problem is that VitalPBX recognize always the principal number and never the others.
The number are serial starting from xxxx900 to xxxx949.
If I call, for example, xxxx949 never reach inbound route on his number. Always routed to xxxx900.
Debugging SIP you can see (below) that the TO header have xxxxx949 as contatted number.
Can anyone help me ?
Regards
PasqualeSIP/2.0 100 Trying
Via: SIP/2.0/UDP 83.211.227.21:5060;branch=z9hG4bK4005.31fd3927.1;received=83.211.227.21
Via: SIP/2.0/UDP 62.94.8.72:5060;branch=z9hG4bK.iIiIiI.c0a8aa4b.e7aeb8b5.Acc0270.Tra00070
Record-Route: <sip:83.211.227.21;lr;ftag=fb2e98c2.Acc270.B2b129;did=dcf.6cc83b94>
Record-Route: <sip:AcCaPl0270.AcCpNo0629.AcCiSt0129.DiAgOuT00.Outgoing.ModB2B.xCSCF-ApLz00284.PnO00602.IsT00263.iIiIiI.c0a8aa4b@62.94.8.72:5060;transport=UDP;lr>
From: “xxxxxxxxxx” <sip:xxxxxxxxxx@clouditalia769.it;user=phone>;tag=fb2e98c2.Acc270.B2b129
To: <sip:xxxxxx949@83.211.227.21:5060;user=phone>
Call-ID: 3bf11822cbb654d0-Acc270-B2b129@10.15.15.8
CSeq: 1 INVITE
Server: VitalPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:xxxxxx900@x.x.x.x:5060>
Content-Length: 00
- Replies
-
- August 9, 2018 at 4:28 pm
- August 9, 2018 at 5:11 pm
Thanks for fast reply. Below the trace from full log with sip debugging set to on. I make hidden the ip address of the server, the CID and at end the called number with xxxxx20949 and trunk principal number with xxxxx20900
Thank you in advance
Pasquale[2018-08-09 18:59:39] VERBOSE[17592] chan_sip.c:
<— SIP read from UDP:83.211.227.21:5060 —>
INVITE sip:xxxxx20900@192.168.200.131:5060 SIP/2.0
Record-Route: <sip:83.211.227.21;lr;ftag=1dff4b84.Acc269.B2b409;did=94e1.d38c3e24>
Via: SIP/2.0/UDP 83.211.227.21:5060;branch=z9hG4bK08ef.c8e0d7c5.1
Via: SIP/2.0/UDP 62.94.8.72:5060;branch=z9hG4bK.iIiIiI.c0a8aa4a.5e8ec29a.Acc0269.Tra00635
Call-ID: 5b6b3c7cd4e27722-Acc269-B2b409@10.15.15.8
From: “xxxxxxxxxx” <sip:xxxxxxxxxx@clouditalia769.it;user=phone>;tag=1dff4b84.Acc269.B2b409
To: <sip:xxxxx20949@83.211.227.21:5060;user=phone>
CSeq: 1 INVITE
Contact: <sip:xxxxxxxxxx.iIiIiI.0a0f0e09.@62.94.8.72:5060>
Record-Route: <sip:AcCaPl0269.AcCpNo0629.AcCiSt0409.DiAgOuT00.Outgoing.ModB2B.xCSCF-ApLz00286.PnO00602.IsT00483.iIiIiI.c0a8aa4a@62.94.8.72:5060;transport=UDP;lr>
Max-Forwards: 16
Accept: application/sdp, application/isup, application/xml
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, MESSAGE, NOTIFY, REGISTER, INFO, REFER, SUBSCRIBE, PUBLISH, UPDATE
Content-Length: 327
Content-Type: application/sdpv=0
o=- 0 1 IN IP4 62.94.199.42
s=-
c=IN IP4 62.94.199.42
t=0 0
m=audio 56624 RTP/AVP 18 0 3 100 8 101
a=ptime:20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:100 CLEARMODE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,32-35
a=sendrecv
a=silenceSupp:off – – – –
a=sqn:0
a=cdsc: 1 image udptl t38
<————->
[2018-08-09 18:59:39] VERBOSE[17592] chan_sip.c: — (15 headers 16 lines) —
[2018-08-09 18:59:39] VERBOSE[17592] chan_sip.c: Sending to 83.211.227.21:5060 (no NAT)
[2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Sending to 83.211.227.21:5060 (no NAT)
[2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Using INVITE request as basis request – 5b6b3c7cd4e27722-Acc269-B2b409@10.15.15.8
[2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Found peer ‘xxxxx20900’ for ‘xxxxxxxxxx’ from 83.211.227.21:5060
[2018-08-09 18:59:39] VERBOSE[17592][C-00000010] netsock2.c: Using SIP RTP TOS bits 184
[2018-08-09 18:59:39] VERBOSE[17592][C-00000010] netsock2.c: Using SIP RTP CoS mark 5
[2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Found RTP audio format 18
[2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Found RTP audio format 0
[2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Found RTP audio format 3
[2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Found RTP audio format 100
[2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Found RTP audio format 8
[2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Found RTP audio format 101
[2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Found audio description format G729 for ID 18
[2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Found unknown media description format CLEARMODE for ID 100
[2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Found audio description format telephone-event for ID 101
[2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Capabilities: us – (ulaw|alaw|g729), peer – audio=(ulaw|gsm|alaw|g729)/video=(nothing)/text=(nothing), combined – (ulaw|alaw|g729)
[2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Non-codec capabilities (dtmf): us – 0x1 (telephone-event|), peer – 0x1 (telephone-event|), combined – 0x1 (telephone-event|)
[2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Peer audio RTP is at port 62.94.199.42:56624
[2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c: Looking for xxxxx20900 in default-trunk (domain 192.168.200.131)
[2018-08-09 18:59:39] VERBOSE[17592][C-00000010] sip/route.c: sip_route_dump: route/path hop: <sip:83.211.227.21;lr;ftag=1dff4b84.Acc269.B2b409;did=94e1.d38c3e24>
[2018-08-09 18:59:39] VERBOSE[17592][C-00000010] sip/route.c: sip_route_dump: route/path hop: <sip:AcCaPl0269.AcCpNo0629.AcCiSt0409.DiAgOuT00.Outgoing.ModB2B.xCSCF-ApLz00286.PnO00602.IsT00483.iIiIiI.c0a8aa4a@62.94.8.72:5060;transport=UDP;lr>
[2018-08-09 18:59:39] VERBOSE[17592][C-00000010] chan_sip.c:
<— Transmitting (no NAT) to 83.211.227.21:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 83.211.227.21:5060;branch=z9hG4bK08ef.c8e0d7c5.1;received=83.211.227.21
Via: SIP/2.0/UDP 62.94.8.72:5060;branch=z9hG4bK.iIiIiI.c0a8aa4a.5e8ec29a.Acc0269.Tra00635
Record-Route: <sip:83.211.227.21;lr;ftag=1dff4b84.Acc269.B2b409;did=94e1.d38c3e24>
Record-Route: <sip:AcCaPl0269.AcCpNo0629.AcCiSt0409.DiAgOuT00.Outgoing.ModB2B.xCSCF-ApLz00286.PnO00602.IsT00483.iIiIiI.c0a8aa4a@62.94.8.72:5060;transport=UDP;lr>
From: “xxxxxxxxxx” <sip:xxxxxxxxxx@clouditalia769.it;user=phone>;tag=1dff4b84.Acc269.B2b409
To: <sip:xxxxx20949@83.211.227.21:5060;user=phone>
Call-ID: 5b6b3c7cd4e27722-Acc269-B2b409@10.15.15.8
CSeq: 1 INVITE
Server: VitalPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:xxxxx20900@xx.xx.xx.xx:5060>
Content-Length: 0<————>
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [xxxxx20900@default-trunk:1] Gosub(“SIP/xxxxx20900-0000000e”, “sub-check-blacklist,s,1(ebe61b6b5987db2a,xxxxxxxxxx)”) in new stack
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s@sub-check-blacklist:1] NoOp(“SIP/xxxxx20900-0000000e”, “Testing if xxxxxxxxxx is in Black List”) in new stack
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s@sub-check-blacklist:2] GotoIf(“SIP/xxxxx20900-0000000e”, “0?banned”) in new stack
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s@sub-check-blacklist:3] Return(“SIP/xxxxx20900-0000000e”, “”) in new stack
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [xxxxx20900@default-trunk:2] Gosub(“SIP/xxxxx20900-0000000e”, “sub-setup-call-type,s,1(incoming)”) in new stack
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s@sub-setup-call-type:1] NoOp(“SIP/xxxxx20900-0000000e”, “Determinating Call Type”) in new stack
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s@sub-setup-call-type:2] GotoIf(“SIP/xxxxx20900-0000000e”, “0?return”) in new stack
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s@sub-setup-call-type:3] Gosub(“SIP/xxxxx20900-0000000e”, “s-incoming,1()”) in new stack
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s-incoming@sub-setup-call-type:1] NoOp(“SIP/xxxxx20900-0000000e”, “Incoming Call”) in new stack
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s-incoming@sub-setup-call-type:2] Set(“SIP/xxxxx20900-0000000e”, “__CALL_TYPE=2”) in new stack
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s-incoming@sub-setup-call-type:3] Return(“SIP/xxxxx20900-0000000e”, “”) in new stack
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s@sub-setup-call-type:4] Set(“SIP/xxxxx20900-0000000e”, “__CALL_TYPE_CONFIGURED=yes”) in new stack
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s@sub-setup-call-type:5] Set(“SIP/xxxxx20900-0000000e”, “CDR(calltype)=2”) in new stack
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s@sub-setup-call-type:6] Return(“SIP/xxxxx20900-0000000e”, “”) in new stack
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [xxxxx20900@default-trunk:3] Goto(“SIP/xxxxx20900-0000000e”, “incoming-
calls,xxxxx20900,1”) in new stack
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx_builtins.c: Goto (incoming-calls,xxxxx20900,1)
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Channel ‘SIP/xxxxx20900-0000000e’ sent to invalid extension: context,exten,priority=incoming-calls,xxxxx20900,1
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [i@incoming-calls:1] NoCDR(“SIP/xxxxx20900-0000000e”, “”) in new stack
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [i@incoming-calls:2] Goto(“SIP/xxxxx20900-0000000e”, “invalid-dest,s,1”) in new stack
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx_builtins.c: Goto (invalid-dest,s,1)
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s@invalid-dest:1] NoOp(“SIP/xxxxx20900-0000000e”, “Invalid Route to Dial”) in new stack
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] pbx.c: Executing [s@invalid-dest:2] Playback(“SIP/xxxxx20900-0000000e”, “im-sorry&no-route-exists-to-dest&vm-goodbye”) in new stack
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] chan_sip.c: Audio is at 18750
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] chan_sip.c: Adding codec ulaw to SDP
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] chan_sip.c: Adding codec alaw to SDP
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] chan_sip.c: Adding codec g729 to SDP
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] chan_sip.c:
<— Reliably Transmitting (no NAT) to 83.211.227.21:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 83.211.227.21:5060;branch=z9hG4bK08ef.c8e0d7c5.1;received=83.211.227.21
Via: SIP/2.0/UDP 62.94.8.72:5060;branch=z9hG4bK.iIiIiI.c0a8aa4a.5e8ec29a.Acc0269.Tra00635
Record-Route: <sip:83.211.227.21;lr;ftag=1dff4b84.Acc269.B2b409;did=94e1.d38c3e24>
Record-Route: <sip:AcCaPl0269.AcCpNo0629.AcCiSt0409.DiAgOuT00.Outgoing.ModB2B.xCSCF-ApLz00286.PnO00602.IsT00483.iIiIiI.c0a8aa4a@62.94.8.72:5060;transport=UDP;lr>
From: “xxxxxxxxxx” <sip:xxxxxxxxxx@clouditalia769.it;user=phone>;tag=1dff4b84.Acc269.B2b409
To: <sip:xxxxx20949@83.211.227.21:5060;user=phone>;tag=as619b41ad
Call-ID: 5b6b3c7cd4e27722-Acc269-B2b409@10.15.15.8
CSeq: 1 INVITE
Server: VitalPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:xxxxx20900@xx.xx.xx.xx:5060>
Content-Type: application/sdp
Content-Length: 321v=0
o=root 86332321 86332321 IN IP4 xx.xx.xx.xx
s=Asterisk PBX 13.21.0
c=IN IP4 xx.xx.xx.xx
t=0 0
m=audio 18750 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv<————>
[2018-08-09 18:59:39] VERBOSE[17592] chan_sip.c:
<— SIP read from UDP:83.211.227.21:5060 —>
ACK sip:xxxxx20900@192.168.200.131:5060 SIP/2.0
Record-Route: <sip:83.211.227.21;lr;ftag=1dff4b84.Acc269.B2b409>
Via: SIP/2.0/UDP 83.211.227.21:5060;branch=z9hG4bK08ef.c8e0d7c5.3
Via: SIP/2.0/UDP 62.94.8.72:5060;branch=z9hG4bK.iIiIiI.c0a8aa4a.5e8ec29c.Acc0269.Tra00189
Call-ID: 5b6b3c7cd4e27722-Acc269-B2b409@10.15.15.8
From: “xxxxxxxxxx” <sip:xxxxxxxxxx@clouditalia769.it;user=phone>;tag=1dff4b84.Acc269.B2b409
To: <sip:xxxxx20949@83.211.227.21:5060;user=phone>;tag=as619b41ad
CSeq: 1 ACK
Contact: <sip:xxxxxxxxxx.iIiIiI.0a0f0e09.@62.94.8.72:5060>
Max-Forwards: 16
Content-Length: 0
P-hint: rr-enforced<————->
[2018-08-09 18:59:39] VERBOSE[17592] chan_sip.c: — (12 headers 0 lines) —
[2018-08-09 18:59:39] VERBOSE[32030][C-00000010] file.c: <SIP/xxxxx20900-0000000e> Playing ‘im-sorry.ulaw’ (language ‘it’)
[2018-08-09 18:59:41] VERBOSE[32030][C-00000010] file.c: <SIP/xxxxx20900-0000000e> Playing ‘no-route-exists-to-dest.ulaw’ (language ‘it’)
[2018-08-09 18:59:43] VERBOSE[32030][C-00000010] file.c: <SIP/xxxxx20900-0000000e> Playing ‘vm-goodbye.ulaw’ (language ‘it’)
[2018-08-09 18:59:44] VERBOSE[32030][C-00000010] pbx.c: Executing [s@invalid-dest:3] Hangup(“SIP/xxxxx20900-0000000e”, “”) in new stack
[2018-08-09 18:59:44] VERBOSE[32030][C-00000010] pbx.c: Spawn extension (invalid-dest, s, 3) exited non-zero on ‘SIP/xxxxx20900-0000000e’
[2018-08-09 18:59:44] VERBOSE[32030][C-00000010] chan_sip.c: Scheduling destruction of SIP dialog ‘5b6b3c7cd4e27722-Acc269-B2b409@10.15.15.8’ in 6400 ms (Method: ACK)
[2018-08-09 18:59:44] VERBOSE[32030][C-00000010] chan_sip.c: set_destination: Parsing <sip:83.211.227.21;lr;ftag=1dff4b84.Acc269.B2b409;did=94e1.d38c3e24> for address/port to send to
[2018-08-09 18:59:44] VERBOSE[32030][C-00000010] chan_sip.c: set_destination: set destination to 83.211.227.21:5060
[2018-08-09 18:59:44] VERBOSE[32030][C-00000010] chan_sip.c: Reliably Transmitting (no NAT) to 83.211.227.21:5060:
BYE sip:xxxxxxxxxx.iIiIiI.0a0f0e09.@62.94.8.72:5060 SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK590d774f
Route: <sip:83.211.227.21;lr;ftag=1dff4b84.Acc269.B2b409;did=94e1.d38c3e24>,<sip:AcCaPl0269.AcCpNo0629.AcCiSt0409.DiAgOuT00.Outgoing.ModB2B.xCSCF-ApLz00286.PnO00602.IsT00483.iIiIiI.c0a8aa4a@62.94.8.72:5060;transport=UDP;lr>
Max-Forwards: 70
From: <sip:xxxxx20949@83.211.227.21:5060;user=phone>;tag=as619b41ad
To: “xxxxxxxxxx” <sip:xxxxxxxxxx@clouditalia769.it;user=phone>;tag=1dff4b84.Acc269.B2b409
Call-ID: 5b6b3c7cd4e27722-Acc269-B2b409@10.15.15.8
CSeq: 102 BYE
User-Agent: VitalPBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0—
[2018-08-09 18:59:45] VERBOSE[17592] chan_sip.c:
<— SIP read from UDP:83.211.227.21:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.200.131:5060;branch=z9hG4bK590d774f
Call-ID: 5b6b3c7cd4e27722-Acc269-B2b409@10.15.15.8
From: <sip:xxxxx20949@83.211.227.21:5060;user=phone>;tag=as619b41ad
To: “xxxxxxxxxx” <sip:xxxxxxxxxx@clouditalia769.it;user=phone>;tag=1dff4b84.Acc269.B2b409
CSeq: 102 BYE
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, MESSAGE, NOTIFY, REGISTER, INFO, REFER, SUBSCRIBE, PUBLISH, UPDATE
P-Charging-Vector: icid-value=”000dda9f7191c54-002b-0286-0000-0000@10.15.14.9″;icid-generated-at=10.15.11.5;orig-ioi=clouditalia769.it
Content-Length: 0<————->
[2018-08-09 18:59:45] VERBOSE[17592] chan_sip.c: — (9 headers 0 lines) —
[2018-08-09 18:59:45] VERBOSE[17592][C-00000010] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived
[2018-08-09 18:59:45] VERBOSE[17592] chan_sip.c: Really destroying SIP dialog ‘5b6b3c7cd4e27722-Acc269-B2b409@10.15.15.8’ Method: ACK
[2018-08-09 18:59:45] DEBUG[17592] chan_sip.c:
———- SIP HISTORY for ‘5b6b3c7cd4e27722-Acc269-B2b409@10.15.15.8’
[2018-08-09 18:59:45] DEBUG[17592] chan_sip.c: * SIP Call
[2018-08-09 18:59:45] DEBUG[17592] chan_sip.c: 001. Rx INVITE / 1 INVITE / sip:xxxxx20900@192.168.200.131:5060
[2018-08-09 18:59:45] DEBUG[17592] chan_sip.c: 002. NewChan Channel SIP/xxxxx20900-0000000e – from 5b6b3c7cd4e27722-Acc269-
[2018-08-09 18:59:45] DEBUG[17592] chan_sip.c: 003. TxResp SIP/2.0 / 1 INVITE – 100 Trying
[2018-08-09 18:59:45] DEBUG[17592] chan_sip.c: 004. TxRespRel SIP/2.0 / 1 INVITE – 200 OK
[2018-08-09 18:59:45] DEBUG[17592] chan_sip.c: 005. Rx ACK / 1 ACK / sip:xxxxx20900@192.168.200.131:5060
[2018-08-09 18:59:45] DEBUG[17592] chan_sip.c: 006. Hangup Cause Normal Clearing
[2018-08-09 18:59:45] DEBUG[17592] chan_sip.c: 007. RTCPaudio Quality:ssrc=642619996;themssrc=130626313;lp=0;rxjitter=0.00000
[2018-08-09 18:59:45] DEBUG[17592] chan_sip.c: 008. TxRe
qRel BYE / 102 BYE – BYE
[2018-08-09 18:59:45] DEBUG[17592] chan_sip.c: 009. SchedDestroy 6400 ms
[2018-08-09 18:59:45] DEBUG[17592] chan_sip.c: 010. Rx SIP/2.0 / 102 BYE / 200 OK
[2018-08-09 18:59:45] DEBUG[17592] chan_sip.c: 011. NeedDestroy Setting needdestroy because transaction completed
[2018-08-09 18:59:45] DEBUG[17592] chan_sip.c:
———- END SIP HISTORY for ‘5b6b3c7cd4e27722-Acc269-B2b409@10.15.15.8’0- August 9, 2018 at 5:15 pm
- August 9, 2018 at 5:23 pm
Ok, I disabled SIP debugging and now there are only the information requested. The important think is that I called xxxxx20949 not xxxxx20900
pbx.c: Executing [xxxxx20900@default-trunk:1] Gosub("SIP/xxxxx20900-0000000f", "sub-check-blacklist,s,1(ebe61b6b5987db2a,xxxxxxxxxx)") in new stack
pbx.c: Executing [s@sub-check-blacklist:1] NoOp("SIP/xxxxx20900-0000000f", "Testing if xxxxxxxxxx is in Black List") in new stack
pbx.c: Executing [s@sub-check-blacklist:2] GotoIf("SIP/xxxxx20900-0000000f", "0?banned") in new stack
pbx.c: Executing [s@sub-check-blacklist:3] Return("SIP/xxxxx20900-0000000f", "") in new stack
pbx.c: Executing [xxxxx20900@default-trunk:2] Gosub("SIP/xxxxx20900-0000000f", "sub-setup-call-type,s,1(incoming)") in new stack
pbx.c: Executing [s@sub-setup-call-type:1] NoOp("SIP/xxxxx20900-0000000f", "Determinating Call Type") in new stack
pbx.c: Executing [s@sub-setup-call-type:2] GotoIf("SIP/xxxxx20900-0000000f", "0?return") in new stack
pbx.c: Executing [s@sub-setup-call-type:3] Gosub("SIP/xxxxx20900-0000000f", "s-incoming,1()") in new stack
pbx.c: Executing [s-incoming@sub-setup-call-type:1] NoOp("SIP/xxxxx20900-0000000f", "Incoming Call") in new stack
pbx.c: Executing [s-incoming@sub-setup-call-type:2] Set("SIP/xxxxx20900-0000000f", "__CALL_TYPE=2") in new stack
pbx.c: Executing [s-incoming@sub-setup-call-type:3] Return("SIP/xxxxx20900-0000000f", "") in new stack
pbx.c: Executing [s@sub-setup-call-type:4] Set("SIP/xxxxx20900-0000000f", "__CALL_TYPE_CONFIGURED=yes") in new stack
pbx.c: Executing [s@sub-setup-call-type:5] Set("SIP/xxxxx20900-0000000f", "CDR(calltype)=2") in new stack
pbx.c: Executing [s@sub-setup-call-type:6] Return("SIP/xxxxx20900-0000000f", "") in new stack
pbx.c: Executing [xxxxx20900@default-trunk:3] Goto("SIP/xxxxx20900-0000000f", "incoming-calls,xxxxx20900,1") in new stack
pbx_builtins.c: Goto (incoming-calls,xxxxx20900,1)
pbx.c: Channel 'SIP/xxxxx20900-0000000f' sent to invalid extension: context,exten,priority=incoming-calls,xxxxx20900,1
pbx.c: Executing [i@incoming-calls:1] NoCDR("SIP/xxxxx20900-0000000f", "") in new stack
pbx.c: Executing [i@incoming-calls:2] Goto("SIP/xxxxx20900-0000000f", "invalid-dest,s,1") in new stack
pbx_builtins.c: Goto (invalid-dest,s,1)
pbx.c: Executing [s@invalid-dest:1] NoOp("SIP/xxxxx20900-0000000f", "Invalid Route to Dial") in new stack
pbx.c: Executing [s@invalid-dest:2] Playback("SIP/xxxxx20900-0000000f", "im-sorry&no-route-exists-to-dest&vm-goodbye") in new stack
file.c: <SIP/xxxxx20900-0000000f> Playing 'im-sorry.ulaw' (language 'it')
file.c: <SIP/xxxxx20900-0000000f> Playing 'no-route-exists-to-dest.ulaw' (language 'it')
file.c: <SIP/xxxxx20900-0000000f> Playing 'vm-goodbye.ulaw' (language 'it')
pbx.c: Executing [s@invalid-dest:3] Hangup("SIP/xxxxx20900-0000000f", "") in new stack
pbx.c: Spawn extension (invalid-dest, s, 3) exited non-zero on 'SIP/xxxxx20900-0000000f'0- August 9, 2018 at 5:27 pm
- August 9, 2018 at 5:36 pm
- August 9, 2018 at 5:59 pm
- August 9, 2018 at 6:22 pm
- August 9, 2018 at 6:28 pm
- August 9, 2018 at 6:30 pm
- August 9, 2018 at 6:36 pm
I don’t want annoyng you, maybe is my poor English and I don’t ask in correct way and don’t understand better what you told me.
You first told me “They are sending the DID number in the asterisk header, specifically in the “To” parameter.” And in the To parameter there is the correct number .
After told me “but is not being sending as DID Number”.
My question is: where must exist the DID number ?
Please can you give me a little example.
Regards
Pasquale0- August 9, 2018 at 6:47 pm
- August 9, 2018 at 7:13 pm
Also my question is simple: where ? What is the header parameter ? INVITE ?
I found a similar problem with a FreePBX based installation at link
But I’m new to VitalPBX and I don’t know where put the modification.
I’m surrending. I’d like VitalPBX and I also wanted to buy some addons, but if I not solve this problem not is the right product for me.
Thanks in any case.
Regards
0- August 9, 2018 at 7:28 pm
I told you, the DID number is not send as usally, is send as Header Parameters:
From: “xxxxxxxxxx” <sip:xxxxxxxxxx@clouditalia769.it;user=phone>;tag=1dff4b84.Acc269.B2b409
To: <sip:xxxxx20949@83.211.227.21:5060;user=phone>The kind of behavior that is described in the link that you sent, can be implemented in future versions of VitalPBX
0- August 10, 2018 at 11:50 am
Hi,
I found that in FreePBX the workaround is very simple. Put context=from-pstn-toheader into trunk definition and all work fine. This call
[from-pstn-toheader]
exten => _.,1,Goto(from-pstn,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1)I hope that this simple solution will be adopted as soon as possible in your product. In the while I must use other PBX solutions.
Regards
Pasquale0
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