vicidial.org

Hello ,,,

My inbound is basicly working.

Just the CID Lookup is not working at all , , ill tryed serverel differnt methods ,,


also to clean the cid nummer with the first 4 digits L1 L2 L3 L4 or R10 cause my numbers are separatet from the country code which come 00xx number in in

the debug is

[Oct 31 11:49:55] Using INVITE request as basis request - 4b9c6b50228d-635fa852-19d564d1-35bae200-1e90de-04-UASession-66o1WvM9-R-UASession-EW9FbD!grK
[Oct 31 11:49:55] Found peer 'SIPtrunk100' for '00436811XXXX' from 193.xx.xx.xx:5060
[Oct 31 11:49:55] == Using SIP RTP CoS mark 5
[Oct 31 11:49:55] Found RTP audio format 8
[Oct 31 11:49:55] Found RTP audio format 101
[Oct 31 11:49:55] Found audio description format PCMA for ID 8
[Oct 31 11:49:55] Found audio description format telephone-event for ID 101
[Oct 31 11:49:55] Capabilities: us - (alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
[Oct 31 11:49:55] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Oct 31 11:49:55] > 0x7f9f040fc840 -- Strict RTP learning after remote address set to: 193.84.65.161:24060
[Oct 31 11:49:55] Peer audio RTP is at port 193.84.65.161:24060
[Oct 31 11:49:55] Looking for 0043720XXXXXX0 in trunkinbound (domain 116.xxx.xxx.xx)
[Oct 31 11:49:55] sip_route_dump: route/path hop: <sip:193.xx.xx.xx;lr;ftag=0A889EBA26BC010000471104038Au5958007524222A98;x-rtpp=1>
[Oct 31 11:49:55]
[Oct 31 11:49:55] <--- Transmitting (NAT) to 193.xx.xx.xx:5060 --->
[Oct 31 11:49:55] SIP/2.0 100 Trying
[Oct 31 11:49:55] Via: SIP/2.0/UDP 193.xx.xx.xx;branch=z9hG4bK7c41.1ee6e8d539612e70abaa948d9b2bb99c.0;received=193.xx.xx.xx;rport=5060
[Oct 31 11:49:55] Via: SIP/2.0/UDP 192.168.46.235:5083;branch=z9hG4bKZZRSDgejsV
[Oct 31 11:49:55] Record-Route: <sip:193.xx.xx.xx;lr;ftag=0A889EBA26BC010000471104038Au5958007524222A98;x-rtpp=1>
[Oct 31 11:49:55] From: "00436811XXXX" <sip:00436811XXXX@bt.sixxxx.xx;x-dno=A902>;tag=0A889EBA26BC010000471104038Au5958007524222A98
[Oct 31 11:49:55] To: <sip:+43720XXXXXX0@192.168.46.235;x-sin=102>
[Oct 31 11:49:55] Call-ID: 4b9c6b50228d-635fa852-19d564d1-35bae200-1e90de-04-UASession-66o1WvM9-R-UASession-EW9FbD!grK
[Oct 31 11:49:55] CSeq: 1 INVITE
[Oct 31 11:49:55] Server: Asterisk PBX 13.29.2-vici
[Oct 31 11:49:55] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Oct 31 11:49:55] Supported: replaces, timer
[Oct 31 11:49:55] Session-Expires: 1800;refresher=uas
[Oct 31 11:49:55] Contact: <sip:0043720XXXXXX0@116.xxx.xxx.xx:5060>
[Oct 31 11:49:55] Content-Length: 0
[Oct 31 11:49:55]
[Oct 31 11:49:55]
[Oct 31 11:49:55] <------------>
[Oct 31 11:49:55] -- Executing [0043720XXXXXX0@trunkinbound:1] AGI("SIP/SIPtrunk100-000033df", "agi-DID_route.agi") in new stack
[Oct 31 11:49:55] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[Oct 31 11:49:55] -- <SIP/SIPtrunk100-000033df>AGI Script agi-DID_route.agi completed, returning 0
[Oct 31 11:49:55] -- Executing [99909*1***DID@default:1] Answer("SIP/SIPtrunk100-000033df", "") in new stack
[Oct 31 11:49:55] Audio is at 13310
[Oct 31 11:49:55] Adding codec alaw to SDP
[Oct 31 11:49:55] Adding non-codec 0x1 (telephone-event) to SDP
[Oct 31 11:49:55]
[Oct 31 11:49:55] <--- Reliably Transmitting (NAT) to 193.xx.xx.xx:5060 --->
[Oct 31 11:49:55] SIP/2.0 200 OK
[Oct 31 11:49:55] Via: SIP/2.0/UDP 193.xx.xx.xx;branch=z9hG4bK7c41.1ee6e8d539612e70abaa948d9b2bb99c.0;received=193.xx.xx.xx;rport=5060
[Oct 31 11:49:55] Via: SIP/2.0/UDP 192.168.46.235:5083;branch=z9hG4bKZZRSDgejsV
[Oct 31 11:49:55] Record-Route: <sip:193.xx.xx.xx;lr;ftag=0A889EBA26BC010000471104038Au5958007524222A98;x-rtpp=1>
[Oct 31 11:49:55] From: "00436811XXXX" <sip:00436811XXXX@bt.sixxxx.xx;x-dno=A902>;tag=0A889EBA26BC010000471104038Au5958007524222A98
[Oct 31 11:49:55] To: <sip:+43720XXXXXX0@192.168.46.235;x-sin=102>;tag=as78c00f49
[Oct 31 11:49:55] Call-ID: 4b9c6b50228d-635fa852-19d564d1-35bae200-1e90de-04-UASession-66o1WvM9-R-UASession-EW9FbD!grK
[Oct 31 11:49:55] CSeq: 1 INVITE
[Oct 31 11:49:55] Server: Asterisk PBX 13.29.2-vici
[Oct 31 11:49:55] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Oct 31 11:49:55] Supported: replaces, timer
[Oct 31 11:49:55] Session-Expires: 1800;refresher=uas
[Oct 31 11:49:55] Contact: <sip:0043720XXXXXX0@116.xxx.xxx.xx:5060>
[Oct 31 11:49:55] Content-Type: application/sdp
[Oct 31 11:49:55] Require: timer
[Oct 31 11:49:55] Content-Length: 261
[Oct 31 11:49:55]
[Oct 31 11:49:55] v=0
[Oct 31 11:49:55] o=root 1352178161 1352178161 IN IP4 116.xxx.xxx.xx
[Oct 31 11:49:55] s=Asterisk PBX 13.29.2-vici
[Oct 31 11:49:55] c=IN IP4 116.xxx.xxx.xx
[Oct 31 11:49:55] t=0 0
[Oct 31 11:49:55] m=audio 13310 RTP/AVP 8 101
[Oct 31 11:49:55] a=rtpmap:8 PCMA/8000
[Oct 31 11:49:55] a=rtpmap:101 telephone-event/8000
[Oct 31 11:49:55] a=fmtp:101 0-16
[Oct 31 11:49:55] a=ptime:20
[Oct 31 11:49:55] a=maxptime:150
[Oct 31 11:49:55] a=sendrecv
[Oct 31 11:49:55]
[Oct 31 11:49:55] <------------>
[Oct 31 11:49:55]
[Oct 31 11:49:55] <--- SIP read from UDP:193.xx.xx.xx:5060 --->
[Oct 31 11:49:55] ACK sip:0043720XXXXXX0@116.xxx.xxx.xx:5060 SIP/2.0
[Oct 31 11:49:55] Via: SIP/2.0/UDP 193.xx.xx.xx;branch=z9hG4bK7c41.8febb0293b904233c4b56b80b4a2c5d7.0
[Oct 31 11:49:55] Via: SIP/2.0/UDP 192.168.46.235:5083;branch=z9hG4bKRvYvvmTE5e
[Oct 31 11:49:55] From: <sip:00436811XXXX@bt.sixxxx.xx;x-dno=A902>;tag=0A889EBA26BC010000471104038Au5958007524222A98
[Oct 31 11:49:55] To: <sip:+43720XXXXXX0@192.168.46.235;x-sin=102>;tag=as78c00f49
[Oct 31 11:49:55] Call-ID: 4b9c6b50228d-635fa852-19d564d1-35bae200-1e90de-04-UASession-66o1WvM9-R-UASession-EW9FbD!grK
[Oct 31 11:49:55] CSeq: 1 ACK
[Oct 31 11:49:55] Max-Forwards: 60
[Oct 31 11:49:55] Contact: <sip:192.168.46.235:5083>
[Oct 31 11:49:55] Content-Length: 0
[Oct 31 11:49:55]
[Oct 31 11:49:55] <------------->
[Oct 31 11:49:55] --- (10 headers 0 lines) ---
[Oct 31 11:49:55] > 0x7f9f040fc840 -- Strict RTP switching to RTP target address 193.84.65.161:24060 as source
[Oct 31 11:49:55] -- Executing [99909*1***DID@default:2] AGI("SIP/SIPtrunk100-000033df", "agi-VDAD_ALL_inbound.agi") in new stack
[Oct 31 11:49:55] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Oct 31 11:49:55] -- <SIP/SIPtrunk100-000033df> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 31 11:49:55] -- <SIP/SIPtrunk100-000033df> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 31 11:49:57] -- Started music on hold, class 'default', on channel 'SIP/SIPtrunk100-000033df'
[Oct 31 11:49:58] Really destroying SIP dialog 'qEKjquwCFeR4wkfxMYOYIg..' Method: REGISTER
[Oct 31 11:50:00] > 0x7f9f040fc840 -- Strict RTP learning complete - Locking on source address 193.84.65.161:24060
[Oct 31 11:50:00] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 31 11:50:00] -- Called 58600052@default
[Oct 31 11:50:00] -- Executing [58600052@default:1] MeetMe("Local/58600052@default-0000563e;2", "8600052,Fmq") in new stack
[Oct 31 11:50:00] -- Local/58600052@default-0000563e;1 answered
[Oct 31 11:50:00] -- Executing [8309@default:1] Answer("Local/58600052@default-0000563e;1", "") in new stack
[Oct 31 11:50:00] -- Executing [8309@default:2] Monitor("Local/58600052@default-0000563e;1", "wav,20221031-115000_00436811XXXX") in new stack
[Oct 31 11:50:00] -- Executing [8309@default:3] Wait("Local/58600052@default-0000563e;1", "3600") in new stack
[Oct 31 11:50:00] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 31 11:50:00] -- Called 116*xxx*xxx*xxx*78600052@default
[Oct 31 11:50:00] -- Executing [116*xxx*xxx*xxx*78600052@default:1] Goto("Local/116*xxx*xxx*xxx*78600052@default-0000563f;2", "default,78600052,1") in new stack
[Oct 31 11:50:00] -- Goto (default,78600052,1)
[Oct 31 11:50:00] -- Executing [78600052@default:1] MeetMe("Local/116*xxx*xxx*xxx*78600052@default-0000563f;2", "8600052,Fq") in new stack
[Oct 31 11:50:00] -- Local/116*xxx*xxx*xxx*78600052@default-0000563f;1 answered
[Oct 31 11:50:00] -- Executing [83047777777777@vicidial-auto:1] Answer("Local/116*xxx*xxx*xxx*78600052@default-0000563f;1", "") in new stack
[Oct 31 11:50:00] -- Executing [83047777777777@vicidial-auto:2] Playback("Local/116*xxx*xxx*xxx*78600052@default-0000563f;1", "ding") in new stack
[Oct 31 11:50:00] -- <Local/116*xxx*xxx*xxx*78600052@default-0000563f;1> Playing 'ding.gsm' (language 'en')
[Oct 31 11:50:00] -- Executing [83047777777777@vicidial-auto:3] Hangup("Local/116*xxx*xxx*xxx*78600052@default-0000563f;1", "") in new stack
[Oct 31 11:50:00] == Spawn extension (vicidial-auto, 83047777777777, 3) exited non-zero on 'Local/116*xxx*xxx*xxx*78600052@default-0000563f;1'
[Oct 31 11:50:00] WARNING[17492][C-0003034e]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Oct 31 11:50:00] -- Executing [h@vicidial-auto:1] AGI("Local/116*xxx*xxx*xxx*78600052@default-0000563f;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[Oct 31 11:50:00] -- <Local/116*xxx*xxx*xxx*78600052@default-0000563f;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ----------) completed, returning 0
[Oct 31 11:50:00] == Spawn extension (default, 78600052, 1) exited non-zero on 'Local/116*xxx*xxx*xxx*78600052@default-0000563f;2'
[Oct 31 11:50:00] WARNING[17493][C-0003034d]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Oct 31 11:50:00] -- Executing [h@default:1] AGI("Local/116*xxx*xxx*xxx*78600052@default-0000563f;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[Oct 31 11:50:00] -- <Local/116*xxx*xxx*xxx*78600052@default-0000563f;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ----------) completed, returning 0
[Oct 31 11:50:00] Really destroying SIP dialog '247c1294-88919fa3-ec9a208@193.xx.xx.xx' Method: OPTIONS
[Oct 31 11:50:01] -- Stopped music on hold on SIP/SIPtrunk100-000033df
[Oct 31 11:50:01] -- <SIP/SIPtrunk100-000033df> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 31 11:50:01] -- <SIP/SIPtrunk100-000033df> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 31 11:50:01] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 31 11:50:01] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 31 11:50:01] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 31 11:50:01] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 31 11:50:01] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 31 11:50:01] -- <SIP/SIPtrunk100-000033df> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 31 11:50:01] -- <SIP/SIPtrunk100-000033df> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 31 11:50:01] -- <SIP/SIPtrunk100-000033df> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 31 11:50:01] -- <SIP/SIPtrunk100-000033df> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 31 11:50:02] -- <SIP/SIPtrunk100-000033df>AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
[Oct 31 11:50:02] -- Executing [116*xxx*xxx*xxx*8600052@default:1] Goto("SIP/SIPtrunk100-000033df", "default,8600052,1") in new stack
[Oct 31 11:50:02] -- Goto (default,8600052,1)
[Oct 31 11:50:02] -- Executing [8600052@default:1] MeetMe("SIP/SIPtrunk100-000033df", "8600052,F") in new stack



any idea whats wrong ?

regards Speedmaker


VERSION: 2.14-718a
BUILD: 190902-0839
© 2019 ViciDial Group

Hi,

Try using L0043 in the Clean CID box and test again.

hello , ,

yeah ,, ill tryed ,, but no change ,,

Ensure your server entry has AGI Output set to FILE or BOTH, wait a few minutes if you have to change it, then call in again, after the call, on the cli run the following command and post the output here inside code tags.

Code: Select all

cat /var/log/astguiclient/agiout.2022-10-31 | grep 'agi-DID_route.agi'

it is on FILE ,,

2022-10-31 14:44:08|agi-DID_route.agi||INSERT INTO vicidial_did_log SET uniqueid='1667223848.58467',channel='SIP/SIPtrunk100-000034b9',server_ip='116.203.235.54',caller_id_number='00436811xxxxx',caller_id_name='0043681xxxxxxx',extension='004372070xxxxx',call_date='2022-10-31 14:44:08',did_id='1',did_route='IN_GROUP';|
2022-10-31 14:44:08|agi-DID_route.agi|-- DID LOG : |1|INSERT INTO vicidial_did_log SET uniqueid='1667223848.58467',channel='SIP/SIPtrunk100-000034b9',server_ip='116.203.235.54',caller_id_number='00436811xxxxxx',caller_id_name='004368xxxxx',extension='00437207xxxxxx',call_date='2022-10-31 14:44:08',did_id='1',did_route='IN_GROUP';|
2022-10-31 14:44:08|agi-DID_route.agi|-- CALL LOG : |1|INSERT INTO call_log SET uniqueid='1667223848.58467', channel='SIP/SIPtrunk100-000034b9', channel_group='DID_INBOUND', server_ip='116.203.xxx.xx', type='SIP', extension='004372xxxxxxx', number_dialed='00437207xxxxxx', caller_code='0043681xxxxxx', start_time='2022-10-31 14:44:08', start_epoch='1667223848';|
2022-10-31 14:44:08|agi-DID_route.agi|exiting the DID app, transferring call to 99909*1***DID @ default

You're likely facing a bug that was patched in SVN 3381.
http://www.vicidial.org/VICIDIALmantis/view.php?id=1247

The patch that fixed it was:

Revision: 3381
Author: mattf
Date: Monday, March 15, 2021 11:44:29
Message:
Fix for multiple filter_clean_cid_number actions in the DID routing AGI script, also added 'T' filter action, Issue #1247
----
Modified : /agc_2-X/trunk/agi/agi-DID_route.agi
Modified : /agc_2-X/trunk/www/vicidial/help_documentation.txt


You'll need to update your server to resolve it.

better use the dialplan to discard the 0043 before sending to AGI(agi-DID_route.agi)

vi /etc/asterisk/extensions.conf

add the below diaplan entry under trunkinbound context along with default entries
make sure the number stored in list should match 720XXXXXX0 and CID lookup set to CIDLOOKUP or CIDLOOKUPRL or CIDLOOKUPRC

[trunkinbound]

exten => _0043X.,1,Goto(trunkinbound,${EXTEN:4},1)

; DID call routing process
; exten => _XXXXXXXXXX,1,AGI(agi-DID_route.agi) ; use this one instead of the one below if you are having delay issues, and match to number of received digits
exten => _X.,1,AGI(agi-DID_route.agi)

; FastAGI for VICIDIAL/astGUIclient call logging
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

regards
striker24x7
https://www.striker24x7.com

hello striker

my exension conf looks now like that

#include extensions-vicidial.conf

;---Outbound ---
SIPAddHeader("+4372xxxx: <sip:dreameter@bt.sxxx.xx>")
Dial(SIP/thenumbertodial@yuxxxxl)

[trunkinbound]
; DID call routing process
exten => _0043X.,1,Goto(trunkinbound,${EXTEN:4},1)
;exten => _XXXXXXXXXX,1,AGI(agi-DID_route.agi) ; use this one instead of the one below if you are having delay issues, and match to number of received digits
;exten => _X.,1,AGI(agi-DID_route.agi)
exten => _X.,1,AGI(agi-DID_route.agi)
exten => _+437205XXXX,1,AGI(agi-DID_route.agi)
exten => _X.,n,Hangup()
exten => _+4X.,1,Goto(trunkinbound,${EXTEN:1},1)
exten => _X.,1,AGI(agi-DID_route.agi)
;exten => _X.,n,Hangup()
; If you have DIDs that arrive with a plus sign at the beginning then uncomment
exten => _+X.,1,AGI(agi-DID_route.agi)
exten => _+X.,n,Hangup()
; If you have DIDs that arrive with a plus and 1 at the beginning that you want to filter out, then uncomment
;exten => _+2X.,1,Goto(trunkinbound,${EXTEN:2},1)
;exten => _43720xxxxxx,1,AGI(agi-DID_route.agi)
;exten => _XXXXXXXXXX,1,AGI(agi-DID_route.agi)
;exten => _X.,n,Hangup()

; FastAGI for VICIDIAL/astGUIclient call logging
;exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME}-----${HANGUPCAUSE(${HANGUPCAUSE_KEYS()},tech)}))
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

but still same result ,,

any idea

Try this instead at the top of trunkinbound

exten => _+43X.,1,Set(CALLERID(num)=${CALLERID(num):4})
same => n,Goto(trunkinbound,${EXTEN:1},1)

This will strip the first 4 digits off the received caller id and the + from the destination number before passing it to the normal processing.

Remember to issue a dialplan reload in asterisk after updating extensions.conf. (rasterisk -x 'dialplan reload')
There's no reason though the feature designed to do this for you shouldn't work on the latest SVN version, I have it in place on nearly every instance I manage and it works fine using L[countrycode] in the clean caller id number field. Eg. L1 for US/Canada, L44 for UK, L34 for Spain and they all work as expected.

hello ,, yeahh ,, ill tryed ,, but still no difference , ,

it seems to ignore whatever i write into the dialplan ,, im confused ,,

<--- SIP read from UDP:193.xx.xx.xx:5060 --->
[Nov 1 22:28:57] INVITE sip:004372xxxxxxxxx@116.2xx.xxx.xx:5060 SIP/2.0
[Nov 1 22:28:57] Record-Route: <sip:193.xx.xx.xx;lr;ftag=0A889EBA26BC010000471103038Au2AC6035082EBA962;x-rtpp=1>
[Nov 1 22:28:57] Via: SIP/2.0/UDP 193.xx.xx.xx;branch=z9hG4bKae38.40973e983b7d8c087d29d4637aa29410.0
[Nov 1 22:28:57] Via: SIP/2.0/UDP 192.168.46.235:5083;branch=z9hG4bKJo!L8X9wcU
[Nov 1 22:28:57] From: "00436811xxxxxxx" <sip:00436811xxxxxxx@bt.sipxxxx.at;x-dno=A903>;tag=0A889EBA26BC010000471103038Au2AC6035082EBA962
[Nov 1 22:28:57] To: <sip:+4372xxxxxxxxx@192.168.46.235;x-sin=102>
[Nov 1 22:28:57] Call-ID: 48ab6b502c8d-63618f99-441b9abb-c51d6d0-214fc8c-03-UASession-vwfIBceB3h-UASession-d-YnWxXyaW
[Nov 1 22:28:57] CSeq: 1 INVITE
[Nov 1 22:28:57] Max-Forwards: 60
[Nov 1 22:28:57] Supported: timer
[Nov 1 22:28:57] Unsupported: refer
[Nov 1 22:28:57] Allow: INVITE,ACK,CANCEL,BYE,INFO,REGISTER,NOTIFY
[Nov 1 22:28:57] Contact: <sip:192.168.46.235:5083>
[Nov 1 22:28:57] Content-Length: 265
[Nov 1 22:28:57] Content-Type: application/sdp
[Nov 1 22:28:57] Allow-Events: talk
[Nov 1 22:28:57] Accept: application/sdp
[Nov 1 22:28:57]
[Nov 1 22:28:57] v=0
[Nov 1 22:28:57] o=- 1032811322356587645 1 IN IP4 193.84.xx.xxx
[Nov 1 22:28:57] s=TELES-SBC
[Nov 1 22:28:57] c=IN IP4 193.84.xx.xxx
[Nov 1 22:28:57] t=0 0
[Nov 1 22:28:57] m=audio 26694 RTP/AVP 8 101
[Nov 1 22:28:57] a=silenceSupp:off - - - -
[Nov 1 22:28:57] a=rtpmap:8 PCMA/8000
[Nov 1 22:28:57] a=rtpmap:101 telephone-event/8000
[Nov 1 22:28:57] a=fmtp:101 0-15
[Nov 1 22:28:57] a=sendrecv
[Nov 1 22:28:57] a=rtcp:26695
[Nov 1 22:28:57] a=ptime:20
[Nov 1 22:28:57] <------------->
[Nov 1 22:28:57] --- (17 headers 13 lines) ---
[Nov 1 22:28:57] Sending to 193.xx.xx.xx:5060 (NAT)
[Nov 1 22:28:57] Sending to 193.xx.xx.xx:5060 (NAT)
[Nov 1 22:28:57] Using INVITE request as basis request - 48ab6b502c8d-63618f99-441b9abb-c51d6d0-214fc8c-03-UASession-vwfIBceB3h-UASession-d-YnWxXyaW
[Nov 1 22:28:57] Found peer 'SIPtrunk100' for '00436811xxxxxxx' from 193.xx.xx.xx:5060
[Nov 1 22:28:57] == Using SIP RTP CoS mark 5
[Nov 1 22:28:57] Found RTP audio format 8
[Nov 1 22:28:57] Found RTP audio format 101
[Nov 1 22:28:57] Found audio description format PCMA for ID 8
[Nov 1 22:28:57] Found audio description format telephone-event for ID 101
[Nov 1 22:28:57] Capabilities: us - (alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
[Nov 1 22:28:57] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Nov 1 22:28:57] > 0x7f9f040fc840 -- Strict RTP learning after remote address set to: 193.84.xx.xxx:26694
[Nov 1 22:28:57] Peer audio RTP is at port 193.84.xx.xxx:26694
[Nov 1 22:28:57] Looking for 004372xxxxxxxxx in trunkinbound (domain 116.2xx.xxx.xx)
[Nov 1 22:28:57] sip_route_dump: route/path hop: <sip:193.xx.xx.xx;lr;ftag=0A889EBA26BC010000471103038Au2AC6035082EBA962;x-rtpp=1>
[Nov 1 22:28:57]
[Nov 1 22:28:57] <--- Transmitting (NAT) to 193.xx.xx.xx:5060 --->
[Nov 1 22:28:57] SIP/2.0 100 Trying
[Nov 1 22:28:57] Via: SIP/2.0/UDP 193.xx.xx.xx;branch=z9hG4bKae38.40973e983b7d8c087d29d4637aa29410.0;received=193.xx.xx.xx;rport=5060
[Nov 1 22:28:57] Via: SIP/2.0/UDP 192.168.46.235:5083;branch=z9hG4bKJo!L8X9wcU
[Nov 1 22:28:57] Record-Route: <sip:193.xx.xx.xx;lr;ftag=0A889EBA26BC010000471103038Au2AC6035082EBA962;x-rtpp=1>
[Nov 1 22:28:57] From: "00436811xxxxxxx" <sip:00436811xxxxxxx@bt.sipxxxx.at;x-dno=A903>;tag=0A889EBA26BC010000471103038Au2AC6035082EBA962
[Nov 1 22:28:57] To: <sip:+4372xxxxxxxxx@192.168.46.235;x-sin=102>
[Nov 1 22:28:57] Call-ID: 48ab6b502c8d-63618f99-441b9abb-c51d6d0-214fc8c-03-UASession-vwfIBceB3h-UASession-d-YnWxXyaW
[Nov 1 22:28:57] CSeq: 1 INVITE
[Nov 1 22:28:57] Server: Asterisk PBX 13.29.2-vici
[Nov 1 22:28:57] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Nov 1 22:28:57] Supported: replaces, timer
[Nov 1 22:28:57] Session-Expires: 1800;refresher=uas
[Nov 1 22:28:57] Contact: <sip:004372xxxxxxxxx@116.2xx.xxx.xx:5060>
[Nov 1 22:28:57] Content-Length: 0
[Nov 1 22:28:57]
[Nov 1 22:28:57]
[Nov 1 22:28:57] <------------>
[Nov 1 22:28:57] -- Executing [004372xxxxxxxxx@trunkinbound:1] Goto("SIP/SIPtrunk100-000034cb", "trunkinbound,72xxxxxxxxx,1") in new stack
[Nov 1 22:28:57] -- Goto (trunkinbound,72xxxxxxxxx,1)
[Nov 1 22:28:57] -- Executing [72xxxxxxxxx@trunkinbound:1] AGI("SIP/SIPtrunk100-000034cb", "agi-DID_route.agi") in new stack
[Nov 1 22:28:57] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[Nov 1 22:28:57] -- <SIP/SIPtrunk100-000034cb>AGI Script agi-DID_route.agi completed, returning 0
[Nov 1 22:28:57] -- Executing [99909*1***DID@default:1] Answer("SIP/SIPtrunk100-000034cb", "") in new stack
[Nov 1 22:28:57] Audio is at 16062
[Nov 1 22:28:57] Adding codec alaw to SDP
[Nov 1 22:28:57] Adding non-codec 0x1 (telephone-event) to SDP
[Nov 1 22:28:57]
[Nov 1 22:28:57] <--- Reliably Transmitting (NAT) to 193.xx.xx.xx:5060 --->
[Nov 1 22:28:57] SIP/2.0 200 OK
[Nov 1 22:28:57] Via: SIP/2.0/UDP 193.xx.xx.xx;branch=z9hG4bKae38.40973e983b7d8c087d29d4637aa29410.0;received=193.xx.xx.xx;rport=5060
[Nov 1 22:28:57] Via: SIP/2.0/UDP 192.168.46.235:5083;branch=z9hG4bKJo!L8X9wcU
[Nov 1 22:28:57] Record-Route: <sip:193.xx.xx.xx;lr;ftag=0A889EBA26BC010000471103038Au2AC6035082EBA962;x-rtpp=1>
[Nov 1 22:28:57] From: "00436811xxxxxxx" <sip:00436811xxxxxxx@bt.sipxxxx.at;x-dno=A903>;tag=0A889EBA26BC010000471103038Au2AC6035082EBA962
[Nov 1 22:28:57] To: <sip:+4372xxxxxxxxx@192.168.46.235;x-sin=102>;tag=as4310c1b7
[Nov 1 22:28:57] Call-ID: 48ab6b502c8d-63618f99-441b9abb-c51d6d0-214fc8c-03-UASession-vwfIBceB3h-UASession-d-YnWxXyaW
[Nov 1 22:28:57] CSeq: 1 INVITE
[Nov 1 22:28:57] Server: Asterisk PBX 13.29.2-vici
[Nov 1 22:28:57] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Nov 1 22:28:57] Supported: replaces, timer
[Nov 1 22:28:57] Session-Expires: 1800;refresher=uas
[Nov 1 22:28:57] Contact: <sip:004372xxxxxxxxx@116.2xx.xxx.xx:5060>
[Nov 1 22:28:57] Content-Type: application/sdp
[Nov 1 22:28:57] Require: timer
[Nov 1 22:28:57] Content-Length: 259
[Nov 1 22:28:57]
[Nov 1 22:28:57] v=0
[Nov 1 22:28:57] o=root 731354311 731354311 IN IP4 116.2xx.xxx.xx
[Nov 1 22:28:57] s=Asterisk PBX 13.29.2-vici
[Nov 1 22:28:57] c=IN IP4 116.2xx.xxx.xx
[Nov 1 22:28:57] t=0 0
[Nov 1 22:28:57] m=audio 16062 RTP/AVP 8 101
[Nov 1 22:28:57] a=rtpmap:8 PCMA/8000
[Nov 1 22:28:57] a=rtpmap:101 telephone-event/8000
[Nov 1 22:28:57] a=fmtp:101 0-16
[Nov 1 22:28:57] a=ptime:20
[Nov 1 22:28:57] a=maxptime:150
[Nov 1 22:28:57] a=sendrecv
[Nov 1 22:28:57]
[Nov 1 22:28:57] <------------>
[Nov 1 22:28:57]
[Nov 1 22:28:57] <--- SIP read from UDP:193.xx.xx.xx:5060 --->
[Nov 1 22:28:57] ACK sip:004372xxxxxxxxx@116.2xx.xxx.xx:5060 SIP/2.0
[Nov 1 22:28:57] Via: SIP/2.0/UDP 193.xx.xx.xx;branch=z9hG4bKae38.a2f0ceb5c85f18afeab93f5499c409fc.0
[Nov 1 22:28:57] Via: SIP/2.0/UDP 192.168.46.235:5083;branch=z9hG4bKRD43D_-6JZ
[Nov 1 22:28:57] From: <sip:00436811xxxxxxx@bt.sipxxxx.at;x-dno=A903>;tag=0A889EBA26BC010000471103038Au2AC6035082EBA962
[Nov 1 22:28:57] To: <sip:+4372xxxxxxxxx@192.168.46.235;x-sin=102>;tag=as4310c1b7
[Nov 1 22:28:57] Call-ID: 48ab6b502c8d-63618f99-441b9abb-c51d6d0-214fc8c-03-UASession-vwfIBceB3h-UASession-d-YnWxXyaW
[Nov 1 22:28:57] CSeq: 1 ACK
[Nov 1 22:28:57] Max-Forwards: 60
[Nov 1 22:28:57] Contact: <sip:192.168.46.235:5083>
[Nov 1 22:28:57] Content-Length: 0
[Nov 1 22:28:57]
[Nov 1 22:28:57] <------------->
[Nov 1 22:28:57] --- (10 headers 0 lines) ---
[Nov 1 22:28:57] > 0x7f9f040fc840 -- Strict RTP switching to RTP target address 193.84.xx.xxx:26694 as source
[Nov 1 22:28:57] -- Executing [99909*1***DID@default:2] AGI("SIP/SIPtrunk100-000034cb", "agi-VDAD_ALL_inbound.agi") in new stack
[Nov 1 22:28:57] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Nov 1 22:28:57] -- <SIP/SIPtrunk100-000034cb> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Nov 1 22:28:57] -- <SIP/SIPtrunk100-000034cb> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Nov 1 22:28:58] Retransmitting #2 (NAT) to 209.159.155.182:19860:
[Nov 1 22:28:58] SIP/2.0 401 Unauthorized
[Nov 1 22:28:58] Via: SIP/2.0/UDP 109.50.75.111:5060;branch=z9hG4bK-524287-1---fnjafv35me3ctd6e;received=209.159.155.182;rport=19860
[Nov 1 22:28:58] From: <sip:109@116.2xx.xxx.xx;transport=UDP>;tag=x4bz5s6a
[Nov 1 22:28:58] To: <sip:11110390237920793@116.2xx.xxx.xx;transport=UDP>;tag=as03e1f98e
[Nov 1 22:28:58] Call-ID: BflGlDZLBW4HcPXTlJvlQE..
[Nov 1 22:28:58] CSeq: 1 INVITE
[Nov 1 22:28:58] Server: Asterisk PBX 13.29.2-vici
[Nov 1 22:28:58] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Nov 1 22:28:58] Supported: replaces, timer
[Nov 1 22:28:58] WWW-Authenticate: Digest algorithm=MD5, realm="dialer.ddnss.org", nonce="3496a4e3"
[Nov 1 22:28:58] Content-Length: 0

Post your entire extensions.conf inside code tags so it's easier to read. Something isn't matching up, the sip dialog says it's to +43... but it's looking for 0043..., if you added what I said at the top of trunkinbound then it should be looking for 43... in trunkinbound you'll need the did configured with 43 in the number.

Did you reload the dialplan after saving the changes before? If you didn't, asterisk isn't aware of the changes you made to extensions.conf.

[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp ; Console interface for demo
TRUNK=DAHDI/r1 ; Trunk interface
TRUNKX=DAHDI/r2 ; 2nd trunk interface
TRUNKIAX=IAX2/ASTtest1:test@10.10.10.16:4569 ; IAX trunk interface
TRUNKIAX1=IAX2/ASTtest1:test@10.10.10.16:4569 ; IAX trunk interface
TRUNKBINFONE=IAX2/1112223333:PASSWORD@iax.binfone.com ; IAX trunk interface
SIPtrunk=SIP/1234:PASSWORD@sip.provider.net ; SIP trunk

#include extensions-vicidial.conf

;---Outbound ---
SIPAddHeader("+437207xxxxxx: <sip:dreaxxxx@bt.sipxxx.xxxx>")
Dial(SIP/thenumbertodial@xxxtel)

[trunkinbound]
; DID call routing process
exten => _+43X.,1,Set(CALLERID(NUM)=${CALLERID(NUM):4})
same => n,Goto(trunkinbound,${EXTEN:1},1)
exten => _0043X.,1,Goto(trunkinbound,${EXTEN:4},1)
;exten => _XXXXXXXXXX,1,AGI(agi-DID_route.agi) ; use this one instead of the one below if you are having delay issues, and match to number of received digits
;exten => _X.,1,AGI(agi-DID_route.agi)
exten => _X.,1,AGI(agi-DID_route.agi)
;exten => _+437205XXXX,1,AGI(agi-DID_route.agi)
exten => _X.,n,Hangup()
exten => _+4X.,1,Goto(trunkinbound,${EXTEN:1},1)
exten => _X.,1,AGI(agi-DID_route.agi)
;exten => _X.,n,Hangup()
; If you have DIDs that arrive with a plus sign at the beginning then uncomment
exten => _+X.,1,AGI(agi-DID_route.agi)
exten => _+X.,n,Hangup()
; If you have DIDs that arrive with a plus and 1 at the beginning that you want to filter out, then uncomment
;exten => _+2X.,1,Goto(trunkinbound,${EXTEN:2},1)
;exten => _43720xxxxxx,1,AGI(agi-DID_route.agi)
;exten => _XXXXXXXXXX,1,AGI(agi-DID_route.agi)
;exten => _X.,n,Hangup()


; FastAGI for VICIDIAL/astGUIclient call logging
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME}-----${HANGUPCAUSE(${HANGUPCAUSE_KEYS()},tech)}))

hello,

yeahh , , i reloaded all - but no difference ,,

speedmaker wrote:[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp ; Console interface for demo
TRUNK=DAHDI/r1 ; Trunk interface
TRUNKX=DAHDI/r2 ; 2nd trunk interface
TRUNKIAX=IAX2/ASTtest1:test@10.10.10.16:4569 ; IAX trunk interface
TRUNKIAX1=IAX2/ASTtest1:test@10.10.10.16:4569 ; IAX trunk interface
TRUNKBINFONE=IAX2/1112223333:PASSWORD@iax.binfone.com ; IAX trunk interface
SIPtrunk=SIP/1234:PASSWORD@sip.provider.net ; SIP trunk

#include extensions-vicidial.conf

;---Outbound ---
SIPAddHeader("+437207xxxxxx: <sip:dreaxxxx@bt.sipxxx.xxxx>")
Dial(SIP/thenumbertodial@xxxtel)

[trunkinbound]
; DID call routing process
exten => _+43X.,1,Set(CALLERID(NUM)=${CALLERID(NUM):4})
same => n,Goto(trunkinbound,${EXTEN:1},1)
exten => _0043X.,1,Goto(trunkinbound,${EXTEN:4},1)
;exten => _XXXXXXXXXX,1,AGI(agi-DID_route.agi) ; use this one instead of the one below if you are having delay issues, and match to number of received digits
;exten => _X.,1,AGI(agi-DID_route.agi)
exten => _X.,1,AGI(agi-DID_route.agi)
;exten => _+437205XXXX,1,AGI(agi-DID_route.agi)
exten => _X.,n,Hangup()
exten => _+4X.,1,Goto(trunkinbound,${EXTEN:1},1)
exten => _X.,1,AGI(agi-DID_route.agi)
;exten => _X.,n,Hangup()
; If you have DIDs that arrive with a plus sign at the beginning then uncomment
exten => _+X.,1,AGI(agi-DID_route.agi)
exten => _+X.,n,Hangup()
; If you have DIDs that arrive with a plus and 1 at the beginning that you want to filter out, then uncomment
;exten => _+2X.,1,Goto(trunkinbound,${EXTEN:2},1)
;exten => _43720xxxxxx,1,AGI(agi-DID_route.agi)
;exten => _XXXXXXXXXX,1,AGI(agi-DID_route.agi)
;exten => _X.,n,Hangup()


; FastAGI for VICIDIAL/astGUIclient call logging
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME}-----${HANGUPCAUSE(${HANGUPCAUSE_KEYS()},tech)}))

ohh ,, iff somebody know - i would also pay for a solution ,,


thx speed

These lines:

SIPAddHeader("+437207xxxxxx: <sip:dreaxxxx@bt.sipxxx.xxxx>")
Dial(SIP/thenumbertodial@xxxtel)

are not valid syntax, perhaps they are causing the reload to fail. Anyways, if you want, reach out to me on Skype and I'll have a look. live:support_71847

well thats outbound ,, for whatever ,,

and its doing absolutly nothing

Resolved after a skype session with @speedmaker.

The call was going to 0043XXXXXXXX but the DID wasn't set up, instead it was going to default which was pointing at ingroup but no clean cid number set.

Reverted default back to the standard default, set up the DID with 0043 rather than +43, route set to ingroup with L0043 in the clean cid number field and it works perfectly.

Pleasure doing business with you!

thx for help ! was a pleasure !