Hallo zusammen,
habe hier einen interessanten Effekt zwischen Asterisk (1.2.10) und einem Cisco Call Manager Express (CCME) bzw Unity Express.
Folgendes passiert:
- Anruf über SIP von 124 an 83104
- Es geht niemand ran und das Telefon leitet auf die Voicemail um => SIP 302 Paket mit der Info [email protected] (Unity) anzurufen
- Asterisk nimmt die Nummer und ruft im Kontext from-sip an, also 2100@from-sip
Das Telefon mit der 124 ist über MISDN angebunden (HiPath 4000), lasst euch davon nicht irritieren.
Ich verstehe nicht ganz warum Asterisk im Kontext from-sip anruft und nicht einfach den auf die IP Adresse anruft.
Hat das was damit zu tun das in der sip.conf ein context=from-sip steht?
In der aktuellen Konfiguration (s.u.) fange ich die 2100 ab und starte einen völlig neuen Anruf an [email protected], das entspricht natürlich nicht dem weitergeleiteten Anruf an die Voicebox.
Ist das ein Bug in Asterisk (Ignorieren der IP Information im SIP Paket) oder muss ich da noch etwas entsprechend konfigurieren?
Ich kann mir nur vorstellen das es eben etwas mit der Kontextdefinition in der sip.conf zu tun hat, habe aber keine Idee wie ich das umbauen müsste...
Vielen Dank schonmal fürs lesen
stephan
Noch die extensions.conf:
habe hier einen interessanten Effekt zwischen Asterisk (1.2.10) und einem Cisco Call Manager Express (CCME) bzw Unity Express.
Folgendes passiert:
- Anruf über SIP von 124 an 83104
- Es geht niemand ran und das Telefon leitet auf die Voicemail um => SIP 302 Paket mit der Info [email protected] (Unity) anzurufen
- Asterisk nimmt die Nummer und ruft im Kontext from-sip an, also 2100@from-sip
Code:
voip1*CLI>
P[ 1] I IND :SETUP oad:124 dad:83104
P[ 1] read_config: Getting Config
P[ 1] config_jb: Called
P[ 1] CONTEXT:from-misdn
P[ 1] I SEND:PROCEEDING oad:124 dad:83104
-- Executing Macro("mISDN/1-2", "to-ccm|SIP|[email protected]") in new stack
-- Executing NoOp("mISDN/1-2", "Macro to-CCM gestartet") in new stack
-- Executing Dial("mISDN/1-2", "SIP/[email protected]|30") in new stack
-- Called [email protected]
-- SIP/10.1.1.67-087f3bb8 is ringing
P[ 1] * IND : Indication [3] from s
P[ 1] --> * IND : ringing pid:1014
P[ 1] I SEND:ALERTING oad:124 dad:83104
P[ 1] --> incoming_early_audio off
P[ 1] --> * SEND: State Ring pid:1014
-- Got SIP response 302 "Moved Temporarily" back from 10.1.1.67
-- Now forwarding mISDN/1-2 to 'Local/2100@from-sip' (thanks to SIP/10.1.1.67-087f3bb8)
-- Executing Dial("Local/2100@from-sip-486f,2", "SIP/[email protected]|20|rT") in new stack
-- Called [email protected]
-- Local/2100@from-sip-486f,1 is ringing
Jun 14 10:59:57 NOTICE[6398]: channel.c:1917 ast_read: Dropping incompatible voice frame on Local/2100@from-sip-486f,2 of format slin since our native format has changed to alaw
Jun 14 10:59:57 NOTICE[6398]: channel.c:1917 ast_read: Dropping incompatible voice frame on Local/2100@from-sip-486f,2 of format slin since our native format has changed to alaw
Jun 14 10:59:57 NOTICE[6398]: channel.c:1917 ast_read: Dropping incompatible voice frame on Local/2100@from-sip-486f,2 of format slin since our native format has changed to alaw
Jun 14 10:59:57 NOTICE[6398]: channel.c:1917 ast_read: Dropping incompatible voice frame on Local/2100@from-sip-486f,2 of format slin since our native format has changed to alaw
Jun 14 10:59:57 NOTICE[6398]: channel.c:1917 ast_read: Dropping incompatible voice frame on Local/2100@from-sip-486f,2 of format slin since our native format has changed to alaw
Jun 14 10:59:57 NOTICE[6398]: channel.c:1917 ast_read: Dropping incompatible voice frame on Local/2100@from-sip-486f,2 of format slin since our native format has changed to alaw
Jun 14 10:59:57 NOTICE[6398]: channel.c:1917 ast_read: Dropping incompatible voice frame on Local/2100@from-sip-486f,2 of format slin since our native format has changed to alaw
Jun 14 10:59:57 NOTICE[6398]: channel.c:1917 ast_read: Dropping incompatible voice frame on Local/2100@from-sip-486f,2 of format slin since our native format has changed to alaw
Jun 14 10:59:57 NOTICE[6398]: channel.c:1917 ast_read: Dropping incompatible voice frame on Local/2100@from-sip-486f,2 of format slin since our native format has changed to alaw
Jun 14 10:59:57 NOTICE[6398]: channel.c:1917 ast_read: Dropping incompatible voice frame on Local/2100@from-sip-486f,2 of format slin since our native format has changed to alaw
Jun 14 10:59:57 NOTICE[6398]: channel.c:1917 ast_read: Dropping incompatible voice frame on Local/2100@from-sip-486f,2 of format slin since our native format has changed to alaw
Jun 14 10:59:57 NOTICE[6398]: channel.c:1917 ast_read: Dropping incompatible voice frame on Local/2100@from-sip-486f,2 of format slin since our native format has changed to alaw
Jun 14 10:59:57 NOTICE[6398]: channel.c:1917 ast_read: Dropping incompatible voice frame on Local/2100@from-sip-486f,2 of format slin since our native format has changed to alaw
Jun 14 10:59:57 NOTICE[6398]: channel.c:1917 ast_read: Dropping incompatible voice frame on Local/2100@from-sip-486f,2 of format slin since our native format has changed to alaw
-- SIP/10.1.1.68-087f3bb8 is ringing
Jun 14 10:59:57 NOTICE[6398]: channel.c:1917 ast_read: Dropping incompatible voice frame on Local/2100@from-sip-486f,2 of format slin since our native format has changed to alaw
Jun 14 10:59:57 NOTICE[6398]: channel.c:1917 ast_read: Dropping incompatible voice frame on Local/2100@from-sip-486f,2 of format slin since our native format has changed to alaw
-- SIP/10.1.1.68-087f3bb8 answered Local/2100@from-sip-486f,2
-- Local/2100@from-sip-486f,1 stopped sounds
P[ 1] * IND : Indication [-1] from s
P[ 1] --> * IND : -1! (stop indication) pid:1014
P[ 1] Tone Indicate:
P[ 1] --> None
-- Local/2100@from-sip-486f,1 answered mISDN/1-2
P[ 1] * ANSWER:
P[ 1] I SEND:CONNECT oad:124 dad:83104
P[ 1] I IND :CONNECT_ACKNOWLEDGE oad:124 dad:83104
== Spawn extension (from-sip, 2100, 1) exited non-zero on 'Local/2100@from-sip-486f,2'
Jun 14 10:59:58 NOTICE[6368]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on c lient if possible. Client IP: 10.1.1.68
P[ 1] I IND :DISCONNECT oad:124 dad:83104
P[ 1] I SEND:RELEASE oad:124 dad:83104
P[ 1] I IND :RELEASE_COMPLETE oad:124 dad:83104
P[ 1] Trying to Release bc with l3id: 201bf
P[ 1] * RELEASING CHANNEL pid:1014 ctx:macro-to-ccm dad:s oad:124 state: CONNECTED
== Spawn extension (macro-to-ccm, s, 2) exited non-zero on 'mISDN/1-2' in macro 'to-ccm'
== Spawn extension (macro-to-ccm, s, 2) exited non-zero on 'mISDN/1-2'
P[ 1] I IND :CLEAN_UP oad: dad:
Das Telefon mit der 124 ist über MISDN angebunden (HiPath 4000), lasst euch davon nicht irritieren.
Ich verstehe nicht ganz warum Asterisk im Kontext from-sip anruft und nicht einfach den auf die IP Adresse anruft.
Hat das was damit zu tun das in der sip.conf ein context=from-sip steht?
In der aktuellen Konfiguration (s.u.) fange ich die 2100 ab und starte einen völlig neuen Anruf an [email protected], das entspricht natürlich nicht dem weitergeleiteten Anruf an die Voicebox.
Ist das ein Bug in Asterisk (Ignorieren der IP Information im SIP Paket) oder muss ich da noch etwas entsprechend konfigurieren?
Ich kann mir nur vorstellen das es eben etwas mit der Kontextdefinition in der sip.conf zu tun hat, habe aber keine Idee wie ich das umbauen müsste...
Vielen Dank schonmal fürs lesen
stephan
Noch die extensions.conf:
Code:
[general]
;
static=yes
writeprotect=yes
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
;#include "filename.conf"
[globals]
[ccm-sc]
exten => _82.,1,Macro(to-ccm,SIP,${EXTEN}@10.1.3.10)
[ccm-ma]
exten => _83.,1,Macro(to-ccm,SIP,${EXTEN}@10.1.1.67)
; uninteressant da falscher kontext:
;exten => 2100,1,Dial(SIP/[email protected])
;exten => 2100,2,Hangup
[ccm-ct]
exten => _84.,1,Macro(to-ccm,SIP,${EXTEN}@10.1.1.33)
[to-sip-phones]
; SNOM Testphone
[hipath-main]
;; This is the step between Asterisk and the Hipath.
;; If you want to intercept things like 85999 to do
;; an echo test with asterisk do this here.
; SIP phone for testing purposes
exten => 85999,1,Dial(SIP/599,40)
exten => 85999,2,Hangup
; echo test
exten => 85998,1,Playback(demo-echotest) ; Let them know what's going on
exten => 85998,n,Echo ; Do the echo test
exten => 85998,n,Playback(demo-echodone) ; Let them know it's over
exten => 85998,n,Goto(s,6) ; Start over
; forward anything else to the hipath and cut the first two digits (85124 => 124)
exten => _85.,1,Dial(misdn/1/${EXTEN:2},20)
exten => _85.,2,Hangup
[from-sip]
;; here goes anything coming from SIP phones and the Call Manager
include => to-sip-phones
include => ccm-sc
include => ccm-ma
include => ccm-ct
include => hipath-main
;; Abfangen von Voicebox Umleitungen in MA und CT (2100@from-sip)
;exten => 2100,1,Playback(were-sorry&were-sorry&were-sorry)
exten => 2100,1,Dial(SIP/[email protected],20,rT);
exten => 2100,2,Hangup
; other extensions with 4 digits
exten => _XXXX,1,Wait(1)
exten => _XXXX,2,Dial(misdn/2/${EXTEN},40)
exten => _XXXX,3,Hangup
; other extensions with 3 digits
exten => _XXX,1,Wait(1)
exten => _XXX,2,Dial(misdn/2/${EXTEN},40)
exten => _XXX,3,Hangup
exten => _00.,1,Wait(1)
exten => _00.,2,Dial(misdn/2/${EXTEN},60)
exten => _00.,3,Hangup
[from-misdn]
;; here goes everything coming from the HiPath
include => to-sip-phones
;; macro for sip
exten => 82999,1,Macro(to-sip,599)
; echotest from HiPath
exten => 82998,1,Playback(demo-echotest) ; Let them know what's going on
exten => 82998,n,Echo ; Do the echo test
exten => 82998,n,Playback(demo-echodone) ; Let them know it's over
exten => 82998,n,Goto(s,6) ; Start over
include => ccm-sc
include => ccm-ma
include => ccm-ct
exten => _85.,1,Dial(SIP/${EXTEN:2},60)
exten => _85.,2,Hangup
exten => _86.,1,Dial(SIP/599,10)
exten => _86.,2,Hangup
exten => _87.,1,Dial(SIP/599,10)
exten => _87.,2,Hangup
exten => _88.,1,Dial(SIP/599,10)
exten => _88.,2,Hangup
;3 Digit Extensions from HiPath mapping to SIP Phones
exten => _XXX,1,Macro(sipvm,SIP,${EXTEN})
;;;;;**********************
;;;;; macros
; ausgelassen!