Asterisk (ankommend) funktioniert in sipgate.de aber nicht in sipgate.at !?!?!?

strubinsky

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Meine Tochter lebt in der BRD und Meine Eltern in Österreich. Ich in den USA. Deshalb habe ich mire zwei sipgate accounts besorgt (sipgate.at und sipgate.de)

Das Misteriöse bei der Sache ist, daß die deutsche nummer super funktioniert, während es beim österreichischen nur ausgehend (von sipgate.at eine Wiener Nummer wählen) klappt. Eingehend sehe ich dass das telefon von asterisk abgehoben wird und die background cmds abgespielt werden. Aber es ist absolut nichts zu hören.

Ich habe die Musterkonfiguration von hier kopiert und: ja, das klappt. Es läutet und wenn ich abhebe kann ich sprechen (ich rufe meine sipgate nr via POTS von den usa aus an). Aber nix anderes geht. Keine voicemail, keine music on hold, keine background Sprachprompts. All das funktioniert aber wunderbar via sipgate.de.

Hier der relevante Teil meiner sip.conf:



[020142637436]
;register => 2637436:<pwde>@sipgate.de/020142637436
type=friend
context=incoming-voip
username=2637436
fromuser=2637436
authuser=2637436
disable=all
allow=alaw
allow=ulaw
allow=g729
allow=gsm
allow=slinear
;dtmfmode=inband ; Choices are inband, rfc2833, or info
;dtmf=rfc2833
insecure=very ; otherwise I get authentication errors
nat=yes
fromdomain=sipgate.de
secret=<pwde>
host=sipgate.de
qualify=yes
;callerid="guenter strubinsky" <020142637436>
mailbox=3484@default
srvlookup=yes
canreinvite=no

[19628879]
type=friend
;register => 9628879:<pwat>@sipgate.at/9628879
context=incoming-voip
username=9628879
fromuser=9628879
authuser=9628879
disable=all
allow=alaw
allow=ulaw
allow=g729
allow=gsm
allow=slinear
;dtmfmode=inband ; Choices are inband, rfc2833, or info
;dtmf=rfc2833
insecure=very ; otherwise I get authentication errors
nat=yes
fromdomain=sipgate.at
secret=<pwat>
host=sipgate.at
qualify=yes
;callerid="guenter strubinsky" <19628879>
mailbox=3484@default
srvlookup=yes
canreinvite=no



und die extensions.conf:

...

;----------------------------------------------------------------------------------------
[macro-stdexten]
;----------------------------------------------------------------------------------------
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;

exten => s,1,NoOp("stdXt (1) ${ARG1}")
exten => s,n,NoOp("stdXt (2) ${CALLERIDNUM}")

exten => s,n,GotoIf($[${ARG1} = ${guenti}]?callGuenti)

exten => s,n,GotoIf($[${LEN(${CALLERIDNUM})} < 7]?internalCall:externalCall)

exten => s,n(internalCall),GotoIf($[${LEN(${ARG1})} < 7]?callInside:callOutside)

exten => s,n(callOutside),NoOp(callOutside)
exten => s,n,Playtones(ring)
exten => s,n,SetCallerId("Strubinsky guenter" <402-403-3113>,a)

exten => s,n,ChanIsAvail(${outLine1})
exten => s,n,NoOp(${AVAILORIGCHAN})
exten => s,n,Dial(${AVAILORIGCHAN}/${ARG1},120,${insecwarn})
;exten => s,n,Dial(IAX2/0611932438@14028172455/${ARG1},120,A(channel-insecure-warn))
exten => s,n,Goto(s-${DIALSTATUS},1) ; Jump based on status
; (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s,n(callInside),NoOp(callInside)
exten => s,n,Dial(SIP/${ARG1},20,o)
exten => s,n,Goto(s-${DIALSTATUS},1) ; Jump based on status
; (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s,n(externalCall),NoOp(External Call)
exten => s,n,Set(LastCalled=${DB(callers/${CALLERIDNUM}/lastcall)})
exten => s,n,Background(channel-insecure-warn)
exten => s,n,SetMusicOnHold(default)
exten => s,n,Dial(SIP/${ARG1},20,om)
exten => s,n,Goto(s-${DIALSTATUS},1) ; Jump based on status
; (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s,n(callGuenti),Macro(guenticall,${ARG1},${ARG2})
exten => s,n,hangup()

exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(dir,411,1) ; If they press #, return to start

exten => s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(dir,411,1) ; If they press #, return to start

exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer

exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain

;----------------------------------------------------------------------------------------
[gotExtension]
;----------------------------------------------------------------------------------------
exten => 1,1,NoOp(${CALLERIDNUM}) ; SayNumber(${LEN(${CALLERIDNUM})})
exten => 1,2,GotoIf($[${LEN(${CALLERIDNUM})} < 7]?1,8)
exten => 1,n,Background(silence/1) ; Wait a second, just for fun
exten => 1,n,Set(DB(callers/${CALLERIDNUM}/lastcall)=${toDial})
exten => 1,n,Background(extension)
exten => 1,n,SayNumber(${toDial})
exten => 1,n,Background(transfer)
exten => 1,n,Macro(stdexten,${toDial},${${dialName}})

exten => #,1,Voicemail(b${toDial})


...

;----------------------------------------------------------------------------------------
[incoming]
;----------------------------------------------------------------------------------------
exten => 1,1,Answer()
exten => 1,2,NoOp(incoming=(${CALLERIDNUM}))
exten => 1,3,Background(silence/1)
exten => 1,4,Set(toDial=411${DB(callers/${CALLERIDNUM}/lastcall)})
exten => 1,5,GotoIf($[${toDial}=411]?1,wtr0:)
;exten => 1,6,Set(toDial=${toDial:3})
exten => 1,6,Set(toDial=3536)
exten => 1,7,Background(the-num-i-have-is)
exten => 1,8,System(/etc/asterisk/filexists /var/spool/asterisk/voicemail/default/${toDial}/greet.gsm) ; test for the existance of this path as a recording
exten => 1,n,Background(/var/spool/asterisk/voicemail/default/${toDial}/greet)
exten => 1,n(aftername),Background(to-call-num-press)
exten => 1,n,Background(pound)
exten => 1,n,Wait(1)
exten => 1,n,Goto(repeat)

exten => 1,109,SayNumber(${toDial})
exten => 1,n,Goto(aftername)
sip.conf:

exten => 1,n(wtr0),Set(toDial=411)
exten => 1,n(wtr),Background(thanks-for-calling-today)
exten => 1,n(repeat),Background(if-u-know-ext-dial)
exten => 1,n,Background(to-dial-by-name)
exten => 1,n,Background(press-9)
exten => 1,n,Background(vm-reachoper)
exten => 1,n,Background(star-for-menu-again)
exten => 1,n(reentry),NoOp()
;----------------------------------------------------------------------------------------
;[singleInput]
;----------------------------------------------------------------------------------------
exten => 1,n(singleInput),NoOp(Single input called)
exten => 1,n,WaitExten()
exten => 1,n,Background(one-moment-please)
exten => 1,n,Set(chosen=0)
;exten => 1,n,Goto(dir,411,1)
exten => 1,n,goto(1,repeat)

exten => i,1,Set(chosen=0)
exten => i,n,background(i-dont-understand)
exten => i,n,background(please-try-again)
exten => i,n,goto(1,repeat)

exten => _0!,1,NoOp(0)
exten => _0!,n,Set(chosen=${EXTEN})
exten => _0!,n,goto(connect,oper) ; hangup

exten => _#!,1,NoOp(pound)
exten => _#!,n,Set(chosen=${EXTEN})
exten => _#!,n,goto(chosen,1) ; hangup

exten => _9!,1,NoOp(9)
exten => _9!,n,Set(chosen=${EXTEN})
exten => _9!,n,goto(dir,411,1) ; hangup

exten => _*!,1,NoOp(star)
exten => _*!,n,Set(chosen=${EXTEN})
exten => _*!,n,goto(1,1) ; hangup

exten => n,1,sayphonetic(n)
exten => n,n,Set(chosen=${EXTEN})
exten => n,n,goto(chosen,1) ; hangup

exten => a,1,NoOp(a) ; * pressed
exten => a,n,Set(chosen=${EXTEN})
exten => a,n,goto(chosen,1) ; hangup

exten => o,1,NoOp(o) ; operator / 0
exten => o,n,Set(chosen=${EXTEN})
exten => o,n,goto(chosen,1) ; hangup

exten => t,1,NoOp(t) ; Invalid
exten => t,n,Set(chosen=${EXTEN})
exten => t,n,goto(chosen,1) ; hangup

exten => h,1,NoOp(h) ; hangup
exten => h,n,background(i-dont-understand)
exten => h,n,background(please-try-again)
exten => h,n,goto(chosen,1) ; hangup

exten => s,1,NoOp(s) ; start
exten => s,n,goto(chosen,1) ; hangup

exten => _3XXX!,1,Set(toDial=${EXTEN})
exten => _3XXX!,n,Goto(gotExtension,1,1)

exten => _8XXX!,1,Goto(default,${EXTEN},1)

exten => chosen,1,NoOp(${chosen})
exten => chosen,n,GotoIf(1,$[${chosen}="*"]?1,repeat)
exten => chosen,n,GotoIf(1,$[${chosen}="9"]?connect,411)
exten => chosen,n,GotoIf(1,$[${chosen}="0"]?connect,oper)
exten => chosen,n,GotoIf(1,$[${chosen}="#"]?gotExtension,1,1)
exten => chosen,n,goto(1,repeat)

exten => connect,1,NoOp(connect entered)
exten => connect,n(411),Goto(dir,411,1)
exten => connect,n(oper),Set(toDial=${oper})
exten => connect,n,Goto(gotExtension,1,1)

;----------------------------------------------------------------------------------------
[incoming-voip]
;----------------------------------------------------------------------------------------

exten => _1XXXXXXXXXX!,1,Set(LANGUAGE()=en)
;exten => _1XXXXXXXXXX!,n,Answer()
;exten => _1XXXXXXXXXX!,n,gotoIf($[${EXTEN} = "4022926801"]?susans,1,1)
exten => _1XXXXXXXXXX!,n,goto(incoming,1,1)
exten => _X.,1,Set(LANGUAGE()=de)
;exten => _X.,n,Answer()
exten => _X.,n,Background(international-call)
exten => _X.,n,goto(incoming,1,1)



wenn es nirgends klappen würde, dann hätte ich einfach ein parr Fehler. Die kann ich ja debuggen.

ABER;
Es funktioniert wenn ich
- lokal (us ortsgespräch - (ankommend/ausgehend),
- us ferngespräch - (ankommend/ausgehend),
- brd orts und ferngespräch - (ankommend/ausgehend)
- at orts und ferngespräch - (AUSGEHEND)

habe, nur nicht wenn ich AT eingehend habe. Da -wie oben erwähnt- kann ich sehen, wie die Backgroundansagen abgesspielt werden (höre aber nix), die Wartemusik gestarted wird (höre aber nix).

Wenn wer interessiert ist, kann ich die notwendigen files komplett zippen und attachen. Hat jemand ähnlicher Erfahrungen gemacht?

Ich habe sogar mit Ethereal die RTP blöcke verfolgt. Alles scheint durchzugehen!!! Die einzige Erklärung die ich habe, ist daß die falschen Codecs für den sound verwendet werden (sipgate anderer als asterisk-sipura spa 2100), aber auch diese habe ich geprüft und es scheint identisch zu sein.

Ich bin am Ende mit meinem Latein!!!! :noidea: :noidea:
Kann mir wer helfen?

günter

p.s. Um es einfacher zu machen:
sip.conf:
[020142637436]
register => 9628879:<pwat>@sipgate.at/9628879
type=friend
;context=incoming-voip
context=ankommend
...

extensions.conf
[ankommend]
exten => 9628879,1,Answer()
exten => 9628879,n,Background(international-call)
exten => 9628879,n,Dial(SIP/3536,20)

Absolut dasselbe Verhalten(ankommend):
sipfgate.de :rock:
sipgate.at :mad:
 
Zuletzt bearbeitet:
Dein Problem ist ganze einfach erklärt :D

Code:
register => 9628879:<pwat>@sipgate.at/9628879

register Einträge dürfen niemals in einem anderen Kontext stehen, als in [general] :!: Das dürfte Dein Hauptfehler sein. Du darfst sie nicht in die einzelnen Kontexte der Accounts schreiben. Und "type=friend" ist in aktuellen Asterisk-Versionen auch problematisch.

Übrigens gibt es hier im Forum auch eine Musterkonfiguration für Sipgate-Accounts auf Asterisk.
 
betateilchen schrieb:
Dein Problem ist ganze einfach erklärt :D

Code:
register => 9628879:<pwat>@sipgate.at/9628879

register Einträge dürfen niemals in einem anderen Kontext stehen, als in [general] :!: Das dürfte Dein Hauptfehler sein. Du darfst sie nicht in die einzelnen Kontexte der Accounts schreiben. Und "type=friend" ist in aktuellen Asterisk-Versionen auch problematisch.

Übrigens gibt es hier im Forum auch eine Musterkonfiguration für Sipgate-Accounts auf Asterisk.

Tut mir Leid für das Mißverständnis ß-Teilchen; der "Register" Teil ist in [general] . Ich habe es lediglich (als Kommentar) unter der ID stehen.

Ich dachte, friend ist rein-raus, user ist raus and peer ist rein.

Okay, ich hab's gefunden, man sollte rein und raus separat behandeln!

Ich habe es so gemacht, wie Dein Template und bin zu folgendem Resultat gekommen:

Code:
[9628879]
type=user
username=9628879
fromuser=9628879
secret=<pwat>
host=sipgate.at
fromdomain=sipgate.at
insecure=very
canreinvite=no
nat=no
disallow=all
allow=ulaw

[sipgate_at_in]
type=peer
fromdomain=sipgate.at
;context=incoming-voip
host=sipgate.at
context=ankommend

Version 1:
Code:
[ankommend]
exten => 9628879,1,Answer()
exten => 9628879,n,Background(international-call)
exten => 9628879,n,Dial(SIP/3536,20)

Resultat:
  • Ich sehe, daß das telefon beantwortet wird (höre aber nichts am anrufenden Aparat -Schweigen-)
  • Ich sehe, daß die international-call Meldung backgrounded wird (höre aber nichts am anrufenden Aparat -Schweigen-)
  • Ich sehe den Dial Befehl
  • Nebenstelle 3536 läutet
  • ich hebe ab und höre immer noch nichts, (an beiden Seiten) auch wenn ich rede, sowohl am ausgehenden wie ankommenden Apparat.
  • Die Leitung wird mit einem TimeOut automatisch getrennt.

Protokoll:
Code:
<-- SIP read from 8.11.1.5:5060:
OPTIONS sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 8.11.1.5:5060;branch=z9hG4bK068525a0;rport
From: "asterisk" <sip:[email protected]>;tag=as462e8898
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Viatalk SIP
Max-Forwards: 70
Date: Thu, 23 Feb 2006 16:18:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
strubinsky*CLI>

--- (12 headers 0 lines)---
Looking for 14024033113 in default (domain 68.227.169.158)
Transmitting (no NAT) to 8.11.1.5:5060:<-- SIP read from 217.116.119.252:5060:
INVITE sip:[email protected] SIP/2.0
Record-Route: <sip:[email protected];ftag=as1b712083;lr=on>
Max-Forwards:  9
Record-Route: <sip:[email protected];ftag=as1b712083;lr=on>
Via: SIP/2.0/UDP 217.116.119.252;branch=z9hG4bK548c.55118bd6.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK548c.75e12503.0
Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK78d44b8d
From: "0414022926801" <sip:[email protected]>;tag=as1b712083
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: sipgate asterisk
Date: Thu, 23 Feb 2006 16:39:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 411

v=0
o=root 32613 32613 IN IP4 217.10.66.71
s=session
c=IN IP4 217.10.66.71
t=0 0
m=audio 12290 RTP/AVP 8 0 3 97 18 2 4 5 110 7 10
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:110 speex/8000
a=rtpmap:7 LPC/8000
a=rtpmap:10 L16/8000
a=silenceSupp:off - - - -

--- (17 headers 18 lines)---
Using INVITE request as basis request - [email protected]
Sending to 217.116.119.252 : 5060 (non-NAT)
Found peer 'sipgate_at_in'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 5
Found RTP audio format 110
Found RTP audio format 7
Found RTP audio format 10
Peer audio RTP is at port 217.10.66.71:12290
Found description format PCMA
Found description format PCMU
Found description format GSM
Found description format iLBC
Found description format G729
Found description format G726-32
Found description format G723
Found description format DVI4
Found description format speex
Found description format LPC
Found description format L16
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x7ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Looking for 9628879 in ankommend (domain 68.227.169.158)
list_route: hop: <sip:[email protected];ftag=as1b712083;lr=on>
list_route: hop: <sip:[email protected];ftag=as1b712083;lr=on>
Transmitting (no NAT) to 217.116.119.252:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.116.119.252;branch=z9hG4bK548c.55118bd6.1;received=217.116.119.252
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK548c.75e12503.0
Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK78d44b8d
From: "0414022926801" <sip:[email protected]>;tag=as1b712083
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0


---
    -- Executing Set("SIP/217.10.66.71-09d10448", "apparat=3536") in new stack
    -- Executing Dial("SIP/217.10.66.71-09d10448", "SIP/3536|20") in new stack
We're at 192.168.11.69 port 18882
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
14 headers, 16 lines
Reliably Transmitting (no NAT) to 192.168.11.202:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.69:5060;branch=z9hG4bK2da50bf2;rport
From: "0414022926801" <sip:[email protected]>;tag=as51f3487a
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "0414022926801" <sip:[email protected]>;privacy=off;screen=no
Date: Thu, 23 Feb 2006 16:39:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 367

v=0
o=root 2558 2558 IN IP4 192.168.11.69
s=session
c=IN IP4 192.168.11.69
t=0 0
m=audio 18882 RTP/AVP 0 3 8 111 18 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Called 3536
strubinsky*CLI>
<-- SIP read from 192.168.11.202:5060:
SIP/2.0 100 Trying
To: <sip:[email protected]:5060>
From: "0414022926801" <sip:[email protected]>;tag=as51f3487a
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.11.69:5060;branch=z9hG4bK2da50bf2
Server: Sipura/SPA2100-2.0.5(d)
Content-Length: 0


--- (8 headers 0 lines)---

<-- SIP read from 192.168.11.202:5060:
SIP/2.0 180 Ringing
To: <sip:[email protected]:5060>;tag=ba310c92db31bcc1i0
rom: "0414022926801" <sip:[email protected]>;tag=as51f3487a
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.11.69:5060;branch=z9hG4bK2da50bf2
Server: Sipura/SPA2100-2.0.5(d)
Content-Length: 0


--- (8 headers 0 lines)---
    -- SIP/3536-443b is ringing
Transmitting (no NAT) to 217.116.119.252:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 217.116.119.252;branch=z9hG4bK548c.55118bd6.1;received=217.116.119.252
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK548c.75e12503.0
Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK78d44b8d
From: "0414022926801" <sip:[email protected]>;tag=as1b712083
To: <sip:[email protected]>;tag=as218b8918
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0


---
strubinsky*CLI>
<-- SIP read from 192.168.11.202:5060:
SIP/2.0 200 OK
To: <sip:[email protected]:5060>;tag=ba310c92db31bcc1i0
From: "0414022926801" <sip:[email protected]>;tag=as51f3487a
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.11.69:5060;branch=z9hG4bK2da50bf2
Contact: 3536 <sip:[email protected]:5060>
Server: Sipura/SPA2100-2.0.5(d)
Content-Length: 241
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 19758508 19758508 IN IP4 192.168.11.202
s=-
c=IN IP4 192.168.11.202
t=0 0
m=audio 16386 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

--- (12 headers 12 lines)---
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.11.202:16386
Found description format PCMU
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x51e (gsm|ulaw|alaw|g726|g729|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:[email protected]:5060>
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 192.168.11.202, port 5060
Transmitting (no NAT) to 192.168.11.202:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.69:5060;branch=z9hG4bK50c9bee7;rport
From: "0414022926801" <sip:[email protected]>;tag=as51f3487a
To: <sip:[email protected]:5060>;tag=ba310c92db31bcc1i0
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "0414022926801" <sip:[email protected]>;privacy=off;screen=no
Content-Length: 0


---
    -- SIP/3536-443b answered SIP/217.10.66.71-09d10448
We're at 68.227.169.158 port 15630
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Reliably Transmitting (no NAT) to 217.116.119.252:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.116.119.252;branch=z9hG4bK548c.55118bd6.1;received=217.116.119.252
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK548c.75e12503.0
Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK78d44b8d
Record-Route: <sip:[email protected];ftag=as1b712083;lr=on>
Record-Route: <sip:[email protected];ftag=as1b712083;lr=on>
From: "0414022926801" <sip:[email protected]>;tag=as1b712083
To: <sip:[email protected]>;tag=as218b8918
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 209

v=0
o=root 2558 2558 IN IP4 68.227.169.158
s=session
c=IN IP4 68.227.169.158
t=0 0
m=audio 15630 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -

---
    -- Attempting native bridge of SIP/217.10.66.71-09d10448 and SIP/3536-443b
strubinsky*CLI>
<-- SIP read from 217.116.119.252:5060:
ACK sip:[email protected] SIP/2.0
Record-Route: <sip:[email protected];ftag=as1b712083;lr=on>
Max-Forwards:  9
Via: SIP/2.0/UDP 217.116.119.252;branch=0
Via: SIP/2.0/UDP 217.10.79.8;branch=0
Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK11696cd9
From: "0414022926801" <sip:[email protected]>;tag=as1b712083
To: <sip:[email protected]>;tag=as218b8918
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: sipgate asterisk
Content-Length: 0


--- (13 headers 0 lines)---
Destroying call '[email protected]'
Destroying call '[email protected]'
Destroying call '[email protected]'
strubinsky*CLI>
<-- SIP read from 192.168.11.202:5060:
BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.11.202:5060;branch=z9hG4bK-cf15f509
From: <sip:[email protected]:5060>;tag=ba310c92db31bcc1i0
To: "0414022926801" <sip:[email protected]>;tag=as51f3487a
Call-ID: [email protected]
CSeq: 101 BYE
Max-Forwards: 70
User-Agent: Sipura/SPA2100-2.0.5(d)
Content-Length: 0

SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 8.11.1.5:5060;branch=z9hG4bK068525a0;rport;received=8.11.1.5
From: "asterisk" <sip:[email protected]>;tag=as462e8898
To: <sip:[email protected]>;tag=as39e1815e
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:68.227.169.158>
Accept: application/sdp
Content-Length: 0


---
Destroying call '[email protected]'
strubinsky*CLI>
<-- SIP read from 217.116.119.252:5060:
INVITE sip:[email protected] SIP/2.0
Record-Route: <sip:[email protected];ftag=as2590f98a;lr=on>
Max-Forwards:  9
Record-Route: <sip:[email protected];ftag=as2590f98a;lr=on>
Via: SIP/2.0/UDP 217.116.119.252;branch=z9hG4bK84d8.8e344746.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK84d8.e9167642.0
Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK1d285581
From: "043" <sip:[email protected]>;tag=as2590f98a
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: sipgate asterisk
Date: Thu, 23 Feb 2006 16:31:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 411

v=0
o=root 31839 31839 IN IP4 217.10.66.71
s=session
c=IN IP4 217.10.66.71
t=0 0
m=audio 10408 RTP/AVP 8 0 3 97 18 2 4 5 110 7 10
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:110 speex/8000
a=rtpmap:7 LPC/8000
a=rtpmap:10 L16/8000
a=silenceSupp:off - - - -

--- (17 headers 18 lines)---
Using INVITE request as basis request - [email protected]
Sending to 217.116.119.252 : 5060 (non-NAT)
Found peer 'sipgate_at_in'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 5
Found RTP audio format 110
Found RTP audio format 7
Found RTP audio format 10
Peer audio RTP is at port 217.10.66.71:10408
Found description format PCMA
Found description format PCMU
Found description format GSM
Found description format iLBC
Found description format G729
Found description format G726-32
Found description format G723
Found description format DVI4
Found description format speex
Found description format LPC
Found description format L16
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x7ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Looking for 9628879 in ankommend (domain 68.227.169.158)
list_route: hop: <sip:[email protected];ftag=as2590f98a;lr=on>
list_route: hop: <sip:[email protected];ftag=as2590f98a;lr=on>
Transmitting (no NAT) to 217.116.119.252:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.116.119.252;branch=z9hG4bK84d8.8e344746.1;received=217.116.119.252
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK84d8.e9167642.0
Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK1d285581
From: "043" <sip:[email protected]>;tag=as2590f98a
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0


---
    -- Executing Answer("SIP/217.10.66.71-09d159b8", "") in new stack
We're at 68.227.169.158 port 18038
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Reliably Transmitting (no NAT) to 217.116.119.252:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.116.119.252;branch=z9hG4bK84d8.8e344746.1;received=217.116.119.252
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK84d8.e9167642.0
Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK1d285581
Record-Route: <sip:[email protected];ftag=as2590f98a;lr=on>
Record-Route: <sip:[email protected];ftag=as2590f98a;lr=on>
From: "043" <sip:[email protected]>;tag=as2590f98a
To: <sip:[email protected]>;tag=as281a85bc
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 209

v=0
o=root 2558 2558 IN IP4 68.227.169.158
s=session
c=IN IP4 68.227.169.158
t=0 0
m=audio 18038 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -

---
Feb 23 10:31:14 WARNING[12593]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x9d00bf0', 10 retries!
    -- Executing BackGround("SIP/217.10.66.71-09d159b8", "international-call") in new stack
    -- Playing 'international-call' (language 'en')
strubinsky*CLI>
<-- SIP read from 217.116.119.252:5060:
ACK sip:[email protected] SIP/2.0
Record-Route: <sip:[email protected];ftag=as2590f98a;lr=on>
Max-Forwards:  9
Via: SIP/2.0/UDP 217.116.119.252;branch=0
Via: SIP/2.0/UDP 217.10.79.8;branch=0
Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK135ce12e
From: "043" <sip:[email protected]>;tag=as2590f98a
To: <sip:[email protected]>;tag=as281a85bc
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: sipgate asterisk
Content-Length: 0


--- (13 headers 0 lines)---
    -- Executing Dial("SIP/217.10.66.71-09d159b8", "SIP/3536|20") in new stack
We're at 192.168.11.69 port 10884
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
14 headers, 16 lines
Reliably Transmitting (no NAT) to 192.168.11.202:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.69:5060;branch=z9hG4bK4d9a92cd;rport
From: "043" <sip:[email protected]>;tag=as298b4c04
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "043" <sip:[email protected]>;privacy=off;screen=no
Date: Thu, 23 Feb 2006 16:31:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 367

v=0
o=root 2558 2558 IN IP4 192.168.11.69
s=session
c=IN IP4 192.168.11.69
t=0 0
m=audio 10884 RTP/AVP 0 3 8 111 18 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Called 3536
strubinsky*CLI>
<-- SIP read from 192.168.11.202:5060:
SIP/2.0 100 Trying
To: <sip:[email protected]:5060>
From: "043" <sip:[email protected]>;tag=as298b4c04
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.11.69:5060;branch=z9hG4bK4d9a92cd
Server: Sipura/SPA2100-2.0.5(d)
Content-Length: 0


--- (8 headers 0 lines)---

<-- SIP read from 192.168.11.202:5060:
SIP/2.0 180 Ringing
To: <sip:[email protected]:5060>;tag=fab189c59911f0e4i0
From: "043" <sip:[email protected]>;tag=as298b4c04
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.11.69:5060;branch=z9hG4bK4d9a92cd
Server: Sipura/SPA2100-2.0.5(d)
Content-Length: 0


--- (8 headers 0 lines)---
    -- SIP/3536-b3bd is ringing
Destroying call '[email protected]'
Destroying call '[email protected]'
Destroying call '[email protected]'
Destroying call '[email protected]'
Destroying call '[email protected]'
Destroying call '[email protected]'
strubinsky*CLI>

ANDERS hingegen:
Code:
[ankommend]
;exten => 9628879,1,Answer()
;exten => 9628879,n,Background(international-call)
exten => 9628879,1,Set(apparat=3536)
exten => 9628879,n,Dial(SIP/${apparat},20)
exten => 9628879,n,Goto(1-${DIALSTATUS},1)			; Jump based on status 
;									  (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => 1-NOANSWER,1,Voicemail(u${${apparat}})		; If unavailable, send to voicemail w/ unavail announce

Das Telefon läutet (im Gegensatz zu oben wo nichtsa zu hören ist kann ich am anrufenden Apparat das Freizeichen hören), ich kann auf beiden Seiten sprechen und hören. Wenn ich nicht abhebe dann kickt NICHT meine lokale Voicemail ein sondern die von Sipgate.

Code:
Bitte beachte das ALLES in Sipgate.de funktioniert !!!!!!
Protokoll (bei abheben):
Code:
<-- SIP read from 217.116.119.252:5060:
INVITE sip:[email protected] SIP/2.0
Record-Route: <sip:[email protected];ftag=as1b712083;lr=on>
Max-Forwards:  9
Record-Route: <sip:[email protected];ftag=as1b712083;lr=on>
Via: SIP/2.0/UDP 217.116.119.252;branch=z9hG4bK548c.55118bd6.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK548c.75e12503.0
Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK78d44b8d
From: "0414022926801" <sip:[email protected]>;tag=as1b712083
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: sipgate asterisk
Date: Thu, 23 Feb 2006 16:39:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 411

v=0
o=root 32613 32613 IN IP4 217.10.66.71
s=session
c=IN IP4 217.10.66.71
t=0 0
m=audio 12290 RTP/AVP 8 0 3 97 18 2 4 5 110 7 10
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:110 speex/8000
a=rtpmap:7 LPC/8000
a=rtpmap:10 L16/8000
a=silenceSupp:off - - - -

--- (17 headers 18 lines)---
Using INVITE request as basis request - [email protected]
Sending to 217.116.119.252 : 5060 (non-NAT)
Found peer 'sipgate_at_in'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 5
Found RTP audio format 110
Found RTP audio format 7
Found RTP audio format 10
Peer audio RTP is at port 217.10.66.71:12290
Found description format PCMA
Found description format PCMU
Found description format GSM
Found description format iLBC
Found description format G729
Found description format G726-32
Found description format G723
Found description format DVI4
Found description format speex
Found description format LPC
Found description format L16
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x7ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Looking for 9628879 in ankommend (domain 68.227.169.158)
list_route: hop: <sip:[email protected];ftag=as1b712083;lr=on>
list_route: hop: <sip:[email protected];ftag=as1b712083;lr=on>
Transmitting (no NAT) to 217.116.119.252:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.116.119.252;branch=z9hG4bK548c.55118bd6.1;received=217.116.119.252
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK548c.75e12503.0
Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK78d44b8d
From: "0414022926801" <sip:[email protected]>;tag=as1b712083
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0


---
    -- Executing Set("SIP/217.10.66.71-09d10448", "apparat=3536") in new stack
    -- Executing Dial("SIP/217.10.66.71-09d10448", "SIP/3536|20") in new stack
We're at 192.168.11.69 port 18882
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
14 headers, 16 lines
Reliably Transmitting (no NAT) to 192.168.11.202:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.69:5060;branch=z9hG4bK2da50bf2;rport
From: "0414022926801" <sip:[email protected]>;tag=as51f3487a
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "0414022926801" <sip:[email protected]>;privacy=off;screen=no
Date: Thu, 23 Feb 2006 16:39:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 367

v=0
o=root 2558 2558 IN IP4 192.168.11.69
s=session
c=IN IP4 192.168.11.69
t=0 0
m=audio 18882 RTP/AVP 0 3 8 111 18 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Called 3536
strubinsky*CLI>
<-- SIP read from 192.168.11.202:5060:
SIP/2.0 100 Trying
To: <sip:[email protected]:5060>
From: "0414022926801" <sip:[email protected]>;tag=as51f3487a
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.11.69:5060;branch=z9hG4bK2da50bf2
Server: Sipura/SPA2100-2.0.5(d)
Content-Length: 0


--- (8 headers 0 lines)---

<-- SIP read from 192.168.11.202:5060:
SIP/2.0 180 Ringing
To: <sip:[email protected]:5060>;tag=ba310c92db31bcc1i0
rom: "0414022926801" <sip:[email protected]>;tag=as51f3487a
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.11.69:5060;branch=z9hG4bK2da50bf2
Server: Sipura/SPA2100-2.0.5(d)
Content-Length: 0


--- (8 headers 0 lines)---
    -- SIP/3536-443b is ringing
Transmitting (no NAT) to 217.116.119.252:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 217.116.119.252;branch=z9hG4bK548c.55118bd6.1;received=217.116.119.252
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK548c.75e12503.0
Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK78d44b8d
From: "0414022926801" <sip:[email protected]>;tag=as1b712083
To: <sip:[email protected]>;tag=as218b8918
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0


---
strubinsky*CLI>
<-- SIP read from 192.168.11.202:5060:
SIP/2.0 200 OK
To: <sip:[email protected]:5060>;tag=ba310c92db31bcc1i0
From: "0414022926801" <sip:[email protected]>;tag=as51f3487a
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.11.69:5060;branch=z9hG4bK2da50bf2
Contact: 3536 <sip:[email protected]:5060>
Server: Sipura/SPA2100-2.0.5(d)
Content-Length: 241
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 19758508 19758508 IN IP4 192.168.11.202
s=-
c=IN IP4 192.168.11.202
t=0 0
m=audio 16386 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

--- (12 headers 12 lines)---
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.11.202:16386
Found description format PCMU
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x51e (gsm|ulaw|alaw|g726|g729|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:[email protected]:5060>
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 192.168.11.202, port 5060
Transmitting (no NAT) to 192.168.11.202:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.69:5060;branch=z9hG4bK50c9bee7;rport
From: "0414022926801" <sip:[email protected]>;tag=as51f3487a
To: <sip:[email protected]:5060>;tag=ba310c92db31bcc1i0
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "0414022926801" <sip:[email protected]>;privacy=off;screen=no
Content-Length: 0


---
    -- SIP/3536-443b answered SIP/217.10.66.71-09d10448
We're at 68.227.169.158 port 15630
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Reliably Transmitting (no NAT) to 217.116.119.252:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.116.119.252;branch=z9hG4bK548c.55118bd6.1;received=217.116.119.252
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK548c.75e12503.0
Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK78d44b8d
Record-Route: <sip:[email protected];ftag=as1b712083;lr=on>
Record-Route: <sip:[email protected];ftag=as1b712083;lr=on>
From: "0414022926801" <sip:[email protected]>;tag=as1b712083
To: <sip:[email protected]>;tag=as218b8918
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 209

v=0
o=root 2558 2558 IN IP4 68.227.169.158
s=session
c=IN IP4 68.227.169.158
t=0 0
m=audio 15630 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -

---
    -- Attempting native bridge of SIP/217.10.66.71-09d10448 and SIP/3536-443b
strubinsky*CLI>
<-- SIP read from 217.116.119.252:5060:
ACK sip:[email protected] SIP/2.0
Record-Route: <sip:[email protected];ftag=as1b712083;lr=on>
Max-Forwards:  9
Via: SIP/2.0/UDP 217.116.119.252;branch=0
Via: SIP/2.0/UDP 217.10.79.8;branch=0
Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK11696cd9
From: "0414022926801" <sip:[email protected]>;tag=as1b712083
To: <sip:[email protected]>;tag=as218b8918
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: sipgate asterisk
Content-Length: 0


--- (13 headers 0 lines)---
Destroying call '[email protected]'
Destroying call '[email protected]'
Destroying call '[email protected]'
strubinsky*CLI>
<-- SIP read from 192.168.11.202:5060:
BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.11.202:5060;branch=z9hG4bK-cf15f509
From: <sip:[email protected]:5060>;tag=ba310c92db31bcc1i0
To: "0414022926801" <sip:[email protected]>;tag=as51f3487a
Call-ID: [email protected]
CSeq: 101 BYE
Max-Forwards: 70
User-Agent: Sipura/SPA2100-2.0.5(d)
Content-Length: 0

Vielen Dank für Deine Mühe!

günter
 
Aaaahhhh!!!

Ich habe die Situation analysiert und bin zu diesem Schluß gekommen:

Wenn ich Diale ohne sound vorher zu haben, dann läutet das telefon und ich kann auf beiden Seiten sprechen und hören, das Freizeichen wird vom anrufenden apparat gehört

Wenn Play, background, Musiconhold vor dem Dial kommen wird im trace zwar der jeweilige command abgespielt, aber es gibt keinen Ton (kein Freizeichen, absolut kein Ton). Kommt danach ein Dial, so läutet zwar das telefon aber wenn ich abhebe, dann kann weiterhin kein Ton gehört werden und der Dial Cmd timed out als ob ich gar nicht abgehoben hätte.

Das sieht mir ganz so aus, als ob die beiden codecs nicht miteinander kompatibel sind.

Sowohl mein Cisco 7960 als auch das Softphone funktionieren, was nicht so überraschend ist, da diese ja keinen Sound vor dem abheben machen. Das abheben scheint die richtigen codecs zu starten und zu synchronisieren.

Kann wer mit einem sipgate.at account den Versuch machen:

Code:
exten => <Deine#>,1,Set(LANGUAGE()=de)
exten => <Deine#>,n,Background(international-call)
exten => <Deine#>,n,Dial(<Nebenstelle>)

... und rausfinden, ob "international call" zu hören ist. ( international-call.gsm ist ein standard soundfile in /asterisk-sounds-1.2.1/sounds/ =. Wenn sich das Verhaslten duplizieren läßt, dann haben die Österreicher was verbockt. Wenn nicht, dann muß ich weitersuchen (obwohl ich nicht mehr weiter weiß als in die C source zu gehen, was unglaublich aufwendig und zeitraubend ist; außer hier weiß jemand einen anderen Weg den ich versuchen kann bevor ich 'ans Eingemachte' gehe! )
 
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