SIP endpoint with two lines: dialout always from line 1


#1

Hello folks,

I run Asterisk for two years with PJSIP trunks and SCCP und PJSIP extensions without problems and upgraded recently to 16.2.1 from 14. Now I added a Cisco 8851 endpoint with two lines (extensions 80 and 81) in the sip.conf (with cisco-usecallmanager patch, thanks to Gareth Palmer) configured and everytime I try to call from the second configured line the call is made from the first line. It doesn’t matter if the outgoing call is for internal or external numbers. Inbound calls from internal or external callers are routed to the correct line and are working as desired.

PJSIP is using tcp/5060, SIP is using tcp/5062.

This is the SIP debug from one try (internal call from extension 81 to 10):

<--- SIP read from TCP:10.10.10.49:51955 --->
NOTIFY sip:81@10.10.10.207 SIP/2.0
Via: SIP/2.0/TCP 10.10.10.49:51955;branch=z9hG4bK39b9a7c0
To: "Test_2" <sip:81@10.10.10.207>
From: "Test_2" <sip:81@10.10.10.207>;tag=b000b4bfa4ee00483455e17b-6421261f
Call-ID: 48ca354e-55d4fcc8@10.10.10.49
Session-ID: 028ae2f300105000a000b000b4bfa4ee;remote=00000000000000000000000000000000
Date: Sat, 16 Mar 2019 17:06:45 GMT
CSeq: 33 NOTIFY
Event: dialog
Subscription-State: active
Max-Forwards: 70
Contact: <sip:81@10.10.10.49:51955;transport=tcp>;+u.sip!devicename.ccm.cisco.com="SEPB000B4FFAAEE"
Authorization: Digest username="80",realm="asterisk",uri="",response="a71c2b39b0cd0c1478c6653f89291237",nonce="4e8cd689",algorithm=MD5
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 357
Content-Type: application/dialog-info+xml
Content-Disposition: session;handling=required

<?xml version="1.0" encoding="UTF-8" ?>
<dialog-info xmlns:call="urn:x-cisco:parmams:xml:ns:dialog-info:dialog:callinfo-dialog" version="18" state="partial" entity="sip:81@10.10.10.49">
<dialog id="24" call-id="b000b4bf-a4ee0018-002310b3-591e303d@10.10.10.49" local-tag="b000b4bfa4ee0047769bdbf9-20b94109"><state>trying</state></dialog>
</dialog-info>
<------------->
--- (17 headers 4 lines) ---
Sending to 10.10.10.49:51955 (no NAT)

<--- Transmitting (no NAT) to 10.10.10.49:51955 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.10.10.49:51955;branch=z9hG4bK39b9a7c0;received=10.10.10.49
From: "Test_2" <sip:81@10.10.10.207>;tag=b000b4bfa4ee00483455e17b-6421261f
To: "Test_2" <sip:81@10.10.10.207>;tag=as27cba850
Call-ID: 48ca354e-55d4fcc8@10.10.10.49
CSeq: 33 NOTIFY
Server: Asterisk PBX 16.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '48ca354e-55d4fcc8@10.10.10.49' in 32000 ms (Method: NOTIFY)

<--- SIP read from TCP:10.10.10.49:51955 --->
INVITE sip:10@10.10.10.207;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.10.10.49:51955;branch=z9hG4bK4c5b5190
From: "Anonymous" <sip:Anonymous@10.10.10.207>;tag=b000b4bfa4ee0047769bdbf9-20b94109
To: <sip:10@10.10.10.207>
Call-ID: b000b4bf-a4ee0018-002310b3-591e303d@10.10.10.49
Max-Forwards: 70
Session-ID: 6ec056e400105000a000b000b4bfa4ee;remote=00000000000000000000000000000000
Date: Sat, 16 Mar 2019 17:06:45 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP8851/12.5.1
Contact: <sip:Anonymous@10.10.10.49:51955;user=phone;transport=tcp>;+u.sip!devicename.ccm.cisco.com="SEPB000B4FFAAEE"
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "Test_2" <sip:81@10.10.10.207>;party=calling;id-type=subscriber;privacy=full;screen=yes
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.5.1
Allow-Events: dialog
Recv-Info: conference
Recv-Info: x-cisco-conference
Authorization: Digest username="80",realm="asterisk",uri="sip:10@10.10.10.207;user=phone",response="a81dd83c80d3e0f4c92f6e136be9f0ca",nonce="4e8cd689",algorithm=MD5
Content-Length: 354
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 11312 0 IN IP4 10.10.10.49
s=SIP Call
b=AS:4064
t=0 0
m=audio 24478 RTP/AVP 0 8 116 18 101
c=IN IP4 10.10.10.49
b=TIAS:64000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (23 headers 17 lines) ---
Sending to 10.10.10.49:51955 (no NAT)
Sending to 10.10.10.49:51955 (no NAT)
Using INVITE request as basis request - b000b4bf-a4ee0018-002310b3-591e303d@10.10.10.49
Found peer '80' for 'Anonymous' from 10.10.10.49:51955

<--- Reliably Transmitting (no NAT) to 10.10.10.49:51955 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP 10.10.10.49:51955;branch=z9hG4bK4c5b5190;received=10.10.10.49
From: "Anonymous" <sip:Anonymous@10.10.10.207>;tag=b000b4bfa4ee0047769bdbf9-20b94109
To: <sip:10@10.10.10.207>;tag=as08851586
Call-ID: b000b4bf-a4ee0018-002310b3-591e303d@10.10.10.49
CSeq: 101 INVITE
Server: Asterisk PBX 16.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0e0899e0"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'b000b4bf-a4ee0018-002310b3-591e303d@10.10.10.49' in 32000 ms (Method: INVITE)

<--- SIP read from TCP:10.10.10.49:51955 --->
ACK sip:10@10.10.10.207;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.10.10.49:51955;branch=z9hG4bK4c5b5190
From: "Anonymous" <sip:Anonymous@10.10.10.207>;tag=b000b4bfa4ee0047769bdbf9-20b94109
To: <sip:10@10.10.10.207>;tag=as08851586
Call-ID: b000b4bf-a4ee0018-002310b3-591e303d@10.10.10.49
Session-ID: 6ec056e400105000a000b000b4bfa4ee;remote=00000000000000000000000000000000
Max-Forwards: 70
Date: Sat, 16 Mar 2019 17:06:45 GMT
CSeq: 101 ACK
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from TCP:10.10.10.49:51955 --->
INVITE sip:10@10.10.10.207;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.10.10.49:51955;branch=z9hG4bK00c16698
From: "Anonymous" <sip:Anonymous@10.10.10.207>;tag=b000b4bfa4ee0047769bdbf9-20b94109
To: <sip:10@10.10.10.207>
Call-ID: b000b4bf-a4ee0018-002310b3-591e303d@10.10.10.49
Max-Forwards: 70
Session-ID: 6ec056e400105000a000b000b4bfa4ee;remote=00000000000000000000000000000000
Date: Sat, 16 Mar 2019 17:06:45 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP8851/12.5.1
Contact: <sip:Anonymous@10.10.10.49:51955;user=phone;transport=tcp>;+u.sip!devicename.ccm.cisco.com="SEPB000B4FFAAEE"
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "Test_2" <sip:81@10.10.10.207>;party=calling;id-type=subscriber;privacy=full;screen=yes
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.5.1
Allow-Events: dialog
Recv-Info: conference
Recv-Info: x-cisco-conference
Authorization: Digest username="80",realm="asterisk",uri="sip:10@10.10.10.207;user=phone",response="6c9982b8826ce9660be980bd1fdf2a47",nonce="0e0899e0",algorithm=MD5
Content-Length: 354
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 11312 0 IN IP4 10.10.10.49
s=SIP Call
b=AS:4064
t=0 0
m=audio 24478 RTP/AVP 0 8 116 18 101
c=IN IP4 10.10.10.49
b=TIAS:64000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (23 headers 17 lines) ---
Sending to 10.10.10.49:51955 (no NAT)
Using INVITE request as basis request - b000b4bf-a4ee0018-002310b3-591e303d@10.10.10.49
Found peer '80' for 'Anonymous' from 10.10.10.49:51955
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 116
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format iLBC for ID 116
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (g729|g722|alaw), peer - audio=(ulaw|alaw|g729|ilbc)/video=(nothing)/text=(nothing), combined - (g729|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
       > 0x7fcff4056bf0 -- Strict RTP learning after remote address set to: 10.10.10.49:24478
Peer audio RTP is at port 10.10.10.49:24478
Looking for 10 in Telekom_01_out (domain 10.10.10.207)
sip_route_dump: route/path hop: <sip:Anonymous@10.10.10.49:51955;user=phone;transport=tcp>

<--- Transmitting (no NAT) to 10.10.10.49:51955 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.10.10.49:51955;branch=z9hG4bK00c16698;received=10.10.10.49
From: "Anonymous" <sip:Anonymous@10.10.10.207>;tag=b000b4bfa4ee0047769bdbf9-20b94109
To: <sip:10@10.10.10.207>
Call-ID: b000b4bf-a4ee0018-002310b3-591e303d@10.10.10.49
CSeq: 102 INVITE
Server: Asterisk PBX 16.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Contact: <sip:10@10.10.10.207:5062;transport=tcp>
Call-Info: <urn:x-cisco-remotecc:callinfo>; security=NotAuthenticated
Content-Length: 0


<------------>
    -- Executing [10@Telekom_01_out:1] Dial("SIP/80-0000002c", "SCCP/10") in new stack
    -- SCCP: Asterisk request to call SCCP/10-0000001F (dest:10, timeout: 0)
    -- 10: Asterisk request to call SCCP/10-0000001F
    -- SCCP/10-0000001F: (sccp_pbx_call) Returning: 0
    -- Called SCCP/10

<--- Transmitting (no NAT) to 10.10.10.49:51955 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 10.10.10.49:51955;branch=z9hG4bK00c16698;received=10.10.10.49
From: "Anonymous" <sip:Anonymous@10.10.10.207>;tag=b000b4bfa4ee0047769bdbf9-20b94109
To: <sip:10@10.10.10.207>;tag=as32a4190a
Call-ID: b000b4bf-a4ee0018-002310b3-591e303d@10.10.10.49
CSeq: 102 INVITE
Server: Asterisk PBX 16.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Contact: <sip:10@10.10.10.207:5062;transport=tcp>
Remote-Party-ID: "Room 1" <sip:10@10.10.10.207>;party=called;privacy=off;screen=no
Call-Info: <urn:x-cisco-remotecc:callinfo>; orientation=to
Content-Length: 0


<------------>
    -- SCCP/10-0000001F is ringing

<--- Transmitting (no NAT) to 10.10.10.49:51955 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 10.10.10.49:51955;branch=z9hG4bK00c16698;received=10.10.10.49
From: "Anonymous" <sip:Anonymous@10.10.10.207>;tag=b000b4bfa4ee0047769bdbf9-20b94109
To: <sip:10@10.10.10.207>;tag=as32a4190a
Call-ID: b000b4bf-a4ee0018-002310b3-591e303d@10.10.10.49
CSeq: 102 INVITE
Server: Asterisk PBX 16.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Contact: <sip:10@10.10.10.207:5062;transport=tcp>
Call-Info: <urn:x-cisco-remotecc:callinfo>; security=NotAuthenticated
Content-Length: 0


<------------>

<--- SIP read from TCP:10.10.10.49:51955 --->
NOTIFY sip:81@10.10.10.207 SIP/2.0
Via: SIP/2.0/TCP 10.10.10.49:51955;branch=z9hG4bK5e253403
To: "Test_2" <sip:81@10.10.10.207>
From: "Test_2" <sip:81@10.10.10.207>;tag=b000b4bfa4ee0049083f727c-658bcfc8
Call-ID: 4b6e20e3-1dbe7adc@10.10.10.49
Session-ID: 028ae2f300105000a000b000b4bfa4ee;remote=00000000000000000000000000000000
Date: Sat, 16 Mar 2019 17:06:52 GMT
CSeq: 34 NOTIFY
Event: dialog
Subscription-State: active
Max-Forwards: 70
Contact: <sip:81@10.10.10.49:51955;transport=tcp>;+u.sip!devicename.ccm.cisco.com="SEPB000B4FFAAEE"
Authorization: Digest username="80",realm="asterisk",uri="",response="ad31225738e21b6905afb6c580a0862b",nonce="0e0899e0",algorithm=MD5
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 361
Content-Type: application/dialog-info+xml
Content-Disposition: session;handling=required

<?xml version="1.0" encoding="UTF-8" ?>
<dialog-info xmlns:call="urn:x-cisco:parmams:xml:ns:dialog-info:dialog:callinfo-dialog" version="19" state="partial" entity="sip:81@10.10.10.49">
<dialog id="24" call-id="b000b4bf-a4ee0018-002310b3-591e303d@10.10.10.49" local-tag="b000b4bfa4ee0047769bdbf9-20b94109"><state>terminated</state></dialog>
</dialog-info>
<------------->
--- (17 headers 4 lines) ---
Sending to 10.10.10.49:51955 (no NAT)

<--- Transmitting (no NAT) to 10.10.10.49:51955 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.10.10.49:51955;branch=z9hG4bK5e253403;received=10.10.10.49
From: "Test_2" <sip:81@10.10.10.207>;tag=b000b4bfa4ee0049083f727c-658bcfc8
To: "Test_2" <sip:81@10.10.10.207>;tag=as2ebb34ed
Call-ID: 4b6e20e3-1dbe7adc@10.10.10.49
CSeq: 34 NOTIFY
Server: Asterisk PBX 16.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '4b6e20e3-1dbe7adc@10.10.10.49' in 32000 ms (Method: NOTIFY)

<--- SIP read from TCP:10.10.10.49:51955 --->
CANCEL sip:10@10.10.10.207;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.10.10.49:51955;branch=z9hG4bK00c16698
From: "Anonymous" <sip:Anonymous@10.10.10.207>;tag=b000b4bfa4ee0047769bdbf9-20b94109
To: <sip:10@10.10.10.207>
Call-ID: b000b4bf-a4ee0018-002310b3-591e303d@10.10.10.49
Max-Forwards: 70
Session-ID: 6ec056e400105000a000b000b4bfa4ee;remote=00000000000000000000000000000000
Date: Sat, 16 Mar 2019 17:06:52 GMT
CSeq: 102 CANCEL
User-Agent: Cisco-CP8851/12.5.1
Content-Length: 0
Authorization: Digest username="80",realm="asterisk",uri="sip:10@10.10.10.207;user=phone",response="6aacde32da8efd45b60051986a617a76",nonce="0e0899e0",algorithm=MD5

<------------->
--- (12 headers 0 lines) ---
Sending to 10.10.10.49:51955 (no NAT)

<--- Reliably Transmitting (no NAT) to 10.10.10.49:51955 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/TCP 10.10.10.49:51955;branch=z9hG4bK00c16698;received=10.10.10.49
From: "Anonymous" <sip:Anonymous@10.10.10.207>;tag=b000b4bfa4ee0047769bdbf9-20b94109
To: <sip:10@10.10.10.207>;tag=as32a4190a
Call-ID: b000b4bf-a4ee0018-002310b3-591e303d@10.10.10.49
CSeq: 102 INVITE
Server: Asterisk PBX 16.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Call-Info: <urn:x-cisco-remotecc:callinfo>; security=NotAuthenticated
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 10.10.10.49:51955 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.10.10.49:51955;branch=z9hG4bK00c16698;received=10.10.10.49
From: "Anonymous" <sip:Anonymous@10.10.10.207>;tag=b000b4bfa4ee0047769bdbf9-20b94109
To: <sip:10@10.10.10.207>;tag=as32a4190a
Call-ID: b000b4bf-a4ee0018-002310b3-591e303d@10.10.10.49
CSeq: 102 CANCEL
Server: Asterisk PBX 16.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Call-Info: <urn:x-cisco-remotecc:callinfo>; security=NotAuthenticated
Content-Length: 0


<------------>
[Mar 16 18:06:52] NOTICE[20373][C-00000038]: sccp_channel.c:1129 sccp_channel_closeAllMediaTransmitAndReceive: SCCP/10-0000001F: (closeAllMediaTransmitAndReceive) called without a valid device
  == Spawn extension (Telekom_01_out, 10, 1) exited non-zero on 'SIP/80-0000002c'

<--- SIP read from TCP:10.10.10.49:51955 --->
ACK sip:10@10.10.10.207;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.10.10.49:51955;branch=z9hG4bK64b51a31
From: "Anonymous" <sip:Anonymous@10.10.10.207>;tag=b000b4bfa4ee0047769bdbf9-20b94109
To: <sip:10@10.10.10.207>
Call-ID: b000b4bf-a4ee0018-002310b3-591e303d@10.10.10.49
Max-Forwards: 70
Session-ID: 6ec056e400105000a000b000b4bfa4ee;remote=00000000000000000000000000000000
Date: Sat, 16 Mar 2019 17:06:52 GMT
CSeq: 102 ACK
User-Agent: Cisco-CP8851/12.5.1
Remote-Party-ID: "Test_2" <sip:81@10.10.10.207>;party=calling;id-type=subscriber;privacy=full;screen=yes
Content-Length: 0
Recv-Info: conference
Recv-Info: x-cisco-conference
Authorization: Digest username="80",realm="asterisk",uri="sip:10@10.10.10.207;user=phone",response="7b9d45284bbc8ca436505625810839a0",nonce="0e0899e0",algorithm=MD5

<------------->
--- (15 headers 0 lines) ---
Really destroying SIP dialog '48ca354e-55d4fcc8@10.10.10.49' Method: NOTIFY
[Mar 16 18:07:24] WARNING[8045]: chan_sip.c:4304 retrans_pkt: Retransmission timeout reached on transmission b000b4bf-a4ee0018-002310b3-591e303d@10.10.10.49 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
Really destroying SIP dialog '4b6e20e3-1dbe7adc@10.10.10.49' Method: NOTIFY
Really destroying SIP dialog 'b000b4bf-a4ee0018-002310b3-591e303d@10.10.10.49' Method: CANCEL

What I find strange is tthe fact that the Digest username is “80” in the first sip notify message…

sip.conf


;****************************************************
;******************   Templates    ******************
;****************************************************

[extension](!)
type=friend
context=internal
host=dynamic
nat=no
trustrpid=no
sendrpid=rpid
rpid_update=yes
rpid_immediate=yes
parkinglot=default
allowsubscribe=yes
notifyhold=no
callcounter=yes
videosupport=no
disallow=all
allow=g729,g722,alaw

[cisco-usecallmanager](!,extension)
transport=tcp
cisco_usecallmanager=yes
cisco_pickupnotify_alert=from,to
cisco_pickupnotify_timer=5
cisco_keep_conference=no
cisco_multiadmin_conference=yes
dndbusy=yes
huntgroup_default=no

[general]
tcpenable=yes
bindaddr=0.0.0.0:5062
tlscipher=AES128-SHA:AES128-SHA256:AES256-SHA:AES256-SHA256:NULL
;tlsenable=no                   ; Enable server for incoming TLS (secure) connections (default is no)
;tlsbindaddr=0.0.0.0:5063

[insecure-mode](!)
transport=tcp

[authenticated-mode](!)
transport=tls

[encrypted-mode](!)
transport=tls
; The res_srtp module needs to be loaded. 
; encryption_taglen is set per-model below
encryption=yes

[cisco-8841](!,cisco-usecallmanager)
busylevel=4
call-limit=5
encryption_taglen=80

[cisco-8865](!,cisco-usecallmanager)
busylevel=4
call-limit=5
; <videoCapability> also needs to be enabled in SEPMAC.cnf.xml
videosupport=yes
; Allow the video codec
allow=h264
encryption_taglen=80

[cisco-9951](!,cisco-usecallmanager)
busylevel=5
call-limit=6
; <videoCapability> also needs to be enabled in SEPMAC.cnf.xml
videosupport=yes
; Allow the video codec
allow=h264
encryption_taglen=80


;****************************************************
;****************************************************


[80](cisco-8841,insecure-mode)
context=Telekom_01_out
secret=12345
callerid="Test_1" <80>
description=Test_1
callgroup=1
pickupgroup=1
; mailbox=80@default
; Second line (lineIndex=2 in SEPMAC.cnf.xml)
register=81

[81](cisco-8841,insecure-mode)
context=Telekom_02_out
secret=67890
callerid="Test_2" <81>
description=Test_2

In my sip.conf I refer to the contexts Telekom_01_out (for line 1) and Telekom_02_out for (line 2) in extensions.conf but in the SIP messages there’s only Telekom_01_out to be found.

extensions.conf

[general]
static=yes
writeprotect=yes
autofallthrough=yes
extenpatternmatchnew=no
clearglobalvars=no
userscontext=unspecified

;***********************************************************************************

[globals]
;CONSOLE=Console/dsp				; Console interface for demo
;CONSOLE=DAHDI/1
;CONSOLE=Phone/phone0
;IAXINFO=guest					; IAXtel username/password
;IAXINFO=myuser:mypass
;TRUNK=DAHDI/G2					; Trunk interface

;***********************************************************************************
;
;***********************************************************************************

[Telekom_in] 
;
; Anrufe von extern via Telekom
; 
; ACHTUNG: die Reihenfolge der Einträge im Dialplan (PJSIP/22&SCCP/11,30)
;    NICHT ändern. Ansonsten erfolgt keine Übergabe an die Voicebox!
; 
;
; 30 Sekunden klingen
exten => 01234567890,1,Dial(PJSIP/20&SCCP/10&SCCP/27&SIP/80,30) 
; danach auf die Mailbox umleiten
exten => 01234567890,n,Wait(1)
exten => x,n,VoiceMail(20@voicemail)
exten => 01234567890,n,Hangup()

; 30 Sekunden klingen
exten => 01234567891,1,Dial(PJSIP/21&SCCP/29&SIP/81,30) 
; danach auf die Mailbox umleiten
exten => 01234567891,n,Wait(1)
exten => 01234567891,n,VoiceMail(29@voicemail)
exten => 01234567891,n,Hangup()

; 30 Sekunden klingen
exten => 01234567892,1,Dial(PJSIP/22&SCCP/11,30) 
; danach auf die Mailbox umleiten
exten => 01234567892,n,Wait(1)
exten => 01234567892,n,VoiceMail(11@voicemail)
exten => 01234567892,n,Hangup()


[sipgate_01-in]
exten => _X.,1,Noop(Processing an incoming call)
same => n,Dial(PJSIP/23&SCCP/12,30)
same => n,VoiceMail(12@voicemail)
same => n,Hangup()

;***********************************************************************************

[Telekom_01_out]
include => internal
include => Telekom_01

[Telekom_02_out]
include => internal
include => Telekom_02

[Telekom_03_out]
include => internal
include => Telekom_03

[sipgate_01_out]
include => internal
include => sipgate_01

;***********************************************************************************

[internal]

; direkt einzelne User anwaehlen
exten => 10,1,Dial(SCCP/10)		;
exten => 11,1,Dial(SCCP/11)		;
exten => 12,1,Dial(SCCP/12)		;
exten => 13,1,Dial(SCCP/13)		;
exten => 14,1,Dial(SCCP/14)		;
exten => 15,1,Dial(SCCP/15)		;

exten => 20,1,Dial(PJSIP/20)	;
exten => 21,1,Dial(PJSIP/21)	;
exten => 22,1,Dial(PJSIP/22)	;
exten => 23,1,Dial(PJSIP/23)	;

exten => 27,1,Dial(SCCP/27)		;
exten => 28,1,Dial(SCCP/28)		;
exten => 29,1,Dial(SCCP/29)		;
exten => 30,1,Dial(SCCP/30)		;
exten => 31,1,Dial(PJSIP/31)	;
exten => 32,1,Dial(PJSIP/32)	;

exten => 80,1,DIAL(SIP/80)		;Test_01
exten => 81,1,DIAL(SIP/81)		;Test_02


exten => 97,1,Dial(PJSIP/98)	;Test_03
exten => 98,1,Dial(PJSIP/98)	;Test_04

;Mailboxabfrage von intern ohne PIN
exten => 99,1,Answer
exten => 99,2,Wait(2)
exten => 99,3,VoiceMailMain(${CALLERID(num)}@voicemail,s)
;exten => 99,3,VoicemailMain() 
exten => 99,4,Hangup 

;****************************** 01234567890******************************
[Telekom_01]

;Notrufe gehen immer
exten => 110,1,Dial(PJSIP/Telekom_01_out/sip:110@tel.t-online.de,60)
exten => 110,n,Hangup() 
exten => 112,1,Dial(PJSIP/Telekom_01_out/sip:112@tel.t-online.de,60)
exten => 112,n,Hangup()

; Ortsnetz
exten => _Z.,1,Dial(PJSIP/Telekom_01_out/sip:02252${EXTEN}@tel.t-online.de,60)
exten => _Z.,n,Hangup() 

;nationale Gespräche
exten => _+49X.,1,Dial(PJSIP/Telekom_01_out/sip:0${EXTEN:3}@tel.t-online.de,60)
exten => _+49X.,n,Hangup()
exten => _0Z.,1,Dial(PJSIP/Telekom_01_out/sip:${EXTEN}@tel.t-online.de,60)
exten => _0Z.,n,Hangup() 

;Sperren von internationalen Gesprächen 
exten => _+X.,1,Hangup() 
exten => _00X.,1,Hangup() 

;Sperren von 0900er Rufnummern
exten => _0900.,1,Hangup() 
exten => _00900.,1,Hangup() 

;****************************** 01234567891******************************

[Telekom_02]

;Notrufe gehen immer
exten => 110,1,Dial(PJSIP/Telekom_02_out/sip:110@tel.t-online.de,60)
exten => 110,n,Hangup() 
exten => 112,1,Dial(PJSIP/Telekom_02_out/sip:112@tel.t-online.de,60)
exten => 112,n,Hangup()

; Ortsnetz
exten => _Z.,1,Dial(PJSIP/Telekom_02_out/sip:02252${EXTEN}@tel.t-online.de,60)
exten => _Z.,n,Hangup() 

;nationale Gespräche
exten => _+49X.,1,Dial(PJSIP/Telekom_02_out/sip:0${EXTEN:3}@tel.t-online.de,60)
exten => _+49X.,n,Hangup()
exten => _0Z.,1,Dial(PJSIP/Telekom_02_out/sip:${EXTEN}@tel.t-online.de,60)
exten => _0Z.,n,Hangup() 

;Sperren von internationalen Gesprächen 
exten => _+X.,1,Hangup() 
exten => _00X.,1,Hangup() 

;Sperren von 0900er Rufnummern
exten => _0900.,1,Hangup() 
exten => _00900.,1,Hangup() 


;****************************** 01234567892******************************
[Telekom_03]

;Notrufe gehen immer
exten => 110,1,Dial(PJSIP/Telekom_03_out/sip:110@tel.t-online.de,60)
exten => 110,n,Hangup() 
exten => 112,1,Dial(PJSIP/Telekom_03_out/sip:112@tel.t-online.de,60)
exten => 112,n,Hangup()

; Ortsnetz
exten => _Z.,1,Dial(PJSIP/Telekom_03_out/sip:02252${EXTEN}@tel.t-online.de,60)
exten => _Z.,n,Hangup() 

;nationale Gespräche
exten => _+49X.,1,Dial(PJSIP/Telekom_03_out/sip:0${EXTEN:3}@tel.t-online.de,60)
exten => _+49X.,n,Hangup()
exten => _0Z.,1,Dial(PJSIP/Telekom_03_out/sip:${EXTEN}@tel.t-online.de,60)
exten => _0Z.,n,Hangup() 

;Sperren von internationalen Gesprächen 
exten => _+X.,1,Hangup() 
exten => _00X.,1,Hangup() 

;Sperren von 0900er Rufnummern
exten => _0900.,1,Hangup() 
exten => _00900.,1,Hangup() 

;***********************************************************************************

[sipgate_01]

exten => _X.,1,Noop(Processing an outgoing call)
same => n,Dial(PJSIP/${EXTEN}@sipgate_01)
same => n,Hangup()
 	
;***********************************************************************************


[unspecified]
; wer hier landet ist entweder schlecht konfiguriert oder hat keine "Rechte"
exten => _X.,1,Answer()
exten => _X.,2,Verbose(D E F A U L T ==> ${CALLERID(num)} kam um ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} in UNSPECIFIED an als er versuchte die Nummer ${EXTEN} anzurufen.)
exten => _X.,3,Hangup()

I kindly ask for any hint or solution what I’m missing.

Thanks in advance!

Joerg


#2

How do you expect Asterisk to distinguish between the two “lines”, given that the From user isn’t uisable and TCP doesn’t use a fixed source port?


#3

Hi and thanks for your reply.

As I did it in sccp.conf with the “context=…” entries in sip.conf.

If this is the wrong method please advise how to do it right.

Joerg


#4

I was asking because one needs to be able to answer the question before one can hope to program the system to distinguish between the two lines. If there is nothing in the protocol exchange that can be used, there is nothing you can do.


#5

This is how I planned to do this. The contexts are defined in extensions.conf and are working perfectly with the SCCP phones as defined in sccp.conf

sip.conf (excerpt only!)

[80](cisco-8841,insecure-mode)
context=Telekom_01_out
secret=12345
callerid="Test_1" <80>
description=Test_1
callgroup=1
pickupgroup=1
; mailbox=80@default
; Second line (lineIndex=2 in SEPMAC.cnf.xml)
register=81

[81](cisco-8841,insecure-mode)
context=Telekom_02_out
secret=67890
callerid="Test_2" <81>
description=Test_2

Otherwise, how would you do this if these two lines on the phone would be two seperate phones?


#6

That configuration is incomplete; there is no host. Also register does not make sense without host=actual address.

You also don’t have type.

To make this work, type will have to be friend (not generally advisable in other cases) and the phones MUST be configured so that the second Anonymous is replaced by the sip.conf section name in:

From: "Anonymous" <sip:Anonymous@10.10.10.207>;tag=b000b4bfa4ee0047769bdbf9-20b94109

#7

Thanks a lot for your answer.

That configuration is incomplete; there is no host. Also register does not make sense without host=actual address.

I have to disagree. I set host and type:

Excerpt of sip.conf:

[extension](!)
type=friend
context=internal
host=dynamic
nat=no
...

The complete sip.conf I use I have posted in my very first post in this thread. The sip.conf you refer to is only a small part of the actual sip.conf. My apologies for not marking the last post as excerpt.

…and the phones MUST be configured so that the second Anonymous is replaced by the sip.conf section name in:

I’m pretty sure I did… The section names are [80] and [81] in my sip.conf. The part of the phone config file related to the sip lines is as follows:

<sipLines>
      <line button="1" lineIndex="1">
        <featureID>9</featureID>
        <featureLabel>Familie</featureLabel>
        <proxy>USECALLMANAGER</proxy>
        <port>5062</port>
        <name>80</name>
        <displayName>Test_01</displayName>
        <autoAnswer>
          <autoAnswerEnabled>0</autoAnswerEnabled>
        </autoAnswer>
        <callWaiting>3</callWaiting>
        <authName></authName>
        <authPassword>12345</authPassword>
        <contact></contact>
        <sharedLine>false</sharedLine>
        <messageWaitingLampPolicy>3</messageWaitingLampPolicy>
        <messageWaitingAMWI>0</messageWaitingAMWI>
        <messagesNumber></messagesNumber>
        <ringSettingIdle>4</ringSettingIdle>
        <ringSettingActive>5</ringSettingActive>
        <forwardCallInfoDisplay>
          <callerName>true</callerName>
          <callerNumber>true</callerNumber>
          <redirectedNumber>false</redirectedNumber>
          <dialedNumber>true</dialedNumber>
        </forwardCallInfoDisplay>
        <maxNumCalls>5</maxNumCalls>
        <busyTrigger>4</busyTrigger>
      </line>
      <line button="2" lineIndex="2">
        <featureID>9</featureID>
        <featureLabel>Privat</featureLabel>
        <proxy>USECALLMANAGER</proxy>
        <port>5062</port>
        <name>81</name>
        <displayName>Test_02</displayName>
        <autoAnswer>
          <autoAnswerEnabled>0</autoAnswerEnabled>
        </autoAnswer>
        <callWaiting>3</callWaiting>
        <!--<authName></authName>
        <authPassword>12345</authPassword> -->
        <contact></contact>
        <sharedLine>false</sharedLine>
        <messageWaitingLampPolicy>3</messageWaitingLampPolicy>
        <messageWaitingAMWI>0</messageWaitingAMWI>
        <messagesNumber></messagesNumber>
        <ringSettingIdle>4</ringSettingIdle>
        <ringSettingActive>5</ringSettingActive>
        <forwardCallInfoDisplay>
          <callerName>true</callerName>
          <callerNumber>true</callerNumber>
          <redirectedNumber>false</redirectedNumber>
          <dialedNumber>true</dialedNumber>
        </forwardCallInfoDisplay>
        <maxNumCalls>5</maxNumCalls>
        <busyTrigger>4</busyTrigger>
      </line>
</sipLines>

#8

Hi folks,

I solved the problem by reinstalling Asterisk in a new virtual machine and created all config files new.
But I still didn’t know what causes the strange behaviour.

Best regards,

Joerg