rendered paste body----------from extensions.conf---------------
[general]
static = yes
writeprotect = no
clearglobalvars = no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc
[globals]
PHONE = SIP/101&SIP/102&SIP/103&SIP/104 ;Zap/2& ... &Skinny/501@501
CONSOLE = Console/dsp
#include "extensions.d/*.conf"
[phones]
#include "extensions-phones.d/*.conf"
; Only incoming audio, no outgoing audio:
[fromgizmo]
exten => gizmoguy,1,Answer
exten => gizmoguy,n,Wait(1)
exten => gizmoguy,n,SendDTMF(1)
exten => gizmoguy,n,Dial(${PHONE})
;; Works fine:
;exten => gizmoguy,1,Dial(${PHONE})
---------------------------------------------
-----------from extensions-phones.d/101.conf-
exten => 101,1,Dial(SIP/101) ;I think you can guess what the others look like
---------------------------------------------
-----------from sip-phones.d/101.conf--------
[101]
type=friend
secret=omgshhhhhh
context=default
callerid=Ext 101 <101>
host=dynamic
nat=yes
canreinvite=no
insecure=very
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc
---------------------------------------------
-----------from sip.conf---------------------
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
nat=yes
localnet=10.4.20.0/255.0.0.0
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc
;----------------------------------------- NAT SUPPORT ------------------------
; The externip, externhost and localnet settings are used if you use Asterisk
; behind a NAT device to communicate with services on the outside.
externip = my.home.address
; For global parameters and 'register' entries:
#include "sip-reg.d/*.conf"
; For peers, each in its own section:
#include "sip-phones.d/*.conf"
[authentication]
---------------------------------------------
----------from sip-phones.d/gizmo.conf-------
[gizmophone]
type=friend
context=fromgizmo
dtmfmode=info
host=proxy01.sipphone.com
fromdomain=proxy01.sipphone.com
insecure=very
username=gizmoguy
fromuser=gizmoguy
authuser=gizmoguy
secret=omgshhhhh
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc
----------------------------------------------
----------------A call------------------------
<--- SIP read from 198.65.166.131:5060 --->
INVITE sip:gizmoguy@XXX.XXX.XXX.XXX SIP/2.0
Record-Route: <sip:198.65.166.131;ftag=1111285111;lr>
Max-Forwards: 16
To: <sip:XXXXXXXXXXX@63.80.235.198:5060>
Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bKc287.ae1c27a3.0
Via: SIP/2.0/UDP 63.80.235.211:5061;rport=5061;branch=z9hG4bK634373567
From: <sip:XXXXXXXXXX@63.80.235.211>;tag=1111285111
Call-ID: 150389463@63.80.235.211
CSeq: 6168 INVITE
User-Agent: YATE/1.2.0
Contact: <sip:XXXXXXXXXX@63.80.235.211:5061>
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
Content-Type: application/sdp
Content-Length: 185
RemoteIP: 63.80.235.211
v=0
o=yate 1192718172 1192718172 IN IP4 63.80.235.211
s=SIP Call
c=IN IP4 63.80.235.211
t=0 0
m=audio 20600 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
<------------->
--- (15 headers 8 lines) ---
Sending to 198.65.166.131 : 5060 (NAT)
Using INVITE request as basis request - 150389463@63.80.235.211
Found peer 'gizmophone'
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 63.80.235.211:20600
Found description format PCMU for ID 0
Found description format telephone-event for ID 101
Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 63.80.235.211:20600
Looking for gizmoguy in fromgizmo (domain XXX.XXX.XXX.XXX)
list_route: hop: <sip:198.65.166.131;ftag=1111285111;lr>
<--- Transmitting (NAT) to 198.65.166.131:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bKc287.ae1c27a3.0;received=198.65.166.131
Via: SIP/2.0/UDP 63.80.235.211:5061;rport=5061;branch=z9hG4bK634373567
From: <sip:XXXXXXXXXX@63.80.235.211>;tag=1111285111
To: <sip:XXXXXXXXXXX@63.80.235.198:5060>
Call-ID: 150389463@63.80.235.211
CSeq: 6168 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:gizmoguy@XXX.XXX.XXX.XXX>
Content-Length: 0
<------------>
-- Executing [gizmoguy@fromgizmo:1] Set("SIP/XXXXXXXXXXX-081e1078", "CALLERID(all)=From AZ Number <XXXXXXXXXX>") in new stack
-- Executing [gizmoguy@fromgizmo:2] Answer("SIP/XXXXXXXXXXX-081e1078", "") in new stack
Audio is at XXX.XXX.XXX.XXX port 19672
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Itchy*CLI>
<--- Reliably Transmitting (NAT) to 198.65.166.131:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bKc287.ae1c27a3.0;received=198.65.166.131
Via: SIP/2.0/UDP 63.80.235.211:5061;rport=5061;branch=z9hG4bK634373567
Record-Route: <sip:198.65.166.131;ftag=1111285111;lr>
From: <sip:XXXXXXXXXX@63.80.235.211>;tag=1111285111
To: <sip:XXXXXXXXXXX@63.80.235.198:5060>;tag=as07456761
Call-ID: 150389463@63.80.235.211
CSeq: 6168 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:gizmoguy@XXX.XXX.XXX.XXX>
Content-Type: application/sdp
Content-Length: 228
v=0
o=root 31186 31186 IN IP4 XXX.XXX.XXX.XXX
s=session
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 19672 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv
<------------>
-- Executing [gizmoguy@fromgizmo:3] Wait("SIP/XXXXXXXXXXX-081e1078", "1") in new stack
Itchy*CLI>
<--- SIP read from 198.65.166.131:5060 --->
ACK sip:gizmoguy@XXX.XXX.XXX.XXX:5060 SIP/2.0
Record-Route: <sip:198.65.166.131;ftag=1111285111;lr>
Via: SIP/2.0/UDP 198.65.166.131;branch=0
Via: SIP/2.0/UDP 63.80.235.211:5061;rport=5061;branch=z9hG4bK1367767871
From: <sip:XXXXXXXXXX@63.80.235.211>;tag=1111285111
To: <sip:XXXXXXXXXXX@63.80.235.198:5060>;tag=as07456761
Call-ID: 150389463@63.80.235.211
CSeq: 6168 ACK
Max-Forwards: 16
Contact: <sip:XXXXXXXXXX@63.80.235.211:5061>
User-Agent: YATE/1.2.0
Content-Length: 0
P-hint: rr-enforced
<------------->
--- (13 headers 0 lines) ---
-- Executing [gizmoguy@fromgizmo:4] SendDTMF("SIP/XXXXXXXXXXX-081e1078", "1") in new stack
set_destination: Parsing <sip:198.65.166.131;ftag=1111285111;lr> for address/port to send to
set_destination: set destination to 198.65.166.131, port 5060
Reliably Transmitting (NAT) to 198.65.166.131:5060:
INFO sip:XXXXXXXXXX@63.80.235.211:5061 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK4ac39bc8;rport
Route: <sip:198.65.166.131;ftag=1111285111;lr>
From: <sip:XXXXXXXXXXX@63.80.235.198:5060>;tag=as07456761
To: <sip:XXXXXXXXXX@63.80.235.211>;tag=1111285111
Contact: <sip:gizmoguy@XXX.XXX.XXX.XXX>
Call-ID: 150389463@63.80.235.211
CSeq: 102 INFO
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Type: application/dtmf-relay
Content-Length: 24
Signal=1
Duration=250
---
Itchy*CLI>
<--- SIP read from 198.65.166.131:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK4ac39bc8;rport=5060
Record-Route: <sip:198.65.166.131;ftag=as07456761;lr>
From: <sip:XXXXXXXXXXX@63.80.235.198:5060>;tag=as07456761
To: <sip:XXXXXXXXXX@63.80.235.211>;tag=1111285111
Call-ID: 150389463@63.80.235.211
CSeq: 102 INFO
Server: YATE/1.2.0
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog INFO arrived
-- Executing [gizmoguy@fromgizmo:5] Dial("SIP/XXXXXXXXXXX-081e1078", "SIP/101&SIP/102&SIP/103&SIP/104") in new stack
Really destroying SIP dialog '583a25980e3a4e211314e85a7c5c9371@127.0.1.1' Method: INVITE
[Oct 18 07:35:55] WARNING[10596]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
Really destroying SIP dialog '785d42116fae55d234fa3841172a4d45@127.0.1.1' Method: INVITE
[Oct 18 07:35:55] WARNING[10596]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
Audio is at 10.4.20.111 port 19592
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.4.20.190:5060:
INVITE sip:103@10.4.20.190:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK25caf26d;rport
From: "From AZ Number" <sip:XXXXXXXXXX@10.4.20.111>;tag=as618b585e
To: <sip:103@10.4.20.190:5060;user=phone>
Contact: <sip:XXXXXXXXXX@10.4.20.111>
Call-ID: 1e174c353d7b9ef87163a8ec572edd2b@10.4.20.111
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 18 Oct 2007 14:35:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 318
v=0
o=root 31186 31186 IN IP4 10.4.20.111
s=session
c=IN IP4 10.4.20.111
t=0 0
m=audio 19592 RTP/AVP 0 8 3 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv
---
-- Called 103
Really destroying SIP dialog '47f4809462d24a3f5360151869b9eeac@127.0.1.1' Method: INVITE
[Oct 18 07:35:55] WARNING[10596]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
Itchy*CLI>
<--- SIP read from 10.4.20.190:5060 --->
SIP/2.0 100 Trying
Call-ID: 1e174c353d7b9ef87163a8ec572edd2b@10.4.20.111
From: "From AZ Number"<sip:XXXXXXXXXX@10.4.20.111>;tag=as618b585e
To: <sip:103@10.4.20.190:5060;user=phone>;tag=3f194-9fa7a352
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK25caf26d;rport
Contact: Cordless<sip:103@10.4.20.190:5060;user=phone>
Server: DTA SIP/0.12.35 DNLD/1.1.13
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Itchy*CLI>
<--- SIP read from 10.4.20.190:5060 --->
SIP/2.0 180 Ringing
Call-ID: 1e174c353d7b9ef87163a8ec572edd2b@10.4.20.111
From: "From AZ Number"<sip:XXXXXXXXXX@10.4.20.111>;tag=as618b585e
To: <sip:103@10.4.20.190:5060;user=phone>;tag=3f194-9fa7a352
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK25caf26d;rport
Contact: Cordless<sip:103@10.4.20.190:5060;user=phone>
Server: DTA SIP/0.12.35 DNLD/1.1.13
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
-- SIP/103-081d9ea0 is ringing
Really destroying SIP dialog '6a8b89cb27b9cbf52e8b114418c26b2d@127.0.1.1' Method: REGISTER
Really destroying SIP dialog '437de217558dfe9452b918a5756584bd@127.0.1.1' Method: REGISTER
Itchy*CLI>
<--- SIP read from 10.4.20.190:5060 --->
SIP/2.0 200 OK
Call-ID: 1e174c353d7b9ef87163a8ec572edd2b@10.4.20.111
From: "From AZ Number"<sip:XXXXXXXXXX@10.4.20.111>;tag=as618b585e
To: <sip:103@10.4.20.190:5060;user=phone>;tag=3f194-9fa7a352
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK25caf26d;rport
Contact: Cordless<sip:103@10.4.20.190:5060;user=phone>
Supported: timer,100rel
Allow: REFER,UPDATE,INFO,MESSAGE,OPTIONS,PRACK
Server: DTA SIP/0.12.35 DNLD/1.1.13
Content-Type: application/sdp
Content-Length: 132
v=0
o=103 0 0 IN IP4 10.4.20.190
s=-
c=IN IP4 10.4.20.190
t=0 0
m=audio 8190 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
<------------->
--- (12 headers 7 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 10.4.20.190:8190
Found description format telephone-event for ID 101
Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.4.20.190:8190
list_route: hop: <sip:103@10.4.20.190:5060;user=phone>
set_destination: Parsing <sip:103@10.4.20.190:5060;user=phone> for address/port to send to
set_destination: set destination to 10.4.20.190, port 5060
Transmitting (NAT) to 10.4.20.190:5060:
ACK sip:103@10.4.20.190:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK1231da7f;rport
From: "From AZ Number" <sip:XXXXXXXXXX@10.4.20.111>;tag=as618b585e
To: <sip:103@10.4.20.190:5060;user=phone>;tag=3f194-9fa7a352
Contact: <sip:XXXXXXXXXX@10.4.20.111>
Call-ID: 1e174c353d7b9ef87163a8ec572edd2b@10.4.20.111
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/103-081d9ea0 answered SIP/XXXXXXXXXXX-081e1078
-- Packet2Packet bridging SIP/XXXXXXXXXXX-081e1078 and SIP/103-081d9ea0
Itchy*CLI>
<--- SIP read from 10.4.20.190:5060 --->
H
<------------->
--- (1 headers 0 lines) ---
Itchy*CLI>
<--- SIP read from 198.65.166.131:5060 --->
BYE sip:gizmoguy@XXX.XXX.XXX.XXX:5060 SIP/2.0
Record-Route: <sip:198.65.166.131;ftag=1111285111;lr>
Call-ID: 150389463@63.80.235.211
From: <sip:XXXXXXXXXX@63.80.235.211>;tag=1111285111
To: <sip:XXXXXXXXXXX@63.80.235.198:5060>;tag=as07456761
Reason: SIP;text="IVR"
Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK2097.28d3a433.0
Via: SIP/2.0/UDP 63.80.235.211:5061;rport=5061;branch=z9hG4bK2121699409
CSeq: 6173 BYE
User-Agent: YATE/1.2.0
Max-Forwards: 16
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
Content-Length: 0
RemoteIP: 63.80.235.211
P-hint: rr-enforced
<------------->
--- (15 headers 0 lines) ---
Sending to 198.65.166.131 : 5060 (NAT)
<--- Transmitting (NAT) to 198.65.166.131:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK2097.28d3a433.0;received=198.65.166.131
Via: SIP/2.0/UDP 63.80.235.211:5061;rport=5061;branch=z9hG4bK2121699409
Record-Route: <sip:198.65.166.131;ftag=1111285111;lr>
From: <sip:XXXXXXXXXX@63.80.235.211>;tag=1111285111
To: <sip:XXXXXXXXXXX@63.80.235.198:5060>;tag=as07456761
Call-ID: 150389463@63.80.235.211
CSeq: 6173 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:gizmoguy@XXX.XXX.XXX.XXX>
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1e174c353d7b9ef87163a8ec572edd2b@10.4.20.111' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:103@10.4.20.190:5060;user=phone> for address/port to send to
set_destination: set destination to 10.4.20.190, port 5060
Reliably Transmitting (NAT) to 10.4.20.190:5060:
BYE sip:103@10.4.20.190:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK53a33c6f;rport
From: "From AZ Number" <sip:XXXXXXXXXX@10.4.20.111>;tag=as618b585e
To: <sip:103@10.4.20.190:5060;user=phone>;tag=3f194-9fa7a352
Contact: <sip:XXXXXXXXXX@10.4.20.111>
Call-ID: 1e174c353d7b9ef87163a8ec572edd2b@10.4.20.111
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
== Spawn extension (fromgizmo, gizmoguy, 5) exited non-zero on 'SIP/XXXXXXXXXXX-081e1078'
Really destroying SIP dialog '150389463@63.80.235.211' Method: BYE
Itchy*CLI>
<--- SIP read from 10.4.20.190:5060 --->
SIP/2.0 200 OK
Call-ID: 1e174c353d7b9ef87163a8ec572edd2b@10.4.20.111
From: "From AZ Number"<sip:XXXXXXXXXX@10.4.20.111>;tag=as618b585e
To: <sip:103@10.4.20.190:5060;user=phone>;tag=3f194-9fa7a352
CSeq: 103 BYE
Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK53a33c6f;rport
Contact: Cordless<sip:103@10.4.20.190:5060;user=phone>
Server: DTA SIP/0.12.35 DNLD/1.1.13
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '1e174c353d7b9ef87163a8ec572edd2b@10.4.20.111' Method: INVITE