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Summary:ASTERISK-15742: SIP-Provider without "SIP/2.0 180 Ringing" makes trouble with call file
Reporter:fa_bian (fa_bian)Labels:
Date Opened:2010-03-04 06:15:26.000-0600Date Closed:2011-06-07 14:05:22
Priority:MinorRegression?No
Status:Closed/CompleteComponents:General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:I make a auto dial via call file.

If I use a SIP-Provider who sends a "SIP/2.0 180 Ringing" all is fine!

If I use a SIP-Provider who DOESN'T send a "SIP/2.0 180 Ringing" Asterisk DOESN'T proceed in context.

****** ADDITIONAL INFORMATION ******

SIP-Provider who sends a "SIP/2.0 180 Ringing":
SIP/2.0 100 Giving a try
SIP/2.0 183 Session Progress
SIP/2.0 180 Ringing
...


SIP-Provider who DOESN'T send a "SIP/2.0 180 Ringing":
SIP/2.0 100 Trying
SIP/2.0 183 Session progress
SIP/2.0 200 Ok
...
Comments:By: Leif Madsen (lmadsen) 2010-03-04 10:00:54.000-0600

Uhhhhh.... ok.

Additional information is required here in order to move this issue forward. Please provide, at a minimum, the following:

* SIP trace from Asterisk console
* Asterisk console output showing the issue working and not working (with debug level loggin)
* Relevant sip.conf configuration information
* SIP history (enabled via sip.conf)
By: fa_bian (fa_bian) 2010-03-04 17:11:13.000-0600

Hello Imadsen, thank you for your fast reply! I will try to give you all needed information. Fabian

Summary:
- A normal call via provider 1 is fine!
- A normal call via provider 2 is fine too, but without "SIP/2.0 180 Ringing"!
- A call-file call via provider 1 is fine!
- A call-file call via provider 2 is not ok, after answering I can't hear my Asterisk [Playback()], because imho Asterisk can not recognize that the call was picked up, so the dialplan stopped!


By: fa_bian (fa_bian) 2010-03-04 17:12:01.000-0600

############
### sip.conf ###
############

[general]
tcpbindaddr=0.0.0.0
udpbindaddr=0.0.0.0
tcpenable=no
srvlookup=yes
defaultexpiry=300
maxexpirey=900
allowoverlap=no
dtmfmode=auto
language=de
qualify=no
directmedia=no
directrtpsetup=no
disallow=all
allow=alaw
allow=ulaw
context=ext_all_in

register => 112233:******@sip.provider1.tld
register => 445566:******@sip.provider2.tld

[provider1_out] ; WITH "180 Ringing"
type=peer
host=sip.provider1.tld
defaultuser=112233
fromuser=112233
remotesecret=******
secret=******
[provider2_out] ; WITHOUT "180 Ringing"
type=peer
host=sip.provider2.tld
defaultuser=445566
fromuser=445566
remotesecret=******
secret=******

[1234]
type=friend
host=dynamic
nat=never
qualify=yes
accountcode=1234
callerid="1234" <1234>
secret=********
context=ext_1234_out


By: fa_bian (fa_bian) 2010-03-04 19:05:40.000-0600

####################
### call file (sip-history) ###
####################

### provider 1, WITH ringing ###
1. NewChan         Channel SIP/provider1_out-000002f9 - from 62e013a51855090d
2. TxReqRel        INVITE / 102 INVITE - INVITE
3. Rx              SIP/2.0 / 102 INVITE / 407 Proxy Authentication Required
4. TxReq           ACK / 102 ACK - ACK
5. AuthResp        Auth response sent for 112233 in realm sip.provider1.tld - nc 1
6. TxReqRel        INVITE / 103 INVITE - INVITE
7. Rx              SIP/2.0 / 103 INVITE / 100 Giving a try
8. Rx              SIP/2.0 / 103 INVITE / 183 Session Progress
9. Rx              SIP/2.0 / 103 INVITE / 200 OK
10. TxReq           ACK / 103 ACK - ACK
11. Masq            Old channel: Local/00491712345678@ext_all_out-b67c;1<ZOMBIE>
12. Masq (cont)     ...new owner: SIP/provider1_out-000002f9

### provider 2, WITHOUT ringing ###
1. NewChan         Channel SIP/provider2_out-000002fb - from 596bba4d26bba5f11
2. TxReqRel        INVITE / 102 INVITE - INVITE
3. Rx              SIP/2.0 / 102 INVITE / 401 Unauthorized
4. TxReq           ACK / 102 ACK - ACK
5. AuthResp        Auth response sent for 445566 in realm sip.provider2.tld - nc 1
6. TxReqRel        INVITE / 103 INVITE - INVITE
7. Rx              SIP/2.0 / 103 INVITE / 100 Trying
8. Rx              SIP/2.0 / 103 INVITE / 183 Session progress
9. Rx              SIP/2.0 / 103 INVITE / 200 Ok
10. TxReq           ACK / 103 ACK - ACK


By: fa_bian (fa_bian) 2010-03-04 19:13:33.000-0600

#######################
### normal call (sip-history) ###
#######################

### provider 1, WITH ringing ###
1. Rx              INVITE / 101 INVITE / sip:01712345678@123.123.123.123;user=
2. AuthChal        Auth challenge sent for  - nc 0
3. TxRespRel       SIP/2.0 / 101 INVITE - 401 Unauthorized
4. SchedDestroy    32000 ms
5. Rx              ACK / 101 ACK / sip:01712345678@123.123.123.123;user=phone
6. Rx              INVITE / 102 INVITE / sip:01712345678@123.123.123.123;user=
7. CancelDestroy
8. Invite          New call: 0006283e-03f60058-0f9a4154-3a12d06e@192.168.0.100
9. AuthOK          Auth challenge succesful for 1234
10. NewChan         Channel SIP/1234-00000300 - from 0006283e-03f60058-0f9
11. TxResp          SIP/2.0 / 102 INVITE - 100 Trying
12. TxResp          SIP/2.0 / 102 INVITE - 183 Session Progress
13. TxResp          SIP/2.0 / 102 INVITE - 180 Ringing
14. TxRespRel       SIP/2.0 / 102 INVITE - 200 OK
15. Rx              ACK / 102 ACK / sip:01712345678@123.123.123.123

### provider 2, WITHOUT ringing ###
1. Rx              INVITE / 101 INVITE / sip:01712345678@123.123.123.123;user=
2. AuthChal        Auth challenge sent for  - nc 0
3. TxRespRel       SIP/2.0 / 101 INVITE - 401 Unauthorized
4. SchedDestroy    18496 ms
5. Rx              ACK / 101 ACK / sip:01712345678@123.123.123.123;user=phone
6. Rx              INVITE / 102 INVITE / sip:01712345678@123.123.123.123;user=
7. CancelDestroy
8. Invite          New call: 0006283e-03f60057-0d0b087c-5b73c322@192.168.0.100
9. AuthOK          Auth challenge succesful for 1234
10. NewChan         Channel SIP/1234-000002fe - from 0006283e-03f60057-0d0b
11. TxResp          SIP/2.0 / 102 INVITE - 100 Trying
12. TxResp          SIP/2.0 / 102 INVITE - 183 Session Progress
13. TxRespRel       SIP/2.0 / 102 INVITE - 200 OK
14. Rx              ACK / 102 ACK / sip:01712345678@123.123.123.123


By: fa_bian (fa_bian) 2010-03-04 21:16:39.000-0600

#######################
### call file (sip-trace)    ###
### provider 1, WITH ringing ###
#######################

INVITE sip:00491712345678@sip.provider1.tld SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK2cdca6b8;rport
Max-Forwards: 70
From: "49301234567" <sip:112233@123.123.123.123>;tag=as6ec65564
To: <sip:00491712345678@sip.provider1.tld>
Contact: <sip:112233@123.123.123.123>
Call-ID: 3df2e69d5319f26668e00eae14319fa5@123.123.123.123
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.2
Date: Fri, 05 Mar 2010 01:37:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 2138980316 2138980316 IN IP4 123.123.123.123
s=Asterisk PBX 1.6.2.2
c=IN IP4 123.123.123.123
t=0 0
m=audio 14422 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---


<--- SIP read from UDP:34.34.34.34:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK2cdca6b8;rport=5060
From: "49301234567" <sip:112233@123.123.123.123>;tag=as6ec65564
To: <sip:00491712345678@sip.provider1.tld>;tag=4fa8f7eb71cc68cca91a14abea886308.16dc

Call-ID: 3df2e69d5319f26668e00eae14319fa5@123.123.123.123
CSeq: 102 INVITE


Proxy-Authenticate: Digest realm="sip.provider1.tld", nonce="4b906174eb50796385f55a94796d29fa73bd824d"
Content-Length: 0


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

ACK sip:00491712345678@sip.provider1.tld SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK2cdca6b8;rport
Max-Forwards: 70
From: "49301234567" <sip:112233@123.123.123.123>;tag=as6ec65564
To: <sip:00491712345678@sip.provider1.tld>;tag=4fa8f7eb71cc68cca91a14abea886308.16dc
Contact: <sip:112233@123.123.123.123>
Call-ID: 3df2e69d5319f26668e00eae14319fa5@123.123.123.123
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.2
Content-Length: 0

INVITE sip:00491712345678@sip.provider1.tld SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK0d6698d9;rport
Max-Forwards: 70
From: "49301234567" <sip:112233@123.123.123.123>;tag=as6ec65564
To: <sip:00491712345678@sip.provider1.tld>
Contact: <sip:112233@123.123.123.123>
Call-ID: 3df2e69d5319f26668e00eae14319fa5@123.123.123.123
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.2.2
Proxy-Authorization: Digest username="112233", realm="sip.provider1.tld", algorithm=MD5, uri="sip:00491712345678@sip.provider1.tld", nonce="4b906174eb50796385f55a94796d29fa73bd824d", response="ac3e7d021dfbf8a384686cd702010d7f"
Date: Fri, 05 Mar 2010 01:37:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 2138980316 2138980317 IN IP4 123.123.123.123
s=Asterisk PBX 1.6.2.2
c=IN IP4 123.123.123.123
t=0 0
m=audio 14422 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:34.34.34.34:5060 --->
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK0d6698d9;rport=5060
From: "49301234567" <sip:112233@123.123.123.123>;tag=as6ec65564
To: <sip:00491712345678@sip.provider1.tld>

Call-ID: 3df2e69d5319f26668e00eae14319fa5@123.123.123.123
CSeq: 103 INVITE


Content-Length: 0


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

<--- SIP read from UDP:34.34.34.34:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK0d6698d9;rport=5060
Record-Route: <sip:35.35.35.35;lr=on>
Record-Route: <sip:34.34.34.34;lr=on;ftag=as6ec65564>
From: "49301234567" <sip:112233@123.123.123.123>;tag=as6ec65564
To: <sip:00491712345678@sip.provider1.tld>;tag=as003d1501

Call-ID: 3df2e69d5319f26668e00eae14319fa5@123.123.123.123
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2400491712345678@36.36.36.36>
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 16906 16906 IN IP4 36.36.36.36
s=session
c=IN IP4 36.36.36.36
t=0 0
m=audio 18484 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

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

<--- SIP read from UDP:34.34.34.34:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK0d6698d9;rport=5060
Record-Route: <sip:35.35.35.35;lr=on>
Record-Route: <sip:34.34.34.34;lr=on;ftag=as6ec65564>
From: "49301234567" <sip:112233@123.123.123.123>;tag=as6ec65564
To: <sip:00491712345678@sip.provider1.tld>;tag=as003d1501
Contact: <sip:2400491712345678@36.36.36.36>
Call-ID: 3df2e69d5319f26668e00eae14319fa5@123.123.123.123
CSeq: 103 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 16906 16907 IN IP4 36.36.36.36
s=session
c=IN IP4 36.36.36.36
t=0 0
m=audio 18484 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

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

ACK sip:2400491712345678@36.36.36.36 SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK5ad9fbda;rport
Route: <sip:34.34.34.34;lr=on;ftag=as6ec65564>,<sip:35.35.35.35;lr=on>
Max-Forwards: 70
From: "49301234567" <sip:112233@123.123.123.123>;tag=as6ec65564
To: <sip:00491712345678@sip.provider1.tld>;tag=as003d1501
Contact: <sip:112233@123.123.123.123>
Call-ID: 3df2e69d5319f26668e00eae14319fa5@123.123.123.123
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.2.2
Content-Length: 0


---







<--- SIP read from UDP:34.34.34.34:5060 --->
BYE sip:112233@123.123.123.123 SIP/2.0
Via: SIP/2.0/UDP 34.34.34.34:5060;branch=z9hG4bKf30f.1c23c1b2.0
Via: SIP/2.0/UDP 35.35.35.35;branch=z9hG4bKf30f.1c23c1b2.0
Via: SIP/2.0/UDP 36.36.36.36:5060;branch=z9hG4bK559edaa9;rport=5060
Max-Forwards: 68
From: <sip:00491712345678@sip.provider1.tld>;tag=as003d1501
To: "49301234567" <sip:112233@123.123.123.123>;tag=as6ec65564
Call-ID: 3df2e69d5319f26668e00eae14319fa5@123.123.123.123
CSeq: 102 BYE



X-hint: rr-enforced


Content-Length: 0



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

<--- Transmitting (no NAT) to 34.34.34.34:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 34.34.34.34:5060;branch=z9hG4bKf30f.1c23c1b2.0;received=34.34.34.34
Via: SIP/2.0/UDP 35.35.35.35;branch=z9hG4bKf30f.1c23c1b2.0
Via: SIP/2.0/UDP 36.36.36.36:5060;branch=z9hG4bK559edaa9;rport=5060
From: <sip:00491712345678@sip.provider1.tld>;tag=as003d1501
To: "49301234567" <sip:112233@123.123.123.123>;tag=as6ec65564

Call-ID: 3df2e69d5319f26668e00eae14319fa5@123.123.123.123
CSeq: 102 BYE
Server: Asterisk PBX 1.6.2.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


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


By: fa_bian (fa_bian) 2010-03-04 21:21:15.000-0600

##########################
### call file (sip-trace) ###
### provider 2, WITHOUT ringing ###
##########################

INVITE sip:00491712345678@sip.provider2.tld SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK2de9667e;rport
Max-Forwards: 70
From: "49301234567" <sip:445566@123.123.123.123>;tag=as2f441153
To: <sip:00491712345678@sip.provider2.tld>
Contact: <sip:445566@123.123.123.123>
Call-ID: 0b7cb2ad625457b03835a8c60214c934@123.123.123.123
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.2
Date: Fri, 05 Mar 2010 01:34:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 293346299 293346299 IN IP4 123.123.123.123
s=Asterisk PBX 1.6.2.2
c=IN IP4 123.123.123.123
t=0 0
m=audio 19358 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---


<--- SIP read from UDP:12.12.12.12:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK2de9667e;rport
From: "49301234567" <sip:445566@123.123.123.123>;tag=as2f441153
To: <sip:00491712345678@sip.provider2.tld>
Contact: sip:00491712345678@12.12.12.12:5060
Call-ID: 0b7cb2ad625457b03835a8c60214c934@123.123.123.123
CSeq: 102 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
WWW-Authenticate: Digest realm="sip.provider2.tld",nonce="3323756578",algorithm=MD5
Content-Length: 0


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

ACK sip:00491712345678@sip.provider2.tld SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK2de9667e;rport
Max-Forwards: 70
From: "49301234567" <sip:445566@123.123.123.123>;tag=as2f441153
To: <sip:00491712345678@sip.provider2.tld>
Contact: <sip:445566@123.123.123.123>
Call-ID: 0b7cb2ad625457b03835a8c60214c934@123.123.123.123
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.2
Content-Length: 0

INVITE sip:00491712345678@sip.provider2.tld SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK16921415;rport
Max-Forwards: 70
From: "49301234567" <sip:445566@123.123.123.123>;tag=as2f441153
To: <sip:00491712345678@sip.provider2.tld>
Contact: <sip:445566@123.123.123.123>
Call-ID: 0b7cb2ad625457b03835a8c60214c934@123.123.123.123
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.2.2
Authorization: Digest username="445566", realm="sip.provider2.tld", algorithm=MD5, uri="sip:00491712345678@sip.provider2.tld", nonce="3323756578", response="595cb53585ff6e80d41456757e8228f9"
Date: Fri, 05 Mar 2010 01:34:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 293346299 293346300 IN IP4 123.123.123.123
s=Asterisk PBX 1.6.2.2
c=IN IP4 123.123.123.123
t=0 0
m=audio 19358 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:12.12.12.12:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK16921415;rport
From: "49301234567" <sip:445566@123.123.123.123>;tag=as2f441153
To: <sip:00491712345678@sip.provider2.tld>
Contact: sip:00491712345678@12.12.12.12:5060
Call-ID: 0b7cb2ad625457b03835a8c60214c934@123.123.123.123
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0


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

<--- SIP read from UDP:12.12.12.12:5060 --->
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK16921415;rport

From: "49301234567" <sip:445566@123.123.123.123>;tag=as2f441153
To: <sip:00491712345678@sip.provider2.tld>;tag=120113ac4b5da89a85a5c2
Contact: sip:00491712345678@12.12.12.12:5060
Call-ID: 0b7cb2ad625457b03835a8c60214c934@123.123.123.123
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE


Content-Type: application/sdp
Content-Length: 201

v=0
o=445566 1267752884 1267752884 IN IP4 13.13.13.13
s=SIP Call
c=IN IP4 13.13.13.13
t=0 0
m=audio 25570 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000


a=ptime:20

<------------->
<--- SIP read from UDP:12.12.12.12:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK16921415;rport


From: "49301234567" <sip:445566@123.123.123.123>;tag=as2f441153
To: <sip:00491712345678@sip.provider2.tld>;tag=120113ac4b5da89a85a5c2
Contact: sip:00491712345678@12.12.12.12:5060
Call-ID: 0b7cb2ad625457b03835a8c60214c934@123.123.123.123
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE

Content-Type: application/sdp
Content-Length: 201

v=0
o=445566 1267752901 1267752901 IN IP4 13.13.13.13
s=SIP Call
c=IN IP4 13.13.13.13
t=0 0
m=audio 25570 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000


a=ptime:20


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

ACK sip:00491712345678@12.12.12.12:5060 SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK21e476bc;rport

Max-Forwards: 70
From: "49301234567" <sip:445566@123.123.123.123>;tag=as2f441153
To: <sip:00491712345678@sip.provider2.tld>;tag=120113ac4b5da89a85a5c2
Contact: <sip:445566@123.123.123.123>
Call-ID: 0b7cb2ad625457b03835a8c60214c934@123.123.123.123
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.2.2
Content-Length: 0


---

OPTIONS sip:sip.provider2.tld SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK28767e3a;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@123.123.123.123>;tag=as02d3179c
To: <sip:sip.provider2.tld>
Contact: <sip:asterisk@123.123.123.123>
Call-ID: 0c8d900b1816ea6229d62a09565a8982@123.123.123.123
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.2
Date: Fri, 05 Mar 2010 01:35:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<--- SIP read from UDP:12.12.12.12:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK28767e3a;rport
From: "asterisk" <sip:asterisk@123.123.123.123>;tag=as02d3179c
To: <sip:sip.provider2.tld>
Contact: sip:12.12.12.12:5060
Call-ID: 0c8d900b1816ea6229d62a09565a8982@123.123.123.123
CSeq: 102 OPTIONS
Supported: foo
User-Agent: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Accept: application/sdp


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



BYE sip:00491712345678@12.12.12.12:5060 SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK6a6f395a;rport


Max-Forwards: 70
From: "49301234567" <sip:445566@123.123.123.123>;tag=as2f441153
To: <sip:00491712345678@sip.provider2.tld>;tag=120113ac4b5da89a85a5c2
Call-ID: 0b7cb2ad625457b03835a8c60214c934@123.123.123.123
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.6.2.2

Authorization: Digest username="445566", realm="sip.provider2.tld", algorithm=MD5, uri="sip:00491712345678@12.12.12.12:5060", nonce="3323756578", response="8d0d81af3175f24f0b9a3053b7d349f1"

X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---


<--- SIP read from UDP:12.12.12.12:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK6a6f395a;rport


From: "49301234567" <sip:445566@123.123.123.123>;tag=as2f441153
To: <sip:00491712345678@sip.provider2.tld>;tag=120113ac4b5da89a85a5c2
Contact: sip:00491712345678@12.12.12.12:5060
Call-ID: 0b7cb2ad625457b03835a8c60214c934@123.123.123.123
CSeq: 104 BYE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE

Content-Length: 0


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


By: fa_bian (fa_bian) 2010-03-04 22:35:19.000-0600

#######################
### call file (CLI) ###
### provider 1, WITH ringing ###
#######################
...
   -- Called 00491712345678@provider1_out
   -- SIP/provider1_out-00000309 is making progress passing it to Local/00491712345678@ext_all_out-a91a;2
   -- SIP/provider1_out-00000309 is ringing
   -- SIP/provider1_out-00000309 answered Local/00491712345678@ext_all_out-a91a;2
      > Channel SIP/provider1_out-00000309 was answered.
   -- Executing [h@macro-cfx_out:1] NoOp("Local/00491712345678@ext_all_out-a91a;2", "...") in new stack
   -- Executing [491712345678@ext_cb1:1] Set("SIP/provider1_out-00000309", "cbPEER=491712345678") in new stack
   -- Executing [491712345678@ext_cb1:2] Goto("SIP/provider1_out-00000309", "ext_cb2,s,1") in new stack
   -- Goto (ext_cb2,s,1)
   -- Executing [s@ext_cb2:1] Set("SIP/provider1_out-00000309", "NR=") in new stack
   -- Executing [s@ext_cb2:2] Wait("SIP/provider1_out-00000309", "1") in new stack
 == Spawn extension (macro-cfx_out, s, 68) exited non-zero on 'Local/00491712345678@ext_all_out-a91a;2' in macro 'cfx_out'
 == Spawn extension (ext_all_out, 00491712345678, 52) exited non-zero on 'Local/00491712345678@ext_all_out-a91a;2'
   -- Executing [s@ext_cb2:3] BackGround("SIP/provider1_out-00000309", "privacy-prompt") in new stack
   -- <SIP/provider1_out-00000309> Playing 'privacy-prompt.gsm' (language 'de')
   -- Executing [s@ext_cb2:4] Set("SIP/provider1_out-00000309", "TIMEOUT(response)=10") in new stack
   -- Response timeout set to 10.000
   -- Executing [s@ext_cb2:5] Set("SIP/provider1_out-00000309", "TIMEOUT(digit)=10") in new stack
   -- Digit timeout set to 10.000
   -- Executing [s@ext_cb2:6] WaitExten("SIP/provider1_out-00000309", "") in new stack
   -- Timeout on SIP/provider1_out-00000309, going to 't'
   -- Executing [t@ext_cb2:1] Playback("SIP/provider1_out-00000309", "vm-goodbye") in new stack
   -- <SIP/provider1_out-00000309> Playing 'vm-goodbye.gsm' (language 'de')
   -- Executing [t@ext_cb2:2] Hangup("SIP/provider1_out-00000309", "") in new stack
 == Spawn extension (ext_cb2, t, 2) exited non-zero on 'SIP/provider1_out-00000309'
[Mar  5 04:57:31] NOTICE[11784]: pbx_spool.c:349 attempt_thread: Call completed to Local/00491712345678@ext_all_out
By: fa_bian (fa_bian) 2010-03-04 22:36:45.000-0600

##########################
### call file (CLI) ###
### provider 2, WITHOUT ringing ###
##########################
...
   -- Called 00491712345678@provider2_out
   -- SIP/provider2_out-00000307 is making progress passing it to Local/00491712345678@ext_all_out-2a67;2
   -- SIP/provider2_out-00000307 answered Local/00491712345678@ext_all_out-2a67;2
   -- Executing [h@macro-cfx_out:1] NoOp("Local/00491712345678@ext_all_out-2a67;2", "...") in new stack
[Mar  5 04:54:09] NOTICE[11758]: pbx_spool.c:339 attempt_thread: Call failed to go through, reason (3) Remote end Ringing
 == Spawn extension (macro-cfx_out, s, 68) exited non-zero on 'Local/00491712345678@ext_all_out-2a67;2' in macro 'cfx_out'
 == Spawn extension (ext_all_out, 00491712345678, 52) exited non-zero on 'Local/00491712345678@ext_all_out-2a67;2'
By: Paul Belanger (pabelanger) 2010-04-28 15:47:58

Please retest using the latest version of 1.6.2. Be sure to *attach* your trace logs to the mantis issue if the problem is still there.
By: Paul Belanger (pabelanger) 2010-05-01 11:44:31

Use the following document to generate you debug log.

http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt
By: Paul Belanger (pabelanger) 2010-05-15 18:55:39

Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested.

Further information can be found at http://www.asterisk.org/developers/bug-guidelines