Hello,
Prerequisites:
-
asterisk setup tested with softphones and found to be working
-
using Asterisk-IM 1.4 and Openfire 3.4.5
Problems:
-
can receive calls on Spark , but after 2 or 3 seconds the call hangs up.
-
can dial on Spark, but after 2 or 3 seconds the call hangs up.
Setup:
sip.conf :
context=default
recordhistory=yes
bindaddr = 0.0.0.0
bindport = 5060
disallow=all
allow=gsm
allow=ulaw
allow=alaw
progressinband = yes
canreinvite = no
useragent=asterisk
qualify = yes
;register test accounts
register => 40341569039:mypassword@mysipgateway/40341569039
disallow=all
allow=gsm
allow=ulaw
type = friend
insecure = very
username = 40341569039
fromuser = 40341569039
secret = wirme6me
host = sip.netmaster.ro
fromdomain = netmaster.ro
canreinvite = yes
qualify = no
nat = yes
context = smartcall-incoming
type=friend
;auth=md5
dtmfmode = rfc2833
username=40341569039
secret=test
callerid=“40341569039” <40341569039>
host=dynamic
canreinvite=yes
disallow=all
allow=gsm
allow=ulaw
allow=alaw
defaultip = 192.168.0.148
nat = yes
qualify = yes
context = localphones
extensions.conf:
exten => _X.,1,Dial(SIP/$@smartcall,60)
;exten => _X.,2,Congestion
;exten => _X.,3,Busy
exten => _X.,2,Hangup
exten => _X.,1,Dial(SIP/40341569039,60)
;exten => _X.,2,Congestion
;exten => _X.,3,Busy
exten => _X.,2,Hangup
Logs:
Incoming call (from outside the asterisk box) :
<— SIP read from 193.16.148.244:5060 —>
INVITE sip:40341569039@89.33.6.124 SIP/2.0
Record-Route: <sip:193.16.148.244;ftag=f4a8705af1b5a0cfcb58f45fb01f6f32;lr>
Via: SIP/2.0/UDP 193.16.148.244;branch=z9hG4bKafcc.e595c43513ec0d792b25d00ca3cc5037.0
Via: SIP/2.0/UDP 193.16.148.244:5061;branch=z9hG4bK4b29a54e3c600206b059c3524d9723e4;rport=5061
Max-Forwards: 16
From: “40341569059” <sip:40341569059@193.16.148.244>;tag=f4a8705af1b5a0cfcb58f45fb01f6f32
To: <sip:40341569039@193.16.148.244>
Call-ID: ZTRkY2U1NDRiNTU4ZmZkMzkzYmMyYTI1ZWVlZDU0YjA.
CSeq: 200 INVITE
Contact: Anonymous <sip:193.16.148.244:5061>
Expires: 300
User-Agent: Sippy
cisco-GUID: 954648260-3819049436-2228224023-2760872502
h323-conf-id: 954648260-3819049436-2228224023-2760872502
Content-Length: 351
Content-Type: application/sdp
v=0
o=Sippy 146938380 0 IN IP4 193.16.148.244
s=Zoiper_user
t=0 0
m=audio 50732 RTP/AVP 3 110 98 8 0 101
c=IN IP4 193.16.148.244
a=rtpmap:3 GSM/8000
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=direction:active
<----
>
— (16 headers 16 lines) —
Sending to 193.16.148.244 : 5060 (no NAT)
Using INVITE request as basis request - ZTRkY2U1NDRiNTU4ZmZkMzkzYmMyYTI1ZWVlZDU0YjA.
Found peer ‘smartcall’
Found RTP audio format 3
Found RTP audio format 110
Found RTP audio format 98
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 193.16.148.244:50732
Found description format GSM for ID 3
Found description format speex for ID 110
Found description format iLBC for ID 98
Found description format PCMA for ID 8
Found description format PCMU for ID 0
Found description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - (gsm|alaw|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 193.16.148.244:50732
Looking for 40341569039 in smartcall-incoming (domain 89.33.6.124)
list_route: hop: <sip:193.16.148.244;ftag=f4a8705af1b5a0cfcb58f45fb01f6f32;lr>
<— Transmitting (NAT) to 193.16.148.244:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 193.16.148.244;branch=z9hG4bKafcc.e595c43513ec0d792b25d00ca3cc5037.0;received=1 93.16.148.244
Via: SIP/2.0/UDP 193.16.148.244:5061;branch=z9hG4bK4b29a54e3c600206b059c3524d9723e4;rport=5061
From: “40341569059” <sip:40341569059@193.16.148.244>;tag=f4a8705af1b5a0cfcb58f45fb01f6f32
To: <sip:40341569039@193.16.148.244>
Call-ID: ZTRkY2U1NDRiNTU4ZmZkMzkzYmMyYTI1ZWVlZDU0YjA.
CSeq: 200 INVITE
User-Agent: asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:40341569039@89.33.6.124>
Content-Length: 0
<----
>
Audio is at 192.168.0.254 port 19718
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.0.148:5060:
INVITE sip:40341569039@192.168.0.148 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK643a1fed;rport
From: “40341569059” <sip:40341569059@192.168.0.254>;tag=as0d1458a0
To: <sip:40341569039@192.168.0.148>
Contact: <sip:40341569059@192.168.0.254>
Call-ID: 4b5a809e58a5ad0d2fad1384528a1502@192.168.0.254
CSeq: 102 INVITE
User-Agent: asterisk
Max-Forwards: 70
Date: Mon, 25 Feb 2008 13:12:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 289
v=0
o=root 73314 73314 IN IP4 192.168.0.254
s=session
c=IN IP4 192.168.0.254
t=0 0
m=audio 19718 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Retransmitting #1 (NAT) to 192.168.0.148:5060:
INVITE sip:40341569039@192.168.0.148 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK643a1fed;rport
From: “40341569059” <sip:40341569059@192.168.0.254>;tag=as0d1458a0
To: <sip:40341569039@192.168.0.148>
Contact: <sip:40341569059@192.168.0.254>
Call-ID: 4b5a809e58a5ad0d2fad1384528a1502@192.168.0.254
CSeq: 102 INVITE
User-Agent: asterisk
Max-Forwards: 70
Date: Mon, 25 Feb 2008 13:12:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 289
v=0
o=root 73314 73314 IN IP4 192.168.0.254
s=session
c=IN IP4 192.168.0.254
t=0 0
m=audio 19718 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Retransmitting #2 (NAT) to 192.168.0.148:5060:
INVITE sip:40341569039@192.168.0.148 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK643a1fed;rport
From: “40341569059” <sip:40341569059@192.168.0.254>;tag=as0d1458a0
To: <sip:40341569039@192.168.0.148>
Contact: <sip:40341569059@192.168.0.254>
Call-ID: 4b5a809e58a5ad0d2fad1384528a1502@192.168.0.254
CSeq: 102 INVITE
User-Agent: asterisk
Max-Forwards: 70
Date: Mon, 25 Feb 2008 13:12:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 289
v=0
o=root 73314 73314 IN IP4 192.168.0.254
s=session
c=IN IP4 192.168.0.254
t=0 0
m=audio 19718 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Retransmitting #3 (NAT) to 192.168.0.148:5060:
INVITE sip:40341569039@192.168.0.148 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK643a1fed;rport
From: “40341569059” <sip:40341569059@192.168.0.254>;tag=as0d1458a0
To: <sip:40341569039@192.168.0.148>
Contact: <sip:40341569059@192.168.0.254>
Call-ID: 4b5a809e58a5ad0d2fad1384528a1502@192.168.0.254
CSeq: 102 INVITE
User-Agent: asterisk
Max-Forwards: 70
Date: Mon, 25 Feb 2008 13:12:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 289
v=0
o=root 73314 73314 IN IP4 192.168.0.254
s=session
c=IN IP4 192.168.0.254
t=0 0
m=audio 19718 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
mail*CLI>
<— SIP read from 192.168.0.148:5061 —>
<----
>
— (0 headers 1 lines) —
Retransmitting #4 (NAT) to 192.168.0.148:5060:
INVITE sip:40341569039@192.168.0.148 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK643a1fed;rport
From: “40341569059” <sip:40341569059@192.168.0.254>;tag=as0d1458a0
To: <sip:40341569039@192.168.0.148>
Contact: <sip:40341569059@192.168.0.254>
Call-ID: 4b5a809e58a5ad0d2fad1384528a1502@192.168.0.254
CSeq: 102 INVITE
User-Agent: asterisk
Max-Forwards: 70
Date: Mon, 25 Feb 2008 13:12:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 289
v=0
o=root 73314 73314 IN IP4 192.168.0.254
s=session
c=IN IP4 192.168.0.254
t=0 0
m=audio 19718 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
NOTICE[73314]: chan_sip.c:2879 auto_congest: Auto-congesting SIP/40341569039-08758000
Scheduling destruction of SIP dialog ‘4b5a809e58a5ad0d2fad1384528a1502@192.168.0.254’ in 32000 ms (Method: INVITE)
Scheduling destruction of SIP dialog ‘ZTRkY2U1NDRiNTU4ZmZkMzkzYmMyYTI1ZWVlZDU0YjA.’ in 32000 ms (Method: INVITE)
<— Reliably Transmitting (NAT) to 193.16.148.244:5060 —>
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 193.16.148.244;branch=z9hG4bKafcc.e595c43513ec0d792b25d00ca3cc5037.0;received=1 93.16.148.244
Via: SIP/2.0/UDP 193.16.148.244:5061;branch=z9hG4bK4b29a54e3c600206b059c3524d9723e4;rport=5061
From: “40341569059” <sip:40341569059@193.16.148.244>;tag=f4a8705af1b5a0cfcb58f45fb01f6f32
To: <sip:40341569039@193.16.148.244>;tag=as18799e0f
Call-ID: ZTRkY2U1NDRiNTU4ZmZkMzkzYmMyYTI1ZWVlZDU0YjA.
CSeq: 200 INVITE
User-Agent: asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:40341569039@89.33.6.124>
Content-Length: 0
<----
>
mail*CLI>
<— SIP read from 193.16.148.244:5060 —>
ACK sip:40341569039@89.33.6.124 SIP/2.0
Via: SIP/2.0/UDP 193.16.148.244;branch=z9hG4bKafcc.e595c43513ec0d792b25d00ca3cc5037.0
From: “40341569059” <sip:40341569059@193.16.148.244>;tag=f4a8705af1b5a0cfcb58f45fb01f6f32
Call-ID: ZTRkY2U1NDRiNTU4ZmZkMzkzYmMyYTI1ZWVlZDU0YjA.
To: <sip:40341569039@193.16.148.244>;tag=as18799e0f
CSeq: 200 ACK
User-Agent: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0
<----
>
— (8 headers 0 lines) —
mail*CLI>
Outgoing call (from Spark to outside number):
Audio is at 192.168.0.254 port 12774
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.0.148:5060:
INVITE sip:40341569039@192.168.0.148 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK778e15c6;rport
From: “40341569039” <sip:40341569039@192.168.0.254>;tag=as521f8f44
To: <sip:40341569039@192.168.0.148>
Contact: <sip:40341569039@192.168.0.254>
Call-ID: 4725fa8231d615141660e16101ae8d6c@192.168.0.254
CSeq: 102 INVITE
User-Agent: asterisk
Max-Forwards: 70
Date: Mon, 25 Feb 2008 13:17:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 289
v=0
o=root 73314 73314 IN IP4 192.168.0.254
s=session
c=IN IP4 192.168.0.254
t=0 0
m=audio 12774 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Retransmitting #1 (NAT) to 192.168.0.148:5060:
INVITE sip:40341569039@192.168.0.148 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK778e15c6;rport
From: “40341569039” <sip:40341569039@192.168.0.254>;tag=as521f8f44
To: <sip:40341569039@192.168.0.148>
Contact: <sip:40341569039@192.168.0.254>
Call-ID: 4725fa8231d615141660e16101ae8d6c@192.168.0.254
CSeq: 102 INVITE
User-Agent: asterisk
Max-Forwards: 70
Date: Mon, 25 Feb 2008 13:17:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 289
v=0
o=root 73314 73314 IN IP4 192.168.0.254
s=session
c=IN IP4 192.168.0.254
t=0 0
m=audio 12774 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Retransmitting #2 (NAT) to 192.168.0.148:5060:
INVITE sip:40341569039@192.168.0.148 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK778e15c6;rport
From: “40341569039” <sip:40341569039@192.168.0.254>;tag=as521f8f44
To: <sip:40341569039@192.168.0.148>
Contact: <sip:40341569039@192.168.0.254>
Call-ID: 4725fa8231d615141660e16101ae8d6c@192.168.0.254
CSeq: 102 INVITE
User-Agent: asterisk
Max-Forwards: 70
Date: Mon, 25 Feb 2008 13:17:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 289
v=0
o=root 73314 73314 IN IP4 192.168.0.254
s=session
c=IN IP4 192.168.0.254
t=0 0
m=audio 12774 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
mail*CLI>
<— SIP read from 192.168.0.148:5061 —>
<----
>
— (0 headers 1 lines) —
Retransmitting #3 (NAT) to 192.168.0.148:5060:
INVITE sip:40341569039@192.168.0.148 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK778e15c6;rport
From: “40341569039” <sip:40341569039@192.168.0.254>;tag=as521f8f44
To: <sip:40341569039@192.168.0.148>
Contact: <sip:40341569039@192.168.0.254>
Call-ID: 4725fa8231d615141660e16101ae8d6c@192.168.0.254
CSeq: 102 INVITE
User-Agent: asterisk
Max-Forwards: 70
Date: Mon, 25 Feb 2008 13:17:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 289
v=0
o=root 73314 73314 IN IP4 192.168.0.254
s=session
c=IN IP4 192.168.0.254
t=0 0
m=audio 12774 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Retransmitting #4 (NAT) to 192.168.0.148:5060:
INVITE sip:40341569039@192.168.0.148 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK778e15c6;rport
From: “40341569039” <sip:40341569039@192.168.0.254>;tag=as521f8f44
To: <sip:40341569039@192.168.0.148>
Contact: <sip:40341569039@192.168.0.254>
Call-ID: 4725fa8231d615141660e16101ae8d6c@192.168.0.254
CSeq: 102 INVITE
User-Agent: asterisk
Max-Forwards: 70
Date: Mon, 25 Feb 2008 13:17:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 289
v=0
o=root 73314 73314 IN IP4 192.168.0.254
s=session
c=IN IP4 192.168.0.254
t=0 0
m=audio 12774 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
NOTICE[73314]: chan_sip.c:2879 auto_congest: Auto-congesting SIP/40341569039-08746000
Scheduling destruction of SIP dialog ‘4725fa8231d615141660e16101ae8d6c@192.168.0.254’ in 32000 ms (Method: INVITE)
mail*CLI>
It seems that no matter what number I am dialling from within Spark, it gets translated by Asterisk-IM into my own extension number:
From: “40341569039” <sip:40341569039@192.168.0.254>;tag=as521f8f44
To: <sip:40341569039@192.168.0.148>
Here is another log while using X-Lite and the same 40341569039 extension:
<— SIP read from 193.16.148.244:5060 —>
SIP/2.0 100 trying – your call is important to us
Via: SIP/2.0/UDP 89.33.6.124:5060;branch=z9hG4bK603ded74;rport=5060
From: “40341569039” <sip:40341569039@netmaster.ro>;tag=as70426038
To: <sip:40341569059@sip.netmaster.ro>
Call-ID: 1cc5f0b447a76e27347573432ec45af0@netmaster.ro
CSeq: 102 INVITE
Server: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0
Is my setup wrong somewhere, or is it a problem with Asterisk-IM?
Dan
PS: sorry for the large post