Dear All,
I’m having some problems with setting up my Asterisk 1.0.9 box with a VoIP SIP provider which apparently uses a equipment called “HuaweiSoft X3000”. I was able to successfully register my asterisk box with them; however, whenever asterisk receives an incoming call and answers it, the call will be dropped immediately, and this happens all the time. If I use X-Lite to connect to this SIP Provider, everything works fine as expected, so I guess it’s either my configuration problem or an incompatibility bug in Asterisk with “Huawei” equipment?
Also please be noted that my Asterisk box is indeed a working one, because i also have an account with BroadVoice, and my box works just fine with BroadVoice, both incoming and outgoing.
Here is my sip.conf:
register => 35011111:123456:35011111@202.0.179.3
[202.0.179.3]
type=peer
user=phone
host=202.0.179.3
fromdomain=202.0.179.3
fromuser=35011111
secret=123456
username=35011111
insecure=very
context=default
authname=35011111
dtmfmode=inband
dtmf=inband
canreinvite=no
And here is my extensions.conf:
exten => 35011111,1,Answer()
exten => 35011111,2,Wait,1
exten => 35011111,3,DigitTimeout,5
exten => 35011111,4,ResponseTimeout,10
exten => 35011111,5,Background(intro-clip)
Here is a SIP Debug dump when I receive a call from the SIP Provider, notice how the call gets dropped unexpectedly with the server sending me a “BYE” message:
Sip read:
INVITE sip:35011111@67.93.49.57:19161;user=phone SIP/2.0
From: sip:OutOfArea@202.0.179.3;user=phone;tag=d5903845
To: sip:35011111@67.93.49.57;user=phone
CSeq: 1 INVITE
Call-ID: dc161cbe2e9b78274b304c37218d82f8@sx3000
Via: SIP/2.0/UDP 202.0.179.3:5060;branch=z9hG4bK4f95d72d6
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER
Max-Forwards: 70
Supported: 100rel
Contact: sip:OutOfArea@202.0.179.3:5060;user=phone
Content-Length: 294
Content-Type: application/sdp
v=0
o=HuaweiSoftX3000 2102168 2102168 IN IP4 10.0.1.36
s=Sip Call
c=IN IP4 202.0.179.3
t=0 0
m=audio 19134 RTP/AVP 8 0 18 4 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=fmtp:18 annexb=yes
12 headers, 13 lines
Using latest request as basis request
Sending to 202.0.179.3 : 5060 (non-NAT)
Found peer '202.0.179.3’
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 97
Peer audio RTP is at port 202.0.179.3:19134
Found description format PCMA
Found description format PCMU
Found description format G729
Found description format G723
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
Looking for 35011111 in default
list_route: hop: sip:OutOfArea@202.0.179.3:5060;user=phone
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 202.0.179.3:5060;branch=z9hG4bK4f95d72d6
From: sip:OutOfArea@202.0.179.3;user=phone;tag=d5903845
To: sip:35011111@67.93.49.57;user=phone
Call-ID: dc161cbe2e9b78274b304c37218d82f8@sx3000
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:35011111@192.168.0.110
Content-Length: 0
to 202.0.179.3:5060
– Executing Answer(“SIP/35011111-05a8”, “”) in new stack
We’re at 192.168.0.110 port 13622
Answering with capability 0x2 (gsm)
Answering with capability 0x4 (ulaw)
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 202.0.179.3:5060;branch=z9hG4bK4f95d72d6
From: sip:OutOfArea@202.0.179.3;user=phone;tag=d5903845
To: sip:35011111@67.93.49.57;user=phone;tag=as5c6e715b
Call-ID: dc161cbe2e9b78274b304c37218d82f8@sx3000
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:35011111@192.168.0.110
Content-Type: application/sdp
Content-Length: 260
v=0
o=root 8538 8538 IN IP4 192.168.0.110
s=session
c=IN IP4 192.168.0.110
t=0 0
m=audio 13622 RTP/AVP 3 0 8 97
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=silenceSupp:off - - - -
to 202.0.179.3:5060
– Executing Wait(“SIP/35011111-05a8”, “1”) in new stack
Sip read:
ACK sip:35011111@67.93.49.57:19161;user=phone SIP/2.0
From: sip:OutOfArea@202.0.179.3;user=phone;tag=d5903845
To: sip:35011111@67.93.49.57;user=phone;tag=as5c6e715b
CSeq: 1 ACK
Call-ID: dc161cbe2e9b78274b304c37218d82f8@sx3000
Via: SIP/2.0/UDP 202.0.179.3:5060;branch=z9hG4bKdc7345f2a
Max-Forwards: 70
Content-Length: 0
8 headers, 0 lines
Sip read:
BYE sip:35011111@67.93.49.57:19161;user=phone SIP/2.0
From: sip:OutOfArea@202.0.179.3;user=phone;tag=d5903845
To: sip:35011111@67.93.49.57;user=phone;tag=as5c6e715b
CSeq: 2 BYE
Call-ID: dc161cbe2e9b78274b304c37218d82f8@sx3000
Via: SIP/2.0/UDP 202.0.179.3:5060;branch=z9hG4bKca96826cb
Reason: Q.850;cause=“100”;text="unknown"
Max-Forwards: 70
Content-Length: 0
9 headers, 0 lines
Sending to 202.0.179.3 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 202.0.179.3:5060;branch=z9hG4bKca96826cb
From: sip:OutOfArea@202.0.179.3;user=phone;tag=d5903845
To: sip:35011111@67.93.49.57;user=phone;tag=as5c6e715b
Call-ID: dc161cbe2e9b78274b304c37218d82f8@sx3000
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:35011111@192.168.0.110
Content-Length: 0
to 202.0.179.3:5060
== Spawn extension (default, 35011111, 2) exited non-zero on 'SIP/35011111-05a8’
Destroying call ‘dc161cbe2e9b78274b304c37218d82f8@sx3000’
Sorry for the long message… but any help will be greatly appreciated. Thanks!!