[asterisk-users] Re: Asterisk + Huawei

Sun Oct 22 18:59:01 MST 2006

Thanks for your answer, here is some more debug information, if is a codec interrupt issue, how can i fix it?

My Sipura uses UID 1234. The huawei softswitch IP address is 10.220.0.2. The Asterisk IP address  is 10.223.6.98.

The Sipura is registered to the Asterisk box and the Asterisk box is registered to the Huawei softswitch. 

Thanks a lot for your help,

Carlos Andres Medina

------------------- INCOMING ------------------------------------------

    -- Executing Macro("SIP/10.220.0.2-08191e48", "incoming|SIP/1234") in new stack
    -- Executing Dial("SIP/10.220.0.2-08191e48", "SIP/1234|30") in new stack
We're at 10.223.6.98 port 19404
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (no NAT) to 10.223.6.99:5150:
INVITE sip:1234 at 10.223.6.99:5150 SIP/2.0
Via: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK3eb83872;rport
From: "Anonymous" <sip:Anonymous at 10.223.6.98>;tag=as448023d0
To: <sip:1234 at 10.223.6.99:5150>
Contact: <sip:Anonymous at 10.223.6.98>
Call-ID: 7addcc5162ce1ee57bc58cba40828bbd at 10.223.6.98
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 19 Oct 2006 01:56:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 236

v=0
o=root 1760 1760 IN IP4 10.223.6.98
s=session
c=IN IP4 10.223.6.98
t=0 0
m=audio 19404 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
-- Called 1234

<-- SIP read from 10.223.6.99:5150:
SIP/2.0 100 Trying
To: <sip:1234 at 10.223.6.99:5150>
From: "Anonymous" <sip:Anonymous at 10.223.6.98>;tag=as448023d0
Call-ID: 7addcc5162ce1ee57bc58cba40828bbd at 10.223.6.98
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK3eb83872
Server: Sipura/SPA2000-2.0.10(e)
Content-Length: 0

--- (8 headers 0 lines)---

<-- SIP read from 10.223.6.99:5150:
SIP/2.0 180 Ringing
To: <sip:1234 at 10.223.6.99:5150>;tag=e2a724add55f408bi0
From: "Anonymous" <sip:Anonymous at 10.223.6.98>;tag=as448023d0
Call-ID: 7addcc5162ce1ee57bc58cba40828bbd at 10.223.6.98
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK3eb83872
Server: Sipura/SPA2000-2.0.10(e)
Content-Length: 0

--- (8 headers 0 lines)---
    -- SIP/1234-08197388 is ringing
Transmitting (no NAT) to 10.220.0.2:5061:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.220.0.2:5061;branch=z9hG4bK94161ad88;received=10.220.0.2
From: Anonymous<sip:Anonymous at 10.220.0.2>;tag=961d1a68
To: <sip:4875129 at 10.223.6.98;user=phone>;tag=as40afbad8
Call-ID: 436fedbce988d7eea66f167d06a0558b at 10.220.0.2
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:4875129 at 10.223.6.98>
Content-Length: 0

<-- SIP read from 10.223.6.99:5150:
SIP/2.0 200 OK
To: <sip:1234 at 10.223.6.99:5150>;tag=e2a724add55f408bi0
From: "Anonymous" <sip:Anonymous at 10.223.6.98>;tag=as448023d0
Call-ID: 7addcc5162ce1ee57bc58cba40828bbd at 10.223.6.98
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK3eb83872
Contact: <sip:1234 at 10.223.6.99:5150>
Server: Sipura/SPA2000-2.0.10(e)
Content-Length: 229
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 78549 78549 IN IP4 10.223.6.99
s=-
c=IN IP4 10.223.6.99
t=0 0
m=audio 21101 RTP/AVP 8 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

--- (12 headers 12 lines)---
Found RTP audio format 8
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 10.223.6.99:21101
Found description format PCMA
Found description format NSE
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:1234 at 10.223.6.99:5150>
set_destination: Parsing <sip:1234 at 10.223.6.99:5150> for address/port to send to
set_destination: set destination to 10.223.6.99, port 5150
Transmitting (no NAT) to 10.223.6.99:5150:
ACK sip:1234 at 10.223.6.99:5150 SIP/2.0
Via: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK403f58ec;rport
From: "Anonymous" <sip:Anonymous at 10.223.6.98>;tag=as448023d0
To: <sip:1234 at 10.223.6.99:5150>;tag=e2a724add55f408bi0
Contact: <sip:Anonymous at 10.223.6.98>
Call-ID: 7addcc5162ce1ee57bc58cba40828bbd at 10.223.6.98
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
    -- SIP/1234-08197388 answered SIP/10.220.0.2-08191e48
We're at 10.223.6.98 port 15322
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.220.0.2:5061:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.220.0.2:5061;branch=z9hG4bK94161ad88;received=10.220.0.2
From: Anonymous<sip:Anonymous at 10.220.0.2>;tag=961d1a68
To: <sip:4875129 at 10.223.6.98;user=phone>;tag=as40afbad8
Call-ID: 436fedbce988d7eea66f167d06a0558b at 10.220.0.2
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:4875129 at 10.223.6.98>
Content-Type: application/sdp
Content-Length: 233

v=0
o=root 1760 1760 IN IP4 10.223.6.98
s=session
c=IN IP4 10.223.6.98
t=0 0
m=audio 15322 RTP/AVP 0 8 97
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=silenceSupp:off - - - -
    -- Attempting native bridge of SIP/10.220.0.2-08191e48 and SIP/1234-08197388

<-- SIP read from 10.220.0.2:5061:
ACK sip:4875129 at 10.223.6.98 SIP/2.0
Via: SIP/2.0/UDP 10.220.0.2:5061;branch=z9hG4bKf4540cd33
Call-ID: 436fedbce988d7eea66f167d06a0558b at 10.220.0.2
From: Anonymous<sip:Anonymous at 10.220.0.2>;tag=961d1a68
To: <sip:4875129 at 10.223.6.98;user=phone>;tag=as40afbad8
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

--- (8 headers 0 lines)---

<-- SIP read from 10.220.0.2:5061:
BYE sip:4875129 at 10.223.6.98 SIP/2.0
Via: SIP/2.0/UDP 10.220.0.2:5061;branch=z9hG4bK412198644
Call-ID: 436fedbce988d7eea66f167d06a0558b at 10.220.0.2
From: Anonymous<sip:Anonymous at 10.220.0.2>;tag=961d1a68
To: <sip:4875129 at 10.223.6.98;user=phone>;tag=as40afbad8
CSeq: 2 BYE
Reason: Q.850;cause=100;text="Invalid information element contents"
Max-Forwards: 70
Content-Length: 0

--- (9 headers 0 lines)---

--------------------- OUTGOING ------------------------------------------------------

<-- SIP read from 10.220.0.2:5060:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK32b32640;rport=5060
Call-ID: 6365afa34dc3ae2318bac62b13945272 at 10.223.6.98
From: "4875129"<sip:4875129 at 10.223.6.98>;tag=as1a151f0f
To: <sip:6024042 at 10.220.0.2>;tag=b1d10bb9
CSeq: 102 INVITE
Reason: Q.850;cause=98;text="Message not compatible with call state or message type non-existent or not implemented"
Content-Length: 0

--- (8 headers 0 lines)---
-- Got SIP response 503 "Service Unavailable" back from 10.220.0.2
Transmitting (no NAT) to 10.220.0.2:5060:
ACK sip:6024042 at 10.220.0.2 SIP/2.0
Via: SIP/2.0/UDP 10.223.6.98:5060;branch=z9hG4bK32b32640;rport
From: "4875129" <sip:4875129 at 10.223.6.98>;tag=as1a151f0f
To: <sip:6024042 at 10.220.0.2>;tag=b1d10bb9
Contact: <sip:4875129 at 10.223.6.98>
Call-ID: 6365afa34dc3ae2318bac62b13945272 at 10.223.6.98
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
-- SIP/epmbogota-08194768 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'SIP/1234-0818f228' status is 'CONGESTION'
Transmitting (no NAT) to 10.223.6.99:5150:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.223.6.99:5150;branch=z9hG4bK-8808b7d3;received=10.223.6.99
From: <sip:1234 at 10.223.6.98>;tag=1fdd8f37d10c2e33o0
To: <sip:6024042 at 10.223.6.98>;tag=as0d9917db
Call-ID: b4719687-fc4f4f23 at 10.223.6.99
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6024042 at 10.223.6.98>
Content-Length: 0
X-Asterisk-HangupCause: Circuit/channel congestion

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