[asterisk-users] calling in-out

Wed Aug 16 12:33:28 MST 2006

Hello people, I am having some issues with my new SIP provider.

The sip provider gives me only an IP address to configure my sip account,
since they do allow by IP address and not by username password.

This all configuration appears to work well, since I can originate a call
and it will ring the destination, and I can originate a call from PSTN and *
will "see" it. But none of both call difections will be stabilished.

If I originate a call from * to a PSTN number, with a sip debug I get:

Destroying call '086247827f2ab1c83b4f39a5389b17bc at 200.59.45.210'
pbx*CLI>
<-- SIP read from 200.123.190.50:5060:
SIP/2.0 500 Server Internal Error
To: <sip:44242904851 at 200.123.190.50>;tag=3364745030-621025
From: "CrossFone" <sip:Unknown at 200.59.45.210>;tag=as4d4398b9
Call-ID: 67d7050e602585775316c78b17d9ea4a at 200.59.45.210
CSeq: 102 INVITE
Contact: sip:44242904851 at 200.123.190.50:5060
Via: SIP/2.0/UDP 200.59.45.210:5060;branch=z9hG4bK6c94441d;rport
Content-Length: 0

--- (8 headers 0 lines)---
    -- Got SIP response 500 "Server Internal Error" back from 200.123.190.50
Transmitting (no NAT) to 200.123.190.50:5060:
ACK sip:44242904851 at 200.123.190.50 SIP/2.0
Via: SIP/2.0/UDP 200.59.45.210:5060;branch=z9hG4bK6c94441d;rport
From: "CrossFone" <sip:Unknown at 200.59.45.210>;tag=as4d4398b9
To: <sip:44242904851 at 200.123.190.50>;tag=3364745030-621025
Contact: <sip:Unknown at 200.59.45.210>
Call-ID: 67d7050e602585775316c78b17d9ea4a at 200.59.45.210
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
    -- SIP/CrossFone-087b3e40 is circuit-busy
Destroying call '67d7050e602585775316c78b17d9ea4a at 200.59.45.210'
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing Goto("SIP/1501-087acbf8", "s-CONGESTION|1") in new stack
    -- Goto (macro-dialout-trunk,s-CONGESTION,1)
    -- Executing NoOp("SIP/1501-087acbf8", "Dial failed due to CONGESTION")
in new stack
    -- Executing Macro("SIP/1501-087acbf8", "outisbusy|") in new stack
    -- Executing Playback("SIP/1501-087acbf8", "all-circuits-busy-now") in
new stack

If I make a call to my SIP number, it will ring till I pickup the phone,
when I pickup the phone, I get:

<-- SIP read from 201.216.206.221:62477:

--- (0 headers 1 lines)---
pbx*CLI>
<-- SIP read from 200.123.190.50:5060:
INVITE sip:1159174200 at 200.59.45.210 SIP/2.0
Max-Forwards: 70
Session-Expires: 3600;Refresher=uac
Supported: timer
To: 1159174200 <sip:1159174200 at 200.123.190.50>
From: <sip:1152184829 at 200.123.190.50:5060>;tag=3364745421-27664
Call-ID: 53757-3364745421-27638 at msw1.subnet32.net
CSeq: 1 INVITE
Via: SIP/2.0/UDP 200.123.190.50:5060;branch=b86b531fb60caa03195a218a6e8947fe
Contact: sip:1152184829 at 200.123.190.50:5060
Content-Type: application/sdp
Content-Length: 170

v=0
o=NexTone-MSW 1234 0 IN IP4 200.123.190.53
s=sip call
c=IN IP4 200.123.190.53
t=0 0
m=audio 21660 RTP/AVP 18 4 4 0
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes

--- (12 headers 8 lines)---
Using INVITE request as basis request -
53757-3364745421-27638 at msw1.subnet32.net
Sending to 200.123.190.50 : 5060 (non-NAT)
Found peer 'CrossFone'
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 4
Found RTP audio format 0
Peer audio RTP is at port 200.123.190.53:21660
Found description format G729
Capabilities: us - 0x10f (g723|gsm|ulaw|alaw|g729), peer - audio=0x105
(g723|ulaw|g729)/video=0x0 (nothing), combined - 0x105 (g723|ulaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing),
combined - 0x0 (nothing)
Looking for 1159174200 in from-sip-external (domain 200.59.45.210)
list_route: hop: <sip:1152184829 at 200.123.190.50:5060>
Transmitting (no NAT) to 200.123.190.50:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
200.123.190.50:5060;branch=b86b531fb60caa03195a218a6e8947fe;received=200.123
.190.50
From: <sip:1152184829 at 200.123.190.50:5060>;tag=3364745421-27664
To: 1159174200 <sip:1159174200 at 200.123.190.50>
Call-ID: 53757-3364745421-27638 at msw1.subnet32.net
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1159174200 at 200.59.45.210>
Content-Length: 0

---
    -- Executing NoOp("SIP/5060-087ace18", "Received incoming SIP connection
from unknown peer to 1159174200") in new stack
    -- Executing Set("SIP/5060-087ace18", "DID=1159174200") in new stack
    -- Executing Goto("SIP/5060-087ace18", "s|1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing GotoIf("SIP/5060-087ace18", "0?from-trunk|1159174200|1") in
new stack
    -- Executing Set("SIP/5060-087ace18", "TIMEOUT(absolute)=15") in new
stack
    -- Channel will hangup at 2006-08-16 19:27:24 UTC.
    -- Executing Answer("SIP/5060-087ace18", "") in new stack
We're at 200.59.45.210 port 19920
Adding codec 0x100 (g729) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 200.123.190.50:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
200.123.190.50:5060;branch=b86b531fb60caa03195a218a6e8947fe;received=200.123
.190.50
From: <sip:1152184829 at 200.123.190.50:5060>;tag=3364745421-27664
To: 1159174200 <sip:1159174200 at 200.123.190.50>;tag=as7635cbf2
Call-ID: 53757-3364745421-27638 at msw1.subnet32.net
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1159174200 at 200.59.45.210>
Content-Type: application/sdp
Content-Length: 231

v=0
o=root 2815 2815 IN IP4 200.59.45.210
s=session
c=IN IP4 200.59.45.210
t=0 0
m=audio 19920 RTP/AVP 18 4 0
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -

---
    -- Executing Wait("SIP/5060-087ace18", "2") in new stack
pbx*CLI>
<-- SIP read from 200.123.190.50:5060:
ACK sip:1159174200 at 200.59.45.210 SIP/2.0
Max-Forwards: 70
To: 1159174200 <sip:1159174200 at 200.123.190.50>;tag=as7635cbf2
From: <sip:1152184829 at 200.123.190.50:5060>;tag=3364745421-27664
Call-ID: 53757-3364745421-27638 at msw1.subnet32.net
CSeq: 1 ACK
Via: SIP/2.0/UDP 200.123.190.50:5060;branch=b86b531fb60caa03195a218a6e8947fe
Contact: sip:1152184829 at 200.123.190.50:5060
Content-Length: 0

--- (9 headers 0 lines)---
pbx*CLI>
<-- SIP read from 201.216.206.221:62364:

any idea?