Postdial delay


#1

I have problem when placing calls over a SIP trunk to a Callmanager phone from Asterisk. It takes approximately 20 second from the call is placed until Asterisk phone hears ringing. The problem seems to be on the Asterisk side. After Asterisk recieves an INVITE from 172.31.1.20 1), it waits approximately 20 seconds to retransmit the INVITE to the Callmanager server, see 2).

Asterisk server ip = 10.0.1.11
Asterisk endpoint ip = 172.31.1.20
callmanager server ip = 172.31.1.55
callmanager endpoint = 10.0.1.13

DebianACLI> sip debug
SIP Debugging enabled
DebianA
CLI>
<-- SIP read from 10.0.1.13:5060:

  1. — (0 headers 0 lines) Nat keepalive —
    DebianA*CLI>
    <-- SIP read from 172.31.1.20:5060:
    INVITE sip:4210@10.0.1.11 SIP/2.0
    Via: SIP/2.0/UDP 172.31.1.20:5060;rport;branch=z9hG4bK34C24C0585ED4E51A9BC34D9AFBF7D4A
    From: bej sip:301@10.0.1.11;tag=3489620998
    To: sip:4210@10.0.1.11
    Contact: sip:301@172.31.1.20:5060
    Call-ID: 288785DE-F0D5-482A-B37E-007BDA5CBFB6@172.31.1.20
    CSeq: 16898 INVITE
    Max-Forwards: 70
    Content-Type: application/sdp
    User-Agent: X-Lite release 1105x
    Content-Length: 281

v=0
o=301 169880924 169880960 IN IP4 172.31.1.20
s=X-Lite
c=IN IP4 172.31.1.20
t=0 0
m=audio 8000 RTP/AVP 0 8 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

— (11 headers 13 lines)—
Using INVITE request as basis request - 288785DE-F0D5-482A-B37E-007BDA5CBFB6@172.31.1.20
Sending to 172.31.1.20 : 5060 (non-NAT)
Found user '301’
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 172.31.1.20:8000
Found description format pcmu
Found description format pcma
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x60c (ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 4210 in internal (domain 10.0.1.11)
list_route: hop: sip:301@172.31.1.20:5060
Transmitting (no NAT) to 172.31.1.20:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.31.1.20:5060;rport;branch=z9hG4bK34C24C0585ED4E51A9BC34D9AFBF7D4A;received=172.31.1.20
From: bej sip:301@10.0.1.11;tag=3489620998
To: sip:4210@10.0.1.11
Call-ID: 288785DE-F0D5-482A-B37E-007BDA5CBFB6@172.31.1.20
CSeq: 16898 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: sip:4210@10.0.1.11
Content-Length: 0


-- Executing Dial("SIP/301-860f", "SIP/1#2138724210@172.31.1.55|10|t") in new stack

DebianA*CLI>
<-- SIP read from 172.31.1.20:5060:

— (0 headers 0 lines) Nat keepalive —
DebianA*CLI>
<-- SIP read from 10.0.1.13:5060:

— (0 headers 0 lines) Nat keepalive —
DebianA*CLI>
<-- SIP read from 10.0.1.13:5060:

  1. — (0 headers 0 lines) Nat keepalive —
    We’re at 10.0.1.11 port 18926
    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 172.31.1.55:5060:
    INVITE sip:1#2138724210@172.31.1.55 SIP/2.0
    Via: SIP/2.0/UDP 10.0.1.11:5060;branch=z9hG4bK721340df;rport
    From: “bej” sip:301@10.0.1.11;tag=as2f45d4f4
    To: sip:1#2138724210@172.31.1.55
    Contact: sip:301@10.0.1.11
    Call-ID: 2a571eed495a14a57aaa9f6c05de6caa@10.0.1.11
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Date: Thu, 13 Apr 2006 17:48:45 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Content-Type: application/sdp
    Content-Length: 232

v=0
o=root 1115 1115 IN IP4 10.0.1.11
s=session
c=IN IP4 10.0.1.11
t=0 0
m=audio 18926 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 1#2138724210@172.31.1.55

DebianA*CLI>
<-- SIP read from 172.31.1.55:49460:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.1.11:5060;branch=z9hG4bK721340df;rport
From: “bej” sip:301@10.0.1.11;tag=as2f45d4f4
To: sip:1#2138724210@172.31.1.55;tag=16777304
Date: Thu, 13 Apr 2006 17:48:45 GMT
Call-ID: 2a571eed495a14a57aaa9f6c05de6caa@10.0.1.11
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0

— (9 headers 0 lines)—
DebianA*CLI>
<-- SIP read from 172.31.1.55:49460:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.1.11:5060;branch=z9hG4bK721340df;rport
From: “bej” sip:301@10.0.1.11;tag=as2f45d4f4
To: sip:1#2138724210@172.31.1.55;tag=16777304
Date: Thu, 13 Apr 2006 17:48:45 GMT
Call-ID: 2a571eed495a14a57aaa9f6c05de6caa@10.0.1.11
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK
Allow-Events: telephone-event
Remote-Party-ID: sip:4210@172.31.1.55;party=called;screen=no;privacy=off
Contact: sip:1#2138724210@172.31.1.55:5060
Content-Length: 0

— (12 headers 0 lines)—
– SIP/172.31.1.55-501e is ringing
Transmitting (no NAT) to 172.31.1.20:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.31.1.20:5060;rport;branch=z9hG4bK34C24C0585ED4E51A9BC34D9AFBF7D4A;received=172.31.1.20
From: bej sip:301@10.0.1.11;tag=3489620998
To: sip:4210@10.0.1.11;tag=as618a94d9
Call-ID: 288785DE-F0D5-482A-B37E-007BDA5CBFB6@172.31.1.20
CSeq: 16898 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: sip:4210@10.0.1.11
Content-Length: 0


#2

Problem fixed by disabling the X-window display manager on startup.


#3

WOW…seem like a very weak machine then ?

Anyway, tnx for reporting back, so others with the same problem know where to look :smile: