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Jake Wicke

2008-Feb-01 17:34 UTC

[asterisk-users] Asterisk 1.6 - Problems with SIP/REFER

I am having issues with transfers (SIP/REFER) using Asterisk 1.6.  You will find
the SIP debug below.

There are three phones in this setup.  5253 and 5258 are Aastra 53i telephones,
101 is a standard phone connected through an Audiocodes gateway.  All phones are
registered in context "phones" and are set to not allow reinvites. 
All phones can dial each other directly.  The dialplan looks as follows:

[phones]
Exten => 5253,1,Dial(SIP/5253,10)
Exten => 5878,1,Dial(SIP/5878,10)
Exten => 101,1,Dial(SIP/101 at audiocodes,10)

Transfer fails regardless of the order (101 calls 5878, 5878 transfers to 5253
or 5878 calls 5253, 5253 transfers to 101, etc)

I do not understand the message "Spawn Extension (phones, 101, 0) exited
non-zero" in the debug - there is no "priority zero" in a
dialplan - priority should start at 1.  What is this message telling me?

What do I need to do to allow these phones to transfer calls between each other?
Any help is greatly appreciated!

Here is the debug:

*CLI>   == Using SIP RTP CoS mark 5
  == Using UDPTL CoS mark 5
    -- Executing [5878 at phones:1] Dial("SIP/5253-0823eab0",
"SIP/5878") in new stack
  == Using SIP RTP CoS mark 5
  == Using UDPTL CoS mark 5
Audio is at 10.7.10.1 port 19968
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.7.10.51:5060:
INVITE sip:5878 at 10.7.10.51:5060 SIP/2.0
Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK38d90448;rport
Max-Forwards: 70
From: "5253" <sip:5253 at 10.7.10.1>;tag=as05a48c1a
To: <sip:5878 at 10.7.10.51:5060>
Contact: <sip:5253 at 10.7.10.1:5060>
Call-ID: 7903ae4900c136a43e6ef74f29c582a5 at 10.7.10.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0-beta2
Remote-Party-ID: "5253" <sip:5253 at
10.7.10.1>;privacy=off;screen=no
Date: Wed, 30 Jan 2008 01:12:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 864806723 864806723 IN IP4 10.7.10.1
s=Asterisk PBX 1.6.0-beta2
c=IN IP4 10.7.10.1
t=0 0
m=audio 19968 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 5878

<--- SIP read from UDP://10.7.10.51:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
10.7.10.1:5060;branch=z9hG4bK38d90448;rport=5060;received=10.7.10.1
From: "5253" <sip:5253 at 10.7.10.1>;tag=as05a48c1a
To: <sip:5878 at 10.7.10.51:5060>;tag=694417843
Call-ID: 7903ae4900c136a43e6ef74f29c582a5 at 10.7.10.1
CSeq: 102 INVITE
Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK,
SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Call-Info: <sip:10.7.10.1>;appearance-index=1
Contact: 5878 <sip:5878 at 10.7.10.51:5060>
Server: Aastra 53i/2.1.0.2145
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
    -- SIP/5878-08250098 is ringing

<--- SIP read from UDP://10.7.10.51:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.7.10.1:5060;branch=z9hG4bK38d90448;rport=5060;received=10.7.10.1
From: "5253" <sip:5253 at 10.7.10.1>;tag=as05a48c1a
To: <sip:5878 at 10.7.10.51:5060>;tag=694417843
Call-ID: 7903ae4900c136a43e6ef74f29c582a5 at 10.7.10.1
CSeq: 102 INVITE
Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK,
SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Call-Info: <sip:10.7.10.1>;appearance-index=1
Contact: 5878 <sip:5878 at 10.7.10.51:5060>
Server: Aastra 53i/2.1.0.2145
Supported: timer, replaces
Content-Type: application/sdp
Content-Length: 313

v=0
o=MxSIP 0 0 IN IP4 10.7.10.51
s=SIP Call
c=IN IP4 10.7.10.51
t=0 0
m=audio 3000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:ZXJ1UmhNLDFmQGNHYGAnRlpKbjEudk9Gfjh8blo/
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<------------->
--- (14 headers 13 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 10.7.10.51:3000
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.7.10.51:3000
list_route: hop: <sip:5878 at 10.7.10.51:5060>
set_destination: Parsing <sip:5878 at 10.7.10.51:5060> for address/port to
send to
set_destination: set destination to 10.7.10.51, port 5060
Transmitting (NAT) to 10.7.10.51:5060:
ACK sip:5878 at 10.7.10.51:5060 SIP/2.0
Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK6476d991;rport
Max-Forwards: 70
From: "5253" <sip:5253 at 10.7.10.1>;tag=as05a48c1a
To: <sip:5878 at 10.7.10.51:5060>;tag=694417843
Contact: <sip:5253 at 10.7.10.1:5060>
Call-ID: 7903ae4900c136a43e6ef74f29c582a5 at 10.7.10.1
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0-beta2
Remote-Party-ID: "5253" <sip:5253 at
10.7.10.1>;privacy=off;screen=no
Content-Length: 0


---
    -- SIP/5878-08250098 answered SIP/5253-0823eab0
    -- Packet2Packet bridging SIP/5253-0823eab0 and SIP/5878-08250098

<--- SIP read from UDP://10.7.10.51:5060 --->
INVITE sip:5253 at 10.7.10.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.7.10.51:5060;branch=z9hG4bKd79bb4c662d65595a
Max-Forwards: 70
From: <sip:5878 at 10.7.10.51:5060>;tag=694417843
To: "5253" <sip:5253 at 10.7.10.1>;tag=as05a48c1a
Call-ID: 7903ae4900c136a43e6ef74f29c582a5 at 10.7.10.1
CSeq: 20367 INVITE
Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK,
SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Contact: 5878 <sip:5878 at 10.7.10.51:5060>
Supported: timer, 100rel, replaces
User-Agent: Aastra 53i/2.1.0.2145
Content-Type: application/sdp
Content-Length: 278

v=0
o=MxSIP 0 1 IN IP4 10.7.10.51
s=SIP Call
c=IN IP4 10.7.10.51
t=0 0
m=audio 3000 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:on - - - -
a=fmtp:101 0-15
a=ptime:30
a=sendonly

<------------->
--- (14 headers 14 lines) ---
Sending to 10.7.10.51 : 5060 (NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 10.7.10.51:3000
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.7.10.51:3000

<--- Transmitting (NAT) to 10.7.10.51:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.7.10.51:5060;branch=z9hG4bKd79bb4c662d65595a;received=10.7.10.51
From: <sip:5878 at 10.7.10.51:5060>;tag=694417843
To: "5253" <sip:5253 at 10.7.10.1>;tag=as05a48c1a
Call-ID: 7903ae4900c136a43e6ef74f29c582a5 at 10.7.10.1
CSeq: 20367 INVITE
User-Agent: Asterisk PBX 1.6.0-beta2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:5253 at 10.7.10.1:5060>
Content-Length: 0


<------------>
Audio is at 10.7.10.1 port 19968
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to 10.7.10.51:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.7.10.51:5060;branch=z9hG4bKd79bb4c662d65595a;received=10.7.10.51
From: <sip:5878 at 10.7.10.51:5060>;tag=694417843
To: "5253" <sip:5253 at 10.7.10.1>;tag=as05a48c1a
Call-ID: 7903ae4900c136a43e6ef74f29c582a5 at 10.7.10.1
CSeq: 20367 INVITE
User-Agent: Asterisk PBX 1.6.0-beta2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:5253 at 10.7.10.1:5060>
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 864806723 864806724 IN IP4 10.7.10.1
s=Asterisk PBX 1.6.0-beta2
c=IN IP4 10.7.10.1
t=0 0
m=audio 19968 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=recvonly

<------------>
    -- Started music on hold, class 'default', on SIP/5253-0823eab0

<--- SIP read from UDP://10.7.10.51:5060 --->
ACK sip:5253 at 10.7.10.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.7.10.51:5060;branch=z9hG4bKb0a1c610b19e25613
Max-Forwards: 70
From: <sip:5878 at 10.7.10.51:5060>;tag=694417843
To: "5253" <sip:5253 at 10.7.10.1>;tag=as05a48c1a
Call-ID: 7903ae4900c136a43e6ef74f29c582a5 at 10.7.10.1
CSeq: 20367 ACK
User-Agent: Aastra 53i/2.1.0.2145
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---

<--- SIP read from UDP://10.7.10.51:5060 --->
REFER sip:5253 at 10.7.10.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.7.10.51:5060;branch=z9hG4bKb535a71447a137a4e
Max-Forwards: 70
From: <sip:5878 at 10.7.10.51:5060>;tag=694417843
To: "5253" <sip:5253 at 10.7.10.1>;tag=as05a48c1a
Call-ID: 7903ae4900c136a43e6ef74f29c582a5 at 10.7.10.1
CSeq: 20368 REFER
Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK,
SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Contact: 5878 <sip:5878 at 10.7.10.51:5060>
Refer-To: 101 <sip:101 at 10.7.10.1:5060>
Referred-By: <sip:5878 at 10.7.10.1>
Supported: timer
User-Agent: Aastra 53i/2.1.0.2145
Content-Length: 0


<------------->
--- (15 headers 0 lines) ---
Call 7903ae4900c136a43e6ef74f29c582a5 at 10.7.10.1 got a SIP call transfer from
caller: (REFER)!
SIP transfer to extension 101 at phones by 5878 at 10.7.10.1

<--- Transmitting (NAT) to 10.7.10.51:5060 --->
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP
10.7.10.51:5060;branch=z9hG4bKb535a71447a137a4e;received=10.7.10.51
From: <sip:5878 at 10.7.10.51:5060>;tag=694417843
To: "5253" <sip:5253 at 10.7.10.1>;tag=as05a48c1a
Call-ID: 7903ae4900c136a43e6ef74f29c582a5 at 10.7.10.1
CSeq: 20368 REFER
User-Agent: Asterisk PBX 1.6.0-beta2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:5253 at 10.7.10.1:5060>
Content-Length: 0


<------------>
set_destination: Parsing <sip:5878 at 10.7.10.51:5060> for address/port to
send to
set_destination: set destination to 10.7.10.51, port 5060
Reliably Transmitting (NAT) to 10.7.10.51:5060:
NOTIFY sip:5878 at 10.7.10.51:5060 SIP/2.0
Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK23e5b645;rport
Max-Forwards: 70
From: "5253" <sip:5253 at 10.7.10.1>;tag=as05a48c1a
To: <sip:5878 at 10.7.10.51:5060>;tag=694417843
Contact: <sip:5253 at 10.7.10.1:5060>
Call-ID: 7903ae4900c136a43e6ef74f29c582a5 at 10.7.10.1
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX 1.6.0-beta2
Remote-Party-ID: "5253" <sip:5253 at
10.7.10.1>;privacy=off;screen=no
Event: refer;id=20368
Subscription-state: active
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 21

SIP/2.0 183 Ringing

---
    -- Stopped music on hold on SIP/5253-0823eab0
set_destination: Parsing <sip:5878 at 10.7.10.51:5060> for address/port to
send to
set_destination: set destination to 10.7.10.51, port 5060
Reliably Transmitting (NAT) to 10.7.10.51:5060:
NOTIFY sip:5878 at 10.7.10.51:5060 SIP/2.0
Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK1431d66c;rport
Max-Forwards: 70
From: "5253" <sip:5253 at 10.7.10.1>;tag=as05a48c1a
To: <sip:5878 at 10.7.10.51:5060>;tag=694417843
Contact: <sip:5253 at 10.7.10.1:5060>
Call-ID: 7903ae4900c136a43e6ef74f29c582a5 at 10.7.10.1
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX 1.6.0-beta2
Remote-Party-ID: "5253" <sip:5253 at
10.7.10.1>;privacy=off;screen=no
Event: refer;id=20368
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 16

SIP/2.0 200 Ok

---
Scheduling destruction of SIP dialog '7903ae4900c136a43e6ef74f29c582a5 at
10.7.10.1' in 32000 ms (Method: REFER)
  == Spawn extension (phones, 101, 0) exited non-zero on
'SIP/5253-0823eab0'

<--- SIP read from UDP://10.7.10.51:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.7.10.1:5060;branch=z9hG4bK23e5b645;rport=5060;received=10.7.10.1
From: "5253" <sip:5253 at 10.7.10.1>;tag=as05a48c1a
To: <sip:5878 at 10.7.10.51:5060>;tag=694417843
Call-ID: 7903ae4900c136a43e6ef74f29c582a5 at 10.7.10.1
CSeq: 103 NOTIFY
Server: Aastra 53i/2.1.0.2145
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP://10.7.10.51:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.7.10.1:5060;branch=z9hG4bK1431d66c;rport=5060;received=10.7.10.1
From: "5253" <sip:5253 at 10.7.10.1>;tag=as05a48c1a
To: <sip:5878 at 10.7.10.51:5060>;tag=694417843
Call-ID: 7903ae4900c136a43e6ef74f29c582a5 at 10.7.10.1
CSeq: 104 NOTIFY
Server: Aastra 53i/2.1.0.2145
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
Retransmitting #1 (NAT) to 10.7.10.51:5060:
NOTIFY sip:5878 at 10.7.10.51:5060 SIP/2.0
Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK23e5b645;rport
Max-Forwards: 70
From: "5253" <sip:5253 at 10.7.10.1>;tag=as05a48c1a
To: <sip:5878 at 10.7.10.51:5060>;tag=694417843
Contact: <sip:5253 at 10.7.10.1:5060>
Call-ID: 7903ae4900c136a43e6ef74f29c582a5 at 10.7.10.1
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX 1.6.0-beta2
Remote-Party-ID: "5253" <sip:5253 at
10.7.10.1>;privacy=off;screen=no
Event: refer;id=20368
Subscription-state: active
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 21

SIP/2.0 183 Ringing

---

<--- SIP read from UDP://10.7.10.51:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.7.10.1:5060;branch=z9hG4bK23e5b645;rport=5060;received=10.7.10.1
From: "5253" <sip:5253 at 10.7.10.1>;tag=as05a48c1a
To: <sip:5878 at 10.7.10.51:5060>;tag=694417843
Call-ID: 7903ae4900c136a43e6ef74f29c582a5 at 10.7.10.1
CSeq: 103 NOTIFY
Server: Aastra 53i/2.1.0.2145
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Retransmitting #2 (NAT) to 10.7.10.51:5060:
NOTIFY sip:5878 at 10.7.10.51:5060 SIP/2.0
Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK23e5b645;rport
Max-Forwards: 70
From: "5253" <sip:5253 at 10.7.10.1>;tag=as05a48c1a
To: <sip:5878 at 10.7.10.51:5060>;tag=694417843
Contact: <sip:5253 at 10.7.10.1:5060>
Call-ID: 7903ae4900c136a43e6ef74f29c582a5 at 10.7.10.1
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX 1.6.0-beta2
Remote-Party-ID: "5253" <sip:5253 at
10.7.10.1>;privacy=off;screen=no
Event: refer;id=20368
Subscription-state: active
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 21

SIP/2.0 183 Ringing

---

<--- SIP read from UDP://10.7.10.51:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.7.10.1:5060;branch=z9hG4bK23e5b645;rport=5060;received=10.7.10.1
From: "5253" <sip:5253 at 10.7.10.1>;tag=as05a48c1a
To: <sip:5878 at 10.7.10.51:5060>;tag=694417843
Call-ID: 7903ae4900c136a43e6ef74f29c582a5 at 10.7.10.1
CSeq: 103 NOTIFY
Server: Aastra 53i/2.1.0.2145
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Retransmitting #3 (NAT) to 10.7.10.51:5060:
NOTIFY sip:5878 at 10.7.10.51:5060 SIP/2.0
Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK23e5b645;rport
Max-Forwards: 70
From: "5253" <sip:5253 at 10.7.10.1>;tag=as05a48c1a
To: <sip:5878 at 10.7.10.51:5060>;tag=694417843
Contact: <sip:5253 at 10.7.10.1:5060>
Call-ID: 7903ae4900c136a43e6ef74f29c582a5 at 10.7.10.1
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX 1.6.0-beta2
Remote-Party-ID: "5253" <sip:5253 at
10.7.10.1>;privacy=off;screen=no
Event: refer;id=20368
Subscription-state: active
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 21

SIP/2.0 183 Ringing

---

<--- SIP read from UDP://10.7.10.51:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.7.10.1:5060;branch=z9hG4bK23e5b645;rport=5060;received=10.7.10.1
From: "5253" <sip:5253 at 10.7.10.1>;tag=as05a48c1a
To: <sip:5878 at 10.7.10.51:5060>;tag=694417843
Call-ID: 7903ae4900c136a43e6ef74f29c582a5 at 10.7.10.1
CSeq: 103 NOTIFY
Server: Aastra 53i/2.1.0.2145
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
[Jan 29 19:12:53] NOTICE[19010]: chan_sip.c:8869 sip_reregister:    --
Re-registration for  6087294353 at sip.broadvoice.com@sip.broadvoice.com
[Jan 29 19:12:53] NOTICE[19010]: chan_sip.c:14782 handle_response_register:
Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling
reregistration in 23 s)
Retransmitting #4 (NAT) to 10.7.10.51:5060:
NOTIFY sip:5878 at 10.7.10.51:5060 SIP/2.0
Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK23e5b645;rport
Max-Forwards: 70
From: "5253" <sip:5253 at 10.7.10.1>;tag=as05a48c1a
To: <sip:5878 at 10.7.10.51:5060>;tag=694417843
Contact: <sip:5253 at 10.7.10.1:5060>
Call-ID: 7903ae4900c136a43e6ef74f29c582a5 at 10.7.10.1
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX 1.6.0-beta2
Remote-Party-ID: "5253" <sip:5253 at
10.7.10.1>;privacy=off;screen=no
Event: refer;id=20368
Subscription-state: active
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 21

SIP/2.0 183 Ringing

---

<--- SIP read from UDP://10.7.10.51:5060 --->
SIP/2.0 500 CSeq Number Out of order
Via: SIP/2.0/UDP
10.7.10.1:5060;branch=z9hG4bK23e5b645;rport=5060;received=10.7.10.1
From: "5253" <sip:5253 at 10.7.10.1>;tag=as05a48c1a
To: <sip:5878 at 10.7.10.51:5060>;tag=694417843
Call-ID: 7903ae4900c136a43e6ef74f29c582a5 at 10.7.10.1
CSeq: 103 NOTIFY
Server: Aastra 53i/2.1.0.2145
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---


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Grey Man

2008-Feb-02 20:59 UTC

head link

[asterisk-users] Asterisk 1.6 - Problems with SIP/REFER

> ----- Original Message ----
> From: Jake Wicke <jake at nxtphase.net>
> To: "asterisk-users at lists.digium.com" <asterisk-users at
lists.digium.com>
> Sent: Friday, 1 February, 2008 5:34:12 PM
> Subject: [asterisk-users] Asterisk 1.6 - Problems with SIP/REFER
>  
> I am having issues with transfers (SIP/REFER) using Asterisk 1.6.  You will
find the SIP debug below.
> 
>  There are three phones in this setup.  5253 and 5258 are Aastra 53i
telephones, 101 is a standard phone connected through an Audiocodes gateway. 
All phones are registered in context ?phones? and are set to not allow
reinvites.  All phones can dial each other directly.  The dialplan looks as
follows:
 >  
> [phones]
 Exten => 5253,1,Dial(SIP/5253,10)

 Exten => 5878,1,Dial(SIP/5878,10)

 Exten => 101,1,Dial(SIP/101 at audiocodes,10)

  
> Transfer fails regardless of the order (101 calls 5878, 5878 transfers to
5253 or 5878 calls 5253, 5253 transfers to 101, etc)
  
> I do not understand the message ?Spawn Extension (phones, 101, 0) exited
non-zero? in the debug ? there is no ?priority zero? in a dialplan ? priority
should start at 1.  What is this message telling me?
>
>
> What do I need to do to allow these phones to transfer calls between each
other?  Any help is greatly appreciated!
  

Hi Jake,

I don't have the answer but I did look at your trace and something about the
way the transfer is being done from your phones is not quite right. You're
calling 5253, then calling 5878 and then requesting a blind transfer of 5253 to
5878. However at that stage 5878 is already on the phone. I suspect the transfer
should be being requested as an attended one not a blind one.

Regards,

Greyman.




      Get the name you always wanted with the new y7mail email address.
www.yahoo7.com.au/y7mail

Johansson Olle E

2008-Feb-02 21:27 UTC

head link

[asterisk-users] Asterisk 1.6 - Problems with SIP/REFER

1 feb 2008 kl. 18.34 skrev Jake Wicke:
> I am having issues with transfers (SIP/REFER) using Asterisk 1.6.   
> You will find the SIP debug below.
>
When you have issues, it's always a good idea to check the bug  
tracker. There might be other people having the same issues, in some  
cases, there's also a solution. If you don't find it on first search,  
make sure you also search in resolved and closed issues.

For this case, we do have open bug reports. The same issue exists in  
1.4 and we're working on a solution.  Stay tuned.

Have a nice weekend!

/Olle

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