I'm using the Debian Sarge package of Asterisk - 1.0.7 + bristuff.
When
I call from a zap channel or a SIP phone to another SIP phone,
then dial
*8 from a third SIP phone, I get 503 Service Unavailable on the
third phone and I get this at the Asterisk console:
Jun 28 09:01:24 DEBUG[16774]: res_features.c:1709 ast_pickup_call:
No call pickup possible...
Jun 28 09:01:24 NOTICE[16774]: chan_sip.c:7402 handle_request:
Nothing to pick up
I'd appreciate hearing from anyone that has this working.
Here's my sip.conf, features.conf, and zapata.conf:
# < zapata.conf sed 's/;.*//g' | grep -v ^$
[trunkgroups]
[channels]
context=default
switchtype=national
signalling=em_w
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
callerid=asreceived
callprogress=yes
musiconhold=default
channel => 1-24
# < features.conf sed 's/;.*//g' | grep -v ^$
[general]
parkext => 700
parkpos => 701-720
context => parkedcalls
pickupexten = *8
# < sip.conf sed 's/;.*//g' | grep -v ^$ | grep -v '^[ ]' | sed s/
secret=.*/secret=donttell/g
[general]
context=default
callgroup=1
pickupgroup=1
port=5060
bindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw
allow=alaw
allow=g723.1
allow=g729
callgroup=1
pickupgroup=1
context=default
nat=no
canreinvite=yes
dtmfmode=rfc2833
incominglimit=4
[1310]
username=1310
secret=donttell
type=friend
host=dynamic
callerid=Grandstream SIP <1310>
[EMAIL PROTECTED]
[i1310]
username=i1310
secret=donttell
type=friend
host=dynamic
callerid=Grandstream SIP <1310>
[1311]
username=1311
secret=donttell
type=friend
host=dynamic
callerid=John Jacob Jingleheime <1311>
[EMAIL PROTECTED]
[1312]
username=1312
secret=donttell
type=friend
host=dynamic
callerid=Cisco 7960G Test <1312>
[EMAIL PROTECTED]
FWIW, I get identical behavior with callgroup=/pickupgroup= specified
for each extension. Here's some sanitized verbose output with SIP
debugging enabled:
-- Starting simple switch on 'Zap/24-1'
Jun 28 10:43:18 DEBUG[16774]: chan_sip.c:771 __sip_autodestruct:
Auto destroying call 'a01052a-13c4-42c104ea-371e-1957'
Destroying call 'a01052a-13c4-42c104ea-371e-1957'
Jun 28 10:43:19 DEBUG[17450]: chan_zap.c:4242 zt_read: DTMF digit:
1 on Zap/24-1
Jun 28 10:43:19 DEBUG[17450]: chan_zap.c:4242 zt_read: DTMF digit:
3 on Zap/24-1
Jun 28 10:43:19 DEBUG[17450]: chan_zap.c:4242 zt_read: DTMF digit:
1 on Zap/24-1
Jun 28 10:43:20 DEBUG[17450]: chan_zap.c:4242 zt_read: DTMF digit:
2 on Zap/24-1
Jun 28 10:43:20 DEBUG[17450]: chan_zap.c:1381 zt_enable_ec:
Enabled echo cancellation on channel 24
-- Executing Macro("Zap/24-1", "stdexten|1312|SIP/1312") in
new stack
-- Executing Dial("Zap/24-1", "SIP/1312|20") in new stack
Jun 28 10:43:20 DEBUG[17450]: chan_sip.c:1309 create_addr: Setting
NAT on RTP to 0
Jun 28 10:43:20 DEBUG[17450]: chan_sip.c:1487 sip_call: Outgoing
Call for 1312
Jun 28 10:43:20 DEBUG[17450]: chan_sip.c:1620 update_user_counter:
Call from user '1312' is 1 out of 0
We're at asterisk.server.ip.addr port 19630
Answering/Requesting with root capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with preferred capability 0x1 (g723)
Answering with preferred capability 0x100 (g729)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 13 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED]:5061 SIP/2.0
Via: SIP/2.0/UDP asterisk.server.ip.addr:5060;branch=z9hG4bK359ec760
From: "asterisk"
<sip:[EMAIL PROTECTED]>;tag=as61d8a13d
To: <sip:[EMAIL PROTECTED]:5061>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 28 Jun 2005 17:43:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 284
v=0
o=root 17450 17450 IN IP4 asterisk.server.ip.addr
s=session
c=IN IP4 asterisk.server.ip.addr
t=0 0
m=audio 19630 RTP/AVP 0 8 4 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(no NAT) to called.phone.ip.addr:5061
-- Called 1312
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP asterisk.server.ip.addr:5060;branch=z9hG4bK359ec760
From: "asterisk"
<sip:[EMAIL PROTECTED]>;tag=as61d8a13d
To: <sip:[EMAIL PROTECTED]:5061>
Call-ID: [EMAIL PROTECTED]
Date: Tue, 28 Jun 2005 17:43:20 GMT
CSeq: 102 INVITE
Server: CSCO/7
Contact: <sip:[EMAIL PROTECTED]:5061>
Content-Length: 0
10 headers, 0 lines
Jun 28 10:43:20 DEBUG[16774]: chan_sip.c:872 __sip_semi_ack:
(Provisional) Stopping retransmission (but retaining packet) on
'[EMAIL PROTECTED]' Request
102: Found
Sip read:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP asterisk.server.ip.addr:5060;branch=z9hG4bK359ec760
From: "asterisk"
<sip:[EMAIL PROTECTED]>;tag=as61d8a13d
To: <sip:[EMAIL PROTECTED]:
5061>;tag=001280b9cebf00025bfd45ed-7102ff29
Call-ID: [EMAIL PROTECTED]
Date: Tue, 28 Jun 2005 17:43:20 GMT
CSeq: 102 INVITE
Server: CSCO/7
Contact: <sip:[EMAIL PROTECTED]:5061>
Content-Length: 0
10 headers, 0 lines
Jun 28 10:43:20 DEBUG[16774]: chan_sip.c:872 __sip_semi_ack:
(Provisional) Stopping retransmission (but retaining packet) on
'[EMAIL PROTECTED]' Request
102: Found
-- SIP/1312-c824 is ringing
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP pickup.phone.ip.addr;branch=z9hG4bKa04377a7f5578f3c
From: "Test SIP" <sip:[EMAIL PROTECTED]>;tag=5eba6d75ff7e1e47
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]>
Supported: replaces, timer
Call-ID: [EMAIL PROTECTED]
CSeq: 48200 INVITE
User-Agent: Grandstream GXP2000 1.0.1.9
Max-Forwards: 70
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRAC
K
Content-Type: application/sdp
Content-Length: 302
v=0
o=1310 8000 8000 IN IP4 pickup.phone.ip.addr
s=SIP Call
c=IN IP4 pickup.phone.ip.addr
t=0 0
m=audio 5004 RTP/AVP 0 8 3 4 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
13 headers, 15 lines
Using latest request as basis request
Sending to pickup.phone.ip.addr : 5060 (non-NAT)
Jun 28 10:43:23 DEBUG[16774]: chan_sip.c:5441 check_user_full:
Setting NAT on RTP to 0
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP pickup.phone.ip.addr;branch=z9hG4bKa04377a7f5578f3c
From: "Test SIP" <sip:[EMAIL PROTECTED]>;tag=5eba6d75ff7e1e47
To: <sip:[EMAIL PROTECTED]>;tag=as114aad8b
Call-ID: [EMAIL PROTECTED]
CSeq: 48200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Proxy-Authenticate: Digest realm="asterisk", nonce="30b68bfa"
Content-Length: 0
to pickup.phone.ip.addr:5060
Scheduling destruction of call
'[EMAIL PROTECTED]' in 15000 ms
Found user '1310'
Sip read:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP pickup.phone.ip.addr;branch=z9hG4bKa04377a7f5578f3c
From: "Test SIP" <sip:[EMAIL PROTECTED]>;tag=5eba6d75ff7e1e47
To: <sip:[EMAIL PROTECTED]>;tag=as114aad8b
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 48200 ACK
User-Agent: Grandstream GXP2000 1.0.1.9
Max-Forwards: 70
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRAC
K
Content-Length: 0
11 headers, 0 lines
Jun 28 10:43:23 DEBUG[16774]: chan_sip.c:840 __sip_ack: Stopping
retransmission on '[EMAIL PROTECTED]' of
Response 48200: Found
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP pickup.phone.ip.addr;branch=z9hG4bKb828ead3d3936e08
From: "Test SIP" <sip:[EMAIL PROTECTED]>;tag=5eba6d75ff7e1e47
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]>
Supported: replaces, timer
Proxy-Authorization: Digest username="1310", realm="asterisk",
algorithm=MD5, uri="sip:[EMAIL PROTECTED]", nonce="30b68bfa",
response="********************************"
Call-ID: [EMAIL PROTECTED]
Seq: 48201 INVITE
User-Agent: Grandstream GXP2000 1.0.1.9
Max-Forwards: 70
Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRAC
K
Content-Type: application/sdp
Content-Length: 302
v=0
o=1310 8000 8001 IN IP4 pickup.phone.ip.addr
s=SIP Call
c=IN IP4 pickup.phone.ip.addr
t=0 0
m=audio 5004 RTP/AVP 0 8 3 4 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
14 headers, 15 lines
Using latest request as basis request
Sending to pickup.phone.ip.addr : 5060 (non-NAT)
Jun 28 10:43:23 DEBUG[16774]: chan_sip.c:5441 check_user_full:
Setting NAT on RTP to 0
Found user '1310'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port pickup.phone.ip.addr:5004
Jun 28 10:43:23 DEBUG[16774]: chan_sip.c:2711 process_sdp: Peer
audio RTP is at port pickup.phone.ip.addr:5004
Found description format PCMU
Found description format PCMA
Found description format GSM
Found description format G723
Found description format G729
Found description format telephone-event
Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x10f
(g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10d
(g723|ulaw|alaw|g729)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723),
combined - 0x1 (g723)
Jun 28 10:43:23 DEBUG[16774]: chan_sip.c:7329 handle_request:
Check for res for 1310
Jun 28 10:43:23 DEBUG[16774]: chan_sip.c:1620 update_user_counter:
Call from user '1310' is 1 out of 0
Looking for *8 in default
Jun 28 10:43:23 DEBUG[16774]: chan_sip.c:4650 build_route:
build_route: Contact hop: <sip:[EMAIL PROTECTED]>
list_route: hop: <sip:[EMAIL PROTECTED]>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP pickup.phone.ip.addr;branch=z9hG4bKb828ead3d3936e08
From: "Test SIP" <sip:[EMAIL PROTECTED]>;tag=5eba6d75ff7e1e47
To: <sip:[EMAIL PROTECTED]>;tag=as23dd6dfb
Call-ID: [EMAIL PROTECTED]
CSeq: 48201 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
to pickup.phone.ip.addr:5060
Jun 28 10:43:23 DEBUG[16774]: res_features.c:1709 ast_pickup_call:
No call pickup possible...
Jun 28 10:43:23 NOTICE[16774]: chan_sip.c:7402 handle_request:
Nothing to pick up
Reliably Transmitting (no NAT):
SIP/2.0 503 Unavailable
Via: SIP/2.0/UDP pickup.phone.ip.addr;branch=z9hG4bKb828ead3d3936e08
From: "Test SIP" <sip:[EMAIL PROTECTED]>;tag=5eba6d75ff7e1e47
To: <sip:[EMAIL PROTECTED]>;tag=as23dd6dfb
Call-ID: [EMAIL PROTECTED]
CSeq: 48201 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
to pickup.phone.ip.addr:5060
Please also let me know if any other information would help to
troubleshoot this.
Robert Woodcock
Sr. Network Engineer
Print, Inc.
(425) 629-2424
http://www.printinc.com
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