I'm having a tough time getting call pickup to work on *. Here's my
configuration:
X100P with T-1, channels 1-4 voice <---> * <---CISCO 7960 with SIP 6.0 Image
A call comes in, and * picks up and presents a menu. Caller chooses
extension, (in this case ext 103, SIP/wsmith)
Wsmith is sitting in my office, hears his phone ringing, picks up my phone,
gets dial tone, and presses *8. He gets a reorder (fast busy) on my phone,
and his phone continues to ring (he then curses loudly, and goes racing down
the hall to try to catch the call)
In * , I get a
Jun 9 15:45:14 NOTICE[229391]: chan_sip.c:5417 handle_request: Nothing to
pick up
I turned on SIP debugging, cleaned out all the Sip register messages that
were flying about while debugging, and present the logs here. My version is
CVS-05/24/04
My zapata.conf looks like:
group=1
callgroup=1
pickupgroup=1-4
context=NuFone-Outgoing
signalling = fxs_ks
callprogress=no
callerid="Radiance Technologies" <(251)-445-0045>
usecallerid=yes
My SIP.conf looks like:
sip.conf [----] 0 L:[105+37 142/142] *(3505/3516b)= c 99 0x63
dtmfmode=inband
mailbox=102
context=Outgoing
callerid="Dean Li" <102>
username=dli
secret=rad1ance
pickupgroup=1
;the ringing SIP phone:
[wsmith]
type=friend
host=dynamic
nat=yes
canreinvite=no
qualify=1000
;defaultip=192.168.30.108
dtmfmode=inband
mailbox=103
context=Outgoing
callerid="Walter Smith" <103>
username=wsmith
secret=******
pickupgroup=1-4
;The phone attempting the *8
[nmartin]
type=friend
host=dynamic
insecure=no
nat=yes
canreinvite=no
qualify=1000
;defaultip=192.168.30.100
dtmfmode=inband
mailbox=105
context=Outgoing
callerid="Nik Martin" <105>
username=nmartin
secret=******
pickupgroup=1-4
callgroup=1
The SIP debug:
pbxMobile*CLI>
-- Starting simple switch on 'Zap/1-1'
pbxMobile*CLI>
-- Executing Wait("Zap/1-1", "3") in new stack
pbxMobile*CLI>
-- Executing Answer("Zap/1-1", "") in new stack
pbxMobile*CLI>
-- Executing NoOp("Zap/1-1", ""MOBILE, AL" <xxxxxxxxx>") in new stack
pbxMobile*CLI>
-- Executing Wait("Zap/1-1", "1") in new stack
pbxMobile*CLI>
Jun 9 15:45:02 WARNING[2211866]: chan_zap.c:3073 zt_handle_event:
Ring/Off-hook in strange state 6 on channel 1
pbxMobile*CLI>
-- Executing BackGround("Zap/1-1", "radiancewelcome") in new stack
pbxMobile*CLI>
-- Playing 'radiancewelcome' (language 'en')
pbxMobile*CLI>
11 headers, 2 lines
8 headers, 0 lines
pbxMobile*CLI>
== CDR updated on Zap/1-1
pbxMobile*CLI>
-- Executing Dial("Zap/1-1", "SIP/wsmith|20|tT") in new stack
pbxMobile*CLI>
We're at 172.31.30.3 port 15418
pbxMobile*CLI>
Answering with preferred capability 4
pbxMobile*CLI>
Answering with preferred capability 2
pbxMobile*CLI>
12 headers, 9 lines
pbxMobile*CLI>
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP
172.31.30.3:5060;branch=z9hG4bK5cc08566 From: "MOBILE, AL"
<sip:[EMAIL PROTECTED]>;tag=as05f4b37a To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]> Call-ID:
[EMAIL PROTECTED] CSeq: 102 INVITE User-Agent:
Asterisk PBX Date: Wed, 09 Jun 2004 20:45:09 GMT Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 181 v=0
o=root 20260 20260 IN IP4 172.31.30.3 s=session c=IN IP4 172.31.30.3 t=0 0
m=audio 15418 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - - (NAT) to 172.31.30.11:5060
pbxMobile*CLI>
-- Called wsmith
pbxMobile*CLI>
Sip read:
SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.31.30.3:5060;branch=z9hG4bK5cc08566
From: "MOBILE, AL" <sip:[EMAIL PROTECTED]>;tag=as05f4b37a To:
<sip:[EMAIL PROTECTED]> Call-ID:
[EMAIL PROTECTED] Date: Wed, 09 Jun 2004 20:48:00
GMT CSeq: 102 INVITE Server: CSCO/6 Contact: <sip:[EMAIL PROTECTED]:5060>
Content-Length: 0
10 headers, 0 lines
pbxMobile*CLI>
Sip read:
SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.31.30.3:5060;branch=z9hG4bK5cc08566
From: "MOBILE, AL" <sip:[EMAIL PROTECTED]>;tag=as05f4b37a To:
<sip:[EMAIL PROTECTED]>;tag=000b46e9ae7e485f2abff4bc-43940b23 Call-ID:
[EMAIL PROTECTED] Date: Wed, 09 Jun 2004 20:48:00
GMT CSeq: 102 INVITE Server: CSCO/6 Contact: <sip:[EMAIL PROTECTED]:5060>
Content-Length: 0
10 headers, 0 lines
pbxMobile*CLI>
-- SIP/wsmith-7e27 is ringing
pbxMobile*CLI>
pbxMobile*CLI>
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP
172.31.30.7:5060;branch=z9hG4bK2408959c From: "105 - Nik Martin"
<sip:[EMAIL PROTECTED]>;tag=003094c4481f49565aff56ad-226e8953 To:
<sip:[EMAIL PROTECTED]> Call-ID:
[EMAIL PROTECTED] Date: Wed, 09 Jun 2004
20:47:30 GMT CSeq: 101 INVITE User-Agent: CSCO/6 Contact:
<sip:[EMAIL PROTECTED]:5060> Expires: 180 Content-Type: application/sdp
Content-Length: 244 Accept: application/sdp Remote-Party-ID: "105 - Nik
Martin"
<sip:[EMAIL PROTECTED]>;party=calling;id-type=subscriber;privacy=off;scree
n=no v=0 o=Cisco-SIPUA 24482 2915 IN IP4 172.31.30.7 s=SIP Call c=IN IP4
172.31.30.7 t=0 0 m=audio 26676 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=fmtp:101 0-15
14 headers, 11 lines
Using latest request as basis request
Sending to 172.31.30.7 : 5060 (non-NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 6, them - 268/0, combined - 4
Non-codec capabilities: us - 1, them - 1, combined - 1
Reliably Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP
172.31.30.7:5060;branch=z9hG4bK2408959c;received=172.31.30.7 From: "105 -
Nik Martin" <sip:[EMAIL PROTECTED]>;tag=003094c4481f49565aff56ad-226e8953
To: <sip:[EMAIL PROTECTED]>;tag=as6f213426 Call-ID:
[EMAIL PROTECTED] CSeq: 101 INVITE User-Agent:
Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact:
<sip:[EMAIL PROTECTED]> Proxy-Authenticate: Digest realm="asterisk",
nonce="2152fdb4" Content-Length: 0
to 172.31.30.7:5060
pbxMobile*CLI>
Sip read:
ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP
172.31.30.7:5060;branch=z9hG4bK2408959c From: "105 - Nik Martin"
<sip:[EMAIL PROTECTED]>;tag=003094c4481f49565aff56ad-226e8953 To:
<sip:[EMAIL PROTECTED]>;tag=as6f213426 Call-ID:
[EMAIL PROTECTED] Date: Wed, 09 Jun 2004
20:47:30 GMT CSeq: 101 ACK Content-Length: 0
8 headers, 0 lines
pbxMobile*CLI>
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP
172.31.30.7:5060;branch=z9hG4bK77078702 From: "105 - Nik Martin"
<sip:[EMAIL PROTECTED]>;tag=003094c4481f49565aff56ad-226e8953 To:
<sip:[EMAIL PROTECTED]> Call-ID:
[EMAIL PROTECTED] Date: Wed, 09 Jun 2004
20:47:30 GMT CSeq: 102 INVITE User-Agent: CSCO/6 Contact:
<sip:[EMAIL PROTECTED]:5060> Proxy-Authorization: Digest
username="nmartin",realm="asterisk",uri="sip:172.31.30.3",response="31288731
f7791b64666a923ebe8a16f3",nonce="2152fdb4",algorithm=md5 Expires: 180
Content-Type: application/sdp Content-Length: 244 Remote-Party-ID: "105 -
Nik Martin"
<sip:[EMAIL PROTECTED]>;party=calling;id-type=subscriber;privacy=off;scree
n=no v=0 o=Cisco-SIPUA 24482 2915 IN IP4 172.31.30.7 s=SIP Call c=IN IP4
172.31.30.7 t=0 0 m=audio 26676 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=fmtp:101 0-15
14 headers, 11 lines
Using latest request as basis request
Sending to 172.31.30.7 : 5060 (NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 6, them - 268/0, combined - 4
Non-codec capabilities: us - 1, them - 1, combined - 1
Looking for *8 in Outgoing
list_route: hop: <sip:[EMAIL PROTECTED]:5060>
Transmitting (NAT):
SIP/2.0 100 Trying Via: SIP/2.0/UDP
172.31.30.7:5060;branch=z9hG4bK77078702;received=172.31.30.7 From: "105 -
Nik Martin" <sip:[EMAIL PROTECTED]>;tag=003094c4481f49565aff56ad-226e8953
To: <sip:[EMAIL PROTECTED]>;tag=as200f8b5c Call-ID:
[EMAIL PROTECTED] CSeq: 102 INVITE User-Agent:
Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact:
<sip:[EMAIL PROTECTED]> Content-Length: 0
to 172.31.30.7:5060
Jun 9 15:45:14 NOTICE[229391]: chan_sip.c:5417 handle_request: Nothing to
pick up
Reliably Transmitting (NAT):
SIP/2.0 503 Unavailable Via: SIP/2.0/UDP
172.31.30.7:5060;branch=z9hG4bK77078702;received=172.31.30.7 From: "105 -
Nik Martin" <sip:[EMAIL PROTECTED]>;tag=003094c4481f49565aff56ad-226e8953
To: <sip:[EMAIL PROTECTED]>;tag=as200f8b5c Call-ID:
[EMAIL PROTECTED] CSeq: 102 INVITE User-Agent:
Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact:
<sip:[EMAIL PROTECTED]> Content-Length: 0
to 172.31.30.7:5060
pbxMobile*CLI>
Sip read:
ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP
172.31.30.7:5060;branch=z9hG4bK77078702 From: "105 - Nik Martin"
<sip:[EMAIL PROTECTED]>;tag=003094c4481f49565aff56ad-226e8953 To:
<sip:[EMAIL PROTECTED]>;tag=as200f8b5c Call-ID:
[EMAIL PROTECTED] Date: Wed, 09 Jun 2004
20:47:30 GMT CSeq: 102 ACK Content-Length: 0
8 headers, 0 lines
pbxMobile*CLI>
Reliably Transmitting:
CANCEL sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP
172.31.30.3:5060;branch=z9hG4bK5cc08566 From: "MOBILE, AL"
<sip:[EMAIL PROTECTED]>;tag=as05f4b37a To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]> Call-ID:
[EMAIL PROTECTED] CSeq: 102 CANCEL User-Agent:
Asterisk PBX Content-Length: 0 (NAT) to 172.31.30.11:5060
== Spawn extension (default, 103, 1) exited non-zero on 'Zap/1-1'
-- Executing Hangup("Zap/1-1", "") in new stack
== Spawn extension (default, h, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
pbxMobile*CLI>
Sip read:
SIP/2.0 200 OK Via: SIP/2.0/UDP 172.31.30.3:5060;branch=z9hG4bK5cc08566
From: "MOBILE, AL" <sip:[EMAIL PROTECTED]>;tag=as05f4b37a To:
<sip:[EMAIL PROTECTED]>;tag=000b46e9ae7e485f2abff4bc-43940b23 Call-ID:
[EMAIL PROTECTED] Date: Wed, 09 Jun 2004 20:48:19
GMT CSeq: 102 CANCEL Server: CSCO/6 Content-Length: 0
9 headers, 0 lines
pbxMobile*CLI>
Sip read:
SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP
172.31.30.3:5060;branch=z9hG4bK5cc08566 From: "MOBILE, AL"
<sip:[EMAIL PROTECTED]>;tag=as05f4b37a To:
<sip:[EMAIL PROTECTED]>;tag=000b46e9ae7e485f2abff4bc-43940b23 Call-ID:
[EMAIL PROTECTED] Date: Wed, 09 Jun 2004 20:48:19
GMT CSeq: 102 INVITE Server: CSCO/6 Contact: <sip:[EMAIL PROTECTED]:5060>
Content-Length: 0
10 headers, 0 lines
Transmitting:
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP
172.31.30.3:5060;branch=z9hG4bK5cc08566 From: "MOBILE, AL"
<sip:[EMAIL PROTECTED]>;tag=as05f4b37a To:
<sip:[EMAIL PROTECTED]>;tag=000b46e9ae7e485f2abff4bc-43940b23 Contact:
<sip:[EMAIL PROTECTED]> Call-ID:
[EMAIL PROTECTED] CSeq: 102 ACK User-Agent:
Asterisk PBX Content-Length: 0 (NAT) to 172.31.30.11:5060
pbxMobile*CLI> sip debugexitsip no debug pbxMobile*CLI>
SIP Debugging Disabled
pbxMobile*CLI> exit [EMAIL PROTECTED]:~# logout
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