thr3ads.net - asterisk users - [Asterisk-Users] Trying to get *8 call pickup to work [Jun 2005]

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Robert Woodcock

2005-Jun-28 14:04 UTC

[Asterisk-Users] Trying to get *8 call pickup to work

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>
mailbox=1310@default
[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>
mailbox=1311@default
[1312]
username=1312
secret=donttell
type=friend
host=dynamic
callerid=Cisco 7960G Test <1312>
mailbox=1312@default

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:1312@called.phone.ip.addr:5061 SIP/2.0
Via: SIP/2.0/UDP asterisk.server.ip.addr:5060;branch=z9hG4bK359ec760
From: "asterisk"
<sip:asterisk@asterisk.server.ip.addr>;tag=as61d8a13d
To: <sip:1312@called.phone.ip.addr:5061>
Contact: <sip:asterisk@asterisk.server.ip.addr>
Call-ID: 4b29d0401b599b130e70f1604398cbf4@asterisk.server.ip.addr
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:asterisk@asterisk.server.ip.addr>;tag=as61d8a13d
To: <sip:1312@called.phone.ip.addr:5061>
Call-ID: 4b29d0401b599b130e70f1604398cbf4@asterisk.server.ip.addr
Date: Tue, 28 Jun 2005 17:43:20 GMT
CSeq: 102 INVITE
Server: CSCO/7
Contact: <sip:1312@called.phone.ip.addr: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
'4b29d0401b599b130e70f1604398cbf4@asterisk.server.ip.addr' 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:asterisk@asterisk.server.ip.addr>;tag=as61d8a13d
To:
<sip:1312@called.phone.ip.addr:5061>;tag=001280b9cebf00025bfd45ed-7102ff29
Call-ID: 4b29d0401b599b130e70f1604398cbf4@asterisk.server.ip.addr
Date: Tue, 28 Jun 2005 17:43:20 GMT
CSeq: 102 INVITE
Server: CSCO/7
Contact: <sip:1312@called.phone.ip.addr: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
'4b29d0401b599b130e70f1604398cbf4@asterisk.server.ip.addr' Request 102:
Found
    -- SIP/1312-c824 is ringing


Sip read:
INVITE sip:*8@asterisk-server SIP/2.0
Via: SIP/2.0/UDP pickup.phone.ip.addr;branch=z9hG4bKa04377a7f5578f3c
From: "Test SIP" <sip:1310@asterisk-server>;tag=5eba6d75ff7e1e47
To: <sip:*8@asterisk-server>
Contact: <sip:1310@pickup.phone.ip.addr>
Supported: replaces, timer
Call-ID: faa98dd842d016fd@pickup.phone.ip.addr
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,PRACK
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:1310@asterisk-server>;tag=5eba6d75ff7e1e47
To: <sip:*8@asterisk-server>;tag=as114aad8b
Call-ID: faa98dd842d016fd@pickup.phone.ip.addr
CSeq: 48200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:*8@asterisk.server.ip.addr>
Proxy-Authenticate: Digest realm="asterisk",
nonce="30b68bfa"
Content-Length: 0


 to pickup.phone.ip.addr:5060
Scheduling destruction of call 'faa98dd842d016fd@pickup.phone.ip.addr'
in 15000 ms
Found user '1310'


Sip read:
ACK sip:*8@asterisk-server SIP/2.0
Via: SIP/2.0/UDP pickup.phone.ip.addr;branch=z9hG4bKa04377a7f5578f3c
From: "Test SIP" <sip:1310@asterisk-server>;tag=5eba6d75ff7e1e47
To: <sip:*8@asterisk-server>;tag=as114aad8b
Contact: <sip:1310@pickup.phone.ip.addr>
Call-ID: faa98dd842d016fd@pickup.phone.ip.addr
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,PRACK
Content-Length: 0


11 headers, 0 lines
Jun 28 10:43:23 DEBUG[16774]: chan_sip.c:840 __sip_ack: Stopping retransmission
on 'faa98dd842d016fd@pickup.phone.ip.addr' of Response 48200: Found


Sip read:
INVITE sip:*8@asterisk-server SIP/2.0
Via: SIP/2.0/UDP pickup.phone.ip.addr;branch=z9hG4bKb828ead3d3936e08
From: "Test SIP" <sip:1310@asterisk-server>;tag=5eba6d75ff7e1e47
To: <sip:*8@asterisk-server>
Contact: <sip:1310@pickup.phone.ip.addr>
Supported: replaces, timer
Proxy-Authorization: Digest username="1310",
realm="asterisk", algorithm=MD5,
uri="sip:*8@asterisk-server", nonce="30b68bfa",
response="********************************"
Call-ID: faa98dd842d016fd@pickup.phone.ip.addr
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,PRACK
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:1310@pickup.phone.ip.addr>
list_route: hop: <sip:1310@pickup.phone.ip.addr>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP pickup.phone.ip.addr;branch=z9hG4bKb828ead3d3936e08
From: "Test SIP" <sip:1310@asterisk-server>;tag=5eba6d75ff7e1e47
To: <sip:*8@asterisk-server>;tag=as23dd6dfb
Call-ID: faa98dd842d016fd@pickup.phone.ip.addr
CSeq: 48201 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:*8@asterisk.server.ip.addr>
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:1310@asterisk-server>;tag=5eba6d75ff7e1e47
To: <sip:*8@asterisk-server>;tag=as23dd6dfb
Call-ID: faa98dd842d016fd@pickup.phone.ip.addr
CSeq: 48201 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:*8@asterisk.server.ip.addr>
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

Tony Nichols

2005-Jun-29 07:16 UTC

head link

[Asterisk-Users] Trying to get *8 call pickup to work

I have been unable to get it to pickup sip-sip calls.... but if an
incoming zap rings I can hit *8# and it works.
My config is the same as yours:
zapata has callgroup = 1
and in sip.conf I have 
pickupgroup = 1

I'm also using Grandstreams.

t o n y

On 6/28/05, Robert Woodcock <rwoodcock@printinc.com>
wrote:> 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>
> mailbox=1310@default
> [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>
> mailbox=1311@default
> [1312]
> username=1312
> secret=donttell
> type=friend
> host=dynamic
> callerid=Cisco 7960G Test <1312>
> mailbox=1312@default
> 
> 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:1312@called.phone.ip.addr:5061 SIP/2.0
> Via: SIP/2.0/UDP asterisk.server.ip.addr:5060;branch=z9hG4bK359ec760
> From: "asterisk"
<sip:asterisk@asterisk.server.ip.addr>;tag=as61d8a13d
> To: <sip:1312@called.phone.ip.addr:5061>
> Contact: <sip:asterisk@asterisk.server.ip.addr>
> Call-ID: 4b29d0401b599b130e70f1604398cbf4@asterisk.server.ip.addr
> 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:asterisk@asterisk.server.ip.addr>;tag=as61d8a13d
> To: <sip:1312@called.phone.ip.addr:5061>
> Call-ID: 4b29d0401b599b130e70f1604398cbf4@asterisk.server.ip.addr
> Date: Tue, 28 Jun 2005 17:43:20 GMT
> CSeq: 102 INVITE
> Server: CSCO/7
> Contact: <sip:1312@called.phone.ip.addr: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
'4b29d0401b599b130e70f1604398cbf4@asterisk.server.ip.addr' 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:asterisk@asterisk.server.ip.addr>;tag=as61d8a13d
> To:
<sip:1312@called.phone.ip.addr:5061>;tag=001280b9cebf00025bfd45ed-7102ff29
> Call-ID: 4b29d0401b599b130e70f1604398cbf4@asterisk.server.ip.addr
> Date: Tue, 28 Jun 2005 17:43:20 GMT
> CSeq: 102 INVITE
> Server: CSCO/7
> Contact: <sip:1312@called.phone.ip.addr: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
'4b29d0401b599b130e70f1604398cbf4@asterisk.server.ip.addr' Request 102:
Found
>     -- SIP/1312-c824 is ringing
> 
> 
> Sip read:
> INVITE sip:*8@asterisk-server SIP/2.0
> Via: SIP/2.0/UDP pickup.phone.ip.addr;branch=z9hG4bKa04377a7f5578f3c
> From: "Test SIP"
<sip:1310@asterisk-server>;tag=5eba6d75ff7e1e47
> To: <sip:*8@asterisk-server>
> Contact: <sip:1310@pickup.phone.ip.addr>
> Supported: replaces, timer
> Call-ID: faa98dd842d016fd@pickup.phone.ip.addr
> 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,PRACK
> 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:1310@asterisk-server>;tag=5eba6d75ff7e1e47
> To: <sip:*8@asterisk-server>;tag=as114aad8b
> Call-ID: faa98dd842d016fd@pickup.phone.ip.addr
> CSeq: 48200 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:*8@asterisk.server.ip.addr>
> Proxy-Authenticate: Digest realm="asterisk",
nonce="30b68bfa"
> Content-Length: 0
> 
> 
>  to pickup.phone.ip.addr:5060
> Scheduling destruction of call
'faa98dd842d016fd@pickup.phone.ip.addr' in 15000 ms
> Found user '1310'
> 
> 
> Sip read:
> ACK sip:*8@asterisk-server SIP/2.0
> Via: SIP/2.0/UDP pickup.phone.ip.addr;branch=z9hG4bKa04377a7f5578f3c
> From: "Test SIP"
<sip:1310@asterisk-server>;tag=5eba6d75ff7e1e47
> To: <sip:*8@asterisk-server>;tag=as114aad8b
> Contact: <sip:1310@pickup.phone.ip.addr>
> Call-ID: faa98dd842d016fd@pickup.phone.ip.addr
> 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,PRACK
> Content-Length: 0
> 
> 
> 11 headers, 0 lines
> Jun 28 10:43:23 DEBUG[16774]: chan_sip.c:840 __sip_ack: Stopping
retransmission on 'faa98dd842d016fd@pickup.phone.ip.addr' of Response
48200: Found
> 
> 
> Sip read:
> INVITE sip:*8@asterisk-server SIP/2.0
> Via: SIP/2.0/UDP pickup.phone.ip.addr;branch=z9hG4bKb828ead3d3936e08
> From: "Test SIP"
<sip:1310@asterisk-server>;tag=5eba6d75ff7e1e47
> To: <sip:*8@asterisk-server>
> Contact: <sip:1310@pickup.phone.ip.addr>
> Supported: replaces, timer
> Proxy-Authorization: Digest username="1310",
realm="asterisk", algorithm=MD5,
uri="sip:*8@asterisk-server", nonce="30b68bfa",
response="********************************"
> Call-ID: faa98dd842d016fd@pickup.phone.ip.addr
> 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,PRACK
> 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:1310@pickup.phone.ip.addr>
> list_route: hop: <sip:1310@pickup.phone.ip.addr>
> Transmitting (no NAT):
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP pickup.phone.ip.addr;branch=z9hG4bKb828ead3d3936e08
> From: "Test SIP"
<sip:1310@asterisk-server>;tag=5eba6d75ff7e1e47
> To: <sip:*8@asterisk-server>;tag=as23dd6dfb
> Call-ID: faa98dd842d016fd@pickup.phone.ip.addr
> CSeq: 48201 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:*8@asterisk.server.ip.addr>
> 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:1310@asterisk-server>;tag=5eba6d75ff7e1e47
> To: <sip:*8@asterisk-server>;tag=as23dd6dfb
> Call-ID: faa98dd842d016fd@pickup.phone.ip.addr
> CSeq: 48201 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:*8@asterisk.server.ip.addr>
> 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
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
> 

-- 
A.G. (Tony) Nichols
I.S. Manager

Robert Woodcock

2005-Jun-30 07:12 UTC

head link

[Asterisk-Users] Trying to get *8 call pickup to work

Thanks for the suggestions, everyone!

I'd previously tried PickUP() but it'd always give me a 503 as well,
with this familiar error:

chan_sip.c:7402 handle_request: Nothing to pick up

Almost as if I still had pickupchan= defined in features.conf, but I did
make sure to comment that out and restart asterisk entirely before
putting PickUP() in extensions.conf.

app_intercept and app_pickupchan give similar and altogether very
interesting results. I put this in extensions.conf:

exten => _*8XXXX,1,PickUpChan(SIP/${EXTEN:2:4})

When I call in from a zap line to x1311, and on x1310 punch in *81311,
x1310 does answer the call. However, x1311 keeps ringing, even after I
hang up x1310! If I then try to answer x1311, it indicates busy.
I tried adding:

exten => _*8XXXX,2,Answer

But that doesn't do anything. Any ideas?

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Klaus-Peter
Junghanns
Sent: Thursday, June 30, 2005 12:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Trying to get *8 call pickup to work

Hi,

app_pickup, app_pickupchan, app_pickdown, app_steal are your friend
in BRIstuff. ;)

best regards

Klaus

Am Mittwoch, den 29.06.2005, 10:09 -0500 schrieb Brian
West:> Go get app_intercept from www.pbxfreeware.org
> 
> /b
> ---
> Anakin: ?You?re either with me, or you?re my enemy.?
> Obi-Wan: ?Only a Sith could be an absolutist.?
> 
> On Jun 29, 2005, at 9:16 AM, Tony Nichols wrote:
> 
> > I have been unable to get it to pickup sip-sip calls.... but if an
> > incoming zap rings I can hit *8# and it works.
> > My config is the same as yours:
> > zapata has callgroup = 1
> > and in sip.conf I have
> > pickupgroup = 1
> >
> > I'm also using Grandstreams.
> >
> > t o n y

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