[Asterisk-Users] Trying to get *8 call pickup to work
Klaus-Peter Junghanns
kpj at junghanns.net
Thu Jun 30 00:20:23 MST 2005
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
> >
> > On 6/28/05, Robert Woodcock <rwoodcock at 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 at 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 at default
> >> [1312]
> >> username=1312
> >> secret=donttell
> >> type=friend
> >> host=dynamic
> >> callerid=Cisco 7960G Test <1312>
> >> mailbox=1312 at 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 at called.phone.ip.addr:5061 SIP/2.0
> >> Via: SIP/2.0/UDP asterisk.server.ip.addr:5060;branch=z9hG4bK359ec760
> >> From: "asterisk"
> >> <sip:asterisk at asterisk.server.ip.addr>;tag=as61d8a13d
> >> To: <sip:1312 at called.phone.ip.addr:5061>
> >> Contact: <sip:asterisk at asterisk.server.ip.addr>
> >> Call-ID: 4b29d0401b599b130e70f1604398cbf4 at 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 at asterisk.server.ip.addr>;tag=as61d8a13d
> >> To: <sip:1312 at called.phone.ip.addr:5061>
> >> Call-ID: 4b29d0401b599b130e70f1604398cbf4 at asterisk.server.ip.addr
> >> Date: Tue, 28 Jun 2005 17:43:20 GMT
> >> CSeq: 102 INVITE
> >> Server: CSCO/7
> >> Contact: <sip:1312 at 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 at 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 at asterisk.server.ip.addr>;tag=as61d8a13d
> >> To: <sip:1312 at called.phone.ip.addr:
> >> 5061>;tag=001280b9cebf00025bfd45ed-7102ff29
> >> Call-ID: 4b29d0401b599b130e70f1604398cbf4 at asterisk.server.ip.addr
> >> Date: Tue, 28 Jun 2005 17:43:20 GMT
> >> CSeq: 102 INVITE
> >> Server: CSCO/7
> >> Contact: <sip:1312 at 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 at asterisk.server.ip.addr' Request
> >> 102: Found
> >> -- SIP/1312-c824 is ringing
> >>
> >>
> >> Sip read:
> >> INVITE sip:*8 at asterisk-server SIP/2.0
> >> Via: SIP/2.0/UDP pickup.phone.ip.addr;branch=z9hG4bKa04377a7f5578f3c
> >> From: "Test SIP" <sip:1310 at asterisk-server>;tag=5eba6d75ff7e1e47
> >> To: <sip:*8 at asterisk-server>
> >> Contact: <sip:1310 at pickup.phone.ip.addr>
> >> Supported: replaces, timer
> >> Call-ID: faa98dd842d016fd at 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,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:1310 at asterisk-server>;tag=5eba6d75ff7e1e47
> >> To: <sip:*8 at asterisk-server>;tag=as114aad8b
> >> Call-ID: faa98dd842d016fd at pickup.phone.ip.addr
> >> CSeq: 48200 INVITE
> >> User-Agent: Asterisk PBX
> >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> >> Contact: <sip:*8 at 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 at pickup.phone.ip.addr' in 15000 ms
> >> Found user '1310'
> >>
> >>
> >> Sip read:
> >> ACK sip:*8 at asterisk-server SIP/2.0
> >> Via: SIP/2.0/UDP pickup.phone.ip.addr;branch=z9hG4bKa04377a7f5578f3c
> >> From: "Test SIP" <sip:1310 at asterisk-server>;tag=5eba6d75ff7e1e47
> >> To: <sip:*8 at asterisk-server>;tag=as114aad8b
> >> Contact: <sip:1310 at pickup.phone.ip.addr>
> >> Call-ID: faa98dd842d016fd at 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,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 'faa98dd842d016fd at pickup.phone.ip.addr' of
> >> Response 48200: Found
> >>
> >>
> >> Sip read:
> >> INVITE sip:*8 at asterisk-server SIP/2.0
> >> Via: SIP/2.0/UDP pickup.phone.ip.addr;branch=z9hG4bKb828ead3d3936e08
> >> From: "Test SIP" <sip:1310 at asterisk-server>;tag=5eba6d75ff7e1e47
> >> To: <sip:*8 at asterisk-server>
> >> Contact: <sip:1310 at pickup.phone.ip.addr>
> >> Supported: replaces, timer
> >> Proxy-Authorization: Digest username="1310", realm="asterisk",
> >> algorithm=MD5, uri="sip:*8 at asterisk-server", nonce="30b68bfa",
> >> response="********************************"
> >> Call-ID: faa98dd842d016fd at 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,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:1310 at pickup.phone.ip.addr>
> >> list_route: hop: <sip:1310 at 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 at asterisk-server>;tag=5eba6d75ff7e1e47
> >> To: <sip:*8 at asterisk-server>;tag=as23dd6dfb
> >> Call-ID: faa98dd842d016fd at pickup.phone.ip.addr
> >> CSeq: 48201 INVITE
> >> User-Agent: Asterisk PBX
> >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> >> Contact: <sip:*8 at 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 at asterisk-server>;tag=5eba6d75ff7e1e47
> >> To: <sip:*8 at asterisk-server>;tag=as23dd6dfb
> >> Call-ID: faa98dd842d016fd at pickup.phone.ip.addr
> >> CSeq: 48201 INVITE
> >> User-Agent: Asterisk PBX
> >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> >> Contact: <sip:*8 at 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 at 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
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at 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
> >
>
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