[Asterisk-Users] Trying to get *8 call pickup to work
Tony Nichols
tony.nichols at gmail.com
Wed Jun 29 07:16:44 MST 2005
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,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 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,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 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,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 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
>
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--
A.G. (Tony) Nichols
I.S. Manager
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