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

Brian West brian.west at mac.com
Wed Jun 29 08:09:31 MST 2005 

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
>>
>> _______________________________________________
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>>
>>
>
>
> -- 
> A.G. (Tony) Nichols
> I.S. Manager
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
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