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
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
[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