[Nov 10 17:34:17] [Nov 10 17:34:17] <-------------> [Nov 10 17:34:19]  Creating Stasis app 'attendant' [Nov 10 17:34:19]  == WebSocket connection from '192.168.210.111:60709' for protocol '' accepted using version '13' [Nov 10 17:34:23] [Nov 10 17:34:23] <--- SIP read from UDP:192.168.210.111:5060 ---> [Nov 10 17:34:23] INVITE sip:6100@192.168.210.132 SIP/2.0 [Nov 10 17:34:23] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK80519034d1a5e6118a3f991006d58e79;rport [Nov 10 17:34:23] From: "PhonerLite" ;tag=2804143771 [Nov 10 17:34:23] To: [Nov 10 17:34:23] Call-ID: 80519034-D1A5-E611-8A3E-991006D58E79@192.168.210.111 [Nov 10 17:34:23] CSeq: 12 INVITE [Nov 10 17:34:23] Contact: [Nov 10 17:34:23] Content-Type: application/sdp [Nov 10 17:34:23] Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE [Nov 10 17:34:23] Max-Forwards: 70 [Nov 10 17:34:23] Supported: 100rel, replaces, from-change [Nov 10 17:34:23] P-Early-Media: supported [Nov 10 17:34:23] User-Agent: SIPPER for PhonerLite [Nov 10 17:34:23] P-Preferred-Identity: [Nov 10 17:34:23] Content-Length: 449 [Nov 10 17:34:23] [Nov 10 17:34:23] v=0 [Nov 10 17:34:23] o=- 61621854 1 IN IP4 192.168.210.111 [Nov 10 17:34:23] s=SIPPER for PhonerLite [Nov 10 17:34:23] c=IN IP4 192.168.210.111 [Nov 10 17:34:23] t=0 0 [Nov 10 17:34:23] m=audio 5062 RTP/AVP 107 8 0 2 3 97 110 111 9 101 [Nov 10 17:34:23] a=rtpmap:107 opus/48000/2 [Nov 10 17:34:23] a=rtpmap:8 PCMA/8000 [Nov 10 17:34:23] a=rtpmap:0 PCMU/8000 [Nov 10 17:34:23] a=rtpmap:2 G726-32/8000 [Nov 10 17:34:23] a=rtpmap:3 GSM/8000 [Nov 10 17:34:23] a=rtpmap:97 iLBC/8000 [Nov 10 17:34:23] a=rtpmap:110 speex/8000 [Nov 10 17:34:23] a=rtpmap:111 speex/16000 [Nov 10 17:34:23] a=rtpmap:9 G722/8000 [Nov 10 17:34:23] a=rtpmap:101 telephone-event/8000 [Nov 10 17:34:23] a=fmtp:101 0-16 [Nov 10 17:34:23] a=ssrc:2536842769 [Nov 10 17:34:23] a=sendrecv [Nov 10 17:34:23] <-------------> [Nov 10 17:34:23] --- (15 headers 19 lines) --- [Nov 10 17:34:23] Sending to 192.168.210.111:5060 (no NAT) [Nov 10 17:34:23] Sending to 192.168.210.111:5060 (no NAT) [Nov 10 17:34:23] Using INVITE request as basis request - 80519034-D1A5-E611-8A3E-991006D58E79@192.168.210.111 [Nov 10 17:34:23] Found peer '290' for '290' from 192.168.210.111:5060 [Nov 10 17:34:23] [Nov 10 17:34:23] <--- Reliably Transmitting (no NAT) to 192.168.210.111:5060 ---> [Nov 10 17:34:23] SIP/2.0 401 Unauthorized [Nov 10 17:34:23] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK80519034d1a5e6118a3f991006d58e79;received=192.168.210.111;rport=5060 [Nov 10 17:34:23] From: "PhonerLite" ;tag=2804143771 [Nov 10 17:34:23] To: ;tag=as65387709 [Nov 10 17:34:23] Call-ID: 80519034-D1A5-E611-8A3E-991006D58E79@192.168.210.111 [Nov 10 17:34:23] CSeq: 12 INVITE [Nov 10 17:34:23] Server: Asterisk PBX GIT-master-0d85f18 [Nov 10 17:34:23] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Nov 10 17:34:23] Supported: replaces, timer [Nov 10 17:34:23] WWW-Authenticate: Digest algorithm=MD5, realm="pc_dany_asterisk", nonce="06ee15f4" [Nov 10 17:34:23] Content-Length: 0 [Nov 10 17:34:23] [Nov 10 17:34:23] [Nov 10 17:34:23] <------------> [Nov 10 17:34:23] Scheduling destruction of SIP dialog '80519034-D1A5-E611-8A3E-991006D58E79@192.168.210.111' in 32000 ms (Method: INVITE) [Nov 10 17:34:23] [Nov 10 17:34:23] <--- SIP read from UDP:192.168.210.111:5060 ---> [Nov 10 17:34:23] ACK sip:6100@192.168.210.132 SIP/2.0 [Nov 10 17:34:23] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK80519034d1a5e6118a3f991006d58e79;rport [Nov 10 17:34:23] From: "PhonerLite" ;tag=2804143771 [Nov 10 17:34:23] To: ;tag=as65387709 [Nov 10 17:34:23] Call-ID: 80519034-D1A5-E611-8A3E-991006D58E79@192.168.210.111 [Nov 10 17:34:23] CSeq: 12 ACK [Nov 10 17:34:23] Content-Length: 0 [Nov 10 17:34:23] [Nov 10 17:34:23] <-------------> [Nov 10 17:34:23] --- (7 headers 0 lines) --- [Nov 10 17:34:23] [Nov 10 17:34:23] <--- SIP read from UDP:192.168.210.111:5060 ---> [Nov 10 17:34:23] INVITE sip:6100@192.168.210.132 SIP/2.0 [Nov 10 17:34:23] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK80519034d1a5e6118a40991006d58e79;rport [Nov 10 17:34:23] From: "PhonerLite" ;tag=2804143771 [Nov 10 17:34:23] To: [Nov 10 17:34:23] Call-ID: 80519034-D1A5-E611-8A3E-991006D58E79@192.168.210.111 [Nov 10 17:34:23] CSeq: 13 INVITE [Nov 10 17:34:23] Contact: [Nov 10 17:34:23] Authorization: Digest username="290", realm="pc_dany_asterisk", nonce="06ee15f4", uri="sip:6100@192.168.210.132", response="f2a12cc27a425c8cc67fa212fe58be5c", algorithm=MD5 [Nov 10 17:34:23] Content-Type: application/sdp [Nov 10 17:34:23] Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE [Nov 10 17:34:23] Max-Forwards: 70 [Nov 10 17:34:23] Supported: 100rel, replaces, from-change [Nov 10 17:34:23] P-Early-Media: supported [Nov 10 17:34:23] User-Agent: SIPPER for PhonerLite [Nov 10 17:34:23] P-Preferred-Identity: [Nov 10 17:34:23] Content-Length: 449 [Nov 10 17:34:23] [Nov 10 17:34:23] v=0 [Nov 10 17:34:23] o=- 61621854 1 IN IP4 192.168.210.111 [Nov 10 17:34:23] s=SIPPER for PhonerLite [Nov 10 17:34:23] c=IN IP4 192.168.210.111 [Nov 10 17:34:23] t=0 0 [Nov 10 17:34:23] m=audio 5062 RTP/AVP 107 8 0 2 3 97 110 111 9 101 [Nov 10 17:34:23] a=rtpmap:107 opus/48000/2 [Nov 10 17:34:23] a=rtpmap:8 PCMA/8000 [Nov 10 17:34:23] a=rtpmap:0 PCMU/8000 [Nov 10 17:34:23] a=rtpmap:2 G726-32/8000 [Nov 10 17:34:23] a=rtpmap:3 GSM/8000 [Nov 10 17:34:23] a=rtpmap:97 iLBC/8000 [Nov 10 17:34:23] a=rtpmap:110 speex/8000 [Nov 10 17:34:23] a=rtpmap:111 speex/16000 [Nov 10 17:34:23] a=rtpmap:9 G722/8000 [Nov 10 17:34:23] a=rtpmap:101 telephone-event/8000 [Nov 10 17:34:23] a=fmtp:101 0-16 [Nov 10 17:34:23] a=ssrc:2536842769 [Nov 10 17:34:23] a=sendrecv [Nov 10 17:34:23] <-------------> [Nov 10 17:34:23] --- (16 headers 19 lines) --- [Nov 10 17:34:23] Sending to 192.168.210.111:5060 (no NAT) [Nov 10 17:34:23] Using INVITE request as basis request - 80519034-D1A5-E611-8A3E-991006D58E79@192.168.210.111 [Nov 10 17:34:23] Found peer '290' for '290' from 192.168.210.111:5060 [Nov 10 17:34:23]  == Using SIP RTP CoS mark 5 [Nov 10 17:34:23] Found RTP audio format 107 [Nov 10 17:34:23] Found RTP audio format 8 [Nov 10 17:34:23] Found RTP audio format 0 [Nov 10 17:34:23] Found RTP audio format 2 [Nov 10 17:34:23] Found RTP audio format 3 [Nov 10 17:34:23] Found RTP audio format 97 [Nov 10 17:34:23] Found RTP audio format 110 [Nov 10 17:34:23] Found RTP audio format 111 [Nov 10 17:34:23] Found RTP audio format 9 [Nov 10 17:34:23] Found RTP audio format 101 [Nov 10 17:34:23] Found audio description format opus for ID 107 [Nov 10 17:34:23] Found audio description format PCMA for ID 8 [Nov 10 17:34:23] Found audio description format PCMU for ID 0 [Nov 10 17:34:23] Found audio description format G726-32 for ID 2 [Nov 10 17:34:23] Found audio description format GSM for ID 3 [Nov 10 17:34:23] Found audio description format iLBC for ID 97 [Nov 10 17:34:23] Found audio description format speex for ID 110 [Nov 10 17:34:23] Found audio description format speex for ID 111 [Nov 10 17:34:23] Found audio description format G722 for ID 9 [Nov 10 17:34:23] Found audio description format telephone-event for ID 101 [Nov 10 17:34:23] Capabilities: us - (alaw), peer - audio=(ulaw|gsm|alaw|g722|ilbc|opus|speex|speex16)/video=(nothing)/text=(nothing), combined - (alaw) [Nov 10 17:34:23] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 10 17:34:23] Peer audio RTP is at port 192.168.210.111:5062 [Nov 10 17:34:23] Peer doesn't provide T.140 [Nov 10 17:34:23] Looking for 6100 in LocalSets (domain 192.168.210.132) [Nov 10 17:34:23] sip_route_dump: route/path hop: [Nov 10 17:34:23] [Nov 10 17:34:23] <--- Transmitting (no NAT) to 192.168.210.111:5060 ---> [Nov 10 17:34:23] SIP/2.0 100 Trying [Nov 10 17:34:23] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK80519034d1a5e6118a40991006d58e79;received=192.168.210.111;rport=5060 [Nov 10 17:34:23] From: "PhonerLite" ;tag=2804143771 [Nov 10 17:34:23] To: [Nov 10 17:34:23] Call-ID: 80519034-D1A5-E611-8A3E-991006D58E79@192.168.210.111 [Nov 10 17:34:23] CSeq: 13 INVITE [Nov 10 17:34:23] Server: Asterisk PBX GIT-master-0d85f18 [Nov 10 17:34:23] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Nov 10 17:34:23] Supported: replaces, timer [Nov 10 17:34:23] Contact: [Nov 10 17:34:23] Content-Length: 0 [Nov 10 17:34:23] [Nov 10 17:34:23] [Nov 10 17:34:23] <------------> [Nov 10 17:34:23]  -- Executing [6100@LocalSets:1] NoOp("SIP/290-00000001", "") in new stack [Nov 10 17:34:23]  -- Executing [6100@LocalSets:2] Answer("SIP/290-00000001", "") in new stack [Nov 10 17:34:23] Audio is at 11276 [Nov 10 17:34:23] Adding codec alaw to SDP [Nov 10 17:34:23] Adding non-codec 0x1 (telephone-event) to SDP [Nov 10 17:34:23] [Nov 10 17:34:23] <--- Reliably Transmitting (no NAT) to 192.168.210.111:5060 ---> [Nov 10 17:34:23] SIP/2.0 200 OK [Nov 10 17:34:23] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK80519034d1a5e6118a40991006d58e79;received=192.168.210.111;rport=5060 [Nov 10 17:34:23] From: "PhonerLite" ;tag=2804143771 [Nov 10 17:34:23] To: ;tag=as6a078b2a [Nov 10 17:34:23] Call-ID: 80519034-D1A5-E611-8A3E-991006D58E79@192.168.210.111 [Nov 10 17:34:23] CSeq: 13 INVITE [Nov 10 17:34:23] Server: Asterisk PBX GIT-master-0d85f18 [Nov 10 17:34:23] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Nov 10 17:34:23] Supported: replaces, timer [Nov 10 17:34:23] Contact: [Nov 10 17:34:23] Content-Type: application/sdp [Nov 10 17:34:23] Content-Length: 257 [Nov 10 17:34:23] [Nov 10 17:34:23] v=0 [Nov 10 17:34:23] o=root 1298631169 1298631169 IN IP4 192.168.210.132 [Nov 10 17:34:23] s=Asterisk PBX GIT-master-0d85f18 [Nov 10 17:34:23] c=IN IP4 192.168.210.132 [Nov 10 17:34:23] t=0 0 [Nov 10 17:34:23] m=audio 11276 RTP/AVP 8 101 [Nov 10 17:34:23] a=rtpmap:8 PCMA/8000 [Nov 10 17:34:23] a=rtpmap:101 telephone-event/8000 [Nov 10 17:34:23] a=fmtp:101 0-16 [Nov 10 17:34:23] a=maxptime:150 [Nov 10 17:34:23] a=sendrecv [Nov 10 17:34:23] [Nov 10 17:34:23] <------------> [Nov 10 17:34:23] [Nov 10 17:34:23] <--- SIP read from UDP:192.168.210.111:5060 ---> [Nov 10 17:34:23] ACK sip:6100@192.168.210.132:5060 SIP/2.0 [Nov 10 17:34:23] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK80519034d1a5e6118a41991006d58e79;rport [Nov 10 17:34:23] From: "PhonerLite" ;tag=2804143771 [Nov 10 17:34:23] To: ;tag=as6a078b2a [Nov 10 17:34:23] Call-ID: 80519034-D1A5-E611-8A3E-991006D58E79@192.168.210.111 [Nov 10 17:34:23] CSeq: 13 ACK [Nov 10 17:34:23] Contact: [Nov 10 17:34:23] Authorization: Digest username="290", realm="pc_dany_asterisk", nonce="06ee15f4", uri="sip:6100@192.168.210.132:5060", response="57e3f971192c0354b74079013a30bf74", algorithm=MD5 [Nov 10 17:34:23] Max-Forwards: 70 [Nov 10 17:34:23] Content-Length: 0 [Nov 10 17:34:23] [Nov 10 17:34:23] <-------------> [Nov 10 17:34:23] --- (10 headers 0 lines) --- [Nov 10 17:34:23]  -- Executing [6100@LocalSets:3] Stasis("SIP/290-00000001", "attendant") in new stack [Nov 10 17:34:28] set_destination: Parsing for address/port to send to [Nov 10 17:34:28] set_destination: set destination to 192.168.210.111:5060 [Nov 10 17:34:28] Reliably Transmitting (no NAT) to 192.168.210.111:5060: [Nov 10 17:34:28] REFER sip:290@192.168.210.111:5060 SIP/2.0 [Nov 10 17:34:28] Via: SIP/2.0/UDP 192.168.210.132:5060;branch=z9hG4bK1ad4c226;rport [Nov 10 17:34:28] Max-Forwards: 70 [Nov 10 17:34:28] From: ;tag=as6a078b2a [Nov 10 17:34:28] To: "PhonerLite" ;tag=2804143771 [Nov 10 17:34:28] Contact: [Nov 10 17:34:28] Call-ID: 80519034-D1A5-E611-8A3E-991006D58E79@192.168.210.111 [Nov 10 17:34:28] CSeq: 102 REFER [Nov 10 17:34:28] User-Agent: Asterisk PBX GIT-master-0d85f18 [Nov 10 17:34:28] Date: Thu, 10 Nov 2016 16:34:28 GMT [Nov 10 17:34:28] Refer-To: [Nov 10 17:34:28] Referred-By: [Nov 10 17:34:28] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Nov 10 17:34:28] Supported: replaces, timer [Nov 10 17:34:28] Content-Length: 0 [Nov 10 17:34:28] [Nov 10 17:34:28] [Nov 10 17:34:28] --- [Nov 10 17:34:28] [Nov 10 17:34:28] <--- SIP read from UDP:192.168.210.111:5060 ---> [Nov 10 17:34:28] SIP/2.0 202 Accepted [Nov 10 17:34:28] Via: SIP/2.0/UDP 192.168.210.132:5060;branch=z9hG4bK1ad4c226;rport=5060 [Nov 10 17:34:28] From: ;tag=as6a078b2a [Nov 10 17:34:28] To: "PhonerLite" ;tag=2804143771 [Nov 10 17:34:28] Call-ID: 80519034-D1A5-E611-8A3E-991006D58E79@192.168.210.111 [Nov 10 17:34:28] CSeq: 102 REFER [Nov 10 17:34:28] Contact: [Nov 10 17:34:28] Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE [Nov 10 17:34:28] Server: SIPPER for PhonerLite [Nov 10 17:34:28] Content-Length: 0 [Nov 10 17:34:28] [Nov 10 17:34:28] <-------------> [Nov 10 17:34:28] --- (10 headers 0 lines) --- [Nov 10 17:34:28] [Nov 10 17:34:28] <--- SIP read from UDP:192.168.210.111:5060 ---> [Nov 10 17:34:28] INVITE sip:102@192.168.210.71 SIP/2.0 [Nov 10 17:34:28] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK00428b37d1a5e6118a42991006d58e79;rport [Nov 10 17:34:28] From: "PhonerLite" ;tag=1192990926 [Nov 10 17:34:28] To: [Nov 10 17:34:28] Call-ID: 00428B37-D1A5-E611-8A41-991006D58E79@192.168.210.111 [Nov 10 17:34:28] CSeq: 14 INVITE [Nov 10 17:34:28] Contact: [Nov 10 17:34:28] Content-Type: application/sdp [Nov 10 17:34:28] Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE [Nov 10 17:34:28] Max-Forwards: 70 [Nov 10 17:34:28] Supported: 100rel, replaces, from-change [Nov 10 17:34:28] P-Early-Media: supported [Nov 10 17:34:28] User-Agent: SIPPER for PhonerLite [Nov 10 17:34:28] P-Preferred-Identity: [Nov 10 17:34:28] Content-Length: 450 [Nov 10 17:34:28] [Nov 10 17:34:28] v=0 [Nov 10 17:34:28] o=- 461280829 1 IN IP4 192.168.210.111 [Nov 10 17:34:28] s=SIPPER for PhonerLite [Nov 10 17:34:28] c=IN IP4 192.168.210.111 [Nov 10 17:34:28] t=0 0 [Nov 10 17:34:28] m=audio 5064 RTP/AVP 107 8 0 2 3 97 110 111 9 101 [Nov 10 17:34:28] a=rtpmap:107 opus/48000/2 [Nov 10 17:34:28] a=rtpmap:8 PCMA/8000 [Nov 10 17:34:28] a=rtpmap:0 PCMU/8000 [Nov 10 17:34:28] a=rtpmap:2 G726-32/8000 [Nov 10 17:34:28] a=rtpmap:3 GSM/8000 [Nov 10 17:34:28] a=rtpmap:97 iLBC/8000 [Nov 10 17:34:28] a=rtpmap:110 speex/8000 [Nov 10 17:34:28] a=rtpmap:111 speex/16000 [Nov 10 17:34:28] a=rtpmap:9 G722/8000 [Nov 10 17:34:28] a=rtpmap:101 telephone-event/8000 [Nov 10 17:34:28] a=fmtp:101 0-16 [Nov 10 17:34:28] a=ssrc:2905527381 [Nov 10 17:34:28] a=sendrecv [Nov 10 17:34:28] <-------------> [Nov 10 17:34:28] --- (15 headers 19 lines) --- [Nov 10 17:34:28] Sending to 192.168.210.111:5060 (no NAT) [Nov 10 17:34:28] Sending to 192.168.210.111:5060 (no NAT) [Nov 10 17:34:28] Using INVITE request as basis request - 00428B37-D1A5-E611-8A41-991006D58E79@192.168.210.111 [Nov 10 17:34:28] Found peer '290' for '290' from 192.168.210.111:5060 [Nov 10 17:34:28] [Nov 10 17:34:28] <--- Reliably Transmitting (no NAT) to 192.168.210.111:5060 ---> [Nov 10 17:34:28] SIP/2.0 401 Unauthorized [Nov 10 17:34:28] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK00428b37d1a5e6118a42991006d58e79;received=192.168.210.111;rport=5060 [Nov 10 17:34:28] From: "PhonerLite" ;tag=1192990926 [Nov 10 17:34:28] To: ;tag=as00a36550 [Nov 10 17:34:28] Call-ID: 00428B37-D1A5-E611-8A41-991006D58E79@192.168.210.111 [Nov 10 17:34:28] CSeq: 14 INVITE [Nov 10 17:34:28] Server: Asterisk PBX GIT-master-0d85f18 [Nov 10 17:34:28] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Nov 10 17:34:28] Supported: replaces, timer [Nov 10 17:34:28] WWW-Authenticate: Digest algorithm=MD5, realm="pc_dany_asterisk", nonce="34dee277" [Nov 10 17:34:28] Content-Length: 0 [Nov 10 17:34:28] [Nov 10 17:34:28] [Nov 10 17:34:28] <------------> [Nov 10 17:34:28] Scheduling destruction of SIP dialog '00428B37-D1A5-E611-8A41-991006D58E79@192.168.210.111' in 32000 ms (Method: INVITE) [Nov 10 17:34:28] [Nov 10 17:34:28] <--- SIP read from UDP:192.168.210.111:5060 ---> [Nov 10 17:34:28] ACK sip:102@192.168.210.71 SIP/2.0 [Nov 10 17:34:28] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK00428b37d1a5e6118a42991006d58e79;rport [Nov 10 17:34:28] From: "PhonerLite" ;tag=1192990926 [Nov 10 17:34:28] To: ;tag=as00a36550 [Nov 10 17:34:28] Call-ID: 00428B37-D1A5-E611-8A41-991006D58E79@192.168.210.111 [Nov 10 17:34:28] CSeq: 14 ACK [Nov 10 17:34:28] Content-Length: 0 [Nov 10 17:34:28] [Nov 10 17:34:28] <-------------> [Nov 10 17:34:28] --- (7 headers 0 lines) --- [Nov 10 17:34:28] [Nov 10 17:34:28] <--- SIP read from UDP:192.168.210.111:5060 ---> [Nov 10 17:34:28] INVITE sip:102@192.168.210.71 SIP/2.0 [Nov 10 17:34:28] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK00428b37d1a5e6118a43991006d58e79;rport [Nov 10 17:34:28] From: "PhonerLite" ;tag=1192990926 [Nov 10 17:34:28] To: [Nov 10 17:34:28] Call-ID: 00428B37-D1A5-E611-8A41-991006D58E79@192.168.210.111 [Nov 10 17:34:28] CSeq: 15 INVITE [Nov 10 17:34:28] Contact: [Nov 10 17:34:28] Authorization: Digest username="290", realm="pc_dany_asterisk", nonce="34dee277", uri="sip:102@192.168.210.71", response="9facd3c2c164923b6ae9b51b4f0c2f80", algorithm=MD5 [Nov 10 17:34:28] Content-Type: application/sdp [Nov 10 17:34:28] Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE [Nov 10 17:34:28] Max-Forwards: 70 [Nov 10 17:34:28] Supported: 100rel, replaces, from-change [Nov 10 17:34:28] P-Early-Media: supported [Nov 10 17:34:28] User-Agent: SIPPER for PhonerLite [Nov 10 17:34:28] P-Preferred-Identity: [Nov 10 17:34:28] Content-Length: 450 [Nov 10 17:34:28] [Nov 10 17:34:28] v=0 [Nov 10 17:34:28] o=- 461280829 1 IN IP4 192.168.210.111 [Nov 10 17:34:28] s=SIPPER for PhonerLite [Nov 10 17:34:28] c=IN IP4 192.168.210.111 [Nov 10 17:34:28] t=0 0 [Nov 10 17:34:28] m=audio 5064 RTP/AVP 107 8 0 2 3 97 110 111 9 101 [Nov 10 17:34:28] a=rtpmap:107 opus/48000/2 [Nov 10 17:34:28] a=rtpmap:8 PCMA/8000 [Nov 10 17:34:28] a=rtpmap:0 PCMU/8000 [Nov 10 17:34:28] a=rtpmap:2 G726-32/8000 [Nov 10 17:34:28] a=rtpmap:3 GSM/8000 [Nov 10 17:34:28] a=rtpmap:97 iLBC/8000 [Nov 10 17:34:28] a=rtpmap:110 speex/8000 [Nov 10 17:34:28] a=rtpmap:111 speex/16000 [Nov 10 17:34:28] a=rtpmap:9 G722/8000 [Nov 10 17:34:28] a=rtpmap:101 telephone-event/8000 [Nov 10 17:34:28] a=fmtp:101 0-16 [Nov 10 17:34:28] a=ssrc:2905527381 [Nov 10 17:34:28] a=sendrecv [Nov 10 17:34:28] <-------------> [Nov 10 17:34:28] --- (16 headers 19 lines) --- [Nov 10 17:34:28] Sending to 192.168.210.111:5060 (no NAT) [Nov 10 17:34:28] Using INVITE request as basis request - 00428B37-D1A5-E611-8A41-991006D58E79@192.168.210.111 [Nov 10 17:34:28] Found peer '290' for '290' from 192.168.210.111:5060 [Nov 10 17:34:28]  == Using SIP RTP CoS mark 5 [Nov 10 17:34:28] Found RTP audio format 107 [Nov 10 17:34:28] Found RTP audio format 8 [Nov 10 17:34:28] Found RTP audio format 0 [Nov 10 17:34:28] Found RTP audio format 2 [Nov 10 17:34:28] Found RTP audio format 3 [Nov 10 17:34:28] Found RTP audio format 97 [Nov 10 17:34:28] Found RTP audio format 110 [Nov 10 17:34:28] Found RTP audio format 111 [Nov 10 17:34:28] Found RTP audio format 9 [Nov 10 17:34:28] Found RTP audio format 101 [Nov 10 17:34:28] Found audio description format opus for ID 107 [Nov 10 17:34:28] Found audio description format PCMA for ID 8 [Nov 10 17:34:28] Found audio description format PCMU for ID 0 [Nov 10 17:34:28] Found audio description format G726-32 for ID 2 [Nov 10 17:34:28] Found audio description format GSM for ID 3 [Nov 10 17:34:28] Found audio description format iLBC for ID 97 [Nov 10 17:34:28] Found audio description format speex for ID 110 [Nov 10 17:34:28] Found audio description format speex for ID 111 [Nov 10 17:34:28] Found audio description format G722 for ID 9 [Nov 10 17:34:28] Found audio description format telephone-event for ID 101 [Nov 10 17:34:28] Capabilities: us - (alaw), peer - audio=(ulaw|gsm|alaw|g722|ilbc|opus|speex|speex16)/video=(nothing)/text=(nothing), combined - (alaw) [Nov 10 17:34:28] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Nov 10 17:34:28] Peer audio RTP is at port 192.168.210.111:5064 [Nov 10 17:34:28] Peer doesn't provide T.140 [Nov 10 17:34:28] Looking for 102 in LocalSets (domain 192.168.210.71) [Nov 10 17:34:28] sip_route_dump: route/path hop: [Nov 10 17:34:28] [Nov 10 17:34:28] <--- Transmitting (no NAT) to 192.168.210.111:5060 ---> [Nov 10 17:34:28] SIP/2.0 100 Trying [Nov 10 17:34:28] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK00428b37d1a5e6118a43991006d58e79;received=192.168.210.111;rport=5060 [Nov 10 17:34:28] From: "PhonerLite" ;tag=1192990926 [Nov 10 17:34:28] To: [Nov 10 17:34:28] Call-ID: 00428B37-D1A5-E611-8A41-991006D58E79@192.168.210.111 [Nov 10 17:34:28] CSeq: 15 INVITE [Nov 10 17:34:28] Server: Asterisk PBX GIT-master-0d85f18 [Nov 10 17:34:28] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Nov 10 17:34:28] Supported: replaces, timer [Nov 10 17:34:28] Contact: [Nov 10 17:34:28] Content-Length: 0 [Nov 10 17:34:28] [Nov 10 17:34:28] [Nov 10 17:34:28] <------------> [Nov 10 17:34:28]  -- Executing [102@LocalSets:1] Dial("SIP/290-00000002", "SIP/102") in new stack [Nov 10 17:34:28] WARNING[6974][C-00000003]: app_dial.c:2528 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) [Nov 10 17:34:28]  == Everyone is busy/congested at this time (1:0/0/1) [Nov 10 17:34:28]  -- Executing [102@LocalSets:2] SayDigits("SIP/290-00000002", "102") in new stack [Nov 10 17:34:28]  --  Playing 'digits/1.gsm' (language 'en') [Nov 10 17:34:28] [Nov 10 17:34:28] <--- SIP read from UDP:192.168.210.111:5060 ---> [Nov 10 17:34:28] BYE sip:6100@192.168.210.132:5060 SIP/2.0 [Nov 10 17:34:28] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK00428b37d1a5e6118a44991006d58e79;rport [Nov 10 17:34:28] From: "PhonerLite" ;tag=2804143771 [Nov 10 17:34:28] To: ;tag=as6a078b2a [Nov 10 17:34:28] Call-ID: 80519034-D1A5-E611-8A3E-991006D58E79@192.168.210.111 [Nov 10 17:34:28] CSeq: 16 BYE [Nov 10 17:34:28] Contact: [Nov 10 17:34:28] Authorization: Digest username="290", realm="pc_dany_asterisk", nonce="06ee15f4", uri="sip:6100@192.168.210.132:5060", response="81d36c616f097bcef59d24de6e31fb97", algorithm=MD5 [Nov 10 17:34:28] Max-Forwards: 70 [Nov 10 17:34:28] User-Agent: SIPPER for PhonerLite [Nov 10 17:34:28] Content-Length: 0 [Nov 10 17:34:28] [Nov 10 17:34:28] <-------------> [Nov 10 17:34:28] --- (11 headers 0 lines) --- [Nov 10 17:34:28] Sending to 192.168.210.111:5060 (no NAT) [Nov 10 17:34:28] Scheduling destruction of SIP dialog '80519034-D1A5-E611-8A3E-991006D58E79@192.168.210.111' in 32000 ms (Method: BYE) [Nov 10 17:34:28] [Nov 10 17:34:28] <--- Transmitting (no NAT) to 192.168.210.111:5060 ---> [Nov 10 17:34:28] SIP/2.0 200 OK [Nov 10 17:34:28] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK00428b37d1a5e6118a44991006d58e79;received=192.168.210.111;rport=5060 [Nov 10 17:34:28] From: "PhonerLite" ;tag=2804143771 [Nov 10 17:34:28] To: ;tag=as6a078b2a [Nov 10 17:34:28] Call-ID: 80519034-D1A5-E611-8A3E-991006D58E79@192.168.210.111 [Nov 10 17:34:28] CSeq: 16 BYE [Nov 10 17:34:28] Server: Asterisk PBX GIT-master-0d85f18 [Nov 10 17:34:28] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Nov 10 17:34:28] Supported: replaces, timer [Nov 10 17:34:28] Content-Length: 0 [Nov 10 17:34:28] [Nov 10 17:34:28] [Nov 10 17:34:28] <------------> [Nov 10 17:34:28] Really destroying SIP dialog '71e3b8034c88188778e225ef025b4dde@127.0.1.1:5060' Method: INVITE [Nov 10 17:34:29]  --  Playing 'digits/0.gsm' (language 'en') [Nov 10 17:34:29] [Nov 10 17:34:29] <--- SIP read from UDP:192.168.210.40:5062 ---> [Nov 10 17:34:29] [Nov 10 17:34:29] [Nov 10 17:34:29] <-------------> [Nov 10 17:34:30]  --  Playing 'digits/2.gsm' (language 'en') [Nov 10 17:34:31]  -- Auto fallthrough, channel 'SIP/290-00000002' status is 'CHANUNAVAIL' [Nov 10 17:34:31] [Nov 10 17:34:31] <--- Reliably Transmitting (no NAT) to 192.168.210.111:5060 ---> [Nov 10 17:34:31] SIP/2.0 503 Service Unavailable [Nov 10 17:34:31] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK00428b37d1a5e6118a43991006d58e79;received=192.168.210.111;rport=5060 [Nov 10 17:34:31] From: "PhonerLite" ;tag=1192990926 [Nov 10 17:34:31] To: ;tag=as7f7c4912 [Nov 10 17:34:31] Call-ID: 00428B37-D1A5-E611-8A41-991006D58E79@192.168.210.111 [Nov 10 17:34:31] CSeq: 15 INVITE [Nov 10 17:34:31] Server: Asterisk PBX GIT-master-0d85f18 [Nov 10 17:34:31] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Nov 10 17:34:31] Supported: replaces, timer [Nov 10 17:34:31] X-Asterisk-HangupCause: Subscriber absent [Nov 10 17:34:31] X-Asterisk-HangupCauseCode: 20 [Nov 10 17:34:31] Content-Length: 0 [Nov 10 17:34:31] [Nov 10 17:34:31] [Nov 10 17:34:31] <------------> [Nov 10 17:34:31] [Nov 10 17:34:31] <--- SIP read from UDP:192.168.210.111:5060 ---> [Nov 10 17:34:31] ACK sip:102@192.168.210.71 SIP/2.0 [Nov 10 17:34:31] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK00428b37d1a5e6118a43991006d58e79;rport [Nov 10 17:34:31] From: "PhonerLite" ;tag=1192990926 [Nov 10 17:34:31] To: ;tag=as7f7c4912 [Nov 10 17:34:31] Call-ID: 00428B37-D1A5-E611-8A41-991006D58E79@192.168.210.111 [Nov 10 17:34:31] CSeq: 15 ACK [Nov 10 17:34:31] Authorization: Digest username="290", realm="pc_dany_asterisk", nonce="34dee277", uri="sip:102@192.168.210.71", response="9facd3c2c164923b6ae9b51b4f0c2f80", algorithm=MD5 [Nov 10 17:34:31] Content-Length: 0 [Nov 10 17:34:31] [Nov 10 17:34:31] <-------------> [Nov 10 17:34:31] --- (8 headers 0 lines) --- [Nov 10 17:34:31] [Nov 10 17:34:31] <--- SIP read from UDP:192.168.210.111:5060 ---> [Nov 10 17:34:31] ACK sip:102@192.168.210.71 SIP/2.0 [Nov 10 17:34:31] Via: SIP/2.0/UDP 192.168.210.111:5060;branch=z9hG4bK00428b37d1a5e6118a43991006d58e79;rport [Nov 10 17:34:31] From: "PhonerLite" ;tag=1192990926 [Nov 10 17:34:31] To: ;tag=as7f7c4912 [Nov 10 17:34:31] Call-ID: 00428B37-D1A5-E611-8A41-991006D58E79@192.168.210.111 [Nov 10 17:34:31] CSeq: 15 ACK [Nov 10 17:34:31] Authorization: Digest username="290", realm="pc_dany_asterisk", nonce="34dee277", uri="sip:102@192.168.210.71", response="9facd3c2c164923b6ae9b51b4f0c2f80", algorithm=MD5 [Nov 10 17:34:31] Content-Length: 0 [Nov 10 17:34:31] [Nov 10 17:34:31] <-------------> [Nov 10 17:34:31] --- (8 headers 0 lines) --- [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable! [Nov 10 17:34:31] WARNING[6974][C-00000003]: channel.c:1187 __ast_queue_frame: Unable to write to alert pipe on SIP/290-00000002 (qlen = 0): Resource temporarily unavailable!