Summary: | ASTERISK-24762: not able to make outbound call from Asterisk/1.8.13.1 | ||
Reporter: | Ashish (Ashish) | Labels: | |
Date Opened: | 2015-02-05 23:14:59.000-0600 | Date Closed: | 2015-02-06 09:33:33.000-0600 |
Priority: | Major | Regression? | |
Status: | Closed/Complete | Components: | Channels/chan_sip/General |
Versions: | 1.8.13.1 | Frequency of Occurrence | Constant |
Related Issues: | |||
Environment: | == Using SIP RTP CoS mark 5 -- Executing [61452377795@outgoing:1] NoOp("SIP/61500-00000018", "outgoing CALLERID(all) "ASHISH" <61500> calling EXTEN 61452377795") in new stack -- Auto fallthrough, channel 'SIP/61500-00000018' status is 'UNKNOWN' | Attachments: | |
Description: | {noformat}
== Using SIP RTP CoS mark 5 -- Executing [61452377795@outgoing:1] NoOp("SIP/61500-00000018", "outgoing CALLERID(all) "ASHISH" <61500> calling EXTEN 61452377795") in new stack -- Auto fallthrough, channel 'SIP/61500-00000018' status is 'UNKNOWN' {noformat} | ||
Comments: | By: Ashish (Ashish) 2015-02-05 23:31:39.387-0600 {noformat} =~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2015.02.06 15:57:40 =~=~=~=~=~=~=~=~=~=~=~= [0KReliably Transmitting (NAT) to 10.61.0.101:5060: OPTIONS sip:61500@10.61.0.101:5060 SIP/2.0 Via: SIP/2.0/UDP 10.61.0.4:5060;branch=z9hG4bK6f1253a9;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@10.61.0.4>;tag=as646db1b8 To: <sip:61500@10.61.0.101:5060> Contact: <sip:asterisk@10.61.0.4:5060> Call-ID: 1cc3fdb05f09b56939445076518e7cc6@10.61.0.4:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 Date: Fri, 06 Feb 2015 04:57:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Kipbx-mlb*CLI> [0K <--- SIP read from UDP:10.61.0.101:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.61.0.4:5060;branch=z9hG4bK6f1253a9;rport=5060 From: "asterisk" <sip:asterisk@10.61.0.4>;tag=as646db1b8 To: <sip:61500@10.61.0.101:5060>;tag=80b95d422aace411923a46763858d8a5 Call-ID: 1cc3fdb05f09b56939445076518e7cc6@10.61.0.4:5060 CSeq: 102 OPTIONS Contact: <sip:61500@10.61.0.101:5060> Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Server: SIPPER for PhonerLite Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Kipbx-mlb*CLI> [0KReally destroying SIP dialog '1cc3fdb05f09b56939445076518e7cc6@10.61.0.4:5060' Method: OPTIONS [Kipbx-mlb*CLI> [0KReally destroying SIP dialog '802E0436-4EAB-E411-9D91-16652760B7C0@10.61.0.110' Method: REGISTER [Kipbx-mlb*CLI> [0KReally destroying SIP dialog '005D132D-4EAB-E411-B4FA-7201B2F660D5@10.61.0.104' Method: REGISTER [Kipbx-mlb*CLI> [0K <--- SIP read from UDP:10.61.0.101:5060 ---> INVITE sip:61452377795@10.61.0.4 SIP/2.0 Via: SIP/2.0/UDP 10.61.0.101:5060;branch=z9hG4bK8097975c2aace411923b46763858d8a5;rport From: "PhonerLite" <sip:61500@10.61.0.4>;tag=1643736679 To: <sip:61452377795@10.61.0.4> Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5@10.61.0.101 CSeq: 108 INVITE Contact: <sip:61500@10.61.0.101:5060> Content-Type: application/sdp Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Max-Forwards: 70 Supported: 100rel, replaces, from-change P-Early-Media: supported User-Agent: SIPPER for PhonerLite P-Preferred-Identity: <sip:61500@10.61.0.4> Content-Length: 302 v=0 o=- 4097900355 1 IN IP4 10.61.0.101 s=SIPPER for PhonerLite c=IN IP4 10.61.0.101 t=0 0 m=audio 5062 RTP/AVP 8 0 3 97 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ssrc:2997880776 a=sendrecv <-------------> --- (15 headers 14 lines) --- [Kipbx-mlb*CLI> [0KSending to 10.61.0.101:5060 (no NAT) Using INVITE request as basis request - 8097975C-2AAC-E411-923A-46763858D8A5@10.61.0.101 [Kipbx-mlb*CLI> [0KFound peer '61500' for '61500' from 10.61.0.101:5060 [Kipbx-mlb*CLI> [0K <--- Reliably Transmitting (NAT) to 10.61.0.101:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.61.0.101:5060;branch=z9hG4bK8097975c2aace411923b46763858d8a5;received=10.61.0.101;rport=5060 From: "PhonerLite" <sip:61500@10.61.0.4>;tag=1643736679 To: <sip:61452377795@10.61.0.4>;tag=as43203775 Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5@10.61.0.101 CSeq: 108 INVITE Server: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Kipbx-mlb*CLI> [0KSupported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1c5808b3" Content-Length: 0 <------------> [Kipbx-mlb*CLI> [0KScheduling destruction of SIP dialog '8097975C-2AAC-E411-923A-46763858D8A5@10.61.0.101' in 6400 ms (Method: INVITE) [Kipbx-mlb*CLI> [0K <--- SIP read from UDP:10.61.0.101:5060 ---> ACK sip:61452377795@10.61.0.4 SIP/2.0 Via: SIP/2.0/UDP 10.61.0.101:5060;branch=z9hG4bK8097975c2aace411923b46763858d8a5;rport From: "PhonerLite" <sip:61500@10.61.0.4>;tag=1643736679 To: <sip:61452377795@10.61.0.4>;tag=as43203775 Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5@10.61.0.101 CSeq: 108 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- [Kipbx-mlb*CLI> [0K <--- SIP read from UDP:10.61.0.101:5060 ---> INVITE sip:61452377795@10.61.0.4 SIP/2.0 Via: SIP/2.0/UDP 10.61.0.101:5060;branch=z9hG4bK8097975c2aace411923c46763858d8a5;rport From: "PhonerLite" <sip:61500@10.61.0.4>;tag=1643736679 To: <sip:61452377795@10.61.0.4> Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5@10.61.0.101 CSeq: 109 INVITE Contact: <sip:61500@10.61.0.101:5060> Authorization: Digest username="61500", realm="asterisk", nonce="1c5808b3", uri="sip:61452377795@10.61.0.4", response="d13bba2f0e2554ca11820f9b66786718", algorithm=MD5 Content-Type: application/sdp Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Max-Forwards: 70 Supported: 100rel, replaces, from-change P-Early-Media: supported User-Agent: SIPPER for PhonerLite P-Preferred-Identity: <sip:61500@10.61.0.4> Content-Length: 302 v=0 o=- 4097900355 1 IN IP4 10.61.0.101 s=SIPPER for PhonerLite c=IN IP4 10.61.0.101 t=0 0 m=audio 5062 RTP/AVP 8 0 3 97 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ssrc:2997880776 a=sendrecv <-------------> --- (16 headers 14 lines) --- [Kipbx-mlb*CLI> [0KSending to 10.61.0.101:5060 (NAT) Using INVITE request as basis request - 8097975C-2AAC-E411-923A-46763858D8A5@10.61.0.101 Found peer '61500' for '61500' from 10.61.0.101:5060 [Kipbx-mlb*CLI> [0K == Using SIP RTP CoS mark 5 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 [Kipbx-mlb*CLI> [0KFound audio description format GSM for ID 3 Found audio description format iLBC for ID 97 Found audio description format telephone-event for ID 101 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x6 (gsm|ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.61.0.101:5062 [Kipbx-mlb*CLI> [0KLooking for 61452377795 in outgoing (domain 10.61.0.4) [Kipbx-mlb*CLI> [0Klist_route: hop: <sip:61500@10.61.0.101:5060> [Kipbx-mlb*CLI> [0K <--- Transmitting (NAT) to 10.61.0.101:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.61.0.101:5060;branch=z9hG4bK8097975c2aace411923c46763858d8a5;received=10.61.0.101;rport=5060 From: "PhonerLite" <sip:61500@10.61.0.4>;tag=1643736679 To: <sip:61452377795@10.61.0.4> Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5@10.61.0.101 CSeq: 109 INVITE Server: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:61452377795@10.61.0.4:5060> Content-Length: 0 <------------> [Kipbx-mlb*CLI> [0K -- Executing [61452377795@outgoing:1] [1;36mNoOp[0m("[1;35mSIP/61500-00000017[0m", "[1;35moutgoing CALLERID(all) "ASHISH" <61500> calling EXTEN 61452377795[0m") in new stack [Kipbx-mlb*CLI> [0K -- Auto fallthrough, channel 'SIP/61500-00000017' status is 'UNKNOWN' [Kipbx-mlb*CLI> [0KScheduling destruction of SIP dialog '8097975C-2AAC-E411-923A-46763858D8A5@10.61.0.101' in 6400 ms (Method: INVITE) [Kipbx-mlb*CLI> [0K <--- Reliably Transmitting (NAT) to 10.61.0.101:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 10.61.0.101:5060;branch=z9hG4bK8097975c2aace411923c46763858d8a5;received=10.61.0.101;rport=5060 From: "PhonerLite" <sip:61500@10.61.0.4>;tag=1643736679 To: <sip:61452377795@10.61.0.4>;tag=as105d9377 Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5@10.61.0.101 CSeq: 109 INVITE Server: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Kipbx-mlb*CLI> [0K <--- SIP read from UDP:10.61.0.101:5060 ---> ACK sip:61452377795@10.61.0.4 SIP/2.0 Via: SIP/2.0/UDP 10.61.0.101:5060;branch=z9hG4bK8097975c2aace411923c46763858d8a5;rport From: "PhonerLite" <sip:61500@10.61.0.4>;tag=1643736679 To: <sip:61452377795@10.61.0.4>;tag=as105d9377 Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5@10.61.0.101 CSeq: 109 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- [Kipbx-mlb*CLI> [0KReliably Transmitting (no NAT) to 10.61.0.110:5060: OPTIONS sip:61509@10.61.0.110:5060 SIP/2.0 Via: SIP/2.0/UDP 10.61.0.4:5060;branch=z9hG4bK26ced09d Max-Forwards: 70 From: "asterisk" <sip:asterisk@10.61.0.4>;tag=as78edb60e To: <sip:61509@10.61.0.110:5060> Contact: <sip:asterisk@10.61.0.4:5060> Call-ID: 3f03ecf16b30b999040cda492f2a6100@10.61.0.4:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 Date: Fri, 06 Feb 2015 04:58:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Kipbx-mlb*CLI> [0KRetransmitting #1 (no NAT) to 10.61.0.110:5060: OPTIONS sip:61509@10.61.0.110:5060 SIP/2.0 Via: SIP/2.0/UDP 10.61.0.4:5060;branch=z9hG4bK26ced09d Max-Forwards: 70 From: "asterisk" <sip:asterisk@10.61.0.4>;tag=as78edb60e To: <sip:61509@10.61.0.110:5060> Contact: <sip:asterisk@10.61.0.4:5060> Call-ID: 3f03ecf16b30b999040cda492f2a6100@10.61.0.4:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 Date: Fri, 06 Feb 2015 04:58:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Kipbx-mlb*CLI> [0KReliably Transmitting (NAT) to 10.61.0.104:5060: OPTIONS sip:61504@10.61.0.104:5060 SIP/2.0 Via: SIP/2.0/UDP 10.61.0.4:5060;branch=z9hG4bK37ee2d9b;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@10.61.0.4>;tag=as6a5cfaad To: <sip:61504@10.61.0.104:5060> Contact: <sip:asterisk@10.61.0.4:5060> Call-ID: 4afe1d1e7fe19bab7f80b02218709b96@10.61.0.4:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 Date: Fri, 06 Feb 2015 04:58:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Kipbx-mlb*CLI> [0K <--- SIP read from UDP:10.61.0.104:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.61.0.4:5060;branch=z9hG4bK37ee2d9b;rport=5060 From: "asterisk" <sip:asterisk@10.61.0.4>;tag=as6a5cfaad To: <sip:61504@10.61.0.104:5060>;tag=802c52672aace411b6857201b2f660d5 Call-ID: 4afe1d1e7fe19bab7f80b02218709b96@10.61.0.4:5060 CSeq: 102 OPTIONS Contact: <sip:61504@10.61.0.104:5060> Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Server: SIPPER for PhonerLite Content-Length: 0 <-------------> --- (10 headers 0 lines) --- [Kipbx-mlb*CLI> [0KReally destroying SIP dialog '4afe1d1e7fe19bab7f80b02218709b96@10.61.0.4:5060' Method: OPTIONS [Kipbx-mlb*CLI> [0KRetransmitting #2 (no NAT) to 10.61.0.110:5060: OPTIONS sip:61509@10.61.0.110:5060 SIP/2.0 Via: SIP/2.0/UDP 10.61.0.4:5060;branch=z9hG4bK26ced09d Max-Forwards: 70 From: "asterisk" <sip:asterisk@10.61.0.4>;tag=as78edb60e To: <sip:61509@10.61.0.110:5060> Contact: <sip:asterisk@10.61.0.4:5060> Call-ID: 3f03ecf16b30b999040cda492f2a6100@10.61.0.4:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 Date: Fri, 06 Feb 2015 04:58:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Kipbx-mlb*CLI> [0KRetransmitting #3 (no NAT) to 10.61.0.110:5060: OPTIONS sip:61509@10.61.0.110:5060 SIP/2.0 Via: SIP/2.0/UDP 10.61.0.4:5060;branch=z9hG4bK26ced09d Max-Forwards: 70 From: "asterisk" <sip:asterisk@10.61.0.4>;tag=as78edb60e To: <sip:61509@10.61.0.110:5060> Contact: <sip:asterisk@10.61.0.4:5060> Call-ID: 3f03ecf16b30b999040cda492f2a6100@10.61.0.4:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 Date: Fri, 06 Feb 2015 04:58:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Kipbx-mlb*CLI> sip set debug on [0KRetransmitting #4 (no NAT) to 10.61.0.110:5060: OPTIONS sip:61509@10.61.0.110:5060 SIP/2.0 Via: SIP/2.0/UDP 10.61.0.4:5060;branch=z9hG4bK26ced09d Max-Forwards: 70 From: "asterisk" <sip:asterisk@10.61.0.4>;tag=as78edb60e To: <sip:61509@10.61.0.110:5060> Contact: <sip:asterisk@10.61.0.4:5060> Call-ID: 3f03ecf16b30b999040cda492f2a6100@10.61.0.4:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 Date: Fri, 06 Feb 2015 04:58:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Feb 6 04:58:11] [1;33mNOTICE[0m[17885]: [1;37mchan_sip.c[0m:[1;37m26267[0m [1;37msip_poke_noanswer[0m: Peer '61509' is now UNREACHABLE! Last qualify: 1 Really destroying SIP dialog '3f03ecf16b30b999040cda492f2a6100@10.61.0.4:5060' Method: OPTIONS [Kipbx-mlb*CLI> sip set debug on[Kf [0KReally destroying SIP dialog '8097975C-2AAC-E411-923A-46763858D8A5@10.61.0.101' Method: ACK [Kipbx-mlb*CLI> sip set debug off ipbx-mlb*CLI> [0KSIP Debugging Disabled [Kipbx-mlb*CLI> {noformat} By: Ashish (Ashish) 2015-02-05 23:32:02.760-0600 Global Settings: ---------------- UDP Bindaddress: 0.0.0.0:5060 TCP SIP Bindaddress: Disabled TLS SIP Bindaddress: Disabled Videosupport: No Textsupport: No Ignore SDP sess. ver.: No AutoCreate Peer: No Match Auth Username: No Allow unknown access: Yes Allow subscriptions: Yes Allow overlap dialing: Yes Allow promisc. redir: No Enable call counters: No SIP domain support: Yes Realm. auth: No Our auth realm asterisk Use domains as realms: No Call to non-local dom.: Yes URI user is phone no: No Always auth rejects: Yes Direct RTP setup: No User Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 SDP Session Name: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3 SDP Owner Name: root Reg. context: (not set) Regexten on Qualify: No Legacy userfield parse: No Caller ID: asterisk From: Domain: Record SIP history: On Call Events: Off Auth. Failure Events: Off T.38 support: No T.38 EC mode: Unknown T.38 MaxDtgrm: -1 SIP realtime: Disabled Qualify Freq : 60000 ms Q.850 Reason header: No Store SIP_CAUSE: No Network QoS Settings: --------------------------- IP ToS SIP: CS0 IP ToS RTP audio: CS0 IP ToS RTP video: CS0 IP ToS RTP text: CS0 802.1p CoS SIP: 4 802.1p CoS RTP audio: 5 802.1p CoS RTP video: 6 802.1p CoS RTP text: 5 Jitterbuffer enabled: No Network Settings: --------------------------- SIP address remapping: Enabled using externaddr Externhost: <none> Externaddr: 110.142.242.138:0 Externrefresh: 10 Localnet: 10.61.0.0/255.255.192.0 Global Signalling Settings: --------------------------- Codecs: 0x80000008050e (gsm|ulaw|alaw|g729|ilbc|h263|testlaw) Codec Order: alaw:20,gsm:20,g729:20,ilbc:30 Relax DTMF: No RFC2833 Compensation: No Symmetric RTP: No Compact SIP headers: No RTP Keepalive: 0 (Disabled) RTP Timeout: 0 (Disabled) RTP Hold Timeout: 0 (Disabled) MWI NOTIFY mime type: application/simple-message-summary DNS SRV lookup: Yes Pedantic SIP support: No Reg. min duration 60 secs Reg. max duration: 3600 secs Reg. default duration: 1800 secs Outbound reg. timeout: 20 secs Outbound reg. attempts: 0 Notify ringing state: Yes Include CID: No Notify hold state: Yes SIP Transfer mode: open Max Call Bitrate: 384 kbps Auto-Framing: No Outb. proxy: <not set> Session Timers: Accept Session Refresher: uas Session Expires: 1800 secs Session Min-SE: 90 secs Timer T1: 500 Timer T1 minimum: 100 Timer B: 32000 No premature media: Yes Max forwards: 70 Default Settings: ----------------- Allowed transports: UDP Outbound transport: UDP Context: default Force rport: No DTMF: rfc2833 Qualify: 0 Use ClientCode: No Progress inband: Never Language: MOH Interpret: default MOH Suggest: Voice Mail Extension: asterisk By: Matt Jordan (mjordan) 2015-02-06 09:33:27.484-0600 Thanks for your comments. This does not appear to be a bug report and we are closing it. We appreciate the difficulties you are facing, but it would make more sense to raise your question in the support tracker, http://www.asterisk.org/support. |