Jump to content

Obed Alba

Members
  • Posts

    4
  • Joined

  • Last visited

Obed Alba's Achievements

Newbie

Newbie (1/14)

0

Reputation

  1. In the extension level say: Miscellaneous: Block outgoing caller-ID: Yes I changed to No and try again and does the same, always busy at the outgoing calls
  2. The thing that I noticed in my system is that the snom one can not "communicate" with the sangoma card, for this reason the outbound calls can not be made. I can receive calls, but can not make calls. Any other idea? Thanks
  3. Could you eplain in plain english? I'm new in snom and don not know what is turned clip on
  4. Hello All I have a System running good for a while, but recently seems so weird, My setup is a Sangoma B600D with Snom One Unlimited and for some reason can receive calls, but can not make outoging calls it always sound busy,internal calls between extensions works fine in both ways. Any ideas what might be causing this, or where to look for clues? Thanks. This is my log [9] 2013/02/28 15:34:23: Resolve 1682: aaaa udp 192.168.137.108 5060 [9] 2013/02/28 15:34:23: Resolve 1682: a udp 192.168.137.108 5060 [9] 2013/02/28 15:34:23: Resolve 1682: udp 192.168.137.108 5060 [5] 2013/02/28 15:34:25: SIP Rx udp:192.168.137.154:2048: INVITE sip:7261247@192.168.137.254;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.137.154:2048;branch=z9hG4bK-em3z2eypz8y4;rport From: <sip:208@192.168.137.254>;tag=66vco17bd1 To: <sip:7261247@192.168.137.254;user=phone> Call-ID: 3c269177ceb5-a1za6z3kewa1 CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:208@192.168.137.154:2048;line=x83vrg9s>;reg-id=1 X-Serialnumber: 00041335C884 P-Key-Flags: keys="3" User-Agent: snom320/8.4.35 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Call-Info: <sip:192.168.137.254>;appearance-index=1 Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 481 v=0 o=root 1638961721 1638961721 IN IP4 192.168.137.154 s=call c=IN IP4 192.168.137.154 t=0 0 m=audio 56492 RTP/AVP 0 8 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:hYBafrqHq8h5F5E2TNCH8xfBRVgk8N57f86tv9mf a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [8] 2013/02/28 15:34:25: Allocating for call port 122, SIP call id 3c269177ceb5-a1za6z3kewa1 [9] 2013/02/28 15:34:25: UDP(IPv4): Opening socket on 0.0.0.0:53328 [9] 2013/02/28 15:34:25: UDP(IPv4): Opening socket on 0.0.0.0:53329 [8] 2013/02/28 15:34:25: Could not find a trunk (1 trunks) [9] 2013/02/28 15:34:25: Resolve 1683: aaaa udp 192.168.137.154 2048 [9] 2013/02/28 15:34:25: Resolve 1683: a udp 192.168.137.154 2048 [9] 2013/02/28 15:34:25: Resolve 1683: udp 192.168.137.154 2048 [5] 2013/02/28 15:34:25: SIP Tx udp:192.168.137.154:2048: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.137.154:2048;branch=z9hG4bK-em3z2eypz8y4;rport=2048 From: <sip:208@192.168.137.254>;tag=66vco17bd1 To: <sip:7261247@192.168.137.254;user=phone>;tag=c38522c82f Call-ID: 3c269177ceb5-a1za6z3kewa1 CSeq: 1 INVITE Content-Length: 0 [9] 2013/02/28 15:34:25: Resolve 1684: aaaa udp 192.168.137.154 2048 [9] 2013/02/28 15:34:25: Resolve 1684: a udp 192.168.137.154 2048 [9] 2013/02/28 15:34:25: Resolve 1684: udp 192.168.137.154 2048 [5] 2013/02/28 15:34:25: SIP Tx udp:192.168.137.154:2048: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 192.168.137.154:2048;branch=z9hG4bK-em3z2eypz8y4;rport=2048 From: <sip:208@192.168.137.254>;tag=66vco17bd1 To: <sip:7261247@192.168.137.254;user=phone>;tag=c38522c82f Call-ID: 3c269177ceb5-a1za6z3kewa1 CSeq: 1 INVITE User-Agent: snomONE/4.5.0.1090 Epsilon Geminids WWW-Authenticate: Digest realm="192.168.137.254",nonce="8a3ad8f16e25a19e70270697b124c7d4",domain="sip:7261247@192.168.137.254;user=phone",algorithm=MD5 Content-Length: 0 [5] 2013/02/28 15:34:26: SIP Rx udp:192.168.137.154:2048: ACK sip:7261247@192.168.137.254;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.137.154:2048;branch=z9hG4bK-em3z2eypz8y4;rport From: <sip:208@192.168.137.254>;tag=66vco17bd1 To: <sip:7261247@192.168.137.254;user=phone>;tag=c38522c82f Call-ID: 3c269177ceb5-a1za6z3kewa1 CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:208@192.168.137.154:2048;line=x83vrg9s>;reg-id=1 Content-Length: 0 [5] 2013/02/28 15:34:26: SIP Rx udp:192.168.137.154:2048: INVITE sip:7261247@192.168.137.254;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.137.154:2048;branch=z9hG4bK-crzvtm8g5izl;rport From: <sip:208@192.168.137.254>;tag=66vco17bd1 To: <sip:7261247@192.168.137.254;user=phone> Call-ID: 3c269177ceb5-a1za6z3kewa1 CSeq: 2 INVITE Max-Forwards: 70 Contact: <sip:208@192.168.137.154:2048;line=x83vrg9s>;reg-id=1 X-Serialnumber: 00041335C884 P-Key-Flags: keys="3" User-Agent: snom320/8.4.35 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Call-Info: <sip:192.168.137.254>;appearance-index=1 Session-Expires: 3600;refresher=uas Min-SE: 90 Authorization: Digest username="208",realm="192.168.137.254",nonce="8a3ad8f16e25a19e70270697b124c7d4",uri="sip:7261247@192.168.137.254;user=phone",response="5d37764e2b30417637b84556437b2258",algorithm=MD5 Content-Type: application/sdp Content-Length: 481 v=0 o=root 1638961721 1638961721 IN IP4 192.168.137.154 s=call c=IN IP4 192.168.137.154 t=0 0 m=audio 56492 RTP/AVP 0 8 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:hYBafrqHq8h5F5E2TNCH8xfBRVgk8N57f86tv9mf a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [8] 2013/02/28 15:34:26: Tagging request with existing tag [7] 2013/02/28 15:34:26: Set packet length to 20 [6] 2013/02/28 15:34:26: Call-leg 122: Sending RTP for 3c269177ceb5-a1za6z3kewa1 to 192.168.137.154:56492, codec not set yet [8] 2013/02/28 15:34:26: Incoming call: Request URI sip:7261247@192.168.137.254;user=phone, To is <sip:7261247@192.168.137.254;user=phone> [8] 2013/02/28 15:34:26: Call from an user 208 [9] 2013/02/28 15:34:26: Resolve 1685: aaaa udp 192.168.137.154 2048 [9] 2013/02/28 15:34:26: Resolve 1685: a udp 192.168.137.154 2048 [9] 2013/02/28 15:34:26: Resolve 1685: udp 192.168.137.154 2048 [5] 2013/02/28 15:34:26: SIP Tx udp:192.168.137.154:2048: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.137.154:2048;branch=z9hG4bK-crzvtm8g5izl;rport=2048 From: <sip:208@192.168.137.254>;tag=66vco17bd1 To: <sip:7261247@192.168.137.254;user=phone>;tag=c38522c82f Call-ID: 3c269177ceb5-a1za6z3kewa1 CSeq: 2 INVITE Content-Length: 0 [8] 2013/02/28 15:34:26: To is <sip:7261247@192.168.137.254;user=phone>, user 0, domain 1 [8] 2013/02/28 15:34:26: From user 208 [8] 2013/02/28 15:34:26: Set the To domain based on From user 208@pbx.company.com [8] 2013/02/28 15:34:26: Call state for call object 57: idle [7] 2013/02/28 15:34:26: Call port 122: set_codecs for 3c269177ceb5-a1za6z3kewa1 codecs "", codec_preference count 7 [9] 2013/02/28 15:34:26: Dialplan: Evaluating !^(\+?[0-9]*)@.*!sip:\1@\r;user=phone!i against 7261247@192.168.137.254 [5] 2013/02/28 15:34:26: Dialplan "Standar": Match 7261247@192.168.137.254 to sip:7261247@please.change;user=phone on trunk PSTN 8FXO [9] 2013/02/28 15:34:26: Generating ht header using {to} [9] 2013/02/28 15:34:26: Generating hpai header using {trunk} [8] 2013/02/28 15:34:26: Play audio_moh/noise.wav, caching true [8] 2013/02/28 15:34:26: Allocating for call port 123, SIP call id 10fe8e01@pbx [9] 2013/02/28 15:34:26: UDP(IPv4): Opening socket on 0.0.0.0:55444 [9] 2013/02/28 15:34:26: UDP(IPv4): Opening socket on 0.0.0.0:55445 [7] 2013/02/28 15:34:26: Call port 123: set_codecs for 10fe8e01@pbx codecs "", codec_preference count 7 [8] 2013/02/28 15:34:26: call port 123: state code from 0 to 100 [9] 2013/02/28 15:34:26: Call port 123: update_codecs for 10fe8e01@pbx: adding codec pcmu/8000 to available list [9] 2013/02/28 15:34:26: Call port 123: update_codecs for 10fe8e01@pbx: adding codec pcma/8000 to available list [9] 2013/02/28 15:34:26: Call port 123: update_codecs for 10fe8e01@pbx: adding codec g722/8000 to available list [9] 2013/02/28 15:34:26: Call port 123: update_codecs for 10fe8e01@pbx: adding codec g729/8000 to available list [9] 2013/02/28 15:34:26: Call port 123: update_codecs for 10fe8e01@pbx: adding codec g726-32/8000 to available list [9] 2013/02/28 15:34:26: Call port 123: update_codecs for 10fe8e01@pbx: adding codec gsm/8000 to available list [9] 2013/02/28 15:34:26: Call port 123: update_codecs for 10fe8e01@pbx: codec_preference size 7, available codecs size 7 [9] 2013/02/28 15:34:26: Resolve 1686: url sip:192.168.137.254:5066 [9] 2013/02/28 15:34:26: Resolve 1686: udp 192.168.137.254 5066 [5] 2013/02/28 15:34:26: SIP Tx udp:192.168.137.254:5066: INVITE sip:7261247@please.change;user=phone SIP/2.0 Via: SIP/2.0/UDP 216.150.32.145:5060;branch=z9hG4bK-0e14ba8c5e81c99046bac1b75c04418a;rport From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=646572893 To: <sip:7261247@pbx.company.com;user=phone> Call-ID: 10fe8e01@pbx CSeq: 98 INVITE Max-Forwards: 70 Contact: <sip:anonymous@216.150.32.145:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.0.1090 Epsilon Geminids P-Asserted-Identity: <sip:please.change> Privacy: id Content-Type: application/sdp Content-Length: 386 v=0 o=- 672703835 672703835 IN IP4 216.150.32.145 s=- c=IN IP4 216.150.32.145 t=0 0 m=audio 55444 RTP/AVP 0 8 9 18 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [5] 2013/02/28 15:34:26: SIP Rx udp:192.168.137.254:5066: SIP/2.0 100 Trying Via: SIP/2.0/UDP 216.150.32.145:5060;branch=z9hG4bK-0e14ba8c5e81c99046bac1b75c04418a;rport=5060;received=192.168.137.254 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=646572893 To: <sip:7261247@pbx.company.com;user=phone>;tag=ds-35dee0d-9e9b8a88 Call-ID: 10fe8e01@pbx CSeq: 98 INVITE Content-Length: 0 Server: Netborder Express Gateway/4.3.3 Contact: <sip:NetborderExpressGateway@192.168.137.254:5066;transport=udp> [9] 2013/02/28 15:34:26: Message repetition, packet dropped [8] 2013/02/28 15:34:26: call port 122: state code from 0 to 183 [7] 2013/02/28 15:34:26: Set packet length to 20 [9] 2013/02/28 15:34:26: Call port 122: update_codecs for 3c269177ceb5-a1za6z3kewa1: adding codec pcmu/8000 to available list [9] 2013/02/28 15:34:26: Call port 122: update_codecs for 3c269177ceb5-a1za6z3kewa1: adding codec pcma/8000 to available list [9] 2013/02/28 15:34:26: Call port 122: update_codecs for 3c269177ceb5-a1za6z3kewa1: adding codec g722/8000 to available list [9] 2013/02/28 15:34:26: Call port 122: update_codecs for 3c269177ceb5-a1za6z3kewa1: adding codec g729/8000 to available list [9] 2013/02/28 15:34:26: Call port 122: update_codecs for 3c269177ceb5-a1za6z3kewa1: adding codec g726-32/8000 to available list [9] 2013/02/28 15:34:26: Call port 122: update_codecs for 3c269177ceb5-a1za6z3kewa1: adding codec gsm/8000 to available list [9] 2013/02/28 15:34:26: Call port 122: update_codecs for 3c269177ceb5-a1za6z3kewa1: codec_preference size 7, available codecs size 7 [6] 2013/02/28 15:34:26: Call-leg 122: Codec pcmu/8000 is chosen for call id 3c269177ceb5-a1za6z3kewa1 [5] 2013/02/28 15:34:26: set codec: codec pcmu/8000 is set to call-leg 122 [9] 2013/02/28 15:34:26: Resolve 1687: aaaa udp 192.168.137.154 2048 [9] 2013/02/28 15:34:26: Resolve 1687: a udp 192.168.137.154 2048 [9] 2013/02/28 15:34:26: Resolve 1687: udp 192.168.137.154 2048 [5] 2013/02/28 15:34:26: SIP Tx udp:192.168.137.154:2048: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.137.154:2048;branch=z9hG4bK-crzvtm8g5izl;rport=2048 From: <sip:208@192.168.137.254>;tag=66vco17bd1 To: <sip:7261247@192.168.137.254;user=phone>;tag=c38522c82f Call-ID: 3c269177ceb5-a1za6z3kewa1 CSeq: 2 INVITE Contact: <sip:208@216.150.32.145:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.0.1090 Epsilon Geminids Require: 100rel RSeq: 1 Content-Type: application/sdp Content-Length: 400 v=0 o=- 428979860 428979860 IN IP4 216.150.32.145 s=- c=IN IP4 216.150.32.145 t=0 0 m=audio 53328 RTP/AVP 0 8 9 18 99 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:99 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [5] 2013/02/28 15:34:26: SIP Rx udp:216.150.32.145:44407: PRACK sip:208@216.150.32.145:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.137.154:2048;branch=z9hG4bK-uwp1uuf0hy3v;rport From: <sip:208@192.168.137.254>;tag=66vco17bd1 To: <sip:7261247@192.168.137.254;user=phone>;tag=c38522c82f Call-ID: 3c269177ceb5-a1za6z3kewa1 CSeq: 3 PRACK Max-Forwards: 70 Contact: <sip:208@192.168.137.154:2048;line=x83vrg9s>;reg-id=1 RAck: 1 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Content-Length: 0 [9] 2013/02/28 15:34:26: Resolve 1688: udp 216.150.32.145 44407 [5] 2013/02/28 15:34:26: SIP Tx udp:216.150.32.145:44407: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.137.154:2048;branch=z9hG4bK-uwp1uuf0hy3v;rport=44407;received=216.150.32.145 From: <sip:208@192.168.137.254>;tag=66vco17bd1 To: <sip:7261247@192.168.137.254;user=phone>;tag=c38522c82f Call-ID: 3c269177ceb5-a1za6z3kewa1 CSeq: 3 PRACK Contact: <sip:208@216.150.32.145:5060> User-Agent: snomONE/4.5.0.1090 Epsilon Geminids Content-Length: 0 [6] 2013/02/28 15:34:26: Call-leg 122: Sending RTP for 3c269177ceb5-a1za6z3kewa1 to 216.150.32.145:30126, codec pcmu/8000 [5] 2013/02/28 15:34:26: SIP Rx udp:192.168.137.254:5066: SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 216.150.32.145:5060;branch=z9hG4bK-0e14ba8c5e81c99046bac1b75c04418a;rport=5060;received=192.168.137.254 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=646572893 To: <sip:7261247@pbx.company.com;user=phone>;tag=ds-35dee0d-9e9b8a88 Call-ID: 10fe8e01@pbx CSeq: 98 INVITE Content-Length: 0 Server: Netborder Express Gateway/4.3.3 CPD-Result: Busy Contact: <sip:192.168.137.254:5066;transport=udp> [7] 2013/02/28 15:34:26: Call 10fe8e01@pbx: Clear last INVITE [9] 2013/02/28 15:34:26: Resolve 1689: url sip:192.168.137.254:5066 [9] 2013/02/28 15:34:26: Resolve 1689: udp 192.168.137.254 5066 [5] 2013/02/28 15:34:26: SIP Tx udp:192.168.137.254:5066: ACK sip:7261247@please.change;user=phone SIP/2.0 Via: SIP/2.0/UDP 216.150.32.145:5060;branch=z9hG4bK-0e14ba8c5e81c99046bac1b75c04418a;rport From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=646572893 To: <sip:7261247@pbx.company.com;user=phone>;tag=ds-35dee0d-9e9b8a88 Call-ID: 10fe8e01@pbx CSeq: 98 ACK Max-Forwards: 70 Contact: <sip:anonymous@216.150.32.145:5060;transport=udp> P-Asserted-Identity: <sip:please.change> Privacy: id Content-Length: 0 [5] 2013/02/28 15:34:26: INVITE Response 486 Busy Here: Terminate 10fe8e01@pbx [8] 2013/02/28 15:34:26: Clearing call port 123, SIP call id 10fe8e01@pbx [8] 2013/02/28 15:34:26: call port 122: state code from 183 to 486 [9] 2013/02/28 15:34:26: Resolve 1690: aaaa udp 192.168.137.154 2048 [9] 2013/02/28 15:34:26: Resolve 1690: a udp 192.168.137.154 2048 [9] 2013/02/28 15:34:26: Resolve 1690: udp 192.168.137.154 2048 [5] 2013/02/28 15:34:26: SIP Tx udp:192.168.137.154:2048: SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 192.168.137.154:2048;branch=z9hG4bK-crzvtm8g5izl;rport=2048 From: <sip:208@192.168.137.254>;tag=66vco17bd1 To: <sip:7261247@192.168.137.254;user=phone>;tag=c38522c82f Call-ID: 3c269177ceb5-a1za6z3kewa1 CSeq: 2 INVITE Contact: <sip:208@216.150.32.145:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.0.1090 Epsilon Geminids Content-Length: 0 [8] 2013/02/28 15:34:26: Remove leg 124: call port 123, SIP call id 10fe8e01@pbx [8] 2013/02/28 15:34:26: Hangup: Call 123 not found [5] 2013/02/28 15:34:26: SIP Rx udp:192.168.137.154:2048: ACK sip:7261247@192.168.137.254;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.137.154:2048;branch=z9hG4bK-crzvtm8g5izl;rport From: <sip:208@192.168.137.254>;tag=66vco17bd1 To: <sip:7261247@192.168.137.254;user=phone>;tag=c38522c82f Call-ID: 3c269177ceb5-a1za6z3kewa1 CSeq: 2 ACK Max-Forwards: 70 Contact: <sip:208@192.168.137.154:2048;line=x83vrg9s>;reg-id=1 Content-Length: 0 [8] 2013/02/28 15:34:26: Clearing call port 122, SIP call id 3c269177ceb5-a1za6z3kewa1 [8] 2013/02/28 15:34:26: Remove leg 123: call port 122, SIP call id 3c269177ceb5-a1za6z3kewa1
×
×
  • Create New...