Andrea Deltacom
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Hi everyone, my ISP changed his platform and I can't anymore call with snom one PBX. My configuration IS: # Trunk 20 in domain localhost Name: Deltacom_Hidden_1 Type: register To: sip RegPass: ******** Direction: Disabled: false Global: false Display: 509091480XXX RegAccount: 509091480XXX RegRegistrar: sip.deltacomsrl.it RegKeep: RegUser: 509091480XXX Icid: Require: OutboundProxy: sip.deltacomsrl.it Ani: 509091480XXX DialExtension: 15 Prefix: Trusted: false AcceptRedirect: false RfcRtp: false Analog: false SendEmail: UseUuid: false Ring180: false Failover: never Privacy: pai Glob: RequestTimeout: Codecs: CodecLock: true Expires: 3600 FromUser: Tel: true TranscodeDtmf: false AssociatedAddresses: InterOffice: false DialPlan: Colines: DialogPermission: And then log SIP [5] 2011/07/20 12:11:47: SIP Rx udp:192.168.14.23:32256: INVITE sip:090774XXX@192.168.14.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.14.23:32256;branch=z9hG4bK-d8754z-016650393159f53f-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:13@192.168.14.23:32256> To: "090774XXX"<sip:090774XXX@192.168.14.254> From: "13"<sip:13@192.168.14.254>;tag=ce18420b Call-ID: ZDc0NTQzMmFhN2Y2ZjFiYmY3MDI0NDdiMjI1YmVkNmM. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1103k stamp 53621 Content-Length: 368 v=0 o=- 9 2 IN IP4 192.168.14.23 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.14.23 t=0 0 m=audio 52344 RTP/AVP 107 0 8 101 a=alt:1 3 : rxGTiS92 oGbOd6pG 192.168.76.1 52344 a=alt:2 2 : A35uDg1B 9Y0EP7mf 192.168.209.1 52344 a=alt:3 1 : KExkKgb7 WmV4o3jd 192.168.14.23 52344 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv [5] 2011/07/20 12:11:47: Last message repeated 2 times [5] 2011/07/20 12:11:47: SIP Tx udp:192.168.14.23:32256: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.14.23:32256;branch=z9hG4bK-d8754z-016650393159f53f-1---d8754z-;rport=32256 From: "13" <sip:13@192.168.14.254>;tag=ce18420b To: "090774XXX" <sip:090774XXX@192.168.14.254>;tag=23f4d20a83 Call-ID: ZDc0NTQzMmFhN2Y2ZjFiYmY3MDI0NDdiMjI1YmVkNmM. CSeq: 1 INVITE Content-Length: 0 [5] 2011/07/20 12:11:47: SIP Tx udp:192.168.14.23:32256: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 192.168.14.23:32256;branch=z9hG4bK-d8754z-016650393159f53f-1---d8754z-;rport=32256 From: "13" <sip:13@192.168.14.254>;tag=ce18420b To: "090774XXX" <sip:090774XXX@192.168.14.254>;tag=23f4d20a83 Call-ID: ZDc0NTQzMmFhN2Y2ZjFiYmY3MDI0NDdiMjI1YmVkNmM. CSeq: 1 INVITE User-Agent: snomOne-PBX/2011-4.2.0.3981 WWW-Authenticate: Digest realm="192.168.14.254",nonce="595c3a4ced41b174ea57f9ef78dc6b1b",domain="sip:090774581@192.168.14.254",algorithm=MD5 Content-Length: 0 [5] 2011/07/20 12:11:47: SIP Rx udp:192.168.14.23:32256: ACK sip:090774581@192.168.14.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.14.23:32256;branch=z9hG4bK-d8754z-016650393159f53f-1---d8754z-;rport To: "090774581" <sip:090774581@192.168.14.254>;tag=23f4d20a83 From: "13"<sip:13@192.168.14.254>;tag=ce18420b Call-ID: ZDc0NTQzMmFhN2Y2ZjFiYmY3MDI0NDdiMjI1YmVkNmM. CSeq: 1 ACK Content-Length: 0 [5] 2011/07/20 12:11:47: SIP Rx udp:192.168.14.23:32256: INVITE sip:090774581@192.168.14.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.14.23:32256;branch=z9hG4bK-d8754z-fe7b77615b04f54e-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:13@192.168.14.23:32256> To: "090774581"<sip:090774581@192.168.14.254> From: "13"<sip:13@192.168.14.254>;tag=ce18420b Call-ID: ZDc0NTQzMmFhN2Y2ZjFiYmY3MDI0NDdiMjI1YmVkNmM. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1103k stamp 53621 Authorization: Digest username="13",realm="192.168.14.254",nonce="595c3a4ced41b174ea57f9ef78dc6b1b",uri="sip:090774581@192.168.14.254",response="2b76fcecc75621c0f28c02fb0d59e682",algorithm=MD5 Content-Length: 368 v=0 o=- 9 2 IN IP4 192.168.14.23 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.14.23 t=0 0 m=audio 52344 RTP/AVP 107 0 8 101 a=alt:1 3 : rxGTiS92 oGbOd6pG 192.168.76.1 52344 a=alt:2 2 : A35uDg1B 9Y0EP7mf 192.168.209.1 52344 a=alt:3 1 : KExkKgb7 WmV4o3jd 192.168.14.23 52344 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv [6] 2011/07/20 12:11:47: Sending RTP for ZDc0NTQzMmFhN2Y2ZjFiYmY3MDI0NDdiMjI1YmVkNmM. to 192.168.14.23:52344, codec not set yet [5] 2011/07/20 12:11:47: SIP Tx udp:192.168.14.23:32256: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.14.23:32256;branch=z9hG4bK-d8754z-fe7b77615b04f54e-1---d8754z-;rport=32256 From: "13" <sip:13@192.168.14.254>;tag=ce18420b To: "090774XXX" <sip:090774XXX@192.168.14.254>;tag=23f4d20a83 Call-ID: ZDc0NTQzMmFhN2Y2ZjFiYmY3MDI0NDdiMjI1YmVkNmM. CSeq: 2 INVITE Content-Length: 0 [7] 2011/07/20 12:11:47: set_codecs: for ZDc0NTQzMmFhN2Y2ZjFiYmY3MDI0NDdiMjI1YmVkNmM. codecs "", codec_preference count 7 [5] 2011/07/20 12:11:47: Dialplan "Delta Test": Match 090774XXX@192.168.14.254 to <sip:090774XXX@sip.deltacomsrl.it;user=phone> on trunk Deltacom_Hidden_1 [7] 2011/07/20 12:11:47: Cannot convert number 50909148XXXX into global format [7] 2011/07/20 12:11:47: Last message repeated 2 times [7] 2011/07/20 12:11:47: set_codecs: for 47771b31@pbx codecs "", codec_preference count 7 [6] 2011/07/20 12:11:47: Codec pcma/8000 is chosen for call id ZDc0NTQzMmFhN2Y2ZjFiYmY3MDI0NDdiMjI1YmVkNmM. [5] 2011/07/20 12:11:47: SIP Tx udp:192.168.14.23:32256: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.14.23:32256;branch=z9hG4bK-d8754z-fe7b77615b04f54e-1---d8754z-;rport=32256 From: "13" <sip:13@192.168.14.254>;tag=ce18420b To: "090774XXX" <sip:090774XXX@192.168.14.254>;tag=23f4d20a83 Call-ID: ZDc0NTQzMmFhN2Y2ZjFiYmY3MDI0NDdiMjI1YmVkNmM. CSeq: 2 INVITE Contact: <sip:13@192.168.14.253:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomOne-PBX/2011-4.2.0.3981 Content-Type: application/sdp Content-Length: 265 v=0 o=- 726816646 726816646 IN IP4 192.168.14.253 s=- c=IN IP4 192.168.14.253 t=0 0 m=audio 51834 RTP/AVP 8 0 101 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [5] 2011/07/20 12:11:47: SIP Tx tcp:77.239.128.7:5060: INVITE sip:090774XXX@sip.deltacomsrl.it;user=phone SIP/2.0 Via: SIP/2.0/TCP 192.168.14.253:35758;branch=z9hG4bK-1966c3fd6b5bdaa31520421ef93c3597;rport From: "Andrea Ciccio'" <sip:509091480041@localhost;user=phone>;tag=1528124281 To: <sip:090774XXX@sip.deltacomsrl.it;user=phone> Call-ID: 47771b31@pbx CSeq: 11223 INVITE Max-Forwards: 70 Contact: <sip:509091480041@192.168.14.253:35758;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomOne-PBX/2011-4.2.0.3981 P-Asserted-Identity: "509091480041" <sip:509091480041@sip.deltacomsrl.it> Content-Type: application/sdp Content-Length: 388 v=0 o=- 1142207063 1142207063 IN IP4 192.168.14.253 s=- c=IN IP4 192.168.14.253 t=0 0 m=audio 57912 RTP/AVP 8 18 0 2 3 9 101 a=rtpmap:8 pcma/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:0 pcmu/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:9 g722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [5] 2011/07/20 12:11:47: SIP Rx tcp:77.239.128.7:5060: SIP/2.0 100 Trying Via: SIP/2.0/TCP 192.168.14.253:35758;received=178.236.173.178;branch=z9hG4bK-1966c3fd6b5bdaa31520421ef93c3597;rport=51950 From: "Andrea Ciccio'" <sip:509091480041@localhost;user=phone>;tag=1528124281 To: <sip:090774XXX@sip.deltacomsrl.it;user=phone> Call-ID: 47771b31@pbx CSeq: 11223 INVITE Content-Length: 0 [5] 2011/07/20 12:11:47: SIP Rx tcp:77.239.128.7:5060: SIP/2.0 403 Unknown User/Endpoint Not Allowed Via: SIP/2.0/TCP 192.168.14.253:35758;received=178.236.173.178;branch=z9hG4bK-1966c3fd6b5bdaa31520421ef93c3597;rport=51950 From: "Andrea Ciccio'" <sip:509091480041@localhost;user=phone>;tag=1528124281 To: <sip:090774XXX@sip.deltacomsrl.it;user=phone>;tag=aprqrjmtc-oulmvc300oaed Call-ID: 47771b31@pbx CSeq: 11223 INVITE Content-Length: 0 [7] 2011/07/20 12:11:47: Call 47771b31@pbx: Clear last INVITE [5] 2011/07/20 12:11:47: SIP Tx tcp:77.239.128.7:5060: ACK sip:090774XXX@sip.deltacomsrl.it;user=phone SIP/2.0 Via: SIP/2.0/TCP 192.168.14.253:35758;branch=z9hG4bK-1966c3fd6b5bdaa31520421ef93c3597;rport From: "Andrea Ciccio'" <sip:509091480041@localhost;user=phone>;tag=1528124281 To: <sip:090774XXX@sip.deltacomsrl.it;user=phone>;tag=aprqrjmtc-oulmvc300oaed Call-ID: 47771b31@pbx CSeq: 11223 ACK Max-Forwards: 70 Contact: <sip:509091480041@192.168.14.253:35758;transport=tcp> P-Asserted-Identity: "509091480041" <sip:509091480041@sip.deltacomsrl.it> Content-Length: 0 [5] 2011/07/20 12:11:47: INVITE Response 403 Unknown User/Endpoint Not Allowed: Terminate 47771b31@pbx [5] 2011/07/20 12:11:47: SIP Tx udp:192.168.14.23:32256: SIP/2.0 403 Unknown User/Endpoint Not Allowed Via: SIP/2.0/UDP 192.168.14.23:32256;branch=z9hG4bK-d8754z-fe7b77615b04f54e-1---d8754z-;rport=32256 From: "13" <sip:13@192.168.14.254>;tag=ce18420b To: "090774XXX" <sip:090774XXX@192.168.14.254>;tag=23f4d20a83 Call-ID: ZDc0NTQzMmFhN2Y2ZjFiYmY3MDI0NDdiMjI1YmVkNmM. CSeq: 2 INVITE Contact: <sip:13@192.168.14.253:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomOne-PBX/2011-4.2.0.3981 Content-Length: 0 [5] 2011/07/20 12:11:47: SIP Rx udp:192.168.14.23:32256: ACK sip:090774XXX@192.168.14.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.14.23:32256;branch=z9hG4bK-d8754z-fe7b77615b04f54e-1---d8754z-;rport To: "090774XXX" <sip:090774XXX@192.168.14.254>;tag=23f4d20a83 From: "13"<sip:13@192.168.14.254>;tag=ce18420b Call-ID: ZDc0NTQzMmFhN2Y2ZjFiYmY3MDI0NDdiMjI1YmVkNmM. CSeq: 2 ACK Content-Length: 0 what can be? Any idea?
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Thanks for Reply, I'm not sure how to make dial plan in Asterisk I tried: exten => XXX,1,Dial(Zap/1) [dialout] exten => XXXXXXXXX,1,Dial(Zap/1,100,r) but isn't working
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I'm trying to communicate a snom one installation with Asterisk. I have in the same machine, snom one with sip 5060, and asterisk with sip 5090. I configured in snom one a Sip Gateway with ip address 127.0.0.1:5090 and this seems ok. In Asterisk I configured an ISDN TE BRI pci card, there is 2 problem: 1. the incoming call, ring in Asterisk but don't go in Snom One extensions 2. the outgoing call, go in Asterisk but don't use idsn trunk Can you help me?
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yup i know, if i can't find another driver maybe i do an asterisk trunk for isdn
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Hi, I'm trying to configure an OpenVOX BRI B100 ISDN PCI Card. I should installed it with zaptel drivers, but i'm not sure how configure to work with snom one. Specially what data i need to put in Trunk Configuration. Thanks Andrea
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we need 1 Port BRI, internal or external. The system is Centos 5 so if internal we need the drivers
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Hi, I need to use a ISDN trunk with Snom ONE. What device can I install? Maybe an ISDN Moden PCI? I need a cheap devices. Thanks Andrea
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yes, it's ok. ty
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this file isn't working: http://downloads.snom.net/snomONE/centos64/snomone-CentOS5-x64.bin # snom m9 firmware update. # Registrations to third-party devices (e.g., SIP phones, SIP cameras, softphones, etc.) have been increased: snom ONE free supports 5 third-party devices, snom ONE yellow supports 10, and blue supports 40. # New addition of snom ONE is available: snom ONE green (includes call support for third-party devices). this seem realy great!!
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ok tnx, i disabled this.
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My router is draytek 2820, it says 15.000 nat table entry. btw before i rebooted it, it was going good, after adsl went up again I can't anymore use my trunks
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Hi, I have 13 trunks (same provider) with 1 ISDN Adapter. Today after a router reboot, they aren't go on anymore with 408 error timeout message. So after many tries, I disabled some of them, and then registering one by one. So this make them work. What could be happened? A SIP provider problem (flood deny?) or a Snom ONE problem? I'm using Version: 4.0.0.3343 (Linux) License: snomONE - 15 Happy New Year Andrea
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Hi, I'm using linksys ATA PAP2T to send and receives fax with snom one, but something isn't going well. Sometime I need to send 2-3 times before I got a positive response, what parameters I can tweak?
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Are you selling CS410 with snom one? and there will be a version with ISDN or GSM ports? Ty Andrea