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Andrea Deltacom

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About Andrea Deltacom

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  1. 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?
  2. there are news about this feature? I need too to disable Automatic Flag with sip phone, or set both in boolean mode thanks
  3. 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
  4. 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?
  5. yup i know, if i can't find another driver maybe i do an asterisk trunk for isdn
  6. 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
  7. we need 1 Port BRI, internal or external. The system is Centos 5 so if internal we need the drivers
  8. 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
  9. 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!!
  10. 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
  11. 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
  12. 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?
  13. Andrea Deltacom

    info

    Are you selling CS410 with snom one? and there will be a version with ISDN or GSM ports? Ty Andrea
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