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AG1

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About AG1

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  • Birthday 05/15/1970

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    Montana
  1. AG1

    Ringtones

    Read this and I dont understand the part about creating a folder. http://wiki.snomone.com/index.php?title=Customized_Ringtones Where do you create this folder on the pbx (Snom One) or on your computer? If Snom One, how and where do you create this? Do I need root access? Figured I would mess around with something else until I can get help for the PA1s that dont work.
  2. AG1

    Snom PA1 Problems

    I started this thread almost 2 weeks ago, and the company I ordered these units from (Telephony Ware) will not answer their phone or assist in getting these replaced. I have numerous phone calls into them as well as emails and NOTHING. LOTS of NOTHING going on around Snom it seems, which is why I am migrating away from their phones and my next move today is to start researching new PBXs. Gotta get something I can get help on since I am not a programmer like most of you guys.
  3. AG1

    Snom PA1 Problems

    None of that makes any sense. If you click on the link you provided it takes me to yet another firmware update that I can place in the update software window, click on load and NOTHING. Then if I click the "recovery update description" it tells me how to do it on a phone with buttons. My Snom PA1 has no buttons so once again another couple hours lost while going in a circle.
  4. AG1

    iPad Question

    Thanks for the input on apps Stefano and Carlos I tried out the Bria app and it worked great, but now i will check out the 3CX one as well. Thanks again for the feedback
  5. AG1

    Snom PA1 Problems

    They looked factory sealed and the MAC addresses were all close to the same. Like I said a couple of them worked flawlessly and the other 8 suck. Still nothing from Snom on emails I sent or posts on here. I still have the versions I can at least talk to with versions 8.4.23 and one other version and those PA1s will not let me upgrade the firmware, you click load and the screen refreshes and nothing. Very frustrating when things are put into the market place and fixed afterwards.
  6. AG1

    iPad Question

    Does anyone know of an app or Snom download that I could install on an iPad to make it an extension of my Snom One PBX? Basically it would be a wireless device I could pair a blue tooth headset to? Thanks
  7. AG1

    Snom PA1 Problems

    I ordered 10 of the Snom PA1 units for 2 separate paging projects and I have spent the last 18 hours wondering why I bought this product. 3 of them are completely unresponsive, I cannot reset them using any of the steps on the snom wiki or other internet sights, they just bricked out of the box. 2 of them came up and worked like they should have, and I easily upgraded the firm ware to the 8.7.3.25 version they work just fine and I never had a problem with them. The other 5 are in various states of disarray. Most of them have different versions of firmware, some get an IP through DHCP some dont, some let me access the web interface some dont. On the couple that I can actually use the web interface they will not let me upgrade the firmware. I have loaded the 8.7.3.25bin file into the firmware upgrade location and clicked on load but nothing happens. I have tried internet explorer, chrome and firefox browsers, tried doing the factory default, tried configuring by just using a PoE switch not connected to my network. I would be surprised if anyone can come up with a scenario that I havent tried and they just will not work. Any suggestions before I mail them back and get new ones? This should have taken about an hour to set up half of these, instead its going on 2 work days and I still only have 2 working units. Is this normal with this product?
  8. AG1

    870 Phones Part 2

    Thanks, I got it working the way I want and now the 870s work as well.
  9. AG1

    870 Phones Part 2

    Yes I have 4 CO lines configured on the phone, I deleted them on one of the phones and now the phone will call out on the Nextiva trunk. Now I have a question about BLF, the reason I had co lines configured was so the operator and 4 other people could see lines ringing into the building and answer them even though they are not in the hunt group. If I use BLF to configure a button to monitor incoming calls would I put "3000" in the parameters box since that is my hunt group? If not how would I make 4 buttons you can monitor and answer on the virtual buttons?
  10. AG1

    870 Phones Part 2

    INVITE sip:914062094291@192.168.100.151;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-utv4lgeoggk3;rport From: "Conference Room" <sip:2203@192.168.100.151>;tag=ngf4lmqvqg To: <sip:914062094291@192.168.100.151;user=phone> Call-ID: 57ca55528438-mg2rh2cjxzgr CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:2203@192.168.100.93:3372;transport=tls;line=27lf1wbz>;reg-id=1 X-Serialnumber: 0004134150E1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom870/8.7.3.19 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 Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Require: buttons-snom870 Content-Type: application/sdp Content-Length: 517 v=0 o=root 974376235 974376235 IN IP4 192.168.100.93 s=call c=IN IP4 192.168.100.93 t=0 0 m=audio 55204 RTP/AVP 9 0 8 99 108 18 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:sX/0lap2QdSW3yfqJoTZSpNIji/HGcd9jq4vyQfE a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:108 AAL2-G726-32/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt a=sendrecv [8] 2013/10/09 15:22:11: Packet authenticated by transport layer [8] 2013/10/09 15:22:11: Allocating for call port 389, SIP call id 57ca55528438-mg2rh2cjxzgr [8] 2013/10/09 15:22:11: Could not find a trunk (3 trunks) [5] 2013/10/09 15:22:11: SIP Tx tls:192.168.100.93:3372: SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-utv4lgeoggk3;rport=3372 From: "Conference Room" <sip:2203@192.168.100.151>;tag=ngf4lmqvqg To: <sip:914062094291@192.168.100.151;user=phone>;tag=14dc02e81d Call-ID: 57ca55528438-mg2rh2cjxzgr CSeq: 1 INVITE Content-Length: 0 [8] 2013/10/09 15:22:11: Call port 389: Added predefined codec 6 (mapped to 9) [8] 2013/10/09 15:22:11: Call port 389: Added predefined codec 2 (mapped to 0) [8] 2013/10/09 15:22:11: Call port 389: Added predefined codec 3 (mapped to 8) [8] 2013/10/09 15:22:11: Call port 389: Added predefined codec 7 (mapped to 18) [8] 2013/10/09 15:22:11: Call port 389: Added rtpmap codec 5 (mapped to 99) [8] 2013/10/09 15:22:11: Call port 389: Added rtpmap codec 10 (mapped to 108) [8] 2013/10/09 15:22:11: Call port 389: Added rtpmap codec 1 (mapped to 101) [7] 2013/10/09 15:22:11: Set packet length to 20 [6] 2013/10/09 15:22:11: Call-leg 389: Sending RTP for 57ca55528438-mg2rh2cjxzgr to 192.168.100.93:55204, codec not set yet [8] 2013/10/09 15:22:11: Incoming call: Request URI sip:914062094291@192.168.100.151;user=phone, To is <sip:914062094291@192.168.100.151;user=phone> [8] 2013/10/09 15:22:11: Call from an user 2203 [8] 2013/10/09 15:22:11: To is <sip:914062094291@192.168.100.151;user=phone>, user 0, domain 2 [8] 2013/10/09 15:22:11: From user 2203 [8] 2013/10/09 15:22:11: Set the To domain based on From user 2203@192.168.100.151 [8] 2013/10/09 15:22:11: Call state for call object 337: idle [7] 2013/10/09 15:22:11: Call port 389: Set codecs to "" preference count 3 [7] 2013/10/09 15:22:11: Skipping pattern match because CO-line is not available for trunk Nextiva [5] 2013/10/09 15:22:11: Dialplan "Standard Plan": Match 914062094291@192.168.100.151 to sip:914062094291@192.168.100.151:5066;user=phone on trunk Netborder Express [5] 2013/10/09 15:22:11: Sending IM from "Stacy Kober" <sip:2006@192.168.100.151> to "Stacy Kober" <sip:2006@192.168.100.151> (1 destinations) [8] 2013/10/09 15:22:11: Allocating for call port 390, SIP call id 50aa13fa@pbx [7] 2013/10/09 15:22:11: Call port 390: Set codecs to "0" preference count 2 [5] 2013/10/09 15:22:11: Sending IM from "Lee Candee" <sip:2012@192.168.100.151> to "Lee Candee" <sip:2012@192.168.100.151> (1 destinations) [8] 2013/10/09 15:22:11: Call port 390: state code from 0 to 100 [5] 2013/10/09 15:22:11: Sending IM from "Conference Room" <sip:2203@192.168.100.151> to "Conference Room" <sip:2203@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:11: Sending IM from "Lynette Hoon" <sip:2018@192.168.100.151> to "Lynette Hoon" <sip:2018@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:11: Sending IM from "Sherri Dardis" <sip:2014@192.168.100.151> to "Sherri Dardis" <sip:2014@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:11: Sending IM from "Brandon Roth" <sip:2021@192.168.100.151> to "Brandon Roth" <sip:2021@192.168.100.151> (1 destinations) [8] 2013/10/09 15:22:11: Play audio_moh/noise.wav, caching true [5] 2013/10/09 15:22:11: SIP Tx udp:192.168.100.151:5066: INVITE sip:914062094291@192.168.100.151:5066;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-735047f4d4c642fb68012d418c450f00;rport From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=2073723721 To: <sip:914062094291@192.168.100.151;user=phone> Call-ID: 50aa13fa@pbx CSeq: 29245 INVITE Max-Forwards: 70 Contact: <sip:anonymous@192.168.100.151:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/5.0.10 P-Asserted-Identity: "Netborder Express" <sip:192.168.100.151:5066> Privacy: id Content-Type: application/sdp Content-Length: 243 v=0 o=- 154740586 154740586 IN IP4 192.168.100.151 s=- c=IN IP4 192.168.100.151 t=0 0 m=audio 60494 RTP/AVP 0 101 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] 2013/10/09 15:22:11: SIP Rx udp:192.168.100.151:5066: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-735047f4d4c642fb68012d418c450f00;rport=5060 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=2073723721 To: <sip:914062094291@192.168.100.151;user=phone>;tag=ds-515242e6-e9eac574 Call-ID: 50aa13fa@pbx CSeq: 29245 INVITE Content-Length: 0 Server: Netborder Express Gateway/4.3.13 Contact: <sip:NetborderExpressGateway@192.168.100.151:5066;transport=udp> [8] 2013/10/09 15:22:11: Call port 389: state code from 0 to 183 [8] 2013/10/09 15:22:11: Call port 389: Added predefined codec 6 (mapped to 9) [8] 2013/10/09 15:22:11: Call port 389: Added predefined codec 2 (mapped to 0) [8] 2013/10/09 15:22:11: Call port 389: Added predefined codec 3 (mapped to 8) [8] 2013/10/09 15:22:11: Call port 389: Added predefined codec 7 (mapped to 18) [8] 2013/10/09 15:22:11: Call port 389: Added rtpmap codec 5 (mapped to 99) [8] 2013/10/09 15:22:11: Call port 389: Added rtpmap codec 10 (mapped to 108) [8] 2013/10/09 15:22:11: Call port 389: Added rtpmap codec 1 (mapped to 101) [7] 2013/10/09 15:22:11: Set packet length to 20 [6] 2013/10/09 15:22:11: Call-leg 389: Codec PCMU/8000 is chosen for call id 57ca55528438-mg2rh2cjxzgr [5] 2013/10/09 15:22:11: set codec: codec PCMU/8000 is set to call-leg 389 [5] 2013/10/09 15:22:11: SIP Tx tls:192.168.100.93:3372: SIP/2.0 183 Session Progress Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-utv4lgeoggk3;rport=3372 From: "Conference Room" <sip:2203@192.168.100.151>;tag=ngf4lmqvqg To: <sip:914062094291@192.168.100.151;user=phone>;tag=14dc02e81d Call-ID: 57ca55528438-mg2rh2cjxzgr CSeq: 1 INVITE Contact: <sip:2203@192.168.100.151:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/5.0.10 Require: 100rel RSeq: 1 Content-Type: application/sdp Content-Length: 386 v=0 o=- 639170658 639170658 IN IP4 192.168.100.151 s=- c=IN IP4 192.168.100.151 t=0 0 m=audio 49472 RTP/AVP 0 18 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:+mUYvsDtufVA2IS4U3Q3DZB12+ZzLjFbdKTFyIiv a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no 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] 2013/10/09 15:22:11: SIP Rx tls:192.168.100.93:3372: PRACK sip:2203@192.168.100.151:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-az5g906p0x93;rport From: "Conference Room" <sip:2203@192.168.100.151>;tag=ngf4lmqvqg To: <sip:914062094291@192.168.100.151;user=phone>;tag=14dc02e81d Call-ID: 57ca55528438-mg2rh2cjxzgr CSeq: 2 PRACK Max-Forwards: 70 Contact: <sip:2203@192.168.100.93:3372;transport=tls;line=27lf1wbz>;reg-id=1 RAck: 1 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Proxy-Require: buttons-snom870 Content-Length: 0 [8] 2013/10/09 15:22:11: Packet authenticated by transport layer [5] 2013/10/09 15:22:11: SIP Tx tls:192.168.100.93:3372: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-az5g906p0x93;rport=3372 From: "Conference Room" <sip:2203@192.168.100.151>;tag=ngf4lmqvqg To: <sip:914062094291@192.168.100.151;user=phone>;tag=14dc02e81d Call-ID: 57ca55528438-mg2rh2cjxzgr CSeq: 2 PRACK Contact: <sip:2203@192.168.100.151:5061;transport=tls> User-Agent: snomONE/5.0.10 Content-Length: 0 [8] 2013/10/09 15:22:11: Packet authenticated by transport layer [8] 2013/10/09 15:22:12: Last message repeated 3 times [8] 2013/10/09 15:22:12: SMTP: Connect to 74.125.140.108:25 [8] 2013/10/09 15:22:13: SMTP: Received 220 mx.google.com ESMTP v22sm64056817yhn.12 - gsmtp [8] 2013/10/09 15:22:13: SMTP: Send EHLO localhost [8] 2013/10/09 15:22:13: Packet authenticated by transport layer [8] 2013/10/09 15:22:13: SMTP: Received 250-mx.google.com at your service, [216.228.51.194] 250-SIZE 35882577 250-8BITMIME 250-STARTTLS 250-ENHANCEDSTATUSCODES 250 CHUNKING [8] 2013/10/09 15:22:13: SMTP: Send STARTTLS [8] 2013/10/09 15:22:13: SMTP: Received 220 2.0.0 Ready to start TLS [8] 2013/10/09 15:22:13: SMTP: Send EHLO localhost [4] 2013/10/09 15:22:13: Certificate for Equifax Secure Certificate Authority not available [5] 2013/10/09 15:22:13: Certificate for smtp.gmail.com could not be verified against [8] 2013/10/09 15:22:13: Play audio_en/aa_no_answer.wav space10 audio_en/aa_receive_callback.wav audio_en/aa_leave_message.wav audio_en/aa_offer_cellphone.wav space50, caching false [8] 2013/10/09 15:22:13: Call port 379: state code from 200 to 200 [8] 2013/10/09 15:22:13: Trunk 4: Preparing for re-registration [8] 2013/10/09 15:22:13: Trunk Nextiva: Sending registration to 208.73.146.95 [8] 2013/10/09 15:22:13: Trunk 4: setup callback to send re-registration after 38 seconds [5] 2013/10/09 15:22:13: SMTP: Connection to 74.125.140.108:25 failed [8] 2013/10/09 15:22:14: Packet authenticated by transport layer [8] 2013/10/09 15:22:16: Last message repeated 4 times [5] 2013/10/09 15:22:16: SIP Rx udp:192.168.100.151:5066: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-735047f4d4c642fb68012d418c450f00;rport=5060 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=2073723721 To: <sip:914062094291@192.168.100.151;user=phone>;tag=ds-515242e6-e9eac574 Call-ID: 50aa13fa@pbx CSeq: 29245 INVITE Content-Length: 238 Content-Type: application/sdp Server: Netborder Express Gateway/4.3.13 Contact: <sip:NetborderExpressGateway@192.168.100.151:5066;transport=udp> v=0 o=Sangoma-Tech 1381353736 1381353785 IN IP4 192.168.100.151 s=SIP Call c=IN IP4 192.168.100.151 t=0 0 m=audio 14574 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv [8] 2013/10/09 15:22:16: Call port 390: Added predefined codec 2 (mapped to 0) [8] 2013/10/09 15:22:16: Call port 390: Added rtpmap codec 1 (mapped to 101) [7] 2013/10/09 15:22:16: Set packet length to 20 [6] 2013/10/09 15:22:16: Call-leg 390: Codec PCMU/8000 is chosen for call id 50aa13fa@pbx [6] 2013/10/09 15:22:16: Call-leg 390: Sending RTP for 50aa13fa@pbx to 192.168.100.151:14574, codec PCMU/8000 [5] 2013/10/09 15:22:16: set codec: codec PCMU/8000 is set to call-leg 390 [5] 2013/10/09 15:22:16: Sending IM from "Stacy Kober" <sip:2006@192.168.100.151> to "Stacy Kober" <sip:2006@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Lee Candee" <sip:2012@192.168.100.151> to "Lee Candee" <sip:2012@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Conference Room" <sip:2203@192.168.100.151> to "Conference Room" <sip:2203@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Lynette Hoon" <sip:2018@192.168.100.151> to "Lynette Hoon" <sip:2018@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Sherri Dardis" <sip:2014@192.168.100.151> to "Sherri Dardis" <sip:2014@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Brandon Roth" <sip:2021@192.168.100.151> to "Brandon Roth" <sip:2021@192.168.100.151> (1 destinations) [8] 2013/10/09 15:22:16: Call state for call object 337: alerting [8] 2013/10/09 15:22:16: Call port 389: state code from 183 to 183 [8] 2013/10/09 15:22:16: Last message repeated 2 times [7] 2013/10/09 15:22:16: 57ca55528438-mg2rh2cjxzgr: RTP pass-through mode [7] 2013/10/09 15:22:16: 50aa13fa@pbx: RTP pass-through mode [5] 2013/10/09 15:22:16: SIP Rx udp:192.168.100.151:5066: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-735047f4d4c642fb68012d418c450f00;rport=5060 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=2073723721 To: <sip:914062094291@192.168.100.151;user=phone>;tag=ds-515242e6-e9eac574 Call-ID: 50aa13fa@pbx CSeq: 29245 INVITE Content-Length: 238 Content-Type: application/sdp Contact: <sip:NetborderExpressGateway@192.168.100.151:5066;transport=udp> Server: Netborder Express Gateway/4.3.13 v=0 o=Sangoma-Tech 1381353736 1381353786 IN IP4 192.168.100.151 s=SIP Call c=IN IP4 192.168.100.151 t=0 0 m=audio 14574 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv [7] 2013/10/09 15:22:16: Call 50aa13fa@pbx: Clear last INVITE [8] 2013/10/09 15:22:16: Call port 390: Added predefined codec 2 (mapped to 0) [8] 2013/10/09 15:22:16: Call port 390: Added rtpmap codec 1 (mapped to 101) [7] 2013/10/09 15:22:16: Set packet length to 20 [5] 2013/10/09 15:22:16: SIP Tx udp:192.168.100.151:5066: ACK sip:NetborderExpressGateway@192.168.100.151:5066;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-b0002f2d1f9910b246232d9f43d1c0b0;rport From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=2073723721 To: <sip:914062094291@192.168.100.151;user=phone>;tag=ds-515242e6-e9eac574 Call-ID: 50aa13fa@pbx CSeq: 29245 ACK Max-Forwards: 70 Contact: <sip:anonymous@192.168.100.151:5060;transport=udp> P-Asserted-Identity: "Netborder Express" <sip:192.168.100.151:5066> Privacy: id Content-Length: 0 [7] 2013/10/09 15:22:16: Determine pass-through mode after receiving response [8] 2013/10/09 15:22:16: Call state for call object 337: connected [5] 2013/10/09 15:22:16: Sending IM from "Stacy Kober" <sip:2006@192.168.100.151> to "Stacy Kober" <sip:2006@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Lee Candee" <sip:2012@192.168.100.151> to "Lee Candee" <sip:2012@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Conference Room" <sip:2203@192.168.100.151> to "Conference Room" <sip:2203@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Lynette Hoon" <sip:2018@192.168.100.151> to "Lynette Hoon" <sip:2018@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Sherri Dardis" <sip:2014@192.168.100.151> to "Sherri Dardis" <sip:2014@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Brandon Roth" <sip:2021@192.168.100.151> to "Brandon Roth" <sip:2021@192.168.100.151> (1 destinations) [8] 2013/10/09 15:22:16: Call port 390: state code from 100 to 200 [8] 2013/10/09 15:22:16: Call port 389: state code from 183 to 200 [5] 2013/10/09 15:22:16: SIP Tx tls:192.168.100.93:3372: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-utv4lgeoggk3;rport=3372 From: "Conference Room" <sip:2203@192.168.100.151>;tag=ngf4lmqvqg To: <sip:914062094291@192.168.100.151;user=phone>;tag=14dc02e81d Call-ID: 57ca55528438-mg2rh2cjxzgr CSeq: 1 INVITE Contact: <sip:2203@192.168.100.151:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/5.0.10 Content-Type: application/sdp Content-Length: 386 v=0 o=- 639170658 639170658 IN IP4 192.168.100.151 s=- c=IN IP4 192.168.100.151 t=0 0 m=audio 49472 RTP/AVP 0 18 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:+mUYvsDtufVA2IS4U3Q3DZB12+ZzLjFbdKTFyIiv a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no 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] 2013/10/09 15:22:16: SIP Rx tls:192.168.100.93:3372: ACK sip:2203@192.168.100.151:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-3ojncdnhtw1j;rport From: "Conference Room" <sip:2203@192.168.100.151>;tag=ngf4lmqvqg To: <sip:914062094291@192.168.100.151;user=phone>;tag=14dc02e81d Call-ID: 57ca55528438-mg2rh2cjxzgr CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:2203@192.168.100.93:3372;transport=tls;line=27lf1wbz>;reg-id=1 Proxy-Require: buttons-snom870 Content-Length: 0 [8] 2013/10/09 15:22:16: Packet authenticated by transport layer [8] 2013/10/09 15:22:18: Last message repeated 6 times [8] 2013/10/09 15:22:18: SMTP: Connect to 74.125.140.108:25 [8] 2013/10/09 15:22:19: SMTP: Received 421 4.4.5 Server busy, try again later. (mx.google.com) s21sm64052493yhk.9 - gsmtp [5] 2013/10/09 15:22:19: SMTP Server returned 421 [8] 2013/10/09 15:22:19: Packet authenticated by transport layer [8] 2013/10/09 15:22:22: Last message repeated 9 times [8] 2013/10/09 15:22:22: rtp_hangup: call port 387, too early to disconnect [8] 2013/10/09 15:22:22: rtp_hangup: call port 388, too early to disconnect [8] 2013/10/09 15:22:23: Packet authenticated by transport layer [8] 2013/10/09 15:22:26: Last message repeated 6 times [8] 2013/10/09 15:22:26: Play audio_en/aa_no_answer.wav space10 audio_en/aa_receive_callback.wav audio_en/aa_leave_message.wav audio_en/aa_offer_cellphone.wav space50, caching false [8] 2013/10/09 15:22:26: Call port 379: state code from 200 to 200 [8] 2013/10/09 15:22:29: Packet authenticated by transport layer [8] 2013/10/09 15:22:35: Last message repeated 12 times [5] 2013/10/09 15:22:35: Identify trunk (IP address/port and domain match) 3 [8] 2013/10/09 15:22:36: Packet authenticated by transport layer [5] 2013/10/09 15:22:36: SIP Rx tls:192.168.100.93:3372: BYE sip:2203@192.168.100.151:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-u7zgtvycra3f;rport From: "Conference Room" <sip:2203@192.168.100.151>;tag=ngf4lmqvqg To: <sip:914062094291@192.168.100.151;user=phone>;tag=14dc02e81d Call-ID: 57ca55528438-mg2rh2cjxzgr CSeq: 3 BYE Max-Forwards: 70 Contact: <sip:2203@192.168.100.93:3372;transport=tls;line=27lf1wbz>;reg-id=1 User-Agent: snom870/8.7.3.19 RTP-RxStat: Total_Rx_Pkts=1242,Rx_Pkts=1227,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=1229,Tx_Pkts=1229,Remote_Tx_Pkts=6 Proxy-Require: buttons-snom870 Content-Length: 0 [8] 2013/10/09 15:22:36: Packet authenticated by transport layer [5] 2013/10/09 15:22:36: SIP Tx tls:192.168.100.93:3372: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-u7zgtvycra3f;rport=3372 From: "Conference Room" <sip:2203@192.168.100.151>;tag=ngf4lmqvqg To: <sip:914062094291@192.168.100.151;user=phone>;tag=14dc02e81d Call-ID: 57ca55528438-mg2rh2cjxzgr CSeq: 3 BYE Contact: <sip:2203@192.168.100.151:5061;transport=tls> User-Agent: snomONE/5.0.10 Content-Length: 0 [7] 2013/10/09 15:22:36: 50aa13fa@pbx: Media-aware pass-through mode [8] 2013/10/09 15:22:36: Clearing call port 389, SIP call id 57ca55528438-mg2rh2cjxzgr [8] 2013/10/09 15:22:36: Call port 390: state code from 200 to 486 [5] 2013/10/09 15:22:36: SIP Tx udp:192.168.100.151:5066: BYE sip:NetborderExpressGateway@192.168.100.151:5066;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-3f99471547e748b3382e3a011de7b126;rport From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=2073723721 To: <sip:914062094291@192.168.100.151;user=phone>;tag=ds-515242e6-e9eac574 Call-ID: 50aa13fa@pbx CSeq: 29246 BYE Max-Forwards: 70 Contact: <sip:anonymous@192.168.100.151:5060;transport=udp> P-Asserted-Identity: "Netborder Express" <sip:192.168.100.151:5066> Privacy: id Content-Length: 0 [5] 2013/10/09 15:22:36: SIP Rx udp:192.168.100.151:5066: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-3f99471547e748b3382e3a011de7b126;rport=5060 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=2073723721 To: <sip:914062094291@192.168.100.151;user=phone>;tag=ds-515242e6-e9eac574 Call-ID: 50aa13fa@pbx CSeq: 29246 BYE Content-Length: 0
  11. AG1

    870 Phones Part 2

    What do you mean send an invite from the pbx?
  12. AG1

    870 Phones Part 2

    I have a Nextiva SIP trunk, I set my dial plan to dial that trunk (9*) when 9 is pushed then the number This works on the Snom 821, Snom 720 and Grandstream GXP 2200 I can select the Nextiva trunk to dial out. I also have a Sangoma NBE gateway that is the default if a 9 is not dialed. You can dial out on the 821,720, 2200 and 870 phones I have in the building. The 870 phones will not dial out on the Nextiva trunk when you dial 9 then the number. The message that comes back is "you must first dial a 1 when calling this number" 91406209xxxx is what I am dialing I have one standard dial plan set up in my PBX 101- Nextiva - 9* 102 - NBE - 911 103 - NBE - * 104 - NBE - 411 105 - NBE - 811 106 - NBE - ^([2-9][0-9]{6})@.* 107 - NBE - ^([0-9]{10})@.* 108 - NBE - ^(1[0-9]{10})@.* All of the phones are assigned to this dial plan but the 870s will not dial out on my Nextiva trunk Any ideas.......................I am probably just going to sell the 5 - 870 phones and get the GXP2200 instead since this is phone related.
  13. AG1

    Phones wont Register

    Nevermind, when the PBX rebooted it came back up with a number in a field there shouldnt have been a number, I deleted the number now it registers all phones
  14. AG1

    Phones wont Register

    I have rebooted each phone as well as the PBX and each time the PBX reboot only 2 phones register. Every time it is a different 2 phones. Any ideas
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