chrispopp

So these are the steps I took in order to provision this phone: 1. Added the mac address of the phone to the extension I want to provision. 2. Opened MAC based provisioning under that extension. 3. Added the address "http://www.server.com" to Setting URL 4. Made PNP config "OFF" 5. Reboot phone These steps work great for Snom 720 and 760. PBX Version: 5.2.4 Phone Version: snom710 8.7.3.25 Phone Code: 1/1/1970 00:00:48 [NOTICE] PHN: Fast read of /snom/snomlang/web_lang_BS.xml successful. 1/1/1970 00:00:48 [NOTICE] PHN: Fast read of /snom/snomlang/web_lang_DA.xml successful. 1/1/1970 00:00:48 [NOTICE] PHN: Fast read of /snom/snomlang/web_lang_DE.xml successful. 1/1/1970 00:00:48 [NOTICE] PHN: Fast read of /snom/snomlang/web_lang_CS.xml successful. 1/1/1970 00:00:48 [NOTICE] PHN: Fast read of /snom/snomlang/web_lang_EN.xml successful. 1/1/1970 00:00:48 [NOTICE] PHN: Fast read of /snom/snomlang/web_lang_ES.xml successful. 1/1/1970 00:00:48 [NOTICE] PHN: Fast read of /snom/snomlang/web_lang_FI.xml successful. 1/1/1970 00:00:48 [NOTICE] PHN: Fast read of /snom/snomlang/web_lang_FR.xml successful. 1/1/1970 00:00:48 [NOTICE] PHN: Fast read of /snom/snomlang/web_lang_HE.xml successful. 1/1/1970 00:00:48 [NOTICE] PHN: Fast read of /snom/snomlang/web_lang_HU.xml successful. 1/1/1970 00:00:48 [NOTICE] PHN: Fast read of /snom/snomlang/web_lang_IT.xml successful. 1/1/1970 00:00:48 [NOTICE] PHN: Fast read of /snom/snomlang/web_lang_LT.xml successful. 1/1/1970 00:00:48 [NOTICE] PHN: Fast read of /snom/snomlang/web_lang_NL.xml successful. 1/1/1970 00:00:48 [NOTICE] PHN: Fast read of /snom/snomlang/web_lang_NO.xml successful. 1/1/1970 00:00:48 [NOTICE] PHN: Fast read of /snom/snomlang/web_lang_PT.xml successful. 1/1/1970 00:00:48 [NOTICE] PHN: Fast read of /snom/snomlang/web_lang_SI.xml successful. 1/1/1970 00:00:48 [NOTICE] PHN: Fast read of /snom/snomlang/web_lang_RU.xml successful. 1/1/1970 00:00:48 [NOTICE] PHN: Fast read of /snom/snomlang/web_lang_SV.xml successful. 1/1/1970 00:00:48 [NOTICE] PHN: Fast read of /snom/snomlang/web_lang_TR.xml successful. 1/1/1970 00:00:48 [NOTICE] PHN: Using web lang English at index:4 from: /snom/snomlang/web_lang_EN.xml 1/1/1970 00:00:48 [NOTICE] PHN: SIP: Udp listener connected 1/1/1970 00:00:49 [NOTICE] PHN: Setting server prio 1, type redirection, url: >http://www.server.com/prov/snom710-{mac}.htm< 1/1/1970 00:00:49 [NOTICE] PHN: Fetching URL: http://www.server.com:80/prov/snom710-00041374####.htm 1/1/1970 00:00:49 [NOTICE] LID: Opening TCP socket on port 7547 1/1/1970 00:00:49 [NOTICE] LID: Opening TCP socket on port 843 1/1/1970 00:00:49 [FATAL ] PHN: Phone::uboot_version:1.1.5-IFX-05.01.12 1/1/1970 00:00:49 [NOTICE] CFG: read_xml_settings: found setting-files XML header 1/1/1970 00:00:49 [NOTICE] CFG: read_xml_settings: found one byte encoding: 1 1/1/1970 00:00:49 [NOTICE] CFG: read_setting_file_list: added URL: http://www.server.com:80/prov/snom_3xx_fw.xml?model=snom710 1/1/1970 00:00:49 [NOTICE] CFG: read_setting_file_list: added URL: https://www.server.com:443/prov/snom_820_phone.xml?model=snom710 1/1/1970 00:00:49 [NOTICE] CFG: read_setting_file_list: added URL: https://www.server.com:443/prov/snom_3xx_fkeys.xml?model=snom710 1/1/1970 00:00:49 [NOTICE] CFG: read_setting_file_list: added URL: https://www.server.com:443/prov/snom_web_lang.xml?model=snom710 1/1/1970 00:00:49 [NOTICE] CFG: read_setting_file_list: added URL: https://www.server.com:443/prov/snom_gui_lang.xml?model=snom710 1/1/1970 00:00:49 [NOTICE] PHN: Config setup: found xml style settings 1/1/1970 00:00:49 [NOTICE] PHN: last prov successful:1; uri: >http://www.server.com/prov/snom710-{mac}.htm<; default uri: >http://provisioning.snom.com/snom710/snom710.php?mac={mac}< 1/1/1970 00:00:49 [NOTICE] PHN: Fetching URL: http://www.server.com:80/prov/snom_3xx_fw.xml?model=snom710 1/1/1970 00:00:49 [ALERT ] PHN: Config setup: code: 404, uri: http://www.server.com:80/prov/snom_3xx_fw.xml?model=snom710 1/1/1970 00:00:49 [NOTICE] PHN: Fetching URL: https://www.server.com:443/prov/snom_820_phone.xml?model=snom710 1/1/1970 00:00:50 [ERROR ] PHN: Warning: Certificate verification omitted. TLS Server authentication is switched off! 1/1/1970 00:00:50 [ALERT ] PHN: Config setup: code: 404, uri: https://www.server.com:443/prov/snom_820_phone.xml?model=snom710 1/1/1970 00:00:50 [NOTICE] PHN: Fetching URL: https://www.server.com:443/prov/snom_3xx_fkeys.xml?model=snom710 1/1/1970 00:00:50 [ERROR ] PHN: Warning: Certificate verification omitted. TLS Server authentication is switched off! 1/1/1970 00:00:51 [ALERT ] PHN: Config setup: code: 404, uri: https://www.server.com:443/prov/snom_3xx_fkeys.xml?model=snom710 1/1/1970 00:00:51 [NOTICE] PHN: Fetching URL: https://www.server.com:443/prov/snom_web_lang.xml?model=snom710 1/1/1970 00:00:51 [ERROR ] PHN: Warning: Certificate verification omitted. TLS Server authentication is switched off! 1/1/1970 00:00:51 [ALERT ] PHN: Config setup: code: 404, uri: https://www.server.com:443/prov/snom_web_lang.xml?model=snom710 1/1/1970 00:00:51 [NOTICE] PHN: Fetching URL: https://www.server.com:443/prov/snom_gui_lang.xml?model=snom710 1/1/1970 00:00:52 [ERROR ] PHN: Warning: Certificate verification omitted. TLS Server authentication is switched off! 1/1/1970 00:00:52 [ALERT ] PHN: Config setup: code: 404, uri: https://www.server.com:443/prov/snom_gui_lang.xml?model=snom710 1/1/1970 00:00:52 [NOTICE] PHN: Fetching URL: http://127.0.0.1:80/dummy.htm 1/1/1970 00:00:52 [ERROR ] PHN: TPL: Socket Error: 17/23/connecting, connect_cb -> No such file or directory (2) 1/1/1970 00:00:52 [NOTICE] PHN: webclient::on_tcp_close conn_id:7 1/1/1970 00:00:52 [ALERT ] PHN: Config setup: code: 500, uri: http://127.0.0.1:80/dummy.htm 1/1/1970 00:00:52 [NOTICE] PHN: Fetching FW URL: http://www.server.com:80/prov/snom710-firmware.htm 1/1/1970 00:00:52 [NOTICE] PHN: Fetching URL: http://www.server.com:80/prov/snom710-firmware.htm 1/1/1970 00:00:52 [NOTICE] PHN: Last provisioning was successful, thus stop here! 1/1/1970 00:00:52 [NOTICE] PHN: Go to wizzard if all settings have been read. 1/1/1970 00:00:52 [NOTICE] PHN: SetProvisioningDone PBX Code: [8] 2014/10/28 13:16:58: HTTP: Received request for file snom710-00041374####.htm from 171.255.255.252 [8] 2014/10/28 13:16:58: Provisioning file snom710-00041374####.htm looking for MAC 00041374#### [8] 2014/10/28 13:16:58: PnP: Using the credentials of 987@www.server.com for file snom_710.xml [8] 2014/10/28 13:16:58: HTTP: file snom710-00041374####.htm based on template snom_710.xml is sent to 171.255.255.252 [8] 2014/10/28 13:16:59: HTTP: Received request for file snom_3xx_fw.xml from 171.255.255.252 [7] 2014/10/28 13:16:59: HTTP: Error finding snom_3xx_fw.xml, Send back 404 Not Found to 171.255.255.252 [8] 2014/10/28 13:17:01: HTTP: Received request for file snom710-firmware.htm from 171.255.255.252 [7] 2014/10/28 13:17:01: HTTP: Error finding snom710-firmware.htm, Send back 404 Not Found to 171.255.255.252

Is there an import function or some sort of wizard that would help us with the transition? We have way too many clients that need to be exported and doing it manually is not an option.

I have a few domains on version pbxnsip 3.4, and would like to move them to SnomOne version 4.5. My problem is that if I export each domain individually, the button profiles don't transfer properly. They either get jumbled or missing altogether. Secondly, I'm having issues with the trunks. Although they register correctly, the variables for the outbound caller id, completely changes. And lastly, the extensions no longer provision after the move. They need to be removed and re-created in order for them to provision again, and upgrade the firmware on the phones (7.3.30 to 8.4.35.) Any ideas on how I can make the transfer smoother? The reason I'm doing this, to to upgrade my legacy pbxes to 4.5 and then upgrade them all to version 5. I want all of them to be on the same version prior to the latest upgrade. Hence, I would like to do this as efficiently as possible. Thanks.

When's the new release coming out? I rather wait, since this is just a tester.

Another issue that seems to be here with the templates is that the reg_footer does not reflect the changes on the admin page. But the changes are reflected in the dom, and usr, which is fine, maybe something to look at.

Which part needs to change, this is my content of reg_texts: <html> <head> <title># sysadmin</title> <meta http-equiv="Content-Type" content="text/html;charset=utf-8" /> <link href="twocol.css" type="text/css" rel="stylesheet" /> <script src="pbx.js" type="text/javascript"></script> <script src="webtempl.js" type="text/javascript"></script> <script src="reg_format.js" type="text/javascript"></script> </head> <body> <table class="maintable"> <tr id="topbanner"></tr> <tr> <td id="navbar"></td> <td id="sepbar"></td> <td id="contbar"> <h1><span>#title</span> <span>help:AWEtem1</span></h1> <p>#expl</p> <p># cust_templates</p> <table class="toggtbl" id="ct"> </table> <div> <p># select</p> <form name="doc" method="post" onsubmit="return savePage(this.page, this.content, this.save);"> <p> <span>#custtype</span> <select name="ttype" id="ttype" onchange="showTemplates();"> <option value="">#templ</option> <option value="webr">#webr</option> <option value="webd">#webd</option> <option value="webu">#webu</option> <option value="email">#email</option> <option value="phone">#phone</option> </select> <span>#custpage</span> <select name="page" id="page" onchange="selectPage(this);"> <option value=""># page</option> </select> </p> <p id="welcome"> <textarea cols="80" rows="40" name="content" id="content"></textarea> </p> <p><input type="submit" name="templates" id="save" /></p> </form> </div> </td> </tr> </table> <script type="text/javascript"> // <!-- translateFiles["reg_domains.htm"] = 1; translateFiles["client_side_scripts.js"] = 1; setupRegEnvironment(); document.getElementById("save").value = lang("button", "save"); loadChangedTemplates(); // --> </script> </body> </html>

Can you inform me asap. This is quite a severe bug IMO. I may roll back to 5.1 if this doesn't work.

Absolutely not, this was working correctly in version 5.1, decided to upgrade, and now I'm getting this. All extensions are either in alpha or 3 digit numeric.

Using v5.2 (CentOS) and when i got to Administrator templates and select login.htm, I find that it's blank. Whatever I modify in it, does not modify the login page. Any ideas, or is this the wrong file to edit in order to change the main screen.

I’m currently testing with a Vodia pbx version 5.2.0 (CentOS32) I’m having an issue where none of my auto-attendants’ Extension input option “When extension matches” does not work correctly. It provides the message “this extension does not exist” after the second digit is entered, even though all my extensions are 3 digits. Changing it to “3 digits” works correctly. [7] 9:22:03.321 APP: Received call from cell phone 4169459999, but VPA is turned off for extension 205@officetest.testinternal.com [8] 9:22:03.321 APP: Call state for call object 22: connected [8] 9:22:03.323 APP: Play audio_moh/noise.wav, caching true [8] 9:22:04.328 APP: Attendant: Timeout (wait) [8] 9:22:04.328 APP: Play recordings/att680.wav space20, caching false [6] 9:22:08.348 APP: Received DTMF 2, call type attendant [6] 9:22:08.868 APP: Received DTMF 0, call type attendant [8] 9:22:08.868 APP: Play audio_en/aa_not_existing.wav space20, caching false [8] 9:22:08.869 APP: Attendant: Ignoring the DTMF 0 in the state not_existing [6] 9:22:09.428 APP: Received DTMF 5, call type attendant [8] 9:22:09.429 APP: Play audio_en/aa_not_existing.wav space20, caching false [8] 9:22:09.429 APP: Attendant: Ignoring the DTMF 5 in the state not_existing

I tried getting the paging out part of the CS410 unit to work. The internal paging port is set to 2040 under PSTN Gateway Configuration. Created a paging account. Streaming Mode: Unicast destination : 1.1.1.2:2040 Source: * Display Name: blank Record message: live playback Permissions to monitor: blank

I'm also very interested in this matter. We were also hit.

My apologies, it is only phone related. I will go to the snom forum

I am looking to push the configuration XML files directly to the Snom 300, 320, 360 and 370 phones. My xml files work great under the Advanced > Update menu (Upload Setting File manually:), but I would like to push it directly to the phone. Also I want it based on IP addressing not MAC (which I presume is a pull method anyways...). Any help will be greatly appreciated.

Any body else got this to work?

Yes I tried all of them...

Yes, seems to work fine on the PBX side. Have you ever encountered anyone using Asterisk to create a SIP Gateway with a PBXnSIP? If so, would you please post the trunk contexts? I've been trying to get this to work for about a month now...

I added an username and password, both "chris" in this case. I'm getting the same result, this is a SIP Gateway tho, I'm not used to adding credentials to it. I usually create a gateway using the IP address being trusted on both sides. Where: 69.39.221.107 - asterisk pbx 69.39.221.108 - pbxnsip domain.pbxnsip.com - domain name (valid!) 4168885555 - number trying to dial 71.19.171.195 - phone external ip 192.168.1.110 - phone internal ip [9] 2011/01/26 16:29:08: SIP Rx udp:71.19.171.195:54864: INVITE sip:4168885555@domain.pbxnsip.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-2d66df632a075225-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:101@192.168.1.110:54864> To: "4168885555"<sip:4168885555@domain.pbxnsip.com> From: "101"<sip:101@domain.pbxnsip.com>;tag=8b16475d Call-ID: NGY4OTQ2NzkxN2I0MGE4ZTFiYjQ3YjIwNjlmMDU2NDg. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 216 v=0 o=- 5 2 IN IP4 192.168.1.110 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.110 t=0 0 m=audio 31742 RTP/AVP 107 0 8 101 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv [9] 2011/01/26 16:29:08: UDP: Opening socket on 0.0.0.0:57496 [9] 2011/01/26 16:29:08: UDP: Opening socket on 0.0.0.0:57497 [9] 2011/01/26 16:29:08: UDP: Opening socket on [::]:57496 [9] 2011/01/26 16:29:08: UDP: Opening socket on [::]:57497 [8] 2011/01/26 16:29:08: Could not find a trunk (98 trunks) [8] 2011/01/26 16:29:08: Using outbound proxy sip:71.19.171.195:54864;transport=udp because UDP packet source did not match the via header [9] 2011/01/26 16:29:08: Resolve 57819391: udp 71.19.171.195 54864 [9] 2011/01/26 16:29:08: SIP Tx udp:71.19.171.195:54864: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-2d66df632a075225-1---d8754z-;rport=54864;received=71.19.171.195 From: "101" <sip:101@domain.pbxnsip.com>;tag=8b16475d To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=2c0f1a95dc Call-ID: NGY4OTQ2NzkxN2I0MGE4ZTFiYjQ3YjIwNjlmMDU2NDg. CSeq: 1 INVITE Content-Length: 0 [9] 2011/01/26 16:29:08: Resolve 57819392: udp 71.19.171.195 54864 [9] 2011/01/26 16:29:08: SIP Tx udp:71.19.171.195:54864: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-2d66df632a075225-1---d8754z-;rport=54864;received=71.19.171.195 From: "101" <sip:101@domain.pbxnsip.com>;tag=8b16475d To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=2c0f1a95dc Call-ID: NGY4OTQ2NzkxN2I0MGE4ZTFiYjQ3YjIwNjlmMDU2NDg. CSeq: 1 INVITE User-Agent: Shanticom-PBX/3.4.0.3201 WWW-Authenticate: Digest realm="domain.pbxnsip.com",nonce="adde8e1d5f44cfea2b9611dfaf9d361d",domain="sip:4168885555@domain.pbxnsip.com",algorithm=MD5 Content-Length: 0 [9] 2011/01/26 16:29:08: SIP Rx udp:71.19.171.195:54864: ACK sip:4168885555@domain.pbxnsip.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-2d66df632a075225-1---d8754z-;rport To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=2c0f1a95dc From: "101"<sip:101@domain.pbxnsip.com>;tag=8b16475d Call-ID: NGY4OTQ2NzkxN2I0MGE4ZTFiYjQ3YjIwNjlmMDU2NDg. CSeq: 1 ACK Content-Length: 0 [9] 2011/01/26 16:29:08: SIP Rx udp:71.19.171.195:54864: INVITE sip:4168885555@domain.pbxnsip.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-8908ca05bb1ea447-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:101@192.168.1.110:54864> To: "4168885555"<sip:4168885555@domain.pbxnsip.com> From: "101"<sip:101@domain.pbxnsip.com>;tag=8b16475d Call-ID: NGY4OTQ2NzkxN2I0MGE4ZTFiYjQ3YjIwNjlmMDU2NDg. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1104o stamp 56125 Authorization: Digest username="101",realm="domain.pbxnsip.com",nonce="adde8e1d5f44cfea2b9611dfaf9d361d",uri="sip:4168885555@domain.pbxnsip.com",response="e4bed3811e9b62ac982d4c98f257f5b3",algorithm=MD5 Content-Length: 216 v=0 o=- 5 2 IN IP4 192.168.1.110 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.110 t=0 0 m=audio 31742 RTP/AVP 107 0 8 101 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv [8] 2011/01/26 16:29:08: Tagging request with existing tag [6] 2011/01/26 16:29:08: Sending RTP for NGY4OTQ2NzkxN2I0MGE4ZTFiYjQ3YjIwNjlmMDU2NDg.#2c0f1a95dc to 192.168.1.110:31742 [9] 2011/01/26 16:29:08: Resolve 57819393: udp 71.19.171.195 54864 [9] 2011/01/26 16:29:08: SIP Tx udp:71.19.171.195:54864: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-8908ca05bb1ea447-1---d8754z-;rport=54864;received=71.19.171.195 From: "101" <sip:101@domain.pbxnsip.com>;tag=8b16475d To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=2c0f1a95dc Call-ID: NGY4OTQ2NzkxN2I0MGE4ZTFiYjQ3YjIwNjlmMDU2NDg. CSeq: 2 INVITE Content-Length: 0 [8] 2011/01/26 16:29:08: Play audio_moh/noise.wav [9] 2011/01/26 16:29:08: UDP: Opening socket on 0.0.0.0:50634 [9] 2011/01/26 16:29:08: UDP: Opening socket on 0.0.0.0:50635 [9] 2011/01/26 16:29:08: UDP: Opening socket on [::]:50634 [9] 2011/01/26 16:29:08: UDP: Opening socket on [::]:50635 [9] 2011/01/26 16:29:08: Resolve 57819394: url sip:69.39.221.107 [9] 2011/01/26 16:29:08: Resolve 57819394: udp 69.39.221.107 5060 [9] 2011/01/26 16:29:08: SIP Tx udp:69.39.221.107:5060: INVITE sip:14168885555@69.39.221.107;user=phone SIP/2.0 Via: SIP/2.0/UDP 69.39.221.108:5060;branch=z9hG4bK-32227ba0f22c581d832378af1cf7a841;rport From: <sip:chris@69.39.221.107>;tag=555707179 To: <sip:14168885555@69.39.221.107;user=phone> Call-ID: 3fbbdc88@pbx CSeq: 32252 INVITE Max-Forwards: 70 Contact: <sip:chris@69.39.221.108:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: Shanticom-PBX/3.4.0.3201 Remote-Party-ID: "TEST2" <sip:6384858569@69.39.221.107;user=phone>;party=calling;screen=yes Content-Type: application/sdp Content-Length: 249 v=0 o=- 1042925789 1042925789 IN IP4 69.39.221.108 s=- c=IN IP4 69.39.221.108 t=0 0 m=audio 50634 RTP/AVP 0 18 101 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=sendrecv [9] 2011/01/26 16:29:08: SIP Rx udp:69.39.221.107:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 69.39.221.108:5060;branch=z9hG4bK-32227ba0f22c581d832378af1cf7a841;received=69.39.221.108;rport=5060 From: <sip:chris@69.39.221.107>;tag=555707179 To: <sip:14168885555@69.39.221.107;user=phone>;tag=as69468af9 Call-ID: 3fbbdc88@pbx CSeq: 32252 INVITE User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1b52bade" Content-Length: 0 [8] 2011/01/26 16:29:08: Answer challenge with username chris [9] 2011/01/26 16:29:08: Resolve 57819395: udp 69.39.221.107 5060 udp:1 [9] 2011/01/26 16:29:08: SIP Tx udp:69.39.221.107:5060: ACK sip:14168885555@69.39.221.107;user=phone SIP/2.0 Via: SIP/2.0/UDP 69.39.221.108:5060;branch=z9hG4bK-32227ba0f22c581d832378af1cf7a841;rport From: <sip:chris@69.39.221.107>;tag=555707179 To: <sip:14168885555@69.39.221.107;user=phone>;tag=as69468af9 Call-ID: 3fbbdc88@pbx CSeq: 32252 ACK Max-Forwards: 70 Content-Length: 0 [9] 2011/01/26 16:29:08: Resolve 57819396: udp 69.39.221.107 5060 udp:1 [9] 2011/01/26 16:29:08: SIP Tx udp:69.39.221.107:5060: INVITE sip:14168885555@69.39.221.107;user=phone SIP/2.0 Via: SIP/2.0/UDP 69.39.221.108:5060;branch=z9hG4bK-7525da7025674f3209d59439aaa205b0;rport From: <sip:chris@69.39.221.107>;tag=555707179 To: <sip:14168885555@69.39.221.107;user=phone> Call-ID: 3fbbdc88@pbx CSeq: 32253 INVITE Max-Forwards: 70 Contact: <sip:chris@69.39.221.108:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: Shanticom-PBX/3.4.0.3201 Remote-Party-ID: "TEST2" <sip:6384858569@69.39.221.107;user=phone>;party=calling;screen=yes Authorization: Digest realm="asterisk",nonce="1b52bade",response="00af5db02b5683f75aaa6b4b6abaa868",username="chris",uri="sip:14168885555@69.39.221.107;user=phone",algorithm=MD5 Content-Type: application/sdp Content-Length: 249 v=0 o=- 1042925789 1042925789 IN IP4 69.39.221.108 s=- c=IN IP4 69.39.221.108 t=0 0 m=audio 50634 RTP/AVP 0 18 101 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=sendrecv [9] 2011/01/26 16:29:08: Message repetition, packet dropped [6] 2011/01/26 16:29:08: Send codec pcmu/8000 [9] 2011/01/26 16:29:08: Resolve 57819397: udp 71.19.171.195 54864 [9] 2011/01/26 16:29:08: SIP Tx udp:71.19.171.195:54864: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-8908ca05bb1ea447-1---d8754z-;rport=54864;received=71.19.171.195 From: "101" <sip:101@domain.pbxnsip.com>;tag=8b16475d To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=2c0f1a95dc Call-ID: NGY4OTQ2NzkxN2I0MGE4ZTFiYjQ3YjIwNjlmMDU2NDg. CSeq: 2 INVITE Contact: <sip:101@69.39.221.108:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: Shanticom-PBX/3.4.0.3201 Content-Type: application/sdp Content-Length: 224 v=0 o=- 643243467 643243467 IN IP4 69.39.221.108 s=- c=IN IP4 69.39.221.108 t=0 0 m=audio 57496 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [9] 2011/01/26 16:29:08: SIP Rx udp:69.39.221.107:5060: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 69.39.221.108:5060;branch=z9hG4bK-7525da7025674f3209d59439aaa205b0;received=69.39.221.108;rport=5060 From: <sip:chris@69.39.221.107>;tag=555707179 To: <sip:14168885555@69.39.221.107;user=phone>;tag=as69468af9 Call-ID: 3fbbdc88@pbx CSeq: 32253 INVITE User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 [7] 2011/01/26 16:29:08: Call 3fbbdc88@pbx#555707179: Clear last INVITE [9] 2011/01/26 16:29:08: Resolve 57819398: url sip:69.39.221.107 [9] 2011/01/26 16:29:08: Resolve 57819398: udp 69.39.221.107 5060 [9] 2011/01/26 16:29:08: SIP Tx udp:69.39.221.107:5060: ACK sip:14168885555@69.39.221.107;user=phone SIP/2.0 Via: SIP/2.0/UDP 69.39.221.108:5060;branch=z9hG4bK-7525da7025674f3209d59439aaa205b0;rport From: <sip:chris@69.39.221.107>;tag=555707179 To: <sip:14168885555@69.39.221.107;user=phone>;tag=as69468af9 Call-ID: 3fbbdc88@pbx CSeq: 32253 ACK Max-Forwards: 70 Contact: <sip:chris@69.39.221.108:5060;transport=udp> Remote-Party-ID: "TEST2" <sip:6384858569@69.39.221.107;user=phone>;party=calling;screen=yes Content-Length: 0 [5] 2011/01/26 16:29:08: INVITE Response 403 Forbidden: Terminate 3fbbdc88@pbx [7] 2011/01/26 16:29:08: Other Ports: 19 [7] 2011/01/26 16:29:08: Call Port: 0406c8b0115619be07d4a54162592a73@69.39.221.107#c1a15e31c5 [7] 2011/01/26 16:29:08: Call Port: 0c30e7c20f4069af55e343c81824ca9e@69.39.221.107#8da065db10 [7] 2011/01/26 16:29:08: Call Port: 1b3599ef66a7e4567fa425996a543e61@69.39.221.107#08a8fe570c [7] 2011/01/26 16:29:08: Call Port: 306de8f13f896b5f720367a96baa0414@69.39.221.107#7e8c0f8c53 [7] 2011/01/26 16:29:08: Call Port: 385a31350d24d8562e3d82110bf04fd4@69.39.221.107#2e1b3e63a7 [7] 2011/01/26 16:29:08: Call Port: 3c269413038d-boh36yl7bz54#03bcbecf66 [7] 2011/01/26 16:29:08: Call Port: 3c26e15be1ad-zczh8x7jb536#e59c99473a [7] 2011/01/26 16:29:08: Call Port: 3c2927e27a18-hm45ek6vgh3z#83f759db21 [7] 2011/01/26 16:29:08: Call Port: 3c32b54b9ba2-rdxxbojm84pm#12cb5af3d7 [7] 2011/01/26 16:29:08: Call Port: 4de90dff1531fcac2a40a4743cb85d7d@69.39.221.107#2cbb247a6d [7] 2011/01/26 16:29:08: Call Port: 72579d5c6d76566371675a317073ef73@69.39.221.107#60863285c4 [7] 2011/01/26 16:29:08: Call Port: 74a86d7b@pbx#2030213476 [7] 2011/01/26 16:29:08: Call Port: 97656269@pbx#1234486799 [7] 2011/01/26 16:29:08: Call Port: NGY4OTQ2NzkxN2I0MGE4ZTFiYjQ3YjIwNjlmMDU2NDg.#2c0f1a95dc [7] 2011/01/26 16:29:08: Call Port: a294e6ec@pbx#551257555 [7] 2011/01/26 16:29:08: Call Port: ad6bb336@pbx#1558453222 [7] 2011/01/26 16:29:08: Call Port: b274dc3f@pbx#1490165311 [7] 2011/01/26 16:29:08: Call Port: bb903a2c9d68a855#8614b5139f [7] 2011/01/26 16:29:08: Call Port: f02665db@pbx#907681560 [9] 2011/01/26 16:29:08: Resolve 57819399: udp 71.19.171.195 54864 [9] 2011/01/26 16:29:08: SIP Tx udp:71.19.171.195:54864: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-8908ca05bb1ea447-1---d8754z-;rport=54864;received=71.19.171.195 From: "101" <sip:101@domain.pbxnsip.com>;tag=8b16475d To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=2c0f1a95dc Call-ID: NGY4OTQ2NzkxN2I0MGE4ZTFiYjQ3YjIwNjlmMDU2NDg. CSeq: 2 INVITE Contact: <sip:101@69.39.221.108:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: Shanticom-PBX/3.4.0.3201 Content-Length: 0 [9] 2011/01/26 16:29:09: SIP Rx udp:71.19.171.195:54864: ACK sip:4168885555@domain.pbxnsip.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-8908ca05bb1ea447-1---d8754z-;rport To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=2c0f1a95dc From: "101"<sip:101@domain.pbxnsip.com>;tag=8b16475d Call-ID: NGY4OTQ2NzkxN2I0MGE4ZTFiYjQ3YjIwNjlmMDU2NDg. CSeq: 2 ACK Content-Length: 0 [7] 2011/01/26 16:29:09: Other Ports: 18 [7] 2011/01/26 16:29:09: Call Port: 0406c8b0115619be07d4a54162592a73@69.39.221.107#c1a15e31c5 [7] 2011/01/26 16:29:09: Call Port: 0c30e7c20f4069af55e343c81824ca9e@69.39.221.107#8da065db10 [7] 2011/01/26 16:29:09: Call Port: 1b3599ef66a7e4567fa425996a543e61@69.39.221.107#08a8fe570c [7] 2011/01/26 16:29:09: Call Port: 306de8f13f896b5f720367a96baa0414@69.39.221.107#7e8c0f8c53 [7] 2011/01/26 16:29:09: Call Port: 385a31350d24d8562e3d82110bf04fd4@69.39.221.107#2e1b3e63a7 [7] 2011/01/26 16:29:09: Call Port: 3c269413038d-boh36yl7bz54#03bcbecf66 [7] 2011/01/26 16:29:09: Call Port: 3c26e15be1ad-zczh8x7jb536#e59c99473a [7] 2011/01/26 16:29:09: Call Port: 3c2927e27a18-hm45ek6vgh3z#83f759db21 [7] 2011/01/26 16:29:09: Call Port: 3c32b54b9ba2-rdxxbojm84pm#12cb5af3d7 [7] 2011/01/26 16:29:09: Call Port: 4de90dff1531fcac2a40a4743cb85d7d@69.39.221.107#2cbb247a6d [7] 2011/01/26 16:29:09: Call Port: 72579d5c6d76566371675a317073ef73@69.39.221.107#60863285c4 [7] 2011/01/26 16:29:09: Call Port: 74a86d7b@pbx#2030213476 [7] 2011/01/26 16:29:09: Call Port: 97656269@pbx#1234486799 [7] 2011/01/26 16:29:09: Call Port: a294e6ec@pbx#551257555 [7] 2011/01/26 16:29:09: Call Port: ad6bb336@pbx#1558453222 [7] 2011/01/26 16:29:09: Call Port: b274dc3f@pbx#1490165311 [7] 2011/01/26 16:29:09: Call Port: bb903a2c9d68a855#8614b5139f [7] 2011/01/26 16:29:09: Call Port: f02665db@pbx#907681560 [9] 2011/01/26 16:29:09: SIP Rx udp:174.119.245.84:60929:

The call should go thru the DP which is forced onto the Asterisk <-> PBXnSIP gateway trunk...

This is what happens when I use a domain instead of the localhost. Where: 69.39.221.107 - asterisk pbx 69.39.221.108 - pbxnsip domain.pbxnsip.com - domain name (valid!) 4168885555 - number trying to dial 71.19.171.195 - phone external ip 192.168.1.110 - phone internal ip `[7] 2011/01/26 15:45:21: SIP Rx udp:71.19.171.195:54864: INVITE sip:4168885555@domain.pbxnsip.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-5066863afe2ea563-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:101@192.168.1.110:54864> To: "4168885555"<sip:4168885555@domain.pbxnsip.com> From: "101"<sip:101@domain.pbxnsip.com>;tag=a0040960 Call-ID: NzI2NGRiNmVmOGJhYzRiMDFlYmNlM2JmNzI0YTdjNTM. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 216 v=0 o=- 0 2 IN IP4 192.168.1.110 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.110 t=0 0 m=audio 50634 RTP/AVP 107 0 8 101 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv [8] 2011/01/26 15:45:21: Could not find a trunk (98 trunks) [8] 2011/01/26 15:45:21: Using outbound proxy sip:71.19.171.195:54864;transport=udp because UDP packet source did not match the via header [7] 2011/01/26 15:45:21: SIP Tx udp:71.19.171.195:54864: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-5066863afe2ea563-1---d8754z-;rport=54864;received=71.19.171.195 From: "101" <sip:101@domain.pbxnsip.com>;tag=a0040960 To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=d202343f2f Call-ID: NzI2NGRiNmVmOGJhYzRiMDFlYmNlM2JmNzI0YTdjNTM. CSeq: 1 INVITE Content-Length: 0 [7] 2011/01/26 15:45:21: SIP Tx udp:71.19.171.195:54864: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-5066863afe2ea563-1---d8754z-;rport=54864;received=71.19.171.195 From: "101" <sip:101@domain.pbxnsip.com>;tag=a0040960 To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=d202343f2f Call-ID: NzI2NGRiNmVmOGJhYzRiMDFlYmNlM2JmNzI0YTdjNTM. CSeq: 1 INVITE User-Agent: Shanticom-PBX/3.4.0.3201 WWW-Authenticate: Digest realm="domain.pbxnsip.com",nonce="2b07b5d9db75c91fe4a1361672d5bb3d",domain="sip:4168885555@domain.pbxnsip.com",algorithm=MD5 Content-Length: 0 [7] 2011/01/26 15:45:21: SIP Rx udp:71.19.171.195:54864: ACK sip:4168885555@domain.pbxnsip.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-5066863afe2ea563-1---d8754z-;rport To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=d202343f2f From: "101"<sip:101@domain.pbxnsip.com>;tag=a0040960 Call-ID: NzI2NGRiNmVmOGJhYzRiMDFlYmNlM2JmNzI0YTdjNTM. CSeq: 1 ACK Content-Length: 0 [7] 2011/01/26 15:45:21: SIP Rx udp:71.19.171.195:54864: INVITE sip:4168885555@domain.pbxnsip.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-0926993c2a5b1d0b-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:101@192.168.1.110:54864> To: "4168885555"<sip:4168885555@domain.pbxnsip.com> From: "101"<sip:101@domain.pbxnsip.com>;tag=a0040960 Call-ID: NzI2NGRiNmVmOGJhYzRiMDFlYmNlM2JmNzI0YTdjNTM. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1104o stamp 56125 Authorization: Digest username="101",realm="domain.pbxnsip.com",nonce="2b07b5d9db75c91fe4a1361672d5bb3d",uri="sip:4168885555@domain.pbxnsip.com",response="afa5cada40725938b15d1899a0f8be23",algorithm=MD5 Content-Length: 216 v=0 o=- 0 2 IN IP4 192.168.1.110 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.110 t=0 0 m=audio 50634 RTP/AVP 107 0 8 101 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv [8] 2011/01/26 15:45:21: Tagging request with existing tag [6] 2011/01/26 15:45:21: Sending RTP for NzI2NGRiNmVmOGJhYzRiMDFlYmNlM2JmNzI0YTdjNTM.#d202343f2f to 192.168.1.110:50634 [7] 2011/01/26 15:45:21: SIP Tx udp:71.19.171.195:54864: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-0926993c2a5b1d0b-1---d8754z-;rport=54864;received=71.19.171.195 From: "101" <sip:101@domain.pbxnsip.com>;tag=a0040960 To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=d202343f2f Call-ID: NzI2NGRiNmVmOGJhYzRiMDFlYmNlM2JmNzI0YTdjNTM. CSeq: 2 INVITE Content-Length: 0 [7] 2011/01/26 15:45:21: SIP Tx udp:69.39.221.107:5060: INVITE sip:14168885555@69.39.221.107;user=phone SIP/2.0 Via: SIP/2.0/UDP 69.39.221.108:5060;branch=z9hG4bK-bfb7d342eef2c1b067d371c1b04acb80;rport From: "TEST2" <sip:6384858569@69.39.221.107;user=phone>;tag=1266699933 To: <sip:14168885555@69.39.221.107;user=phone> Call-ID: 634192b0@pbx CSeq: 11026 INVITE Max-Forwards: 70 Contact: <sip:6384858569@69.39.221.108:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: Shanticom-PBX/3.4.0.3201 Remote-Party-ID: "TEST2" <sip:6384858569@69.39.221.107;user=phone>;party=calling;screen=yes Content-Type: application/sdp Content-Length: 249 v=0 o=- 2124256490 2124256490 IN IP4 69.39.221.108 s=- c=IN IP4 69.39.221.108 t=0 0 m=audio 48642 RTP/AVP 0 18 101 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=sendrecv [8] 2011/01/26 15:45:21: Play audio_moh/noise.wav [7] 2011/01/26 15:45:21: SIP Rx udp:69.39.221.107:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 69.39.221.108:5060;branch=z9hG4bK-bfb7d342eef2c1b067d371c1b04acb80;received=69.39.221.108;rport=5060 From: "TEST2" <sip:6384858569@69.39.221.107;user=phone>;tag=1266699933 To: <sip:14168885555@69.39.221.107;user=phone>;tag=as2c68f520 Call-ID: 634192b0@pbx CSeq: 11026 INVITE User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7440bca6" Content-Length: 0 [7] 2011/01/26 15:45:21: Call 634192b0@pbx#1266699933: Clear last INVITE [7] 2011/01/26 15:45:21: SIP Tx udp:69.39.221.107:5060: ACK sip:14168885555@69.39.221.107;user=phone SIP/2.0 Via: SIP/2.0/UDP 69.39.221.108:5060;branch=z9hG4bK-bfb7d342eef2c1b067d371c1b04acb80;rport From: "TEST2" <sip:6384858569@69.39.221.107;user=phone>;tag=1266699933 To: <sip:14168885555@69.39.221.107;user=phone>;tag=as2c68f520 Call-ID: 634192b0@pbx CSeq: 11026 ACK Max-Forwards: 70 Contact: <sip:6384858569@69.39.221.108:5060;transport=udp> Remote-Party-ID: "TEST2" <sip:6384858569@69.39.221.107;user=phone>;party=calling;screen=yes Content-Length: 0 [5] 2011/01/26 15:45:21: INVITE Response 401 Unauthorized: Terminate 634192b0@pbx [7] 2011/01/26 15:45:21: Other Ports: 26 [7] 2011/01/26 15:45:21: Call Port: 13ccde2c-b65f537a@localhost#aa6c08a0f1 [7] 2011/01/26 15:45:21: Call Port: 154d3f0c@pbx#1008184428 [7] 2011/01/26 15:45:21: Call Port: 3c26992c6726-eu3cfqmr3aaa#d9f059f594 [7] 2011/01/26 15:45:21: Call Port: 3c27bb5785a3-844ug2wf59xq#b1d600805f [7] 2011/01/26 15:45:21: Call Port: 3c27dbc54ecc-8a0txagwtu0c#36f3df207b [7] 2011/01/26 15:45:21: Call Port: 3c2fcaf65979-9e1r50rqa8om#bb2b24f3c5 [7] 2011/01/26 15:45:21: Call Port: 3c32acca4ec5-07rrezqxi0lj#af0a49a366 [7] 2011/01/26 15:45:21: Call Port: 3c3621cf732d-2bgalyw4iq1a#cde4b05959 [7] 2011/01/26 15:45:21: Call Port: 3c3e5313785b-2yqj8rn1fmsx#655803ba5b [7] 2011/01/26 15:45:21: Call Port: 3c786c9e47bb-3ihpx3ca835r#9e907d2e63 [7] 2011/01/26 15:45:21: Call Port: 3c92cf90a4a6-u9pjbehd2t50#a0a42bee81 [7] 2011/01/26 15:45:21: Call Port: 4169e4ea@pbx#1720057541 [7] 2011/01/26 15:45:21: Call Port: 47678ca033aa336e69f6d1bf7e94783e@69.39.221.107#dc142f7561 [7] 2011/01/26 15:45:21: Call Port: 5dadb756@pbx#2108676965 [7] 2011/01/26 15:45:21: Call Port: 643386be@pbx#896059230 [7] 2011/01/26 15:45:21: Call Port: 6e80920f6f6840490ab2826670af2f07@69.39.221.107#d5697e696f [7] 2011/01/26 15:45:21: Call Port: 74e7e54c38acf7c8602305190862a751@69.39.221.107#eecc166ad8 [7] 2011/01/26 15:45:21: Call Port: 7c6e218f@pbx#887543378 [7] 2011/01/26 15:45:21: Call Port: 8014479f@pbx#1820803417 [7] 2011/01/26 15:45:21: Call Port: 86e63a85@pbx#1919082329 [7] 2011/01/26 15:45:21: Call Port: 91ec0f63@pbx#280178641 [7] 2011/01/26 15:45:21: Call Port: NzI2NGRiNmVmOGJhYzRiMDFlYmNlM2JmNzI0YTdjNTM.#d202343f2f [7] 2011/01/26 15:45:21: Call Port: a0f3dd41@pbx#182301236 [7] 2011/01/26 15:45:21: Call Port: c1f21b1eeff01a3d#241ad7d39c [7] 2011/01/26 15:45:21: Call Port: caf298fa@pbx#290526904 [7] 2011/01/26 15:45:21: Call Port: ffea8091@pbx#965068692 [6] 2011/01/26 15:45:21: Send codec pcmu/8000 [7] 2011/01/26 15:45:21: SIP Tx udp:71.19.171.195:54864: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-0926993c2a5b1d0b-1---d8754z-;rport=54864;received=71.19.171.195 From: "101" <sip:101@domain.pbxnsip.com>;tag=a0040960 To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=d202343f2f Call-ID: NzI2NGRiNmVmOGJhYzRiMDFlYmNlM2JmNzI0YTdjNTM. CSeq: 2 INVITE Contact: <sip:101@69.39.221.108:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: Shanticom-PBX/3.4.0.3201 Content-Type: application/sdp Content-Length: 226 v=0 o=- 1929781437 1929781437 IN IP4 69.39.221.108 s=- c=IN IP4 69.39.221.108 t=0 0 m=audio 57948 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [7] 2011/01/26 15:45:21: SIP Tx udp:71.19.171.195:54864: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-0926993c2a5b1d0b-1---d8754z-;rport=54864;received=71.19.171.195 From: "101" <sip:101@domain.pbxnsip.com>;tag=a0040960 To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=d202343f2f Call-ID: NzI2NGRiNmVmOGJhYzRiMDFlYmNlM2JmNzI0YTdjNTM. CSeq: 2 INVITE Contact: <sip:101@69.39.221.108:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: Shanticom-PBX/3.4.0.3201 Content-Length: 0 [7] 2011/01/26 15:45:21: SIP Rx udp:71.19.171.195:54864: ACK sip:4168885555@domain.pbxnsip.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.110:54864;branch=z9hG4bK-d8754z-0926993c2a5b1d0b-1---d8754z-;rport To: "4168885555" <sip:4168885555@domain.pbxnsip.com>;tag=d202343f2f From: "101"<sip:101@domain.pbxnsip.com>;tag=a0040960 Call-ID: NzI2NGRiNmVmOGJhYzRiMDFlYmNlM2JmNzI0YTdjNTM. CSeq: 2 ACK Content-Length: 0 [7] 2011/01/26 15:45:21: Other Ports: 25 [7] 2011/01/26 15:45:21: Call Port: 13ccde2c-b65f537a@localhost#aa6c08a0f1 [7] 2011/01/26 15:45:21: Call Port: 154d3f0c@pbx#1008184428 [7] 2011/01/26 15:45:21: Call Port: 3c26992c6726-eu3cfqmr3aaa#d9f059f594 [7] 2011/01/26 15:45:21: Call Port: 3c27bb5785a3-844ug2wf59xq#b1d600805f [7] 2011/01/26 15:45:21: Call Port: 3c27dbc54ecc-8a0txagwtu0c#36f3df207b [7] 2011/01/26 15:45:21: Call Port: 3c2fcaf65979-9e1r50rqa8om#bb2b24f3c5 [7] 2011/01/26 15:45:21: Call Port: 3c32acca4ec5-07rrezqxi0lj#af0a49a366 [7] 2011/01/26 15:45:21: Call Port: 3c3621cf732d-2bgalyw4iq1a#cde4b05959 [7] 2011/01/26 15:45:21: Call Port: 3c3e5313785b-2yqj8rn1fmsx#655803ba5b [7] 2011/01/26 15:45:21: Call Port: 3c786c9e47bb-3ihpx3ca835r#9e907d2e63 [7] 2011/01/26 15:45:21: Call Port: 3c92cf90a4a6-u9pjbehd2t50#a0a42bee81 [7] 2011/01/26 15:45:21: Call Port: 4169e4ea@pbx#1720057541 [7] 2011/01/26 15:45:21: Call Port: 47678ca033aa336e69f6d1bf7e94783e@69.39.221.107#dc142f7561 [7] 2011/01/26 15:45:21: Call Port: 5dadb756@pbx#2108676965 [7] 2011/01/26 15:45:21: Call Port: 643386be@pbx#896059230 [7] 2011/01/26 15:45:21: Call Port: 6e80920f6f6840490ab2826670af2f07@69.39.221.107#d5697e696f [7] 2011/01/26 15:45:21: Call Port: 74e7e54c38acf7c8602305190862a751@69.39.221.107#eecc166ad8 [7] 2011/01/26 15:45:21: Call Port: 7c6e218f@pbx#887543378 [7] 2011/01/26 15:45:21: Call Port: 8014479f@pbx#1820803417 [7] 2011/01/26 15:45:21: Call Port: 86e63a85@pbx#1919082329 [7] 2011/01/26 15:45:21: Call Port: 91ec0f63@pbx#280178641 [7] 2011/01/26 15:45:21: Call Port: a0f3dd41@pbx#182301236 [7] 2011/01/26 15:45:21: Call Port: c1f21b1eeff01a3d#241ad7d39c [7] 2011/01/26 15:45:21: Call Port: caf298fa@pbx#290526904 [7] 2011/01/26 15:45:21: Call Port: ffea8091@pbx#965068692 [7] 2011/01/26 15:45:21: SIP Rx tcp:174.89.48.178:2116:

I also tried it with a Snom 360... if that matters with the exact same results. I'm using a gateway with no user name or password associated with a global trunk. This is the context in Asterisk for outgoing: type=user host=64.34.222.110 context=from-pstn I have quite a few gateway trunks in asterisk, and they all work, but it seems to just absolutely fail with pbxnsip. BTW i'm running version 3.4 debian

Where: 4168885555 is the external number to call. 64.34.222.111 is the PBXNSIP pbx 79.10.172.171 is my local IP what I don't see is the attempt to see where it connects to the asterisk server. [9] 20110126113603: SIP Rx udp:79.10.172.171:46140: INVITE sip:4168885555@64.34.222.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.110:46140;branch=z9hG4bK-d8754z-6a011e61de02ae31-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:40@79.10.172.171:46140> To: "4168885555"<sip:4168885555@64.34.222.111> From: "40"<sip:40@64.34.222.111>;tag=8b5ecf62 Call-ID: ZTJhOWM4ZjI1ZjA3ZjE5OTY5Njc2MzQ0NmUxNmY1NDk. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 265 v=0 o=- 1 2 IN IP4 192.168.1.110 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.110 t=0 0 m=audio 1586 RTP/AVP 107 0 8 101 a=alt:1 1 : N7XPQWhc NjvjTWRo 192.168.1.110 1586 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv [9] 20110126113603: UDP: Opening socket on 0.0.0.0:61238 [9] 20110126113603: UDP: Opening socket on 0.0.0.0:61239 [9] 20110126113603: UDP: Opening socket on [::]:61238 [9] 20110126113603: UDP: Opening socket on [::]:61239 [8] 20110126113603: Could not find a trunk (98 trunks) [8] 20110126113603: Using outbound proxy sip:79.10.172.171:46140;transport=udp because UDP packet source did not match the via header [9] 20110126113603: Resolve 57573222: udp 79.10.172.171 46140 [9] 20110126113603: SIP Tx udp:79.10.172.171:46140: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.110:46140;branch=z9hG4bK-d8754z-6a011e61de02ae31-1---d8754z-;rport=46140;received=79.10.172.171 From: "40" <sip:40@64.34.222.111>;tag=8b5ecf62 To: "4168885555" <sip:4168885555@64.34.222.111>;tag=7cdd5e20d0 Call-ID: ZTJhOWM4ZjI1ZjA3ZjE5OTY5Njc2MzQ0NmUxNmY1NDk. CSeq: 1 INVITE Content-Length: 0 [9] 20110126113603: Resolve 57573223: udp 79.10.172.171 46140 [9] 20110126113603: SIP Tx udp:79.10.172.171:46140: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 192.168.1.110:46140;branch=z9hG4bK-d8754z-6a011e61de02ae31-1---d8754z-;rport=46140;received=79.10.172.171 From: "40" <sip:40@64.34.222.111>;tag=8b5ecf62 To: "4168885555" <sip:4168885555@64.34.222.111>;tag=7cdd5e20d0 Call-ID: ZTJhOWM4ZjI1ZjA3ZjE5OTY5Njc2MzQ0NmUxNmY1NDk. CSeq: 1 INVITE User-Agent: Shanticom/3.4.0.3201 WWW-Authenticate: Digest realm="64.34.222.111",nonce="19be2cdcd227ea0e1467de7350f15c06",domain="sip:4168885555@64.34.222.111",algorithm=MD5 Content-Length: 0

It does happen if you have two windows open on the same pbx. Use different browsers....

I'm also having the same issue. I tried using authentication as well as without, still not working.

Thank you for the prompt response. Do you think an AA with a timeout of 0 with no voice recording forwarded to the extension is a good idea?