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Leonmeijer

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Everything posted by Leonmeijer

  1. The available upstream is about 10 Mbit and the downstream is 120 Mbit (the server itself has a +/- 400Mbit upstream). I checked my snom 360 it actually has "better" audio then the softphone but the softphone (the snom one) didn't want to install because the visual C++ was "missing" and the VC++ setup told me that a newer version is allready installed so I just renamed all the files and ran it... Connections from a mobile phone to the pbx (from mobile to my voip provider to the pbx) are fine so the problem should be in my LAN then. I will check the router and probably I'll get it fixed. Thanks for the help and I'll be back here if i'm having issues fixing it.
  2. Set these values by editing the pbx.xml file, restarted it and it seems to work again which is great, thanks, but now the audio sounds crackling instead of hearing a voice when e.g. calling the mailbox. Guess this might be related to codec preference? What is the default order for codecs and which ones can be better removed from the list?
  3. Thanks for the info, there's no such file pnp.xml but in pbx.xml there's: "<min_expires/><max_expires/>" (no value, so both are probably set to 0), what are correct values for these properties? nat_tcp is set to 180 there. The same LAN gives the same result even using the same machine through 127.0.0.1 / localhost doen't do anything which I guess should 'always' work...
  4. Hello I'm running PBXnSIP 4.5.1.1107 Zeta Perseids (Win64) And extensions are just unable to connect; Not from my home network. Not from any smartphone. Not from the Win 2008 server where the PBX runs on. I'm getting quite mad about it because I've been trying a few years now and just registered the phones to the sip registars directly instaed of using the product. I'm not going to upgrade to v5 because the license costs and it's a pitty there is no newer v4.5 version around. Registrationg goes OK, but after 15 secs times out (there is no 15 second timeout set anywhere on the system). Directly when I try to call after registration to 8100 (voicemail) it goes: [6] 2013/05/31 14:06:18: SIP TCP/TLS timeout on my.external.ip.address:49984, closing connection and this one: [9] 2013/05/31 14:06:12: Registration for account "Test Phone" <sip:100@localhost> expired, removing contact <sip:100@my.internal.ip.address:49984;transport=tcp;line=j8766z>;reg-id=1;+sip.instance="<urn:uuid:396b5946-418e-4f9c-b571-45e238ed2424>" TLS is disabled, tried both UDP and TCP. Firewall settings are OK even without firewall I have no luck. Can someone PLEASE get this fixed? [5] 2013/05/31 14:06:07: SIP Tx tcp:my.external.ip.address:49984: SIP/2.0 401 Authentication Required Via: SIP/2.0/TCP my.internal.ip.address:49984;branch=z9hG4bK-4jmrch;rport=49984;received=my.external.ip.address From: "100" <sip:100@localhost>;tag=na1r0k To: "100" <sip:100@localhost>;tag=f0163d3e69 Call-ID: gkjhts0d@snom CSeq: 1474 SUBSCRIBE User-Agent: snomONE/4.5.1.1107 Zeta Perseids WWW-Authenticate: Digest realm="localhost",nonce="52c1ffa043d16f26daaf05da27c81027",domain="sip:100@localhost",algorithm=MD5 Content-Length: 0 [5] 2013/05/31 14:06:07: SIP Rx tcp:my.external.ip.address:49984: SUBSCRIBE sip:100@localhost SIP/2.0 Via: SIP/2.0/TCP my.internal.ip.address:49984;branch=z9hG4bK-qlbtdt;rport From: "100" <sip:100@localhost>;tag=na1r0k To: "100" <sip:100@localhost> Call-ID: gkjhts0d@snom CSeq: 1475 SUBSCRIBE Max-Forwards: 70 Contact: <sip:100@my.internal.ip.address:49984;transport=tcp;line=j8766z>;reg-id=1;+sip.instance="<urn:uuid:396b5946-418e-4f9c-b571-45e238ed2424>" Supported: outbound, gruu Event: message-summary Accept: application/simple-message-summary User-Agent: snom-YYY/9.3.7XXX Authorization: Digest realm="localhost",nonce="52c1ffa043d16f26daaf05da27c81027",response="d98297037105ea646ae9719dee7d37ce",username="100",uri="sip:100@localhost",algorithm=MD5 Expires: 3600 Content-Length: 0 [5] 2013/05/31 14:06:07: SIP Tx tcp:my.external.ip.address:49984: SIP/2.0 200 Ok Via: SIP/2.0/TCP my.internal.ip.address:49984;branch=z9hG4bK-qlbtdt;rport=49984;received=my.external.ip.address From: "100" <sip:100@localhost>;tag=na1r0k To: "100" <sip:100@localhost>;tag=f0163d3e69 Call-ID: gkjhts0d@snom CSeq: 1475 SUBSCRIBE Contact: <sip:the.ip.of.the.pbx:5060;transport=tcp> Require: outbound Date: Fri, 31 May 2013 12:06:07 GMT Server: snomONE/4.5.1.1107 Zeta Perseids Expires: 0 Content-Length: 0 [9] 2013/05/31 14:06:12: Registration for account "Test Phone" <sip:100@localhost> expired, removing contact <sip:100@my.internal.ip.address:49984;transport=tcp;line=j8766z>;reg-id=1;+sip.instance="<urn:uuid:396b5946-418e-4f9c-b571-45e238ed2424>" [6] 2013/05/31 14:06:18: SIP TCP/TLS timeout on my.external.ip.address:49984, closing connection [5] 2013/05/31 14:06:18: Registration 8vjv97sl@snom closed connection, removing it [5] 2013/05/31 14:06:18: Registration gkjhts0d@snom closed connection, removing it [8] 2013/05/31 14:06:18: Release SIP thread 4 [8] 2013/05/31 14:06:56: Received SIP connection 5 from my.external.ip.address:49991 [5] 2013/05/31 14:06:56: SIP Rx tcp:my.external.ip.address:49991:
  5. Okay, thanks for the explaination, will remain on v4.5.1 here, it's actually good enough except that I miss the fax2mail feature which I requested many times
  6. very nice but... my v4 license is not supported anymore? I've started at PBXnSIP v1, never had to buy a new license but from 4 to 5 seems to be a bit of a problem. Can't imagine an upgrade should be payed for these days?!
  7. Allright, yeah then I will wait for version 6. I suggested such a scripting languages a while ago, at that point it wasn't planned. Great to hear that it might become available in v6. What I want to do is when a call comes down the line, check the number (lookup in a database) then play some wave file and redirect the call to the destination in the database. Regards, Leon
  8. Hello, Any updates on this one? just upgraded to SNOM one and trying to play a wave file using a SOAP response. <destionation>100</destination> <play>recordings/rec0001.wav</play> but it doesn't work, how to get this working? I can't create a 1000 IVR nodes (for each wave file another IVR) unless you give me a free license . Regards, Leon
  9. Hello, I want to route annonymouse calls away from 1 trunk before the call enters the agent group (I need the agent group because of the name on the display of the phone and queue settings). Now I guess I can put: !0!8100! in the FROM based routing (IF annonymouse (=0) goto voicemail of 100. !E!100! in the DTMF list for end of wave file playback (AND NOT annonymouse) What I want is, to notify the caller about something, then filter annonymouse to voicemail and other just to the extension (agent group). (in the case, I din't put in the agent group but an extension number but it's just the example;)). Can someone tell me if this is correct or what todo if not.
  10. Ah didn't notice the 20sec thing, well here we go: [5] 7/5/2010 17:36:18: Applying Settings... [6] 7/5/2010 17:36:18: Attempt to write the file /snomconfig/config.xml which has not changed. [6] 7/5/2010 17:36:18: Attempt to write the file /snomconfig/firmware.xml which has not changed. [6] 7/5/2010 17:36:18: Attempt to write the file /snomconfig/fkeys.xml which has not changed. [6] 7/5/2010 17:36:18: Attempt to write the file /snomconfig/tbook.xml which has not changed. [6] 7/5/2010 17:36:18: Attempt to write the file /snomconfig/dialplan.xml which has not changed. (just put another password in, clicked save, wait 20sec and.... NOT saved)
  11. Here is the phone's log which shows it sets the password (password not shown) I can't find wheater it is saved or not, but in the pbxnsip I can see that it isn't [6] 7/5/2010 17:27:21: new Tag: >70bmx126qn5qyfld< Filename: log.htm [9] 7/5/2010 17:27:22: Timer: webserver/webserver.cpp (192): [close] [6] 7/5/2010 17:27:23: webserver::request 77: GET /line_login.htm?l=1 HTTP/1.1 [6] 7/5/2010 17:27:23: Basic authentication. [6] 7/5/2010 17:27:23: new Tag: >elvtzq3n6bl1klib< Filename: line_login.htm [9] 7/5/2010 17:27:25: Timer: webserver/webserver.cpp (192): [close] [9] 7/5/2010 17:27:33: Timer: sip/registrar.cpp (720): [set_nr_timeout] [9] 7/5/2010 17:27:33: Timer: sip/sip_send.cpp (1421): [send_packet] [9] 7/5/2010 17:27:36: Timer: sip/network.cpp (751): [message_repetition] [6] 7/5/2010 17:27:36: webserver::request 78: POST /line_login.htm?l=1 HTTP/1.1 [6] 7/5/2010 17:27:36: Basic authentication. [7] 7/5/2010 17:27:36: settings::apply_value: user_pass = '******', set.need_apply: 0, finished: 1, need reboot to apply: 0 [8] 7/5/2010 17:27:36: Routing to outbound proxy sip:ip2.lmeijer.nl [8] 7/5/2010 17:27:36: route_pending_packet 1003384: entry=url ? sip:ip2.lmeijer.nl [8] 7/5/2010 17:27:36: DNS cache_lookup: naptr ip2.lmeijer.nl -> [8] 7/5/2010 17:27:36: route_pending_packet 1003384: entry=srv tls _sips._tcp.ip2.lmeijer.nl [8] 7/5/2010 17:27:36: DNS cache_lookup: srv _sips._tcp.ip2.lmeijer.nl -> [8] 7/5/2010 17:27:36: route_pending_packet 1003384: entry=srv tcp _sip._tcp.ip2.lmeijer.nl [8] 7/5/2010 17:27:36: DNS cache_lookup: srv _sip._tcp.ip2.lmeijer.nl -> [8] 7/5/2010 17:27:36: route_pending_packet 1003384: entry=srv udp _sip._udp.ip2.lmeijer.nl [8] 7/5/2010 17:27:36: DNS cache_lookup: srv _sip._udp.ip2.lmeijer.nl -> [8] 7/5/2010 17:27:36: route_pending_packet 1003384: entry=a udp ip2.lmeijer.nl 5060 [8] 7/5/2010 17:27:36: DNS cache_lookup: a ip2.lmeijer.nl -> 80.69.83.97 [8] 7/5/2010 17:27:36: route_pending_packet 1003384: entry=udp 80.69.83.97 5060 [8] 7/5/2010 17:27:36: Send Packet REGISTER [9] 7/5/2010 17:27:36: result of get_ip_adr:80.69.83.97 192.168.1.110 [9] 7/5/2010 17:27:36: result of get_ip_adr:80.69.83.97 192.168.1.110 [5] 7/5/2010 17:27:36: sip::send_register: reregister timer for line 0 set to 300s [8] 7/5/2010 17:27:36: Routing to outbound proxy sip:ip2.lmeijer.nl [8] 7/5/2010 17:27:36: route_pending_packet 1003385: entry=url ? sip:ip2.lmeijer.nl [8] 7/5/2010 17:27:36: DNS cache_lookup: naptr ip2.lmeijer.nl -> [8] 7/5/2010 17:27:36: route_pending_packet 1003385: entry=srv tls _sips._tcp.ip2.lmeijer.nl [8] 7/5/2010 17:27:36: DNS cache_lookup: srv _sips._tcp.ip2.lmeijer.nl -> [8] 7/5/2010 17:27:36: route_pending_packet 1003385: entry=srv tcp _sip._tcp.ip2.lmeijer.nl [8] 7/5/2010 17:27:36: DNS cache_lookup: srv _sip._tcp.ip2.lmeijer.nl -> [8] 7/5/2010 17:27:36: route_pending_packet 1003385: entry=srv udp _sip._udp.ip2.lmeijer.nl [8] 7/5/2010 17:27:36: DNS cache_lookup: srv _sip._udp.ip2.lmeijer.nl -> [8] 7/5/2010 17:27:36: route_pending_packet 1003385: entry=a udp ip2.lmeijer.nl 5060 [8] 7/5/2010 17:27:36: DNS cache_lookup: a ip2.lmeijer.nl -> 80.69.83.97 [8] 7/5/2010 17:27:36: route_pending_packet 1003385: entry=udp 80.69.83.97 5060 [7] 7/5/2010 17:27:36: Trusted IP Addresses: udp:80.69.83.97 [6] 7/5/2010 17:27:36: new Tag: >cb160p5fvnfej1d8< Filename: line_login.htm [9] 7/5/2010 17:27:37: Timer: sip/sip_send.cpp (791): [route_pending_packet] [8] 7/5/2010 17:27:37: Send Packet REGISTER [9] 7/5/2010 17:27:37: result of get_ip_adr:80.69.83.97 192.168.1.110 [9] 7/5/2010 17:27:37: result of get_ip_adr:80.69.83.97 192.168.1.110 [9] 7/5/2010 17:27:38: Timer: webserver/webserver.cpp (192): [close] [8] 7/5/2010 17:27:38: Sending Keepalive to Watchdog [9] 7/5/2010 17:27:38: Timer: sip/sip_send.cpp (620): [resend_packet] [8] 7/5/2010 17:27:38: Send Packet REGISTER [9] 7/5/2010 17:27:38: result of get_ip_adr:80.69.83.97 192.168.1.110 [9] 7/5/2010 17:27:38: result of get_ip_adr:80.69.83.97 192.168.1.110 [6] 7/5/2010 17:27:38: webserver::request 79: GET /log.htm HTTP/1.1 [6] 7/5/2010 17:27:38: Basic authentication. [6] 7/5/2010 17:27:38: new Tag: >pikheruujxtml6nx< Filename: log.htm And here the system information from the web interface: System Information: Phone Type: snom360-SIP MAC-Address: 00041329827D IP-Address: 192.168.1.110 Firmware-Version: snom360-SIP 8.2.29 20821 Firmware-URL: Production Information: Mac:00041329827D;Version:Standard;Hardware:snom360 (H: R2A);Date:15/05/08;Copyright(C) snom technology AG Uptime: 1 days, 19 hours, 4 minutes LCS: 1 days, 19 hours, 4 minutes (0) Memfree: 440 K CPU: 0.01 0.02 0.00 1/10 78 Bootloader-Version: 1.1.3-u Provisioning: snom Provisioning SIP Identity Status: Identity 1 Status: 100@localhost: Network Failure Identity 2 Status: Identity 3 Status: Identity 4 Status: Identity 5 Status: Identity 6 Status: Identity 7 Status: Identity 8 Status: Identity 9 Status: Identity 10 Status: Identity 11 Status: Identity 12 Status: Ethernet Status: Net Port: Connection Type: 100 Mbit Full Duplex Status: connected PC Port: Connection Type: 100 Mbit Full Duplex Status: connected
  12. Well, I have played arround with a lot of things, other pc's compatibilty view (IE8) saving and rebooting using the phone, webinterface and the plug. Tried to reset all to default but the phone is just "ignoring" / " not saving" the password. The log doesn't show anything related to saving the extension (phone log). I ran out of options on this one...
  13. Well I didn't reboot at all, but now I did it both pulling the plug and via the maintenance menu but nothing, still the password is blank. Other settings are saved correctly btw.
  14. It looks like my snom won't save the password, I've set the PBXnSIP password to blank and... there you go " 200 OK" it's registered. So it's a snom related error, now it's registered it only won't call e.g. the mailbox the phone keeps saying "calling" but that might be a codec or firewall related error i gues...
  15. I've upgraded to the latest v8 firmware, and double-checked control+c, control+v'ed a new password (copied from notepad, pasted in both pbx and phone password fields). I still get these 401 auth required messages... I also tried a new extension but didn't work too... I did a brand new pbxnsip installation... if you have any ideas let me know, i'll keep on trying some settings with the @domain thing (btw, no I only have a "localhost" domain).
  16. Hello, I ran my PBXnSIP always locally but moved the server to a datacenter now. I've configured the phone (a snom 360) as follows: Identity Active: ON. Account: 100 (pbxnsip extension). Password: the password Registar: localhost (pbxnsip domain). Outbound proxy: IP.of.external.host Auth. username: 100 (pbxnsip extension). Then, when I check the PBXnSIP I see the following log (level 9). REGISTER sip:localhost SIP/2.0 Via: SIP/2.0/UDP 10.0.0.250:2051;branch=z9hG4bK-2s6f0e0n3kfe;rport From: "100" <sip:100@localhost>;tag=84y005zlh7 To: "100" <sip:100@localhost> Call-ID: 3c26700ccf85-rdt8ol2pksam@snom360-00041329827D CSeq: 1 REGISTER Max-Forwards: 70 Contact: <sip:100@10.0.0.250:2051;line=q2o58ags>;q=1.0;flow-id=1;+sip.instance="<urn:uuid:b4909cfe-3dd3-4353-b064-addd22cead39>";audio;mobility="fixed";duplex="full";description="snom360";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom360/6.5.13 Supported: gruu Allow-Events: dialog X-Real-IP: 10.0.0.250 WWW-Contact: <http://10.0.0.250:80> WWW-Contact: <https://10.0.0.250:443> Expires: 3600 Content-Length: 0 [9] 2010/05/05 18:16:51: SIP Tr udp:my.own.external.ip:2051: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 10.0.0.250:2051;branch=z9hG4bK-2s6f0e0n3kfe;rport=2051;received=195.241.108.161 From: "100" <sip:100@localhost>;tag=84y005zlh7 To: "100" <sip:100@localhost>;tag=3b5e38fe8e Call-ID: 3c26700ccf85-rdt8ol2pksam@snom360-00041329827D CSeq: 1 REGISTER User-Agent: pbxnsip-PBX/4.0.1.3499 WWW-Authenticate: Digest realm="localhost",nonce="3ba3badbfee7cb13a1ef832e9ae76784",domain="sip:localhost",algorithm=MD5 Content-Length: 0 It appears that the phone sends it's "Real-ip" as the local ip of the phone. The phone isn't behind any firewall and I din't change the password(s) or any other setting. Can anyone help me out?
  17. I use the agent group to place calls in a queue if the number is busy, and I need to record a name for annonymous calls, so... how can I do this? I suggest (seen the above) a feature in agent group "forward annonymous calls to" or can I do it by adding an IVR node "before" the agent group wich does a from based routing annonymous to extension blabla, but... in the wiki I can't find a DTMF for annonymous (maybe A?) so I guess when use: !0!100! (0 is empty/nothing/annonymous I guess) in the FROM based routing of an IVR will do or not?
  18. Oke, would be nice to put the name in the recording (or save the name). I've also just tested how the name prompting works. - I called annonymously to my phone number from another phone (outside the pbxnsip). - The call went from the trunk to the agent group and, yes yes from the agent group directly to the extention WITHOUT the name prompt... oops, that's not right... so, is that a version 4 bug or not included yet? There must be prompted for a name if the caller-id is none or annonymous in ANY case;) Regards, Leon
  19. Hello, Someone annonymous keeps on calling me but, of cource when I'm not in, they also don't leave a message behind on the voicemail, so now I set the Ask for name feature on annonymous calls, today when I came back I saw 1 missed call from "Annonymous (none)" but now I'm unable to find the recorded name, if it's recorded (but I guess so because the call got through...). When the calls come in to the PBX (trunk) they go to an agent group which transfers the call to the extension, i've set the name agent group to the phone number of which they come (trunk 1 or 2) with the remote party ID included so I can see to which number is being called. Can someone tell me where to find the recorded names or where to specifiy the location to store them? Another question I have, maybe I need to create another topic for it, is it possible to record calls on the agent group from the point they enter the queue (so you can listen back who was in that queue). Regards, Leon
  20. Yeah but in some cases people don't want to wait, but some important information could be told then they start pressing arround to get rid of the message, which in most systems doesn't do anything and just keep playing the message, in some cases only # can stop playing the sound (that's my experience) I would like to just ignore input but else, I will just use an expression that redirects when [0-9] or * / # is pressed.
  21. I'm currently using the latest 4 beta, if the user presses a key, the wave stops playing and the time out kicks in... it doesn't just "ignore" user input which it should do... (only # is allowed to stop the wave and forward). Is that a v4 problem or.... do I something wrong?
  22. Just about to ask the same question, I really would appriciate a changelog / release notes to see what's changed, to see at least which bugs / issues are fixed.... could be just a textfile but would be nice
  23. Hello, I want to setup an IVR node which e.g. goes to extenions 1 if the user input is a star * and dials extension 2 when the user presses #. When another input is detected the PBX should continue playing the wave file and ignoring it, how to do this?
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