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Everything posted by dslepnev

  1. Hello, We found one strange issue with pickup function. Maybe somebody saw it, and know how to solve... Scenario: Call from SIP provider to Pbxnsip: 555-555-555 --> 200 (extension on pbxnsip). While 200 phone ringing, another user (202) pick up the call, which ringing on 200. Pickup working ok. But, in Web interface of pbxnsip -> Active calls we see that this call active between 555-555-555 and 200! It's strange because 202 already did pickup and speaking with 555-555-55. There is should be active call between 555-555-55 and 202... Version: (Linux)
  2. Hello, Fom time to time we hear from customers that they have a problems with NAT traversal (no audio in both or one direction). To solve this problem we recommend to use STUN servers with pbxnsip. In most cases it helps. Questions is: have pbxnsip any plans to implement STUN inside the pbxnsip code? It's will be very useful, and from my point of vew will not take a long time for development...
  3. Yes, I tryed - without success. Only one way to add record with speed dial code - it's to add all the lines at the same time. First, last name, phone number, speed dial code. When it works.
  4. Hello, We found the folowing issue with Domain Address book: 1. Initially we create regular record: First name, Last name, phone number, and press "Create" button. We don't put speed dial number at this stage, plan to do it later. 2. Ok, we see the new record in the Address Book, with created user. Press Edit (we would like to add Speed dial number now). 3. Type Speed dial number, and press Save. Ok, let's check. Nothing changes. Still the line with name of user, phone number, but without Speed dial code entered later. If we create new user and fill all lines with Speed dial, and press Create - everything is ok. The problem only with Edit mode. Version: (Win32)
  5. OS: Windows XP, pbxnsip version (Win32). Call scenario: PSTN --E1-->AC Mediant 1000 --SIP-->pbxnsip AA (111) --> Hunt group (444). from Mediant it's looks like a call to specified phone number of AA (111).
  6. I have checked XML files in the cdr folder. Here is all information collected correctly. Everything is ok, but not with CSV.
  7. Hello, I have configured CDR's output to CSV file. But I found that not all records are stored in the file. For example: Below call from extension to PSTN. It's ok. 20090321101329, 213, 989267901766, 0 Here is incoming call from trunk gateway to pbxnsip. Here is no sourse number and called number. 20090321110155, , , 1 Ok, I have modified CDR_FORMAT value in pbx.xml to "$w$20e$20c$5d$10c$10i$20F$20T$o$v$x$y" It helps but, not clear: 20090408203235, , , 3, ,ea19da5462,345@test;user=phone>, <sip:111@test>,,attendant As you can see when I try to take sourse and destination number from SIP headers they have also additional data with phone number. Advice please, how I need to configure cdr_format to see in the CDR call from extension and call from PSTN (Trunk) onto extension?
  8. It's still doesn't work as well. The same - instead of custom zone name from timezones.xml I see in the blank line. About additional fixes for RU: 1. This version include few new timezones (Samara, Ekaterinburg, etc). Fisrt of all - this zones named in English. Strange to see it when whole interface translated to Russian. 2. Small mistake: change please Yakrutks to Yakutsk. It's typo.
  9. Translation file on the same folder. Everything is ok. OS: Windows XP Tryed on 3.1 and on <?xml version="1.0" encoding="utf-8"?> <language name="ru"> <file> <file name="timezones.xml"> <item id="GMT+6">Russia, Yekaterinburg</item> </file> </language>
  10. Hello, timezones.xml file already in /html directory. Code: <?xml version="1.0" encoding="utf-8"?> <timezones dict="timezones.xml"> <zone name="GMT+6"> <description>Russia, Yekaterinburg</description> <gmt_offset>-32400</gmt_offset> <dst_offset>3600</dst_offset> <dst_start_day_of_week>1</dst_start_day_of_week> <dst_start_month>4</dst_start_month> <dst_start_time>02:00</dst_start_time> <dst_start_week_of_month>1</dst_start_week_of_month> <dst_stop_day_of_week>1</dst_stop_day_of_week> <dst_stop_month>10</dst_stop_month> <dst_stop_time>02:00</dst_stop_time> <dst_stop_week_of_month>Last</dst_stop_week_of_month> </zone> </timezones> After that added timezone item to lang_ru.xml at the same directory: <file name="timezones.xml"> <item id="GMT+6">Russia, Yekaterinburg</item> Restarted the service, found message that timezone found: [7] 20090312122134: Found time zones GMT+6 But when I trying to set this timezone in web-interface - I see blank line instead of zone name and error message in the log: 5] 20090312122330: Dictionary: Item timezones.xml GMT+6 ru not found What's wrong? Advice pease!
  11. Thanks in advice! Now it's ok.
  12. I tryed to use this feature. The system cut part of destination numbers. Part of CSV : 20090217182847, 202,254495xxxx, 9 202 my extension number really dialed number was 81097254495xxxx
  13. I sent the trace already to support@pbxnsip.com
  14. SIP messages reaching ip phones. I checked that home router do not blocking the traffic from pbx. Pbxnsip send media traffic to internal IP address, and this media can't be delivered to the ip phone. This is the problem. Sorry, I can't attach the trace "Upload failed. You are not permitted to upload this type of file".
  15. Hi, We trying to configure pbxnsip to work with remote users who would like to work at home. Most of remote users have internet connection behind NAT. It causes problem when users don't hear audio at all. What happens: This is INVITE from remote user. This INVITE coming from real address. 1 0.000000 SIP/SDP Request: INVITE sip:112@tw;user=phone, with session description pbxnsip answering to this address. Good. 2 0.003732 SIP Status: 100 Trying 3 0.010382 SIP/SDP Status: 200 Ok, with session description and trying to send RTP to private address. This address taken by pbxnsip from INVITE. 4 0.021485 RTP Payload type=ITU-T G.711 PCMU, SSRC=1237099670, Seq=24774, Time=3241200, Mark 6 0.041015 RTP Payload type=ITU-T G.711 PCMU, SSRC=1237099670, Seq=24775, Time=3241360 Sure that pbxnsip working fully according SIP RFC - sends RTP to IP address from INVITE/SDP. Quesion is: maybe possible to configure pbxnsip to send RTP to addres from INVITE received? As part of Mini-SBC function..
  16. Correct! Offset value 18000, zone description "Russia, Ekaterinburg".
  17. Hello, We would like to use GTM+5 (Ekaterinburg, DST) timezone in pbxnsip, but there is no such setting. Advice please: it's possible to add manually support of this timezone? Thanks!
  18. Thanks, but CC don't do what we want. Sometimes ANI which comes to pbx was broken. Idea is: call-back initiated from www (by click to dial). It's can help us to create web-based address book, with call-back initiated from web-site to sip phone and mobile (everywhere). Can we hope that feature when click to dial event start also call to mobile phone can be done in next versions?
  19. Hello, Inside of my extension settings I set my mobile phone number. When I make test call from another one extension to my - both phones (ip-phone and mobile) are ringing at the same time, as configured. But when I trying to initiate Click to Dial with my username and extension - the system calling back my IP-phone _only_, and did'nt tryed to reach mobile phone number. Advice please, it's possible in general? Version: (Win32) Thanks!
  20. Thank you! One more question: how I can separate two regular expressions? I tryed with space, but it does'nt work "!([005411])!\1!t!110! !([0-9]{6})!\1!t!112!".
  21. Hello, I trying to use Extended mode inside the Trunk settings. Referring to the "http://wiki.pbxnsip.com/index.php/Inbound_Calls_on_Trunk" I put "![0-9]{7}([0-9]*)!\1!t!112" into "Send calls to extension" and send the call from PSTN gateway with 7 digits phone number. As result I got SIP 404 "not found message". Checked almost everything, and did'nt found where is the problem. Version: (Win32) Advice please! Why I tryed to use that feature: If call coming from ITSP (always with the same B number (0050001)) - I need to send this call to Auto-Attendant (112), If call from the same ITSP coming with any other number - then I need to send it to another one Auto-Attendant (110). I starded from "simpe way" (ot the top of topic), but got a problem.
  22. Advice please: how to turn off "Rewrite global numbers" function? We don't use +/0/00 at all, but the system add "+" before ANI, and trying to use 0 and 00 prefixes on DNIS. Version: (Win32)
  23. Hi, We using pbxnsip v. with snom ip-phones (320, 360, 370). Today I saw in the pbxnsip log the folowing messages: [5] 20080903110133: Web Server: File snom_web_lang.xml not found [5] 20080903110133: Web Server: File snom_gui_lang.xml not found Could somebody advice me where to get this xml files? Thanks!
  24. No, there is no redirects. Just IVR node with greeting, no records in CDR. It happens when I use "Send call to extension:" option only.
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