dslepnev
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Posts posted by dslepnev
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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...
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Tested the same thing on .3177 here (Windows version). Did see the problem. Any more details? Are you repeating the speed dial star code?
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.
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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: 3.3.1.3177 (Win32)
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Seem to work ok here. Can you tell me the OS and call scenario, say, whether it is coming thought the AA or Hunt/Agent group etc?
OS: Windows XP, pbxnsip version 3.2.0.3144 (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).
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Can you first verify whether you have the correct data under <pbx install dir>/cdr? This folder contains CDRs in the XML format. You should be able to locate the corresponding file and verify. If that file shows the correct data, then we may have some issue in the CSV format.
I have checked XML files in the cdr folder. Here is all information collected correctly. Everything is ok, but not with CSV.
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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?
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If you like, please try http://pbxnsip.com/protect/pbxctrl-3.3.0.3160.exe. That might also contain additional fixes for RU.
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.
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The translation is missing? Any change to try an upgrade? If so, what operating system?
Translation file on the same folder. Everything is ok.
OS: Windows XP
Tryed on 3.1 and on 3.2.0.314
<?xml version="1.0" encoding="utf-8"?>
<language name="ru">
<file>
<file name="timezones.xml">
<item id="GMT+6">Russia, Yekaterinburg</item>
</file>
</language>
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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!
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Looks like too many digits to me. Probably you have to increase the number of digits in the string. For example, change the string "$w$5e$10c$5d" to something like "$w$5e$20c$5d". It is in the setting "cdr_format", check out http://wiki.pbxnsip.com/index.php/Global_Configuration_File for more information.
Thanks in advice! Now it's ok.
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Yea, we removed it. If you run version 3.2, you can just put "file:cdr.txt" into the SOAP CDR URL field and you'll get nicely formatted files in the file system.
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
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You can send the trace to support@pbxnsip.com
I sent the trace already to support@pbxnsip.com
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I have faced the same situation, to resolve I didi the following:
1- use a router that support DMZ and set to the adress of your PBX server - work fine
I need to set routers which supports DMZ on all user's locations. Not best way. SIP ALG?
OR
2- set your router to allow ports for your PBX - all of them RTP is transporting audio
There is problem not in ports. Problem in addresses, which coming inside SIP messages to the PBX.
you can also set your PBX to use the following route table :
under Admin>Ports set "SIP IP Replacement List" to first IP of your PBX and IP of your router - something like 192.168.0.1/xxx.xxx.xxx.xxx -> PBXIP/RouterIP
restart PBX and It should work fine
50/50. It can help in case when static IP used. Most of ISP's changes external IP addresses of home users (DSL) once a day. I need to seek this addresses....
This run fine on my side
BTW you should have a very very very good Internet connection if you want to keep audio quality for all external registration on your PBX otherwise it will be more like a cell phone quality, forcing codec might help.
G.711 used only for tests. Sure we will use G.729 in future.
hope it help!
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Based on the above log, it looks like SIP messages are not reaching the phones. I do not see an ACK from the phone for the 200 OK. Please make sure that the home router is not blocking the traffic from PBX. Generally, SIP/RTP uses UDP. So you have to make sure that port UDP 5060 is not blocked and for RTP, 49152-64512 are not blocked. Please refer Admin->settings->ports page to see these values.
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".
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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 195.146.74.104 192.168.22.43 SIP/SDP Request: INVITE sip:112@tw;user=phone, with session description
pbxnsip answering to this address. Good.
2 0.003732 192.168.22.43 195.146.74.104 SIP Status: 100 Trying
3 0.010382 192.168.22.43 195.146.74.104 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 192.168.22.43 192.168.1.25 RTP Payload type=ITU-T G.711 PCMU, SSRC=1237099670, Seq=24774, Time=3241200, Mark
6 0.041015 192.168.22.43 192.168.1.25 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..
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We have Moscow:
<zone name="GMT+3"> <description>Russia, Moscow</description> <gmt_offset>10800</gmt_offset> <dst_offset>3600</dst_offset> <dst_start_day_of_week>1</dst_start_day_of_week> <dst_start_month>3</dst_start_month> <dst_start_time>02:00</dst_start_time> <dst_start_week_of_month>Last</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>
So the only thing we have to change would be the gmt_offset?
Correct! Offset value 18000, zone description "Russia, Ekaterinburg".
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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!
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Yea, at the moment that is the intended behavior. If you want to get a callback on your cell phone, consider using the callback feature (see http://wiki.pbxnsip.com/index.php/Calling_Card).
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?
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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: 3.1.1.3091 (Win32)
Thanks!
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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!".
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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: 3.1.1.3091 (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.
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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: 3.1.1.3091 (Win32)
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Hi,
We using pbxnsip v. 3.0.0.2998 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!
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Is that call being redirected somewhere? Does the final transferred call show up in the CDR?
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.
Pickup issue
in Web Interface
Posted
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: 3.4.0.3201 (Linux)