Vodia PBX
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i am using a PRI card with pbxnsip latest version but i have an issue where a caller calls in and does not hear any audio untill the caller picks up or it goes to voicemail...im assuming it is not playing the ringback tone but i checked the audio folder and the ringback.wav file is there.
Some trunks cannot deal with early media. Check if you have the flag "Ringback" set to "Message 180".
You can easily see if the PBX sends media by using Wireshark. That should further help to nail the problem.
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After about 3 hours of testing we figured it out. We had other IPs that where on the NIC so our data center could ping all of our IPs per their requirements. For some reason pbxnsip binded incoming audio to the snom phone to that IP but all traffic wasn't suppose to be on that pictular IP. We removed and everything was back to normal.
Just updated http://wiki.pbxnsip.com/index.php/One-way_Audio.
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%GMT_TIME_ZONE%:{tz gmt-hour 13}
Try changing the "13" to "12". Maybe that was the problem.
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No dice. I have found the section edited it and I am still an hour off. My time is 1:02pm and the phones say 2:02pm. I have daylight savings time enabled on the pbx but something is still wrong.
Maybe just take the below file and modify the time section (call it snom_m3.cfg). That should fix your problem for the moment (create and put it into the html directory).
//-----------Network Settings------------
%SIP_RPORT_ENABLE%:1 // A way around NAT
%SIP_STUN_ENABLE%:0 // 0-disable, 1-enable stun services
%NETWORK_STUN_SERVER%: "stun.com" // insert name of STUN Server here
%SIP_STUN_BINDTIME_GUARD%:80
%SIP_STUN_BINDTIME_DETERMINE%:0
%SIP_STUN_KEEP_ALIVE_TIME%:90
%SIP_RTP_PRIORITY%:{global tos_rtp}
%NETWORK_VLAN_ID%:{parameter NETWORK_VLAN_ID}
%NETWORK_VLAN_USER_PRIORITY%:{parameter NETWORK_VLAN_USER_PRIORITY}
%VOIP_SETTINGS_PIN_CODE%:{parameter VOIP_SETTINGS_PIN_CODE}
//------------Network Time---------------
%NETWORK_SNTP_SERVER%:"{tz ntp-adr}" // insert a local NTP Server here
%NETWORK_SNTP_SERVER_UPDATE_TIME%:0xFF //SNTP Update in seconds
%GMT_TIME_ZONE%:{tz gmt-hour 13}
%DST_ENABLE%:{tz dst-enable}
%DST_FIXED_DAY_ENABLE%:{tz dst-fixed}
%DST_START_MONTH%:{tz dst-start-month}
%DST_START_DATE%:{tz dst-start-day0}
%DST_START_TIME%:{tz dst-start-hour}
%DST_START_DAY_OF_WEEK%:{tz dst-start-wday}
%DST_START_WDAY_LAST_IN_MONTH%:{tz dst-start-last}
%DST_STOP_MONTH%:{tz dst-stop-month}
%DST_STOP_DATE%:{tz dst-stop-day0}
%DST_STOP_TIME%:{tz dst-stop-hour}
%DST_STOP_DAY_OF_WEEK%:{tz dst-stop-wday}
%DST_STOP_WDAY_LAST_IN_MONTH%:{tz dst-stop-last}
//--------Provisioning Server------------
%NETWORK_FWU_SERVER%: "provisioning.snom.com" // Firmware update server IP or FQDN
%FWU_TFTP_SERVER_PATH%:"/m3/firmware/" //subdir to find Firmware updates
%VOIP_LOG_AUTO_UPLOAD%:0 //0-no uploading, 1-upload bootlog only,2-upload everything
{loop-start 0 8}
// ------------ SIP Server ---------------
%SRV_{lc}_SIP_UA_DATA_DOMAIN%:"{domain}"
%SRV_{lc}_SIP_URI_DOMAIN_CONFIG%:0
%SRV_{lc}_SIP_UA_DATA_SERVER_PORT%:{sip-udp-port}
%SRV_{lc}_SIP_UA_DATA_PROXY_PORT%:{sip-udp-port}
%SRV_{lc}_SIP_UA_DATA_SERVER_TYPE%:0
%SRV_{lc}_SIP_UA_DATA_SERVER_IS_LOCAL%:1
%SRV_{lc}_SIP_UA_DATA_REREG_TIME%:600
%SRV_{lc}_SIP_UA_DATA_PROXY_ADDR%:"{ip-adr}"
%SRV_{lc}_DTMF_SIGNALLING%:2
%SRV_{lc}_SIP_UA_CODEC_PRIORITY%:0, 1, 2, 4, 255
// ------------- Registration -------------------
%SUBSCR_{lc}_SIP_UA_DATA_SIP_NAME%:"{account}"
%SUBSCR_{lc}_UA_DATA_DISP_NAME%:"{display-name}"
%SUBSCR_{lc}_SIP_UA_DATA_SIP_NAME_ALIAS%:"{account}"
%SUBSCR_{lc}_SIP_UA_DATA_VOICE_MAILBOX_NAME%:"{account}"
%SUBSCR_{lc}_SIP_UA_DATA_VOICE_MAILBOX_NUMBER%:"{account}"
%SUBSCR_{lc}_UA_DATA_AUTH_NAME%:"{account}"
%SUBSCR_{lc}_UA_DATA_AUTH_PASS%:"{password}"
{loop-end}
// ----------Emergency Primary Line----------------
END_OF_FILE
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Is there anything i can try?
Well, we need to get the initial seed for the random number generator sorted out. Depending on your operating system the answer will differ. For example in Linux, usually the OS will inject some randomness (e.g. by measuring the interrupt time). Maybe we also need to do something like taking some configuration-based data to add more randomness to the system; but first lets try to find out what exactly is causing this surprising non-randomness.
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Since the upgrade to 3.1.2.3120, we have had an issue with the vm message indicator not shutting off. There are no new messages, even after a *99 the light will go off for a while then all of a sudden start blinking again, it makes everyone check their vm's over and over again.
Make sure that all user agents subscribe for the MWI event. The PBX tries to do a favor and send MWI also out if there is no subscription. But as soon as one registered device has a MWI subscription it sticks to the subscriptions. I guess this is a topic that we need to clean up (no more favors).
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i have two PBXnSIP server trunked together for a company that has one office in NJ and one in NY.
although internal callers in each location are able to dial the extension of the other office without a problem i do have a number of issues.
1) when a caller calls in from the outside and reaches the auto attendant in NY and they try to enter a NJ extension it tells them "this extension does not exist" is there any way to setup the possibility of callers being able to do that without a problem?
2) when i tried to use the hot desking feature from one pbx to the other it does not work but when i forward all calls it works fine...is there any reason for this and how (if possible) can i resolve this?
1) Yea, that is a common problem in this kind of setup. As long as you have less than 10 destinations you can use the direct destinations feature to work around this.
2) Hot desking is like changing the registration. The call redirection is sending out new calls. That is why hot desking cannot work across different PBX.
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We are a hosted environment, meaning multiple domains on multiple servers. The main reason behind the migration is that we have found we can have more domains per server on the Linux servers with less impact than on the Win2003 servers. Please see the attached pdf.
Seems we had a similar finding, see http://wiki.pbxnsip.com/index.php/Performa...perating_System.
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Since I changed to V 2.1 I find records like this in the log data:
Dialplan default: Match xxx@_cpempty_ to <sip:xxx@192.168.26.140;user=phone> on trunk voxip
The PBX works fine but I am still curious what "_cpempty_" stands for, as I can't remember to have seen this in older logs.
Hmm. That looks like the PBX must have picked something up from the trunk... You are sure that this was not the domain name??? Anyway, there is no deeper meaning.
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Windows (one is 2000 server the other one is XP Pro)
As a side note, when I originally created the trunk, the problem did not exist. It was not until the first reboot after the trunk was created that the problem popped up.
It is strange that they share the same call-ID. That looks like there is a problem with the initialization of the random number generator.
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Recently our company buyed PBXnSIP 3.0. I have problem with some patterns. For example, I need write pattern for range 9150000000-9154999999. I think that pattern (915[0-4]*) will right. But this rule send only first four digits on trunk.
Help me to write pattern for this range 9150000000-9154999999.
And when I may to read about pattern syntax?
Try (915[0-4][0-9]*)@.* - the match must also include the domain name...
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If I want to use a particular extension like a guest room, is their any simple process to reset completelly the mailbox without deleting the extension and recreate it? Something like an hospitality system!
There is a SOAP request for this (especially for hospitality purposes). Not sure if that is "simple"...
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Whow the Call-ID are the same! Is that Windows or Linux?
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Would you explain the reference to combinging? Does this mean in any service flag field you can place to service flags to monitor? Would you simply place the service flags 701 and 702 in the service flag field? Then if either is set it goes to the Night Account.
Also with so much aligned on 6 minute increments, can we reliably test service flags with time periods less than 6 minutes?
Well, in the night mode checking area, you can list more than one service flag. E.g. you have one manual first, and if that is not set then take the automatic. That could help solving the problem.
Service flags are mit second-precise but reasonable (I would say minute-precise).
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Sanity check to address a lingering anomoly about dynamically resetting a scheduled service flag. The idea to provide a client the ability to easily stay late or to have an emergency phone solution is to send all calls to an ACD with the option #L option to redirect all calls to a normal AA for normal operations. In the event the business want's to be open after hours or the weekends, have an extension log into the ACD group. The would intercept normal operations and provide a fork to open after normal business hours....
Maybe you can combine one automatic flag with another manual flag. The ACD checks the manual flag first; if it is set, the call gets redirected somewhere else (maybe the "overtime" ACD). Otherwise the standard rules apply for 9-5 automatic flag.
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I encountered problem that i cannot create the calling card account, on top of that if i delete any ext , i cannot recreate again please advise, thanks
I already upgraded to latest version which is ver 3.1x
That sounds like a license problem. Anything in the LOG?
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so <tos_rtp>cs5</tos_rtp> would be correct?
No. The value is what the PBX passed down to the OS. It is just an integer in the range between 0 and 255, I believe in steps of 4. A value of "184" is the default, corresponding to the value "B8" (hexadecimal), corresponding to B8 >> 2 = DSCP 2E. Not sure what value CS5 has...
If you look in Wireshark on a IPv4 packet, you see the bits right there. A lot of talk about 8 bits in the IP header.
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It seems odd that pbxnsip goes 98% of the way to OCS integration and doesnt get the presense complete as you would expect OCS to behave (on call presense indication)
Well the question is what should the PBX then do with the presence information?
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how do i set this in linux version?
will the PnP setting send this to the phone as well?
There is a global setting called "tos_rtp". This gets propagated to the phones as well (if the PnP file supports it).
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I want to use an auto attendant, but need other prompts that are not available in the drop down list.
Is it possibility to add the wav files myself and will they be shown or should you replace/rename the current prompts.
If you are a pragmatic guy, just overwrite one of the prompts with what you need. Adding it "in style" would be a lot of work.
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It's actually not the address that changes but the port number. This is a totaly wireless deployment within a resort. Evertime a phone re-registers it is aasigned a new port by the remote WIFI radio which is routing and NAT'ing actually at this point we are double NAT'ing. As such all PbxnSip is doing is saying for example is that Source address for sip:2036@172.32.5.1 has changed to udp:172.32.5.2:31989. This was appearing in the log file but stopped when I turned off all logging selections. I just don't want to receive an email evertime this happens. How can I stop this from happening.
Changing the port number is as bad as changing the IP address. You have a lot of "black spots" in the availability of the device, don't be surprised if you have "random" unavailability of the device.
Anyway, if you don't want to receive those email messages, set "email_address_change" to "false" (see http://wiki.pbxnsip.com/index.php/Global_Configuration_File).
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I'm using Zoiper as a soft phone and dilaing out works great. When I first installed it I was able to receive incoming calls. After a day it stopped working. I have done a network trace on my local pc and don't see any incomign SIP traffic when I call in from and outside line, through a SIP trunk to our PBX.
My next step is to do a trace on the PBX itself and see if there is any traffic being sent to my laptop.
http://wiki.pbxnsip.com/index.php/Troubles..._Trunk_Problems and http://wiki.pbxnsip.com/index.php/One-way_Audio would be locations that I would check.
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Using User Name & Password seems to work. The problem I have now is that I am receiving SIP Address Change notifications even though I have logging turned off.
Those messages are serious and a indication something is not right with that extension. If the address keeps changing probably the router is not suitable for stable VoIP. You can turn the notification of with a special setting, check out your pbx.xml file for "email_address_change".
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I think I must be missing something that would fix this issue. I want the ability to record calls on any extension I choose on a per call basis. I want to start recording before the call begins and end it when the call ends. I do not want to record all calls to or from an extension. This seems very simple so I think I am missing something in the setup. Right now I can get it to record all calls so I know the recording feature works.
For per-call basis the user needs to start the recording from the phone. You can either use a special code for that (see the feature codes in the domain) or use a special key on the phone. Then the recording does not land in the file system, it lands in the mailbox as if someone left a voicemail.
NO RINGBACK TONE
in General Setup
Posted
Looks like you need Wireshark to see what is going on.