Vodia PBX
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I am setting up my first Polycom phone with pbxnsip. Have used Snom before. On pbxnsip I am using both the SIP IP Replacement List and IP Routing List. This enabled good calls both internally and calls from clients connected from the internet using xLite. Ive setup the Polycom 330 and is currently logged into pbxnsip. When making calls though, no audio in either direction is heard. When I remove both lines (SIP IP Replacement List and IP Routing List) audio is restored, but only one way with outgoing audio working. I need the tow line (SIP IP Replacement List and IP Routing List) as this resolved some issues I had where when hanging up a call did not terminate the other side etc. And it works fine with the Snom.
Am I missing a setting that needs to be added to the Polycom?
I believe that the replacement list is not 100 % clean. Probably the snom is not so picky with the Contact headers like the Polycom, and the conversation works also with this flaw. You will see if the PBX SIP messages contain a Contact header that can be reached from the phones.
The other thing that would come to my mind would be the RTP port range which could be different between snom and Polycom.
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What's wrong with this statement to alter TLS email sending on a CS410 with this version 3.0.0.2993 (Linux)
We thought 3.x supported this option or must we update?
localhost/reg_status.htm?save=save&smtp_starttls=TRUE
or
localhost/reg_status.htm?save=save&smtp_starttls=FALSE
The symptom we see is the browser goes to a unknown page.
"TRUE" is not "true"... It is case sensitive.
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1. UPTIME - since last restart - Minutes-hours-days
(We would detect this value dropping to near zero as a indicator the system has rebooted for known or unknown reasons) Plus as a an indicator in our client reports. (A Restart email could do this too, for those not using a NMS)
Okay, added as OID .16.
2. TRUNK CALL COUNTER (RESETTING THIS VALUE during midnight processes)(We would set thresholds that would detect no changes in this value in a preset period as a possible error state)
14.13.1 could be daily call count
(Add a Total Call Count with the nightly CDR email)
That is not so easy... Workaround: If you want to check if there is traffic on the trunk you can also use the trunk calls OID.
3. TRUNK REGISTRATION STATUS 1 = registered 0 = Not Registered(14.13) is call counts for trunk 13, (14.13.2) could be status
Yes, added as OID .15.x. It will return the last registration status (e.g. 200 for "200 Ok"). See http://wiki.pbxnsip.com/index.php/SNMP. A build is available at http://pbxnsip.com/protect/pbxctrl-3.2.0.3141.exe.
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Oh oh. There was a bug in OID.14.x. Please get the latest & greatest from http://pbxnsip.com/protect/pbxctrl-3.2.0.3140.exe.
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is there anyway to set it up in the dialplan that when you dial a 10 digit number it should add the number 1 in front of it?
im also havignt he same problem that when i dial from the missed call log, it doesnt have the 1 in front of it
or maybe can we have PBXnSIP add a number 1 in front of the number when display the caller ID? it should only add it on USA 10 digit numbers (so it wont add it on international calls)
Did you put the country code into the domain settings? That makes the PBX aware about the location and then it tries to be smart about the number representation.
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I try to change the wave file for service flags "activated", "deactivated" now I hear just a beep sound but it's not clear for everybody to hear when it's on or off so I would like to change the wave file to "Activated!" and "Deactivated!" but can't find the beep wave file, can someone tell me where it's located?
Those files are in the audio_moh directory. Check "bi_gong1.wav" and "bi_gong2.wav".
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You are close. I would use something like this
http://$pbx/snom/adrbook.xml?user=$index&auth=md5("$username:$password")You also need to know the index in the users table. I know that was stupid, but at the moment that's the way it is. And of course the hash should be something more "random"...
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Is there any way to restore a pbxnsip configuration from backup without having to restart the pbxnsip service?
the problem we're having is that we're doing some simple failover but when we rsync the config files found in this script:
tar cvfz $filename acds adrbook attendants autocallback button_lists buttons callingcards cdr colines conferences
dial_plan dial_plan_entry domain_alias domains extensions generated hoots hunts ivrnodes lamps messages mohs orbits
pnp_parms recordings regidx registrations schedules shared_lines spool srvflags tftp trunks user_alias users pbx.xml
we have to restart pbxnsip on the second server for the changes to come up.
At the moment that is the only way. The PBX is not aware of file system changes in the configuration. We are working on improvements regarding failover; but it will take some time until we have what we want...
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yes i use in my dial plan the 9 to make a call to my isdn gateway but it's just a digits that is not use for the number , in the exemple that i use i want to call to 069 665262
Does the gateway give you any log? Maybe there is some dial plan missing on the gateway. Or it does not trust the IP address of the PBX. On the PBX the setup seems to be fine.
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No ideas?
The only weird thing that I see is that the SUBSCRIBE packets for TAPI are sent twice. Maybe something "smart" on the PC that doubles the packets?
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I'm afraid this is more a feature request than a question...
I have two sites (A e B ) with two pbxnsip, they are connected with a fiber-optic cable, so I don't have any problem in connecting an extension in site B to the pbx in site A, but to be more "fault tolerant" I'd like to use a pbx in each site.
My problem is that I use Hunt Groups and Agent Groups, in site A, that should have site B's extensions in them.
Am I missing something or in this moment it's impossibile to to put a phone connected to a trunk in a group (agent/hunt)?
by the way, is it possibile for an agent logged in TWO different groups (on the same PBX, now :-D) to logout/login to one of the two groups separately?
This second question is because we support two families of products, so sometime I support the first or the last, depending on the day of the week...
If these two things are not possible, as I wrote upper, this is a feature request...
What you can do easily is register the agents phones in both locations.
Having a phone number associated with the agent will easily result in loops. For example, agent A redirects all calls to agent B; who redirects all calls to C; who redirects all calls to A again. This will generate nice traffic storms, and you'll have a hard time figuring out why which agent redirected a call to which other agent. And you cannot stop them from doing this.
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Our XML trunk file is 13, does this overide the newly added 13?
Nono, use .14.x, where x is the trunk number. This OID has one more digit than the other OID.
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multi domain - it is not registrered any where.....
Do you find the trunk in the trunks/xxx.xml files? If the answer is yes, something went wrong with the deletion.
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Version: 3.2.0.3130 (Win32)
Can the once a day graph be scheduled to run every 1, 5 ,10 or more minutes? (testing purposes only) to provide a smaller performance window.
At the moment it is pretty much hardcoded to 240 samples per day (that is every 6 minutes). Regarding the maximum, you don't miss anything - it precisely calculates the maximum within that 6 minute window.
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The great majority of PBXnSIP installations that we would be installing would almost always have just 1 trunk. If they had more than 1 trunk we would still consider that trunk as making calls and knowing total out-in calls vs. total calls that might include internal calls would be best.
Okay, we added 13: Number calls (not call legs) and 14: Number of calls on a specific trunk. The trunk number must be the index of the trunk (to locate it see the XML file name in the trunks directory). For example, if the file name is 25.xml then the OID would be 1.3.6.1.4.1.25060.1.14.25.
Check out http://pbxnsip.com/protect/pbxctrl-3.2.0.3138.exe for a build that has these two additional OID. Would be great if you can verify this.
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strXML &= "<env:Envelope xmlns:env='http://schemas.xmlsoap.org/soap/envelope/' xmlns:sns='http://www.pbxnsip.com/soap/pbx'>"
Maybe it really checks the www.pbxnsip.com URL. There is no document
and maybe that is a problem for ASP.NET. -
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ACTIVE TRUNK CALLS
I am not the big expert on SNMP... Is it possible to pass a string to the GET request? Then the SNMP tool could tell the PBX what trunk it is interested in. That would simplify the setup. Otherwise we would have to assign a OID to every trunk. That would not simplify the setup.
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Look at this one P 71-74 This is an extract of the manual
Okay, that helps.
Obviously, one way to use it is to set it up to the corporate Active Directory. You don't need the PBX for that.
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I see in the general settings the Inband DTMF detection: ON;
But when I tried to do an inbound call with Inband-DTMF, the pbxnsip cannot detect this.
What could be wrong? Also I did Log Level 9 for general logging and log media events ON, but can't get the logs I would like to analyze for Inband-DTMF.
When I used DTMF-rfc2833 on my device, the DTMF works! and I can get some logs (example: [6] 2009/01/16 10:02:46: Received DTMF 1 ).
A difficult topic. I believe if the user agent advertizes RFC2833 (the new number is actually RFC4733) the PBX has no motivation to burn CPU resources on analyzing it. Yes, you should see "DTMF: Power:" on log level 9 (Media).
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The secondary server will have all of the same information the primary does. Should I use the same IP address? That way if server 1 fails I would power on server 2. But the phone would all need to register again, correct? But a feature in a new reales to make this easier would be great, especially as we try to roll this out to mission critical places.
Regarding the IP address there are esentially two possibilities. The first is to use DNS SRV entries; in this case it is the SIP device's job to locate the secondary server. The second is to give the new server the same IP address; this avoids the requirement for DNS SRV. However it has the problem that this setup can easily result in an IP address conflict - espcially when the primary server restarts and the old IP address is still active (which is very probable in the case of a sudden shutdown).
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I know this has been discussed with hunt groups. but I ran into a call center who wants to have remote ACD agents via PSTN. so they login and the queue calls the PSTN extension and the agent presses 1 to accept the call.
Ran into a local MLM wanting to do this.
Yes, that is on the list.
I am sure that once that we have the feature people will try it out, have a lot of problems, they will pick up calls and nobody connectd (because someone else picked the call up), their phone bill will explode (talking to a lot of mailboxes all the time and waiting for the "1" to connect the call), cell phone mailboxes will be full, their lines will be busy all the time (because one incoming call triggers multiple outgoing calls), the QoS to the ITSP will be hell (because practically nobody has a call limitation) and so on.
But if customers are demanding it, we will put it in. Just want to keep the expecation level reasonable.
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the only information I can find is from LinkSys web Site at:
http://www-ca.linksys.com/servlet/Satellit...d=1519652913B14
Hmm. A search for "LDAP" does not return a result...
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I entered them in the following ways, tried many different ones...
708;709
email@address.com;email2@address.com
708;709;email@address.com;email2@address.com
They are fine. So you are saying other emails in the domain are working fine? What version and OS are you running? I remember there was a bug in this area, maybe you just need a newer version.
The second part is after you schedule a meeting you can pull up the Meeting file, the one Outlook receives once the email is working... It has the proper schedule time, and in the body of the appointment it has the conference Access Code... But in the description it just had the extension if I left the Tel Alias blank, and if I had the Tel Alias added it just had the Phone number... I wanted the appointment to have both the phone number and the extension to dial so the Appointment has all the information the people need to call the conference...In version 2, the PBX will present the tel:alias number. This number is supposed to be the number than outside parties dial. The internal number can be "embedded" in other places, for example the conference name (okay, that's a workaround).
and maybe that is a problem for ASP.NET.
330 no audio
in Polycom Phones
Posted
If you have a public IP address and the possibility to put the PBX into a DMZ (transparent, no NAT) then such a setup is a lot easier. You can just configure one NIC for this public interface and the other NIC with a private IP address.