Vodia PBX
-
Posts
11,130 -
Joined
-
Last visited
Content Type
Profiles
Forums
Events
Posts posted by Vodia PBX
-
-
Looks like you are using the wrong boot rom? Should be something starting with 4.
Also, if you are using the phone from a remote location, make sure that you are using HTTP. TFTP is not very NAT-friendly!
-
It is ringning for 2-3 minuts. - correct.
It drops out when calling BUT also when in the queue
If the phone is ringing for so long it becomes a major problem. Most phones don't ring so long and they send a response back that looks like the user rejected the call. I would suggest to escalate the call after 90 seconds into another queue or even to a hunt group or a cell phone. Eventually someone has to pick up the phone!
-
We have customer with only two agents - and sometimes, they have a lot of calls.
When people are waiting in the agent group, after apx. 2-3 mins. the call dropped.
Does the agent's phone ring for 2-3 minutes? Or do callers drop out while they are in the queue?
-
Hello. We have 7 offices with PBXNSIP on all of them and we have them interconnected. We have most of our PSTN traffic setup to go out our Audio Codes Mediant 1000 which has 2 trunks. One with a mexican carrier and another with a US carrier. Each carriers deleivers 100 DID's in both countries and they match to 1500 - 1599. We have our Remote Party/Privacy Indication: on the trunk set to P Prefered Identity in order to have our ANI's displayed correctly as we do alot of call forwarding of some of the DID's to different mobiles and land lines and we need the remote party number displayed. It has all been working great for sometime know.
Just last week we came into the need of having our user dialing into calling card DID or auto attendent DID and having them dial out via the PBX for cost and control reasons. We noticed that the called party was being displayed the orginal ANI which was coming from a mobile or remote land line. In order for us to get PBXNSIP to respect the ANI of the extension being used we have change the Remote Party/Privacy Indication:on the trunk to P Asserted Identity. With this the 2nd scenario works great but we loose the first scenario.
Is there any way to have more control over Remote Party/Privacy Indication: on a trunk? Maybe something where we can control via the extension/DID?
Remote-Party-ID is a old, obsolete Cisco proposal for indicating the remote party. It expired a couple of years ago. Everyone in this industry should better stop using it. We are too nice and we still offer it. I believe even Cisco does not offer it any more in the latest versions.
P-Asserted-Identity and P-Preferred-Identity made it to the RFC state. This is the way to go. AudioCodes as one of the better products in this industry fully supports it.
Regarding the cell phone redirection, there is the old problem: How can the gateway "trust" the PBX that this is not a spoofed caller-ID? Especially for ITSP that is a huge problem. Because of this, we introduced a new header called "Related-Call-ID" that indicates the original caller-ID. Not RFC yet, but at least CallCentric said they are going to support it and then the original caller-ID will show up on the cell phone. Bravo!
-
Hello. We are seeing an issue with the Caller ID match when a call comes in to the auto attendent. We have our mobile phones setup on a user extension profile with a prefix infront of the number so it can fork the call to the mobile number. the problem is that when you match the caller id of the incoming call you dont recognize the number because of the prefix. if i remove the prefix it works fine for the incoming call to get authenticated but the forking stops working.
all our call have either a prefix of 8 for USA calls which send to our audio codes mediant 1000 for the specific trunk to the USA and a 9 for mexico.
example of mobile number 7631599 but we have setup as 87631599 on the mobile setting.
i know we could do manual routes on the dial plan to add the 8 infront but then that makes us modify all our dialplans.
my suggestion is to read the caller id from right to left until you match the 7 digits and that it could be configurable per the system admin.
That algorithm does not work properly. For example, there are people in this world that have the same cell phone number like their home phone (e.g. +49-171-8028055 and +49-30-8028055).
If you are in USA, use the country code "1". That will tell the PBX that the caller-ID can be 10 or 11 digits and it will automatically concert it into a global format. If there is no country code the PBX will not change the numbers and use them literally. Then it is very likely you are running into problems with the matching.
Also on the trunk, you can tell the PBX how the gateway or service provider wants it. Then also for incoming calls, it will convert the number into the right global format.
Make a backup before you change the country code. The code changes a lot, also changes the address book entries. If something goes wrong, you want to be able to revert to the previous state.
-
I am still having this problem... so does anyone have a fix? on the phones, i have ' answer after policy' always.
enable intercom on
i don't know what these settings were prior to latest firmware update.
this was a very nice feature. Can someone please help?
I would factory-reset the phone and provision it automatically. There is no need to change the configuration; the PBX will signal the intercom feature in the INVITE.
-
I have download and install new 3.3.0.3147 version and so fare I loe the new features but I found that something is not working related to cell phone transfert.
Incoming tranfer call to cell phone is not receving audio on the cell phone but cell phone transmit audio no problem.
I went back to 3.2 and everything seams to be OK.
Hard to say what the problem is. Only a trace would tell. But 3.3 also has a new feature that locks the codec, that usually helps to sort this kind of problems out.
-
Is there any way that you could setup PBXnSIP so that it will allow calling between 2 trunks. I am Interested in allowing one branch office to connect through it's sip gateway trunk to the main office's PBXnSIP, and then out the main office's Sip Registration trunk to the ITSP. This would be possible if you could assign a dialplan to the Sip gateway trunk that links the PBXnSIPs together. I know it is possible by using a sip registration from the remote office to the main office as an extension. However PBXnSIP is very problematic when doing this, especially when it comes to sending callerID, and the short registration time that is necessary, leads to excessive load, and poor call quality on the main office's server.
One key might be that you can register the branch office to the main office; then assign all DID numbers to the ANI field in the registered extension (seperated by space). Then when the branch office places a call, the PBX would pick the matching ANI for the outbound call.
For inbound, you can assign the DID numbers as alias names in the extension in the main office.
-
But what is the meaning/function of this particular item?
"When the allocation of a new CO-line failed and a call was rejected because of this:"
If you have CO-lines on a trunk and the PBX would like to use this trunk for an inbound or outbound calls, but all CO-lines are already busy and the call has been rejected.
-
I am wondering what version will be able to utilize multicore technologies so we can have more than 75-80 concurrent calls per server. We're wanting to be able to have at least 250+ before having to deploy more servers.
Also looking for in the future would be able to manage all domains from one management interface whether 2 or more servers. It would be nice to have the directory stored on a SAN that multiple servers can pull from and also making the service load balance.
Don't know if PBXnSIP is looking to do this kind of development or stay more towards the smaller ITSPs.
As long as you are able to split the user group up into domains with 75 calls per domain then you can do that today. You need to run several processes; each one bound to a seperate core (leave one core for the OS). Each process has a different working directory. And you either bind each process to another IP address or you use different ports. In Linux/BSD this is relatively straightforward. In Windows, you cannot use the standard installation procedure but must use another mechanism to set the services up.
In any case, the key is to be able to split users up into smaller pieces. Then you can scale along cores and servers.
-
What are the chances for an upgrade? Maybe the problem is already "automatically" fixed in a newer version.
-
Using the latest firmware of the audiocodes mp-118 fxo and latest copy of pbxnsip i was able to get everything to work as it should except for one glitch. It seems the volume levels are really low even when i up the voice settings in the audiocodes. if i up the settings too much, i get static. What else is unusal is when i dial out, the first ring is extremely loud, but the rings after the 1st one is very low and it also seems that when the second ring starts that is when i get connected. any ideas on what i can do?
I remember Bill H recommended to use an analog measurement tool to se what is going on. There is also other gear available that amplifies the signal.
-
if the extension your trying to record the message is 5002 just dial *985002*0 etc
I you require that users have to enter their PIN code when going to their mailbox, then they also need to punch in their PIN code when they want to record a call. Then they must have a PIN code - no PIN code, no recording.
-
Today, a user had the "stuck" MWI issue. She had saved a message but did not delete one so the MWI sticks on, occasionally. No messages existed in any group she was in. *99 did not clear it. So I sent her a test voicemail, instructing her to delete it as this has cleared stuck MWI in the past.
As soon as I sent the message, the MWI cleared (good but unexpected). Email was generated about the voicemail I left (good). However the email states that the message length was 0 seconds (bad) and no voicemail was available (very bad).
This is not an easy thing for me to duplicate as MWI sticks only rarely.
Running 3.2.0.3143.
Consider moving to a 3.3 build; there were cases when the speed of the file system caused problems with the mailbox message.
-
I see the problem here.
The CS-410 is using its local LAN Address (192.168.1.99) in the SDP and that is an unroutable address from the remote stations standpoint.
I have resolved this problem with other products using STUN.
The customer has only 1 Public IP Address.
This is from the WIKI:
If you don't have a second IP address things are getting a little bit trickier, but you can still deal with this.
You can add an additional IP routing entry in the PBX service (not in the operating system) that will tell the PBX that routing to a specific IP address will result in a specific IP address.
This is done in the setting "IP Routing List" which is just below the "SIP IP Replacement List".
Is this something I should be seeing in the CS-410???
---------------------------------------------------------------------
So it sounds like I would plug the LAN of the PBXNSIP into the router and give it STATIC 192.168.1.99 / 255.255.255.0 / 192.168.1.1 for IP Assignments
and also plug the WAN of PBXNSIP into the same router and give it STATIC Public IP Address / Public IP Addresses SNMask / Public IP Addresses Router asignments.
I would not use STUN. It works in many cases, but not in all cases. STUN is extremly support intensive; there are so many routers out there that have different implementations of NAT. Some of them work "most of the time", then when your customer calls you you spend hours on the phone trying to figure out what the problem is.
Also, running the PBX on a not-routable IP address creates a lot of grief if you want to register phones to that address. Yes, there are "tricks" to get this working as well (described in that Wiki page). My bottom line is: If you want to connect remote users, get over it and get a public IP address.
Fortunately, most service providers understand that trunks mostly come from private IP addresses and they offer some kind of SBC service. So if you register a trunk to a service provider, you usally don't need a public IP address. I assume here that the service provider has a routable IP address!
On day when IPv6 is out, we'll all laugh about these problems.
-
anyone using snmp with either op manager or prtg?
if so, any input on getting the snmp to work and read the mibs?
We are using PRTG. Setting up the OID did not require a MIB. Pretty straightforward; though the version is already a couple of years old.
-
I am using the Park and Pickup reminder and it works nicely , but the customers want to be able to tell between a new incoming call and a Pickup reminder without having to rely on caller ID on the phone ,
would it be possible to add another option for different ring type or something to indicate when park reminders are being issued , .. mainly looking for Snom or Polycom usage ,
Since park and pickup is getting more sophisticated a new topic may be in order ,
Hmm, interesting. We need to put the whole ring tone topic on the agenda for version 4, including the callbacks (also for camp on).
-
I`ll check and i have no file with --busy-det....
do you have a other solution??
In analog PSTN lines, the other side play a tone to signal that the call has been disconnected and the subscriber should disconnect the call. More than hundred years ago people were happy when they could hear the other party.
Today, we have tones coming from such strange sources like cars (when you open the door). The PBX might believe that this is the disconnect signal! We also had cases where the carrier was playing such useful hints like "your call has been disconnected, please hang up". You need real intelligence in a PBX if you want to deal with that.
Proper hang-up detection is really difficult in the analog FXO world of Alexander Graham Bell. We tried to put some intelligence into the FXO trunk on the PBX side (check "trunk requires hangup detection"). If you feel it is too agressive, consider turning it off. However, then callers might hang in the auto attendant until the hangup-timeout kicks in.
I love SIP trunking. Reason #1 is a clear disconnect signal.
-
I am connecting to a CS-410 as a remote station.
I can see the remote stations Remote IP Address (Ex. 71.211.22.33) on the Account Registration page
At the remote station I can call a cellphone and they can hear me, but I can't hear them.
When I looked at the SIP Trace the Session Discription Protocol shows the remote device with the Inernal IP address of the PBXNSIP not the external IP addreess I found.
How do I correct it be the correct External IP Address??
The CS_410 is connect to the customers LAN on its LAN Port.
If you have both a private and a public IP address configured, then you probably have a problem with the default gateway. If you tell the PBX to use a private address as the default gateway, it will send it there and then you will also see the private IP address in the SDP.
If you have only a private IP address things will get tricky. Then you should study http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses. If you don't like what you read there get a public IP address...
-
Ok in my office I have about 62 Snom M3 phones. We just went live today. I have received about 6 or 7 reports of calls dropping in midsentance. The calls are not on mute and I am running a permanent key. These calls go from the PBX to Audiocodes FXO gateways. It is very sporadic. Theres is also no NAT that goes on in this situation. My cpu usage is very low and I am working on getting a wiresharp capture the next time it is reported. I am running version 3.1.2.3120. Does anyone have any ideas?
You should definitevely turn on the email reporting of such events. Then you get an email when a gets dropped by the PBX.
Also, you better use version 3.2, 3.1 has a ugly bug in the web interface.
What you can do is writing a log file of the SIP traffic (only "other" message types) to the file system. Don't write all the other stuff, it just makes the system drown in messages. In the BYE message, the PBX reports how many packets have been sent and received. If there is something out of balance, then you get a hint.
In the 3.3 version we added sending of email messages when the user hangs up during a one-way audio situation. If the problem persists, you might have to (temporaily) move to version 3.3 to get better information. Make a backup so that you can move back to the 3.2 version after the problem has been identified!
-
We use Firefox exclusively.
Any chance to quick try to IE? Just for the same to find out if that is the problem.
-
So why is the codec list on the admin/ports web page empty? And how do we populate it?
It should not be empty! What browser are you using?
-
We are using hot desking *70 on a multiple domain environment. Using extension 116 and with *70 login as extension 720.
We nog get reactions from a customer from another domain that the name of her extension, which also 116, has changed to 720. Also it happens that when this person calls internally the calls are routed to our domain instead of the domain se works in. It doesn't happen all the time. This is also what the call log shows me.
Hot desking had a bug in 3.1.1/3.2.0 with multiple domains. If you need this feature you must use one of the head builds (what OS?).
-
I have a question on the codec settings field in the System Admin >> Ports >> RTP >> Codec Preferences. The list here is not populated with codecs and does not give you the ability to do so (see screen shot attachment). How does this field relate to the codec settings in the pbx.xml file, namely <codec_preference>0 8 9 18 2 3</codec_preference>?? Are the settings in the pbx.xml file used even though they do not show in the System Administration Web Interface? Are the codec settings that would be listed in the System Admin >> Ports >> RTP >> Codec Preferences fields downstream, upstream or both? Also how are these settings (pbx.xml) different from the Override Codec Preference settings in the Trunks?
The new codec selection field is just a glorified JavaScript input form for the gool old list. If you look at the pbx.xml file, you will still see the old list of codec numbers. The change was made to make it easier to select the codec preference.
If there is no codec selected (e.g. on the trunk) then that means that the system default should be used. Only in the admin/ports web page there must be at least one codec selected (because that is the default codec preference).
HP ML series, FreeBSD and XenServer
in FreeBSD-related topics
Posted
If you can, avoid virtual machines. They might end up in resource conflicts and yes then it is not clear who gets the CPU.
Though I heared there is some progress on VM. I would use it in a test setup, but I don't feel safe in a deployment yet.