Vodia PBX
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We have a user that is getting 2 emails when he misses a call. When he gets the voicemail in his email that is only 1. No one else is having this problem and all use our smtp info. Any ideas would be great.
You mean the same email twice? Or one email for the missed call and another one for the email CDR?
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Thanks, but not quite sure what you mean by username@domain form for the account.
Sorry that was just the forum editing tool that tricked me. The text should be just username@domain, nothing around it.
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This happens every couple days--when someone uses DND.
This is on a snom360.
What can i do to stop this from happening?
Are you using the button "dnd" for this with the mass provisioning?
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I've been burned by routers in at least 3 and possibly 4 out of about 10 locations. Do you also see such a high percentage of "will never work" routers? I've lost 2-3 hours with a linksys ADSL/wifi router and 1-2 hours with every other.
Is there a list "best practices" for router configuration than I can follow before giving up?
Lastly for people that ask me "what router should I buy to play nice with SIP?" is there some feature they should be checking before buying?
I just disposed my private router collection. I tried to make a statistics on whoich router is using what type; but in the end the result was: Every router as it's own type. There is a interesting IETF document at http://tools.ietf.org/html/draft-jennings-...test-results-04 that tries to classify routers. IMHO just another reason to finally give up on NAT and move straight ahead to IPv6.
Or use TLS and encrypt the traffic between the phone and the PBX so that no "smart" routers can mess the SIP packets up! Security against vendors that want to do you something good, but unfortunately are unable to do so from their skill set.
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and this will not work as the BOOTROM cannot authenticate?? But the SIP app can and does auth properly so it can reprovision the config on a ALREADY working phone correctly but not on a virgin.
Whow... I can't try this out right now... That would explain the problems. Does that apply only to "older" bootloaders or also the the latest and greatest? Can someone verify?
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I am trying to auto provision a snom 360 with firmware 7.3.14. My phone has a DHCP address of 172.16.x.x my server has na ip address of 172.31.x.x The server sits in the DMZ.
DMZ - are you trying to provision the phone from the "outside world" or from the LAN? The PBX has to generate a lot of links for the various files that have to be downloaded during the provisioning process. If it just puts 172.16.x.x there and the phone is somewhere at home behind a router that address cannot be resolved and we are talking about a tricky routing setup here.
I can see the phone requesting the file snom360.htm. I have the file below named as such and it is in the tftp and html directories. How ever each time the phone requests the file I see an authentication error. In the PBX under settings-->PNP I have the snom admin password set to 1234.That is just the PIN code if you want to switch from user mode to admin mode on the phone. Has nothing to do with the provisioning itself.
Then under the domain settings--->provisioning parameters I have the password set to 1234. In the phone under security I have the password set to 1234 with a blank username, I have also tried using my account username and password here as well, no dice.The username must identify your account. if you have multiple domains, you must use the form user@domain; if you have just one domain then just user may also work (play safe and use the user@domain form). The password is the web password of the account; or you can put the domain provisioning password there. Play safe and use the user web password here...
Under the update section on the snom I have it seto to update automatically. The setting URL is http://172.31.x.x/provisioning/snom360.htm Subscribe and PnP config are both off. And I get an authorization error in Wireshark everytime. Any help would be great.If you have an authorization problem, then you should focus on the right username and password. Also if you plan to use DMZ, then try it out first without DMZ in the LAN.
Also where is the proper placement of this file! Thank you all and by the way we are running pbxnsip version 3.3.1.3177 (Win32)Don't place this file anywhere. Also if you have a pnp.xml file, move it away. Also for the sake of getting it working, move PnP files away that you might have in the html directory. After you get it working, you can move them back one by one; then you might see which one causes the grief.
Sorry for all this trouble; but the 7.3.14 firmware is a little bit "bitchy" if you want to provision a phone without major security holes.
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If the caller wants to call an extension, it is the best choice to use a extension for this on the IP-PBX. Then all the features that do exist for this application work nicely. The only workaround is that you cannot use a "native" SIP phone, but use a PSTN gateway to call the extension on a legacy PBX. So far so good.
The first catch is the setup. If you have hundreds of extensions, setting that up can be a major problem. But if you take a look at the XML files in the working directories it should be relatively easy to copy them into other extensions with a shell script.
The other catch might be licensing. If you have 600 extensions then that solution can become more expensive that a single trunk setup. But I would say this is a business problem, and you should try to make a special deal for this case.
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I am trying to setup my agent group to failover to an external answering service of ours. I input the full number to route the Q caller to after 30 seconds, but when that timer goes off the caller receives an operator message stating the call is prohibited. How does one route a Q call to an external phone number properly? Thanks a ton.
It could be that the dial plan that you assigned to the agent group does not permit to place such a call.
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Is it possible to have an export of the Domain Address Book ?
I would like to have this option to be able copy the contents from the Adress Book to another running installation.
Nope, the only workaround is the copy & paste from the web interface and re-format it to CSV format. The other possibility would be to clone the whole domain and then after restoring on another server delete what you don't want to carry over.
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We are using the PBXnSIP hosted platform (ver 3.3.0.3165) and I am trying to configure SNOM handsets buttons via the PBXnSIP interface (domain settings/buttons).
Can anyone tell me if this feature works on the the hosted platform and what needs to be set on the PBX and handsets to get this working?
Yes this also works. In this environment you must use the username@domain form for the account. The password can either be the web password for that user or the domain provisioning password (if set).
The URL that you have to put into the phone must be in the form http://pbx-address/provisioning/snomxxx.htm (no MAC address).
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It is unbelieveable what kind of devices are out there in the router market. I have seen routers that have no expiration of NAT ports, and they just use 32 ports in a FIFO fashion. That router in a busy office was a nightmare troubleshooting why some phone calls just wont work!!!
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You actually have to do more if you don't want to lose the recordings for the extensions that have DIDs in them. If you just clone the domain and use it when creating a new one you will have to re-record everything over again. I've figure out that you can remove the DIDs from the original domain and then clone and put back in the new domain the DIDs to the extension and all records don't get erased. This is because the system will not allow you to create extension with multiple DIDs and therefore it would strip them when importing the template.
I worked a couple of hours on playing with this and got it working this way with out having to re-record all greatings and AAs.
You are right. DID are a problem. Because they are global, they get lost when you want to restore them in a domain.
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not related to hunt group...but i've thot it would be nice to have the music on hold with the marketing messages mixed (like agent group) for normal on hold stream.
how hard/likely would that be?
Hmm. You have a point here. Maybe the way the MoH works right now should be changed. The MoH source itself should already insert the messages, instead of having the ACD do that. Then you could pick the stream that you want for the occasion that you want.
Would not be too hard (I believe). But would be a change that is not backward compatible and that is always causing an uproar.
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so, if remote phones do not authenticate based on MAC. does that mean for remote phones we no longer need to put the MAC into the provisioning tab? dont know why you would if you are now using http auth.
Right. If you do HTTP authentication then there is no more need for the MAC. The only point of the MAC is to group extensions, so that it is possible to provision multiple extensions on the same handset. Only if you want to do that you need to put the MAC into the extensions that should be provisoned together.
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has this been added yet? with the option to press 1 and accept the call...
Not yet. But is a core component of the next major release.
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I would either clone the whole configuration of the PBX (if it was a single domain setup) or get a domain backup and then restore the config on the new server or create a new domain from the domain backup. Then after the restoration, delete what is not needed any more.
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what firmware is now used for 3.3.1?
snom: 7.3.14
Polycom: 3.1.2
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I have a dhcp server in both vlans and the server is in the vlan that I have tagged the phone for.. I don't think the pbx is in both vlans though??
Yea, VLAN setup is still very tricky. The problem is that when the phone boots up (after a factory reset), it is not in any VLAN and needs to get there. If there would be a way to tell the phone to switch into a VLAN then things would be straight forward. Maybe LLDP is the answer?
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My firmware is 7.3.10a.
Can you upgrade one phone to 7.3.14 and see if that changes the picture?
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I am trying to find a way to send a different outbound caller ID only when someone dials 911. According to Nexvortex, the way that they do E911 when you have multiple E911 locations with them is based upon the ANI you send out. The problem is since they require a SIP registration, the only way I can change the ANI is to send it out a different trunk, however I only have one trunk. The other thing I had thought about is to have it go through another PBXnSIP machine, however I would prefer not to need an additional machine to manage, and have found that as you get multiple PBXnSIP registrations to eachother, performance tends to degrade quickly, so I want to avoid that at all costs. What is the best way to acomplish this?
In 3.3 we introduced the "EPID" (endpoint identifier). When someone calls 911 on a emergency trunk, then the PBX will put the EPID into the contact. When the emergency center calls back, the call goes directly to the extension. even if there is usually an auto attendant in the middle.
This goes beyond what good old TDM was able to do. VoIP and SIP can do more than emulating circuits.
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Is this in the works?
Not yet, we are still collecting items for 4.0!
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I remember at some point last year, when I initially bought and configured my CS410, inbound caller-id was working. Not anymore. All calls on the call log appear as from "Anonymous (anonymous@localhost)"
I connected a phone directly to the CO line to test it out, and called-id is indeed active and working perfectly, yet my CS410 is not grabbing the info.
Any ideas what to look for?
Running the latest (3.2.0.3143)
What country are you in and what carrier are you using?
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A customer is trying to get pbnxsip working with OCS and the mediation server for a project. He is getting calls working in both directions but he does not see the presence information or the call status (if he is busy) the OCS does not see this. So if you pick the IP phone up and make a call to the voip provider the OCS softphone does not see this. And if they are away on the counterpath softphone it does not send this information to OCS. Is it possible to do this with pbxnsip?
Unfortunately, presence is not presence. Microsoft uses their own way of presenting presence, which is AFAIK now compatible to the presence that counterpath is using. The PBX in the middle does not translate the presence, it does not even generate the presence.
As OCS is becoming more popular, we'll have to see if the PBX can "synthesize" presence information and pass it on to OCS.
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I am not looking to do major recording on it, just the occasional call. Can you give some more information how I could setup basic recording through the system?
I am just trying to see if there is any way I can get occasional recording to work or I guess I will have to look into a new system if I can't do it through the CS410 at all.
From the CPU perspective there is no difference between occasional and permanent recording. You don't want to have stuttering voice when a user records a couple of seconds; and you can never know if several users press the recording button at the same time.
I would look into a new system...
Limit calls using dial plan
in Dial Plan Setup
Posted
That looks okay... Try turning the logging on and see how the PBX processes the dial plan when the user dials a number. Maybe wrong assignment of the dial plan to the user or maybe the wrong trunk in the dial plan.