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Ryan

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Posts posted by Ryan

  2. Call recording location / which user did what

    in Call Recording

    Posted

    Hello,

     

    I have my recordings saved to /pbx/recordings. In there, of course I have a bunch of .wav files which are the actual recordings. I'm going to be making a PHP script to be able to play back a specific recording from a specific number. How can I find out information about a specific .wav file in my recordings directory, and who/what it goes with?

     

    Ryan

  3. Cross-Domain calling

    in Trunk Setup

    Posted

    Hello,

     

    Thanks for the response. These domains are actually on the same server. We will have one physical server with multiple domains. With the approach you talked about, wouldn't we have to create a new dial plan with ALL accounts on the server, in order for all domains to know about each other?

     

    Ryan

     

     

    HI;

     

    if both sites have access to the internet there is no problem trunking the two pbx's together.

    all you have to do is create a trunk on each end and put in the public ip address or FQDN in the domain field of the trunk.

     

    the simplest way to set it up is to keep the extension numbers different i.e SITE1 should be 1XX and SITE2 should be 2XX.

     

    as far as putting in an alias with a full 10 digit number im not 100% sure but there should be a way to create the dial plan to first check if SITE2 has such a phone number and if not then dial out using a SIP trunk or whatever you will be using for that.

     

    if you need any help please let me know.

  4. Cross-Domain calling

    in Trunk Setup

    Posted

    Hello,

     

    We are going to have several domains set up in our installation, let's say site1.com and site2.com. Site 1 and 2 have their own accounts, and their own numbers, let's say they both have accounts 101 and 101.

     

    What I need to know is, can site1.com on acct. 101 call site2.com on their acct. 101, or any account for that matter? Basically, can site1.com and site2.com communicate between each other.

     

    Among this, we want to make some kind of alias for '101'@site1.com to be say, a fake 10-digit number. So site2.com could call site1 and not even know it, since they are just calling this 10-digit number.

     

    This may be simple, I'm not sure about where to start with this. Any ideas?

     

     

    Thanks in advance,

     

    Ryan

  9. Web-based or Linux Shell API

    in General Setup

    Posted

    I see...basically I just need to create a signup page for our customers to have an account created instantly.

    I suppose cURL would be the way to go but I'm not sure what URL params I can pass to the domain setup script, etc

     

    That is not true... There is a SOAP-based API, e.g. for settings extension parameters while the system is running. We don't promote it too much, it is very support intensive and the number of users that really need it is quite low.

  11. Web-based or Linux Shell API

    in General Setup

    Posted

    Hello,

     

    I am trying to find out if Pbxnsip has any API features where I can create new accounts on the fly, via a URL or possibly a Linux shell script.

    Is this possible? I tried searching the forums for anything with "API" but it wouldn't let me.

     

    Thanks in advance,

     

    Ryan

  12. NAT Traversal

    in General Setup

    Posted

    I'm just looking for all possible options.

     

    We have an Asterisk server we set up a while back and with a phone on the same router / setup, it works fine back and forth. And we all know Asterisk uses port 5060 by default, just like pbxnsip. Which is why I am asking about other reasons beside the firewall. I don't want my clients when we go live to have to figure out how to port forward on their routers unless absolutely necessary.

     

     

    Do you want to hear that it isn't the firewall, when all evidence says it is? We specify Xyzel routers and we have yet to have a 1-way audio problem. Zywall Plus+2 makes a fine choice.

     

    RINGING occurs as a result of a TCP message using SIP

    Talking occurs on a UDP Stream once handed off between the two endpoints talking SIP to each other.

    The firewall of choice is clobbering the UDP streams but passing the SIP messages as you already know since you have 5060 forwarded.

     

    PLease post you firewall choices so others will know...(perhaps a firmware upgrade from vendor will help?)

  13. NAT Traversal

    in General Setup

    Posted

    I had heard not to use STUN before, and we haven't been. It's been off.

     

    ALSO:: I tried testing a SNOM 105 phone and had the same problem.

     

    So can anyone else tell me why the phone will ring, but I can't hear on either end? Besides a firewall issue?

     

    Thanks.

     

    Do NOT use STUN.

    Do NOT use STUN.

    Do NOT use STUN.

    Do NOT use STUN.

    Do NOT use STUN.

    Do NOT use STUN.

  14. NAT Traversal

    in General Setup

    Posted

    Hello,

     

    After setting up a grandstream phone, It seems that some routers work and others don't. If I open port 5060 on the router, I can hear both ways, yet if not specifically opened, the phone will only ring, and when picked up, you cannot hear either way.

     

    What could possibly cause this?

     

    Thanks

  17. Area Code problem

    in General Setup

    Posted

    Hello,

     

    I have set in domain settings the 1 and the 916 area code. Yet when clients on this domain call out, they still need to dial a 1916 before calling; this is not feasible because our customers need to be able to call in their local area code without dialing it first.

     

    Any ideas?

     

    Thank you,

     

    Ryan

  19. Backups for seperate domains

    in General Setup

    Posted

    Hello,

     

    I understand that you can just tar the entire pbxnsip directory (/usr/local/pbxnsip), but I am wondering what would need to be done to create backups per domain, so our customers can go back to say Wednesday's backup, and restore what they had. If i just tarred the whole pbxnsip directory each day, that would be in essence an full backup of the system and this is not what i need.

     

    Thanks in advance,

     

    Ryan

  20. Can make calls, but cannot hear receiving voice

    in Grandstream Phones

    Posted

    On the phone I was testing with, it was a Grandstream BT100, rev. 02. I had the same problem with another cheaper grandstream setup as well.

    I don't have the option of using the G.711 codec on this phone.

     

    The default codec, which we are using, is G.722 (wide band). Other options are

    G.723.1, G.729A/B, PCMU, PCMA, iLBC, and G.726-32.

     

    Any other ideas? I appreciate the help.

     

    Ryan

     

    Make sure you use G.711u on all codec choice and do not use STUN or any nat traversal feature. What is the version of pbxnsip?

     

    Digisoft VOIP

  21. Outgoing calls - cannot hear

    in General Setup

    Posted

    The Phone is a Grandstream phone.

     

    I can call the phone from my cell phone, and can hear and talk back and forth fine...

    but when I try to make a call from the phone, I cannot hear the other person talking, yet they can hear me.

     

    The fact that it works fine on receiving but not on sending makes no sense. It sounds like a firewall issue, but we are set up fine.

     

    Any ideas?

     

    Ryan

  22. Can make calls, but cannot hear receiving voice

    in Grandstream Phones

    Posted

    Thanks for your reply, Detlef.

     

    I have checked most of that stuff out and to no avail. It seems to be a pbxnsip issue as we've had this issue before.

    Not sure why it is but I'm going to also post in the main.

     

    Ryan

     

     

    You registered the Grandstream as extension?

     

    I would check what codec it is using, routing and connectivity between pbx and Grandstream, maybe a firewall inbetween, STUN settings...

     

    I had something similar with audio in only one direction (not with a Grandstream) and it was because of the low bandwidth codec g.729 (18) which has to be purchased separately in pbxnsip.

     

    Detlef

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