Grandstream Gxp1615 Manual

[Francesca Cruiz]

Iam new to free Pbx, I have been trying to set it up for my office which has 4 users. - Its a simple yet easy install and I have managed to get Zoiper Softphone with Iax working no problem. The softphone app MizuDroid with Sip works no problem. the incoming two lines with the VoIP service provider using sip are working with no problem. Ivr is working, I can call the system and the IVr will pick up and direct the calls. The internal Softphone users can call each other with no problems. ( a few fine tweaks are still needed to make it fully functional, but so far all is good)

However, I have hit a snag with grandstream desk phones. I have been at it for two weeks and I am at wits end. My issue is relating to the IP desk phones that we use in the office. The desk phones are Grandstream Gx-1625.

My configuration is as follows:

Freepbx running in a Virtual container on our server in Datacentre - Assigned public ip (41.x.x.x) and with an internal nated ip of 192.168.2.154 - we control the Firewall running Openwrt which sits between the Metal server and the internet. - All the clients/users remote in from their own internet connections which are all Dnat or behind some sort of firewall and have. None of the clients have fixed Ipaddress including our office connection. Ip are Dynamically assigned.

Please can someone explain to me how I can use SIP or someway of automating the provisioning of the office desk phone devices so that when a user is connected or attempts to connect their phone is provisioned with the correct settings, its exhausting and challenging to get grand stream phones to work with SIP and manual configurations ?.

nat= is really two different options for working round peers with broken NAT. The yes and no options that set both of them together have been deprecated since around Asterisk 1.6 or 1.8. Both exist in chan_pjsip, as first class options, rather than sub-options, although the name for comedia equivalent is slightly plainer English, so you need to read the documentation.

(off topic) I have also had issues with comcast (cable modem) customers where everything works fine for months and then all of a sudden no matter what I do one of the phones will not work on that network. I install a new phone and its fine, move the old phone to any other customer (different network) and its fine. Almost like the MAC of the phone is stuck in some cache on the router. Tried everything, finally I just give up and install a new phone and recycle the old one into my inventory. Has happened several times now in the last 2 years.

I have Finally go the grandstream phone to work partially - forwarded the ports, did a *60 speaking clock test and it worked now *43 for the echo test, but I have no outgoing mic audio from the handset to the FREEPBX server.

I am using Chan_PJSIP , - I have managed to get the phones to register and I can hear incoming audio, but my voice (outgoing audio is not working ) - when I run the echo test, I cannot hear my voice. I think its got to do with the fact that the outgoing messages from the client device is not routing correctly.

to me that definitely sounds like a NAT issue. You can try chan_sip just to be sure (under advanced click "switch to Chan_sip), save the changes, then go back into advanced and find NAT change to YES (FORCED), then in EPM, delete the extension and re-add it. On the phone you will need to reprovision so that it switches to the correct port numbers. Personally with the grandstream it seems like it works better if you factory reset, enter the PBX IP again and click provision. Also, what version firmware is the phone running? I have found 1.0.11.10 to be the most stable with freepbx 14/15. Havent tried 16 yet. If you have firmware updates enabled and pick the latest version in EPM I find the phone will reboot and attempt to update and after a really long time (like over an hour) may or may not ever finish the firmware update so I always do it directly on the phone and turn off firmware management in EPM for the extension.

If that works then your issue is the same that I have with pjsip which I have not been able to get working unless I change the default port numbers for pjsip (has to do with something the ISP is doing).

It will not work on Chan_sip. Call drops and dont get any traffic, I have now gone on to the default ports and trying to get it to work with the legacy ports. This must be the most complicated IT thing I have ever tried to do myself.

I think the routers on both sides are preventing the UDP packets.

I am using the OPENWRT Firewall on both routers, at my client side and at the server side.

I am using phone GXP-1625 with Firmware 1.0.4.99

PBX is 15.0.17.64

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