Hi Peter Ok, I got the net changed in acl.conf.xml. I then tried setting console_loglevel, but I don't see any output on the console, it could very well be because it's a FreeBSD, and has very limited console. But after a restart it registers! So for some reason it needed a nudge there, very interesting. So now I have a local extension registered, and a provider registered, now I just need them to communicate. As I understand it, that's the dialplan I have to look at. I only have one provider hooked up, so a dial should be simple, right? It as Order 001 Condition ^0(.\d+)$ action bridge As I understand it, it should react on the 0 (for outside dialing) and then strip it And the action bridges the call to the outside. But I guess I'm missing something, because I just get a 'temporarily unavailable' shown in the xlite.
06-12-2009 kl. 15:14 skrev Peter P GMX <Prometheus001@gmx.net> i meddelelsen <4B1BBC52.3050106@gmx.net>:
Concerning,
Which I'm kinda confused about, I don't have any 192.168 net here?? I think, this is a default entry in the acl.conf.xml. Please check the entries there. But normally this shouldn't stop freeswitch from working and handling requests.
Can you set the console_log_level to "debug" in vars.xml and post you console output when the phone tries to register? You may also grep the network traffic on port 5060 (e.g. ngrep -d any port 5060 -W byline) on your machine, to see what's wrong. Best regards Peter mailinglist schrieb:
Hi Adam
Excellent first steps! Thankyou for the hint. Now I hope somebody can tell me what I'm doing wrong next...
I've gotten it to register to the testprovider here (musimi.dk), but I get an error when I create an account for testing with the X-Lite phone.
It displays 403 forbidden in the display.
I've created an account on FreeSwitch
extension 1001 password 1001 mailbox 1001 voicemail password 1001 account code 1001 Effective Caller ID Name Fribert Effective Caller ID Number 4692xxxx (the Musimi number) Voicemail Mail To <my address> Voicemail Attach File true User Context default Call Group <> Enabled true Extension Description Test number
In the X-Lite Display Name Fribert User name 1001 Password 1001 Autorization user name 1001 Domain LAN-IP-OF-pfSense
Check in Register with domain and receive incoming calls Check in domain.
That's about it.
Looking on the status page, I can see these lines in the log: 2009-12-06 09:55:50.310829 [NOTICE] switch_core.c:1064 Adding 192.168.42.0/24 (deny) to list lan 2009-12-06 09:55:50.310842 [NOTICE] switch_core.c:1064 Adding 192.168.42.42/32 (allow) to list lan 2009-12-06 09:55:50.311256 [DEBUG] mod_event_socket.c:2291 Socket up listening on 0.0.0.0:8021 2009-12-06 09:55:50.311256 [DEBUG] mod_event_socket.c:2291 Socket up listening on 0.0.0.0:8021
Which I'm kinda confused about, I don't have any 192.168 net here??? But as it also primarily forbids it, except .42 to allow, I'm wondering if it could be something internal?
Best regards
05-12-2009 kl. 18:52 skrev "Adam Ford" <lists@redbonez.net> i meddelelsen <0E0013F55E224674A1361329CF7A85F0@redbonez>:
I used the pfSense FreeSWITCH for awhile, as it is the only GUI FreeSWITCH I have found with a stable release. It was very easy to use, I would recommend it if you just want a quick base system with standard features. Though, I ended up switching to a compiled version of FreeSWITCH in order to make the customizations I needed for my office.
http://doc.pfsense.org/index.php/FreeSWITCH
-AF
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*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *mailinglist *Sent:* Saturday, December 05, 2009 2:47 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Setting up a FreeSwitch system on a pfSensefirewall???
Has anybody done this?
I'm completely at a loss, having tinkered very little with Asterisk, and giving up on that, I wonder if there's any help to be found on FreeSwitch?
Anybody that can give pointers to a good step-by-step instruction?
I want to have it handle my two sip-phones (siemens dect ip and spa 901), and handle a sip account at my provider.
Of course transferring calls between the two, as well as group calls would be a nice benefit.
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