Topic: Incoming calls with sail ?

[portedaix]

Hello,

How do I manage to receive incoming calls ? I am a newbie with asterisk. It is certainly very simple but I did not catch the idea.
Console tells me :
"Call from '0033484123456' to extension '0033484123456' rejected because extension not found." And I have a beautifull message in french, from asterisk, telling me "the number you have dialed is not registred" as an approximate traduction. I guess when asterisk receives the call it does not know what to do with it. And I just don't know where to give it the correct instructions.

By the way, it took me a hudge amount of time to find out the correct software version : sail-3.0 is not compatible with dolibarr (redirection loop is created to dolibarr) I had to install sail-2.5. And asterisk-1.6 did not function with sail (that was a question of '|' instead of ';' in hte trunk login sent to the sip provider, in sip.conf or something like that, I forgot the details but found other posts on that, problem known).

Thanks for your help
Olivier

[SARK devs]

Hello Olivier

To receive your call you need to create an entry in trunks to "catch" it and then direct it to where you want it to go.    Create a trunk of type PTT_DiD_Group and in the start and end box put 0033484123456.   This will catch your inbound call.  You can now edit the trunk you have created (by changing the open/closed boxes) to send your call to an extension or to a call group, IVR or queue.

To fix redirect loops either set dynamic proxy to off in globals panel or add the dolibar local url to the proxy ignore list in globals panel.

sail docs here - http://sarkpbx.com/twiki

Sail does not run with Asterisk 1.6; only 1.4 right now.

Kind Regards

S

[portedaix]

Hello again,

Thanks a lot for your help. I have inbound calls functionning nicely now. Fantastic 
I read many times this excellent sark admin guide, but did not catch this PTT_DID_Group trunk idea. asterisk and sail are really powerful.

I have three other points to really clarify my ideas. If you could help me again, thanks in advance.

• I disabled the dynamic proxy. asterisk is still functionning nicely. Actually, what is the meaning of having it "on" ? Twiki says "to proxy through the server manager panel to individual telephones". Does it mean to dial internal calls with names instead of numbers, or... ?
• I understood that with this dynamic proxy now, I could run sail-3.0 without having loop problems with dolibarr. As everything works fine with sail-2.5, I am just wondering if I should do it. 3.0 is definitely looking much better. I saw some differences, like 3 numbers extensions instead of 4 for ver2.5, no country identifier in global anymore (automatic ?). So what is the advantage of upgrading to 3.0 ?
• My sip provider is ovh, france. I have this line "chan_sip.c:7289 determine_firstline_parts: Bad request protocol Packet" coming up in asterisk cli. I saw other adsl box provider sip lines creating this message (freephonie france). It seems to be harmless to asterisk, just a package which should be droped silently. asterisk-1.6 did not display it. The only fix I found is to enter the rule 'iptables -I INPUT -p udp --src 123.123.123.123 --dport 5060 -m string --algo bm --string "Cirpack KeepAlive" -j DROP Packet'. But with sme and its templates, I do not know how to fix it. Any idea not to see this message again ?
        "

Personal comments : I tried different soft phones. The best I found in term of sound quality and voice delay is from 3CX, a commercial company but with some free softs like their softphone (fancy iphone look). SJPhone was not so good, I had to disable activeX to have the microphone functionning (?), and no mic funtionning at all with XLite (video and others OK). All set as general sip device, maybe that's the problem. Still looking, as I would like ldap available. Next I test is ekiga.

Good day to all contribs reader
Olivier