Started by Ad_Hominem, July 11, 2011, 05:37:59 PM
Quote from: Ad_Hominem on March 14, 2012, 04:39:47 PMI've made some updates to these instructions, including adding information about how to specify the dialing delay and how to decide whether 911 calls are routed directly to the FXO (bypassing your PBX) or to force 911 calls to go through the PBX.
Quote from: Ad_Hominem on March 28, 2012, 11:54:49 PMMichigan,I have no idea why context=from-trunk did not appear in my original instructions, as I have that in my own trunk settings and in my "internal" documentation. I've corrected it. Thank you for bringing that to my attention.
Quote from: Ad_Hominem on March 28, 2012, 11:54:49 PMSince I subscribe to your blog, I was notified when you posted the idea of using a voice gateway to handle FXO calls and creating a SIP URI trunk in FreePBX to handle outgoing calls, and I was intrigued. I suspect that many users (myself included) only have an FXO for emergency calling and to keep my longstanding phone number tied to a provider that has no risk of going out of business. In fact, I receive no incoming calls at all on my FXO because they all forward to VOIP provider instead. I use the FXO for local outbound and 911 calls only, and so I'd simply ignore the Caller ID related issues. Attempting to integrate your instructions into mine is on my list of things to do!
Quote from: Ad_Hominem on March 28, 2012, 11:54:49 PMAlong those same lines, I urge you to take a close look at the modifications that I've made to the instructions recently, particularly regarding the digit map's handling of 911 calls, the dial plan, the use of X_UseRefer, and the means to enable message waiting, and to incorporate those into your instructions as well.
Quote from: Ad_Hominem on April 01, 2012, 09:53:54 PMForget the UseRefer option. Leave it unchecked. Change the MaxSessions in the SP for the FXS to something higher. I changed it to 99, and now the Obi will forward a call (just like any other endpoint) and still let me make and receive other calls.If you want to use the UseRefer, you have to understand how it works. 1. Place or receive the first call.2. Hit Flash.3. Dial the second call. The second person will receive a call from the ATA.4. Hang-up. The call from the ATA to the second caller will end.5. The Obi will send a REFER message and the original call will be re-routed to the second caller.6. The second person will receive the referred call, i.e. a new call from the original caller. 7. Since this all happens very quickly, you may be occupying more than 1 channel on the second person's endpoint, and if they can only handle one, then the call in step #6 will reach voicemail.Note that if, in step #4, you hit flash instead, you'll be in a conference. If you then hang-up before the first party (i.e., the party you called in step 1) hangs-up, the REFER message will get sent and steps 5 and 6 will occur anyway.Since these effects are a bit kludgy and unintended, I recommend that you use the new method of allowing the OBI to handle the transfer and increasing the maximum number of channels it can handle at once using the MaxSessions option in the Service Provider configuration. I've updated my instructions, above, to change this setting to 10, which is an arbitrary number that should meet most people's needs.
Quote from: Stewart on April 01, 2012, 07:58:18 PMTurning off ReferAOR might help.
Quote from: Stewart on April 01, 2012, 07:58:18 PMThough I know very little about Asterisk, you may find a clue by comparing the behavior of the Obi (using SIP Debug) with Zoiper (running Wireshark on the Zoiper PC). Asterisk's log may also show an error or other useful info.