I will admit I don’t understand the DTMF thing so well. I saw an article on another site or forum that mentioned it so I had to ask. On my Grandstream phones the settings are “in audio” and “via RTP”.
I also thought codec issues might be a problem so I have tested with and without g729.
I don’t know how this turned into a distro debate. In the initial post I clearly stated that I had tried with CENTOS and Ubuntu. I didn’t post this to make a statment about Ubuntu. I haven’t done much with 1.6 and I just wanted some help. I have done at least 100 installs with 1.4 and never had this problem.
I do NOT set phones inside the NAT to “nat=yes”. My 1st client had 30 remote phones outside his NAT, and 20 inside. I understand that setting and learned all the ways you could go wrong with a NAT from him setting up new extensions and phones when I wasn’t available.
I think I am also clear on the NAT settings for Freepbx. I have done this with both static and dynamic IPs on the outside.
I’ll be the 1st to admit this is probably a stupid, little mistake I am making somewhere in the install of 1.6x but it’s hard to swallow that when I can do the exact same install with no errors or issues on 1.4x.
Since you’re unwilling to consider the fish as a possible issue your next step is to break out wireshark and see what is happening.
I would grab a tcpdumb on the box running Asterisk and analyze it offline. You will need a hub or a switch that supports port mirroring (spanning in Cisco speak. Don’t forget to increase the packet capture “snarl” length or you will only grab headers and not payload.
Since you are in uncharted waters with regard to this issue you have to be willing to do a deep dive.
RFC2833 encodes the DTMF in the RTP stream, it is decoded at the other end. Inband simply digitizes the tones with all the other audio.
1 - DTMF method must match, don’t use auto
2 - RFC2833 or SIP INFO method (like 2833 but uses SIP messages to send DTMF data)
must be used on all but full bit rate CODEC’s
The DTMF issue has nothing to do with the can’t pick up the phone issue.
Any hub will work, or a managed switch with port mirroring. I also suggest you take a quick network 101 refresher.
Ok, so I didn’t refresh my cache. As far as judging male or fish attractiveness, that task I will defer.
I looked at what you installed. Asterisk doesn’t depend on anything to process calls. You are barking up the wrong tree.
Did you try the IAX thing yet? It’s a very important data point. Load Zoiper softphone it takes 5 minutes.
If you build Asterisk 1.4 on the same machine with no other changes does it run? Do you know how to build asterisk to another directory so you can have both versions on the same machine?
I have never been able to get any Asterisk 1.6 install to do anything on Freepbx if I did NOT “make samples”.
Without I get a loop where asterisk restarts with an error code “1” when I do ./start_asterisk start and can’t proceed with the install. If I install the “samples” it goes on but I still can’t answer an inbound call. BUT,…I wonder if something in the samples is hosing the whole config.
BTW SKO,…I can’t get my Hard IAX phone or Zoiper to register. I checked and double checked the settings I created for the IAX extension on Freepbx to make sure I didn’t make a type-O and bad match.
I have used “Zoiper” before with no problems on my working, Asterisk 1.4x machines.
You hinted that this could lead to an important clue SKO/Shaggy. What do you make of that?
Someone mentioned checking in with my service provider. Here’s what they had to say:
May 10, 2010 08:20 AM : Customer service
Hello,
Please add the following lines into your Callcentric trunk:
disallow=all
allow=ulaw
allow=alaw
Additionally, within your Callcentric trunk please change “username=17772789999” to “defaultuser=17772789999”. After doing so please restart Asterisk.
I did that,…still no worky. I wrote back and this was their response.
May 10, 2010 02:03 PM : Customer service
Hello,
We have reviewed your calling history, and on a few occasions we do see that a few of of your incoming calls are dropping specifically due to a “media timeout” which basically means that our system did not detect any audio being transmitted in either direction (our system drops these types of calls automatically, as a way to drop “rouge” calls). If you would, within your PBX’s configurations, please try adding the line “session-timers=refuse” to your sip.conf file (or if you are using freePBX in conduction with your PBX, your sip_general_custom.conf file). Once you have included the line above, you may need to restart the entire Asterisk service in order for the changes to take into effect.
I did a very careful walk through of my script manually and added the “session-timers=refuse” to sip_general_custom.conf.
Thanks for everyone’s input. Philippe’s suggestion that it might be a carrier issue never occurred to me since I have never had any issues with Callcentric. They have excellent support and I am sorry I didn’t call them sooner.
first off don’tput it in sip_general_additional.conf, it will get automatically removed.
If I’m not mistaken, this is a trunk specific setting so it may be better off there. If that is not the case, then it should go into sip_general_custom.conf (as they suggested) though if you are using the Asterisk SIP Settings module, you can define it in that module and not have to mess with the configuration file.
As far as the session-timers=refuse fix, there was a bug at one point in Asterisk that triggered this problem:
I just upgraded my home system to 1.6x with what I learned from Callcentric. Both of their suggested updates ended up being necessary evils.
Step 1 --------------------------
May 10, 2010 08:20 AM : Customer service
Hello,
Please add the following lines into your Callcentric trunk:
disallow=all
allow=ulaw&alaw
Additionally, within your Callcentric trunk please change “username=17772789999” to “defaultuser=1777278999”. After doing so please restart Asterisk.
Step 2 ---------------------------
Add session-timers=refuse to your sip configuration,… “Using the Asterisk SIP Settings module” (as per Phillipe), bottom of the page, “Other SIP settings”
Now my home system is running like a top under Ubuntu 10.4, Dahdi 2.3.0 complete and Asterisk 1.6.27. Touche to the Ubuntu bashers.
I appreciate your help and input Sky King, “Point Taken”, but I’ve made so many posts about this in so many places that I forgot to mention I could make outbound calls and ext to ext. I thought I had said it (inbound pickup) was the only thing I couldn’t do,… but I just reviewed the whole thread and see that I did NOT word it that way.
I mentioned the details of my OS/distro and the whole thread went off the rails. Apparently those people didn’t read the 1st entry of this post where I said I had also done the build with Centos 5.4.