One-Way Sound and No Call Recordings after Upgrade to Asterisk 1.8!

I’m a newbie, and still decided to go through the painstaking process of upgrading from Asterisk 1.6 to 1.8 because I thought it would make the ilbc or gsm codec work.

The ilbc and gsm codecs still don’t work, and now I have 2 new problems!! Hours later and I’m 8 steps back! UGH!

1----> On all calls, the person on the other end can hear me, but I hear nothing but dead air (one-way sound)

2----> There are no more call recordings showing up! The extensions are all still setup to always record.

Any help would be GREATLY appreciated!! Thanks!

More info is needed. What version of FreePBX are you using? What type of trunking for voice.

This type of problem is typically a firewall issue if you are using SIP trunks.

GSM is a supported CODEC, nothing has to be done. It has been in Asterisk since pre version 1.0.

iLBC is not included by default, I have never used this CODEC, is there s pecific reason why the g.729 CODEC is not sufficient for low bandwidth environment?

1 way audio is a networking NAT issue, not CODEC related.

The previous poster asked you to post your complete information. Since we have no idea what you did, what system you are working on and what steps you take specific advice is not possible.

I updated freePBX from 2.7 to 2.8 to 2.9, and then i IMMEDIATELY upgraded Asterisk 1.6 to 1.8. (i know i should have tested feePBX 2.9 before upgrading Asterisk to isolate any potential problems).

I am using SIP trunks, but everything worked 100% ok in freePBX 2.7 with Asterisk 1.6, so 1 of them must have messed with firewall settings! Do you think both problems are firewall-related, or just the one-way voice problem? I have UDP ports 5060 and 10000-20000 forwarded, and i tried forwarding ports 5061-5065 too and putting my PBX in the DMZ, but it didn’t help.

Sometimes I will hear half of the 1st ringtone when making a call, but no more incoming sound after that! There are never any call recordings even if I do get half a ringtone incoming sound, making me think these 2 problems aren’t at all related.

I encountered so many hiccups in the process of going from Asterisk 1.6 to 1.8 (finding and downloading missing package after missing package with yum, etc), that I just assumed it has to do with asterisk, but of course it could be the freePBX upgrade.

I’m contemplating biting the bullet and doing a complete re-install of the downgraded versions of Asterisk and FreePBX and then buying g729 licenses, or god forbid switching to 3CX for windows. My whole reason for doing the upgrade was that g711 works like garbage over my internet connection (always has). I heard that the ILBC codec was fixed in Asterisk 1.8 (wrong), and hoped that maybe the GSM codec was fixed in 1.8 (also wrong), or that the free g729 codec files would work (wrong #3)!

I updated freePBX from 2.7 to 2.8 to 2.9, and then i IMMEDIATELY upgraded Asterisk 1.6 to 1.8. (i know i should have tested feePBX 2.9 before upgrading Asterisk to isolate any potential problems).

I am using SIP trunks, but everything worked 100% ok in freePBX 2.7 with Asterisk 1.6, so 1 of them must have messed with firewall settings! Do you think both problems are firewall-related, or just the one-way voice problem? I have UDP ports 5060 and 10000-20000 forwarded, and i tried forwarding ports 5061-5065 too and putting my PBX in the DMZ, but it didn’t help.

Sometimes I will hear half of the 1st ringtone when making a call, but no more incoming sound after that! There are never any call recordings even if I do get half a ringtone incoming sound, making me think these 2 problems aren’t at all related.

I encountered so many hiccups in the process of going from Asterisk 1.6 to 1.8 (finding and downloading missing package after missing package with yum, etc), that I just assumed it has to do with asterisk, but of course it could be the freePBX upgrade.

I’m contemplating biting the bullet and doing a complete re-install of the downgraded versions of Asterisk and FreePBX and then buying g729 licenses, or god forbid switching to 3CX for windows. My whole reason for doing the upgrade was that g711 works like garbage over my internet connection (always has). I heard that the ILBC codec was fixed in Asterisk 1.8 (wrong), and hoped that maybe the GSM codec was fixed in 1.8 (also wrong), or that the free g729 codec files would work (wrong #3)!

You still did not tell us how you installed this. Did you download everything from source? Why not use a distro that takes care of this for you.

What do you mean GSM codec is broke? It works fine. It is a nice compromise for a free, low bandwidth CODEC. It is a shame most providers don’t support it.

If you installed our distro you would have a working FreePBX 2.9 and Asterisk 1.8 system. Certainly a closed source soltion is not an answer.

Asterisk 1.8 changes many of the NAT and media processing commands. Sounds like your media is getting “invited” off your box by a downstream SIP peer and your router can’t handle the NAT (the half ringing points to this).

GSM never worked either! Only g711 has ever worked. My SIP provider supports g711a/g711u/g729a/GSM/ILBC (g729 is g729a-only)

g729a isn’t preferred, because it’s not low bandwidth that is the problem for me at all; it’s high-jitter, where g729a isn’t much better than g711 and perhaps worse when there’s plenty of bandwidth, and jitter is the only problem. Perhaps g729a is more easily corrected with a jitter buffer though? Also, every 729 channel needs to be licensed. GSM is natural to most humans, and I’d be 100% fine with it, and ILBC is pretty much perfect for my connection seeing that jitter is my only problem!

Whenever I set my trunk to only use GSM, or g729, or ILBC, I get “all circuits busy” when trying to dial out, and my PBX immediately hangs up on incoming calls saying “unsupported codec” when I monitored the channels.

I used asterisk’s distro called AsteriskNOW 1.7.1, which is still the most current version even though it only has Asterisk 1.6 and FreePBX 2.7… Should I start over with the FreePBX distro? Is there a distro that takes care of this??!

GSM fails in exactly the same manner as g729 and ILBC do… I have no idea why unless my provider is lying when they say they support it! The phone I’m using as my extension does not support GSM, so I can’t test my provider’s support of GSM.

SkykingOH---- Your diagnosis sounds SPOT-ON. Would using FreePBX’s distro to reinstall solve this one-way sound problem, or is there an easier fix? Do I need to format the whole drive and start 100% fresh? I installed the X Window system, so perhaps there’s a slightly easier way to whipe Asterisk away and start fresh with the FreePBX distro?

Let’s start from the beigning, your assertion that GSM is broke is not correct, it is your provider or configuration that is preventing using the CODEC.

No amount of jitter buffer or CODEC changes is going to make a connection usable that is unusable.

Asterisk Now is out of date, and in my opionion not well taken care of (sorry Qwell).

Installing xwindows on a phone system is a bit much, it certainly did not solve your problems.

Installing a new distribution is not going to solve a misconfiguration.

I’m going to save the GSM problem discuss for a different thread, because this is NOT a problem resulting from the software upgrades (the topic of the thread).

ON THAT FRONT… GREAT NEWS… I fixed BOTH my problems at once!!! (i guess they were somehow oddly related) The problem is that I don’t know how I fixed it, and I can’t replicate the error! This bothers me and makes fixing it bittersweet. My best guess, is that I added “externip=50.12.X.XXX” (x’s being the actual numbers of my public IP address) to my trunk’s settings. However, I then removed this line and reloaded FreePBX, and everything still works though. While I was upgrading asterisk, I had rebooted my modem, so my ISP may have given me a different IP. Perhaps I can’t replicate the error by simply removing externip=50.12.X.XXX, because now my SIP provider still knows where to find me? (: I’d have to change my IP address again and if I can replicate the error!

==FIREWALL HOUSEKEEPING QUESTIONS==
Right now I only have 1 extension (1 phone), because I haven’t deployed the PBX yet, so another device can’t be stealing the connection. I would like to change port usage around to make everything well a little more “kosher”… Right now EVERYTHING… my SIP extension phone and all 3 of my trunks use 5060 as the control port, and they’re all from the same provider, so same SIP server IP. I don’t like this, and I want to have 1 trunk use 5060, 1 use 5061, 1 use 5062, etc. Then, can I have all my extension phones share a single port that’s not used by a trunk (say 5071).

I’m currently forwarding UDP ports 69, 5060-5070 and 10000-20000 to the PBX, and TCP port 10000 only. I want to narrow the UDP 10000-20000 down to 10000-16383, but then I would need to manually edit “etc/asterisk/rtp.conf” and tell asterisk only to use 10001-16383 instead of the default 10001-20000, so asterisk doesn’t try to use ports over 16383 for trunk communications. My extension phones use ports 16384+ to communicate with the PBX, so I don’t want those ports in the usable range for the trunk.

Did I get all of that right??

Could the problem with GSM be that my PBX isn’t transcoding?

I have searched high and low, and cannot find an answer to where to find that setting! My endpoints (phones) DO NOT support GSM. I want to use ALAW for phone>pbx. I’m still getting “all circuits busy” when trying to dial out. HOWEVER, GSM seems to work fine when the endpoints aren’t involved (the IVR is picking up now with the trunk configured as GSM only). This has to be an issue with transcoding turned off. How do I turn it on?!

First, don’t add externip to a peer it is a general setting. You should use the sip settings module. You will need to use exterhost and a dynamic DNS service to keep up with your dynamic outside IP.

Second, the SIP settings module allows you to define CODEC priorities. Sounds like you have a conflict between your peers and the SIP general settings CODEC priorities.

You are way overthinking the ports. Don’t worry about these issues, I doubt you will run out of entries in your routers translation table. Make sure you have any type of SIP ALG or helper turned off in your firewall/router.

Why do you have tftp (udp port 69) opened? Anyone can download your phone configs in the clear.

Why do you have TCP 10,000 opened? Do you really need to expose Webmin? It is horribly insecure.

If you are going to expose SIP to public make sure you have very string peer secret’s and some type of IDS such as Fail2ban or BFD.

Could the problem with GSM be that my PBX isn’t transcoding?

I have searched high and low, and cannot find an answer to where to find that setting! My endpoints (phones) DO NOT support GSM. I want to use ALAW for phone>pbx. I’m still getting “all circuits busy” when trying to dial out. HOWEVER, GSM seems to work fine when the endpoints aren’t involved (the IVR is picking up now with the trunk configured as GSM only). This has to be an issue with transcoding turned off. How do I turn it on?!