This is my first post on here. I try not to as for help too much, but this one has me stumped.
The problem is this, my inbound Sip Trunk calls work only about half the time while outbound works 100% of the time.
I’m running FreePBX 2.5.2.3 behind a Fios router with UDP ports 5060-5082 and 10000-20000 forwarded to my PBX’s IP. Nat is also setup with my public IP and internal network of 192.168.1.0/255.255.255.0 in sip_nat.conf.
that comes form a version of Asterisk that has a bug in it, it’s possible to override it with a sip setting but I don’t recall the setting.
However, typically if that were the problem it would fail all the time not just part of the time. None the less, you will want it fixed and then you can see if it has any impact on your issue.
Well, i’m not sure what was wrong, but i went and grabbed some trunks from SipStation and they work perfectly without my having to do ANYTHING. The Firewall test even checked out.
I think I’ll just be canceling my voipvoip account. Thanks for the help guys.
Thanks for posting back. I’ll look forward to your post this evening.
I am sure you’re probably right…
At least, I feel like it should be a simple fix (e.g. “just add _____ to the _____.conf file”, or “just check the ______ box”).
But I can’t figure out where I’m going wrong.
I have done the SIP Station firewall test and it comes back as “pass”
Side Note (possibly related??):
I have checked with my ISP and I don’t think they are the issue… though the set up is a little odd.
Our internet service is provided by our local utility (FiberNet - http://www.morristownutilities.org/Technology.html) and they say that our IP address is assigned through DHCP, but that it won’t change (static characteristics), i.e. that it locks onto the MAC address of our router. Since two-way voice works on outbound calls I don’t really think that they are the problem… But I wondered if I would need to change my PBX settings since it’s not truly a static IP (at least in the way I’m familiar).
I don’t know if these screen shots will help anything, but here are some further details (3 pics inside the FreePBX web control – Asterisk Inf, and Sys. Info)
It sounds to me like you don’t have your internal/external ip address set in the system.I’m at work right now, but I’ll try to get the info for you when I get home.
One good way to test to see, is by running the freepbx sip trunk firewall test through the module.
I have a similar problem (outgoing call no problem, incoming callers can hear me, but I can’t hear them)
Forwarding 5060-5070, and 10000-20000 UDP to my FreePBX.
I performed a fresh install 4 days ago (CentOS 5.5, FreePBX 2.8.0.4, Asterisk 1.8.2.2)
All modules are up to date (including SIP Station) and I am using (or trying to use) two SIP Station trunks.
Name : fpbx-1-6104d9aa
Secret :
MD5Secret :
Remote Secret:
Context : from-pstn
Subscr.Cont. :
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
MOH Suggest :
Mailbox :
VM Extension : *97
LastMsgsSent : 32767/65535
Call limit : 0
Max forwards : 0
Dynamic : No
Callerid : “” <>
MaxCallBR : 384 kbps
Expire : -1
Insecure : port,invite
Force rport : Yes
ACL : No
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: -1
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : Yes
Send RPID : Yes
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost : trunk1.freepbx.com
Addr->IP : 216.82.225.24:5060
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 6104d9aa
SIP Options : path precondition replaces replace timer
Codecs : 0x104 (ulaw|g729)
Codec Order : (ulaw:20,g729:20)
Auto-Framing : No
100 on REG : No
Status : OK (79 ms)
Useragent :
Reg. Contact :
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
> doing dnsmgr_lookup for 'trunk2.freepbx.com'
> doing dnsmgr_lookup for 'trunk1.freepbx.com'
> doing dnsmgr_lookup for 'trunk2.freepbx.com'
> doing dnsmgr_lookup for 'trunk1.freepbx.com'
== Manager ‘admin’ logged on from 127.0.0.1
== Manager ‘admin’ logged off from 127.0.0.1
> doing dnsmgr_lookup for ‘trunk2.freepbx.com’
> doing dnsmgr_lookup for ‘trunk1.freepbx.com’
> doing dnsmgr_lookup for ‘trunk2.freepbx.com’
> doing dnsmgr_lookup for ‘trunk1.freepbx.com’
== Manager ‘admin’ logged on from 127.0.0.1
== Manager ‘admin’ logged off from 127.0.0.1
== Manager ‘admin’ logged on from 127.0.0.1
== Manager ‘admin’ logged off from 127.0.0.1
> doing dnsmgr_lookup for ‘trunk2.freepbx.com’
> doing dnsmgr_lookup for ‘trunk1.freepbx.com’
== Manager ‘admin’ logged on from 127.0.0.1
== Manager ‘admin’ logged off from 127.0.0.1
– Remote UNIX connection
– Remote UNIX connection disconnected
> doing dnsmgr_lookup for ‘trunk2.freepbx.com’
> doing dnsmgr_lookup for ‘trunk1.freepbx.com’
> doing dnsmgr_lookup for ‘trunk2.freepbx.com’
> doing dnsmgr_lookup for 'trunk1.freepbx.com’
pbx*CLI>
The last few lines (doing dnsmgr_lookup, etc.) just cycle about every minute or two.
I notice I have the same "Expire: -1"
But don’t know where I adjust that setting (and obviously, the solution of switching to SIP Station trunks won’t solve my problem
Any insight would be appreciated.
Thanks,
Lawrence
