Trunk Registration

I have 2 trunks online. The trunk registration goes from 1-2 frequently.I can place about 5 outbound calls then something goes
haywire and no outbound calls can be placeed even after phone reboot.I’m not getting any inbound calls at all,but will go to voicemail somtimes

0418071651|copy ||03|Server ‘10.0.0.189’ said ‘provisioning/p.php/2345-12360-001.sip.ld’ is not present
0418071651|cfg |
|03|Prov|Starting to update sip.ld

I have been using sip.cfg on the server and database that was working before but not sure why the phone is looking for sip.ld now?

I’m getting these to logs from the phon often
000011.058|app1 |*|03| is out of range, using 3600
000007.640|log |4|03|syslog tcp connect FAILED to (10.0.0.181) port 1468 for transport tcp

port 1468 is open on my firewall

I also found this in asterisk log

[2013-04-18 02:10:27] VERBOSE[10636] pbx.c: – Executing [[email protected]:1] GotoIf(“SIP/fpbx-2-0e8c70b0-0000000d”, “0?blacklisted”) in new stack
[2013-04-18 02:10:27] VERBOSE[10636] pbx.c: – Executing [[email protected]:2] Set(“SIP/fpbx-2-0e8c70b0-0000000d”, “CALLED_BLACKLIST=1”) in new stack

[2013-04-18 02:10:27] VERBOSE[10636] pbx.c: – Executing [[email protected]:9] Dial(“SIP/fpbx-2-0e8c70b0-0000000d”, “SIP/fpbx-1-0e8c70b0/5035038775960,300,Tt”) in new stack
[2013-04-18 02:10:27] WARNING[10636] app_dial.c: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)

sorry for the long post

sip.ld is the software, that’s not your problem.

You have a NAT issue of some sort. Since I don’t know your network topology hard to comment any further.

thanks,
are there any tools like tcpdump i can use to help figure this out.?

You can install tcpdump, would not be the first place I would start.

You didn’t tell us anything about your topology, so again can’t say anything specific.

If the system was working I would focus on what changed.

what is topololgy?
can you point me in the directtion of nat config file or somthing?

Topology, as in topographic. What is the layout of the routers, are the phones on the same network as the servers, van, firewalls etc.

No files to look at if running a current FreePBX. It’s in sip settings.

there’s one router.Everything is all on the same network the phone is connected to
my labtop. The server and labtop are both using wireless internet.I’m thinking i should check the firewall on the labtop to see if it may be causing a problem

You are using ICS on the laptop to bridge the connections? If so that is probably the issue:

[2013-04-18 02:10:27] WARNING[10636] app_dial.c: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)

The above error indicates to me that within a matter of milliseconds Asterisk had connection then lost connection to the phone. Usually NAT, UDP timers on WAN’s. Since you are on a LAN I would look at layer 2 issues such as spanning-tree blocking.

If you are using ICS between the wireless and wired interface on the laptop I am surprised it ever worked.

Skyking
I did not manually bridge anything, the labtop did it automatically.
I remember seeing ics as a service that was running recently so it may have been
ics as you stated. Both the phone and labtop would message "duplicate ip"the i would reboot the phone and all would be good for about 5 calls.

Anyhow,thanks for your help. I’m sure you’ve been wondering all along why
did i go wireless to cause all this trouble.So i will tell you my issue with the
direct connect was, I dont pay the bill and its shared internet among our tenants so people got upset when i drew so much bandwidth and kicked me off the direct connection to the router stating i could have all the wireless i wanted.

I am pleased to say my problem with this will be over tomorrow as i just got a brand new clear Motorola modem and signing up with there service in the morning

I also just installed a second Ethernet card that i have tested with the modem
with success it gets loop back connection and seems to work already

Jeff

Ok, the ICS will never work right.

You are not going to be happy with the service from Clear. I am in the Cleveland market. We moved our office and the installation of our fiber was delayed. I had three Clear modems, one for a VPN for the servers, one for surfing and one for phones. The service was very inconsistent. I experienced frequent 10-20 second gaps in voice as the wireless buffers.

DSL is generally the best quality for IP phones (bandwidth is not shared until you get back to the CO), then cable modems (DOCSIS 3, 2 was marginal, the networks are so overbuilt latency and jitter is almost non-existent). None of the wireless services LTE (Cell Carriers) or WiMax (clear) are worth a damn for voice IMHO.

Your market may be better, let us know.

BTW Clear of course is Clearwire the Sprint Pre-LTE data fiasco.