Traveling Man -- Can't call remote extension...

[dandy_don]

I have traveling man working with several softphones, but only one way dialing. I can call into incredible pbx from the outside world, call extensions behind the firewall and make outside calls. The audio both ways is fine (but with an appreciable delay).

However, I can NOT ever reach the remote softphone from any of the extensions behind the firewall. When bring the softphone back behind the firewall and access the server directly, I can two-way calling works fine. Whenever I'm outside the firewall, it is one-way calling only back into the server...

Although the calling works for one-way calls, it takes anywhere from 5-30 seconds before the call placed from the softphone actually goes through and starts ringing.

Here is some output from the server
root@pbx:~ $ tail -f /var/log/asterisk/full
[2011-08-10 23:11:32] NOTICE[6001] chan_sip.c: Peer '501' is now Reachable. (759ms / 2000ms)
[2011-08-10 23:18:01] VERBOSE[5645] asterisk.c: -- Remote UNIX connection
[2011-08-10 23:18:01] VERBOSE[21587] asterisk.c: -- Remote UNIX connection disconnected
[2011-08-10 23:18:42] NOTICE[6001] chan_sip.c: Peer '501' is now Lagged. (3385ms / 2000ms)
[2011-08-10 23:18:56] NOTICE[6001] chan_sip.c: Peer '501' is now UNREACHABLE! Last qualify: 3385
[2011-08-10 23:19:07] NOTICE[6001] chan_sip.c: Peer '715' is now Reachable. (4ms / 2000ms)
[2011-08-10 23:19:38] VERBOSE[5645] asterisk.c: -- Remote UNIX connection
[2011-08-10 23:19:38] VERBOSE[21842] asterisk.c: -- Remote UNIX connection disconnected
[2011-08-10 23:19:38] VERBOSE[5645] asterisk.c: -- Remote UNIX connection
[2011-08-10 23:19:38] VERBOSE[21845] asterisk.c: -- Remote UNIX connection disconnected
[2011-08-10 23:27:01] VERBOSE[5645] asterisk.c: -- Remote UNIX connection
[2011-08-10 23:27:01] VERBOSE[21929] asterisk.c: -- Remote UNIX connection disconnected
[2011-08-10 23:36:01] VERBOSE[5645] asterisk.c: -- Remote UNIX connection
[2011-08-10 23:36:01] VERBOSE[22027] asterisk.c: -- Remote UNIX connection disconnected

Asterisk = ONLINE | Dahdi = ONLINE | MySQL = ONLINE │
│ SSH = ONLINE | Apache = ONLINE | Iptables = ONLINE │
│ Fail2ban = ONLINE | Internet = ONLINE | Ip6Tables = ONLINE │
│ BlueTooth = ONLINE | Hidd = ONLINE | NTPD = ONLINE │
│ SendMail = ONLINE | Samba = OFFLINE | Webmin = ONLINE │
│ Ethernet0 = ONLINE | Ethernet1 = N/A | Wlan0 = N/A │
│ │
│ PBX in a Flash Version = 1.7.5.6 │
│ FreePBX Version = 2.8.1.4 │
│ Running Asterisk Version = 1.8.4 │
│ Asterisk Source Version = 1.8.4 │
│ Dahdi Source Version = 2.4.1.2+2.4.1 │
│ Libpri Source Version = 1.4.11.5 │
│ IP Address = XX.XX.XXX.XXX on eth0 │
│ Operating System = CentOS release 5.6 (Final) │
│ Kernel Version = 2.6.18-238.9.1.el5 - 32 Bit


Any help is greatly appreciated.

Thanks,
Don

 

[dandy_don]

Anyone else having this problem? Anyone have any idea how to fix this?

Thanks,
Don

 

[dandy_don]

OK -- I've managed to solve some of the latency issues with inbound call placement by tweaking some of the client settings in the softphone. Calls into the system work fairly quickly now and audio delay isn't too bad.

I still can't dial outward to the softphone. The error I get is 503 and the message played by the server to the local phone attempting to make the outbound call to the softphone is, "Your call can not be completed as dialed..." It is as if the server doesn't have an outbound route to reach the softphone?!? However, the softphone is registered and makes calls inward just fine...

Is there an outbound route that needs to be setup in order to make outbound calls to the softphone?

Thanks,
Don

 

[dandy_don]

PLEASE! Can't someone offer some suggestions? I've searched high & low, etc., and can't figure this out. Isn't there someone who can share some insight?

Thanks,
Don

 

[dandy_don]

Please -- Anyone -- offer some helpful suggestions?!? Please... I really want to get this to work and I've gone through everything I can find and to no avail... At this point I am willing to pay for support to get this going (if that helps).

I've followed all of the instructions, clues, etc., that I can find and this still doesn't work.

Thanks,
Don

 

[dallas]

This will almost certainly be a firewall issue. Try disabling the firewall and retest. If it works re-enable the firewall, increase the logging verbosity, (asterisk -v...vr) and look at the SIP signalling.
Do you have nat=yes in the extension config?

Dallas

Another question... do you have a firewall on you travelling man PC?

 

[dandy_don]

Dallas -- Thanks for responding!
I turned off the firewall on the laptop but it didn't make any difference. I do have nat=yes in the extension config. I have the port forwarded to the server and inbound calls are just fine. I just can't contact the softphone outside the firewall.

I'll try the verbose logging and report back with the findings.

Thanks!
Don

 

[dandy_don]

Dallas,

I dialed in from the softphone (outside the firewall) to extension 703 inside the firewall. It worked fine... I hung up and tried calling the softphone (ext 501) from extension 703. Here is the output. I'm still a noob here. Thanks,
Don


== Spawn extension (from-internal, 6781234567, 8) exited non-zero on 'Local/6781234567@from-internal-c7b9;2'
-- Executing [h@from-internal:1] Macro("Local/6781234567@from-internal-c7b9;2", "hangupcall") in new stack
-- Executing [s@macro-dial-one:38] ExecIf("SIP/703-00000010", "0?Set(DIALSTATUS=)") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("Local/6781234567@from-internal-c7b9;2", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-dial-one:39] GosubIf("SIP/703-00000010", "0?s-CONGESTION,1") in new stack
-- Executing [s@macro-hangupcall:4] GotoIf("Local/6781234567@from-internal-c7b9;2", "1?skipblkvm") in new stack
-- Executing [s@macro-dial-one:40] MacroExit("SIP/703-00000010", "") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("Local/6781234567@from-internal-c7b9;2", "1?theend") in new stack
-- Executing [s@macro-exten-vm:10] GotoIf("SIP/703-00000010", "0?exit") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-exten-vm:11] Set("SIP/703-00000010", "SV_DIALSTATUS=CONGESTION") in new stack
-- Executing [s@macro-exten-vm:12] GosubIf("SIP/703-00000010", "0?docfu,1") in new stack
-- Executing [s@macro-exten-vm:13] GosubIf("SIP/703-00000010", "0?docfb,1") in new stack
-- Executing [s@macro-exten-vm:14] Set("SIP/703-00000010", "DIALSTATUS=CONGESTION") in new stack
-- Executing [s@macro-exten-vm:15] NoOp("SIP/703-00000010", "Voicemail is '501'") in new stack
-- Executing [s@macro-exten-vm:16] GotoIf("SIP/703-00000010", "0?s-CONGESTION,1") in new stack
-- Executing [s@macro-exten-vm:17] NoOp("SIP/703-00000010", "Sending to Voicemail box 501") in new stack
-- Executing [s@macro-exten-vm:18] Macro("SIP/703-00000010", "vm,501,CONGESTION,") in new stack
-- Executing [s@macro-vm:1] Macro("SIP/703-00000010", "user-callerid,SKIPTTL") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/703-00000010", "AMPUSER=703") in new stack
-- Executing [s@macro-hangupcall:9] Hangup("Local/6781234567@from-internal-c7b9;2", "") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'Local/6781234567@from-internal-c7b9;2' in macro 'hangupcall'
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/703-00000010", "0?report") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'Local/6781234567@from-internal-c7b9;2'
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/703-00000010", "0?Set(REALCALLERIDNUM=703)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/703-00000010", "AMPUSER=703") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/703-00000010", "AMPUSERCIDNAME=703") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/703-00000010", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/703-00000010", "AMPUSERCID=703") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/703-00000010", "CALLERID(all)="703" <703>") in new stack
-- Executing [s@macro-user-callerid:9] ExecIf("SIP/703-00000010", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/703-00000010", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] Set("SIP/703-00000010", "CALLERID(number)=703") in new stack
-- Executing [s@macro-user-callerid:20] Set("SIP/703-00000010", "CALLERID(name)=703") in new stack
-- Executing [s@macro-user-callerid:21] NoOp("SIP/703-00000010", "Using CallerID "703" <703>") in new stack
-- Executing [s@macro-vm:2] Set("SIP/703-00000010", "VMGAIN=""") in new stack
-- Executing [s@macro-vm:3] GotoIf("SIP/703-00000010", "1?vmx,1") in new stack
-- Goto (macro-vm,vmx,1)
-- Executing [vmx@macro-vm:1] Set("SIP/703-00000010", "MEXTEN=501") in new stack
-- Executing [vmx@macro-vm:2] Set("SIP/703-00000010", "MMODE=CONGESTION") in new stack
-- Executing [vmx@macro-vm:3] Set("SIP/703-00000010", "RETVM=") in new stack
-- Executing [vmx@macro-vm:4] Set("SIP/703-00000010", "MODE=unavail") in new stack
-- Executing [vmx@macro-vm:5] GotoIf("SIP/703-00000010", "0?chknomsg") in new stack
-- Executing [vmx@macro-vm:6] Set("SIP/703-00000010", "VM_OPTS=") in new stack
-- Executing [vmx@macro-vm:7] GotoIf("SIP/703-00000010", "0?s-CONGESTION,1") in new stack
-- Executing [vmx@macro-vm:8] GotoIf("SIP/703-00000010", "1?notdirect") in new stack
-- Goto (macro-vm,vmx,10)
-- Executing [vmx@macro-vm:10] NoOp("SIP/703-00000010", "Checking if ext 501 is enabled: disabled") in new stack
-- Executing [vmx@macro-vm:11] GotoIf("SIP/703-00000010", "1?s-CONGESTION,1") in new stack
-- Goto (macro-vm,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-vm:1] Macro("SIP/703-00000010", "get-vmcontext,501") in new stack
-- Executing [s@macro-get-vmcontext:1] Set("SIP/703-00000010", "VMCONTEXT=default") in new stack
-- Executing [s@macro-get-vmcontext:2] GotoIf("SIP/703-00000010", "0?200:300") in new stack
-- Goto (macro-get-vmcontext,s,300)
-- Executing [s@macro-get-vmcontext:300] NoOp("SIP/703-00000010", "") in new stack
-- Executing [s-CONGESTION@macro-vm:2] VoiceMail("SIP/703-00000010", "501@default,u""") in new stack
-- <SIP/703-00000010> Playing 'vm-theperson.gsm' (language 'en')
== Spawn extension (macro-vm, s-CONGESTION, 2) exited non-zero on 'SIP/703-00000010' in macro 'vm'
== Spawn extension (macro-exten-vm, s, 18) exited non-zero on 'SIP/703-00000010' in macro 'exten-vm'
== Spawn extension (from-internal, 501, 1) exited non-zero on 'SIP/703-00000010'
-- Executing [h@from-internal:1] Macro("SIP/703-00000010", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/703-00000010", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/703-00000010", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/703-00000010", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup("SIP/703-00000010", "") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/703-00000010' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/703-00000010'

 

[phonebuff]

So,

In the original post you have documented the issue for Extension 501..

[2011-08-10 23:11:32] NOTICE[6001] chan_sip.c: Peer '501' is now Reachable. (759ms / 2000ms)
[2011-08-10 23:18:01] VERBOSE[5645] asterisk.c: -- Remote UNIX connection
[2011-08-10 23:18:01] VERBOSE[21587] asterisk.c: -- Remote UNIX connection disconnected
[2011-08-10 23:18:42] NOTICE[6001] chan_sip.c: Peer '501' is now Lagged. (3385ms / 2000ms)
[2011-08-10 23:18:56] NOTICE[6001] chan_sip.c: Peer '501' is now UNREACHABLE! Last qualify: 3385

For a call to go to an extension Asterisk has to be able to "reach" it. at 23:11 501 is reachable although the trip times are extensive, at 23:18 seven minutes latter the path from the Server to the Extension is no longer discoverable and the extension is declared offline / unreachable.

http://www.voip-info.org/wiki/view/Asterisk+sip+qualify

To me this would indicate either a routing or Firewall issue. Since I have not deployed a traveling man configuration I can't say for sure which. But I would bet on a routing issue since you say it works when the connection is on the Local Network segment.. ping and tracert can be your friends here as the OS level and "sip show peer[s| 501] at the Asterisk level..

Good Luck..

-----------------------------------------

 

[dallas]

[quote
-- Executing [s@macro-hangupcall:1] GotoIf("Local/6781234567@from-internal-c7b9;2", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-dial-one:39] GosubIf("SIP/703-00000010", "0?s-CONGESTION,1") in new stack
-- Executing [s@macro-hangupcall:4] GotoIf("Local/6781234567@from-internal-c7b9;2", "1?skipblkvm") in new stack
-- Executing [s@macro-dial-one:40] MacroExit("SIP/703-00000010", "") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("Local/6781234567@from-internal-c7b9;2", "1?theend") in new stack
-- Executing [s@macro-exten-vm:10] GotoIf("SIP/703-00000010", "0?exit") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-exten-vm:11] Set("SIP/703-00000010", "SV_DIALSTATUS=CONGESTION") in new stack
-- Executing [s@macro-exten-vm:12] GosubIf("SIP/703-00000010", "0?docfu,1") in new stack
-- Executing [s@macro-exten-vm:13] GosubIf("SIP/703-00000010", "0?docfb,1") in new stack
-- Executing [s@macro-exten-vm:14] Set("SIP/703-00000010", "DIALSTATUS=CONGESTION") in new stack
-- Executing [s@macro-exten-vm:15] NoOp("SIP/703-00000010", "Voicemail is '501'") in new stack
-- Executing [s@macro-exten-vm:16] GotoIf("SIP/703-00000010", "0?s-CONGESTION,1") in new stack
-- Executing [s@macro-exten-vm:17] NoOp("SIP/703-00000010", "Sending to Voicemail box 501") in new stack
-- Executing [s@macro-exten-vm:18] Macro("SIP/703-00000010", "vm,501,CONGESTION,") in new stack
-- Executing [s@macro-vm:1] Macro("SIP/703-00000010", "user-callerid,SKIPTTL") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/703-00000010", "AMPUSER=703") in new stack
-- Executing [s@macro-hangupcall:9] Hangup("Local/6781234567@from-internal-c7b9;2", "") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'Local/6781234567@from-internal-c7b9;2' in macro 'hangupcall'
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/703-00000010", "0?report") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'Local/6781234567@from-internal-c7b9;2'
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/703-00000010", "0?Set(REALCALLERIDNUM=703)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/703-00000010", "AMPUSER=703") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/703-00000010", "AMPUSERCIDNAME=703") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/703-00000010", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/703-00000010", "AMPUSERCID=703") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/703-00000010", "CALLERID(all)="703" <703>") in new stack
-- Executing [s@macro-user-callerid:9] ExecIf("SIP/703-00000010", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/703-00000010", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] Set("SIP/703-00000010", "CALLERID(number)=703") in new stack
-- Executing [s@macro-user-callerid:20] Set("SIP/703-00000010", "CALLERID(name)=703") in new stack
-- Executing [s@macro-user-callerid:21] NoOp("SIP/703-00000010", "Using CallerID "703" <703>") in new stack
-- Executing [s@macro-vm:2] Set("SIP/703-00000010", "VMGAIN=""") in new stack
-- Executing [s@macro-vm:3] GotoIf("SIP/703-00000010", "1?vmx,1") in new stack
-- Goto (macro-vm,vmx,1)
-- Executing [vmx@macro-vm:1] Set("SIP/703-00000010", "MEXTEN=501") in new stack
-- Executing [vmx@macro-vm:2] Set("SIP/703-00000010", "MMODE=CONGESTION") in new stack
-- Executing [vmx@macro-vm:3] Set("SIP/703-00000010", "RETVM=") in new stack
-- Executing [vmx@macro-vm:4] Set("SIP/703-00000010", "MODE=unavail") in new stack
-- Executing [vmx@macro-vm:5] GotoIf("SIP/703-00000010", "0?chknomsg") in new stack
-- Executing [vmx@macro-vm:6] Set("SIP/703-00000010", "VM_OPTS=") in new stack
-- Executing [vmx@macro-vm:7] GotoIf("SIP/703-00000010", "0?s-CONGESTION,1") in new stack
-- Executing [vmx@macro-vm:8] GotoIf("SIP/703-00000010", "1?notdirect") in new stack
-- Goto (macro-vm,vmx,10)
-- Executing [vmx@macro-vm:10] NoOp("SIP/703-00000010", "Checking if ext 501 is enabled: disabled") in new stack
-- Executing [vmx@macro-vm:11] GotoIf("SIP/703-00000010", "1?s-CONGESTION,1") in new stack
-- Goto (macro-vm,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-vm:1] Macro("SIP/703-00000010", "get-vmcontext,501") in new stack
-- Executing [s@macro-get-vmcontext:1] Set("SIP/703-00000010", "VMCONTEXT=default") in new stack
-- Executing [s@macro-get-vmcontext:2] GotoIf("SIP/703-00000010", "0?200:300") in new stack
-- Goto (macro-get-vmcontext,s,300)
-- Executing [s@macro-get-vmcontext:300] NoOp("SIP/703-00000010", "") in new stack
-- Executing [s-CONGESTION@macro-vm:2] VoiceMail("SIP/703-00000010", "501@default,u""") in new stack
-- <SIP/703-00000010> Playing 'vm-theperson.gsm' (language 'en')
== Spawn extension (macro-vm, s-CONGESTION, 2) exited non-zero on 'SIP/703-00000010' in macro 'vm'
== Spawn extension (macro-exten-vm, s, 18) exited non-zero on 'SIP/703-00000010' in macro 'exten-vm'
== Spawn extension (from-internal, 501, 1) exited non-zero on 'SIP/703-00000010'
-- Executing [h@from-internal:1] Macro("SIP/703-00000010", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/703-00000010", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/703-00000010", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/703-00000010", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup("SIP/703-00000010", "") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/703-00000010' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/703-00000010'
[/quote]

You haven't gone back far enough in the log but phonebuff is on the money... your latency is too great for reliable conversation. Aside from that, your remote extension was not registered to your asterisk at the time of the call.

I don't have travelling man either so I don't know what is installed on your laptop. Have you tried using a softphone (say X-lite or Zoiper) from the laptop?

Dallas

 

[dandy_don]

I've been using Twinkle, on an ubuntu laptop (64-bit, 8GB ram, dual core...).

I'll try another softphone and report findings.

I've got Skype working of this server and it works great as do all of the other phones located inside the firewall.

Thanks,
Don

 

[dandy_don]

What ports do I need to have open on the linux laptop? This is the host for the external softphone?

Thanks,
Don

 

[dandy_don]

Perhaps this is related...

If I add externip or externhost in the sip_custom.conf file, then all outbound calls fail and skype doesn't work any longer.

Don

 

[dandy_don]

Well -- I installed Zoiper today but still had the exact same results. Phoning into the system works fine, audio both ways, etc., but still can not dial out to the external softphone. I tried this on different laptop, also with Zoiper and then again with Twinkle.

I then tried four other wifi hotspots and still the exact performance.

I also experimented with settings on the firewall, but again, nothing I tried fixed the problem.

Are there any gurus that I could pay to help me figure this thing out?!?

Thanks,
Don

 

[dandy_don]

OK -- I did some additional testing this afternoon... From my laptop, I started firestarter, then accessed Travling Man, started zoiper, registered, and made several successful calls into the phones behind the firewall, and made a successful call from the softphone to my cell phone...

There were never any network requests from the server -- I understand that to "qualify" this extension, the server will send out up to 7 qualify queries, then repeat the process every 60 seconds. This never happened.. There did not ever appear to be any requests from the server to the softphone.


I'm stumped. I followed Ward's instructions, have switched to an approved DLink firewall, forwarded the appropriate ports, but can't dial the remote extension. I read through all of the postings that I can find, tried everything that seemed appropriate, but I have not run across this problem before. I've always had two-way audio, but I simply can't dial the external sip softphone...

Now -- PLEASE -- Would Someone/Anyone chime in. I need help resolving this. I've been asking for help. I'm even willing to pay for it. There are no responses to that either...

Certainly there are folks that must have some ideas they could share -- please do. I realize my questions are probably plenty stupid and I've been begging for help...


Thanks...

 

[saladman02]

Interesting - I have a nearly identical problem. I am using a nortel 1535 phone as the remote extension. It can call anyone it likes. But should you try to call it, I get a congestion error. Something is up indeed.

 

[dandy_don]

Saladman -- Thanks for posting. I was starting to think I was the only one with this problem.

The odd thing is that it works very well (one way). I have Skype running as well and it works great in and out.

I've been going through all of the suggested settings, etc., for those folks with one-way audio (which isn't the problem). If configure either sip_nat.conf or sip_custom.conf as suggested with my external ip address, I end up breaking outward Skype connectivity.

I've played with just about every setting on the firewall but it didn't ever fix it. This weekend I'm going to shut everything down and connect the PBX server directly to the cable modem (without any firewall hardware) and do some testing between the remote softphone and my cell phone. If I can dial out to the softphone, then that would indicate the there is something amiss with the firewall. If it doesn't make any difference, that should suggest that the problem is somewhere with the server configuration.

So far I haven't gotten very much input which has been somewhat frustrating...

I'll post my results.

Thanks,
Don

 

[wardmundy]

Hate to say it but we've seen this with some DSL providers. It's almost like they're blocking a port... but we all know that would not be legal.

 

[dandy_don]

Ward,

I'm using a timewarner cable modem... I had att DSL but changed over approximately two years ago.

I haven't tried getting into the cable modem configuration -- Do you think the cable modem might be blocking ports? Everything else just works fine...

Thanks,
Don

 

[Fala]

I'm having the same problem as you outlined. I had this working with "Running Asterisk Version = 1.8.4", before upgrading(New Install) to "Running Asterisk Version = 1.8.5.0". I'm using Comcast Broadband connection...not DSL!