All Outbound calls dropped after 32 seconds

[beshara1]

All calls are dropped after exactly 32 seconds. Voice mail included. The app runs normally, but the call just beeps and drops. This started happening randomly. Whats going on?

Also, starting today, I can no longer hear ringing when I dial someone's number. I can only hear when they pick up. Suggestions?

[beshara1]

I disabled the firewall on my router and now calls last longer than 32 seconds.

I still have the problem of not hearing outgoing calls until 5 seconds after the connection is made. I don't hear any ringing or anything in between.

[Phoneuser]

I have the same problem: Every call drop after 30-32 sec

This happen since the installation of the new version of Freephoneline.
The old version always worked good.
I tried to reinstall the old version, but I cant get it to connect to the server.

This is very annoying.

Any real solution, simple?

Thx

[tlon]

Problem here: no sound after 30 SECONDS. (counter still continues so the call is not completely dropped.) This, after fongo support solved the problem of no audio-in sound (by forcing registration on their side, whatever that means.)
If anyone got this fixed and is willing to share here, please do. Or does this have to get solved by fongo on their side? (while I'm awaiting fongo support's 3rd response by PM - after the audio-in problem now solved.) Thanks. P.S. also no ringing when dialing out... but I can live with that (just look at the counter instead)... I'm happy if the the call through at all, as many times they don't... (BTW. as implied by "looking at the counter"... using the desktop software only. no ATA.)

[Fongo Support]

Problem here: no sound after 30 SECONDS. (counter still continues so the call is not completely dropped.) This, after fongo support solved the problem of no audio-in sound (by forcing registration on their side, whatever that means.)
If anyone got this fixed and is willing to share here, please do. Or does this have to get solved by fongo on their side? (while I'm awaiting fongo support's 3rd response by PM - after the audio-in problem now solved.) Thanks. P.S. also no ringing when dialing out... but I can live with that (just look at the counter instead)... I'm happy if the the call through at all, as many times they don't... (BTW. as implied by "looking at the counter"... using the desktop software only. no ATA.)

Hi there,
I am looking into this and I just send u a quick note.

Thanks,

[balsamboy]

I have had this happening since the changover a month ago. Calling computer to computer is fine but computer to landline will only last 31 seconds. I have tried everything, the free phone worked like a charm before the "change" .

Too bad..............

[Funkytown]

I have had this happening since the changover a month ago. Calling computer to computer is fine but computer to landline will only last 31 seconds. I have tried everything, the free phone worked like a charm before the "change" .

Too bad..............

Sounds to me like a firewall/port issue?

[tony_qc]

I have already answered those questions in the past, the drop calls after 32 sec are a MODEM problem.

http://forum.fongo.com/viewtopic.php?f=8&t=1845
or
http://forum.fongo.com/viewtopic.php?f=10&t=971

[tony_qc]

I have received many emails about this and here is what happen:

You want a router with SIP ALG disabled to make voip calls: Netgear router are very good for that. (they have the option From Wan Setup Menu "Disable SIP ALG")

Here a short list of enabled SIP ALG router:

Dlink
Models: DIR-655

Motorola
Models: SBG6580 (SurfBoard Extreme Wireless Cable Modem Gateway)

No Registeration possible behind NAT as the device changes Call-ID and causes the responses to be discarded by SIP clients/ATAs
No Solution at this time (SIP ALG, called SIP Pass Through, can not be disabled) .
Must disable NAT and put the device in bridge mode.


SpeedTouch
Models: ST530 v6 (firmware >= 5.4.0.13) comes with SIP ALG enabled by default.
NAT type: symmetrical
Issues:

No incoming calls.
It replaces the private IP appearing in SIP headers with the public IP using a dumb text replacement. If for example the private IP appears in the "Call-ID" it replaces it too (that it's completely unnecessary).

To disable SIP ALG:
~# telnet router
-> connection unbind application=SIP port=5060
-> saveall

Model: speestream 6520

- Click the Start button, then select Run.
- Type: telnet 192.168.254.254 (IP of Your Speedsream but im using 192.168.2.1 because I can use my old file for the hide QOS ) then select OK
- Type username: admin and password admin
- Type: cfg upro{ac2{qos=y
- Type: cfg save
- Type: cfg napt{dsip=y
- Type: cfg save
- Type: do reboot

Zyxel
Models: 660 family comes with SIP ALG enabed by default.
NAT type: symmetrical
Issues:

No incoming calls.
SIP protocol broken making 50% of outgoing calls impossible because the wrong values are inserted into SIP headers.

To disable SIP ALG:
~# telnet router
Menu option "24. System Maintenance".
Menu option "8. Command Interpreter Mode".
ip nat service sip active 0


Netgear
Models: WGR614v9 Wireless-G Router, DGN2000 Wireless-N ADSL2+ Modem Router
Firmware V1.0.18_8.0.9NA
To disable SIP ALG: From Wan Setup Menu, NAT Filtering, uncheck the box next to "Disable SIP ALG"


SMC
Models: ToDo
NAT type: No symmetrical
Issues:

The ALG doesn't replace the private address in "Call-ID" header (that is correct) but it does replace the "call-id" value in "Refer-To" header so SIP transfer is broken.

To disable SIP ALG: ToDo no ALG related options found via web and telnet. No idea of how to dissable it.


Linksys
Models: WRV200, WRT610N
NAT type: Symmetrical
Issues:

The ALG replaces the private address in "Call-ID" header (not needed at all). Some phones (as Linksys with latest firmware) encode the "Call-ID" value in the "Refer-To" header (by escaping the dots) so the private IP appearing there is not replaced with the public IP. This causes that the call transfer fails since the proxy/PBX/endpoint will not recognize the dialog info.

To disable SIP ALG: ToDo no ALG related options found via web and telnet. No idea of how to disable it.
To disable SIP ALG on WRT610N: Web Interface: Administration, Management, under side heading 'Advanced Features' SIP ALG, can be disabled.

Peplink Multi-WAN routers
Models: All multi-WAN models
To disable SIP ALG, go to http://<router.LAN.IP>/cgi-bin/MANGA/support.cgi
Click the "Disable" button under "SIP ALG Support"
Issues:

I'm not aware of any SIP ALG issues, but if you just want to turn it off, here you go.



Fortinet
Models: All models come with SIP Helper enabled by default

To disable SIP helper:
~# telnet firewall
config system settings
set sip-helper disable
set sip-nat-trace disable
end

config system session-helper
delete 12
end


The preferred solution is to configure the SIP ALG. Policies that use the SIP ALG will not use SIP helper. Full documentation at http://docs.fortinet.com then pick FortiOS for the version on your device, then VoIP solutions: SIP.


Cisco
Models: 800 Series

To disable the NAT services for SIP in IOS, just run these commands:


no ip nat service sip tcp port 5060
no ip nat service sip udp port 5060


Juniper / Netscreen
Models: SSG Series

To disable SIP ALG:
In the Web interface: Security -> ALG

[tony_qc]

Also, the PAP2T need to be configured this way since 1 mount:

System:
HostName: PAP2T
Primary NTP Server: 64.90.182.55
Secondary NTP Server: 192.43.244.18

SIP:
RTP Packet Size: 0.020

Regional:
Ring Waveform: sinusoid
Ring Tone Frequency: 52 (if no ring from your phone try between 10-52)
Ring Tone Voltage: 90 (if no ring from your phone try between 40-90)
Time Zone: GMT -5:00 ( for est time)
Daylight Saving Time Rule: start=3/8/7/2:00;end=11/1/7/2:00;save=1
FXS Port Input Gain: default: -3 (Volume)
FXS Port Output Gain: default: -3 (Volume)

Line1:
Line Enable: Yes
Proxy: voip.freephoneline.ca
Use Outbound Proxy: NO
Outbound Proxy: blank
Registration Expires: 3600
Display Name: **************** (your name)
User ID: 1********** (your FreePhoneLine number ( 11 digits show in http://freephoneline.ca/myDetails))
Password: ******** (your password show in http://freephoneline.ca/showSipSettings )
Preferred Codec: G711u
Use Pref Codec Only: NO
Dial Plan: (911|[2-9]xxxxxxxxx|1xxxxxxxxxx|011xxxxxxxxxxxx.|98*|[6-7]x*xxxxxxxxxxx.)

http://forum.fongo.com/viewtopic.php?f=15&t=310