Fongo Community Forums

I got perfectly "working" SPA3000 however there is one problem that I can't get rid of

Sometimes incoming call (only incoming) gets cut after 15-30 sec.
Looks like this will happen only if you use the VOIP number and after first call received from PSTN
All calls received after the first disconnection to PSTN are fine.
Next call received to VOIP registered provider will be cut but after that all calls to VOIP will work fine.

Any ideas what could be the problem?
The ATA is temporary in DMZ so this shouldn't be firewall problem

Thanks

I am sorry for the delay in responding, you can always try my suggestions here http://forum.fongo.com/viewtopic.php?f=6&t=7944

Good Luck

Thanks , will do

1. First suggestion is to check inside the modem/router for a setting "keep alive" - Enable this, it's basically telling the devices connected to the modem/router that there still here and alive and not to worry.
Enabled (10sec)

2. Second suggestion you might have to upgrade the modem/router firmware.
Latest

3. Third suggestion is to reduce the firewalls to minimum in the modem/router.
DMZ enabled

4. Fourth suggestion check your modem/router for a SIP ALG setting - Disable this.
Disabled but shouldn't matter since DMZ enabled

lifeisfun wrote:Enabled (10sec)

Try (60sec) this should work.

Unfortunately no change, will try to use sip debug.

Any ideas why it cut off the call ?
Thanks




[0:5060]<<ip:5060
ACK sip:2222222222@ip:5060 SIP/2.0

Via: SIP/2.0/UDP ip:5060;branch=z9hG4bK-d8754z-6941bd0134481612-1---d8754z-;rport

Via: SIP/2.0/UDP ip:5061;rport=5061;branch=z9hG4bK-b7ixozcd7tsgxh5m

Max-Forwards: 69

To: <sip:222222222222@ip>;tag=3ba38fee197cfa3di0

From: "222222222222"<sip:22222222222@ip>;tag=mge3kqcby2k5avmo.o

Call-ID: 11111111111@192.168.1.236~o

CSeq: 232 ACK

User-Agent: Sippy

Content-Length: 0




CC:Connected
[0:0]ENC INIT 0
[0:0]RTP Tx Up (pt=0->43d4085d:16452)
[0:0]RTCP Tx Up
[0:0]RTP Rx 1st PKT @16428(3)
[0:0]DEC INIT 0
[0]On Hook
[0:0]AUD Rel Call
[0:5060]->ip:5060
BYE sip:ip:5061 SIP/2.0

Via: SIP/2.0/UDP ip:5060;branch=z9hG4bK-e9dc9a22;rport

From: <sip:22222222222@ip>;tag=3ba38fee197cfa3di0

To: "3438832130" <sip:222222222@ip>;tag=mge3kqcby2k5avmo.o

Call-ID: 111111111111@192.168.1.236~o

CSeq: 101 BYE

Max-Forwards: 70

Route: <sip:ip:5060;lr>, <sip:ip:5060;lr;transport=UDP>

User-Agent: Sipura/SPA3000-3.1.7(GWc)

Content-Length: 0




[0:5060]<<ip:5060
SIP/2.0 200 OK

Via: SIP/2.0/UDP ip:5060;branch=z9hG4bK-e9dc9a22;rport=5060

To: "22222222222"<sip:22222222222@ip>;tag=mge3kqcby2k5avmo.o

From: <sip:222222222222@ip>;tag=3ba38fee197cfa3di0

Call-ID: 11111111111@192.168.1.236~o

CSeq: 101 BYE

Server: Sippy

Content-Length: 0




DLG Terminated
Sess Terminated
RSE_DEBUG: reference domain:voip.provider.ca
[0:5060]->ip:5060
NOTIFY sip:voip.provider.ca SIP/2.0

Via: SIP/2.0/UDP ip:5060;branch=z9hG4bK-8b6266c0;rport

From: 222222222 <sip:2222222222@voip.freephoneline.ca>;tag=19a51be41bfe29d1o0

To: <sip:voip.freephoneline.ca>

Call-ID: 11111111111@10.0.0.130

CSeq: 6831 NOTIFY

Max-Forwards: 70

Event: keep-alive

User-Agent: Sipura/SPA3000-3.1.7(GWc)

Content-Length: 0




[0:5060]<<ip:5060
SIP/2.0 200 OK

Via: SIP/2.0/UDP ip:5060;branch=z9hG4bK-8b6266c0;rport

From: 2222222222 <sip:222222222@voip.provider.ca>;tag=19a51be41bfe29d1o0

To: <sip:voip.provider.ca>;tag=deadbeef

Call-ID: 1111111111111111@10.0.0.130

CSeq: 6831 NOTIFY

Content-Length: 0




RSE_DEBUG: unref domain, voip.provider.ca
RSE_DEBUG: last unref for domain voip.provider.ca
CC:Clean Up
--- OBJ POOL STAT ---
OP:RTPRXB = 96 ( 96 192) OP:RTPREB = 40 ( 40 48)
OP:RTPTXB = 64 ( 64 108) OP:TIMEOU = 110 (120 40)
OP:SIPCOR = 0 ( 1 28) OP:SIPCTS = 32 ( 32 564)
OP:SIPSTS = 32 ( 32 3452) OP:SIPAUS = 2 ( 8 588)
OP:SIPDLG = 10 ( 10 140) OP:SIPSES = 12 ( 12 7920)
OP:SIPREG = 2 ( 4 252) OP:SIPLIN = 0 ( 2 128)
OP:STUNTS = 16 ( 16 68)
RSE_DEBUG: reference domain:voip.provider.ca
[0:5060]->ip:5060
NOTIFY sip:voip.provider.ca SIP/2.0

Via: SIP/2.0/UDP ip:5060;branch=z9hG4bK-509ee45d;rport

From: 2222222222 <sip:2222222222@voip.provider.ca>;tag=19a51be41bfe29d1o0

To: <sip:voip.provider.ca>

Call-ID: 111111111111@10.0.0.130

CSeq: 6832 NOTIFY

Max-Forwards: 70

Event: keep-alive

User-Agent: Sipura/SPA3000-3.1.7(GWc)

Content-Length: 0