10 seconds delay to make a call, receiving call is OK.

[llrf]

The delay of making a call is around 10 seconds. There is no delay to receive a call. I am using 8M cable internet, TP link WR740N router. Voip device is an old one from CIK telecom, I don't know exact the model number.


Here is the record:

[APP] >chan 0 is handed up,status:0

[APP] user fast dial number

[APP] chan 0 fast dial number

[APP] >chan 0 gets the key=1 down,status=5

[APP] >chan 0 gets the key=5 down,status=5

[APP] >chan 0 gets the key=1 down,status=5

[APP] >chan 0 gets the key=4 down,status=5

[APP] >chan 0 gets the key=3 down,status=5

[APP] >chan 0 gets the key=4 down,status=5

[APP] >chan 0 gets the key=0 down,status=5

[APP] >chan 0 gets the key=4 down,status=5

[APP] >chan 0 gets the key=7 down,status=5

[APP] >chan 0 gets the key=1 down,status=5

[APP] >chan 0 gets the key=1 down,status=5

[APP] Start to search dial rule table

[APP] Finish the lookup job

[APP] we choose---------.protocol:-1

[APP] PNum->number:15143404711

[APP] Starting Sip Call

[APP] pub sip register num get:15145583655

[SIP] sipCallStart:appchan:1,chan:0

[SIP] callee: 15143404711[sip:15143404711@voip.freephoneline.ca]

[SIP] calling: 15145583655

[SIP] add g729 in sip client

[SIP] add g711a in sip client

[SIP] add g711u in sip client

[SIP] add dtmf2833 in sip client

[SIP] SIP1 register user name:15145583655

[SIP] SIP1 informative:100

[SIP] SIP1 register user name:15145583655

[SIP] SIP1 informative:100

[SIP] SIP1 register user name:15145583655

[SIP] SIP1 informative:183

[SIP] 183,so we try to connectd the rtp channel

[SIP] invite accept

[SIP] Sip Rtp:Local port:10006,Peer Port:54802:sipChan->localPort:10006

[APP] ============Sip://we will start the talk with cap:18 ===the audio list=== 4->g723 8->g711a 0->g711u 18->g729 ========

[DSP] 729 payload len change to 20

[SIP] invite accept

[SIP] Sip Rtp:Local port:10006,Peer Port:54802:sipChan->localPort:10006

[SIP] teletalkstart:app:1;chan:0;lport:10006;rport:54802

[APP] chan 0 ivr send prepare

[APP] ============Sip://we will start the talk with cap:18 ===the audio list=== 4->g723 8->g711a 0->g711u 18->g729 ========

[DSP] 729 payload len change to 20

[SIP] teletalkstart:app:1;chan:0;lport:10006;rport:54802

[SIP] process SIP1_Register

[SIP] end process

[APP] >chan 0 is handed down

[APP] >chan 0 is handed up,status:0

[APP] user fast dial number

[APP] chan 0 fast dial number

[APP] >chan 0 is handed down

[mysaurin]

I followed the steps from this link - http://forum.fongo.com/viewtopic.php?f=15&t=7962 and it solved my recent problems of delay while incoming & outgoing calls and caller ID
Thanks

[llrf]

After I changed to Grandstream HT502, all of these problems disapper.