Call not patching from outer network

Asterisk Asterisk Support
1

Greetings Everyone…
I searrched the forum and got related topics but still in sticky situation…

My router has public IP and asterisk plus cdma gateway is running on a static ip the When a call comes from a voip to router -> asterisk -> cdma gateway -> the caller tone is heard on the softphone(outside local network)…but after 1 or 2 seconds of call pathcing the call hangs up and cli says to visit hte sip retransmittin url…well the voice is audiable on the sofphone for the 1-2sec…after which the call is cut…
my sip.conf settings
[general]
disallow=all
allow=alaw
allow=ulaw
allow=all
qualify=yes
port=5060

I am using asterisk 11.6.0

Any idea what I am missing…I did all the router nat settings…

2

You are missing the SIP trace that shows which packet’s retransmission is timing out.

You are also missing the peer entries from sip.conf.

Your allow lines need cleaning up.

My guess, in this case, is that something doesn’t like COLP (connected line presentation) updates.

3

hello David…
my peer entry :
[400]
type=friend
secret=1234
host=dynamic
context=default

[401]
type=friend
secret=1234
host=dynamic
context=default

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.105:5060;branch=z9hG4bK47e1f128;received=x.x.x.x
Contact: sip:401@x.x.x.x:5060;transport=UDP
To: sip:401@x.x.x.x;transport=UDP;tag=b89e6308
From: sip:919509162345@x.x.x.x;transport=UDP;tag=as50622577
Call-ID: YzUzYTgxNzcwNTgzYjdiNzllN2Y1NjA0N2E2MmIxODY.
CSeq: 102 BYE
User-Agent: Zoiper r20647
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘269674511b18719a7612e7903f524fea@192.168.0.105:5060’ Method: INVITE

4

You are transmitting the 200 OK with a private use contact address. That suggests that there is NAT in your network, but you have configured no way for Asterisk to discover its public address.

5

Hello david …

ITS solved
I added following line in my sip.conf…and its working for now…
thanks for guidance…

bindport=5060
bindaddr=0.0.0.0
rtpkeepalive=60
localnet=192.168.0.0/255.255.0.0
localnet=10.0.0.0/255.0.0.0
localnet=172.16.0.0/12
localnet=169.254.0.0/255.255.0.0

But still I am confused how it happened…plz elaborate david …what happened…
thanks again.

6

I don’t know enough about your network topology to answer that, and I’m not sure that it is worth working out the answer, even if I did know.

7

Greetings again…
jut in case if anyone is faacing problem of not registering sip phones from particual country…check for the sip port…in some countries the default 5060 …it is blocked

8

Typically, if it is blocked, it is either illegal to use VoIP over WANs in that country, or it is a breach of the ISP’s terms of service.

9

Greetinngs everybody…

Well what I have gained from the above fact is "you should first collect all the port and protocol information for the area you are plannig to plant your solution …before buying any gateway or any other hardware else consider the alternatives if SIP port is blocked ".
Well there are ohter alternatives…if SIP port is blocked…
1.use proxy / try different port(instead of default 5060).
2.use VPN.