SIP Calls Not Working

Asterisk Asterisk Support
1

Hello,

I have two sip phones (zoiper). Earlier these used to communicate using the settings below for sip.conf and extensions.conf and now we asterisk 1.8.29.0, so these phones have stopped communicating. My question is that does 1.8.29.0 release require any more changes to be done to the sip.conf and extensions.conf to make the below work ?

The sip.conf contains following enteries

[100]
type=friend
username=100
secret=100
host=dynamic
port=5060
dtmfmode=rfc2833
fromdomain=dynamic
nat=no
canreinvite=false
context=exten-100

[101]
type=friend
username=101
secret=101
host=dynamic
port=5060
dtmfmode=rfc2833
fromdomain=dynamic
nat=no
canreinvite=false
context=exten-101

The extensions.conf contains

[exten-100]
exten => 101,1,Dial(SIP/101)
;exten => echo,1,Echo()
;exten => busytone,1,Playback(moh)
;exten => 101,n,Hangup()
exten => 100,1,Answer()
exten => 100,n,Hangup()

[exten-101]
exten => 101,1,Answer()
exten => 101,n,Hangup()
exten => 100,1,Dial(SIP/100)
;exten => _x.,1,Playback(moh)

2

I don’t think any of the deprecated names you are using have actually been discontinued in that version, and, although probably useless, username (defautluser) and fromdomain are probably not doing any harm. Therefore, I think you are going to have to provide sufficient logging to demonstrate the exact problem.

Better practice: type=peer

Deprecated: username (defaultuser); canreinvite (directmedia)

Probably meaningless: fromdomain=dynamic

Bad practice: guessable device names.

3

Logging Info

== Using SIP RTP CoS mark 5
– Executing [100@exten-101:1] Dial(“SIP/101-00000014”, “SIP/100”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/100
– Registered SIP ‘101’ at 115.252.66.70:55258
[Sep 1 18:10:20] NOTICE[4629]: chan_sip.c:25735 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 101
[Sep 1 18:10:43] WARNING[4629]: chan_sip.c:3982 retrans_pkt: Retransmission timeout reached on transmission 5f0235b842799d285a70eb2d452974fb@dynamic for seqno 102 (Critical Request) – See wiki.asterisk.org/wiki/display/ … nsmissions
Packet timed out after 32000ms with no response
[Sep 1 18:10:43] WARNING[4629]: chan_sip.c:4011 retrans_pkt: Hanging up call 5f0235b842799d285a70eb2d452974fb@dynamic - no reply to our critical packet (see wiki.asterisk.org/wiki/display/ … nsmissions).
– SIP/100-00000015 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Auto fallthrough, channel ‘SIP/101-00000014’ status is ‘CONGESTION’

4

Hi David55,

We changed type=friend to type=peer in sip.conf and we are getting error 403 now.

Regards

Deepak

5

The 403 means the device never registered, or given the other symptoms, registered with a contact address that was not the same as the actual address. Put back the type=friend, for the moment and concentrate on getting registrations to work.

The retransmission could also be be because a firewall is blocking the INVITE or the response to it.

You need to enable SIP debugging.

(The reason that using type=peer is better is that type=friend allows anyone to make calls if they know th user and password, with not even changing registrations signalling an attack, and probably more important, calls from other sources, presenting the device name as their caller ID will be rejected, because the the user name will take precedence over the IP address of the trunk, and the call will authenticated against the extension that matches the caller ID, not against the trunk, and will fail, because the ITSP will not know the local device’s password.)