Call disconnected after two minutes

Asterisk Asterisk Support
1

Hello,
first i want to say sorry for my bad english. So now i want to describe my problem. I have an asterisk 1.8 running on an router. All is fine i can make calls but every time after two minutes the call will be disconnected! I have no idear what is the problem with my configuration.

My sip.conf:

[general]
port=5060
bindport=5060
bindaddr=0.0.0.0
svrlookup=no
register => VORWAHLRUFNUMMER:PASSWORT:VORWAHLRUFNUMMER@VORWAHL.sip.arcor.de/VORWAHLRUFNUMMER

[arcor]
context=ankommend
type=peer
username=0xxxxxxxxxxx
fromuser=0xxxxxxxxxxx
authuser=0xxxxxxxxxxx
domain=0xxxx.sip.arcor.de
host=0xxxx.sip.arcor.de
fromdomain=0xxxx.sip.arcor.de
secret=pw
dtmfmode=rfc2833
nat=no
canreinvite=no
insecure=invite

My extensions.conf

[ankommend]
exten => _0xxxxxxxxxx,1,Dial(SIP/${EXTEN},30,r)

and the output from the console:

  == Using SIP RTP CoS mark 5
    -- Executing [0xxxxxxxxxxxx@ankommend:1] Dial("SIP/arcor-00000002", "SIP/0xxxxxxxxxxxx,30,r") in new stack
  == Using SIP RTP CoS mark 5
[Sep  9 04:34:35] ERROR[2416]: netsock2.c:245 ast_sockaddr_resolve: getaddrinfo("0xxxxxxxxxxxx", "(null)", ...): Name or service not known
[Sep  9 04:34:35] WARNING[2416]: chan_sip.c:5223 create_addr: No such host: 0xxxxxxxxxxxx
[Sep  9 04:34:35] WARNING[2416]: acl.c:708 ast_ouraddrfor: Cannot connect
[Sep  9 04:34:35] WARNING[2416]: chan_sip.c:3280 __sip_xmit: sip_xmit of 0x5f89c0 (len 768) to (null) returned -1: Invalid argument
    -- Called 0xxxxxxxxxxxx
[Sep  9 04:34:36] WARNING[2412]: chan_sip.c:3280 __sip_xmit: sip_xmit of 0x5f89c0 (len 768) to (null) returned -1: Invalid argument
[Sep  9 04:34:37] WARNING[2412]: chan_sip.c:3280 __sip_xmit: sip_xmit of 0x5f89c0 (len 768) to (null) returned -1: Invalid argument
[Sep  9 04:34:39] WARNING[2412]: chan_sip.c:3280 __sip_xmit: sip_xmit of 0x5f89c0 (len 768) to (null) returned -1: Invalid argument
  == Spawn extension (ankommend, 0xxxxxxxxxxxx, 1) exited non-zero on 'SIP/arcor-00000002'
[Sep  9 04:34:43] WARNING[2412]: chan_sip.c:3280 __sip_xmit: sip_xmit of 0x5f89c0 (len 768) to (null) returned -1: Invalid argument
[Sep  9 04:34:51] WARNING[2412]: chan_sip.c:3280 __sip_xmit: sip_xmit of 0x5f89c0 (len 768) to (null) returned -1: Invalid argument
[Sep  9 04:35:07] WARNING[2412]: chan_sip.c:3280 __sip_xmit: sip_xmit of 0x5f89c0 (len 768) to (null) returned -1: Invalid argument
[Sep  9 04:35:07] WARNING[2412]: chan_sip.c:3551 retrans_pkt: Retransmission timeout reached on transmission 293d0c471edf9d9a47a370fc70f99bee@(null) for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response

PLEASE can anyone try to help me? Thanks a lot

2

It should have failed much faster!

There is no 0xxxxxxxxxxxx section in sip.conf and that is not a valid sub-domain of your default domain.

More likely you meant to use Dial(SIP/[color=#FF0000]arcor/[/color]0xxxxxxxxxxxx)

3

Hey,
thank you for your answere. So i don’t under stand what i have to do? Is the error in the sip.conf or is the error in the dial-syntax?

0xxxxxxxxxxxx is my sip-user so what do you mean with “there is no section in sip.conf”?
What i have found in the internet is this syntax for the dial:
exten => SIPUSER,1,Dial(SIP/30)
this one i have tried with the same result.
and you mean i have to use:
exten => SIPUSER,1,Dial(SIP/arcor/SIPUSER) right?

Sorry that i dont unterstand waht you mean and thanks for your help.

4

I can’t tell for certain whether the error is in sip.conf or extensions.conf, although common sense suggests that it is unlikely that you would have a SIP peer named after a PSTN number.

The syntax for SIP dial strings is:

SIP/{||}[/]

{||} choose one of the options.
[] optional
<> meta-name.
means a device identified by a sip.conf or users.conf section name.

There is also an alternative syntax, using “@”, with the forwarded digits before the device.

5

Try replacing the existing “exten => _0xxxxxxxxxx,1…” entry with this:

exten => _0xxxxxxxxxx,1,Dial(SIP/arcor/${EXTEN},30,r)