How to active PRACK/UPDATE for SIP

[celsoannes]

Hi,
I am using Asterisk 13.6.0, and I have a SIP TRUNK with an ISP.

The calls are hangup in the middle of the call, the ISP told me that I have to active on the INVITE support to the 100rel, PRACK and UPDATE.

I dont know how to do that, at moment my sip.conf is like that:

[general]
context=interno
allowguest=no
bindport=5060
binaddr=0.0.0.0
language=pt_BR
disallow=all
allow=alaw
dtmfmode=rfc2833
nat=force_rport,comedia
limitonpeer=yes
srvlookup=no
rtptimeout=60
rtpholdtimeout=300
tos_sip=cs3
tos_audio=ef
tos_video=af41
t38pt_udptl=yes
ignoresdpversion=yes

[voxip]
type=peer
host=10.150.66.16
outboundproxy=10.150.66.16
fromuser=5430468800
insecure=invite,port
qualify=yes
port=5060
nat=no
disallow=all
allow=g729,alaw
prack=yes
reinvite=yes
canreinvite=no
username=5430468800
secret=5430468800
context=entrada-voxip
rtptimeout=10

I didn’t see the option prack on the sip.conf, but the ISP told me that I have to include this on my sip.conf.

The ISP sent me an image showing that my Asterisk isn’t sent the UPDATE, what is needed to keep the call going on otherwise the ISP drop the call.

[jcolp]

The chan_sip module does not support PRACK. It is supported in chan_pjsip.

[david551]

The ITSP isn’t SIP compliant as support for UPDATE isn’t mandatory, nor is 100Rel support.

chan_sip is deprecated, doesn’t have full UPDATE support, so does not advertise it, and does not have 100Rel support.

There is also no reinvite configuration option and canreinvite is an obsolete name for directmedia.

Why are you disable port number checks?

chan_pjsip does have 100Rel support. I don’t know if it advertises UPDATE support.

[jcolp]

I believe it does advertise it, but don’t know for certain.

[celsoannes]

@jcolp

The chan_sip module does not support PRACK. It is supported in chan_pjsip.

I have none experience on chan_pjsip, could you please, remake this configuration for pjsip.conf for me.

I thank you in advance.

[jcolp]

The project provides numerous wiki pages[1] talking about configuration, as well as a tool that can be used to convert a sip.conf to a pjsip.conf.

[1] https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip

[celsoannes]

What do you mean with?

Why are you disable port number checks?

[johnkiniston]

David is referencing this line of your configuration file where you are disabling the port number check.

@celsoannes

insecure=invite,port

[celsoannes]

Thanks for clarifying.

The template was given to me by the ISP.

If the concern is about safety, the connection to the ISP is made locally, there is a modem with connect directly to the server network interface. It don’t go through internet.

If it was mentioned for other reason, please, clarify to me.

[david551]

ISPs have a habit of handing out confiugrations that are about ten years out of date and are designed to avoid support calls rather than make systems secure. With anything with a name like insecure, you should only be disabling checks that you actually need to disable, and unless the ITSP originates calls from ports other than 5060, you don’t need insecure=port.

[celsoannes]

Thanks for the clarification.