SIP Credentials and PJSIP (FreePBX)

SbS

Hi,

I'm trying to use the SIP credentials provided by Swisscom for an inOne line with a FreePBX running asterisk 13.22 and using chan_pjsip (and not chan_sip).

I'm struggling to find the configuration for the trunk because of the bad username containing the @ sign.

Has anyone managed to use the trunk with a pjsip trunk?

Thanks

Giuseppe

fabinode

Salut!

Je suis aussi dessus le même problème, j'arrive à recevoir un appel, mais pas à en passer...

As-tu trouver une solution?

gravasio

Hi,

are you using chan_sip or chan_pjsip?

fabinode

J'utilise CHAN-SIP mais si il y a une solution avec PJSIP, je suis prenneur!

gravasio

chan_sip is working without problems with this setup:

OUTGOING:

user=+$NUMBER
type=peer
srvlookup=yes
secret=$SECRET
outboundproxy=fs1.ims.swisscom.ch
nat=yes
insecure=invite, port
host=swisscom.ch
fromuser=+$NUMBER
fromdomain=swisscom.ch
dtmfmode=auto
disallow=all
defaultuser=$LOGIN@swisscom.ch
canreinvite=no
allow=alaw & ulaw & g729 & gsm & slinear & ulaw

INCOMING:

type=peer
host=fs1.ims.swisscom.ch
fromdomain=swisscom.ch
disallow=all
allow=alaw & ulaw & g729 & gsm & slinear & ulaw

REGISTRATION STRING:

+$NUMBER@swisscom.ch:$SECRET:$LOGIN@fs1.ims.swisscom.ch/+$NUMBER

Obviously you must set the Variables with your line SIP credentials

It's imperative that chan_sip is using the 5060 port, otherwise the trunk will not work fully, due to a swisscom bug!

Good luck 🙂

fabinode

yes thank you!

I have make a test with FreePBX 14 and asterisk 13 and 16 but same problem.

My config:

create a trunk CHAN SIP

OUTGOING:

user=+4121xxxxx
type=peer
srvlookup=yes
secret=xxxxxxxxxxxxxxxx
outboundproxy=fs1.ims.swisscom.ch
nat=yes
insecure=invite, port
host=swisscom.ch
fromuser=+4121xxxxxxxxxx
fromdomain=swisscom.ch
dtmfmode=auto
disallow=all
defaultuser=NCxxxxxxxxxxxxxx@swisscom.ch
canreinvite=no

INCOMING:

type=peer
host=fs1.ims.swisscom.ch
fromdomain=swisscom.ch
disallow=all
allow=alaw&ulaw&g729&gsm&slinear&ulaw

+4121xxxxx@swisscom.ch:mypasswordxxxx:NCxxxxxxxxxxxxxxxxx@swisscom.ch@fs1.ims.swisscom.ch/+4121xxxxxxxx

And i have modifiy in Settings - ASTERISK SIP SETTINGS - Registration Default Expiry 120 default to 180

All incomming call are stable and good quality, but my problem is for outgoing.

I have a response in log:

-- Called SIP/TR1_Sw_OUT/5478966
[2019-10-19 20:09:37] WARNING[4807][C-00000001]: chan_sip.c:24055 handle_response_invite: Received response: "Forbidden" from '<sip:+412*******@swisscom.ch:5160>;tag=as7aa9544d'
\== Everyone is busy/congested at this time (1:0/0/1)

why port 5160?

fabinode

YES,

I have reconfigured my FreePBX and apply the port 5060 to CHAN SIP and the 5160 to the PJSIP and it's work now.

thank you for your help.

I am now going to do some long-term testing and see how stability is going...

I'll be back in a while.

gravasio

Hi, the port is the key problem here.

In freepbx 14 the default sip driver is PJSIP that is configured to use the default SIP port (5060) and the old chan_sip is using the alternate 5060.

You need to change the configuration in "Asterisk SIP settings" but be aware that this will impact all the endpoints that you are using.

I looked a lot into finding a solution to have the trunk working with chan_pjsip but as of today I haven't found anithing working!

gravasio

I'm using this configuration in production from more than 6 months now all is working flawlessly.

fabinode

OK thanks.

Trunk use CHAN-SIP with port 5060 for extern ( swisscom force ).

All phone and extension with pjsip with SIP port 5160

But only for new install, or reconfigure all phone..

fabinode

Before SIP credential, i have test and it's work fine with:

FreePBX CHAN_SIP trunk to SIP with Centro Business V2 of Swisscom

FreePBX CHAN_SIP trunk to SIP Fritzbox 7690

FreePBX PJSIP_SIP trunk to SIP Fritzbox 7690

FreePBX CHAN_SIP trunk to SIP Internet BOX V2 Swisscom

FreePBX PJSIP_SIP trunk to SIP Internet BOX V2 Swisscom