Etisalat SIP Line - Cannot make outbound calls but incoming works fine

Asterisk Asterisk SIP
1

Hello,

Etisalat SIP Line - Cannot make outbound calls but incoming works fine.

@asterisknooob
I saw your post PJSIP Dialplan For Outbound & Inbound and configured the same way but just cannot get it working.

Please help.

2

Help requires showing the logs and describing what actually happens.

3

Incoming calls work fine but outgoing call error “All circuits are busy now, please try again later”

Please let me know which log file to upload.

4

Enable the full log. Make sure verbosity is at least three. Enable detailed logging for your SIP channel driver (e.g “pjsip set logger on”, for chan_pjsip). Then provide the contents of /var/log/asterisk/full

5

Hello can you show me your settings so I can figure out.

6

Hello,

My system here is FreePBX/Asterisk.

I spoke to Etisalat technical support yesterday and they said, my REGISTER and INVITE requests are not going to the same IP and that’s the reason outbound calls are failing.
The SIP proxy address vims-siptrunk.etisalat.ae resolves to the following IPs.

vims-siptrunk.etisalat.ae. 36911 IN A 10.238.70.33
vims-siptrunk.etisalat.ae. 36911 IN A 10.238.70.97
vims-siptrunk.etisalat.ae. 36911 IN A 10.238.70.201
vims-siptrunk.etisalat.ae. 36911 IN A 10.238.69.209
vims-siptrunk.etisalat.ae. 36911 IN A 10.238.69.33
vims-siptrunk.etisalat.ae. 36911 IN A 10.238.66.57

I have also attached a detailed SIP trunk registration document from Etisalat.

Here is my configuration:

[Etisalat-SIP]
type=registration
transport=0.0.0.0-udp
outbound_auth=Etisalat-SIP
retry_interval=60
fatal_retry_interval=30
forbidden_retry_interval=30
max_retries=10000
expiration=3600
auth_rejection_permanent=no
line=yes
endpoint=Etisalat-SIP
contact_user=+9714XXXXX00
server_uri=sip:ims.etisalat.ae:5060
client_uri=sip:+9714XXXXX00@ims.etisalat.ae:5060
outbound_proxy=sip:vims-siptrunk.etisalat.ae:5060

[Etisalat-SIP]
type=auth
auth_type=userpass
password=**********
username=+9714XXXXX00@ims.etisalat.ae

[Etisalat-SIP]
type=aor
qualify_frequency=60
contact=sip:+9714XXXXX00@ims.etisalat.ae:5060
outbound_proxy=sip:vims-siptrunk.etisalat.ae:5060

[Etisalat-SIP]
type=identify
endpoint=Etisalat-SIP
match=vims-siptrunk.etisalat.ae

[Etisalat-SIP]
type=endpoint
transport=0.0.0.0-udp
context=from-pstn
disallow=all
allow=alaw,ulaw,gsm,g726,g722,g723,g729
aors=Etisalat-SIP
send_connected_line=no
rtp_keepalive=0
language=en
outbound_proxy=sip:vims-siptrunk.etisalat.ae:5060
outbound_auth=Etisalat-SIP
from_domain=ims.etisalat.ae
contact_user=+9714XXXXX00
user_eq_phone=no
t38_udptl=no
t38_udptl_ec=none
fax_detect=no
trust_id_inbound=no
t38_udptl_nat=no
send_pai=yes
direct_media=no
rtp_symmetric=yes
dtmf_mode=auto

7

Sorry attaching missed Etisalat SIP trunk registration document.
CPE_Directions_List_Update_15_09_2021.zip (369.2 KB)

8

Alright you are using FreePBX right?
So it’s GUI based you can share screenshots with hide sensitive info.

9

PJSIP conforms to the SIP specification for locating SIP servers, which means it doesn’t guarantee that the same IP address used for a registration will be used for calls. There’s no option to disable this currently, so you’d have to force it to use a single IP address in configuration or through local DNS records.

10

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