Twilio outbound call - endpoint issue

Asterisk Asterisk SIP
1

I am trying to trigger a make outbound call using Twilio sip trunk. Done Tried multiple options to sort the issue every time I am making changes giving me a different error.

Getting below error:

Unable to create PJSIP channel - endpoint
Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)

Error from logs

 -- Executing [+1234567890@twlext:2] Dial("PJSIP/User1-00000000", "PJSIP/+1234567890@twl") in new stack
[2020-12-02 10:57:53] ERROR[26856]: chan_pjsip.c:2699 request: Unable to create PJSIP channel - endpoint 'twl' was not found
[2020-12-02 10:57:53] WARNING[26921][C-00000001]: app_dial.c:2576 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
    -- No devices or endpoints to dial (technology/resource)
    -- Executing [+1234567890@twlext:3] Answer("PJSIP/User1-00000000", "") in new stack
[2020-12-02 10:57:53] ERROR[26856]: pjproject: <?>: 	   icess0x7f67b894b038 ...Error sending STUN request: Operation not permitted
[2020-12-02 10:57:53] ERROR[26871]: pjproject: <?>: 	   icess0x7f67b894b038 ..Error sending STUN request: Operation not permitted
    -- Executing [+1234567890@twlext:4] Hangup("PJSIP/User1-00000000", "") in new stack
  == Spawn extension (twlext, +1234567890, 4) exited non-zero on 'PJSIP/User1-00000000'
[2020-12-02 11:02:47] WARNING[26870]: res_stun_monitor.c:138 stun_monitor_request: STUN poll failed. Re-evaluating STUN server address

Here is my pjsip.conf file

[twl]
type=aor
max_contacts=10
qualify_frequency=120
remove_existing=yes
contact=sip:NUMBER@SUB_DOMAIN.pstn.twilio.com

[twl]
type=endpoint
context=twl
username=USERNAME
secret=PASSWORD
host=SUB_DOMAIN.pstn.twilio.com
canreinvite=no
disallow=all
allow=ulaw
insecure=port,invite
fromdomain=SUB_DOMAIN.pstn.twilio.com
context=from-trunk
callerid="NAME" <+REGISTER_NUMBER>


[twl]
type=identify
endpoint=twl
match=SUB_DOMAIN.pstn.twilio.com
match=sip.twilio.com
2

These are some options that are invalid. There may be more, but these stand out. Are you sure that the whole section wasn’t thrown out as a result.

canreinvite isn’t even a current option for chan_sip, although I believe it is still accepted as an alias for a current one. chan_psjip never had the problem that existed in earlier versions of chan_sip and which required insecure=invite. It allows authentication to be controlled independently in each direction. Most people don’t need insecure=port, even in chan_sip, and I don’t know if it has any equivalent in chan_pjsip. fromdomain is incorrectly punctuated. Authentication is handled by type=auth sections, not by the use of secret in the type=endpoint one.

callerid="NAME" <+REGISTER_NUMBER>

You would have a strange system for this to make sense. Normally people want to receive the real originator’s caller ID from ITSPs.

You have two different context’s!

3

hey thank you for your quick response and finding where I was doing wrong david551

I was able to resolve my error by removing everything from the pjsip config and made another which resolved my error.

But there is another problem I am getting which is, I will add that in another post because thats not related this this subject.

thank you once again.

4

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