meirjdd
February 29, 2024, 9:59pm
1
Hi,
I want to use my TELNYX SIP TRUNK to originate calls.
This SIP TRUNK is associated to 10 phone numbers.
When I use ZOIPER FREE with:
DOMAIN
USERNAME
PASSWORD
CALLER ID NAME
I can receive and emit calls with TELNYX which use one of the 10 avaible numbers.
Then, I added this connection to PJSIP:
;TELNYX
[telnyx]
type=registration
outbound_auth=telnyx
server_uri=sip:DOMAIN
client_uri=sip:USERNAME@DOMAIN
retry_interval=5
fatal_retry_interval=5
forbidden_retry_interval=20
;auth_rejection_permanent=yes
[telnyx]
type=auth
auth_type=userpass
password=PASSWORD
username=USERNAME
;keep_alive_interval=20
[telnyx]
type=aor
contact=sip:DOMAIN
max_contacts=1
qualify_frequency=20
qualify_timeout=5
[telnyx]
type=endpoint
context=router
disallow=all
allow=alaw,ulaw;,slin,gsm,g729,g723
outbound_auth=telnyx
aors=telnyx
direct_media=yes
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
ice_support=yes
media_use_received_transport=yes
[telnyx]
type=identify
endpoint=telnyx
match=DOMAIN
Why, the originate commands does not work?
the PJSIP configuration is not good? I use the same for OVH numbers and it works…
Thanks a lot!!
You forgot the logs and the details of the Originate command.
Hi
await this.localChannel.originate({
endpoint: `PJSIP/NUMTOCALL@telnyx`,
formats: this.opts.format,
extension: 's',
context: 'incoming',
priority: 1,
timeout: 30,
app: this.ariAppname,
callerId: 'ME',
appArgs: 'dialed',
})
About the log, there is nothing instead of
Channel XXX joined ‘simple bridge’…
Channel XXX left ‘simple bridge’…
Connected to Asterisk 20.4.0 currently running on ip55 (pid = 1279)
ip55*CLI>
Creating Stasis app 'externalMedia_MYID'
Shutting down application 'externalMedia_LAST'
Destroying Stasis app externalMedia_LAST
-- Remove stasis-externalMedia_LAST/h/1, registrar=res_stasis; con=stasis-externalMedia_LAST(0x7f6004006080); con->root=0x7f6004006470
== WebSocket connection from '127.0.0.1:45688' for protocol '' accepted using version '13'
-- Remove stasis-externalMedia_LAST/_./1, registrar=res_stasis; con=stasis-externalMedia_LAST(0x7f6004006080); con->root=0x7f6004006470
-- Called NUMTOCALL@telnyx
> 0x7f6008018080 -- Strict RTP learning after remote address set to: 127.0.0.1:6900
-- Called 127.0.0.1:6900
-- UnicastRTP/127.0.0.1:6900-0x7f6008012b40 answered
> Launching Stasis(externalMedia_MYID) on UnicastRTP/127.0.0.1:6900-0x7f6008012b40
-- Channel Snoop/cd2fc7dd-f837-47a5-a5bc-ec05da7ef2d9-00000012 joined 'simple_bridge' stasis-bridge <d2df488c-348a-4a40-9123-44b4715ddc98>
-- Channel Snoop/cd2fc7dd-f837-47a5-a5bc-ec05da7ef2d9-00000013 joined 'simple_bridge' stasis-bridge <78559588-347f-499b-8b21-d9b57848d896>
-- Channel UnicastRTP/127.0.0.1:6900-0x7f6008012b40 joined 'simple_bridge' stasis-bridge <9972ceb0-58cb-4a78-9c64-4745141d8cdf>
-- Channel UnicastRTP/127.0.0.1:6900-0x7f6008012b40 left 'simple_bridge' stasis-bridge <9972ceb0-58cb-4a78-9c64-4745141d8cdf>
-- Channel UnicastRTP/127.0.0.1:6900-0x7f6008012b40 joined 'simple_bridge' stasis-bridge <d2df488c-348a-4a40-9123-44b4715ddc98>
-- Channel Snoop/cd2fc7dd-f837-47a5-a5bc-ec05da7ef2d9-00000013 left 'simple_bridge' stasis-bridge <78559588-347f-499b-8b21-d9b57848d896>
Deactivating Stasis app 'externalMedia_MYID'
== WebSocket connection from '127.0.0.1:45688' closed
-- Channel Snoop/cd2fc7dd-f837-47a5-a5bc-ec05da7ef2d9-00000012 left 'simple_bridge' stasis-bridge <d2df488c-348a-4a40-9123-44b4715ddc98>
-- Channel UnicastRTP/127.0.0.1:6900-0x7f6008012b40 left 'simple_bridge' stasis-bridge <d2df488c-348a-4a40-9123-44b4715ddc98>
Thanks
In addition, from the asterisk CLI
originate PJSIP/MYNUMBER@OVH_SIP application Playback hello-world → originate the call
originate PJSIP/MYNUMBER@telnyx application Playback hello-world → do not originate the call
if I turn on pjsip logger, I have a D35 error and 403 “Caller Origination Number is Invalid”
For MYNUMBER I tried +336, 00336, 06…and always this error…
Pooh
March 5, 2024, 9:22am
5
On Tuesday 05 March 2024 at 01:02:28, meirjdd via Asterisk Community wrote:
403 “Caller Origination Number is Invalid” For MYNUMBER I tried +336, 00336,
06…and always this error…
You need to ask your upstream provider what number/s, and in what format/s,
you have to present Caller ID for them to accept the call. We cannot know
this.
Antony.
–
RTFM may be the appropriate reply, but please specify exactly which FM to R.
Please reply to the list;
please *don't* CC me.
Thanks! Using ARI I managed to specify the CallerID and it works
Thanks a lot
system
Closed
April 4, 2024, 10:12am
7
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