My problem is that in the invite value I get the username from the sip provider, and in the “To” field I get the numbers purchased from the provider. Naturally routing by DID does not work. Incoming I need to distribute to different clients.
CDR says outbound
@AlexZah what you were dialling? DID or outbound number. If DID then what is added in DID forwarding ?
Please share screenshot if possible.
Thanks.
Gateway config
| rtp-ip | localipastpp | |
|---|---|---|
| dialplan | XML | |
| user-agent-string | ASTPP | |
| debug | 0 | |
| sip-trace | no | |
| tls | false | |
| inbound-reg-force-matching-username | true | |
| disable-transcoding | true | |
| all-reg-options-ping | false | |
| unregister-on-options-fail | true | |
| log-auth-failures | true | |
| status | 0 | |
| inbound-bypass-media | false | |
| inbound-proxy-media | false | |
| disable-transfer | true | |
| enable-100rel | false | |
| rtp-timeout-sec | 60 | |
| dtmf-duration | 2000 | |
| manual-redirect | false | |
| aggressive-nat-detection | false | |
| enable-timer | false | |
| minimum-session-expires | 120 | |
| session-timeout-pt | 1800 | |
| auth-calls | true | |
| apply-inbound-acl | default | |
| inbound-codec-prefs | PCMU,PCMA,G729 | |
| outbound-codec-prefs | PCMU,PCMA,G729 | |
| inbound-late-negotiation | false | |
| rtp-start-port | 12001 | |
| rtp-end-port | 13000 | |
| ext-sip-ip | extipastpp | |
| ext-rtp-ip | extipastpp | |
| caller-id-type | pid | |
| context | default |
Hello @AlexZah ,
The screenshots says its an outbound call. Is it?
Please explain call flow is it sip → destination number.
or PSTN → DID → forwarding to somewhere.
As its an confusion here. DID to PSTN is that?
Thanks.
Hello @palak
I’m making a call to a number we’re renting from a telephony provider. From the provider’s sip server, the call goes to my astpp server. But astpp for some reason processes this incoming call as an outgoing call. And for some reason, the called number has the trunk login, and not the number specified in the TO field and in the PAI field. The DID for the number is specified in ASTPP.
inside ASTPP DID is routed via SIP-DID
it seems that the problem is that the called number is the login and the “pai” field is not taken into account in any way. Prompt, how it is possible to force to route a call on the field “to” or on the field “pai”?
Hello @AlexZah ,
It seems you are dialling inbound calls but it is routing as outbound call.
Please log your issue here- https://jira.astppbilling.org/
Thanks.
Thanks for the help. my problem was that the call was routed by the invite header, and this is the trunk login, not the phone number, since there was no suitable route in the incoming context, the call fell into the outgoing context. By changing the value of the variable destination_number = params:getHeader(“variable_sip_to_user”), I was able to get the correct incoming flow.
You can fix this variable in /opt/ASTPP/freeswitch/scripts/astpp/scripts
Oh no, thanks =) I wrote this for those who will be looking for an answer in a similar situation
Hi hope this post is still alive, I have the same issue, but I can’t fine where to change teh variable you are refering too, I set my DID when I call the DID number, my server treated as a outbound call and tehn ome ring and busy signal, outbount calls work fine, I have it set it up for DOD-local I even name the xtensionas the DID number. in conclusion I’m unable to recieve calls using my DID
ASTPP is considering ‘Caller-Destination-Number’ Info Channel Variable as destination number and logic is comparing whatever value got from DID provider in that variable with system wide DID numbers and further checking