Why asterisk ignores some headers

<--- Received SIP response (1221 bytes) from TLS:57.144.211.157:5061 --->
SIP/2.0 403 Business Initiated Connected Call Per Day Limit Hit: Unknown error. Details - facebook::wa_biz_api::messaging::MessagingException: Business initiated calls daily limit hit, limit: 5. Next allowed unix epoch time in seconds: 1759081634, error is Non-retryable, classified as UserError
Via: SIP/2.0/TLS xx.xx.xxx.xx:5061;rport=19388;received=2803:6081:70e8:6ee0:6fa5:4860:400:0;branch=z9hG4bKPj987f36ec-1368-45fd-84d9-22c619a561ee;alias
Record-Route: <sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr>
Record-Route: <sip:wa.meta.vc;transport=tls;lr>
Call-ID: 35c89fb5-e868-45af-91ef-8dc9dabf06d3
From: <sip:xxxxxxxxxxxx@example.example.com>;tag=05664f1a-f946-4d76-8288-a12e0548cf80
To: <sip:+91xxxxxxxxxxxx@wa.meta.vc>;tag=2665f834-e1db-41d9-9eae-42dcad1e94c0
CSeq: 5627 INVITE
Allow: INVITE, ACK, BYE, CANCEL, NOTIFY, OPTIONS
X-FB-External-Domain: wa.meta.vc
Warning: 399 wa.meta.vc "Business Initiated Connected Call Per Day Limit Hit: Unknown error. Details - facebook::wa_biz_api::messaging::MessagingException: Business initiated calls daily limit hit, limit: 5. Next allowed unix epoch time in seconds: 1759081634, error is Non-retryable, classified as UserError"
Content-Length:  0

<--- Transmitting SIP response (530 bytes) to WSS:103.5.134.34:34646 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/WSS 3ogb7n0caft2.invalid;rport=34646;received=103.5.134.34;branch=z9hG4bK6523670
Call-ID: ies9k0vsq9a7uohaik1q
From: <sip:xxxxxxxxxxxx@example.example.com>;tag=qni6lk19d8
To: <sip:+91xxxxxxxxxxxx@wa.meta.vc>;tag=dd62a1e8-1473-4fd2-a966-f15443be5ae7
CSeq: 2 INVITE
Server: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE, INFO
Reason: Q.850;cause=21
Content-Length:  0

I am making an outbound calls from my client code through asterisk to meta but when meta sends warning headers in response ,asterisk send only the status code but i also want to send the warning header too but it is not transmistted to my wss

Asterisk is a multi-technology, back to back, user agent, not a SIP proxy. The status code isn’t actually copied through, but rather converted into the nearest equivalent ISDN code and converted the other way if the other channel is also SIP. The overall conversion is not one to one, and the code can sometimes change.

As this is a response, I don’t think you can pass through the headers that are not of interest to Asterisk. If it were an INVITE, you could read the header on the incoming channel and set it on the outgoing channel.

Nothing is implemented to support this. New functionality would have to be coded.

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