call.in-progress.error-sip-outbound-call-failed-to...
# support
l
Hello Vapi team 👋 I’m currently trying to connect my on-premise Asterisk server with Vapi using a SIP trunk and a BYO phone number. What I'm trying to achieve: When someone calls my GSM line (connected via a GOIP gateway), the call should be received by Asterisk and then forwarded to a Vapi assistant. I've attached two files: sip_traffic.pcap – a SIP traffic capture recorded during a failed call. vapi_asterisk_conf.txt – the relevant parts of my pjsip.conf and extensions.conf. Current issue: The call reaches Vapi and appears in the Call Logs as "Queued", But then fails shortly after with the error: call.in-progress.error-sip-outbound-call-failed-to-connect The assistant doesn't speak, and the call disconnects within 1 second. I would appreciate any help or insights — especially if you can review the SIP traffic and see what's wrong in the dialplan or signaling. Thanks in advance! @Shubham Bajaj @Subrat Sagar https://cdn.discordapp.com/attachments/1388272496759017512/1388272497623040094/vapi_asterisk_conf.txt?ex=688896c4&is=68874544&hm=2a223df0f66612af0ebed52d58d65236e449ca4ebb4b254bd466785a898e8fae& https://cdn.discordapp.com/attachments/1388272496759017512/1388272497992007871/sip_traffic.pcap?ex=688896c4&is=68874544&hm=57f1b37eec2b2fd0853bbfe6a92e6af6e5107504f2d21932601ddf82a3eec3ec&
k
Update your dialplan to use Dial (PJSIP/endpoint-vapi/sip:your_user@sip.vapi.ai) with matching SIP credentials or IP-based trunking, and ensure media/RTP settings are correct..
s
Hey Leo! I looked into the shared pcap file, but I don't see any association with Vapi. You have also mentioned you have a call id. So if you could share the call id, I could try to find out the association and what's actually going wrong. But looking at pcap file and txt file now I couldn't pass any conclusion. So I need the call id.
l
a06f9e03-5028-4dc7-b7d2-f797c1393fc0
s
Hey can you book a call with my colleague