[asterisk-bugs] [JIRA] (ASTERISK-29461) Contact header format for inbound calls. One way audio inbound calls behind NAT.
Maxim Grechikhin (JIRA)
noreply at issues.asterisk.org
Wed Jun 2 11:10:17 CDT 2021
Maxim Grechikhin created ASTERISK-29461:
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Summary: Contact header format for inbound calls. One way audio inbound calls behind NAT.
Key: ASTERISK-29461
URL: https://issues.asterisk.org/jira/browse/ASTERISK-29461
Project: Asterisk
Issue Type: Information Request
Security Level: None
Components: Resources/res_pjsip
Affects Versions: 18.3.0
Reporter: Maxim Grechikhin
Dear community,
I faced a situation of one way audio at inbound calls using PJSIP channel on Asterisk 18.3.0 installation. Previously I had the same situation when I was using SIP (chan_sip & sip.conf) channel and my SIP provider explained that they require Contact header in the following format: <sip:callee_number at external_ip:port> in transmitted portion of SIP headers to them. That time the issue was caused by private IP address used by Asterisk to form contact header and as I remember it has been fixed by "from_domain=" parameter in a trunk description. Here I would like to tell that despite the fact that Asterisk is located behind NAT, the firewall is configured correctly and allows inbound RTP sessions. Since migration to PJSIP channel the contact header looks like <sip: external_ip_addr:5060> and we do not have audio signal from remote parties. Is there a way to modify any settings in order to affect the format of contact header? Or any other related suggestions much appreciated.
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