[asterisk-bugs] [JIRA] (ASTERISK-28287) Pjsip rewrite port in from/to headers on reply
Asterisk Team (JIRA)
noreply at issues.asterisk.org
Wed Feb 13 07:18:47 CST 2019
[ https://issues.asterisk.org/jira/browse/ASTERISK-28287?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=246209#comment-246209 ]
Asterisk Team commented on ASTERISK-28287:
------------------------------------------
Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.
A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.
Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].
Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur.
> Pjsip rewrite port in from/to headers on reply
> ----------------------------------------------
>
> Key: ASTERISK-28287
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-28287
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_pjsip
> Affects Versions: 16.0.0
> Environment: CentOS Linux release 7.5.1804 (Core)
> Asterisk 16.0.0
> PJPROJECT version currently running against: 2.7.2
> Reporter: Aleksandr Khomutov
> Severity: Minor
> Labels: pjsip
>
> Hello! I have problems configuring chan_pjsip. In response to requests from my softswitch, the asterisk rewrites the headers from and to the port, removes the port from there. Tell me how to make it so that the asterisk does not do this? This problem is not observed for the chan_sip driver.
> ```
> 2019/02/13 10:31:09.351381 85.12.217.3:5060 -> 10.227.228.4:5061
> INVITE sip:73432875561 at 10.227.228.4:5061;user=phone SIP/2.0
> Via: SIP/2.0/UDP 85.12.217.3:5060;rport;branch=z9hG4bK-8ae2b6262f5011e99bf22c44fd84c5e0;sig=35ee398c
> Via: SIP/2.0/UDP 85.12.217.3:5060;rport;branch=z9hG4bK-8ae2b0362f5011e99bf22c44fd84c5e0;sig=e4f91997
> Via: SIP/2.0/UDP 85.12.217.3:5062;rport=5062;branch=z9hG4bK-8ae293762f5011e99bf22c44fd84c5e0
> From: <sip:73432786060 at 85.12.217.3:5062;user=phone>;tag=8ae26efa2f5011e99bf22c44fd84c5e0
> To: <sip:73432875561 at 10.227.228.4:5061;user=phone>
> Call-ID: 8ae26f222f5011e99bf22c44fd84c5e0 at 85.12.217.3
> CSeq: 1 INVITE
> Contact: <sip:73432786060 at 85.12.217.3:5062;user=phone>
> Content-Type: application/sdp
> Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE
> Max-Forwards: 69
> User-Agent: TS-v4.6.0-07a
> Cisco-Guid: 2329864784-793776617-2616339524-4253337056
> Content-Length: 312
> Record-Route: <sip:AQEAEOAgaLvZEqxL3Z9L9avyD5MDAAQ6GnBx at 85.12.217.3;lr>
> Record-Route: <sip:AQEAEEiS5FS8Xjpm4lEkLYOwprwDAARDhHft at 85.12.217.3;lr>
> v=0
> o=- 1550035858 1550035858 IN IP4 85.12.217.17
> s=-
> c=IN IP4 85.12.217.17
> t=0 0
> m=audio 27044 RTP/AVP 0 8 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> a=sendrecv
> a=silenceSupp:off - - - -
> 2019/02/13 10:31:09.352290 10.227.228.4:5061 -> 85.12.217.3:5060
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 85.12.217.3:5060;rport=5060;received=85.12.217.3;branch=z9hG4bK-8ae2b6262f5011e99bf22c44fd84c5e0;sig=35ee398c
> Via: SIP/2.0/UDP 85.12.217.3:5060;rport;branch=z9hG4bK-8ae2b0362f5011e99bf22c44fd84c5e0;sig=e4f91997
> Via: SIP/2.0/UDP 85.12.217.3:5062;rport=5062;branch=z9hG4bK-8ae293762f5011e99bf22c44fd84c5e0
> Record-Route: <sip:AQEAEOAgaLvZEqxL3Z9L9avyD5MDAAQ6GnBx at 85.12.217.3;lr>
> Record-Route: <sip:AQEAEEiS5FS8Xjpm4lEkLYOwprwDAARDhHft at 85.12.217.3;lr>
> Call-ID: 8ae26f222f5011e99bf22c44fd84c5e0 at 85.12.217.3
> From: <sip:73432786060 at 85.12.217.3;user=phone>;tag=8ae26efa2f5011e99bf22c44fd84c5e0
> To: <sip:73432875561 at 10.227.228.4;user=phone>;tag=z9hG4bK-8ae2b6262f5011e99bf22c44fd84c5e0
> CSeq: 1 INVITE
> WWW-Authenticate: Digest realm="centrex",nonce="1550035869/1ef0271eb3e86940bce3fc40fc143ef5",opaque="6f75f93b26037d9b",algorithm=md5,qop="auth"
> Server: centrex
> Content-Length: 0
> ```
> Full config:
> ```
> [global]
> type=global
> user_agent=centrex
> default_from_user=centrex
> default_realm=centrex
> [transport-udp]
> type=transport
> protocol=udp ;udp,tcp,tls,ws,wss
> bind=0.0.0.0:5061
> [opensips-endpoint](!)
> type=endpoint
> send_pai=no
> send_rpid=no
> ;send_rpid=yes
> rpid_immediate=no
> trust_id_outbound=no
> trust_id_inbound=no
> context=from_opensips
> from_domain=sbc.profintel.ru
> sdp_session=Centrex v.1.0
> disallow=all
> allow=alaw
> allow=ulaw
> t38_udptl_maxdatagram=20
> t38_udptl=yes
> ;---nat
> rtp_symmetric=yes
> force_rport=yes
> rewrite_contact=yes
> [opensips-auth](!)
> type=auth
> auth_type=userpass
> realm=centrex
> [opensips-aor](!)
> type=aor
> max_contacts=1
> [opensips_sbc1](opensips-aor)
> [opensips_sbc1](opensips-auth)
> password=oemae8angoc4utoh2aeP
> username=opensips_sbc1
> [opensips_sbc1](opensips-endpoint)
> auth=opensips_sbc1
> aors=opensips_sbc1
> accountcode = opensips_sbc1
> [opensips_sbc2](opensips-aor)
> [opensips_sbc2](opensips-auth)
> password=oemae8angoc4utoh2aeP
> username=opensips_sbc2
> [opensips_sbc2](opensips-endpoint)
> auth=opensips_sbc2
> aors=opensips_sbc2
> accountcode = opensips_sbc2
> [opensips_sbc2]
> type=identify
> endpoint=opensips_sbc2
> match=10.227.228.18
> [opensips_sbc1]
> type=identify
> endpoint=opensips_sbc1
> match=10.227.228.169
> ```
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