Description I'm running Kamailio 5.2.0, whenever I relay an invite via Kamailio, my original contact header is changed from the original: sip:+XXXXXXXXX@YYY.YYY.YYY.YYYY:5060;transport=udp;gw=netvision To sip:YYY.YYY.YYY.YYY;line=sr-.n274i0TMfsHMcCAlVyAMV5IlgCXvV8n4edQlgIUlgKQ4Vk74gI1LmZ69NM79FZAR3JC.cDNLfHUj3Wnp3MP9nd*. Even when I set it correctly on my route, using: remove_hf("Contact"); append_hf("Contact: sip:$tU@YYY.YYY.YYY.YYY:5060;transport=udp;gw=netvision\r\n", "Contact");
It still ends up being modified. What can I do to keep the contact header as it is?
SIP Traffic U YYY.YYY.YYY.YYY:5060 -> 81.24.193.248:5060 INVITE sip:+442033202609@81.24.193.248:5060 SIP/2.0. Record-Route: sip:YYY.YYY.YYY.YYY;r2=on;lr=on;ftag=mp0S9yH11vryH;vsf=AAAAAAAAAAAAAAAAAAAAAAADCAEACRgPGB8AAy4xNjM-. Record-Route: sip:2YYY.YYY.YYY.YYY;line=sr-.n274V8TlidQMVyUlg87vV8nlfDXlVHB9VD0.VHB9VDoLGtNRbH7ltlTEkyQl3jXEky1LNMoRkt5ukt5ukt5ukt5ukt5ukt5ukt5ukt5W8M5WktckoLuWAsduktTMm64pCAD. Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bK53e9.308596b23ed683368534c9d609dce0f6.0. Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKsr-kA2uvfsUlerJWtIylgCXvV8n4edQlgIUlgKQ4Vk74gI1.oJVjb2njburlgCXvV8n4edQlgIUlgKQ4HWsRbWPLVDX.GrXLcAHlcC74nZXKbTVpcHh4b6mMGZvJgLsLcJTkgtGu2PGgy**. Max-Forwards: 67. From: +18702935016 sip:+18702935016@YYY.YYY.YYY.YYY;tag=mp0S9yH11vryH. To: sip:+442033202609@81.24.193.248. Call-ID: !!:jbkfMgyHjbsDjblfjqAQlVldvgyHKolDlcIHlcknKVIdjcIT. CSeq: 18182498 INVITE. Contact: sip:YYY.YYY.YYY.YYY;line=sr-.n274i0TMfsHMcCAlVyAMV5IlgCXvV8n4edQlgIUlgKQ4Vk74gI1LmZ69NM79FZAR3JC.cDNLfHUj3Wnp3MP9nd*. User-Agent: FreeSWITCH-mod_sofia/1.9.0+git~20180706T160334Z~de3df8dc0e~64bit. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, path, replaces. Allow-Events: talk, hold, conference, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 224. Remote-Party-ID: "+18702935016" sip:+18702935016@YYY.YYY.YYY.YYY;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1585439298 1585439299 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ. s=FreeSWITCH. c=IN IP4 ZZZ.ZZZ
Description I'm running Kamailio 5.2.0, whenever I relay an invite via Kamailio, my original contact header is changed from the original: sip:+XXXXXXXXX@YYY.YYY.YYY.YYYY:5060;transport=udp;gw=netvision To sip:YYY.YYY.YYY.YYY;line=sr-.n274i0TMfsHMcCAlVyAMV5IlgCXvV8n4edQlgIUlgKQ4Vk74gI1LmZ69NM79FZAR3JC.cDNLfHUj3Wnp3MP9nd*. Even when I set it correctly on my route, using: remove_hf("Contact"); append_hf("Contact: sip:$tU@YYY.YYY.YYY.YYY:5060;transport=udp;gw=netvision\r\n", "Contact");
It still ends up being modified. What can I do to keep the contact header as it is?
SIP Traffic U YYY.YYY.YYY.YYY:5060 -> 81.24.193.248:5060 INVITE sip:+442033202609@81.24.193.248:5060 SIP/2.0. Record-Route: sip:YYY.YYY.YYY.YYY;r2=on;lr=on;ftag=mp0S9yH11vryH;vsf=AAAAAAAAAAAAAAAAAAAAAAADCAEACRgPGB8AAy4xNjM-. Record-Route: sip:2YYY.YYY.YYY.YYY;line=sr-.n274V8TlidQMVyUlg87vV8nlfDXlVHB9VD0.VHB9VDoLGtNRbH7ltlTEkyQl3jXEky1LNMoRkt5ukt5ukt5ukt5ukt5ukt5ukt5ukt5W8M5WktckoLuWAsduktTMm64pCAD. Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bK53e9.308596b23ed683368534c9d609dce0f6.0. Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKsr-kA2uvfsUlerJWtIylgCXvV8n4edQlgIUlgKQ4Vk74gI1.oJVjb2njburlgCXvV8n4edQlgIUlgKQ4HWsRbWPLVDX.GrXLcAHlcC74nZXKbTVpcHh4b6mMGZvJgLsLcJTkgtGu2PGgy**. Max-Forwards: 67. From: +18702935016 sip:+18702935016@YYY.YYY.YYY.YYY;tag=mp0S9yH11vryH. To: sip:+442033202609@81.24.193.248. Call-ID: !!:jbkfMgyHjbsDjblfjqAQlVldvgyHKolDlcIHlcknKVIdjcIT. CSeq: 18182498 INVITE. Contact: sip:YYY.YYY.YYY.YYY;line=sr-.n274i0TMfsHMcCAlVyAMV5IlgCXvV8n4edQlgIUlgKQ4Vk74gI1LmZ69NM79FZAR3JC.cDNLfHUj3Wnp3MP9nd*. User-Agent: FreeSWITCH-mod_sofia/1.9.0+git~20180706T160334Z~de3df8dc0e~64bit. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, path, replaces. Allow-Events: talk, hold, conference, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 224. Remote-Party-ID: "+18702935016" sip:+18702935016@YYY.YYY.YYY.YYY;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1585439298 1585439299 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ. s=FreeSWITCH. c=IN IP4 ZZZ.ZZZ
Hello,
Quoting from the bug report, where Daniel already replied:
"It looks like you are using topoh module, which has the purpose of changing the headers that contain ip addresses. Remove it from your config."
Have you tried already to deactivate this module in your cfg?
Cheers,
Henning
-- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.comhttps://gilawa.com/
From: sr-users sr-users-bounces@lists.kamailio.org On Behalf Of Edward Romanenco Sent: Sunday, March 29, 2020 2:51 PM To: sr-users@lists.sip-router.org Subject: [SR-Users] Kamailio breaks RFC's 'Contact Header'
Description I'm running Kamailio 5.2.0, whenever I relay an invite via Kamailio, my original contact header is changed from the original: sip:+XXXXXXXXX@YYY.YYY.YYY.YYYY:5060;transport=udp;gw=netvision To sip:YYY.YYY.YYY.YYY;line=sr-.n274i0TMfsHMcCAlVyAMV5IlgCXvV8n4edQlgIUlgKQ4Vk74gI1LmZ69NM79FZAR3JC.cDNLfHUj3Wnp3MP9nd*. Even when I set it correctly on my route, using: remove_hf("Contact"); append_hf("Contact: sip:$tU@YYY.YYY.YYY.YYY:5060;transport=udp;gw=netvision\r\n", "Contact");
It still ends up being modified. What can I do to keep the contact header as it is?
SIP Traffic U YYY.YYY.YYY.YYY:5060 -> 81.24.193.248:5060 INVITE sip:+442033202609@81.24.193.248:5060 SIP/2.0. Record-Route: sip:YYY.YYY.YYY.YYY;r2=on;lr=on;ftag=mp0S9yH11vryH;vsf=AAAAAAAAAAAAAAAAAAAAAAADCAEACRgPGB8AAy4xNjM-. Record-Route: sip:2YYY.YYY.YYY.YYY;line=sr-.n274V8TlidQMVyUlg87vV8nlfDXlVHB9VD0.VHB9VDoLGtNRbH7ltlTEkyQl3jXEky1LNMoRkt5ukt5ukt5ukt5ukt5ukt5ukt5ukt5W8M5WktckoLuWAsduktTMm64pCAD. Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bK53e9.308596b23ed683368534c9d609dce0f6.0. Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKsr-kA2uvfsUlerJWtIylgCXvV8n4edQlgIUlgKQ4Vk74gI1.oJVjb2njburlgCXvV8n4edQlgIUlgKQ4HWsRbWPLVDX.GrXLcAHlcC74nZXKbTVpcHh4b6mMGZvJgLsLcJTkgtGu2PGgy**. Max-Forwards: 67. From: +18702935016 sip:+18702935016@YYY.YYY.YYY.YYY;tag=mp0S9yH11vryH. To: sip:+442033202609@81.24.193.248. Call-ID: !!:jbkfMgyHjbsDjblfjqAQlVldvgyHKolDlcIHlcknKVIdjcIT. CSeq: 18182498 INVITE. Contact: sip:YYY.YYY.YYY.YYY;line=sr-.n274i0TMfsHMcCAlVyAMV5IlgCXvV8n4edQlgIUlgKQ4Vk74gI1LmZ69NM79FZAR3JC.cDNLfHUj3Wnp3MP9nd*. User-Agent: FreeSWITCH-mod_sofia/1.9.0+git~20180706T160334Z~de3df8dc0e~64bit. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, path, replaces. Allow-Events: talk, hold, conference, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 224. Remote-Party-ID: "+18702935016" sip:+18702935016@YYY.YYY.YYY.YYY;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1585439298 1585439299 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ. s=FreeSWITCH. c=IN IP4 ZZZ.ZZZ
Yes I did actually, I didn't remove it completely but gave an exception in the vein of -
event_route[topoh:msg-outgoing] { if($sndto(ip)=="10.1.1.10") { drop; }
Thanks Daniel and Henning!
Edward ________________________________ From: Henning Westerholt hw@skalatan.de Sent: Monday, March 30, 2020 9:40 PM To: Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org; sr-users@lists.sip-router.org sr-users@lists.sip-router.org Cc: Edward Romanenco edward@telemessage.com Subject: RE: Kamailio breaks RFC's 'Contact Header'
Hello,
Quoting from the bug report, where Daniel already replied:
“It looks like you are using topoh module, which has the purpose of changing the headers that contain ip addresses. Remove it from your config.”
Have you tried already to deactivate this module in your cfg?
Cheers,
Henning
--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.comhttps://gilawa.com/
From: sr-users sr-users-bounces@lists.kamailio.org On Behalf Of Edward Romanenco Sent: Sunday, March 29, 2020 2:51 PM To: sr-users@lists.sip-router.org Subject: [SR-Users] Kamailio breaks RFC's 'Contact Header'
Description
I'm running Kamailio 5.2.0, whenever I relay an invite via Kamailio, my original contact header is changed from the original:
sip:+XXXXXXXXX@YYY.YYY.YYY.YYYY:5060;transport=udp;gw=netvision
To
sip:YYY.YYY.YYY.YYY;line=sr-.n274i0TMfsHMcCAlVyAMV5IlgCXvV8n4edQlgIUlgKQ4Vk74gI1LmZ69NM79FZAR3JC.cDNLfHUj3Wnp3MP9nd*.
Even when I set it correctly on my route, using:
remove_hf("Contact");
append_hf("Contact: sip:$tU@YYY.YYY.YYY.YYY:5060;transport=udp;gw=netvision\r\n", "Contact");
It still ends up being modified.
What can I do to keep the contact header as it is?
SIP Traffic
U YYY.YYY.YYY.YYY:5060 -> 81.24.193.248:5060
INVITE sip:+442033202609@81.24.193.248:5060 SIP/2.0.
Record-Route: sip:YYY.YYY.YYY.YYY;r2=on;lr=on;ftag=mp0S9yH11vryH;vsf=AAAAAAAAAAAAAAAAAAAAAAADCAEACRgPGB8AAy4xNjM-.
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bK53e9.308596b23ed683368534c9d609dce0f6.0.
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKsr-kA2uvfsUlerJWtIylgCXvV8n4edQlgIUlgKQ4Vk74gI1.oJVjb2njburlgCXvV8n4edQlgIUlgKQ4HWsRbWPLVDX.GrXLcAHlcC74nZXKbTVpcHh4b6mMGZvJgLsLcJTkgtGu2PGgy**.
Max-Forwards: 67.
From: +18702935016 sip:+18702935016@YYY.YYY.YYY.YYY;tag=mp0S9yH11vryH.
To: sip:+442033202609@81.24.193.248.
Call-ID: !!:jbkfMgyHjbsDjblfjqAQlVldvgyHKolDlcIHlcknKVIdjcIT.
CSeq: 18182498 INVITE.
User-Agent: FreeSWITCH-mod_sofia/1.9.0+git~20180706T160334Z~de3df8dc0e~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 224.
Remote-Party-ID: "+18702935016" sip:+18702935016@YYY.YYY.YYY.YYY;party=calling;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1585439298 1585439299 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ.
s=FreeSWITCH.
c=IN IP4 ZZZ.ZZZ