Hello,
I am proxying all RTP through RTPEngine. Everything works fine until about
5 seconds into the call, when rtpengine enters kernelization, after which
all RTP forwarding ceases. I've checked the required iptables entries, and
all looks good.
Here is a description of my environment:
# cat os-release
PRETTY_NAME="Debian GNU/Linux 11 (bullseye)"
NAME="Debian GNU/Linux"
VERSION_ID="11"
VERSION="11 (bullseye)"
VERSION_CODENAME=bullseye
ID=debian
HOME_URL="https://www.debian.org/"
SUPPORT_URL="https://www.debian.org/support"
BUG_REPORT_URL="https://bugs.debian.org/"
# kamailio -v
version: kamailio 5.6.2 (x86_64/linux) 54a9c1
flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE,
USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC,
TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT,
USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST,
HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED
ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024,
BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: 54a9c1
compiled on 16:02:49 Dec 1 2022 with gcc 10.2.1
# rtpengine -v
Version: 11.1.1.3-1~bpo11+1
Michel Pelletier
Hello ,
im using kamailio with two interfaces external and internal.
i need a way either to :
* enable topoh hiding only when outgoing interface is external ( mask contact and Via ip only when ougoing interface is external
* or if it is not possible to enable it only in one direction.i want to know how to configure dynamic(for example in xavp or avp) ip to put in Contact and Via when topoh is enabled.
i see that the 'mask_ip' parameter of topoh module is a string. so we can not set a dynamic value here unfortunately.
Thanks
Hello,
I'm continuing investigations on Kamailio and stress test. Got it again in
the state where it's not accepting any new TCP/TLS connections (UDP still
works though), but all looks good from lsof/netnstat part, like system is
not reporting any zombie connections. Restart of Kamailio process helps
This time I got output of kamctl trap
Put it here: https://pastebin.com/iYrNZ8U9
kamailio --version
version: kamailio 5.6.2 (x86_64/linux) 54a9c1
flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE,
USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC,
TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT,
USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST,
HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024,
BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: 54a9c1
compiled on 14:01:01 Oct 18 2022 with gcc 4.8.5
It's statically linked with tlsa pointing on openssl-1.1.1q
Settings related to TLS are:
fork=yes
children=4
tcp_children=12
enable_tls=yes
tcp_accept_no_cl=yes
tcp_max_connections=63536
tls_max_connections=63536
tcp_accept_aliases=no
tcp_async=yes
tcp_connect_timeout=10
tcp_conn_wq_max=63536
tcp_crlf_ping=yes
tcp_delayed_ack=yes
tcp_fd_cache=yes
tcp_keepalive=yes
tcp_keepcnt=3
tcp_keepidle=30
tcp_keepintvl=10
tcp_linger2=30
tcp_rd_buf_size=80000
tcp_send_timeout=10
tcp_wq_blk_size=2100
tcp_wq_max=10485760
open_files_limit=63536
Can you please help to read gdb output and understand where I missed in
config?
Thanks in advance!
--
Best regards,
Ihor (Igor)
Dear List,
We’re using Kamailio to load balance MRCP requests to multiple backend groups with a configuration as follows:
# kamailio.cfg
...
...
route[DISPATCH] {
if($ua=="mrcp_backend_1") {
if(!ds_select_dst("1", "4")) {
send_reply("404", "No destination");
exit;
}
}
if($ua=="mrcp_backend_2") {
if(!ds_select_dst("2", "4")) {
send_reply("404", "No destination");
exit;
}
}
xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n");
t_on_failure("RTF_DISPATCH");
route(RELAY);
exit;
}
...
...
# dispatcher.list
1 sip:mrcp01.server.int:8060;transport=tcp
1 sip:mrcp02.server.int:8060;transport=tcp
2 sip:mrcp03.server.int:8060;transport=tcp
2 sip:mrcp04.server.int:8060;transport=tcp
With this configuration, Kamailio load balances the initial SIP INVITE among the MRCP servers. After the INVITE, the service communicates directly to the MRCP servers via SIP (for hanging up the call), MRCPv2 (for sending speech control messages), and RTP (for sending audio).
We would like to implement a configurable number of retries, so that if a particular backend times out, Kamailio would retry X times to other backend(s). In short, something equivalent to HAProxy’s retries, but for Kamailio. This probably implies having Kamailio always as part of our communication (not just load balancing the initial SIP INVITE).
I haven’t been able to find much information about this, could someone provide some pointers?
Thank you so much
With best wishes,
Unai Rodriguez
Hello All
I have below setup :
SBC -> kamailio -> Media server(remote) -> web-client
Call comes from sbc to kamilio and it relays to Remote server to end user,
In Ringing state if remote server goes down then in kamailio there is any
way to Recover that ringing call like we can generate new invite for other
Remote server which is active state, for relaying call i am using
dispatcher module..
I tried with tm.cancel RPC command but it works only if the remote server
is running , when I crash the remote server at that time tm.cancel is not
generating cancel to remote server.
Please suggest if there is any other way to generate INVITE.
Regards
Devang Dhandhalya
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Hello!
I have a problem with writing the BYE message from the ACC module to the database, because the messages are generated locally by the ims_charging module at the end of the credit and the flag will not work.
You can help catch such local messages and make a record in the database.
Hello, I have some trouble to figure out how routing works when the
request coming from the kamailio itself (used as a user agent per example)
.I try to use presence PUA... modules. The issue I have is the NOTIFY
requests aren't sent to the good destination; it doesn't seems to obey to
the contact or other informations when the device did a SUBSCRIBE.
I don't want to ask to troubleshoot the presence module directly
I just want to know how Kamailio deals with request coming from Kamailio
itself (in the case of a NOTIFY generated by PUA_... modules
My mains questions are:
Does the pua modules when generating request like NOTIFY will pass through
request_route ?
Does the pua modules when generating request like NOTIFY will be sent
only when t_relay() is used or UA kamailio based request bypass all the
logic of the Kamailio proxy behavior ?
I'm trying to understand if I can modify the NOTIFY request to change
destination before the "routing decision" is made by Kamailio
Kamailio 5.6.0 used as :
-proxy to remote pbx for call dialog
-registrar/location server
-media relaying and nat management
-...and I try to use it as a presence server instead to use the pbx behind
kamailio
Hi Brandon
Apologies if you have provided this but do you have a PCAP of the INVITE / 400 Bad Request rather than the Pastebin of the INVITE
The last time I had this was because of a missing /r/n on the SDP granted it was not with teams but still difficult to track down.
Also, what is the MTU Size, have you checked to see if the is causing packet fragmentation and manifesting in a Bad Request.
Are there any clues in the Teams - Call Analytics.
Regards
Lewis
-----Original Message-----
From: sr-users-request(a)lists.kamailio.org <sr-users-request(a)lists.kamailio.org>
Sent: 29 December 2022 20:17
To: sr-users(a)lists.kamailio.org
Subject: sr-users Digest, Vol 211, Issue 67
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Today's Topics:
1. Re: Direct Routing, SIP, INVITE TO TEAMS (400 BAD REQUEST)
(Brandon Armstead)
2. Re: Direct Routing, SIP, INVITE TO TEAMS (400 BAD REQUEST)
(Brandon Armstead)
----------------------------------------------------------------------
Message: 1
Date: Thu, 29 Dec 2022 12:15:16 -0800
From: Brandon Armstead <brandon(a)cryy.com>
Subject: [SR-Users] Re: Direct Routing, SIP, INVITE TO TEAMS (400 BAD
REQUEST)
To: Henning Westerholt <hw(a)gilawa.com>
Cc: "Kamailio (SER) - Users Mailing List"
<sr-users(a)lists.kamailio.org>
Message-ID:
<CABAX3EoF+hfruLiV2Nh_g117_q8diaKxVEyp5hMFO=wyKbQidQ(a)mail.gmail.com>
Content-Type: multipart/alternative;
boundary="0000000000005692d905f0fd2538"
Henning,
There was no reply (reason) available in my case.
- Brandon
On Thu, Dec 29, 2022 at 11:48 AM Henning Westerholt <hw(a)gilawa.com> wrote:
> Hello,
>
>
>
> as mentioned before, have a look to the 400 reply reason phrase. This
> was done quite good from their side, better than many other vendors.
>
> Usually, Teams is not too bad if you are using a defined
> infrastructure that do not introduce too many variables. We are
> usually using a B2BUA in our customer projects for that purpose.
>
>
>
> Cheers,
>
>
>
> Henning
>
>
>
> --
>
> Henning Westerholt -
> https://eur03.safelinks.protection.outlook.com/?url=https%3A%2F%2Fskal
> atan.de%2Fblog%2F&data=05%7C01%7Clewis.hutchinson%40missionlabs.co.uk%
> 7Ce1670c8a21444edf0ed408dae9f91fac%7C97c26f550a7a4661bd8f7b43b50d3f2b%
> 7C0%7C0%7C638079553431377130%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwM
> DAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdat
> a=QU%2BzOgPdbmRYh51a9PvptIvVjnTZrGxT8z1G7b2RRGQ%3D&reserved=0
>
> Kamailio services -
> https://eur03.safelinks.protection.outlook.com/?url=https%3A%2F%2Fgila
> wa.com%2F&data=05%7C01%7Clewis.hutchinson%40missionlabs.co.uk%7Ce1670c
> 8a21444edf0ed408dae9f91fac%7C97c26f550a7a4661bd8f7b43b50d3f2b%7C0%7C0%
> 7C638079553431377130%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQI
> joiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdata=PuDqkZ
> QCj2IlxvRsLQGoKFv19K3uVE5RBxvcPKPODwQ%3D&reserved=0
>
>
>
> *From:* Brandon Armstead <brandon(a)cryy.com>
> *Sent:* Thursday, December 29, 2022 7:52 PM
> *To:* Kamailio (SER) - Users Mailing List
> <sr-users(a)lists.kamailio.org>
> *Subject:* [SR-Users] Re: Direct Routing, SIP, INVITE TO TEAMS (400
> BAD
> REQUEST)
>
>
>
> Alex & Kaufman,
>
>
>
> Appreciate the feedback. The pastebin is one example of about
> 1000+ iterations literally of changes. So I'm finally reaching out
> :). I haven't tested the socket attribute in record-route change, I
> will try this now. As for the angle brackets I've tried with and
> without, etc (already) to no effect. The SDP unknown media type seems
> to be a Poly thing, so I
> *want* to rule this out as its not in my other SDP's and still receive
> 400 bad request.
>
> One more iteration here we go... please feel free to let me know if
> you have any other thoughts on the matter :).
>
>
>
> Thanks!
>
>
>
> - Brandon
>
>
>
> On Thu, Dec 29, 2022 at 8:08 AM Alex Balashov
> <abalashov(a)evaristesys.com>
> wrote:
>
> Yeah, the grammar says that rr-params are just generic-params, in
> which case it's not. I didn't see that -- nicely spotted!
>
> I think that may be the basis of the 400 Bad Request. I'd be shocked
> if it weren't.
>
> The broken clock of "Microsoft SIP" can still be right twice a day.
>
> > On Dec 29, 2022, at 10:31 AM, Kaufman <bkaufman(a)bcmone.com> wrote:
> >
> > In your top Record-Route you have:
> >
> > socket=;
> >
> > Not sure if that is legal.
> >
> > Kaufman
> >
> > -----Original Message-----
> > From: Alex Balashov <abalashov(a)evaristesys.com>
> > Sent: Thursday, December 29, 2022 9:02 AM
> > To: Kamailio (SER) - Users Mailing List
> > <sr-users(a)lists.kamailio.org>
> > Subject: [SR-Users] Re: Direct Routing, SIP, INVITE TO TEAMS (400
> > BAD
> REQUEST)
> >
> > Sorry to hear you're having to interoperate with Teams. It's a
> > unique
> form of sadism I wouldn't wish upon anyone.
> >
> > A few theories:
> >
> > 1) Microsoft doesn't like the "bare" Contact header-value here:
> >
> > Contact:
> sip:+MY_FROM_PHONE_NUMBER_HERE@MY_FQDN_WAS_HERE:5061;transport=tls
> >
> > Unlike the careted one right above:
> >
> > P-Asserted-Identity:
> > <sip:+MY_FROM_PHONE_NUMBER_HERE@MY_FQDN_WAS_HERE>
> >
> > A bare URI absent other header-params is of course completely legal,
> > but
> I'm really trying to get inside the imaginary world of antisocial
> "Microsoft SIP" here.
> >
> > 2) Could it be that antisocial "Microsoft SIP" sends 400 as a way of
> objecting to something in the SDP, e.g. where a non-demented SIP stack
> would send "488 Not Acceptable Here" or "415 Unsupported Media Type"?
> >
> > I know you've said you tried multiple clients to rule that out, but
> > I
> wonder if the thing it's objecting to hasn't been ruled out that way.
> >
> > 3) I saw this media line in the SDP:
> >
> > m=application 41356 <unknown media type>
> >
> > What's that?
> >
> > -- Alex
> >
> >> On Dec 29, 2022, at 9:51 AM, Brandon Armstead <brandon(a)cryy.com> wrote:
> >>
> >> Outbound calls from my SBC into Teams (Polycom -> SBC -> Teams)
> >> always
> result in a 400 BAD REQUEST.
> >> Example invite below:
> >> https://eur03.safelinks.protection.outlook.com/?url=https%3A%2F%2Fn
> >> am11.safelinks.protection.outlook.com%2F%3Furl%3Dhttps%253A%252F%25
> >> 2Fpast&data=05%7C01%7Clewis.hutchinson%40missionlabs.co.uk%7Ce1670c
> >> 8a21444edf0ed408dae9f91fac%7C97c26f550a7a4661bd8f7b43b50d3f2b%7C0%7
> >> C0%7C638079553431377130%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDA
> >> iLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sda
> >> ta=5v20CvNnmdiHwfWMBX6tgjMlE1i2aKpFgTnuVFfpyl0%3D&reserved=0
> >> ebin.com%2FF1G1Ce59&data=05%7C01%7Cbkaufman%40bcmone.com%7C55672dbf
> >> 739
> >> a4ff447f008dae9b0c9b2%7Cafc1818e7b6848568913201b9396c4fc%7C1%7C0%7C
> >> 638
> >> 079242759152531%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjo
> >> iV2
> >> luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdata=Cbff6B4g
> >> Nr8
> >> A9zs89smJ129y8IyNM9%2B3zVkGhlzpa54%3D&reserved=0
> >> I've taken care to make sure numbers are all E.164 format in
> From/To/Contact. I've also taken care to make sure that FQDN is used
> in Contact and Record-Route header.
> >> I've tried many different variations and have followed the SIP
> information here:
> >> https://eur03.safelinks.protection.outlook.com/?url=https%3A%2F%2Fn
> >> am11.safelinks.protection.outlook.com%2F%3Furl%3Dhttps%253A%252F%25
> >> 2Flear&data=05%7C01%7Clewis.hutchinson%40missionlabs.co.uk%7Ce1670c
> >> 8a21444edf0ed408dae9f91fac%7C97c26f550a7a4661bd8f7b43b50d3f2b%7C0%7
> >> C0%7C638079553431377130%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDA
> >> iLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sda
> >> ta=ELk0UD1Pleq%2FG9j1%2BT18tiuFe0yjflimR9ztnW%2F7oPM%3D&reserved=0
> >> n.microsoft.com%2Fen-us%2Fmicrosoftteams%2Fdirect-routing-protocols
> >> -si
> >> p&data=05%7C01%7Cbkaufman%40bcmone.com%7C55672dbf739a4ff447f008dae9
> >> b0c
> >> 9b2%7Cafc1818e7b6848568913201b9396c4fc%7C1%7C0%7C638079242759152531
> >> %7C
> >> Unknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6I
> >> k1h
> >> aWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdata=GO26A9FOOoS42yMCWPPGo5PkKo
> >> 75c
> >> kjSJSkjAeYRsU0%3D&reserved=0 I've also tried several different
> >> clients (Bria, Polycom CCX 600, Grandstream, etc) to see if maybe
> >> it was
> something in the SDP or otherwise causing an issue.
> >> SIP Transport is TLS, RTP is SRTP
> >> I might also add that OPTION pings are active and Direct Routing
> Endpoint is active, so this is successful. I also am able to receive
> calls FROM teams to my IP phone(s) without issue. It is only when I
> try and call INTO teams (INVITE -> Microsoft Teams) that I always
> receive a 400 BAD REQUEST to my INVITE.
> >> Any help is appreciated, thank you!
> >>
> >> - Brandon
> >>
> >> __________________________________________________________
> >> Kamailio - Users Mailing List - Non Commercial Discussions To
> >> unsubscribe send an email to sr-users-leave(a)lists.kamailio.org
> >> Important: keep the mailing list in the recipients, do not reply
> >> only
> to the sender!
> >> Edit mailing list options or unsubscribe:
> >
> > --
> > Alex Balashov | Principal | Evariste Systems LLC
> >
> > Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
> > Web:
> https://eur03.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.e
> varistesys.com%2F&data=05%7C01%7Clewis.hutchinson%40missionlabs.co.uk%
> 7Ce1670c8a21444edf0ed408dae9f91fac%7C97c26f550a7a4661bd8f7b43b50d3f2b%
> 7C0%7C0%7C638079553431377130%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwM
> DAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdat
> a=hbww6wEEztE85YqxcN92bpMR%2BkiI9%2BE3jF1i6fQpjEk%3D&reserved=0,
>
> https://eur03.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.c
> srpswitch.com%2F&data=05%7C01%7Clewis.hutchinson%40missionlabs.co.uk%7
> Ce1670c8a21444edf0ed408dae9f91fac%7C97c26f550a7a4661bd8f7b43b50d3f2b%7
> C0%7C0%7C638079553431377130%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMD
> AiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdata
> =4E7a5ZxSvttBiMxl1VTtolmTvS4T5VGZHA00CsbPR2c%3D&reserved=0
> >
> > __________________________________________________________
> > Kamailio - Users Mailing List - Non Commercial Discussions To
> unsubscribe send an email to sr-users-leave(a)lists.kamailio.org
> > Important: keep the mailing list in the recipients, do not reply
> > only to
> the sender!
> > Edit mailing list options or unsubscribe:
> > __________________________________________________________
> > Kamailio - Users Mailing List - Non Commercial Discussions To
> > unsubscribe send an email to sr-users-leave(a)lists.kamailio.org
> > Important: keep the mailing list in the recipients, do not reply
> > only to
> the sender!
> > Edit mailing list options or unsubscribe:
>
> --
> Alex Balashov | Principal | Evariste Systems LLC
>
> Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
> Web:
> https://eur03.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.e
> varistesys.com%2F&data=05%7C01%7Clewis.hutchinson%40missionlabs.co.uk%
> 7Ce1670c8a21444edf0ed408dae9f91fac%7C97c26f550a7a4661bd8f7b43b50d3f2b%
> 7C0%7C0%7C638079553431377130%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwM
> DAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdat
> a=hbww6wEEztE85YqxcN92bpMR%2BkiI9%2BE3jF1i6fQpjEk%3D&reserved=0,
> https://eur03.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.c
> srpswitch.com%2F&data=05%7C01%7Clewis.hutchinson%40missionlabs.co.uk%7
> Ce1670c8a21444edf0ed408dae9f91fac%7C97c26f550a7a4661bd8f7b43b50d3f2b%7
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Outbound calls from my SBC into Teams (Polycom -> SBC -> Teams) always
result in a 400 BAD REQUEST.
Example invite below:
https://pastebin.com/F1G1Ce59
I've taken care to make sure numbers are all E.164 format in
From/To/Contact. I've also taken care to make sure that FQDN is used in
Contact and Record-Route header.
I've tried many different variations and have followed the SIP information
here:
https://learn.microsoft.com/en-us/microsoftteams
/direct-routing-protocols-sip
I've also tried several different clients (Bria, Polycom CCX 600,
Grandstream, etc) to see if maybe it was something in the SDP or otherwise
causing an issue.
SIP Transport is TLS, RTP is SRTP
I might also add that OPTION pings are active and Direct Routing Endpoint
is active, so this is successful. I also am able to receive calls FROM
teams to my IP phone(s) without issue. It is only when I try and call INTO
teams (INVITE -> Microsoft Teams) that I always receive a 400 BAD REQUEST
to my INVITE.
Any help is appreciated, thank you!
- Brandon
Hi out there!
While experimenting with the listen and adverize 'hostname' option I
came across a voice switch, which I suppose fails if the Via header is
not an IPv4 address.
There is a record_route_advertised_address(address) in the RR module,
to set a customer Record-Route Header.
Is there something similar for the Via header so I could try if having
an IP address instead of a hostname in the top-most Via would solve the
issue?
Or is there a way to remove all via header? Remove_hf("Via") in the
branch route trigger garbles the From: HF.
Mit freundlichen Grüssen
-Benoît Panizzon-
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