Hi there,
I was wondering if there's a way to log various core error events in some
DB.
For example, below error occurred
Jan 29 11:43:56 kamailio[11076]: {1 11440 INVITE
28b653bd-8f62-4085-a5ef-f2a4ac0f393d } ERROR: <core>
[core/parser/sdp/sdp_helpr_funcs.c:499]: extract_mediaip(): no `IP[4|6]'
address in `c=' field
Jan 29 11:43:56 kamailio[11076]: {1 11440 INVITE
28b653bd-8f62-4085-a5ef-f2a4ac0f393d } ERROR: <core>
[core/parser/sdp/sdp.c:430]: parse_sdp_session(): can't extract common
media IP from the message
Thanks in advance. Regards,
--Sergiu
Hello community,
Please your support, I tell you that I am integrating Kamailio with
MsTeams, after some time reviewing this, I finally achieved the connection
from my Kamailio server to MsTeams and I can validate that the connection
to MsTeams is in AP.
[root @ kamailio-server kamailio] # kamcmd dispatcher.list | egrep "URI |
FLAGS"
URI: sip: sip.pstnhub.microsoft.com;
transport = tls
FLAGS: AP
However, from the admin panel of MsTeams (Direct Routing) I see that the
connection to my sbc "sbc.netvoiceperu.com" is with TLS connectivity status
in "Active" but the SIP options status is in "Warning".
I have made calls from MsTeams thinking that the SIP options status would
change to "active" but it is still in "Warning" state. On the other hand, I
have enabled a siptrace in Kamailio and verify that the SIP OPTIONS from
kamailio are being sent in the following format to MsTeams.
OPTIONS sip: sip.pstnhub.microsoft.com; transport = tls SIP / 2.0
Via: SIP / 2.0 / TLS 161.35.44.66:5061
;branch=z9hG4bKea07.52224687000000000000000000000000.0
To: <sip: sip.pstnhub.microsoft.com; transport = tls>
From: <sip: sbc.netvoiceperu.com>; tag =
d3569c818b500aeb8c373426e76c2884-81763c71
CSeq: 10 OPTIONS
Call-ID: 13ea237a751e0c48-9148(a)161.35.44.66
Max-Forwards: 70
Content-Length: 0
User-Agent: kamailio (5.4.0 (x86_64 / linux))
As you can see, the SIP OPTIONS sent from Kamailio to MsTeams does not
contain the "Contact" field, which in theory said "Contact" field should
have been added by Kamailio according to the configuration added in
kamailio.cfg
event_route [tm: local-request] {
sip_trace ();
if (is_method ("OPTIONS") && $ ru = ~ "pstnhub.microsoft.com") {
append_hf ("Contact: <sip: sbc.netvoiceperu.com: 5061;
transport = tls> \ r \ n");
}
xlog ("L_INFO", "Sent out tm request: $ mb \ n");
}
As additional information, I inform you that I also managed to observe the
SIP OPTIONS that MsTeams sends to Kamailio.
OPTIONS sip: sbc.netvoiceperu.com: 5061; transport = tls SIP / 2.0
FROM: <sip: sip-du-a-eu.pstnhub.microsoft.com: 5061>; tag =
f1bdeb5f-662f-4544-a436-e9aa9ad78da4
TO: <sip: sbc.netvoiceperu.com>
CSEQ: 1 OPTIONS
CALL-ID: c47e2782-16c3-49cb-8931-24e9709d260a
MAX-FORWARDS: 70
VIA: SIP / 2.0 / TLS 52.114.75.24:5061;branch=z9hG4bK48b0e6be
CONTACT: <sip: sip-du-a-eu.pstnhub.microsoft.com: 5061>
CONTENT-LENGTH: 0
USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2021.1.15.7 i.EUWE.10
ALLOW: INVITE, ACK, OPTIONS, CANCEL, BYE, NOTIFY
However I don't see the 200 OK SIP responses from Kamailio to MsTeams.
I think this may be the reason why I see the SIP OPTIONS status in
"Warning" from the MsTeams panel. Maybe the contact field is not being
added in the SIP OPTIONS messages that Kamailio sends to MsTeams and for
that reason I don't see 200OK responses from MsTeams.
Could you help me solve this please.
Cheers
Saludos Cordiales
--
*Willy Valles Rios*
*Unified Communications Specialist*
phone: +51955747343
em@il: willyvalles17(a)gmail.com
Hello,
I am trying to configure Kamailio as both an inbound and outbound proxy. Inbound requests flow Sip Trunk -> Kamailio -> Asterisk. Outbound requests flow Asterisk -> Kamailio -> Sip Trunk. Inbound traffic is sent to port 5060 on kamailio which listens on the PRIVATE_IP:5060 and advertises PUBLIC_IP:5060. How can we ensure that when messages are sent to the Sip Trunk they have Record-Route headers with the PUBLIC_IP and when messages are sent to asterisk they have Record-Route headers with the PRIVATE_IP?
Hello,
I got some time Ack packets looping on my servers during 4 hours if I don’t restart the system ( 4 hours is my dialog timeout).
Jan 27 16:59:02 vma-2 kamailio[4544]: WARNING: { 1 # SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020 * } dialog [dlg_handlers.c:1322]: dlg_onroute(): unable to find dialog for ACK with route param 'dc2.f24' [717:1071] and call-id 'SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020'
Jan 27 16:59:02 vma-2 kamailio[4545]: WARNING: { 1 # SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020 * } dialog [dlg_handlers.c:1322]: dlg_onroute(): unable to find dialog for ACK with route param 'dc2.f24' [717:1071] and call-id 'SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020'
Jan 27 16:59:02 vma-2 kamailio[4545]: WARNING: { 1 # SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020 * } dialog [dlg_handlers.c:1322]: dlg_onroute(): unable to find dialog for ACK with route param 'dc2.f24' [717:1071] and call-id 'SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020'
Jan 27 16:59:02 vma-2 kamailio[4544]: WARNING: { 1 # SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020 * } dialog [dlg_handlers.c:1322]: dlg_onroute(): unable to find dialog for ACK with route param 'dc2.f24' [717:1071] and call-id 'SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020'
Jan 27 16:59:02 vma-2 kamailio[4545]: WARNING: { 1 # SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020 * } dialog [dlg_handlers.c:1322]: dlg_onroute(): unable to find dialog for ACK with route param 'dc2.f24' [717:1071] and call-id 'SDvl91b03-576e9e12b1569f0361fceb3625955037-agvh8j0020'
It generates approximatively 2 millions packets per loop.
On the last occurrence I noticed a ‘SIP/2.0 200 canceling’ generated per Kamailio before the loop started (which seems uncomon on this host)
Bellow is a screenshot of the beginning of the loop and may dialog and topos config.
Has anyone encountered such behavour ?
Regards,
David
[cid:image001.png@01D6F4E2.F68B5140]
modparam("dialog", "dlg_flag", 1)
modparam("dialog", "hash_size", 1024)
modparam("dialog", "default_timeout", 14500)
modparam("dialog", "event_callback", "ksr_dialog_event")
modparam("dialog", "db_url", DBURL_DLG)
modparam("dialog", "db_update_period", 15)
modparam("dialog", "db_mode", 1)
modparam("topos", "storage", "redis")
modparam("topos", "dialog_expire", 14500)
modparam("topos", "contact_host", "PUBLICIP")
modparam("topos_redis", "serverid", "topos")
Hey group,
I was wondering if there is a way to control the websocket buffer size to
support large SIP messages. I received the following error when sending
content-length of 65000
2021-01-26T17:32:19.251408+00:00 sjomainkama50 kamailio[31871]: 109(32008)
WARNING: websocket [*ws_frame.c*:482]: decode_and_validate_ws_frame():
message is too long for our buffer size (65535 / 66151)
I am assuming this is controlled by the tcp_rd_buf params which I have set
to 131k:
*tcp_*rd_buf_size=131072
and this is my tcp_wr_buf params:
*tcp_*wq_blk_size=131072
*tcp_*conn_wq_max=131072
Please advise.
Thanks.
--
Andy Chen
Sr. Telephony Lead Engineer
achen@ <achen(a)thinkingphones.com>fuze.com
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