Hi,
I am trying to send calls to a provider that is switching to Nokia SBCs. They dont like seeing any IP other than our kamailio SBC in the messaging, I changed the contact but I cant find a way to change the owner line in the SDP
I have try mangler and SDPOPS to change but neither seem to work on changing or removing the owner line
Any ideas?
Nick
Hi All:
I am new to the list, I am struggling to find out Kamailio's capabilities
with SIP-I/ISUP. I see that there is SIP-T module in kamailio and ss7ops
module, but I need to get a hold on how to start. I would appreciate if
someone could guide me.
previously, I tried to use OpenSIPS SIP-I to extract ISUP from incoming
calls, i was able to extract ISUP parameters in SDP body as
xlog("Called Number: $isup_param_str(Called Party Number|Address
signal)\r\n");
xlog("Calling Number: $isup_param_str(Calling Party Number|Address
signal)\r\n");
xlog("Location is: $isup_param_str(Location Number)\r\n");
Location parameter is appearing as 0x04971400330268 which is wrongly
interpreted. its proper interpretation should appear as 4100332086.
I also tried below:
xlog("Location is: $isup_param_str(Location Number|Address signal)\r\n");
but this subfield is not present.
The transformation didn't make any difference:
xlog("Location NUM: $(var(isup_body){isup.param, Location Number}
Help pleeez ?
Hi!
I'm using Kamaio in front of multiple Asterisk instances. At this
moment it works as a SIP over Websocket proxy, with rtpengine, for
browser clients to connect to Asterisk using WebRTC. I do not use the
registration module of Kamailio, as each backend Asterisk is
independent and handles its own registrations.
Everything works great when making calls from the browser, and the
routing is correctly executed by Kamailio based on the request SIP
domain. We have an internal routing API that it calls to discover
which backend Asterisk to route the calls.
The issue I have is when a call initiates from that backend Asterisk,
trying to reach a contact that is connected in Kamailio via the
websocket. The Asterisk sends the message to the proxy, and Kamailio
must route it to the corresponding websocket.
I've tried a few approaches:
- using add_contact_alias + handle_ruri_alias: I have the alias with
alias=<ip>~<port>~ws in the contact registration, but for some reason
handle_ruri_alias cannot use it
- using the Path module on Asterisk, so when registering, the path is
recorded and sent back from Asterisk, Kamailio is also not respecting
that
- Using contact_param_encode and contact_param_encode and
contact_param_decode_ruri, but the encoded sip address is always the
invalid websocket, like sip:58c0ktrg@5hp0nn5hqqv9.invalid;transport=ws
None with success. Any hints on that can be wrong? I can share more
detailed information.
Greetings,
Vinicius
Hi. Is there an option for tcp timeout in kamailio?
In my setup we have tcp peers and kamailio pings them every 5 seconds,
but when tcp session stucks kamailio not close this session for 15
minutes. I tried many tcp options kamailio supports, but any of them
helps. I think kamailio can use TCP_USER_TIMEOUT on linux to detect
such tcp sessions and reestablish failed connections?
Hello Community,
There is a module dispatcher and sharing algorithm 11 that features
the ability to redistribute load when the host is enabled or disabled
that is quite valuable for me.
In the test lab I faced unexpected behavior:
- when I add to group <=25 hosts algorithm behaves as per description.
- when a group includes 26 or more hosts, and 100 calls are done for
instance, then about 25 go to the first host and 75 are distributed
among the rest of hosts.
Expected behavior: all hosts of the group having the same rweight
value must receive the same amount of calls no mater what number of
hosts is in the group.
Could you please to help to figure out what is wrong and why when the
dispatcher group consists of 26+ hosts dispatching algorithm 11
behaves differently than when group consists of 25 or less of hosts?
Some configuration notes:
# module configs
modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list")
modparam("dispatcher", "flags", 2)
modparam("dispatcher", "ds_ping_method", "OPTIONS")
modparam("dispatcher", "ds_probing_threshold", 3)
modparam("dispatcher", "ds_inactive_threshold", 10)
modparam("dispatcher", "ds_probing_mode", 3)
modparam("dispatcher", "ds_ping_interval", 10)
modparam("dispatcher", "ds_ping_reply_codes", "501,403,404,400,200")
modparam("dispatcher", "ds_ping_from",DS_PING_FROM_PARAM)
modparam("dispatcher", "use_default", 0)
# dispatching algorithm call
if ( ds_is_from_list("101")) {
sl_send_reply("100","My calls");
ds_select_dst("100", "11");
return;
}
# dispatcher list group config
100 sip:10.60.27.123:7000 0 10 rweight=50
100 sip:10.60.27.123:7001 0 10 rweight=50
100 sip:10.60.27.123:7002 0 10 rweight=50
100 sip:10.60.27.123:7003 0 10 rweight=50
100 sip:10.60.27.123:7004 0 10 rweight=50
100 sip:10.60.27.123:7005 0 10 rweight=50
100 sip:10.60.27.123:7006 0 10 rweight=50
100 sip:10.60.27.123:7007 0 10 rweight=50
100 sip:10.60.27.123:7008 0 10 rweight=50
100 sip:10.60.27.123:7009 0 10 rweight=50
100 sip:10.60.27.123:7010 0 10 rweight=50
100 sip:10.60.27.123:7011 0 10 rweight=50
100 sip:10.60.27.123:7012 0 10 rweight=50
100 sip:10.60.27.123:7013 0 10 rweight=50
100 sip:10.60.27.123:7014 0 10 rweight=50
100 sip:10.60.27.123:7015 0 10 rweight=50
100 sip:10.60.27.123:7016 0 10 rweight=50
100 sip:10.60.27.123:7017 0 10 rweight=50
100 sip:10.60.27.123:7018 0 10 rweight=50
100 sip:10.60.27.123:7019 0 10 rweight=50
100 sip:10.60.27.123:7020 0 10 rweight=50
100 sip:10.60.27.123:7021 0 10 rweight=50
100 sip:10.60.27.123:7022 0 10 rweight=50
100 sip:10.60.27.123:7023 0 10 rweight=50
100 sip:10.60.27.123:7024 0 10 rweight=50
Kamailio version:
kamailio -v
version: kamailio 5.4.4 (x86_64/linux) e16352
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_BLACKLIST, 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: e16352
compiled on 15:56:46 Feb 15 2021 with gcc 4.8.5
System info
CentOS Linux release 7.7.1908 (Core)
Linux test-carrier-1.loc 3.10.0-1062.4.1.el7.x86_64 #1 SMP Fri Oct 18
17:15:30 UTC 2019 x86_64 x86_64 x86_64 GNU/Linux
Hello,
I need to do the simple communication between two Kamailio, ex:
[Side A - K1 (10.0.0.0/24)- UE numbers 500] <----------> [Side B - K2 (192.168.0.0/24) UE numbers 400].
I already looked in the documentation, and I was surprised that I didn't find anything like that.
I don't know which module or configuration to use.
Thanks
Gustavo (BR)
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Dear Team,
I am using SBC as proxy server and I am using Freeswitch for providing
RBT/MOH. Please guide how can I play RBT from Freeswitch while using SBC as
proxy server.
Thanks & Regards,
Amit Sharma
(Sr. Team Leader)
(An ISO 9001:2008 company)
Mobile: <tel:9891612004> tel:9891612004
PH: +91 120 2595870
Ext.: <tel:870> tel:870
Email : <mailto:amitsharma@coraltele.com> amitsharma(a)coraltele.com
Web : <blocked::http://www.coraltele.com> www.coraltele.com
Hello.
We faced an issue that $T_branch_idx called from failure_route returned a
strange result.
Few words about our setup.
We use branch indexes like key in avp when we store via-branch value for
rtpengine.
route[RTPMANAGE]
...
if ( is_request() and is_method("INVITE") and !has_totag() and
t_is_branch_route() )
{
$avp(viabranch$T_branch_idx) = $branch_str;
}
$avp(extra_branch_id) = $avp(viabranch$T_branch_idx);
$var(rtpkey) = $var(rtpkey) + " via-branch=extra";
......
rtpengine_manage($var(rtpkey));
Call this route in branch_route and in failure_route.
In branch_route T_branch_idx is 0, in failure_route is 1.
In documentation:
the index (starting with 1 for the first branch) of the branch for which is
executed the branch_route[]. If used outside of branch_route[] block, the
value is '0'.
But in source code behavior is different. (modules/tmx/t_var.c:480):
for BRANCH_FAILURE_ROUTE and BRANCH_ROUTE it is idx = tcx->branch_index,
but for FAILURE_ROUTE it is idx = t->nr_of_outgoings + nr_branches;
Can you describe why for failure_route it is different and what we can do
in our case? And can you update documentation for this variable if it is
wrong?
Hi,
The behaviour of the sdpops module is not fully clear to me.
Lets say in the INVITE contains in the m= line the payloads 0 99 but has no a= line for codec 0 / PCMU, Would the sdp_keep_codecs_by_name(“PCMU”) keep PCMU ?
What about the case that the INVITE contains in the m=line only the payload 99 but shows an additional a=line for codec 0 / PCMU. Would the sdp_keep_codecs_by_name(“PCMU”) keep PCMU ? If yes, would the extra mtype parameter prevent that? e.g. sdp_keep_codecs_by_name(“PCMU”, mtype)
What happens when several m= lines are present? Does sdp_keep_codecs_by_name consider all m=lines ? Are the treated separately?
What happens to payloads which are not in the hard-coded list in sdpops_data.c ? Can they be used with the sdpops functions?
Thanks for your hints.
Best Gerry
codecsmap_t sdpops_codecsmap_table[] = {
{ {"PCMU", 4}, {"0", 1} },
{ {"GSM", 3}, {"3", 1} },
{ {"G723", 4}, {"4", 1} },
{ {"DVI4", 4}, {"5,6,16,17", 9} },
{ {"LPC", 3}, {"7", 1} },
{ {"PCMA", 4}, {"8", 1} },
{ {"G722", 4}, {"9", 1} },
{ {"L16", 3}, {"10,11", 5} },
{ {"QCELP", 5}, {"12", 2} },
{ {"CN", 2}, {"13", 5} },
{ {"MPA", 3}, {"14", 2} },
{ {"G728", 4}, {"15", 2} },
{ {"G729", 4}, {"18", 2} },
{ {0, 0}, {0, 0} }
};
