Hi,
I am wondering why callee info is not visible in dialog details:
Here is the case:
client-1 > kamailio > two asterisks > kamailio > client-2
registration is on asterisk.
dialog module is enabled and I want to see the asterisk IP that is determined by dispatcher module.
I have tried to put dlg_manage function even after dispatch function, but did not help.
I have also tried to update to header before dlg_manage function, it didn't help either.
Any way to have asterisk IP in dialog details.
It shows only this:
{
      h_entry: 483
      h_id: 8644
      ref: 1
      call-id: v489LdfOFD5mnlHupQHOvw..
      from_uri: sip:1515@12.12.12.12;transport=UDP
      to_uri: sip:2222@12.12.12.12
      state: 5
      start_ts: 0
      init_ts: xxxx
      end_ts: xxxx
      duration: xxxx
      timeout: 0
      lifetime: 3600
      dflags: 512
      sflags: 0
      iflags: 0
      caller: {
            tag: d0283007
            contact: sip:1515@1.1.1.1:64966;transport=UDP
            cseq: 1
            route_set:
            socket: udp:172.22.10.10:5060
      }
      callee: {
            tag: <null string>
            contact: <null string>
            cseq: <null string>
            route_set: <null string>
            socket: <null string>
      }
      profiles: {
      }
      variables: {
      }
}
Regards
Get Outlook for Android<https://aka.ms/AAb9ysg>
Hello!
In my architecture, SIP SUBSCRIBE messages can reach the Kamailio
Presence server in several ways.
And I noticed that re-SUBSCRIBEs messages do not update the record_route
field in the active_watchers table (MySQL), so subsequent SIP NOTIFY
messages do not inherit it and have the routes set of the initial SIP
SUBSCRIBE message.
Is there any way to change this behavior?
modparam("presence", "db_url", DBURL)
modparam("presence", "subs_db_mode", 3)
modparam("presence", "timeout_rm_subs", 0)
modparam("presence", "expires_offset", 0)
modparam("presence", "max_expires", 1800)
modparam("presence", "db_update_period", 30)
modparam("presence", "clean_period", 180)
modparam("presence", "send_fast_notify", 1)
modparam("presence", "pres_htable_size", 32)
modparam("presence", "subs_htable_size", 32)
modparam("presence", "publ_cache", 0)
modparam("presence", "notifier_processes", 0)
# kamailio -v
version: kamailio 5.1.2 (x86_64/linux)
--
BR,
Denys Pozniak
Hi,
I would like to move app_lua_sr to kamailio-archive but I would like to hear if anyone out there has a solid reason not to do it. Since we have app_lua as KEMI for a long time I think is time to retire the old remains of app_lua
Cheers
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| ,''`. Victor Seva |
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| `- Debian Developer |
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Hi everybody,
I'm just testing Kamailio 5.4.1 with dialog replication over DMQ. This
seems to work very good. Dialogs are replicated without problems.
When I'm restarting one node I would have expected, that all dialogs are
synced again, just like in dmq_usrloc.
But this does not happen. After a restart the nodes dialog-list is empty.
Did I miss somethin? Is there a special parameter that I have to set?
BR, Björn
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Hello Everyone.
I am looking for documentation to have multiple RTP Engine servers
connected with kamailio using db url and balance the load accordingly.
Thanks
I need to move a few dozen FreePBXen with some commercial modules running in individual VMs to a new data center.
I’m trying to work out a plan to move the PBXes to the new data center in a way that will be transparent to the endpoints, or at the very least with the absolute minimum of downtime.
Some of the installations are rather old, and there’s a handful of peculiarities on each, so the typical FreePBX backup/restore process hasn’t gone smoothly, at least in our tests.
My current train of thought is to put a clone the VMs in the new data center and use Kamailio to route the SIP traffic to the new servers/IPs. I’ve never used it before, so I may be barking up the wrong tree, but it /appears/ to do what I’m suggesting.
If so, I’m thinking I can install Kamailio on each VM, point it to the local asterisk/FreePBX initially, and clone the VM. Then, after the new instance is up and running, point Kamailio on the original VM to the cloned VM's asterisk, after which I can make appropriate DNS changes.
Another option would be to stand up a single Kamailio server and redirect SIP traffic destined for individual asterisk servers to it at the router.
Endpoints are mostly Yealink, and I’m not sure if they’ll feel the need to restart when the registration/SIP server’s IP changes, but a quick bounce when not in use isn’t the end of the world. I’d very much like to avoid having to send an update to each phone manually, but I can script a SIP NOTIFY if required.
Anything stupid, wrong, ignorant, or just smelly about this tactic?
Or, for that matter, any other suggestions?
Hello,
A bit generic question, does Kamailio supports CRLF keepalive per
https://www.rfc-editor.org/rfc/rfc5626#section-4.4.1 ?
It's more for tracking TCP connection states, as seems tcp_keepalive_enable
is not super reliable especially in mobile networks.
--
Best regards,
Ihor (Igor)
Hi List
I am just wondering...
When I am sending the initial INVITE to a customer CPE, this goes
throug the whole location lookup and through a branch route in which I
make some last adjustments to the headers, like removing header the
customer shall not get (like P-Asserted-Identity which would reveal the
caller identity on a callerid restricted call).
On a Re-Invite (session timer refresh) the call is being routed by
loose_route() and immediately sent to RELAY.
uac_replace and uac_restore seem to work fine for stuff like To and
From Header. But how do I prevent unwanted header to disclose
information to the customer which should ne be disclosed?
Do I need to re-arm the branch trigger also for within dialog calls?
Mit freundlichen Grüssen
-Benoît Panizzon-
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Hi
https://lists.kamailio.org/pipermail/sr-users/2020-July/109801.html
I have the exact same issue.
When the B side is starting a new transaction (UPDATE to refresh the
session in my case) without topos enabled, that transaction contains
one or multiple route header and a to_tag.
Therefore loose_route() is true and the call is more or less sent to
route(RELAY) immediately without further checks.
If topos is enabled, all record-route header are removed and only one
Via sent to the CPE.
So when the CPE replies, there is no Route header.
Therefore loose_route returns false.
This makes me wonder, at which stage is topos restoring the route
headers? Shouldn't that happen before loose_route is evaluated?
Mit freundlichen Grüssen
-Benoît Panizzon-
--
I m p r o W a r e A G - Leiter Commerce Kunden
______________________________________________________
Zurlindenstrasse 29 Tel +41 61 826 93 00
CH-4133 Pratteln Fax +41 61 826 93 01
Schweiz Web http://www.imp.ch
______________________________________________________