Sirs:
We cannot get our domain to configure to the iptel SIP Service.
I have attached a copy of how we've tried setting the DNS, but we
cannot find anywhere that gives us info on the "SIP" and/or
"protocol UDP".
Can you assist us so that we can use our own domain
(http://supercel.tel) so that we can send SIP calls across this
domain?
Sincerely,
Frank Riley, CTO
SuperCel.tel
HelloOur kamailio proxy (version 1.5.2 ) crashed today due to a segmentation fault:Feb 22 16:52:10 AW-PROXY-01 kernel: kamailio[3667]: segfault at 000000000000000b rip 00002ab2b47bf93f rsp 00007ffff62dd270 error 6Apparently due to lack of private memory:[root@AW-PROXY-01 log]# cat proxy.log | grep ERROR | grep dbFeb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:db_unixodbc:db_unixodbc_convert_rows: no private memory left
Feb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:db_unixodbc:db_unixodbc_convert_result: converting rows failed
Feb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:db_unixodbc:db_unixodbc_store_result: failed to convert result
Feb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:core:db_do_query: error while storing result
Feb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:db_unixodbc:db_unixodbc_fetch_result: no private memory left
Feb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:htable:ht_db_load_table: Error while fetching result
I was hoping to see a core file somewhere but I haven't found one yet. I did not have the line "disable_core_dump=no" set on the config file because supposedly is set to no by default:
>From the online guide for my version:
////////////////////
Disable_core_dumpCan be 'yes' or 'no'. By default core dump limits are set to unlimited or
a high enough value. Set this config variable to 'yes' to disable core dump-ing
(will set core limits to 0).Default value is 'no'.
///////////////////How can I make sure that a core file is created after a situation like this? Where will it be created?
How can I check the status of the private memory? I use the "kamctl fifo get_statistics all" but I don't know which one relates to "private memory":
Example:shmem:total_size = 2147483648
shmem:used_size = 16860928
shmem:real_used_size = 17033968
shmem:max_used_size = 19174528
shmem:free_size = 2130449680
shmem:fragments = 1850I know that with so little information is hard to diagnose, but if anybody has an idea as to what else I can do I greatly appreciate it.
Thank you in advancefborot
Hello all,
i am currently confused by using the avpops function avp_delete. When i
run avp_delete without the flag-value \g only the last value will be
unset (like an undo). Is this the wanted behaviour?
EXAMPLE:
$avp(test) = "test"
if( $avp(test)){xlog("L_NOTICE","result1:$avp(test)\n");}
$avp(test) = "nooo";
if( $avp(test)){xlog("L_NOTICE","result2:$avp(test)\n");}
avp_delete("$avp(test)");
if( $avp(test)){xlog("L_NOTICE","result3:$avp(test)\n");}
RESULT:
result1:test
result2:nooo
result3:test
Could anybody helps me to understand that?
thanks in advance,
Sven
Hi,
I need to add a parameter to the contact header. suppose to look like:
Contact: <sip:anonymous@domain:port;param1=XXX;param2=YYY>
How do I add the parameters?
Should i concatenate the $ct with the parameters?
Or is there a better way?
BR,
Uri
Greetings,
I'm not sure if I found a bug, or if I just have something completely misconfigured... I'm a total newb with Kamailio, working on a proof of concept design.
Here's my configuration:
provider -> nat firewall -> kamailio/rtpproxy -> asterisk
For outbound calls from a phone registered to asterisk via kamailio, I'm trying to use fix_nated_sdp("2", "10.50.50.8") to rewrite the media ip address to resolve my audio issues, where 10.50.50.8 is the address outside my firewall. What I'm running into is the 'c=' line doesn't get re-written properly... it inserts the specified address in front of the existing address, and I end up with the following line in my INVITE:
c=IN IP4 10.50.50.810.0.10.10
I have the fix_nated_sdp command under route[sipout], because I only want to use it on calls being sent outside the nat firewall.
Here's the sip invite without the 'fix_nated_sdp' command:
--------------------------------------------------------------------------------------------------------------
INVITE sip:19165551212@xxx.xxx.xxx.xxx SIP/2.0
Record-Route: <sip:10.0.10.10;lr=on;ftag=as5498b77e;nat=yes>
Via: SIP/2.0/UDP 10.50.50.8.;branch=z9hG4bK4b3a.960f6466.0
Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bK145db73e;rport=5060
Max-Forwards: 69
From: "1009" <sip:1009@10.0.10.11>;tag=as5498b77e
To: <sip:19165551212@xxx.xxx.xxx.xxx>
Contact: <sip:1009@10.0.10.11:5060>
Call-ID: 06b8bb1b7dd7801d7b3b9c917fcb9b12@10.0.10.11:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-1.8-r356107
Date: Wed, 22 Feb 2012 03:06:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 309
P-hint: outbound
v=0
o=root 604360056 604360056 IN IP4 10.0.10.10
s=Asterisk PBX SVN-branch-1.8-r356107
c=IN IP4 10.0.10.10
t=0 0
m=audio 9702 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=nortpproxy:yes
--------------------------------------------------------------------------------------------------------------
Here's the sip invite with the 'fix_nated_sdp' command:
--------------------------------------------------------------------------------------------------------------
INVITE sip:19167828326@xxx.xxx.xxx.xxx SIP/2.0
Record-Route: <sip:10.0.10.10;lr=on;ftag=as49e00c81;nat=yes>
Via: SIP/2.0/UDP 10.50.50.8.;branch=z9hG4bK1eab.800c4724.0
Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bK20d28324;rport=5060
Max-Forwards: 69
From: "1009" <sip:1009@10.0.10.11>;tag=as49e00c81
To: <sip:19167828326@xxx.xxx.xxx.xxx>
Contact: <sip:1009@10.0.10.11:5060>
Call-ID: 4def5539675b6f644b99bb300e8ec8d6@10.0.10.11:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-1.8-r356107
Date: Wed, 22 Feb 2012 03:18:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 347
P-hint: outbound
v=0
o=root 1009117068 1009117068 IN IP4 10.0.10.10
s=Asterisk PBX SVN-branch-1.8-r356107
c=IN IP4 10.50.50.8.10.0.10.10
t=0 0
m=audio 13540 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=oldmediaip:10.0.10.11
a=nortpproxy:yes
--------------------------------------------------------------------------------------------------------------
Is this a bug, or is it likely I have something else screwed up?
Thank you in advance for your assistance - this list is an incredible resource!
-Ric
Hi All,
Does it possible on tls module require certificates only some hosts?
My kamailio comunicate with several UA clients (Softphones and IP phones)
and one SIP gateway. I want require certificate only gateway SIP, it's
possible?
The parameter *modparam("tls", "require_certificate", 1)* require
certificate of all clients...
Best Regards
Hi folks,
I have a topology like this:
UA -> kamailio -> PSTN GW.
When I'm placing calls to a PSTN I have to append prefixes so the PSTN
GW knows how to route the call. So the request-uri is
<prefix><number>@<gateway_ip> e.g. 999123456(a)192.168.1.1. When the UA
calls to the GW a "200 OK" message contains a contact header with
999123456(a)192.168.1.1. This is a problem because I don't want my
customers to see gateway prefixes.
The Topoh module only hides IP addresses or the domain part. Is there
any way to remove prefixes. This document
(http://www.kamailio.org/events/2011-Cluecon/DCM-kamailio-security.pdf)
stated "encoding IP and prefixes can be set via parameters". Can I
remove prefixes from the contact in any other way than some contact
header substitutions?
Thanks
Efelin
Dear List
After upgrading from V3.1.5 to V3.2. Kamailio crashes with the below error
messages:
Feb 20 16:43:59 sipproxy /usr/local/sbin/kamailio[11705]: ERROR: <core>
[db.c:392]: invalid number of rows received: 4, dialog_vars
Feb 20 16:43:59 sipproxy /usr/local/sbin/kamailio[11705]: ERROR: <core>
[db.c:419]: querying version for table dialog_vars
Feb 20 16:43:59 sipproxy /usr/local/sbin/kamailio[11705]: ERROR: dialog
[dlg_db_handler.c:153]: error during dialog-vars version check.
Feb 20 16:43:59 sipproxy /usr/local/sbin/kamailio[11705]: ERROR: dialog
[dialog.c:646]: failed to initialize the DB support
Feb 20 16:43:59 sipproxy /usr/local/sbin/kamailio[11705]: ERROR: <core>
[sr_module.c:932]: init_mod(): Error while initializing module dialog
(/usr/local/lib/kamailio/modules_k/dialog.so)
My dialog paramaters are as follows:
# ----------------- DIALOG MODULE
PARAMETERS----------------------------------#
modparam("dialog", "enable_stats", 1)
modparam("dialog", "hash_size", 4096)
modparam("dialog", "rr_param", "did")
modparam("dialog", "dlg_flag", 4)
modparam("dialog", "timeout_avp", "$avp(i:10)")
modparam("dialog", "default_timeout", 21600)
modparam("dialog", "dlg_extra_hdrs", "NULL")
modparam("dialog", "dlg_match_mode", 1)
modparam("dialog", "detect_spirals", 1)
modparam("dialog", "db_url", "mysql://openser:openserrw@localhost/openser")
modparam("dialog", "db_mode", 1)
modparam("dialog", "db_update_period", 60)
modparam("dialog", "db_fetch_rows", 500)
modparam("dialog", "table_name", "dialog")
modparam("dialog", "from_uri_column", "from_uri")
modparam("dialog", "from_tag_column", "from_tag")
modparam("dialog", "to_uri_column", "to_uri")
modparam("dialog", "to_tag_column", "to_tag")
modparam("dialog", "h_id_column", "hash_id")
modparam("dialog", "h_entry_column", "hash_entry")
modparam("dialog", "state_column", "state")
modparam("dialog", "start_time_column", "start_time")
modparam("dialog", "timeout_column", "timeout")
modparam("dialog", "sflags_column", "sflags")
modparam("dialog", "bridge_controller", "sip:controller@kamailio.org")
modparam("dialog", "default_timeout", 7200)
i realised that when i set the
"modparam("dialog", "db_mode", 0)" kamailio starts fine, seems to be a
problem connecting to the database.
Does anyone know what is wrong?
Thanking you in advance
Phillip
Hello,
we have a Kamailio proxy which gets the call from PSTN gw and does some
call forwarding (serial forking) to several destinations through our sbc
The call flow I am looking at is:
- Kamailio sends INVITE to branch_1.
- branch_1 sends 180 with to-tag*, proxy relays it to the gw
* 180 meets the requirements for dialog creating 18x responses in
sections 12.1, 12.1.1 because it contains to-tag, contact and mirrored
record-route.
- After some seconds Kamailio sends a CANCEL to branch_1.
- And sends the INVITE to branch_2.
- branch_1 replies 200 for the CANCEL and 487 for the INVITE.
- branch_2 replies 180 and 200 for the INVITE.
- When PSTN gw receives that it sees it still needs to cancel other
early dialog established by 180 from branch_1.
- The PSTN gw sends a BYE with to-tag of branch_1 to cancel this
specific early dialog.
SIP allows early dialogs to individually released while other dialogs
continue, as written in RFC, section 15:
"The BYE request is used to terminate a specific session or attempted
session. In this case, the specific session is the one with the peer UA
on the other side of the dialog. (...). The caller’s UA MAY send a BYE
for either confirmed or early dialogs, and the callee’s UA MAY send a
BYE on confirmed dialogs, but MUST NOT send a BYE on early dialogs."
The BYE follows the loose routing path, proxy gets 481 from the sbc and
forwards that response back to PSTN gw, which somehow breaks it. AFAICS
it's not specified in RFC what should the behavior look like when
getting both a 200 and error-class response for the INVITE (quotes are
most welcome!).
IMO it would be more correct to absorb BYE in proxy but I see a big
problem here: branch_2 can even ring for 5 minutes and it's not feasible
for proxy to have a wt-timer that long. Also it's not possible to inform
the gw that early dialog has cleared as soon as we receive 200/487 from
branch_1.
So I'm not sure which party is at fault and if we can workaround that
somehow in the Kamailio. Any thoughts?
