Hello,
I am trying to generate sms status report using smsops module but without
success. I couldn't find any example on the net. Can somebody help me?
According to https://en.wikipedia.org/wiki/GSM_03.40 TP-MTI (tpdu(type) in
kamailio) should be "1 0" for status report. Delivery is "0 0" and submit
"0 1". According to the provided examples in smsops module:
tpdu(type) = 4 is delivery
tpdu(type) = 1 is submit
So what's the number for status? I could't find the mapping in the code.
Only this:
// Types of the PDU-Message
typedef enum _pdu_message_type {
DELIVER = 0x00,
SUBMIT = 0x01,
COMMAND = 0x02,
ANY = 0x03,
} pdu_message_type_t;
So any hint or example is appreciated.
Thanks,
Pavel Siderov
Hello
I want to use ka_add_destination function of KA module but i am getting
this error :
[root@SBC-4-1] kamailio -c -f /usr/local/etc/kamailio/kamailio.cfg
loading modules under config path: /usr/local/lib64/kamailio/modules/
0(48016) ERROR: <core> [core/cfg.y:3451]: yyparse(): cfg. parser: failed
to find command ka_add_destination (params 1)
0(48016) CRITICAL: <core> [core/cfg.y:3592]: yyerror_at(): parse error in
config file /usr/local/etc/kamailio/kamailio.cfg, line 1449, column 58:
unknown command, missing loadmodule?
I know that 5.3.3 version does not support this function but 5.4.0 should
as per documentation :
https://kamailio.org/docs/modules/5.4.x/modules/keepalive.html#keepalive.ka…
Keepalive module is compiled and loaded in cfg.
--
Thanks,
Sagar
Hi,
Is it possible to make it work without giving listen IP in pcscf.cfg,
icscf.cfg, scscf.cfg files since i am configuring it as kubernetes pods and
automating all configurations.
Due to this, I am unable to start the kamailio service without giving the
proper listen IP.
Is there any alternative way. Kindly Help
Thanks,
Pavithra
Anyone know where to grab an example config for getting web socket-sip working.
I tried this one but its failing on kamailio restart it might be some very small thing.
https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg
wss://sip.kamailio-box.com:8080
Client = sip.js (webrtc)
SIP server = (mitel) PBX (straight sip)
Dear Daniel/Team,
I had raised one question in "Workshop 3 - Kamailio" at Cluecon 2020(Last
Night), i.e. Can Progress Call(Ringing Calls) be recovered in case of
redundancy with Kamailio. You were told me that straight way it is not
possible but try with hash table once. I had tried following link
https://wazo-platform.org/blog/kamailio-ha-dispatcher-and-dmq and able to
recover Call in progress within 2-3 nodes.
1. My one question is that either this approach will work in production
or not.
2. I have been using Freeswitch for last 6-7 years but "Call in
Progress Recovery in Redundancy" is not possible there in "Freeswitch", So I
tried Kamailio and got success. My Second question is that can it be
possible that Call established on Kamailio and after call set up Kamailio
leave that call and handed over it to Freeswitch for further processing(Like
Re-homing available in OpenSIPS). This will save years of time that I have
invested building features around Freeswitch.
Please suggest me the best way possible to achieve this.
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
Dear Team,
I am glad to work with kamailio Team,
I am using kamailio freeswitch integration. I want to work like this.
My all subscriber register at kamailio. My application server is freeswitch.
Now I want ask is this possible at kamailio?
"when subscriber establish their call then kamailio handed over its call
control to freeswitch" i.e. when kamailio got "ACK" (call patch signal) of
two subscriber then call control will be transferred to freeswitch.
Here I am Using Kamailio for its excellent feature recovery of progress call
(by DMQ and HTable) in Highly Availability(HA) . Freeswitch doesn't have
this feature. But freeswitch we are using as application server.
I have configure Kamailio like "KMUser1==>FS==> KAMUser2" . but in that case
progress call (KMUser1 Get alert of KMUser2) has not recover on HA. So call
will be on kamailio till alerting after connect call will be transferred to
Freeswitch.
I want to know by which module we can achieve this. It will be great help
for me if provided sample .
Thanks
Amit Pal
Coral Telecom Ltd
Hello,
I have kamailio with only one interface. Is it possible to set one RR - lan
ip address to INVITEs (to dispatcher list members) and 127.0.0.1 to replies
200OK (to softphone)?
I can't use record_route() function in onreply route because: Command
cannot be used in the block.
--
Aydar A. Kamalov
What is the point of dns_naptr_ignore_rfc default value:
dns_naptr_ignore_rfc
If the DNS lookup should ignore the remote side's protocol preferences,
as indicated by the Order field in the NAPTR records and mandated by RFC
2915.
dns_naptr_ignore_rfc = yes | no (default yes)
In my (and RFC 2915's) opinion the default should be 'no'.
-- Juha
Hello,
We are using a few carries where we are sending calls to, each of them is
using a special prefix in the From header in order to manage the callerid.
Some of them are not recognizing headers like P-Asserted-Identity or
Remote-Party-ID, so for them we have to replace the From header with the
corresponding prefix. The way we do - it's by using the uac_replace_from
function, which works fine.
The problem is when we are sending the same call to the failover trunk, and
that trunk is using a different prefix. We have to call again the
uac_replace_from function which is appending the new prefix to the existing
one, something like:
From: <sip:12345@domain.localsip:+12345@domain.local>;tag=as404cb50c
I was trying to use uac_restore_from() function and call again
uac_replace_from, but the result is the same.
Is there any way to call the uac_restore_from function twice for the same
call?
Thank you.