Hi,
Authentication for users is working fine for me, I add a user to the db and the phone can register. I am looking to connect a SIP trunk but there doesn't seem to be any authentication? What I want is to allow users to connect from any IP but only allow SIP trunks to connect from certain IPs and authenticate.
Thanks,
Keith
Hi list,
I've been trying to make my Yealink phone to give BLF indications but I
haven't been able to achieve this successfully yet so I need some expert
advise here.
My Yealink phone, as soon as it registers to Asterisk, gives me BLF lights.
The same phone registering to Kamailio sends SUBSCRIBE and nothing happens.
The Presence Handling is enabled in configurations and yet no
dialog-info+xml NOTIFY is received.
Please help.
Best Regards,
Sammy Go.
When I make REGISTER request to server Kamailio. Kamailio sometimes return
me REGISTER 200 OK response with many Contacts in Contact header field
Contact:
<sip:user1@1.1.1.1:58492
;transport=TLS;ob>;expires=29;received="sip:1.1.1.1:58492;transport=TLS",
<sip:user1@3.3.3.3:58520;transport=TLS;ob>;expires=244,
<sip:user1@1.1.1.1:58529
;transport=TLS;ob>;expires=284;received="sip:1.1.1.1:58529;transport=TLS",
<sip:user1@3.3.3.3:58548;transport=TLS;ob>;expires=329,
<sip:user1@3.3.3.3:58562;transport=TLS;ob>;expires=393,
<sip:user1@1.1.1.1:58571
;transport=TLS;ob>;expires=483;received="sip:1.1.1.1:58571;transport=TLS",
<sip:user1@2.2.2.2:58588;transport=TLS;ob>;expires=538,
<sip:user1@1.1.1.1:58600
;transport=TLS;ob>;expires=587;received="sip:1.1.1.1:58600;transport=TLS",
<sip:user1@2.2.2.2:58611;transport=TLS;ob>;expires=630,
<sip:user1@1.1.1.1:58624
;transport=TLS;ob>;expires=670;received="sip:1.1.1.1:58624;transport=TLS",
<sip:user1@2.2.2.2:58632;transport=TLS;ob>;expires=706,
<sip:user1@1.1.1.1:58650
;transport=TLS;ob>;expires=826;received="sip:1.1.1.1:58650;transport=TLS",
<sip:user1@2.2.2.2:58661
;transport=TLS;ob>;expires=900;+sip.instance="<urn:uuid:00000000-0000-0000-0000-00007dtrf0a4c>";reg-id=1
Why is that ?
--
Khoa Pham
HCMC University of Science
www.fantageek.com
Hi
I have the same issue with the CLI on unconditional call forwarding being
set to the original caller's number as in this post from March 2012
http://lists.sipwise.com/pipermail/spce-user/2012-March/001149.html
In that post it suggests that version 2.6 might have a fix. I am on 2.8 and
have tried various settings but can't see a way to make the From: header
contain the subscriber's CLI.
Is this still not possible without a custom tweak?
-Barry Flanagan
Never mind...
Problem on my end.... not between keyboard and chair.
Rgds,
Gertjan
From: sr-users-bounces(a)lists.sip-router.org
[mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Gertjan Wolzak
Sent: donderdag 30 mei 2013 11:39
To: sr-users(a)lists.sip-router.org
Subject: [SR-Users] git branch moved?
Gents,
Trying to install Kamailio 4.0 from GIT, but it looks like only the master
is still there...
Something gone wrong ??
root@sip:/usr/local/src/kamailio-4.0/kamailio# git status
# On branch master
nothing to commit (working directory clean)
root@sip:/usr/local/src/kamailio-4.0/kamailio# git checkout -b 4.0
origin/4.0
fatal: git checkout: updating paths is incompatible with switching branches.
Did you intend to checkout 'origin/4.0' which can not be resolved as commit?
root@sip:/usr/local/src/kamailio-4.0/kamailio# git branch
* master
root@sip:/usr/local/src/kamailio-4.0/kamailio# git show
show show-branch
root@sip:/usr/local/src/kamailio-4.0/kamailio# git show
show show-branch
root@sip:/usr/local/src/kamailio-4.0/kamailio# git show-branch
[master] documentation: Rebuild all modified READMEs
root@sip:/usr/local/src/kamailio-4.0/kamailio# git checkout
HEAD master origin/HEAD origin/master
root@sip:/usr/local/src/kamailio-4.0/kamailio# git checkout
Regards,
Gertjan Wolzak
Gents,
Trying to install Kamailio 4.0 from GIT, but it looks like only the master
is still there...
Something gone wrong ??
root@sip:/usr/local/src/kamailio-4.0/kamailio# git status
# On branch master
nothing to commit (working directory clean)
root@sip:/usr/local/src/kamailio-4.0/kamailio# git checkout -b 4.0
origin/4.0
fatal: git checkout: updating paths is incompatible with switching branches.
Did you intend to checkout 'origin/4.0' which can not be resolved as commit?
root@sip:/usr/local/src/kamailio-4.0/kamailio# git branch
* master
root@sip:/usr/local/src/kamailio-4.0/kamailio# git show
show show-branch
root@sip:/usr/local/src/kamailio-4.0/kamailio# git show
show show-branch
root@sip:/usr/local/src/kamailio-4.0/kamailio# git show-branch
[master] documentation: Rebuild all modified READMEs
root@sip:/usr/local/src/kamailio-4.0/kamailio# git checkout
HEAD master origin/HEAD origin/master
root@sip:/usr/local/src/kamailio-4.0/kamailio# git checkout
Regards,
Gertjan Wolzak
Hello:
I followed the step by step guide (http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb) that describe the realtime integration
between Kamailio and Asterisk. I have no problem with registration but when I try a call from 106 to 107 I get the followng error :
1. in the asterisk console: " Unresolvable destination (478/SL)"
2. in the kamailio log:
May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: <core> [resolve.c:1540]: ERROR: sip_hostport2su: could not resolve hostname: "(null)"
May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: tm [ut.h:327]: failed to resolve "(null)"
May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: tm [t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches
May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: sl [sl_funcs.c:371]: ERROR: sl_reply_error used: Unresolvable destination (478/SL)
Any idea about the cause of this problem?
ps:kamailio and asterisk are running in the same machine.
ps:attachment is Asterisk CLI log
Best Regards,
zhengyw
郑友闻
医疗终端部
东软熙康健康有限公司
沈阳市浑南新区新秀街2号，A1楼216
Postcode：110179
Tel：+86 24 83662278
Mobile：15242493836
Email : zhengyw(a)neusoft.com
http://www.neusoft.com
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Dear list
I'm trying to route using the loose_route() method for an ACK message of an
INVITE.
the ACK message contains a Route header field with the outbound proxy
address (which is the server processing this message - aka "myself") and a
flow token to the UAC. but the routing is failed because the
process_outbound() function in loose.c deliberatly ignores my flow-token
because of the condition in loose.c:522 which is:
else if (!ip_addr_cmp(&rcv->src_ip, &_m->rcv.src_ip || rcv->src_port !=
_m->rcv.src_port).
I don't understand why i cannot use this flow token to send the ACK back to
the UAC...
Thanks in advance...
Hi Friends,
I am using Kamailio 4 with SIPML5 HTML client.
When I subscriber for presence I see that data get filled into the
"presentity" table.
But no into the "active_watchers".
When I use QuteCom SIP client
The "active_watchers" table get filled with the contacts in the QuteCom.
Can we fix this for SIPML HTML client ?
Best Regards,
Roy.
