> > The problem I am having specifically is for outbound calls flowing
> > through a dispatcher, I want to separate calls into a profile for each
> > SIP domain that I send calls to. I'm not sure how do accomplish this.
> > I see in kamctl fifo dlg_list, there is 'caller_bind_addr::
> > udp:10.10.12.24:5060'. Is there a way to export that into a variable
> > that I can use as the 'profile' for the dialog?
>
> The domain part of the request URI is stored in a pseudovariable
> called $rd. I would just use that as your profile 'value'.
>
> --
> Alex Balashov - Principal
Thanks Alex, after I walked away from it for a bit, that came to mind.
Really appreciate the feedback.
Thanks.
JR
>> I recently interconnected to an upstream carrier using Kamailio 3.0,
>> working fine. We have configure 2 SIP trunks for failover/redundnacy.
>> I'm using dispatcher module to round robin calls to the carrier. I
>> wanted to monitor trunk usage between us. I was reading in the devel
>> 3.2 about some native monitoring in the new dispatcher module without
>> the need to load the dialog module.
>>
>> Can I use the new module in version 3.0 or will I need to upgrade the
>> core to 3.2?
>>
>> I typically will not use dev software in production, so can anyone
>> suggest a light method of trunk usage monitoring for ver 3.0? I'm not
>> opposed to using the dialog module, I'm just not familiar with it and
>> not sure the best way to integrate it for usage tracking. I don't
>> need to do billing or capture cdr's on the SIP-Routers so the simpler
>> the better.
>
> Honestly, going the dialog module route is your easiest bet.
>
> --
> Alex Balashov - Principal
Thanks Alex,
Would there be any usage examples of dialog module? I'm not sure if I
really need profiles and [values] and when to set and unset. A push
in the right direction would be helpful.
Thanks.
JR
--
JR Richardson
Engineering for the Masses
Hi All,
I'm trying to configure the dialog module with profiles for outbound
calls to different Dispatcher domains. I can get the dialog module
configured for inbound calls and pull statistics and separate calls
into specific profiles, that part is working well.
The problem I am having specifically is for outbound calls flowing
through a dispatcher, I want to separate calls into a profile for each
SIP domain that I send calls to. I'm not sure how do accomplish this.
I see in kamctl fifo dlg_list, there is 'caller_bind_addr::
udp:10.10.12.24:5060'. Is there a way to export that into a variable
that I can use as the 'profile' for the dialog?
Thanks.
JR
--
JR Richardson
Engineering for the Masses
Hello all,
Have this issue with my ser config. I try to load a fwdbusy value from db
using the $fu class, but it's complaining..
config:
...
if (avp_db_load("$fu", "s:fwdbusy")) {
...
log:
0(20749) ERROR: <core> [pvapi.c:516]: error searching pvar "fu"
0(20749) ERROR: <core> [pvapi.c:720]: wrong char [u/117] in [$fu] at [2 (0)]
0(20749) ERROR: avpops [avpops.c:275]: bad param 1; expected :
$pseudo-variable or int/str value
What's wrong with that? Should it not considered as a ser avp? Not K one. Or
that avp_db_load is an exlusively K function?
Thank you,
Claudio
Dear all,
I'm trying to register Kamailio (version 3.1.3) to different VoIP
provider but I always receive a 401 unauthorized even if I use valid
credentials.
I post below two different capture of the registration to the same
VoIP provider using the same credentials (SIP domain 94.23.56.123, SIP
auth username 1028), with a Blink softphone (successful) and with
Kamailio 3.1.3 (denied). The public IP address of my Kamailio is
anonymized with XXXXXXXX:
------ Blink softphone --------
REGISTER sip:94.23.56.123 SIP/2.0
Via: SIP/2.0/UDP
192.168.2.1:49186;rport;branch=z9hG4bKPjGtybPeuocgJbvOZlRNoWG3RPL0peCs31
Max-Forwards: 70
From: "ItaliADSL" <sip:1028@94.23.56.123>;tag=Tid9EUBxo3Bc5Eo0xzaMg82Vk8q7J57j
To: "ItaliADSL" <sip:1028@94.23.56.123>
Contact: <sip:alybrkhq@192.168.2.115:49186>
Call-ID: cdr3Rr05eEs0voSSZ275zLxVF.Eo74mo
CSeq: 1 REGISTER
Expires: 600
User-Agent: Blink 0.22.2 (MacOSX)
Content-Length: 0
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.2.1:49186;branch=z9hG4bKPjGtybPeuocgJbvOZlRNoWG3RPL0peCs31;received=95.248.232.55;rport=49186
From: "ItaliADSL" <sip:1028@94.23.56.123>;tag=Tid9EUBxo3Bc5Eo0xzaMg82Vk8q7J57j
To: "ItaliADSL" <sip:1028@94.23.56.123>;tag=as5a82d867
Call-ID: cdr3Rr05eEs0voSSZ275zLxVF.Eo74mo
CSeq: 1 REGISTER
User-Agent: MOR Softswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6afe07ca"
Content-Length: 0
REGISTER sip:94.23.56.123 SIP/2.0
Via: SIP/2.0/UDP
192.168.2.1:49186;rport;branch=z9hG4bKPj.JA-9Zf0dpKV6v0JmKtOGSMhyNw5ep1w
Max-Forwards: 70
From: "ItaliADSL" <sip:1028@94.23.56.123>;tag=Tid9EUBxo3Bc5Eo0xzaMg82Vk8q7J57j
To: "ItaliADSL" <sip:1028@94.23.56.123>
Contact: <sip:alybrkhq@192.168.2.115:49186>
Call-ID: cdr3Rr05eEs0voSSZ275zLxVF.Eo74mo
CSeq: 2 REGISTER
Expires: 600
User-Agent: Blink 0.22.2 (MacOSX)
Authorization: Digest username="1028", realm="asterisk",
nonce="6afe07ca", uri="sip:94.23.56.123",
response="f3489b3c8510f39bcf3d1fd96b8836fb", algorithm=MD5
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.2.1:49186;branch=z9hG4bKPj.JA-9Zf0dpKV6v0JmKtOGSMhyNw5ep1w;received=95.248.232.55;rport=49186
From: "ItaliADSL" <sip:1028@94.23.56.123>;tag=Tid9EUBxo3Bc5Eo0xzaMg82Vk8q7J57j
To: "ItaliADSL" <sip:1028@94.23.56.123>;tag=as5a82d867
Call-ID: cdr3Rr05eEs0voSSZ275zLxVF.Eo74mo
CSeq: 2 REGISTER
User-Agent: MOR Softswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Expires: 600
Contact: <sip:alybrkhq@192.168.2.115:49186>;expires=600
Date: Sun, 08 May 2011 20:16:58 GMT
Content-Length: 0
----------- Kamailio 3.1.3 ----------------
REGISTER sip:94.23.56.123 SIP/2.0
Via: SIP/2.0/UDP XXXXXXXXXXX;branch=z9hG4bK58b4.3ac3b834.0
To: sip:1028@94.23.56.123
From: sip:1028@94.23.56.123;tag=464474ecb420f5357c1361be147933f0-45fd
CSeq: 10 REGISTER
Call-ID: 6efffe60-27704@XXXXXXXXXXXX
Content-Length: 0
User-Agent: kamailio (3.1.3 (i386/linux))
Contact: <sip:1028@XXXXXXXXXX>
Expires: 360
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP XXXXXXXXXX;branch=z9hG4bK58b4.3ac3b834.0;received=XXXXXXXXXX
From: sip:1028@94.23.56.123;tag=464474ecb420f5357c1361be147933f0-45fd
To: sip:1028@94.23.56.123;tag=as322a1728
Call-ID: 6efffe60-27704@XXXXXXXXXXX
CSeq: 10 REGISTER
User-Agent: MOR Softswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5f3ee3e8"
Content-Length: 0
REGISTER sip:94.23.56.123 SIP/2.0
Via: SIP/2.0/UDP XXXXXXXXXX;branch=z9hG4bK7666.c1b2e93.0
To: sip:1028@94.23.56.123
From: sip:1028@94.23.56.123;tag=464474ecb420f5357c1361be147933f0-1f9c
CSeq: 10 REGISTER
Call-ID: 6efffe60-27697@XXXXXXXXXXX
Content-Length: 0
User-Agent: kamailio (3.1.3 (i386/linux))
Contact: <sip:1028@XXXXXXXXXXX>
Expires: 360
Authorization: Digest username="1028", realm="asterisk",
nonce="5f3ee3e8", uri="sip:94.23.56.123",
response="df6ebfa9b5870b8b912d1e3b53eedc62", algorithm=MD5
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP XXXXXXXXX;branch=z9hG4bK7666.c1b2e93.0;received=XXXXXXXXX
From: sip:1028@94.23.56.123;tag=464474ecb420f5357c1361be147933f0-1f9c
To: sip:1028@94.23.56.123;tag=as529fc11d
Call-ID: 6efffe60-27697@XXXXXXXXXX
CSeq: 10 REGISTER
User-Agent: MOR Softswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="41f3fe77"
Content-Length: 0
--------------------------------------
As you can see from the captures, Blink softphone successfully
registrates with VoIP provider while Kamailio's request is rejected. I
tried to analize the two flows of SIP messages but I didn't notice any
difference between them so I can't understand why Kamailio is not able
to register.
Can anybody help me to solve this issue?
greetings,
Pierpaolo
I want kamailio to authenticate itself to a host if it is sent a 401, just as that host is expected to authenticate if kamailio sends it one. I am not finding much in the online probably because I am not searching for the right terms. Does anyone have any experience in this?
Thanks!
-Eric
Hi, I've been trying to configure kamailio for enabling BLF for Aastra
phones. Sincerely, it's been a nightmare, till now I'm not sure if I have to
enable pua_dialoginfo or presence_dialoginfo module or both for handling
dialog events. Besides that is not clear which changes should be made to the
route configuration nor which parameters should be configured. I would
appreciate if anyone can enlighten me or could point me to the right
documentation page.
Thanks in advance.
Lucas Alvarez
Hello to all!
I need a little help with our ser installation (ser-2.0.0-rc1).
Due some network restrictions, we need to force every call to pass trought rtpproxy.
Could someone point me to the right place where to force it?
Please an example based on our config would be really helpful, but in two scenarios:
1. ALL CALLS trought rtpproxy
2. ONLY CALLS from a given network trought rtpproxy
We are in doubt on what choose between the two solutions. Having both examples could me help to
implement the needs faster :)
Thank's a lot for your help!
Simon
Dear All,
I'm trying the following and i get error:
$var(new_uri) = "sip:" +$avp(s:term_prefix) +$rU +"@" +$avp(s:gw_address);
rewriteuri=($var(new_uri)); OR append_branch($var(new_uri));
But it seems that it does not like the variable. The documentation says that
rewriteuri get a string so i don't understand why does not work.
Do you have any ideas on this ?
Cheers
Alex
Hi All,
I recently interconnected to an upstream carrier using Kamailio 3.0,
working fine. We have configure 2 SIP trunks for failover/redundnacy.
I'm using dispatcher module to round robin calls to the carrier. I
wanted to monitor trunk usage between us. I was reading in the devel
3.2 about some native monitoring in the new dispatcher module without
the need to load the dialog module.
Can I use the new module in version 3.0 or will I need to upgrade the
core to 3.2?
I typically will not use dev software in production, so can anyone
suggest a light method of trunk usage monitoring for ver 3.0? I'm not
opposed to using the dialog module, I'm just not familiar with it and
not sure the best way to integrate it for usage tracking. I don't
need to do billing or capture cdr's on the SIP-Routers so the simpler
the better.
Any guidance will be much appreciated.
Thanks.
JR
--
JR Richardson
Engineering for the Masses
