Hi, Guys...
I'm challenged to implement a routing that consider the FROM header. That
is, if call goes to X and comes from user U, then route to GW G1, else route
to GW2.
In version 0.9.6, how to find the "from user" value?
Edson
test
On Sun, 2006-11-12 at 12:00 +0000, serusers-request(a)lists.iptel.org
wrote:
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> 1. SER not sending 486 Busy Here (Ravi *****)
> 2. Re: SER pre-release - PRESENCE ISSUES (SER LIST)
> 3. Suggestion (Eddie Johnson Jr)
> 4. SER pre-release - Presence and NOTIFY (SER LIST)
> 5. SER pre-release ser_ctl (SER LIST)
> 6. Re: SER pre-release ser_ctl (Alexandr Dubovikov)
> 7. Re: SER pre-release - PRESENCE ISSUES (Michal Matyska)
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> ----------------------------------------------------------------------
>
> Message: 1
> Date: Sat, 11 Nov 2006 14:15:03 +0200
> From: "Ravi *****" <mravikreddy(a)gmail.com>
> Subject: [Serusers] SER not sending 486 Busy Here
> To: serusers(a)lists.iptel.org
> Message-ID:
> <cfbb389b0611110415l33701327vc3fd09fd9d571e10(a)mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi Users ,
>
> I am Using SER-0.96 , When I make a call from
> SER to Pstn it works fine I mean is all "INV"BYE"CAN".
>
> But when I recieved a call from Pstn I cannot cancel the call because
> X-lite is sending to pstn gateway 480 temp. service not available instead
> of 486 Busy Here.
>
> I am using default config. but I dont understand where
> it went wrong .
>
> Any Idea or help greatly appreciated
>
> Thank You.
> Regards,
> Ravi.
>
On Sun, 2006-11-12 at 12:00 +0000, serusers-request(a)lists.iptel.org
wrote:
> Send Serusers mailing list submissions to
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>
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> or, via email, send a message with subject or body 'help' to
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>
> You can reach the person managing the list at
> serusers-owner(a)lists.iptel.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of Serusers digest..."
>
>
> Today's Topics:
>
> 1. SER not sending 486 Busy Here (Ravi *****)
> 2. Re: SER pre-release - PRESENCE ISSUES (SER LIST)
> 3. Suggestion (Eddie Johnson Jr)
> 4. SER pre-release - Presence and NOTIFY (SER LIST)
> 5. SER pre-release ser_ctl (SER LIST)
> 6. Re: SER pre-release ser_ctl (Alexandr Dubovikov)
> 7. Re: SER pre-release - PRESENCE ISSUES (Michal Matyska)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Sat, 11 Nov 2006 14:15:03 +0200
> From: "Ravi *****" <mravikreddy(a)gmail.com>
> Subject: [Serusers] SER not sending 486 Busy Here
> To: serusers(a)lists.iptel.org
> Message-ID:
> <cfbb389b0611110415l33701327vc3fd09fd9d571e10(a)mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi Users ,
>
> I am Using SER-0.96 , When I make a call from
> SER to Pstn it works fine I mean is all "INV"BYE"CAN".
>
> But when I recieved a call from Pstn I cannot cancel the call because
> X-lite is sending to pstn gateway 480 temp. service not available instead
> of 486 Busy Here.
>
> I am using default config. but I dont understand where
> it went wrong .
>
> Any Idea or help greatly appreciated
>
> Thank You.
> Regards,
> Ravi.
>
Hello Guys
Is there a way to force proxy accounting on radius without/before use
t_relay()?
My problem is:
I have a SIP URI that is received by the proxy but should change that
URI before forward it. For example:
Proxy receive an INVITE: sip:123@teste.com
Proxy should send leg to radius
Proxy change uri to: sip:098@example.com
Proxy shoud foward INVITE (t_relay)
I saw acc_rad_request ("forced") on ser.cfg, but Im not sure about what
it does.
Thanks for any help.
Gustavo
Hi,
Today I found a UAC which is *not* located behind NAT (public IP
1.2.3.4) and sends this Via-Header, which seems perfectly valid
according to RFC3261:
SIP/2.0/UDP VINNASUP06C:5060;maddr=1.2.3.4;branch=z9hG4bK-2198d2
I used to check for nated clients using nat_uac_test("3"), which detects
NAT in this case, because the host-part doesn't match the
received-address. So is the test-flag "2" useless, since the host-part
can be "hostname / IPv4address / IPv6reference", or should this
particular test be extended to also check for the maddr-parameter?
In the meanwhile, I've changed my nat-test to "17" for only testing
Contact and Via-Port instead of Contact and Via-Address, but it's still
not optimal.
Any opinions on this?
Andreas
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Hello,
When I invite new member to instance message conference I get the
message that doesn't contain header:
Content-Type: text/plain
This couses me little bit problems...Should this header include to the
reply?
MESSAGE sip:xx SIP/2.0.
Via: SIP/2.0/UDP xx;branch=z9hG4bK74fb.067ba117.0.
To: sip:xx.
From:sip:x;tag=0c3c711902a8bbb20a1e767e10f39a78-96ab.
CSeq: 10 MESSAGE.
Call-ID: 7bdbafcb-32586@x.
Content-Length: 38.
User-Agent: OpenSer (1.2.0-dev6-notls (i386/linux)).
.
x@x joined room.. left
Best Regards,
Henri Keski-Sikkila
Hello
I'm trying to compile OpenSER for an arm platform (Linux).
The compiler is arm-linux-gcc
I tried the following command:
*$ make CC=arm-linux-gcc*
And there is the following errors:
*cc1: error: invalid option `inline-all-stringops'
cc1: error: bad value (athlon) for -mcpu= switch
*
The arm-linux-gcc -v gives:
*Reading specs from /usr/local/lib/gcc-lib/arm-linux/3.3.3/specs
Configured with: ../configure --target=arm-linux --disable-shared
--disable-threads --with-gnu-as --with-gnu-ld --enable-multilib
--enable-languages=c
Thread model: single
gcc version 3.3.3
*
I also have to modified the libs.
Is there a simple way to do this porting or do I have to check all
compilation and link options and modified or re-write the Makefiles?
Regards,
Michel
Hello- I am running CVS version of ser w/ presence. All seems to be working
well, except that I see a lots of packets being returned by SER indicating
"Subscription does not exist".
SIP/2.0 481 Subscription does not exist.
From: <sip:111@xx.com>;tag=3ffe8a226de90fb471c542cb1065d20a.
To: <sip:111@xx.com>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-9321.
Call-Id:
ssc-3d959a71f819794a430784374ee6f664@4bfc7569f5fad1a1a28a85dcb86badbb.
Cseq: 11 SUBSCRIBE.
Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK01ee.cbc45cb3.0.
Via: SIP/2.0/UDP
192.168.1.3:5065;received=67.X.X.X;branch=z9hG4bK-ecfc3a30c3920b9e;rport=5065.
Server: Sip EXpress router (0.10.99-dev57-tm_fixes (i386/linux)).
Content-Length: 0.
These are in response to a SUBSCRIBE request with Event: presence.winfo -
The packet above was sent by SER in response to:
SUBSCRIBE sip:X.X.X.X:5060 SIP/2.0.
Record-Route: <sip:X.X.X.X;ftag=3ffe8a226de90fb471c542cb1065d20a;lr>.
From: <sip:111@xx.com>;tag=3ffe8a226de90fb471c542cb1065d20a.
To: <sip:111@xx.com>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-9321.
Call-Id:
ssc-3d959a71f819794a430784374ee6f664@4bfc7569f5fad1a1a28a85dcb86badbb.
Cseq: 11 SUBSCRIBE.
Contact: sip:111@sip.tpad.com.
Event: presence.winfo.
Expires: 600.
Date: Tue, 13 Feb 2007 21:18:18 GMT.
Max-Forwards: 16.
User-Agent: SomUserAgent
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, INFO, SUBSCRIBE, NOTIFY,
MESSAGE.
Supported: replaces.
Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK01ee.cbc45cb3.0.
Via: SIP/2.0/UDP
192.168.1.3:5065;received=67.X.X.X;branch=z9hG4bK-ecfc3a30c3920b9e;rport=5065.
Content-Length: 0.
P-hint: Loose Routed.
Any suggestions?
Thanks in advance?
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Hi All,
I,m use openser 1.0.1
We imagined the following setting:
1- It Receive the connections of a pstn that sends all of the requisition
for my one openser
2- it authenticate the coming calls of that IP
3- It Receive the PIN, authenticate and direct the call for the routes
The question is like do that verification of the PIN´s and send the calls
after that verification
It plan is:
It receives itself 0800 123 123 of an ONLY IP, through the
src_ip==xxx.xxx.xxx.xxx sends for a route
I obtained to remove with the strip I number him 0800 123 123 and with the
prefix (""); send with the rewritehostport for a route
I need that be attended, authenticated the PIN and receive of the side "A" I
number of fate and sent for the route of smaller cost
How to made this????
Please help-me!!
Thanks in advanced!!
Jobson Andrade
Projetos & Desenvolvimento
Obelisk - The Asterisk & VoIP Experts
phone/fax: (11) 2164-4808 ext. 115
cell Phone: (11) 8175-9916 / 8271-0480
email: <mailto:jandrade@obelisknet.com.br> jandrade(a)obelisknet.com.br