I want caller-id to work for a subscriber who has a non-numeric
username.
Subject: Re: [Serusers] caller-id with raius using sip-rpid
There is a Remote-Party-ID header in your message. What else do you
want ? Please be more specific and provide some examples, your
terminology is
hard to follow for me, sorry.
Jan.
On 03-10 16:05, Steve Dolloff wrote:
> Calling from one ata to another, proxied and registered through ser,
> here is my ngrep port 5060 output. Remote-party-id is being set but
not
> the call-id.
>
> U 219.242.10.153:5060 -> 226.145.234.80:5060
> INVITE sip:256@226.145.234.80:5060;user=phone;transport=udp
> SIP/2.0..Max-Fo
> rwards: 10..Record-Route:
> <sip:256@219.242.10.153;ftag=2441840520;lr=on>..V
> ia: SIP/2.0/UDP 219.242.10.153;branch=z9hG4bK41f3.9e2c3976.0..Via:
> SIP/2.0/
> UDP 226.145.234.113:5060;rport=5060..From:
> sip:test@voip2.test.net;tag=24418
> 40520..To: <sip:256@voip2.test.net;user=phone>..Call-ID:
> 2303820380(a)226.145.
> 234.113..CSeq: 2 INVITE..Contact:
> <sip:test@226.145.234.113:5060;transport=
> udp>..User-Agent: Cisco ATA 186 v2.16.2 ata18x
> (030909a)..Authorization: D
> igest
> username="test",realm="voip2.test.net",nonce="3f7de542de99d4e832372e6b
>
>
812a73562d346e6f",uri="sip:256@voip2.test.net",response="1a4552ce6c65333
> 439b
> 6a97672f415a5"..Expires: 300..Content-Length: 271..Content-Type:
> applicatio
> n/sdp..Remote-Party-ID:
> 222;party=calling;id-type=subscriber;screen=no....v
> =0..o=test 5648 5648 IN IP4 226.145.234.113..s=ATA186 Call..c=IN IP4
> 226.14
> 5.234.113..t=0 0..m=audio 16384 RTP/AVP 0 4 8 101..a=rtpmap:0
> PCMU/8000/1..
> a=rtpmap:4 G723/8000/1..a=rtpmap:8 PCMA/8000/1..a=rtpmap:101
> telephone-even
> t/8000..a=fmtp:101 0-15..a=direction:active..
> #
> U 226.145.234.80:5060 -> 219.242.10.153:5060
> SIP/2.0 100 Trying..Via: SIP/2.0/UDP
> 219.242.10.153;branch=z9hG4bK41f3.9e2c
> 3976.0..Via: SIP/2.0/UDP 226.145.234.113:5060;rport=5060..From:
> sip:test@vo
> ip2.test.net;tag=2441840520..To:
> <sip:256@voip2.test.net;user=phone>;tag=2077
> 559661..Call-ID: 2303820380@226.145.234.113..CSeq: 2 INVITE..Server:
> Cisco
> ATA 186 v2.16.2 ata18x (030909a)..Content-Length: 0....
>
>
> Subject: Re: [Serusers] caller-id with raius using sip-rpid
>
> Is there any Remote-Party-ID header field in outgoing INVITE ? If so
> then your gateway/whatever ignores it.
>
> Jan.
>
> On 02-10 15:18, Steve Dolloff wrote:
> > Any phone that supports caller-id. But I haven't gotten to work.
I'm
> > seeing the username instead.
> >
> >
> > Subject: RE: [Serusers] caller-id with raius using sip-rpid
> >
> > Which phone device should I use to see caller ID
> > information when using the ATA186?
> >
> > Alejandro
> >
> > --- Steve Dolloff <sdolloff(a)noc.test.net> escreveu: >
> > At the top of the config I have the following
> > > statement....
> > >
> > > if (method=="INVITE") {
> > > record_route();
> > > if (!radius_www_authorize("")) {
> > > log(1,"radius auth
> > > failure");
> > >
> > > www_challenge("voip2.test.net","0");
> > > break;
> > > };
> > > append_rpid_hf();
> > > };
> > >
> > > If I call a phone with caller-id and my username is
> > > 256, it shows 256 on
> > > the phone, if I call with a username of test, it
> > > shows 4534 on the
> > > handset even though I have the sip-rpid set to 222.
> > >
> > > Stephen
> > >
> > >
> > > Subject: Re: [Serusers] caller-id with raius using
> > > sip-rpid
> > >
> > > Retrieving caller-id when users register doesn't
> > > work because REGISTER
> > > messages are processed by the server and the server
> > > generates a reply
> > > only.
> > >
> > > What you need is to insert Remote-Party-ID header
> > > field into INVITE
> > > message, to make it work you must authenticate also
> > > INVITE messages.
> > > My guess is that you do not authenticate INVITE
> > > messages and therefore
> > > there is nothing append_rpid_hf can add.
> > >
> > > Jan.
> > >
> > > On 02-10 11:35, Steve Dolloff wrote:
> > > > I want to retrieve the caller-id of a user from
> > > radius when they
> > > > register and use it when I redirect an invite
> > > request to the voicemail
> > > > system so that the voicemail system can route the
> > > call based on
> > > > caller-id. I also want to be able to send it to
> > > the sip gateway when
> > > I
> > > > am routing calls to the pstn.
> > > >
> > > > I have this in my config currently and it doesn't
> > > appear to set the
> > > > caller-id in the invite message.
> > > >
> > > >
> > > rewritehostport("219.242.10.153:5061");
> > > > append_rpid_hf();
> > > > t_relay();
> > > >
> > > >
> > > > Subject: Re: [Serusers] caller-id with raius using
> > > sip-rpid
> > > >
> > > > append_rpid_hf has no parameters. Version with 2
> > > parameters is in
> > > > unstable branch in the CVS only. What exactly do
> > > you want to do ?
> > > >
> > > > Jan.
> > > >
> > > > On 02-10 11:26, Steve Dolloff wrote:
> > > > > I found this example in a previous Serusers
> > > message.
> > > > >
> > > > > append_rpid_hf("<sip:+",
> > > > >
> > > > >
> > > >
> > >
> >
>
"@zettou.net>;party=calling;id-type=subscriber;privacy=off;screen=yes");
> > > > >
> > > > > I tried calling append_rpid_hf();, but it does
> > > nothing.
> > > > >
> > > > > I also tried
> > > > >
> > > >
> > >
> >
>
append_rpid_hf("party=calling;id-type=subscriber;privacy=off;screen=yes"
> > > > > ); but I get an unknown command presumably
> > > because I am not using
> > > the
> > > > > right number of parameters.
> > > > >
> > > > > I only want to modify the calling-id info.
> > > > >
> > > > > Can anyone provide an example?
> > > > >
> > > > > -----Original Message-----
> > > > > From: Jan Janak [mailto:jan@iptel.org]
> > > > > Sent: Thursday, October 02, 2003 11:09 AM
> > > > > To: Steve Dolloff
> > > > > Cc: serusers(a)lists.iptel.org
> > > > > Subject: Re: [Serusers] caller-id with raius
> > > using sip-rpid
> > > > >
> > > > > If you want to add Remote-Party-ID header field
> > > then you have to
> > > call
> > > > > append_rpid_hf function in your script.
> > > > >
> > > > > Jan.
> > > > >
> > > > > On 02-10 11:07, Steve Dolloff wrote:
> > > > > > OK, I am setting up a Voicemail system (using
> > > asterisk) and I am
> > > > > > currently doing a rewritehostport(ip:port) and
> > > then trelay() to
> > > send
> > > > > it
> > > > > > to the voicemail system if an invite fails.
> > > > > >
> > > > > > Should I change something? See my ser.cfg and
> > > output from the call
> > > > to
> > > > > > the vm.
> > > > > >
> > > > > > Here is the code from ser.cfg
> > > > > >
> > > > > >
> > > > > rewritehostport("219.242.10.153:5061");
> > > > > > t_relay();
> > > > > >
> > > > > > Here is the sip info from asterisk.
> > > > > >
> > > > > > INVITE sip:200@219.242.10.153:5061;user=phone
> > > SIP/2.0
> > > > > > Max-Forwards: 10
> > > > > > Record-Route:
> > > <sip:200@219.242.10.153;ftag=2236658534;lr=on>
> > > > > > Via: SIP/2.0/UDP
> > > 219.242.10.153;branch=z9hG4bK13de.31901234.0
> > > > > > Via: SIP/2.0/UDP
> > > 226.145.234.113:5060;rport=5060
> > > > > > From: sip:test@voip2.test.net;tag=2236658534
> > > > > > To: <sip:200@voip2.test.net;user=phone>
> > > > > > Call-ID: 2218108971(a)226.145.234.113
> > > > > > CSeq: 1 INVITE
> > > > > > Contact:
> > > <sip:test@226.145.234.113:5060;transport=udp>
> > > > > > User-Agent: Cisco ATA 186 v2.16.2 ata18x
> > > (030909a)
> > > > > > Expires: 300
> > > > > > Content-Length: 271
> > > > > > Content-Type: application/sdp
> > > > > >
> > > > > >
> > > > > >
> > > > > > Subject: Re: [Serusers] caller-id with raius
> > > using sip-rpid
> > > > > >
> > > > > > Remote-Party-ID header field is inserted into
> > > forwarded requests,
> > > > not
> > > > > > responses.
> > > > > >
> > > > > > Jan.
> > > > > >
> > > > > > On 02-10 10:53, Steve Dolloff wrote:
> > > > > > > Ser doesn't appear to be passing the
> > > Caller-id to the ata at
> > > auth
> > > > or
> > > > > I
> > > > > > > am doing something wrong. Can anyone point
> > > me in the right
> > > > > direction?
> > > > > > >
> > > > > > > Thanks,
> > > > > > >
> > > > > > > Stephen
> > > > > > >
> > > > > > > I have the following entry in my freeradius
> > > users file.
> > > > > > >
> > > > > > > test(a)voip2.test.net Auth-Type := Digest,
> > > User-Password == "test"
> > > > > > > Reply-Message = "Hello, test with
> > > digest", Sip-Rpid =
> > > > > > > "8472222222"
> > > > > > >
> > > > > > > When I run a radclient test I get the
> > > correct info..
> > > > > > >
> > > > > > > radclient -f digest.test 219.242.10.153:1812
> > > auth testing
> > > > > > > Received response ID 134, code 2, length =
> > > 57
> > >
> > === message truncated ===
> >
> > Yahoo! Mail - o melhor webmail do Brasil
> > http://mail.yahoo.com.br
Where do I register the login for login on ./www.html/admin/index.php ?
I just can not figuer it out:-( I hade a thoute it was al-right. But no....
Regardes Håkan
Hi folks,
I've updated rtpproxy tarball to the latest version. New features include:
- support for IPv6, courtesy of Jan Janak <jan(a)iptel.org>. This new
feature allows rtpproxy to be used as IPv4/IPv6 RTP bridge.
- rtpproxy now allocates only even ports for RTP. This is necessary to
ensure that RTCP packets generated by one session will not interfere
with another RTP session.
Latest version is available for downloading from:
http://www.portaone.com/~sobomax/rtpproxy.tar
Enjoy!
-Maxim
Hi
I m using RADIATOR with ser0.8.11, while ser is running it doesnt send req to radius, following are snapd of log generated at ser (ngrep o/p)
#
U 202.133.64.66:5060 -> 202.133.64.71:5060
REGISTER sip:voice.cooking.com.pk SIP/2.0..Via: SIP/2.0/UDP 202.133.64.66;branch=z9hG4bKnp1730137267-43b5a45e202.133.64.6
6..From: <sip:12321@voice.cooking.com.pk>;tag=671fccba..To: <sip:12321@voice.cooking.com.pk>..Call-ID: 1969536413-43c1d03
a@1969536416-43c1d037..Contact: <sip:12321@202.133.64.66>;expires=600;q=0.500..Expires: 600..CSeq: 22 REGISTER..Content-
Length: 0..User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13....
#
U 202.133.64.71:5060 -> 202.133.64.66:5060
SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 202.133.64.66;branch=z9hG4bKnp1730137267-43b5a45e202.133.64.66..From: <sip:123
;tag=671fccba..To">21(a)voice.cooking.com.pk>;tag=671fccba..To: <sip:12321@voice.cooking.com.pk>;tag=b27e1a1d33761e85846fc98f5f3a7e58.bdca..Ca
ll-ID: 1969536413-43c1d03a@1969536416-43c1d037..CSeq: 22 REGISTER..WWW-Authenticate: Digest realm="cooking.com.pk", nonce
="3f7d19d62902cb25358e2c666df77d1369d90974"..Server: Sip EXpress router (0.8.11 (i386/linux))..Content-Length: 0..Warning
: 392 202.133.64.71:5060 "Noisy feedback tells: pid=13339 req_src_ip=202.133.64.66 req_src_port=5060 in_uri=sip:voice.co
oking.com.pk out_uri=sip:voice.cooking.com.pk via_cnt==1"....
While here is my ser.cfg
# mysql.so must be loaded !
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_radius.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
#modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
#modparam("auth_dbs", "calculate_ha1", yes)
modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf")
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#
#modparam("auth_db", "password_column", "password")
modparam("auth_radius", "service_type", 15)
:
:
:
:
# Uncomment this if you want to use digest authentication
if (!radius_www_authorize("cooking.com.pk")) {
www_challenge("cooking.com.pk", "0");
break;
};
Any hint?
JF
---------------------------------
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I just found out about this thing called cuphone!
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They let you setup a gateway at home where you can call in from anywhere
in the world, using their software. I have unlimited long distance
service, so I can actually call in from my school using my laptop, and
make unlimited long distance calls through my phone line! If the voice
quality is really good with their USB phone hardware, I can let my
friends call from my laptop to anywhere in the US, provided they buy me
lunch. This is a good way to put POTS to work.
SIP = Sexy Internet Product !
Shahriar Chowdhury
http://freewareCDs.com - The OpenSource Solution Provider.
I see from a 2-week old press release that packet8.net and sipphone.com
are supposed to be compatible, but what are the updates on that? I have
packet8.net (really good service for $20) and wish to get a sipphone for
my girlfriend so that we can talk to each other for free, without
requiring a separate number from packet8. FWD is also offering the
same phone for much cheaper price, so I might just get that. The only
reason I would get sipphone.com's phone will be because of packet8
compatibility.
Did anyone notice the superb possibilities of sipphone in customer
service on VOIP? http://fwd.pulver.com/callme.php?userid=**7474745000
I can use this activeX program to dial FWD or sipphone.com customers.
This also means if I have a sipphone, my customers can call me! Since
FWD comes with free voicemail now, I think it has really nice future in
click-to-talk customer service sector.
Shahriar Chowdhury
http://freewareCDs.com - The OpenSource Solution Provider.
OK, I think this conversation must have been confused by geographic
barriers. In north America, when you call someone your phone number
shows up on their phone. The Caller-ID or ANI. Strangely, the username
of the from address is showing up in my caller-id window.... if the
username is 256(a)test.com, 256 shows up as the caller-id. If the
username is test(a)test.com, 4534 shows up as the caller-id. I want it to
show up as 1XXXXXXXXXX.
Maybe there is a telephony guru on the list that is more familiar with
the north American conventions that can help me to explain it better.
This field is mandatory for voice services here to identify the caller
to emergency services.
Subject: Re: [Serusers] caller-id with raius using sip-rpid
What is caller-id ? Is is a header field or what ? Example, please....
Jan.
On 03-10 16:21, Steve Dolloff wrote:
> I want caller-id to work for a subscriber who has a non-numeric
> username.
>
> Subject: Re: [Serusers] caller-id with raius using sip-rpid
>
> There is a Remote-Party-ID header in your message. What else do you
> want ? Please be more specific and provide some examples, your
> terminology is
> hard to follow for me, sorry.
>
> Jan.
>
> On 03-10 16:05, Steve Dolloff wrote:
> > Calling from one ata to another, proxied and registered through ser,
> > here is my ngrep port 5060 output. Remote-party-id is being set but
> not
> > the call-id.
> >
> > U 219.242.10.153:5060 -> 226.145.234.80:5060
> > INVITE sip:256@226.145.234.80:5060;user=phone;transport=udp
> > SIP/2.0..Max-Fo
> > rwards: 10..Record-Route:
> > <sip:256@219.242.10.153;ftag=2441840520;lr=on>..V
> > ia: SIP/2.0/UDP 219.242.10.153;branch=z9hG4bK41f3.9e2c3976.0..Via:
> > SIP/2.0/
> > UDP 226.145.234.113:5060;rport=5060..From:
> > sip:test@voip2.test.net;tag=24418
> > 40520..To: <sip:256@voip2.test.net;user=phone>..Call-ID:
> > 2303820380(a)226.145.
> > 234.113..CSeq: 2 INVITE..Contact:
> > <sip:test@226.145.234.113:5060;transport=
> > udp>..User-Agent: Cisco ATA 186 v2.16.2 ata18x
> > (030909a)..Authorization: D
> > igest
> >
username="test",realm="voip2.test.net",nonce="3f7de542de99d4e832372e6b
> >
> >
>
812a73562d346e6f",uri="sip:256@voip2.test.net",response="1a4552ce6c65333
> > 439b
> > 6a97672f415a5"..Expires: 300..Content-Length: 271..Content-Type:
> > applicatio
> > n/sdp..Remote-Party-ID:
> > 222;party=calling;id-type=subscriber;screen=no....v
> > =0..o=test 5648 5648 IN IP4 226.145.234.113..s=ATA186 Call..c=IN
IP4
> > 226.14
> > 5.234.113..t=0 0..m=audio 16384 RTP/AVP 0 4 8 101..a=rtpmap:0
> > PCMU/8000/1..
> > a=rtpmap:4 G723/8000/1..a=rtpmap:8 PCMA/8000/1..a=rtpmap:101
> > telephone-even
> > t/8000..a=fmtp:101 0-15..a=direction:active..
> > #
> > U 226.145.234.80:5060 -> 219.242.10.153:5060
> > SIP/2.0 100 Trying..Via: SIP/2.0/UDP
> > 219.242.10.153;branch=z9hG4bK41f3.9e2c
> > 3976.0..Via: SIP/2.0/UDP 226.145.234.113:5060;rport=5060..From:
> > sip:test@vo
> > ip2.test.net;tag=2441840520..To:
> > <sip:256@voip2.test.net;user=phone>;tag=2077
> > 559661..Call-ID: 2303820380@226.145.234.113..CSeq: 2
INVITE..Server:
> > Cisco
> > ATA 186 v2.16.2 ata18x (030909a)..Content-Length: 0....
> >
> >
> > Subject: Re: [Serusers] caller-id with raius using sip-rpid
> >
> > Is there any Remote-Party-ID header field in outgoing INVITE ? If so
> > then your gateway/whatever ignores it.
> >
> > Jan.
> >
> > On 02-10 15:18, Steve Dolloff wrote:
> > > Any phone that supports caller-id. But I haven't gotten to work.
> I'm
> > > seeing the username instead.
> > >
> > >
> > > Subject: RE: [Serusers] caller-id with raius using sip-rpid
> > >
> > > Which phone device should I use to see caller ID
> > > information when using the ATA186?
> > >
> > > Alejandro
> > >
> > > --- Steve Dolloff <sdolloff(a)noc.test.net> escreveu: >
> > > At the top of the config I have the following
> > > > statement....
> > > >
> > > > if (method=="INVITE") {
> > > > record_route();
> > > > if (!radius_www_authorize("")) {
> > > > log(1,"radius auth
> > > > failure");
> > > >
> > > > www_challenge("voip2.test.net","0");
> > > > break;
> > > > };
> > > > append_rpid_hf();
> > > > };
> > > >
> > > > If I call a phone with caller-id and my username is
> > > > 256, it shows 256 on
> > > > the phone, if I call with a username of test, it
> > > > shows 4534 on the
> > > > handset even though I have the sip-rpid set to 222.
> > > >
> > > > Stephen
> > > >
> > > >
> > > > Subject: Re: [Serusers] caller-id with raius using
> > > > sip-rpid
> > > >
> > > > Retrieving caller-id when users register doesn't
> > > > work because REGISTER
> > > > messages are processed by the server and the server
> > > > generates a reply
> > > > only.
> > > >
> > > > What you need is to insert Remote-Party-ID header
> > > > field into INVITE
> > > > message, to make it work you must authenticate also
> > > > INVITE messages.
> > > > My guess is that you do not authenticate INVITE
> > > > messages and therefore
> > > > there is nothing append_rpid_hf can add.
> > > >
> > > > Jan.
> > > >
> > > > On 02-10 11:35, Steve Dolloff wrote:
> > > > > I want to retrieve the caller-id of a user from
> > > > radius when they
> > > > > register and use it when I redirect an invite
> > > > request to the voicemail
> > > > > system so that the voicemail system can route the
> > > > call based on
> > > > > caller-id. I also want to be able to send it to
> > > > the sip gateway when
> > > > I
> > > > > am routing calls to the pstn.
> > > > >
> > > > > I have this in my config currently and it doesn't
> > > > appear to set the
> > > > > caller-id in the invite message.
> > > > >
> > > > >
> > > > rewritehostport("219.242.10.153:5061");
> > > > > append_rpid_hf();
> > > > > t_relay();
> > > > >
> > > > >
> > > > > Subject: Re: [Serusers] caller-id with raius using
> > > > sip-rpid
> > > > >
> > > > > append_rpid_hf has no parameters. Version with 2
> > > > parameters is in
> > > > > unstable branch in the CVS only. What exactly do
> > > > you want to do ?
> > > > >
> > > > > Jan.
> > > > >
> > > > > On 02-10 11:26, Steve Dolloff wrote:
> > > > > > I found this example in a previous Serusers
> > > > message.
> > > > > >
> > > > > > append_rpid_hf("<sip:+",
> > > > > >
> > > > > >
> > > > >
> > > >
> > >
> >
>
"@zettou.net>;party=calling;id-type=subscriber;privacy=off;screen=yes");
> > > > > >
> > > > > > I tried calling append_rpid_hf();, but it does
> > > > nothing.
> > > > > >
> > > > > > I also tried
> > > > > >
> > > > >
> > > >
> > >
> >
>
append_rpid_hf("party=calling;id-type=subscriber;privacy=off;screen=yes"
> > > > > > ); but I get an unknown command presumably
> > > > because I am not using
> > > > the
> > > > > > right number of parameters.
> > > > > >
> > > > > > I only want to modify the calling-id info.
> > > > > >
> > > > > > Can anyone provide an example?
> > > > > >
> > > > > > -----Original Message-----
> > > > > > From: Jan Janak [mailto:jan@iptel.org]
> > > > > > Sent: Thursday, October 02, 2003 11:09 AM
> > > > > > To: Steve Dolloff
> > > > > > Cc: serusers(a)lists.iptel.org
> > > > > > Subject: Re: [Serusers] caller-id with raius
> > > > using sip-rpid
> > > > > >
> > > > > > If you want to add Remote-Party-ID header field
> > > > then you have to
> > > > call
> > > > > > append_rpid_hf function in your script.
> > > > > >
> > > > > > Jan.
> > > > > >
> > > > > > On 02-10 11:07, Steve Dolloff wrote:
> > > > > > > OK, I am setting up a Voicemail system (using
> > > > asterisk) and I am
> > > > > > > currently doing a rewritehostport(ip:port) and
> > > > then trelay() to
> > > > send
> > > > > > it
> > > > > > > to the voicemail system if an invite fails.
> > > > > > >
> > > > > > > Should I change something? See my ser.cfg and
> > > > output from the call
> > > > > to
> > > > > > > the vm.
> > > > > > >
> > > > > > > Here is the code from ser.cfg
> > > > > > >
> > > > > > >
> > > > > > rewritehostport("219.242.10.153:5061");
> > > > > > > t_relay();
> > > > > > >
> > > > > > > Here is the sip info from asterisk.
> > > > > > >
> > > > > > > INVITE sip:200@219.242.10.153:5061;user=phone
> > > > SIP/2.0
> > > > > > > Max-Forwards: 10
> > > > > > > Record-Route:
> > > > <sip:200@219.242.10.153;ftag=2236658534;lr=on>
> > > > > > > Via: SIP/2.0/UDP
> > > > 219.242.10.153;branch=z9hG4bK13de.31901234.0
> > > > > > > Via: SIP/2.0/UDP
> > > > 226.145.234.113:5060;rport=5060
> > > > > > > From: sip:test@voip2.test.net;tag=2236658534
> > > > > > > To: <sip:200@voip2.test.net;user=phone>
> > > > > > > Call-ID: 2218108971(a)226.145.234.113
> > > > > > > CSeq: 1 INVITE
> > > > > > > Contact:
> > > > <sip:test@226.145.234.113:5060;transport=udp>
> > > > > > > User-Agent: Cisco ATA 186 v2.16.2 ata18x
> > > > (030909a)
> > > > > > > Expires: 300
> > > > > > > Content-Length: 271
> > > > > > > Content-Type: application/sdp
> > > > > > >
> > > > > > >
> > > > > > >
> > > > > > > Subject: Re: [Serusers] caller-id with raius
> > > > using sip-rpid
> > > > > > >
> > > > > > > Remote-Party-ID header field is inserted into
> > > > forwarded requests,
> > > > > not
> > > > > > > responses.
> > > > > > >
> > > > > > > Jan.
> > > > > > >
> > > > > > > On 02-10 10:53, Steve Dolloff wrote:
> > > > > > > > Ser doesn't appear to be passing the
> > > > Caller-id to the ata at
> > > > auth
> > > > > or
> > > > > > I
> > > > > > > > am doing something wrong. Can anyone point
> > > > me in the right
> > > > > > direction?
> > > > > > > >
> > > > > > > > Thanks,
> > > > > > > >
> > > > > > > > Stephen
> > > > > > > >
> > > > > > > > I have the following entry in my freeradius
> > > > users file.
> > > > > > > >
> > > > > > > > test(a)voip2.test.net Auth-Type := Digest,
> > > > User-Password == "test"
> > > > > > > > Reply-Message = "Hello, test with
> > > > digest", Sip-Rpid =
> > > > > > > > "8472222222"
> > > > > > > >
> > > > > > > > When I run a radclient test I get the
> > > > correct info..
> > > > > > > >
> > > > > > > > radclient -f digest.test 219.242.10.153:1812
> > > > auth testing
> > > > > > > > Received response ID 134, code 2, length =
> > > > 57
> > > >
> > > === message truncated ===
> > >
> > > Yahoo! Mail - o melhor webmail do Brasil
> > > http://mail.yahoo.com.br
Calling from one ata to another, proxied and registered through ser,
here is my ngrep port 5060 output. Remote-party-id is being set but not
the call-id.
U 219.242.10.153:5060 -> 226.145.234.80:5060
INVITE sip:256@226.145.234.80:5060;user=phone;transport=udp
SIP/2.0..Max-Fo
rwards: 10..Record-Route:
<sip:256@219.242.10.153;ftag=2441840520;lr=on>..V
ia: SIP/2.0/UDP 219.242.10.153;branch=z9hG4bK41f3.9e2c3976.0..Via:
SIP/2.0/
UDP 226.145.234.113:5060;rport=5060..From:
sip:test@voip2.test.net;tag=24418
40520..To: <sip:256@voip2.test.net;user=phone>..Call-ID:
2303820380(a)226.145.
234.113..CSeq: 2 INVITE..Contact:
<sip:test@226.145.234.113:5060;transport=
udp>..User-Agent: Cisco ATA 186 v2.16.2 ata18x
(030909a)..Authorization: D
igest
username="test",realm="voip2.test.net",nonce="3f7de542de99d4e832372e6b
812a73562d346e6f",uri="sip:256@voip2.test.net",response="1a4552ce6c65333
439b
6a97672f415a5"..Expires: 300..Content-Length: 271..Content-Type:
applicatio
n/sdp..Remote-Party-ID:
222;party=calling;id-type=subscriber;screen=no....v
=0..o=test 5648 5648 IN IP4 226.145.234.113..s=ATA186 Call..c=IN IP4
226.14
5.234.113..t=0 0..m=audio 16384 RTP/AVP 0 4 8 101..a=rtpmap:0
PCMU/8000/1..
a=rtpmap:4 G723/8000/1..a=rtpmap:8 PCMA/8000/1..a=rtpmap:101
telephone-even
t/8000..a=fmtp:101 0-15..a=direction:active..
#
U 226.145.234.80:5060 -> 219.242.10.153:5060
SIP/2.0 100 Trying..Via: SIP/2.0/UDP
219.242.10.153;branch=z9hG4bK41f3.9e2c
3976.0..Via: SIP/2.0/UDP 226.145.234.113:5060;rport=5060..From:
sip:test@vo
ip2.test.net;tag=2441840520..To:
<sip:256@voip2.test.net;user=phone>;tag=2077
559661..Call-ID: 2303820380@226.145.234.113..CSeq: 2 INVITE..Server:
Cisco
ATA 186 v2.16.2 ata18x (030909a)..Content-Length: 0....
Subject: Re: [Serusers] caller-id with raius using sip-rpid
Is there any Remote-Party-ID header field in outgoing INVITE ? If so
then your gateway/whatever ignores it.
Jan.
On 02-10 15:18, Steve Dolloff wrote:
> Any phone that supports caller-id. But I haven't gotten to work. I'm
> seeing the username instead.
>
>
> Subject: RE: [Serusers] caller-id with raius using sip-rpid
>
> Which phone device should I use to see caller ID
> information when using the ATA186?
>
> Alejandro
>
> --- Steve Dolloff <sdolloff(a)noc.test.net> escreveu: >
> At the top of the config I have the following
> > statement....
> >
> > if (method=="INVITE") {
> > record_route();
> > if (!radius_www_authorize("")) {
> > log(1,"radius auth
> > failure");
> >
> > www_challenge("voip2.test.net","0");
> > break;
> > };
> > append_rpid_hf();
> > };
> >
> > If I call a phone with caller-id and my username is
> > 256, it shows 256 on
> > the phone, if I call with a username of test, it
> > shows 4534 on the
> > handset even though I have the sip-rpid set to 222.
> >
> > Stephen
> >
> >
> > Subject: Re: [Serusers] caller-id with raius using
> > sip-rpid
> >
> > Retrieving caller-id when users register doesn't
> > work because REGISTER
> > messages are processed by the server and the server
> > generates a reply
> > only.
> >
> > What you need is to insert Remote-Party-ID header
> > field into INVITE
> > message, to make it work you must authenticate also
> > INVITE messages.
> > My guess is that you do not authenticate INVITE
> > messages and therefore
> > there is nothing append_rpid_hf can add.
> >
> > Jan.
> >
> > On 02-10 11:35, Steve Dolloff wrote:
> > > I want to retrieve the caller-id of a user from
> > radius when they
> > > register and use it when I redirect an invite
> > request to the voicemail
> > > system so that the voicemail system can route the
> > call based on
> > > caller-id. I also want to be able to send it to
> > the sip gateway when
> > I
> > > am routing calls to the pstn.
> > >
> > > I have this in my config currently and it doesn't
> > appear to set the
> > > caller-id in the invite message.
> > >
> > >
> > rewritehostport("219.242.10.153:5061");
> > > append_rpid_hf();
> > > t_relay();
> > >
> > >
> > > Subject: Re: [Serusers] caller-id with raius using
> > sip-rpid
> > >
> > > append_rpid_hf has no parameters. Version with 2
> > parameters is in
> > > unstable branch in the CVS only. What exactly do
> > you want to do ?
> > >
> > > Jan.
> > >
> > > On 02-10 11:26, Steve Dolloff wrote:
> > > > I found this example in a previous Serusers
> > message.
> > > >
> > > > append_rpid_hf("<sip:+",
> > > >
> > > >
> > >
> >
>
"@zettou.net>;party=calling;id-type=subscriber;privacy=off;screen=yes");
> > > >
> > > > I tried calling append_rpid_hf();, but it does
> > nothing.
> > > >
> > > > I also tried
> > > >
> > >
> >
>
append_rpid_hf("party=calling;id-type=subscriber;privacy=off;screen=yes"
> > > > ); but I get an unknown command presumably
> > because I am not using
> > the
> > > > right number of parameters.
> > > >
> > > > I only want to modify the calling-id info.
> > > >
> > > > Can anyone provide an example?
> > > >
> > > > -----Original Message-----
> > > > From: Jan Janak [mailto:jan@iptel.org]
> > > > Sent: Thursday, October 02, 2003 11:09 AM
> > > > To: Steve Dolloff
> > > > Cc: serusers(a)lists.iptel.org
> > > > Subject: Re: [Serusers] caller-id with raius
> > using sip-rpid
> > > >
> > > > If you want to add Remote-Party-ID header field
> > then you have to
> > call
> > > > append_rpid_hf function in your script.
> > > >
> > > > Jan.
> > > >
> > > > On 02-10 11:07, Steve Dolloff wrote:
> > > > > OK, I am setting up a Voicemail system (using
> > asterisk) and I am
> > > > > currently doing a rewritehostport(ip:port) and
> > then trelay() to
> > send
> > > > it
> > > > > to the voicemail system if an invite fails.
> > > > >
> > > > > Should I change something? See my ser.cfg and
> > output from the call
> > > to
> > > > > the vm.
> > > > >
> > > > > Here is the code from ser.cfg
> > > > >
> > > > >
> > > > rewritehostport("219.242.10.153:5061");
> > > > > t_relay();
> > > > >
> > > > > Here is the sip info from asterisk.
> > > > >
> > > > > INVITE sip:200@219.242.10.153:5061;user=phone
> > SIP/2.0
> > > > > Max-Forwards: 10
> > > > > Record-Route:
> > <sip:200@219.242.10.153;ftag=2236658534;lr=on>
> > > > > Via: SIP/2.0/UDP
> > 219.242.10.153;branch=z9hG4bK13de.31901234.0
> > > > > Via: SIP/2.0/UDP
> > 226.145.234.113:5060;rport=5060
> > > > > From: sip:test@voip2.test.net;tag=2236658534
> > > > > To: <sip:200@voip2.test.net;user=phone>
> > > > > Call-ID: 2218108971(a)226.145.234.113
> > > > > CSeq: 1 INVITE
> > > > > Contact:
> > <sip:test@226.145.234.113:5060;transport=udp>
> > > > > User-Agent: Cisco ATA 186 v2.16.2 ata18x
> > (030909a)
> > > > > Expires: 300
> > > > > Content-Length: 271
> > > > > Content-Type: application/sdp
> > > > >
> > > > >
> > > > >
> > > > > Subject: Re: [Serusers] caller-id with raius
> > using sip-rpid
> > > > >
> > > > > Remote-Party-ID header field is inserted into
> > forwarded requests,
> > > not
> > > > > responses.
> > > > >
> > > > > Jan.
> > > > >
> > > > > On 02-10 10:53, Steve Dolloff wrote:
> > > > > > Ser doesn't appear to be passing the
> > Caller-id to the ata at
> > auth
> > > or
> > > > I
> > > > > > am doing something wrong. Can anyone point
> > me in the right
> > > > direction?
> > > > > >
> > > > > > Thanks,
> > > > > >
> > > > > > Stephen
> > > > > >
> > > > > > I have the following entry in my freeradius
> > users file.
> > > > > >
> > > > > > test(a)voip2.test.net Auth-Type := Digest,
> > User-Password == "test"
> > > > > > Reply-Message = "Hello, test with
> > digest", Sip-Rpid =
> > > > > > "8472222222"
> > > > > >
> > > > > > When I run a radclient test I get the
> > correct info..
> > > > > >
> > > > > > radclient -f digest.test 219.242.10.153:1812
> > auth testing
> > > > > > Received response ID 134, code 2, length =
> > 57
> >
> === message truncated ===
>
> Yahoo! Mail - o melhor webmail do Brasil
> http://mail.yahoo.com.br
Any phone that supports caller-id. But I haven't gotten to work. I'm
seeing the username instead.
Subject: RE: [Serusers] caller-id with raius using sip-rpid
Which phone device should I use to see caller ID
information when using the ATA186?
Alejandro
--- Steve Dolloff <sdolloff(a)noc.dls.net> escreveu: >
At the top of the config I have the following
> statement....
>
> if (method=="INVITE") {
> record_route();
> if (!radius_www_authorize("")) {
> log(1,"radius auth
> failure");
>
> www_challenge("voip2.test.net","0");
> break;
> };
> append_rpid_hf();
> };
>
> If I call a phone with caller-id and my username is
> 256, it shows 256 on
> the phone, if I call with a username of test, it
> shows 4534 on the
> handset even though I have the sip-rpid set to 222.
>
> Stephen
>
>
> Subject: Re: [Serusers] caller-id with raius using
> sip-rpid
>
> Retrieving caller-id when users register doesn't
> work because REGISTER
> messages are processed by the server and the server
> generates a reply
> only.
>
> What you need is to insert Remote-Party-ID header
> field into INVITE
> message, to make it work you must authenticate also
> INVITE messages.
> My guess is that you do not authenticate INVITE
> messages and therefore
> there is nothing append_rpid_hf can add.
>
> Jan.
>
> On 02-10 11:35, Steve Dolloff wrote:
> > I want to retrieve the caller-id of a user from
> radius when they
> > register and use it when I redirect an invite
> request to the voicemail
> > system so that the voicemail system can route the
> call based on
> > caller-id. I also want to be able to send it to
> the sip gateway when
> I
> > am routing calls to the pstn.
> >
> > I have this in my config currently and it doesn't
> appear to set the
> > caller-id in the invite message.
> >
> >
> rewritehostport("209.242.10.153:5061");
> > append_rpid_hf();
> > t_relay();
> >
> >
> > Subject: Re: [Serusers] caller-id with raius using
> sip-rpid
> >
> > append_rpid_hf has no parameters. Version with 2
> parameters is in
> > unstable branch in the CVS only. What exactly do
> you want to do ?
> >
> > Jan.
> >
> > On 02-10 11:26, Steve Dolloff wrote:
> > > I found this example in a previous Serusers
> message.
> > >
> > > append_rpid_hf("<sip:+",
> > >
> > >
> >
>
"@zettou.net>;party=calling;id-type=subscriber;privacy=off;screen=yes");
> > >
> > > I tried calling append_rpid_hf();, but it does
> nothing.
> > >
> > > I also tried
> > >
> >
>
append_rpid_hf("party=calling;id-type=subscriber;privacy=off;screen=yes"
> > > ); but I get an unknown command presumably
> because I am not using
> the
> > > right number of parameters.
> > >
> > > I only want to modify the calling-id info.
> > >
> > > Can anyone provide an example?
> > >
> > > -----Original Message-----
> > > From: Jan Janak [mailto:jan@iptel.org]
> > > Sent: Thursday, October 02, 2003 11:09 AM
> > > To: Steve Dolloff
> > > Cc: serusers(a)lists.iptel.org
> > > Subject: Re: [Serusers] caller-id with raius
> using sip-rpid
> > >
> > > If you want to add Remote-Party-ID header field
> then you have to
> call
> > > append_rpid_hf function in your script.
> > >
> > > Jan.
> > >
> > > On 02-10 11:07, Steve Dolloff wrote:
> > > > OK, I am setting up a Voicemail system (using
> asterisk) and I am
> > > > currently doing a rewritehostport(ip:port) and
> then trelay() to
> send
> > > it
> > > > to the voicemail system if an invite fails.
> > > >
> > > > Should I change something? See my ser.cfg and
> output from the call
> > to
> > > > the vm.
> > > >
> > > > Here is the code from ser.cfg
> > > >
> > > >
> > > rewritehostport("219.242.10.153:5061");
> > > > t_relay();
> > > >
> > > > Here is the sip info from asterisk.
> > > >
> > > > INVITE sip:200@219.242.10.153:5061;user=phone
> SIP/2.0
> > > > Max-Forwards: 10
> > > > Record-Route:
> <sip:200@219.242.10.153;ftag=2236658534;lr=on>
> > > > Via: SIP/2.0/UDP
> 219.242.10.153;branch=z9hG4bK13de.31901234.0
> > > > Via: SIP/2.0/UDP
> 226.145.234.113:5060;rport=5060
> > > > From: sip:test@voip2.test.net;tag=2236658534
> > > > To: <sip:200@voip2.test.net;user=phone>
> > > > Call-ID: 2218108971(a)226.145.234.113
> > > > CSeq: 1 INVITE
> > > > Contact:
> <sip:test@226.145.234.113:5060;transport=udp>
> > > > User-Agent: Cisco ATA 186 v2.16.2 ata18x
> (030909a)
> > > > Expires: 300
> > > > Content-Length: 271
> > > > Content-Type: application/sdp
> > > >
> > > >
> > > >
> > > > Subject: Re: [Serusers] caller-id with raius
> using sip-rpid
> > > >
> > > > Remote-Party-ID header field is inserted into
> forwarded requests,
> > not
> > > > responses.
> > > >
> > > > Jan.
> > > >
> > > > On 02-10 10:53, Steve Dolloff wrote:
> > > > > Ser doesn't appear to be passing the
> Caller-id to the ata at
> auth
> > or
> > > I
> > > > > am doing something wrong. Can anyone point
> me in the right
> > > direction?
> > > > >
> > > > > Thanks,
> > > > >
> > > > > Stephen
> > > > >
> > > > > I have the following entry in my freeradius
> users file.
> > > > >
> > > > > test(a)voip2.test.net Auth-Type := Digest,
> User-Password == "test"
> > > > > Reply-Message = "Hello, test with
> digest", Sip-Rpid =
> > > > > "8472222222"
> > > > >
> > > > > When I run a radclient test I get the
> correct info..
> > > > >
> > > > > radclient -f digest.test 219.242.10.153:1812
> auth testing
> > > > > Received response ID 134, code 2, length =
> 57
>
=== message truncated ===
Yahoo! Mail - o melhor webmail do Brasil
http://mail.yahoo.com.br
Hi all,
I get a memory error while starting ser with sems.
I'm running a Linux red Hat 9.0 with all updates, I have 512 MB ram, ser
0.8.11 and the latest sems rpm.
I'm using the cfg file for voicemail described in the admin guide 0.8.11
Is there other conf to apply?
thanks for any help
Cordialement
Philippe
_______________________