SER will try to resolve using DNS SRV and A lookups unless you specify IP
addresses, so:
route[2]
{
#Host and port where Asterisk is listening, sip.conf
rewritehostport("192.168.1.5:5070");
route(1);
break;
}
g-)
Antonio Cano damas wrote:
> Hi,
>
> That seems to start working :). I've made the next changes:
> listen="redstar.organization.org"
> alias="192.168.1.5"
> alias="organization.org"
> alias="redstar.organization.org"
>
> check_via=no # (cmd. line: -v)
> dns=no # (cmd. line: -r)
> rev_dns=no # (cmd. line: -R)
>
> And the proper updates into MySQL Database for domain like
> 'redstar.organization.org'.
>
> Now I've got happening something strange, my UAs x-lite and
> grandstream can register but Asterisk doesn't (at first look).
> Looking the URIS I've seen that X-lite generate
> @192.168.1.5redstar.organization.org while Asterisk generate
> @redstar.organization.org. Finally I've got registering the Asterisk
> using fromdomain= 192.168.1.5restar.organization.org :-/
>
> My next step is try to make a call from X-Lite to one extension of
> Asterisk. In the SER.cfg:
> A) Into the INVITE process section:
> if (uri =~ "^sip:(2[0-9][0-9])@*" ) {
> log(1,"Al asterisk\n");
> route(2);
> break;
> };
> B)
> route[2]
> {
> #Host and port where Asterisk is listening, sip.conf
> rewritehostport("192.168.1.5redstar.organization.org:5070");
> route(1);
> break;
> }
>
> route[1]
> {
> # if client or server know to be behind a NAT, enable relay
> if (isflagset(6)) {
> #log(1, "Pasando por force_rtp_proxy\n");
> append_hf("P-Hint: Pasando por RTP_PROXY\r\n");
> force_rtp_proxy();
> };
>
> # labeled all transaction for accounting
> #setflag(4);
>
> # send it out now; use stateful forwarding as it works reliably
> # even for UDP2TCP
> if (!t_relay()) {
> sl_reply_error();
> };
> }
>
> With that the INVITE petition gets an Request Timeout response,
> making a little of ngrep:
>
> U IP_Client_UA:5060 -> 192.168.1.5:5060
> INVITE sip:213@192.168.1.5redstar.organization.org SIP/2.0..Via:
> SIP/2.0/UDP
> 192.168.0.6:5060;rport;branch=z9hG4bK4F9444CAD6AD11D9B421000A95A55E26..From:
> Anto
> nio F. Cano <sip:10101@redstar.organization.org>;tag=1817289323..To:
> <sip:213@redstar.organization.org>..Contact:
> <sip:10101@192.168.0.6:5060>..Call-ID: 4F13C7CB-D
> 6AD-11D9-B421-000A95A55E26@192.168.0.6..CSeq: 3733
> INVITE..Max-Forwards: 70..Content-Type: application/sdp..User-Agent:
> X-Lite release 1103m..Content-Len
> gth: 263....v=0..o=10101 5457996 5458244 IN IP4
> 192.168.0.6..s=X-Lite..c=IN IP4 192.168.0.6..t=0 0..m=audio 8000
> RTP/AVP 3 0 8 98 101..a=rtpmap:0 pcmu/80
> 00..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98
> iLBC/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101
> 0-15..
> #
>
> U 192.168.1.5:5060 -> IP_DNS_SERVER:5070
> INVITE sip:213@192.168.1.5redstar.organization.org:5070
> SIP/2.0..Record-Route:
> <sip:213@192.168.1.5;ftag=1817289323;lr=on>..Via: SIP/2.0/UDP
> 192.168.1.5;branc h=z9hG4bK151f.afc77ca4.0..Via: SIP/2.0/UDP
> 192.168.0.6:5060;received=IP_Client_UA;rport=5060;branch=z9hG4bK4F9444CAD6AD11D9B421000A95A55E26..From:
> Antonio
> F. Cano <sip:10101@redstar.organization.org>;tag=1817289323..To:
> <sip:213@redstar.organization.org>..Contact:
> <sip:10101@IP_Client_UA:5060>..Call-ID: 4F13C7CB-D6AD-1
> 1D9-B421-000A95A55E26@192.168.0.6..CSeq: 3733 INVITE..Max-Forwards:
> 69..Content-Type: application/sdp..User-Agent: X-Lite release
> 1103m..Content-Length:
> 283..P-Hint: Nat uac_test=3 ..P-Hint: Metodo INVITE corrigiendo
> SDP..P-Hint: Pasando por RTP_PROXY....v=0..o=10101 5457996 5458244 IN
> IP4 192.168.0.6..s=
> X-Lite..c=IN IP4 192.168.0.6..t=0 0..m=audio 8000 RTP/AVP 3 0 8 98
> 101..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3
> gsm/8000..a=rtpmap:98 iLBC
> /8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101
> 0-15..a=direction:active..
>
> Why SER sends the message to IP_DNS_SERVER in place of
> 192.168.1.5redstar.organization.org?
>
> What's happen here? I don't understand nothing. What I make wrong? The
> @192.168.1.5redstar.organization.org doesn't seem to be so elegant is
> there any way to make it like redstar.organization.org
>
> Kind regards and thanks for your time,
> Antonio F. Cano
>
> Greger V. Teigre wrote:
>
>> You can turn off SRV in X-Lite and Grandstream (Use DNS SRV option)
>> And use SIP proxy: fully qualified domain name (same as outbound)
>> g-)
>> Antonio Cano damas wrote:
>>
>>> Hello all,
>>>
>>> First of all excuses for my bad english. I was looking into the list
>>> and doesn't find nothing similar to my problem maybe I didn't use
>>> the correct keywords.
>>>
>>> I've got a Proxy that is behaind and Public IP and I've got a
>>> subdomain name pointing to this machine. But the DNS server isn't
>>> mine and I can't add a SRV record, I've redirected all petitions to
>>> this server and in that way if I make a petition of anykind service
>>> (ssh, web, ...) to the subdomain.name.com it comes into the
>>> machine. The problem is that all de UA i try (X-Lite, Grandstream and
>>> Asterisk) makes DNS SRV queries and i don't know how to avoid this.
>>>
>>> For example:
>>>
>>> My subdomain is redstar.organization.org and I'd like to use the
>>> realm organization.org
>>>
>>> + File /etc/hosts
>>> 192.168.1.5 Public_IP_Of_RedStar redstar.organization.org
>>> redstar
>>>
>>> + In /etc/init.d/ser I've got this params line:
>>> PARAMS="-P $PIDFILE -u root -g root -l
>>> redstar.organization.org" --->I try using -l
>>> organization.org, but the server said to me: 0(0)
>>> ERROR: udp_init: bind(5, 0x80c8cc0, 16) on IP_Organization.org:
>>> Cannot assign requested address
>>>
>>> + In /etc/ser/ser.cfg added this:
>>> alias="organization.org"
>>> alias="redstar.organization.org"
>>>
>>> + MySQL tables the domain field value is 'organization.org'
>>>
>>> When I start the SER it tells me:
>>> Restarting ser: serListening on
>>> Public_IP_Of_Redstar [192.168.1.5]:5060
>>> Aliases: redstar:5060 redstar.organization.org:*
>>> organization.org:* The UA Grandstream config is:
>>> SIP Server: organization.org
>>> Outbound Proxy: redstar.organization.org
>>>
>>> And the received answer is a 478 Unresolveable destination. Can
>>> anyone help me? What can I do? I'm a little desesperated :(
>>>
>>> Thanks in advance,
>>> Antonio F. Cano
>>>
>>> Ngrep log result into redstar.organization.org machine:
>>>
>>> U UA_Public_IP:62303 -> 192.168.1.5:5060
>>> REGISTER sip:192.168.1.5organization.org SIP/2.0..Via: SIP/2.0/UDP
>>> 192.168.0.9:62303;branch=z9hG4bK083fbe
>>> 2c60e48e32..From: "Antonio F. Cano (iGT)"
>>> <sip:10101@organization.org;user=phone>;tag=b511698681c5146b..T
>>> o: <sip:10101@organization.org;user=phone>..Contact:
>>> <sip:10101@192.168.0.9:62303;user=phone>..Call-ID: 3
>>> 4ee28e5dad286d9@192.168.0.9..CSeq: 100 REGISTER..Expires:
>>> 3600..User-Agent: Grandstream HT487 1.0.5.
>>> 18..Max-Forwards: 70..Allow:
>>> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Leng
>>> th:
>>> 0....
>>>
>>> #
>>>
>>> Why this 'sip:192.168.1.5organization.org'?
>>>
>>> U 192.168.1.5:32863 -> 195.235.113.3:53
>>>
>>> Z............_sip._udp.192.168.1.5organization.org..!..
>>>
>>>
>>> ######################################################################
>>> U UA_Public_IP:62303 -> 192.168.1.5:5060
>>> REGISTER sip:192.168.1.5organization.org SIP/2.0..Via: SIP/2.0/UDP
>>> 192.168.0.9:62303;branch=z9hG4bK083fbe
>>> 2c60e48e32..From: "Antonio F. Cano (iGT)"
>>> <sip:10101@organization.org;user=phone>;tag=b511698681c5146b..T
>>> o: <sip:10101@organization.org;user=phone>..Contact:
>>> <sip:10101@192.168.0.9:62303;user=phone>..Call-ID: 3
>>> 4ee28e5dad286d9@192.168.0.9..CSeq: 100 REGISTER..Expires:
>>> 3600..User-Agent: Grandstream HT487 1.0.5.
>>> 18..Max-Forwards: 70..Allow:
>>> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Leng
>>> th:
>>> 0....
>>>
>>> #########################################################
>>> U 195.235.113.3:53 -> 192.168.1.5:32863
>>>
>>> Z............_sip._udp.192.168.1.5organization.org..!...)......*0.=.a.gtld-servers.).nstld.verisign-grs.c
>>>
>>>
>>> om.B..@..........:.....
>>>
>>> #
>>> U 192.168.1.5:32863 -> 195.235.113.3:53
>>>
>>> Z............_sip._udp.192.168.1.5organization.org..!..
>>>
>>> ##########
>>> U 195.235.113.3:53 -> 192.168.1.5:32863
>>>
>>> Z............_sip._udp.192.168.1.5organization.org..!...)......*0.=.a.gtld-servers.).nstld.verisign-grs.c
>>>
>>>
>>> om.B..@..........:.....
>>>
>>> ##################################################################################################################
>>>
>>> U UA_Public_IP:62303 -> 192.168.1.5:5060
>>> REGISTER sip:192.168.1.5organization.org SIP/2.0..Via: SIP/2.0/UDP
>>> 192.168.0.9:62303;branch=z9hG4bK083fbe
>>> 2c60e48e32..From: "Antonio F. Cano (iGT)"
>>> <sip:10101@organization.org;user=phone>;tag=b511698681c5146b..T
>>> o: <sip:10101@organization.org;user=phone>..Contact:
>>> <sip:10101@192.168.0.9:62303;user=phone>..Call-ID: 3
>>> 4ee28e5dad286d9@192.168.0.9..CSeq: 100 REGISTER..Expires:
>>> 3600..User-Agent: Grandstream HT487 1.0.5.
>>> 18..Max-Forwards: 70..Allow:
>>> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Leng
>>> th:
>>> 0....
>>>
>>> ###########################################################################################################################
>>>
>>> U 192.168.1.5:5060 -> UA_Public_IP:62303
>>> SIP/2.0 478 Unresolveable destination (478/TM)..Via: SIP/2.0/UDP
>>> 192.168.0.9:62303;branch=z9hG4bK083
>>> fbe2c60e48e32;rport=62303;received=IP_Client_UA..From: "Antonio F.
>>> Cano (iGT)" <sip:10101@organization.org;
>>> user=phone>;tag=b511698681c5146b..To:
>>> <sip:10101@organization.org;user=phone>;tag=1b908f8c725dfd3e50cbc74
>>> f548dfb0d-588b..Call-ID: 34ee28e5dad286d9@192.168.0.9..CSeq: 100
>>> REGISTER..Server: Sip EXpress route
>>> r (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392
>>> 80.38.244.122:5060 "Noisy feedback tells:
>>> pid=8023 req_src_ip=IP_Client_UA req_src_port=62303
>>> in_uri=sip:192.168.1.5organization.org out_uri=sip:192.
>>> 168.1.5organization.org
>>> via_cnt==1"....
>>>
>>>
>>> _______________________________________________
>>> Serusers mailing list
>>> serusers(a)lists.iptel.org
>>> http://lists.iptel.org/mailman/listinfo/serusers
I have recently found out that our current gateway that we use for our
pstn will not do both h323 and sip at the same time. I am looking for a
way to possible sip trunk from my Cisco Callmanger to ser or just use a
cisco gateway to do it. I am not so sure about the cisco gateway
supporting both h323 and sip. I would appreciate any advice on this
situation. Thanks in advance.
Wercs Communications
Clay Bryan
Network Administrator
Wercs Communications
400 East First
Casper, WY 82601
<http://maps.yahoo.com/py/maps.py?Pyt=Tmap&addr=400+East+First&csz=Caspe
r%2C+WY++82601&country=us>
CBryan(a)wercs.com <mailto:CBryan@wercs.com>
tel:
fax:
mobile:
307-233-8359
307-233-8701
307-258-7371
Add me to your address book...
<https://www.plaxo.com/add_me?u=21475170292&v0=646557&k0=282754785>
Want a signature like this? <http://www.plaxo.com/signature>
I heard somebody wanted to use LDAP instead of MySQL, currently Im
developing a module to replace MySQL database with an LDAP directory. If
you are interested please check the following URL:
http://rocksteady.cs.fiu.edu/SER-LDAP/
Regards,
Alberto
Hi all.
I just have my SER running on a FreeBSD machine. I put the export
SIP_DOMAIN= 10.0.1.10 line in all the .profile and profile files I found but
I still have the same "You need to set...."phrase when triyng to run serctl
I also tried to do export from the console but I got export:command not
found.
The shell is csh.
Any help will be appreciated.
Juan Ferrari
Hello all,
I got SER working, to add voicemail service. I need a
help which one is better and why Sems-voicemail or
Asterisk-voicemail.
Thanks
Ahmed
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
Mediaproxy will be between the clients for RTP only. Signaling will be
relayed by SER.
-----Original Message-----
From: a cairo [mailto:a.cairo@scsitalia.it]
Sent: Saturday, June 04, 2005 6:14 AM
To: Giudice, Salvatore; bogdan aux; serusers(a)lists.iptel.org
Subject: RE: [Serusers] Mediaproxy in LAN (?)
Hi all,
i don't understand architecture of mediaproxy with ser !
Is mediaproxy between client and ser?
Thanks
-----Original Message-----
From: "Giudice, Salvatore" <Salvatore.Giudice(a)FMR.COM>
To: "bogdan aux" <aux1d(a)yahoo.com>, <serusers(a)lists.iptel.org>
Date: Tue, 17 May 2005 20:10:46 -0400
Subject: RE: [Serusers] Mediaproxy in LAN (?)
> I am trying to route a call from a non-routable 10.x.x.x network to a
> phone on the internet using ser 0.9.0 and mediaproxy 1.31. Aka: PSTN
to
> SIP from my asterisk to an x-lite
>
> I found some references indicating that I should call:
>
>
> if ( src_ip = IP_GATEWAY)
> {
> force_rport();
> fix_contact();
> use_media_proxy();
> };
>
> Supposedly, you need the force_rport and the fix_contact to put media
> proxy in the middle of the two rtp sessions, but I have et to see this
> actually work. Has anyone been able to pass a call through media proxy
> from a GW on a non-routable to a routable network, or even between two
> ip's on the same network as the mediaproxy?
>
> If so, please share your configs.
>
> -----Original Message-----
> From: bogdan aux [mailto:aux1d@yahoo.com]
> Sent: Friday, May 13, 2005 6:16 AM
> To: serusers(a)lists.iptel.org
> Subject: [Serusers] Mediaproxy in LAN (?)
>
> I want to route RTP packets between 2 computers in the
> same network through Mediaproxy. I know this may have
> no sense, but it's just a test configuration.
> The call use_media_proxy() seems to have no effect,
> the RTP packets go directly from one computer to
> another.
> My guess is mediaproxy makes a test on the IPs (they
> are 192.168.147.2 and 192.168.147.3) and refuses to
> proxy the call.
> Is mediaproxy working just between different networks
> or it's an error in my configuration file?
>
>
>
>
> Yahoo! Mail
> Stay connected, organized, and protected. Take the tour:
> http://tour.mail.yahoo.com/mailtour.html
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Is it possible to rewrite a sip status received from a gateway? I
mean, if I get a 183 Session Progress I want to rewrite it to a 180
Ringing. I've tried using the replace function but it only works for
the "From" and "To" fiels.
This is what I've used:
onreply_route[1] {
if (status =~ "183") {
log(1, "183 received");
replace("183 Session Progress", "180 Ringing");
}
Thank you
Juan
