Re: [OpenSIPS-Users] WARNING - dialog module in trunk
Hello Bogdan, If the Dialog module is in inconsistent / non-functional state, Then why documentation of all the versions starting from 1.2.x, 1.3.x, 1.4.x and 1.5.x shows that the Dialog module is stable !!! I just want to use use the dialog module for the functionality check. But its not working. The dialog should be created at the initial INVITE request. And it should be stored in the DB. But its not working. Dialog is created successfully but at the same time it gets destroyed. Thanks for your attention. -Urmi On Wed, Aug 19, 2009 at 6:52 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi, The dialog module in trunk is under heavy changes and right now it is in an inconsistent / non-functional state. Please do not use it or report bug about it. The work on the module will be finished in the following days. Thanks for understanding, Bogdan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] WARNING - dialog module in trunk
Urmi,I think Bogdan was referring to the trunk code. Dialog is stable in the other releases. I use it pretty heavily on many servers. With your issue, the dialog is created and since the relay fails, the dialog is destroyed. This happens *so fast* that you simply won't catch it unless you are watching the mysql sock or have a query log running. I really think that's what's going on for you. -Brett On Thu, Aug 20, 2009 at 1:08 AM, urmi lakkad urmi.lak...@gmail.com wrote: Hello Bogdan, If the Dialog module is in inconsistent / non-functional state, Then why documentation of all the versions starting from 1.2.x, 1.3.x, 1.4.x and 1.5.x shows that the Dialog module is stable !!! I just want to use use the dialog module for the functionality check. But its not working. The dialog should be created at the initial INVITE request. And it should be stored in the DB. But its not working. Dialog is created successfully but at the same time it gets destroyed. Thanks for your attention. -Urmi On Wed, Aug 19, 2009 at 6:52 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi, The dialog module in trunk is under heavy changes and right now it is in an inconsistent / non-functional state. Please do not use it or report bug about it. The work on the module will be finished in the following days. Thanks for understanding, Bogdan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Dialog information tracing in opensips Issue
Hello Brett, Thank you for support. I am firing a SIPp call of more than 20 seconds and when the call is going on, I m checking in the DB also after call completion I am checking in the DB. But the Dialog entry is not there in dialog table. Am I doing right or not ? If not, please tell me the correct way. One more thing, Is my configuration is correct or not ?? Thank you very much. -Urmi On Thu, Aug 20, 2009 at 6:24 PM, Brett Nemeroff br...@nemeroff.com wrote: Have you created a test scenario that keeps the dialog open for a while? Then checked? On Thu, Aug 20, 2009 at 12:37 AM, urmi lakkad urmi.lak...@gmail.comwrote: Hello Brett, Thank you very much for your attention. I have continuously checked my DB. but Dialog is *not* added to the DB. In log it shows the dialog is created successfully, and then it is destroyed as well. Dont know y its behaving like this !! Once again Thank you for your support. -Urmi On Wed, Aug 19, 2009 at 7:38 PM, Brett Nemeroff br...@nemeroff.comwrote: Is it possible that your dialog is being destroyed so quickly you don't see that it was added to the db and then subsequently removed? Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog 0x2d55af90 failed (negative reply) On Wed, Aug 19, 2009 at 8:40 AM, urmi lakkad urmi.lak...@gmail.comwrote: Hello, I am using OpenSIPs-1.5.1 version. I am trying to implement the dialog module of it. When I am firing the call, its dialog should be created, and it should be stored in the database. Right !! But my problem is that calls are working fine. The dialog is created successfully but its not strong the dialog informations into the database. I have done the following configuration in opensips.cfg file. Here I have also attached opensips log. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Dialog information tracing in opensips Issue
2009/8/20 urmi lakkad urmi.lak...@gmail.com: Am I doing right or not ? If not, please tell me the correct way. One more thing, Is my configuration is correct or not ?? It looks like your call doesn't even get accepted: Aug 19 17:46:27 [6060] DBG:dialog:next_state_dlg: dialog 0x2d55af90 changed from state 1 to state 5, due event 1 Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog 0x2d55af90 failed (negative reply) Maybe you require authentication, or something else? Just take care of the call not failing first. So far it's rejected before an OK answer (state 1 is after sending an INVITE, state 5 is deleted - more or less). Capture the traffic and see what's going on. -- Kind regards, Stanisław Pitucha, Gradwell Voip Engineer T: 01225 800 831 | F: 01225 800 801 | E: s...@gradwell.net | www.gradwell.com Gradwell – Internet for Business People Phone Services | Business Broadband | Email Website Hosting Can switching to VoIP today put some change in your pocket? Registered Address: 26 Cheltenham Street, Bath, BA2 3EX, UK. Company Number: 3673235 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Dialog information tracing in opensips Issue
Hello Stanisław Pitucha, Thank you for support. No, My call is established perfectly and is running for the specified duration without fail. I m firing the call using SIPp. Also, the dialog state gives me 1. Thanks for ur attention. -Urmi 2009/8/20 Stanisław Pitucha virap...@gmail.com 2009/8/20 urmi lakkad urmi.lak...@gmail.com: Am I doing right or not ? If not, please tell me the correct way. One more thing, Is my configuration is correct or not ?? It looks like your call doesn't even get accepted: Aug 19 17:46:27 [6060] DBG:dialog:next_state_dlg: dialog 0x2d55af90 changed from state 1 to state 5, due event 1 Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog 0x2d55af90 failed (negative reply) Maybe you require authentication, or something else? Just take care of the call not failing first. So far it's rejected before an OK answer (state 1 is after sending an INVITE, state 5 is deleted - more or less). Capture the traffic and see what's going on. -- Kind regards, Stanisław Pitucha, Gradwell Voip Engineer T: 01225 800 831 | F: 01225 800 801 | E: s...@gradwell.net | www.gradwell.com Gradwell – Internet for Business People Phone Services | Business Broadband | Email Website Hosting Can switching to VoIP today put some change in your pocket? Registered Address: 26 Cheltenham Street, Bath, BA2 3EX, UK. Company Number: 3673235 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Dialog information tracing in opensips Issue
Urmi,You log shows the call having failed. I'm not sure why you think it runs for the proper duration. But as far as OpenSIPs is concerned, the call failed. It's likely a problem in your sipp scenario. It's very possible that sipp thinks the call is up, but the proxy does not. In any case, OpenSIPs is behaving as expected, the call fails, the dialog is destroyed. Aug 19 17:46:27 [6060] DBG:dialog:next_state_dlg: dialog 0x2d55af90 changed from state 1 to state 5, due event 1 Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog 0x2d55af90 failed (negative reply) BTW, a negative reply is =400 (or may also include = 300, can't remember). Check your traces, see where that comes from. -Brett On Thu, Aug 20, 2009 at 9:02 AM, urmi lakkad urmi.lak...@gmail.com wrote: Hello Stanisław Pitucha, Thank you for support. No, My call is established perfectly and is running for the specified duration without fail. I m firing the call using SIPp. Also, the dialog state gives me 1. Thanks for ur attention. -Urmi 2009/8/20 Stanisław Pitucha virap...@gmail.com 2009/8/20 urmi lakkad urmi.lak...@gmail.com: Am I doing right or not ? If not, please tell me the correct way. One more thing, Is my configuration is correct or not ?? It looks like your call doesn't even get accepted: Aug 19 17:46:27 [6060] DBG:dialog:next_state_dlg: dialog 0x2d55af90 changed from state 1 to state 5, due event 1 Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog 0x2d55af90 failed (negative reply) Maybe you require authentication, or something else? Just take care of the call not failing first. So far it's rejected before an OK answer (state 1 is after sending an INVITE, state 5 is deleted - more or less). Capture the traffic and see what's going on. -- Kind regards, Stanisław Pitucha, Gradwell Voip Engineer T: 01225 800 831 | F: 01225 800 801 | E: s...@gradwell.net | www.gradwell.com Gradwell – Internet for Business People Phone Services | Business Broadband | Email Website Hosting Can switching to VoIP today put some change in your pocket? Registered Address: 26 Cheltenham Street, Bath, BA2 3EX, UK. Company Number: 3673235 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Kamailio Opensser Call forwarding
HAHAHA! Well said! On Thu, Aug 20, 2009 at 12:18 PM, Iñaki Baz Castillo i...@aliax.net wrote: El Jueves, 20 de Agosto de 2009, happyalways escribió: Hiii..I installed mysql5.o...and Kamailio 1.5 succesfully...Authentication is working properly. Next i'm going through blind call forwarding. I need your help in configuring. Please provide me the configuartion file for blind call forwarding. Sure, but first I need you to provide me some amount of money. Thanks. -- Iñaki Baz Castillo i...@aliax.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Kamailio Opensser Call forwarding
Iñaki Baz Castillo wrote: El Jueves, 20 de Agosto de 2009, happyalways escribió: Hiii..I installed mysql5.o...and Kamailio 1.5 succesfully...Authentication is working properly. Next i'm going through blind call forwarding. I need your help in configuring. Please provide me the configuartion file for blind call forwarding. Sure, but first I need you to provide me some amount of money. Thanks. And after that amount is provided, please provide me the name and number of your principal so I can have a conversation with him about cheating on your homework assignments. -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] UAC_REDIRECT : get_redirectsdoesnot rewritehost
Hi, I am now able to handle SIP/2.0 302 redirection corrrectly, without
setdsturi($ru) there was a problem internaly not sure wy but if I add this
Opensips behave correctly
failure_route[1] {
sip_trace();
end_media_session();
if (t_check_status(302)) {
xlog(L_NOTICE, ***FAILURE ROUTE - REDIRECT ru[$ru]**\n);
get_redirects(6:2,redirect);
xlog(L_NOTICE, ***FAILURE ROUTE - REDIRECTING TO:
[$ru]**\n);
setdsturi($ru);
t_relay();
exit;
}
}
From: users-boun...@lists.opensips.org on behalf of Julien Chavanton
Sent: Wed 19/08/2009 5:01 PM
To: OpenSIPS users mailling list; OpenSIPS users mailling list; OpenSIPS users
mailling list
Subject: Re: [OpenSIPS-Users] UAC_REDIRECT : get_redirectsdoesnot rewritehost
Hi, I found a fix to this problem, in some circumstances the contact header is
already parsed when hiting shmcontact2dset()
however it is not parsed by parse_contact(), I have set a logging entry in
there.
My fix/workaround is simply to force it to parse again :
// start modification
int* b=0;
contact_hdr-parsed =(void*)b;
// end modification
/* parse the body of contact header */
if (contact_hdr-parsed==0) {
if ( parse_contact(contact_hdr)0 ) {
LM_ERR(contact hdr parse failed\n);
ret = -1;
goto restore;
}
if (dup==0)
dup = 1;
}
Could we find out why it is parsed in a way that result in this confusion
between
Contact and VIA header, this would be better.
From: users-boun...@lists.opensips.org on behalf of Julien Chavanton
Sent: Tue 18/08/2009 12:06 AM
To: OpenSIPS users mailling list; OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] UAC_REDIRECT : get_redirectsdoesnot rewritehost
Hi Bogdan, I installed a test server with
1.5.1
1.5.2
and the SVN
I have the same behavior, where the IP adress is not taken from the contact
header but from somewhere else (maybe the VIA hdr)
The setup I have in my lab :
X-Lite(caller computer) and another X-Lite(called computer).
I set X-Lite(called computer), to forward the call to another target
new_tar...@1.1.1.1 for example.
Maybe I should test with gateway instead of registered phones.
Regards
From: users-boun...@lists.opensips.org on behalf of Bogdan-Andrei Iancu
Sent: Fri 14/08/2009 8:51 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] UAC_REDIRECT : get_redirects doesnot rewritehost
Hi Julien,
Could you please try the latest 1.5 from SVN (1.5.2) ? I remember there
were some fixes in the uac_redirect part and I want to be sure we are
not troubleshooting it again.
What is really strange is that the IP in the printed contacts is
actually the IP + port from VIA hdr :-/..
If upgrading is an issue for you, I can prepare some debuging patch to
print more stuff and to see where the change comes from.
Regards,
Bogdan
Julien Chavanton wrote:
I have activated full debug mode, sort_contacts() seems to find the
host somewhere else ?
Maybe this will help you understand what is going on
#
U 10.0.1.73:57226 - 10.0.5.10:5060
SIP/2.0 302 Moved Temporarily.
Via: SIP/2.0/UDP 10.0.5.10;branch=z9hG4bK2714.0ee86a21.0.
Via: SIP/2.0/UDP
10.0.1.73:57226;received=10.0.1.73;branch=z9hG4bK-d8754z-7048034193434959-1---d8754z-;rport=57226.
Contact: sip:new_tar...@10.0.1.1:5060.
To: 777sip:7...@osip.domain.com;tag=944d064a.
From: 777sip:7...@osip.domain.com;tag=d650bd17.
Call-ID: YmI3ZmQ0Mzk2MDE3OGY0M2FlNjVkNGVlY2EwZmI5NzM..
CSeq: 1 INVITE.
User-Agent: X-Lite release 1103d stamp 53117.
Content-Length: 0.
Aug 13 21:23:27 osip /usr/local/sbin/opensips[16925]: ***FAILURE ROUTE
- REDIRECT ru[sip:7...@10.0.1.73:57226;rinstance=c699f35c276783d3]**
Aug 13 21:23:27 osip /usr/local/sbin/opensips[16925]:
DBG:uac_redirect:get_redirect: resume branch=0
Aug 13 21:23:27 osip /usr/local/sbin/opensips[16925]:
DBG:uac_redirect:get_redirect: checking branch=0 (added=0)
Aug 13 21:23:27 osip /usr/local/sbin/opensips[16925]:
DBG:uac_redirect:get_redirect: branch=0 is a redirect (added=0)
Aug 13 21:23:27 osip /usr/local/sbin/opensips[16925]:
DBG:uac_redirect:sort_contacts: sort_contacts:
sip:new_tar...@10.0.1.73:57226 q=10
Aug 13 21:23:27 osip /usr/local/sbin/opensips[16925]:
DBG:uac_redirect:shmcontact2dset: adding contact
sip:new_tar...@10.0.1.73:57226
Aug 13 21:23:27 osip /usr/local/sbin/opensips[16925]:
DBG:core:parse_headers: flags=78
Aug 13 21:23:27 osip /usr/local/sbin/opensips[16925]: ACC: request
accounted:
timestamp=1250195007;method=INVITE;from_tag=d61cb370;to_tag=;call_id=NDA2ZmI4YmNmMjJmMGNjYjI4YjUxODUyNTNkZmUyNzQ.;code=;reason=redirect
Aug 13 21:23:27 osip
[OpenSIPS-Users] Module Path and function loose_route
Hello,
I'm testing with the path module.
I'm using this scheme:
User1 - REGISTER - opensips - REGISTER - registrar.
Then, when a call comes into the registrar to User1 I'm trying to send
the INVITE to User1 via opensips.
In my cfg I have added a test to see if an out-of-dialog request
contains a valid route set:
#if there is a pre-loaded route set, then it is a request to
be sent to a user
if(loose_route()) {
...relay the request
}
However, loose_route() is returning false.
This is in accordance to
http://www.opensips.org/html/docs/modules/1.5.x/rr.html where we can
read:
There is only one exception: If the request is out-of-dialog (no
to-tag) and there is only one Route: header indicating the local
proxy, then the Route: header is removed and the function returns
FALSE.
But why does it return FALSE?
I'm trying with the following INVITE:
INVITE sip:eywhywhdyhdyw...@192.168.4.6 SIP/2.0
Via: SIP/2.0/UDP
192.168.2.121:5070;rport;branch=z9hG4bKPjJ5ldIJPcArMnJ0soPgyRWi3zGyA9TYBt
Max-Forwards: 70
From: sip:p...@ldilsjeodlskekdksl.com;tag=QumeNtxM4Cb9Qj77vumy5ujr8vHoJBfS
To: sip:eywhywhdyhdyw...@ldilsjeodlskekdksl.com
Contact: sip:tes...@192.168.2.121:5070
Call-ID: Vw3Cvf6utvMuzj-K2WGOc.YYvYWfORGB
CSeq: 28223 INVITE
Route: sip:192.168.2.100:5060;lr
Allow: SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK
Supported: replaces, 100rel
Content-Type: application/sdp
Content-Length: 453
[SDP ommited]
So, how should the INVITE be sent for loose_route to succeed?
Or should I put something else in my cfg?
regards,
takeshi
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Re: [OpenSIPS-Users] Module Path and function loose_route
El Jueves, 20 de Agosto de 2009, mayamatakeshi escribió: There is only one exception: If the request is out-of-dialog (no to-tag) and there is only one Route: header indicating the local proxy, then the Route: header is removed and the function returns FALSE. But why does it return FALSE? Because if an initial request (no To-tag) has a single Route header pointing to the proxy handling it, it's useless. -- Iñaki Baz Castillo i...@aliax.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] presence_dialoginfo context
opensips, not 'kamailio'. I write it out of habit. dlublink wrote: Hello, I have three different groups of extensions on my kamailio I want to be able to separate them, so I prefixed a name to the extensions, so I have : 1. group1.101 2. group1.102 3. group2.101 4. group2.102 5. group3.102 6. group3.103. The phones from different groups can not call each other, I found a pseudo variable that I use to rewrite the destination url, so if user group1.101 dials 102 I rewrite it to group1.102. I want to do the same thing for presence_dialog info, how can I rewrite the subscribe, presence and and notify messages to append the appropriate prefix ? Thanks, David ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Creating core tables failed
Hii...I installed Mysql 5.0.37 and Kamailio 1.5.1 with tls...They successfully installed. The problem is when i'm creating the databse for openser, I gave the command kamdbctl create. the database 'openser' is created but the tables are not being created. The error occureed is [r...@localhost sbin]# kamdbctl create MySQL password for root: database engine 'mysql' loaded INFO: test server charset INFO: creating database openser ... Creating core table: standard /usr/local/lib/kamailio/kamctl/kamdbctl.mysql: line 155: ./mysql/standard-create.sql: No such file or directory ERROR: Creating core tables failed! Please help me Thanks in advance:) -- View this message in context: http://n2.nabble.com/Creating-core-tables-failed-tp3456674p3456674.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Possible problem in osipsconsole
In each create db function this if - else is exist:
if ( -d $DATA_DIR./mysql ) {
$DB_SCHEMA=$DATA_DIR./postgres;
} else {
$DB_SCHEMA=./postgres;
}
If you DONT install 'mysql' or this directory for a reason is not in
$DATA_DIR, DB_SCHEMA will fallback to current workdir without notice, which
is bad.
Suggested solution:
move if - else out from all functions that checks for the $DB_SCHEMA value,
and call ONE function to handle this logic. I.e. read and analyse the output
from DATA_DIR and return the exact directory or local workdir with an
informative warning.
e.g.
sub validate_datadir{
my $type = shift;
if( -d $DATA_DIR./$type ) {
return $DATA_DIR./$type;
}
print Warn fallback to local workdir ./$type\n;
return ./$type;
}
--
View this message in context:
http://n2.nabble.com/Possible-problem-in-osipsconsole-tp3458993p3458993.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
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[OpenSIPS-Users] SIP Trunking
Hello, I'm brand new to OpenSIPS, just going through the make process now. I need to configure OpenSIPS to act like a SBC for some SIP trunks coming off a VoIP switch. Where should I look for Documentation/Examples of a working config? Here is my scenario: OpenSIPS has two interfaces, private public. VoIP Gateway is on private LAN with no gateway configured (it can only talk to local machines, no routing) End user has an Asterisk server on a private lan behind their firewall (NAT) I need to configure OpenSIPS to listen for SIP messages on :5060 from the end user firewall. It then need to rewrite the SIP message and send it to the Gateway. The Gateway would see the messages coming from the internal IP of the OpenSIPS server. Once all of the SIP messages get processed I then need the OpenSIPS server to proxy the RTP streams (plan on using mediaproxy) between the Asterisk server and VoIP Gateway. Any helpful hints on where to look? -Matt -- Matthew S. Crocker President Crocker Communications, Inc. PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.com P: 413-746-2760 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP Trunking
Matthew, Look for the mediaproxy module. That said, do be aware that a proxy is, by definition, not like an SBC. SBCs have many other capabilities a proxy does not; a proxy is a relatively thin interoperation layer. Perhaps the recently introduced b2bua module is brought to bear on that somewhat, but classically, OpenSIPS is a proxy. -- Alex Matthew S. Crocker wrote: Hello, I'm brand new to OpenSIPS, just going through the make process now. I need to configure OpenSIPS to act like a SBC for some SIP trunks coming off a VoIP switch. Where should I look for Documentation/Examples of a working config? Here is my scenario: OpenSIPS has two interfaces, private public. VoIP Gateway is on private LAN with no gateway configured (it can only talk to local machines, no routing) End user has an Asterisk server on a private lan behind their firewall (NAT) I need to configure OpenSIPS to listen for SIP messages on :5060 from the end user firewall. It then need to rewrite the SIP message and send it to the Gateway. The Gateway would see the messages coming from the internal IP of the OpenSIPS server. Once all of the SIP messages get processed I then need the OpenSIPS server to proxy the RTP streams (plan on using mediaproxy) between the Asterisk server and VoIP Gateway. Any helpful hints on where to look? -Matt -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Module Path and function loose_route
Iñaki Baz Castillo wrote: El Jueves, 20 de Agosto de 2009, mayamatakeshi escribió: There is only one exception: If the request is out-of-dialog (no to-tag) and there is only one Route: header indicating the local proxy, then the Route: header is removed and the function returns FALSE. But why does it return FALSE? Because if an initial request (no To-tag) has a single Route header pointing to the proxy handling it, it's useless. That's correct - initial INVITEs (and all initial requests) are different than in-dialog requests (requests arising within a dialog created by the initial requests). They are routed manually, not using loose_route() in any way. -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Module Path and function loose_route
El Jueves, 20 de Agosto de 2009, Alex Balashov escribió: Iñaki Baz Castillo wrote: El Jueves, 20 de Agosto de 2009, mayamatakeshi escribió: There is only one exception: If the request is out-of-dialog (no to-tag) and there is only one Route: header indicating the local proxy, then the Route: header is removed and the function returns FALSE. But why does it return FALSE? Because if an initial request (no To-tag) has a single Route header pointing to the proxy handling it, it's useless. That's correct - initial INVITEs (and all initial requests) are different than in-dialog requests (requests arising within a dialog created by the initial requests). They are routed manually, not using loose_route() in any way. In fact, in case of PATH usage, the registrar should receive the request for a registered user, add Route header pointing to the inbound/outbound proxy of the registered user and change the RURI with the real location of the registered user (or mapped public address in case of NAT), route the request to it, and the inbound/outbound proxy should remove the Route header and route the request based on the RURI as usual. -- Iñaki Baz Castillo i...@aliax.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP Trunking
I understand that OpenSIPS is not a full blown SBC (I can't afford an ACMEPacket). Will it perform the functions to proxy the SIP RTP streams (via mediaproxy) between my end users and my internal gateway? At some point I plan on increasing the use of openSIPS to handle registration, presence, routing, etc. -Matt - Alex Balashov abalas...@evaristesys.com wrote: From: Alex Balashov abalas...@evaristesys.com To: OpenSIPS users mailling list users@lists.opensips.org Sent: Thursday, August 20, 2009 1:58:48 PM GMT -05:00 US/Canada Eastern Subject: Re: [OpenSIPS-Users] SIP Trunking Matthew, Look for the mediaproxy module. That said, do be aware that a proxy is, by definition, not like an SBC. SBCs have many other capabilities a proxy does not; a proxy is a relatively thin interoperation layer. Perhaps the recently introduced b2bua module is brought to bear on that somewhat, but classically, OpenSIPS is a proxy. -- Alex Matthew S. Crocker wrote: Hello, I'm brand new to OpenSIPS, just going through the make process now. I need to configure OpenSIPS to act like a SBC for some SIP trunks coming off a VoIP switch. Where should I look for Documentation/Examples of a working config? Here is my scenario: OpenSIPS has two interfaces, private public. VoIP Gateway is on private LAN with no gateway configured (it can only talk to local machines, no routing) End user has an Asterisk server on a private lan behind their firewall (NAT) I need to configure OpenSIPS to listen for SIP messages on :5060 from the end user firewall. It then need to rewrite the SIP message and send it to the Gateway. The Gateway would see the messages coming from the internal IP of the OpenSIPS server. Once all of the SIP messages get processed I then need the OpenSIPS server to proxy the RTP streams (plan on using mediaproxy) between the Asterisk server and VoIP Gateway. Any helpful hints on where to look? -Matt -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Matthew S. Crocker President Crocker Communications, Inc. PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.com P: 413-746-2760 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP Trunking
Matthew, While I'm no Mediaproxy expert, I have seen many conversations on this list where Mediaproxy is described as a part of a far-end NAT solution. It was not designed to have a private IP attached to it. For that, you most likely will want to look at the rtpproxy application. It sounds like you are constructing a local ALG to connect private and public networks. You don't necessarily need a full-blown Acme for that. I've had great luck with Edgewater Networks' Edgemarc devices, for example. That's just one. There are many. - Jeff On 8/20/09 2:49 PM, Matthew S. Crocker matt...@corp.crocker.com wrote: I understand that OpenSIPS is not a full blown SBC (I can't afford an ACMEPacket). Will it perform the functions to proxy the SIP RTP streams (via mediaproxy) between my end users and my internal gateway? At some point I plan on increasing the use of openSIPS to handle registration, presence, routing, etc. -Matt - Alex Balashov abalas...@evaristesys.com wrote: From: Alex Balashov abalas...@evaristesys.com To: OpenSIPS users mailling list users@lists.opensips.org Sent: Thursday, August 20, 2009 1:58:48 PM GMT -05:00 US/Canada Eastern Subject: Re: [OpenSIPS-Users] SIP Trunking Matthew, Look for the mediaproxy module. That said, do be aware that a proxy is, by definition, not like an SBC. SBCs have many other capabilities a proxy does not; a proxy is a relatively thin interoperation layer. Perhaps the recently introduced b2bua module is brought to bear on that somewhat, but classically, OpenSIPS is a proxy. -- Alex Matthew S. Crocker wrote: Hello, I'm brand new to OpenSIPS, just going through the make process now. I need to configure OpenSIPS to act like a SBC for some SIP trunks coming off a VoIP switch. Where should I look for Documentation/Examples of a working config? Here is my scenario: OpenSIPS has two interfaces, private public. VoIP Gateway is on private LAN with no gateway configured (it can only talk to local machines, no routing) End user has an Asterisk server on a private lan behind their firewall (NAT) I need to configure OpenSIPS to listen for SIP messages on :5060 from the end user firewall. It then need to rewrite the SIP message and send it to the Gateway. The Gateway would see the messages coming from the internal IP of the OpenSIPS server. Once all of the SIP messages get processed I then need the OpenSIPS server to proxy the RTP streams (plan on using mediaproxy) between the Asterisk server and VoIP Gateway. Any helpful hints on where to look? -Matt -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP Trunking
Media Proxy will work with public internet addresses but, and this is just my understanding, was not built for use with private IPs let alone bridging from one subnet to another. If your keeping the gateways on a private network is for security purposes, you may consider giving them public ips on the same subnet as your opensips and mediaproxy setup, but not specifying a default gateway. Essentially, this would allow the media proxy to do its job relaying the audio, while still preventing 99% of any unwanted traffic to your gateways. Couple that will firehol or some other cool iptables app (or manually configure it if you like) and you'd be sitting pretty secure I would think. Really depends on what you've designed (and why). - Darren On Aug 20, 2009, at 12:49 PM, Matthew S. Crocker wrote: I understand that OpenSIPS is not a full blown SBC (I can't afford an ACMEPacket). Will it perform the functions to proxy the SIP RTP streams (via mediaproxy) between my end users and my internal gateway? At some point I plan on increasing the use of openSIPS to handle registration, presence, routing, etc. -Matt - Alex Balashov abalas...@evaristesys.com wrote: From: Alex Balashov abalas...@evaristesys.com To: OpenSIPS users mailling list users@lists.opensips.org Sent: Thursday, August 20, 2009 1:58:48 PM GMT -05:00 US/Canada Eastern Subject: Re: [OpenSIPS-Users] SIP Trunking Matthew, Look for the mediaproxy module. That said, do be aware that a proxy is, by definition, not like an SBC. SBCs have many other capabilities a proxy does not; a proxy is a relatively thin interoperation layer. Perhaps the recently introduced b2bua module is brought to bear on that somewhat, but classically, OpenSIPS is a proxy. -- Alex Matthew S. Crocker wrote: Hello, I'm brand new to OpenSIPS, just going through the make process now. I need to configure OpenSIPS to act like a SBC for some SIP trunks coming off a VoIP switch. Where should I look for Documentation/Examples of a working config? Here is my scenario: OpenSIPS has two interfaces, private public. VoIP Gateway is on private LAN with no gateway configured (it can only talk to local machines, no routing) End user has an Asterisk server on a private lan behind their firewall (NAT) I need to configure OpenSIPS to listen for SIP messages on :5060 from the end user firewall. It then need to rewrite the SIP message and send it to the Gateway. The Gateway would see the messages coming from the internal IP of the OpenSIPS server. Once all of the SIP messages get processed I then need the OpenSIPS server to proxy the RTP streams (plan on using mediaproxy) between the Asterisk server and VoIP Gateway. Any helpful hints on where to look? -Matt -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Matthew S. Crocker President Crocker Communications, Inc. PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.com P: 413-746-2760 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP Trunking
It's possible also to use RTPproxy, because it's designed to work in bridging mode where it can forward the traffic from the internal network towards external one. Regards On Thu, 2009-08-20 at 12:55 -0600, Darren Sessions wrote: Media Proxy will work with public internet addresses but, and this is just my understanding, was not built for use with private IPs let alone bridging from one subnet to another. If your keeping the gateways on a private network is for security purposes, you may consider giving them public ips on the same subnet as your opensips and mediaproxy setup, but not specifying a default gateway. Essentially, this would allow the media proxy to do its job relaying the audio, while still preventing 99% of any unwanted traffic to your gateways. Couple that will firehol or some other cool iptables app (or manually configure it if you like) and you'd be sitting pretty secure I would think. Really depends on what you've designed (and why). - Darren On Aug 20, 2009, at 12:49 PM, Matthew S. Crocker wrote: I understand that OpenSIPS is not a full blown SBC (I can't afford an ACMEPacket). Will it perform the functions to proxy the SIP RTP streams (via mediaproxy) between my end users and my internal gateway? At some point I plan on increasing the use of openSIPS to handle registration, presence, routing, etc. -Matt - Alex Balashov abalas...@evaristesys.com wrote: From: Alex Balashov abalas...@evaristesys.com To: OpenSIPS users mailling list users@lists.opensips.org Sent: Thursday, August 20, 2009 1:58:48 PM GMT -05:00 US/Canada Eastern Subject: Re: [OpenSIPS-Users] SIP Trunking Matthew, Look for the mediaproxy module. That said, do be aware that a proxy is, by definition, not like an SBC. SBCs have many other capabilities a proxy does not; a proxy is a relatively thin interoperation layer. Perhaps the recently introduced b2bua module is brought to bear on that somewhat, but classically, OpenSIPS is a proxy. -- Alex Matthew S. Crocker wrote: Hello, I'm brand new to OpenSIPS, just going through the make process now. I need to configure OpenSIPS to act like a SBC for some SIP trunks coming off a VoIP switch. Where should I look for Documentation/Examples of a working config? Here is my scenario: OpenSIPS has two interfaces, private public. VoIP Gateway is on private LAN with no gateway configured (it can only talk to local machines, no routing) End user has an Asterisk server on a private lan behind their firewall (NAT) I need to configure OpenSIPS to listen for SIP messages on :5060 from the end user firewall. It then need to rewrite the SIP message and send it to the Gateway. The Gateway would see the messages coming from the internal IP of the OpenSIPS server. Once all of the SIP messages get processed I then need the OpenSIPS server to proxy the RTP streams (plan on using mediaproxy) between the Asterisk server and VoIP Gateway. Any helpful hints on where to look? -Matt -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Matthew S. Crocker President Crocker Communications, Inc. PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.com P: 413-746-2760 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP Trunking
Can mediaproxy glue two RTP streams together (CallerA to CallerB)? Can mediaproxy glue two RTP streams together from different interfaces/IPs (eth0 eth1) ? If so then it should be able to glue two calls together between public IP (eth0) and private IP (eth1). If the two RTP streams have to be on the same interface for mediaproxy to work then I would expect to run into issues. EndUser - (eth0) MediaProxy (eth1) - SIP Gateway - Jeff Pyle jp...@fidelityvoice.com wrote: From: Jeff Pyle jp...@fidelityvoice.com To: OpenSIPS users mailling list users@lists.opensips.org Sent: Thursday, August 20, 2009 2:52:35 PM GMT -05:00 US/Canada Eastern Subject: Re: [OpenSIPS-Users] SIP Trunking Matthew, While I'm no Mediaproxy expert, I have seen many conversations on this list where Mediaproxy is described as a part of a far-end NAT solution. It was not designed to have a private IP attached to it. For that, you most likely will want to look at the rtpproxy application. It sounds like you are constructing a local ALG to connect private and public networks. You don't necessarily need a full-blown Acme for that. I've had great luck with Edgewater Networks' Edgemarc devices, for example. That's just one. There are many. - Jeff On 8/20/09 2:49 PM, Matthew S. Crocker matt...@corp.crocker.com wrote: I understand that OpenSIPS is not a full blown SBC (I can't afford an ACMEPacket). Will it perform the functions to proxy the SIP RTP streams (via mediaproxy) between my end users and my internal gateway? At some point I plan on increasing the use of openSIPS to handle registration, presence, routing, etc. -Matt - Alex Balashov abalas...@evaristesys.com wrote: From: Alex Balashov abalas...@evaristesys.com To: OpenSIPS users mailling list users@lists.opensips.org Sent: Thursday, August 20, 2009 1:58:48 PM GMT -05:00 US/Canada Eastern Subject: Re: [OpenSIPS-Users] SIP Trunking Matthew, Look for the mediaproxy module. That said, do be aware that a proxy is, by definition, not like an SBC. SBCs have many other capabilities a proxy does not; a proxy is a relatively thin interoperation layer. Perhaps the recently introduced b2bua module is brought to bear on that somewhat, but classically, OpenSIPS is a proxy. -- Alex Matthew S. Crocker wrote: Hello, I'm brand new to OpenSIPS, just going through the make process now. I need to configure OpenSIPS to act like a SBC for some SIP trunks coming off a VoIP switch. Where should I look for Documentation/Examples of a working config? Here is my scenario: OpenSIPS has two interfaces, private public. VoIP Gateway is on private LAN with no gateway configured (it can only talk to local machines, no routing) End user has an Asterisk server on a private lan behind their firewall (NAT) I need to configure OpenSIPS to listen for SIP messages on :5060 from the end user firewall. It then need to rewrite the SIP message and send it to the Gateway. The Gateway would see the messages coming from the internal IP of the OpenSIPS server. Once all of the SIP messages get processed I then need the OpenSIPS server to proxy the RTP streams (plan on using mediaproxy) between the Asterisk server and VoIP Gateway. Any helpful hints on where to look? -Matt -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Matthew S. Crocker President Crocker Communications, Inc. PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.com P: 413-746-2760 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP Trunking
Matthew S. Crocker wrote: Can mediaproxy glue two RTP streams together (CallerA to CallerB)? Can mediaproxy glue two RTP streams together from different interfaces/IPs (eth0 eth1) ? Yes. -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Using LDAP for Authorization
Hi, I am currently trying to use LDAP authorization for my opensips server. I have setup an LDAP server using openldap in the same machine. I created an ldap config file and set the ldap_server_url = ip-address of teh machine:389 Now when I try to authorize using LDAP. I get the following error message Aug 20 12:48:17 localhost /usr/local/sbin/opensips[4229]: ERROR:ldap:ldap_connect: [sipaccounts]: ldap_initialize (10.100.104.28:389) failed: Bad parameter to an ldap routine Aug 20 12:48:17 localhost /usr/local/sbin/opensips[4229]: ERROR:ldap:child_init: [sipaccounts]: failed to connect to LDAP host(s) Aug 20 12:48:17 localhost /usr/local/sbin/opensips[4229]: ERROR:core:init_mod_child: failed to initializing module ldap, rank 2 followed by more error messages similar to this. Could anyone guide me on what I could have done wrong? This is my ldap.cfg file [sipaccounts] ldap_server_url = 127.0.0.1:389 ldap_bind_dn = cn=root, dc=example, dc=com ldap_bind_password = root123 ldap_network_timeout = 500 ldap_client_bind_timeout = 500 Thanks, Harish ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP Trunking
Hello, I tried to use mediaproxy, it includes two softwares (dispatcher relay), I tried a lot to run more than one relay on the same server in order to bind them to different interfaces. But unfortunately this didn't work and I think it's not possible. I recommend using RTPProxy which is designed to work in bridging mode between two networks and you can run multiple instance of RTPProxy on the same server. Regards. On Thu, 2009-08-20 at 15:48 -0400, Matthew S. Crocker wrote: Can mediaproxy glue two RTP streams together (CallerA to CallerB)? Can mediaproxy glue two RTP streams together from different interfaces/IPs (eth0 eth1) ? If so then it should be able to glue two calls together between public IP (eth0) and private IP (eth1). If the two RTP streams have to be on the same interface for mediaproxy to work then I would expect to run into issues. EndUser - (eth0) MediaProxy (eth1) - SIP Gateway - Jeff Pyle jp...@fidelityvoice.com wrote: From: Jeff Pyle jp...@fidelityvoice.com To: OpenSIPS users mailling list users@lists.opensips.org Sent: Thursday, August 20, 2009 2:52:35 PM GMT -05:00 US/Canada Eastern Subject: Re: [OpenSIPS-Users] SIP Trunking Matthew, While I'm no Mediaproxy expert, I have seen many conversations on this list where Mediaproxy is described as a part of a far-end NAT solution. It was not designed to have a private IP attached to it. For that, you most likely will want to look at the rtpproxy application. It sounds like you are constructing a local ALG to connect private and public networks. You don't necessarily need a full-blown Acme for that. I've had great luck with Edgewater Networks' Edgemarc devices, for example. That's just one. There are many. - Jeff On 8/20/09 2:49 PM, Matthew S. Crocker matt...@corp.crocker.com wrote: I understand that OpenSIPS is not a full blown SBC (I can't afford an ACMEPacket). Will it perform the functions to proxy the SIP RTP streams (via mediaproxy) between my end users and my internal gateway? At some point I plan on increasing the use of openSIPS to handle registration, presence, routing, etc. -Matt - Alex Balashov abalas...@evaristesys.com wrote: From: Alex Balashov abalas...@evaristesys.com To: OpenSIPS users mailling list users@lists.opensips.org Sent: Thursday, August 20, 2009 1:58:48 PM GMT -05:00 US/Canada Eastern Subject: Re: [OpenSIPS-Users] SIP Trunking Matthew, Look for the mediaproxy module. That said, do be aware that a proxy is, by definition, not like an SBC. SBCs have many other capabilities a proxy does not; a proxy is a relatively thin interoperation layer. Perhaps the recently introduced b2bua module is brought to bear on that somewhat, but classically, OpenSIPS is a proxy. -- Alex Matthew S. Crocker wrote: Hello, I'm brand new to OpenSIPS, just going through the make process now. I need to configure OpenSIPS to act like a SBC for some SIP trunks coming off a VoIP switch. Where should I look for Documentation/Examples of a working config? Here is my scenario: OpenSIPS has two interfaces, private public. VoIP Gateway is on private LAN with no gateway configured (it can only talk to local machines, no routing) End user has an Asterisk server on a private lan behind their firewall (NAT) I need to configure OpenSIPS to listen for SIP messages on :5060 from the end user firewall. It then need to rewrite the SIP message and send it to the Gateway. The Gateway would see the messages coming from the internal IP of the OpenSIPS server. Once all of the SIP messages get processed I then need the OpenSIPS server to proxy the RTP streams (plan on using mediaproxy) between the Asterisk server and VoIP Gateway. Any helpful hints on where to look? -Matt -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Module Path and function loose_route
On Fri, Aug 21, 2009 at 3:43 AM, Iñaki Baz Castilloi...@aliax.net wrote: El Jueves, 20 de Agosto de 2009, Alex Balashov escribió: Iñaki Baz Castillo wrote: El Jueves, 20 de Agosto de 2009, mayamatakeshi escribió: There is only one exception: If the request is out-of-dialog (no to-tag) and there is only one Route: header indicating the local proxy, then the Route: header is removed and the function returns FALSE. But why does it return FALSE? Because if an initial request (no To-tag) has a single Route header pointing to the proxy handling it, it's useless. That's correct - initial INVITEs (and all initial requests) are different than in-dialog requests (requests arising within a dialog created by the initial requests). They are routed manually, not using loose_route() in any way. In fact, in case of PATH usage, the registrar should receive the request for a registered user, add Route header pointing to the inbound/outbound proxy of the registered user and change the RURI with the real location of the registered user (or mapped public address in case of NAT), route the request to it, Iñaki, Alex, thanks. But I'm still confused about this. The above is what I believe I'm doing in my tests: I am setting the RURI to the real location of the registered user and I'm passing the URI of opensips in the sole Route header. and the inbound/outbound proxy should remove the Route header and route the request based on the RURI as usual. How my cfg should accomplish this then? If loose_route returned TRUE, I would simply relay the request to the address in the RURI. Do you mean I should not be using loose_route() for this? Should I perform the checking in another way? Well, of course I could come up with something else, but it seems to me loose_route() would be the simplest way to do it if there was not that exception clause that I really don't understand why is there. regards, takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
