Re: [OpenSIPS-Users] Transfer issue
Hi Jeff,
Jeff Kronlage wrote:
Anca,
Thanks for the quick reply. I tried as you suggested, on a development box,
and while it didn't work, it did look a lot more like what I was expecting to
see.
However, I guess I am looking for more of a tack on solution for what I've
already developed. The right answer may be to not support REFERs. Is
there any way to modify the script scenario to -just- pick up from the REFER
and still bridge in the first call leg? It's OK if the answer is 'no', I
just need to know what my options are.
I am sorry, but the answer is really no. The B2BUA must be in the middle
of the call from the beginning.
Regards,
Anca
Thanks,
Jeff K
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Anca Vamanu
Sent: Monday, November 09, 2009 7:17 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Transfer issue
Hi Jeff,
Jeff Kronlage wrote:
Anca,
The key pieces of my config file are:
Loadmodule tm.so
loadmodule b2b_entities.so
loadmodule b2b_logic.so
modparam(tm, pass_provisional_replies, 1)
modparam(b2b_entities,server_address,sip:opens...@myproxyabc.com)
modparam(b2b_logic, script_scenario,
/usr/local/etc/opensips/refer_script.xml)
modparam(b2b_entities, script_req_route, b2b_request)
modparam(b2b_entities, script_reply_route, b2b_reply)
...and down in route[1], just prior to where I would normally call t_relay():
if (is_method(REFER)) {
b2b_init_request(refer);
exit;
}
No, no, no, you should no call b2b_init_request for REFER but for the
initial INVITE - since the B2BUA must but itself in the middle of the
call from the beginning.
The contents of /usr/local/etc/opensips/refer_script.xml:
?xml version=1.0?
scenario id=refer name=Handle refer at server param=0 type=script
init
bridge
server
idserver1/id
/server
client
idclient1/id
typemessage/type
destination
value type=initialserver1/value
/destination
/client
/bridge
/init
rules
request
refer
rule id=1
action
send_reply
code202/code
reasonAccepted/reason
/send_reply
end_dialog_leg/
bridge
client
peer/
/client
client
idclient2/id
destination
value type=headerRefer-To/value
/destination
/client
/bridge
/action
/rule
/refer
/request
/rules
/scenario
So I believe I've done everything you've suggested? The only thing that was
a little strange is that when I compiled from the svn, I had to edit
/parser/parse_fline.h. I had found items such as INVITE_LEN in there, and
my compiler complained that REFER_LEN, as well as several other variables,
were undefined. I modified this section of parse_fline.h to the following:
#define INVITE INVITE
#define CANCEL CANCEL
#define ACK ACK
#define BYE BYE
#define INFOINFO
#define PRACK PRACK
#define REFER REFER
#define SUBSCRIBE SUBSCRIBE
#define NOTIFY NOTIFY
#define MESSAGE MESSAGE
#define INVITE_LEN 6
#define CANCEL_LEN 6
#define ACK_LEN 3
#define BYE_LEN 3
#define INFO_LEN 4
#define PRACK_LEN 5
#define REFER_LEN 5
#define SUBSCRIBE_LEN 9
#define NOTIFY_LEN 6
#define MESSAGE_LEN 7
I forgot to commit the parse_fline.h file on friday. I have commited it
today. You can delete yours and update from svn.
Regards,
Anca
Please advise,
Jeff
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Anca Vamanu
Sent: Monday, November 09, 2009 2:34 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Transfer issue
Hi Jeff,
It seems that the b2b module just does simple forward of REFER request
in your setup.
Have you loaded the refer scenario? You can find it here:
http://www.opensips.org/Resources/B2buaTutorial#toc15. You have to put
in in a file and then set the 'script_scenario' parameter to the path of
the file:
modparam(b2b_logic, script_scenario, path_to_scenario_refer.xml)
The you have to call the b2b_init_request function with the refer
parameter:
b2b_init_request(refer);
Regards,
Anca
Jeff Kronlage wrote:
Anca,
Thanks again for your work on this. I've gotten the b2b modules working
and I'm attempting to use the REFER scenario, but I'm having some confusion
regarding how a REFER with a B2BUA should be handled.
My test environment looks like this:
UA1 (softphone) ---INVITE-- [Opensips] ---PSTN_GATEWAY(UA2)-- POTS Phone
. (Session progress/OK/etc)
UA1
Re: [OpenSIPS-Users] IMC Commands
Indiver, X-Lite might sent those commands not in plain text, but HTML embedded. The _very first_ character in the body of the message must be a sharp (i.e., #). As HTML starts , it is not considered as a command. Try using sipsak or some client where you can force plain text to create the channel. Hope that helps. Sebastian -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Indiver Sent: Tuesday, 10. November 2009 07:52 To: users@lists.opensips.org Subject: [OpenSIPS-Users] IMC Commands [...] But the problem is there is no chat room created and none of the following imc commands are working from x-lite. But i can succesfully chat between to two sip xlite phones. #create chat-000 private #join sip:chat-...@opensips.org ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] IMC Commands
Hi Sebastian, Thanks for quick reply. As per your suggestion i tried from sipsak the #create chat-000 private command. But it is taking it as '' only it is not recognizing #. Is the above code enough for creating chat rooms. If so from where i can send the join,create chat commands.Thanks in advance. Schumann Sebastian wrote: Indiver, X-Lite might sent those commands not in plain text, but HTML embedded. The _very first_ character in the body of the message must be a sharp (i.e., #). As HTML starts , it is not considered as a command. Try using sipsak or some client where you can force plain text to create the channel. Hope that helps. Sebastian -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Indiver Sent: Tuesday, 10. November 2009 07:52 To: users@lists.opensips.org Subject: [OpenSIPS-Users] IMC Commands [...] But the problem is there is no chat room created and none of the following imc commands are working from x-lite. But i can succesfully chat between to two sip xlite phones. #create chat-000 private #join sip:chat-...@opensips.org ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- View this message in context: http://n2.nabble.com/IMC-Commands-tp3978084p3979283.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
Re: [OpenSIPS-Users] pidf-manipulation - Openxcap spits error when saving pidf.
There is really one one param that is needed to be configured in opensiops and openxcap. And I did it correctly on both. That is the xmlrpc port. After I restart openxcap and opensips I see no error on oenxcap now. 63.148.166.3 'PUT /xcap-root/pidf-manipulation/users/ msivara...@smithmicro.com/pidf.xml HTTP/1.1' 401 0 141 'MultiPlatform Networking API - WinInet v1.0' - Starting factory twisted.web.xmlrpc._QueryFactory instance at 0x8b5924c 63.148.166.3 'PUT /xcap-root/pidf-manipulation/users/ msivara...@smithmicro.com/pidf.xml HTTP/1.1' 200 0 - 'MultiPlatform Networking API - WinInet v1.0' 5f75f52ab58d49c8319137f03fd5d6f1 Stopping factory twisted.web.xmlrpc._QueryFactory instance at 0x8b5924c I get the logs like above, which means that openxcap is able to save the pidf in the mysql database correctly. OPenxcap sends the Notify pidf to the watcher with only offline status. It is not sending the full pidf with all the tuple and person info. It sends a second Notify with no pidf payload and with subscription status active and expires value = 1222. Note that I'm saving the pidf via xcap after the UnRegisteration is completed and that is infact saved in the mysql database correctly. Because If I point my IE and get the pidf I can see the saved pidf correctly. Please let me know what else I need to configure in opensips to make pidf-manipulation working. On Mon, Nov 9, 2009 at 6:46 PM, Iñaki Baz Castillo i...@aliax.net wrote: 2009/11/9 mani sivaraman mani.opens...@gmail.com: I'm able to save the pidf.xml to Openxcap server and it sends 200 OK for the PUT pidf. But pidf-manipulation never worked for me, so I ran openxcap in no-fork mode and got the follwoing error spit from openxcap when I try to PUT pidf to it. Coudl any one please point out what is wrong ? in openxcap or my pidf ? I think you have not configured properly mi_xmlrpc in opensips so OpenXCAP cannot contact it. -- 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
[OpenSIPS-Users] static nat and 1 to 1 nat
I'm curious... Do static or 1 to 1 natted locations produce a null resort in the received column of the location table? Some natted locations show received as the outside ip and contact with the inside ip which is usually an internal ip behind a firewall. Some others show null as the received and a the contact has a public address followed by a port. These devices are obviously natted because I see many of the same ip with different ports. Is the difference b/w these 2 types a static nat port or is it something else? Thanks! ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] BYE does not work / Opensips stops working
Hello, I am installing the new 1.6 and getting a strange behavior. When the call ended on one side the other side does not get disconnected, and the log shows: Nov 10 18:40:38 en4 /sbin/opensips[20610]: WARNING:dialog:dlg_onroute: unable to find dialog for BYE with route param 'a94.cd4c751' Nov 10 18:40:39 en4 /sbin/opensips[20609]: WARNING:dialog:dlg_onroute: unable to find dialog for BYE with route param 'a94.cd4c751' Nov 10 18:40:41 en4 /sbin/opensips[20611]: WARNING:dialog:dlg_onroute: unable to find dialog for BYE with route param 'a94.cd4c751' Nov 10 18:40:45 en4 /sbin/opensips[20609]: WARNING:dialog:dlg_onroute: unable to find dialog for BYE with route param 'a94.cd4c751' I did a ngrep and the SIP Route: shows the correct param. Eventually the Opensips starts not to accept INVITEs or creates a very long setup time for INVITEs (20-40s). I think I am doing something wrong, could someone please check my config below. Thanks, Alberto ### Global Parameters # debug=3 log_stderror=no log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = /usr/local/etc/opensips/tls/user/user-cert.pem #tls_private_key = /usr/local/etc/opensips/tls/user/user-privkey.pem #tls_ca_list = /usr/local/etc/opensips/tls/user/user-calist.pem port=5060 /* uncomment and configure the following line if you want opensips to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:192.168.1.2:5060 ### Modules Section #set module path mpath=//lib/opensips/modules/ /* uncomment next line for MySQL DB support */ loadmodule db_mysql.so loadmodule signaling.so loadmodule sl.so loadmodule tm.so loadmodule rr.so loadmodule maxfwd.so loadmodule usrloc.so loadmodule registrar.so loadmodule textops.so loadmodule mi_fifo.so loadmodule uri.so loadmodule xlog.so loadmodule acc.so loadmodule aaa_radius.so loadmodule auth.so loadmodule auth_db.so #loadmodule alias_db.so #loadmodule domain.so #loadmodule presence.so #loadmodule presence_xml.so loadmodule dialog.so # # - nat_traversal #** loadmodule nat_traversal.so #** modparam(nat_traversal, keepalive_interval, 90) modparam(nat_traversal, keepalive_method, OPTIONS) modparam(nat_traversal, keepalive_from, sip:keepal...@mydomain.com) # # -- nathelper #** loadmodule nathelper.so #** modparam(nathelper, natping_interval, 60) modparam(nathelper, ping_nated_only, 1) modparam(nathelper, sipping_bflag, 7) modparam(nathelper, received_avp, $avp(i:801)) modparam(nathelper, sipping_from, sip:pin...@mydomain.com) # #** loadmodule mediaproxy.so #** #--- Default Values modparam(mediaproxy, mediaproxy_socket, /var/run/mediaproxy/dispatcher.sock) modparam(mediaproxy, mediaproxy_timeout, 500) modparam(mediaproxy, signaling_ip_avp, $avp(s:nat_ip)) modparam(mediaproxy, media_relay_avp, $avp(s:media_relay)) loadmodule drouting.so loadmodule permissions.so # - permissions params - modparam(permissions, db_url, mysql://opensips:opensip...@127.0.0.1/opensips) loadmodule call_control.so # - setting module-specific parameters --- # - mi_fifo params - modparam(mi_fifo, fifo_name, /tmp/opensips_fifo) # - rr params - # add value to ;lr param to cope with most of the UAs modparam(rr, enable_full_lr, 1) # do not append from tag to the RR (no need for this script) modparam(rr, append_fromtag, 0) # - registrar params - /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam(registrar, max_contacts, 10) # - usrloc params - #modparam(usrloc, db_mode, 0) /* uncomment the following lines if you want to enable DB persistency for location entries */ modparam(usrloc, db_mode, 2) modparam(usrloc, db_url, mysql://opensips:opensip...@localhost/opensips) modparam(usrloc, nat_bflag, 6) # - uri params - modparam(uri, use_uri_table, 0) # - acc params - /* what sepcial events should be accounted ? */ #modparam(acc, early_media, 1) #modparam(acc, report_ack, 1) #modparam(acc, report_cancels, 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable
[OpenSIPS-Users] how to test if xmlrpc is working bet openxcap and opensips
I want to know if xmlrpc is working between openxcap and opensips. OPenxcap accepts all the data I'm trying to save from the client and sends 200 OK. OPensips is listening to the correct xmlrpc port that I configured on opensips.cfg. On opensips.cfg I have : loadmodule mi_xmlrpc.so modparam(mi_xmlrpc, log_file, /var/log/openser-xmlrpc.log) modparam(mi_xmlrpc, port, 9997) And on openxcap config.ini I have the following: [OpenSIPS] ; The address and port of the xml-rpc management interface xmlrpc_url = http://sips01.mydomain.com:9997 I also see that on /var/log/openser-xmlrpc.log the pidf POST 200 336 statement is logged when I PUT pidf.xml to openxcap. Whch all indicates that xmlrpc is working. But I don;t see any xmlrpc debug logs on openspis console when pidf is posted. Please tell me what to look for and how to make sure that openxcap is fact communicating with openspis. Thakns in Advance. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] pidf-manipulation - Openxcap spits error when saving pidf.
2009/11/10 mani sivaraman mani.opens...@gmail.com: OPenxcap sends the Notify pidf to the watcher with only offline status. It is not sending the full pidf with all the tuple and person info. It sends a second Notify with no pidf payload and with subscription status active and expires value = 1222. First you should study how XCAP works as a XCAP server doesn't send a presence NOTIFY to watchers. Instead it notifies to OpenSIPS presence server (via mi_xmlrpc) about a pidf change for a specific user so OpenSIPS presence server re-read the pidf document stored in XCAP table and generates the new NOTIFY merging presentity uploaded by that user via SIP (PUBLISH) and presentity uploaded by the client via XCAP (HTTP). -- Iñaki Baz Castillo i...@aliax.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Calling from one Subnet to another using Prefix
Hi there
I got 2 Subnets with the addresses 192.168.1.0 and 172.31.10.0
Between those two OpenSips is a Linux Debian Gateway installed with IP
forwarding and 2 Network Interfaces.
OpenSips1 Gateway OpenSips2
My Goal is to config the OpenSips that i can call from Subnet 1 to
Subnet 2 using Prefixes.
I got a Sipphone at Subnet 1 (Callnumber 1000) and another at subnet 2
(1001)
My configutarion of opensips.cfg for prefix calling is:
if (uri=~sip:01[0-9]+@) {
strip(2);
rewritehost(192.168.1.2); // ip from opensips1
t_relay();
exit;
}
-- if I dial 01-1000 (01 Prefix) from agent 2 than i should get a
ring at agent 1. but it doesnt work (time out)
is this code correct? did i forget something? or is there another
possibility to do such calls?
Thanks for your help in advance.
Regards
Duc
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Re: [OpenSIPS-Users] Calling from one Subnet to another using Prefix
So this might be a stupid question, but is OpenSIPS1 able to ping
OpenSIPS2??? I am wondering if you Network Gateway/Router is the issue. If
you can ping then what do you see on OpenSIPS2 when you perform an ngrep for
the SIP packets?
On Tue, Nov 10, 2009 at 8:27 PM, Duc-Lam Tran dltran1...@gmail.com wrote:
Hi there
I got 2 Subnets with the addresses 192.168.1.0 and 172.31.10.0
Between those two OpenSips is a Linux Debian Gateway installed with IP
forwarding and 2 Network Interfaces.
OpenSips1 Gateway OpenSips2
My Goal is to config the OpenSips that i can call from Subnet 1 to
Subnet 2 using Prefixes.
I got a Sipphone at Subnet 1 (Callnumber 1000) and another at subnet 2
(1001)
My configutarion of opensips.cfg for prefix calling is:
if (uri=~sip:01[0-9]+@) {
strip(2);
rewritehost(192.168.1.2); // ip from opensips1
t_relay();
exit;
}
-- if I dial 01-1000 (01 Prefix) from agent 2 than i should get a
ring at agent 1. but it doesnt work (time out)
is this code correct? did i forget something? or is there another
possibility to do such calls?
Thanks for your help in advance.
Regards
Duc
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--
--
*--*--*--*--*--*
Duane
*--*--*--*--*--*
--
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