Re: [OpenSIPS-Users] RURI domain on NAT'd endpoints
All very good points. I've largely been able to avoid NAT up until now, so
I'm afraid I'm a bit too green to offer any clarification.
- Jeff
On 9/8/09 1:08 AM, Thomas Gelf tho...@gelf.net wrote:
Thomas Gelf wrote:
Jeff Pyle wrote:
if (client_nat_test(3)) {
force_rport();
$avp(s:received_uri) = $source_uri;
if (!is_method(REGISTER)) fix_contact();
setbflag(7);
}# nat_keepalive() further down after some pref checks
One more thing: I'm doing client_nat_test(1) in reply routes (should
make sense), client_nat_test(7) for REGISTERs and client_nat_test(3)
for other requests. While I'm pretty sure regarding the reply_route part
(you cannot use test 2 and 4 as Via headers are nothing but copies from
Request), I'm wondering whether the latter distinction between REGISTER
and other request is making sense.
Googling and having a look to Flavios book shows that also others are
doing so - why?? If test 4 matches, the request would for sure also
trigger test 2 wouldn't it? So, may I completely skip test 4?
There is one special case that comes to my mind: does test 2 somehow
respect the rport parameter? I mean, if topmost Via has a private IP
and rport set - is test 2 then still triggered?
Best regards,
Thomas Gelf
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[OpenSIPS-Users] Checksum incorrect maybe caused by UDP checksum offload
Hello, Could anybody tell me what is the problem with such a configuration: --- (UAC)192.168.10.1 |=== |192.168.10.10 [OpenSIPS] 192.168.20.20|=== |(UAC)192.168.20.1 | --- The mysterious point is that all packets orginating from the interface (192.168.20.20) towards UAC(192.168.20.1) have Bad checksum in the UDP level. This happened with/without RTPproxy, so i'm wondering whether I should care about this problem or not because the sound is conveyed and everything seems okay. Regards. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Checksum incorrect maybe caused by UDP checksum offload
How are you making the packet captures? If tcpdump, did you use -s0? -- /Saúl http://www.saghul.net | http://www.sipdoc.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Checksum incorrect maybe caused by UDP checksum offload
I'm using Wireshark Saúl Ibarra sag...@gmail.com a écrit : How are you making the packet captures? If tcpdump, did you use -s0? -- /Saúl http://www.saghul.net | http://www.sipdoc.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] Checksum incorrect maybe caused by UDP checksum offload
The mysterious point is that all packets orginating from the interface (192.168.20.20) towards UAC(192.168.20.1) have Bad checksum in the UDP level. This happened with/without RTPproxy, so i'm wondering whether I should care about this problem or not because the sound is conveyed and everything seems okay. I guess everything is fine. Try to disable checksum offloading (using ethtool), your capture should look fine then. Cheers, Thomas -- mail: tho...@gelf.net web: http://thomas.gelf.net/ ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] CDRTool query to media_sessions table
Dan, Unfortunately I'm not finding any references to escapaing characters in Freeradius. I realize this isn't your home turf, but having said that, do you have any pointers? Thanks, Jeff On 9/7/09 8:02 AM, Dan Pascu d...@ag-projects.com wrote: On 5 Sep 2009, at 23:01, Jeff Pyle wrote: Hello, On CDRTool v6.8.0 and Mediaproxy 2.3.6 I notice sometimes I get ³No information available² when I click on the Media Information link. In one particular example, the query looks for ³SDjui5c99-324092c7=2B1=2B2b6b0008=2B96cdd538² but the database has ³SDjui5c99-324092c7+1+2b6b0008+96cdd538². Is there a way to encourage CDRTool to query with the non-modified version of the characters? Or, is another solution preferable? CDRTool queries with the information available from the radius accounting table. You should configure your radius server (I presume freeradius) not to escape those characters. The difference appears from the fact that the information is inserted unmodified in the media sessions table, but the radius server escapes certain characters when inserting it in the accounting table. - Jeff ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Dan ___ 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] Checksum incorrect maybe caused by UDP checksum offload
2009/9/8 ghaith.alkay...@telecom-bretagne.eu: The mysterious point is that all packets orginating from the interface (192.168.20.20) towards UAC(192.168.20.1) have Bad checksum in the UDP level. If you're capturing at the machine that has the 192.168.20.20 interface, then everything is ok. You're seeing wrong checksum on your traces because it's calculated on your network card instead - so when you capture the packet the checksum is probably still filled with zeros. Just ignore it, or disable the highlighting of bad checksums. If you disable the offloading, you'll see the result you expect, but you'll also give your CPU a bit more work on every packet, so for RTP that's impacting both latency and throughput. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Access to Radius AVP's and radius_extra parameter
I have been experimenting with Radius authentication on an older version of OpenSIPS (actually my test unit has OpenSER v1.3.2, but I don't think it will be much different to 1.5.x for this). I want to be able to read several values returned from the Radius server. Is this what the modparam radius_extra is for? Or is it to allow extra values to be passed to the Radius server for accounting? My Radius server is configured to return three AVP's like this: Reply-Message = Opensips rules!, Sip-RPId = sip:123...@mysip.com, SIP-AVP = #14:012...@mynumber.biz Using Wireshark, I can see that the values are all being sent back to the OpenSIPS server, but I have only had success in accessing the value in SIP-AVP. Is there a trick to this or are those other values unavailable in the OpenSIPS script? Is the functionality I'm looking for only available in the new AAA modules? John Quick Smartvox Limited Web: www.smartvox.co.uk Smartvox is a limited company, registered in England and Wales, number 5005263. Registered office: Spectrum House, Dunstable Road, Redbourn, St.Albans, Herts AL3 7PR ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] New AAA module: Which release of Opensips?
Hi John, The new AAA/RADIUS support is present only in 1.6 (current devel version). Options are: 1) see if the old RADIUS support does not fit your needs somehow - use opensips 1.5 2) go for the new AAA/RADIUS support and use 1.6 (even if devel, you can take the chance to use it - there are other people doing that) 3) go for the new AAA/RADIUS support, but wait for the official release of 1.6 in mid October. Best regards, Bogdan John Quick wrote: Hi, I want to use the new AAA/Radius module, but it is for a business critical service so am reluctant to move to release 1.6 if it is not quite bedded in yet. What choices are available? John Quick Smartvox Limited Web: www.smartvox.co.uk Smartvox is a limited company, registered in England and Wales, number 5005263. Registered office: Spectrum House, Dunstable Road, Redbourn, St.Albans, Herts AL3 7PR ___ 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] Nextone - Proxy-Authorization and AuthorizationHeader
Hi Julien,
It most of the cases, your approach is correct , but to be 100% safe,
use remove_credentials() instead of remove_hf() - remove_credentials
will remove only the credentials/headers that were using on the local
authentication, which is useful when a requests carries more than one
auth sets (multi stage authentication)
Regards,
Bogdan
Julien Chavanton wrote:
We need to deal with the fact that the remote Nextone UA does not
reply when Proxy-Authorization or Authorization is present in the body
In this example, removing the Proxy-Authorization after authentication
as been done on Opensips seems to work.
# local and authenticated user ? route the call, else challenge
authentication
if(is_from_local()){
if (proxy_authorize(, subscriber)){
remove_hf(Proxy-Authorization);
t_relay();
exit;
}
else{
proxy_challenge(, 1);
exit;
}
}
*From:* users-boun...@lists.opensips.org on behalf of Julien Chavanton
*Sent:* Mon 07/09/2009 4:06 PM
*To:* Users@lists.opensips.org
*Subject:* [OpenSIPS-Users] Nextone - Proxy-Authorization and
AuthorizationHeader
Hi, when forwarding call to Nextone SBC if there is
Proxy-Authorization or Authorization header it does not reply to INVITE
Can I configure Opensip not to forward the authentication information
used with his clients softphones ?
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Re: [OpenSIPS-Users] Regarding Uac_replace_from
Hi Ashwini, I do not know about the Call-Control module too much, but can you also push to it the new FROM value from the script ? or you cannot influence the info that is sent to CC module ? Regards, Bogdan ASHWINI NAIDU wrote: Hi Bogdan, The whole scenario is i want the new from(after uac_replace_from) header value to be used by the call controller. is there any way i can do it. On Fri, Sep 4, 2009 at 6:14 PM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi Ashwini, The variables that provide information from the SIP message (like $fu) get the values from the original received message, disregarding whatever changes you do .The only exception is $ru (and family) vars. Regards, Bogdan ASHWINI NAIDU wrote: Hi all, I am trying to replace the from header using the *uac_replace_from* in uac module. the thing i noticed the replacement is happening in the* sip headers*. After using the *uac_replace_from* and try to print the *$fu and $fU *the previous values before replacement can be seen . can anyone tell me what is the problem? -- Thanking You, Ashwini BR Naidu ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Thanking You, Ashwini BR Naidu ___ 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] Call Control module flexibility (renamed)
Hi Ashwini, I see you point - the question is how flexible is the Call Control module in order to allow you to customize the billing account (and not to use all the time the FROM URI). Maybe somebody involved in this module devel can answer you. Regards, Bogdan ASHWINI NAIDU wrote: Hi Bogdan, I have a scenario where one user Ashwini can have 2 phnos as her accounts. When Ashwini calls from any of there 2 phnos or Ashwini the value shd be deducted from the ashw...@.com mailto:ashw...@.com which is provided. *Example: Ashwini has 91992 and 919xxx19 If i call from 91992 my debit balance should be retrieved from ashwini * This is the whole scenario. The call-controller is picking the original From header value. other than this is there any way to achieve the above scenario. On Fri, Sep 4, 2009 at 7:36 PM, Brett Nemeroff br...@nemeroff.com mailto:br...@nemeroff.com wrote: Why not just use whatever value you are using as a parameter for uac_replace_from ? On Fri, Sep 4, 2009 at 8:07 AM, ASHWINI NAIDU ashwini.na...@gmail.com mailto:ashwini.na...@gmail.com wrote: Hi Bogdan, The whole scenario is i want the new from(after uac_replace_from) header value to be used by the call controller. is there any way i can do it. On Fri, Sep 4, 2009 at 6:14 PM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi Ashwini, The variables that provide information from the SIP message (like $fu) get the values from the original received message, disregarding whatever changes you do .The only exception is $ru (and family) vars. Regards, Bogdan ASHWINI NAIDU wrote: Hi all, I am trying to replace the from header using the *uac_replace_from* in uac module. the thing i noticed the replacement is happening in the* sip headers*. After using the *uac_replace_from* and try to print the *$fu and $fU *the previous values before replacement can be seen . can anyone tell me what is the problem? -- Thanking You, Ashwini BR Naidu ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Thanking You, Ashwini BR Naidu ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Thanking You, Ashwini BR Naidu ___ 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] New AAA module: Which release of Opensips?
Hi, I want to use the new AAA/Radius module, but it is for a business critical service so am reluctant to move to release 1.6 if it is not quite bedded in yet. What choices are available? John Quick Smartvox Limited Web: www.smartvox.co.uk Smartvox is a limited company, registered in England and Wales, number 5005263. Registered office: Spectrum House, Dunstable Road, Redbourn, St.Albans, Herts AL3 7PR ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] explizit handling auf replyto
Hi Uwe, Uwe Kastens wrote: Hi Bogdan, Seems that my question was not very clear. I would expect that reply messages would be handled automatically, if I use t_relay. whatever forwarding function you are using (forward / t_relay), the replies are automatically routed back by opensips core without any help from script. This seems not to happen in my setup. could you post a sip trace of a failing reply ? I need to route this replys with an reply_route and forward them explicitly to the pstn gateway. and how do you do reply routing ?? replies are automatically routed based on VIA stack and you cannot influence this from script. Regards, Bogdan This not as it should be? BR Uwe using forward() forces a stateless behaviour of OpenSIPS. To be able to catch replies for a requests (what on_reply route does) it requires a statefull approach. This is the reason it does not work for you. If you use t_relay() instead of forward(), you should be able to use failure and onreply routes. Regards, Bogdan Uwe Kastens wrote: Hello, I am using opensips 1.5.1 and I have the problem, that busy messages are not passed to the mediagw. Anything else is working (ACK etc.pp.). If I route that message via forward(IP:port) to the media gw in on_replyto, busy is processed correctly. If not, the busy is not send to the mediagw. So I was wondering if I had to handle some replyto messages? BR uwe ___ 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] b2bua top hiding + authentication
Hello, What impact does authentication have on the b2bua modules? When I put b2b_init_request(top hiding) into my otherwise functioning script, very strange things happen. In order to achieve topology hiding, is it as simple as inserting this init line at some point, or is there more to it? - Jeff ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] rpid_avp and NULL values
Il giorno 22/lug/09, alle ore 11:09, Bogdan-Andrei Iancu ha scritto: Hi Carlo, you must set the line : modparam(auth_db, load_credentials, $avp(s:rpid)=rpid) This line instructs opensips to load at db auth time tht rpid field into the $avp(s:rpid) variable. Hi Bogdan, Last july I have modified the opensips.cfg adding modparam(auth_db, load_credentials, $avp(s:rpid)=rpid) and all worked. Now I have executed some test after upgrading to 1.5.3 and I'm experiencing with the same problem (if rpid field of user A is NULL, after some calls user A takes the rpid of another user B (that is not null)) In addition, user A has a different domain of user B. What do you think about this behaviour? Thanks and Regards, Carlo Dimaggio ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Access to Radius AVP's and radius_extra parameter
Hello John, On Tue, Sep 8, 2009 at 5:43 PM, John Quick john.qu...@smartvox.co.ukwrote: I have been experimenting with Radius authentication on an older version of OpenSIPS (actually my test unit has OpenSER v1.3.2, but I don't think it will be much different to 1.5.x for this). I want to be able to read several values returned from the Radius server. Is this what the modparam radius_extra is for? Or is it to allow extra values to be passed to the Radius server for accounting? The radius_extra parameter only allows you to send extra values to the Radius server, to be logged for accounting. You cannot fetch any value from an accounting reply. My Radius server is configured to return three AVP's like this: Reply-Message = Opensips rules!, Sip-RPId = sip:123...@mysip.com sip%3a123...@mysip.com, SIP-AVP = #14:012...@mynumber.biz 14%3a012...@mynumber.biz Using Wireshark, I can see that the values are all being sent back to the OpenSIPS server, but I have only had success in accessing the value in SIP-AVP. Is there a trick to this or are those other values unavailable in the OpenSIPS script? Is the functionality I'm looking for only available in the new AAA modules? The functionality you are looking for is provided by the new AAA_RADIUS module. Using the functions exported by this module you can yield directly from the script any type of Radius query and also inspect replies for certain attributes. For more information, check out the tutorialhttp://www.opensips.org/Resources/DocsTutRadiusand the documentationhttp://www.opensips.org/html/docs/modules/devel/aaa_radius.html . If you have any other uncertainties, please let me know. Any other suggestions are welcome as well. John Quick Smartvox Limited Web: www.smartvox.co.uk Smartvox is a limited company, registered in England and Wales, number 5005263. Registered office: Spectrum House, Dunstable Road, Redbourn, St.Albans, Herts AL3 7PR ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users Best regards, Irina Stanescu ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] explizit handling auf replyto
Hi Bogdan,
I need to route this
replys with an reply_route and forward them explicitly to the pstn gateway.
and how do you do reply routing ?? replies are automatically routed
based on VIA stack and you cannot influence this from script.
...
if (!t_relay()) {
sl_reply_error();
};
t_on_reply(1);
}
onreply_route[1] {
xlog(L_DBG,== route [1] t_on_failure msg-len from $si:$sp:
$ml \n$mb\n ==\n);
forward(10.20.30.101:5100);
}
BR
Uwe
Regards,
Bogdan
This not as it should be?
BR
Uwe
using forward() forces a stateless behaviour of OpenSIPS. To be able to
catch replies for a requests (what on_reply route does) it requires a
statefull approach. This is the reason it does not work for you.
If you use t_relay() instead of forward(), you should be able to use
failure and onreply routes.
Regards,
Bogdan
Uwe Kastens wrote:
Hello,
I am using opensips 1.5.1 and I have the problem, that busy messages are
not passed to the mediagw. Anything else is working (ACK etc.pp.). If I
route that message via forward(IP:port) to the media gw in on_replyto,
busy is processed correctly. If not, the busy is not send to the mediagw.
So I was wondering if I had to handle some replyto messages?
BR
uwe
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Re: [OpenSIPS-Users] Newbie To OpenSIPS
Hi Lars, Any guidance would be appreciated from the community that has done this already. I installed current revision. Could be helpfull to know what you want to do with opensips :-) BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] explizit handling auf replyto
Replies are automatically routed only if they are statefully routed.
Uwe Kastens wrote:
Hi Bogdan,
I need to route this
replys with an reply_route and forward them explicitly to the pstn gateway.
and how do you do reply routing ?? replies are automatically routed
based on VIA stack and you cannot influence this from script.
...
if (!t_relay()) {
sl_reply_error();
};
t_on_reply(1);
}
onreply_route[1] {
xlog(L_DBG,== route [1] t_on_failure msg-len from $si:$sp:
$ml \n$mb\n ==\n);
forward(10.20.30.101:5100);
}
BR
Uwe
Regards,
Bogdan
This not as it should be?
BR
Uwe
using forward() forces a stateless behaviour of OpenSIPS. To be able to
catch replies for a requests (what on_reply route does) it requires a
statefull approach. This is the reason it does not work for you.
If you use t_relay() instead of forward(), you should be able to use
failure and onreply routes.
Regards,
Bogdan
Uwe Kastens wrote:
Hello,
I am using opensips 1.5.1 and I have the problem, that busy messages are
not passed to the mediagw. Anything else is working (ACK etc.pp.). If I
route that message via forward(IP:port) to the media gw in on_replyto,
busy is processed correctly. If not, the busy is not send to the mediagw.
So I was wondering if I had to handle some replyto messages?
BR
uwe
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Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
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Re: [OpenSIPS-Users] explizit handling auf replyto
Hi, Replies are automatically routed only if they are statefully routed. Statefull = t_relay() ? BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] explizit handling auf replyto
Uwe Kastens wrote: Replies are automatically routed only if they are statefully routed. Statefull = t_relay() ? 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
Re: [OpenSIPS-Users] explizit handling auf replyto
Uwe Kastens wrote: Hi, Replies are automatically routed only if they are statefully routed. Statefull = t_relay() ? Keep in mind, you can't really mix stateful and stateless handling for what are hopefully obvious reasons. If you statelessly direct a reply, the stateful transaction layer used to open the transaction from the initial request will not know that you did so, and vice versa, etc etc. -- 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] explizit handling auf replyto
Hello,
Ok. I need to have that forward() in my configuration the get answers
like 404, 486, 487 back to my asterisk. Reading your statements this
should not be possible since I use t_relay for the requests and the
replys should be routed by default.
I will make a trace and post it to the list. One with forward and the
other without.
BR and Thanks in advance
Uwe
Bogdan-Andrei Iancu schrieb:
Uwe,
forward() is a function exclusivly used for REQUESTS - for replies,
nothing needs to be done as OpenSIPS will do it automatically:
1) if the requests was statefully forwarded (via t_relay() ), the
transaction will contain all the info to route back the reply
2) if the requests was statelessly forwarded (via forward() ), the VIA
stack (in received reply) will contain all the info to route back the reply
Regards,
Bogdan
Uwe Kastens wrote:
Hi Bogdan,
I need to route this
replys with an reply_route and forward them explicitly to the pstn gateway.
and how do you do reply routing ?? replies are automatically routed
based on VIA stack and you cannot influence this from script.
...
if (!t_relay()) {
sl_reply_error();
};
t_on_reply(1);
}
onreply_route[1] {
xlog(L_DBG,== route [1] t_on_failure msg-len from $si:$sp:
$ml \n$mb\n ==\n);
forward(10.20.30.101:5100);
}
BR
Uwe
Regards,
Bogdan
This not as it should be?
BR
Uwe
using forward() forces a stateless behaviour of OpenSIPS. To be able to
catch replies for a requests (what on_reply route does) it requires a
statefull approach. This is the reason it does not work for you.
If you use t_relay() instead of forward(), you should be able to use
failure and onreply routes.
Regards,
Bogdan
Uwe Kastens wrote:
Hello,
I am using opensips 1.5.1 and I have the problem, that busy messages are
not passed to the mediagw. Anything else is working (ACK etc.pp.). If I
route that message via forward(IP:port) to the media gw in on_replyto,
busy is processed correctly. If not, the busy is not send to the mediagw.
So I was wondering if I had to handle some replyto messages?
BR
uwe
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Re: [OpenSIPS-Users] explizit handling auf replyto
Uwe,
forward() is a function exclusivly used for REQUESTS - for replies,
nothing needs to be done as OpenSIPS will do it automatically:
1) if the requests was statefully forwarded (via t_relay() ), the
transaction will contain all the info to route back the reply
2) if the requests was statelessly forwarded (via forward() ), the VIA
stack (in received reply) will contain all the info to route back the reply
Regards,
Bogdan
Uwe Kastens wrote:
Hi Bogdan,
I need to route this
replys with an reply_route and forward them explicitly to the pstn gateway.
and how do you do reply routing ?? replies are automatically routed
based on VIA stack and you cannot influence this from script.
...
if (!t_relay()) {
sl_reply_error();
};
t_on_reply(1);
}
onreply_route[1] {
xlog(L_DBG,== route [1] t_on_failure msg-len from $si:$sp:
$ml \n$mb\n ==\n);
forward(10.20.30.101:5100);
}
BR
Uwe
Regards,
Bogdan
This not as it should be?
BR
Uwe
using forward() forces a stateless behaviour of OpenSIPS. To be able to
catch replies for a requests (what on_reply route does) it requires a
statefull approach. This is the reason it does not work for you.
If you use t_relay() instead of forward(), you should be able to use
failure and onreply routes.
Regards,
Bogdan
Uwe Kastens wrote:
Hello,
I am using opensips 1.5.1 and I have the problem, that busy messages are
not passed to the mediagw. Anything else is working (ACK etc.pp.). If I
route that message via forward(IP:port) to the media gw in on_replyto,
busy is processed correctly. If not, the busy is not send to the mediagw.
So I was wondering if I had to handle some replyto messages?
BR
uwe
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Re: [OpenSIPS-Users] explizit handling auf replyto
Hi, you do not need to do any routing in onreply route at all, in none of the case (stateless or statefull) I will make a trace and post it to the list. One with forward and the other without. make a trace and opensips logs. I have attached opensips.log with debug=9. w_forward_an.gz = with forward wo_forward_an.gz = without forward BR Uwe -- kiste lat: 54.322684, lon: 10.13586 w_forward_an.gz Description: GNU Zip compressed data wo_forward_an.gz Description: GNU Zip compressed data ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Newbie To OpenSIPS
Thank you for the response UK. We are looking at using OpenSIPS on Linux to act as a SIP Router to handoff SIP calls to other carriers that are also using SIP, and explore the use of the agent registration server functionality. I have already previously installed and debugged Asterisk on CentOS to act as a branded appliance issuing DHCP and DNS to VOIP phones. My understanding is that you can route your Asterisk systems to OpenSIPS as a SIP-Proxy-where they can then get router to other SIP or SS7 carriers, or your own Class 4 or 5 environment. Maybe I am misunderstanding as far as what OpenSIPS can actually do? Thanks. [cid:image001.jpg@01CA307D.1602ADC0] -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens Sent: Tuesday, September 08, 2009 11:32 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS Hi Lars, Any guidance would be appreciated from the community that has done this already. I installed current revision. Could be helpfull to know what you want to do with opensips :-) BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users inline: image001.jpg___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Newbie To OpenSIPS
Hi, Opensips is a SIP router not a media gateway. So far you will need something that will take care of comvert TDM/PSTN to sip. There should be lot of examples for this kind of setup. BR Uwe Kemp, Larry schrieb: Thank you for the response UK. We are looking at using OpenSIPS on Linux to act as a SIP Router to handoff SIP calls to other carriers that are also using SIP, and explore the use of the agent registration server functionality. I have already previously installed and debugged Asterisk on CentOS to act as a branded appliance issuing DHCP and DNS to VOIP phones. My understanding is that you can route your Asterisk systems to OpenSIPS as a SIP-Proxy-where they can then get router to other SIP or SS7 carriers, or your own Class 4 or 5 environment. Maybe I am misunderstanding as far as what OpenSIPS can actually do? Thanks. -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens Sent: Tuesday, September 08, 2009 11:32 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS Hi Lars, Any guidance would be appreciated from the community that has done this already. I installed current revision. Could be helpfull to know what you want to do with opensips :-) BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ 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 -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] explizit handling auf replyto
Bogdan-Andrei Iancu wrote:
2) if the requests was statelessly forwarded (via forward() ), the VIA
stack (in received reply) will contain all the info to route back the reply
I think the question is whether stateless forwarding can be used to
override default processing of Via and route the reply somewhere else.
You can, for example, do this (whether in stateless or stateful request
forwarding mode):
onreply_route[1] {
drop;
}
... I think it's along that general train of thought.
--
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Evariste Systems
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Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
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Re: [OpenSIPS-Users] Newbie To OpenSIPS
Hi, Good question and not easy to answer. ACME is expensive AND you will need somebody to configure it in a way you will need it. So as an redundant option your talking about 100-150K. To buy a big name won't prevent you from implementing, bugsearching. My personal opinion: Take less money, look for good consultants and try it with opensource. BR Uwe Kemp, Larry schrieb: Certainly. If I just wanted to pass my SIP to other carriers or have them connect to my SIP customers could I use OpenSIPS for that alone, or would I still need some other sort of session border controller? Larry Kemp -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens Sent: Tuesday, September 08, 2009 1:31 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS Hi, Yes, this could be an option. But a very expensive one :-) BR Uwe Kemp, Larry schrieb: So I would use OpenSIPS behind say like an Acme Packet http://www.acmepacket.com/ Session Border Controller or a MetaSwitch http://www.metaswitch.com/ connecting to the PSTN, then use OpenSIPS to talk to those Gateways via SIP and route my customer's VOIP traffic from their Asterisk PBX's to those devices that speak SIP, right? Lars -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens Sent: Tuesday, September 08, 2009 12:19 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS Hi, Opensips is a SIP router not a media gateway. So far you will need something that will take care of comvert TDM/PSTN to sip. There should be lot of examples for this kind of setup. BR Uwe Kemp, Larry schrieb: Thank you for the response UK. We are looking at using OpenSIPS on Linux to act as a SIP Router to handoff SIP calls to other carriers that are also using SIP, and explore the use of the agent registration server functionality. I have already previously installed and debugged Asterisk on CentOS to act as a branded appliance issuing DHCP and DNS to VOIP phones. My understanding is that you can route your Asterisk systems to OpenSIPS as a SIP-Proxy-where they can then get router to other SIP or SS7 carriers, or your own Class 4 or 5 environment. Maybe I am misunderstanding as far as what OpenSIPS can actually do? Thanks. -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens Sent: Tuesday, September 08, 2009 11:32 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS Hi Lars, Any guidance would be appreciated from the community that has done this already. I installed current revision. Could be helpfull to know what you want to do with opensips :-) BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ 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 -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Newbie To OpenSIPS
Can OpenSIPS be used as a Session Border controller sitting at my edge passing and receiving SIP traffic to others I SIP peer with? If not, what other open-source would anyone suggest to act as SBC's? I too would rather do it via open-source and x86 or 64bit chip, less costly. Thanks. Lars -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens Sent: Tuesday, September 08, 2009 1:50 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS Hi, Good question and not easy to answer. ACME is expensive AND you will need somebody to configure it in a way you will need it. So as an redundant option your talking about 100-150K. To buy a big name won't prevent you from implementing, bugsearching. My personal opinion: Take less money, look for good consultants and try it with opensource. BR Uwe Kemp, Larry schrieb: Certainly. If I just wanted to pass my SIP to other carriers or have them connect to my SIP customers could I use OpenSIPS for that alone, or would I still need some other sort of session border controller? Larry Kemp -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens Sent: Tuesday, September 08, 2009 1:31 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS Hi, Yes, this could be an option. But a very expensive one :-) BR Uwe Kemp, Larry schrieb: So I would use OpenSIPS behind say like an Acme Packet http://www.acmepacket.com/ Session Border Controller or a MetaSwitch http://www.metaswitch.com/ connecting to the PSTN, then use OpenSIPS to talk to those Gateways via SIP and route my customer's VOIP traffic from their Asterisk PBX's to those devices that speak SIP, right? Lars -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens Sent: Tuesday, September 08, 2009 12:19 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS Hi, Opensips is a SIP router not a media gateway. So far you will need something that will take care of comvert TDM/PSTN to sip. There should be lot of examples for this kind of setup. BR Uwe Kemp, Larry schrieb: Thank you for the response UK. We are looking at using OpenSIPS on Linux to act as a SIP Router to handoff SIP calls to other carriers that are also using SIP, and explore the use of the agent registration server functionality. I have already previously installed and debugged Asterisk on CentOS to act as a branded appliance issuing DHCP and DNS to VOIP phones. My understanding is that you can route your Asterisk systems to OpenSIPS as a SIP-Proxy-where they can then get router to other SIP or SS7 carriers, or your own Class 4 or 5 environment. Maybe I am misunderstanding as far as what OpenSIPS can actually do? Thanks. -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens Sent: Tuesday, September 08, 2009 11:32 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS Hi Lars, Any guidance would be appreciated from the community that has done this already. I installed current revision. Could be helpfull to know what you want to do with opensips :-) BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ 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 -- kiste lat: 54.322684, lon: 10.13586 ___ 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] Newbie To OpenSIPS
Kemp, Larry wrote: Can OpenSIPS be used as a Session Border controller sitting at my edge passing and receiving SIP traffic to others I SIP peer with? If not, what other open-source would anyone suggest to act as SBC's? I too would rather do it via open-source and x86 or 64bit chip, less costly. Thanks. This is mostly a semantic issue. OpenSIPS is not an SBC in the way that commercial SBCs are SBCs, and lacks a number of key aspects, including ASIC-assisted processing in the higher-end ones. But it can be used for subscriber or carrier-facing edge duty in the same way SBCs are often used (unnecessarily so). -- 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] Newbie To OpenSIPS
Personally I prefer the Sonus GSX9000 gear for a PSTN gateway. I even put one in my kid's room. - Jeff On 9/8/09 1:30 PM, Uwe Kastens ki...@kiste.org wrote: Hi, Yes, this could be an option. But a very expensive one :-) BR Uwe Kemp, Larry schrieb: So I would use OpenSIPS behind say like an Acme Packet http://www.acmepacket.com/ Session Border Controller or a MetaSwitch http://www.metaswitch.com/ connecting to the PSTN, then use OpenSIPS to talk to those Gateways via SIP and route my customer's VOIP traffic from their Asterisk PBX's to those devices that speak SIP, right? Lars -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens Sent: Tuesday, September 08, 2009 12:19 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS Hi, Opensips is a SIP router not a media gateway. So far you will need something that will take care of comvert TDM/PSTN to sip. There should be lot of examples for this kind of setup. BR Uwe Kemp, Larry schrieb: Thank you for the response UK. We are looking at using OpenSIPS on Linux to act as a SIP Router to handoff SIP calls to other carriers that are also using SIP, and explore the use of the agent registration server functionality. I have already previously installed and debugged Asterisk on CentOS to act as a branded appliance issuing DHCP and DNS to VOIP phones. My understanding is that you can route your Asterisk systems to OpenSIPS as a SIP-Proxy-where they can then get router to other SIP or SS7 carriers, or your own Class 4 or 5 environment. Maybe I am misunderstanding as far as what OpenSIPS can actually do? Thanks. -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens Sent: Tuesday, September 08, 2009 11:32 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS Hi Lars, Any guidance would be appreciated from the community that has done this already. I installed current revision. Could be helpfull to know what you want to do with opensips :-) BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ 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] Newbie To OpenSIPS
Hi, Yes, this could be an option. But a very expensive one :-) BR Uwe Kemp, Larry schrieb: So I would use OpenSIPS behind say like an Acme Packet http://www.acmepacket.com/ Session Border Controller or a MetaSwitch http://www.metaswitch.com/ connecting to the PSTN, then use OpenSIPS to talk to those Gateways via SIP and route my customer's VOIP traffic from their Asterisk PBX's to those devices that speak SIP, right? Lars -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens Sent: Tuesday, September 08, 2009 12:19 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS Hi, Opensips is a SIP router not a media gateway. So far you will need something that will take care of comvert TDM/PSTN to sip. There should be lot of examples for this kind of setup. BR Uwe Kemp, Larry schrieb: Thank you for the response UK. We are looking at using OpenSIPS on Linux to act as a SIP Router to handoff SIP calls to other carriers that are also using SIP, and explore the use of the agent registration server functionality. I have already previously installed and debugged Asterisk on CentOS to act as a branded appliance issuing DHCP and DNS to VOIP phones. My understanding is that you can route your Asterisk systems to OpenSIPS as a SIP-Proxy-where they can then get router to other SIP or SS7 carriers, or your own Class 4 or 5 environment. Maybe I am misunderstanding as far as what OpenSIPS can actually do? Thanks. -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens Sent: Tuesday, September 08, 2009 11:32 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS Hi Lars, Any guidance would be appreciated from the community that has done this already. I installed current revision. Could be helpfull to know what you want to do with opensips :-) BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ 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 -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Newbie To OpenSIPS
Certainly. If I just wanted to pass my SIP to other carriers or have them connect to my SIP customers could I use OpenSIPS for that alone, or would I still need some other sort of session border controller? Larry Kemp -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens Sent: Tuesday, September 08, 2009 1:31 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS Hi, Yes, this could be an option. But a very expensive one :-) BR Uwe Kemp, Larry schrieb: So I would use OpenSIPS behind say like an Acme Packet http://www.acmepacket.com/ Session Border Controller or a MetaSwitch http://www.metaswitch.com/ connecting to the PSTN, then use OpenSIPS to talk to those Gateways via SIP and route my customer's VOIP traffic from their Asterisk PBX's to those devices that speak SIP, right? Lars -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens Sent: Tuesday, September 08, 2009 12:19 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS Hi, Opensips is a SIP router not a media gateway. So far you will need something that will take care of comvert TDM/PSTN to sip. There should be lot of examples for this kind of setup. BR Uwe Kemp, Larry schrieb: Thank you for the response UK. We are looking at using OpenSIPS on Linux to act as a SIP Router to handoff SIP calls to other carriers that are also using SIP, and explore the use of the agent registration server functionality. I have already previously installed and debugged Asterisk on CentOS to act as a branded appliance issuing DHCP and DNS to VOIP phones. My understanding is that you can route your Asterisk systems to OpenSIPS as a SIP-Proxy-where they can then get router to other SIP or SS7 carriers, or your own Class 4 or 5 environment. Maybe I am misunderstanding as far as what OpenSIPS can actually do? Thanks. -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Uwe Kastens Sent: Tuesday, September 08, 2009 11:32 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Newbie To OpenSIPS Hi Lars, Any guidance would be appreciated from the community that has done this already. I installed current revision. Could be helpfull to know what you want to do with opensips :-) BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ 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 -- kiste lat: 54.322684, lon: 10.13586 ___ 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] Newbie To OpenSIPS
First time compiling OpenSIPS; installing on CentOS. Downloaded source as well as RPM's from http://centos.leurent.eu. I was wondering if any of you that have already done this successfully and had any resources or notes from your deployment that you might be willing to share, detailed how-to info for database and making/installing gui, or could recommend which of the many books that exist to read on this. Info seems splintered since the OpenSER split. To date I have: Compiled started /usr/local/sbin/opensips Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 172.20.30.184 [172.20.30.184]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 172.20.30.184 [172.20.30.184]:5060 Aliases: tcp: localhost:5060 tcp: localhost.localdomain:5060 tcp: hostname:5060 tcp: hostname.domain.com:5060 udp: localhost:5060 udp: localhost.localdomain:5060 udp: hostname:5060 udp: hostname.domain.com:5060 Any guidance would be appreciated from the community that has done this already. I installed current revision. Lars ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] questions about log?
Hello everybody, dose anyone know where the log file is?___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Regarding Uac_replace_from
Hi Bogdan, That problem is solved by using the diverter_avp modparam in call control module. thanks for replying On Tue, Sep 8, 2009 at 8:37 PM, Bogdan-Andrei Iancu bog...@voice-system.rowrote: Hi Ashwini, I do not know about the Call-Control module too much, but can you also push to it the new FROM value from the script ? or you cannot influence the info that is sent to CC module ? Regards, Bogdan ASHWINI NAIDU wrote: Hi Bogdan, The whole scenario is i want the new from(after uac_replace_from) header value to be used by the call controller. is there any way i can do it. On Fri, Sep 4, 2009 at 6:14 PM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi Ashwini, The variables that provide information from the SIP message (like $fu) get the values from the original received message, disregarding whatever changes you do .The only exception is $ru (and family) vars. Regards, Bogdan ASHWINI NAIDU wrote: Hi all, I am trying to replace the from header using the *uac_replace_from* in uac module. the thing i noticed the replacement is happening in the* sip headers*. After using the *uac_replace_from* and try to print the *$fu and $fU *the previous values before replacement can be seen . can anyone tell me what is the problem? -- Thanking You, Ashwini BR Naidu ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org mailto:Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Thanking You, Ashwini BR Naidu ___ 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 -- Thanking You, Ashwini BR Naidu ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] questions about log?
By default the logging of opensips will be done in */var/log/syslog* in debian systems and */var/log/messages* in redhat based systems 2009/9/9 zhangchao1 zhangchao...@163.com Hello everybody, dose anyone know where the log file is? -- 中国制造,讲述中国60年往事http://news.163.com/madeinchina/index.html?from=mailfooter ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Thanking You, Ashwini BR Naidu ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
