Re: [OpenSIPS-Users] backport for dialog profile for 1.5
Even a backport wouldn't help you, as profiles, vars and flags can change multiple times during a dialog they are not stored to db unless you restart OpenSIPS - to let dialogs cleanly survive a stop /start sequence. What you want to retrieve is however available via MI-modules, I'm for example preferring the XML-RPC one. Best regards, Thomas Gelf Uwe Kastens schrieb: Hello, I miss the option to find out via db how much calls are online for defined dialog profiles. There is an option in 1.6 where profiles is usable via database. Would there be a backport to 1.5 for that field? BR Uwe ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Multi-homed systems
I've setup a multi-homed OpenSIPS system with two ethernet interfaces (eth0 eth1). The problem I've got, is that regardless of which physical interface the packets leave the box, they always have the same source IP address. I.e. Packets leaving eth0 have the IP address of eth1. Is there any way to control this, and either tell OpenSIPS to use the interface IP address, or to specify, for this route, use this source IP address ? GTG ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] backport for dialog profile for 1.5
Thomas, Thomas Gelf schrieb: Even a backport wouldn't help you, as profiles, vars and flags can change multiple times during a dialog they are not stored to db unless you restart OpenSIPS - to let dialogs cleanly survive a stop /start sequence. Good point. Ok so a backport make no sense. What you want to retrieve is however available via MI-modules, I'm for example preferring the XML-RPC one. I will try this out. Looks simple from the docs - is it that simple? 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] Multi-homed systems
2009/7/7 Gordon Ross gr...@ucs.cam.ac.uk: I've setup a multi-homed OpenSIPS system with two ethernet interfaces (eth0 eth1). The problem I've got, is that regardless of which physical interface the packets leave the box, they always have the same source IP address. I.e. Packets leaving eth0 have the IP address of eth1. Is there any way to control this, and either tell OpenSIPS to use the interface IP address, or to specify, for this route, use this source IP address ? It shouldn't happen. Do you have Iptables rules? -- 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] Multi-homed systems
On 07/07/2009 08:43, Iñaki Baz Castillo i...@aliax.net wrote: It shouldn't happen. Do you have Iptables rules? It shouldn't happen, but it does. Ethereal/wireshark doesn't usually lie. No IP tables rules. Nothing. GTG ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Multi-homed systems
Hi, set mhomed=1 ? BR Uwe Gordon Ross schrieb: I've setup a multi-homed OpenSIPS system with two ethernet interfaces (eth0 eth1). The problem I've got, is that regardless of which physical interface the packets leave the box, they always have the same source IP address. I.e. Packets leaving eth0 have the IP address of eth1. Is there any way to control this, and either tell OpenSIPS to use the interface IP address, or to specify, for this route, use this source IP address ? GTG ___ 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] backport for dialog profile for 1.5
Uwe Kastens wrote:
What you want to retrieve is however available via MI-modules, I'm
for example preferring the XML-RPC one.
I will try this out. Looks simple from the docs - is it that simple?
It is. Short example using PHP with ZF libraries:
$client = new Zend_XmlRpc_Client('http://sip.proxy.tld:8000/RPC2');
$proxy = $client-getProxy();
$dialog_list = $proxy-dlg_list();
Should return more or less what you get on commandline running
opensipsctl fifo dlg_list
The rest is just a text-parsing task, you'll not get structured data.
Cheers,
Thomas Gelf
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Re: [OpenSIPS-Users] Multi-homed systems
On 07/07/2009 08:57, Uwe Kastens ki...@kiste.org wrote: Hi, set mhomed=1 ? Star ! Thanks, GTG ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Transfering hold event from OpenSIPS to SEMS for music on hold
Indeed, but you should be also able to do onhold with OpenSIPS + RTPproxy only. The rtpproxy has some new functions to inject RTP streams: rtpproxy_stream2uac - http://www.opensips.org/html/docs/modules/1.5.x/nathelper.html#rtpproxy_stream2xxx The idea will be to start streaming via RTPproxy when you detect the on hold (0.0.0.0 IP) and to stop it when off-hold is done (valid ip in SDP). Regards, Bogdan Victor Gamov wrote: looks like b2bua module announced in 1.6 will resolve this situation http://lists.opensips.org/pipermail/users/2009-July/006669.html On 23.06.2009 15:57, Yehavi Bourvine wrote: Unfortunately it seems that Stefan's answer is the correct one... Thanks anyway! __Yehavi: 2009/6/23 Stefan Sayer stefan.sa...@iptego.com mailto:stefan.sa...@iptego.com Hello, o Yehavi Bourvine [06/21/09 08:47]: Hello, We want to implement music on hold. With OpenSIPS we can catch the INVITE used for hold (recognize if by the sendonly voice attribute) and would like to forward it to the SEMS so it can play the music. The problem is that this invite is in the middle of a dialog and cannot be just redirected to SEMS (as SEMS will try to look for an existing dialog and fail). How do I do it? Is there any sample config code for this? not easily. Have a look at this response to the same question: http://lists.iptel.org/pipermail/sems/2009-June/002930.html ___ 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] CDRTool - increasing max_allowed_packet
Dan I did not see this yet but it sounds like a PHP setting, you may want to check your php.ini. Adrian On Jul 6, 2009, at 12:49 PM, DanB wrote: Hey Guys, I have recently discovered a problem with one of my CDRTool installations. I have about 2.2 mil destinations in cdrtool.destinations table and when trying to load them into cdrtool memory, the process fails with: Got a packet bigger than 'max_allowed_packet' bytes (1153). Can u please advice how to increase this value? Ta, DanB ___ 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] Transfering hold event from OpenSIPS to SEMS for music on hold
Bogdan,Can you provide some examples on the wiki for this? :) I've wanted to implement this in opensips for some time now. -Brett On Tue, Jul 7, 2009 at 3:19 AM, Bogdan-Andrei Iancu bog...@voice-system.rowrote: Indeed, but you should be also able to do onhold with OpenSIPS + RTPproxy only. The rtpproxy has some new functions to inject RTP streams: rtpproxy_stream2uac - http://www.opensips.org/html/docs/modules/1.5.x/nathelper.html#rtpproxy_stream2xxx The idea will be to start streaming via RTPproxy when you detect the on hold (0.0.0.0 IP) and to stop it when off-hold is done (valid ip in SDP). Regards, Bogdan Victor Gamov wrote: looks like b2bua module announced in 1.6 will resolve this situation http://lists.opensips.org/pipermail/users/2009-July/006669.html On 23.06.2009 15:57, Yehavi Bourvine wrote: Unfortunately it seems that Stefan's answer is the correct one... Thanks anyway! __Yehavi: 2009/6/23 Stefan Sayer stefan.sa...@iptego.com mailto:stefan.sa...@iptego.com Hello, o Yehavi Bourvine [06/21/09 08:47]: Hello, We want to implement music on hold. With OpenSIPS we can catch the INVITE used for hold (recognize if by the sendonly voice attribute) and would like to forward it to the SEMS so it can play the music. The problem is that this invite is in the middle of a dialog and cannot be just redirected to SEMS (as SEMS will try to look for an existing dialog and fail). How do I do it? Is there any sample config code for this? not easily. Have a look at this response to the same question: http://lists.iptel.org/pipermail/sems/2009-June/002930.html ___ 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] Is opensips a front end to asterisk?
I think I'd like to jump on the bandwagon here also As said by some of the other members on the list, this IS a very big project. The fact that you are looking for an ISO to test it out, suggests that you don't really understand how it all works. Without going into too much detail, let me just say that OpenSIPs won't do anything for you. You have to configure it from the ground up. So even if you had an ISO, it would give you binaries and modules that essentially wouldn't do anything at all. And honestly the compilation, installation bits are the easy part! As for this whole Asterisk redundancy. This has been discussed for a long time and there are dozens of ways to do that. Everything from using UCARP, Linux HA, Asterisk redundancy patches, database backending, programming backup proxy IP in UACs. One thing I can tell you with reasonably certainty is that anything you do to make redundancy with asterisk is almost certainly a hack. That being said, I can also say that many many people (myself included) have come up with methods that seem to work to provide redundancy in a way that is reasonably painless for the end user. But there are always something like: 1. the backup server doesn't have the primary server voicemails synced 2. phones on the backup server can't call phones still on the primary server 3. call picked, user presence doesn't work from primary to backup Just keep in mind that Asterisk itself really wasn't ever designed with the concept that another server would be it's redundant pair. As for pairing OpenSIPs and Asterisk. There are a dozen ways to do that as well.. You can do it on the front end (phones register to opensips), on the back end (phones register to asterisk, calls go out to opensips), or somewhere in between even (ie: Asterisk as just a media server). It's all really dependent on how you are going to leverage the technologies of the individual platforms. I personally try to make asterisk do as absolutely little as possible. I think you'll find scaling OpenSIPs much more enjoyable than Asterisk. ;) That all being said, you're really only limited by your creativity and skill level with the platforms. It's best to think of them complementing each other. Pick the features you want to use from each. As for NAT. Sure you can run these platforms behind NAT, but you're asking for a world of pain. :) Just don't do it. That's my opinion. Far end NAT is tricky enough to deal with to also have Near end NAT issues as well. http://en.wikipedia.org/wiki/KISS_principle :) -Brett On Wed, Jul 1, 2009 at 10:52 PM, li...@grounded.net li...@grounded.netwrote: I've come across this project a few times but have been having a bit of a time confirming just what the project does. I thought perhaps the best way would be to join the list and ask. My task is to put together a scalable asterisk based pbx system. Because the boxes will initially have more than they really should installed on them, we need to limit the number of users per box to perhaps 50. Right now, the plan calls for every box to have a second one for redundancy. I was planning on manually redirecting connections (for now) but it sounds like opensips could take care of a number of issues. I have multiple providers (WANs) at one location but was thinking that for highest reliability, that I might have three locations to be safe unless there are better ideas. One would be the location where the initial user connection is made, such as a proxy/load balancer. Then, two separate physical locations and networks for redundancy. The front end could use both sites as needed but if something went down, could re-route users/sessions to the redundant location. This of course is where my questions about opensips come in. -From what I can tell, opensips could act as a pbx on it's own but it can act as a proxy/load balancer/gateway to asterisk systems as well. -If this is the case, would there be a way of creating a distributed environment, like as in a web server farm, making scaling quite easy. -Does opensips handle only new incoming connections or could it actually move sessions from a down server to another which is still up? -Would there be any control, or even any need depending on how the back end can be set up, by which to control which pbx/pair that someone registers to? -Would I have some method of controlling how many people can register on any one box? Thank you very much for this information as it will help to first understand what the project can do. Mike ___ 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] SEMS 1.1.1 released
(apologies if you receive multiple copies of this mail) Hello, the 111 version of SEMS, the SIP Express Media Server, is now available for download at http://ftp.iptel.org/pub/sems/1.1/1.1.1/ http://ftp.iptel.org/pub/sems/1.1/1.1.1/src/sems-1.1.1.tar.gz 4a6422d09ddadaf9eacd8cae8f0848d5 sems-1.1.1.tar.gz Debian packages for lenny64 are available in the repository: deb http://ftp.iptel.org/pub/sems/debian lenny free deb-src http://ftp.iptel.org/pub/sems/debian lenny free and etch64 in http://ftp.iptel.org/pub/sems/1.1/1.1.1/packages . This is a bugfix release in the 1.1 branch which accumulates fixes for bugs found in 1.1.0 so far. Specifically, this is - fixed Via HF missing the port number in ACK to 200 reply - do not try to scale too short RTP packets - fixed initialization of SSL - caused random crashing of xmlrpc server - fix size() for AmArg struct type - authenticate on both 401 and 407 reply in click2dial - fixed ssl build dependency for DIAMETER client in deb Many Thanks to everyone who contributed with bug reporting and fixes. More information, documentation etc about SEMS can be found at its homepage: http://iptel.org/sems Best Regards Stefan Sayer -- Stefan Sayer VoIP Services stefan.sa...@iptego.com www.iptego.com IPTEGO GmbH Wittenbergplatz 1 10789 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Mediaproxy ver 2.3.4 - Conntrack meets Music on Hold
On 6 Jul 2009, at 19:47, Alex Hermann wrote: On Thursday 02 July 2009 12:33:14 Dan Pascu wrote: Another false assumption is that the receiving endpoint would accept a media stream coming from a source that was not negotiated in the current session. How is the source IP:port negotiated in SDP? AFAIK only the destination IP:port is negotiated, an UA only specifies the address is it willing to _receive_ RTP on. The RTP stream contains both a sequence number and a signature. The signature is the same for the life of the stream, while the sequence number is incremented with each packet. The destination may choose to learn the signature from the first packet and discard any packet not having the same signature, so a newly negotiated RTP stream outside of the dialog will fail. It will also ignore any packet with a sequence number that is lower than the last sequence number, because it thinks it already did process that chunk. The destination may even employ security measures similar to mediaproxy: it can learn the IP address from the 1st packet and refuse any packet from a different IP until the stream is renegotiated using a re-INVITE. Overall, all these issues do not exist if the MOH is delivered through a stream negotiated inside the same dialog using a re-INVITE, but they pose a problem if the MOH comes via a stream from a different dialog. It's lottery: it may work with some particular devices, it may not with others. -- Dan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Is opensips a front end to asterisk?
As said by some of the other members on the list, this IS a very big project. The fact that you are looking for an ISO to test it out, suggests that you don't really understand how it all works. I have it installed and running, just wondered if someone had put out an ISO. ISO's are a great way to see things in action, often helping to get a better view of things when first jumping in. much detail, let me just say that OpenSIPs won't do anything for you. You have to configure it from the ground up. So even if you had an ISO, it would give you binaries and modules that essentially wouldn't do anything at all. And honestly the compilation, installation bits are the easy part! As with anyone that's new to a product, have to ask questions to get a better sense of things. It's a complex project, why would I not ask plenty of questions before spending time on it :). As for this whole Asterisk redundancy. This has been discussed for a long time and there are dozens of ways to do that. Everything from using UCARP, What I'm really trying to get a handle on is how it ties together. I've read a lot of material, have a good sense of what it is, so trying to put the pieces together by asking. What I've not been able to get enough information on is something I read this in voip-info.org; It is flexible and highly configurable but cannot be used to provide media services as voicemail, announcements or conferencing. For such services, Asterisk is the most suitable open source product. I have to assume that once the user is passed on to one of the asterisk servers, that there, they CAN get their media services, as if they connected directly to any other asterisk system. In other words, once they are connected to an asterisk box on the back end, they get the usual services, so what ever we have on those boxes, vm, conf, faxing, etc. Is this correct and I hope I'm explaining this correctly. (You've mentioned something farther in this thread that is interesting, about being better than asterisk) That being said, I can also say that many many people (myself included) have come up with methods that seem to work to provide Right, by sharing tips, tricks and information :). As for pairing OpenSIPs and Asterisk. There are a dozen ways to do that as well.. You can do it on the front end (phones register to opensips), on the My hope would be to simply get things up and running, become familiar and then figure out a longer term picture. Hard to pre-plan everything 100% when starting to use something like this. Like every other technology anyone ever takes on, you learn as you go and you fine tune it as you get to know it better. Sounds like it's very flexible so that's great news of course. I personally try to make asterisk do as absolutely little as possible. I think you'll find scaling OpenSIPs much more enjoyable than Asterisk. ;) Well, this is where I am also trying to get a better handle on what it is/does. It's a media gateway, it's not a pbx, so it always needs to have asterisk in the mix. Can you expand on what you mean above then please. That all being said, you're really only limited by your creativity and skill level with the platforms. It's best to think of them complementing each other. Pick the features you want to use from each. Well, the limits will expand once I get some time with this, ask questions, learn by trying things out etc. How long that takes, hard to tell since this isn't the only thing I'll have going on and my little ol brain can only handle so much stuffing at a time. As for NAT. Sure you can run these platforms behind NAT, but you're asking for a world of pain. :) Just don't do it. That's my opinion. Far end NAT is tricky enough to deal with to also have Near end NAT issues as well. http://en.wikipedia.org/wiki/KISS_principle :) Thought I saw another reply or thread on the list this morning saying that it works just fine using NAT? I've not run anything on a public IP cept routers and firewalls for a heck of a long time, not sure why such a mature project would not work on NAT or how this would be accomplished, safely. Mike ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Is opensips a front end to asterisk?
We all started at the some point as you do now and there was never a magical simple answer that answered the kind of complex question you ask. There is plenty of room to innovate here. Adrian On Jul 7, 2009, at 6:45 PM, li...@grounded.net wrote: I love how joining pretty much any new mailing list and asking initial questions leads to the typical 'you should realize how difficult this is' replies. That's nothing new since there are countless folks who have aspirations without the follow through but not everyone. And really, all of you learned the same way, asking sometimes stupid, but a lot of questions, reading, playing with and getting to know, the software. Well, maybe not the developers :). Anyhow, I'd still love to see some feedback on my original question. Mike ___ 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] Fwd: opensips+asterisk call dropping in 20 seconds
Hi In continuation with the subject when i call intiated from Opensips the call drop in 20seconds but when i register directly from * box i dont see the call drop even for 20-30min of talk any suggestions Ram On Tue, Jun 30, 2009 at 8:35 PM, ram talk2...@gmail.com wrote: On Tue, Jun 30, 2009 at 5:20 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Ram, I found your email on the Asterisk mailing list also ;) So, to answer here also: do you get any reply back from Asterisk ? Hi Bogdan thanks for the reply I have made a quick Fix, iam not sure how far its good. Just put coment in secret , in the Asterisk Additional_a2billing_sip.conf. rather doing twise authentication. But i have another problem here with the Dispatcher, dispatcher sending calls round robin, 1 st call to 1st * 2nd call to 2nd * 3 call to 3rd * if 2nd Asterisk fails to respond still Dispatcher module sending calls to 2nd asterisk how can i fix this issue with Dispatcher, if any one of * box not reachable it should detect and send call to 3rd * if 2nd comes back in to network and live, it should send to 2nd * how can i achive this ? Ram ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Fwd: opensips+asterisk call dropping in 20 seconds
In my opinion the 20 sec drop call is due to a NAT issue, check your NAT setup and or configuration On Tue, Jul 7, 2009 at 1:13 PM, ram-2 (via Nabble) ml-user+92105-174774...@n2.nabble.comml-user%2b92105-174774...@n2.nabble.com wrote: Hi In continuation with the subject when i call intiated from Opensips the call drop in 20seconds but when i register directly from * box i dont see the call drop even for 20-30min of talk any suggestions Ram On Tue, Jun 30, 2009 at 8:35 PM, ram talk2...@...http://n2.nabble.com/user/SendEmail.jtp?type=nodenode=3220575i=0 wrote: On Tue, Jun 30, 2009 at 5:20 PM, Bogdan-Andrei Iancu bog...@...http://n2.nabble.com/user/SendEmail.jtp?type=nodenode=3220575i=1 wrote: Hi Ram, I found your email on the Asterisk mailing list also ;) So, to answer here also: do you get any reply back from Asterisk ? Hi Bogdan thanks for the reply I have made a quick Fix, iam not sure how far its good. Just put coment in secret , in the Asterisk Additional_a2billing_sip.conf. rather doing twise authentication. But i have another problem here with the Dispatcher, dispatcher sending calls round robin, 1 st call to 1st * 2nd call to 2nd * 3 call to 3rd * if 2nd Asterisk fails to respond still Dispatcher module sending calls to 2nd asterisk how can i fix this issue with Dispatcher, if any one of * box not reachable it should detect and send call to 3rd * if 2nd comes back in to network and live, it should send to 2nd * how can i achive this ? Ram ___ Users mailing list us...@...http://n2.nabble.com/user/SendEmail.jtp?type=nodenode=3220575i=2 http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- View message @ http://n2.nabble.com/Fwd%3A-opensips%2Basterisk-call-dropping-in-20-seconds-tp3220575p3220575.html To unsubscribe from OpenSIPS (Open SIP Server), click here (link removed) =. -- View this message in context: http://n2.nabble.com/Fwd%3A-opensips%2Basterisk-call-dropping-in-20-seconds-tp3220575p3220589.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] Fwd: opensips+asterisk call dropping in 20 seconds
On Tue, Jul 7, 2009 at 10:46 PM, cesar.fiestas fiestas.ce...@gmail.comwrote: In my opinion the 20 sec drop call is due to a NAT issue, check your NAT setup and or configuration All are Public IP's any other suggestions Ram ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] pua_xmpp xmpp2simple presence -help
Hi Anca I get the following error in tm module when the xmpp client buddy changes state. As a result the sip client does not get presence notify. This could be a simple config bug or error. I followd exactly what;s in the example config. THe SIP client is promptly getting NOTIFIES for the xmpp buddies's watcher.info all the time. The only problem is that it is not getting the 'preesnce' notify. PLEASE I'm trying to get this working for almost a MONTH now. Any help is appreciated. Just tell me what to look for in the debug msg to make sure the presence 'subscribes' fot the xmpp buddies are going out. WIth the latest code MESSAGES are not woring and even sip 2 xmpp presence is not working. I used the uri as you told. Jul 7 12:13:51 [5046] ERROR:core:parse_uri: uri too short: (0) Jul 7 12:13:51 [5046] ERROR:tm:uri2proxy: bad_uri: Jul 7 12:13:51 [5046] ERROR:tm:uri2su: failed create a dst proxy Jul 7 12:13:51 [5046] ERROR:tm:t_uac: no socket found Jul 7 12:13:51 [5046] ERROR:pua:send_subscribe: while sending request with t_request ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Fwd: opensips+asterisk call dropping in 20 seconds
Hi, You are missing some ACKs in one direction. Looks like you missed some record_route loose_route entries in your config? Wireshark/ngrep is your best friend :-) Good luck BR Uwe ram schrieb: On Tue, Jul 7, 2009 at 10:46 PM, cesar.fiestas fiestas.ce...@gmail.com mailto:fiestas.ce...@gmail.com wrote: In my opinion the 20 sec drop call is due to a NAT issue, check your NAT setup and or configuration All are Public IP's any other suggestions Ram ___ 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] Is opensips a front end to asterisk?
You are right. We all started from the same point and asked questions to learn a lot. The more specific the question is, the better the answer would match. I think your setup is not new, but it depends on your requirement and your setup. BTW: What was the initial question? :) BR Uwe li...@grounded.net schrieb: I love how joining pretty much any new mailing list and asking initial questions leads to the typical 'you should realize how difficult this is' replies. That's nothing new since there are countless folks who have aspirations without the follow through but not everyone. And really, all of you learned the same way, asking sometimes stupid, but a lot of questions, reading, playing with and getting to know, the software. Well, maybe not the developers :). Anyhow, I'd still love to see some feedback on my original question. Mike ___ 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] Is opensips a front end to asterisk?
Specific and well-parameterised questions really are the key. -- Sent from mobile device On Jul 7, 2009, at 2:00 PM, Uwe Kastens ki...@kiste.org wrote: You are right. We all started from the same point and asked questions to learn a lot. The more specific the question is, the better the answer would match. I think your setup is not new, but it depends on your requirement and your setup. BTW: What was the initial question? :) BR Uwe li...@grounded.net schrieb: I love how joining pretty much any new mailing list and asking initial questions leads to the typical 'you should realize how difficult this is' replies. That's nothing new since there are countless folks who have aspirations without the follow through but not everyone. And really, all of you learned the same way, asking sometimes stupid, but a lot of questions, reading, playing with and getting to know, the software. Well, maybe not the developers :). Anyhow, I'd still love to see some feedback on my original question. Mike ___ 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] Fwd: opensips+asterisk call dropping in 20 seconds
On Tue, Jul 7, 2009 at 11:25 PM, Uwe Kastens ki...@kiste.org wrote: Hi, You are missing some ACKs in one direction. Looks like you missed some record_route loose_route entries in your config? Wireshark/ngrep is your best friend :-) thanks for the suggestions iam doing network trace and comparing sure some where i did mistake in the config Ram ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] sip2pstn: P-Asserted-Identity and P-Preferred-Identity
Hi list, This is not exactly a opensips issue. I don't if anybody give me a hint. Until today I was very sure the the P-Asserted-Identity is trusted and the P-Preferred-Identity is untrusted. So it is wise to map the asserted to the pstn number which is the carrier trusted (network provided) and the preferred is a number for clip no screening. I discussed with a vendor which will send me a ddi-number for a pbx as asserted and the main number as preferred. The RFC is not very clear in that point - or did I read the wrong ones. 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] Is opensips a front end to asterisk?
On Tue, 7 Jul 2009 14:02:11 -0400, Alex Balashov wrote: Specific and well-parameterised questions really are the key. I'll certainly do that, once I start understanding the product but for now, I'm just trying to get a handle on basics, not deep in depth understandings, just enough to formulate a plan. Ok, so we all started somewhere, I'm starting here and very much appreciate the input I'll be getting. I'm not going to suck the list dry and leave it, I do want to learn, first, if this is what I need to be working on, and second, learning to use it if it is. I had two questions which I posted. One was asking about the viability of using opensips on ESXi. Because of how easy it is to use snapshots, backup and so on, this would be the best working environment. So my question was, does opensips have any hardware timing requirement issues such as asterisk does. If timing is not critical, as a voip server is, then opensips must run nicely in a virtual manner. Second, (still at the top of this thread). I've come across this project a few times but have been having a bit of a time confirming just what the project does. I thought perhaps the best way would be to join the list and ask. A 'general' question, to assess whether this is what I am looking for or not. Reading is one thing, getting a little input from it's users is the best. I'm not asking for detailed operations, I can get that on the opensips site, just looking for general input. I don't have any numbers to work with, which is why I say scalable. I'm looking for something which can help me to scale a voip based application to many users. So let's say hundreds of users so that we have a number. I know many of you are running many thousands so this should be a good starting point. Right now, the plan calls for every box to have a second one for redundancy. I was planning on manually redirecting connections (for now) but it sounds like opensips could take care of a number of issues. This is how I would have approached this, until I started looking for a sip gateway/load balancer. I have multiple providers (WANs) at one location but was thinking that for highest reliability, that I might have three locations to be safe unless there are better ideas. This should be pretty straight forward to those who have pro setups and want as much reliability as possible. I want to have two separate locations so that I can fail over, simple as that really. One would be the location where the initial user connection is made, such as a proxy/load balancer. -From what I can tell, opensips could act as a pbx on it's own but it can act as a proxy/load balancer/gateway to asterisk systems as well. This is what I asked about in this thread a couple of times now. It's not fully clear to me, even after reading. It sometimes sounds like opensips can be a voip server though it does not provide other media services such as voice mail and so on. I get that it is a gateway but I'm trying to get a better understanding of FROM that point on. From the opensips gateway; -are users registering for services such as in buying services or registering as in authenticating their voip devices, etc. -are users forwarded to asterisk servers (or what ever someone wants to use it for, but in my case, asterisk services), where they can then get their media services such as voice mail, faxing, etc. From what someone posted earlier, I get that you can build what ever you want behind it, in what ever way you wish, such as individual asterisk servers or a distributed (again, your choice of how you do it) farm of them. -Does opensips handle only new incoming connections or could it actually move sessions from a down server to another which is still up? In other words, does it forward the user to say an asterisk server, but, does it handle sessions for example, so that if that asterisk server goes down, opensips can send that user to another asterisk server. -Would I have some method of controlling how many people can register on any one box? What this means is that the media services I'll have on the box bloats the box. It'll have voice mail, faxing, even silly video, but that's what it is, so be it. That means I can't have too many users on any one box at any time so need to be able to limit the number of connections to any one asterisk box. I'm not sure how else to ask these properly formatted questions because I don't quite have all the pieces yet but with some input, I will be starting to. Thanks. Mike ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Is opensips a front end to asterisk?
I'm sorry if I gave you the impression that I was sending you away or telling you it's too hard to handle. That certainly wasn't my intention. Yeah, we all learned the same way. Even now, I have a decent idea what I'm doing and I ask stupid questions all the time! :) The important thing I was trying to iterate is that opensips, asterisk, freeswitch, yate, etc, they are all toolkits. And there isn't just one way to do it. By far the best thing I think you can do is to gain familiarity with the pieces on their own, so you can make a good decision about what parts make sense for you to use for your specific application. The opensips community on the mailing list is a really helpful and friendly bunch. So I hope you continue to post your questions and I hope you can find solutions for your applications. As always you're always more likely to get better help with a more specific question. On Tue, Jul 7, 2009 at 11:45 AM, li...@grounded.net li...@grounded.netwrote: I love how joining pretty much any new mailing list and asking initial questions leads to the typical 'you should realize how difficult this is' replies. That's nothing new since there are countless folks who have aspirations without the follow through but not everyone. And really, all of you learned the same way, asking sometimes stupid, but a lot of questions, reading, playing with and getting to know, the software. Well, maybe not the developers :). Anyhow, I'd still love to see some feedback on my original question. Mike ___ 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] Opensips on OS X?
Hi all, Just a quick question. Before I try to convince some guys that Debian would be MUCH easier for Opensips. Anyone tried OS X, was it easy installing/compiling Opensips for OSX? Regards, Matti Zemack, Stockholm, Sweden ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Can OpenXCAP run on Debina Lenny? (I get an error)
Hi, I'm trying to run OpenXCAP in Debian Lenny 64 bits. Due to dependency
version requeriments I had to install two Python libraries from
Debian Sid, but they were installed succesfully.
However, when I start OpenXCAP I get the following error:
--
Jul 7 23:03:55 myserver openxcap[2657]: Log opened.
Jul 7 23:03:55 myserver openxcap[2657]: Starting OpenXCAP 1.1.0
Jul 7 23:03:56 myserver openxcap[2657]: error: Traceback (most recent call
last):
Jul 7 23:03:56 myserver openxcap[2657]: error: File /usr/bin/openxcap,
line 55, in module
Jul 7 23:03:56 myserver openxcap[2657]: error: from xcap.server import
XCAPServer
Jul 7 23:03:56 myserver openxcap[2657]: error: File
/usr/lib/pymodules/python2.5/xcap/server.py, line 17, in module
Jul 7 23:03:56 myserver openxcap[2657]: error: from xcap import
authentication
Jul 7 23:03:56 myserver openxcap[2657]: error: File
/usr/lib/pymodules/python2.5/xcap/authentication.py, line 22, in module
Jul 7 23:03:56 myserver openxcap[2657]: error: from xcap.appusage import
getApplicationForURI, namespaces
Jul 7 23:03:56 myserver openxcap[2657]: error: File
/usr/lib/pymodules/python2.5/xcap/appusage/__init__.py, line 54, in module
Jul 7 23:03:56 myserver openxcap[2657]: error:
configuration.read_settings('Server', ServerConfig)
Jul 7 23:03:56 myserver openxcap[2657]: error: File
/usr/lib/pymodules/python2.5/xcap/config.py, line 108, in read_settings
Jul 7 23:03:56 myserver openxcap[2657]: error: setattr(cls, prop, value)
Jul 7 23:03:56 myserver openxcap[2657]: error: File
/usr/lib/pymodules/python2.5/application/configuration/__init__.py, line 70,
in __setattr__
Jul 7 23:03:56 myserver openxcap[2657]: error:
cls.__settings__[attr].__set__(None, value)
Jul 7 23:03:56 myserver openxcap[2657]: error: File
/usr/lib/pymodules/python2.5/application/configuration/__init__.py, line 29,
in __set__
Jul 7 23:03:56 myserver openxcap[2657]: error: value = self.type(value)
Jul 7 23:03:56 myserver openxcap[2657]: error: File
/usr/lib/pymodules/python2.5/xcap/appusage/__init__.py, line 37, in __new__
Jul 7 23:03:56 myserver openxcap[2657]: error: value = value.lower()
Jul 7 23:03:56 myserver openxcap[2657]: error: AttributeError: 'module' object
has no attribute 'lower'
--
I've installed OpenXCAP via apt by following instructions in
http://openxcap.org/wiki/Installation.
The doc says:
---
OpenXCAP has been tested on Debian unstable with the following software
versions:
libxml2: 2.6.32.dfsg-2
python: 2.5.2-1
python-application: 1.1.0
python-gnutls: 1.1.8
python-lxml: 2.0.7-1
python-sqlobject: 0.10.1-1
python-twisted-core: 8.0.1-1
python-twisted-web: 8.0.0-1
python-twisted-web2: 8.0.1-1
python-zopeinterface: 3.3.1-6
All those packages have a greater version in Debian Lenny so
I don't see issues with it.
These are my Python related installed packages:
-
ii python 2.5.2-3
ii python-application 1.1.2
ii python-central 0.6.8
ii python-codespeak-lib0.9.1-3
ii python-crypto
2.0.1+dfsg1-2.3+lenny0
ii python-dns 2.3.3-2
ii python-dnspython1.6.0-1.1
ii python-docutils 0.5-2
ii python-elementtree 1.2.6-12
ii python-eventlet 0.8.10
ii python-formencode 1.0.1-1
ii python-fpconst 0.7.2-4
ii python-gnutls 1.1.8-1
ii python-lxml 2.1.1-2.1
ii python-minimal 2.5.2-3
ii python-msrplib 0.10.0
ii python-mysqldb 1.2.2-7
ii python-openssl 0.7-2
ii python-pam 0.4.2-12
ii python-pkg-resources0.6c8-4
ii python-pyopenssl0.7-2
ii python-roman0.5-2
ii python-serial 2.3-1
ii
Re: [OpenSIPS-Users] Can OpenXCAP run on Debina Lenny? (I get an error)
The error seems related to .lower() method: error: File /usr/lib/pymodules/python2.5/xcap/appusage/__init__.py, line 37, in __new__ error: value = value.lower() error: AttributeError: 'module' object has no attribute 'lower' Line 37 is: --- 34 class Backend(object): 35 Configuration datatype, used to select a backend module from the configuration file. 36 def __new__(typ, value): 37 value = value.lower() --- However, /usr/bin/openxcap uses /usr/bin/python which, in my system, is a link to /usr/bin/python2.6. Anyhow, QwQsASweWEW.lower() works perfectly by executing it with python2.5 or python2.6, so I can't get the issue. -- 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] Can OpenXCAP run on Debina Lenny? (I get an error)
El Martes, 7 de Julio de 2009, Iñaki Baz Castillo escribió: However, /usr/bin/openxcap uses /usr/bin/python which, in my system, is a link to /usr/bin/python2.6. Anyhow, QwQsASweWEW.lower() works perfectly by executing it with python2.5 or python2.6, so I can't get the issue. Sorry, I meant: -- However, /usr/bin/openxcap uses /usr/bin/python which, in my system, is a link to /usr/bin/python2.5. Anyhow, QwQsASweWEW.lower() works perfectly by executing it with python2.5 or python2.4, so I can't get the issue. - However, if I run OpenXCAP with python2.4 (2.4.6-1) it runs !!! Doesn't it require Python 2.5? -- Iñaki Baz Castillo i...@aliax.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] [OpenXCAP] Problem with EyeBeam: Document selector does not contain context
Hi, OpenXCAP root is: root = http://xcap.mydomain.net/xcap-root I've configured EyeBem (1.5.19) with a SIP account sip:eyeb...@mydomain.net to use this URL for XCAP: http://xcap.mydomain.net:443/xcap-r...@mydomain.net/ I've chosen it by following the Configuration section in: http://openxcap.org/wiki/Installation I do know that this is a good workaround to avoid the need of a SSL certificate for each domain in a multidomain SIP server. However, the requests sent by Eyebem look like: GET /xcap-r...@mydomain.net/eyebeam.mydomain.net.pres-rules HTTP/1.1 GET /xcap-r...@oversip.net/eyebeam.mydomain.net.resource-list.xml HTTP/1.1 so the server replies a 404 with body: Document selector does not contain context: 'eyebeam.mydomain.net.pres-rules' By readin XCAP related RFC's, I understand that the URL sent by EyeBeam is wrong since it should be something as: GET /xcap-r...@mydomain.net/users/eyebeam/pres-rules HTTP/1.1 or GET /xcap-r...@mydomain.net/users/eyeb...@mydomain.net/pres-rules HTTP/1.1 Am I wrong? is this implementation of XCAP wrong in this version of EyeBeam? Thanks. -- 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] [OpenXCAP] Problem with EyeBeam: Document selector does not contain context
El Miércoles, 8 de Julio de 2009, Iñaki Baz Castillo escribió: By readin XCAP related RFC's, I understand that the URL sent by EyeBeam is wrong since it should be something as: GET /xcap-r...@mydomain.net/users/eyebeam/pres-rules HTTP/1.1 or GET /xcap-r...@mydomain.net/users/eyeb...@mydomain.net/pres-rules HTTP/1.1 Am I wrong? is this implementation of XCAP wrong in this version of EyeBeam? And now I try Zoiper and sends the following request: GET /xcap-r...@mydomain.net/ HTTP/1.1 So OpenXCAP replies: Document selector does not contain auid But... does nobody read the fuck*** specifications before implementing something??? -- Iñaki Baz Castillo i...@aliax.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] [OpenXCAP] About root parameter
Hi, if I set: root = http://xcap.mydomain.net/xcap-root then the UA could be configured with: XCAP: http://xcap.mydomain.net/xcap-r...@mydomain.net (for the cases in which the UA justs sets /users/username) But, what about if I set in OpenXCAP the following?: root = http://xcap.mydomain.net Could then configure the UA wiht: XCAP: http://xcap.mydomain.net/@mydomain.net ? -- 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] [OpenXCAP] Problem with EyeBeam: Document selector does not contain context
El Miércoles, 8 de Julio de 2009, Iñaki Baz Castillo escribió: However, the requests sent by Eyebem look like: GET /xcap-r...@mydomain.net/eyebeam.mydomain.net.pres-rules HTTP/1.1 GET /xcap-r...@oversip.net/eyebeam.mydomain.net.resource-list.xml HTTP/1.1 Opss, by error I selected WebDAV storage instead of XCAP... -- Iñaki Baz Castillo i...@aliax.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] [OpenXCAP] /users/sip:al...@domain or /users/al...@domain ?
Hi, when EyeBeam is configured to use SIP credentials for XCAP it generates an URL like: /xcap-root/resource-lists/users/sip:al...@domain/resource-list.xml but I think I've seen in some RFC's the following: /xcap-root/resource-lists/users/al...@domain/resource-list.xml Would OpenXCAP allow both cases? -- 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] Is NAPTR query always necessary?
Hi Bogdan, I use multi-domain support, but if I comment the multi-domain setting in opensips.cfg the looping still happen. Would I add something in my opensips.cfg to avoid the looping? Thanks, Eason Bogdan-Andrei Iancu wrote: Hi Eason , the looping happens because your OpenSIPS does not recognize the domain from RURI as local domain (to be processed locally) and simply fwds it based on DNS. In your script, do you use multi-domain support? Is it based on the default opensips cfg ? Regards, Bogdan hsuan wrote: Hi Bogdan, Thanks for your feedback. Yes, I can find the sip request to itself on LO interface. But I don't know what setting in opensips server will cause SIP looping. Could you spell out more precisely? Why the sip looping only happens in the domain is not numeric? Do you mean it's not the DNS issue? I am a little confused. Best regards, Eason Bogdan-Andrei Iancu wrote: Hi Eason, according to RFC3263 ( SIP: Locating SIP Servers), if the explicit protocol and port is contain by the destination SIP URI, the proxy must try to do NAPTR and SRV lookups to discover what protocol and port to use... If these records are missing, the default value ( UDP + 5060) are assumed. these DNS lookups are done each time a requests is routed out (if not using IPs). What happens in your case is a SIP looping on the proxy (sending the request to itself). You can check this with tcpdump/ngrep/wireshark , watching the traffic on LO interface. Regards, Bogdan hsuan wrote: Hi all, I am newbie on opensips, I wonder if the NAPTR is necessary in some case? When my sip domain is set to FQDN, the opensips server will send NAPTR query to DNS server. But there is no NAPTR supported DNS in my environment, although the opensips server will get the ip and port eventaully. Why the opensips server will send NAPTR query again and again even though it already get the ip and port from DNS? When the opensips server get into the NAPTR loop, the user will get 483 too many hops finally and will not able to register to opensips server. Once if the sip domain is set to ip address, the user will register successfully. Is there any way to disable the NAPTR query in opensips server? Please advice. Best regards, Eason ___ 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 -- View this message in context: http://n2.nabble.com/Is-NAPTR-query-always-necessary--tp3187819p3222991.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] Database Authentication Problem
On Wed, Jul 8, 2009 at 7:16 AM, Nathaniel L Keeling
keel...@akan-tech.comwrote:
I am having a problem with database authentication and would like some
help. I have everything setup for the database and have added a user.
The user was added using opensipsctl. The setting for the database is to
not use plain text passwords. If I add the password to the database
entry, the user will authenticate. I feel that there is a configuration
problem but don't know where. Here is the extract from my config file
/* uncomment the following lines if you want to enable the DB based
authentication */
modparam(auth_db, calculate_ha1, yes)
modparam(auth_db, password_column, password)
modparam(auth_db, db_url,
postgres://opensips:opensip...@localhost/opensips)
modparam(auth_db, load_credentials, )
/* uncomment the following line if you want to enable multi-domain support
in the modules (dafault off) */
modparam(alias_db|auth_db|usrloc|uri_db, use_domain, 1)
From opensipsctlrc and osipsconsolerc
## do (1) or don't (0) store plaintext passwords
## in the subscriber table - default '1'
STORE_PLAINTEXT_PW=0
Hi
apart from that you need to edit some more information like
# - 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)
# - uri_db params -
/* by default we disable the DB support in the module as we do not need it
in this configuration */
modparam(uri_db, use_uri_table, 0)
modparam(uri_db, db_url, mysql://opensips:opensip...@localhost
/opensips)
and
# generated by local subscriber (domain from FROM URI is local)
if (!(method==REGISTER) from_uri==myself) /*no multidomain
version*/
if (!(method==REGISTER) is_from_local()) /*multidomain
version*/
{
if (!proxy_authorize(, subscriber)) {
proxy_challenge(, 0);
exit;
}
if (!check_from()) {
sl_send_reply(403,Forbidden auth ID);
exit;
}
consume_credentials();
# caller authenticated
}
Or else post full config to suggest better
Ram
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