Hello, is there a way in FS to determine the IP address of the sender of a SIP message. I'm not talking about IP addresses or domain names that are sent on application layer (i. e. within SIP) but on network layer. I need this to determine if one network element pretends to have several (SIP-)identities. Alois Komenda Fraunhofer-Einrichtung für Systeme der Kommunikationstechnik ESK
This is available in the dial-plan as the network_addr. Please note that if this first goes through a proxy, this will show the proxy address (the address that we actually get the sip packet from on the ip layer). Mike On Apr 7, 2008, at 11:39 AM, Alois Komenda wrote:
Hello,
is there a way in FS to determine the IP address of the sender of a SIP message. I'm not talking about IP addresses or domain names that are sent on application layer (i. e. within SIP) but on network layer. I need this to determine if one network element pretends to have several (SIP-)identities.
Alois Komenda Fraunhofer-Einrichtung für Systeme der Kommunikationstechnik ESK
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If you do use a proxy, you can populate a sip X-header with the IP and that will be available to you via the channel variables in FS. On Mon, Apr 7, 2008 at 10:48 AM, Michael Jerris <mike@jerris.com> wrote:
This is available in the dial-plan as the network_addr. Please note that if this first goes through a proxy, this will show the proxy address (the address that we actually get the sip packet from on the ip layer). Mike
On Apr 7, 2008, at 11:39 AM, Alois Komenda wrote:
Hello,
is there a way in FS to determine the IP address of the sender of a SIP message. I'm not talking about IP addresses or domain names that are sent on application layer (i. e. within SIP) but on network layer. I need this to determine if one network element pretends to have several (SIP-)identities.
Alois Komenda Fraunhofer-Einrichtung für Systeme der Kommunikationstechnik ESK
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Check out the caller profile, theres a ton of info in there including the IP address of the UA/Proxy sending your box the messages K From: Alois Komenda <alois.komenda@esk.fraunhofer.de> Reply-To: <freeswitch-users@lists.freeswitch.org> Date: Mon, 7 Apr 2008 17:39:42 +0200 To: <freeswitch-users@lists.freeswitch.org> Subject: [Freeswitch-users] Determine IP address of sender Hello, is there a way in FS to determine the IP address of the sender of a SIP message. I'm not talking about IP addresses or domain names that are sent on application layer (i. e. within SIP) but on network layer. I need this to determine if one network element pretends to have several (SIP-)identities. Alois Komenda Fraunhofer-Einrichtung für Systeme der Kommunikationstechnik ESK _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi all, I have got one more question, I'm not able to find in docs: How could I dynamicaly add/change params/remove SIP gateways I have to register with (and the others - I shouldn't register with - too :-)? Like I see, gateways are defined in Sofia profiles - XML files. Are these files read periodically (at best on all call processing)? Or is there a way how to get mod_xml_curl into play? Or something completly different? :-) Any hints are very appreciated! Regards, kokoska.rokoska
Hi Kukoska, I did this using mod_xml_curl, which worked fine: you also need to poke appropriate commands in to FreeSWITCH to have it reload the profiles for whichever gateways you change. Cheers -- Dave
Hi all,
I have got one more question, I'm not able to find in docs:
How could I dynamicaly add/change params/remove SIP gateways I have to register with (and the others - I shouldn't register with - too :-)?
Like I see, gateways are defined in Sofia profiles - XML files. Are these files read periodically (at best on all call processing)? Or is there a way how to get mod_xml_curl into play? Or something completly different? :-)
Any hints are very appreciated!
Regards,
kokoska.rokoska
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note that reloading a profile will abort all active calls in that profile. On Mon, Apr 7, 2008 at 7:52 PM, David Knell <dave@3c.co.uk> wrote:
Hi Kukoska,
I did this using mod_xml_curl, which worked fine: you also need to poke appropriate commands in to FreeSWITCH to have it reload the profiles for whichever gateways you change.
Cheers --
Dave
Hi all,
I have got one more question, I'm not able to find in docs:
How could I dynamicaly add/change params/remove SIP gateways I have to register with (and the others - I shouldn't register with - too :-)?
Like I see, gateways are defined in Sofia profiles - XML files. Are these files read periodically (at best on all call processing)? Or is there a way how to get mod_xml_curl into play? Or something completly different? :-)
Any hints are very appreciated!
Regards,
kokoska.rokoska
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Jonas Gauffin napsal(a):
note that reloading a profile will abort all active calls in that profile.
Thank you very much, Jonas, for your answer! If reload abort active calls, than it is useless for me. With - may be - hundreds of active calls and tens of gw changes per minute I need to find another solution. Any hint is very appreciated :-) BTW: Even if I kill only one active call per year it is not good practice :-) Best regards, kokoska.rokoska
On Apr 7, 2008, at 3:57 PM, kokoska rokoska wrote:
f reload abort active calls, than it is useless for me. With - may be - hundreds of active calls and tens of gw changes per minute I need to find another solution. Any hint is very appreciated :-)
What could possibly cause that many gateway changes? Mike
Michael Jerris napsal(a):
On Apr 7, 2008, at 3:57 PM, kokoska rokoska wrote:
f reload abort active calls, than it is useless for me. With - may be - hundreds of active calls and tens of gw changes per minute I need to find another solution. Any hint is very appreciated :-)
What could possibly cause that many gateway changes?
Mike
Thousands of users which define (add/edit/remove) their own connections to various VoIP operators. I know this is not very comfortable for the server, but I can do nothing with it. For now the peak "frequency" of gw changes is slightly below 2 changes per minute, but I'm trying to be prepared for the future :-) Best regards, kokoska.rokoska
The only way to do what you want is to make one sip profile per customer. FreeSWITCH sofia profiles are meant as an instance of a single UA and the gateways are alternated credential sets for that UA so in a hosted scenario if you want to let customers have their own gateways, you give them each their own FQHN, setup SRV records for this domain and store all of his gateways in that user's user directory. That means 1 sofia profile with 1 ip/port. This is the only way what you want will be supported because you can restart each sofia profile independently. There is really no discussion as to an alternative because this is by design and I don't intend to change it. On Mon, Apr 7, 2008 at 4:41 PM, kokoska rokoska <kokoska.rokoska@post.cz> wrote:
Michael Jerris napsal(a):
On Apr 7, 2008, at 3:57 PM, kokoska rokoska wrote:
f reload abort active calls, than it is useless for me. With - may be - hundreds of active calls and tens of gw changes per minute I need to find another solution. Any hint is very appreciated :-)
What could possibly cause that many gateway changes?
Mike
Thousands of users which define (add/edit/remove) their own connections to various VoIP operators. I know this is not very comfortable for the server, but I can do nothing with it.
For now the peak "frequency" of gw changes is slightly below 2 changes per minute, but I'm trying to be prepared for the future :-)
Best regards,
kokoska.rokoska
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Anthony Minessale napsal(a):
The only way to do what you want is to make one sip profile per customer. FreeSWITCH sofia profiles are meant as an instance of a single UA and the gateways are alternated credential sets for that UA so in a hosted scenario if you want to let customers have their own gateways, you give them each their own FQHN, setup SRV records for this domain and store all of his gateways in that user's user directory.
Thank you, Anthony, for very useful help! I have two more questions: 1. How about if I want to allow to some group of users to share gateways? If I catch it correctly, I should create faked user and map gws to him. Am I rigthy? Or exist another/better approach? 2. Registers (and recieves from/ sonds to) Fresswitch with this user-gateways even if the user isn't registerd with Freeswitch? It is not clear to me...
That means 1 sofia profile with 1 ip/port.
OK. But what do you think about, let say, 10.000 of sofia profiles? Isn't it impacted to negatively performance of Freeswitch? I have no idea whats happen, so I ask...
This is the only way what you want will be supported because you can restart each sofia profile independently.
This is exactly what I need. Thank you!
There is really no discussion as to an alternative because this is by design and I don't intend to change it.
OK. I should live with it :-) <off-topic> BTW: Freeswitch is, from my little knowledge, amazing piece of SW and I like the philosiphy much more than Asterisk's one I'm using now. And - in addition - nearly all my changes to Asterisk more or less "emulate" Freeswitch behaviour. But I made them in the past when I know nothing about Freeswitch :-) Whay talk about it? Because the only one thing I'm now in trouble with Freeswitch is "dynamic gateways". And I hope that above outlined solution will work for me and I can utilize all Freeswitch power I see in. </off-topic> Best regards, kokoska.rokoska
I would also love to be able to switch gateways at runtime without restarting the profiles. Anthony: wont a bounty help either? On Tue, Apr 8, 2008 at 9:03 AM, kokoska rokoska <kokoska.rokoska@post.cz> wrote:
Anthony Minessale napsal(a):
The only way to do what you want is to make one sip profile per customer. FreeSWITCH sofia profiles are meant as an instance of a single UA and the gateways are alternated credential sets for that UA so in a hosted scenario if you want to let customers have their own gateways, you give them each their own FQHN, setup SRV records for this domain and store all of his gateways in that user's user directory.
Thank you, Anthony, for very useful help!
I have two more questions: 1. How about if I want to allow to some group of users to share gateways? If I catch it correctly, I should create faked user and map gws to him. Am I rigthy? Or exist another/better approach? 2. Registers (and recieves from/ sonds to) Fresswitch with this user-gateways even if the user isn't registerd with Freeswitch? It is not clear to me...
That means 1 sofia profile with 1 ip/port.
OK. But what do you think about, let say, 10.000 of sofia profiles? Isn't it impacted to negatively performance of Freeswitch? I have no idea whats happen, so I ask...
This is the only way what you want will be supported because you can restart each sofia profile independently.
This is exactly what I need. Thank you!
There is really no discussion as to an alternative because this is by design and I don't intend to change it.
OK. I should live with it :-)
<off-topic> BTW: Freeswitch is, from my little knowledge, amazing piece of SW and I like the philosiphy much more than Asterisk's one I'm using now. And - in addition - nearly all my changes to Asterisk more or less "emulate" Freeswitch behaviour. But I made them in the past when I know nothing about Freeswitch :-) Whay talk about it? Because the only one thing I'm now in trouble with Freeswitch is "dynamic gateways". And I hope that above outlined solution will work for me and I can utilize all Freeswitch power I see in. </off-topic>
Best regards,
kokoska.rokoska
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Jonas Gauffin napsal(a):
I would also love to be able to switch gateways at runtime without restarting the profiles.
Anthony: wont a bounty help either?
I like the concept of independent profiles, but switching gws without restarting profile will be very cool. If bounty will be setted up, I try to unpocket some money :-) Best regards, kokoska.rokoska
My concern is with detribalizing things. The idea of a bounty does hold some weight because things I don't want to do are the exact things that should have a bounty. There are a lot of moving parts in a gateway with many threads involved which is why it's a tricky situation. To what extent do you want to manipulate them? There are a lot of parameters to a gateway so 1 giant api call to create them seems daunting. Re-parsing the config assuming there are new ones has the issue of where to look because we have the gateways that live in the users and the ones that live in the profile and when you do that how do you know when to erase them. If anything, all I can see is being able to add more by rereading the config or maybe a very elaborate FSAPI call but only being able to *remove* them 1 by 1 via FSAPI. See the issue? P.S. What kind of bounty are we talking about? On Tue, Apr 8, 2008 at 2:38 AM, kokoska rokoska <kokoska.rokoska@post.cz> wrote:
Jonas Gauffin napsal(a):
I would also love to be able to switch gateways at runtime without restarting the profiles.
Anthony: wont a bounty help either?
I like the concept of independent profiles, but switching gws without restarting profile will be very cool. If bounty will be setted up, I try to unpocket some money :-)
Best regards,
kokoska.rokoska
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Anthony Minessale napsal(a):
My concern is with detribalizing things.
The idea of a bounty does hold some weight because things I don't want to do are the exact things that should have a bounty.
:-)
There are a lot of moving parts in a gateway with many threads involved which is why it's a tricky situation.
I don't know how exactly Freeswitch internaly works, so thank you, Anthony, for explanation.
To what extent do you want to manipulate them? There are a lot of parameters to a gateway so 1 giant api call to create them seems daunting.
I agree.
Re-parsing the config assuming there are new ones has the issue of where to look because we have the gateways that live in the users and the ones that live in the profile and when you do that how do you know when to erase them.
One possibility is to introduce mandatory unique identifier for each gw in config.
If anything, all I can see is being able to add more by rereading the config or maybe a very elaborate FSAPI call but only being able to *remove* them 1 by 1 via FSAPI.
Removing doesn't help me :-)
See the issue?
Yes. ---------------------------- The main reason I talk about it is that I don't want to stop calls during profile reload/restart/etc. Because it practicaly eliminates possibility of sharing one profile between more users. If there is/will be a way how to reload/restart sip profile and keep current calls intact, I don't care about I could not setup new ones until reload/restart finishes. The same way I have tried to outline I use lcr module from OpenSER http://www.openser.org/docs/modules/1.3.x/lcr.html And, from my opinion, even Asterisk doesn't interrupt calls in case of chan_sip reload. All at all, I just want to manipulate with sip gws and keep calls alive :-) If you be so kind, explain to me please, if it is possible to incorporate it into Freeswitch or not. I have no idea :-) BTW: If this feature breaks Freeswitch clean desing (or generally you see it as ugly hack) it is better - from my point of view - to leave thing as they are... Thanks once more, Anthony, for your clarification! Bets regards, kokoska.rokoska
Brian West napsal(a):
The gateway name has to already be unique.
Of course :-) But the names could change between config reloads - I think it is Anthony talked about. So gws could be recognized by names and the names of gws could not be changed or let's add another constant indetifier :-) Best regards, kokoska.rokoska
Asterisk does let you reload, and it *tries* not to interrupt the calls but it's like a game of Russian Roulette, because at some point for sure you *will* deadlock and/or seg fault when you reload. There is an actual unavoidable race condition in many of the things that they made reloadable that I have identified myself as a long time developer for the project and to this day still exist and are a major bullet point to address on their roadmap for future releases. 1.8 maybe? Our philosophy on reload is simple and strict. Each module is responsible independently for it's own ability to reload. In the case of sofia, it was a year after it was written until I introduced the profile restart command because it's quite complicated and introduces a great challenge in stability. Nearly all of the elements in a sofia profile are things you must stop the profile to change anyway. There are a few innocuous params that could be changed while it's running so I will say there is a possibility to make a sofia reload and a sofia profile reload that unlike profile restart looks for profiles that do not yet exist and brings them online and when they do exist only changes the params that do not require a full restart of the sip engine (context to use, dialplan, moh prefs vs bind url, sip specific options that cannot be changed). As part of this process the profiles could be rescanned for new gateways that do not already exist and bring them online if they do not already exist. Removing them would require a full restart of that profile. This idea I am willing to entertain but in the light of my horribly busy schedule and the fact that the patch needs near surgical precision to avoid tainting our release candidate state of stability. Not to mention I have coded nearly 20,000 lines of mod_sofia pro bono providing every SIP feature anyone can ask for, I would like to see the bounty for it first. =D On Wed, Apr 9, 2008 at 7:16 AM, kokoska rokoska <kokoska.rokoska@post.cz> wrote:
Brian West napsal(a):
The gateway name has to already be unique.
Of course :-) But the names could change between config reloads - I think it is Anthony talked about. So gws could be recognized by names and the names of gws could not be changed or let's add another constant indetifier :-)
Best regards,
kokoska.rokoska
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Anthony Minessale napsal(a):
Asterisk does let you reload, and it *tries* not to interrupt the calls but it's like a game of Russian Roulette, because at some point for sure you *will* deadlock and/or seg fault when you reload.
Yes, Anthony, I heard/red about it a lot :-) But form general user point of view it is better to have a chance it will work (calls stay alive) than have guarantee it will not work (calls go down). The worst can happen is to end up in same situation - all calls are dropped.
There is an actual unavoidable race condition in many of the things that they made reloadable that I have identified myself as a long time developer for the project and to this day still exist and are a major bullet point to address on their roadmap for future releases. 1.8 maybe?
Yes, I know you were/are Asterisk developer. And I realise that Asterisk without your work on ARA remain a toy. And, of course, I'm very thankful to you!
Our philosophy on reload is simple and strict. Each module is responsible independently for it's own ability to reload. In the case of sofia, it was a year after it was written until I introduced the profile restart command because it's quite complicated and introduces a great challenge in stability. Nearly all of the elements in a sofia profile are things you must stop the profile to change anyway.
OK. Thank you very much, Anthony, for explanation. I didn't know that.
There are a few innocuous params that could be changed while it's running so I will say there is a possibility to make a sofia reload and a sofia profile reload that unlike profile restart looks for profiles that do not yet exist and brings them online and when they do exist only changes the params that do not require a full restart of the sip engine (context to use, dialplan, moh prefs vs bind url, sip specific options that cannot be changed).
Thank you for explanation again! But I'm interested only in "sip specific" parts of profile :-(
As part of this process the profiles could be rescanned for new gateways that do not already exist and bring them online if they do not already exist. Removing them would require a full restart of that profile.
OK. Thanks for the info!
This idea I am willing to entertain but in the light of my horribly busy schedule and the fact that the patch needs near surgical precision to avoid tainting our release candidate state of stability. Not to mention I have coded nearly 20,000 lines of mod_sofia pro bono providing every SIP feature anyone can ask for
Anthony, I realy respect all work you have done on Asterisk, FreeSWITCH, Sofia etc and I appreciate your help to me and all users too. And, in particular, I just try to become FreeSWITCH user and thus looking for improvements helpful for everybody, I'm not your enemy.
I would like to see the bounty for it first. =D
OK. Give me a while :-) Best regards, kokoska.rokoska
Hi all again :-) I would like to ask if is there any way how to "propagate" variables (variables setted up on curl directory lookup, sip headers etc.) to dialplan xml_curl post. All my processing before dialplan generation couldn't be done without them. Any tips are welcome! Best regards, kokoska.rokoska
All of the variables you care to know about are in the post-data on the http request. On Mon, Apr 7, 2008 at 3:04 PM, kokoska rokoska <kokoska.rokoska@post.cz> wrote:
Hi all again :-)
I would like to ask if is there any way how to "propagate" variables (variables setted up on curl directory lookup, sip headers etc.) to dialplan xml_curl post. All my processing before dialplan generation couldn't be done without them.
Any tips are welcome!
Best regards,
kokoska.rokoska
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Anthony Minessale napsal(a):
All of the variables you care to know about are in the post-data on the http request.
Thank you very much, Anthony, for your answer! I didn't try it and there are no docs discribing what variables - except basic ones - are send. Or, better to say, I wasn't able to find right document. Let me appologize for my stupid question. For near future I will remember: try twice and than ask :-) Thanks again for notable help! Best regards, kokoska.rokoska
if you have routed a call to the "info" app, the data seen there is almost identical to what you will get as the post data. On Mon, Apr 7, 2008 at 4:47 PM, kokoska rokoska <kokoska.rokoska@post.cz> wrote:
Anthony Minessale napsal(a):
All of the variables you care to know about are in the post-data on the http request.
Thank you very much, Anthony, for your answer!
I didn't try it and there are no docs discribing what variables - except basic ones - are send. Or, better to say, I wasn't able to find right document. Let me appologize for my stupid question. For near future I will remember: try twice and than ask :-)
Thanks again for notable help!
Best regards,
kokoska.rokoska
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Anthony Minessale napsal(a):
if you have routed a call to the "info" app, the data seen there is almost identical to what you will get as the post data.
Thank you, Anthony, for very useful tip! "Info" app is my best friend and now it will be even better :-) Best regards, kokoska.rokoska
Hi all, Freeswitch has ipv6 support, I was unable to find any information? Thanks
David Knell napsal(a):
Hi Kukoska,
I did this using mod_xml_curl, which worked fine: you also need to poke appropriate commands in to FreeSWITCH to have it reload the profiles for whichever gateways you change.
Cheers --
Dave
Thank you very much, David, for your answer! I'm afraid I didn't catch it :-) Could you explain it to me a little bit more detailed? In what case (and when) Fresswitch do the curl lookup on gateways? On every call? And how this https-post lookes like? Thanks once more, regards kokoska.rokoska
The variable ${remote_addr} has the true network address that send the request. /b On Apr 7, 2008, at 10:39 AM, Alois Komenda wrote:
Hello,
is there a way in FS to determine the IP address of the sender of a SIP message. I'm not talking about IP addresses or domain names that are sent on application layer (i. e. within SIP) but on network layer. I need this to determine if one network element pretends to have several (SIP-)identities.
Alois Komenda Fraunhofer-Einrichtung für Systeme der Kommunikationstechnik ESK
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Brian West sip:brian@freeswitch.org
Yah I mean to say network_addr and not remote_addr.. I have apache on the brain. /b On Apr 7, 2008, at 10:39 AM, Alois Komenda wrote:
Hello,
is there a way in FS to determine the IP address of the sender of a SIP message. I'm not talking about IP addresses or domain names that are sent on application layer (i. e. within SIP) but on network layer. I need this to determine if one network element pretends to have several (SIP-)identities.
Alois Komenda Fraunhofer-Einrichtung für Systeme der Kommunikationstechnik ESK
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Brian West sip:brian@freeswitch.org
<badjoke>Because you've been... scalped?</badjoke> On Mon, 2008-04-07 at 10:56 -0500, Brian West wrote:
Yah I mean to say network_addr and not remote_addr.. I have apache on the brain.
/b
On Apr 7, 2008, at 10:39 AM, Alois Komenda wrote:
Hello,
is there a way in FS to determine the IP address of the sender of a SIP message. I'm not talking about IP addresses or domain names that are sent on application layer (i. e. within SIP) but on network layer. I need this to determine if one network element pretends to have several (SIP-)identities.
Alois Komenda Fraunhofer-Einrichtung für Systeme der Kommunikationstechnik ESK
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Brian West sip:brian@freeswitch.org
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participants (11)
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Alois Komenda
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Anthony Minessale
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Brian West
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Dale Thatcher
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David Knell
-
echenique
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Jonas Gauffin
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Ken Rice
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kokoska rokoska
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Michael Jerris
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Shannon Price