Re: [SR-Users] Syntax issue?
Additionally, there's no other way than implementing dialog module to keep a variable between the beginning and the end of a call? Regards, Igor. De : Igor Potjevlesch [mailto:igor.potjevle...@gmail.com] Envoyé : lundi 16 février 2015 18:36 À : 'Kamailio (SER) - Users Mailing List' Objet : RE: [SR-Users] Syntax issue? Thank you guys, I will try this. I misunderstood the notion of transaction. I was thinking that it was the whole call-flow. Regards, Igor. De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de Muhammad Shahzad Envoyé : lundi 16 février 2015 18:27 À : Kamailio (SER) - Users Mailing List Objet : Re: [SR-Users] Syntax issue? As far as i know AVPs are transaction specific only. So they will be deleted as soon as transaction is over, i.e. 200 OK for INVITE is received for example. They will not be available in in-dialog transactions such as ACK, or BYE etc. What you need is to set dialog variable instead, see more info here, http://kamailio.org/docs/modules/4.2.x/modules/dialog.html#idp4202736 Thank you. On Mon, Feb 16, 2015 at 6:08 PM, Igor Potjevlesch igor.potjevle...@gmail.com mailto:igor.potjevle...@gmail.com wrote: Hello, I'm looking for a way to track a call by using basic AVP like this: Into a route that treats INVITE: $avp(s:state)=call_start; Then, if I test this AVP into WITHINDLG route: if($avp(s:state)!=call_start) ; the test fails. Did I miss something? The goal is to update this AVP during the life of the transaction. Regards, Igor. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Interpretation of Contact header
Hello, if you know that the contact address is valid and should be used for opening connections towards UA, then do not call fix_nated_register() for REGISTER request. Unfortunately UA behind NAT using STUN can lead to public address in Via/Contact/etc... but with a wrong port, therefore we have many tests to decide if UA should be considered behind NAT or no. In your case, should not be considered behind nat. The received parameter in Via header was fixed, but I am quite sure 3.2.x doesn't have it, that's quite old. You have to upgrade to a more recent version. Cheers, Daniel On 13/02/15 17:55, Joachim Büchse wrote: Good day, I’m experiencing some problems with our VoiP providers handling of REGISTER requests. We are using a Gigaset PRO N720 as UAC behind a Juniper SSG 140 with SIP-Alg enabled. This setup kind of works with UDP but our provider wants us to use TCP. With TCP enforced incoming calls don’t work. I’ve done some wire tracing and to me it seems that the providers configuration is to blame, but then - there are many RFCs out there and many NAT and UAC bug workarounds. Anyway, I wanted to get the opinion of “the experts about how the requests send to the UAS SHOULD be correctly interpreted. The REGISTER requests/responses look like this (outside of the firewall): Protocol TCP! client port 19091 - server port 5060 REGISTER sip:pbx.peoplefone.ch SIP/2.0 Via: SIP/2.0/TCP 212.126.160.92:6717;rport;branch=z9hG4bKc1375589832468de63a719eac31156ec From: Michel sip:90780408...@pbx.peoplefone.ch;tag=2153084485 To: Michel sip:90780408...@pbx.peoplefone.ch Call-ID: 2825358480@10_10_128_10 CSeq: 1 REGISTER Contact: sip:90780408050@212.126.160.92:6717;transport=tcp Max-Forwards: 70 User-Agent: N720-DM-PRO/70.089.00.000.000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 212.126.160.92:6717;rport=19091;branch=z9hG4bKc1375589832468de63a719eac31156ec From: Michel sip:90780408...@pbx.peoplefone.ch;tag=2153084485 To: Michel sip:90780408...@pbx.peoplefone.ch;tag=a0440f545f39b2694d387b475a5f6bc9.b8fc Call-ID: 2825358480@10_10_128_10 CSeq: 1 REGISTER WWW-Authenticate: Digest realm=pbx.peoplefone.ch, nonce=VNqJBVTah9m57ZGGs8b5XCTM3GyaExDy Server: kamailio (3.2.1 (x86_64/linux)) Content-Length: 0 REGISTER sip:pbx.peoplefone.ch SIP/2.0 Via: SIP/2.0/TCP 212.126.160.92:6717;rport;branch=z9hG4bK9c27afea96e2af4baab2f8d144a588e0 From: Michel sip:90780408...@pbx.peoplefone.ch;tag=2153084485 To: Michel sip:90780408...@pbx.peoplefone.ch Call-ID: 2825358480@10_10_128_10 CSeq: 2 REGISTER Contact: sip:90780408050@212.126.160.92:6717;transport=tcp Authorization: Digest username=90780408050, realm=pbx.peoplefone.ch, uri=sip:pbx.peoplefone.ch, nonce=VNqJBVTah9m57ZGGs8b5XCTM3GyaExDy, response=764f371a08d258157a249f8d1b852514 Max-Forwards: 70 User-Agent: N720-DM-PRO/70.089.00.000.000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/TCP 212.126.160.92:6717;rport=19091;branch=z9hG4bK9c27afea96e2af4baab2f8d144a588e0 From: Michel sip:90780408...@pbx.peoplefone.ch;tag=2153084485 To: Michel sip:90780408...@pbx.peoplefone.ch;tag=a0440f545f39b2694d387b475a5f6bc9.6bda Call-ID: 2825358480@10_10_128_10 CSeq: 2 REGISTER Contact: sip:90780408050@212.126.160.92:6717;transport=tcp;q=0;expires=180;received=sip:212.126.160.92:19091;transport=TCP Server: kamailio (3.2.1 (x86_64/linux)) Content-Length: 0 The ip:port the firewall is sending those requests from is ip 212.126.160.92 port 19091. So this does NOT match the port from the Contact header. For TCP this seems rather logical to me, as one cant be listening on a TCP port and use it for sending at the same time. The UAC closes this “register connection” with TCP FIN after the register, and so does the firewall. However unfortunately subsequent requests from the provider (ie UAS) come in on port 19091 (not port 6717 from the Contact header) and the firewall simply drops them. Observations: - the server does NOT include received=212.126.160.92 in the Via of the reponse. According to RFC3581 this is mandatory when rport is present in the request, so this is probably an error in the server. - the server does include received=sip:212.126.160.92:19091;transport=TCP” in the Contact of the response. I didnt see this in any RFC (which means nothing;-) but it could be an error. - after the client received the 200 OK it closes the TCP connection. - the server tries several times to re-contact the client (incoming TCP SYN). However not on port 6717 (defined in the Contact header) but on port 19091 (where the REGISTER came from). RFC3581 defines special behaviour when “rport” is defined in the request (i.e. response should go to the same port the request came from) - however
Re: [SR-Users] rtpproxy bridge ie ei behind NAT (like in aws EC2)
On 16/02/15 12:39 PM, Muhammad Shahzad wrote: I haven't done something like that myself but i think if you use RTPEngine with media-address set correctly in offer and answer functions, you can easily achieve this. Simply check if request/reply is coming from FS or the end-user and adjust the media appropriately without even invoking auto-bridge etc. Or perhaps with multiple interface specifications, binding to the same local address but with different advertised addresses, as suggested in the OP. Cheers ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] rtpproxy bridge ie ei behind NAT (like in aws EC2)
dear Kamailians, I have Kamailio+rtpproxy in front of FreeSWITCH. Kamailio and FreeSWITCH are on the same private network. Public Internet IP address ports are redirected to Kamailio and rtpproxy (same situation as in Amazon EC2). Clients comes from Internet, and make calls to Internet, SIP signaling passing through FreeSWITCH (eg: A leg incoming INVITE, FreeSWITCH originate an outbound B leg INVITE, and then bridge the legs). Using rtpproxy with -A advertise patch from Daniel, this topology works fine in a traditional telco way: rtp goes from caller to rtpproxy to callee, and viceversa. Now I want to maintain FreeSWITCH in the middle of rtp flow all the time, in a pure b2bua way, so it can control and analyze the media streams. So, I need rtpproxy to act paying attention to direction, as in caller-rtpproxy-freeswitch-rtpproxy-callee (and viceversa). Normally I would use Kamailio multihomed and rtpproxy in bridging mode. But I cannot have a NIC on the public address. How I can use the ie ei feature of rtpproxy in an Amazon-EC2 like environment? (eg: no public address attached to machine, but ports redirection from public address). I read this trick from Hugh Waite: I have used rtpproxy (with the advertised address patch) in Amazon to bridge media between internet facing and private subnets in a VPC. I found that I couldn’t use different advertised addresses depending on which direction the signalling was going on a single private IP address. I worked around this by allocating a second private ip address to the instance and used that in the ‘bridge’. -A 54.86.X.X/10.0.1.15 –l 10.0.1.10/10.0.1.15 Can you explain how to use this trick, or another way (without additional addresses is gladly accepted!) to reach the same result (rtp always passing through FreeSWITCH) ? Thank you all in advance, -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Syntax issue?
Hello, I'm looking for a way to track a call by using basic AVP like this: Into a route that treats INVITE: $avp(s:state)=call_start; Then, if I test this AVP into WITHINDLG route: if($avp(s:state)!=call_start) ; the test fails. Did I miss something? The goal is to update this AVP during the life of the transaction. Regards, Igor. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Syntax issue?
Hello, avps are lasting for the duration of the transaction. In route withindlg you handle already another transaction than the initial invite, so the avp is gone. Try to use $dlg_var(...) for this case -- check also if there is no $dlg(...) var that returns the state of the dialog and you can reuse. Cheers, Daniel On 16/02/15 18:08, Igor Potjevlesch wrote: Hello, I'm looking for a way to track a call by using basic AVP like this: Into a route that treats INVITE: $avp(s:state)=call_start; Then, if I test this AVP into WITHINDLG route: if($avp(s:state)!=call_start) ; the test fails. Did I miss something? The goal is to update this AVP during the life of the transaction. Regards, Igor. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] rtpproxy bridge ie ei behind NAT (like in aws EC2)
Hello, rtpproxy doing bridging requires two network interfaces to work with. You can try one of the following: - let freeswitch advertise the public ip for media and skip rtpproxy completely - use the second parameter of rtpproxy_manage() to set the advertised ip address for media and don't configure rtpproxy in bridge mode Cheers, Daniel On 16/02/15 17:30, Giovanni Maruzzelli wrote: dear Kamailians, I have Kamailio+rtpproxy in front of FreeSWITCH. Kamailio and FreeSWITCH are on the same private network. Public Internet IP address ports are redirected to Kamailio and rtpproxy (same situation as in Amazon EC2). Clients comes from Internet, and make calls to Internet, SIP signaling passing through FreeSWITCH (eg: A leg incoming INVITE, FreeSWITCH originate an outbound B leg INVITE, and then bridge the legs). Using rtpproxy with -A advertise patch from Daniel, this topology works fine in a traditional telco way: rtp goes from caller to rtpproxy to callee, and viceversa. Now I want to maintain FreeSWITCH in the middle of rtp flow all the time, in a pure b2bua way, so it can control and analyze the media streams. So, I need rtpproxy to act paying attention to direction, as in caller-rtpproxy-freeswitch-rtpproxy-callee (and viceversa). Normally I would use Kamailio multihomed and rtpproxy in bridging mode. But I cannot have a NIC on the public address. How I can use the ie ei feature of rtpproxy in an Amazon-EC2 like environment? (eg: no public address attached to machine, but ports redirection from public address). I read this trick from Hugh Waite: I have used rtpproxy (with the advertised address patch) in Amazon to bridge media between internet facing and private subnets in a VPC. I found that I couldn’t use different advertised addresses depending on which direction the signalling was going on a single private IP address. I worked around this by allocating a second private ip address to the instance and used that in the ‘bridge’. -A 54.86.X.X/10.0.1.15 –l 10.0.1.10/10.0.1.15 Can you explain how to use this trick, or another way (without additional addresses is gladly accepted!) to reach the same result (rtp always passing through FreeSWITCH) ? Thank you all in advance, -giovanni -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] rtpproxy bridge ie ei behind NAT (like in aws EC2)
I haven't done something like that myself but i think if you use RTPEngine with media-address set correctly in offer and answer functions, you can easily achieve this. Simply check if request/reply is coming from FS or the end-user and adjust the media appropriately without even invoking auto-bridge etc. Thank you. On Mon, Feb 16, 2015 at 5:30 PM, Giovanni Maruzzelli gmar...@gmail.com wrote: dear Kamailians, I have Kamailio+rtpproxy in front of FreeSWITCH. Kamailio and FreeSWITCH are on the same private network. Public Internet IP address ports are redirected to Kamailio and rtpproxy (same situation as in Amazon EC2). Clients comes from Internet, and make calls to Internet, SIP signaling passing through FreeSWITCH (eg: A leg incoming INVITE, FreeSWITCH originate an outbound B leg INVITE, and then bridge the legs). Using rtpproxy with -A advertise patch from Daniel, this topology works fine in a traditional telco way: rtp goes from caller to rtpproxy to callee, and viceversa. Now I want to maintain FreeSWITCH in the middle of rtp flow all the time, in a pure b2bua way, so it can control and analyze the media streams. So, I need rtpproxy to act paying attention to direction, as in caller-rtpproxy-freeswitch-rtpproxy-callee (and viceversa). Normally I would use Kamailio multihomed and rtpproxy in bridging mode. But I cannot have a NIC on the public address. How I can use the ie ei feature of rtpproxy in an Amazon-EC2 like environment? (eg: no public address attached to machine, but ports redirection from public address). I read this trick from Hugh Waite: I have used rtpproxy (with the advertised address patch) in Amazon to bridge media between internet facing and private subnets in a VPC. I found that I couldn’t use different advertised addresses depending on which direction the signalling was going on a single private IP address. I worked around this by allocating a second private ip address to the instance and used that in the ‘bridge’. -A 54.86.X.X/10.0.1.15 –l 10.0.1.10/10.0.1.15 Can you explain how to use this trick, or another way (without additional addresses is gladly accepted!) to reach the same result (rtp always passing through FreeSWITCH) ? Thank you all in advance, -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Event reg - questions
Hello, afaik, the pua_reginfo is for publishing details of location records to another sip server node, main purpose being location replication. I don't think it is something for an end UA. If you want publishing online/offline states for an user based on its registration state, look at pua_usrloc module. On 12/02/15 15:17, Stefan Ljung wrote: I made a test configuration for trying the pua_regEvent module. There were two things I want to ask about. First – when the SUBSCRIBE to event ‘reg’ following a UEs registration to the Kamailio is responded with a NOTIFY – there was no body in the NOTIFY. I expected a XML body with registration status. Second – When the UE re-registers to Kamailio – I expected a NOTIFY to be send on the existing ‘reg’ subscription, but nothing happened. Or am I missing some configuration ? Here’s some of the conf script: [..] modparam(pua_reginfo, publish_reginfo, 0) Btw, with the above parameter value, the module is no longer sending PUBLISH requests, in other words, it is not informing the other peers that there is some update to location records. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] rtpproxy bridge ie ei behind NAT (like in aws EC2)
BTW, if nothing works, you can always use network:msg event route to find / replace any part of the SIP request and response, including media IP in SDP. ;-) http://kamailio.org/docs/modules/4.2.x/modules/corex.html#async.evr.network_io Thank you. On Mon, Feb 16, 2015 at 6:39 PM, Muhammad Shahzad shaherya...@gmail.com wrote: I haven't done something like that myself but i think if you use RTPEngine with media-address set correctly in offer and answer functions, you can easily achieve this. Simply check if request/reply is coming from FS or the end-user and adjust the media appropriately without even invoking auto-bridge etc. Thank you. On Mon, Feb 16, 2015 at 5:30 PM, Giovanni Maruzzelli gmar...@gmail.com wrote: dear Kamailians, I have Kamailio+rtpproxy in front of FreeSWITCH. Kamailio and FreeSWITCH are on the same private network. Public Internet IP address ports are redirected to Kamailio and rtpproxy (same situation as in Amazon EC2). Clients comes from Internet, and make calls to Internet, SIP signaling passing through FreeSWITCH (eg: A leg incoming INVITE, FreeSWITCH originate an outbound B leg INVITE, and then bridge the legs). Using rtpproxy with -A advertise patch from Daniel, this topology works fine in a traditional telco way: rtp goes from caller to rtpproxy to callee, and viceversa. Now I want to maintain FreeSWITCH in the middle of rtp flow all the time, in a pure b2bua way, so it can control and analyze the media streams. So, I need rtpproxy to act paying attention to direction, as in caller-rtpproxy-freeswitch-rtpproxy-callee (and viceversa). Normally I would use Kamailio multihomed and rtpproxy in bridging mode. But I cannot have a NIC on the public address. How I can use the ie ei feature of rtpproxy in an Amazon-EC2 like environment? (eg: no public address attached to machine, but ports redirection from public address). I read this trick from Hugh Waite: I have used rtpproxy (with the advertised address patch) in Amazon to bridge media between internet facing and private subnets in a VPC. I found that I couldn’t use different advertised addresses depending on which direction the signalling was going on a single private IP address. I worked around this by allocating a second private ip address to the instance and used that in the ‘bridge’. -A 54.86.X.X/10.0.1.15 –l 10.0.1.10/10.0.1.15 Can you explain how to use this trick, or another way (without additional addresses is gladly accepted!) to reach the same result (rtp always passing through FreeSWITCH) ? Thank you all in advance, -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Log Media IP
Hello,
you should be able to extract it with {line} and {subst} transformations
applied to sdp body.
Cheers,
Daniel
On 10/02/15 23:42, Ryan Brindley wrote:
Hey community,
What's the best way to pull out the media ip from the SIP INVITE body
(for logging)?
Ryan Brindley
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Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
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Re: [SR-Users] Syntax issue?
As far as i know AVPs are transaction specific only. So they will be deleted as soon as transaction is over, i.e. 200 OK for INVITE is received for example. They will not be available in in-dialog transactions such as ACK, or BYE etc. What you need is to set dialog variable instead, see more info here, http://kamailio.org/docs/modules/4.2.x/modules/dialog.html#idp4202736 Thank you. On Mon, Feb 16, 2015 at 6:08 PM, Igor Potjevlesch igor.potjevle...@gmail.com wrote: Hello, I'm looking for a way to track a call by using basic AVP like this: Into a route that treats INVITE: $avp(s:state)=call_start; Then, if I test this AVP into WITHINDLG route: if($avp(s:state)!=call_start) ; the test fails. Did I miss something? The goal is to update this AVP during the life of the transaction. Regards, Igor. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Syntax issue?
Thank you guys, I will try this. I misunderstood the notion of transaction. I was thinking that it was the whole call-flow. Regards, Igor. De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de Muhammad Shahzad Envoyé : lundi 16 février 2015 18:27 À : Kamailio (SER) - Users Mailing List Objet : Re: [SR-Users] Syntax issue? As far as i know AVPs are transaction specific only. So they will be deleted as soon as transaction is over, i.e. 200 OK for INVITE is received for example. They will not be available in in-dialog transactions such as ACK, or BYE etc. What you need is to set dialog variable instead, see more info here, http://kamailio.org/docs/modules/4.2.x/modules/dialog.html#idp4202736 Thank you. On Mon, Feb 16, 2015 at 6:08 PM, Igor Potjevlesch igor.potjevle...@gmail.com mailto:igor.potjevle...@gmail.com wrote: Hello, I'm looking for a way to track a call by using basic AVP like this: Into a route that treats INVITE: $avp(s:state)=call_start; Then, if I test this AVP into WITHINDLG route: if($avp(s:state)!=call_start) ; the test fails. Did I miss something? The goal is to update this AVP during the life of the transaction. Regards, Igor. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Syntax issue?
Well, you can also put them in some storage backend e.g. MySQL, PGSQL using AVPOPS or memory caches such as Redis etc. Another way is to set it as record-route parameter using RR module. (not recommended) http://kamailio.org/docs/modules/4.2.x/modules/rr.html#add-rr-param-id Thank you. On Mon, Feb 16, 2015 at 6:42 PM, Igor Potjevlesch igor.potjevle...@gmail.com wrote: Additionally, there's no other way than implementing dialog module to keep a variable between the beginning and the end of a call? Regards, Igor. *De :* Igor Potjevlesch [mailto:igor.potjevle...@gmail.com] *Envoyé :* lundi 16 février 2015 18:36 *À :* 'Kamailio (SER) - Users Mailing List' *Objet :* RE: [SR-Users] Syntax issue? Thank you guys, I will try this. I misunderstood the notion of transaction. I was thinking that it was the whole call-flow. Regards, Igor. *De :* sr-users [mailto:sr-users-boun...@lists.sip-router.org sr-users-boun...@lists.sip-router.org] *De la part de* Muhammad Shahzad *Envoyé :* lundi 16 février 2015 18:27 *À :* Kamailio (SER) - Users Mailing List *Objet :* Re: [SR-Users] Syntax issue? As far as i know AVPs are transaction specific only. So they will be deleted as soon as transaction is over, i.e. 200 OK for INVITE is received for example. They will not be available in in-dialog transactions such as ACK, or BYE etc. What you need is to set dialog variable instead, see more info here, http://kamailio.org/docs/modules/4.2.x/modules/dialog.html#idp4202736 Thank you. On Mon, Feb 16, 2015 at 6:08 PM, Igor Potjevlesch igor.potjevle...@gmail.com wrote: Hello, I'm looking for a way to track a call by using basic AVP like this: Into a route that treats INVITE: $avp(s:state)=call_start; Then, if I test this AVP into WITHINDLG route: if($avp(s:state)!=call_start) ; the test fails. Did I miss something? The goal is to update this AVP during the life of the transaction. Regards, Igor. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] rtpproxy bridge ie ei behind NAT (like in aws EC2)
On 16/02/15 01:00 PM, Virmantas Variakojis wrote: Hi, There pathch with -A can be found or it is allready implemented into specific rtpengine version? Latest master from git. The command line syntax is a bit different from rtpproxy, but the basic idea is the same. Cheers ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Syntax issue?
Why not an RR parameter? It's probably the most reliable way to store some dialog-persistent data, since it doesn't depend on any in-memory/runtime state to be kept by the proxy itself, instead using the SIP messaging itself as a persistence layer. The only trouble with this approach is that it relies on correct RR behaviour by both endpoints and of course neither hides the value from the endpoints not prevents them from manipulating it.If the latter aspects are a concern, $dlg vars are probably the way to go. If you don't want to use the dialog module, use an 'htable' with Call-ID + From-tag as key.--SentfrommyBlackBerry.Pleaseexcuseerrorsandbrevity.From: Muhammad ShahzadSent: Monday, February 16, 2015 12:55 PMTo: Kamailio (SER) - Users Mailing ListReply To: Kamailio (SER) - Users Mailing ListSubject: Re: [SR-Users] Syntax issue?Well, you can also put them in some storage backend e.g. MySQL, PGSQL using AVPOPS or memory caches such as Redis etc.Another way is to set it as record-route parameter using RR module. (not recommended)http://kamailio.org/docs/modules/4.2.x/modules/rr.html#add-rr-param-idThank you.On Mon, Feb 16, 2015 at 6:42 PM, Igor Potjevlesch igor.potjevle...@gmail.com wrote:Additionally, there's no other way than implementing dialog module to keep a variable between the beginning and the end of a call?Regards,Igor.De: Igor Potjevlesch [mailto:igor.potjevle...@gmail.com] Envoyé: lundi 16 février 2015 18:36À: 'Kamailio (SER) - Users Mailing List'Objet: RE: [SR-Users] Syntax issue?Thank you guys, I will try this.I misunderstood the notion of "transaction". I was thinking that it was the whole call-flow.Regards,Igor.De: sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de Muhammad ShahzadEnvoyé: lundi 16 février 2015 18:27À: Kamailio (SER) - Users Mailing ListObjet: Re: [SR-Users] Syntax issue?As far as i know AVPs are transaction specific only. So they will be deleted as soon as transaction is over, i.e. 200 OK for INVITE is received for example. They will not be available in in-dialog transactions such as ACK, or BYE etc. What you need is to set dialog variable instead, see more info here,http://kamailio.org/docs/modules/4.2.x/modules/dialog.html#idp4202736Thank you.On Mon, Feb 16, 2015 at 6:08 PM, Igor Potjevlesch igor.potjevle...@gmail.com wrote:Hello,I'm looking for a way to track a call by using basic AVP like this:Into a route that treats INVITE:$avp(s:state)="call_start";Then, if I test this AVP into WITHINDLG route:if($avp(s:state)!="call_start") ; the test fails.Did I miss something?The goal is to update this AVP during the life of the transaction.Regards,Igor.___SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Syntax issue?
Indeed, RR could do the job. But it will not be easy to get the value after. It could be possible with regex I guess. I will look at htable too. It's looks to be easier than dialog. For AVPOPS, why not. I'm just afraid with the delay. Many thanks for all these suggestions. Regards, Igor. De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de Alex Balashov Envoyé : lundi 16 février 2015 19:05 À : Muhammad Shahzad Objet : Re: [SR-Users] Syntax issue? Why not an RR parameter? It's probably the most reliable way to store some dialog-persistent data, since it doesn't depend on any in-memory/runtime state to be kept by the proxy itself, instead using the SIP messaging itself as a persistence layer. The only trouble with this approach is that it relies on correct RR behaviour by both endpoints and of course neither hides the value from the endpoints not prevents them from manipulating it. If the latter aspects are a concern, $dlg vars are probably the way to go. If you don't want to use the dialog module, use an 'htable' with Call-ID + From-tag as key. -- Sent from my BlackBerry. Please excuse errors and brevity. From: Muhammad Shahzad Sent: Monday, February 16, 2015 12:55 PM To: Kamailio (SER) - Users Mailing List Reply To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Syntax issue? Well, you can also put them in some storage backend e.g. MySQL, PGSQL using AVPOPS or memory caches such as Redis etc. Another way is to set it as record-route parameter using RR module. (not recommended) http://kamailio.org/docs/modules/4.2.x/modules/rr.html#add-rr-param-id Thank you. On Mon, Feb 16, 2015 at 6:42 PM, Igor Potjevlesch igor.potjevle...@gmail.com mailto:igor.potjevle...@gmail.com wrote: Additionally, there's no other way than implementing dialog module to keep a variable between the beginning and the end of a call? Regards, Igor. De : Igor Potjevlesch [mailto:igor.potjevle...@gmail.com mailto:igor.potjevle...@gmail.com ] Envoyé : lundi 16 février 2015 18:36 À : 'Kamailio (SER) - Users Mailing List' Objet : RE: [SR-Users] Syntax issue? Thank you guys, I will try this. I misunderstood the notion of transaction. I was thinking that it was the whole call-flow. Regards, Igor. De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de Muhammad Shahzad Envoyé : lundi 16 février 2015 18:27 À : Kamailio (SER) - Users Mailing List Objet : Re: [SR-Users] Syntax issue? As far as i know AVPs are transaction specific only. So they will be deleted as soon as transaction is over, i.e. 200 OK for INVITE is received for example. They will not be available in in-dialog transactions such as ACK, or BYE etc. What you need is to set dialog variable instead, see more info here, http://kamailio.org/docs/modules/4.2.x/modules/dialog.html#idp4202736 Thank you. On Mon, Feb 16, 2015 at 6:08 PM, Igor Potjevlesch igor.potjevle...@gmail.com mailto:igor.potjevle...@gmail.com wrote: Hello, I'm looking for a way to track a call by using basic AVP like this: Into a route that treats INVITE: $avp(s:state)=call_start; Then, if I test this AVP into WITHINDLG route: if($avp(s:state)!=call_start) ; the test fails. Did I miss something? The goal is to update this AVP during the life of the transaction. Regards, Igor. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Syntax issue?
I tried with $sht(myhash=$ci::state) = call_start. It works fine!! Many thanks. Is that could work too: $sht(myhash=$ci::$ft::state) = call_start? To delete this, can I do sht_rm_name_re(myhash=$ci);? I want to be sure that after the call ends, everything is cleared. Regards, Igor. De : Igor Potjevlesch [mailto:igor.potjevle...@gmail.com] Envoyé : lundi 16 février 2015 19:38 À : 'Kamailio (SER) - Users Mailing List' Objet : RE: [SR-Users] Syntax issue? I just tried with RR but it didn't really match what I want to do. HTAble with Call-ID+From-tag is a really interesting idea. I start reading the documentation of the module. Have you an example of what this might look like? Regards, Igor. De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de Alex Balashov Envoyé : lundi 16 février 2015 19:22 À : Igor Potjevlesch Objet : Re: [SR-Users] Syntax issue? It's pretty straightforward using the right transformations on $hdr(Record-Route). Have a look at the transformations docs. -- Sent from my BlackBerry. Please excuse errors and brevity. From: Igor Potjevlesch Sent: Monday, February 16, 2015 1:17 PM To: 'Kamailio (SER) - Users Mailing List' Reply To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Syntax issue? Indeed, RR could do the job. But it will not be easy to get the value after. It could be possible with regex I guess. I will look at htable too. It's looks to be easier than dialog. For AVPOPS, why not. I'm just afraid with the delay. Many thanks for all these suggestions. Regards, Igor. De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de Alex Balashov Envoyé : lundi 16 février 2015 19:05 À : Muhammad Shahzad Objet : Re: [SR-Users] Syntax issue? Why not an RR parameter? It's probably the most reliable way to store some dialog-persistent data, since it doesn't depend on any in-memory/runtime state to be kept by the proxy itself, instead using the SIP messaging itself as a persistence layer. The only trouble with this approach is that it relies on correct RR behaviour by both endpoints and of course neither hides the value from the endpoints not prevents them from manipulating it. If the latter aspects are a concern, $dlg vars are probably the way to go. If you don't want to use the dialog module, use an 'htable' with Call-ID + From-tag as key. -- Sent from my BlackBerry. Please excuse errors and brevity. From: Muhammad Shahzad Sent: Monday, February 16, 2015 12:55 PM To: Kamailio (SER) - Users Mailing List Reply To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Syntax issue? Well, you can also put them in some storage backend e.g. MySQL, PGSQL using AVPOPS or memory caches such as Redis etc. Another way is to set it as record-route parameter using RR module. (not recommended) http://kamailio.org/docs/modules/4.2.x/modules/rr.html#add-rr-param-id Thank you. On Mon, Feb 16, 2015 at 6:42 PM, Igor Potjevlesch igor.potjevle...@gmail.com mailto:igor.potjevle...@gmail.com wrote: Additionally, there's no other way than implementing dialog module to keep a variable between the beginning and the end of a call? Regards, Igor. De : Igor Potjevlesch [mailto:igor.potjevle...@gmail.com mailto:igor.potjevle...@gmail.com ] Envoyé : lundi 16 février 2015 18:36 À : 'Kamailio (SER) - Users Mailing List' Objet : RE: [SR-Users] Syntax issue? Thank you guys, I will try this. I misunderstood the notion of transaction. I was thinking that it was the whole call-flow. Regards, Igor. De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de Muhammad Shahzad Envoyé : lundi 16 février 2015 18:27 À : Kamailio (SER) - Users Mailing List Objet : Re: [SR-Users] Syntax issue? As far as i know AVPs are transaction specific only. So they will be deleted as soon as transaction is over, i.e. 200 OK for INVITE is received for example. They will not be available in in-dialog transactions such as ACK, or BYE etc. What you need is to set dialog variable instead, see more info here, http://kamailio.org/docs/modules/4.2.x/modules/dialog.html#idp4202736 Thank you. On Mon, Feb 16, 2015 at 6:08 PM, Igor Potjevlesch igor.potjevle...@gmail.com mailto:igor.potjevle...@gmail.com wrote: Hello, I'm looking for a way to track a call by using basic AVP like this: Into a route that treats INVITE: $avp(s:state)=call_start; Then, if I test this AVP into WITHINDLG route: if($avp(s:state)!=call_start) ; the test fails. Did I miss something? The goal is to update this AVP during the life of the transaction. Regards, Igor. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org
Re: [SR-Users] rtpproxy bridge ie ei behind NAT (like in aws EC2)
Could you provide us a little example? For examlple i have kamailio with three interfaces: two interfaces (vlan's look at two different providers) and third interface looks at sip clients. Thank's in advance! 2015 vas. 16 20:04 Richard Fuchs rfu...@sipwise.com rašė: On 16/02/15 01:00 PM, Virmantas Variakojis wrote: Hi, There pathch with -A can be found or it is allready implemented into specific rtpengine version? Latest master from git. The command line syntax is a bit different from rtpproxy, but the basic idea is the same. Cheers ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- __ CONFIDENTIALITY NOTICE: The information contained in the present message (including any information contained in attachments herein) may be confidential and privileged. It may be read, copied and used only by the intended recipient. If you have received it in error please contact the sender (by return e-mail) immediately and delete this message. Any unauthorized use or dissemination of this message in whole or in parts is strictly prohibited. Print this message only if sharp necessary. УВЕДОМЛЕНИЕ О КОНФИДЕНЦИАЛЬНОСТИ: Информация, содержащаяся в настоящем сообщении (включая любое вложение) может быть конфиденциальной и охраняться действующим законодательством. Сообщение может быть прочитано, скопировано и использовано исключительно лицом, которому сообщение предназначается. Если Вы получили настоящее сообщение по ошибке, пожалуйста, незамедлительно сообщите об этом отправителю (ответным письмом по электронной почте). Любое несанкционированное использование или распространение информации, содержащейся в настоящем сообщении в целом или в части, строго запрещены. Не распечатывайте настоящее сообщение, если в этом нет крайней необходимости. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Syntax issue?
It's pretty straightforward using the right transformations on $hdr(Record-Route). Have a look at the transformations docs.--SentfrommyBlackBerry.Pleaseexcuseerrorsandbrevity.From: Igor PotjevleschSent: Monday, February 16, 2015 1:17 PMTo: 'Kamailio (SER) - Users Mailing List'Reply To: Kamailio (SER) - Users Mailing ListSubject: Re: [SR-Users] Syntax issue?Indeed, RR could do the job. But it will not be easy to get the value after. It could be possible with regex I guess.I will look at htable too. It's looks to be easier than dialog.For AVPOPS, why not. I'm just afraid with the delay.Many thanks for all these suggestions.Regards,Igor.De: sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de Alex BalashovEnvoyé: lundi 16 février 2015 19:05À: Muhammad ShahzadObjet: Re: [SR-Users] Syntax issue?Why not an RR parameter? It's probably the most reliable way to store some dialog-persistent data, since it doesn't depend on any in-memory/runtime state to be kept by the proxy itself, instead using the SIP messaging itself as a persistence layer. The only trouble with this approach is that it relies on correct RR behaviour by both endpoints and of course neither hides the value from the endpoints not prevents them from manipulating it.If the latter aspects are a concern, $dlg vars are probably the way to go. If you don't want to use the dialog module, use an 'htable' with Call-ID + From-tag as key.--SentfrommyBlackBerry.Pleaseexcuseerrorsandbrevity.From: Muhammad ShahzadSent: Monday, February 16, 2015 12:55 PMTo: Kamailio (SER) - Users Mailing ListReply To: Kamailio (SER) - Users Mailing ListSubject: Re: [SR-Users] Syntax issue?Well, you can also put them in some storage backend e.g. MySQL, PGSQL using AVPOPS or memory caches such as Redis etc.Another way is to set it as record-route parameter using RR module. (not recommended)http://kamailio.org/docs/modules/4.2.x/modules/rr.html#add-rr-param-idThank you.On Mon, Feb 16, 2015 at 6:42 PM, Igor Potjevlesch igor.potjevle...@gmail.com wrote:Additionally, there's no other way than implementing dialog module to keep a variable between the beginning and the end of a call?Regards,Igor.De: Igor Potjevlesch [mailto:igor.potjevle...@gmail.com] Envoyé: lundi 16 février 2015 18:36À: 'Kamailio (SER) - Users Mailing List'Objet: RE: [SR-Users] Syntax issue?Thank you guys, I will try this.I misunderstood the notion of "transaction". I was thinking that it was the whole call-flow.Regards,Igor.De: sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de Muhammad ShahzadEnvoyé: lundi 16 février 2015 18:27À: Kamailio (SER) - Users Mailing ListObjet: Re: [SR-Users] Syntax issue?As far as i know AVPs are transaction specific only. So they will be deleted as soon as transaction is over, i.e. 200 OK for INVITE is received for example. They will not be available in in-dialog transactions such as ACK, or BYE etc. What you need is to set dialog variable instead, see more info here,http://kamailio.org/docs/modules/4.2.x/modules/dialog.html#idp4202736Thank you.On Mon, Feb 16, 2015 at 6:08 PM, Igor Potjevlesch igor.potjevle...@gmail.com wrote:Hello,I'm looking for a way to track a call by using basic AVP like this:Into a route that treats INVITE:$avp(s:state)="call_start";Then, if I test this AVP into WITHINDLG route:if($avp(s:state)!="call_start") ; the test fails.Did I miss something?The goal is to update this AVP during the life of the transaction.Regards,Igor.___SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users___SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] rtpproxy bridge ie ei behind NAT (like in aws EC2)
On 16/02/15 01:12 PM, Virmantas Variakojis wrote: Could you provide us a little example? For examlple i have kamailio with three interfaces: two interfaces (vlan's look at two different providers) and third interface looks at sip clients. You would define two interfaces with different names, for example --interface=public/10.0.1.15!54.86.X.X for outside media and --interface=local/10.0.1.15 for local media. You would then use two direction=... options in the offer to determine where A side and B side are located, respectively. You can also call the interfaces external and internal and use these flags instead, which mirrors rtpproxy's behaviour. Cheers ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Syntax issue?
Yes for the same reasons as you mentioned, it adds dependency on external entities in your setup and may not be suitable for any sensitive data (e.g. related to billing etc.). Thank you. On Mon, Feb 16, 2015 at 7:05 PM, Alex Balashov abalas...@evaristesys.com wrote: Why not an RR parameter? It's probably the most reliable way to store some dialog-persistent data, since it doesn't depend on any in-memory/runtime state to be kept by the proxy itself, instead using the SIP messaging itself as a persistence layer. The only trouble with this approach is that it relies on correct RR behaviour by both endpoints and of course neither hides the value from the endpoints not prevents them from manipulating it. If the latter aspects are a concern, $dlg vars are probably the way to go. If you don't want to use the dialog module, use an 'htable' with Call-ID + From-tag as key. -- Sent from my BlackBerry. Please excuse errors and brevity. *From: *Muhammad Shahzad *Sent: *Monday, February 16, 2015 12:55 PM *To: *Kamailio (SER) - Users Mailing List *Reply To: *Kamailio (SER) - Users Mailing List *Subject: *Re: [SR-Users] Syntax issue? Well, you can also put them in some storage backend e.g. MySQL, PGSQL using AVPOPS or memory caches such as Redis etc. Another way is to set it as record-route parameter using RR module. (not recommended) http://kamailio.org/docs/modules/4.2.x/modules/rr.html#add-rr-param-id Thank you. On Mon, Feb 16, 2015 at 6:42 PM, Igor Potjevlesch igor.potjevle...@gmail.com wrote: Additionally, there's no other way than implementing dialog module to keep a variable between the beginning and the end of a call? Regards, Igor. *De :* Igor Potjevlesch [mailto:igor.potjevle...@gmail.com] *Envoyé :* lundi 16 février 2015 18:36 *À :* 'Kamailio (SER) - Users Mailing List' *Objet :* RE: [SR-Users] Syntax issue? Thank you guys, I will try this. I misunderstood the notion of transaction. I was thinking that it was the whole call-flow. Regards, Igor. *De :* sr-users [mailto:sr-users-boun...@lists.sip-router.org sr-users-boun...@lists.sip-router.org] *De la part de* Muhammad Shahzad *Envoyé :* lundi 16 février 2015 18:27 *À :* Kamailio (SER) - Users Mailing List *Objet :* Re: [SR-Users] Syntax issue? As far as i know AVPs are transaction specific only. So they will be deleted as soon as transaction is over, i.e. 200 OK for INVITE is received for example. They will not be available in in-dialog transactions such as ACK, or BYE etc. What you need is to set dialog variable instead, see more info here, http://kamailio.org/docs/modules/4.2.x/modules/dialog.html#idp4202736 Thank you. On Mon, Feb 16, 2015 at 6:08 PM, Igor Potjevlesch igor.potjevle...@gmail.com wrote: Hello, I'm looking for a way to track a call by using basic AVP like this: Into a route that treats INVITE: $avp(s:state)=call_start; Then, if I test this AVP into WITHINDLG route: if($avp(s:state)!=call_start) ; the test fails. Did I miss something? The goal is to update this AVP during the life of the transaction. Regards, Igor. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Syntax issue?
Yes, and yes.--SentfrommyBlackBerry.Pleaseexcuseerrorsandbrevity.From: Igor PotjevleschSent: Monday, February 16, 2015 2:12 PMTo: 'Kamailio (SER) - Users Mailing List'Reply To: Kamailio (SER) - Users Mailing ListSubject: Re: [SR-Users] Syntax issue?I tried with $sht(myhash=$ci::state) = "call_start". It works fine!! Many thanks. Is that could work too: $sht(myhash=$ci::$ft::state) = "call_start"?To delete this, can I do sht_rm_name_re("myhash=$ci");? I want to be sure that after the call ends, everything is cleared.Regards,Igor.De: Igor Potjevlesch [mailto:igor.potjevle...@gmail.com] Envoyé: lundi 16 février 2015 19:38À: 'Kamailio (SER) - Users Mailing List'Objet: RE: [SR-Users] Syntax issue?I just tried with RR but it didn't really match what I want to do.HTAble with Call-ID+From-tag is a really interesting idea. I start reading the documentation of the module.Have you an example of what this might look like?Regards,Igor.De: sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de Alex BalashovEnvoyé: lundi 16 février 2015 19:22À: Igor PotjevleschObjet: Re: [SR-Users] Syntax issue?It's pretty straightforward using the right transformations on $hdr(Record-Route). Have a look at the transformations docs.--SentfrommyBlackBerry.Pleaseexcuseerrorsandbrevity.From: Igor PotjevleschSent: Monday, February 16, 2015 1:17 PMTo: 'Kamailio (SER) - Users Mailing List'Reply To: Kamailio (SER) - Users Mailing ListSubject: Re: [SR-Users] Syntax issue?Indeed, RR could do the job. But it will not be easy to get the value after. It could be possible with regex I guess.I will look at htable too. It's looks to be easier than dialog.For AVPOPS, why not. I'm just afraid with the delay.Many thanks for all these suggestions.Regards,Igor.De: sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de Alex BalashovEnvoyé: lundi 16 février 2015 19:05À: Muhammad ShahzadObjet: Re: [SR-Users] Syntax issue?Why not an RR parameter? It's probably the most reliable way to store some dialog-persistent data, since it doesn't depend on any in-memory/runtime state to be kept by the proxy itself, instead using the SIP messaging itself as a persistence layer. The only trouble with this approach is that it relies on correct RR behaviour by both endpoints and of course neither hides the value from the endpoints not prevents them from manipulating it.If the latter aspects are a concern, $dlg vars are probably the way to go. If you don't want to use the dialog module, use an 'htable' with Call-ID + From-tag as key.--SentfrommyBlackBerry.Pleaseexcuseerrorsandbrevity.From: Muhammad ShahzadSent: Monday, February 16, 2015 12:55 PMTo: Kamailio (SER) - Users Mailing ListReply To: Kamailio (SER) - Users Mailing ListSubject: Re: [SR-Users] Syntax issue?Well, you can also put them in some storage backend e.g. MySQL, PGSQL using AVPOPS or memory caches such as Redis etc.Another way is to set it as record-route parameter using RR module. (not recommended)http://kamailio.org/docs/modules/4.2.x/modules/rr.html#add-rr-param-idThank you.On Mon, Feb 16, 2015 at 6:42 PM, Igor Potjevlesch igor.potjevle...@gmail.com wrote:Additionally, there's no other way than implementing dialog module to keep a variable between the beginning and the end of a call?Regards,Igor.De: Igor Potjevlesch [mailto:igor.potjevle...@gmail.com] Envoyé: lundi 16 février 2015 18:36À: 'Kamailio (SER) - Users Mailing List'Objet: RE: [SR-Users] Syntax issue?Thank you guys, I will try this.I misunderstood the notion of "transaction". I was thinking that it was the whole call-flow.Regards,Igor.De: sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de Muhammad ShahzadEnvoyé: lundi 16 février 2015 18:27À: Kamailio (SER) - Users Mailing ListObjet: Re: [SR-Users] Syntax issue?As far as i know AVPs are transaction specific only. So they will be deleted as soon as transaction is over, i.e. 200 OK for INVITE is received for example. They will not be available in in-dialog transactions such as ACK, or BYE etc. What you need is to set dialog variable instead, see more info here,http://kamailio.org/docs/modules/4.2.x/modules/dialog.html#idp4202736Thank you.On Mon, Feb 16, 2015 at 6:08 PM, Igor Potjevlesch igor.potjevle...@gmail.com wrote:Hello,I'm looking for a way to track a call by using basic AVP like this:Into a route that treats
Re: [SR-Users] Syntax issue?
Thank you Alex. I'm not sure to understand the parameter size associated to the hashtable. I have setup 4. So, I understand that I can have 2^4 entries. Does it mean that, if the table is composed with $ci+$ft, I can have 16 concurrent calls store into the table? Regards, Igor. De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de Alex Balashov Envoyé : lundi 16 février 2015 21:24 À : Igor Potjevlesch Objet : Re: [SR-Users] Syntax issue? Yes, and yes. -- Sent from my BlackBerry. Please excuse errors and brevity. From: Igor Potjevlesch Sent: Monday, February 16, 2015 2:12 PM To: 'Kamailio (SER) - Users Mailing List' Reply To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Syntax issue? I tried with $sht(myhash=$ci::state) = call_start. It works fine!! Many thanks. Is that could work too: $sht(myhash=$ci::$ft::state) = call_start? To delete this, can I do sht_rm_name_re(myhash=$ci);? I want to be sure that after the call ends, everything is cleared. Regards, Igor. De : Igor Potjevlesch [mailto:igor.potjevle...@gmail.com] Envoyé : lundi 16 février 2015 19:38 À : 'Kamailio (SER) - Users Mailing List' Objet : RE: [SR-Users] Syntax issue? I just tried with RR but it didn't really match what I want to do. HTAble with Call-ID+From-tag is a really interesting idea. I start reading the documentation of the module. Have you an example of what this might look like? Regards, Igor. De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de Alex Balashov Envoyé : lundi 16 février 2015 19:22 À : Igor Potjevlesch Objet : Re: [SR-Users] Syntax issue? It's pretty straightforward using the right transformations on $hdr(Record-Route). Have a look at the transformations docs. -- Sent from my BlackBerry. Please excuse errors and brevity. From: Igor Potjevlesch Sent: Monday, February 16, 2015 1:17 PM To: 'Kamailio (SER) - Users Mailing List' Reply To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Syntax issue? Indeed, RR could do the job. But it will not be easy to get the value after. It could be possible with regex I guess. I will look at htable too. It's looks to be easier than dialog. For AVPOPS, why not. I'm just afraid with the delay. Many thanks for all these suggestions. Regards, Igor. De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de Alex Balashov Envoyé : lundi 16 février 2015 19:05 À : Muhammad Shahzad Objet : Re: [SR-Users] Syntax issue? Why not an RR parameter? It's probably the most reliable way to store some dialog-persistent data, since it doesn't depend on any in-memory/runtime state to be kept by the proxy itself, instead using the SIP messaging itself as a persistence layer. The only trouble with this approach is that it relies on correct RR behaviour by both endpoints and of course neither hides the value from the endpoints not prevents them from manipulating it. If the latter aspects are a concern, $dlg vars are probably the way to go. If you don't want to use the dialog module, use an 'htable' with Call-ID + From-tag as key. -- Sent from my BlackBerry. Please excuse errors and brevity. From: Muhammad Shahzad Sent: Monday, February 16, 2015 12:55 PM To: Kamailio (SER) - Users Mailing List Reply To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Syntax issue? Well, you can also put them in some storage backend e.g. MySQL, PGSQL using AVPOPS or memory caches such as Redis etc. Another way is to set it as record-route parameter using RR module. (not recommended) http://kamailio.org/docs/modules/4.2.x/modules/rr.html#add-rr-param-id Thank you. On Mon, Feb 16, 2015 at 6:42 PM, Igor Potjevlesch igor.potjevle...@gmail.com mailto:igor.potjevle...@gmail.com wrote: Additionally, there's no other way than implementing dialog module to keep a variable between the beginning and the end of a call? Regards, Igor. De : Igor Potjevlesch [mailto:igor.potjevle...@gmail.com mailto:igor.potjevle...@gmail.com ] Envoyé : lundi 16 février 2015 18:36 À : 'Kamailio (SER) - Users Mailing List' Objet : RE: [SR-Users] Syntax issue? Thank you guys, I will try this. I misunderstood the notion of transaction. I was thinking that it was the whole call-flow. Regards, Igor. De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de Muhammad Shahzad Envoyé : lundi 16 février 2015 18:27 À : Kamailio (SER) - Users Mailing List Objet : Re: [SR-Users] Syntax issue? As far as i know AVPs are transaction specific only. So they will be deleted as soon as transaction is over, i.e. 200 OK for INVITE is received for example. They will not be available in in-dialog transactions such as ACK, or BYE etc. What you need is to set dialog variable instead, see more info here,
Re: [SR-Users] rtpproxy bridge ie ei behind NAT (like in aws EC2)
You could simply let the RTP traffic to flow directly between FS and endpoints (no need for rtpproxy). All you need to do is: - forward the appropriate RTP ports to FS; - fix the private IP in SDP by replacing it with the public IP for the inbound rtp streams (to FS). -ovidiu On Mon, Feb 16, 2015 at 11:30 AM, Giovanni Maruzzelli gmar...@gmail.com wrote: dear Kamailians, I have Kamailio+rtpproxy in front of FreeSWITCH. Kamailio and FreeSWITCH are on the same private network. Public Internet IP address ports are redirected to Kamailio and rtpproxy (same situation as in Amazon EC2). Clients comes from Internet, and make calls to Internet, SIP signaling passing through FreeSWITCH (eg: A leg incoming INVITE, FreeSWITCH originate an outbound B leg INVITE, and then bridge the legs). Using rtpproxy with -A advertise patch from Daniel, this topology works fine in a traditional telco way: rtp goes from caller to rtpproxy to callee, and viceversa. Now I want to maintain FreeSWITCH in the middle of rtp flow all the time, in a pure b2bua way, so it can control and analyze the media streams. So, I need rtpproxy to act paying attention to direction, as in caller-rtpproxy-freeswitch-rtpproxy-callee (and viceversa). Normally I would use Kamailio multihomed and rtpproxy in bridging mode. But I cannot have a NIC on the public address. How I can use the ie ei feature of rtpproxy in an Amazon-EC2 like environment? (eg: no public address attached to machine, but ports redirection from public address). I read this trick from Hugh Waite: I have used rtpproxy (with the advertised address patch) in Amazon to bridge media between internet facing and private subnets in a VPC. I found that I couldn’t use different advertised addresses depending on which direction the signalling was going on a single private IP address. I worked around this by allocating a second private ip address to the instance and used that in the ‘bridge’. -A 54.86.X.X/10.0.1.15 –l 10.0.1.10/10.0.1.15 Can you explain how to use this trick, or another way (without additional addresses is gladly accepted!) to reach the same result (rtp always passing through FreeSWITCH) ? Thank you all in advance, -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- VoIP Embedded, Inc. http://www.voipembedded.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] TLS certificates per domain
Hello, the SNI (server name indication) support was available in kamailio v1.5 and then lost when the code was integrated with ser. It was on my to-do to re-add it but no time for it in the past. I just pushed a partial patch that allows to set a server_name for each TLS server domain (context) configured in the tls.cfg, like: [server:127.0.0.1:5061] method = TLSv1 ... server_name = localhost.loc [server:127.0.0.1:5061] method = TLSv1 ... server_name = localhost1.loc So far I had the time to add only for server side -- when Kamailio is accepting a TLS connection, should be able to select the context with server_name matching the one advertised by the client. Soon I will add the option to set the server name for connections that are opened by kamailio towards other tls nodes. Because it is impossible to know if the client will present a SNI, kamailio first selects the context based only on ip:port matching and once the SNI callback is executed, will switch to the appropriate one. Given that there can be more contexts for same ip:port, the last one matching in tls.cfg is selected first time. If no server name is matching after SNI callback, the the 'default' server context is selected. I did just basic testing so far with SIP registration, therefore proper testing would be required on your side and feedback will be very appreciated. Cheers, Daniel On 12/02/15 15:15, Muhammad Shahzad wrote: Hi, I want to deploy a kamailio v4.2.x setup with multiple domains, all resolve to same IPv4 address kamailio is listening on. I am bit confused about how to configure TLS certificates using tls config file as mentioned here, http://kamailio.org/docs/modules/4.2.x/modules/tls.html#tls.p.config The documentation states that, -- If set the TLS module will load a special config file or config files from config directory, in which different TLS parameters can be specified on a per role (server or client) and domain basis (*for now only IPs*). The corresponding module parameters will be ignored. -- since all domains resolve single IP, so i assume i can specify only one section in tls config file with pair of key/pem file path. How can i specify more server certificates for same ip but with different domains? Thank you. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] selective logging with debugger module
Hello, I followed the documentation from http://kamailio.org/docs/modules/4.2.x/modules/debugger.html#idp84752. I have the global debug flag at 9. modparam(debugger, cfgtrace, 1) modparam(debugger, mod_level_mode, 1) modparam(debugger, mod_level, core=3) My Kamailio complains with a parsing error of some sort and fails to start. - dbg_mod_level_param(): cannot store parameter: core=3 My intention is to turn off/reduce the core module tracing while keeping the script module at debug. Any suggestions ? I am running version 4.1.4. thanks Sharath ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] rtpproxy bridge ie ei behind NAT (like in aws EC2)
Could you show the revelant codes in rtpproxy and kamailio.cfg? I am unable to get the audio pass through from extranet to intranet as private IP address is used after rtpproxy. I use Kamailio 4.2 and rtpproxy in Debian wheezy. Both are installed from repository. On Tue, Feb 17, 2015 at 8:27 AM, Ovidiu Sas o...@voipembedded.com wrote: You could simply let the RTP traffic to flow directly between FS and endpoints (no need for rtpproxy). All you need to do is: - forward the appropriate RTP ports to FS; - fix the private IP in SDP by replacing it with the public IP for the inbound rtp streams (to FS). -ovidiu On Mon, Feb 16, 2015 at 11:30 AM, Giovanni Maruzzelli gmar...@gmail.com wrote: dear Kamailians, I have Kamailio+rtpproxy in front of FreeSWITCH. Kamailio and FreeSWITCH are on the same private network. Public Internet IP address ports are redirected to Kamailio and rtpproxy (same situation as in Amazon EC2). Clients comes from Internet, and make calls to Internet, SIP signaling passing through FreeSWITCH (eg: A leg incoming INVITE, FreeSWITCH originate an outbound B leg INVITE, and then bridge the legs). Using rtpproxy with -A advertise patch from Daniel, this topology works fine in a traditional telco way: rtp goes from caller to rtpproxy to callee, and viceversa. Now I want to maintain FreeSWITCH in the middle of rtp flow all the time, in a pure b2bua way, so it can control and analyze the media streams. So, I need rtpproxy to act paying attention to direction, as in caller-rtpproxy-freeswitch-rtpproxy-callee (and viceversa). Normally I would use Kamailio multihomed and rtpproxy in bridging mode. But I cannot have a NIC on the public address. How I can use the ie ei feature of rtpproxy in an Amazon-EC2 like environment? (eg: no public address attached to machine, but ports redirection from public address). I read this trick from Hugh Waite: I have used rtpproxy (with the advertised address patch) in Amazon to bridge media between internet facing and private subnets in a VPC. I found that I couldn’t use different advertised addresses depending on which direction the signalling was going on a single private IP address. I worked around this by allocating a second private ip address to the instance and used that in the ‘bridge’. -A 54.86.X.X/10.0.1.15 –l 10.0.1.10/10.0.1.15 Can you explain how to use this trick, or another way (without additional addresses is gladly accepted!) to reach the same result (rtp always passing through FreeSWITCH) ? Thank you all in advance, -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- VoIP Embedded, Inc. http://www.voipembedded.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] dialog module 4.2 REFER
Just to add that the Contact address used for bridging can be changed via module parameter: - http://kamailio.org/docs/modules/stable/modules/dialog.html#idp1855568 Should be changed to reflect local IP of the server. Cheers, Daniel On 15/02/15 18:02, Ben Langfeld wrote: The REFER's contact header should be the referring party, and is used as the destination for NOTIFYing progress of the refer. The party to refer *to* is stated in the ReferTo header. In what way does the refer fail? Maybe you could provide logs... On 15 February 2015 at 10:38, Uri Shacked ushac...@gmail.com mailto:ushac...@gmail.com wrote: Hi, I am trying to use dlg_refer. I set the side to refer and the final destination. But, the contact header of the refer stays contro...@kamailio.org mailto:contro...@kamailio.org. So, the refer fails. Is it a bug or should i change the contact header by myself before doing the dlg_refer? same with the dlg_bridge. Cheers, Uri ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
