- sr-users - kamailio.org

Re: [Kamailio-Users] Users Digest, Vol 42, Issue 11
by luzango mfupe 04 Nov '08

04 Nov '08

> 9. Re: the sip router project (Bogdan-Andrei Iancu) > > > Mmmh! guys, i am missing some points here, does this has anything to do > with OPENSIPS-YATE Collabo? a counter attack move of some sort??? > > > ------------------------------ > > Message: 9 > Date: Tue, 04 Nov 2008 17:24:15 +0200 > From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> > Subject: Re: [Kamailio-Users] the sip router project > To: Johansson Olle E <oej(a)edvina.net> > Cc: devel <devel(a)lists.kamailio.org>, serusers(a)lists.iptel.org, > sr-dev(a)lists.sip-router.org, users <users(a)lists.kamailio.org>, SER > developer mailing list <serdev(a)iptel.org>, > business(a)lists.kamailio.org > Message-ID: <4910691F.9020509(a)voice-system.ro> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi Olle, > > Thank you for your thoughts. On a first view, it looks interesting, but I'm > missing some points here (important points): > > 1) as OpenSER was forked from SER because different views (and the OpenSER > view proved to be a very popular and successful one), I wonder > why, Kamilio is getting back to SER? not sharing any more the OpenSER view > as claimed? because such merging will definitely have a great > impact on the dynamical and openness of the projects (like releases, > contributions, driving the project) > > 2) this major change of perspective (at least for kamilio) was a backstage > decision, kept secret from the community - shouldn't be in the > interest of the community to say if going back to the roots (merging into > SER) is something wanted or not? it somehow contradicts the self > existence of OpenSER, right? > > 3) the benefits you mentions are mainly optimization of the internal > project activities and not optimizations of the outcome - what the > project will deliver. And I guess this is the most important. We already > went though the experience of large devel community, frameworks, etc but > with no outcome for more than 2 years... > > >From my personal perspective, the new project looks more like SER > absorbing Kamilio (considering the sizes, the companies behind each > project, the resources, and the man-power behind each project). > > And at the moment I would like preserve the OpenSER vision and to have an > open source project (a standalone one), far away from the "control" of any > Big Brother ;) > > Regards, > Bogdan > > > > Johansson Olle E wrote: > > Congratulations to both projects! > > > > I think this is very beneificial for both communities as well as for > > all the companies out there using the two products. > > > > I am very happy with this result and hope that the third kid in the > > SER family - OpenSIPS - will consider joining the effort too. > > > > Best regards, > > /Olle > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > > > > > > ------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > End of Users Digest, Vol 42, Issue 11 > ************************************* > -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

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[Kamailio-Users] the sip router project
by Daniel-Constantin Mierla 04 Nov '08

04 Nov '08

Hello Everybody, We are pleased to announce to you the SIP Router project. It aims to build a solid open source SIP routing platform, based on collaboration of the SIP Express Router (SER) and Kamailio (OpenSER) teams. Developers of these two projects believe that an united and non-conflicting environment will bring many benefits to them, community members and companies: * bring together the developers and user communities of both projects * reduce maintenance overhead * avoid duplicated efforts in development * develop a core framework that is flexible, extensible and scalable * promote and build a solid open source SIP server project * ensure business credibility * make future forking undesirable, this harms everybody, affects credibility and business You are welcome to join! Visit the web site at: http://sip-router.org There is a meeting in Karlsruhe, Germany, on Nov 10, 2008, hosted by 1&1, where the developers and community members have the chance to discuss and tune the last aspects of the new project. We are looking to see many of you there: http://sip-router.org/index.php/meeting/ We hope this is a great news for you, thanks to the effort of main developers and management teams of the two projects. We invite you to join the new mailing list for further discussions: http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev Daniel, Jiri Kamailio Management Team SER Management Team

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[Kamailio-Users] Unknown column 'from_uri' in 'field list'
by Kaeman Chris 04 Nov '08

04 Nov '08

Hello, modparam("acc", "db_extra", "from_uri=$fu; to_uri=$tu") added 2 new fields in the acc table. Logs giving following errors while making calls Nov 4 07:14:11 [4567] ERROR:db_mysql:db_mysql_submit_query: driver error on query: Unknown column 'from_uri' in 'field list' Nov 4 07:14:11 [4567] ERROR:core:db_do_insert: error while submitting query Nov 4 07:14:11 [4567] ERROR:acc:acc_db_request: failed to insert into database Is this error due to the added fields? Thanks, KChris Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/

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[Kamailio-Users] SER config file adaptation to Kamailio ?
by Samuel Muller 04 Nov '08

04 Nov '08

Hello all, I tried during one week to adapt my old SER configuration file (ser.cfg, for SER 2.0.0), who works really great, to Kamailio v1.4.2. I used to change all the stuff needed (many functions, onsend route, ... disappeared or was replaced, ...). That's ok, concepts are the same - registration are ok, NAT (seems to) works with rtpproxy, and I'm using radius server (Radiator). MySQL is only here to store locations, domains, and trusted stuff like gateways. Everything looks fine (load kamalio : no pbm, REGISTER, OPTIONS methods and SIP ping ok, ...) Right now, RTP does not work, and no SIP phone (linksys spa9xx, thomson st2030, ...) can't call - there's no ringing, even if all the sip phones are registered and known by OpenSER ... It seems OpenSER can't send the SIP requests to the callee ??? Did anyone had the same problem and resolved it ? Or, eventually anyone does have some config sample config file (as you can have in SER examples stuff) ? I would like to let here a debug log of a call from a linksys spa942 to a sjphone but it's a bit enormous (debug log + config file) ... so i let it in attached file. the scenario is : 0123451011 (sjphone) calls 0123451012 (linksys spa9xx) the 2 UACs are on the same LAN, through the same router, with the same realm. there're registered and are stored into the location table in mySQL. but the problem is present in all the case we can have (different realms, different WAN connections, different SIP UACs ...) Thanks in advance ! -- Samuel MULLER

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[Kamailio-Users] Remote-Party-ID header
by ALAEDDINE abbech 04 Nov '08

04 Nov '08

hi all, Please, what is the role of Remote-Party-ID header? thanks

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[Kamailio-Users] PUA API send to certain IP
by Schumann Sebastian 04 Nov '08

04 Nov '08

Anca, All, I have the following scenario UE - OpenSER Proxy/Presence Server - OpenSER Registrar The proxy acts as proxy and presence server. If the UE sends a subscribe to user@registrar, it handles this subscription. If it publishes its state user@registrar it does as well. All registration/call control is proxied to the registrar. I try now to use the PUA module API to create PUBLISH on the Proxy/Presence Server. This works fine, the only issue is that it has the request-uri user@registrar, like UE generated subscriptions. This is actually not an issue, but wanted. The problem is that the proxy forwards the package to the registrar acc. R-URI. I want it sent to another IP. Is there the possibility to define the IP where the package should be sent to, i.e. presence server IP, to handle the packet there? Thanks! Best regards, Sebastian

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[Serusers] How to use SRV records
by Someone beneath the blue sky 04 Nov '08

04 Nov '08

Hello all, I'm pretty new to the SIP world and appreciate if somebody can help me out. In the SerWeb login page I noticed a link "Have-My-Domain!", and it looks like that the result would be having something similar to email forwarding service. Suppose that I have my own domain (e.g. mydomain.com), and I have added an SRV record to it and pointed it to iptel's sip server. Then following the link I was able to create a new username (e.g. user01). Now based on what I've understood, whenever somebody calls the sip uri user01(a)mydomain.com my soft phone that is registered with the above account should ring (my soft phone is using user01 for username, mydomain.com for server and iptel.org for registerar and it seems that it is registered without any problem). My questions: 1. What prevents others from impersonating a new username in my domain ? It seems that if somebody knows that mydomain.com has an SRV record pointed to iptel.org, can create another username (user02(a)mydomain.com) and use it without my knowledge, what is worse is that my DNS provider doesn't have any logging options for DNS requests, so I'll never find out about it. 2. In my account in iptel ( user01(a)mydomain.com ), there is another sip uri ( something like 123456(a)mydomain.com ), which I guess is supposed to be used by somebody who uses a device without alphabets (for example if somebody wants to call me via sipbroker gateways), I've already tried *478 - 123456 ( *478 is the code for iptel), but it doesn't work and it goes to voice mail, how can I use sipbroker to call user01(a)mydomain.com ? Thanks a lot in advance and sorry for the long message ! Someone

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[Kamailio-Users] Proxy IAX ?
by BERGANZ François 04 Nov '08

04 Nov '08

Is there a proxy IAX ? Openser can do it ? Thank you Cordialement BERGANZ François

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[Serusers] problem with SER not proxying messages
by amar mahmoud 03 Nov '08

03 Nov '08

>> Hi all, >> i am new to use SER. i want to build a test bed using SER as proxy server and SIPp as user agents to make some measurements as follows: >> >> A -------------------------- SER ---------------------------------- B >> SIPp SIPp >> >> >> i have a problem using SER : >> I did registration for the user agents with the server and i can see them in location table in SER database, when A start a call to B the server respond to the INVITE with 100 trying then can't relay the message to B until A gets 408 request timeout , it seems that the server cannot retrieve the contact of B from the database. >> >> the Request URI i am using is sip:B@192.168.165.18 which is the IP of the server and i tried also using alias like ser.org so it became sip:B@ser.org but still have the same problem. >> >> i m putting below my config file ser.cfg >> >> i have also some questions: >> >> 1/ the domain field of location table in database is empty, so is this normal situation, actually i did set it manually to my domain. >> 2/ also i have noticed that when using the command : serctl moni , i got the following message: >> root@amar-desktop:~# serctl moni >> [: 1074: ==: unexpected operator >> [: 1074: 0: unexpected operator >> >> despite that ser is starting by using the command : serctl start, and i can see it is starting and even there is no error in /var/log/syslog. >> >> 3/ is there a way to view the sip messages traversing the server, i mean to monitor it., and is /var/log/syslog the only place to see the logs. >> >> i would be thankful if someone can help me with that cause i have task to do and i got stuck beside having very limited time. >> >> >> Amar >> >> >> >> >> _________________________________________________________________ >> You live life beyond your PC. So now Windows goes beyond your PC. >> http://clk.atdmt.com/MRT/go/115298556/direct/01/ > > _________________________________________________________________ > You live life beyond your PC. So now Windows goes beyond your PC. > http://clk.atdmt.com/MRT/go/115298556/direct/01/ _________________________________________________________________ When your life is on the go—take your life with you. http://clk.atdmt.com/MRT/go/115298558/direct/01/

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[Serusers] problem with SER not proxying messages
by amar mahmoud 03 Nov '08

03 Nov '08

> > Hi all, > i am new to use SER. i want to build a test bed using SER as proxy server and SIPp as user agents to make some measurements as follows: > > A -------------------------- SER ---------------------------------- B > SIPp SIPp > > > i have a problem using SER : > I did registration for the user agents with the server and i can see them in location table in SER database, when A start a call to B the server respond to the INVITE with 100 trying then can't relay the message to B until A gets 408 request timeout , it seems that the server cannot retrieve the contact of B from the database. > > the Request URI i am using is sip:B@192.168.165.18 which is the IP of the server and i tried also using alias like ser.org so it became sip:B@ser.org but still have the same problem. > > i m putting below my config file ser.cfg > > i have also some questions: > > 1/ the domain field of location table in database is empty, so is this normal situation, actually i did set it manually to my domain. > 2/ also i have noticed that when using the command : serctl moni , i got the following message: > root@amar-desktop:~# serctl moni > [: 1074: ==: unexpected operator > [: 1074: 0: unexpected operator > > despite that ser is starting by using the command : serctl start, and i can see it is starting and even there is no error in /var/log/syslog. > > 3/ is there a way to view the sip messages traversing the server, i mean to monitor it., and is /var/log/syslog the only place to see the logs. > > i would be thankful if someone can help me with that cause i have task to do and i got stuck beside having very limited time. > > > Amar > > > > > _________________________________________________________________ > You live life beyond your PC. So now Windows goes beyond your PC. > http://clk.atdmt.com/MRT/go/115298556/direct/01/ _________________________________________________________________ You live life beyond your PC. So now Windows goes beyond your PC. http://clk.atdmt.com/MRT/go/115298556/direct/01/

1 0

[Serusers] problem with SER not proxying messages
by amar mahmoud 03 Nov '08

03 Nov '08

Hi all, i am new to use SER. i want to build a test bed using SER as proxy server and SIPp as user agents to make some measurements as follows: A -------------------------- SER ---------------------------------- B SIPp SIPp i have a problem using SER : I did registration for the user agents with the server and i can see them in location table in SER database, when A start a call to B the server respond to the INVITE with 100 trying then can't relay the message to B until A gets 408 request timeout , it seems that the server cannot retrieve the contact of B from the database. the Request URI i am using is sip:B@192.168.165.18 which is the IP of the server and i tried also using alias like ser.org so it became sip:B@ser.org but still have the same problem. i m putting below my config file ser.cfg i have also some questions: 1/ the domain field of location table in database is empty, so is this normal situation, actually i did set it manually to my domain. 2/ also i have noticed that when using the command : serctl moni , i got the following message: root@amar-desktop:~# serctl moni [: 1074: ==: unexpected operator [: 1074: 0: unexpected operator despite that ser is starting by using the command : serctl start, and i can see it is starting and even there is no error in /var/log/syslog. 3/ is there a way to view the sip messages traversing the server, i mean to monitor it., and is /var/log/syslog the only place to see the logs. i would be thankful if someone can help me with that cause i have task to do and i got stuck beside having very limited time. Amar _________________________________________________________________ You live life beyond your PC. So now Windows goes beyond your PC. http://clk.atdmt.com/MRT/go/115298556/direct/01/

1 0

[Kamailio-Users] irc devel meeting
by Daniel-Constantin Mierla 03 Nov '08

03 Nov '08

Hello, being the time for a new IRC devel meeting, I propose Thursday, Nov 6, 2008, at 15:00UTC, on channel #kamailio, server irc.freenode.net The focus is to sync the developers related the work since the last major release, roadmap to next major release as well as minor releases scheduling. Please add to the agenda at: http://www.kamailio.org/dokuwiki/doku.php/development:irc-meeting-agenda-dr… Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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Re: [Kamailio-Users] re mote clients don't contact other remote clients
by Ricky Gutierrez 03 Nov '08

03 Nov '08

________________________________ Monday, November 3, 2008 3:28:30 AM arif.zaman Wrote: > Did you check the location table whether callee is registered before > forwarding any call? that is strange that I see in the table location it, registered : +----+----------+------------------------------------------+------------------------+------------------------------+ | id | username | contact | received | user_agent | +----+----------+------------------------------------------+------------------------+------------------------------+ | 1 | 122 | sip:122@192.168.0.60:5065 | sip:190.184.35.4:5065 | Linksys/SPA942-5.2.8 | | 2 | 201 | sip:201@192.168.11.30:5060;transport=udp | sip:192.168.11.30:5060 | Grandstream BT200 1.1.4.18 | | 4 | 202 | sip:202@192.168.10.30:5064;transport=udp | sip:190.212.5.155:5064 | Grandstream GXP2020 1.1.6.16 | +----+----------+------------------------------------------+------------------------+------------------------------+ regardsss rickygm ________________________________

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[Kamailio-Users] INVITE accounting problem
by Kaeman Chris 03 Nov '08

03 Nov '08

Hello, I have a problem in accounting INVITE. When the call is connected with duration the INVITE, ACK, BYE are accounted in db. But the problem is when the call is disconnected before it is actually attempting to connect. i am getting only CANCEL accounted whereas INVITE is not accounted for the cancelled call. I have the xlogs showing INVITE and CANCEL both for this call only no accounting is performed for INVITE. following is the config for INVITE & CANCEL; if (is_method("INVITE")) { setflag(1); setflag(2); # MISSED if (allow_trusted()) { rewritehostport("xx.xxx.xy.xyz:5060"); route(1); } else { sl_send_reply("401", "Unauthorized"); route(1); exit; } } # CANCEL processing if (is_method("CANCEL")) { setflag(1); setflag(3); if (t_check_trans()) t_relay(); exit; } t_check_trans(); Please let me know what should be changed so that for the Call that is cancelled before it actually connects can have INVITE also accounted along w/ CANCEL. Thanks, KChris. Connect with friends all over the world. Get Yahoo! India Messenger at http://in.messenger.yahoo.com/?wm=n/

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[Kamailio-Users] restrict calls from gateway to some extensions
by muhammad akl 03 Nov '08

03 Nov '08

Hey all , I want to make some ACL for the INVITES come from a PSTN-GW to certain users on my OpenSER server. for ex: I have the following extensions on my OpenSER server : 11001 11002 | | | | 11111 What I want to implement is something similar to this : Caller <-------> PSTN-GW <------->OpenSER <----> (complete this call to this extension ) 11001 Caller <-------> PSTN-GW <------->OpenSER <----> (reject this call for this user ) 11002 and so on for other extensions . any way to implement this feature in openser ?

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[Kamailio-Users] remote clients don't contact other remote clients
by Ricky Gutierrez 03 Nov '08

03 Nov '08

Hello list, I have another question and a very strange detail, I have a couple of connected remote clients to my kamailio 1.3.2 and mediaproxy 1.9.1, when the client 122 calls to the client 202 this very contacts or, if I call a minute I contact him very well, but if I call to the client 122 after some minutes I don't contact with that client and the jumps me to the voicemail . I see in the chart location the address ip of the UAC that it can be happening? regards rickygm 122 UAC <-> NAT <-> internet==== kamailio 1.3.2 / mediaproxy ==== internet <-> ADSL <-> NAT <-> UAC 202 public ip: x.x.x.x.x interface: any filter: (ip) and ( port 5060 ) # U +0.890415 190.212.35.223:5064 -> 190.184.22.152:5060 INVITE sip:122@190.184.22.152 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK1eff324647d9fef9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@192.168.10.30:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 390 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 238 v=0 o=202 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.10.30 t=0 0 m=audio 5004 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.000286 190.184.22.152:5060 -> 190.212.35.223:5064 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK1eff324647d9fef9;received=190.212.35.223 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152>;tag=329cfeaa6ded039da25ff8cbb8668bd2.ea16 Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 390 INVITE Proxy-Authenticate: Digest realm="190.184.22.152", nonce="490eb5583ea2f71836195c92a7e7df100eec3075" Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 # U +0.197976 190.212.35.223:5064 -> 190.184.22.152:5060 ACK sip:122@190.184.22.152 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK1eff324647d9fef9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152>;tag=329cfeaa6ded039da25ff8cbb8668bd2.ea16 Contact: <sip:202@192.168.10.30:5064;transport=udp> Supported: path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 390 ACK User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 # U +0.073962 190.212.35.223:5064 -> 190.184.22.152:5060 INVITE sip:122@190.184.22.152 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@192.168.10.30:5064;transport=udp> Supported: replaces, timer, path Proxy-Authorization: Digest username="202", realm="190.184.22.152", algorithm=MD5, uri="sip:122@190.184.22.152", nonce="490eb5583ea2f71836195c92a7e7df100eec3075", response="aa82c7aaba810bcc0b46d7e7ea442603" Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 238 v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.10.30 t=0 0 m=audio 5004 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.003823 190.184.22.152:5060 -> 190.212.35.223:5064 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK7288d140481a11e9;rport=5064;received=190.212.35.223 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 # U +0.000053 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.485936 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.999801 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +1.999612 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +3.999219 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +3.999220 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +4.001208 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +3.599535 190.184.22.152:5060 -> 192.168.11.1:5070 INVITE sip:u122@192.168.11.1:5070 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.1 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.000362 190.184.22.152:5070 -> 190.184.22.152:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.1;received=190.184.22.152 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:u122@190.184.22.152:5070> Content-Length: 0 # U +0.000283 190.184.22.152:5070 -> 190.184.22.152:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.1;received=190.184.22.152 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152>;tag=as6ee3962f Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:u122@190.184.22.152:5070> Content-Type: application/sdp Content-Length: 289 v=0 o=root 9069 9069 IN IP4 190.184.22.152 s=session c=IN IP4 190.184.22.152 t=0 0 m=audio 15334 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv # U +0.002152 190.184.22.152:5060 -> 190.212.35.223:5064 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152>;tag=as6ee3962f Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:u122@190.184.22.152:5070> Content-Type: application/sdp Content-Length: 289 P-hint: onreply_route|usemediaproxy v=0 o=root 9069 9069 IN IP4 190.184.22.152 s=session c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv # U +0.236892 190.212.35.223:5064 -> 190.184.22.152:5060 ACK sip:u122@190.184.22.152:5070 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK0fd7eaebf7c6f6c6 Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152>;tag=as6ee3962f Contact: <sip:202@192.168.10.30:5064;transport=udp> Supported: path Proxy-Authorization: Digest username="202", realm="190.184.22.152", algorithm=MD5, uri="sip:122@190.184.22.152", nonce="490eb5583ea2f71836195c92a7e7df100eec3075", response="aa82c7aaba810bcc0b46d7e7ea442603" Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 ACK User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 # U +0.000248 190.184.22.152:5060 -> 190.184.22.152:5070 ACK sip:u122@190.184.22.152:5070 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.3 Via: SIP/2.0/UDP 192.168.10.30:5064;received=190.212.35.223;branch=z9hG4bK0fd7eaebf7c6f6c6 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152>;tag=as6ee3962f Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: path Proxy-Authorization: Digest username="202", realm="190.184.22.152", algorithm=MD5, uri="sip:122@190.184.22.152", nonce="490eb5583ea2f71836195c92a7e7df100eec3075", response="aa82c7aaba810bcc0b46d7e7ea442603" Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 ACK User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 P-hint: LR|fixcontact,setflag6 exit 40 received, 0 dropped

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Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by Ricky Gutierrez 02 Nov '08

02 Nov '08

Hi list, always looking for as solving my audio problem with mediaproxy asterisk and openser, there will be some form of telling to the openser that when he comes from the from sip:asterisk@192.168.10.1:5070 that doesn't use the mediaproxy or the onreply_route[1] , SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK36b7f619;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as42edbc9b;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as42edbc9b To: <sip:113@192.168.10.1>;tag=6d45d2188218c8ef Call-ID: 5a1c60382f18bd832fa3bdc54dc6ab13(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 212 P-hint: Onreply-route - fixcontact P-hint: onreply_route|usemediaproxy v=0 o=113 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.1.64 t=0 0 m=audio 35064 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 making other tests if I change this form the onreply_route[1], I have audio in the openser extension, but the one that this behind the pstn doesn't have audio or he doesn't listen to me onreply_route[1] { # #-- On-replay block routing -- # if (client_nat_test("1")) { append_hf("P-hint: Onreply-route - fixcontact \r\n"); fix_nated_contact(); }; if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (search("^Content-Type:[ ]*application/sdp")) { append_hf("P-hint: onreply_route|usemediaproxy \r\n"); use_media_proxy(); }; }; exit; } my best regardss rickygm

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Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by Ricky Gutierrez 31 Oct '08

31 Oct '08

________________________________ Friday, October 31, 2008 2:59:58 AMDaniel-Constantin Mierla wrote: > do you have to bridge the RTP between the two network interfaces? >I do not know mediaproxy, but rtpproxy (nathelper module) has bridge mode >that can be used in such scenarios. Hi Daniel, I don't have a bridge with the two net cards, I had read some rtpproxy that can be used in bridge mode, but I didn't find alone examples some comments, but he gave me an error and I could not make it work this problem has me sad... and I am not seeing an exit.. regards .. rickygm

1 0

Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by Ricky Gutierrez 31 Oct '08

31 Oct '08

Thank you Luciano for your answer but doesn't it still work me, do the calls enter me of the pstn and don't I listen anything... where I can look for? , there will be some of form so that the IP of where the calls come from asterisk does not use mediaproxy, somebody has an example. any help is appreciated.. thank you rickygm ________________________________ From: luzango mfupe <luzango.mfupe(a)gmail.com> Sent: Friday, October 31, 2008 5:19:25 AM Subject: Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy Hi Ricky, I think you should increase the mediaproxy natping interval to 30 and then add use_media_proxy() line after you setflag(6). check below:--- ##Detect requests in the dialog behind NAT and flag with 6 if(nat_uac_test("19") || search("^Route:.*;nat=yes")){ append_hf("P-hint: LR|fixcontact,setflag6\r\n"); fix_contact(); setbflag(6); USE MEDIA PROXY HERE---> use_media_proxy(); }; route(1); } else { sl_send_reply("404", "Not here"); } exit; } On Fri, Oct 31, 2008 at 5:21 AM, Ricky Gutierrez <xserverlinux(a)yahoo.com> wrote: I show you here my openser.cfg my best regards rickygm loadmodule "nathelper.so" loadmodule "mediaproxy.so" modparam("registrar", "received_avp", "$avp(i:42)") modparam("usrloc", "db_mode", 2) modparam("usrloc", "nat_bflag", 4) modparam("nathelper","rtpproxy_sock", "") modparam("nathelper","natping_interval", 0) modparam("nathelper","received_avp", "$avp(i:42)") modparam("mediaproxy","natping_interval",20) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy","sip_asymmetrics","/etc/openser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/ect/openser/rtp-clients") route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!is_method("REGISTER")) { if(nat_uac_test("19")){ record_route(";nat=yes"); } else { record_route(); }; }; if (has_totag()) { # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { if(method=="BYE") { #Account BYE transactions setflag(2); }; if (method=="INVITE" && (!allow_trusted())) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Forbidden, use From=ID"); exit; }; }; if(method=="BYE" || method=="CANCEL") { end_media_session(); }; ##Detect requests in the dialog behind NAT and flag with 6 if(nat_uac_test("19") || search("^Route:.*;nat=yes")){ append_hf("P-hint: LR|fixcontact,setflag6\r\n"); fix_contact(); setbflag(6); }; route(1); } else { sl_send_reply("404", "Not here"); } exit; } ##Loose_route packets if (loose_route()) { if(!has_totag()){ sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; }; route(1); }; #CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { end_media_session(); t_relay(); }; exit; } t_check_trans(); if (method=="REGISTER"){ route(2); } else { route(3); }; } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP t_on_reply("1"); t_on_failure("1"); if (!t_relay()) { ; if (method=="INVITE" || method=="ACK") { ; end_media_session(); ; }; # sl_reply_error(); }; exit; } route[2] { # # -- Register request handler -- # if (is_uri_host_local()) { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; }; if (!check_to()) { sl_send_reply("403", "Forbidden"); exit; }; sl_send_reply("100", "Trying"); if(!search("^Contact:[ ]*\*") && client_nat_test("7")) { setbflag(6); fix_nated_register(); force_rport(); }; save("location"); exit; } else if { sl_send_reply("403", "Forbidden"); }; } route[3] { # # -- INVITE request handler -- # if (method=="INVITE") { # Account INVITE packets setflag(2); # Account Missed calls setflag(3); }; if (is_from_local()){ # From an internal domain -> check the credentials and the FROM if(!allow_trusted()){ if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Forbidden, use From=ID"); exit; }; }; if (client_nat_test("3")) { append_hf("P-hint: route(3)|setflag7,forcerport,fix_contact\r\n"); setbflag(7); force_rport(); fix_contact(); }; if(avp_db_load("$ru/username","$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); route(4); route(1); exit; } consume_credentials(); # Verify aliases lookup("aliases"); if (is_uri_host_local()) { # -- Inbound to Inbound route(10); } else { # -- Inbound to outbound route(11); }; } else { # From an external domain -> do not check credentials #Verify aliases, if found replace R-URI. lookup("aliases"); if (is_uri_host_local()) { #-- Outbound to inbound route(12); } else { # -- Outbound to outbound route(13); }; }; } route[6] { # # -- NAT handling -- # if (isbflagset(6) || isbflagset(7)) { append_hf("P-hint: Route[6]: mediaproxy \r\n"); use_media_proxy(); }; } route[10] { #from an internal domain -> inbound #Native SIP destinations are handled using the location table append_hf("P-hint: inbound->inbound \r\n"); if (uri=~"^sip:[2346578][0-9]{6}@.*") { if (is_user_in("credentials", "local")){ # rewritehostport("192.168.10.1:5070"); route(6); route(4); exit; } else { sl_send_reply("403", "No tienes permiso para llamadas locales"); exit; }; }; if (!lookup("location")) { if (does_uri_exist()) { revert_uri(); prefix("u"); rewritehostport("192.168.10.1:5070"); route(6); route(1); } else { sl_send_reply("404", "Not Found"); exit; }; sl_send_reply("404", "Not Found"); exit; }; route(6); route(1); } route[11] { # from an internal domain -> outbound # Simply route the call outbound using DNS search append_hf("P-hint: inbound->outbound \r\n"); route(1); } route[12] { # From an external domain -> inbound # Verify aliases, if found replace R-URI. lookup("aliases"); if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; route(1); } route[13] { #From an external domain outbound #we are not accepting these calls append_hf("P-hint: outbound->inbound \r\n"); sl_send_reply("403", "Forbidden"); exit; } route[4] { rewritehostport("192.168.10.1:5070"); route(1); } ### aqui se detecta cunado una llamada es no disponible , ocupada o falla failure_route[1] { if (t_was_cancelled()) { exit; }; if (t_check_status("486")) { revert_uri(); prefix("b"); xlog("L_ERR","Stepped into the 486 ruri=<$ru>"); rewritehostport("192.168.10.1:5070"); append_branch(); route(1); exit; }; if (t_check_status("408") || t_check_status("480")) { revert_uri(); prefix("u"); xlog("L_ERR","Stepped into the 480 ruri=<$ru>"); rewritehostport("192.168.10.1:5070"); append_branch(); route(1); exit; }; } onreply_route[1] { # #-- On-replay block routing -- # if (client_nat_test("1")) { append_hf("P-hint: Onreply-route - fixcontact \r\n"); fix_nated_contact(); }; if ((isbflagset(6) || isbflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (search("^Content-Type:[ ]*application/sdp")) { append_hf("P-hint: onreply_route|usemediaproxy \r\n"); use_media_proxy(); }; }; exit; } -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

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[Serusers] SER/FIFO not accessible
by Shruti Sanadhya 31 Oct '08

31 Oct '08

Hi all, I am trying to setup a SER to support calls on Kphone. I am trying to add users to the mysql SER database using: serctl 1000 1000 1000(a)mydomain.com but I am getting the 'SER/FIFO not accessible' error. I tried it using both the SER and root password for MySql, but no luck. I have also added 'fifo_mode=0777" in the configuration file, but the error still persists. Please advise, Thanks!!

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Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by luzango mfupe 31 Oct '08

31 Oct '08

Hi Ricky, I think you should increase the mediaproxy natping interval to 30 and then add use_media_proxy() line after you setflag(6). check below:--- ##Detect requests in the dialog behind NAT and flag with 6 if(nat_uac_test("19") || search("^Route:.*;nat=yes")){ append_hf("P-hint: LR|fixcontact,setflag6\r\n"); fix_contact(); setbflag(6); *USE MEDIA PROXY HERE---> use_media_proxy(); *}; route(1); } else { sl_send_reply("404", "Not here"); } exit; } On Fri, Oct 31, 2008 at 5:21 AM, Ricky Gutierrez <xserverlinux(a)yahoo.com>wrote: > > I show you here my openser.cfg > > my best regards > > rickygm > > loadmodule "nathelper.so" > loadmodule "mediaproxy.so" > > > modparam("registrar", "received_avp", "$avp(i:42)") > modparam("usrloc", "db_mode", 2) > modparam("usrloc", "nat_bflag", 4) > > modparam("nathelper","rtpproxy_sock", "") > > modparam("nathelper","natping_interval", 0) > modparam("nathelper","received_avp", "$avp(i:42)") > modparam("mediaproxy","natping_interval",20) > modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") > modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") > > modparam("mediaproxy","sip_asymmetrics","/etc/openser/sip-clients") > modparam("mediaproxy","rtp_asymmetrics","/ect/openser/rtp-clients") > > > route{ > # initial sanity checks -- messages with > # max_forwards==0, or excessively long requests > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483","Too Many Hops"); > exit; > }; > if (msg:len >= 2048 ) { > sl_send_reply("513", "Message too big"); > exit; > }; > # we record-route all messages -- to make sure that > # subsequent messages will go through our proxy; that's > # particularly good if upstream and downstream entities > # use different transport protocol > if (!is_method("REGISTER")) { > if(nat_uac_test("19")){ > record_route(";nat=yes"); > } else { > record_route(); > }; > }; > > if (has_totag()) { > # subsequent messages withing a dialog should take the > # path determined by > record-routing > if (loose_route()) { > if(method=="BYE") { > #Account BYE transactions > setflag(2); > }; > if (method=="INVITE" && (!allow_trusted())) { > if (!proxy_authorize("","subscriber")) { > proxy_challenge("","0"); > exit; > } else if (!check_from()) { > sl_send_reply("403", "Forbidden, use From=ID"); > exit; > > }; > > }; > if(method=="BYE" || method=="CANCEL") { > end_media_session(); > }; > > ##Detect requests in the dialog behind NAT and flag with 6 > if(nat_uac_test("19") || search("^Route:.*;nat=yes")){ > append_hf("P-hint: LR|fixcontact,setflag6\r\n"); > fix_contact(); > setbflag(6); > }; > route(1); > } else { > sl_send_reply("404", "Not here"); > } > exit; > } > > ##Loose_route packets > if > (loose_route()) { > if(!has_totag()){ > sl_send_reply("403", "Initial Loose-Routing Rejected"); > exit; > }; > > > route(1); > }; > > > #CANCEL processing > if (is_method("CANCEL")) { > if (t_check_trans()) { > end_media_session(); > t_relay(); > }; > exit; > } > t_check_trans(); > > if (method=="REGISTER"){ > route(2); > } else { > route(3); > }; > } > > route[1] { > # send it out now; use stateful forwarding as it works reliably > # even for UDP2TCP > t_on_reply("1"); > t_on_failure("1"); > if (!t_relay()) { > ; if (method=="INVITE" || method=="ACK") { > ; end_media_session(); > ; }; > > > # sl_reply_error(); > }; > exit; > } > route[2] { > # > # -- Register request handler -- > # > if (is_uri_host_local()) { > if (!www_authorize("", "subscriber")) { > > www_challenge("", "0"); > exit; > }; > if (!check_to()) { > sl_send_reply("403", "Forbidden"); > exit; > }; > sl_send_reply("100", "Trying"); > if(!search("^Contact:[ ]*\*") && client_nat_test("7")) { > setbflag(6); > fix_nated_register(); > force_rport(); > }; > > save("location"); > exit; > } else if { > sl_send_reply("403", "Forbidden"); > }; > } > > route[3] { > # > # -- INVITE request handler -- > # > if (method=="INVITE") { > # Account INVITE packets > setflag(2); > # Account Missed calls > setflag(3); > }; > if (is_from_local()){ > # From an internal domain -> check the credentials and the FROM > if(!allow_trusted()){ > if (!proxy_authorize("","subscriber")) { > proxy_challenge("","0"); > exit; > } else if (!check_from()) { > sl_send_reply("403", "Forbidden, use From=ID"); > exit; > > }; > > }; > if (client_nat_test("3")) { > append_hf("P-hint: route(3)|setflag7,forcerport,fix_contact\r\n"); > setbflag(7); > force_rport(); > fix_contact(); > }; > > if(avp_db_load("$ru/username","$avp(s:callfwd)")) { > avp_pushto("$ru", "$avp(s:callfwd)"); > route(4); > route(1); > > exit; > } > > consume_credentials(); > > # Verify aliases > lookup("aliases"); > if (is_uri_host_local()) { > # -- Inbound to Inbound > > route(10); > } else { > # -- Inbound to outbound > route(11); > }; > > } else { > # From an external domain -> do not check credentials > #Verify aliases, if found replace R-URI. > lookup("aliases"); > if (is_uri_host_local()) { > #-- Outbound to inbound > route(12); > } else { > # -- Outbound to outbound > route(13); > }; > }; > } > route[6] { > > # > # -- NAT handling -- > # > if (isbflagset(6) || isbflagset(7)) { > append_hf("P-hint: Route[6]: mediaproxy \r\n"); > use_media_proxy(); > }; > } > > route[10] { > #from an internal domain -> inbound > #Native SIP destinations are handled using the location table > append_hf("P-hint: inbound->inbound \r\n"); > if (uri=~"^sip:[2346578][0-9]{6}@.*") { > if (is_user_in("credentials", "local")){ > # rewritehostport("192.168.10.1:5070"); > route(6); > route(4); > exit; > } else { > sl_send_reply("403", "No tienes permiso para llamadas locales"); > exit; > }; > }; > > if (!lookup("location")) { > if (does_uri_exist()) { > revert_uri(); > prefix("u"); > rewritehostport("192.168.10.1:5070"); > route(6); > route(1); > } else { > sl_send_reply("404", "Not Found"); > exit; > }; > sl_send_reply("404", "Not Found"); > exit; > }; > > > route(6); > route(1); > } > > route[11] { > # from an internal domain -> outbound > # Simply route the call outbound using DNS search > append_hf("P-hint: inbound->outbound \r\n"); > route(1); > } > route[12] { > # From an external domain -> inbound > # Verify aliases, if found replace R-URI. > lookup("aliases"); > if (!lookup("location")) { > sl_send_reply("404", "Not Found"); > exit; > }; > route(1); > } > route[13] { > #From an external domain outbound > #we are not accepting these calls > append_hf("P-hint: outbound->inbound \r\n"); > sl_send_reply("403", "Forbidden"); > exit; > } > > > route[4] { > rewritehostport("192.168.10.1:5070"); > route(1); > > } > > ### aqui se detecta cunado una llamada es no disponible , ocupada o falla > failure_route[1] { > if (t_was_cancelled()) { > exit; > }; > > if (t_check_status("486")) { > revert_uri(); > > prefix("b"); > xlog("L_ERR","Stepped into the 486 ruri=<$ru>"); > rewritehostport("192.168.10.1:5070"); > append_branch(); > route(1); > exit; > }; > if (t_check_status("408") || t_check_status("480")) { > revert_uri(); > prefix("u"); > xlog("L_ERR","Stepped into the 480 ruri=<$ru>"); > rewritehostport("192.168.10.1:5070"); > append_branch(); > route(1); > exit; > }; > } > onreply_route[1] { > # > #-- On-replay block routing -- > # > if (client_nat_test("1")) { > append_hf("P-hint: Onreply-route - fixcontact \r\n"); > fix_nated_contact(); > }; > > if ((isbflagset(6) || isbflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { > if (search("^Content-Type:[ ]*application/sdp")) { > append_hf("P-hint: onreply_route|usemediaproxy \r\n"); > use_media_proxy(); > }; > }; > > exit; > } > > > > > > > -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

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[Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by Ricky Gutierrez 31 Oct '08

31 Oct '08

Hi list is making tests with openser 1.3.2 and mediaproxy to solve the nat, I have gotten myself an ip it public with my supplier, I have two network cards in the pc that I am using for openser and mediaproxy together with asterisk, making tests with mediaproxy 1.9.1 when I receive a call from the pstn through asterisk I don't have audio, if I call to the pstn they listen to me well . From: "Ventas" <sip:112@192.168.10.1>;tag=69451218021829df To: <sip:2685249@192.168.10.1>;tag=329cfeaa6ded039da25ff8cbb8668bd2.b1b2 Contact: <sip:112@192.168.10.30:5060;transport=udp> Supported: path Call-ID: fb5f5dac83056f72(a)192.168.10.30 CSeq: 7492 ACK User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 # U +0.022110 192.168.10.30:5060 -> 192.168.10.1:5060 INVITE sip:2685249@192.168.10.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKf428b928c25dad03 From: "Ventas" <sip:112@192.168.10.1>;tag=69451218021829df To: <sip:2685249@192.168.10.1> Contact: <sip:112@192.168.10.30:5060;transport=udp> Supported: replaces, timer, path Proxy-Authorization: Digest username="112", realm="192.168.10.1", algorithm=MD5, uri="sip:2685249@192.168.10.1", nonce="4907ac8cb6dc757eb6ba5522e0fdb9786b4c3d6e", response="c40a9387fdf5de29115c1edadc7f79db" Call-ID: fb5f5dac83056f72(a)192.168.10.30 CSeq: 7493 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 358 v=0 o=112 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.10.30 t=0 0 m=audio 5004 RTP/AVP 0 18 3 97 2 9 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:2 G726-32/8000 a=rtpmap:9 G722/16000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.003938 192.168.10.1:5060 -> 192.168.10.30:5060 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKf428b928c25dad03;rport=5060 From: "Ventas" <sip:112@192.168.10.1>;tag=69451218021829df To: <sip:2685249@192.168.10.1> Call-ID: fb5f5dac83056f72(a)192.168.10.30 CSeq: 7493 INVITE Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 # U +0.000115 192.168.10.1:5060 -> 192.168.10.1:5070 INVITE sip:2685249@192.168.10.1:5070 SIP/2.0 Record-Route: <sip:192.168.10.1;lr=on;ftag=69451218021829df;nat=yes> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK400f.b93e5c35.0 Via: SIP/2.0/UDP 192.168.10.30:5060;rport=5060;branch=z9hG4bKf428b928c25dad03 From: "Ventas" <sip:112@192.168.10.1>;tag=69451218021829df To: <sip:2685249@192.168.10.1> Contact: <sip:112@192.168.10.30:5060;transport=udp> Supported: replaces, timer, path Call-ID: fb5f5dac83056f72(a)192.168.10.30 CSeq: 7493 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 358 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=112 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.1.64 t=0 0 m=audio 35040 RTP/AVP 0 18 3 97 2 9 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:2 G726-32/8000 a=rtpmap:9 G722/16000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.000471 192.168.10.1:5070 -> 192.168.10.1:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK400f.b93e5c35.0;received=192.168.10.1 Via: SIP/2.0/UDP 192.168.10.30:5060;rport=5060;branch=z9hG4bKf428b928c25dad03 Record-Route: <sip:192.168.10.1;lr=on;ftag=69451218021829df;nat=yes> From: "Ventas" <sip:112@192.168.10.1>;tag=69451218021829df To: <sip:2685249@192.168.10.1> Call-ID: fb5f5dac83056f72(a)192.168.10.30 CSeq: 7493 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:2685249@192.168.10.1:5070> Content-Length: 0 I don't have a lot of experience with mediaproxy, and I have some doubts that such you see they can help me to clarify, inside the file mediaproxy.ini some options appear which I have configured them but I am not sure if it is the best way. my scenario is the following one: <-> UAC<-> NAT <-> ADSL <-> Internet <-> eth0 wan (public ip x.x.x.x ) <- openser/mediaproxy/asterisk -> eth1 lan (192.168.11.1) <-> UAC [MediaProxy] start = yes socket = /var/run/mediaproxy.sock group = openser listen = None allow = None proxyIP = x.x.x.x (public ip) ;portRange = 60000:65000 portRange = 35000:65000 TOS = 0xb8 idleTimeout = 60 holdTimeout = 3600 forceClose = 0 [Accounting] ; one of none, radius or database accounting = none [Database] user = dbuser password = dbpass host = dbhost database = radius table = radacct [Radius] secret = secret server = localhost authport = 1812 acctport = 1813 dictionaries = /etc/radiusclient-ng/dictionary, /etc/openser/radius/dictionary, /usr/share/mediaproxy/dictionary retries = 2 timeout = 3 this couple of you line inside the openser, I don't still understand them according to the guide of ser getting started they are for asymmetric clients, but I don't find an example modparam("mediaproxy","sip_asymmetrics","/etc/openser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/ect/openser/rtp-clients") somebody that can give me a good help... regards rickygm

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Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by Ricky Gutierrez 31 Oct '08

31 Oct '08

I don't have problems when I make calls to the pstn I listen well and people listen to me well, the problem is when I receive a call from the pstn I don't listen anything and they don't listen to me, inside the sip.conf already has configured the values nat, externip localnet . I believe that the problem is that openser detects as nat the ip of my asterisk, eye > "I have the openser and the mediaproxy with asterisk in the same pc" ### Sip Log Asterisk #### <--- SIP read from 192.168.10.1:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK5839d960;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as4f7a434f;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as4f7a434f To: <sip:113@192.168.10.1>;tag=a72df908ec08f63d Call-ID: 1661965d0c28650d517426333e4a6ae5(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 P-hint: Onreply-route - fixcontact <-------------> --- (12 headers 0 lines) --- -- SIP/openser-08c0ea58 is ringing xserver*CLI> <--- SIP read from 192.168.10.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK5839d960;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as4f7a434f;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as4f7a434f To: <sip:113@192.168.10.1>;tag=a72df908ec08f63d Call-ID: 1661965d0c28650d517426333e4a6ae5(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 212 P-hint: Onreply-route - fixcontact P-hint: onreply_route|usemediaproxy v=0 o=113 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.1.64 t=0 0 m=audio 35004 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (15 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.64:35004 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.64:35004 list_route: hop: <sip:192.168.10.1;lr=on;ftag=as4f7a434f;nat=yes> set_destination: Parsing <sip:192.168.10.1;lr=on;ftag=as4f7a434f;nat=yes> for address/port to send to set_destination: set destination to 192.168.10.1, port 5060 Transmitting (NAT) to 192.168.10.1:5060: ACK sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK32b6019c;rport Route: <sip:192.168.10.1;lr=on;ftag=as4f7a434f;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as4f7a434f To: <sip:113@192.168.10.1>;tag=a72df908ec08f63d Contact: <sip:asterisk@192.168.10.1:5070> Call-ID: 1661965d0c28650d517426333e4a6ae5(a)192.168.10.1 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/openser-08c0ea58 answered Zap/4-1 xserver*CLI> <--- SIP read from 192.168.10.1:5060 ---> BYE sip:asterisk@192.168.10.1:5070 SIP/2.0 Record-Route: <sip:192.168.10.1;lr=on;ftag=a72df908ec08f63d> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK71b2.32123901.0 Via: SIP/2.0/UDP 192.168.10.30:5062;branch=z9hG4bK02603cb0e798dac0 From: <sip:113@192.168.10.1>;tag=a72df908ec08f63d To: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as4f7a434f Supported: path Call-ID: 1661965d0c28650d517426333e4a6ae5(a)192.168.10.1 CSeq: 9793 BYE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 P-hint: LR|fixcontact,setflag6 <-------------> --- (14 headers 0 lines) --- Sending to 192.168.10.1 : 5060 (NAT) <--- Transmitting (NAT) to 192.168.10.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK71b2.32123901.0;received=192.168.10.1 Via: SIP/2.0/UDP 192.168.10.30:5062;branch=z9hG4bK02603cb0e798dac0 Record-Route: <sip:192.168.10.1;lr=on;ftag=a72df908ec08f63d> From: <sip:113@192.168.10.1>;tag=a72df908ec08f63d To: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as4f7a434f Call-ID: 1661965d0c28650d517426333e4a6ae5(a)192.168.10.1 CSeq: 9793 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:asterisk@192.168.10.1:5070> Content-Length: 0 ________________________________ From: luzango mfupe <luzango.mfupe(a)gmail.com> Hi RickyI should have seen how you handle NAT in kamaiilo.conf but you can also edit sip.conf in Asterisk and try to put Nat=yes Rgds,

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[Kamailio-Users] Fw: version cdrtool mediaproxy 1.9.1
by Ricky Gutierrez 30 Oct '08

30 Oct '08

OK thank you Nuno to answer, but I continue thinking that that version is not compatible with mediaproxy 1.9.1, for comments that read by the web, but better we ask to ag-projects Dan .. my best regards rickygm ________________________________ From: Nuno Marques <yangsengopenser(a)gmail.com> To: Ricky Gutierrez <xserverlinux(a)yahoo.com> Cc: users(a)lists.kamailio.org Sent: Wednesday, October 29, 2008 5:24:35 PM Subject: Re: version cdrtool mediaproxy 1.9.1 Hi Ricky, I'm using CDRTool 6.6.10 (latest). I still can't establish connection between CDRTool and mediaproxy database... but i'm almost there ! :) Nuno 2008/10/29 Ricky Gutierrez <xserverlinux(a)yahoo.com> Hi nuno, excuse me but that version of cdrtool you used for mediaproxy 1.9.1? , I am trying to do the same a server billing with openser and asterisk .. best regardss rickygm

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[Kamailio-Users] Dimensioning a telephony system based on openser!
by Nuno Marques 30 Oct '08

30 Oct '08

Hi, I've sucessfully completed an Openser 1.3.2 + Mediaproxy 1.9.1 + Asterisk 1.4 + CDRTool with freeradius telephony system. Asterisk is used only for voice mail and redirectioning calls. Every calls should pass through mediaproxy so that i can account them. The goal was to create a simple prototype of what could be a VoIP provider. Now i need to dimensioning this system to work with this requisites: - 10000 users; - 100 VoIP to VoIP calls simultaneously capacity; - 30 VoIP to PSTN calls simultaneously capacity; Can anyone point me some ideas of how can i design such a system (how many servers, how to distribute the services among them, etc.). I have this prototype mounted with VMWare, so i think that even making tests with sipp aren't going to be reliable. Thanks in advance, Nuno

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Re: [Kamailio-Users] Kamailio/Openser - improve the performance with children process
by Daniel Cardoso Alves 30 Oct '08

30 Oct '08

Hi Henning Westerholt, Thanks for you reply... But I would like to know if there is a number that improve the performance... For example: If come 100 INVITE at the same moment, is there a NUMBER OF CHILDREN that has the best perfomance? The number of children depends on something? Thanks --- Em qua, 29/10/08, Henning Westerholt <henning.westerholt(a)1und1.de> escreveu: De: Henning Westerholt <henning.westerholt(a)1und1.de> Assunto: Re: [Kamailio-Users] Kamailio/Openser - improve the performance with children process Para: users(a)lists.kamailio.org Cc: dcardosoa(a)yahoo.com.br Data: Quarta-feira, 29 de Outubro de 2008, 11:25 On Wednesday 29 October 2008, Daniel-Constantin Mierla wrote: > > Is there a number of children that the Kamailio/Openser for improve > > the performance? > > > > Each process takes care of a request? > > yes, each process takes care of a SIP message (request/reply). Hi Daniel Cardoso, take a look to the core cookbook in our wiki [1] for more informations how you can control the number of childrens per protocol (UDP, TCP..). Cheers, Henning [1] http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:devel#children Novos endereços, o Yahoo! que você conhece. Crie um email novo com a sua cara @ymail.com ou @rocketmail.com. Novos endereços, o Yahoo! que você conhece. Crie um email novo com a sua cara @ymail.com ou @rocketmail.com. http://br.new.mail.yahoo.com/addresses

4 3

[OpenSER-Users] Openser1.3.2 + snmp
by Ricardo Alves Ferreira Silva 30 Oct '08

30 Oct '08

Hi all, Trying to install Openser1.3.2.... everything is OK and fine... but when I try to add support for snmp there are errors.. Dig dep into google, recheck all dependencies and didn't succeed... My env is FreeBSD 7.0-STABLE i386 The full net-snmp package is installed: atila# pkg_info | grep net-snmp net-snmp-5.3.2_3 An extendable SNMP implementation These are the error messages.... make modules/libsnmpstats modules.... Compiling alarm_checks.c gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -DMOD_NAME='"snmpstats"' -DNAME='"openser"' -DVERSION='"1.3.2-notls"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 4.2.1"' -D__CPU_i386 -D__OS_freebsd -D__SMP_no -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -DHAVE_TIMEGM -DHAVE_NETINET_IN_SYSTM -DHAVE_KQUEUE -DHAVE_SELECT -c alarm_checks.c -o alarm_checks.o In file included from alarm_checks.c:38: openserObjects.h:41:38: error: net-snmp/net-snmp-config.h: No such file or directory openserObjects.h:42:40: error: net-snmp/net-snmp-includes.h: No such file or directory openserObjects.h:43:52: error: net-snmp/agent/net-snmp-agent-includes.h: No such file or directory If I edit each file and hardcode the path ... /usr/local/include/net-snmp/... there are no errors... What am I missing in making sure the path to the include files are OK... Tried a couple options but didn't succeed... tehre are tons of include files into the src.... Regards Ricardo Ferreira Yes, we do run Openser www.vipway.com.br Enterprise VoIP Services

5 10

[Serusers] Error in jabber module - XJAB:extract_aor: Error while parsing URI
by Tomasz Zieleniewski 30 Oct '08

30 Oct '08

Hi, We are using ser 2.0 We encountered the following error: 7(407) route[PRESENCE_SUBSCRIBE](SUBSCRIBE cseq:933): (handle_subscription(jabber)) 7(407) XJAB:extract_aor: Error while parsing URI But still SER sends an 200 OK response and after then a NOTIFY. Here is the message received by ser: SUBSCRIBE sip:lkl%touk.pl@jabber.rd.touk.pl SIP/2.0 Via: SIP/2.0/UDP 192.168.0.165;rport;branch=z9hG4bKxakietyi Max-Forwards: 70 Proxy-Authorization: Digest username="tzl",realm="touk.pl",nonce="490987cb2702bb9db8d631e016825ad1c9cd12ac",uri="sip:lkl%touk.pl@jabber.rd.touk.pl",response="bd4474cf8af3d1ec8a7246872c0ecb46",algorithm=MD5,cnonce="mk57n39293",qop=auth,nc=00000001 To: <sip:lkl%touk.pl@jabber.rd.touk.pl> From: "Tomasz Zieleniewski" <sip:tzl@sip.rd.touk.pl>;tag=cejrf Call-ID: vxiulpsynbnidjo(a)sen.rd.touk.pl CSeq: 933 SUBSCRIBE Contact: <sip:tzl@192.168.0.165> Accept: application/pidf+xml Event: presence Expires: 3600 User-Agent: Twinkle/1.2 Content-Length: 0 Jabber config: # -- jabber params --- modparam("jabber", "jaddress", "touk.pl") modparam("jabber", "jport", 5222) modparam("jabber", "registrar", "serimgw(a)sip.rd.touk.pl") modparam("jabber","workers",2) modparam("jabber","max_jobs",10) modparam("jabber","cache_time",200) modparam("jabber","delay_time",60) modparam("jabber","jdomain","jabber.rd.touk.pl=%") Please give us a hint what could be wrong. Kind regards Tomasz

1 0

[Serusers] Configuring SER as an outbound proxy for multiple Ekigas on a LAN
by Enrico Maria Crisostomo 30 Oct '08

30 Oct '08

Hi all. I'm a first-time would-be user of SER and I have the following problem: I was planning to use Ekiga on a LAN with several workstations and I was looking for documentation about the options I have and the best setup I can achieve. How to solve the problem of NAT traversal for multiple Ekigas was answered by Ekiga documentation: use a SIP proxy. The workstations are all equipped with Solaris 10 and I have a small Solaris 10 server too, so I thought it was natural for me to use SER as a SIP proxy. Unfortunately I'm finding no documentation about how to configure it as a simple outbound proxy and to say the truth I found very little documentation about configuring a SIP proxy in general. My idea would be just having SER behave as an outbound SIP proxy and use ekiga.net accounts for everybody (no need for a PBX as Asterisk). It seems a kind of basic configuration but I have no idea where to start from. Do you have suggestions and pointers to documentation to help me achieve this goal? Thank you in advance, Enrico. -- Ελευθερία ή θάνατος GPG key: 1024D/FD2229AF fpr: 9E07 D40E 33A5 5993 6FC5 09A8 5BCF B1F2 FD22 29AF

2 1

Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by Ricky Gutierrez 30 Oct '08

30 Oct '08

Hi luzano thank you for your help and time, this it is my full ngrep. I have asterisk, mediaproxy and openser together in the same pc I have the doubt that when a incoming call from the PSTN by asterisk, sends it to an extension of openser, to which I do not request authentication to him within invites, I believe that there it is where I have problems with mediaproxy that it leaves you want to see of the openser.cfg, or everything? regards .. interface: any filter: (ip) and ( port 5060 ) # U +0.063740 192.168.10.1:5070 -> 192.168.10.1:5060 INVITE sip:113@192.168.10.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1> Contact: <sip:asterisk@192.168.10.1:5070> Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 29 Oct 2008 16:26:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 238 v=0 o=root 9850 9850 IN IP4 192.168.10.1 s=session c=IN IP4 192.168.10.1 t=0 0 m=audio 14750 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv # U +0.004203 192.168.10.1:5060 -> 192.168.10.1:5070 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1> Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 # U +0.000275 192.168.10.1:5060 -> 192.168.10.30:5062 INVITE sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1> Contact: <sip:asterisk@192.168.10.1:5070> Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Wed, 29 Oct 2008 16:26:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 238 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=root 9850 9850 IN IP4 192.168.10.1 s=session c=IN IP4 192.168.1.64 t=0 0 m=audio 35058 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv # U +0.026845 192.168.10.30:5062 -> 192.168.10.1:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1> Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Content-Length: 0 # U +0.009886 192.168.10.30:5062 -> 192.168.10.1:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 # U +0.000169 192.168.10.1:5060 -> 192.168.10.1:5070 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 P-hint: Onreply-route - fixcontact # U +0.000140 190.184.35.4:5060 -> 192.168.1.64:5064 OPTIONS sip:130@192.168.1.64:5064 SIP/2.0 Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282;rport From: "asterisk" <sip:asterisk@190.184.35.4>;tag=as35db6300 To: <sip:130@192.168.1.64:5064> Contact: <sip:asterisk@190.184.35.4> Call-ID: 16e287fc2b24afb97eb1759952eff3d3(a)190.184.35.4 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 29 Oct 2008 16:26:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 # U +0.000017 190.184.35.4:5060 -> 192.168.10.19:5064 OPTIONS sip:130@192.168.1.64:5064 SIP/2.0 Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282;rport From: "asterisk" <sip:asterisk@190.184.35.4>;tag=as35db6300 To: <sip:130@192.168.1.64:5064> Contact: <sip:asterisk@190.184.35.4> Call-ID: 16e287fc2b24afb97eb1759952eff3d3(a)190.184.35.4 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 29 Oct 2008 16:26:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 # U +0.008208 192.168.10.19:5064 -> 190.184.35.4:5060 SIP/2.0 200 OK To: <sip:130@192.168.1.64:5064>;tag=9611a90af8ab9321i1 From: "asterisk" <sip:asterisk@190.184.35.4>;tag=as35db6300 Call-ID: 16e287fc2b24afb97eb1759952eff3d3(a)190.184.35.4 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282 Server: Linksys/SPA942-5.2.8 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces # U +0.000014 192.168.1.64:5064 -> 190.184.35.4:5060 SIP/2.0 200 OK To: <sip:130@192.168.1.64:5064>;tag=9611a90af8ab9321i1 From: "asterisk" <sip:asterisk@190.184.35.4>;tag=as35db6300 Call-ID: 16e287fc2b24afb97eb1759952eff3d3(a)190.184.35.4 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 190.184.35.4:5060;branch=z9hG4bK715ee282 Server: Linksys/SPA942-5.2.8 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces # U +0.956449 192.168.10.19:5064 -> 190.184.35.4:5060 NOTIFY sip:190.184.35.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219 From: <sip:130@190.184.35.4>;tag=7c3557f6145bd125o1 To: <sip:190.184.35.4> Call-ID: ce8ea9b2-baee4a99(a)192.168.10.19 CSeq: 12 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA942-5.2.8 Content-Length: 0 # U +0.000027 192.168.1.64:5064 -> 190.184.35.4:5060 NOTIFY sip:190.184.35.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219 From: <sip:130@190.184.35.4>;tag=7c3557f6145bd125o1 To: <sip:190.184.35.4> Call-ID: ce8ea9b2-baee4a99(a)192.168.10.19 CSeq: 12 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA942-5.2.8 Content-Length: 0 # U +0.156145 190.184.35.4:5060 -> 192.168.1.64:5064 SIP/2.0 489 Bad event Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219;received=192.168.1.64 From: <sip:130@190.184.35.4>;tag=7c3557f6145bd125o1 To: <sip:190.184.35.4>;tag=as2002c003 Call-ID: ce8ea9b2-baee4a99(a)192.168.10.19 CSeq: 12 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 # U +0.000017 190.184.35.4:5060 -> 192.168.10.19:5064 SIP/2.0 489 Bad event Via: SIP/2.0/UDP 192.168.10.19:5064;branch=z9hG4bK-5ade1219;received=192.168.1.64 From: <sip:130@190.184.35.4>;tag=7c3557f6145bd125o1 To: <sip:190.184.35.4>;tag=as2002c003 Call-ID: ce8ea9b2-baee4a99(a)192.168.10.19 CSeq: 12 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 # U +5.895003 192.168.10.30:5062 -> 192.168.10.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 212 v=0 o=113 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.10.30 t=0 0 m=audio 5004 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.001400 192.168.10.1:5060 -> 192.168.10.1:5070 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3716ee27;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 212 P-hint: Onreply-route - fixcontact P-hint: onreply_route|usemediaproxy v=0 o=113 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.1.64 t=0 0 m=audio 35058 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.000557 192.168.10.1:5070 -> 192.168.10.1:5060 ACK sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3f155ae6;rport Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 Contact: <sip:asterisk@192.168.10.1:5070> Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 # U +0.000215 192.168.10.1:5060 -> 192.168.10.30:5062 ACK sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Record-Route: <sip:192.168.10.1;lr=on;ftag=as30de9085;nat=yes> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK7ce3.dd78e5e3.2 Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK3f155ae6;rport=5070 From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as30de9085 To: <sip:113@192.168.10.1>;tag=21c220c2e075d838 Contact: <sip:asterisk@192.168.10.1:5070> Call-ID: 373456b8787e65d9764158381c9da273(a)192.168.10.1 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 69 Content-Length: 0 P-hint: LR|fixcontact,setflag6 # U +0.886545 192.168.10.28:5060 -> 192.168.10.1:5060 ________________________________ From: luzango mfupe <luzango.mfupe(a)gmail.com> To: users(a)lists.kamailio.org Sent: Wednesday, October 29, 2008 5:12:29 AM Subject: Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy Hi Ricky, Where is your Kamailio config?? is this your full ngrep capture?? Rgds, Luzango.

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[Kamailio-Users] kamailio and rtpproxy-no audio
by Fabian Borot 30 Oct '08

30 Oct '08

hello I am trying to use rtpproxy and kamailio. Below please find my kamailio.cfg file. But this is what is happening: 1- ext 101[IP 76.109.183.2] is able to send audio to kamailio's IP [65.111.185.187]but it does not reach ext 100 IP [76.109.15.75]. 2- ext 100 does not send RTP at all. 3- I still see the private IPs in the SDPs of both exts. I am aattaching a wireshark capture Thank you in advance Fabian # $Id: kamailio.cfg 4576 2008-08-05 16:22:38Z klaus_darilion $ # # Kamailion basic configuration script # by Anca Vamanu <anca(a)voice-system.ro> # # Please refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # ####### Global Parameters ######### debug=3 log_stderror=no log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = '/usr/local/etc/kamailio/tls/user/user-cert.pem' #tls_private_key = '/usr/local/etc/kamailio/tls/user/user-privkey.pem' #tls_ca_list = '/usr/local/etc/kamailio/tls/user/user-calist.pem' port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:65.111.185.187:5060 ####### Modules Section ######## #set module path mpath='/usr/local/lib/kamailio/modules/' /* uncomment next line for MySQL DB support */ loadmodule 'db_mysql.so' loadmodule 'sl.so' loadmodule 'tm.so' loadmodule 'rr.so' loadmodule 'maxfwd.so' loadmodule 'usrloc.so' loadmodule 'registrar.so' loadmodule 'textops.so' loadmodule 'mi_fifo.so' loadmodule 'uri_db.so' loadmodule 'uri.so' loadmodule 'xlog.so' loadmodule 'acc.so' /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule 'auth.so' loadmodule 'auth_db.so' /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule 'alias_db.so' /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see 'multi-module params' section ) */ #loadmodule 'domain.so' /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule 'presence.so' #loadmodule 'presence_xml.so' loadmodule 'nathelper.so' # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam('mi_fifo', 'fifo_name', '/tmp/kamailio_fifo') # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam('rr', 'enable_full_lr', 1) # do not append from tag to the RR (no need for this script) modparam('rr', 'append_fromtag', 0) # ----- rr params ----- modparam('registrar', 'method_filtering', 1) /* uncomment the next line to disable parallel forking via location */ # modparam('registrar', 'append_branches', 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam('registrar', 'max_contacts', 10) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam('uri_db', 'use_uri_table', 0) modparam('uri_db', 'db_url', '') # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam('acc', 'early_media', 1) modparam('acc', 'report_ack', 1) modparam('acc', 'report_cancels', 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable 'append_fromtag' in 'rr' module */ modparam('acc', 'detect_direction', 0) /* account triggers (flags) */ modparam('acc', 'failed_transaction_flag', 3) modparam('acc', 'log_flag', 1) modparam('acc', 'log_missed_flag', 2) /* uncomment the following lines to enable DB accounting also */ modparam('acc', 'db_flag', 1) modparam('acc', 'db_missed_flag', 2) # ----- usrloc params ----- modparam('usrloc', 'db_mode', 0) /* uncomment the following lines if you want to enable DB persistency for location entries */ modparam('usrloc', 'db_mode', 2) #modparam('usrloc', 'db_url', # 'mysql://openser:openserrw@localhost/openser') modparam('usrloc', 'nat_bflag', 3) # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam('auth_db', 'calculate_ha1', yes) modparam('auth_db', 'password_column', 'password') modparam('auth_db', 'db_url', 'mysql://openser:openserrw@localhost/openser') modparam('auth_db', 'load_credentials', '') # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam('alias_db', 'db_url', # 'mysql://openser:openserrw@localhost/openser') # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam('domain', 'db_url', # 'mysql://openser:openserrw@localhost/openser') #modparam('domain', 'db_mode', 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam('alias_db|auth_db|usrloc|uri_db', 'use_domain', 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #modparam('presence|presence_xml', 'db_url', # 'mysql://openser:openserrw@localhost/openser') #modparam('presence_xml', 'force_active', 1) #modparam('presence', 'server_address', 'sip:192.168.1.2:5060') # ----- nathelper params ----- modparam('nathelper', 'natping_interval', 10) modparam('nathelper', 'ping_nated_only', 1) modparam('nathelper', 'rtpproxy_sock', 'udp:localhost:8899') modparam('nathelper', 'rtpproxy_disable_tout', 20) modparam('nathelper', 'rtpproxy_tout', 2) modparam('nathelper', 'rtpproxy_retr', 2) modparam('nathelper', 'sipping_from', 'sip:pinger@mykamailio.com') modparam('nathelper', 'sipping_bflag', 7) ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header('10')) { sl_send_reply('483','Too Many Hops'); exit; } if (is_method('REGISTER')) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize('', 'subscriber')) { www_challenge('', '0'); exit; } if (!save('location')) sl_reply_error(); exit; } if (!lookup('location')) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply('404', 'Not Found'); exit; case -2: sl_send_reply('405', 'Method Not Allowed'); exit; } } # when routing via usrloc, log the missed calls also setflag(2); # record routing if (!is_method('REGISTER|MESSAGE')) { record_route(); } # account only INVITEs if (is_method('INVITE')) { setflag(1); # do accounting } # for INVITEs enable some additional helper routes if (is_method('INVITE')) { force_rtp_proxy(); } if (!t_relay()) { sl_reply_error(); }; exit; # CANCEL processing if (is_method('CANCEL|BYE')) { unforce_rtp_proxy(); if (t_check_trans()) t_relay(); exit; } } _________________________________________________________________ Stay organized with simple drag and drop from Windows Live Hotmail. http://windowslive.com/Explore/hotmail?ocid=TXT_TAGLM_WL_hotmail_102008

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[Kamailio-Users] version cdrtool mediaproxy 1.9.1
by Ricky Gutierrez 29 Oct '08

29 Oct '08

Hi nuno, excuse me but that version of cdrtool you used for mediaproxy 1.9.1? , I am trying to do the same a server billing with openser and asterisk .. best regardss rickygm

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Re: [Kamailio-Users] REGISTER port with Xlite 2.0
by ingdavidcespedes＠cable.net.co 29 Oct '08

29 Oct '08

Hi I download Twinkle, and install all the prerequisites, except for Qt. I have some problems with it. First I download Qt for Linux/X11: C++ "qt-x11-opensource-src-4.4.3.tar" I really don't know if it is the one I have to download, because In the Twinkle page, it only says you have to install "Qt for open source (3.3.0 or later, but not Qt4)". Then I do all the steps cd /tmp gunzip qt-x11-opensource-src-4.4.3.tar.gz # uncompress the archive tar xvf qt-x11-opensource-src-4.4.3.tar # unpack it cd /tmp/qt-x11-opensource-src-4.4.3 ./configure #It took more than 6 hours make su -c "make install" Then I retry to install Twinkle entering in Twinkle directory an doing ./configure but I receive the following error: checking for qt-mt >= 3.3.0 qt-mt < 4.0... Package qt-mt was not found in the pkg-config search path. Perhaps you should add the directory containing `qt-mt.pc' to the PKG_CONFIG_PATH environment variable No package 'qt-mt' found Package qt-mt was not found in the pkg-config search path. Perhaps you should add the directory containing `qt-mt.pc' to the PKG_CONFIG_PATH environment variable No package 'qt-mt' found configure: error: Library requirements (qt-mt >= 3.3.0 qt-mt < 4.0) not met; consider adjusting the PKG_CONFIG_PATH environment variable if your libraries are in a nonstandard prefix so pkg-config can find them. So I read the FAQ of Twinke (http://www.xs4all.nl/~mfnboer/twinkle/index.html) an it says that I can skip this error by doing ./configure --disable-qt-check, but now I receive the following error: checking value of $QTDIR... /usr/lib/qt-3.3 checking for qmake... no configure: error: Cannot find qmake in $QTDIR/bin. $QTDIR is incorrect. So I realize that I didn't have libqt-mt so I try to recompile QT with "./configure -thread" but it didn't let me, so I try with "./configure -DQT_THREAD_SUPPORT" and the compilation took also mother tan 6 hours, and I retry to install Twinkle, but without success, I receive the same message. checking value of $QTDIR... /usr/lib/qt-3.3 checking for qmake... no configure: error: Cannot find qmake in $QTDIR/bin. $QTDIR is incorrect. I don't Know what I am doing wrong. Any Help!!!!! please. David Cespedes. ----- Mensaje original ----- De: Iñaki Baz Castillo <ibc(a)aliax.net> Fecha: Martes, Octubre 28, 2008 8:15 am Asunto: Re: [Kamailio-Users] REGISTER port with Xlite 2.0 > 2008/10/28 Daniel-Constantin Mierla <miconda(a)gmail.com>: > > is twinkle able now to display rich presence status? Last time I > tried> was showing only online/offline status. > > For now just online/offline status. > > But it has other great features as: > - Indication of Message Composition for Instant Messaging - RFC 3994 > - PingPong TCP keepalive. > - TCP/UDP > - PRACK > - q-value in registration. > > For me it's the most reliable softphone. > > > -- > Iñaki Baz Castillo > <ibc(a)aliax.net> > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >

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Re: [Kamailio-Users] REGISTER port with Xlite 2.0
by ingdavidcespedes＠cable.net.co 29 Oct '08

29 Oct '08

Excuse me list, it was an error copy to this list, but maybe some one can help me. ----- Mensaje original ----- De: Alex Balashov <abalashov(a)evaristesys.com> Fecha: Miércoles, Octubre 29, 2008 11:31 am Asunto: Re: [Kamailio-Users] REGISTER port with Xlite 2.0 > This is not the Twinkle list. > > ingdavidcespedes(a)cable.net.co wrote: > > > Hi > > > > I download Twinkle, and install all the prerequisites, except for > Qt. I have some problems with it. First I download Qt for > Linux/X11: C++ "qt-x11-opensource-src-4.4.3.tar" I really don't > know if it is the one I have to download, because In the Twinkle > page, it only says you have to install "Qt for open source (3.3.0 > or later, but not Qt4)". Then I do all the steps > > > > cd /tmp > > gunzip qt-x11-opensource-src-4.4.3.tar.gz # > uncompress the archive > > tar xvf qt-x11-opensource-src-4.4.3.tar # unpack it > > cd /tmp/qt-x11-opensource-src-4.4.3 > > ./configure #It took more than 6 > hours> make > > su -c "make install" > > Then I retry to install Twinkle entering in Twinkle directory an > doing ./configure but I receive the following error: > > > > checking for qt-mt >= 3.3.0 qt-mt < 4.0... Package qt-mt was not > found in the pkg-config search path. Perhaps you should add the > directory containing `qt-mt.pc' to the PKG_CONFIG_PATH environment > variable No package 'qt-mt' found Package qt-mt was not found in > the pkg-config search path. Perhaps you should add the directory > containing `qt-mt.pc' to the PKG_CONFIG_PATH environment variable > No package 'qt-mt' found > > configure: error: Library requirements (qt-mt >= 3.3.0 qt-mt < > 4.0) not met; consider adjusting the PKG_CONFIG_PATH environment > variable if your libraries are in a nonstandard prefix so pkg- > config can find them. > > > > So I read the FAQ of Twinke > (http://www.xs4all.nl/~mfnboer/twinkle/index.html) an it says that > I can skip this error by doing ./configure --disable-qt-check, but > now I receive the following error: > > > > checking value of $QTDIR... /usr/lib/qt-3.3 > > checking for qmake... no > > configure: error: Cannot find qmake in $QTDIR/bin. $QTDIR is > incorrect.> > > So I realize that I didn't have libqt-mt so I try to recompile QT > with "./configure -thread" but it didn't let me, so I try with > "./configure -DQT_THREAD_SUPPORT" and the compilation took also > mother tan 6 hours, and I retry to install Twinkle, but without > success, I receive the same message. > > > > checking value of $QTDIR... /usr/lib/qt-3.3 > > checking for qmake... no > > configure: error: Cannot find qmake in $QTDIR/bin. $QTDIR is > incorrect.> > > I don't Know what I am doing wrong. Any Help!!!!! please. > > > > David Cespedes. > > > > ----- Mensaje original ----- > > De: Iñaki Baz Castillo <ibc(a)aliax.net> > > Fecha: Martes, Octubre 28, 2008 8:15 am > > Asunto: Re: [Kamailio-Users] REGISTER port with Xlite 2.0 > > > >> 2008/10/28 Daniel-Constantin Mierla <miconda(a)gmail.com>: > >>> is twinkle able now to display rich presence status? Last time > I > >> tried> was showing only online/offline status. > >> > >> For now just online/offline status. > >> > >> But it has other great features as: > >> - Indication of Message Composition for Instant Messaging - RFC > 3994>> - PingPong TCP keepalive. > >> - TCP/UDP > >> - PRACK > >> - q-value in registration. > >> > >> For me it's the most reliable softphone. > >> > >> > >> -- > >> Iñaki Baz Castillo > >> <ibc(a)aliax.net> > >> _______________________________________________ > >> Users mailing list > >> Users(a)lists.kamailio.org > >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >> > > > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (706) 338-8599 > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >

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[Kamailio-Users] Fwd: How to enable local_route feature ?
by Pascal Maugeri 29 Oct '08

29 Oct '08

Hi Following your instructions I have enable the compilation defined in Makefile.defs and recompile everything after a make clean. One capture of the compilation for you to see the define is enable: [...] gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -Wredundant-decls -DMOD_NAME='"uri_db"' -DNAME='"kamailio"' -DVERSION='"1.4.2-notls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.0.2"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC -DUSE_LOCAL_ROUTE -DSVNREVISION='"2:5135M"' -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -c checks.c -o checks.o [...] When I try to start the proxy it complains about: CRITICAL:core:yyerror: parse error in config file, line 432, column 1-2: syntax error CRITICAL:core:yyerror: parse error in config file, line 432, column 1-2: invalid local_route statement As you announced this feature for OpenSER 1.4 release series, I assume this is available for kamailio v1.4.2 ? Best regards, Pascal ---------- Forwarded message ---------- From: Pascal Maugeri <pascal.maugeri(a)gmail.com> Date: Wed, Oct 29, 2008 at 11:17 AM Subject: How to enable local_route feature ? To: Henning Westerholt <henning.westerholt(a)1und1.de> Cc: devel(a)lists.openser.org, openser users <users(a)lists.openser.org> Hi Following your instructions I have enable the compilation defined in Makefile.defs and recompile everything after a make clean. One capture of the compilation for you to see the define is enable: [...] gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -Wredundant-decls -DMOD_NAME='"uri_db"' -DNAME='"kamailio"' -DVERSION='"1.4.2-notls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.0.2"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC -DUSE_LOCAL_ROUTE -DSVNREVISION='"2:5135M"' -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -c checks.c -o checks.o [...] When I try to start the proxy it complains about: CRITICAL:core:yyerror: parse error in config file, line 432, column 1-2: syntax error CRITICAL:core:yyerror: parse error in config file, line 432, column 1-2: invalid local_route statement As you announced this feature for OpenSER 1.4 release series, I assume this is available for kamailio v1.4.2 ? Best regards, Pascal On Thu, Jun 26, 2008 at 1:11 PM, Henning Westerholt < henning.westerholt(a)1und1.de> wrote: > On Friday 06 June 2008, Bogdan-Andrei Iancu wrote: > > OpenSER has a new type of route: local route. This route is to be > > executed when a brand new request is internally generated (openser being > > the UAC) ; such requests are result of : > > - t_uac_dlg MI command > > - msilo, sms, presence, cpl-c, dialog (and others) modules > > [..] > > Hello all, > > the addition of this feature short before the code freeze caused some > conflicts, which could not resolved sufficently in time for this release. > Thus the OpenSER board was asked to decide in this matter. The majority of > the board voted for decativating this feature in the 1.4 release series. > > For more details please take a look at the developer mailing list at [1], > [2] > and [3]. Details about the "dectivate" commit can be found at [4] > > The local route functionality is still available, but its not activated in > the > default installation. In order to use this feature its necessary to set > the "USE_LOCAL_ROUTE" define in the "Makefile.defs" file and compile your > own > binary from the source. Then it will possible to use this feature. > > In the 1.5.0 release this functionality (or something resemble this) will > be > present, fully usable and supported. > > With best regards, > > Henning Westerholt > > > [1] http://lists.openser.org/pipermail/devel/2008-June/014028.html > [2] http://lists.openser.org/pipermail/devel/2008-June/014122.html > [3] http://lists.openser.org/pipermail/devel/2008-June/014204.html > [4] http://lists.openser.org/pipermail/devel/2008-June/014444.html > > > -- > Henning Westerholt - Development Consumer Products / DSL Core > 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users >

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[Kamailio-Users] Kamailio/Openser - improve the performance with children process
by Daniel Cardoso Alves 29 Oct '08

29 Oct '08

Hi people... Is there a number of children that the Kamailio/Openser for improve the performance? Each process takes care of a request? Thanks in advance. Novos endereços, o Yahoo! que você conhece. Crie um email novo com a sua cara @ymail.com ou @rocketmail.com. http://br.new.mail.yahoo.com/addresses

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Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by luzango mfupe 29 Oct '08

29 Oct '08

Hi Ricky,Where is your Kamailio config?? is this your full ngrep capture?? Rgds, Luzango. > > Message: 2 > Date: Tue, 28 Oct 2008 23:10:12 -0700 (PDT) > From: Ricky Gutierrez <xserverlinux(a)yahoo.com> > Subject: [Kamailio-Users] I don't have asterisk audio to openser - > mediaproxy > To: users(a)lists.kamailio.org > Message-ID: <654712.75990.qm(a)web59903.mail.ac4.yahoo.com> > Content-Type: text/plain; charset="us-ascii" > > Hi list is making tests with openser 1.3.2 and mediaproxy to solve the nat, > I have gotten myself an ip it public with my supplier, I have two network > cards in the pc that I am using for openser and mediaproxy together with > asterisk, making tests with mediaproxy 1.9.1 when I receive a call from the > pstn through asterisk I don't have audio, if I call to the pstn they listen > to me well . > > > From: "Ventas" <sip:112@192.168.10.1 <sip%3A112(a)192.168.10.1> > >;tag=69451218021829df > To: <sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1> > >;tag=329cfeaa6ded039da25ff8cbb8668bd2.b1b2 > Contact: <sip:112@192.168.10.30:5060;transport=udp> > Supported: path > Call-ID: fb5f5dac83056f72(a)192.168.10.30 > CSeq: 7492 ACK > User-Agent: Grandstream GXP2020 1.1.6.16 > Max-Forwards: 70 > Allow: > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE > Content-Length: 0 > > > # > U +0.022110 192.168.10.30:5060 -> 192.168.10.1:5060 > INVITE sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1> SIP/2.0 > Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bKf428b928c25dad03 > From: "Ventas" <sip:112@192.168.10.1 <sip%3A112(a)192.168.10.1> > >;tag=69451218021829df > To: <sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1>> > Contact: <sip:112@192.168.10.30:5060;transport=udp> > Supported: replaces, timer, path > Proxy-Authorization: Digest username="112", realm="192.168.10.1", > algorithm=MD5, uri="sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1>", > nonce="4907ac8cb6dc757eb6ba5522e0fdb9786b4c3d6e", > response="c40a9387fdf5de29115c1edadc7f79db" > Call-ID: fb5f5dac83056f72(a)192.168.10.30 > CSeq: 7493 INVITE > User-Agent: Grandstream GXP2020 1.1.6.16 > Max-Forwards: 70 > Allow: > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE > Content-Type: application/sdp > Content-Length: 358 > > v=0 > o=112 8000 8001 IN IP4 192.168.10.30 > s=SIP Call > c=IN IP4 192.168.10.30 > t=0 0 > m=audio 5004 RTP/AVP 0 18 3 97 2 9 101 > a=sendrecv > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=fmtp:97 mode=20 > a=rtpmap:2 G726-32/8000 > a=rtpmap:9 G722/16000 > a=ptime:20 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-11 > > # > U +0.003938 192.168.10.1:5060 -> 192.168.10.30:5060 > SIP/2.0 100 Giving a try > Via: SIP/2.0/UDP 192.168.10.30:5060 > ;branch=z9hG4bKf428b928c25dad03;rport=5060 > From: "Ventas" <sip:112@192.168.10.1 <sip%3A112(a)192.168.10.1> > >;tag=69451218021829df > To: <sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1>> > Call-ID: fb5f5dac83056f72(a)192.168.10.30 > CSeq: 7493 INVITE > Server: OpenSER (1.3.2-notls (i386/linux)) > Content-Length: 0 > > > # > U +0.000115 192.168.10.1:5060 -> 192.168.10.1:5070 > INVITE sip:2685249@192.168.10.1:5070 SIP/2.0 > Record-Route: <sip:192.168.10.1;lr=on;ftag=69451218021829df;nat=yes> > Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK400f.b93e5c35.0 > Via: SIP/2.0/UDP 192.168.10.30:5060 > ;rport=5060;branch=z9hG4bKf428b928c25dad03 > From: "Ventas" <sip:112@192.168.10.1 <sip%3A112(a)192.168.10.1> > >;tag=69451218021829df > To: <sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1>> > Contact: <sip:112@192.168.10.30:5060;transport=udp> > Supported: replaces, timer, path > Call-ID: fb5f5dac83056f72(a)192.168.10.30 > CSeq: 7493 INVITE > User-Agent: Grandstream GXP2020 1.1.6.16 > Max-Forwards: 69 > Allow: > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE > Content-Type: application/sdp > Content-Length: 358 > P-hint: route(3)|setflag7,forcerport,fix_contact > P-hint: inbound->inbound > P-hint: Route[6]: mediaproxy > > v=0 > o=112 8000 8001 IN IP4 192.168.10.30 > s=SIP Call > c=IN IP4 192.168.1.64 > t=0 0 > m=audio 35040 RTP/AVP 0 18 3 97 2 9 101 > a=sendrecv > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=fmtp:97 mode=20 > a=rtpmap:2 G726-32/8000 > a=rtpmap:9 G722/16000 > a=ptime:20 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-11 > > # > U +0.000471 192.168.10.1:5070 -> 192.168.10.1:5060 > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK400f.b93e5c35.0;received= > 192.168.10.1 > Via: SIP/2.0/UDP 192.168.10.30:5060 > ;rport=5060;branch=z9hG4bKf428b928c25dad03 > Record-Route: <sip:192.168.10.1;lr=on;ftag=69451218021829df;nat=yes> > From: "Ventas" <sip:112@192.168.10.1 <sip%3A112(a)192.168.10.1> > >;tag=69451218021829df > To: <sip:2685249@192.168.10.1 <sip%3A2685249(a)192.168.10.1>> > Call-ID: fb5f5dac83056f72(a)192.168.10.30 > CSeq: 7493 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:2685249@192.168.10.1:5070> > Content-Length: 0 > > > I don't have a lot of experience with mediaproxy, and I have some doubts > that such you see they can help me to clarify, inside the file > mediaproxy.ini some options appear which I have configured them but I am not > sure if it is the best way. > > my scenario is the following one: > > <-> UAC<-> NAT <-> ADSL <-> Internet <-> > eth0 wan (public ip x.x.x.x ) <- openser/mediaproxy/asterisk -> eth1 lan ( > 192.168.11.1) <-> UAC > > > [MediaProxy] > > start = yes > socket = /var/run/mediaproxy.sock > group = openser > listen = None > allow = None > proxyIP = x.x.x.x (public ip) > ;portRange = 60000:65000 > portRange = 35000:65000 > TOS = 0xb8 > idleTimeout = 60 > holdTimeout = 3600 > forceClose = 0 > > [Accounting] > ; one of none, radius or database > accounting = none > > [Database] > user = dbuser > password = dbpass > host = dbhost > database = radius > table = radacct > > [Radius] > secret = secret > server = localhost > authport = 1812 > acctport = 1813 > dictionaries = /etc/radiusclient-ng/dictionary, > /etc/openser/radius/dictionary, /usr/share/mediaproxy/dictionary > retries = 2 > timeout = 3 > > > > > this couple of you line inside the openser, I don't still understand them > according to the guide of ser getting started they are for asymmetric > clients, but I don't find an example > > modparam("mediaproxy","sip_asymmetrics","/etc/openser/sip-clients") > modparam("mediaproxy","rtp_asymmetrics","/ect/openser/rtp-clients") > > somebody that can give me a good help... > > regards > > rickygm > > > > -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

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[Kamailio-Users] Fwd: Re: How to enable local_route feature ?
by Henning Westerholt 29 Oct '08

29 Oct '08

On Wednesday 29 October 2008, Pascal Maugeri wrote: (you can reach the kamailio list at lists.kamailio.org, the lists at openser.org goes to the opensips lists) > Following your instructions I have enable the compilation defined in > Makefile.defs and recompile everything after a make clean. One capture of > the compilation for you to see the define is enable: > > [...] > gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall > -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott > -DF_MALLOC -DUSE_LOCAL_ROUTE -DSVNREVISION='"2:5135M"' -DFAST_LOCK > [..] > checks.c -o checks.o > [...] > > When I try to start the proxy it complains about: > > CRITICAL:core:yyerror: parse error in config file, line 432, column 1-2: > syntax error > CRITICAL:core:yyerror: parse error in config file, line 432, column 1-2: > invalid local_route statement > > As you announced this feature for OpenSER 1.4 release series, I assume this > is available for kamailio v1.4.2 ? Hi Pascal, i just tried it with the trunk (which should show the same behaviour with regards to this feature). My compilation accepts local_route { [..] } in the configuration. Can you quote the relevant part of your configuration? Cheers, Henning

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[Serusers] Error logs for SER
by Shruti Sanadhya 29 Oct '08

29 Oct '08

Hi, I am trying to setup a SER on Ubuntu 8.04 (Hardy heron). I have been following the starter's guide for configuring my SER but I do not know where to look for the error logs. I just get a 'bad config file' error. Where can I get more details? Thanks

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[Kamailio-Users] error when initilazing "uac.so"
by Alex R.S.M 28 Oct '08

28 Oct '08

Hi, I added "uac.so" module to be able to use "uac_replace_from" but got the following error. I tried to initialized the module params I could think of. I am wondering what I am missing . Oct 28 08:58:01 lab-linux01 /sbin/kamailio[14014]: INFO:avpops:avpops_init: initializing... Oct 28 08:58:01 lab-linux01 /sbin/kamailio[14014]: INFO:uac:mod_init: initializing... Oct 28 08:58:01 lab-linux01 /sbin/kamailio[14014]: ERROR:uac:mod_init: 'append_fromtag' RR param is not enabled! - required by AUTO restore mode Oct 28 08:58:01 lab-linux01 /sbin/kamailio[14014]: ERROR:core:init_mod: failed to initialize module uac Oct 28 08:58:01 lab-linux01 /sbin/kamailio[14014]: ERROR:core:main: error while initializing modules Thanks, Alex _________________________________________________________________ When your life is on the go—take your life with you. http://clk.atdmt.com/MRT/go/115298558/direct/01/

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Re: [Kamailio-Users] REGISTER port with Xlite 2.0
by ingdavidcespedes＠cable.net.co 28 Oct '08

28 Oct '08

Thanks I'm now trying Zoiper, and it works perfectly, I have solved lot of problems that I thought that were because of my Proxy configuration, but they were because of my X-Lite. Now I will try Twinkle. Thanks for your comments. David Cespedes ----- Mensaje original ----- De: Victor Pascual Ávila <victor.pascual.avila(a)gmail.com> Fecha: Sábado, Octubre 25, 2008 12:51 pm Asunto: Re: [Kamailio-Users] REGISTER port with Xlite 2.0 > On Sat, Oct 25, 2008 at 12:09 PM, Iñaki Baz Castillo > <ibc(a)aliax.net> wrote: > > El Sábado, 25 de Octubre de 2008, ingdavidcespedes(a)cable.net.co > escribió:> > >> since we don't have an upgrade of X-lite for linux > >> systems, could you recommend me another Softphone. > > > > Always Twinkle softphone, the best one ;) > > I agree-- Twinkle is available for Linux only. > -- > Victor Pascual Ávila > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >

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[Kamailio-Users] My contrib to the project - Simple UI
by Graham Wooden 28 Oct '08

28 Oct '08

Hi list, After seeing that CDRTool was a little too powerful (and complicated for that matter) I decided to take it's CDR gathering and add a "simple" web UI to it allowing me to maintain my SIP users and various settings (some aren't shown in this version right now). Let me preface this - this is very plain, meaning it's not stylized. No CSS, no fancy layouts, etc. Mainly because I have this within my existing UI. However, I stripped it out and is functional as described. Again, for the CDR stuff it does require to have the RADIUS accounting and MySQL setup from CDRTool. Link: https://www.leasedminds.com/ui/ If people see value with this, I will be more than happy to work with someone to make it "stylized" and more featured pack. Enjoy and thanks, -graham

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[Kamailio-Users] Proxying media on reinvite
by CSB 28 Oct '08

28 Oct '08

We have the following scenario: Nated UAC receives INVITE Call is established, we proxy audio Far end sends Re-INVITE (to inform UAC to use UDPTL) We don't proxy audio so call fails So I need to understand how the appropriate way to force us to proxy the audio when the far end of a reinvite is NATed. Relevant parts of openser.cfg (OpenSER 1.3.2): We're using mediaproxy. modparam("usrloc", "nat_bflag", 6) route { # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- #If it's an INVITE & client is NATed, if (method=="INVITE" && client_nat_test("3")) { #Record-route and specify the record-route header explicitly record_route_preset("64.38.1.1:5060;nat=yes"); # insert IP address #xlog("invite and nated so record_route_preset"); #If not Nated and not REGISTER then normal record-route } else if (method!="REGISTER") { record_route(); xlog("not register and not nated so record_route"); }; #if (method!="REGISTER") { # record_route(); #}; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { #Ensure we are dealing with a re-INVITE. Only connected calls have tag= #entry in the To header. So if it's loose routed and doesn't have totag #and it's an invite or reply then something's wrong xlog("loose_route"); if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); xlog("403 in lr"); return; }; if (method=="INVITE") { if (!allow_trusted()) { xlog("!allow_ trusted for r-uri <$ru>"); if (!proxy_authorize("","subscriber")) { proxy_challenge("domain.com","1"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); }; #check client is nated or that we've already identified it's nated if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setbflag(6); xlog("Recipient is nated so setbflag 6 and use mediaproxy: r-uri <$ru>"); use_media_proxy(); }; }; route(1); return; }; } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- #Call reply_route(1) to intercept response messages heading to the client t_on_reply("1"); #Try to relay the message to its destination if (!t_relay()) { #If it can't and it's an INVITE or ACK end the media proxying log(1,"route[1] !t_relay"); if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); } else { log(1,"route[1] t_relay"); }; } onreply_route[1] #Handles message that are returned to the sender i.e. response to caller's original request { xlog("hit onreply_route(1). rs $rs si $si rm $rm ru $ru tu $tu fu $fu rr $rr"); if (isbflagset(6)) { xlog("flag(6) is set for $fu . We're currently in onreply_route(1)"); }; if (isbflagset(7)) { xlog("flag(7) is set for $fu . We're currently in onreply_route(1)"); }; if (status=~"(180)|(183)|2[0-9][0-9]") { xlog("status matches for $fu . We're currently in onreply_route(1)"); }; if ((isbflagset(6) || isbflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { #Check the SDP payload length. Assume if we have something we can call mediaproxy if (!search("^Content-Length:[ ]*0")) { log(1,"using media proxy in onreply_route(1)"); use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } The re-invite looks like this: INVITE sip:6422334455@58.28.1.1:5061 SIP/2.0 Via: SIP/2.0/UDP 202.180.1.1:5060;branch=z9hG4bK251fed3b;rport Route: <sip:147.202.1.1;lr=on;ftag=as45a8ade4> From: "user" <sip:022998877@202.180.1.1>;tag=as45a8ade4 To: <sip:022334455@147.202.1.1>;tag=4c813866c894f124i1 Contact: <sip:022998877@202.180.1.1> Call-ID: 0818dfa10f1da0f25daa4759332d3c55(a)202.180.1.1 CSeq: 103 INVITE User-Agent: UAS Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 349 v=0 o=root 5466 5467 IN IP4 202.180.1.1 s=session c=IN IP4 202.180.1.1 t=0 0 m=image 4724 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:72 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy Our OK looks like this: SIP/2.0 200 OK To: <sip:022334455@147.202.1.1>;tag=4c813866c894f124i1 From: "user" <sip:022998877@202.180.1.1>;tag=as45a8ade4 Call-ID: 0818dfa10f1da0f25daa4759332d3c55(a)202.180.1.1 CSeq: 103 INVITE Via: SIP/2.0/UDP 202.180.1.1:5060;branch=z9hG4bK251fed3b;rport=5060 Record-Route: <sip:147.202.1.1;lr=on;ftag=as45a8ade4> Contact: 6422334455 <sip:6422334455@58.28.1.1:5061> Server: Linksys/SPA2102-5.1.9 Content-Length: 265 Content-Type: application/sdp v=0 o=- 69685 69685 IN IP4 192.168.1.113 s=- c=IN IP4 192.168.1.113 t=0 0 m=image 16386 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:200 a=T38FaxUdpEC:t38UDPRedundancy Unfortunately this code doesn't result in the c record of the OK being updated to the server IP (it is still the private IP of the UAC). In the loose route section, the re-invite doesn't pass (client_nat_test("3") || search("^Route:.*;nat=yes")) so we don't use mediaproxy. In the on-reply route bflag(6) is not set. I don't understand why bflag(6) is not set since the r-uri is to a registered user that has bflag(6) set (at least that's how I understand it). Could anyone point me in the right direction? Thanks Cameron

2 1

[Kamailio-Users] [presence] Modifying NOTIFY headers set
by Pascal Maugeri 27 Oct '08

27 Oct '08

Hi I need to add a SIP header to outgoing NOTIFY messages, just before it is sent to remote users. Is that possible to do that without using the local_route section (using openser 1.3) ? Regards, Pascal

4 7

[Kamailio-Users] [dispatcher] configure the sending of OPTIONS pings
by Pascal Maugeri 27 Oct '08

27 Oct '08

Hi How do you configure dispatcher to send OPTIONS ping to load-balanced proxies ? I'm using kamailio 1.4.2 with this of configuration: [...] modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_interval", 5) modparam("dispatcher", "ds_ping_from", "sip:fend1@10.132.184.33<sip%3Afend1(a)10.132.184.33> ") modparam("dispatcher", "ds_probing_threshhold", 3) [...] The load-balancing works properly : I see the traffic being sent to my proxies. But I don't see any SIP OPTIONS packets. I've read in module doc "This method is only available, if compiled with the probing of failed gateways enabled. " but I can't find how to activate this on compilation time, so I'm not sure if I have this feature or not enable. Regards, Pascal

3 4

[Kamailio-Users] i want to stop second registration request if user already registered
by Mian M Asif 27 Oct '08

27 Oct '08

Hi All, i want to stop second registration request if use already registered i am using below code but getting no any response. please help me how to solve this problem. if (method=="REGISTER") { if (registered("location") ) { xlog("L_NOTICE","Asif: lookup(location) Passed\n"); sl_send_reply("699", "User Not Found"); return; } route(2); return; route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- xlog("L_NOTICE","Danish: Entering route[2]\n"); # if (registered("location")) { # xlog("L_NOTICE","Asif: lookup(location) failed\n"); # sl_send_reply("404", "User Not Found"); # return; # }; xlog("L_NOTICE","Asif: Register message IPsrcaddr [$si], RecvdIPaddr[$Ri]\n"); if (!search("^Contact:[ ]*\*") && nat_uac_test("2")) { xlog("L_NOTICE","Danish: In route[2] Found Nated contact\n"); setflag(6); setbflag(6); # if you want OPTIONS natpings uncomment next setbflag(7); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!radius_www_authorize("abc.com")) { www_challenge("abc.com", "1"); }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } } regards, Asif

4 5

[Kamailio-Users] REGISTER port with Xlite 2.0
by ingdavidcespedes＠cable.net.co 25 Oct '08

25 Oct '08

Hi I'm using Xlite 2.0 release 1105d, and when I try to REGISTER to IP:5070, it doesn't let me. In another Windows computer I have Xlite 3.0 and there you can user an IP:Port Address to register to. Does anyone know If it is a problem from version 2.0 or I am doing something wrong. If it is a problem of 2.0 version, and since we don't have an upgrade of X-lite for linux systems, could you recommend me another Softphone. Thanks David Cespedes

3 2

[Kamailio-Users] Stateless mode for sip load balancer
by Kaeman Chris 25 Oct '08

25 Oct '08

Hello, I have opensip working as a sip load balancer which balances load between 8 asterisk servers running 4 instances on each. The opensip task is just to load balance the incoming requests for 32 asterisk instances on 8 boxes. Can i run it in the stateless mode? Anything that i should be aware about which will prohibit me to run it on the stateless mode. Thanks, KChris Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/

3 2

Re: [Kamailio-Users] kamailio with tls
by ingdavidcespedes＠cable.net.co 24 Oct '08

24 Oct '08

I knowed that you can have TLS connection between UA and other Proxy Servers, but I didn't know that Kamailio supports it, thanks for your comment. So maybe Paulos's problem is that his Client is not trying to REGISTER to the correct port (5061) of his Kamailio. ----- Mensaje original ----- De: Klaus Darilion <klaus.mailinglists(a)pernau.at> Fecha: Viernes, Octubre 24, 2008 12:45 pm Asunto: Re: [Kamailio-Users] kamailio with tls > (cc'ed to the mailinglist) > > ingdavidcespedes(a)cable.net.co schrieb: > > Enable TLS in Kamailio is only for connection between other Kamailio > > (SIP Proxy Servers), it doesn't have anything to do with UA, or > am I > > wrong? . As I understand, you can not implement TLS between UA's and > > Kamailio. > > Of course you can. If the SIP client supports TLS you can also use > TLS > between UA and Kamailio - no problem. > > I use it with eyebeam clients and SNOM phones. Also pjsip supports > TLS, > for example you can test it using QjSimple. > http://www.ipcom.at/index.php?id=560 > > regards > klaus > > > > > > ----- Mensaje original ----- De: Klaus Darilion > > <klaus.mailinglists(a)pernau.at> Fecha: Viernes, Octubre 24, 2008 > 11:04> am Asunto: Re: [Kamailio-Users] kamailio with tls > > > >> > >> paulo leonardo schrieb: > >>> Hi, > >>> > >>> I would like a litle help :D! > >>> > >>> I installed kamailio and everything is ok :D! But i want to use > >> TLS, but > >>> when i setup TLS in kamailio, don't work the REGISTER (i can't > >> register > >>> my sofphone ...). And i compliled kamilio with TLS!!! when i > >> comment the > >>> cofigurantion TLS and use port 5060 works! > >> > >> So, "what" does not work? > >> > >> - Does Kamailio start (ps -Alf|grep kamailio)? - Does K listen to > >> the specified TLS socket (netstat -anp|grep kama)? - which client > >> do you use? - is a TCP connection set up? - is a TLS handshake > >> happening (ssldump)? - what is in the logfile of Kamailio? ..... > >> > >> Your error description is to short! > >> > >> regards klaus > >> > >> > >> > >> > >>> below is my configuration > >>> > >>> ----------------------------------------- disable_tls = no listen > >>> = tls:192.168.170.101:5061 <http://192.168.170.101:5061> > >>> tls_verify_server = 1 tls_verify_client = 1 > >>> tls_require_client_certificate = 1 tls_method = TLSv1 > >>> tls_certificate = > >>> "/usr/local/etc/kamailio/tls/user/user-cert.pem" tls_private_key > >>> = "/usr/local/etc/kamailio/tls/user/user- > >> privkey.pem"> tls_ca_list > >>> --------------------------------------- > >>> > >>> if (!www_authorize("192.168.170.101 <http://192.168.170.101>", > >>> "subscriber")) { www_challenge("192.168.170.101 > >> <http://192.168.170.101>", "0"); > >>> exit; } > >>> > >>> ---------------------------------------- > >>> > >>> root@pst:/usr/local/etc/kamailio# ls -R tls/ tls/: ca.conf > >>> README request.conf rootCA user user.conf > >>> > >>> tls/rootCA: cacert.pem certs index.txt private serial > >>> > >>> tls/rootCA/certs: > >>> > >>> tls/rootCA/private: cakey.pem > >>> > >>> tls/user: user-calist.pem user-cert.pem user-cert_req.pem > >>> user-privkey.pem > >>> > >>> > >>> thanks!!! > >>> > >>> > >>> > >>> ---------------------------------------------------------------- > -- > >>> > >> ------ > >>> _______________________________________________ Users mailing > >>> list Users(a)lists.kamailio.org > >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >> _______________________________________________ Users mailing > list > >> Users(a)lists.kamailio.org > >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >> > > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >

1 0

[Kamailio-Users] Problems with CDRTool installation!
by Nuno Marques 24 Oct '08

24 Oct '08

Hi, I'm trying to follow the INSTALL.txt of the CDRTool to accomplish a nice installation of OpenSER 1.3.2 + Mediaproxy + SerMyAdmin + Asterisk + CDRTool + FreeRadius. All went well until i started CDRTool installation... it's a little confusing for a newbie like me but... all the steps of the installation seem to went well. When i go to the browser to look for the CDRTool portal it gives me this error: Database error: pconnect as cdradmin to sipdb failed. Session halted. I thought it was a mysql users problem... so i went to the shell and tried to enter mysql with login and password for cdradmin, and... No problem. It worked fine. Mysql accepted the user login/password. Is there someone who can tell me where can i look for errors... I'm really stuck on this part... Thanks in advance, Nuno

2 9

[Kamailio-Users] draft sip-outbound
by paulo leonardo 24 Oct '08

24 Oct '08

Hi, I would like to know if kamailio implements draft sip-outbound? Can i use draft sip-outbound? http://tools.ietf.org/html/draft-ietf-sip-outbound-15 thanks!

2 1

Re: [Kamailio-Users] kamailio with tls
by Klaus Darilion 24 Oct '08

24 Oct '08

(cc'ed to the mailinglist) ingdavidcespedes(a)cable.net.co schrieb: > Enable TLS in Kamailio is only for connection between other Kamailio > (SIP Proxy Servers), it doesn't have anything to do with UA, or am I > wrong? . As I understand, you can not implement TLS between UA's and > Kamailio. Of course you can. If the SIP client supports TLS you can also use TLS between UA and Kamailio - no problem. I use it with eyebeam clients and SNOM phones. Also pjsip supports TLS, for example you can test it using QjSimple. http://www.ipcom.at/index.php?id=560 regards klaus > > ----- Mensaje original ----- De: Klaus Darilion > <klaus.mailinglists(a)pernau.at> Fecha: Viernes, Octubre 24, 2008 11:04 > am Asunto: Re: [Kamailio-Users] kamailio with tls > >> >> paulo leonardo schrieb: >>> Hi, >>> >>> I would like a litle help :D! >>> >>> I installed kamailio and everything is ok :D! But i want to use >> TLS, but >>> when i setup TLS in kamailio, don't work the REGISTER (i can't >> register >>> my sofphone ...). And i compliled kamilio with TLS!!! when i >> comment the >>> cofigurantion TLS and use port 5060 works! >> >> So, "what" does not work? >> >> - Does Kamailio start (ps -Alf|grep kamailio)? - Does K listen to >> the specified TLS socket (netstat -anp|grep kama)? - which client >> do you use? - is a TCP connection set up? - is a TLS handshake >> happening (ssldump)? - what is in the logfile of Kamailio? ..... >> >> Your error description is to short! >> >> regards klaus >> >> >> >> >>> below is my configuration >>> >>> ----------------------------------------- disable_tls = no listen >>> = tls:192.168.170.101:5061 <http://192.168.170.101:5061> >>> tls_verify_server = 1 tls_verify_client = 1 >>> tls_require_client_certificate = 1 tls_method = TLSv1 >>> tls_certificate = >>> "/usr/local/etc/kamailio/tls/user/user-cert.pem" tls_private_key >>> = "/usr/local/etc/kamailio/tls/user/user- >> privkey.pem"> tls_ca_list >>> --------------------------------------- >>> >>> if (!www_authorize("192.168.170.101 <http://192.168.170.101>", >>> "subscriber")) { www_challenge("192.168.170.101 >> <http://192.168.170.101>", "0"); >>> exit; } >>> >>> ---------------------------------------- >>> >>> root@pst:/usr/local/etc/kamailio# ls -R tls/ tls/: ca.conf >>> README request.conf rootCA user user.conf >>> >>> tls/rootCA: cacert.pem certs index.txt private serial >>> >>> tls/rootCA/certs: >>> >>> tls/rootCA/private: cakey.pem >>> >>> tls/user: user-calist.pem user-cert.pem user-cert_req.pem >>> user-privkey.pem >>> >>> >>> thanks!!! >>> >>> >>> >>> ------------------------------------------------------------------ >>> >> ------ >>> _______________________________________________ Users mailing >>> list Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> _______________________________________________ Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >

1 0

[Kamailio-Users] kamailio with tls
by paulo leonardo 24 Oct '08

24 Oct '08

Hi, I would like a litle help :D! I installed kamailio and everything is ok :D! But i want to use TLS, but when i setup TLS in kamailio, don't work the REGISTER (i can't register my sofphone ...). And i compliled kamilio with TLS!!! when i comment the cofigurantion TLS and use port 5060 works! below is my configuration ----------------------------------------- disable_tls = no listen = tls:192.168.170.101:5061 tls_verify_server = 1 tls_verify_client = 1 tls_require_client_certificate = 1 tls_method = TLSv1 tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem" tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem" tls_ca_list --------------------------------------- if (!www_authorize("192.168.170.101", "subscriber")) { www_challenge("192.168.170.101", "0"); exit; } ---------------------------------------- root@pst:/usr/local/etc/kamailio# ls -R tls/ tls/: ca.conf README request.conf rootCA user user.conf tls/rootCA: cacert.pem certs index.txt private serial tls/rootCA/certs: tls/rootCA/private: cakey.pem tls/user: user-calist.pem user-cert.pem user-cert_req.pem user-privkey.pem thanks!!!

2 1

[Serusers] Asterisk -> SER one way audio call issue
by Thorsten 24 Oct '08

24 Oct '08

Hi guys does someone have experience about the interaction of asterisk and SER. I have a strange behavior. I terminate a call from an Asterisk to SER and then to a interconnection partner which runs also on asterisk. Whenever I make a call after a long (3-5 mins) idle time I got a one way audio call because the interconnection partner sends it back to SER instead of asterisk. But the call after works like a charm. And this happens all the time. This is my SER config # ----------- global configuration parameters ------------------------ #debug=4 debug=2 fork=yes log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=82.98.89.140:5060 port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/ser/current/lib/ser/modules/sl.so" loadmodule "/usr/local/ser/current/lib/ser/modules/exec.so" loadmodule "/usr/local/ser/current/lib/ser/modules/tm.so" loadmodule "/usr/local/ser/current/lib/ser/modules/rr.so" loadmodule "/usr/local/ser/current/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/ser/current/lib/ser/modules/usrloc.so" loadmodule "/usr/local/ser/current/lib/ser/modules/registrar.so" loadmodule "/usr/local/ser/current/lib/ser/modules/textops.so" loadmodule "/usr/local/ser/current/lib/ser/modules/avpops.so" #loadmodule "/usr/local/ser/current/lib/ser/modules/group.so" loadmodule "/usr/local/ser/current/lib/ser/modules/xlog.so" loadmodule "/usr/local/ser/current/lib/ser/modules/auth.so" #loadmodule "/usr/local/ser/current/lib/ser/modules/nathelper.so" loadmodule "/usr/local/ser/current/lib/ser/modules/uri.so" #loadmodule "/usr/local/ser/current/lib/ser/modules/uri_db.so" # ----------------- setting module-specific parameters --------------- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="ACK") { route(1); break; } if (method=="REGISTER") { save("location"); break; }; # only forward if from our nat ip if (method=="INVITE" && (src_ip==82.98.89.146 || src_ip==82.98.89.147 || src_ip==82.98.89.129)) { #record_route(); if (uri =~ "sip:[0-9]*@*") { #record_route(); route(3); break; } } else { sl_send_reply("401", "Unauthorized"); }; if (method=="CANCEL") { route(1); break; }; if (method=="BYE") { route(1); break; } lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } route[3] { # strip the first 4 characters 1991 #strip(4); #rewritehost("62.134.16.104"); rewritehostport("62.134.16.104:5060"); #t_relay_to_udp("62.134.16.104", "5060"); xlog("L_WARN", "[%Tf]: Call forwarded to BT from: [%fu] to: [%tu] next: [%ds]\r\n"); route(1); break; } The asterisk dialplan looks like this ; route everything to SER exten => _X.,1,Noop(${SIP_HEADER(FROM)}) exten => _X.,n,Set(SIP_CODEC="gsm") exten => _X.,n,Dial(sip/${EXTEN}(a)82.98.89.140) exten => _X.,n,Hangup() The asterisk server has the public IP 82.98.89.146 and SER 82.98.89.140. BT which is our interconnection partner has the IP 62.134.16.104. Could someone help me with this? Thanks Ciao Thorsten Kohlhepp

1 0

[Serusers] registrar: interesting, however rare, bug..
by Alexandre Snarskii 24 Oct '08

24 Oct '08

Hi! This week we saw interesing problem with one of our clients: initially he was not able to configure his CCME to register on our proxy, so we just added static location serctl ul add ..., that's our classic workaround for CCME and another not-so-smart admins. But when he finally configured registration - things gone nuts: clients asks for registration: Date: Fri, 24 Oct 2008 13:16:29 GMT Timestamp: 1224854189 CSeq: 2905600 REGISTER Contact: <sip:NNNNNNN@AA.BBB.CCC.DD:5060> Expires: 3600 Authorization: Digest .... and ser replies with 200 OK, but with Contact: field filled with his static registration: Contact: <sip:NNNNNNN@AA.BBB.CCC.DD>;q=1;expires=4293498731;received="" As you can see, expires value is 4293498731, which means that it will be valid for about 136 years, just sixteen days short than 2^32 seconds... So, by RFC, client's router should wait for ~100 years before attempting registration refresh :) But CCME, i suppose, thinks different, it just adds received expires value to current time, got uint32 overflow and result is now()-16 days. As a result CCME thinks that registration is always expired and tries to re-register immediately... And does so hundredths times per second, creating unnecessary extra load to our proxy and database backend.... Why that is so: modules/registrar/reply.c calculates expires= as int2str((int)(c->expires - act_time), &len) but, c->expires can be less than act_time, so, result will be negative integer (translated to unsigned int on call to int2str, and this unsigned int will have higher bit set on, so it will be big one). Well, there are check if(VALID_CONTACT(c, act_time)) which filters out most of the cases where с->expires <= act_time, but for 'static' registrations these checks are off: ((c->expires > act_time) || (c->flags & FL_PERMANENT)) is always true in case of 'static' contact because of FL_PERMANENT. For now I fixed it with following patch (not perfect, i suppose, it just uses default_expires value for negative expires time, but it was easy to write and it works for me): --- reply.c.orig Thu Jan 13 11:48:08 2005 +++ reply.c Fri Oct 24 14:08:13 2008 @@ -163,7 +163,11 @@ memcpy(p, EXPIRES_PARAM, EXPIRES_PARAM_LEN); p += EXPIRES_PARAM_LEN; - cp = int2str((int)(c->expires - act_time), &len); + if (c->expires > act_time) { + cp = int2str((int)(c->expires - act_time), &len); + } else { + cp = int2str(default_expires, &len); + } memcpy(p, cp, len); p += len; Environment: ser-0.9.6/i386/FreeBSD (ser-2.0 has the same code, so it may be affected too). The only tuning of usrloc and registrar modules is db_url and dbmode = 2.

1 0

Re: [Kamailio-Users] Users Digest, Vol 41, Issue 68
by Mian M Asif 24 Oct '08

24 Oct '08

Dear Abalashov, What do you mean by "no[t] getting any response? is mean this function is not working please let me know i am doing right or wrong, dear Arif, exit also is not working cheers :). regards, Asif > Today's Topics: > > 1. Re: Architecture with load-balancing (dispatcher module) > (Ovidiu Sas) > 2. i want to stop second registration request if user already > registered (Mian M Asif) > 3. Re: i want to stop second registration request if user > already registered (Alex Balashov) > 4. Re: i want to stop second registration request if user > already registered (arif.zaman) > 5. Re: Anyone doing CNAM lookups? (Klaus Darilion) > 6. Kamailio v1.4.2 Released (Daniel-Constantin Mierla) > > > Message: 2 > Date: Thu, 23 Oct 2008 19:49:21 +0500 > From: "Mian M Asif" <asif44pk(a)gmail.com> > Subject: [Kamailio-Users] i want to stop second registration request > if user already registered > To: users(a)lists.kamailio.org > Message-ID: > <b005c33b0810230749v1b00bf9auabb97cfc17e517c6(a)mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > Hi All, > i want to stop second registration request if use already registered i > am using below code but getting no any response. please help me how to > solve this problem. > > if (method=="REGISTER") { > if (registered("location") ) { > xlog("L_NOTICE","Asif: lookup(location) Passed\n"); > sl_send_reply("699", "User Not Found"); > return; > } > route(2); > return; > > route[2] { > > # ----------------------------------------------------------------- > # REGISTER Message Handler > # ---------------------------------------------------------------- > xlog("L_NOTICE","Danish: Entering route[2]\n"); > # if (registered("location")) { > # xlog("L_NOTICE","Asif: lookup(location) failed\n"); > # sl_send_reply("404", "User Not Found"); > # return; > # }; > > xlog("L_NOTICE","Asif: Register message IPsrcaddr [$si], > RecvdIPaddr[$Ri]\n"); > if (!search("^Contact:[ ]*\*") && nat_uac_test("2")) { > xlog("L_NOTICE","Danish: In route[2] Found Nated contact\n"); > setflag(6); > setbflag(6); > # if you want OPTIONS natpings uncomment next > setbflag(7); > fix_nated_register(); > force_rport(); > }; > > sl_send_reply("100", "Trying"); > > if (!radius_www_authorize("abc.com")) { > www_challenge("abc.com", "1"); > }; > consume_credentials(); > > if (!save("location")) { > sl_reply_error(); > }; > > } > } > > regards, > Asif > > > > ------------------------------ > > Message: 3 > Date: Thu, 23 Oct 2008 10:51:30 -0400 > From: Alex Balashov <abalashov(a)evaristesys.com> > Subject: Re: [Kamailio-Users] i want to stop second registration > request if user already registered > To: Mian M Asif <asif44pk(a)gmail.com> > Cc: users(a)lists.kamailio.org > Message-ID: <49008F72.1060700(a)evaristesys.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > What do you mean by "no[t] getting any response?" > > Mian M Asif wrote: > >> Hi All, >> i want to stop second registration request if use already registered i >> am using below code but getting no any response. please help me how to >> solve this problem. >> >> if (method=="REGISTER") { >> if (registered("location") ) { >> xlog("L_NOTICE","Asif: lookup(location) Passed\n"); >> sl_send_reply("699", "User Not Found"); >> return; >> } >> route(2); >> return; >> >> route[2] { >> >> # ----------------------------------------------------------------- >> # REGISTER Message Handler >> # ---------------------------------------------------------------- >> xlog("L_NOTICE","Danish: Entering route[2]\n"); >> # if (registered("location")) { >> # xlog("L_NOTICE","Asif: lookup(location) failed\n"); >> # sl_send_reply("404", "User Not Found"); >> # return; >> # }; >> >> xlog("L_NOTICE","Asif: Register message IPsrcaddr [$si], >> RecvdIPaddr[$Ri]\n"); >> if (!search("^Contact:[ ]*\*") && nat_uac_test("2")) { >> xlog("L_NOTICE","Danish: In route[2] Found Nated contact\n"); >> setflag(6); >> setbflag(6); >> # if you want OPTIONS natpings uncomment next >> setbflag(7); >> fix_nated_register(); >> force_rport(); >> }; >> >> sl_send_reply("100", "Trying"); >> >> if (!radius_www_authorize("abc.com")) { >> www_challenge("abc.com", "1"); >> }; >> consume_credentials(); >> >> if (!save("location")) { >> sl_reply_error(); >> }; >> >> } >> } >> >> regards, >> Asif >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (706) 338-8599 > > > > ------------------------------ > > Message: 4 > Date: Thu, 23 Oct 2008 07:59:45 -0700 (PDT) > From: "arif.zaman" <arif.zaman(a)brotecs.com> > Subject: Re: [Kamailio-Users] i want to stop second registration > request if user already registered > To: users(a)lists.kamailio.org > Message-ID: <20132764.post(a)talk.nabble.com> > Content-Type: text/plain; charset=us-ascii > > > > Mian M Asif wrote: >> >> if (method=="REGISTER") { >> if (registered("location") ) { >> xlog("L_NOTICE","Asif: lookup(location) Passed\n"); >> sl_send_reply("699", "User Not Found"); >> return; >> } >> > > Use "exit;" instead of "return;". > > Cheers, > ARIF >

1 0

[Kamailio-Users] Anyone doing CNAM lookups?
by Brian McCrary 24 Oct '08

24 Oct '08

I've been using OpenSER 1.0.1 for the past couple of years, but I now need to add CNAM lookups. It looks like the presence module might be what I need, although from the looks if it, it's made to respond to lookups, instead of actually generate a query to another server for a lookup. I'm looking at doing CNAM lookups with Verisign which look to interface with a SUBSCRIBE message, in which in return Verising sends back a NOTIFY message with one of the fields being "calling-name" which has the CNAM info in it. Has anyone been succesful in implementing such a lookup with Verisign or another CNAM database provider? Brian

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[Serusers] SER without any fork(), It is possible?
by Samuel Ayet San Andres 24 Oct '08

24 Oct '08

Hi all, I´m trying to use SER (SIP Express Router) in a linux enbedded system with uClinux. In my enbedded processor i dont have any MMU so it cant fork(), it uses vfork() that is a cousin of fork but diferent because qhen father vfork a child, stops until the child make execve() or exit. When I run SER on my embedded plataform in a debug mode it shows me: ERROR: main_loop can´t fork(); Then I try to chage all forks for vfork() but it still not working due to the childs never make execve(). And after the explain of the context, here is the question: can I run SER without any child? Can I run SER without making any fork? I mean that only ONE process (witout deamonize) will run alone. I don´t need to manage more than 1 call...I only want to have one process listening to ONE IP. I have try to change the configuration of the ser 'ser.cfg' asi show: fork=no childrens=0 But still without working and i´m not sure if this is the right way to run SER without forking... Thank you for you support. Samuel. --

1 0

[Kamailio-Users] presence network Agent
by suresh tattikota 24 Oct '08

24 Oct '08

Hi all I am working on Presence network Agent. please let me know any one is working on the same . thank you all suresh

2 1

Re: [Kamailio-Users] Record Route and proxy behind static NAT
by Zappasodi Daniele 23 Oct '08

23 Oct '08

Thanks for your reply. I'm agree with your observations. I had already tried to solve the situations as described in the draft (avoiding spiral) and I have built a configuration that seem to work, at least for the basic call, but my solution needs to much tricks and I'm not sure that what I have done is safe and if it works in more complex call flows. I set the IP address in Record Route depending on position public\private of the callee and I substitute the IP address in Record Route for each reply, when it needs. First of all I suppose to know if the caller and callee is private or public (I check if the Request URI domain with RFC 1918) and I don't know if this is right. Then there is the problem to write the record route. I have moved the record route section in a branch route, because here I know the final destination for each branch. Here there is another doubt: I rewrote the record_route_preset function in order to remove the limit of one record route because in parallel forking I set more than record for each transaction, one record route for each branch. What are the consequences? I risk "only" to loose a call or to kill the proxy? I would prefer to leave the record route section at the beginning, but is it possible to substitute the record route after it is added? I remember that is possible to change only the original message and that there are some limitations on the possibility to change an header added during the message processing (I had some segmentation faults when I tryed with a custom header). Just now I'm looking on what to do with a relay from the failure route to a second destination. Before I proceding on this way I would like to know if I'm going on the wrong direction and if someone has found a simpler solution with this kind of scenarios. Regards, Daniele > Date: Wed, 22 Oct 2008 19:02:35 +0200 > From: " I?aki Baz Castillo " <ibc(a)aliax.net> > Subject: Re: [Kamailio-Users] Record Route and proxy behind static NAT > Cc: users(a)lists.kamailio.org > Message-ID: > <cc1f582e0810221002i7c95570epf585456a248811d1(a)mail.gmail.com> > Content-Type: text/plain; charset=UTF-8 > The solution could be adding two Record-Routes, one with private IP > and the other with public IP, but I'm not sure of how to achieve it > with Kamailio. AFAIK Kamailio *already* does it when the transaction > comes via eth0 and leaves the proxy via eth1, but this is not your > case. > I recommend you reading this draft, and later trying to get it in Kamailio: > http://tools.ietf.org/html/draft-ietf-sip-record-route-fix > (5. Double Record-Routing) ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message inerror. **********************************************************************

2 1

[Kamailio-Users] Kamailio v1.4.2 Released
by Daniel-Constantin Mierla 23 Oct '08

23 Oct '08

Hello everybody, a new release in 1.4 series is out. Kamailio 1.4.2 is based on the latest version of branch 1.4, including many fixes in code and documentation, therefore those running 1.4.0 or 1.4.1 are advised to upgrade. This is a minor release that includes updates since v1.4.1 - configuration file and database compatibility is preserved. Source tarballs are available at: http://www.kamailio.org/pub/kamailio/1.4.2/src/ <http://www.kamailio.org/pub/kamailio/1.4.2/src> Detailed changelog: http://www.kamailio.org/pub/kamailio/1.4.2/ChangeLog Download via SVN: svn co https://openser.svn.sourceforge.net/svnroot/openser/branches/1.4 kamailio Tag for this release can be browsed at: http://openser.svn.sourceforge.net/viewvc/openser/tags/1.4.2/ Project site at SourceForge.net: http://sourceforge.net/projects/openser/ Modules' documentation: http://www.kamailio.org/docs/modules/1.4.x/ <http://www.kamailio.org/docs/modules/1.4.x> What is new in 1.4.x release series is summarized in the announcement of v1.4.0: http://www.kamailio.org/mos/view/KAMAILIO-v1.4.0-Released Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] Architecture with load-balancing (dispatcher module)
by Pascal Maugeri 23 Oct '08

23 Oct '08

Hi I was wondering how to achieve an architecture with two or more active load-balancer nodes (kamailio+dispatcher module). I have read how to setup two dispatcher nodes, one node as a master and the other one as a backup. The backup node doesn't process any traffic until master fails. But how to make the traffic being processed by both load-balancers ? In the case for instance the capacity of single node is not enough to process all incoming traffic. Is there any recommended configuration (eg. using frontal application-level swith) Regards, Pascal

4 4

Re: [Kamailio-Users] Architecture with load-balancing (dispatcher module)
by Daniel-Constantin Mierla 23 Oct '08

23 Oct '08

On 10/23/08 15:05, Pascal Maugeri wrote: > > On Thu, Oct 23, 2008 at 1:47 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > a solution I am using is to do shared IP (vrrp/ucarp). The typical > scenario is active-backup, when one share IP is used. > > But you can do active-active by using two shared IP addresses, so > will be > > server1: active_ip_1 - backup_ip_2 > server2: backup_ip_1 - active_ip_2 > > So each server is active and backs-up the other. This can be > scaled, with more than 2 servers, you need an IP per server, and > you will have a chain of active servers backing up next one. Based > on who is backing up who, you may have one or more servers down > without affecting the service > > You have to take care of balancing the traffic among the IP addresses. > > > OK but how do you share traffic among the IP addresses ? You need a > kind of super load-balancer that load-balance traffic among > load-balancers ? one way you can do it is via DNS. Daniel > -pascal > > > > > Cheers, > Daniel > > > > On 10/23/08 13:16, Alex Balashov wrote: > > Any "frontal application-level switch" would simply have the > same liability. > > Kamailio is a very high-throughput proxy that can handle huge > amounts of call setups per second. I think you can count on that. > > Failing over around the load balancer node to a secondary load > balancer or distributing the traffic among multiple load > balancers is a job best left to the sending endpoint. For > example, a DID origination provider's switch or SBC can be set > up to fail over calls to a different IP endpoint for your SIP > trunk if no response is received within a certain amount of > time. That is how this is typically done. At some point > you've got to say that you've done all you can do, and it's up > to the other side. > > Pascal Maugeri wrote: > > > > Hi > > I was wondering how to achieve an architecture with two or > more active load-balancer nodes (kamailio+dispatcher module). > > I have read how to setup two dispatcher nodes, one node as > a master and the other one as a backup. > > The backup node doesn't process any traffic until master > fails. But how to make the traffic being processed by both > load-balancers ? In the case for instance the capacity of > single node is not enough to process all incoming traffic. > Is there any recommended configuration (eg. using frontal > application-level swith) > > Regards, > Pascal > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > > > > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] Usrloc: cache vs database inconsistencies
by CSB 23 Oct '08

23 Oct '08

I have noticed a consistent difference between the location records OpenSER has cached and those stored in the database. e.g. openserctl ul show --brief database engine 'MYSQL' loaded Control engine 'FIFO' loaded entering fifo_cmd ul_dump brief Domain:: location table=512 records=19 max_slot=2 Compared to: select count(*) from ser.location; +----------+ | count(*) | +----------+ | 10 | +----------+ 1 row in set (0.00 sec) This difference is consistent over time. When I look at the database logs OpenSER is trying to update an existing record in the database which, for some reason, doesn't exist. Our config has the following relevant parameters modparam("usrloc", "db_mode", 2) Openser 1.3.2 Any thoughts on how to prevent this circumstance from occurring and how to synchronise the cache with the db in an orderly manner? I had thought that after 120 seconds (timer_interval default)this would happen but it doesn't seem to. Any advice appreciated. Regards Cameron

2 1

[Kamailio-Users] Hold - core dump
by Marian Piater 23 Oct '08

23 Oct '08

Hello, I tried Kamailio 1.4.1+TLS with Mediaproxy 2.1.0. Everything works fine, but when I place call to Hold (using Hold button), Kamailio sometimes crashed. When I disable mediaproxy, Kamailio works without crash. I tried using engage_media_proxy() without any end_media_proxy or use_media_proxy and I also tried using use_media_proxy and end_media_proxy without engage_media_proxy with that same result. Sometimes Kamailio crash after first place call to hold, sometimes crash after few place call to hold/unhold. Sometimes work fine. OS is Debian Etch. Kamailio generate core dump. Core was generated by `kamailio'. Program terminated with signal 11, Segmentation fault. #0 0xb7d1e19a in strnlen () from /lib/tls/i686/cmov/libc.so.6 (gdb) bt #0 0xb7d1e19a in strnlen () from /lib/tls/i686/cmov/libc.so.6 #1 0xb7cf13f1 in vfprintf () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7d0fb01 in vsnprintf () from /lib/tls/i686/cmov/libc.so.6 #3 0xb7cf6945 in snprintf () from /lib/tls/i686/cmov/libc.so.6 #4 0xb7888e10 in end_media_session (callid=<value optimized out>, from_tag={s = 0x2 "", len = 4}, to_tag= {s = 0xbf9ce1b8 "remove\r\ncall_id: \r\nfrom_tag: ?I?E?\177\017openser", len = 0}) at mediaproxy.c:1478 #5 0xb7889147 in __dialog_ended (dlg=0x6, type=2048, _params=0xb7883cbc) at mediaproxy.c:1545 #6 0xb7865378 in run_dlg_callbacks (type=2048, dlg=0xb59e6398, msg=0x0, dir=0, dlg_data=0x0) at dlg_cb.c:252 #7 0xb7876186 in destroy_dlg_table () at dlg_hash.c:118 #8 0xb7861d8e in mod_destroy () at dialog.c:529 #9 0x080b3d41 in destroy_modules () at sr_module.c:381 #10 0x08069b60 in cleanup (show_status=1) at main.c:333 #11 0x0806aaf1 in handle_sigs () at main.c:523 #12 0x0806ecc8 in main (argc=1, argv=0xbf9cecc4) at main.c:865 DEBUG from Kamailio: Oct 22 16:25:34 [7817] DBG:tm:t_relay_to: new transaction fwd'ed Oct 22 16:25:34 [7856] CRITICAL:core:receive_fd: EOF on 15 Oct 22 16:25:34 [7856] DBG:core:handle_ser_child: dead child 8, pid 7817 (shutting down?) Oct 22 16:25:34 [7856] DBG:core:io_watch_del: io_watch_del (0x8175e80, 15, -1, 0x0) fd_no=26 called Oct 22 16:25:34 [7803] INFO:core:handle_sigs: child process 7817 exited by a signal 11 Oct 22 16:25:34 [7803] INFO:core:handle_sigs: core was generated Oct 22 16:25:34 [7803] INFO:core:handle_sigs: terminating due to SIGCHLD Oct 22 16:25:34 [7856] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 144400, used+overhead=169568, free=879008 max used (+overhead)= 169568 dumping free list: hash = 115 fragments no.: 1, unused: 0 bucket size: 920 - 920 (first 920) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 873576) TOTAL: 6 free fragments = 879152 free bytes ----------------------------- Oct 22 16:25:34 [7853] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 152800, used+overhead=178088, free=870488 max used (+overhead)= 178088 dumping free list: hash = 75 fragments no.: 1, unused: 0 bucket size: 600 - 600 (first 600) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 865376) TOTAL: 6 free fragments = 870632 free bytes ----------------------------- Oct 22 16:25:34 [7850] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 152800, used+overhead=178088, free=870488 max used (+overhead)= 178088 dumping free list: hash = 75 fragments no.: 1, unused: 0 bucket size: 600 - 600 (first 600) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 865376) TOTAL: 6 free fragments = 870632 free bytes ----------------------------- Oct 22 16:25:34 [7847] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 152800, used+overhead=178088, free=870488 max used (+overhead)= 178088 dumping free list: hash = 75 fragments no.: 1, unused: 0 bucket size: 600 - 600 (first 600) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 865376) TOTAL: 6 free fragments = 870632 free bytes ----------------------------- Oct 22 16:25:34 [7844] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 152800, used+overhead=178088, free=870488 max used (+overhead)= 178088 dumping free list: hash = 75 fragments no.: 1, unused: 0 bucket size: 600 - 600 (first 600) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 865376) TOTAL: 6 free fragments = 870632 free bytes ----------------------------- Oct 22 16:25:34 [7829] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102496, used+overhead=127784, free=920792 max used (+overhead)= 127864 dumping free list: hash = 1 fragments no.: 1, unused: 0 bucket size: 8 - 8 (first 8) hash = 10 fragments no.: 1, unused: 0 bucket size: 80 - 80 (first 80) hash = 62 fragments no.: 1, unused: 0 bucket size: 496 - 496 (first 496) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915696) TOTAL: 8 free fragments = 920936 free bytes ----------------------------- Oct 22 16:25:34 [7826] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102496, used+overhead=127776, free=920800 max used (+overhead)= 127784 dumping free list: hash = 1 fragments no.: 1, unused: 0 bucket size: 8 - 8 (first 8) hash = 73 fragments no.: 1, unused: 0 bucket size: 584 - 584 (first 584) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915696) TOTAL: 7 free fragments = 920944 free bytes ----------------------------- Oct 22 16:25:34 [7823] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102496, used+overhead=127776, free=920800 max used (+overhead)= 127784 dumping free list: hash = 1 fragments no.: 1, unused: 0 bucket size: 8 - 8 (first 8) hash = 73 fragments no.: 1, unused: 0 bucket size: 584 - 584 (first 584) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915696) TOTAL: 7 free fragments = 920944 free bytes ----------------------------- Oct 22 16:25:34 [7820] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102496, used+overhead=127768, free=920808 max used (+overhead)= 127768 dumping free list: hash = 75 fragments no.: 1, unused: 0 bucket size: 600 - 600 (first 600) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915696) TOTAL: 6 free fragments = 920952 free bytes ----------------------------- Oct 22 16:25:34 [7814] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102528, used+overhead=128304, free=920272 max used (+overhead)= 132536 dumping free list: hash = 1 fragments no.: 7, unused: 0 bucket size: 8 - 8 (first 8) hash = 2 fragments no.: 3, unused: 0 bucket size: 16 - 16 (first 16) hash = 3 fragments no.: 7, unused: 0 bucket size: 24 - 24 (first 24) hash = 4 fragments no.: 25, unused: 0 bucket size: 32 - 32 (first 32) hash = 5 fragments no.: 11, unused: 0 bucket size: 40 - 40 (first 40) hash = 6 fragments no.: 3, unused: 0 bucket size: 48 - 48 (first 48) hash = 7 fragments no.: 1, unused: 0 bucket size: 56 - 56 (first 56) hash = 9 fragments no.: 1, unused: 0 bucket size: 72 - 72 (first 72) hash = 10 fragments no.: 1, unused: 0 bucket size: 80 - 80 (first 80) hash = 15 fragments no.: 1, unused: 0 bucket size: 120 - 120 (first 120) hash = 17 fragments no.: 1, unused: 0 bucket size: 136 - 136 (first 136) hash = 27 fragments no.: 2, unused: 0 bucket size: 216 - 216 (first 216) hash = 40 fragments no.: 1, unused: 0 bucket size: 320 - 320 (first 320) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) hash = 94 fragments no.: 1, unused: 0 bucket size: 752 - 752 (first 752) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 914936) TOTAL: 68 free fragments = 920416 free bytes ----------------------------- Oct 22 16:25:34 [7811] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102528, used+overhead=128240, free=920336 max used (+overhead)= 132296 dumping free list: hash = 1 fragments no.: 5, unused: 0 bucket size: 8 - 8 (first 8) hash = 2 fragments no.: 4, unused: 0 bucket size: 16 - 16 (first 16) hash = 3 fragments no.: 7, unused: 0 bucket size: 24 - 24 (first 24) hash = 4 fragments no.: 19, unused: 0 bucket size: 32 - 32 (first 32) hash = 5 fragments no.: 12, unused: 0 bucket size: 40 - 40 (first 40) hash = 6 fragments no.: 1, unused: 0 bucket size: 48 - 48 (first 48) hash = 8 fragments no.: 1, unused: 0 bucket size: 64 - 64 (first 64) hash = 9 fragments no.: 1, unused: 0 bucket size: 72 - 72 (first 72) hash = 14 fragments no.: 1, unused: 0 bucket size: 112 - 112 (first 112) hash = 15 fragments no.: 1, unused: 0 bucket size: 120 - 120 (first 120) hash = 16 fragments no.: 2, unused: 0 bucket size: 128 - 128 (first 128) hash = 27 fragments no.: 2, unused: 0 bucket size: 216 - 216 (first 216) hash = 58 fragments no.: 1, unused: 0 bucket size: 464 - 464 (first 464) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915696) TOTAL: 60 free fragments = 920480 free bytes ----------------------------- Oct 22 16:25:34 [7841] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 111616, used+overhead=136744, free=911832 max used (+overhead)= 136744 dumping free list: hash = 118 fragments no.: 1, unused: 0 bucket size: 944 - 944 (first 944) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 907496) TOTAL: 5 free fragments = 911976 free bytes ----------------------------- Oct 22 16:25:34 [7832] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 159496, used+overhead=184608, free=863968 max used (+overhead)= 184608 dumping free list: hash = 129 fragments no.: 1, unused: 0 bucket size: 1032 - 1032 (first 1032) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 858424) TOTAL: 6 free fragments = 864112 free bytes ----------------------------- Oct 22 16:25:34 [7807] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102528, used+overhead=127808, free=920768 max used (+overhead)= 127808 dumping free list: hash = 70 fragments no.: 1, unused: 0 bucket size: 560 - 560 (first 560) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915696) TOTAL: 6 free fragments = 920912 free bytes ----------------------------- Oct 22 16:25:34 [7806] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102528, used+overhead=127808, free=920768 max used (+overhead)= 127808 dumping free list: hash = 70 fragments no.: 1, unused: 0 bucket size: 560 - 560 (first 560) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915696) TOTAL: 6 free fragments = 920912 free bytes ----------------------------- Oct 22 16:25:34 [7805] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102528, used+overhead=127808, free=920768 max used (+overhead)= 127808 dumping free list: hash = 70 fragments no.: 1, unused: 0 bucket size: 560 - 560 (first 560) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915696) TOTAL: 6 free fragments = 920912 free bytes ----------------------------- Oct 22 16:25:34 [7804] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102744, used+overhead=128088, free=920488 max used (+overhead)= 128088 dumping free list: hash = 13 fragments no.: 1, unused: 0 bucket size: 104 - 104 (first 104) hash = 61 fragments no.: 1, unused: 0 bucket size: 488 - 488 (first 488) hash = 101 fragments no.: 1, unused: 0 bucket size: 808 - 808 (first 808) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 915696) TOTAL: 7 free fragments = 920632 free bytes ----------------------------- Oct 22 16:25:34 [7835] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 159496, used+overhead=184608, free=863968 max used (+overhead)= 184608 dumping free list: hash = 129 fragments no.: 1, unused: 0 bucket size: 1032 - 1032 (first 1032) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 858424) TOTAL: 6 free fragments = 864112 free bytes ----------------------------- Oct 22 16:25:34 [7838] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 159496, used+overhead=184608, free=863968 max used (+overhead)= 184608 dumping free list: hash = 129 fragments no.: 1, unused: 0 bucket size: 1032 - 1032 (first 1032) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 144 fragments no.: 1, unused: 0 bucket size: 1152 - 1152 (first 1152) hash = 147 fragments no.: 1, unused: 0 bucket size: 1176 - 1176 (first 1176) hash = 151 fragments no.: 1, unused: 0 bucket size: 1208 - 1208 (first 1208) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 858424) TOTAL: 6 free fragments = 864112 free bytes ----------------------------- Oct 22 16:25:34 [7808] INFO:core:sig_usr: signal 15 received Memory status (pkg): fm_status (0x81a6e20): heap size= 1048576 used= 102600, used+overhead=128496, free=920080 max used (+overhead)= 132720 dumping free list: hash = 1 fragments no.: 7, unused: 0 bucket size: 8 - 8 (first 8) hash = 2 fragments no.: 5, unused: 0 bucket size: 16 - 16 (first 16) hash = 3 fragments no.: 9, unused: 0 bucket size: 24 - 24 (first 24) hash = 4 fragments no.: 31, unused: 0 bucket size: 32 - 32 (first 32) hash = 5 fragments no.: 15, unused: 0 bucket size: 40 - 40 (first 40) hash = 6 fragments no.: 3, unused: 0 bucket size: 48 - 48 (first 48) hash = 8 fragments no.: 2, unused: 0 bucket size: 64 - 64 (first 64) hash = 9 fragments no.: 1, unused: 0 bucket size: 72 - 72 (first 72) hash = 12 fragments no.: 1, unused: 0 bucket size: 96 - 96 (first 96) hash = 14 fragments no.: 1, unused: 0 bucket size: 112 - 112 (first 112) hash = 15 fragments no.: 1, unused: 0 bucket size: 120 - 120 (first 120) hash = 27 fragments no.: 2, unused: 0 bucket size: 216 - 216 (first 216) hash = 46 fragments no.: 1, unused: 0 bucket size: 368 - 368 (first 368) hash = 115 fragments no.: 1, unused: 0 bucket size: 920 - 920 (first 920) hash = 140 fragments no.: 1, unused: 0 bucket size: 1120 - 1120 (first 1120) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 914768) TOTAL: 82 free fragments = 920224 free bytes ----------------------------- Oct 22 16:25:34 [7803] DBG:dialog:destroy_dlg: destroing dialog 0xb5a063a0 Oct 22 16:25:34 [7803] DBG:dialog:run_dlg_callbacks: dialog=0xb5a063a0, type=2048 Any suggestions? With best regards Marian Piater -- Marian Piater LAM plus s.r.o. odd. hlasové komunikace - VoIP 242 413 000 Do práce 605 267 612 Mobilní telefon marian.piater(a)lam.cz LAM plus s.r.o. Mezi vodami 19 143 00 Praha - Modřany P Prosím, uvědomte si svou zodpovědnost za životní prostředí, než vytisknete tento e-mail.

2 1

[Kamailio-Users] OpenSER ActiveWatcher Table
by George Lee 22 Oct '08

22 Oct '08

Hi Expert, When a SUBSCRIBE message is received, OpenSER creates an entries in the ActiveWatcher table. This is used to generate NOTIFY message to watchers. The problem I have seen is that OpenSER keeps on creating new entries to the ActiveWatchers if the SIP UA deregister, register, deregister, register, etc. for several times. This triggers multiple NOTIFY messages to be sent to watchers. How does OpenSER remove entries in the ActiveWatcher table? It deosn't look like a SUBSCRIBE with expires=0 would delete an entry in the table. Thanks, George

3 3

[Kamailio-Users] URI length
by VoIP Forums www.Go4Calls.com 22 Oct '08

22 Oct '08

Hi All, I was trying to make dialplan if dialed number length >= 8 it should go for PSTN call otherwise local call. I did the following but PSTN is working well if length is greater than 8 but when number comes for local call i am getting the error "Message to Big" if ($(rU{s.len}) >= 8) { rewritehostport("89.21.XXX.XX:5060"); route(1); exit; }else{ lookup("location"); route(1); exit; }; Thank psl suggest us how i can do the local call if it is < to 8 thank you _________________________________________________________________ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=…

2 1

[Kamailio-Users] Record Route and proxy behind static NAT
by Zappasodi Daniele 22 Oct '08

22 Oct '08

Hello, I have my proxy behind a NAT (static NAT) with SIP clients that are both on my LAN and on public IP. UAC1 Private IP <--> <--> UAC3 PublicIP Proxy <--> NAT <--> WAN UAC2 Private IP <--> <--> UAC4 PublicIP I'm looking for the correct configuration of record route for this scenario. I looked for examples with proxy behind the NAT and I have always found a config section for the record route like this # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("<MY OPENSER PBULIC IP>:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; In this way the proxy uses "record_route_preset" when the caller is private (client_nat_test("3")) and "record_route" when the caller is public, without consider the callee. Consequently the proxy puts in the record route the public IP when the caller is private and the private IP address when the caller is public. I can't understand how it can work: when a client on an public ip calls a client on an public ip, they find in record route the private ip and they will try to send the SIP method (ACK,BYE, reINVITE ecc) to a private IP, so this messages will be lost. This is the beahaviour that I'm observing in my test. Maybe this configuration isn't adapted for my scenario or I miss something else (I have already added advertised_address/port) or I haven't correctly understand the instructions or the Record Route's usage. ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message inerror. **********************************************************************

2 1

[Kamailio-Users] UUID column in usr_preferences - How unique does it need to be?
by Graham Wooden 22 Oct '08

22 Oct '08

Hi, By looking at the indexes set for the table usr_preferences, it appears that UUID is going to be unique against the attribute column. So if user A has a attribute of custom_parameter_Z and user B has the same attribute, then both UUIDs need to be different, right? UUID can really be anything right? Its set to VARCHAR(64), so heck - a MD5 checksum against say a epoch timestamp would be fine. I just want to confirm how unique each row (in respects to the UUID) it needs to be. Thanks, -graham

2 2

[Kamailio-Users] Registrar performance
by Pascal Maugeri 22 Oct '08

22 Oct '08

Hi I would like to have some rough estimate of the registrar performance (with DB). If there is no figure available, does it make sense to use call performance benchmark like this one http://www.transnexus.com/White%20Papers/OpenSER_RTPproxy_test.htm and extrapolate to registrar performances ? eg. can I say that the SIP REGISTRAR is able to handle >600 registrations per second in the case of TransNexus testbed ? Regards, Pascal

3 3

[Kamailio-Users] Pike module is not working!
by Arif-Uz-Zaman 21 Oct '08

21 Oct '08

Hi, I've configured OpenSER with Pike module but it's not working as expected (doesn't block IP according to the configured time). J Here is a simple configuration by considering the following: ---------------------------------------------------------------------------- --------------- modparam("pike", "sampling_time_unit", 60) #1 minute modparam("pike", "reqs_density_per_unit", 200) #200 requests modparam("pike", "remove_latency", 7200) #Block IP for 2 hours if (!pike_check_req()) xlog("L_CRIT","#########TOO MANY HITS FROM $si##########\n"); exit; } ---------------------------------------------------------------------------- ----------------- If the number of SIP messages from a single IP address to my SIP Proxy exceeds 200 per minute. Action: Block IP for 2 hours. I'm not sure whether I'm doing wrong or is there any problem with Pike module? Please help me with valuable feedback. Thanks in advance! ARIF

1 0

[Kamailio-Users] Pike module is not working!
by Arif-Uz-Zaman 21 Oct '08

21 Oct '08

Hi, I've configured OpenSER with Pike module but it's not working as expected (doesn't block IP according to the configured time). :-) Here is a simple configuration by considering the following: ---------------------------------------------------------------------------- --------------- modparam("pike", "sampling_time_unit", 60) #1 minute modparam("pike", "reqs_density_per_unit", 200) #200 requests modparam("pike", "remove_latency", 7200) #Block IP for 2 hours if (!pike_check_req()) xlog("L_CRIT","#########TOO MANY HITS FROM $si##########\n"); exit; } ---------------------------------------------------------------------------- ----------------- If the number of SIP messages from a single IP address to my SIP Proxy exceeds 200 per minute. Action: Block IP for 2 hours. I'm not sure whether I'm doing wrong or is there any problem with Pike module? Please help me with valuable feedback. Thanks in advance! ARIF

1 0

[Kamailio-Users] Re write any SIP Header Value!
by arif.zaman 21 Oct '08

21 Oct '08

Hi, Is it possible to update any SIP Header (i.e. User-Agent) value before it saving to “location” database? If yes, please help me with sample config. For example: User-Agent value always save as "arif" in location database. Thanks in advance, ARIF -- View this message in context: http://www.nabble.com/Rewrite-any-SIP-Header-Value%21-tp20067253p20067253.h… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 7

[Kamailio-Users] (moved to kamailio)Why PUA_BLA generates a SUBSCRIBE "Event: dialog" that is never supported by OpenSIPS
by Klaus Darilion 21 Oct '08

21 Oct '08

Iñaki Baz Castillo wrote: > El Domingo, 19 de Octubre de 2008, Iñaki Baz Castillo escribió: >>> Hi, I using PUA_BLA and PUA_DIALOGINFO modules. >>> >>> I can't understand why when the proxy receives an INVITE it generates a >>> SUBSCRIBE to itselfs with: >>> Event: dialog >>> >>> Shouldn't it be: >>> Event: dialog;sla >>> ? > > Well, I've realized that it's because PUA_DIALOG info (Event: dialog). > > And more sorry! since it's just in Kamailio. The pua_dialoginfo modules sends SUBSCRIBE requests to itself? It can not send SUBSCRIBE - only PUBLISH. I guess the SUBSCRIBE is coming from pua_bla regards klaus

2 1

[Kamailio-Users] Remove function
by michel freiha 21 Oct '08

21 Oct '08

Dear All, I have the following function: if($rU =~ "^[+].*") { route(8); exit; } I need to remove the + sign from the dilaed number before sending the call to route(8) Regards

2 1

[Serusers] simple cluster setup with asterisk
by Uwe Kastens 21 Oct '08

21 Oct '08

Hello, Can anybody give me a hint how to start with a very simple cluster setup? I just want to route SIP to 2 different asterisk server. There is no need to care about NAT or call routing. I was wondering if its enough to use the dispatch module. Is it possible with the dispatch module to achieve that the system will reuse the information where a user is registered to route the calls to the same system? BR kiste -- kiste lat: 54.322684, lon: 10.13586

1 0

[Kamailio-Users] t_relay("0x02")
by jh＠tutpro.com 20 Oct '08

20 Oct '08

tm readme on t_relay flag "0x02" says: 0x02 - do not internally send a negative reply in case of forward failure (due internal error, bad RURI, bad message, etc). When a forward failure occurs, no SIP request is relayed and therefore no negative reply or timeout will show up on the failure_route (if one is set). It applies only when the transaction is created. By default one negative reply is sent. Useful if you want to implement a serial forking in case of failure. i don't quite understand the statement: It applies only when the transaction is created. what does "It" refer to? i did some experiments with new lcr implementation and if, for example, tcp connection to a gw cannot be established, then t_relay without "0x02" flag generates 477 reply to caller and, indeed, failure route is not executed. so i need test if t_relay("0x02") failed due to "send failed" and handle selection of next gw in the same route block where t_relay("0x02") was called. this is inconvenient, because there already exists a failure route block that does all that and now its code needs to be at least partially duplicated. or am i missing something and it is somehow possible to get a failure route block executed also on this kind of internally generated negative reply? -- juha

2 2

[Kamailio-Users] $rr and $rr are null
by jh＠tutpro.com 20 Oct '08

20 Oct '08

i have if (t_check_status("403")) { xlog("L_INFO", "DBG: status is 403\n, rr is <$rr> and rs is <$rs>\n"); }; and i get to syslog: Oct 18 19:51:37 localhost /usr/sbin/kamailio[18580]: DBG: status is 403 , rr is <<null>> and rs is <<null>> is this expected behavior? -- juha

2 4

Re: [Kamailio-Users] opensips db
by Alex Balashov 20 Oct '08

20 Oct '08

waleed hassn wrote: > On Mon, Oct 20, 2008 at 5:41 AM, Alex Balashov > <abalashov(a)evaristesys.com> wrote: >> waleed hassn wrote: >>> On Mon, Oct 20, 2008 at 4:25 AM, Alex Balashov >>> <abalashov(a)evaristesys.com> wrote: >>>> waleed hassn wrote: >>>>> hi all >>>>> i am new in opensips , how can i set database and add users >>>> Which database, and what users? >>>> >>>> >>>> -- >>>> Alex Balashov >>>> Evariste Systems >>>> Web : http://www.evaristesys.com/ >>>> Tel : (+1) (678) 954-0670 >>>> Direct : (+1) (678) 954-0671 >>>> Mobile : (+1) (706) 338-8599 >>>> >>> sip users and mysql database >>> an error occur when i execute opensipsdb command >> 1) Learn to set up and administer MySQL. >> >> 2) What error? >> >> 3) It might be beneficial to post to the OpenSIPS-Users list. >> >> -- >> Alex Balashov >> Evariste Systems >> Web : http://www.evaristesys.com/ >> Tel : (+1) (678) 954-0670 >> Direct : (+1) (678) 954-0671 >> Mobile : (+1) (706) 338-8599 >> > thanks , can you get me some useful links to begin First result on Google for "mysql tutorial": http://dev.mysql.com/doc/refman/5.0/en/tutorial.html -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

1 0

Re: [Kamailio-Users] opensips db
by Alex Balashov 20 Oct '08

20 Oct '08

waleed hassn wrote: > On Mon, Oct 20, 2008 at 4:25 AM, Alex Balashov > <abalashov(a)evaristesys.com> wrote: >> waleed hassn wrote: >>> hi all >>> i am new in opensips , how can i set database and add users >> Which database, and what users? >> >> >> -- >> Alex Balashov >> Evariste Systems >> Web : http://www.evaristesys.com/ >> Tel : (+1) (678) 954-0670 >> Direct : (+1) (678) 954-0671 >> Mobile : (+1) (706) 338-8599 >> > > sip users and mysql database > an error occur when i execute opensipsdb command 1) Learn to set up and administer MySQL. 2) What error? 3) It might be beneficial to post to the OpenSIPS-Users list. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

1 0

[Kamailio-Users] opensips db
by waleed hassn 20 Oct '08

20 Oct '08

hi all i am new in opensips , how can i set database and add users thanks

2 1

[Kamailio-Users] RLS and presence Server
by mahesh.peddi＠info-spectrum.com 20 Oct '08

20 Oct '08

Hello, I am able to see RLS properly working when RLS and presence functionality is in the same server. after that, I tried to integrate RLS as a separate application server. To forward the message from RLS to presence server, I made below changes in the config file: route[2] { if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("NOTIFY")) { rls_handle_notify(); t_release(); } if(is_method("SUBSCRIBE")) { $var(ret_code)= rls_handle_subscribe(); if($var(ret_code)== 10) #to forward to presence server t_relay("udp:<ip>:5070"); } if(is_method("PUBLISH")) { #to forward to presence server. t_relay("udp:<ip>:5070"); t_release(); } exit; } because of this configuration, i found two problems: 1. when i am forwarding subscribe message (from RLS to presence server), its not sending the subscribe for all the contacts. Here, i am expecting (sort of) output of rls_handle_subscribe() to be sent to the presence server. Which is not happening. 2. When presence server is sending notify message, it is sending directly to scscf..instead of rls.. Can anybody let me know, how can I get the desired functionality of: 1. UE --> RLS --> presence server (subscribe) 2. Presence server --> RLS --> UE (for notify). If there is any other way to forward message from RLS to presence, please let me know Regards, Mahesh Peddi

1 0

Re: [Kamailio-Users] Check if a user is calling from aregistered device and if not, deny the call!
by Daniel-Constantin Mierla 20 Oct '08

20 Oct '08

Hello, On 10/19/08 14:32, Arif-Uz-Zaman wrote: > Hi, > > Here is a tricky sample config to check if a user is calling from a > registered device and if not, deny the call: > > $var(temp) = $ruri; > $ruri = $fu; > if(!registered("location")) { > sl_send_reply("403","Forbidden auth ID"); > xlog("<ARIF> Not Found\n"); > exit; > } > $ruri = $var(temp); > in the trunk (so available with upcoming 1.5.0) you can do more strict checks. While you are sure an user is registered, you cannot detect if he is calling from registered device or a different one. You just know it has a valid contact in location database. Here is an example to check Contact address in INVITE is same as one in location table: http://openser.blogspot.com/2008/10/registrar-enhancements.html As you get access to all attributes stored for a location record, you can do many other tests, like source IP and port, etc. Cheers, Daniel > Thanks, > ARIF > > -----Original Message----- > From: users-bounces(a)lists.kamailio.org > [mailto:users-bounces@lists.kamailio.org] On Behalf Of Juha Heinanen > Sent: Sunday, October 19, 2008 3:44 PM > To: users(a)lists.kamailio.org > Subject: Re: [Kamailio-Users] Check if a user is calling from > aregistereddevice and if not, deny the call! > > Arif-Uz-Zaman writes: > > > I've already tested with the following config but no luck. Since the > > following config check only whether callee is registered or not. Not the > > caller. > > > > if(!registered("location")) { > > sl_send_reply("403","Forbidden auth ID"); > > exit; > > } > > indeed the function should take a pseudo variable that contains aor used > for checking. > > > Is there any other suggestions? > > save r-uri, store caller's aor to r-uri and call the above function. > then restore r-uri. > > -- juha > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > -- Daniel-Constantin Mierla http://www.asipto.com

2 2

[Kamailio-Users] load/next_contacts question
by Klaus Darilion 20 Oct '08

20 Oct '08

Hi Juha and others! I wonder if I understand load/next_contacts(). If I understand it right it can be used for example to make parallel/serial forking after an ENUM lookup (multiple NAPTRs). I think this is the correct usage: if (enum_query()) { if (load_contacts()) { if (next_contacts()) { t_on_failure("1"); t_relay(); } else { sl_reply("500","error adding contacts"); } } else { sl_reply("500","error fetching destination set"); } exit; } But I wonder what if all the branches after enum_query() have the same q value. According to the README: load_contact: If all contacts in the destination set have the same qvalue, load_contacts() does not do anything thus minimizing performance impact of sequential forking capability when it is not needed. Returns 1 if loading of contacts succeeded or there was nothing to do. Thus, next_contacts() will be called although there are no AVPs to fetch. Do I miss something here? regards klaus

2 2

Re: [Kamailio-Users] Check if a user is calling from a registereddevice and if not, deny the call!
by jh＠tutpro.com 20 Oct '08

20 Oct '08

Arif-Uz-Zaman writes: > I've already tested with the following config but no luck. Since the > following config check only whether callee is registered or not. Not the > caller. > > if(!registered("location")) { > sl_send_reply("403","Forbidden auth ID"); > exit; > } indeed the function should take a pseudo variable that contains aor used for checking. > Is there any other suggestions? save r-uri, store caller's aor to r-uri and call the above function. then restore r-uri. -- juha

3 2

[Kamailio-Users] Send MaxCallDuration and UserCredit for freeradius-cdrtool
by Diego Zuaneti Arruda 20 Oct '08

20 Oct '08

Hi all, How I send to freeradius-cdrtool the maxCallDuration and UserCredit? On the /etc/raddb/users I have: 87001(a)mydomain.com.br Auth-Type := Digest, User-Password == "mypassword" Reply-Message = "Authenticated", SIP-AVP = "authentic#1", SIP-AVP = "callMaxDur:60", SIP-AVP = "credit:25" Is this correct ? Thanks. Diego.

3 28

Re: [Kamailio-Users] How to handle DoS attack and OpenSER Pikemodule!
by arif.zaman 20 Oct '08

20 Oct '08

arif.zaman wrote: > > Hello Daniel, > > I've tested according to your suggessions but No Luck. I've configured to > block for 10 minutes and its working but not works for 2 hours. > >> In my case: If the number of SIP messages from a single IP address to >> my SIP Proxy exceeds 200 per minute. Recommended action: Block IP >> for 2 hours. >> > >> modparam("pike", "sampling_time_unit", 60) modparam("pike", >> "reqs_density_per_unit", 200) modparam("pike", "remove_latency", 7200) > Is there any help? Thanks, ARIF -- View this message in context: http://www.nabble.com/How-to-handle-DoS-attack-and-OpenSER-Pike-module%21-t… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

1 0

[Kamailio-Users] lcr gateway can now be given a hostname
by jh＠tutpro.com 19 Oct '08

19 Oct '08

lcr gateway can now be given a hostname that is used in request-uri when request is sent to the gateway. hostname is not, however, used in forwarding the request. forwarding is still based on gateway's ip address, port, and transport protocol. this is in order to be able to keep on supporting from_gw() and to_gw() functions. -- juha

1 0

[Kamailio-Users] Check if a user is calling from a registered device and if not, deny the call!
by Arif-Uz-Zaman 19 Oct '08

19 Oct '08

Hi, I do like to configure OpenSER in such a way so that it can accept incoming call from a registered user only. After receiving SIP INVITE message, OpenSER will check whether caller is registered or not before forwarding that received INVITE message to make sure only registered user will make call through OpenSER. If Caller is not registered, OpenSER will deny the call. Is it possible to check if a user is calling from a registered device without using any Database? Please help me with sample configuration by considering my interest. Thanks, ARIF

1 0

[Kamailio-Users] Check if a user is calling from a registered device and if not, deny the call!
by jh＠tutpro.com 19 Oct '08

19 Oct '08

Arif-Uz-Zaman writes: > OpenSER will check whether caller is registered or not before forwarding > that received INVITE message to make sure only registered user will make > call through OpenSER. If Caller is not registered, OpenSER will deny the > call. > Is it possible to check if a user is calling from a registered device > without using any Database? Please help me with sample configuration by > considering my interest. arif, i'm quite sure that this same question has come up many times before already. please check the mailing list archives. -- juha

1 0

[Kamailio-Users] OpenSER 1.3.2 + Mediaproxy 1.9.2 + Asterisk 1.4 + CDRTool with FreeRadius
by Nuno Marques 18 Oct '08

18 Oct '08

Hi to all, I need to implement a simple prototype of a Voip system with OpenSER 1.3.2 + Mediaproxy (for NAT traversal) + Asterisk 1.4 (for interconnection with PSTN) + CDRTool (for accounting) on a debian 4 etch operating system. I've already configured OpenSER + Mediaproxy + Asterisk with sucess. I'm missing CDRTool. I have two questions, if some one can help me: 1º which is the CDRTool version compatible with openser 1.3.2 and mediaproxy 1.9.2. Can i use the latest (6.6.10 i supose)? 2º I've read the install file for CDRTool but it is a little confusing (maybe because i'm a newbie to all this). Can anyone show me some literature (or tutorials) to accomplish this installation? Thanks in advance to all, Nuno

1 0

[Kamailio-Users] simple question
by Alex R.S.M 18 Oct '08

18 Oct '08

Hi, I want openSER to act ad a B2B UA, replacing TO, FROM and URI fields. I know "rewritehostport" function allow me to modify URI. What are functions can be used to change TO and FROM fields. Thanks, _________________________________________________________________ Get more out of the Web. Learn 10 hidden secrets of Windows Live. http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns!5…

6 25

[Serusers] About a SIP load balancing document involving SER
by Iñaki Baz Castillo 17 Oct '08

17 Oct '08

Hi, I've found a document about SIP load balancing in iptel.org website and I would like to comment an error I've found in the document. The document is this: "Towards Effective SIP load Balancing" http://www.iptel.org/voipsecurity/doc/14%20-%20Kambourakis%20-%20Towards%20… Page 7 says: -------------------- SIP Proxies insert a VIA header in all the incoming SIP requests So… In case the LB (Load Balancer) is implemented as another proxy all SIP responses will pass through that proxy Request that belong to a specific dialog should not pass through LB -------------------- This is completely true, but next page (8) says: -------------------- Transparency for responses Prevent Load Balancer from inserting a VIA header E.g. in SER utilizing the SEND core command Modify the SIP's Proxy core to ignore the VIA-header added by the Load Balancer ------------------- There is an important error that unfortunatelly I've realized it's very common. Section 18.2.2 of RFC 3261 says clearly that the responses are *always* sent through the same nodes the request came from. So the response should always traverse the load balancer. 1) Load balancer --- (SIP UDP) ---> UAS In this case the UAS would always reply to the *real* source IP (if this is different of the Via "sent-by" then UAS adds "received" parameter and replies there). 2) Load balancer --- (SIP TCP/SCTP) ---> UAS By definition a UAS must reply using the incoming TCP connection. So it's extrange for me that a document about SIP load balancing tries to offer solutions that are not SIP compliant and also unfeasible (UAS will always reply to the real source IP regardless of the Via content). Since the document proposes SER based solutions (using "SEND" command) I'd just like to confirm if I'm completely right, or maybe it's common those not SIP compliant methods by some vendors in order to provide a load balancing solution. Thanks a lot for any comment and best regards. -- Iñaki Baz Castillo

5 18

Re: [Kamailio-Users] In-dialog request correlation without loose routing?
by Alex Balashov 17 Oct '08

17 Oct '08

Klaus Darilion wrote: > So do you perform lookup() also for in_dialog requests? When necessary. Otherwise, A endpoint just provides the URI of the Z-end of the signaling path for the domain and I make special exceptions to relay that as long as I have the Call-ID stored somewhere, which I do. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

2 4

Re: [Kamailio-Users] In-dialog request correlation without loose routing?
by Alex Balashov 17 Oct '08

17 Oct '08

Klaus Darilion wrote: > Alex Balashov schrieb: >> BTW, I do think it would be a good idea for the dialog module to >> export these functions directly into the script symbols so they can be >> called that way. I do not like to do loose routing unnecessarily / >> when I have no use for it. > > How does your setup work without loose_route? The proxy sees in-dialog > requests only if you record_route. If you do record_route(), you have to > use loose_route for correct routing. You can't have one without the > other (except you do manual routing also for in-dialog requests) That's generally what I always do - manual routing for in-dialog requests. I was not aware that loose_route() is required for correct routing of in-dialog requests when they are record-routed back through the proxy. I think one of the reasons why this may not be much of an issue for me is because my proxy applications generally always have the proxy as the URI domain - I won't relay for !uri == myself. So, after the initial INVITE is rewritten, the UAC/UAS cores on either side send subsequent in-dialog requests to the same URI they sent the INVITE to. I would, of course, be eager to hear any methodological insights you may offer about what I'm doing incorrectly. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

1 0

[Kamailio-Users] Openser 1.3.2 to Kamailio 1.4.1 Upgrade Problem. Pls HELP.
by George Lee 17 Oct '08

17 Oct '08

Hi, I am upgrading openser 1.3.2 to kamailio 1.4.1 and I am having this problem: Oct 16 18:20:25 dev-ser-01 /sbin/kamailio[6466]: ERROR:core:db_check_table_version: invalid version 2 for table presentity found, expected 3 (check table structure and table "version") Is there any database schema change to the presentity table? If so, how do I upgrade the presentity table? I tried to run the kamdbctl script to upgrade the database and I got access denied with 'root' user. How do I run kamdbctl with 'openser' and 'openserrw' to access mysql database? Also, my database is on a remote host so kamdbctl may not be useful. Thanks, George

2 2

[Kamailio-Users] In-dialog request correlation without loose routing?
by Alex Balashov 17 Oct '08

17 Oct '08

Also, is there a way to get the dialog module to match subsequent in-dialog requests (i.e. re-INVITEs, ACKs, BYEs) without using the dialog correlator field that gets stamped into the Route: header, and therefore requiring the use of loose_route() to read and consume? I thought that switching the matching mode to pure SIP attributes only would do the trick, but it doesn't. They won't match. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

4 24

Re: [Kamailio-Users] Client REGISTERED but "404-NOT FOUND"
by luzango mfupe 17 Oct '08

17 Oct '08

Hi Iak and Victor, The problem is now solved, the Client appeared to be buggy and not my Kamailio config. Cheers, Lu. Message: 4 Date: Thu, 16 Oct 2008 17:26:24 +0200 From: " Victor Pascual ?vila " <victor.pascual.avila(a)gmail.com> Subject: Re: [Kamailio-Users] Client REGISTERED but "404-NOT FOUND" during INVITE To: " I?aki Baz Castillo " <ibc(a)aliax.net> Cc: users(a)lists.kamailio.org Message-ID: <618e24240810160826l700fbca9gb61e52ed43dc479(a)mail.gmail.com> Content-Type: text/plain; charset=UTF-8 On Wed, Oct 15, 2008 at 10:48 AM, I?aki Baz Castillo <ibc(a)aliax.net> wrote: > 2008/10/15 luzango mfupe <luzango.mfupe(a)gmail.com>: >> >> >> Hi Mates, >> I am testing a Client in Nokia E65 using kamailio 1.3.3. I managed to >> successfull get it Registered but when i attempt to make a call i always get >> the "404 Not Found" message in the INVITE. Strangely enough, the Client >> appears to still be ONLINE in the location table. > > Your conclusion is wrong. A user doen't need to be registered in order > to make a PSTN call (except if you add that logic to your script, that > is not the case). > > So, forget yuor user is registered, it's doesn't matter, and debug why > your dialed number is not matched as a number to the PSTN. I?aki is right here. In addition, "404 Not Found" is not the correct reponse when the destination target is not registered, IMO it should be a 480 instead. -- Victor Pascual ?vila ------------------------------ Message: 5 Date: Thu, 16 Oct 2008 18:23:19 +0200 From: " I?aki Baz Castillo " <ibc(a)aliax.net> Subject: Re: [Kamailio-Users] Client REGISTERED but "404-NOT FOUND" during INVITE Cc: users(a)lists.kamailio.org Message-ID: <cc1f582e0810160923h5e6f9ffeq2d6bb8452730646b(a)mail.gmail.com> Content-Type: text/plain; charset=UTF-8 2008/10/16 Victor Pascual ?vila <victor.pascual.avila(a)gmail.com>: > In addition, "404 Not Found" is not the correct reponse when the > destination target is not registered, IMO it should be a 480 instead. Yeah!!! 404 should be returned when the proxy/UAS received a request and it's not responsible for it, for example a non existing user (non existing user != non registered user). When a user is not registered, but it does exist in the server, the server must reply "480". -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......." -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

1 0

[Kamailio-Users] Send me an SMS
by Sunkara RaviPrakash 17 Oct '08

17 Oct '08

1 0

[Kamailio-Users] Send me an SMS
by Sunkara RaviPrakash 17 Oct '08

17 Oct '08

1 0

[Serusers] handing of NAT use MediaProxy
by guolingxi＠126.com 17 Oct '08

17 Oct '08

---------- 转发邮件信息 ---------- 发件人："guolingxi <guolingxi(a)126.com>" 发送日期：2008-10-17 13:48:23 收件人：serusers(a)iptel.org 主题： handing of NAT use MediaProxy Hello: I am a chinese student in beijing University.Recently I intrested in SIP,and I found IPTEL is a good place.I enjoy it very much. I have read your "SER-getting started",it is a very wondful guiding.But i am a beginner,i am confused by some question when I want to use SER to hand of NAT Travesal. Could you give me some advice if I want to use SER to set a serve to do a experiment about NAT Travesal.What hardwares must I need?Now I have only some normal computers ,are they Sufficient？Looking forward your advice or Solution，I will very appreciate if you can provide me some concreted document. You can write to me on you convinience,Thank you for your reading!:) a confused student 2008.10.17

1 0

[Kamailio-Users] lcr module enhancements
by jh＠tutpro.com 16 Oct '08

16 Oct '08

i just committed to kamailio trunk an enhanced version of lcr module. changes from kamailio wiki page http://www.kamailio.org/dokuwiki/doku.php/features:new-in-1.5.x are as follows: * New high-performance implementation that keeps lcr information in in-memory hash table, whose size can be given in a new module parameter 'lcr_hash_size'. See lcr/README for lcr function execution times. * New 'weight' field in 'gw' table that can be used to assign a gateway a weight among gateways of its group. * Support for prefix_mode=1 has been dropped. * lcr_dump MI function has been split into lcr_gw_dump and lcr_lcr_dump functions. * lcr_reload function is now executed under a lock thus minimizing race conditions. * Regular expressions are now Perl 5.x, instead of POSIX, compatible due to use of PCRE regular expression library. testing and feedback is appreciated. -- juha

1 0

[Kamailio-Users] Client REGISTERED but "404-NOT FOUND" during INVITE
by luzango mfupe 16 Oct '08

16 Oct '08

Hi Mates,I am testing a Client in Nokia E65 using kamailio 1.3.3. I managed to successfull get it Registered but when i attempt to make a call i always get the "404 Not Found" message in the INVITE. Strangely enough, the Client appears to still be ONLINE in the location table. ################################ kamailio config below ############################## if(!is_method("REGISTER")){ if(nat_uac_test("19")){ record_route(";nat=yes"); } else { record_route(); }; }; ##Loose_route packets if (has_totag()) { #sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if(method=="BYE") { #Account BYE transactions setflag(2); }; #Check authentication of re-invites if(method=="INVITE" && (!allow_trusted())) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Forbidden, use From=ID"); exit; }; }; if(method=="BYE" || method=="CANCEL") { end_media_session(); }; ##Detect requests in the dialog behind NAT and flag with 6 if(nat_uac_test("19") || search("^Route:.*;nat=yes")){ append_hf("P-hint: LR|fixcontact,setflag6\r\n"); fix_contact(); setflag(6); }; route(1); } else { sl_send_reply("404","Not here"); } exit; } ##Loose_route packets if (loose_route()) { if(!has_totag()){ sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; }; route(1); }; #CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { end_media_session(); t_relay(); }; exit; } t_check_trans(); # # -- 3 -- Determine Request Target # if (method=="REGISTER") { route(2); } else { route(3); }; } route[1] { # # -- 4 -- Forward request to target # # Forward statefully t_on_reply("1"); t_on_failure("1"); if (!t_relay()) { # sl_reply_error(); }; exit; } route[2] { ## Register request handler if (is_uri_host_local()) { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; }; if (!check_to()) { sl_send_reply("403", "Forbidden"); exit; }; if(!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; save("location"); exit; } else if { sl_send_reply("403", "Forbidden"); }; } route[3] { ## Requests handler if (method=="INVITE") { # Account INVITE packets setflag(2); # Account Missed calls setflag(3); # Radius Extra # $avp(s:sip_proxy_ip)="127.0.0.1"; }; if (is_from_local()){ # From an internal domain -> check the credentials and the FROM if(!allow_trusted()){ if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if(!check_from()) { sl_send_reply("403", "Forbidden, use From=ID"); exit; }; }; if (client_nat_test("3")) { append_hf("P-hint: route(3)|setflag7,forcerport,fix_contact\r\n"); setflag(7); force_rport(); fix_contact(); }; #unconditional call forward if(avp_db_load("$ru/username","$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); route(1); exit; } consume_credentials(); #verify aliases, if found replace R-URI. lookup("aliases"); if (is_uri_host_local()) { # -- Inbound to Inbound route(10); } else { # -- Inbound to outbound route(11); }; } else { #From an external domain ->do not check credentials #Verify aliases, if found replace R-URI. lookup("aliases"); if (is_uri_host_local()) { #-- Outbound to inbound route(12); } else { # -- Outbound to outbound route(13); }; }; } route[4] { # routing to the public network rewritehostport("xx.xxx.xxx.xx:5065"); #t_on_failure("2"); if (!t_relay()) { sl_reply_error(); }; exit; } route[6] { # # -- NAT handling -- # if (isflagset(6) || isflagset(7)) { append_hf("P-hint: Route[6]: mediaproxy \r\n"); use_media_proxy(); }; } route[10] { #from an internal domain -> inbound #Native SIP destinations are handled using the location table #Gateway destinations are handled by regular expressions append_hf("P-hint: inbound->inbound \r\n"); if (uri=~"^sip:0[1-9][0-9]*@") { if (is_user_in("credentials","local")) { strip(1); prefix("27"); route(6); route(4); exit; } else { sl_send_reply("403", "No permissions for local calls"); exit; }; }; if (uri=~"^sip:00[1-9][0-9]*@") { if (is_user_in("credentials","int")) { strip(2); route(6); route(4); exit; } else { sl_send_reply("403", "No permissions for international calls"); }; }; if (!lookup("location")) { route(6); route(1); } else { sl_send_reply("404", "Not Found"); exit; }; route(6); route(1); } route[11] { # from an internal domain -> outbound # Simply route the call outbound using DNS search append_hf("P-hint: inbound->outbound \r\n"); route(1); } route[12] { # From an external domain -> inbound # Verify aliases, if found replace R-URI. lookup("aliases"); if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; route(1); } route[13] { #From an external domain outbound #we are not accepting these calls append_hf("P-hint: outbound->inbound \r\n"); sl_send_reply("403", "Forbidden"); exit; } failure_route[1] { ##-- ##-- If cancelled, exit. ##-- if (t_was_cancelled()) { exit; }; ##-- ##-- If busy send to the e-mail server, prefix the "b" ##-- character to indicate busy. ##-- if (t_check_status("486")) { revert_uri(); prefix("b"); rewritehostport("xx.xxx.xxx.xx:5065"); append_branch(); route(1); exit; }; ##-- ##-- If timeout (408) or unavailable temporarily (480), ##-- prefix the uri with the "u"character to indicate ##-- unanswered and send to the e-mail ##-- sever ##-- if (t_check_status("408") || t_check_status("480")) { revert_uri(); prefix("u"); rewritehostport("xx.xxx.xxx.xx:5065"); append_branch(); route(1); exit; }; } failure_route[2] { t_on_failure("2"); t_relay(); } onreply_route[1] { # #-- On-replay block routing -- # if (client_nat_test("1")) { append_hf("P-hint: Onreply-route - fixcontact \r\n"); fix_contact(); }; if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (search("^Content-Type:[ ]*application/sdp")) { append_hf("P-hint: onreply_route|usemediaproxy \r\n"); use_media_proxy(); }; }; exit; } ###################################################################### Captured Invite packets below. ##################################################################### . U 2008/12/12 20:41:19.227089 41.23.215.255:5060 -> xx.xxx.xxx.xx:5060 INVITE sip:0123825710@xx.xxx.xxx.xx SIP/2.0. Via: SIP/2.0/UDP 41.23.215.255:5060 ;branch=z9hG4bK5lvrd249rhhc6dbif4j3i48;rport. From: <sip:1974@10.20.11.62 <sip%3A1974(a)10.20.11.62> >;tag=gm8rd21tl9hc68h6f4j2. To: <sip:0123825710@xx.xxx.xxx.xx>. Contact: <sip:1974@41.23.215.255 <sip%3A1974(a)41.23.215.255>>. Accept-Contact: *;+media=audio. CSeq: 953 INVITE. Call-ID: -u6Sy5l7oIf2mTACTYE46_z8sRn-_f. User-Agent: Tuune Mobile. Max-Forwards: 70. Content-Type: application/sdp. Content-Length: 172. . v=0. o=- 54865261 0 IN IP4 41.23.215.255. s=Tuune Mobile. c=IN IP4 41.23.215.255. t=0 0. m=audio 5000 RTP/AVP 0 97. a=sendrecv. a=rtpmap:0 PCMU/8000. a=rtpmap:97 AMR/8000. U 2008/12/12 20:41:19.227945 xx.xxx.xxx.xx:5060 -> 41.23.215.255:5060 SIP/2.0 404 Not Found. Via: SIP/2.0/UDP 41.23.215.255:5060 ;branch=z9hG4bK5lvrd249rhhc6dbif4j3i48;rport= 5060. From: <sip:1974@10.20.11.62 <sip%3A1974(a)10.20.11.62> >;tag=gm8rd21tl9hc68h6f4j2. To: <sip:0123825710@xx.xxx.xxx.xx >;tag=329cfeaa6ded039da25ff8cbb8668bd2.bf78. CSeq: 953 INVITE. Call-ID: -u6Sy5l7oIf2mTACTYE46_z8sRn-_f. Server:Tuune Mobile Test. Content-Length: 0. . #################################### It is online ############ 41:/# openserctl ul show database engine 'MYSQL' loaded Control engine 'FIFO' loaded entering fifo_cmd ul_dump Domain:: aliases table=512 records=0 max_slot=0 Domain:: location table=512 records=1 max_slot=1 AOR:: 1974 Contact:: sip:1974@41.22.248.8 <sip%3A1974(a)41.22.248.8> Q= Expires:: 932 Callid:: x1ySy9r2oIf1Ogj2Uzeht2FSEcYGxE Cseq:: 957 User-agent:: n/a State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:xx.xxx.xxx.xx:5060 Methods:: 4294967295 FIFO command was: :ul_dump:openser_receiver_15196 I hope some one can give a clue, WBR, Lu. -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

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[Kamailio-Users] Problems installing serMyAdmin0.9 in OpenSER1.3.2
by Nuno Marques 16 Oct '08

16 Oct '08

Hi, I tried to install the new version of sermyadmin - 0.9 - and it gave me problems. Created a new openser database and notice that the sermyadmin didn't even made any changes to it. Tried with the old sermyadmin 0.7 and everything went well. I didn't made any changes to tomcat... or anything else. As anyone experienced the same problem? Does anyone knows where could i get sermyadmin0.8 to test it? Thanks in advance for any help on this Nuno

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[Kamailio-Users] debian and solaris packages for kamailio and openser available
by Henning Westerholt 16 Oct '08

16 Oct '08

Hello all, thanks to Muhammad Akl, packages and binaries for the lastest stable release of openser on solaris 10 (on i386) are now available on www.kamailio.org in the download area [1, 2]. He also build packages for some older releases, like the previous 1.3 and 1.2 branch based releases. I added debian packages for the 1.3.3 release of openser and 1.4.1 release of kamailio for etch (on i386) as well [3, 4]. Packages of kamailio on solaris will be also appear soon there, they are still a few incompatibilites that needs to be sorted out that prevent the building. We're working on it.. If you want to contribute some packages for your favorite distribution or operating system, just contact me per mail or on the #kamailio IRC channel. Cheers, Henning [1] http://www.kamailio.org/pub/openser/1.3.3/bin/solaris/10/ [2] http://www.kamailio.org/pub/openser/1.3.3/packages/solaris/10/ [3] http://www.kamailio.org/pub/openser/1.3.3/packages/debian/etch/ [4] http://www.kamailio.org/pub/kamailio/1.4.1/packages/debian/etch/ -- Henning Westerholt - Development Consumer Products / DSL Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

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[Kamailio-Users] 1.4.x Dialog module vs. :dlg_end_dlg: command
by Martin Vít 16 Oct '08

16 Oct '08

Hello, i'm testing dialog module and trying to account died sessions. I've problem or i've misunderstood the way how to account it. If I send dlg_end_dlg, a dialog session is ended and BYE is sended to both sides. But if the BYE is not confirmed the acc module will never log this. Is there any way to immidiatly log BYE asap dialog ends? thanks for any suggestion MV

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[Kamailio-Users] Change FROM display name
by ALAEDDINE abbech 16 Oct '08

16 Oct '08

Hi, I should change the actual display name in header From with an other name stored in avp(s:display) . I use uac_replace_from("$avp(s:display)","");, but this function add the new name in the end line of From. Exemple after tcpdump: i like change "aladin" with "1111111111" but i have this line in INVITE request, From: aladin <sip:testerser@x.x.x.x:5060>;tag=64204a12d09be0c^M"11111111111" Have you a solution for this pb. Thanks

2 1

[Kamailio-Users] INVITE queue
by sabino frisardi 16 Oct '08

16 Oct '08

Hi all, i'm tryng to enqueue invite messages in a queue and to make the parsing of the invite after the enqueuing. After the parsing i call tmb.t_relay(msg,0,0,0,0,0,0) from a module. The rest of messages of the classic sip call pass trhough the main route. So when i run openser it consume too much memory. Is there any error with the call to t_relay through the binding? Is there any error on the transaction if i call the relay in a module for INVITE message and in main route for NON_INVITE messages? Why the consume of memory increase? Can anyone help me? Thanks for your help. Regards -- Sabino Frisardi

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Re: [Kamailio-Users] Problems installing serMyAdmin0.9 in OpenSER1.3.2
by Nuno Marques 16 Oct '08

16 Oct '08

Hi Muhammad, The only error that appears is when i tried to access http:// <server_ip>:8080/serMyAdmin/ Gives me an error 404 and says that the resource is not available... HTTP Status 404 - ------------------------------ *type* Status report *message* *description* *The requested resource () is not available.* ------------------------------ Apache Tomcat/6.0.18 I've run java -version and my result is: java version "1.5.0_14" Java(TM) 2 Runtime Environment, Standard Edition (build 1.5.0_14-b03) Java HotSpot(TM) Client VM (build 1.5.0_14-b03, mixed mode, sharing) I think my problem is with the version of Java. Going to try to instal java 6. Thanks a lot Muhammad. Nuno 2008/10/16 muhammad akl <muhammad.akl(a)gmail.com> > > Hi Nuno , > > serMyAdmin-0.9 will work with any version of openser and/or kamailio , > > but alot of changes have been added to the NEW BETA version (0.9) and the > changes AFAIK are depending on java 6 could you please append the error that > appeared for you ? > > Regards > > Muhammad >

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[Kamailio-Users] 1.4.x Dialog module vs. :dlg_end_dlg: command
by Martin Vít 16 Oct '08

16 Oct '08

Hello, i'm testing dialog module and trying to account died sessions. I've problem or i've misunderstood the way how to account it. If I send dlg_end_dlg, a dialog session is ended and BYE is sended to both sides. But if the BYE is not confirmed the acc module will never log this. Is there any way to immidiatly log BYE asap dialog ends? thanks for any suggestion MV

1 0

[Kamailio-Users] Dynamic profile creation for dialog module.
by Alex Balashov 16 Oct '08

16 Oct '08

Hello, It'd be really nice to have a way to create profiles used by the dialog module dynamically, ideally in script but MI would work too. The problem is that I am using many different profiles to keep track of the number of open dialogs for a lot of different call formulas, some of whose existence should be user-configurable. But these profiles have to be defined statically as having values using the dialog module's parameter "profiles_with_value," because I do use pseudovariable values with them. Now, of course, one can simply use one or more big profiles and construct pseudovariable values based on what needs to be profiled, e.g. "$rU:bigblend" But whereas the profiles are matched to tracked dialogs using a hash, the values are linearly compared using memcmp() in modules/dialog/dlg_profile.c:is_dlg_in_profile(). So, there is a speed penalty in doing this. Cheers, -- Alex -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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[Kamailio-Users] How to change shared memory size?
by Pablo Bertuzzi 15 Oct '08

15 Oct '08

Hi Forum, kamailio failed to start I believe due to a large carrierrote table (50K prefixes). Kamailio version is 1.4.1, OS is debian, physical memory installed is 4Gb. How can I change the size? Oct 14 15:01:18 sip-proxy /usr/local/sbin/kamailio[3541]: INFO:carrierroute:get_route_tree: found domain 0 Oct 14 15:01:18 sip-proxy /usr/local/sbin/kamailio[3541]: INFO:carrierroute:add_route: found route, now adding Oct 14 15:01:18 sip-proxy /usr/local/sbin/kamailio[3541]: ERROR:carrierroute:add_route_rule: out of shared memory Oct 14 15:01:18 sip-proxy /usr/local/sbin/kamailio[3541]: ERROR:carrierroute:prepare_route_tree: could not load routing data Oct 14 15:01:18 sip-proxy /usr/local/sbin/kamailio[3541]: ERROR:carrierroute:mod_init: could not prepare route tree Oct 14 15:01:18 sip-proxy /usr/local/sbin/kamailio[3541]: ERROR:core:init_mod: failed to initialize module carrierroute Oct 14 15:01:18 sip-proxy /usr/local/sbin/kamailio[3541]: ERROR:core:main: error while initializing modules

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[Kamailio-Users] new feature: non-blocking logging
by Henning Westerholt 15 Oct '08

15 Oct '08

Hi all, i just commited support for non-blocking logging to the trunk. This was implemented using the syslog-async functionality from the dnsmasq server [1] from Simon Kelley. The standard Unix syslog() library routine can block waiting for the syslog daemon. On some systems, using a datagram socket for /dev/log avoids endless waits, but on Linux, even this does not work. Try typing killall -SIGSTOP $pid_of_syslog into a root terminal on a non-critical machine. Then generate traffic to get some logs, eventually, the server will stop. This blocking of the syslog daemon can happens because of bugs, or if the daemon waits for external events, like DNS lookups. Instead of blocking, log-lines are buffered in memory. The buffer size is limited and if the buffer overflows log lines are lost. When lines are lost this fact is logged with a message. This functionality can be enabled with setting the define SYSLOG_ASYNC in the Makefile.defs file. The default is the normal log functionality from the system library, nothing changed if the define is not set. I tested this with our internal tests suite, and also the tests in the trunk. You can get the code from the svn repository, we also provide debian packages for several debian releases on devel.kamailio.org. Cheers, Henning [1] http://www.thekelleys.org.uk/dnsmasq/doc.html -- Henning Westerholt - Development Consumer Products / DSL Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

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[Kamailio-Users] CDRTool 6.6.4 with freeradius-cdrtool
by Diego Zuaneti Arruda 15 Oct '08

15 Oct '08

Hi all, Someone has a patch for use freeradius-cdrtool with the latest version of cdrtool ? The stable versions of cdrtool not have multiple commands that were in earlier versions. Thanks. Diego.

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[Kamailio-Users] I want to have both Presence Network Agent and Resource List Service in my presence network.
by suresh tattikota 15 Oct '08

15 Oct '08

Hi, I want to have both Presence Network Agent and Resource List Service in my presence network. As per the architecture, both presence network agent and resource list server can act as the application servers, and they can communicate with presence server. Can anybody let me know, how to make the communication possible between application server and the presence server? Regards, Suresh

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Re: [Kamailio-Users] Interfaceing Presence server with Presence Network Aget(PNA)
by Daniel-Constantin Mierla 15 Oct '08

15 Oct '08

please cc to mailing list. Daniel On 10/15/08 07:51, suresh tattikota wrote: > hi Daniel thank u > > > *PNA*:presence network agent , it collects the presence information of > the network element like *HSS*(Home Subscriber Server) and > *S-CSCF*(serving call session control function) (i.e registered are > unregistered ).Using '*sh'* interface i am collecting these > information through *SNR/SNA,PNR/PNA*(subscribe notify request and > answer,,push notify request/answer ) after getting the PNR i have to > send PUBLISH message to the presence server through *PEN*(its nothing > but publish message )interface , > > My question is how to connect the presence server with presence > network agent . > and how to attach the presence information to the PUBLISH XML body . > > thank you very much > > > On 10/14/08, *Daniel-Constantin Mierla* <miconda(a)gmail.com > <mailto:miconda@gmail.com>> wrote: > > Hello, > > On 10/13/08 12:30, suresh tattikota wrote: > > > hi all > > I am trying to develop the interface between PNA and PS,that > is PEN interface, > for that i am using the one openser 1.2 as the PNA > Application server and Presence server. > > it is recommended that you use kamailio 1.4.1 as presence server, > it has many improvements in this area, openser 1.2 is rather old > and not supported anymore > > > I already developed sh interface between HSS and the PNA. > After getting the PNR request from HSS,PNA has to send the > PUBLISH message to PS > > so here my doubt is > > 1.How to connect the PNA and PS > 2.how to construct PUBLISH XML body > 3.can i use the opensips/openser as presence network agent > > The problem with the questions you ask is that you use a lot of > abbreviations from IMS specs (as they look) . You won't find good > answers unless you get to more details of what you want to > achieve, preferably at SIP level - nobody will spend time to > understand what PS, PEN, PNA are supposed to do. > > Kamailio (OpenSER) can be used as: > - presence server - can handle PUBLISH/SUBSCRIBE messages and send > NOTIFYs > - presence user agent - initiate PUBLISH requests > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com <http://www.asipto.com/> > > -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] New OpenXCAP release 1.0.5
by Adrian Georgescu 15 Oct '08

15 Oct '08

Hello, There are new releases of the open source OpenXCAP server and a Python XCAP library available. They contain many bug fixes and lessons learnt at SIPIT 23, still in progress in Lannion, France. Changelogs are attached: openxcap (1.0.5) unstable; urgency=low * fixed: openxcap with 'openser' backend could be start if enable_publish_xcapdiff=yes * fixed a couple of potential Internal Server Errors * added python-mysqldb to dependencies * added <entry> node to <uniqueness-failure> error as required by RFC * nicer error messages in several cases openxcap (1.0.4) unstable; urgency=low * Disabled authentication for requests in /global/ context, because it didn't work well. It is safe, because the only use for /global/ now is retrieving xcap-caps. openxcap (1.0.3) unstable; urgency=low * Fixed: requesting XCAP root resulted in 500 (now 404) * Fixed: multiple matches resulted in 500 (now 404) * Improved error responses openxcap (1.0.2) unstable; urgency=high * Fixed and improved logging. it now logs much more then before and can be customized * Fixed bug: xcap-caps application was not available under /global/ tree * Added experimental support for sqlite database * Fixed multiple xcap roots/aliases. having more then one root= entry in the config is legal now * Fixed bug that made OpenXCAP dependent on specific MySQLdb versions (those with 'reconnect' parameter) * Fixed pyxml issue: previously, if pyxml was not installed, OpenXCAP would fail with obscure exception on element operation. Now it refuses to start. * Fixed URI parsing * Better error reporting in many cases * Many more bugs fixes openxcap (1.0.1) unstable; urgency=low * Added proper URI parsing, so att_value may contain slashes now * Fixed schemas not to fetch files from the internet on initialization python-xcaplib (1.0.6) unstable; urgency=low * made "index" default filename for xcapclient as well * made --app unnecessary for all types of requests if -i is present python-xcaplib (1.0.5) unstable; urgency=low * changed default filename from index.xml to index * changed default auth to None (support both basic and digest) * removed loading options from environ (XCAP_ROOT etc) * added -a, --account-name option * added --show-config option python-xcaplib (1.0.4) unstable; urgency=low * added --sip-address option (sip_address in the config file) python-xcaplib (1.0.3) unstable; urgency=low * added support for documents in the global context * added --filename parameter to xcapclient python-xcaplib (1.0.2) unstable; urgency=low * removed setuptools from setup.py completely python-xcaplib (1.0.1) unstable; urgency=low * fixed setup.py to use setuptools only if available How to upgrade or install the software For Debian testing or unstable on an i386 architecture there is an official public repository provided by AG Projects. Install the AG Projects debian software signing key: wget http://download.ag-projects.com/agp-debian-gpg.key apt-key add agp-debian-gpg.key Add these lines in etc/apt/sources.list # AG Projects software deb http://ag-projects.com/debian unstable main deb-src http://ag-projects.com/debian unstable main After that, run: apt-get update apt-get install openxcap python-xcaplib For other operating systems download the tar archive from: http://openxcap.org Kind regards, Adrian Georgescu

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Re: [Kamailio-Users] Client REGISTERED but "404-NOT FOUND"
by luzango mfupe 15 Oct '08

15 Oct '08

> > Hi Castilo, Yea you might be right that my conclusion is wrong but in my case at least a client must be in the ACL in order to be able to make PSTN calls,unregistered clients cant call as well. I tested the same configuration with a X-lite client and it was possible to make calls. > ...debug why your dialed number is not matched as a number to the PSTN. > > Can you explain here a lil bit more?? because the regex part in my config > is working fine it matches all PSTN[Local & Int.] perfectly!! Cheers, Lu. > > > Message: 2 > Date: Wed, 15 Oct 2008 10:48:48 +0200 > From: " I?aki Baz Castillo " <ibc(a)aliax.net> > Subject: Re: [Kamailio-Users] Client REGISTERED but "404-NOT FOUND" > during INVITE > Cc: users(a)lists.kamailio.org > Message-ID: > <cc1f582e0810150148h1ba9b282t94ae8278c3550e27(a)mail.gmail.com> > Content-Type: text/plain; charset=UTF-8 > > 2008/10/15 luzango mfupe <luzango.mfupe(a)gmail.com>: > > > > > > Hi Mates, > > I am testing a Client in Nokia E65 using kamailio 1.3.3. I managed to > > successfull get it Registered but when i attempt to make a call i always > get > > the "404 Not Found" message in the INVITE. Strangely enough, the Client > > appears to still be ONLINE in the location table. > > Your conclusion is wrong. A user doen't need to be registered in order > to make a PSTN call (except if you add that logic to your script, that > is not the case). > > So, forget yuor user is registered, it's doesn't matter, and debug why > your dialed number is not matched as a number to the PSTN. > > > -- > I?aki Baz Castillo > <ibc(a)aliax.net> > > ------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > End of Users Digest, Vol 41, Issue 42 > ************************************* > -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

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[Kamailio-Users] Rouge UAC: location database not updated
by CSB 15 Oct '08

15 Oct '08

I see the following messages in the logs: Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:parse_from_header: bad from header Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:siptrace:sip_trace: cannot parse FROM header Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:get_hdr_field: bad to header Oct 14 15:59:58 beta /sbin/openser[32535]: INFO:core:parse_headers: bad header field Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:maxfwd:is_maxfwd_present: parsing MAX_FORWARD header failed! Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:get_hdr_field: bad to header Oct 14 15:59:58 beta /sbin/openser[32535]: INFO:core:parse_headers: bad header field Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:rr:find_first_route: failed to parse headers Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:parse_from_header: bad from header Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:parse_from_uri: cannot parse FROM header Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:domain:is_from_local: Error while parsing From header Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:get_hdr_field: bad to header Oct 14 15:59:58 beta /sbin/openser[32535]: INFO:core:parse_headers: bad header field Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:get_hdr_field: bad to header Oct 14 15:59:58 beta /sbin/openser[32535]: INFO:core:parse_headers: bad header field Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:build_res_buf_from_sip_req: parse_headers failed Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:sl:sl_send_reply_helper: response building failed Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:get_hdr_field: bad to header Oct 14 15:59:58 beta /sbin/openser[32535]: INFO:core:parse_headers: bad header field Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:pv_get_to_attr: cannot parse To header Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:parse_from_header: bad from header Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:pv_get_from_attr: cannot parse From header Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:parse_from_header: bad from header Oct 14 15:59:58 beta /sbin/openser[32535]: ERROR:core:pv_get_from_attr: cannot parse From header This seems to have been caused by one rogue UAC and has the effect of preventing entries being written to the location database. My question: could anyone suggest how to prevent a UAC which is sending some weird data from causing the server to choke? Thanks Cameron

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Re: [Kamailio-Users] problem starting kamailio
by ingdavidcespedes＠cable.net.co 15 Oct '08

15 Oct '08

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[Kamailio-Users] problem starting kamailio
by Fabian Borot 15 Oct '08

15 Oct '08

hello I installed kamailio, and want to use mysql auth,loaded the mysql module, uncomented the lines needed to use mysql in the .cfg file I ran the openser_mysql.sh script, and when I try to started kamailio and this is what I see in the console: [root@sp1094a sbin]# ./openser_mysql.sh create MySql password for root: Domain (realm) for the default user 'admin': 65.111.185.187 creating database openser ... Install SERWEB tables ?(y/n):n # I tried with and w/o the servweb tables [root@sp1094a sbin]# ./kamctl start database engine 'MYSQL' loaded Control engine 'FIFO' loaded INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed in the var/log/message file this is what I see: Oct 14 21:09:12 sp1094a ./kamailio[29249]: NOTICE:core:main: version: kamailio 1.4.1-notls (i386/linux) Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:core:main: using 32 Mb shared memory Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:core:main: using 1 Mb private memory per process Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:sl:mod_init: Initializing StateLess engine Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:tm:mod_init: TM - initializing... Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:maxfwd:mod_init: initializing... Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:usrloc:ul_init_locks: locks array size 512 Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:registrar:mod_init: initializing... Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:textops:mod_init: initializing... Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:xlog:mod_init: initializing... Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:acc:mod_init: initializing... Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:auth:mod_init: initializing... Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:auth_db:mod_init: initializing... Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 210 kb Oct 14 21:09:12 sp1094a ./kamailio[29253]: ERROR:db_mysql:db_mysql_submit_query: driver error on query: Unknown column 'cflags' in 'field list' Oct 14 21:09:12 sp1094a ./kamailio[29253]: ERROR:core:db_do_query: error while submitting query Oct 14 21:09:12 sp1094a ./kamailio[29253]: ERROR:usrloc:preload_udomain: db_query (1) failed Oct 14 21:09:12 sp1094a ./kamailio[29253]: ERROR:usrloc:child_init: child(1): failed to preload domain 'location' Oct 14 21:09:12 sp1094a ./kamailio[29253]: ERROR:core:init_mod_child: failed to initializing module usrloc, rank 1 Oct 14 21:09:12 sp1094a ./kamailio[29253]: ERROR:core:main_loop: init_child failed for UDP listener Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:core:handle_sigs: child process 29253 exited normally, status=255 Oct 14 21:09:12 sp1094a ./kamailio[29249]: INFO:core:handle_sigs: terminating due to SIGCHLD Oct 14 21:09:12 sp1094a ./kamailio[29264]: INFO:core:sig_usr: signal 15 received Oct 14 21:09:12 sp1094a ./kamailio[29261]: INFO:core:sig_usr: signal 15 received Oct 14 21:09:12 sp1094a ./kamailio[29259]: INFO:core:sig_usr: signal 15 received Oct 14 21:09:12 sp1094a ./kamailio[29257]: INFO:core:sig_usr: signal 15 received Oct 14 21:09:12 sp1094a ./kamailio[29255]: INFO:core:sig_usr: signal 15 received what can that be? txs Fabian <html><DIV> <DIV> </DIV></DIV></html> _________________________________________________________________ Get more out of the Web. Learn 10 hidden secrets of Windows Live. http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns!5…

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Re: [Kamailio-Users] How to handle DoS attack and OpenSER Pike module!
by Daniel-Constantin Mierla 14 Oct '08

14 Oct '08

Hello, On 10/14/08 09:32, Arif-Uz-Zaman wrote: > Hi all, > I need to bother about crazy client by considering “Flood” detection > technique. I can do it by using OpenSER Pike > <http://kamailio.org/docs/modules/1.2.x/pike.html> module which helps > to keep trace of all (or selected ones) incoming request's IP source > and blocks the ones that exceeded some limit. > > In my case: If the number of SIP messages from a single IP address to > my SIP Proxy exceeds *200* per minute. Recommended action: Block IP > for 2 hours. > > I tried with the pike module but I’m little bit confused with > sampling, density, and timeout value. > > Please help me with example configuration by considering my point. > have you tried: modparam("pike", "sampling_time_unit", 60) modparam("pike", "reqs_density_per_unit", 200) modparam("pike", "remove_latency", 7200) Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 1

[Kamailio-Users] Interfaceing Presence server with Presence Network Aget(PNA)
by suresh tattikota 14 Oct '08

14 Oct '08

hi all I am trying to develop the interface between PNA and PS,that is PEN interface, for that i am using the one openser 1.2 as the PNA Application server and Presence server. I already developed sh interface between HSS and the PNA. After getting the PNR request from HSS,PNA has to send the PUBLISH message to PS so here my doubt is 1.How to connect the PNA and PS 2.how to construct PUBLISH XML body 3.can i use the opensips/openser as presence network agent thank you regards suresh

2 1

Re: [Kamailio-Users] what is the best solution NAT problem
by Ricky Gutierrez 14 Oct '08

14 Oct '08

Sunday, October 12, 2008 3:43:16 PM Iñaki Baz Castillo wrote: OpenSer behind a NAT router with dynamic IP? It's really an annoying scenario. Note that when the INVITE/200/(ACK) comes from the LAN 192.168.10.X and goes through OpenSer, you must replace the media IP in the SDP and the IP in "Contact" with the **public** IP of the LAN router, that is a dynamic IP and AFAIK it's not valid to set a domain in the SDP. And when he INVITE/200/(ACK) comes from Internet and goes through OpenSer, you must replace the media IP in the SDP and IP in "Contact" with the eth1 IP of RtpProxy (19.168.10.1). Iñaki I am replacing it for the private ip that I obtain of my router adsl 192.168.1.64, in some examples I read that they replace for the ip it public.. what you try to say is that it replaces the ip that comes from the router for the ip of the lan Also, you need the SIP ports and RtpProxy media ports redirected in the router to the RtpProxy server. that already has it configured in my router ... Anyway, this scenario is not appropiate for a businnes service. you are right, I have a wall of NAT I have open the ports UDP 5060:5065, 10000:20000, 35000:65000 TCP: 5060 Why do you open all these ports? and what do you mean with "open"? don't you mean "redirected"? excuse me, I meant that the ports this redirect to my server PD: A suggestion: Buy an space in a datacenter (a virtual machine could be enough depending on your traffic ammount) and install the OpenSer and RtpProxy decently in a host with public IP. ;) best regards .. rickygm

2 1

[Kamailio-Users] presence module, events
by toly 14 Oct '08

14 Oct '08

Greetings, This is a question to the author: bogdan_iancu, but if anybody can answer, I'd appreciate it. Softphone client is Bria. Here is publish event from ngrep PUBLISH sip:tolyp@webley SIP/2.0. Via: SIP/2.0/TCP 10.50.1.101:45464;branch=z9hG4bK- T 2008/10/13 16:15:38.621949 10.50.1.101:1572 -> 10.50.4.40:5070 [AP] d8754z-697cf1172b7bf515-1---d8754z-;rport. Max-Forwards: 70. Contact: <sip:tolyp@10.50.1.101:1572;transport=TCP>. To: "tolyp"<sip:tolyp@webley>. From: "tolyp"<sip:tolyp@webley>;tag=ca65f37d. Call-ID: YmFlNjUxMzEwM2Q4Nzk0MWRhZGUxZmQwMDA5NmE4Nzc.. CSeq: 1 PUBLISH. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/pidf+xml. User-Agent: Bria release 2.4 stamp 49274. Event: presence. Content-Length: 441. . <?xml version='1.0' encoding='UTF-8'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' xmlns:lt='urn:ietf:params:xml:ns:location-type' entity='sip:tolyp@webley'><tuple id='t5a055764'><status><basic>open</basic></status></tuple><dm:person id='p5e347748'><dm:note>Available</dm:note></dm:person></presence> Here is the code from parser/parse_event.c if ((_e->text.len == PRES_STR_LEN) && !strncasecmp(PRES_STR, tmp.s, _e->text.len)) { _e->parsed = EVENT_PRESENCE; } else if ((_e->text.len == PRES_XCAP_CHANGE_STR_LEN) && !strncasecmp(PRES_XCAP_CHANGE_STR, tmp.s, _e->text.len)) { _e->parsed = EVENT_XCAP_CHANGE; } else if ((_e->text.len == PRES_WINFO_STR_LEN) && !strncasecmp(PRES_WINFO_STR, tmp.s, _e->text.len)) { _e->parsed = EVENT_PRESENCE_WINFO; } else if ((_e->text.len == PRES_SIP_PROFILE_STR_LEN) && !strncasecmp(PRES_SIP_PROFILE_STR, tmp.s, _e->text.len)) { _e->parsed = EVENT_SIP_PROFILE; } else if ((_e->text.len == DIALOG_STR_LEN) && !strncasecmp(DIALOG_STR, tmp.s, _e->text.len)) { _e->parsed = EVENT_DIALOG; } else if ((_e->text.len == MWI_STR_LEN) && !strncasecmp(MWI_STR, tmp.s, _e->text.len)) { _e->parsed = EVENT_MWI; } else { _e->parsed = EVENT_OTHER; } if( (*end)== ';') { str params_str; params_str.s= end+1; end= skip_token(params_str.s, tmp.len- _e->text.len- 1); params_str.len= end- params_str.s; if (parse_params(&params_str, CLASS_ANY, &phooks, &_e->params)<0) return -1; } else _e->params= NULL; It detects presence event correctly, but then if( (*end)== ';') does not find any semicolon and here is the result: 10/13 16:34:57 13030 err ERROR:presence:handle_publish: <publish.c: 508> Missing or unsupported event header field value 10/13 16:34:57 13030 err ERROR:presence:handle_publish: <publish.c: 511> event=[presence] What semicolon it is looking? If I do (*end)== ' ') it still can't parse. Is Bria message has incorrect format? What was the format of message the parser was written for? Thanks a lot, Toly -- View this message in context: http://www.nabble.com/presence-module%2C-events-tp19963494p19963494.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 2

[Kamailio-Users] How to handle DoS attack and OpenSER Pike module!
by Arif-Uz-Zaman 14 Oct '08

14 Oct '08

Hi all, I need to bother about crazy client by considering "Flood" detection technique. I can do it by using OpenSER Pike <http://kamailio.org/docs/modules/1.2.x/pike.html> module which helps to keep trace of all (or selected ones) incoming request's IP source and blocks the ones that exceeded some limit. In my case: If the number of SIP messages from a single IP address to my SIP Proxy exceeds 200 per minute. Recommended action: Block IP for 2 hours. I tried with the pike module but I'm little bit confused with sampling, density, and timeout value. Please help me with example configuration by considering my point. Thanks, ARIF

1 0

[Serusers] ser-0.9.6: modules/postgres bugreport & patch
by Alexandre Snarskii 13 Oct '08

13 Oct '08

Hi! I'm running ser-0.9.6, on FreeBSD 6.1-stable, database backend is postgresql version 8.1.3. Today I got errors in logfiles, saying: messages.2.bz2:May 29 14:35:03 <XXX> /usr/local/sbin/ser[51448]: ERROR:avpops:dbrow2avp: dbrow contains NULL fields The similar problem reported in: http://lists.iptel.org/pipermail/serusers/2005-May/019681.html with much more detailed description of error, database contents and config samples. Patch is trivial, and looks more like a fix to copy'n'paste error: in mysql/val.c function str2val states: if (!_s) { memset(_v, 0, sizeof(db_val_t)); VAL_TYPE(_v) = _t; VAL_NULL(_v) = 1; return 0; } VAL_NULL(_v) = 0; and the last line mentions that 'well, that's value is not NULL'. In postgresql/db_val.c, line 182, function str2valp, the same statement is the: if (!_s) { DLOG("str2valp", "got a null value"); VAL_TYPE(_v) = _t; VAL_NULL(_v) = 1; return 0; } without explicit notification that this is not-NULL value. More than, nowhere else in this function VAL_NULL(_v) not set to 0. So, if a value _v.val contained anyting but 0 at the function start, resulting value will be threated as NULL despite the fact, that _s is NOT NULL. Patch is obvious, just add VAL_NULL(_v)=0; after cited block (line 188) and everyting will be ok. At least for me it's ok for some hours :)

2 3

[Serusers] installing CDRTool on Freebsd
by Mohit Saxena 13 Oct '08

13 Oct '08

Hello Friends, Can any one help me installing the CDR TOOL to integrate with freeradius on freebsd? Any help will be highly appreciated. Thanks Mohit Mohit C. Saxena Sr. Network Administrator Starcomms Plc. www.starcomms.com <blocked::http://www.starcomms.com> Mob: +234-702-800-0709 DISCLAIMER: The information contained in this message (including any attachments) is confidential and may be privileged. If you have received it by mistake please notify the sender by return e-mail and permanently delete this message and any attachments from your system. Any form of dissemination, use, review, distribution, printing or copying of this message in whole or in part is strictly prohibited if you are not the intended recipient of this e-mail. Please note that e-mails are susceptible to change. STARCOMMS PLC shall not be liable for the improper or incomplete transmission of the information contained in this communication nor for any delay in its receipt or damage to your system. STARCOMMS PLC does not guarantee that the integrity of this communication has been maintained or that this communication is free of viruses, interceptions or interferences. STARCOMMS PLC reserves the right to monitor all e-mail communications, whether related to the business of STARCOMMS or not, through its internal or external networks.

5 7

[Kamailio-Users] fr_anything_at_all_timer.
by Alex Balashov 13 Oct '08

13 Oct '08

It would be really nice to have a feature in TM for even faster failover that is turned off when any reply is received at all, provisional or otherwise. The problem with fr_timer and fr_inv_timer is that they throw on lack of final reply. But sometimes gateways have very lengthy, complicated asynchronous operations that must complete before they can return a final OK. If you are doing failover, the most important thing to know is that the gateway is there at all, which the provisional 1xx message accomplishes. What would be really nice is to have a timer that can be diffused as soon as a provisional reply arrives. I don't see how this can be accomplished with any of the existing timers. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

4 8

[Serusers] CDRTool error Table 'cdrtool.radacct200810' doesn't exist
by satish patel 13 Oct '08

13 Oct '08

Hi I have installed CDRTool and i got error on CDR search entry like this Table 'cdrtool.radacct200810' doesn't exist i have read on goolge about it and people tell the problem was if radacct200810 table not in radius database. i have also run rotatetable.php script to rotate database but i got error like this #php /var/www/CDRTool/scripts/SER/rotateTables.php rotateTable(radacct200810,200809,radacct200809) Error: cannot locate mysql creation file i am using 6.6.10 kindly suggest me.. $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/

1 0

[Kamailio-Users] problem with relaying invite
by Asim Riaz 13 Oct '08

13 Oct '08

Hi, I am relaying invites from Kamilio(1.2) to a third proxy, i am receivng Server error occured in SIP messages, I am seeing these messages in log file; Asterisk--------->Kamilio----------------->third-proxy route{ openser.cfg if(method=="INVITE"){ route(4); exit; } route[4] { if(!t_relay("udp:third_proxy_1:5060") || !t_relay("udp:third_proxy_2:5060")) { end_media_session(); sl_reply_error(); } } DEBUG: RFC3261 transaction matching failed DEBUG: t_lookup_request: no transaction found ERROR:tm:t_forward_nonack: no branch for forwarding ERROR:tm:w_t_relay: t_forward_nonack failed DEBUG:sl:sl_reply_error: error text is Server error occurred (1/SL) any idea ? Thanks Asim

3 7

[Kamailio-Users] what is the best solution NAT problem
by Ricky Gutierrez 13 Oct '08

13 Oct '08

Hi list, I am trying to solve my nat problem with openser 1.3.2 and rtpproxy 1.1, I have my openser with 2 net cards. I detail my scenario: (192.168.10.1) LAN-eth1- Server Openser eth0-WAN (192.168.1.64)<-> NAT <-> ADSL dyndns <-> Internet <-> ADSL <-> NAT <-> UAC All my external clients are also behind an ADSL with address private ip and my server openser, I don't have ip it public but register a domain with dyndns and it configures it in my router adsl, I have access from out to my server through dyndns. All my external clients configure them so that they use of address proxy the dyndns domain, my external clients register but when they call to another UAC that this behind the server openser is not audio, and if a UAC that this behind the server openser calls an external client the call doesn't arrive he fails. I have open the ports UDP 5060:5065, 10000:20000, 35000:65000 TCP: 5060 some idea of like I can solve this problem?... best regards all list rickygm ### LOG SIP ### U +1.744617 192.168.10.30:5062 -> 192.168.10.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK2fc9.6c704556.0 Via: SIP/2.0/UDP 192.168.0.60:5063;rport=5063;received=190.184.35.4;branch=z9hG4bK-de71d299 Record-Route: <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3> Record-Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3> From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3 To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978 Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 214 v=0 o=113 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.10.30 t=0 0 m=audio 5004 RTP/AVP 18 101 a=sendrecv a=rtpmap:18 G729/8000 a=ptime:30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.000481 192.168.1.64:5060 -> 190.184.35.4:5063 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.60:5063;rport=5063;received=190.184.35.4;branch=z9hG4bK-de71d299 Record-Route: <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3> Record-Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3> From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3 To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978 Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 232 P-hint: onreply_route|force_rtp_proxy v=0 o=113 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.1.64 t=0 0 m=audio 35006 RTP/AVP 18 101 a=sendrecv a=rtpmap:18 G729/8000 a=ptime:30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=nortpproxy:yes # U +0.219110 190.184.35.4:5063 -> 192.168.1.64:5060 ACK sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.60:5063;branch=z9hG4bK-547056da From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3 To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978 Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60 CSeq: 102 ACK Max-Forwards: 70 Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3>, <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3> Proxy-Authorization: Digest username="122",realm="gnuforever.homelinux.com",nonce="48f260b8529fd8e9d8f0247fa92f734a317f2da5",uri="sip:113@gnuforever.homelinux.com",algorithm=MD5,response="ec45ed24126b924160da73b8ba10e73d" Contact: <sip:122@192.168.0.60:5063> User-Agent: Linksys/SPA942-5.2.8 Content-Length: 0 # U +0.000944 192.168.10.1:5060 -> 192.168.10.30:5062 ACK sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Record-Route: <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3> Record-Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK2fc9.6c704556.2 Via: SIP/2.0/UDP 192.168.0.60:5063;rport=5063;received=190.184.35.4;branch=z9hG4bK-547056da From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3 To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978 Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60 CSeq: 102 ACK Max-Forwards: 69 Proxy-Authorization: Digest username="122",realm="gnuforever.homelinux.com",nonce="48f260b8529fd8e9d8f0247fa92f734a317f2da5",uri="sip:113@gnuforever.homelinux.com",algorithm=MD5,response="ec45ed24126b924160da73b8ba10e73d" Contact: <sip:122@190.184.35.4:5063;nat=yes> User-Agent: Linksys/SPA942-5.2.8 Content-Length: 0 # U +0.140447 192.168.1.64:5060 -> 190.184.35.4:5063 OPTIONS sip:190.184.35.4:5063 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.64:5060;branch=0 From: sip:pinger@192.168.1.64;tag=f7508b75 To: sip:190.184.35.4:5063 Call-ID: 29b7fe71-06901b84-fa6(a)192.168.1.64 CSeq: 1 OPTIONS Content-Length: 0 # U +0.164881 190.184.35.4:5063 -> 192.168.1.64:5060 SIP/2.0 404 Not Found To: sip:190.184.35.4:5063;tag=a04ea06caeb3256i3 From: sip:pinger@192.168.1.64;tag=f7508b75 Call-ID: 29b7fe71-06901b84-fa6(a)192.168.1.64 CSeq: 1 OPTIONS Via: SIP/2.0/UDP 192.168.1.64:5060;branch=0 Server: Linksys/SPA942-5.2.8 Content-Length: 0 # U +1.195009 192.168.10.28:5060 -> 192.168.10.1:5060 # U +0.639899 192.168.10.1:5060 -> 192.168.10.27:5060 OPTIONS sip:192.168.10.27:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1:5060;branch=0 From: sip:pinger@192.168.1.64;tag=08508b75 To: sip:192.168.10.27:5060 Call-ID: 29b7fe71-16901b84-1b6(a)192.168.10.1 CSeq: 1 OPTIONS Content-Length: 0 # U +0.001412 192.168.10.27:5060 -> 192.168.10.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1:5060;branch=0 From: sip:pinger@192.168.1.64;tag=08508b75 To: sip:192.168.10.27:5060;tag=b1e56101b3b09b53 Call-ID: 29b7fe71-16901b84-1b6(a)192.168.10.1 CSeq: 1 OPTIONS User-Agent: Grandstream GXV3000 1.1.3.14 Contact: <sip:120@192.168.10.27:5060> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer, 100rel, path Content-Length: 0 # U +3.393457 190.184.35.4:5063 -> 192.168.1.64:5060 BYE sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.60:5063;branch=z9hG4bK-f7539b7 From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3 To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978 Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60 CSeq: 103 BYE Max-Forwards: 70 Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3>, <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3> Proxy-Authorization: Digest username="122",realm="gnuforever.homelinux.com",nonce="48f260b8529fd8e9d8f0247fa92f734a317f2da5",uri="sip:113@192.168.10.30:5062",algorithm=MD5,response="4416adbfacebc76ed9cc3f002ff958a1" User-Agent: Linksys/SPA942-5.2.8 Content-Length: 0 # U +0.000768 192.168.10.1:5060 -> 192.168.10.30:5062 BYE sip:113@192.168.10.30:5062;transport=udp SIP/2.0 Record-Route: <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3> Record-Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK3fc9.1cba8107.0 Via: SIP/2.0/UDP 192.168.0.60:5063;rport=5063;received=190.184.35.4;branch=z9hG4bK-f7539b7 From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3 To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978 Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60 CSeq: 103 BYE Max-Forwards: 69 Proxy-Authorization: Digest username="122",realm="gnuforever.homelinux.com",nonce="48f260b8529fd8e9d8f0247fa92f734a317f2da5",uri="sip:113@192.168.10.30:5062",algorithm=MD5,response="4416adbfacebc76ed9cc3f002ff958a1" User-Agent: Linksys/SPA942-5.2.8 Content-Length: 0 # U +0.055972 192.168.10.30:5062 -> 192.168.10.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK3fc9.1cba8107.0 Via: SIP/2.0/UDP 192.168.0.60:5063;rport=5063;received=190.184.35.4;branch=z9hG4bK-f7539b7 Record-Route: <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3> Record-Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3> From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3 To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978 Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60 CSeq: 103 BYE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0

3 3

Re: [Kamailio-Users] No audio both ways
by luzango mfupe 13 Oct '08

13 Oct '08

Hi Castillo,I did post my config file showing how i used the Mediaproxy. However i would like to get your opinion on how to use/configure it properly. cheers, Lu. > Message: 3 > Date: Sun, 12 Oct 2008 23:45:35 +0200 > From: I?aki Baz Castillo <ibc(a)aliax.net> > Subject: Re: [Kamailio-Users] No audio both ways > To: users(a)lists.kamailio.org > Message-ID: <200810122345.35371.ibc(a)aliax.net> > Content-Type: text/plain; charset="utf-8" > > El Domingo, 12 de Octubre de 2008, luzango mfupe escribi?: > > I solved the problem by disabling use_mediaproxy() call and did put > nat=yes > > in the Asterisk side. > > This means that you didn't use MediaProxy correctly. If it's properly > configured and the proxy logic script calls use_mediaproxy() when needed, > then you don't need to enable the Comedia mode in Asterisk (nat=yes). > > -- > I?aki Baz Castillo > > > > ------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > End of Users Digest, Vol 41, Issue 33 > ************************************* > -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

2 1

Re: [Kamailio-Users] No audio both ways
by luzango mfupe 12 Oct '08

12 Oct '08

Hi mates, I solved the problem by disabling use_mediaproxy() call and did put nat=yesin the Asterisk side. Many thanks to Daniel and David. Cheers, Lu. -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

2 1

[Kamailio-Users] lcr module prefix_mode=1
by jh＠tutpro.com 12 Oct '08

12 Oct '08

by accident i sent this message to devel list when i intended to send it to users list. i'm thinking to speed up lcr load_gws() implementation by storing prefixes into hash table. it would allow load_gws() to execute (if necessary) in constant time independent of the number of prefixes. also, i'm planing to make the size of hash table controllable by module parameter. the side effect of this is that prefix_mode=1, where prefixes are regular expressions, would have to be dropped, because hashing on regular expressions makes no sense. something close to prefix_mode=1 could be achieved using dialplan module to store the regular expressions and return as attribute a gateway grp_id. that could then be used by new load_gws_from_group(grp_id) to load gws in the given group. any comments on this? is someone depending on prefix_mode=1? by the way, before starting this project, i took a look at opensips drouting module, but unfortunately it does not cover many features of lcr module that i use in my configurations and thus cannot be used to replace lcr module. -- juha

2 3

[Kamailio-Users] How can I validate a field HF of the Request with BD?
by Marcio 10 Oct '08

10 Oct '08

Hi all, I would like to know, how can i validate a field-hf(realm) of the request SIP, in case Register, with a column(realm/domain) of the table, which was created by me, in the mysql... will be that by avp ? or i will have of the create a module, for this...? Any advices welcome! cheers, Marcio

2 1

[Serusers] compiltation error when compiling avpops module in latest SER version from CVS
by Deepak 10 Oct '08

10 Oct '08

Hi, I am getting an error compiling avpops: make[1]: Entering directory `/opt/NEWSER/sip_router/modules/avpops' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -minline-all-stringops -falign-loops -ftree-vectorize -mtune=athlon64 -Wall -DNAME='"ser"' -DVERSION='"2.1.0-dev23-make"' -DARCH='"i386"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' -D__CPU_i386 -D__OS_linux -DSER_VER=2001000 -DCFG_DIR='"/usr/local/etc/ser/"' -DMODS_DIR='"/usr/local/lib/ser/modules/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -c avpops.c -o avpops.o In file included from avpops_impl.h:41, from avpops_parse.h:38, from avpops.c:50: avpops_db.h:66: error: expected â)â before â*â token make[1]: *** [avpops.o] Error 1 make[1]: Leaving directory `/opt/NEWSER/sip_router/modules/avpops' make: *** [modules] Error 1 Any suggestions? Thanks

1 0

[Serusers] serctl moni problems
by psycho psycho 10 Oct '08

10 Oct '08

I haven' noticed but the whole error finally message by executing : sudo serctl moni is: [: 1074: ==: unexpected operator [: 1074: 0: unexpected operator _________________________________________________________________ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+world&mkt=en-US&form=QBRE

1 0

[Kamailio-Users] kamailio "failed to generate nonce"
by mayamatakeshi 10 Oct '08

10 Oct '08

Hello, I'm testing with kamailio 1.4 (just got revision 5050). I'm stressing it using a sipp scenario to verify which population of subscribers kamailio can handle. Using openser 1.3.3 I could easily keep thousands of subscribers online even when running on a VirtualMachine hosted on a desktop computer (last time I tested this it was about 8000 subscribers). But I'm running kamailio on a production machine and when I reach around 700 subscribers, kamailio starts to log this: Oct 9 17:44:20 ipx029 /usr/local/sbin/kamailio[7689]: ERROR:auth:build_auth_hf: no more nonces can be generated Oct 9 17:44:20 ipx029 /usr/local/sbin/kamailio[7689]: ERROR:auth:challenge: failed to generate nonce And the registration fails. Is anybody else seeing this? regards, takeshi

2 3

[Kamailio-Users] No audio both ways.
by luzango mfupe 10 Oct '08

10 Oct '08

Hi mates,I have this setup: Xlite---->Openser---->Asterisk------>VoIP to PTSN Provider I use kamailio 1.3.3(port5060), MediaProxy and Asterisk 1.4 (port 5065) on the same Debian Box in Realtime with no NAT. Asterisk connects calls to the VoIP to PSTN provider. I am able to establish calls towards the PSTN side(Landline & Mobiles) but with no audio. I can hear the ringing tone but when the call connects and the conversation begin i hear nothing so as the Callee side. Below are my configs,the ngrep captured packets and codecs. #################################################################################### route[4] { # routing to the public network rewritehostport("xx.xxx.xxx.xx:5065"); t_on_failure("2"); if (!t_relay()) { sl_reply_error(); }; exit; } route[6] { # # -- NAT handling -- # if (isbflagset(6) || isbflagset(7)) { append_hf("P-hint: Route[6]: mediaproxy \r\n"); use_media_proxy(); }; } route[10] { #from an internal domain -> inbound #Native SIP destinations are handled using the location table #Gateway destinations are handled by regular expressions append_hf("P-hint: inbound->inbound \r\n"); if (uri=~"^sip:0[1-9][0-9]+@.*") { if (is_user_in("credentials","local")) { strip(1); prefix("27"); route(6); route(4); exit; } else { sl_send_reply("403", "No permissions for local calls"); exit; }; }; if (uri=~"^sip:00[1-9][0-9]+@.*") { if (is_user_in("credentials","int")) { strip(2); route(6); route(4); exit; } else { sl_send_reply("403", "No permissions for international calls"); }; }; ################################################################################### This call was from the xlite softphone 1974 towards the Landline 0123825710. ################################################################################### U 2008/12/06 03:38:43.896057 196.212.209.18:46738 -> kamailio IP:5060 INVITE sip:0123825710@KAMAILIO ip SIP/2.0..Via: SIP/2.0/UDP 192.168.0.55: 46738;branch=z9hG4bK-d8754z-2d5e73428b95d262-1---d8754z-;rport..Max-Forward s: 70..Contact: <sip:1974@196.212.209.18:46738>..To: "0123825710"<sip:01238 25710@kamailio IP>..From: <sip:1974@kamailio IP>;tag=353dd217..Call-ID: MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 1 INVITE..Allow: INVIT E, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Cont ent-Type: application/sdp..User-Agent: X-Lite release 1014k stamp 47051..Co ntent-Length: 237....v=0..o=- 1 2 IN IP4 192.168.0.55..s=CounterPath X-Lite 3.0..c=IN IP4 192.168.0.55..t=0 0..m=audio 60782 RTP/AVP 0 8 3 101..a=fmtp :101 0-15..a=rtpmap:101 telephone-event/8000..a=alt:1 1 : ljiQYRpD NiMZsfdZ 192.168.0.55 60782..a=sendrecv.. U 2008/12/06 03:38:43.896350 Kamailio IP:5060 -> 196.212.209.18:46738 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 192.168.0.55:46 738;branch=z9hG4bK-d8754z-2d5e73428b95d262-1---d8754z-;rport=46738;received =196.212.209.18..To: "0123825710"<sip:0123825710@kamailio IP>;tag=329cfea a6ded039da25ff8cbb8668bd2.dcfe..From: <sip:1974@kamailio IP>;tag=353dd217 ..Call-ID: MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 1 INVITE..Pr oxy-Authenticate: Digest realm="41.208.212.97", nonce="4939d8cfeb060ab14354 85eee811cdf644f759a2"..Content-Length: 0.... U 2008/12/06 03:38:44.086313 196.212.209.18:46738 -> kamailio IP:5060 ACK sip:0123825710@kamailio IP SIP/2.0..Via: SIP/2.0/UDP 192.168.0.55:467 38;branch=z9hG4bK-d8754z-2d5e73428b95d262-1---d8754z-;rport..To: "012382571 0"<sip:0123825710@kamailio IP>;tag=329cfeaa6ded039da25ff8cbb8668bd2.dcfe. .From: <sip:1974@41.208.212.97 <sip%3A1974(a)41.208.212.97>>;tag=353dd217..Call-ID: MGRiNzM0ZGYxZTk1ZDI3 ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 1 ACK..Content-Length: 0.... U 2008/12/06 03:38:44.582208 asterisk IP:5065 -> kamailio IP:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP kamailio IP;branch=z9hG4bKd78.7bd576 24.0;received=kamailio IP..Via: SIP/2.0/UDP 192.168.0.55:46738;received=1 96.212.209.18 ;branch=z9hG4bK-d8754z-2aedec27a8ddd96c-1---d8754z-;rport=4673 8..Record-Route: <sip:kamailio IP;lr=on;ftag=353dd217;nat=yes>..From: <si p:1974@kamailio IP>;tag=353dd217..To: "0123825710"<sip:0123825710@41.208. 212.97>..Call-ID: MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 2 INV ITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, RE FER, SUBSCRIBE, NOTIFY..Supported: replaces..Contact: <sip:27123825710@41.2 08.212.97:5065>..Content-Length: 0.... U 2008/12/06 03:38:44.711225 70.42.72.49:5060 -> Asterisk IP:5065 SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP Asterisk IP:5065;branch=z9hG4b K22abec12;rport=5065..From: "1974" <sip:1974@Asterisk IP:5065>;tag=as6a7c b89f..To: <sip:1214650027123825710@70.42.72.49<sip%3A1214650027123825710(a)70.42.72.49>>..Call-ID: 1934f5d443abffe07 c59d0a42215b49c@41.208.212.97..CSeq: 102 INVITE..Server: OpenSER (1.3.2-not ls (i386/solaris))..Content-Length: 0.... U 2008/12/06 03:38:47.206445 70.42.72.49:5060 -> Asterisk IP:5065 SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP Asterisk IP:5065;branch=z9 hG4bK22abec12;rport=5065..From: "1974" <sip:1974@Asterisk IP:5065>;tag=as 6a7cb89f..To: <sip:1214650027123825710@70.42.72.49<sip%3A1214650027123825710(a)70.42.72.49> >;tag=cba-1a6e-48ecbab6.. Call-ID: 1934f5d443abffe07c59d0a42215b49c@Asterisk IP..CSeq: 102 INVITE.. Contact: <sip:1214650027123825710@70.42.72.138<sip%3A1214650027123825710(a)70.42.72.138>>..Date: Wed, 08 Oct 2008 13: 50:49 GMT..Server: BRSIP v2.0.1.2..Record-Route: <sip:70.42.72.49 ;lr=on;fta g=as6a7cb89f>..Content-Type: application/sdp..Content-Length: 212....v=0..o =BRSDP 792898 792898 IN IP4 216.49.201.22..s=BRSDP Session..c=IN IP4 216.49 .201.22..t=0 0..m=audio 27852 RTP/AVP 0 101..a=ptime:20..a=rtpmap:0 PCMU/80 00..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15.. U 2008/12/06 03:38:47.206891 Asterisk IP:5065 -> Kamailio IP:5060 SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP kamailio IP;branch=z9hG4bK d78.7bd57624.0;received=kamailio IP..Via: SIP/2.0/UDP 192.168.0.55:46738; received=196.212.209.18 ;branch=z9hG4bK-d8754z-2aedec27a8ddd96c-1---d8754z-; rport=46738..Record-Route: <sip:41.208.212.97 ;lr=on;ftag=353dd217;nat=yes>. .From: <sip:1974@kamilio IP>;tag=353dd217..To: "0123825710"<sip:01238257 10(a)41.208.212.97>;tag=as4377a96d..Call-ID: MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVl NTQ0NTUwZjg...CSeq: 2 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported: replaces..Conta ct: <sip:27123825710@Asterisk IP:5065>..Content-Type: application/sdp..Co ntent-Length: 287....v=0..o=root 2664 2664 IN IP4 41.208.212.97..s=session. .c=IN IP4 41.208.212.97..t=0 0..m=audio 19202 RTP/AVP 8 0 3 101..a=rtpmap:8 PCMA/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telepho ne-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=se ndrecv.. U 2008/12/06 03:38:47.207106 Kamailio IP:5060 -> 196.212.209.18:46738 SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP 192.168.0.55:46738 ;received= 196.212.209.18 ;branch=z9hG4bK-d8754z-2aedec27a8ddd96c-1---d8754z-;rport=467 38..Record-Route: <sip:kamailio IP;lr=on;ftag=353dd217;nat=yes>..From: <s ip:1974@Kamailio IP>;tag=353dd217..To: "0123825710"<sip:0123825710@kamilioIP>;tag=as4377a96d..Call-ID: MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZ jg...CSeq: 2 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported: replaces..Contact: <sip: 27123825710@41.208.212.97:5065>..Content-Type: application/sdp..Content-Len gth: 287....v=0..o=root 2664 2664 IN IP4 Asterisk IP..s=session..c=IN IP4 41.208.212.97..t=0 0..m=audio 19202 RTP/AVP 8 0 3 101..a=rtpmap:8 PCMA/800 0..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/ 8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv.. ######################################################################################## Sip.conf [general] context=from-trunk bindport=5065 autocreatepeer=yes bindaddr=xx.xxx.xxx.xx disallow=all ;allow=gsm ;allow=amr allow=alaw allow=ulaw allow=gsm ;allow=ilbc ;disallow=all ; ;useragent=Asterisk PBX dtmfmode = rfc2833 domain=xx.xxx.xxx.xx ; Add IP address as local domain [Provider] disallow=all canreinvite=no context=from-trunk allow=all ;allow=ulaw ;allow=gsm host=xx.xx.xx.xx insecure=port,invite type=peer ; we only want to call out, not be call$ dtmfmode=rfc2833 ######################################################################################### Here is my codecs 41*CLI> core show translation Translation times between formats (in milliseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 amr g723 - - - - - - - - - - - - - - gsm - - 2 2 2 2 1 5 - 19 - 2 - 14 ulaw - 5 - 1 2 2 1 5 - 19 - 2 - 14 alaw - 5 1 - 2 2 1 5 - 19 - 2 - 14 g726aal2 - 5 2 2 - 2 1 5 - 19 - 1 - 14 adpcm - 5 2 2 2 - 1 5 - 19 - 2 - 14 slin - 4 1 1 1 1 - 4 - 18 - 1 - 13 lpc10 - 6 3 3 3 3 2 - - 20 - 3 - 15 g729 - - - - - - - - - - - - - - speex - 6 3 3 3 3 2 6 - - - 3 - 15 ilbc - - - - - - - - - - - - - - g726 - 5 2 2 1 2 1 5 - 19 - - - 14 g722 - - - - - - - - - - - - - - amr - 6 3 3 3 3 2 6 - 20 - 3 - - With best regards, Lu. -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

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[Kamailio-Users] Register Kamailio with my service provider
by ingdavidcespedes＠cable.net.co 10 Oct '08

10 Oct '08

Hi! I'm trying to register my Sip Proxy Server (Kamailio) to my service provider, but I have to do it sending my user name and password. Right now I'm trying to make this work with Asterisk acting as a Gateway to my SIP SP, but there most be or should be a way to do it with kamailio, because based in SIP Connect Technical Recommendations the SPS of the client should register directly to the SPS of the Service Provider, without using any gateway. I know there is a way to make it with the UAC module, but only if the service provider match an IP address, not with user name and password. with best regards David Cespedes.

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[Serusers] How can I keep SER from changing reply Response Code from 503 to 500?
by Frank Durda IV 09 Oct '08

09 Oct '08

I have a situation where SER is gumming up the works. The problem is that I have a call where the INVITE fails because a PSTN gateway hasn't authorized the SIP invitee to call a given number, as intended. The PSTN switch is returning a "503 Service Unavailable" as is expected, but SER is then sending the SIP invitee a "500 Service Unavailable" instead of the original message, which is wrong in several ways. Here is the ngrep of what SER received from the PSTN switch (which is the desired answer in this situation): # U 2008/10/09 18:08:45.045015 10.133.0.2:5060 -> 10.133.90.6:5060 SIP/2.0 503 Service Unavailable. Call-ID: 14407805-3432564525-49719(a)sbc1.sometelco.com. CSeq: 1 INVITE. From: <sip:8019995678@209.x.x.x>;tag=3432564525-49731. To: <sip:9179891234@208.x.x.x:5060;user=phone>;tag=000a0285+1+4bae000a+de6c6831. Via: SIP/2.0/UDP 10.133.90.6;branch=z9hG4bKaa0f.96addb37.0. Via: SIP/2.0/UDP 209.x.x.x:5060;branch=z9hG4bKafec84c056d5d9ac97d27cc12d3dbb92. Server: DC-SIP/2.0. Content-Length: 0. Retry-After: 3. . That response code (503 Service Unavailable) is what should be sent back to the call originator, who can cope with it and redirect the call. However, here is the ngrep of what SER transmitted on to the invitee instead: # U 2008/10/09 18:08:45.050440 10.133.90.6:5060 -> 209.x.x.x:5060 SIP/2.0 500 Service Unavailable. To: <sip:9179891234@208.x.x.x:5060;user=phone>;tag=b11e68d9aa35b34c54f995457fe194f9-0069. From: <sip:8019995678@209.x.x.x>;tag=3432564525-49731. Call-ID: 14407805-3432564525-49719(a)sbc1.sometelco.com. CSeq: 1 INVITE. Via: SIP/2.0/UDP 209.x.x.x:5060;branch=z9hG4bKafec84c056d5d9ac97d27cc12d3dbb92. Server: Sip EXpress router (2.0.0-rc1 (x86_64/freebsd)). Content-Length: 0. . And the calling party receives the confusing "503 Service Unavailable". The onreply_route routine in use is a strip-down of what is found in the PSTN sample (page 54 of the SER Getting Started manual), and it makes no attempt to alter response codes: onreply_route["PSTN_REPLY"] { xlog("L_ERR","In onreply_route %rs %rr"); force_send_socket("10.133.90.6"); #emit on em1 fix_nated_contact("208.x.x.x"); /*Fixes Contact:*/ # Apply RTP proxy if necessary, but only for INVITE transactions # and 183 or 2xx replies if (@cseq.method != "INVITE") return; xlog("L_ERR","onreply_route[INVITE_REPLY] R1 %rs %rr"); if ((status =~ "(183)|2[0-9][0-9]") && search("^(Content-Type|c):.*application/sdp")) { xlog("L_ERR","In onreply_route FIXING PROXY"); force_rtp_proxy("e","208.x.x.x"); } return; } From debug messages (show below), the call to force_rtp_proxy() is not being performed, so in theory this routine is left with the 503 still intact. The xlog message shows the 503 value was still there on entry to onreply_route as well as near the end of the routine: Oct 9 18:08:45 ser1 /usr/local/sbin/ser[1820]: In onreply_route 503 Service Unavailable Oct 9 18:08:45 ser1 /usr/local/sbin/ser[1820]: onreply_route[INVITE_REPLY] R1 503 Service Unavailable The "FIXING PROXY' debug message was not reached, as expected. So what is causing: 503 Service Unavailable. to mysteriously turn into 500 Service Unavailable. which isn't even the correct RFC textual response for a 500 code. This is being seen in ser-2.0.0-rc1 and is completely repeatable. Ideas? Suggestions? Thanks in advance!

1 0

[Serusers] serctl moni problems
by psycho psycho 09 Oct '08

09 Oct '08

Hi, I have a problem with the serctl utility. Even though the ser process starts without any problems, when i am trying to use the monitor by executing serctl moni i get the following error [: 1074: 0: unexpected operator. All seems to work (making call etc) but monitor. If the number 1074 points to a line from the serctl then i attach is my serctl source (i don't know if that helps): 1073 cisco_restart) 1074 if [ "$#" -ne 2 ] ; then 1075 usage 1076 exit 1 1077 fi Any suggestions.? Thank you for your time. _________________________________________________________________ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=…

1 0

[Kamailio-Users] Problem with call on hold
by Diego Zuaneti Arruda 09 Oct '08

09 Oct '08

Hi all, I'm having a problem when the user places the call on hold. When the user removes the call on hold, only one side hears the call. Does anyone know what is wrong? The dialog of the call is attached. Thanks. Diego.

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Re: [Kamailio-Users] Users Digest, Vol 41, Issue 24
by luzango mfupe 09 Oct '08

09 Oct '08

Ok David, will give feedback soon.rgds, Lu. On Thu, Oct 9, 2008 at 11:53 AM, <users-request(a)lists.kamailio.org> wrote: > Send Users mailing list submissions to > users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request(a)lists.kamailio.org > > You can reach the person managing the list at > users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. Re: No audio both ways. (David Villasmil) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 9 Oct 2008 11:07:12 +0200 > From: "David Villasmil" <david.villasmil.work(a)gmail.com> > Subject: Re: [Kamailio-Users] No audio both ways. > Cc: users(a)lists.kamailio.org > Message-ID: > <9853f4ff0810090207p11d15bddt8dca8f57370c83c9(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Try adding a: > > nat=yes > > to the kamailio/openser peer definition and test > > dvg > > On Thu, Oct 9, 2008 at 10:58 AM, luzango mfupe <luzango.mfupe(a)gmail.com > >wrote: > > > > > Hi mates,I have this setup: > > Xlite---->Openser---->Asterisk------>VoIP to PTSN Provider > > > > I use kamailio 1.3.3(port5060), MediaProxy and Asterisk 1.4 (port 5065) > on > > the same Debian Box in Realtime with no NAT. Asterisk connects calls to > the > > VoIP to PSTN provider. I am able to establish calls towards the PSTN > > side(Landline & Mobiles) but with no audio. I can hear the ringing tone > but > > when the call connects and the conversation begin i hear nothing so as > the > > Callee side. > > > > Below are my configs,the ngrep captured packets and codecs. > > > > > #################################################################################### > > route[4] { > > # routing to the public network > > rewritehostport("xx.xxx.xxx.xx:5065"); > > t_on_failure("2"); > > if (!t_relay()) { > > sl_reply_error(); > > }; > > exit; > > } > > > > route[6] { > > # > > # -- NAT handling -- > > # > > if (isbflagset(6) || isbflagset(7)) { > > append_hf("P-hint: Route[6]: mediaproxy \r\n"); > > use_media_proxy(); > > }; > > } > > > > route[10] { > > #from an internal domain -> inbound > > #Native SIP destinations are handled using the location table > > #Gateway destinations are handled by regular expressions > > append_hf("P-hint: inbound->inbound \r\n"); > > > > if (uri=~"^sip:0[1-9][0-9]+@.*") { > > if (is_user_in("credentials","local")) { > > strip(1); > > prefix("27"); > > route(6); > > route(4); > > exit; > > } else { > > sl_send_reply("403", "No permissions for local calls"); > > exit; > > }; > > }; > > > > if (uri=~"^sip:00[1-9][0-9]+@.*") { > > if (is_user_in("credentials","int")) { > > strip(2); > > route(6); > > route(4); > > exit; > > } else { > > sl_send_reply("403", "No permissions for international > calls"); > > }; > > }; > > > > > > > ################################################################################### > > This call was from the xlite softphone 1974 towards the Landline > > 0123825710. > > > > > ################################################################################### > > U 2008/12/06 03:38:43.896057 196.212.209.18:46738 -> kamailio IP:5060 > > INVITE sip:0123825710@KAMAILIO ip SIP/2.0..Via: SIP/2.0/UDP > 192.168.0.55 > > : > > > > > 46738;branch=z9hG4bK-d8754z-2d5e73428b95d262-1---d8754z-;rport..Max-Forward > > s: 70..Contact: <sip:1974@196.212.209.18:46738>..To: > > "0123825710"<sip:01238 > > 25710@kamailio IP>..From: <sip:1974@kamailioIP>;tag=353dd217..Call-ID: > > MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 1 INVITE..Allow: > > INVIT > > E, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, > > INFO..Cont > > ent-Type: application/sdp..User-Agent: X-Lite release 1014k stamp > > 47051..Co > > ntent-Length: 237....v=0..o=- 1 2 IN IP4 192.168.0.55..s=CounterPath > > X-Lite > > 3.0..c=IN IP4 192.168.0.55..t=0 0..m=audio 60782 RTP/AVP 0 8 3 > > 101..a=fmtp > > :101 0-15..a=rtpmap:101 telephone-event/8000..a=alt:1 1 : ljiQYRpD > > NiMZsfdZ > > 192.168.0.55 60782..a=sendrecv.. > > > > U 2008/12/06 03:38:43.896350 Kamailio IP:5060 -> 196.212.209.18:46738 > > SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP > > 192.168.0.55:46 > > > > > 738;branch=z9hG4bK-d8754z-2d5e73428b95d262-1---d8754z-;rport=46738;received > > =196.212.209.18..To: "0123825710"<sip:0123825710@kamailioIP > >;tag=329cfea > > a6ded039da25ff8cbb8668bd2.dcfe..From: <sip:1974@kamailioIP > >;tag=353dd217 > > ..Call-ID: MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 1 > > INVITE..Pr > > oxy-Authenticate: Digest realm="41.208.212.97", > > nonce="4939d8cfeb060ab14354 > > 85eee811cdf644f759a2"..Content-Length: 0.... > > > > U 2008/12/06 03:38:44.086313 196.212.209.18:46738 -> kamailio IP:5060 > > ACK sip:0123825710@kamailio IP SIP/2.0..Via: SIP/2.0/UDP > > 192.168.0.55:467 > > 38;branch=z9hG4bK-d8754z-2d5e73428b95d262-1---d8754z-;rport..To: > > "012382571 > > 0"<sip:0123825710@kamailioIP > >;tag=329cfeaa6ded039da25ff8cbb8668bd2.dcfe. > > .From: <sip:1974@41.208.212.97 <sip%3A1974(a)41.208.212.97> < > sip%3A1974(a)41.208.212.97 <sip%253A1974(a)41.208.212.97> > >>;tag=353dd217..Call-ID: > > MGRiNzM0ZGYxZTk1ZDI3 > > ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 1 ACK..Content-Length: 0.... > > > > > > > > U 2008/12/06 03:38:44.582208 asterisk IP:5065 -> kamailio IP:5060 > > SIP/2.0 100 Trying..Via: SIP/2.0/UDP kamailio > IP;branch=z9hG4bKd78.7bd576 > > 24.0;received=kamailio IP..Via: SIP/2.0/UDP 192.168.0.55:46738 > > ;received=1 > > 96.212.209.18 > > ;branch=z9hG4bK-d8754z-2aedec27a8ddd96c-1---d8754z-;rport=4673 > > 8..Record-Route: <sip:kamailio IP;lr=on;ftag=353dd217;nat=yes>..From: > <si > > p:1974@kamailio IP>;tag=353dd217..To: > > "0123825710"<sip:0123825710@41.208. > > 212.97>..Call-ID: MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 2 > > INV > > ITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, > BYE, > > RE > > FER, SUBSCRIBE, NOTIFY..Supported: replaces..Contact: > > <sip:27123825710@41.2 > > 08.212.97:5065>..Content-Length: 0.... > > > > U 2008/12/06 03:38:44.711225 70.42.72.49:5060 -> Asterisk IP:5065 > > SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP Asterisk > IP:5065;branch=z9hG4b > > K22abec12;rport=5065..From: "1974" <sip:1974@AsteriskIP > :5065>;tag=as6a7c > > b89f..To: <sip:1214650027123825710@70.42.72.49<sip%3A1214650027123825710(a)70.42.72.49> > <sip%3A1214650027123825710(a)70.42.72.49<sip%253A1214650027123825710(a)70.42.72.49> > >>..Call-ID: > > 1934f5d443abffe07 > > c59d0a42215b49c@41.208.212.97..CSeq: 102 INVITE..Server: OpenSER > > (1.3.2-not > > ls (i386/solaris))..Content-Length: 0.... > > > > U 2008/12/06 03:38:47.206445 70.42.72.49:5060 -> Asterisk IP:5065 > > SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP Asterisk > IP:5065;branch=z9 > > hG4bK22abec12;rport=5065..From: "1974" <sip:1974@AsteriskIP > :5065>;tag=as > > 6a7cb89f..To: <sip:1214650027123825710@70.42.72.49<sip%3A1214650027123825710(a)70.42.72.49> > <sip%3A1214650027123825710(a)70.42.72.49<sip%253A1214650027123825710(a)70.42.72.49> > > > > >;tag=cba-1a6e-48ecbab6.. > > Call-ID: 1934f5d443abffe07c59d0a42215b49c@Asterisk IP..CSeq: 102 > > INVITE.. > > Contact: <sip:1214650027123825710@70.42.72.138<sip%3A1214650027123825710(a)70.42.72.138> > <sip%3A1214650027123825710(a)70.42.72.138<sip%253A1214650027123825710(a)70.42.72.138> > >>..Date: > > Wed, 08 Oct 2008 13: > > 50:49 GMT..Server: BRSIP v2.0.1.2..Record-Route: <sip:70.42.72.49 > > ;lr=on;fta > > g=as6a7cb89f>..Content-Type: application/sdp..Content-Length: > > 212....v=0..o > > =BRSDP 792898 792898 IN IP4 216.49.201.22..s=BRSDP Session..c=IN IP4 > > 216.49 > > .201.22..t=0 0..m=audio 27852 RTP/AVP 0 101..a=ptime:20..a=rtpmap:0 > > PCMU/80 > > 00..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15.. > > > > U 2008/12/06 03:38:47.206891 Asterisk IP:5065 -> Kamailio IP:5060 > > SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP kamailio > IP;branch=z9hG4bK > > d78.7bd57624.0;received=kamailio IP..Via: SIP/2.0/UDP > 192.168.0.55:46738 > > ; > > received=196.212.209.18 > > ;branch=z9hG4bK-d8754z-2aedec27a8ddd96c-1---d8754z-; > > rport=46738..Record-Route: <sip:41.208.212.97 > > ;lr=on;ftag=353dd217;nat=yes>. > > .From: <sip:1974@kamilio IP>;tag=353dd217..To: > "0123825710"<sip:01238257 > > 10(a)41.208.212.97>;tag=as4377a96d..Call-ID: > > MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVl > > NTQ0NTUwZjg...CSeq: 2 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, > > ACK, > > CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported: > > replaces..Conta > > ct: <sip:27123825710@Asterisk IP:5065>..Content-Type: > > application/sdp..Co > > ntent-Length: 287....v=0..o=root 2664 2664 IN IP4 > > 41.208.212.97..s=session. > > .c=IN IP4 41.208.212.97..t=0 0..m=audio 19202 RTP/AVP 8 0 3 > > 101..a=rtpmap:8 > > PCMA/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 > > telepho > > ne-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - > > -..a=ptime:20..a=se > > ndrecv.. > > > > U 2008/12/06 03:38:47.207106 Kamailio IP:5060 -> 196.212.209.18:46738 > > SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP 192.168.0.55:46738 > > ;received= > > 196.212.209.18 > > ;branch=z9hG4bK-d8754z-2aedec27a8ddd96c-1---d8754z-;rport=467 > > 38..Record-Route: <sip:kamailio IP;lr=on;ftag=353dd217;nat=yes>..From: > <s > > ip:1974@Kamailio IP>;tag=353dd217..To: > > "0123825710"<sip:0123825710@kamilio IP>;tag=as4377a96d..Call-ID: > > MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZ > > jg...CSeq: 2 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, > > CANCEL, > > OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported: replaces..Contact: > > <sip: > > 27123825710@41.208.212.97:5065>..Content-Type: > > application/sdp..Content-Len > > gth: 287....v=0..o=root 2664 2664 IN IP4 Asterisk IP..s=session..c=IN > IP4 > > 41.208.212.97..t=0 0..m=audio 19202 RTP/AVP 8 0 3 101..a=rtpmap:8 > > PCMA/800 > > 0..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 > > telephone-event/ > > 8000..a=fmtp:101 0-16..a=silenceSupp:off - - - > > -..a=ptime:20..a=sendrecv.. > > > > > > > ######################################################################################## > > Sip.conf > > [general] > > context=from-trunk > > bindport=5065 > > autocreatepeer=yes > > bindaddr=xx.xxx.xxx.xx > > > > > > disallow=all > > ;allow=gsm > > ;allow=amr > > allow=alaw > > allow=ulaw > > allow=gsm > > ;allow=ilbc > > ;disallow=all > > ; > > ;useragent=Asterisk PBX > > dtmfmode = rfc2833 > > > > > > domain=xx.xxx.xxx.xx ; Add IP address as local domain > > > > [Provider] > > disallow=all > > canreinvite=no > > context=from-trunk > > allow=all > > ;allow=ulaw > > ;allow=gsm > > host=xx.xx.xx.xx > > insecure=port,invite > > type=peer ; we only want to call out, not > be > > call$ > > dtmfmode=rfc2833 > > > > > ######################################################################################### > > Here is my codecs > > > > 41*CLI> core show translation > > Translation times between formats (in milliseconds) for one > second > > of data > > Source Format (Rows) Destination Format (Columns) > > > > g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc > g726 > > g722 amr > > g723 - - - - - - - - - - - > - > > - - > > gsm - - 2 2 2 2 1 5 - 19 - > 2 > > - 14 > > ulaw - 5 - 1 2 2 1 5 - 19 - > 2 > > - 14 > > alaw - 5 1 - 2 2 1 5 - 19 - > 2 > > - 14 > > g726aal2 - 5 2 2 - 2 1 5 - 19 - > 1 > > - 14 > > adpcm - 5 2 2 2 - 1 5 - 19 - > 2 > > - 14 > > slin - 4 1 1 1 1 - 4 - 18 - > 1 > > - 13 > > lpc10 - 6 3 3 3 3 2 - - 20 - > 3 > > - 15 > > g729 - - - - - - - - - - - > - > > - - > > speex - 6 3 3 3 3 2 6 - - - > 3 > > - 15 > > ilbc - - - - - - - - - - - > - > > - - > > g726 - 5 2 2 1 2 1 5 - 19 - > - > > - 14 > > g722 - - - - - - - - - - - > - > > - - > > amr - 6 3 3 3 3 2 6 - 20 - > 3 > > - - > > > > > > > > With best regards, > > Lu. > > > > -- > > Luzango Mfupe > > TUUNE MOBILE > > Tel:0128440528/0123825710 > > Tshwane-RSA > > > > "...Ships are safe in harbor, but they were never meant to stay > > there......." > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > >

1 0

[Serusers] SIP MESSAGE BODY ENCRYPTION
by Adeyeye Oluwasegun Michael 09 Oct '08

09 Oct '08

Hi, I was wondering if the content or body of a SIP MESSAGE from a SIP client is always encrypted when a TLS-supported SIP proxy for the client registration and the likes. I know that S-MIME supported SIP stack could do that for me, but I was wondering if a TLS-supported proxy could do as well. Thanks, Michael.

2 1

[Kamailio-Users] SST module: Min-SE optional in request
by Schumann Sebastian 09 Oct '08

09 Oct '08

Dear all I might have noticed a bug in SST module: If I send an INVITE with Session-Expires: 300 Supported: timer I get rejected with 422 Session-Timer too small. If a Min-SE: x header with x > set min_se parameter included I do not get the message. This is IMHO wrong behavior, as Min-SE is optional in initial INVITE-requests sent by the phone acc. RFC 4028, table 1. Another thing that I noticed is the response to this message includes the Min-SE header (as it should), but has no linebreak after it (at least in kamailio branch 1.4). The message looks like MIN-SE:120Server: Kamailio (1.4.1-notls (i386/linux)) The phone call cannot be performed. At least one thing is IMHO wrong, as the phone cannot complete the call. It might be wrong the call gets rejected, or the answer might be possible like this, but then the phone cannot extract Min-SE due to wrong format. Any opinions on this? Sebastian

2 5

[Kamailio-Users] !!Urgent Need to setup SIP SERVER like SIP Account Provider as my own Server.Thanks for your very great Help.
by sam 09 Oct '08

09 Oct '08

HI , I want to be a SIP providers like broadvoice gimzo ,where can i get DID numbers ,do we need to have contact with telecommunication Department. Please give me a full details.When we use *kamailio *,then there is no need to get sip account and virtual numbers. -- Cheers K.S.Sampath Kumar

3 2

[Kamailio-Users] Kamailio and Asterisk trunking?
by John D 09 Oct '08

09 Oct '08

Not sure if I'm properly wording this, but I used to have two asterisk servers speaking to each other over SIP, configured via sip.conf. I now need a way to forward calls from my Asterisk instance to my Kamailio server so it can properly distribute the call. At this point however, I cannot find any documentation in order to properly have the two speak to each other. Would anyone be kind enough to point me in the right direction to establish that connection? Thanks.

3 2

[Kamailio-Users] Attended Transfer with asterisk
by ALAEDDINE abbech 08 Oct '08

08 Oct '08

Hello, Does anybody got a working configuration to make an "attended call transfer" with a call through an Asterisk gateway? Example: PSTN --> Asterisk --> SER --+-- A |<---transfer call pstn to B +-- B The call will come from the PSTN Network and will go through "A". A sets the call on "Hold" and calls "B". After A is connected with B, A hangup and B got the call from PSTN. This is not working at the moment. Attended call transfer only with OpenSER and only with SIP-Phones is no Problem. But if the is an Asterisk as PSTN-GW in the game it will not work.

3 4

[Kamailio-Users] ACK without matching transaction discarded
by Alex R.S.M 08 Oct '08

08 Oct '08

Hi, I am using openSER as a proxy for a simple SIP call (UAC <--> openSER <--> UAS). openSER receives ACK from callee but doesn't send it to caller. It seems openSER can not find a matching transaction, so it ignore and discard the message. if ( is_method("ACK") ) { if ( t_check_trans() ) { t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } Here is the message header: Internet Protocol, Src: 10.0.0.174 (10.0.0.174), Dst: 10.0.0.160 (10.0.0.160) ... ACK sip:9134750000@10.0.0.183:5080 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.174:5040;branch=z9hG4bK-12255-1-4 From: <sip:9133750000@10.0.0.174:5040>;tag=1 To: <sip:9134750000@10.0.0.183:5080>;tag=1 Call-ID: 1-12255(a)10.0.0.174 CSeq: 1 ACK Contact: sip:sipp@10.0.0.174:5040 Max-Forwards: 70 Subject: Performance Test Content-Length: 0 Am I missing anything? Thanks, Alex _________________________________________________________________ Want to do more with Windows Live? Learn “10 hidden secrets” from Jamie. http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns!5…

2 2

[Kamailio-Users] Ping both UACs when a call starts
by Cesar Troncoso 08 Oct '08

08 Oct '08

Hello! When a call starts I would like OpenSER to ping (by sending OPTIONS or INFO request) both UAC´s just to know if they are still there or for some reason they crashed or lost connectivity. In case they don't response, the call should be finished. What is the best way to do this? Thanks in advance! Cesar

2 3

[Kamailio-Users] openser 1.3.3 domain module
by toly 07 Oct '08

07 Oct '08

Greetings, In domain module, command is_domain_local - using parameter_fixup where there is a call pkg_malloc, but nowhere function free_fixup is used. This will be memory leak, or I'm missing something. Thanks for the prompt reply, Toly -- View this message in context: http://www.nabble.com/openser-1.3.3-domain-module-tp19863638p19863638.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

3 3

[Kamailio-Users] LCR to Carrierroute with DNS/SRV Support
by Krom 07 Oct '08

07 Oct '08

Hello, I am/will be a newbie to 1.3.x and beyond and have a question regarding a new requirement of two of my upstream providers. We have been running 1.1.1-notls since it was released. We have been in the mindset of, "if it's not broken, don't fix it". But our upstream providers are now telling us to move to DNS/SRV for signaling outbound - for good reason. We have been using LCR for carrier/gateway selection and for URI manipulation prior to relaying to a cluster of session boarder controllers. As you know the DB table for LCR only supports integers in the gw table for ip_addr. I have been reading up on 1.2.x and 1.3.x to upgrade to a solution that would allow us to stay within the same call processing (script) design, but still support the new upstream requirement. I noticed the carrierroute table rewrite_host column is string. Could this be used to satisfy my needs of manipulating the URI for an IP where the carrier still uses a list of gateway IP's to try and use the DNS/SRV domain name where the carrier requires the use for SRV look-up? The session boarder controllers will conduct the SRV look-up after receiving the relayed message from Kamailio, it is just getting the domain name on the URI instead of IP address that I am looking to accomplish. Will the carrierroute module assist in this effort or is there another way that I am totally missing? Figured I would ask the gurus before following Alice down the rabbit hole. -- Thanks, -- Kyle

3 3

[Kamailio-Users] Can openser.cfg lead to pkg memory problem?
by mayamatakeshi 07 Oct '08

07 Oct '08

Hello, we have openser 1.3.3 running in production (current rev.: 4943). For 3 times in 50 days we had to restart openser to correct pkg memory problem. After some time logging messages like this: /openser.log:Aug 19 10:39:18 ipx022 /usr/local/sbin/openser[16991]: ERROR:core:new_credentials: no pkg memory left, openser will eventually run out of pkg memory and refuse all subsequent requests. We are trying to recreate this in our lab so that we can follow memory troubleshooting instructions at http://kamailio.net/dokuwiki/doku.php/troubleshooting:memory, but so far we were unable to do it even when generating millions of calls and registration transactions (we are using SIPp to generate normal call flows and even abnormal call flows detected when reading openser.log, like 'invalid cseq for aor', malformed SIP messages etc). And this is much more than in our production environment, with just 600 subscribers and about 2000 calls a day. The frequency the problem happens is increasing with the number of subscribers, so we are performing periodic restart of openser (actually, what we do is to switch over to the standby server). We already recompiled openser with pkg memory pool size set to 4MB so that this will not have to be done frequently. Since we cannot recreate this in our lab, we suspect there is a situation happening in production that might not be having been properly handled by openser.cfg. So my question is: would it be possible to an overlooked detail in openser.cfg to cause pkg memory problem? In case someone could take a look at it, here's our cfg file: ####### Global Parameters ######### debug=0 log_stderror=no log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" port=5060 /* uncomment and configure the following line if you want openser to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:202.173.5.181:5060 ####### Modules Section ######## #set module path mpath="/usr/local/lib/openser/modules/" /* uncomment next line for MySQL DB support */ loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "uri_db.so" loadmodule "uri.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "carrierroute.so" loadmodule "nathelper.so" loadmodule "dialog.so" loadmodule "snmpstats.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "lcr.so" /* uncomment next line for aliases support NOTE: a DB (like mysql) module must be also loaded */ loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like mysql) module must be also loaded */ #loadmodule "presence.so" #loadmodule "presence_xml.so" loadmodule "uac.so" loadmodule "avpops.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 1) # ----- rr params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ modparam("registrar", "max_contacts", 10) modparam("registrar", "min_expires", 30) modparam("registrar", "max_expires", 40) modparam("registrar", "default_expires", 35) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("uri_db", "use_domain", 1) # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) /* uncomment the following lines to enable DB accounting also */ modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) # ----- usrloc params ----- #modparam("usrloc", "db_mode", 0) /* uncomment the following lines if you want to enable DB persistency for location entries */ modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("usrloc", "use_domain", 1) # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "load_credentials", "$avp(s:rpid)=rpid;$avp(s:blocked)=subscriber_status") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ modparam("alias_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("alias_db", "use_domain", 0) # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ modparam("domain", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #modparam("presence|presence_xml", "db_url", # "mysql://openser:openserrw@localhost/openser") #modparam("presence_xml", "force_active", 1) #modparam("presence", "server_address", "sip:192.168.1.2:5060") # ----- carrieroute params ----- modparam("carrierroute", "db_url", "mysql://openser:openserrw@localhost /openser") modparam("carrierroute", "config_source", "db") modparam("carrierroute", "use_domain", 1) # ----- NatHelper ----- #para versao a partir da versao 1.2 eh necessario esse paramtro para nao dar erro qdo usa a funcao "fix_nated_register();" modparam("nathelper|registrar", "received_avp", "$avp(i:42)") modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- LCR ----- modparam("lcr", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("lcr|tm", "fr_inv_timer_avp", "$avp(i:704)") modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("^auth$|lcr", "rpid_avp", "$avp(i:302)") modparam("lcr", "contact_avp", "$avp(i:711)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "dm_flag", 25) # ----- Dialog ---- modparam("dialog", "dlg_flag", 4) # ----- SnmpStat ----- modparam("snmpstats", "sipEntityType", "registrarServer") modparam("snmpstats", "sipEntityType", "proxyServer") modparam("snmpstats", "MsgQueueMinorThreshold", 2000) modparam("snmpstats", "MsgQueueMajorThreshold", 5000) modparam("snmpstats", "dlg_minor_threshold", 500) modparam("snmpstats", "dlg_major_threshold", 750) modparam("snmpstats", "snmpgetPath","/usr/bin/") modparam("snmpstats", "snmpCommunity","public") ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } ##nat route(2); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accouting ... setflag(3); # ... even if the transaction fails } route(1); } else { /* uncomment the following lines if you want to enable presence */ ##if (is_method("SUBSCRIBE") && $rd == "your.server.ip.address") { ## # in-dialog subscribe requests ## route(2); ## exit; ##} if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server #t_relay(); #exit; route(1); } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests setflag(4); #for dialog statistics # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) route(1); # t_relay(); # exit; } #t_check_trans(); if (is_method("PUBLISH|SUBSCRIBE|REFER|OPTIONS|MESSAGE")) { sl_send_reply("405", "Method not allowed"); exit; } # authenticate if from local subscriber (uncomment to enable auth) if (!(method=="REGISTER") && (!from_gw())) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (!check_from()) { sl_send_reply("403","Forbidden auth ID"); exit; }else if (avp_check("$avp(s:blocked)", "eq/0")) { sl_send_reply("603","Subscriber disabled"); exit; }else if (avp_check("$avp(s:blocked)", "eq/1")) { sl_send_reply("603","Subscriber with outgoing blocked"); exit; } consume_credentials(); # caller authenticated } # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accouting } if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (!check_to()) { sl_send_reply("403","Forbidden auth ID"); exit; }else if (avp_check("$avp(s:blocked)", "eq/0")) { sl_send_reply("403","Subscriber disabled"); exit; } if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); #if the call came from a known gateway it is not authenticated and we cannot use the function check_from() if (from_gw()) { route(4); }else if (!check_from()) {#if the check_from() returns false the call is not from a subscriber route(4); } else {#it is a subscriber, route using flip domain xlog("L_INFO", "routing using carrierroute $rm to $ru\n"); if (!cr_user_rewrite_uri("$fu", "flip")) { t_newtran(); t_reply("404", "No Route"); exit; } #replaces from by it's default DID uac_replace_from("sip:$avp(s:rpid)@$fd"); } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { xlog("L_INFO", "ROUTE_1 $rm to $ru\n"); if (subst_uri('/(sip:.*);nat=yes/\1/')) { setflag(6); }; if (isflagset(5)||isflagset(6)) { route(3); } if (!t_relay()) { sl_reply_error(); }; exit; } route[2] { xlog("L_INFO", "ROUTE_2 $rm to $ru\n"); if (method=="REGISTER") { fix_nated_register(); } else if (!from_gw()){ fix_nated_contact(); }; setflag(5); } route[3] { xlog("L_INFO", "ROUTE_3 $rm to $ru\n"); if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")) { xlog("L_INFO", "FORCE RTP w/ parameter.\n"); force_rtp_proxy("r"); t_on_failure("1"); }; if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); } route[4] { xlog("L_INFO", "uri does exist $rm to $ru \n"); if (alias_db_lookup("dbaliases")){ if (!lookup("location")) { switch ($retcode) { case -1: t_newtran(); t_reply("404", "Subscriber not online"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } }else{#check if did is blocked $rU = "(BLK)" + $rU; if (alias_db_lookup("dbaliases")){ sl_send_reply("403", "DID blocked"); exit; }else{# if it is not a valid DID nor a blocked DID tries to route it using peering domain if (!cr_rewrite_uri("peering", "call_id")) { t_newtran(); t_reply("404", "Peering Not Found"); exit; } } } } failure_route[1] { xlog("L_INFO", "FAILURE $rm to $ru\n"); if (isflagset(6)||isflagset(5)) { unforce_rtp_proxy(); } } onreply_route[1] { xlog("L_INFO", "ONREPLY_1 - Status $rs from $si $rm .\n"); if (is_method("INVITE")) { if ((isflagset(5)||isflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); if (!from_gw()){ #if (isflagset(6)) { xlog("L_INFO", "ONREPLY_1 - ! from gw.\n"); fix_nated_contact(); } exit; } } Regards, takeshi

6 15

Re: [Kamailio-Users] Case Sensitive
by Daniel-Constantin Mierla 07 Oct '08

07 Oct '08

Please cc all the time to mailing lists, so everybody benefits of the outcome or can add comments/suggestions. Also, repeat my question, right now do you need it in some module or for string operations in the config file? Thank you, Daniel On 10/02/08 12:18, michel freiha wrote: > Dear Daniel, > > I prefer doing it in the config file...Is there any function that I > can add and it'll runn the server in case sensitive mode? > > Regards > > On Thu, Oct 2, 2008 at 11:42 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > > On 10/02/08 11:10, michel freiha wrote: > > Dear All, > I would like to ask please about how to force kamailio to be > case sensitive regarding username? > > similar issue reported on the tracker: > http://sourceforge.net/tracker/index.php?func=detail&aid=2047019&group_id=1… > <http://sourceforge.net/tracker/index.php?func=detail&aid=2047019&group_id=1…> > > I guess it is same for you. Right now, do you need it in some > functions or in config file operations? There is need for special > functions for different comparisons, but also might be good to > have case sensitive operations in config file. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com <http://www.asipto.com/> > > -- Daniel-Constantin Mierla http://www.asipto.com

2 7

[Kamailio-Users] Dispatcher hash algorithms.
by Alex Balashov 06 Oct '08

06 Oct '08

I am curious, what exactly is the purpose and philosophy of the various hash algorithms in the dispatcher module? I am referring to the ones that allow the gateway in the route set to be determined through a hash of various SIP headers, such as the From URI, the To URI, the Call-ID GUID, etc. No guarantee of fair distribution - or in fact, any distribution that can be characterised by any describable pattern whatsoever - is implied by these algorithms. Additionally, it seems that without knowing (1) The exact hash algorithm in use; (2) The distribution that this hash algorithm would yield for a given set of possible values of these header fields, which in many cases are specifically intended to be pseudorandom (e.g. Call-ID), there is absolutely no way to determine, from any meaningfully deterministic perspective, which numerical entry in the route set these algorithms would computationally yield. So, I guess my question is: With no implied uniformity or weighting in the distribution whatsoever based on the incidental character of such values, what practical use does it serve to use any algorithms except round-robin or random? Is it expected that the user will plot the hash values against a log of given input strings to determine how the distribution will shape up? Is there some reason why the sort of profoundly lopsided distribution that may create might be desirable? Thanks, -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

4 11

[Kamailio-Users] CDRTools: Error instantiating db class:
by Tiago Rés 06 Oct '08

06 Oct '08

Hello all, When I login to CDRTool, it appears this error ("Error instantiating db class:") in the main page (CDRs). Any suggestion? Thanks! Tiago

1 0

[Serusers] lcr module and cancel
by Deepak 06 Oct '08

06 Oct '08

Hi, I have the following issue: I am using lcr module to select a gw from a set of gw's based upon R-URI prefixes. All is working fine. The issue is with a CANCEL that is sent by the UAC that originates the call before a dialog can be established. This implies that the CANCEL cannot be routed to the correct GW since SER see a CANCEL and simply returns "487 request terminated". How can I forward the CANCEL to the GW the the original INVITE was forwarded to? Any help appreciated.....

2 4

[Kamailio-Users] [offtopic] update vs delete/insert
by Daniel-Constantin Mierla 06 Oct '08

06 Oct '08

Hello, I am trying to get out you knowledge as DBA ... in some discussions in the past was brought out that in some cases doing a delete on primary key followed by a pure insert might be faster than one update. It might be dependent of DB server type and perhaps the question would have better answer in appropriate db-specific mailing lists, but I want to see if some of you has more experience/insights about this and is willing to share or got into same dilemma. Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

3 4

[Kamailio-Users] Carrierroute module: preffix qty?
by Pablo Bertuzzi 06 Oct '08

06 Oct '08

Hi Forum, I want to use carrierroute to terminate 50,000 preffixes to a PSTN gw and block 100,000. So I created 50,000 records with gw ip as rewrite_host. However I only was able to start kamailio with no more than about 2,200 records in carrierroute mysql table. Is it possible to use carrierroute in such a way? Find below log with error: Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: INFO:carrierroute:bind_data_loader: use database as configuration source Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: INFO:carrierroute:load_route_data: name default, id 0, trees: 1 Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: INFO:carrierroute:add_carrier_tree: add carrier default Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: INFO:carrierroute:add_carrier_tree: carrier default not found, add it Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: INFO:carrierroute:add_tree: tree default has internal id 0 Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: INFO:carrierroute:add_carrier_tree: created carrier tree: default, with id 0 and 1 trees Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:db_mysql:db_mysql_convert_row: no private memory left Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:db_mysql:db_mysql_convert_rows: error while converting row #2282 Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:db_mysql:db_mysql_convert_result: error while converting rows Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:db_mysql:db_mysql_store_result: error while converting result Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:core:db_do_query: error while storing result Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:carrierroute:load_route_data: Failed to query database. Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:carrierroute:prepare_route_tree: could not load routing data Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:carrierroute:mod_init: could not prepare route tree Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:core:init_mod: failed to initialize module carrierroute Sep 29 13:36:22 sip-proxy /usr/local/sbin/kamailio[2569]: ERROR:core:main: error while initializing modules

3 2

[Kamailio-Users] Developing Sh interface between the AS(OPENSER/OPENSIPS)and HSS
by suresh tattikota 06 Oct '08

06 Oct '08

hi all I am using the Openser(Opensips) as apllication server . i am developing Sh interface between the AS and HSS. For developing the Sh interface i am using the CDiameterpeer , here i am getiing memory and locking errors. And for using the CDiameterpeer as standalone library what steps i have to follow ... Thanks suresh :)

2 1

Re: [Kamailio-Users] nested redirect
by Daniel-Constantin Mierla 05 Oct '08

05 Oct '08

Hello, your main route block is something loke: route { if(is_method("INVITE")) { .... } exit; } So, apart of INVITE, every other sip request is dropped. See default config file to get a better understanding. kamailio/openser discards by default any sip request received if there is no action to forward for it. Cheers, Daniel Kent Pirlo wrote: > Attached is the full test config I am using.. I have gone back and forth trying a few things, but this is the latest.. If you have a another way to achieve the same result, I am open to change how I'm trying to do it. > > Thanks again for your help. > > Kent > > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Sent: Tuesday, September 23, 2008 5:50 AM > To: Kent Pirlo > Cc: users > Subject: Re: [Kamailio-Users] nested redirect > > Hello, > > sorry, you were right, there was a pcap attachment! The ACK looks ok. > Seems that ACK is quickly discarded by the script. Is the config you > pasted in your first email complete? I need to see it all the understand > how the ack get's routed and where could be dropped by the config. > > Cheers, > Daniel > > > > On 09/17/08 18:42, Kent Pirlo wrote: > >> There should have been a 20k attachment.. redirect.cap.. from tethereal. >> Let me know if you do not have it, and I will re-send it. >> >> kent >> >> -----Original Message----- >> From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] >> Sent: Wednesday, September 17, 2008 6:43 AM >> To: Kent Pirlo >> Cc: users >> Subject: Re: [Kamailio-Users] nested redirect >> >> Hello, >> >> what you sent is the debug from kamailio/openser, which is useful, but I >> would need the sip messages, you can use ngrep for that, do >> ngrep -d any -qt -W byline port 5060 >> >> on your openser/kamailio server and send the output. >> >> Thanks, >> Daniel >> >> >> On 09/17/08 02:22, Kent Pirlo wrote: >> >> >>> ... is my core switch, >>> ... is my openser/Kamailio redirect server >>> ... is a redirect server from a carrier of ours that returns a billing TN in contact header.. >>> >>> >>> Here is the debug of the incoming ACK from the core switch after being sent the 300... >>> >>> >>> Sep 16 15:49:50 [6505] DBG:core:parse_msg: SIP Request: >>> Sep 16 15:49:50 [6505] DBG:core:parse_msg: method: <ACK> >>> Sep 16 15:49:50 [6505] DBG:core:parse_msg: uri: <sip:...@...> >>> Sep 16 15:49:50 [6505] DBG:core:parse_msg: version: <SIP/2.0> >>> Sep 16 15:49:50 [6505] DBG:core:parse_headers: flags=2 >>> Sep 16 15:49:50 [6505] DBG:core:parse_to_param: tag=391854ebdd4d0c63cf50836cbb12191e-6708 >>> Sep 16 15:49:50 [6505] DBG:core:parse_to: end of header reached, state=29 >>> Sep 16 15:49:50 [6505] DBG:core:parse_to: display={}, >>> Sep 16 15:49:50 [6505] DBG:core:get_hdr_field: <To> [75]; >>> Sep 16 15:49:50 [6505] DBG:core:get_hdr_field: to body >>> Sep 16 15:49:50 [6505] DBG:core:get_hdr_field: cseq <CSeq>: <1> <ACK> >>> Sep 16 15:49:50 [6505] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bKdce3ac34730f843a5a9b9c7002690f22>; state=16 >>> Sep 16 15:49:50 [6505] DBG:core:parse_via: end of header reached, state=5 >>> Sep 16 15:49:50 [6505] DBG:core:parse_headers: via found, flags=2 >>> Sep 16 15:49:50 [6505] DBG:core:parse_headers: this is the first via >>> Sep 16 15:49:50 [6505] DBG:core:receive_msg: After parse_msg... >>> Sep 16 15:49:50 [6505] DBG:core:receive_msg: preparing to run routing scripts... >>> Sep 16 15:49:50 [6505] DBG:sl:sl_filter_ACK: to late to be a local ACK! >>> Sep 16 15:49:50 [6505] DBG:core:destroy_avp_list: destroying list (nil) >>> Sep 16 15:49:50 [6505] DBG:core:receive_msg: cleaning up >>> >>> >>> >>> >>> -----Original Message----- >>> From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] >>> Sent: Tuesday, September 16, 2008 5:10 AM >>> To: kentp(a)globalpops.com >>> Cc: users(a)lists.kamailio.org >>> Subject: Re: [Kamailio-Users] nested redirect >>> >>> Hello, >>> >>> can you post here the SIP messages? Will help to spot if there is >>> anything wrong with the ACK or something else in the signaling. >>> >>> Cheers, >>> Daniel >>> >>> >>> On 09/13/08 04:11, kentp(a)globalpops.com wrote: >>> >>> >>> >>>> i have a simple redirect server working: >>>> >>>> Call comes from switch (A) to kamailio redirect server (B).. >>>> (B) runs perl script to determine route >>>> (B) sends Stateless 300 redirect reply to (A). >>>> (A) sends ACK to (B). >>>> >>>> Now, I need to add a nested redirect... >>>> >>>> call comes from sitch (A) to kamailio redirect server (B).. >>>> (B) then needs to send the call to third party proxy (C).. >>>> (C) will return 302 redirect.. >>>> (B) sends ACK to (C) >>>> (B) needs to read the contact header, then run perl script to determine route (using the contact value >>>> returned from (c)) >>>> (B) then sends 300 redirect back to (A). >>>> (A) sends ACK back to (B) >>>> >>>> >>>> so far i have been trying something like this, but the final (A) ACK gets ignored and (B) keeps sending >>>> multiple 300's to (A): >>>> >>>> route { >>>> route(1); >>>> } >>>> >>>> route[1] { >>>> t_on_reply("1"); >>>> t_on_failure("2"); >>>> if (!t_relay("proxy.server.c:5060","0x02")) { >>>> sl_reply_error(); >>>> } >>>> onreply_route[1] { >>>> if(t_check_status("302")){ >>>> $var(lrn_from_bv) = $ct; >>>> } >>>> } >>>> failure_route[2] { >>>> >>>> if (t_check_status("3[0-9][0-9]")) { >>>> route(2); >>>> exit; >>>> } >>>> route[2] { >>>> if(perl_exec("callcheck")){ >>>> avp_pushto("$ru", "$avp(s:routes)/g"); >>>> t_reply("300","Go Here"); >>>> exit; >>>> } >>>> } >>>> >>>> >>>> >>>> -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Serusers] LCR module and CANCEL
by Deepak 04 Oct '08

04 Oct '08

Hi, I have the following issue: I am using lcr module to select a gw from a set of gw's based upon R-URI prefixes. All is working fine. The issue is with a CANCEL that is sent by the UAC that originates the call before a dialog can be established. This implies that the CANCEL cannot be routed to the correct GW since SER see a CANCEL and simply returns "487 request terminated". How can I forward the CANCEL to the GW the the original INVITE was forwarded to? Any help appreciated.....

1 0

[Kamailio-Users] Presence Server returns a 404 Not here msg
by Yoo Chan Jeon 03 Oct '08

03 Oct '08

Hi, I setup the presence server. My sip phone sends a PUBLISH msg. Th presence server sends a 200 msg. My another sip phone sends a SUBSCRIBE msg. The presence server returns 202 msg, and NOTIFY msg. These msgs does not include user in the contact header My sip phone sends a 200 msg I sends a second SUBSCRIBE msg to refresh it. This time my subscribe msg does not include the user in the request msg because the presence server did not include the user in the contact header. The presence server replied back with a 404 messge. So I looked at the codes, and add the user in the contact header when the presence server replys back to a client. But I stll have a same 404 msg. Do I miss something? I always appreciate your help. Thanks. Jeon

2 1

[Kamailio-Users] openser.cfg children=4
by Asim Riaz 03 Oct '08

03 Oct '08

Hi, my openser.cfg is configured for childern=4, but I am seeing child processes? Thanks Asim Riaz

3 6

Re: [Kamailio-Users] sermyadmin setup suse 11
by Ricky Gutierrez 03 Oct '08

03 Oct '08

I have looked for this inside the kamctl and I don't find it, I don't believe that it was that the export error, somebody that has installed sermyadmin in suse 11? regards .. rickygm Wednesday, October 1, 2008 12:24:06 PM, Iñaki Baz Castillo wrote: Subject: Re: [Kamailio-Users] sermyadmin setup suse 11 Please, don't send an image attachment, just copy the console output. This error is very common and is related to "phplib" field. Modify /etc/openserctl and set it to "no" (or something similar..). Look for the exact error in Google for more responses. --

2 1

[Kamailio-Users] Case Sensitive
by michel freiha 02 Oct '08

02 Oct '08

Dear All, I would like to ask please about how to force kamailio to be case sensitive regarding username? Regards

2 1

[Kamailio-Users] registrar module enhancements
by Daniel-Constantin Mierla 02 Oct '08

02 Oct '08

Hello, in the last time several enhancements were added to registrar module to help in dealing with user location records. This correlated with the new modue pv shall give lot of control over the destination set for calls. To get the latest devel version, check out the latest svn - svn co https://openser.svn.sourceforge.net/svnroot/openser/trunk kamailio The news in registrar module are: - ability to unregister all contacts for a SIP addres - there is a new function: unregister(table, uri) - ability to store and maintain single contact per user - the save() function accepts a new value for flags parameter (0x04) to restrict to one single record in location table - ability to access the attributes of all contacts for a user - there are two new functions: reg_fetch_contacts(table, uri, profile) and reg_free_contacts(profile) to make available/release the contacts for a SIP uri. After fetching the contacts, the attributes can be accessed via pseudo variable $ulc(profile=>attr) For more details see the README: http://www.kamailio.org/docs/modules/devel/registrar.html Here are couple of use cases: 1) only one contact per user save("location", "0x04"); Note that if you have couple of phones registering for same user, the one that sent last REGISTER message will be the only ringing. 2) check if a user is calling from a registered device and if not, deny the call if(!reg_fetch_contacts("location", "$fu", "caller")) { sl_send_reply("403", "Please register first"); exit; } $var(i) = 0; $var(found) = 0; $var(contact) = $(ct{nameaddr.uri}); while($var(found) == 0 && $var(i) < $(ulc(caller=>count))) { $if($var(contact)==$(ulc(caller=>addr)[$var(i)])) $var(found) = 1; else $var(i) = $var(i) + 1; } if($var(found) == 0) { sl_send_reply("403", "Please register first"); exit; } Testing and feedback is very much appreciated. For the future: - unregister() to get a $ulc(...) variable as parameter to unregister a specific contact - drop a branch via pv module Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

4 5

[Kamailio-Users] RLS(Resource List Server)
by Yoo Chan Jeon 02 Oct '08

02 Oct '08

Hi, I just want to know how RLS handles SUBSCRIBE messages without using XCAP. I assume that the server already has the list information. I want send the SUBSRIBE message with resource list format: SUBSCRIBE sip:list@domain . . supported:eventlist Accept:application/rlmi+xml Accept:multipart/related I added rls, presence, presence_xml, and pua modlues in the configuration file. I do not have any problems to run the server. Which database should I add list@doamin <list@doamin> to? How do I add the resource list? What kind of format do I need to add a list if I want to add joe@domain, bob@domain, tom@domain ? Do you have some kind of examples? Thanks you. Jeon

2 1

[Kamailio-Users] sermyadmin setup suse 11
by Ricky Gutierrez 01 Oct '08

01 Oct '08

Hi, I am trying to install sermyadmin 0.8 with the patch of security, but when I export the file openser.sql, it shows me this error: INSERT INTO `jsec_user_role_rel` (`id`, `version`, `role_id`, `user_id`) V' at line 1 if I try to export it again, it shows me an error message of having duplicated domain I erase the data of the domain chart, but same the same error ERROR 1062 (23000) at line 6: Duplicate entry '1' for key 1 God blesses, any help is welcome rickygm

2 1

[Serusers] two way routing
by Ivan Zweig 01 Oct '08

01 Oct '08

Hi, we are a small hosted VOIP company. We need a box that can take IN multiple origination providers and route the calls to the appropriate customer instance on our hosted platform. We use switchvox. We would also like to have 2-3 outbound routes to gain price efficiency and redundancy. I need someone that can implement this box and train us how to use it. you can email me to start a dialogue. We are looking for a consultant to help us with this.

1 0

[Kamailio-Users] About presence_dialoginfo - RR in figure 2
by Iñaki Baz Castillo 01 Oct '08

01 Oct '08

Hi, figure 2 of http://www.kamailio.org/docs/modules/devel/presence_dialoginfo.html shows the presence server sending the NOTIFY directly to the watcher. Is it possible to send the NOTIFY through the proxy? Well, since NOTIFY is in-dialog it's just needed that the proxy does RR in the initial SUBSCRIBE before forwarding it to presence server, am I right? Thanks. -- Iñaki Baz Castillo

3 3

[Kamailio-Users] new CDRTool release 6.6.9
by Adrian Georgescu 01 Oct '08

01 Oct '08

Hello, The software contains bug fixes for parallel prepaid calls. Changelog since 6.6.8 cdrtool (6.6.9) unstable; urgency=low * Purge old entries from prepaid_history table * Fixed wrong return to call control engine because sessionDoesNotExist variable was not initialized properly at each request * Print the call id in the syslog when calculating rate The software can be downloaded from: http://download.ag-projects.com/CDRTool/ For people running Debian testing or unstable on an i386 architecture there is an official public repository provided by AG Projects. To use it, add these lines in /etc/apt/sources.list # AG Projects software deb http://ag-projects.com/debian unstable main deb-src http://ag-projects.com/debian unstable main Install the AG Projects debian software signing key: wget http://download.ag-projects.com/agp-debian-gpg.key apt-key add agp-debian-gpg.key After that, run: apt-get update apt-get install cdrtool Regards, Adrian

1 0

[Kamailio-Users] LCR vs Carrierroute and SRV Support Inquirer
by Krom 30 Sep '08

30 Sep '08

Hello, I am/will be a newbie to 1.3.x and beyond and have a question regarding a new requirement of two of my upstream providers. We have been running 1.1.1-notls since it was released. We have been in the mindset of, "if it's not broken, don't fix it". But our upstream providers are now telling us to move to DNS/SRV for signaling outbound - for good reason. We have been using LCR for carrier/gateway selection and for RURI manipulation prior to relaying to a cluster of session boarder controllers. As you know the DB table for LCR only supports integers in the gw table for ip_addr. I have been reading up on 1.2.x and 1.3.x to upgrade to a solution that would allow us to stay within the same call processing (script) design, but still support the new upstream requirement. I noticed the carrierroute table rewrite_host column is string. Could this be used to satisfy my needs of manipulating the RURI for an IP where the carrier still uses a list of gateway IP's to try and use the DNS/SRV domain name where the carrier requires the use for SRV look-up? The session boarder controllers will conduct the SRV look-up after receiving the relayed message from Kamailio, it is just getting the domain name on the RURI instead of IP address that I am looking to accomplish. Will the (new to me) carrierroute module assist in this effort or is there another way that I am totally missing? Figured I would ask the gurus before following Alice down the rabbit hole. -- Thanks, -- Kyle

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[Kamailio-Users] How to prevent - Same username used on multiple sip phones
by Ali Jawad 30 Sep '08

30 Sep '08

Hi On my openser I can logon with the same username on two different softphones. How can I prevent the same username from being used on multiple devices. With Regards

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[Serusers] how-to AVP from radius load
by Samuel Muller 30 Sep '08

30 Sep '08

Hello, I've a single question about AVPs : how can I use them when loaded from radius_load_attr ? example : the 0123452000 is the caller, I want to use an acl to block him. => in my radius user config : 0123451010(a)sip.dot.com User-Password = "0000" User-Name = "0123451010(a)sip.dot.com", SER-UID = "0123451010(a)sip.dot.com", SER-DID = "1", SER-From-UID = "0123451010(a)sip.dot.com", SER-From-DID = "1", SER-Digest-Username = "0123451010", SER-Digest-Realm = "sip.dot.com", SER-Uri-User = "0123451010(a)sip.dot.com", SER-Attr = "acl:pstn", SER-Attr = "asserted-id:0123451010", SER-Attr = "phone:0123451010", SER-Attr = "lang:fr" => in the SER (v2.0.0) : if (!radius_load_attrs("$fu", "@f.uri")) { sl_reply("403", "Radius attributes not found"); exit; } if ( $acl == "pstn" ) { sl_reply("401", "Unauthorized"); exit; } I tried too " $fu.acl ", " set_attr function, ... it seems I can't use SER-Attr as a variable. in the debug (dump_attrs) : ... ser[28670]: r_INBOUND: => dump_attrs() ser[28670]: INFO: avp.c:542: class=GLOBAL ser[28670]: AVP["lang"]="en" ser[28670]: INFO: avp.c:552: track=FROM class=DOMAIN ser[28670]: AVP["did"]="1" ser[28670]: AVP["digest_realm"]="sip.dot.com" ser[28670]: INFO: avp.c:562: track=TO class=DOMAIN ser[28670]: AVP["did"]="1" ser[28670]: AVP["digest_realm"]="sip.dot.com" ser[28670]: INFO: avp.c:572: track=FROM class=USER ser[28670]: AVP["uid"]="0123451010(a)sip.dot.com" ser[28670]: AVP["acl"]="pstn" ser[28670]: AVP["asserted-id"]="0123451010" ser[28670]: AVP["lang"]="fr" ser[28670]: AVP["phone"]="0123451010" ser[28670]: AVP["acl"]="pstn" ser[28670]: AVP["asserted-id"]="0123451010" ser[28670]: AVP["lang"]="fr" ser[28670]: AVP["phone"]="0123451010" ser[28670]: INFO: avp.c:582: track=TO class=USER ser[28670]: AVP["acl"]="pstn" ser[28670]: AVP["asserted-id"]="0123451005" ser[28670]: AVP["lang"]="fr" ser[28670]: AVP["phone"]="0123451005" ser[28670]: AVP["acl"]="pstn" ser[28670]: AVP["asserted-id"]="0123451005" ser[28670]: AVP["lang"]="fr" ser[28670]: AVP["phone"]="0123451005" ser[28670]: INFO: avp.c:592: track=FROM class=URI ser[28670]: AVP["uac_nat"]=1 ser[28670]: AVP["account"]="yes" ser[28670]: INFO: avp.c:602: track=TO class=URI ser[28670]: INFO: No AVP present ... If anyone has tips to help me, many thanks ! I'm sure there's something who replaces the old group_radius module, but which one or how to create SER-Attr $grp ? Many thanks, -- Samuel MULLER sml(a)720.fr

1 0

[Kamailio-Users] new CDRTool release 6.6.8
by Adrian Georgescu 30 Sep '08

30 Sep '08

Hello, The software includes critical bug fixes for parallel prepaid calls. The software can be downloaded from: http://download.ag-projects.com/CDRTool/ Changelog since 6.6.3 cdrtool (6.6.8) unstable; urgency=low * Log debit balance to prepaid_history table cdrtool (6.6.7) unstable; urgency=low * Do not debit balance for inexistent prepaid sessions cdrtool (6.6.5) unstable; urgency=low * Fixed purging of expired prepaid sessions cdrtool (6.6.4) unstable; urgency=low * Added links to Prepaid calls and quota in the main menu * Fixed showFooter() when exporting csv files from rating page * Fixed prepaid table mysql schema Regards, Adrian

1 0

[Kamailio-Users] kamailio 1.4.1 and mediaproxy 1.9.1
by toly 29 Sep '08

29 Sep '08

Greetings, I'm looking for the short answer, most likely no way. Thanks, Toly -- View this message in context: http://www.nabble.com/kamailio-1.4.1-and-mediaproxy-1.9.1-tp19728623p197286… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

1 0

[Kamailio-Users] Disable transfers (REFER) on Kamailio
by Diego Zuaneti Arruda 29 Sep '08

29 Sep '08

Hi all, It's possible to disable any transfers (blind or attended) on Kamailio ? Thanks. Diego.

2 2

Re: [Kamailio-Users] SDP ending with CRLF and a=nortpproxy:yes
by Schober Walter 29 Sep '08

29 Sep '08

I see: * [r4968] modules/nathelper/nathelper.c: - fix the calculation of CRLF length when adding norttptoxy string - patch by Alex Hermann (closes #2117700) And yes, it's that issue and therefore fixed in 1.4.1. > -----Ursprüngliche Nachricht----- > Von: Schober Walter > Gesendet: Samstag, 27. September 2008 14:13 > An: 'miconda(a)gmail.com'; Jerome Martin > Cc: users(a)lists.kamailio.org > Betreff: AW: [Kamailio-Users] SDP ending with CRLF and > a=nortpproxy:yes > > Hi! > > Same issue with 1.4.0. I'm migrating from 1.1.1 to 1.4.0, but > force_rtp_proxy() now adds a sdp line with a beginning 0x00, > so line looks like > 0000 00 61 3d 6e 6f 72 74 70 70 72 6f 78 79 3a 79 65 > .a=nortpproxy:ye > 0010 73 s > But no \r\n\r\n in front anymore in 1.4.0. > > Someone to direct me to the propper source file,pls? > > Walter > > > -----Ursprüngliche Nachricht----- > > Von: users-bounces(a)lists.kamailio.org > > [mailto:users-bounces@lists.kamailio.org] Im Auftrag von > > Daniel-Constantin Mierla > > Gesendet: Montag, 15. September 2008 14:23 > > An: Jerome Martin > > Cc: users(a)lists.kamailio.org > > Betreff: Re: [Kamailio-Users] SDP ending with CRLF and > > a=nortpproxy:yes > > > > Hi Jerome, > > > > On 09/15/08 13:43, Jerome Martin wrote: > > > Hi All, > > > > > > Just FYI, I had to disable the nortpproxy test here because > > it breaks > > > thomson ST2030 endpoints : with that line, they would just plain > > > ignore the messages containing the SDP ... > > was i with malformed SDP (empty lines inside SDP before the > a= line), > > or just the presence of the a=nortpproxy:yes line made the > ST2030 go > > mad? > > > > Cheers, > > Daniel > > > > > > > > Cheers, > > > > > > On Fri, 2008-09-12 at 18:12 +0300, Daniel-Constantin Mierla wrote: > > >> On 09/11/08 16:10, Klaus Darilion wrote: > > >> > A similar bug was also described in opensips. They tried > > to fix it > > >> > but the bug is still present. I wonder if this is > > something new or > > >> > if the bug is thre since the beginning of nathelper > but did not > > >> > occured until now > > >> > > > >> maybe this can be fixed by using the sdp parser available > > now in core. > > >> nathelper has quite limited sdp parsing capabilities now. > > the issue > > >> is to find the proper hook to add the lump with > > >> > > >> a=nortpproxy:yes\r\n" > > >> > > >> Cheers, > > >> Daniel > > >> > > >> > > >> > klaus > > >> > > > >> > Aymeric Moizard schrieb: > > >> > > > >> >> Hi, > > >> >> > > >> >> I just happen to find an issue with a software trying to call > > >> >> through my openser-1.3.1 > > >> >> > > >> >> After inserting "a=nortpproxy:yes\r\n", the message > coming out > > >> >> openser is: > > >> >> > > >> >> [normal sdp packet] > > >> >> a=fmtp:101 0-15\r\n > > >> >> \r\n > > >> >> \r\n > > >> >> "a=nortpproxy:yes\r\n" > > >> >> > > >> >> No matter who is responsible in that specific case (the SDP > > >> >> received by openser with several \r\n looks not > > compliant to me). > > >> >> > > >> >> I think openser should add "a=nortpproxy:yes" in > > different place > > >> >> in the SDP packet: NOT at the end of the SDP body. The > > "a=nortpproxy:yes" > > >> >> should be either put at the global attribute level or > > right below > > >> >> a m= line. > > >> >> > > >> >> This approach would solve such issue and would be much more > > >> >> adequate to the requirement: currently it's not very > > possible to > > >> >> relay an audio stream and not relay a video stream with the > > >> >> current approach. > > >> >> > > >> >> tks, > > >> >> Aymeric MOIZARD / ANTISIP > > >> >> amsip - http://www.antisip.com > > >> >> osip2 - http://www.osip.org > > >> >> eXosip2 - http://savannah.nongnu.org/projects/exosip/ > > >> >> > > >> >> > > >> >> _______________________________________________ > > >> >> Users mailing list > > >> >> Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > > >> >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > >> >> > > >> > > > >> > _______________________________________________ > > >> > Users mailing list > > >> > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > > >> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > >> > > > >> > > > >> > > >> > > > > > > *Jérôme Martin **| **LongPhone* > > > *Responsable Architecture Réseau* > > > 122, rue la Boetie | 75008 Paris > > > Tel : +33 (0)1 56 26 28 44 > > > Fax : +33 (0)1 56 26 28 45 > > > Mail : *jmartin*(a)longphone.fr > > > Web : www.longphone.com <http://www.longphone.com> > > > > > > > > > > > > ---------------------------------------------------------------------- > > > -- > > > > > > _______________________________________________ > > > Users mailing list > > > Users(a)lists.kamailio.org > > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > > > > -- > > Daniel-Constantin Mierla > > http://www.asipto.com > > > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >

2 1

[Kamailio-Users] CDRTool - stored procedures - not creating tables
by Graham Wooden 28 Sep '08

28 Sep '08

All, I have been having a heck of a time getting the radacct tables to be generated from the stored procedures, with the latest CDRTool. I have done and have reviewed them again the steps mentioned at: http://cdrtool.ag-projects.com/browser/doc/INSTALL.txt Can anyone give any guidance. The call entries are showing up (for the most part) in the radacct table, but getting the main error upon viewing the CDR. ³Database error: Invalid SQL: select count(*) as records from radacct200809 where (AcctStartTime >= '2008-09-28 05:34' and AcctStartTime < '2008-09-28 23:55') MySQL error: 1146 (Table 'cdrtool.radacct200809' doesn't exist) 64Session halted.² I am using 1.4 Kamailio Thanks, -graham

1 0

Re: [Kamailio-Users] SDP ending with CRLF and a=nortpproxy:yes
by Schober Walter 27 Sep '08

27 Sep '08

Hi! Same issue with 1.4.0. I'm migrating from 1.1.1 to 1.4.0, but force_rtp_proxy() now adds a sdp line with a beginning 0x00, so line looks like 0000 00 61 3d 6e 6f 72 74 70 70 72 6f 78 79 3a 79 65 .a=nortpproxy:ye 0010 73 s But no \r\n\r\n in front anymore in 1.4.0. Someone to direct me to the propper source file,pls? Walter > -----Ursprüngliche Nachricht----- > Von: users-bounces(a)lists.kamailio.org > [mailto:users-bounces@lists.kamailio.org] Im Auftrag von > Daniel-Constantin Mierla > Gesendet: Montag, 15. September 2008 14:23 > An: Jerome Martin > Cc: users(a)lists.kamailio.org > Betreff: Re: [Kamailio-Users] SDP ending with CRLF and > a=nortpproxy:yes > > Hi Jerome, > > On 09/15/08 13:43, Jerome Martin wrote: > > Hi All, > > > > Just FYI, I had to disable the nortpproxy test here because > it breaks > > thomson ST2030 endpoints : with that line, they would just plain > > ignore the messages containing the SDP ... > was i with malformed SDP (empty lines inside SDP before the > a= line), or just the presence of the a=nortpproxy:yes line > made the ST2030 go mad? > > Cheers, > Daniel > > > > > Cheers, > > > > On Fri, 2008-09-12 at 18:12 +0300, Daniel-Constantin Mierla wrote: > >> On 09/11/08 16:10, Klaus Darilion wrote: > >> > A similar bug was also described in opensips. They tried > to fix it > >> > but the bug is still present. I wonder if this is > something new or > >> > if the bug is thre since the beginning of nathelper but did not > >> > occured until now > >> > > >> maybe this can be fixed by using the sdp parser available > now in core. > >> nathelper has quite limited sdp parsing capabilities now. > the issue > >> is to find the proper hook to add the lump with > >> > >> a=nortpproxy:yes\r\n" > >> > >> Cheers, > >> Daniel > >> > >> > >> > klaus > >> > > >> > Aymeric Moizard schrieb: > >> > > >> >> Hi, > >> >> > >> >> I just happen to find an issue with a software trying to call > >> >> through my openser-1.3.1 > >> >> > >> >> After inserting "a=nortpproxy:yes\r\n", the message coming out > >> >> openser is: > >> >> > >> >> [normal sdp packet] > >> >> a=fmtp:101 0-15\r\n > >> >> \r\n > >> >> \r\n > >> >> "a=nortpproxy:yes\r\n" > >> >> > >> >> No matter who is responsible in that specific case (the SDP > >> >> received by openser with several \r\n looks not > compliant to me). > >> >> > >> >> I think openser should add "a=nortpproxy:yes" in > different place > >> >> in the SDP packet: NOT at the end of the SDP body. The > "a=nortpproxy:yes" > >> >> should be either put at the global attribute level or > right below > >> >> a m= line. > >> >> > >> >> This approach would solve such issue and would be much more > >> >> adequate to the requirement: currently it's not very > possible to > >> >> relay an audio stream and not relay a video stream with the > >> >> current approach. > >> >> > >> >> tks, > >> >> Aymeric MOIZARD / ANTISIP > >> >> amsip - http://www.antisip.com > >> >> osip2 - http://www.osip.org > >> >> eXosip2 - http://savannah.nongnu.org/projects/exosip/ > >> >> > >> >> > >> >> _______________________________________________ > >> >> Users mailing list > >> >> Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > >> >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >> >> > >> > > >> > _______________________________________________ > >> > Users mailing list > >> > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > >> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >> > > >> > > >> > >> > > > > *Jérôme Martin **| **LongPhone* > > *Responsable Architecture Réseau* > > 122, rue la Boetie | 75008 Paris > > Tel : +33 (0)1 56 26 28 44 > > Fax : +33 (0)1 56 26 28 45 > > Mail : *jmartin*(a)longphone.fr > > Web : www.longphone.com <http://www.longphone.com> > > > > > > > ---------------------------------------------------------------------- > > -- > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >

1 0

[Kamailio-Users] towards kamailio v1.5.0
by Daniel-Constantin Mierla 26 Sep '08

26 Sep '08

Hello, as we just released 1.4.1 in the latest stable branch, I just want to give updates on the development version of the project. In less that two months since 1.4.x release series was branched, lot of new things happened - new modules and features, code refactoring and fixes. The outline of the new additions: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-1.5.x A drafted version of the roadmap is available at: http://www.kamailio.org/dokuwiki/doku.php/roadmap:1.5.x I take this occasion to ask all developers to add there the new features already introduced and their plans for 1.5.0 in the above two pages. There are some features introduced on SVN that are not yet listed in Dokuwiki, features:new-in-1.5.x. So please, when you add some major improvement, add a short note about. As shortcut, those links are hosted on the website page: http://www.kamailio.org/mos/view/Roadmap/ For a deeper track of all changes you can browse the SVN repository at Source Forge. http://openser.svn.sourceforge.net/viewvc/openser/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 4

[Kamailio-Users] ip address of VIA header
by Alex R.S.M 26 Sep '08

26 Sep '08

Hi, Is there any function in openSER to return the ip address of VIA header? or return the VIA header? Thanks, alex _________________________________________________________________ Get more out of the Web. Learn 10 hidden secrets of Windows Live. http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns!5…

3 4

[Kamailio-Users] Exported statistics
by Daniel Cardoso Alves 26 Sep '08

26 Sep '08

Hi people... I am trying to use "Exported statistics" of dialog module, but I have the following message: 404 Statistics Not Found I am using the dialog module to finish some calls... And it's work perfect... But I want to get some statistics... I looked on Internet but I dont find a solution... Are there some special configuration to make to use "Exported statistics"??? Thanks in advance... Novos endereços, o Yahoo! que você conhece. Crie um email novo com a sua cara @ymail.com ou @rocketmail.com. http://br.new.mail.yahoo.com/addresses

2 3

[Kamailio-Users] SUBSCRIBE message with a private header
by Yoo Chan Jeon 26 Sep '08

26 Sep '08

Hi, Thank you for your reponse. By using $hdr(HEADER-NAME), it returns "Private-Header:value". Now I have other problems. I would like to check if the header exist, then I want to get the value. How do I get the value from $hdr(HEADER-NAME) ? Thanks. Jeon El Miércoles, 24 de Septiembre de 2008, Yoo Chan Jeon escribió: >* Hi, *>* I have a client sip agent which sends a SUBSCRIBE msg with a private header *>* to the Kamailio presence server. *>* Hwo do I get the private header informtaion? I want to read it and send it *>* with a Notify message. * Read it with $hdr(HEADER-NAME).

2 1

Re: [Kamailio-Users] How to prevent - Same username used on multiple sip phones
by Ali Jawad 26 Sep '08

26 Sep '08

Hi I did try if(is_method("REGISTER")) { if(lookup("location")){ sl_send_reply("401","Unautorized"); exit; } } With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554 ________________________________ From: Sergio Gutierrez [mailto:saguti@gmail.com] Sent: 2008-09-26 14:38 To: Ali Jawad Subject: Re: [Kamailio-Users] How to prevent - Same username used on multiple sip phones Hi Ali. Maybe you could use the function registered() of the module registrar, to check whether the user is already registered. http://kamailio.org/docs/modules/1.4.x/registrar.html#id2506815 Best regards. Sergio G. On 9/26/08, Ali Jawad <ali.jawad(a)splendor.net> wrote: Hi On my openser I can logon with the same username on two different softphones. How can I prevent the same username from being used on multiple devices. With Regards _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users -- This message has been scanned for viruses and dangerous content by MailScanner <http://www.mailscanner.info/> , and is believed to be clean.

1 1

[Kamailio-Users] SUBSCRIBE message with a private header
by Yoo Chan Jeon 26 Sep '08

26 Sep '08

Hi, Now I got the value. Discard the previous message. Thanks. Jeon >Hi, >Thank you for your reponse. >By using $hdr(HEADER-NAME), it returns "Private-Header:value". >Now I have other problems >I would like to check if the header exist, then I want to get the value. >How do I get the value from $hdr(HEADER-NAME) ? >Thanks. >Jeon El Miércoles, 24 de Septiembre de 2008, Yoo Chan Jeon escribió: > Hi, > I have a client sip agent which sends a SUBSCRIBE msg with a private header > to the Kamailio presence server. > Hwo do I get the private header informtaion? I want to read it and send it > with a Notify message. Read it with $hdr(HEADER-NAME).

2 1

[Kamailio-Users] Failed gateway recovery with dispatcher.
by Alex Balashov 25 Sep '08

25 Sep '08

How does one go about resurrecting gateways that have been marked into "probing" mode back into the "active" set once they start replying to the pings that the dispatcher facility sends? I see the OPTIONS pings going through, and I see the UAs generating replies to them whey come back up. But OpenSER does nothing in response to receipt of those replies to mark them as active again. Because the OPTIONS request is not originated from script, there is no way to catch the replies in an ONREPLY-ROUTE, otherwise I would have tried to handle it that way and re-mark the gateways. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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[Kamailio-Users] Kamailio v1.4.1 Released
by Daniel-Constantin Mierla 25 Sep '08

25 Sep '08

Hello, A new minor release is available as Kamailio 1.4.1. It is based on the latest version of branch 1.4, therefore those running 1.4.0 are advised to upgrade. The compatibility of configuration file and database structure with v1.4.0 is preserved. This release includes fixes and improvements to issues discovered since the release of v1.4.0. Source tarballs are available at: http://www.kamailio.org/pub/kamailio/1.4.1/src/ Detailed changelog: http://www.kamailio.org/pub/kamailio/1.4.1/ChangeLog Download via SVN: svn co https://openser.svn.sourceforge.net/svnroot/openser/branches/1.4 kamailio Tag for this release can be browsed at: http://openser.svn.sourceforge.net/viewvc/openser/tags/1.4.1/ Project site at SourceForge.net (still using old name): http://sourceforge.net/projects/openser/ Modules' documentation: http://www.kamailio.org/docs/modules/1.4.x/ <http://www.kamailio.org/docs/modules/1.4.x/> Best regards, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 3

[Kamailio-Users] crazy error OpenSER 1.3.3
by VIP Carrier 25 Sep '08

25 Sep '08

Hello, I'm trying to compile OpenSER 1.3.3 and getting the following error i have run a comand "make all include_modules="mysql"" -------------------------------------------------------------------------------------------------------- dbase.c:29:25: error: mysql/mysql.h: No such file or directory dbase.c:30:26: error: mysql/errmsg.h: No such file or directory dbase.c:31:33: error: mysql/mysql_version.h: No such file or directory In file included from dbase.c:38: my_con.h:38: error: expected specifier-qualifier-list before Б─≤MYSQL_RESБ─≥ dbase.c: In function Б─≤db_mysql_submit_queryБ─≥: dbase.c:65: error: Б─≤struct my_conБ─≥ has no member named Б─≤timestampБ─≥ dbase.c:66: warning: implicit declaration of function Б─≤mysql_pingБ─≥ dbase.c:66: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:70: error: Б─≤struct my_conБ─≥ has no member named Б─≤timestampБ─≥ dbase.c:88: warning: implicit declaration of function Б─≤mysql_queryБ─≥ dbase.c:88: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:91: warning: implicit declaration of function Б─≤mysql_errnoБ─≥ dbase.c:91: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:92: error: Б─≤CR_SERVER_GONE_ERRORБ─≥ undeclared (first use in this function) dbase.c:92: error: (Each undeclared identifier is reported only once dbase.c:92: error: for each function it appears in.) dbase.c:92: error: Б─≤CR_SERVER_LOSTБ─≥ undeclared (first use in this function) dbase.c:96: warning: implicit declaration of function Б─≤mysql_errorБ─≥ dbase.c:96: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:96: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:96: warning: format Б─≤%sБ─≥ expects type Б─≤char *Б─≥, but argument 4 has type Б─≤intБ─≥ dbase.c: In function Б─≤db_mysql_store_resultБ─≥: dbase.c:195: error: Б─≤struct my_conБ─≥ has no member named Б─≤resБ─≥ dbase.c:195: warning: implicit declaration of function Б─≤mysql_store_resultБ─≥ dbase.c:195: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:196: error: Б─≤struct my_conБ─≥ has no member named Б─≤resБ─≥ dbase.c:197: warning: implicit declaration of function Б─≤mysql_field_countБ─≥ dbase.c:197: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:202: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:202: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:202: warning: format Б─≤%sБ─≥ expects type Б─≤char *Б─≥, but argument 4 has type Б─≤intБ─≥ dbase.c:216: warning: implicit declaration of function Б─≤mysql_free_resultБ─≥ dbase.c:216: error: Б─≤struct my_conБ─≥ has no member named Б─≤resБ─≥ dbase.c:223: error: Б─≤struct my_conБ─≥ has no member named Б─≤resБ─≥ dbase.c: In function Б─≤db_mysql_free_resultБ─≥: dbase.c:253: error: Б─≤struct my_conБ─≥ has no member named Б─≤resБ─≥ dbase.c:254: error: Б─≤struct my_conБ─≥ has no member named Б─≤resБ─≥ dbase.c: In function Б─≤db_mysql_fetch_resultБ─≥: dbase.c:361: error: Б─≤struct my_conБ─≥ has no member named Б─≤resБ─≥ dbase.c:361: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:362: error: Б─≤struct my_conБ─≥ has no member named Б─≤resБ─≥ dbase.c:363: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:368: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:368: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:368: warning: format Б─≤%sБ─≥ expects type Б─≤char *Б─≥, but argument 4 has type Б─≤intБ─≥ dbase.c:379: warning: implicit declaration of function Б─≤mysql_num_rowsБ─≥ dbase.c:379: error: Б─≤struct my_conБ─≥ has no member named Б─≤resБ─≥ dbase.c:414: error: Б─≤struct my_conБ─≥ has no member named Б─≤rowБ─≥ dbase.c:414: warning: implicit declaration of function Б─≤mysql_fetch_rowБ─≥ dbase.c:414: error: Б─≤struct my_conБ─≥ has no member named Б─≤resБ─≥ dbase.c:415: error: Б─≤struct my_conБ─≥ has no member named Б─≤rowБ─≥ dbase.c:416: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:416: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ dbase.c:416: warning: format Б─≤%sБ─≥ expects type Б─≤char *Б─≥, but argument 4 has type Б─≤intБ─≥ dbase.c: In function Б─≤db_last_inserted_idБ─≥: dbase.c:649: warning: implicit declaration of function Б─≤mysql_insert_idБ─≥ dbase.c:649: error: Б─≤struct my_conБ─≥ has no member named Б─≤conБ─≥ make[1]: *** [dbase.o] Error 1 make[1]: Leaving directory `/src/openser-1.3.3-notls/modules/mysql' make: *** [modules] Error 2

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[Kamailio-Users] Res: Problems with SerMyAdmin Installation
by Flavio Goncalves 25 Sep '08

25 Sep '08

Hi Brian, I have seen some people complianing about this, but I have not being able to reproduce the problem. Could you send me a copy of your database schema, it is probably the source of the problem. Have you applied the openser.sql in the latest version? Best regards, Flavio E. Goncalves ----- Mensagem original ---- De: Brian Chamberlain <brian(a)asterisk.ie> Para: users(a)lists.kamailio.org Enviadas: Quarta-feira, 10 de Setembro de 2008 13:45:04 Assunto: Re: [Kamailio-Users] Problems with SerMyAdmin Installation Hi, Can you add your users to groups? I just went back to openser 1.2 because of the problems I was facing with sermyadmin adding users to groups and bombing out. I was using 1.33 - but it seems that that is not the problem. I can add groups ok but get the following when adding users to those groups! (GRRRR) Grails Runtime Exception Error Details Message: could not insert: [UserGroup]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [UserGroup] Caused by: org.springframework.dao.DataIntegrityViolationException: could not insert: [UserGroup]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [UserGroup] Class: UserController At Line: [115] Code Snippet: Stack Trace org.codehaus.groovy.runtime.InvokerInvocationException: org.springframework.dao.DataIntegrityViolationException: could not insert: [UserGroup]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [UserGroup] at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:290) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationDispatcher.invoke(ApplicationDispatcher.java:630) at org.apache.catalina.core.ApplicationDispatcher.processRequest(ApplicationDispatcher.java:436) at org.apache.catalina.core.ApplicationDispatcher.doForward(ApplicationDispatcher.java:374) at org.apache.catalina.core.ApplicationDispatcher.forward(ApplicationDispatcher.java:302) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.jsecurity.web.servlet.WebInterceptorFilter.doFilterInternal(WebInterceptorFilter.java:106) at org.jsecurity.web.servlet.SecurityContextFilter.doFilterInternal(SecurityContextFilter.java:93) at org.jsecurity.web.servlet.OncePerRequestFilter.doFilter(OncePerRequestFilter.java:106) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.StandardWrapperValve.invoke(StandardWrapperValve.java:233) at org.apache.catalina.core.StandardContextValve.invoke(StandardContextValve.java:175) at org.apache.catalina.core.StandardHostValve.invoke(StandardHostValve.java:128) at org.apache.catalina.valves.ErrorReportValve.invoke(ErrorReportValve.java:102) at org.apache.catalina.core.StandardEngineValve.invoke(StandardEngineValve.java:109) at org.apache.catalina.connector.CoyoteAdapter.service(CoyoteAdapter.java:286) at org.apache.coyote.http11.Http11Processor.process(Http11Processor.java:844) at org.apache.coyote.http11.Http11Protocol$Http11ConnectionHandler.process(Http11Protocol.java:583) at org.apache.tomcat.util.net.JIoEndpoint$Worker.run(JIoEndpoint.java:447) at java.lang.Thread.run(Thread.java:595) Caused by: org.springframework.dao.DataIntegrityViolationException: could not insert: [UserGroup]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [UserGroup] at UserController$_closure6_closure13.doCall(UserController.groovy:115) at UserController$_closure6.doCall(UserController.groovy:105) at UserController$_closure6.doCall(UserController.groovy) ... 29 more Caused by: org.hibernate.exception.ConstraintViolationException: could not insert: [UserGroup] at $Proxy3.saveOrUpdate(Unknown Source) ... 32 more Caused by: com.mysql.jdbc.exceptions.MySQLIntegrityConstraintViolationException: Duplicate entry 'brian-sip.asterisk.ie-' for key 2 at com.mysql.jdbc.SQLError.createSQLException(SQLError.java:1011) at com.mysql.jdbc.SQLError.createSQLException(SQLError.java:956) at com.mysql.jdbc.MysqlIO.checkErrorPacket(MysqlIO.java:3491) at com.mysql.jdbc.MysqlIO.checkErrorPacket(MysqlIO.java:3423) at com.mysql.jdbc.MysqlIO.sendCommand(MysqlIO.java:1936) at com.mysql.jdbc.MysqlIO.sqlQueryDirect(MysqlIO.java:2060) at com.mysql.jdbc.ConnectionImpl.execSQL(ConnectionImpl.java:2542) at com.mysql.jdbc.PreparedStatement.executeInternal(PreparedStatement.java:1734) at com.mysql.jdbc.PreparedStatement.executeUpdate(PreparedStatement.java:2019) at com.mysql.jdbc.PreparedStatement.executeUpdate(PreparedStatement.java:1937) at com.mysql.jdbc.PreparedStatement.executeUpdate(PreparedStatement.java:1922) at org.apache.tomcat.dbcp.dbcp.DelegatingPreparedStatement.executeUpdate(DelegatingPreparedStatement.java:102) ... 33 more Hello, I solved my problem with Nuno's help. My problem was with the permissions in mysql. The command GRANT ALL PRIVILEGES ON openser.* TO user@'%' IDENTIFIED BY 'Password'; did't add the user properly.. I had to also use GRANT ALL PRIVILEGES ON openser.* TO user@'localhost' IDENTIFIED BY 'password'; because sermyadmin and mysql were running in the same machine. After that I could run sermyadmin and it changed all the tables properly. -- View this message in context: http://www.nabble.com/Problems-with-SerMyAdmin-Installation-tp16965040p1941… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com. _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users

2 1

[Kamailio-Users] problem with emergency calls (short-numbers)
by BERGANZ François 25 Sep '08

25 Sep '08

Hello, I have a problem with my emergency calls if (uri=~"^sip:1[578]{1}@*" || uri=~"^sip:11[29]{1}@*") if I call 170725014, it think that it is 17 (emergency in france) is my "if" syntaxe is correct ? Cordialement BERGANZ François http://www.acropolistelecom.net <http://www.acropolistelecom.net/>

2 3

[Kamailio-Users] BAD URI After $ru and avp
by Ali Jawad 25 Sep '08

25 Sep '08

Hi I am trying to change the Ruri as stated below DB usr_preferences Domain= Value=callfwd Attribute=33179974674 if(avp_db_load("$ruri/username","$avp(s:callfwd)")){ xlog("L_INFO","X0XOXOXOXOXOXOXXOXOXOXXOXOXOXOXOX"); $ru = $avp(s:callfwd); route(8); exit; }; It enters allright and it changes the rURI however It throws the errors below: Sep 25 13:31:23 [24879] DBG:avpops:db_close_query: close avp query Sep 25 13:31:23 [24879] DBG:core:db_free_rows: freeing 1 rows Sep 25 13:31:23 [24879] DBG:core:db_free_rows: row[0]=0x8187160 Sep 25 13:31:23 [24879] ERROR:core:do_action: bad uri <33179974674>, dropping packet Sep 25 13:31:23 [24879] DBG:core:parse_to: display={"9611375554"}, ruri={sip:9611375554@xyz.com} Sep 25 13:31:23 [24879] ERROR:core:parse_uri: bad uri, state 0 parsed: <3317> (4) / <33179974674> (25) Sep 25 13:31:23 [24879] ERROR:core:parse_sip_msg_uri: bad uri <33179974674(a)33.33.33.33> Sep 25 13:31:23 [24879] DBG:core:set_err_info: ec: 1, el: 3, ei: 'error parsing r-uri' Sep 25 13:31:23 [24879] ERROR:tm:new_t: uri invalid With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

2 1

[Kamailio-Users] Error compiling openser 1.2.2 notls on Ubuntu 8.04 x86_64
by Marcello Lupo 25 Sep '08

25 Sep '08

Hi, i'm getting this error on compiling openser-1.2.2-notls on Ubuntu 8.04 x86_64 Can you help me to see wath's is wrong? Same sources compiled perfect on SLES 10 x86_64. Thank you. Bye, Marcello gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops - falign-loops -ftree-vectorize -mtune=opteron -DNAME='"openser"' - DVERSION='"1.2.2-notls"' -DARCH='"x86_64"' -DOS='"linux"' - DCOMPILER='"gcc 4.2.3"' -D__CPU_x86_64 -D__OS_linux -D__SMP_yes - DCFG_DIR='"/data/openser/etc/openser/"' -DPKG_MALLOC -DSHM_MEM - DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE - DHAVE_RESOLV_RES -DSTATISTICS -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT - DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN - DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL - DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT - DHAVE_SELECT -c action.c -o action.o In file included from parser/msg_parser.h:48, from action.h:27, from action.c:44: parser/../config.h:2: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘:’ token In file included from parser/msg_parser.h:48, from action.h:27, from action.c:44: parser/../config.h:2:38: error: invalid digit "8" in octal constant parser/../config.h:2:41: error: invalid suffix "Z" on integer constant parser/../config.h:8: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘can’ parser/../config.h:10: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘version’ parser/../config.h:14: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘even’ parser/../config.h:19: error: expected identifier or ‘(’ before ‘if’ parser/../config.h:26:63: warning: missing terminating ' character parser/../config.h:26: error: missing terminating ' character In file included from action.c:45: config.h:2: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘:’ token In file included from action.c:45: config.h:2:38: error: invalid digit "8" in octal constant config.h:2:41: error: invalid suffix "Z" on integer constant config.h:8: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘can’ config.h:10: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘version’ config.h:14: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘even’ config.h:19: error: expected identifier or ‘(’ before ‘if’ config.h:26:63: warning: missing terminating ' character config.h:26: error: missing terminating ' character In file included from route.h:32, from forward.h:39, from action.c:49: config.h:2: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘:’ token In file included from route.h:32, from forward.h:39, from action.c:49: config.h:2:38: error: invalid digit "8" in octal constant config.h:2:41: error: invalid suffix "Z" on integer constant config.h:8: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘can’ config.h:10: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘version’ config.h:14: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘even’ config.h:19: error: expected identifier or ‘(’ before ‘if’ config.h:26:63: warning: missing terminating ' character config.h:26: error: missing terminating ' character In file included from ut.h:47, from action.c:54: config.h:2: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘:’ token In file included from ut.h:47, from action.c:54: config.h:2:38: error: invalid digit "8" in octal constant config.h:2:41: error: invalid suffix "Z" on integer constant config.h:8: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘can’ config.h:10: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘version’ config.h:14: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘even’ config.h:19: error: expected identifier or ‘(’ before ‘if’ config.h:26:63: warning: missing terminating ' character config.h:26: error: missing terminating ' character In file included from mem/mem.h:35, from action.c:56: mem/../config.h:2: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘:’ token In file included from mem/mem.h:35, from action.c:56: mem/../config.h:2:38: error: invalid digit "8" in octal constant mem/../config.h:2:41: error: invalid suffix "Z" on integer constant mem/../config.h:8: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘can’ mem/../config.h:10: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘version’ mem/../config.h:14: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘even’ mem/../config.h:19: error: expected identifier or ‘(’ before ‘if’ mem/../config.h:26:63: warning: missing terminating ' character mem/../config.h:26: error: missing terminating ' character In file included from mem/mem.h:54, from action.c:56: mem/f_malloc.h:134: warning: ‘struct mem_info’ declared inside parameter list mem/f_malloc.h:134: warning: its scope is only this definition or declaration, which is probably not what you want action.c: In function ‘do_action’: action.c:387: error: ‘ser_error’ undeclared (first use in this function) action.c:387: error: (Each undeclared identifier is reported only once action.c:387: error: for each function it appears in.) action.c:564: error: ‘rlist’ undeclared (first use in this function) make: *** [action.o] Error 1

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[Kamailio-Users] Bind Access To Users to their IPs
by Ali Jawad 25 Sep '08

25 Sep '08

Hi I want some of my users only to be able to logon through a certain IP, is there some ACL mechanism that can be used to do that ? With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

2 4

[Kamailio-Users] SUBSCRIBE message with a private header
by Yoo Chan Jeon 24 Sep '08

24 Sep '08

Hi, I have a client sip agent which sends a SUBSCRIBE msg with a private header to the Kamailio presence server. Hwo do I get the private header informtaion? I want to read it and send it with a Notify message. Thanks. Jeon

2 1

[Kamailio-Users] sipgrep as non-privileged user
by Robert McNaught 24 Sep '08

24 Sep '08

Hi, I am using sipgrep to do sip traces on an openser/asterisk. I am find this great for getting readable traces - previously ngrep was difficult to read. However, to use sipgrep, I need to be logged in as the linux root user. I can change the PATH of an admin user so that they can have access to the command but they cannot use it as they need to be root in order to gain access to see the network traffic. Does anyone know of a way to do this? Is it normal to have engineers continually log in as root to take these traces - it seems awful dangerous? I know you could log in as a non-prileged user and put in sudo sipgrep -f xxx -t xxx, but not sure if people are actually doing this in production systems, or have a workaround? Thanks Robert

3 2

[Kamailio-Users] new feature: generic shared container in config language
by Elena-Ramona Modroiu 24 Sep '08

24 Sep '08

Hi, I just committed a new module: htable http://www.kamailio.org/docs/modules/devel/htable.html It adds a generic hash table container for usage in configuration file. The hash table is in shared memory, therefore the values are global over all kamailio processes. The items in hash table can be accessed via $sht(name). The name can include pseudo-variables that will be evaluated at runtime. In this way, the hash table can be used to store data per user and can offer functionality of dealing with array of values. An example of how to protect against dictionary attacks is in the README (link to online version above). As another example, loading values in hash tables from AVPs and building an array: $var(i) = 0; while($(avp(s:tmp)[$var(i)]) != NULL) { $sht(tmp[$var(i)]) = $(avp(s:tmp)[$var(i)]); $var(i) = $var(i)+1; } As a roadmap for the new module, planned for addition: - MI commands to get/set items in HT - ability to save/load in/from database - ability to init items at startup - auto-expire time for items This completes global variables space along with shared variables ($shv(name)) which are single value and allow only static name, but they are faster to access (see cfgutils readme file for more details on $shv(name)). Regards, Ramona -- Elena-Ramona Modroiu Web: http://www.asipto.com

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