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37519 discussions

[Kamailio-Users] openser.org and project documentation
by Jeremy McNamara 23 Sep '08

23 Sep '08

Whoever has taken over openser.org should do the community as a favor and at the very least HTTP 301 redirect the documentation links over to kamailio.org. There are soo many sites that index URLs like: http://openser.org/docs/modules/ ...however openser.org gives back a 404 not found.

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[Kamailio-Users] About Changing URI's
by Ali Jawad 23 Sep '08

23 Sep '08

Hi I have the following setup Cell Phone "Any Possible Number" ----Calls a DID number-----> DID Provider --> Forwards Call to my Openser --> OpenSer Server--> Forwards Call to Asterisk with A2B --> Forwards it to the cell phone of the user In short, a person can call a DID and it gets forwarded to Openser, if the user is online using his sip phone, the sip phone rings and all is OK. However, if the user is not online I do forward that call to Asterisk and I want to forward it to the cell phone of the user. The problem is that the FROM field and the SOURCE URI keep changing,they are in the from phonenumberOfCaller(a)DID_Service_Provider.com and I do want to charge the user for the call. To do that I need to have the FROM field and the source URI to be the USER ID so that I can charge him for the call. So, I might have to change the source URI and FROM field of the packet and send it to asterisk..it would send the packet back to Openser and I have to change those back to what they were and then send it to the DID provider. Might sound confusing, and it might sound stupid. Please help me set this straight, can it be done ? Maybe not ? Maybe Yes... Any suggestions are welcome. With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

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[Kamailio-Users] un-subscribing a pending subscription is also pending
by jh＠tutpro.com 23 Sep '08

23 Sep '08

a user complained that un-subscribing a pending subscription is also pending, i.e., kamailio responds with 202 instead of 200. i verified the behavior. now the question is, what do the relevant RFCs say about it? it appears logical to me that if a user gives up waiting to get his/her subscription confirmed, he/she should be able to forget the whole thing by sending an un-subscribe. any comments on this? -- juha

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[Kamailio-Users] CDRTool 6.6.3/6.6.4 with freeradius-cdrtool-1.1
by Diego Zuaneti Arruda 23 Sep '08

23 Sep '08

Hi all, it is possible to use CDRTool 6.6.3 or 6.6.4 with freeradius-cdrtool-1.1 ? The problem is in the code of freeradius-cdrtool.py, where in the code have "DebitBalance From=%s To=%s Duration=%s\n" , "MaxSessionTime From=%s To=%s Duration=%s Lock=1\n" and others, but the newer CDRTool need have a gateway parameter and Call Id parameter. Anybody have a sucess with this versions ? Thanks in advanced, Diego.

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[Kamailio-Users] how to change source number
by BERGANZ François 23 Sep '08

23 Sep '08

Hello, I need to change the source number, to have an international format (e164) but when I use : $avp(s:245)=$fu; avp_subst("$avp(s:245)/$avp(s:246)/d", "/sip:/sip:33/"); uac_replace_from("", "$avp(s:246)"); in fact, my phone have in the screen sip:33..., before, it have my name I think that I have to use another function to change the source number, but which function? Thank you Cordialement BERGANZ François http://www.acropolistelecom.net <http://www.acropolistelecom.net/>

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Re: [Kamailio-Users] nested redirect
by Daniel-Constantin Mierla 23 Sep '08

23 Sep '08

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; >> } >> } >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users(a)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

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[Kamailio-Users] Question about redirect if user is not online
by Ali Jawad 23 Sep '08

23 Sep '08

Hi All I do forward calls to a DID number to my openser box, if my softphone is online it rings directly because I am using the DID number as an alias. However I want the call to be forwarded to my asterisk box if the user is not online. What statement should I use ? THanks With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

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[Kamailio-Users] preparing release 1.4.1
by Daniel-Constantin Mierla 22 Sep '08

22 Sep '08

Hello, I propose to release kamailio 1.4.1 next Thursday, Sep 25, 2008. Please send to devel(a)lists.kamailio.org any critical issues you are aware of, to try to fix them. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Serusers] Handling server with broken record-route handling
by Marcus Better 22 Sep '08

22 Sep '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I have problems with a remote server, outside of my control, that seems to have very broken request URI handling. It is possibly suffering from an Asterisk bug [1] or something similar. On my side I have a UAC (IP address 172.16.1.102) and a SER 2.0 proxy. The UAC sends an INVITE, and the remote UAS ends the dialog with a BYE request. However this BYE returns to my proxy with missing route headers, so SER cannot route it. I need to work around the problem with SER on my side and wonder what the best solution is. Here is the relevant part of the conversation as seen on my proxy 80.81.82.3 (slightly massaged, irrelevant parts cut): INVITE sip:00463112341234@proxy.remote.com SIP/2.0. Record-Route: <sip:80.81.82.3;avp=Z3oDBwBhY2NvdW50AwB5ZXM;lr>. Call-ID: f730a3a6a5c450508274739a8133ae7f(a)172.16.1.102. CSeq: 1 INVITE. To: <sip:00463112341234@proxy.remote.com>. From: <sip:17473605948@proxy.remote.com>;tag=2045565953650740630. Via: SIP/2.0/UDP 80.81.82.3;branch=z9hG4bK761b.b1548ac4.0. Via: SIP/2.0/UDP 172.16.1.102:5060;rport=5060;appname=myapp;branch=z9hG4bK74592b710287e6cfa94b4cf9c10e15bf. Contact: "17473605948" <sip:172.16.1.102:5060;transport=udp>. SIP/2.0 200 OK. Via: SIP/2.0/UDP 80.81.82.3;branch=z9hG4bK761b.b1548ac4.0. Via: SIP/2.0/UDP 172.16.1.102:5060;rport=5060;appname=myapp;branch=z9hG4bK74592b710287e6cfa94b4cf9c10e15bf. Record-Route: <sip:123.12.13.14;lr;ftag=2045565953650740630>. Record-Route: <sip:80.81.82.3;avp=Z3oDBwBhY2NvdW50AwB5ZXM;lr>. From: <sip:17473605948@proxy.remote.com>;tag=2045565953650740630. To: <sip:00463112341234@proxy.remote.com>;tag=as438ee174. Call-ID: f730a3a6a5c450508274739a8133ae7f(a)172.16.1.102. CSeq: 1 INVITE. Contact: <sip:011463112341234@130.94.88.93:5060>. ACK sip:01146739835569@130.94.88.90:5060 SIP/2.0. Record-Route: <sip:80.81.82.3;avp=Z3oDBwBhY2NvdW50AwB5ZXM;lr>. Call-ID: f730a3a6a5c450508274739a8133ae7f(a)172.16.1.102. CSeq: 1 ACK. Via: SIP/2.0/UDP 80.81.82.3;branch=0. Via: SIP/2.0/UDP 172.16.1.102:5060;rport=5060;appname=myapp;branch=z9hG4bK96a96b64dfd5aba5c896c37e0de18850. From: <sip:17473605944@proxy.remote.com>;tag=7737826714334905221. To: <sip:0046739835569@proxy.remote.com>;tag=as543c9f3f. Route: <sip:123.12.13.14;lr;ftag=7737826714334905221. Content-Type: application/sdp. BYE sip:80.81.82.3:5060 SIP/2.0. Record-Route: <sip:123.12.13.14;lr;ftag=as438ee174>. Via: SIP/2.0/UDP 123.12.13.14;branch=z9hG4bK100d.d617f387.0. Via: SIP/2.0/UDP 130.94.88.93:5060;branch=z9hG4bK0256ad7c;rport=5060. Route: <sip:80.81.82.3;avp=Z3oDBwBhY2NvdW50AwB5ZXM;lr>. From: <sip:00463112341234@proxy.remote.com>;tag=as438ee174. To: <sip:17473605948@proxy.remote.com>;tag=2045565953650740630. Call-ID: f730a3a6a5c450508274739a8133ae7f(a)172.16.1.102. CSeq: 102 BYE. I think this BYE is incorrect. It should have the originator's Contact (sip:172.16.1.102:5060;transport=udp) in the request URI and the Record-Routes of the original INVITE in the Route header. Or, if it is a strict router, it could have the current request URI but add the Contact as the last Route header. Now it has destroyed the originator's route information. What is the best way to get SER to remember/reconstruct this last hop to the UAC? Cheers, Marcus [1] http://bugs.digium.com/view.php?id=3609 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkjXkbQACgkQXjXn6TzcAQnhrQCeP9T0pR+AlPJjTr8OCS5nNOAA GBIAnR5pw68rqe1PR00jDLvcYh+eQXPP =sdca -----END PGP SIGNATURE-----

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[Kamailio-Users] Kamailio+Asterisk Need Help
by VIP Carrier 22 Sep '08

22 Sep '08

Hi, I'm trying to setup an HA cluster based of 2 Proxy 4 Asterisk 2 DB's Can some one please help me with this.

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