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19 participants
37571 discussions

[Kamailio-Users] rewriteuser( ) with avp
by Alex R.S.M 31 Aug '09

31 Aug '09

Hi, I'm trying to change "username" field of URI in the INT message. When using rewriteuser() function with avp variable as below, the username is changed to $avp(i:8) itself instead of the value of the avp. rewriteuser("$avp(i:8)"); I've tried all sorts of diff formats and none of them replace username with the value of avp variable.: rewriteuser($avp(i:8)); rewriteuser("avp(i:8)"); is this a bug? Thanks, A-RS _________________________________________________________________ Windows Live: Keep your friends up to date with what you do online. http://windowslive.com/Campaign/SocialNetworking?ocid=PID23285::T:WLMTAGL:O…

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[Kamailio-Users] scope of AVPs (AVP defined when execution of script starts)
by jh＠tutpro.com 29 Aug '09

29 Aug '09

i have always thought that AVPs exits per transaction. that does not seem to be the case, however, because when i put in the beginning of my script xlog("L_INFO", "Caller lcr_id $avp(CALLER_LCR_ID_AVP)\n"); this gets printed to syslog: Aug 29 11:34:34 localhost /usr/sbin/sip-router[10210]: INFO: Caller lcr_id 2 is this the indented behavior? do i really need to delete all AVPs in the beginning of the script in order to make sure that they don't have any values left from processing of previous message? -- juha

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[Kamailio-Users] Phones that support TLS with certificate
by catalina oancea 27 Aug '09

27 Aug '09

Hello, I am trying to find the phones that can use tls with certificate, I mean that the phone doesn't accept incoming calls unless the server certificate is signed by a public authority or manually added to a list. For now my conclusions are: -Snom 360 - works with TLS. But although snom wiki states that a certificate is needed, incoming calls are accepted even if the certificate is not uploaded. -SJPhone - same as snom -Bria - this is the only phone I could find that works only if the certificate is loaded. Has anybody used other phones with tls and can give me a piece of advice? also, maybe I am not configuring Snom right and that's why it doesn't need a certificate, any ideas? Thanks Catalina

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[Kamailio-Users] Perl module and memory leak in append_branch
by James Puckett 26 Aug '09

26 Aug '09

I have followed a bit of the discussions regarding the problems with memory management between the perl modules and the core kamailio software. I need to get append_branch working from within the perl module so that it doesn't cause a memory leak. It's been a while since I did any serious straight C programming, but if someone can point me to the correct way to handle memory allocated with the module so that it gets freed in the core I'd be glad to take this problem on. Specifically, I'm using something that looks like: $m->append_branch("<sip:".$cdpn."\@".$peers{$_}."?From=%3Csip:". $sourcenumber."@".$destIp."%3E;tag=".$fromtag.">", $qval); This leaves about 190 bytes per call unfreed. Thanks. James Puckett - Engineers' Consulting Group Phone 229-316-0012 - Fax 229-316-1490

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[Kamailio-Users] What happens with filled htables?
by catalina oancea 26 Aug '09

26 Aug '09

Hello I use the htable module a lot but the only problem, when I add a new entry in a htable, is when I will delete it. My question is: if the hash table is completely filled and I try to add a new value to it, do I get an error or is an old value automatically deleted to be able to write my new value? If an old value was automatically deleted whenever a new value is added, I wouldn't have to bother deleting the values I no longer need. Thanks Catalina

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[Serusers] translate message from TLS to UDP
by qiulei212 26 Aug '09

26 Aug '09

hi everyone: sorry to trouble you again! I use the SER as a proxy between a SIP-softphone and the asterisk. The message between the sip-phone and SER is TLS and i want to translant the message by UDP to my asterisk.HOW would i do? I have tried the forward_udp , t_relay_to_tls ,t_relay_to_udp and no useless. which function should i prefer? thanks!! qiulei212 2009-08-26

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[Serusers] problems with stun using uas
by lorenzo 25 Aug '09

25 Aug '09

hi! i'm the developer of a sip app that happens to be connecting to a sip server that's using SER. i'm having troublewith the way SER handles incoming calls to my stun-enabled ua (which is placed behind a router) I have two clients running on the same machine. one is my app, the other is Gizmo5. when i make an outgoing call from my app to the user i'm logged in with in Gizmo5, everything works. but when i call my app from the Gizmo5 app, sniffing the network traffic with wireshark, the only thing i see happening is the sip server sending a 100 Giving a try to the calling ua. The reason i'm posting this here, is that i know it's not a bug with my app, since everything works fine when i don't use stun. (note that i don't have direct access to ser's configuration) this is my app's REGISTER request (note that it has a *public ip*): REGISTER sip:proxy01.sipphone.com SIP/2.0 Via: SIP/2.0/UDP 93.148.135.54:5060;rport;branch=z9hG4bK70207 Max-Forwards: 70 To: <sip:asymmetric@proxy01.sipphone.com> From: <sip:asymmetric@proxy01.sipphone.com>;tag=z9hG4bK57274073 Call-ID: 557402852938(a)93.148.135.54 CSeq: 2 REGISTER Contact: <sip:asymmetric@93.148.135.54:5060;transport=udp> Expires: 3600 User-Agent: mjsip stack 1.6 Authorization: Digest username="asymmetric", realm="proxy01.sipphone.com", nonce="xxx", uri="sip:proxy01.sipphone.com", algorithm=md5, response="yyy" Content-Length: 0 and the response: SIP/2.0 200 OK Via: SIP/2.0/UDP 93.148.135.54:5060;rport=12810;branch=z9hG4bK70207 To: <sip:asymmetric@proxy01.sipphone.com>;tag=92390300a369f0d75803e369c733575e.50aa From: <sip:asymmetric@proxy01.sipphone.com>;tag=z9hG4bK57274073 Call-ID: 557402852938(a)93.148.135.54 CSeq: 2 REGISTER Contact: <sip:asymmetric@93.148.135.54:5060;transport=udp>;expires=3600 Content-Length: 0 Gizmo5's app REGISTER request: REGISTER sip:proxy01.sipphone.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:64064;branch=z9hG4bK-d87543-bbc1e2712d417352-1--d87543-;rport Max-Forwards: 70 Contact: <sip:17474492586@93.148.135.54:10251> To: <sip:17474492586@proxy01.sipphone.com> From: <sip:17474492586@proxy01.sipphone.com>;tag=f43be43c Call-ID: b1b4a060512d4b0c1250955005@YXN5bW1ldHJpYy5sb2NhbA.. CSeq: 2 REGISTER Expires: 1800 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER User-Agent: MacGizmo/2.0.02 (Gizmo-s2n1) Authorization: Digest username="17474492586",realm="proxy01.sipphone.com",nonce="xxx",uri="sip:proxy01.sipphone.com",response="yyy",algorithm=MD5 Content-Length: 0 and response: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.100:64064;branch=z9hG4bK-d87543-bbc1e2712d417352-1--d87543-;rport=10251;received=93.148.135.54 To: <sip:17474492586@proxy01.sipphone.com>;tag=92390300a369f0d75803e369c733575e.c70f From: <sip:17474492586@proxy01.sipphone.com>;tag=f43be43c Call-ID: b1b4a060512d4b0c1250955005@YXN5bW1ldHJpYy5sb2NhbA.. CSeq: 2 REGISTER P-Behind-NAT: Yes Contact: <sip:17474492586@93.148.135.54:10251>;expires=1800 Content-Length: 0 in the second case, the client gets the P-Behind-NAT flag set to Yes, and for a good reason. But then why is this all i get when trying to call the first ua? : SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.1.100:64064;branch=z9hG4bK-d87543-7fb07a028db5c137-1--d87543-;rport=10251;received=93.148.135.54 To: <sip:asymmetric@proxy01.sipphone.com> From: <sip:17474492586@proxy01.sipphone.com>;tag=bc135372 Call-ID: 88d41c746d51146e1250955219@YXN5bW1ldHJpYy5sb2NhbA.. CSeq: 1 INVITE P-Behind-NAT: Yes Content-Length: 0 whereas the inverse works: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bKf3b1.bc014575.0 Via: SIP/2.0/UDP 93.148.135.54:5060;rport=13171;branch=z9hG4bK82654 Record-Route: <sip:198.65.166.131;lr;ftag=z9hG4bK90319798> Contact: <sip:17474492586@proxy01.sipphone.com> To: <sip:17474492586@proxy01.sipphone.com>;tag=6ae0e759 From: "asymmetric"<sip:asymmetric@proxy01.sipphone.com>;tag=z9hG4bK90319798 Call-ID: 593542951976(a)93.148.135.54 CSeq: 1 INVITE User-Agent: MacGizmo/2.0.02 (Gizmo-s2n1) Content-Length: 0 CQBM: 244 -- thanks a lot! asymmetric

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Presence over NAT
by Andrew O. Zhukov 25 Aug '09

25 Aug '09

Is there any samples let's forwarded Subscribe fix the Contact: I mean Proxy URI not the customer mapped. Let's the followed NOTIFY go over proxy not P2P. Thanks in avance. Andrew O. Zhukov

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[Kamailio-Users] kamailio 3.0 - the time before freezing
by Daniel-Constantin Mierla 25 Aug '09

25 Aug '09

Hello everybody, during the last IRC devel meeting, we pinned end of summer to be the time to enter testing phase for the new major release - codenamed kamailio 3.0. There was quite a lot of brand new features, considering that most of devel effort was directed to integration. A try to collect new items is available at: http://sip-router.org/wiki/features/new-in-devel Of course, all ser features are available as well, resulting a large amount of new stuff in kamailio 3.0 vs kamailio 1.5.x. See: http://sip-router.org/benefits/ http://www.iptel.org/ser/features I propose to start the testing phase sometime next week. Meanwhile, let's see if something important was forgotten from kamailio core or tm. To my knowledge: - path support in tm - Andrei has it on his todo afaik - in a way or another, it must get in next release - drop compatibility - drop reply in reply_route - internally drop and exit are now differentiated by core, needs review and update of handling after running the routes - on my to-do Apart seas, all Kamailio modules are fully ugraded to new core - I started but lack of test env + busy summer resulted in delays - can be done during testing period, being just api updates. If you have discovered something else, let us know. Thanks, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] exchanging CDR data between Asterisk servers
by Klaus Darilion 24 Aug '09

24 Aug '09

Hi! I have the following setup: PSTN<------>Asterisk<-----SIP-->Asterisk GW/LCR \ \ ... \ \ ... \ --SIP-->Asterisk \ ... ------->Asterisk The GW-Asterisk just does the gatewaying stuff and writes the CDRs for the billing system. The other Asterisk servers handle all the services (IVR, REGISTER, ...) Some scenarios require to write some additional data to the CDRs. For outgoing calls this is not a problem (I signal the extra data in a SIP header and set a CDR() variable in the GW asterisk). My problem are incoming calls. The "extra data" is only known to the service Asterisk, but the CDR is written by the GW Asterisk. Does anybody know a method how to signal the "extra data" during the call from the server Asterisk back to the GW Asterisk und put it into a CDR() variable? Regards Klaus

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