sr-users April 2004

sr-users@lists.kamailio.org

138 participants
266 discussions

RE: [Serusers] Call Parking
by Kapil Dhawan 02 Apr '04

02 Apr '04

Thx yes i got this link and thinking of working on it too regards >From: "Franz Edler" <franz.edler(a)utanet.at> >Reply-To: <franz.edler(a)utanet.at> >To: "'Kapil Dhawan'" <sersavvy(a)hotmail.com> >Subject: RE: [Serusers] Call Parking >Date: Fri, 2 Apr 2004 14:05:18 +0200 > >From: Of Kapil Dhawan Sent: Friday, April 02, 2004 8:31 AM > > > > How can i implement Call Parking > >Look at: draft-ietf-sipping-service-examples-06 / 2.16 Call Park >It shows how call parking can be implemented using REFER. > > > How can i implement it in SER. is it already there... > >This is a different role (not SIP proxy) involving UA and media server >functionality. I think this is not the mainstream functionality of SER. >You should better look at Media Servers supporting Call Parking. > >Franz > _________________________________________________________________ Buzz on your screen! Download on your screen. http://www.msn.co.in/Download/screensaver/ Keep yourself smiling!

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[Serusers] BUG: module nathelper and tcp_main_loop: dead child 5
by jimmy huang 02 Apr '04

02 Apr '04

Hi, I would like to try use the module nathelper with rtpproxy to test when INVITE , voice (rtp) can pass nat and firewall, just like (http://www.informatik.uni-bremen.de/~prelle/terena/cookbook/main/ch04s07.ht… 4.7.3.5.1. SIP Express Router mentioned), but ser crashes after receive REGISTER but after REGISTER , ser crashe... ================================================================= syslog: BUG: 15 (23183) tcp_main_loop: dead child 5 0 (23168) child process 23173 existed by a signal 11 0 (23168) core was not generated 0 (23168) INFO:terminating due to SIGCHILD ... ... ================================================================= i install ser0.8.12 , and download newest nathelper , rtpproxy form berlios cvs Makefile rev 1.6 nathelper.c rev 1.51 nathelper.cfg rev 1.2 nathelper_rtpp.cfg rev 1.1 nhelpr_funcs.c rev 1.7 nhelpr_funcs.h rev 1.2 rtpproxy main.c rev 1.17 ... ... My environment is kphone 3.14 -------> Gateway(NAT+Firewall) ----Internet----Ser + rtpproxy kphone use UDP , Symmetric Signaling and Symmetric Media , not use STUN server my ser.cfg is the same as the link (http://www.informatik.uni-bremen.de/~prelle/terena/cookbook/main/ch04s07.ht… 4.7.3.5.1. SIP Express Router mentioned) besides one line if (method == "REGISTER" || ! search("^Record-Route:")) { ===> ser said it is wrong , i don't what's wrong...? so i modify it to if (method == "REGISTER") { My question is : 1. Does the idea (http://www.informatik.uni-bremen.de/~prelle/terena/cookbook/main/ch04s07.ht… 4.7.3.5.1. SIP Express Router mentioned) work? 2. used ser 0.8.12 and only update nathelper module to newest work? Thanks Jimmy

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[Serusers] help me !!!
by Kiem Hung 02 Apr '04

02 Apr '04

Now I'm trying config ser with authenticate module . Can you help me how to do this . Could you tell me step to config ser Thanks alots ________________________________________________________________________ Yahoo! Messenger - Communicate instantly..."Ping" your friends today! Download Messenger Now http://uk.messenger.yahoo.com/download/index.html

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[Serusers] RE: [Serdev] Rmoving to-tags
by Jason Penton 02 Apr '04

02 Apr '04

Hi Jiri Thanks for the advice - it works perfectly now. I must say that out of the many Uas I have played with - more of them screw the tags up than less. Thanks again for the help. Cheers Jason > -----Original Message----- > From: Jiri Kuthan [mailto:jiri@iptel.org] > Sent: 01 April 2004 05:50 PM > To: Jason Penton > Subject: RE: [Serdev] Rmoving to-tags > > At 09:22 AM 4/1/2004, Jason Penton wrote: > > > >> -----Original Message----- > >> From: Jiri Kuthan [mailto:jiri@iptel.org] > >> Sent: 01 April 2004 09:17 AM > >> To: Jason Penton > >> Subject: RE: [Serdev] Rmoving to-tags > >> > >> At 09:10 AM 4/1/2004, Jason Penton wrote: > >> >Hi Jiri > >> > > >> >Yes - I have. BUT I was under the impression that the 180 RINGING > >> >response is not REQUIRED to have a to-tag i.e. to set up an > >> early dialog. > >> > > >> >Is this understanding correct? > >> > >> indeed, to-tags serve the purpose if establishing a dialog. > >> 180 does not establish any, only 183 does. > > > >So then to fix the problem I am having - I could > theoretically remove > >the to-tags from all incoming 180 RINGING responses before > forwarding > >them to the UAC and my problem should be solved. > > you could, but it sounds to me like a too terrible hack. > > >Now my second quesiton is where in > >the SER code can I do this (first assumption is in the tm module > >??????) > > now, message mangling is done in textops. You need to set up > stateful processing (t_on_reply), process replies, look at > which you wish to mangle and eventually mangle them. > > -jiri > > >

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RE: [Sip-implementors] [Serdev] Re: [Serusers] Forking Proxy
by nataraju.alilaghatta＠wipro.com 02 Apr '04

02 Apr '04

Thanks, Nataraju A.B. -----Original Message----- From: sip-implementors-bounces(a)cs.columbia.edu [mailto:sip-implementors-bounces@cs.columbia.edu] On Behalf Of Juha Heinanen Sent: Thursday, April 01, 2004 2:47 PM To: Klaus Darilion Cc: serusers(a)lists.iptel.org; sip-implementors(a)cs.columbia.edu; serdev(a)lists.iptel.org Subject: [Sip-implementors] [Serdev] Re: [Serusers] Forking Proxy Klaus Darilion writes: > UA1 behaves wrong - per RFC 3261 the dialog will be established by the > 200 OK message (not by 180), therefore the to-tag from the 200 OK must > be used. klaus, can you point out the section/paragraph in rfc 3261 that tells that the to-tag from the 200 ok must be used. i scanned the document for a while i didn't find it. the closest i found is this: [ABN] the base line for to-tag handling in UAC is that the new 1xx or 2xx might match for an existing dialog or create a new dialog depending on the to-tag value. If the to-tag in 2xx matches the one in dialog block then the dialog will transitioned to confirmed state and no tag updation happens effectively same tag used in further transactions in that dialog. If the To-Tag does not match any of the existing dialogs for that call then it creates a new dialog and stored in call block then that tag will be used in further transactions in that dialog which in turn part of the call. 13.2.2 Processing INVITE Responses If the dialog identifier in the 2xx response matches the dialog identifier of an existing dialog, the dialog MUST be transitioned to the confirmed state, and the route set for the dialog MUST be recomputed based on the 2xx response using the procedures of Section 12.2.1. but it only mentions about recomputing the route set, not the rest of the dialog state. -- juha _______________________________________________ Sip-implementors mailing list Sip-implementors(a)cs.columbia.edu http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

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[Serusers] Call Parking
by Kapil Dhawan 02 Apr '04

02 Apr '04

How can i implement Call Parking Definition: Call parking allows you to park a call and pick it up at another location. Lets say 111 dialed 222 and now 222 somehow parked the call using some dial code (e.g 150) and now 222 can reach any other phone and can pick up the call by dialing call parking answer code (e.g. 151) + extension from where he parked.... How can i implement it in SER. is it already there... _________________________________________________________________ Easiest Money Transfer to India. Send Money To 6000 Indian Towns. http://go.msnserver.com/IN/42198.asp Easiest Way To Send Money Home!

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[Serusers] Re: SER and Asterisk problem
by Welesley Sibelson Dias 01 Apr '04

01 Apr '04

put notransfer=yes into asterisk e don't work I see the SER log e receive this message t_reply: ACKs are not replied. On Wed, 2004-03-31 at 15:48, Welesley Sibelson Dias wrote: > Hi All. > I'am using Asterisk with SER. I can make call between two internal VoIP > gateways or from na internal to external VoIP gateway. But when I get a > external call, this call hang ups 5 seconds after and I reveive the > following messages > > *CLI> -- Executing Dial("SIP/16008-3d17", > "SIP/16007&SIP/16006|20|tr") in new stack > -- Called 16007 > -- Called 16006 > -- SIP/16007-8c24 is ringing > -- SIP/16007-8c24 answered SIP/16008-3d17 > -- Attempting native bridge of SIP/16008-3d17 and SIP/16007-8c24 Mar > 30 13:53:11 WARNING[1125685952]: chan_sip.c:495 > retrans_pkt: Maximum retries exceeded on call > 2eb06a983415436d4f2845a44dd9df5a(a)192.168.0.252 for seqno 102 > (Request) > Mar 30 13:53:12 WARNING[1125685952]: chan_sip.c:495 > retrans_pkt: Maximum retries exceeded on call > 2eb06a983415436d4f2845a44dd9df5a(a)192.168.0.252 for seqno 102 > (Request) > Mar 30 13:53:12 WARNING[1125685952]: chan_sip.c:495 > retrans_pkt: Maximum retries exceeded on call > D9A3-E113-71478237A5B7568-7@Octtel for seqno 8 (Response) > =3D=3D Spawn extension (sip, 1000, 1) exited non-zero on > 'SIP/16008-3d17' > Jadylson da Rocha Passos Bomfim > > > Redevox Telecom > > Uberlandia +55 34 3234-7813 > > S=E3o Paulo +55 11 5055-6888 > > M=F3vel +55 34 9103-6854 > >

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[Serusers] NAT and multiple proxy
by zeusng 01 Apr '04

01 Apr '04

Consider the following scenario: UA---NAT---Proxy1---Proxy2+UsrLoc User agent UA is behind NAT which send a REGISTER request to Proxy2 but was instructed to use Proxy1 as the first hub. Which of the two proxies should rewrite the contact header. Note that Proxy1 does nothing but to forward all requests to Proxy2, something like: Route[0] { check (max_forward_header & message_len); record_route(); rewritehostport(proxy2:port); t_relay(); } Zeus Ng

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[Serusers] ser -ED option is not working
by siva kumar 01 Apr '04

01 Apr '04

hi friends, my ser -ED option is not working,that means when i am using /etc/ser/ser start is working and my phone is regisrered.but when i am usning "ser -ED" phone is not connecting. plz. tell me what's the problem.before using i used ser -f = ser.cfg also at /etc/ser. thank u _________________________________________________________________ Apply to 50,000 jobs now. http://go.msnserver.com/IN/45531.asp Post your CV on naukri.com today.

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[Serusers] question about domain name
by zhou jie 01 Apr '04

01 Apr '04

Hi, i don't know if it is a bug in ser. I found if i use domain (realm) name with minus char "-", UA can not rigister correctly. i.e. if realm=xx.xx-xxx.xx it can not register. but without "-" it works fine. zhou

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[Serusers] Forking Proxy
by Jason Penton 01 Apr '04

01 Apr '04

Hi All I have a question about the CORRECT operation of UA when making a call via a forking proxy: Lets say UA1 calls 7000(a)sip.com and user 7000(a)sip.com is available at UA2 and UA3 i.e. the Proxy forks the request. UA1 Forking UA2 UA3 Proxxy 1 |------INVITE---------->| | | 2 | |--------INVITE-------------->| | 3 | |---------------------------INVITE------------->| 4 | |<---RINGING(totag=1234)------| | 5 |<---RINGING(totag=1234)| | | 6 | |<------------------RINGING(totag=5768)---------| 7 |<---RINGING(totag=5678)| | | | | | | | | | | NOW UA3 WILL ANSWER | |<----------------200 OK (totag=5678)-----------| 8 |<-200 OK (totag=5678)--| | | 9 |---ACK (totag=1234)--->| | | 10 | |------------------ACK (totag=1234)------------>| AT THIS STAGE UA 3 IGNORES THE ACK AS IT DOES NOT CORRESPOND TO ORIGINATING 200 OK AND THE CALL IS NOT SETUP * UA1 is using the to-tag of the first 180 RINGING it received (frame 5) no matter what the to-tag in the 200 OK is * My question here is: who is in the wrong???? The proxy or UA1? * Should the proxy change the to-tag of the ACK before forewarding it to UA3??? Any help/guidance would be much appreciated Jason Penton Rhodes University Grahamstown South Africa

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[Serusers] Problems with serweb
by Roberto Hiribarne Guedes 01 Apr '04

01 Apr '04

Hi, I configured serweb very well, but when I can't login any valid user ... Neither in admin pages ... It says its an invalid user .. The register pages works until the confirmation ... With this error ... ------- Warning: fopen("/tmp/ser_fifo", "w") - Permission denied in /var/www/html/functions.php on line 206 fie.uclv.edu.cu User Management sorry -- cannot open write fifo ------- My /etc/init.d/ser script doesn't work at boot time ... So a run it by hand ... Do I have to run it as another user .. (a ser user for example ?) My apache is running under the apache user ... What abaut the permissions of /tmp/ser_fifo ... Thanks ... Robert

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[Serusers] RE: [Sip-implementors] FW: Forking Proxy
by sreeram.kanumuri＠wipro.com 01 Apr '04

01 Apr '04

Hi Jason Penton, >>* My question here is: who is in the wrong???? The proxy or UA1? UA1 is wrong. >>* Should the proxy change the to-tag of the ACK before forewarding it to UA3??? Proxy should just Forward the message .Proxy should not change the to-tag. Thanking you, Regards, Sreeram Phone: 8520408 Ex:3208 -----Original Message----- From: sip-implementors-bounces(a)cs.columbia.edu [mailto:sip-implementors-bounces@cs.columbia.edu] On Behalf Of Jason Penton Sent: Thursday, April 01, 2004 1:29 PM To: sip-implementors(a)cs.columbia.edu Cc: serdev(a)lists.iptel.org; serusers(a)lists.iptel.org Subject: [Sip-implementors] FW: Forking Proxy Hi All Sorry about the last post - I have attached the call flow as an image this time :-) I have a question about the CORRECT operation of UA when making a call via a forking proxy: (see attached image for call flow) Lets say UA1 calls 7000(a)sip.com and user 7000(a)sip.com is available at UA2 and UA3 i.e. the Proxy forks the request. * UA1 is using the to-tag of the first 180 RINGING it received (frame 5) no matter what the to-tag in the 200 OK is * My question here is: who is in the wrong???? The proxy or UA1? * Should the proxy change the to-tag of the ACK before forewarding it to UA3??? Jason Penton Rhodes University Grahamstown South Africa Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

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RE: [Serusers] msn messenger is not registering with ser
by Ralph.Wabel＠swisscom.com 01 Apr '04

01 Apr '04

Hi What version of MSN are you using? Because MSN Messenger 5.0 and above does not work with SIP -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Ansari, Fahriz Sent: Mittwoch, 31. März 2004 20:24 To: serusers(a)lists.iptel.org Subject: [Serusers] msn messenger is not registering with ser Hi, I have ser working fine with all other products with authentication. I have problem with MSN messenger. It is sending request to ser more than about 15 times but ser is not registering. SER is even not responds with any type of error message. Can somebody help me please? Thanks, Fahriz Aqeel Ansari

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[Serusers] Removing to-tags
by Jason Penton 01 Apr '04

01 Apr '04

Hi All Is there a way/hack to remove to-tags from incoming '180 Ringing' responses before forwarding them to the UAC. My reason for doing this is that one of the UACs I am using seems to screw the to-tag up when SER acts as a forking proxy. i.e. the calling UA sends an ACK for the 200OK with the to-tag equal to the to-tag received from the FIRST '180 RINGING' received. The net result is that the ACK is sent with a different to-tag than the to-tag of the received 200 OK. The answering UA therefore does not accept the ACK and the call is not completed. Any help/guidance would be much appreciated Jason Penton Rhodes University Grahamstown South Africa

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[Serusers] Asking for advice
by Roberto Hiribarne Guedes 01 Apr '04

01 Apr '04

Hi, I'm new in IP Telephony world and I began from the beginning using SER ... I have found it very nice, simple to configure/use ... Althoug I've had some problems in redhat9. Ok, some days ago I was talking to a friend of mine who was investigating on VoIP and tall me about VOCAL from Vovida.org and Asterisk ... I have read something about Asterisk in this mailing list but I hadn't know of VOCAL ... Could any one tell me about the open source SIP platforms in the todays world ? Their differences ... Etc ... And also the Oparating Systems usually used for this ... (I work with RedHat Linux 9) I'm working for a really good solution for my University VoIP platform ... Thanks, Robert.

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sr-users April 2004