sr-users October 2005

sr-users@lists.kamailio.org

214 participants
457 discussions

[Users] CDRTool call rating "Cannot instantiate non-existent class"

by Cameron Beattie

I have been testing CDRTool and get the following message in the log files when browsing to www.example.com/scripts/ratingEngine.php: Cannot instantiate non-existent class: ratingtables in /var/www/CDRTool/scripts/ratingEngine.php Is this because the non commercial version of CDRTool does not provide rating or is it something I have done wrong in the setup? Any advice would be appreciated. Regards Cameron

18 years, 8 months

[Serusers] rewriting domain

by Maciej Świniarski

i have stupid hardphone, proxy, registrer server is always domain :/ I have configured DNS and ser to use some domain.com as sip domain, but proxy have sip.mydomain.com addres. Registering is no problem because i can tell what domain ser should search but problem is for examplke with aliases i don't want write every time `serctl alias add user(a)domain.com` `serctl alias add user(a)sip.domain.com` if (uri=~".*sip\.my\.domain\.com") { rewritehost("my.somain.com"; } don't work becaus it rewrites requested uri i hope you get my point. Any one know how to solve this problem?

18 years, 8 months

[Serusers] acc+nat

by Maciej Świniarski

hi i have little problem with acc and nat becaus mysql records created with acc module contains ip behind nat. And yes, i use fix_nated_contact befor setting flag for acc module. Any one have some experience with this issue?

18 years, 8 months

Re: [Users] Constructing sip uri from Dialed number and userloc

by Daniel-Constantin Mierla

On 10/12/05 02:54, Gene Willingham wrote: > > > > Not sure if I have all of the terminology right, but I would like to > have Asterisk register with openser. Then when a call comes in I > would like to construct the sip uri using the Dialed number and the > user location. The final uri should be: > > > > Sip:240744XXXX@25.25.25.25. Where the IP address is the user location > of the Asterisk box. > You can play with avpops to get this. You have there avp_write() to backup the original dialed number, or avp_subst() to extract parts of the original dialed number. Then use avp_printf() to concatenate the values, after usrloc lookup (see http://www.voice-system.ro/docs/avpops/) and avp_pushto() to rewrite the r-uri. Cheers, Daniel > > > Any suggestions. > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

18 years, 8 months

[Serusers] BYE method not handled properly

by Sam Lee

I would like to know why does my BYE method are always replied with a 'Call Leg/Transaction does not exist' . How do they compare whether the transaction in the BYE method exist or not ? ( tag? ftag ? ) Are there any thing in the config that might cause this kind of problem ? Just want to highlight that all the calls are made in a good condition, everything except when the call is ending. I have checked that the phones have not received a prior BYE. Any other idea what is wrong ? Here's a more detailed situation :- Caller (PSTN) --> Voice Gateway --> OPENSER --> Callee (UA) When Callee (UA) tried to end the call , OPENSER will forward a copy of the BYE to Voice Gateway to inform him of the BYE. The Gateway , somehow , replied with a 'Call Leg/Transaction Does Not Exist' . The strange thing is, the Caller (PSTN) was somehow informed of the BYE method and terminate the session . Anyone has any idea how to handle these errors ? I will be glad to provide a ngrep for more reference. Please let me know if you dont understand. Regards, Sam

18 years, 8 months

RE: [Users] BYE method accompanied by error

by Sam Lee

I have checked that the phones have not received a prior BYE. Any other idea what is wrong ? Here's a more detailed situation :- Caller (PSTN) --> Voice Gateway --> OPENSER --> Callee (UA) When Callee (UA) tried to end the call , OPENSER will forward a copy of the BYE to Voice Gateway to inform him of the BYE. The Gateway , somehow , replied with a 'Call Leg/Transaction Does Not Exist' . The strange thing is, the Caller (PSTN) was somehow informed of the BYE method and terminate the session . Anyone has any idea how to handle these errors ? I will be glad to provide a ngrep for more reference. Regards, Sam -----Original Message----- From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Iqbal Sent: Tuesday, October 11, 2005 7:35 PM To: Sam Lee Cc: users(a)openser.org Subject: Re: [Users] BYE method accompanied by error Can you check to see if you have already received a BYE for that call, some phones I had were sending there own Bye's after the GW had Iqbal Sam Lee wrote: > Hi all, > > I would like to know why does my BYE method are always replied with a > 'Call Leg/Transaction does not exist' . How do they compare whether > the transaction in the BYE method exist or not ? ( tag? ftag ? ) Are > there any thing in the config that might cause this kind of problem ? > Just want to highlight that all the calls are made in a good > condition, everything except when the call is ending. > > Please let me know if you dont understand. > > Regards, > Sam > >----------------------------------------------------------------------- >- > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

18 years, 8 months

Re: [Users] Constructing sip uri from Dialed number and userloc

by Klaus Darilion

Check out the admin guide! http://www.iptel.org/ser/doc/seruser/seruser.html#URIREWRITINGEXAMPLES You want to use rewritehostport(). You could also use AVPs and avpops module. klaus Gene Willingham wrote: > > > Not sure if I have all of the terminology right, but I would like to > have Asterisk register with openser. Then when a call comes in I would > like to construct the sip uri using the Dialed number and the user > location. The final uri should be: > > > > Sip:240744XXXX@25.25.25.25. Where the IP address is the user location > of the Asterisk box. > > > > Any suggestions. > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users

18 years, 8 months

[Serusers] Forward to PSTN gateway question

by Alejandro G

Hi, I have the following configuration: 192.168.0.101 192.168.0.1 Public IP <---> Asterisk <------------------------------> SER <---> Public IP <-----> SIP Phone as PSTN GW I use mediaproxy for dealing with NAT, everything works fine. We have 2 internet connections: one over Asterisk with low bandwith, and the other over SER with more bandwidth. The SIP phone is Sipura 2100. The routing block I use in INVITE is something liske this: route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("192.168.0.1"); # GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); append_hf("P-hint: GATEWAY\r\n"); # ----------------------------------------------------------------- # RTP Proxy Enabler # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { # flags set by client_nat_test() use_media_proxy(); }; route(1); } When the user with the SIP phone terminates in PSTN via Asterisk, SER forwards to Asterisk but Asterisk then * contacts rtp streams directly to SIP phone bypassing SER and using the low bandwidth connection. My knowledge is poor about SER and SIP issues but I believe that * tries to contact to the UA specified in the Contact header. Is this correct? Is anyway SIP phone could terminate in PSTN connecting using SER and private LAN to connect to asterisk instead of using asterisk's public IP? Sorry, I'm not sure that my questions is clear enough. Thanks. Alejandro Ghergherian

18 years, 8 months

Re: [Serusers] Please help on terminating active sessions!!!

by Ryan Pagquil

Hi Samuel, I'm getting these (To: From: Call-ID) informations from the database, since I'm accounting those messages in mysql database, will it help? Does this method will terminate active sessions for both side? I mean the caller and the callee will be terminated when I do this? Is there any other way to terminate calls instead of sipsak? I want to try all available option just to do this. Please help.... Thanks, Ryan Samuel Osorio Calvo wrote: >I guess you change To:, From: and Call-ID headers every time you want to tear down a concrete session.....these headers change for every session and you will have to take the values from the message exchange creating the session. (INVITE/OK/ACK). >Moreover, it looks weird that the To,From, and Contact represents the same identity while the req-uri is another...... > >Hope you can use these hints to improve the so-wanted pre-paid stuff for SER ;) > >Samuel. > > >Unclassified. > > >>>><rpagquil(a)philonline.com> 10/11/05 12:52PM >>> >>>> >>>> >Hi guys, >I'm having a big problem on how can I terminate active sessions... I'm currently using sipsak to terminate it but all tests are not successfull. > >Here is what I do: >my bye file is like this: > >BYE sip:rpagquil@202.84.24.126:5060 SIP/2.0 >To: <sip:acjeff@sip.philonline.com>;tag=3986313335 >From: <sip:acjeff@sip.philonline.com>;tag=1362465910 >Contact: <sip:acjeff@202.84.24.126:5060> >Call-ID: 4D161E9C-3A79-11DA-BFB7-00E04CAB4AB4(a)10.0.1.144 >CSeq: 7 BYE >Max-Forwards: 16 >User-Agent: X-Lite release 1103m >Content-Length: 0 > >here im trying to terminate these two calls. >and here is what my command in sipsak: > >sipsak -vv -f byemes -s sip:sip.philonline.com > >But those things aren't working with me... any suggestions or any other methods are applicable with what i'm doing? > >I'm doing this so that those users that have no credits left are disconnected automatically...I'm trying to do a prepaid service for SER. > >Please help on this matter.... > >I really need your help... > >Thanks, > >Ryan Pagquil >Infodyne Inc. (www.philonline.com) >Tel. (632)-6870715 > > > > > -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

18 years, 8 months

[Users] Constructing sip uri from Dialed number and userloc

by Gene Willingham

Not sure if I have all of the terminology right, but I would like to have Asterisk register with openser. Then when a call comes in I would like to construct the sip uri using the Dialed number and the user location. The final uri should be: Sip:240744XXXX@25.25.25.25. Where the IP address is the user location of the Asterisk box. Any suggestions.

18 years, 8 months

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sr-users October 2005