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37461 discussions

Re: [Serusers] Caller-ID & Remote-Party-ID example
by Greger V. Teigre 22 Jun '06

22 Jun '06

I'm not very familiar with freeradius, but in general there is only one dictionary, sometimes with some sort of #include-support if you want to split up the dictionary. g-) Kostas Marneris wrote: > thanks for your help. > I hadn't understand all the Radius issues.. > > I should add the following line on FreeRadius Dictionary . > ATTRIBUTE Sip-Rpid 213 string # Proprietary, auth_radius > > Is this the proper way or should I have a separate dictionary file ? > > > thank you. > > > Greger V. Teigre wrote: > >> The SIP-Rpid attribute will be returned as part of the RADIUS auth if >> present (0.9.x). In 0.10.x I believe it is SIP-AVP=rpid:value >> This is the "saved value". >> g-) >> >> Kostas Marneris wrote: >> >> >>> Hello all, >>> >>> I'm trying to get 'caller-id with Remote-Party-ID' work, >>> but I think I'm missing something.. >>> >>> What is the "saved value of the SIP URI received from radius" ? >>> (according to auth/README file). >>> >>> >>> What is supposed to be the RADIUS attribute/value in RADIUS user profile ? >>> >>> >>> Can someone send an example ? >>> >>> >>> thank you. >>> _______________________________________________ >>> Serusers mailing list >>> Serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >>> >>> >> >> > >

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RE: [Serusers] How to send instant message programmatically
by srinivas.sriram＠wipro.com 22 Jun '06

22 Jun '06

Hello Ilker, I am writing to address of the sip message to fifo and from the fifo call sipp instead of mysql. But , I do not now how to send message from sipp when the user is online. can you please help in sending message from sipp to user agent(to address). Thanks & Regards Sriram Srinivas ________________________________ From: İlker Aktuna (Koç.net) [mailto:ilkera@koc.net] Sent: Monday, June 19, 2006 7:06 PM To: Srinivas Sriram (WT01 - IP-Multimedia Carrier & Ent Networks) Subject: RE: [Serusers] How to send instant message programmatically Hi Sriram, I am also interested in sending missed calls as IM when the user becomes online. Do you know how to write missed calls to a mysql table ? If that's done, we can send messages via sipp when the user is online. Please share with me your findings... Regards, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of srinivas.sriram(a)wipro.com Sent: Monday, June 19, 2006 3:37 PM To: serusers(a)iptel.org Subject: RE: [Serusers] How to send instant message programmatically Hi Greger I have checked with sipp , with that we can send messages. But What is my requirement is i have write the code for constructing SIP "MESSAGE" message for a given sip uri. This I have to call form one of the ser module. Total what I am trying is to send a IM message when there is a missed call for a particular SIP uri. In SER there is a code for writing into the data base of the sip uri , when there is a missed call for that Uri. At this point we have to send IM message. Thank you, Regards, Sriram Srinivas. ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Monday, June 19, 2006 5:35 PM To: Srinivas Sriram (WT01 - IP-Multimedia Carrier & Ent Networks) Cc: serusers(a)iptel.org Subject: Re: [Serusers] How to send instant message programmatically You are better off generating the message outside SER. SER is not a UAS. Have a look at sipsak or sipp (both found at sourceforge, I think). g-) srinivas.sriram(a)wipro.com wrote: Hello, I am using SER-0.9.6. How to send instant message programmatically, how to build sip message and which function can by used(like t_uac, t_request). Can you help where and which to be changed in ser code. Thank you, Regards, Sriram Srinivas. 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 proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ________________________________ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers 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 proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com <http://387555.sigclick.mailinfo.com/sigclick/000A0702/02074C00/0305054B/051…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________ 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 proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

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[Users] Any way to steal Media IP/Port from RTPProxy or MediaProxy
by Hoa Thai Duy 22 Jun '06

22 Jun '06

Hi List I want to get to c= and m= value after use_media_proxy or force_rtp_proxy (after real RTP flow between UA and media/rtpproxy) I want this in order to steal this pair of information, and bypass the RTPProxy/MediaProxy and use this information for UA to talk with other application server (eg. Asterisk) Pls. help Brgds

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[Users] Differentiating between SIP and PSTN invites
by Kenny Chua 22 Jun '06

22 Jun '06

Hello, I was wondering, how should I implement this. I would like the user to press the #key first if they would like to dial SIP 2 SIP calls and just leave PSTN calls as it is? How am I suppose to do that? I have this problem too, which whenever I hangup, the other side doesn't gets disconnected. What should I check for to enable hang up on both sides if one sides suddenly hangs up? Thank you. --------------------------------- Want to be your own boss? Learn how on Yahoo! Small Business.

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RE: [Serusers] checking source ip and dest ip
by İlker Aktuna (Koç.net) 22 Jun '06

22 Jun '06

Thank you; that was what I'm looking for. How can I match i.e IP block 10.224.0.0 - 10.255.255.255 ? Is there an easy way or should I list them with || as in : (uri=~"^.*@10\.224\..* <mailto:uri=~^.*@10\.224\..*> " || uri=~"^.*@10\.225\..*" || .....) Thanks, ilker ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Wednesday, June 21, 2006 12:50 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] checking source ip and dest ip Check the request uri after lookup, right before t_relay(). (uri=~"^.*@10\..*" <mailto:^.*@10\..*> ) As long as you don't have nated clients (and thus don't use fix_nated_register()) the request uri will contain the actual IP address. g-) İlker Aktuna (Koç.net) wrote: Hi, In fact my network is not 10.0.0.0 ; that was just an example (bad example). The network addresses are specific public network addresses (lets say 3 different C Class network) and has no relation to NAT. That's why I need to check the IP adrresses (from and to) of the SIP message before deciding if it should use mediaproxy or not. Any idea how to do that ? Soruce address can be checked as : if (src_ip="195.1.1.+") { } but that won't be same for destination address as "dst_ip" is the SER server ip. How can I check the ip address of the client that the SIP message will be forwarded ? Thanks, ilker ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Tuesday, June 20, 2006 10:26 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] checking source ip and dest ip Yes, and your network is in the RFC1918 (private) address space, so you can use the same logic (even without change as you want to proxy when the address is on your network). g-) İlker Aktuna (Koç.net) wrote: In fact, I am not looking for "if address is NATed or not", I need to check if address belongs to my network or not. what do you mean by RFC1918 addresses ? Thanks, ilker ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Tuesday, June 20, 2006 1:40 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] checking source ip and dest ip Why not test for RFC1918 addresses? That's what the onsip.org nat-handling scripts do. A NATed destination is determined by a flag in the location record and the src ip is checked with nat_uac_test(). It will give a more robust NAT handling that testing on IP addresses for parts of the RFC1918 addresses. The problem has already been solved, why not reuse? ;-) g-) İlker Aktuna (Koç.net) wrote: Hi, I need to check source ip of a SIP message and also destination IP to which the message will be sent. i.e. I will use mediaproxy only if the one of the ip addresses match 10.0.0.0 / 255.0.0.0 is it possible ? Thanks, ilker C "-//W3C//DTD HTML 4.0 Transitional//EN"> In fact, I am not looking for "if address is NATed or not", I need to check if address belongs to my network or not. what do you mean by RFC1918 addresses ? Thanks, ilker ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Tuesday, June 20, 2006 1:40 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] checking source ip and dest ip Why not test for RFC1918 addresses? That's what the onsip.org nat-handling scripts do. A NATed destination is determined by a flag in the location record and the src ip is checked with nat_uac_test(). It will give a more robust NAT handling that testing on IP addresses for parts of the RFC1918 addresses. The problem has already been solved, why not reuse? ;-) g-) İlker Aktuna (Koç.net) wrote: Hi, I need to check source ip of a SIP message and also destination IP to which the message will be sent. i.e. I will use mediaproxy only if the one of the ip addresses match 10.0.0.0 / 255.0.0.0 is it possible ? Thanks, ilker C "-//W3C//DTD HTML 4.0 Transitional//EN"> Hi, In fact my network is not 10.0.0.0 ; that was just an example (bad example). The network addresses are specific public network addresses (lets say 3 different C Class network) and has no relation to NAT. That's why I need to check the IP adrresses (from and to) of the SIP message before deciding if it should use mediaproxy or not. Any idea how to do that ? Soruce address can be checked as : if (src_ip="195.1.1.+") { } but that won't be same for destination address as "dst_ip" is the SER server ip. How can I check the ip address of the client that the SIP message will be forwarded ? Thanks, ilker ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Tuesday, June 20, 2006 10:26 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] checking source ip and dest ip Yes, and your network is in the RFC1918 (private) address space, so you can use the same logic (even without change as you want to proxy when the address is on your network). g-) İlker Aktuna (Koç.net) wrote: In fact, I am not looking for "if address is NATed or not", I need to check if address belongs to my network or not. what do you mean by RFC1918 addresses ? Thanks, ilker ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Tuesday, June 20, 2006 1:40 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] checking source ip and dest ip Why not test for RFC1918 addresses? That's what the onsip.org nat-handling scripts do. A NATed destination is determined by a flag in the location record and the src ip is checked with nat_uac_test(). It will give a more robust NAT handling that testing on IP addresses for parts of the RFC1918 addresses. The problem has already been solved, why not reuse? ;-) g-) İlker Aktuna (Koç.net) wrote: Hi, I need to check source ip of a SIP message and also destination IP to which the message will be sent. i.e. I will use mediaproxy only if the one of the ip addresses match 10.0.0.0 / 255.0.0.0 is it possible ? Thanks, ilker C "-//W3C//DTD HTML 4.0 Transitional//EN"> In fact, I am not looking for "if address is NATed or not", I need to check if address belongs to my network or not. what do you mean by RFC1918 addresses ? Thanks, ilker <http://387555.sigclick.mailinfo.com/sigclick/070C0F04/02074D00/0507084D/051…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

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[Users] help me .
by raviprakash sunkara 22 Jun '06

22 Jun '06

hi user... What means " t_newtrans :- transaction already in process " and " I'm terribly sorry, server error occurred ". How to resovle this.... Please help me... -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com <b>

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[Users] OpenSER - one year old
by buxalex＠libero.it 22 Jun '06

22 Jun '06

Hello to everybody, I would like to say.... Happy Birhtday Openser!!!! :O) Sometimes I got answers with delay, sometimes I spent nights trying to understand how to configure a module, or even a single statement, somrtimes I needed to read ten times the same document because it was unclear related to my knolwledge and still I want to thank the community for this great product. Every day I try to read all the emails coming from the community, and every day I learn a little bit more (threads, ideas, suggestions,solutions...). By the way, there is any open source for a sip softphone in windows (minisip even doesnt start on my laptop with windows 2000) :o)) thanks Alessandro

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[Serusers] Mediaproxy has nothing to do with codecs..
by Dave 22 Jun '06

22 Jun '06

Can someone please clarify whether the statement in Subject is correct? Does mediaproxy not simply take a RTP stream and "proxy" it in both directions without really caring about codecs? Codec negotiation is done between endpoints and mediaproxy does not come in the picture. If the above is true, does it mean that if two SIP user agents support video and we use mediaproxy, it won't be an issue? Please educate me the what if any relationship mediaproxy has with codecs... Dave __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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Re: [Users] 499 Not Acceptable here
by Kenny Chua 22 Jun '06

22 Jun '06

Thanks Andreas, that was really the problem. Really got to watch out for codecs. --------------------------------- Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail Beta.

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Re: [Users] keep alive using RE-INVITE
by Javier Ramirez 21 Jun '06

21 Jun '06

Bogdan, I don't understand how to make this with openser. but I understand the keepalive mechanism can you help me? best regards Javier Ramirez ----- Original Message ----- From: Bogdan-Andrei Iancu To: Javier Ramirez Sent: Wednesday, June 21, 2006 1:35 PM Subject: Re: [Users] keep alive using RE-INVITE Javier, I guess the RFC3261 or google (SIP + reINVITE) are the best source for looking into. regards, bogdan Javier Ramirez wrote: > Bogdan, > I read that a lot of person use the RE-INVITE to keepalive the call > but I don´t understand this. > please, can you help me ? > best regards > Javier Ramirez > > > ----- Original Message ----- > *From:* Bogdan-Andrei Iancu <mailto:bogdan@voice-system.ro> > *To:* Javier Ramirez <mailto:jvr_78@yahoo.com.ar> > *Cc:* users(a)openser.org <mailto:users@openser.org> > *Sent:* Thursday, June 08, 2006 7:14 AM > *Subject:* Re: [Users] keep alive using RE-INVITE > > Hi Javier, > > this is quite impossible to do - by sending re-INVITEs from a middle > point, you will break the CSEQ consistency between the end points. > Within the dialog, the cseq must increase from message to message > - if > you send a re-invite to point A with cseq n+1, after a while point > B may > sent a request with the same cseq since it was not aware that some > party > in the middle changed the number. > > the best way to do it is if you have a end2end keepalive mechanism as > Session-Timer. > > regards, > bogdan > > Javier Ramirez wrote: > > > Hi ! > > I looking for a mechanism to keep alive, and I see that is used the > > RE-INVITE, but > > How to use the Re-INVITE to a keep alive mechanism ? > > exist another way to implement another keep alive mechanism ? > > > > best regards > > Javier ramirez > > > >------------------------------------------------------------------------ > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org > >http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > __________ NOD32 1.1586 (20060608) Information __________ > > This message was checked by NOD32 antivirus system. > http://www.eset.com > __________ NOD32 1.1611 (20060620) Information __________ This message was checked by NOD32 antivirus system. http://www.eset.com

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