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37538 discussions

Re: [Users] register more than one username by IP phone
by Klaus Darilion 27 Mar '07

27 Mar '07

forgotten to Cc the list ... Klaus Darilion wrote: > I think what you need are aliases. An alias maps from one SIP URI to > another. > > e.g. sip:kd@domain.com --> sip:klaus.darilion@domain.com > sip:+4315056416@domain.com --> sip:klaus.darilion@domain.com. > > Thus, if openser receives a request, you first lookup the alias and then > lookup the location table. You can either use the aliases table and use > lookup("aliases") or use the ALIAS_DB module. > > regards > klaus > > Bodin Bruno wrote: >> Klaus Darilion a écrit : >>> The only identifier which is REGISTERed is the current location of >>> the SIP client (ipaddress:port). This gets mapped to the SIP AoR (the >>> user's SIP address). >>> >>> A user can use several clients concurrently. Then this user will show >>> up multiple times in the location table - one entry for each SIP client. >>> >>> Hope this helps - I still don't understand what you want to achieve. >>> >>> regards >>> klaus >>> >>> Bodin Bruno wrote: >>>> Hi, >>>> Is it possible to register two (or more) identifiers by user, such >>>> as phone number, username or service ? >>>> Thank >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users(a)openser.org >>>> http://openser.org/cgi-bin/mailman/listinfo/users >> Hi, >> In some steps, >> >> what I do : >> >> step 1. Softphone clients are registered in openser with >> authentification (digest etc....) >> step 2. Openser use radius to check them (with ldap) >> step 3. Openser do save("location") to store softphone address. >> >> What I need : >> >> after step 3, authenticate username, and phone number of client are >> usable to call them. >> >> What I don't need : >> >> currently, the username used to call a softphone is the only choice of >> this user and this choice isn't limited. >> >> I'm sorry for my poor english, i'm french student. >> >> thank for your help >> >> >> >> >

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[Serusers] Jabber w/ Ser
by SER LIST 27 Mar '07

27 Mar '07

Hello, has anyone successfuly worked with Jabber module? Especially forwarding messages to Hotmail, yahoo, etc. I am having issues and wanted to know whether there are issues in implementation of jaber module or not?Thanks in advance _________________________________________________________________ Your Space. Your Friends. Your Stories. Share your world with Windows Live Spaces. http://spaces.live.com/?mkt=en-ca

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[Users] Openser on Sun Ultrasparc T1 Processors - Mismatch architecture detection
by Sergio Gutierrez 27 Mar '07

27 Mar '07

Hi Henning. I tested the compilation and it does not detect the architecture. I think that maybe is because of uname -p outputs sparc, and uname -m outputs sun4v. Maybe, would it be possible to include the fix to this condition at the later if (line 83 at Makefile.defs) # fix sparc -> sparc64 ifeq ($(ARCH),sparc) ifeq ($(shell uname -m),sun4u) ARCH := sparc64 endif endif Kind regards On 3/27/07, Henning Westerholt <henning.westerholt(a)1und1.de> wrote: > > On Tuesday 27 March 2007 17:12, you wrote: > > Hi Henning. > > > > The patch applies OK. > > Hi Sergio, thank you. > And is the right architecture detected, if you do a fresh compile with > this > patch? > > Best regards, > > Henning >

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Re: [Serusers] re: CONTACT field malformed in a 302 reply
by Jiri Kuthan 27 Mar '07

27 Mar '07

Contact from SER is NOT malformed. It is 100% compliant to the RFC3261 spec (see sections 20.10 and 25 of RFC 3261). in the grammar, contact expands to name-addr|addr-spec, with addr-spec being URI. Exapmles in the addr-spec for also appear throughout the spec. Thus I suggest that the softswitch vendor makes himself familiar with this document. As a workaround you may wish trying some manual tricks to form a reply which will make the UAC equipment happy (append_to_reply). -jiri p.s. interestingly, gl_redirect must be your own feature, so if you developed some extra SER code updating the way of generating Contacts in 3xx should not be a big deal. I just occur to think that doing it in public SER doesn't make much sense, because if there are multiple ways to do things in SIP, whichever you choose, you will find someone who is silly enough only to accept the opposite one :-) At 20:57 26/03/2007, Jignesh Gandhi wrote: >Hello, > >I am using SER as a redirect server. >Recently I came across an issue where the 302 sent back by SER is >not liked by a soft switch. Particularly , the format of the CONTACT >field according to the softswitch.. > >Here is an excerpt of the 302 reply send back ... > >Session Initiation Protocol > Status-Line: SIP/2.0 302 MovedTemporarily > Status-Code: 302 > Resent Packet: True > Suspected resend of frame: 73 > Message Header > To: <<mailto:sip:12345@172.20.20.46>sip:12345@172.20.20.46>;tag=b27e1a1d33761e85846fc98f5f3a7e58.dfa5 > SIP to address: <mailto:sip:12345@172.20.20.46>sip:12345@172.20.20.46 > SIP tag: b27e1a1d33761e85846fc98f5f3a7e58.dfa5 > From: <http://172.20.20.46>172.20.20.46<<mailto:sip:12345@172.20.20.46>sip:12345@172.20.20.46>;tag=161dda6e > SIP Display info: <http://172.20.20.46>172.20.20.46 > SIP from address: <mailto:sip:12345@172.20.20.46>sip:12345@172.20.20.46 > SIP tag: 161dda6e > Via: SIP/2.0/UDP <http://10.99.99.140:9585>10.99.99.140:9585;branch=z9hG4bK-d87543-730866470-1--d87543-;rport=9585 > Call-ID: 95239a63f0347c53 > CSeq: 1 INVITE > Contact: sip:12345@172.20.20.37:5060 > Server: Sip EXpress router (0.8.14 (i386/linux)) > Content-Length: 0 > Warning: 392 <http://172.20.20.46:5060>172.20.20.46:5060 "Noisy feedback tells: pid=9550 req_src_ip=<http://10.99.99.140>10.99.99.140 req_src_port=9585 in_uri=<mailto:sip:12345@172.20.20.46>sip:12345@172.20.20.46 out_uri=sip:12345@172.20.20.37:5060 via_cnt==1 > >the softswitch is wanting the contact field with <sip:xxxx@xxx.xxx.xxx.xxx:5060> like a FROM or TO URI. > >Here is ser.cfg part that does the sl_send_reply(); > > # do a stateless redirect, if return code is correct > if (method=="INVITE") > { > xlog("L_INFO", "SourceIP <%is> \n"); > xlog("L_INFO", "From-uri<%fu>, r-uri <%ru> \n"); > if (!gl_redirect()) > { > sl_send_reply("480", "TemporarilyUnavailable"); > xlog("L_WARN", "Sending a 480 response with r-uri <%ru>\n"); > break; > } > else > { > sl_send_reply("302", "MovedTemporarily"); > xlog("L_DBG", "Sending a 302 response with r-uri <%ru>\n"); > break; > } > }; > >Any help is appreciated. > > >thanks, >-- >Jignesh Gandhi ><mailto:jigpgandhi@gmail.com>jigpgandhi(a)gmail.com >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[Users] Openser on Sun Ultrasparc T1 Processors - Mismatch architecture detection
by Sergio Gutierrez 27 Mar '07

27 Mar '07

Hi everybody. I think that there is something missing on Makefile.defs. Sun Machines using the new line Processors (UltraSparc T1) are not correctly detected as a Sparc 64 bit machine. I think that architecture detection should include "sun4v" (uname -m output on a Sun Fire T1000 or T2000) as equivalent to Sun4u. Thanks. Sergio Gutiérrez.

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[Users] fifo,fifo_mode commands in 1.2.0-tls
by Steve Blair 27 Mar '07

27 Mar '07

I'm upgrading a 1.0 system to 1.2.0-tls and I get a few config file errors when trying to start the proxy using the previous openser.cfg file. It seems the fifo, fifo_mode and fifo_db_url commands have changed but I can only find a change log reference for fifo_mode. What are the correct replacements for these commands? Thanks,Steve

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Re: [Users] Not able to access avp variables in reply route
by Asterisk Expert 27 Mar '07

27 Mar '07

On 3/26/07, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: > > > > Asterisk Expert wrote: > > So you mean it is not possible to access values set from one > > dialog(INVITE-200 OK) in (BYE-200OK) ? > > You can use avp_db_query to store a certain value into a table during > INVITE and fetch it back during BYE. Of course you need a key which is > identical during the dialog - e.g. callid+fromtag+totag. But totag will be null before relaying invite request, isn't it? Currently I did it with just call id. Let me know if it will create any problem. regards > klaus > > > Is there any alternative? I must have to use custom CDR not db or > radius. > > > > On 3/26/07, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: > >> > >> > >> > >> Asterisk Expert wrote: > >> > I thought one call is one transaction isn't it? > >> > >> No. > >> > >> A transaction is a request till the final response, e.g. INVITE-200 OK. > >> or OPTIONS-200 Ok. > >> > >> In SIP terminology a call is named "dialog". A dilaog consists of > >> multiple transaction. The first transaction is the "dialog creating > >> transaction" (e.g. INVITE) and the last transaction in this dialog is > >> the "dialog terminating transcation" e.g. BYE. > >> > >> Thus, INVITE and BYE are seperate transactions which belong to the same > >> dialog. > >> > >> Openser is transaction statefull, but not dialog statefull. > >> > >> regards > >> klaus > >> > >> > > >> > On 3/26/07, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: > >> >> > >> >> I guess you are trying to use an AVP created during INVITE in the > BYE > >> >> transaction. This is not possible as the BYE is a different > >> transaction > >> >> and AVP lifetime is limited to a single transaction. > >> >> > >> >> regards > >> >> klaus > >> >> > >> >> Asterisk Expert wrote: > >> >> > Hello, > >> >> > I'm trying to implement custom CDR using OpenSER. What I do is, > >> >> insert a > >> >> > record in cdr table on invite request. I get the id of the > inserted > >> row > >> >> > using LAST_INSERT_ID() function of mysql in avp_db_query and > >> store in > >> >> avp > >> >> > variable rowid. I set connect time in reply route when I receive > 200 > >> OK > >> >> for > >> >> > Invite request. When I get 200 OK for BYE request, I update the > cdr > >> >> record > >> >> > and store connect time, disconnect time, duration, etc. But at > this > >> >> > place(reply route) I get the null rowid so as connect time. > >> >> According to > >> >> tm > >> >> > module's onreply_avp_mode parameter, I should be able to see the > >> values > >> >> set > >> >> > from request route in reply route if I set that parameter to 1 > >> but it > >> >> > has no > >> >> > effect. I can not use either avp variables or script variables as > >> >> > transaction variables to record custom cdr. Can anyone guide me > >> >> > implementing > >> >> > this scenario or tell me what I've done wrong? > >> >> > > >> >> > > >> >> > > >> >> > >> > ------------------------------------------------------------------------ > >> >> > > >> >> > _______________________________________________ > >> >> > Users mailing list > >> >> > Users(a)openser.org > >> >> > http://openser.org/cgi-bin/mailman/listinfo/users > >> >> > >> > > >> > > >> > > >> > > > > > > > -- Regards Ruchir Brahmbhatt

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RE: [Users] load testing presence server
by Toni Barata 27 Mar '07

27 Mar '07

Hi Daniel, In a scenario in witch we have a winfo and presence Subscribe with a 70 sessions per second after a wile (3906 sessions in last test) the openser crashes. Can you take a look in the attached logs from /var/log/openser.log and from core dump (they are on the same attached file)? Thanks, Toni -----Original Message----- From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] Sent: terça-feira, 27 de Março de 2007 10:12 To: Toni Barata Cc: users(a)openser.org Subject: Re: [Users] load testing presence server Hello, On 03/26/07 19:12, Toni Barata wrote: > Hi Daniel, > > Increasing the children from 4 to 16 did turn into same results (I was already using debug=3 and fork=yes). > > But setting debug=0 did real turn in much better results. > looks like lot of syslog messages are printed. Can you check if your syslog is set asynchronous for openser? Cheers, Daniel > Thanks a lot, > Toni > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > Sent: segunda-feira, 26 de Março de 2007 16:53 > To: Toni Barata > Cc: users(a)openser.org > Subject: Re: [Users] load testing presence server > > Hello, > > if you use exactly the same config, then turn debug to a lower level > (e.g., 3), set fork=yes and children to 16 for better results. > > Cheers, > Daniel > > > On 03/26/07 18:48, Toni Barata wrote: > >> Hi Anka, >> >> >> >> In a scenario using Openser as a Presence Sever (using the config file >> example located in >> http://openser.org/dokuwiki/doku.php/presence:configuration-file), the >> execution of load tests (with sipp) with Subscribe Presence WInfo had >> as result a maximum of 20 sessions per second (We are using a Intel(R) >> Xeon(TM) CPU 3.00GHz with 2GB RAM), with lots of retransmissions if we >> increase to values bigger than 20 (Openser dos not crash). Is there >> any configurable parameter in the Presence module that allows >> increasing this value? >> >> >> >> Best regards, >> >> Toni >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > >

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RE: [Users] load testing presence server
by Toni Barata 27 Mar '07

27 Mar '07

Hi Ovidiu, Thanks for the tip. But using the asynchronous mode it was not enough to increase the presence server performance if debug parameter is >= 3. Best Regards, Toni -----Original Message----- From: Ovidiu Sas [mailto:sip.nslu@gmail.com] Sent: segunda-feira, 26 de Março de 2007 22:24 To: Toni Barata Cc: daniel(a)voice-system.ro; users(a)openser.org Subject: Re: [Users] load testing presence server Hi Toni, Maybe this will help you: http://www.voice-system.ro/docs/ser-syslog/ar01s06.html Check the Note at the end of the page: NOTE: The '-' in front of "/var/log/ser.log" is to omit synchronizing the log file after every log message. If you choose to configure "syslog" in synchronous mode you must be aware that it has big impact over performances when the signaling traffic is high. Regards, Ovidiu Sas On 3/26/07, Toni Barata <toni-r-barata(a)ptinovacao.pt> wrote: > Hi Daniel, > > Increasing the children from 4 to 16 did turn into same results (I was already using debug=3 and fork=yes). > > But setting debug=0 did real turn in much better results. > > Thanks a lot, > Toni > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > Sent: segunda-feira, 26 de Março de 2007 16:53 > To: Toni Barata > Cc: users(a)openser.org > Subject: Re: [Users] load testing presence server > > Hello, > > if you use exactly the same config, then turn debug to a lower level > (e.g., 3), set fork=yes and children to 16 for better results. > > Cheers, > Daniel > > > On 03/26/07 18:48, Toni Barata wrote: > > > > Hi Anka, > > > > > > > > In a scenario using Openser as a Presence Sever (using the config file > > example located in > > http://openser.org/dokuwiki/doku.php/presence:configuration-file), the > > execution of load tests (with sipp) with Subscribe Presence WInfo had > > as result a maximum of 20 sessions per second (We are using a Intel(R) > > Xeon(TM) CPU 3.00GHz with 2GB RAM), with lots of retransmissions if we > > increase to values bigger than 20 (Openser dos not crash). Is there > > any configurable parameter in the Presence module that allows > > increasing this value? > > > > > > > > Best regards, > > > > Toni > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > 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 >

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[Users] register more than one username by IP phone
by Bodin Bruno 27 Mar '07

27 Mar '07

Hi, Is it possible to register two (or more) identifiers by user, such as phone number, username or service ? Thank

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