- sr-users - kamailio.org

[OpenSER-Users] [Users] new MediaProxy release 1.8.2
by Pedro Fernandes 23 Oct '07

23 Oct '07

Any news about that mysql issue?i have the same problem! :( mediaproxy mediaproxy]# ./mediaproxy.py --no-fork warning: accounting is enabled but the accounting module is missing. accounting is not available! Listening for commands on local socket `/var/run/mediaproxy.sock' Listening for remote commands on `0.0.0.0:25061' Remote commands are allowed from: 192.168.0.187, 127.0.0.1 Using IP address `192.168.0.187' for the RTP/RTCP proxy

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[OpenSER-Users] Call Failed: Proxy Authentication Failed
by Pedro Fernandes 22 Oct '07

22 Oct '07

hi, iam trying to implement a openser MediaProxy under nat. I use USER1 with 10.0.0.x and media proxy to resolve nat. User 2 have 192.168.0.x. Both clients can register in openser (with mysql accounting), but when i try do call each other, i get error CALL FAILED: Proxy Authentication Required. Iam using xlite sofphone on my clients.

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[Serusers] 200 OK/SDP and ACK/SDP problem - 2
by Alexandr Volkov 22 Oct '07

22 Oct '07

Hi all I have a problem that correlates with problem that was already posted in 2006-Feb (http://lists.iptel.org/pipermail/serusers/2006-February/026841.html) I have a stateful ser with mediaproxy at public IP, client A(SIP B2BUA) with public address and client B(user phone) with private address. 1. SIP dialog already established and client A send re-invite without SDP (reinvite is necessary, problem was described in rfc3725.txt). 2. client B respond OK(200) with NATed SDP 3. Client A send ACK with normal SDP. At step 2 in ser we can determine that we need to use media proxy and we start it. But when we processing ACK, we see normal SDP, and won't to start mediaproxy. Thus we have one-way audio. I tried to use flags, but it seems that when ACK arrives all flags zeroed. (Ser imply that ACK don't belong to invite transaction?) I can't use AVP because commands are unavailable in onreply_route. The only way I found is to use global flags. But this way is worse because foreign ACK can be processed between our OK and ACK from the same transaction... Thanks for your advise. Alex

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[OpenSER-Users] Compiling
by Gerson Matiolli 22 Oct '07

22 Oct '07

Hi, all I am using Fedora 7 and openser-1.2.2-tls I need to change the contents of the file lcr_mod.c (MAX_NO_OF_LCRS 8000) When compiling using the command: make include_modules="mysql" modules Two errors are found. What can be? make[1]: Entrando no diretório `/root/openser-1.2.2-tls/modules/mysql' make[1]: Saindo do diretório `/root/openser-1.2.2-tls/modules/mysql' make[1]: Entrando no diretório `/root/openser-1.2.2-tls/modules/mysql' Compiling dbase.c gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=athlon64 -DNAME='"openser"' -DVERSION='"1.2.2-notls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.1.2"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -I/usr/local/include -I/usr/local/include/mysql -I/usr/local/mysql/include/mysql -I/usr/local/mysql/include -I/usr/include/mysql -c dbase.c -o dbase.o dbase.c:29:25: error: mysql/mysql.h: Arquivo ou diretório não encontrado dbase.c:30:26: error: mysql/errmsg.h: Arquivo ou diretório não encontrado In file included from dbase.c:35: val.h:39: error: expected ) before * token In file included from dbase.c:36: my_con.h:38: error: expected specifier-qualifier-list before MYSQL_RES dbase.c: In function db_mysql_submit_query: dbase.c:63: error: struct my_con has no member named timestamp dbase.c:64: warning: implicit declaration of function mysql_ping dbase.c:64: error: struct my_con has no member named con dbase.c:68: error: struct my_con has no member named timestamp dbase.c:86: warning: implicit declaration of function mysql_query dbase.c:86: error: struct my_con has no member named con dbase.c:89: warning: implicit declaration of function mysql_errno dbase.c:89: error: struct my_con has no member named con dbase.c:90: error: CR_SERVER_GONE_ERROR undeclared (first use in this function) dbase.c:90: error: (Each undeclared identifier is reported only once dbase.c:90: error: for each function it appears in.) dbase.c:90: error: CR_SERVER_LOST undeclared (first use in this function) dbase.c:94: warning: implicit declaration of function mysql_error dbase.c:94: error: struct my_con has no member named con dbase.c:94: error: struct my_con has no member named con dbase.c:94: warning: format %s expects type char *, but argument 3 has type int dbase.c:94: error: struct my_con has no member named con dbase.c:94: warning: format %s expects type char *, but argument 3 has type int dbase.c:94: error: struct my_con has no member named con dbase.c:94: warning: format %s expects type char *, but argument 3 has type int dbase.c:94: error: struct my_con has no member named con dbase.c:94: warning: format %s expects type char *, but argument 3 has type int dbase.c:94: error: struct my_con has no member named con dbase.c:94: warning: format %s expects type char *, but argument 3 has type int dbase.c:94: error: struct my_con has no member named con dbase.c:94: warning: format %s expects type char *, but argument 3 has type int dbase.c:94: error: struct my_con has no member named con dbase.c:94: warning: format %s expects type char *, but argument 3 has type int dbase.c: At top level: dbase.c:134: error: expected ) before * token dbase.c:162: error: expected ) before * token dbase.c:203: error: expected ) before * token dbase.c: In function db_mysql_store_result: dbase.c:329: error: struct my_con has no member named res dbase.c:329: warning: implicit declaration of function mysql_store_result dbase.c:329: error: struct my_con has no member named con dbase.c:330: error: struct my_con has no member named res dbase.c:331: warning: implicit declaration of function mysql_field_count dbase.c:331: error: struct my_con has no member named con dbase.c:336: error: struct my_con has no member named con dbase.c:336: error: struct my_con has no member named con dbase.c:336: warning: format %s expects type char *, but argument 3 has type int dbase.c:336: error: struct my_con has no member named con dbase.c:336: warning: format %s expects type char *, but argument 3 has type int dbase.c:336: error: struct my_con has no member named con dbase.c:336: warning: format %s expects type char *, but argument 3 has type int dbase.c:336: error: struct my_con has no member named con dbase.c:336: warning: format %s expects type char *, but argument 3 has type int dbase.c:336: error: struct my_con has no member named con dbase.c:336: warning: format %s expects type char *, but argument 3 has type int dbase.c:336: error: struct my_con has no member named con dbase.c:336: warning: format %s expects type char *, but argument 3 has type int dbase.c:336: error: struct my_con has no member named con dbase.c:336: warning: format %s expects type char *, but argument 3 has type int dbase.c:350: warning: implicit declaration of function mysql_free_result dbase.c:350: error: struct my_con has no member named res dbase.c:351: error: struct my_con has no member named res dbase.c: In function db_mysql_free_result: dbase.c:373: error: struct my_con has no member named res dbase.c:374: error: struct my_con has no member named res dbase.c: In function db_mysql_query: dbase.c:423: warning: implicit declaration of function db_mysql_print_where dbase.c:423: error: struct my_con has no member named con dbase.c: In function db_mysql_fetch_result: dbase.c:467: error: struct my_con has no member named res dbase.c:467: error: struct my_con has no member named con dbase.c:468: error: struct my_con has no member named res dbase.c:469: error: struct my_con has no member named con dbase.c:474: error: struct my_con has no member named con dbase.c:474: error: struct my_con has no member named con dbase.c:474: warning: format %s expects type char *, but argument 3 has type int dbase.c:474: error: struct my_con has no member named con dbase.c:474: warning: format %s expects type char *, but argument 3 has type int dbase.c:474: error: struct my_con has no member named con dbase.c:474: warning: format %s expects type char *, but argument 3 has type int dbase.c:474: error: struct my_con has no member named con dbase.c:474: warning: format %s expects type char *, but argument 3 has type int dbase.c:474: error: struct my_con has no member named con dbase.c:474: warning: format %s expects type char *, but argument 3 has type int dbase.c:474: error: struct my_con has no member named con dbase.c:474: warning: format %s expects type char *, but argument 3 has type int dbase.c:474: error: struct my_con has no member named con dbase.c:474: warning: format %s expects type char *, but argument 3 has type int dbase.c:487: warning: implicit declaration of function mysql_num_rows dbase.c:487: error: struct my_con has no member named res dbase.c:516: error: struct my_con has no member named row dbase.c:516: warning: implicit declaration of function mysql_fetch_row dbase.c:516: error: struct my_con has no member named res dbase.c:517: error: struct my_con has no member named row dbase.c:518: error: struct my_con has no member named con dbase.c:518: error: struct my_con has no member named con dbase.c:518: warning: format %s expects type char *, but argument 3 has type int dbase.c:518: error: struct my_con has no member named con dbase.c:518: warning: format %s expects type char *, but argument 3 has type int dbase.c:518: error: struct my_con has no member named con dbase.c:518: warning: format %s expects type char *, but argument 3 has type int dbase.c:518: error: struct my_con has no member named con dbase.c:518: warning: format %s expects type char *, but argument 3 has type int dbase.c:518: error: struct my_con has no member named con dbase.c:518: warning: format %s expects type char *, but argument 3 has type int dbase.c:518: error: struct my_con has no member named con dbase.c:518: warning: format %s expects type char *, but argument 3 has type int dbase.c:518: error: struct my_con has no member named con dbase.c:518: warning: format %s expects type char *, but argument 3 has type int dbase.c: In function db_mysql_insert: dbase.c:584: warning: implicit declaration of function db_mysql_print_values dbase.c:584: error: struct my_con has no member named con dbase.c: In function db_mysql_delete: dbase.c:630: error: struct my_con has no member named con dbase.c: In function db_mysql_update: dbase.c:674: warning: implicit declaration of function db_mysql_print_set dbase.c:674: error: struct my_con has no member named con dbase.c:683: error: struct my_con has no member named con dbase.c: In function db_mysql_replace: dbase.c:726: error: struct my_con has no member named con make[1]: ** [dbase.o] Erro 1 make[1]: Saindo do diretório `/root/openser-1.2.2-tls/modules/mysql' make: ** [modules] Erro 2

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[OpenSER-Users] Manipulation of the From header username
by malmic＠alice.it 22 Oct '07

22 Oct '07

Hello there, I need to strip away the first digit "0" from the username of the Header >From on an incoming INVITE message.I had already used the uac_replace_from when adding digits to the from field on outgoing INVITE but stripping a digit seems harder for me. Can somebody help me? Thx Michele

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[Serusers] FW: Get RealTime Online Users status
by roayu 22 Oct '07

22 Oct '07

-----Original Message----- From: roayu Sent: Monday, October 22, 2007 3:13 PM To: 'Michal Matyska' Subject: RE: [Serusers] Get RealTime Online Users status Thanks Michal! I will try on it. Thanks. Cheers, Roa Yu -----Original Message----- From: Michal Matyska [mailto:michal@iptel.org] Sent: Friday, October 19, 2007 11:55 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: RE: [Serusers] Get RealTime Online Users status Looks you have SIP ALG enabled on your NAT, which breaks the call transfer (no surprise, I haven't seen working SIP ALG yet). Check OPTIONS request sent by SJphone and received by SER, check NOTIFY sent by SJphone and received by SER. Disable the SIP ALG and try again. If you can't disable it, change the NAT box itself. Michal On Fri, 2007-10-19 at 14:42 +0800, roayu wrote: > Hi Michal, > > I've attached the config file as well as the network capture so that you can > help me to find out the problem. But I'm not sure whether I will gonna guide > u to the answer 42 or not. :p > > Thanks for your help! > > > Cheers, > Roa Yu > > > -----Original Message----- > From: Michal Matyska [mailto:michal@iptel.org] > Sent: Friday, October 19, 2007 1:17 PM > To: roayu > Cc: Jiri Kuthan; serusers(a)lists.iptel.org > Subject: Re: [Serusers] Get RealTime Online Users status > > Hi, > > unless you post the network capture and your config file (but make sure > you don't disclose anything you want to keep secret) there is nobody who > can help you. > > The only answer is: your script is broken, except the answer 42, but I > don't think your question was the ultimate question about life, universe > etc. ;-) > > Michal > > P.S. If you haven't read "Hitchhikers guide to the universe" don't panic > and just forget the answer 42. > > On Fri, 2007-10-19 at 11:35 +0800, roayu wrote: > > Hi Jiri! > > > > I've captured and analyzed those SIP message dumps. I found that when > > Sjphone A sent a NOTIFY signal to the SER (after Sjphone A accepted the > > REFER), SER didn't pass the NOTIFY to the SPA942 to terminate the call > > between them. > > > > Therefore, Sjphone A unable to INVITE Sjphone B to bridge up the > connection. > > May I know what causes this? > > > > > > Thanks for your precious time and reply! > > > > > > Cheers, > > Roa Yu :) > > > > > > -----Original Message----- > > From: Jiri Kuthan [mailto:jiri@iptel.org] > > Sent: Thursday, October 18, 2007 4:28 PM > > To: roayu > > Cc: serusers(a)lists.iptel.org > > Subject: Re: [Serusers] Get RealTime Online Users status > > > > At 04:59 18/10/2007, roayu wrote: > > >Hi Jiri! > > > > > >Actually, I'm using Sjphone and SPA942 to do the call transfer. I'm > calling > > >using Sjphone A to SPA942, then use the SPA942 to transfer the call to > > >Sjphone B. > > > > > > SJPhone A --> SPA942 --> SJPhone B > > > > > >After a short conversation with SJPhone B, then I press the 'xfer' button > > on > > >the 2nd time so that both SJPhone A and SJPhone B can talk to each other. > > >But the call couldn't be transferred to SJPhone B whereas when I used > > >Asterisk, it's able to do so. > > > > > >I've tried to disable the Digest-qop but it still couldn't perform the > > >transfer properly. What other settings that I need to configure on SER? > > > > I'm afraid you've gotta analyze SIP message dumps to figure out what's > gone > > wrong. -jiri > > > > > > >Thanks and really appreciate on your reply. > > > > > >Cheers, > > >Roa Yu > > > > > > > > >-----Original Message----- > > >From: Jiri Kuthan [mailto:jiri@iptel.org] > > >Sent: Thursday, October 18, 2007 9:16 AM > > >To: roayu > > >Cc: serusers(a)lists.iptel.org > > >Subject: Re: [Serusers] Get RealTime Online Users status > > > > > >At 10:37 10/10/2007, roayu wrote: > > >>Oh, I got it! Thanks Jiri! > > >> > > >>Btw, I have another questions. > > >>1) I would like to know can call transfer being done when the > > >>connection is P2P? > > > > > >you mean without a proxy? sure it can. > > > > > >> When I tried to use relay (mediaproxy) to do the call transfer, > > >it's > > >>able to transfer the call to the other party whereas when the > condition > > >is > > >>P2P, it just can't pass the correct signal. > > > > > >that's apparently unrelated to use of media proxy. you would have to > > >check the SIP dumps what's going wrong. Unfortunately I can't promise > > >I would help to review those -- my to-do-list is endless these days. > > >Perhaps someone else on the list will. > > > > > > > > >>2) Can SER support SPA942 ? Or is there some settings that I need > to > > >>configure on SER ? > > > > > >yes. > > >well for some SPAs you may need to disable digest-qop since they have a > > bug. > > >(don't ask me which though). > > > > > >-jiri > > > > > > > > > > > >>Thanks. > > >> > > >>Cheers, > > >>Roa Yu > > >> > > >>-----Original Message----- > > >>From: Jiri Kuthan [mailto:jiri@iptel.org] > > >>Sent: Wednesday, October 10, 2007 2:56 PM > > >>To: roayu; SIP > > >>Cc: serusers(a)lists.iptel.org > > >>Subject: Re: [Serusers] Get RealTime Online Users status > > >> > > >>yes, set usrloc's database mode to 1. -jiri > > >> > > >>At 03:32 10/10/2007, roayu wrote: > > >> > > >>>Thanks for your reply. I found that it's only update after around 30 > > >>>seconds. Is there anyway that I can fasten the update on the database? > > >>> > > >>>Thanks. > > >>> > > >>>Cheers, > > >>>Roa Yu > > >>> > > >>>-----Original Message----- > > >>>From: SIP [mailto:sip@arcdiv.com] > > >>>Sent: Tuesday, October 09, 2007 9:14 PM > > >>>To: roayu > > >>>Cc: serusers(a)lists.iptel.org > > >>>Subject: Re: [Serusers] Get RealTime Online Users status > > >>> > > >>>roayu wrote: > > >>>> > > >>>> Hi there! > > >>>> > > >>>> Can anyone tell me how to get the RealTime Online user status other > > >>>> than using command '*serctl ul show*'? Or how can I store the > realtime > > >>>> online user status to the MySQL db? > > >>>> > > >>>> Thanks. > > >>>> > > >>>> Cheers, > > >>>> > > >>>> Roa Yu J > > >>>> > > >>>> > > ------------------------------------------------------------------------ > > >>>> > > >>>> _______________________________________________ > > >>>> Serusers mailing list > > >>>> Serusers(a)lists.iptel.org > > >>>> http://lists.iptel.org/mailman/listinfo/serusers > > >>>> > > >>>The location table in the MySQL database stores what are, in effect, > > >>>currently online users. It is, however, only an approximation. When a > UA > > >>>registers, it sends an expiration time on its registration and that > gets > > >>>stored in the location table. If the user disconnects without > cancelling > > >>>his registration (some UAs send am expire register message to 'log out' > > > >>>and some don't), then the data may still be in the table until the > > >>>expire time occurs. Realistically, though, you can get a good idea of > > >>>currenly online users using the location table and while it may not be > > >>>100% accurate, it's close enough for government work, as it were. > > >>> > > >>>N. > > >>>_______________________________________________ > > >>>Serusers mailing list > > >>>Serusers(a)lists.iptel.org > > >>>http://lists.iptel.org/mailman/listinfo/serusers > > >> > > >> > > >> > > >>-- > > >>Jiri Kuthan http://iptel.org/~jiri/ > > >>_______________________________________________ > > >>Serusers mailing list > > >>Serusers(a)lists.iptel.org > > >>http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > >-- > > >Jiri Kuthan http://iptel.org/~jiri/ > > >_______________________________________________ > > >Serusers mailing list > > >Serusers(a)lists.iptel.org > > >http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > -- > > Jiri Kuthan http://iptel.org/~jiri/ > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers >

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[OpenSER-Users] Re: Users Digest, Vol 29, Issue 61
by vallimamod abdullah 20 Oct '07

20 Oct '07

On 10/19/07, users-request(a)openser.org <users-request(a)openser.org> wrote on behalf of Iñaki Baz Castillo <ibc(a)in.ilimit.es>: > > > My openser have a private IP (192.168.5.119) statically natted to > > xx.xx.33.119; so the first via contains the private IP. Is it up to the > > client to figure out how to manage this or is there a way to handle this > > in openser ? > > > Look at the "Via" headres in both cases, they are the same. That's right but the difference is that the second invite reply (from Sipura) arrives to the server and the first one from Thomson don't. > You have an issue because NAT. The Thomson INVITE leaves OpenSer natted but > the Via header added by OpenSer says "192.168.5.119". You should fix the Via > header for messages that will be natted, but sincerely I don't know how. Thank you for this hint. It reminded me a wonderful option in openser core called "advertised_address" where you can set what you want to see in the first Via. So I set my domain name there. It is resolved with the public IP from outside and with the lan IP from the inside and now it works from both sides ! -- Regards, - vma .

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[Users] Re: 200 ok not cought in ONREPLY
by Dan-Cristian Bogos 19 Oct '07

19 Oct '07

Guys, thanks to your support (both Bogdan and Daniel), I found out that there was nothing wrong with the server and t_onreply handling, but the INVITE was taking another routing three than the one I was checking and therefore never hit the t_on_reply section I wanted. This being said, I consider this issue resolved. Cheers, Dan

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[OpenSER-Users] LCR
by Gerson Matiolli 19 Oct '07

19 Oct '07

Hi all I am using Fedora 7 and installed "openser - 1.2.2 - 3.fc7.i386.rpm" and I am using mysql. I need to increase the table LCR, 256 entries for 8000 entries, as do that?

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Re: [OpenSER-Users] Manipulation of the From header username
by Jiri Kuthan 19 Oct '07

19 Oct '07

At 18:30 19/10/2007, samuel wrote: >SIP version 1 (2543) used From header to identify the dialog so it's not recommended to change the From header to be backwards compatible (is there any RFC 2543 SIP endpoint around??...). >SIP Version 2 (RFC 3261) uses tags to identify dialogs so you should be able to change the FRom header except the tag. and mandates backwards compatibility at the same time, which gets you back to square 1. >Recently there was an RFC 4916 that specifies how to identify the connected identity and how to make a change in the From but I think this is far beyond what you are trying to achieve. I'm adraid there are too many fascinating extensions in such quantity that many have stopped taking care of those (sometimes for a good reason) >There are additional headers (Remote_PartyID, and related) that can be used to specify an identify of the calling party.. asserted identity being most adequate nowadays, I guess. -jiri -- Jiri Kuthan http://iptel.org/~jiri/

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Re: [Serusers] Get RealTime Online Users status
by Michal Matyska 19 Oct '07

19 Oct '07

Looks you have SIP ALG enabled on your NAT, which breaks the call transfer (no surprise, I haven't seen working SIP ALG yet). Check OPTIONS request sent by SJphone and received by SER, check NOTIFY sent by SJphone and received by SER. Disable the SIP ALG and try again. If you can't disable it, change the NAT box itself. Michal On Fri, 2007-10-19 at 14:42 +0800, roayu wrote: > Hi Michal, > > I've attached the config file as well as the network capture so that you can > help me to find out the problem. But I'm not sure whether I will gonna guide > u to the answer 42 or not. :p > > Thanks for your help! > > > Cheers, > Roa Yu > > > -----Original Message----- > From: Michal Matyska [mailto:michal@iptel.org] > Sent: Friday, October 19, 2007 1:17 PM > To: roayu > Cc: Jiri Kuthan; serusers(a)lists.iptel.org > Subject: Re: [Serusers] Get RealTime Online Users status > > Hi, > > unless you post the network capture and your config file (but make sure > you don't disclose anything you want to keep secret) there is nobody who > can help you. > > The only answer is: your script is broken, except the answer 42, but I > don't think your question was the ultimate question about life, universe > etc. ;-) > > Michal > > P.S. If you haven't read "Hitchhikers guide to the universe" don't panic > and just forget the answer 42. > > On Fri, 2007-10-19 at 11:35 +0800, roayu wrote: > > Hi Jiri! > > > > I've captured and analyzed those SIP message dumps. I found that when > > Sjphone A sent a NOTIFY signal to the SER (after Sjphone A accepted the > > REFER), SER didn't pass the NOTIFY to the SPA942 to terminate the call > > between them. > > > > Therefore, Sjphone A unable to INVITE Sjphone B to bridge up the > connection. > > May I know what causes this? > > > > > > Thanks for your precious time and reply! > > > > > > Cheers, > > Roa Yu :) > > > > > > -----Original Message----- > > From: Jiri Kuthan [mailto:jiri@iptel.org] > > Sent: Thursday, October 18, 2007 4:28 PM > > To: roayu > > Cc: serusers(a)lists.iptel.org > > Subject: Re: [Serusers] Get RealTime Online Users status > > > > At 04:59 18/10/2007, roayu wrote: > > >Hi Jiri! > > > > > >Actually, I'm using Sjphone and SPA942 to do the call transfer. I'm > calling > > >using Sjphone A to SPA942, then use the SPA942 to transfer the call to > > >Sjphone B. > > > > > > SJPhone A --> SPA942 --> SJPhone B > > > > > >After a short conversation with SJPhone B, then I press the 'xfer' button > > on > > >the 2nd time so that both SJPhone A and SJPhone B can talk to each other. > > >But the call couldn't be transferred to SJPhone B whereas when I used > > >Asterisk, it's able to do so. > > > > > >I've tried to disable the Digest-qop but it still couldn't perform the > > >transfer properly. What other settings that I need to configure on SER? > > > > I'm afraid you've gotta analyze SIP message dumps to figure out what's > gone > > wrong. -jiri > > > > > > >Thanks and really appreciate on your reply. > > > > > >Cheers, > > >Roa Yu > > > > > > > > >-----Original Message----- > > >From: Jiri Kuthan [mailto:jiri@iptel.org] > > >Sent: Thursday, October 18, 2007 9:16 AM > > >To: roayu > > >Cc: serusers(a)lists.iptel.org > > >Subject: Re: [Serusers] Get RealTime Online Users status > > > > > >At 10:37 10/10/2007, roayu wrote: > > >>Oh, I got it! Thanks Jiri! > > >> > > >>Btw, I have another questions. > > >>1) I would like to know can call transfer being done when the > > >>connection is P2P? > > > > > >you mean without a proxy? sure it can. > > > > > >> When I tried to use relay (mediaproxy) to do the call transfer, > > >it's > > >>able to transfer the call to the other party whereas when the > condition > > >is > > >>P2P, it just can't pass the correct signal. > > > > > >that's apparently unrelated to use of media proxy. you would have to > > >check the SIP dumps what's going wrong. Unfortunately I can't promise > > >I would help to review those -- my to-do-list is endless these days. > > >Perhaps someone else on the list will. > > > > > > > > >>2) Can SER support SPA942 ? Or is there some settings that I need > to > > >>configure on SER ? > > > > > >yes. > > >well for some SPAs you may need to disable digest-qop since they have a > > bug. > > >(don't ask me which though). > > > > > >-jiri > > > > > > > > > > > >>Thanks. > > >> > > >>Cheers, > > >>Roa Yu > > >> > > >>-----Original Message----- > > >>From: Jiri Kuthan [mailto:jiri@iptel.org] > > >>Sent: Wednesday, October 10, 2007 2:56 PM > > >>To: roayu; SIP > > >>Cc: serusers(a)lists.iptel.org > > >>Subject: Re: [Serusers] Get RealTime Online Users status > > >> > > >>yes, set usrloc's database mode to 1. -jiri > > >> > > >>At 03:32 10/10/2007, roayu wrote: > > >> > > >>>Thanks for your reply. I found that it's only update after around 30 > > >>>seconds. Is there anyway that I can fasten the update on the database? > > >>> > > >>>Thanks. > > >>> > > >>>Cheers, > > >>>Roa Yu > > >>> > > >>>-----Original Message----- > > >>>From: SIP [mailto:sip@arcdiv.com] > > >>>Sent: Tuesday, October 09, 2007 9:14 PM > > >>>To: roayu > > >>>Cc: serusers(a)lists.iptel.org > > >>>Subject: Re: [Serusers] Get RealTime Online Users status > > >>> > > >>>roayu wrote: > > >>>> > > >>>> Hi there! > > >>>> > > >>>> Can anyone tell me how to get the RealTime Online user status other > > >>>> than using command '*serctl ul show*'? Or how can I store the > realtime > > >>>> online user status to the MySQL db? > > >>>> > > >>>> Thanks. > > >>>> > > >>>> Cheers, > > >>>> > > >>>> Roa Yu J > > >>>> > > >>>> > > ------------------------------------------------------------------------ > > >>>> > > >>>> _______________________________________________ > > >>>> Serusers mailing list > > >>>> Serusers(a)lists.iptel.org > > >>>> http://lists.iptel.org/mailman/listinfo/serusers > > >>>> > > >>>The location table in the MySQL database stores what are, in effect, > > >>>currently online users. It is, however, only an approximation. When a > UA > > >>>registers, it sends an expiration time on its registration and that > gets > > >>>stored in the location table. If the user disconnects without > cancelling > > >>>his registration (some UAs send am expire register message to 'log out' > > > >>>and some don't), then the data may still be in the table until the > > >>>expire time occurs. Realistically, though, you can get a good idea of > > >>>currenly online users using the location table and while it may not be > > >>>100% accurate, it's close enough for government work, as it were. > > >>> > > >>>N. > > >>>_______________________________________________ > > >>>Serusers mailing list > > >>>Serusers(a)lists.iptel.org > > >>>http://lists.iptel.org/mailman/listinfo/serusers > > >> > > >> > > >> > > >>-- > > >>Jiri Kuthan http://iptel.org/~jiri/ > > >>_______________________________________________ > > >>Serusers mailing list > > >>Serusers(a)lists.iptel.org > > >>http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > >-- > > >Jiri Kuthan http://iptel.org/~jiri/ > > >_______________________________________________ > > >Serusers mailing list > > >Serusers(a)lists.iptel.org > > >http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > -- > > Jiri Kuthan http://iptel.org/~jiri/ > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers >

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[OpenSER-Users] segfault in presence moudle of openser-1.2 branch
by Klaus Darilion 19 Oct '07

19 Oct '07

Hi Anca! #0 0x00e030e3 in update_subscription (msg=0x8162b68, subs=0xbfe24ef0, rtag=0xbfe24e08, to_tag_gen=0) at subscribe.c:359 359 record_route.len= strlen(record_route.s); (gdb) (gdb) (gdb) (gdb) bt #0 0x00e030e3 in update_subscription (msg=0x8162b68, subs=0xbfe24ef0, rtag=0xbfe24e08, to_tag_gen=0) at subscribe.c:359 #1 0x00e065f8 in handle_subscribe (msg=0x8162b68, str1=0x0, str2=0x0) at subscribe.c:1294 #2 0x080528d0 in do_action (a=0x815d518, msg=0x8162b68) at action.c:891 #3 0x08054b6c in run_action_list (a=0x815d438, msg=0x8162b68) at action.c:131 #4 0x08054258 in do_action (a=0x815d5c8, msg=0x8162b68) at action.c:815 #5 0x08054b6c in run_action_list (a=0x815ba20, msg=0x8162b68) at action.c:131 #6 0x08054d9f in run_top_route (a=0x815ba20, msg=0x3) at action.c:111 #7 0x0807cf23 in receive_msg ( buf=0x81253e0 "SUBSCRIBE sip:12.34.56.246:5060 SIP/2.0\r\nVia: SIP/2.0/UDP 80.75.44.244;branch=z9hG4bK5eb5 .405f258.0\r\nVia: SIP/2.0/UDP 10.241.131.64:23308;received=194.48.128.75;branch=z9hG4bK-d87543-537e85345d6f75 26-"..., len=601, rcv_info=0xbfe25720) at receive.c:156 #8 0x080a6935 in udp_rcv_loop () at udp_server.c:451 #9 0x0806b966 in main_loop () at main.c:834 #10 0x0806da0e in main (argc=7, argv=0xbfe25904) at main.c:1393 The crash happened when processing a SUSBCRIBE without Record-Route header: SUBSCRIBE sip:12.34.56.246:5060 SIP/2.0 Via: SIP/2.0/UDP 12.34.56.244;branch=z9hG4bK5eb5.405f258.0 Via: SIP/2.0/UDP 10.123.123.64:23308;received=123.45.123.75;branch=z9hG4bK-d87543-537e85345d6f7526-1--d87543-;rport=11609 Max-Forwards: 68 Contact: <sip:a.foobar@123.45.123.75:11609;transport=udp> To: <sip:a.foobar@mydomain.at>;tag=mango.1441.1192697556.19 From: "foobarfoob"<sip:a.foobar@mydomain.at>;tag=5224f96c Call-ID: ZGU0ZGE4NjQzYzRjZGYxZWNhNjM5N2U2MDY0MTZlY2E. CSeq: 4 SUBSCRIBE Expires: 0 User-Agent: foobarfoobarfoobarfoobarfoobarfoobar Event: presence.winfo Content-Length: 0 regards Klaus

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[OpenSER-Users] Strange behavior for openser 1.3 in mhomed mode (automatic mangling of Contact header and c= line of SDP)
by Ovidiu Sas 19 Oct '07

19 Oct '07

Hi all, I was playing with openser 1.3 in a multihomed setup. In this setup, I noticed that by default, openser is mangling the Contact header and the sdp. I stripped down the config to bare minimum to see if it was something related to my config, but it seems that this is not related to the config. Here's the stripped down config that I used: # ----------- global configuration parameters ------------------------ debug=3 fork=yes log_stderror=no check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) auto_aliases=no listen=udp:br0 listen=udp:eth1 children=1 mhomed=1 disable_tcp=yes disable_dns_blacklist=true # ------------------ module loading ---------------------------------- mpath="/opt/lib/openser/modules" loadmodule "sl.so" loadmodule "tm.so" # -- tm params -- modparam("tm", "fr_inv_timer", 100) modparam("tm", "fr_inv_timer_avp", "$avp(i:26)") modparam("tm", "fr_timer", 10) modparam("tm", "fr_timer_avp", "$avp(i:25)") modparam("tm", "noisy_ctimer", 1) modparam("tm", "onreply_avp_mode", 1) loadmodule "rr.so" # ------------------------- request routing logic ------------------- # main routing logic route{ if (!method=="REGISTER") { record_route(); } if (loose_route()) { route(1); }; if (!uri==myself) { route(1); }; if (uri==myself) { sl_send_reply("404", "Not Found"); exit; }; route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } Here's the network setup: <asterisk box> --- < openser > --- <SIP server> <192.168.2.20> --- <192.168.2.1 : EXTERNAL_IP> --- <SIP_SERVER> And here's an ngrep of the initial INVITEs: 23:53:31.439678 IP 192.168.2.20.sip > 192.168.2.1.sip: SIP, length: 819 INVITE sip:callee@SIP_SERVER SIP/2.0 Via: SIP/2.0/UDP 192.168.2.20:5060;branch=z9hG4bK4f70ea70 From: "Ovi Sas" <sip:caller@SIP_SERVER>;tag=as048c5bed To: <sip:callee@SIP_SERVER> Contact: <sip:caller@192.168.2.20> Call-ID: 28aedce563bc57dc2cac78967e69c141@SIP_SERVER CSeq: 102 INVITE User-Agent: asterisk Max-Forwards: 70 Date: Thu, 18 Oct 2007 04:23:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 285 v=0 o=root 8560 8560 IN IP4 192.168.2.20 s=session c=IN IP4 192.168.2.20 t=0 0 m=audio 34028 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 23:53:31.454727 IP EXTERNAL_IP.sip > SIP_SERVER.sip: SIP, length: 1011 INVITE sip:callee@SIP_SERVER SIP/2.0 Record-Route: <sip:EXTERNAL_IP;r2=on;lr;ftag=as048c5bed> Record-Route: <sip:192.168.2.1;r2=on;lr;ftag=as048c5bed> Via: SIP/2.0/UDP EXTERNAL_IP:5060;branch=z9hG4bKea4e.7aacc885.0 Via: SIP/2.0/UDP 192.168.2.20:5060;branch=z9hG4bK4f70ea70 From: "Ovi Sas" <sip:caller@SIP_SERVER>;tag=as048c5bed To: <sip:callee@SIP_SERVER> Contact: <sip:caller@EXTERNAL_IP:5060> Call-ID: 28aedce563bc57dc2cac78967e69c141@SIP_SERVER CSeq: 102 INVITE User-Agent: asterisk Max-Forwards: 70 Date: Thu, 18 Oct 2007 04:23:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 286 v=0 o=root 8560 8560 IN IP4 192.168.2.20 s=session c=IN IP4 EXTERNAL_IP t=0 0 m=audio 34028 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv Please note the Contact header and the c= line inside SDP. Is there a way to bypass this behaviour? Using the nathelper module doesn't seem to help (the Contact header will still carry the EXTERNAL_IP. Regards, Ovidiu Sas

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[OpenSER-Users] Problem with python-application when installing OpenXCAP
by Antoine Fressancourt 19 Oct '07

19 Oct '07

Hello, I try to install OpenXCAP on a Ubuntu server Feisty machine. I am up to the point of installing the python-application 1.0.9, but when I try to install it with python setup.py install, I get the following error : byte-compiling /usr/lib/python2.5/site-packages/application/python/queue.py to queue.pyc File "/usr/lib/python2.5/site-packages/application/python/queue.py", line 9 v v v v v v v ^ SyntaxError: invalid syntax As I have absolutely no experience with Python, I would like to know how I could work around this error. Thank you in advance for your help, Antoine

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[OpenSER-Users] New OpenXCAP release 0.9.7
by Adrian Georgescu 19 Oct '07

19 Oct '07

Hello, There is a new OpenXCAP release available. openxcap (0.9.7) unstable; urgency=low * Prevented Twisted from printing SQL login info in the log * Added some info to INSTALL about tested versions openxcap (0.9.6) unstable; urgency=low * Fixed trusted peer authentication bug against the latest Twisted svn trunk. openxcap (0.9.5) unstable; urgency=low * Changed watcher application to reflect new database column names of OpenSER revision 2923 * Changed default SQL table name from "xcap" to "xcap_xml" openxcap (0.9.4) unstable; urgency=low * Added "watchers" application to query OpenSER presence watchers. * Include debian directory in the source archive. openxcap (0.9.3) unstable; urgency=low * Adjusted OpenXCAP to use python-application >= 1.0.9 * Set TLS mode based on http/https scheme of the root URI. * document_validation defaults to Yes in config.ini.sample openxcap (0.9.2) unstable; urgency=low * Changed the name and structure of xcap_xml table. * Added the 'type' param to refreshWatchers OpenSER command. * Added configuration option to disable XML schema validation. * Updated XML scheme to allow EyeBeam resource lists to pass validation. * Update debian dependencies. The software can be downloaded from: http://openxcap.org Kind regards, Adrian Georgescu

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Re: [Serusers] RFC 3841 support?
by Jiri Kuthan 19 Oct '07

19 Oct '07

At 18:28 18/10/2007, Stefan.Brozinski(a)materna.de wrote: >Jiri, > >does that mean that there are no plans to support RFC 3841? First: what do you mean by "supporting RFC 3841" ... can't you script it? Secondly: whether as scripting or not, my individual plans are not as important as whether there is some individual on this mailing list willing to do it, there is really not a grand plan.... I think the most reasonable next step would be for someone desiring to have it to script it and share it. >This is sad because 3GPP's CSCFs require these headers, and the >OpenIMSCore project uses SER to implement its CSCFs. Well, I am unfortunately not involved in OpenIMSCore. -jiri >Regards >Stefan > > >> -----Original Message----- >> From: Jiri Kuthan [mailto:jiri@iptel.org] >> Sent: Thursday, October 18, 2007 4:03 PM >> To: Brozinski, Stefan; serusers(a)lists.iptel.org >> Subject: Re: [Serusers] RFC 3841 support? >> >> >> There is no explicit support for callerprefs. I think though >> that most of the scenarios >> they are goof for could be achieved using textops and >> selects+AVP processing. >> >> -jiri >> >> At 15:40 18/10/2007, Stefan.Brozinski(a)materna.de wrote: >> >Hello everybody >> > >> >are there any plans to support RFC 3841 in SER? >> > >> >Specifically, I am looking at the 'Accept-Contact' and the >> >'Request-Disposition' header fields. >> > >> >Thanks >> >Stefan >> >_______________________________________________ >> >Serusers mailing list >> >Serusers(a)lists.iptel.org >> >http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> -- >> Jiri Kuthan http://iptel.org/~jiri/ >> >> -- Jiri Kuthan http://iptel.org/~jiri/

3 2

[OpenSER-Users] sanitizing sip requests
by William Quan 19 Oct '07

19 Oct '07

Hi all, I came across a security alert that basically embeds javascript in the display name of the From to initiate cross-site-scripting (XSS) attacks. Here is an example: From: "<script>alert('hack')</script>""user" <sip:user at domain.com <https://lists.grok.org.uk/mailman/listinfo/full-disclosure>>;tag=002a000c Grammatically , I don't see an issue with this. However, under the right circumstances this could get ugly. Do you see value in having openser take a proactive role to detect these and reject calls? Or is this outside the scope of what a proxy should be doing (leave it to the UA to sanitize) ? Looking to get your thoughts- -will

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[OpenSER-Users] OpenSER SIP registrar - third party registration
by Pascal Maugeri 19 Oct '07

19 Oct '07

Hi I would like to make OpenSER SIP REGISTRAR to advertise my SIP server that a user just registered successfully to the domain. My server has to make a call to this user as soon as he's registered. First of all is this called "third party registration" ? In practice how does it work ? OpenSER sends a REGISTER to my server and my server replies with 200 OK ? Which script command should I use ? Thanks in advance for your help Pascal

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[OpenSER-Users] question regarding record-route and via
by vallimamod abdullah 19 Oct '07

19 Oct '07

Hello list, I am trying to understand why the thomson ST2030 hard phone does not work with my openser. I have noticed while ngreping that it seems to not respect the record-route header. I have the following incoming invite: U 2007/10/18 19:31:54.067545 xx.xx.33.119:5060 -> 10.0.0.136:15060 INVITE sip:1920@10.0.0.136:15060;user=phone SIP/2.0. Record-Route: <sip:xx.xx.33.119:5060;nat=yes;lr=on>. Via: SIP/2.0/UDP 192.168.5.119;branch=z9hG4bKae88.af6caa.0. Via: SIP/2.0/UDP 192.168.5.120:5060;branch=z9hG4bK58e93388;rport=5060. From: <sip:6897@192.168.5.120>;tag=as246d1f36. To: <sip:1920@192.168.5.119>. And the phone answers to the IP of the first via instead of the one in record-route: U 2007/10/18 19:31:54.083370 10.0.0.136:15060 -> 192.168.5.119:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.5.119;branch=z9hG4bKae88.af6caa.0. Via: SIP/2.0/UDP 192.168.5.120:5060;branch=z9hG4bK58e93388;rport=5060. From: <sip:6897@192.168.5.120>;tag=as246d1f36. To: <sip:1920@192.168.5.119>. I would like to know the opinion of SIP gourous out there: is this rfc-compliant ? I don't have this issue with other phones so I am wondering if I should ask for a refund... Thank you ! -- Regards, - vma .

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Re: [Serusers] RFC 3841 support?
by Jiri Kuthan 19 Oct '07

19 Oct '07

There is no explicit support for callerprefs. I think though that most of the scenarios they are goof for could be achieved using textops and selects+AVP processing. -jiri At 15:40 18/10/2007, Stefan.Brozinski(a)materna.de wrote: >Hello everybody > >are there any plans to support RFC 3841 in SER? > >Specifically, I am looking at the 'Accept-Contact' and the >'Request-Disposition' header fields. > >Thanks >Stefan >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

4 5

[Serusers] Documentation search problem
by Greger V. Teigre 19 Oct '07

19 Oct '07

It just came to my attention that some pages on iptel.org do not function properly, including the doc search tool. While we try to rectify the problem, here's an alternative link: http://siprouter.onsip.org/doc/docsearch/ g-)

1 0

Re: [Serusers] Get RealTime Online Users status
by roayu 19 Oct '07

19 Oct '07

Hi Jiri! I've captured and analyzed those SIP message dumps. I found that when Sjphone A sent a NOTIFY signal to the SER (after Sjphone A accepted the REFER), SER didn't pass the NOTIFY to the SPA942 to terminate the call between them. Therefore, Sjphone A unable to INVITE Sjphone B to bridge up the connection. May I know what causes this? Thanks for your precious time and reply! Cheers, Roa Yu :) -----Original Message----- From: Jiri Kuthan [mailto:jiri@iptel.org] Sent: Thursday, October 18, 2007 4:28 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Get RealTime Online Users status At 04:59 18/10/2007, roayu wrote: >Hi Jiri! > >Actually, I'm using Sjphone and SPA942 to do the call transfer. I'm calling >using Sjphone A to SPA942, then use the SPA942 to transfer the call to >Sjphone B. > > SJPhone A --> SPA942 --> SJPhone B > >After a short conversation with SJPhone B, then I press the 'xfer' button on >the 2nd time so that both SJPhone A and SJPhone B can talk to each other. >But the call couldn't be transferred to SJPhone B whereas when I used >Asterisk, it's able to do so. > >I've tried to disable the Digest-qop but it still couldn't perform the >transfer properly. What other settings that I need to configure on SER? I'm afraid you've gotta analyze SIP message dumps to figure out what's gone wrong. -jiri >Thanks and really appreciate on your reply. > >Cheers, >Roa Yu > > >-----Original Message----- >From: Jiri Kuthan [mailto:jiri@iptel.org] >Sent: Thursday, October 18, 2007 9:16 AM >To: roayu >Cc: serusers(a)lists.iptel.org >Subject: Re: [Serusers] Get RealTime Online Users status > >At 10:37 10/10/2007, roayu wrote: >>Oh, I got it! Thanks Jiri! >> >>Btw, I have another questions. >>1) I would like to know can call transfer being done when the >>connection is P2P? > >you mean without a proxy? sure it can. > >> When I tried to use relay (mediaproxy) to do the call transfer, >it's >>able to transfer the call to the other party whereas when the condition >is >>P2P, it just can't pass the correct signal. > >that's apparently unrelated to use of media proxy. you would have to >check the SIP dumps what's going wrong. Unfortunately I can't promise >I would help to review those -- my to-do-list is endless these days. >Perhaps someone else on the list will. > > >>2) Can SER support SPA942 ? Or is there some settings that I need to >>configure on SER ? > >yes. >well for some SPAs you may need to disable digest-qop since they have a bug. >(don't ask me which though). > >-jiri > > > >>Thanks. >> >>Cheers, >>Roa Yu >> >>-----Original Message----- >>From: Jiri Kuthan [mailto:jiri@iptel.org] >>Sent: Wednesday, October 10, 2007 2:56 PM >>To: roayu; SIP >>Cc: serusers(a)lists.iptel.org >>Subject: Re: [Serusers] Get RealTime Online Users status >> >>yes, set usrloc's database mode to 1. -jiri >> >>At 03:32 10/10/2007, roayu wrote: >> >>>Thanks for your reply. I found that it's only update after around 30 >>>seconds. Is there anyway that I can fasten the update on the database? >>> >>>Thanks. >>> >>>Cheers, >>>Roa Yu >>> >>>-----Original Message----- >>>From: SIP [mailto:sip@arcdiv.com] >>>Sent: Tuesday, October 09, 2007 9:14 PM >>>To: roayu >>>Cc: serusers(a)lists.iptel.org >>>Subject: Re: [Serusers] Get RealTime Online Users status >>> >>>roayu wrote: >>>> >>>> Hi there! >>>> >>>> Can anyone tell me how to get the RealTime Online user status other >>>> than using command '*serctl ul show*'? Or how can I store the realtime >>>> online user status to the MySQL db? >>>> >>>> Thanks. >>>> >>>> Cheers, >>>> >>>> Roa Yu J >>>> >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> Serusers mailing list >>>> Serusers(a)lists.iptel.org >>>> http://lists.iptel.org/mailman/listinfo/serusers >>>> >>>The location table in the MySQL database stores what are, in effect, >>>currently online users. It is, however, only an approximation. When a UA >>>registers, it sends an expiration time on its registration and that gets >>>stored in the location table. If the user disconnects without cancelling >>>his registration (some UAs send am expire register message to 'log out' >>>and some don't), then the data may still be in the table until the >>>expire time occurs. Realistically, though, you can get a good idea of >>>currenly online users using the location table and while it may not be >>>100% accurate, it's close enough for government work, as it were. >>> >>>N. >>>_______________________________________________ >>>Serusers mailing list >>>Serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >>-- >>Jiri Kuthan http://iptel.org/~jiri/ >>_______________________________________________ >>Serusers mailing list >>Serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers > > > >-- >Jiri Kuthan http://iptel.org/~jiri/ >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

2 1

[OpenSER-Users] General protection error
by Marcello Lupo 18 Oct '07

18 Oct '07

Hi to all, i've got this error in logs and openser crashed: [20670]: force_rtp_proxy2: incorrect port 0 in reply from rtp proxy [20670]: kernel: openser[20670] general protection rip:2b467ccf4150 rsp:7fff2e6d6370 error:0 [20670]: child process 20670 exited by a signal 11 [20670]: core was generated [20630]: INFO: terminating due to SIGCHLD [20640]: INFO: signal 15 received .... .... .... Can you remember me where the core is and if you need it to be analyzed? Openser is 1.2.2 no tls on SLES10 x86_64 . Thanks, Bye, Marcello

1 2

[Serusers] RTPProxy question
by Mitya 18 Oct '07

18 Oct '07

Hi all, I have a few unclarified questions in my head about RTPProxy. I think I know the basic concept, but I'm curious to know a few details. How does the proxy transfer the packets from A to B ? When it gets a packet from A, does it send it immediately to B (when its possible) without any delay or timing ? Or does it have a buffer (few ms buffer), where it re-orders the packets ? Does it parse the RTP stream ? Does it generate RTCP packets ? Can somebody help me in this ? Thanks, Mitya

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[OpenSER-Users] "200 OK" not SDP fixed by RtpProxy... but it works ! ¿?
by Iñaki Baz Castillo 18 Oct '07

18 Oct '07

Hi, user_A with STUN calls to user_B behind NAT with no STUN. onreply_route[1] { if (nat_uac_test("1")) fix_nated_contact(); if (isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") force_rtp_proxy("l"); } In the initial request bflag(6) is up because "location" of user_B so RtpProxy is used in the INVITE and 200-OK. But in re-INVITE bflag(6) is down and is not applied force_rtp_proxy("l"); in the 200-OK. In fact I debug the SIP trace in the re-INVITE in OpenSer: ---------------------------------------------------------------------- # user_A -> OpenSer INVITE sip:user_B@112.121.235.28:5061 SIP/2.0 c=IN IP4 212.121.235.18 # OpenSer -> user_B INVITE sip:806@212.121.235.18:5061 SIP/2.0 c=IN IP4 80.94.0.110 <---- RtpProxy applied (OK) # user_B -> OpenSer SIP/2.0 200 OK c=IN IP4 192.168.1.106 [*1] *1: Needs RtpProxy but bflag(6) is down so not applied. # OpenSer -> user_A SIP/2.0 200 OK c=IN IP4 192.168.1.106 <-- RtpProxy NO applied ---------------------------------------------------------------------- But I can confirm that the audio works after re-INVITE in both directions!!! ¿? I've debugged with tcpdump, the RTP is send to RtpProxy from user_A, how is possible?? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] received registration reply
by Asim Riaz 18 Oct '07

18 Oct '07

Hi, I am relaying all registration to another proxy server using t_relay, any way to save the reply from that proxy to my "location" if the authentication is successful. Regards Riaz

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[Serusers] SER 0.9.6 and MySQL connection failed
by Mitya 18 Oct '07

18 Oct '07

Hi all, I'm running SER 0.9.6 on a Debian Etch, compiled it from source, using etch's libmysqlclient5 to connect a mysql server 5.0.20 (running on a different server, but on the same LAN) >From my ser.cfg: modparam("usrloc", "db_mode", 1) modparam("usrloc", "timer_interval", 10) modparam("usrloc", "use_domain", 1) modparam("usrloc", "db_url", "mysql://ser:xxxxx@x.x.x.x:3306/ser") When I start SER, everything seems to be fine for a while. After about one day, when I unplug my UAC and the registration expires, SER can not remove it from the database because it says: Oct 18 14:47:07 sip ser[21813]: Binding '1111111(a)1.2.3.4','sip:1111111@2.3.4.5:1164' has expired Oct 18 14:47:07 sip ser[21813]: submit_query(): mysql_ping failed Oct 18 14:47:07 sip ser[21813]: submit_query(): MySQL server has gone away Oct 18 14:47:07 sip ser[21813]: db_delete: Error while submitting query Oct 18 14:47:07 sip ser[21813]: db_del_ucontact(): Error while deleting from database Oct 18 14:47:07 sip ser[21813]: wt_timer(): Error while deleting contact from database If I restart SER, everything works fine (for a while) again. I'm sure that _NO_ONE_ was restarted the MySQL server during the test period. I can reproduce it at any time. Is it a known issue, or do I have a special case ? Every comments are welcome. Thanks, Mitya

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[Serusers] RFC 3841 support?
by Stefan.Brozinski＠materna.de 18 Oct '07

18 Oct '07

Hello everybody are there any plans to support RFC 3841 in SER? Specifically, I am looking at the 'Accept-Contact' and the 'Request-Disposition' header fields. Thanks Stefan

1 0

[OpenSER-Users] Regarding on Implementation the Music On Hold With OpenSER as SIP Proxy Server.
by raviprakash sunkara 18 Oct '07

18 Oct '07

---------- Forwarded message ---------- From: RaviPrakash <ravi.prakash(a)troopsoftware.com> Date: Oct 18, 2007 3:25 PM Subject: Regarding on Implementation the Music On Hold With OpenSER as SIP Proxy Server. To: adamlinford(a)oralnet.co.uk Cc: sunkara.raviprakash.feb14(a)gmail.com Hello Regarding on Implementation the Music On Hold with Opener as SIP Proxy. As I know OpenSER is not Media Proxy Server. But I want to implement below call flow, that Bob has to Send the REFER method To Media Server ( Asterisk ). But I Don't want this ( Alice --invite_bob---> OpenSER ---- invite_bob-->Asterisk-----invite_bob---> openser---> Bob ) this scenario will work .. That User Agent has to send the REFER method to OpenSER Proxy through --> Asterisk. Alice Bob Media Proxy.. *Plz let me know, If anything is missed or something is else...* *Thanks & Regards* *Ravi Prakash Sunkara* *Troop Software* -- Thanks &Regards Ravi Prakash Sunkara VoIP Development Tech Lead

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Re: [Serusers] ser 2_0_0 lookup_user($tu.uid, @ruri) and handling of user names with unescaped semicolon.
by Jiri Kuthan 18 Oct '07

18 Oct '07

no, it is saying that "sip:alice;day=<mailto:tuesday@atlanta.com>tuesday(a)atlanta.com" is the username (consistiing of semicolon, equal sign, etc.) A strange but a valid one. The only deficit to my knowledge is not expanding % escapes. (perhaps you want to report it in JIRA?) -jiri p.s. some of the questions you had asked previously are addressed in the new oob.cfg -- feel free to check it out and give feeback. At 11:36 18/10/2007, Tomasz Zieleniewski wrote: >Hi, > >According to rfc 3261: > > > >19.1.3 Example SIP and SIPS URI > > > > sip:alice;day=<mailto:tuesday@atlanta.com> >tuesday(a)atlanta.com > > The last sample URI above has a user field value of > "alice;day=tuesday". The escaping rules defined above allow a > semicolon to appear unescaped in this field. For the purposes of > > this protocol, the field is opaque. The structure of that value is > only useful to the SIP element responsible for the resource. > > >in such case shouldn't the lookup_user($tu.uid, @ruri) method analyze only the username part before semicolon >for the lookup?? > >Waiting for Your Feedback >Bests >Tomasz >_______________________________________________ >Serusers mailing list >Serusers@lists.iptel.orghttp://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[OpenSER-Users] : Regarding on Implementation the Music On Hold With OpenSER as SIP Proxy Server.
by raviprakash sunkara 18 Oct '07

18 Oct '07

Hello Regarding on Implementation the Music On Hold with Opener as SIP Proxy. As I know OpenSER is not Media Proxy Server. But I want to implement below call flow, that Bob has to Send the REFER method To Media Server ( Asterisk ). But I Don't want this ( Alice --invite_bob---> OpenSER ---- invite_bob-->Asterisk-----invite_bob---> openser---> Bob ) this scenario will work .. That User Agent has to send the REFER method to OpenSER Proxy through --> Asterisk. Alice Bob Media Proxy.. *Plz let me know, If anything is missed or something is else...* *Thanks & Regards* *Ravi Prakash Sunkara* VoIP Development Tech Lead

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[Serusers] ser 2_0_0 lookup_user($tu.uid, @ruri) and handling of user names with unescaped semicolon.
by Tomasz Zieleniewski 18 Oct '07

18 Oct '07

Hi, According to rfc 3261: 19.1.3 Example SIP and SIPS URI sip:alice;day=tuesday@atlanta.com The last sample URI above has a user field value of "alice;day=tuesday". The escaping rules defined above allow a semicolon to appear unescaped in this field. For the purposes of this protocol, the field is opaque. The structure of that value is only useful to the SIP element responsible for the resource. in such case shouldn't the lookup_user($tu.uid, @ruri) method analyze only the username part before semicolon for the lookup?? Waiting for Your Feedback Bests Tomasz

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Re: [Serusers] Get RealTime Online Users status
by Jiri Kuthan 18 Oct '07

18 Oct '07

At 04:59 18/10/2007, roayu wrote: >Hi Jiri! > >Actually, I'm using Sjphone and SPA942 to do the call transfer. I'm calling >using Sjphone A to SPA942, then use the SPA942 to transfer the call to >Sjphone B. > > SJPhone A --> SPA942 --> SJPhone B > >After a short conversation with SJPhone B, then I press the 'xfer' button on >the 2nd time so that both SJPhone A and SJPhone B can talk to each other. >But the call couldn't be transferred to SJPhone B whereas when I used >Asterisk, it's able to do so. > >I've tried to disable the Digest-qop but it still couldn't perform the >transfer properly. What other settings that I need to configure on SER? I'm afraid you've gotta analyze SIP message dumps to figure out what's gone wrong. -jiri >Thanks and really appreciate on your reply. > >Cheers, >Roa Yu > > >-----Original Message----- >From: Jiri Kuthan [mailto:jiri@iptel.org] >Sent: Thursday, October 18, 2007 9:16 AM >To: roayu >Cc: serusers(a)lists.iptel.org >Subject: Re: [Serusers] Get RealTime Online Users status > >At 10:37 10/10/2007, roayu wrote: >>Oh, I got it! Thanks Jiri! >> >>Btw, I have another questions. >>1) I would like to know can call transfer being done when the >>connection is P2P? > >you mean without a proxy? sure it can. > >> When I tried to use relay (mediaproxy) to do the call transfer, >it's >>able to transfer the call to the other party whereas when the condition >is >>P2P, it just can't pass the correct signal. > >that's apparently unrelated to use of media proxy. you would have to >check the SIP dumps what's going wrong. Unfortunately I can't promise >I would help to review those -- my to-do-list is endless these days. >Perhaps someone else on the list will. > > >>2) Can SER support SPA942 ? Or is there some settings that I need to >>configure on SER ? > >yes. >well for some SPAs you may need to disable digest-qop since they have a bug. >(don't ask me which though). > >-jiri > > > >>Thanks. >> >>Cheers, >>Roa Yu >> >>-----Original Message----- >>From: Jiri Kuthan [mailto:jiri@iptel.org] >>Sent: Wednesday, October 10, 2007 2:56 PM >>To: roayu; SIP >>Cc: serusers(a)lists.iptel.org >>Subject: Re: [Serusers] Get RealTime Online Users status >> >>yes, set usrloc's database mode to 1. -jiri >> >>At 03:32 10/10/2007, roayu wrote: >> >>>Thanks for your reply. I found that it's only update after around 30 >>>seconds. Is there anyway that I can fasten the update on the database? >>> >>>Thanks. >>> >>>Cheers, >>>Roa Yu >>> >>>-----Original Message----- >>>From: SIP [mailto:sip@arcdiv.com] >>>Sent: Tuesday, October 09, 2007 9:14 PM >>>To: roayu >>>Cc: serusers(a)lists.iptel.org >>>Subject: Re: [Serusers] Get RealTime Online Users status >>> >>>roayu wrote: >>>> >>>> Hi there! >>>> >>>> Can anyone tell me how to get the RealTime Online user status other >>>> than using command '*serctl ul show*'? Or how can I store the realtime >>>> online user status to the MySQL db? >>>> >>>> Thanks. >>>> >>>> Cheers, >>>> >>>> Roa Yu J >>>> >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> Serusers mailing list >>>> Serusers(a)lists.iptel.org >>>> http://lists.iptel.org/mailman/listinfo/serusers >>>> >>>The location table in the MySQL database stores what are, in effect, >>>currently online users. It is, however, only an approximation. When a UA >>>registers, it sends an expiration time on its registration and that gets >>>stored in the location table. If the user disconnects without cancelling >>>his registration (some UAs send am expire register message to 'log out' >>>and some don't), then the data may still be in the table until the >>>expire time occurs. Realistically, though, you can get a good idea of >>>currenly online users using the location table and while it may not be >>>100% accurate, it's close enough for government work, as it were. >>> >>>N. >>>_______________________________________________ >>>Serusers mailing list >>>Serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >>-- >>Jiri Kuthan http://iptel.org/~jiri/ >>_______________________________________________ >>Serusers mailing list >>Serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers > > > >-- >Jiri Kuthan http://iptel.org/~jiri/ >_______________________________________________ >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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Re: [OpenSER-Users] OpenXcap fails to start the HTTP on port
by Cesar Troncoso 18 Oct '07

18 Oct '07

Hi, try telnet to be sure if no one is listening on that port. [root@XXXXX]# telnet yourhost 8000 if you are not running openxcap in debug mode, check out the messages to see if your xcap server is really not listening. Later on just set up the Root URI properly on both, the server and your client. Cheers, Cesar ------------------------------ Message: 2 Date: Wed, 17 Oct 2007 13:11:47 +0200 From: "Gregorio" <goyo.villalba(a)dialcom.com> Subject: [OpenSER-Users] OpenXcap fails to start the HTTP on port To: <users(a)openser.org> Message-ID: <003e01c810ae$825c4c60$8714e520$@villalba@dialcom.com> Content-Type: text/plain; charset="iso-8859-1" Hi all. Finally (buff!!) I could configure openXcap, after executing openxcap start the log says: [-] Log opened. [-] Starting Open XCAP 0.9.4 [-] Supported Root URIs: http://192.168.1.206 [-] twisted.web2.channel.http.HTTPFactory starting on 8000 And nothing else, no more logs displayed. OpenXcap process is running in the machine: [root@XXXXX]# ps -ef|grep xcap root 926 1 0 18:02 ? 00:00:00 /usr/bin/python /etc/rc.d/init.d/openxcap start But no one is listening in port 8000, neither before executing xcap nor after. I havent found any error in any log so I dont know what is going on. Anyone can help me? Thanks. Gregorio

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Re: [Serusers] Get RealTime Online Users status
by roayu 18 Oct '07

18 Oct '07

Hi Jiri! Actually, I'm using Sjphone and SPA942 to do the call transfer. I'm calling using Sjphone A to SPA942, then use the SPA942 to transfer the call to Sjphone B. SJPhone A --> SPA942 --> SJPhone B After a short conversation with SJPhone B, then I press the 'xfer' button on the 2nd time so that both SJPhone A and SJPhone B can talk to each other. But the call couldn't be transferred to SJPhone B whereas when I used Asterisk, it's able to do so. I've tried to disable the Digest-qop but it still couldn't perform the transfer properly. What other settings that I need to configure on SER? Thanks and really appreciate on your reply. Cheers, Roa Yu -----Original Message----- From: Jiri Kuthan [mailto:jiri@iptel.org] Sent: Thursday, October 18, 2007 9:16 AM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Get RealTime Online Users status At 10:37 10/10/2007, roayu wrote: >Oh, I got it! Thanks Jiri! > >Btw, I have another questions. >1) I would like to know can call transfer being done when the >connection is P2P? you mean without a proxy? sure it can. > When I tried to use relay (mediaproxy) to do the call transfer, it's >able to transfer the call to the other party whereas when the condition is >P2P, it just can't pass the correct signal. that's apparently unrelated to use of media proxy. you would have to check the SIP dumps what's going wrong. Unfortunately I can't promise I would help to review those -- my to-do-list is endless these days. Perhaps someone else on the list will. >2) Can SER support SPA942 ? Or is there some settings that I need to >configure on SER ? yes. well for some SPAs you may need to disable digest-qop since they have a bug. (don't ask me which though). -jiri >Thanks. > >Cheers, >Roa Yu > >-----Original Message----- >From: Jiri Kuthan [mailto:jiri@iptel.org] >Sent: Wednesday, October 10, 2007 2:56 PM >To: roayu; SIP >Cc: serusers(a)lists.iptel.org >Subject: Re: [Serusers] Get RealTime Online Users status > >yes, set usrloc's database mode to 1. -jiri > >At 03:32 10/10/2007, roayu wrote: > >>Thanks for your reply. I found that it's only update after around 30 >>seconds. Is there anyway that I can fasten the update on the database? >> >>Thanks. >> >>Cheers, >>Roa Yu >> >>-----Original Message----- >>From: SIP [mailto:sip@arcdiv.com] >>Sent: Tuesday, October 09, 2007 9:14 PM >>To: roayu >>Cc: serusers(a)lists.iptel.org >>Subject: Re: [Serusers] Get RealTime Online Users status >> >>roayu wrote: >>> >>> Hi there! >>> >>> Can anyone tell me how to get the RealTime Online user status other >>> than using command '*serctl ul show*'? Or how can I store the realtime >>> online user status to the MySQL db? >>> >>> Thanks. >>> >>> Cheers, >>> >>> Roa Yu J >>> >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Serusers mailing list >>> Serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >>> >>The location table in the MySQL database stores what are, in effect, >>currently online users. It is, however, only an approximation. When a UA >>registers, it sends an expiration time on its registration and that gets >>stored in the location table. If the user disconnects without cancelling >>his registration (some UAs send am expire register message to 'log out' >>and some don't), then the data may still be in the table until the >>expire time occurs. Realistically, though, you can get a good idea of >>currenly online users using the location table and while it may not be >>100% accurate, it's close enough for government work, as it were. >> >>N. >>_______________________________________________ >>Serusers mailing list >>Serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers > > > >-- >Jiri Kuthan http://iptel.org/~jiri/ >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

1 0

Re: [Serusers] Get RealTime Online Users status
by Jiri Kuthan 18 Oct '07

18 Oct '07

At 10:37 10/10/2007, roayu wrote: >Oh, I got it! Thanks Jiri! > >Btw, I have another questions. >1) I would like to know can call transfer being done when the >connection is P2P? you mean without a proxy? sure it can. > When I tried to use relay (mediaproxy) to do the call transfer, it's >able to transfer the call to the other party whereas when the condition is >P2P, it just can't pass the correct signal. that's apparently unrelated to use of media proxy. you would have to check the SIP dumps what's going wrong. Unfortunately I can't promise I would help to review those -- my to-do-list is endless these days. Perhaps someone else on the list will. >2) Can SER support SPA942 ? Or is there some settings that I need to >configure on SER ? yes. well for some SPAs you may need to disable digest-qop since they have a bug. (don't ask me which though). -jiri >Thanks. > >Cheers, >Roa Yu > >-----Original Message----- >From: Jiri Kuthan [mailto:jiri@iptel.org] >Sent: Wednesday, October 10, 2007 2:56 PM >To: roayu; SIP >Cc: serusers(a)lists.iptel.org >Subject: Re: [Serusers] Get RealTime Online Users status > >yes, set usrloc's database mode to 1. -jiri > >At 03:32 10/10/2007, roayu wrote: > >>Thanks for your reply. I found that it's only update after around 30 >>seconds. Is there anyway that I can fasten the update on the database? >> >>Thanks. >> >>Cheers, >>Roa Yu >> >>-----Original Message----- >>From: SIP [mailto:sip@arcdiv.com] >>Sent: Tuesday, October 09, 2007 9:14 PM >>To: roayu >>Cc: serusers(a)lists.iptel.org >>Subject: Re: [Serusers] Get RealTime Online Users status >> >>roayu wrote: >>> >>> Hi there! >>> >>> Can anyone tell me how to get the RealTime Online user status other >>> than using command '*serctl ul show*'? Or how can I store the realtime >>> online user status to the MySQL db? >>> >>> Thanks. >>> >>> Cheers, >>> >>> Roa Yu J >>> >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Serusers mailing list >>> Serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >>> >>The location table in the MySQL database stores what are, in effect, >>currently online users. It is, however, only an approximation. When a UA >>registers, it sends an expiration time on its registration and that gets >>stored in the location table. If the user disconnects without cancelling >>his registration (some UAs send am expire register message to 'log out' >>and some don't), then the data may still be in the table until the >>expire time occurs. Realistically, though, you can get a good idea of >>currenly online users using the location table and while it may not be >>100% accurate, it's close enough for government work, as it were. >> >>N. >>_______________________________________________ >>Serusers mailing list >>Serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers > > > >-- >Jiri Kuthan http://iptel.org/~jiri/ >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

1 0

[OpenSER-Users] avp_load_radius('callee') returns 'caller' AVPs... and with 'rc_auth failure'
by Edson 18 Oct '07

18 Oct '07

Hi, guys... I'm starting to move my config from a MySQL based one to a Radius one. The AVP are the challenge. Authentication works fine but 'callee' AVP recovery don't. I'm getting "rc_auth failure" and 'caller' AVPs instead of 'callee' AVPs. Bellow are the detailed informations... openser 1.3.0-dev21-tls (i386/linux) svnrevision: 2:2921M running in a VMWare, guested on an OpenSUSE 10.2, on a P4 machine. Why this behavior? Bug? Radius mis-configuration? Or what am I doing something wrong on the scripts? Edson. =============================================================== INVITE message: INVITE sip:1234@domain.test SIP/2.0. Via: SIP/2.0/UDP 172.27.248.6;branch=z9hG4bKac1....00000018;rport. From: "unknown" <sip:8999@domain.test>;tag=2bf01d7afd4. To: <sip:1234@domain.test>. Contact: <sip:8999@172.27.248.6>. This scripts lines: ... avp_load_radius('callee'); avp_printf(); ... Gives this LOGs: ***on /var/log/messages:*** avpops:ops_print_avp: p=0xb59c78e0, flags=0x0003 avpops:ops_print_avp: name=<caller_cli> avpops:ops_print_avp: val_str=<"010101(a)domain.test" / 19> avpops:ops_print_avp: p=0xb59c74d0, flags=0x0003 avpops:ops_print_avp: name=<acc_state> avpops:ops_print_avp: val_str=<call / 4> avpops:ops_print_avp: p=0xb59c7870, flags=0x0003 avpops:ops_print_avp: name=<acc_caller_domain> avpops:ops_print_avp: val_str=<domain.test / 10> avpops:ops_print_avp: p=0xb59c7598, flags=0x0003 avpops:ops_print_avp: name=<acc_caller_user> avpops:ops_print_avp: val_str=<8999 / 4> ***on Radius console ("radius -X"):*** rad_recv: Access-Request packet from host 127.0.0.1:1024, id=31, length=55 User-Name = "1234(a)domain.test" Service-Type = SIP-Callee-AVPs NAS-Port = 0 NAS-IP-Address = 127.0.0.1 Processing the authorize section of radiusd.conf modcall: entering group authorize for request 250 modcall[authorize]: module "preprocess" returns ok for request 250 radius_xlat: '/var/log/radius/radacct/127.0.0.1/auth-detail-20071017' rlm_detail: /var/log/radius/radacct/%{Client-IP-Address}/auth-detail-%Y%m%d expands to /var/log/radius/radacct/127.0.0.1/auth-detail-20071017 modcall[authorize]: module "auth_log" returns ok for request 250 rlm_realm: Looking up realm "domain.test" for User-Name = "1234(a)domain.test" rlm_realm: No such realm "domain.test" modcall[authorize]: module "suffix" returns noop for request 250 modcall[authorize]: module "digest" returns noop for request 250 radius_xlat: '1234(a)domain.test' rlm_sql (sql): sql_set_user escaped user --> '1234(a)domain.test' radius_xlat: 'SELECT id,UserName,Attribute,Value,op FROM radcheck WHERE Username = '1234(a)domain.test' ORDER BY id' rlm_sql (sql): Reserving sql socket id: 4 rlm_sql_mysql: query: SELECT id,UserName,Attribute,Value,op FROM radcheck WHERE Username = '1234(a)domain.test' ORDER BY id radius_xlat: 'SELECT radgroupcheck.id,radgroupcheck.GroupName,radgroupcheck.Attribute,radgroupche ck.Value,radgroupcheck.op FROM radgroupcheck,usergroup WHERE usergroup.Username = '1234(a)domain.test' AND usergroup.GroupName = radgroupcheck.GroupName ORDER BY radgroupcheck.id' rlm_sql_mysql: query: SELECT radgroupcheck.id,radgroupcheck.GroupName,radgroupcheck.Attribute,radgroupche ck.Value,radgroupcheck.op FROM radgroupcheck,usergroup WHERE usergroup.Username = '1234(a)domain.test' AND usergroup.GroupName = radgroupcheck.GroupName ORDER BY radgroupcheck.id rlm_sql: Failed to create the pair: Unknown attribute "Simultaneus-Access" rlm_sql (sql): Error getting data from database radius_xlat: 'SELECT id,UserName,Attribute,Value,op FROM radreply WHERE Username = '1234(a)domain.test' ORDER BY id' rlm_sql_mysql: query: SELECT id,UserName,Attribute,Value,op FROM radreply WHERE Username = '1234(a)domain.test' ORDER BY id radius_xlat: 'SELECT radgroupreply.id,radgroupreply.GroupName,radgroupreply.Attribute,radgrouprep ly.Value,radgroupreply.op FROM radgroupreply,usergroup WHERE usergroup.Username = '1234(a)domain.test' AND usergroup.GroupName = radgroupreply.GroupName ORDER BY radgroupreply.id' rlm_sql_mysql: query: SELECT radgroupreply.id,radgroupreply.GroupName,radgroupreply.Attribute,radgrouprep ly.Value,radgroupreply.op FROM radgroupreply,usergroup WHERE usergroup.Username = '1234(a)domain.test' AND usergroup.GroupName = radgroupreply.GroupName ORDER BY radgroupreply.id rlm_sql (sql): Released sql socket id: 4 modcall[authorize]: module "sql" returns ok for request 250 modcall: leaving group authorize (returns ok) for request 250 auth: type Local auth: No User-Password or CHAP-Password attribute in the request auth: Failed to validate the user. Login incorrect: [1234(a)domain.test] (from client localhost port 0) Delaying request 250 for 1 seconds

1 0

[OpenSER-Users] radius_is_user_in and avps
by Morten Isaksen 17 Oct '07

17 Oct '07

Hi! I have some problems with avps in the radius_is_user_in function. if (radius_is_user_in("Request-URI", "$avp(s:test)")) { xlog("test"); } On the radius server I can see that the string "$avp(s:test)" is passed and not the value of the avp. Does the function not supports avps or am I missing something? -- Morten Isaksen http://www.misak.dk/blog/

1 1

[OpenSER-Users] Is possible to use two "branch_route[ ]" for same branch?
by Iñaki Baz Castillo 17 Oct '07

17 Oct '07

Hi, I use "branch_route[1]" for common branches and want to use "branch_route[3]" just for initial INVITE's. But I'd like "branch_route[1]" is runned after "branch_route[3]" but think is not possible, so I must copy&paste the "branch_route[1]" code at the end of "branch_route[3]". Is it? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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RE: [OpenSER-Users] openxcap Problem
by Gregorio 17 Oct '07

17 Oct '07

Hi. I am not getting the exception now, my libraries related to this problem are: openxcap-0.9.4 libxml2-2.6.30 lxml-1.3.4 lxml-1.3.4 I hope it helps. Gregorio. -----Mensaje original----- De: users-bounces(a)openser.org [mailto:users-bounces@openser.org] En nombre de Gregorio Enviado el: martes, 16 de octubre de 2007 13:03 Para: 'KevinKinnan'; users(a)openser.org Asunto: RE: [OpenSER-Users] openxcap Problem Hi all. I am also facing the same problem, I use a Centos 5, so it is not a Debian specific problem, my libraries: openxcap-0.9.3 libxml2-2.6.29 lxml-1.3.3 Latest version of Twisted Core, Twisted Web and Twisted Web 2. Anyone could help? Cheers. Gregorio. -----Mensaje original----- De: users-bounces(a)openser.org [mailto:users-bounces@openser.org] En nombre de KevinKinnan Enviado el: jueves, 11 de octubre de 2007 16:33 Para: users(a)openser.org Asunto: Re: [OpenSER-Users] openxcap Problem Hi, Sebastian. I met the same problem as yours. I am using Ubuntu 7.04 at the moment. Have you fixed that problem? I dont if the problem comes from xml schema checking via http with proxy. Can you give me some help? THANK U Regards, Kevin Schumann Sebastian wrote: > > Dear all > > I have a big problem with openxcap. I don't get it work :( As I saw on > their webpage, this mailing list can be posted for this concern as well. > > I am using version 0.9.3 and acc. the installation file on > http://www.openxcap.org all current dependencies from the Debian > unstable distribution. > > When I start /usr/bin/openxcap, the following error occurs in the > syslog: > > Sep 25 17:01:21 openxcap openxcap[11876]: [-] Log opened. > > Sep 25 17:01:21 openxcap openxcap[11876]: [-] Starting Open XCAP 0.9.3 > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] Traceback (most recent > call last): > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] File > "/usr/bin/openxcap", line 56, in ? > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap.server > import XCAPServer > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] File > "/usr/lib/python2.4/site-packages/xcap/server.py", line 21, in ? > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap import > authentication > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] File > "/usr/lib/python2.4/site-packages/xcap/authentication.py", line 21, in ? > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap.appusage > import getApplicationForURI > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] File > "/usr/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 466, > in ? > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] applications = > {'xcap-caps': XCAPCapabilitiesApplication(), > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] File > "/usr/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 64, > in __init__ > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] self.xml_schema = > etree.XMLSchema(xml_schema_doc) > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "xmlschema.pxi", > line 67, in etree.XMLSchema.__init__ > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] etree.XMLSchemaParseError: > Document is not valid XML Schema > > > > As I wrote, I followed all instructions from the openxcap webpage but > didn't find any hints how to make it work. > > > > Any help on that would be appreciated. > > > > Best regards > > Sebastian > > > -- > Sebastian Schumann > Diploma Student Architecture and Design Department > > Slovak Telekom, a. s. > T-Com, Operations Unit > Production and Service Division > Innovations and Enterprise Solution Subunit > > Address: Namestie Slobody 6, 817 62 Bratislava, Slovakia > Office: Room 449, Jarabinkova 1, 821 09 Bratislava, Slovakia > > +421 2 588 13332 (tel) > +421 910 643010 (mobile) > +49 175 1925928 (mobile) > > sebastian.schumann(a)t-com.sk > http://www.t-com.sk <http://www.t-com.sk/> > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > -- View this message in context: http://www.nabble.com/openxcap-Problem-tf4515559.html#a13157095 Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com. _______________________________________________ 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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[OpenSER-Users] OpenXcap fails to start the HTTP on port
by Gregorio 17 Oct '07

17 Oct '07

Hi all. Finally (buff!!) I could configure openXcap, after executing openxcap start the log says: [-] Log opened. [-] Starting Open XCAP 0.9.4 [-] Supported Root URIs: http://192.168.1.206 [-] twisted.web2.channel.http.HTTPFactory starting on 8000 And nothing else, no more logs displayed. OpenXcap process is running in the machine: [root@XXXXX]# ps -ef|grep xcap root 926 1 0 18:02 ? 00:00:00 /usr/bin/python /etc/rc.d/init.d/openxcap start But no one is listening in port 8000, neither before executing xcap nor after. I haven´t found any error in any log so I dont know what is going on. Anyone can help me? Thanks. Gregorio

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[OpenSER-Users] RE: Users Digest, Vol 29, Issue 54
by Papadopoulos Georgios 17 Oct '07

17 Oct '07

Hello Inaki, I wasn't aware of this parameter and I am using the deault (matching contact only). However this does not aswer my question. Sipsak has a default expires of 15 sec for the registration. I run sipsak every 2 minutes. So every time that sipsak runs, Openser should not find any valid entries in usrloc. Also note that I am using the default cseq_delay which is 20 sec. Any other ideas? thank you George > Date: Wed, 17 Oct 2007 10:09:51 +0200 > From: I?aki Baz Castillo <ibc(a)in.ilimit.es> > Subject: Re: [OpenSER-Users] using sipsak to check openser > To: users(a)openser.org > Message-ID: <200710171009.51575.ibc(a)in.ilimit.es> > Content-Type: text/plain; charset="ISO-8859-1" > > El Wednesday 17 October 2007 09:51:43 Papadopoulos Georgios escribiσ: > > > sipsak is running on the same server with Openser. Since > the CSeq is 2 > > it means that Openser has received the first REGISTER, has > asked for > > authentication, sipsak replied, but then Openser says "Invalid CSeq > > number". Has anyone seen this happen before and/or has a reasonable > > explanation? > > Not sure, but maybe this can help you: > > > http://www.openser.org/docs/modules/1.3.x/usrloc.html#CONTACT- > MATCHING-ALGS > > > > > -- > Iρaki Baz Castillo > ibc(a)in.ilimit.es > > > Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

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[OpenSER-Users] SRV records
by Antonio Reale 17 Oct '07

17 Oct '07

Hi all, I tried to implement redundancy with SRV records configured with different priority and same weight. If I register the client on the first proxy and then I stop OpenSER on it, the client continues to send requests to the first proxy. Only after sending about ten requests to first proxy it switches on the second proxy. Is it normal? Thanks. Regards, Antonio.

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[Serusers] Rewriting "To" uri
by geethanjali.csg＠wipro.com 17 Oct '07

17 Oct '07

Hi, Can some one tell me if its possible to rewrite the To uri in ser. I am forwarding the request from SER to openSBC which is expecting the same Request URI and To URI fields. So the SBC is looping back the call to SER. Can some give me some pointers to solve this issue. Regards, Geethanjali 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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Re: [Serusers] SER + Linksys SPA942 + Call transfer
by roayu 17 Oct '07

17 Oct '07

Hi there! Can anyone tell me is there anyway to control the client(UA) to send the INVITE signal before sending the NOTIFY signal to the SER ? Or probably configure SER to accept the INVITE then only accept the NOTIFY signal? Thanks. Cheers, Roa Yu :) -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of roayu Sent: Tuesday, October 16, 2007 11:41 AM To: serusers(a)lists.iptel.org Subject: Re: [Serusers] SER + Linksys SPA942 + Call transfer Hi there! I'm using ser0.9.6 to do the call transfer, which part of the configuration that I need to configure in ser.cfg? Do I need to create a new route? Thanks. Cheers, Roa Yu :) -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of roayu Sent: Friday, October 12, 2007 2:45 PM To: kgfleischmann(a)t-online.de Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] SER + Linksys SPA942 + Call transfer Hi Klaus! The UA does not received the REFER sent by SPA942. What should I need to configure to ensure that the UA receive the REFER from the transfer agent(SPA942)? Thanks. Cheers, Roa Yu -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of kgfleischmann(a)t-online.de Sent: Friday, October 12, 2007 2:22 PM To: serusers(a)lists.iptel.org Subject: Re: [Serusers] SER + Linksys SPA942 + Call transfer The NOTIFY is always sent from the UA, which received the REFER. SER only relays it. Did you check this? Cheers Klaus roayu wrote: > Hi there! > > After SPA942(transfer agent) sent REFER to SER, SER does not send a NOTIFY > to SPA942(transfer agent). What's causes this? Can anyone tell me. > > Thanks. > > Cheers, > Roa Yu :) > > -----Original Message----- > From: serusers-bounces(a)lists.iptel.org > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of roayu > Sent: Thursday, October 11, 2007 4:34 PM > To: Atle Samuelsen > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] SER + Linksys SPA942 + Call transfer > > > Hi Atle, > > Let me make my scenario clear on you. > > I have 2 softphone and 1 IP phone(SPA942). I would like to call from > Softphone A to SPA942, then use SPA942 transfer the call to Softphone B. > But, when I do so, the connection still remained on between SPA942 and > Softphone B. > > I managed to get Softphone A onHold and the SPA942 managed to call to > Softphone B. Once I press on the 'xfer' button after calling Softphone B, A > still onHold and no SIP signal to unHold A and to terminate B. > > If I would like to configure the ser.cfg file, which part that I need to > configure? And how do I do that? > > Thanks. > > Cheers, > Roa Yu > > > -----Original Message----- > From: Atle Samuelsen [mailto:clona@cyberhouse.no] > Sent: Thursday, October 11, 2007 3:19 PM > To: roayu > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] SER + Linksys SPA942 + Call transfer > > > Hi Roa,i > * roayu <roayu(a)ctisys.net> [071011 04:01]: >> Hi, there! >> >> I have some questions and help. >> 1) I would like to know can call transfer being done when the >> connection is P2P? When I tried to use relay (mediaproxy) to do the call >> transfer, it's able to transfer the call to the other party whereas when > the >> condition is P2P, it just can't pass the correct signal. > > I'm proberbly only tierd (9:15 am here) but P2P? Can you provide the > signalling and ser.cfg so we can understand what you are trying to > establish? > > I personally think you are trying to do a call transfer from one ua to a > other, where the original call went true mediaproxy, but you do not want > the "new" transferd call to go true it. > >> 2) Can SER support SPA942 ? Or is there some other settings that I need >> to configure on SER ? > > SPA942, SPA962, SPA2102, yea.. all Linksys SPA products work like a > charm with SER (Who has a 942 as his primary phone these days) > > > Best Regards > > ATle >> Thanks. >> >> Cheers, >> Roa Yu >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[OpenSER-Users] using sipsak to check openser
by Papadopoulos Georgios 17 Oct '07

17 Oct '07

Hello all, In order to check that our Openser is running properly I use sipsak periodically. The command is : sipsak -v -N -U -I -s sip:sipsak@213.5.xxx.xxx:5060 -a xxxxxxx This works fine 99% of the time, however sometimes I get the following from sipsak : warning: ignoring -i option when in usrloc mode received: SIP/2.0 400 Bad Request Via: SIP/2.0/UDP 127.0.0.1:54393;branch=z9hG4bK.4d383307;rport=54393;alias;received=213.5 .43.131 From: sip:sipsak@213.5.43.131:5060;tag=44d4f5ef To: sip:sipsak@213.5.43.131:5060;tag=97e44c910458bd56beb40ca4028d7cc8.91db Call-ID: 1154807279(a)127.0.0.1 CSeq: 2 REGISTER P-Registrar-Error: Invalid CSeq number Server: Altec Telecoms SIP Proxy Content-Length: 0 error: didn't received '200 OK' on register (see above). aborting SIP failure sipsak is running on the same server with Openser. Since the CSeq is 2 it means that Openser has received the first REGISTER, has asked for authentication, sipsak replied, but then Openser says "Invalid CSeq number". Has anyone seen this happen before and/or has a reasonable explanation? thank you George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

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[OpenSER-Users] Reply vs. failure events.
by Alex Balashov 16 Oct '07

16 Oct '07

If I am not using a failure_route to catch failed responses to transactional events, but have an onreply_route, does that guarantee that *all* failures will be caught by the onreply_route without being diverted and muffled somewhere? Based on the behaviour I am witnessing, it seems that is correct, but I wanted to verify. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671

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[OpenSER-Users] How to visualice a flag in easy way?
by Iñaki Baz Castillo 16 Oct '07

16 Oct '07

Hi, for now, the only way to test flags is by $bf (not useful if there are enabled some bflags) or doing something as: if (isbflagset(6)) { $var(bflag6) = 1; } xlog("L_INFO","--- $var(bflag6) ----\n"); As a suggestion, couldn't be a way to access to the value of a flag like: xlog("L_INFO","--- $bf[6] ----\n"); This could be useful to when matching flgas with table fields and so. Regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

1 0

[OpenSER-Users] About efficiency in conditionals
by Iñaki Baz Castillo 16 Oct '07

16 Oct '07

Hi, I'm wondering which solution (A or B) is more efficient in case method is NOT an INVITE: A) if ( (method=="INVITE") && (isflagset(5) || isbflagset(6)) ) B) if ( mehtod=="INVITE" ) My question is: Is faster B because does less comparasions? or does OpenSer break A conditional during first condition (method=="INVITE") because it's false so then A and B are equally efficient? And which is more efficient (C or D) when all the conditions are true? C) if ( (method=="INVITE") && (isflagset(5) || isbflagset(6)) ) { .... D) if ( method=="INVITE" ) { if ( (isflagset(5) || isbflagset(6)) ) { ... Thanks for any explanation. Best regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] 500 Server error occurred (1/SL)
by Cseke Tamas 16 Oct '07

16 Oct '07

Hello, I get a similar problem like this: http://www.openser.org/pipermail/devel/2007-February/005984.html I get the following error log, and packets: http://pastebin.com/f112f652a <http://pastebin.com/f112f652a> Thanks any help, Tamas

1 0

[OpenSER-Users] Multidomain and in-dialog REFER auth issue
by Iñaki Baz Castillo 16 Oct '07

16 Oct '07

Hi, I've a multidomain OpenSer and I want to allow REFER (with authenticacion before) just in calls between users of same domain, but it's quite complex: - userA(a)domainA.com INVITE's userB(a)domainB.com. - During the call userA(a)domainA.com does an in-dialog REFER. Because the REFER is in-dialog the message is like: -------------------- REFER sip:userB@80.98.123.23:5060 SIP/2.0 From: sip:userA@domainA.com To: userB(a)domainB.com Refer-To: ... ... -------------------- How can my OpenSer know which domain this REFER goes? of course the "To" header is not valid at all. Could be a solution a SQL query to "location" table looking for the URI (sip:userB@80.98.123.23:5060) and getting the username and domain of this? Any other option? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] Do AVP's really exist in entire transaction?
by Iñaki Baz Castillo 16 Oct '07

16 Oct '07

Hi, in order to not apply RtpPrxy if the called is an Asterisk (that supports "comedia") I load "usr_preferences" table looking for "pbx_comedia" attribute for called domain. If it's, then it's created a avp: avp(s:pbx_comedia)="yes". But when I try to test this AVP in the "on_reply_route" (to not apply RtpProxy) this AVP has null value. In fact it's easy to see: ------------------------------------------------------ route[3] { ; INVITE route $avp(s:test)="hello"; ... route(1); } onreply_route[1] { xlog("L_INFO","----- avp(s:test) = $avp(s:test) ------\n"); } ------------------------------------------------------ The xlog of onreply_route shows: ----- avp(s_test) = null ----- is it normal? aren't AVP's transaction aware? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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Re: [Serusers] SER + Linksys SPA942 + Call transfer
by roayu 16 Oct '07

16 Oct '07

Hi there! I'm using ser0.9.6 to do the call transfer, which part of the configuration that I need to configure in ser.cfg? Do I need to create a new route? Thanks. Cheers, Roa Yu :) -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of roayu Sent: Friday, October 12, 2007 2:45 PM To: kgfleischmann(a)t-online.de Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] SER + Linksys SPA942 + Call transfer Hi Klaus! The UA does not received the REFER sent by SPA942. What should I need to configure to ensure that the UA receive the REFER from the transfer agent(SPA942)? Thanks. Cheers, Roa Yu -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of kgfleischmann(a)t-online.de Sent: Friday, October 12, 2007 2:22 PM To: serusers(a)lists.iptel.org Subject: Re: [Serusers] SER + Linksys SPA942 + Call transfer The NOTIFY is always sent from the UA, which received the REFER. SER only relays it. Did you check this? Cheers Klaus roayu wrote: > Hi there! > > After SPA942(transfer agent) sent REFER to SER, SER does not send a NOTIFY > to SPA942(transfer agent). What's causes this? Can anyone tell me. > > Thanks. > > Cheers, > Roa Yu :) > > -----Original Message----- > From: serusers-bounces(a)lists.iptel.org > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of roayu > Sent: Thursday, October 11, 2007 4:34 PM > To: Atle Samuelsen > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] SER + Linksys SPA942 + Call transfer > > > Hi Atle, > > Let me make my scenario clear on you. > > I have 2 softphone and 1 IP phone(SPA942). I would like to call from > Softphone A to SPA942, then use SPA942 transfer the call to Softphone B. > But, when I do so, the connection still remained on between SPA942 and > Softphone B. > > I managed to get Softphone A onHold and the SPA942 managed to call to > Softphone B. Once I press on the 'xfer' button after calling Softphone B, A > still onHold and no SIP signal to unHold A and to terminate B. > > If I would like to configure the ser.cfg file, which part that I need to > configure? And how do I do that? > > Thanks. > > Cheers, > Roa Yu > > > -----Original Message----- > From: Atle Samuelsen [mailto:clona@cyberhouse.no] > Sent: Thursday, October 11, 2007 3:19 PM > To: roayu > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] SER + Linksys SPA942 + Call transfer > > > Hi Roa,i > * roayu <roayu(a)ctisys.net> [071011 04:01]: >> Hi, there! >> >> I have some questions and help. >> 1) I would like to know can call transfer being done when the >> connection is P2P? When I tried to use relay (mediaproxy) to do the call >> transfer, it's able to transfer the call to the other party whereas when > the >> condition is P2P, it just can't pass the correct signal. > > I'm proberbly only tierd (9:15 am here) but P2P? Can you provide the > signalling and ser.cfg so we can understand what you are trying to > establish? > > I personally think you are trying to do a call transfer from one ua to a > other, where the original call went true mediaproxy, but you do not want > the "new" transferd call to go true it. > >> 2) Can SER support SPA942 ? Or is there some other settings that I need >> to configure on SER ? > > SPA942, SPA962, SPA2102, yea.. all Linksys SPA products work like a > charm with SER (Who has a 942 as his primary phone these days) > > > Best Regards > > ATle >> Thanks. >> >> Cheers, >> Roa Yu >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

Re: [OpenSER-Users] Insert in location table very slow
by Ovidiu Sas 15 Oct '07

15 Oct '07

That's normal. Set the db_mode to 1 if you want instant updates: http://www.openser.org/docs/modules/1.3.x/usrloc.html#AEN285 Regards, Ovidiu Sas On 10/15/07, Gregorio <goyo.villalba(a)dialcom.com> wrote: > Hi, > > Up to 60 seconds in mode 2: 'modparam("usrloc", "db_mode", 2)' > > Gregorio. > > -----Mensaje original----- > De: Alex Balashov [mailto:abalashov@evaristesys.com] > Enviado el: lunes, 15 de octubre de 2007 18:48 > Para: Gregorio > CC: users(a)openser.org > Asunto: Re: [OpenSER-Users] Insert in location table very slow > > > Gregorio, > > How long does it take? > > -- Alex > > On Mon, 15 Oct 2007, Gregorio wrote: > > > Hi all. > > > > > > > > I am using openSer 1.3 with mySQL. My problem is that although > registration > > is fast in SIP protocol terms, the contact binding is written in the > > 'location' table long time after this contact has received a 200 OK to his > > registration. So a user could think that he is registered before he is in > > the data base. > > > > > > > > Could it be a configuration issue? > > > > Can I solve this? > > > > > > > > Thanks! > > > > > > > > This is my config: > > > > > > > > > > > > # ----------- global configuration parameters ------------------------ > > > > > > > > debug=2 # debug level (cmd line: -dddddddddd) > > > > log_facility=LOG_LOCAL7 > > > > fork=yes > > > > log_stderror=no children=4 > > > > port=5060 > > > > > > > > listen=udp:192.168.1.206:5060 > > > > listen=tcp:192.168.1.206:5060 > > > > > > > > # ------------------ module loading ---------------------------------- > > > > > > > > #set module path > > > > mpath="/usr/local/lib/openser/modules/" > > > > > > > > loadmodule "mysql.so" > > > > loadmodule "xlog.so" > > > > loadmodule "sl.so" > > > > loadmodule "tm.so" > > > > loadmodule "rr.so" > > > > loadmodule "maxfwd.so" > > > > loadmodule "usrloc.so" > > > > loadmodule "registrar.so" > > > > loadmodule "textops.so" > > > > loadmodule "auth.so" > > > > loadmodule "auth_db.so" > > > > > > > > loadmodule "presence.so" > > > > > > > > loadmodule "presence_xml.so" > > > > > > > > loadmodule "mi_fifo.so" > > > > > > > > loadmodule "mi_xmlrpc.so" > > > > > > > > # ----------------- setting module-specific parameters --------------- > > > > > > > > > > > > modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") > > > > > > > > # -- usrloc params -- > > > > > > > > modparam("usrloc", "db_mode", 2) > > > > > > > > # -- auth params -- > > > > modparam("auth_db", "calculate_ha1", yes) > > > > modparam("auth_db", "password_column", "password") > > > > > > > > # -- rr params -- > > > > # add value to ;lr param to make some broken UAs happy > > > > modparam("rr", "enable_full_lr", 1) > > > > > > > > > > > > #Agnadido para la Presencia > > > > modparam("presence|usrloc|pua", > > "db_url","mysql://openser:openserrw@127.0.0.1/openser") > > > > modparam("presence", "presentity_table", "presentity") > > > > modparam("presence", "active_watchers_table", "active_watchers") > > > > modparam("presence", "watchers_table", "watchers") > > > > modparam("presence", "clean_period", 100) > > > > modparam("presence", "max_expires", 3600) > > > > modparam("presence", "server_address", "sip:192.168.1.206:5060" ) > > > > > > > > #Presence_XML > > > > modparam("presence_xml", > > "db_url","mysql://openser:openserrw@127.0.0.1/openser") > > > > modparam("presence_xml", "xcap_table", "xcap") > > > > modparam("presence_xml", "force_active", 1) > > > > modparam("presence_xml", "integrated_xcap_server", 1) > > > > modparam("presence_xml", "pidf_manipulation", 1) > > > > modparam("presence_xml", "pidf_manipulation", 1) > > > > > > > > #Mi FIFO > > > > modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") > > > > modparam("mi_fifo", "fifo_mode", 0600) > > > > modparam("mi_fifo", "fifo_group", 0) > > > > modparam("mi_fifo", "fifo_group", "root") > > > > modparam("mi_fifo", "fifo_user", 0) > > > > modparam("mi_fifo", "fifo_user", "root") > > > > modparam("mi_fifo", "reply_dir", "/home/openser/tmp/") > > > > > > > > #Mi XMLRPC > > > > modparam("mi_xmlrpc", "port", 8080) > > > > modparam("mi_xmlrpc", "log_file", "/var/log/openser-xmlrpc.log") > > > > modparam("mi_xmlrpc", "reply_option", 0) > > > > modparam("mi_xmlrpc", "buffer_size", 8192) > > > > > > > > > > > > # ------------------------- request routing logic ------------------- > > > > > > > > # main routing logic > > > > > > > > route{ > > > > > > > > # initial sanity checks -- messages with > > > > # max_forwards==0, or excessively long requests > > > > > > > > if (!mf_process_maxfwd_header("10")) { > > > > sl_send_reply("483","Too Many Hops. Route filter"); > > > > exit; > > > > } > > > > > > > > if (msg:len >= max_len ) { > > > > sl_send_reply("513", "Message too big. Route filter"); > > > > exit; > > > > } > > > > > > > > if (!method=="REGISTER") > > > > record_route(); > > > > > > > > # subsequent messages withing a dialog should take the > > > > # path determined by record-routing > > > > if (loose_route()) { > > > > # mark routing logic in request > > > > append_hf("P-hint: rr-enforced\r\n"); > > > > route(1); > > > > }; > > > > if (uri==myself) { > > > > route(1); > > > > }; > > > > > > > > if (uri==myself) { > > > > # presence handling > > > > if( is_method("PUBLISH|SUBSCRIBE")){ > > > > route(2); > > > > }; > > > > > > > > if (is_present_hf("MyHeader")) > > > > { > > > > t_relay("192.168.1.111"); > > > > exit; > > > > }; > > > > > > > > > > > > if (method=="REGISTER") { > > > > # Uncomment this if you want to use digest > > authentication > > > > if (!www_authorize("openserDialcom", "subscriber")) > > { > > > > www_challenge("openserDialcom", "0"); > > > > exit; > > > > }; > > > > > > > > save("location"); > > > > exit; > > > > }; > > > > > > > > lookup("aliases"); > > > > if (!uri==myself) { > > > > append_hf("P-hint: outbound alias\r\n"); > > > > route(1); > > > > }; > > > > > > > > # native SIP destinations are handled using our USRLOC DB > > > > if (!lookup("location")) { > > > > sl_send_reply("404", "Not Found in openSer"); > > > > exit; > > > > }; > > > > append_hf("P-hint: usrloc applied\r\n"); > > > > }; > > > > > > > > route(1); > > > > } > > > > > > > > route[1] { > > > > # send it out now; use stateful forwarding as it works reliably > > > > # even for UDP2TCP > > > > if (!t_relay()) { > > > > sl_reply_error(); > > > > }; > > > > exit; > > > > } > > > > route[2] > > > > { > > > > # absorb retransmissions > > > > if (! t_newtran()) > > > > { > > > > sl_reply_error(); > > > > exit; > > > > }; > > > > > > > > if(is_method("PUBLISH")) > > > > { > > > > handle_publish(); > > > > t_release(); > > > > } else if( is_method("SUBSCRIBE")) { > > > > handle_subscribe(); > > > > t_release(); > > > > }; > > > > > > > > exit; > > > > } > > > > > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : +1-678-954-0670 > Direct : +1-678-954-0671 > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

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Re: [OpenSER-Users] Insert in location table very slow
by Alex Balashov 15 Oct '07

15 Oct '07

Gregorio, How long does it take? -- Alex On Mon, 15 Oct 2007, Gregorio wrote: > Hi all. > > > > I am using openSer 1.3 with mySQL. My problem is that although registration > is fast in SIP protocol terms, the contact binding is written in the > 'location' table long time after this contact has received a 200 OK to his > registration. So a user could think that he is registered before he is in > the data base. > > > > Could it be a configuration issue? > > Can I solve this? > > > > Thanks! > > > > This is my config: > > > > > > # ----------- global configuration parameters ------------------------ > > > > debug=2 # debug level (cmd line: -dddddddddd) > > log_facility=LOG_LOCAL7 > > fork=yes > > log_stderror=no children=4 > > port=5060 > > > > listen=udp:192.168.1.206:5060 > > listen=tcp:192.168.1.206:5060 > > > > # ------------------ module loading ---------------------------------- > > > > #set module path > > mpath="/usr/local/lib/openser/modules/" > > > > loadmodule "mysql.so" > > loadmodule "xlog.so" > > loadmodule "sl.so" > > loadmodule "tm.so" > > loadmodule "rr.so" > > loadmodule "maxfwd.so" > > loadmodule "usrloc.so" > > loadmodule "registrar.so" > > loadmodule "textops.so" > > loadmodule "auth.so" > > loadmodule "auth_db.so" > > > > loadmodule "presence.so" > > > > loadmodule "presence_xml.so" > > > > loadmodule "mi_fifo.so" > > > > loadmodule "mi_xmlrpc.so" > > > > # ----------------- setting module-specific parameters --------------- > > > > > > modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") > > > > # -- usrloc params -- > > > > modparam("usrloc", "db_mode", 2) > > > > # -- auth params -- > > modparam("auth_db", "calculate_ha1", yes) > > modparam("auth_db", "password_column", "password") > > > > # -- rr params -- > > # add value to ;lr param to make some broken UAs happy > > modparam("rr", "enable_full_lr", 1) > > > > > > #Agnadido para la Presencia > > modparam("presence|usrloc|pua", > "db_url","mysql://openser:openserrw@127.0.0.1/openser") > > modparam("presence", "presentity_table", "presentity") > > modparam("presence", "active_watchers_table", "active_watchers") > > modparam("presence", "watchers_table", "watchers") > > modparam("presence", "clean_period", 100) > > modparam("presence", "max_expires", 3600) > > modparam("presence", "server_address", "sip:192.168.1.206:5060" ) > > > > #Presence_XML > > modparam("presence_xml", > "db_url","mysql://openser:openserrw@127.0.0.1/openser") > > modparam("presence_xml", "xcap_table", "xcap") > > modparam("presence_xml", "force_active", 1) > > modparam("presence_xml", "integrated_xcap_server", 1) > > modparam("presence_xml", "pidf_manipulation", 1) > > modparam("presence_xml", "pidf_manipulation", 1) > > > > #Mi FIFO > > modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") > > modparam("mi_fifo", "fifo_mode", 0600) > > modparam("mi_fifo", "fifo_group", 0) > > modparam("mi_fifo", "fifo_group", "root") > > modparam("mi_fifo", "fifo_user", 0) > > modparam("mi_fifo", "fifo_user", "root") > > modparam("mi_fifo", "reply_dir", "/home/openser/tmp/") > > > > #Mi XMLRPC > > modparam("mi_xmlrpc", "port", 8080) > > modparam("mi_xmlrpc", "log_file", "/var/log/openser-xmlrpc.log") > > modparam("mi_xmlrpc", "reply_option", 0) > > modparam("mi_xmlrpc", "buffer_size", 8192) > > > > > > # ------------------------- request routing logic ------------------- > > > > # main routing logic > > > > route{ > > > > # initial sanity checks -- messages with > > # max_forwards==0, or excessively long requests > > > > if (!mf_process_maxfwd_header("10")) { > > sl_send_reply("483","Too Many Hops. Route filter"); > > exit; > > } > > > > if (msg:len >= max_len ) { > > sl_send_reply("513", "Message too big. Route filter"); > > exit; > > } > > > > if (!method=="REGISTER") > > record_route(); > > > > # subsequent messages withing a dialog should take the > > # path determined by record-routing > > if (loose_route()) { > > # mark routing logic in request > > append_hf("P-hint: rr-enforced\r\n"); > > route(1); > > }; > > if (uri==myself) { > > route(1); > > }; > > > > if (uri==myself) { > > # presence handling > > if( is_method("PUBLISH|SUBSCRIBE")){ > > route(2); > > }; > > > > if (is_present_hf("MyHeader")) > > { > > t_relay("192.168.1.111"); > > exit; > > }; > > > > > > if (method=="REGISTER") { > > # Uncomment this if you want to use digest > authentication > > if (!www_authorize("openserDialcom", "subscriber")) > { > > www_challenge("openserDialcom", "0"); > > exit; > > }; > > > > save("location"); > > exit; > > }; > > > > lookup("aliases"); > > if (!uri==myself) { > > append_hf("P-hint: outbound alias\r\n"); > > route(1); > > }; > > > > # native SIP destinations are handled using our USRLOC DB > > if (!lookup("location")) { > > sl_send_reply("404", "Not Found in openSer"); > > exit; > > }; > > append_hf("P-hint: usrloc applied\r\n"); > > }; > > > > route(1); > > } > > > > route[1] { > > # send it out now; use stateful forwarding as it works reliably > > # even for UDP2TCP > > if (!t_relay()) { > > sl_reply_error(); > > }; > > exit; > > } > > route[2] > > { > > # absorb retransmissions > > if (! t_newtran()) > > { > > sl_reply_error(); > > exit; > > }; > > > > if(is_method("PUBLISH")) > > { > > handle_publish(); > > t_release(); > > } else if( is_method("SUBSCRIBE")) { > > handle_subscribe(); > > t_release(); > > }; > > > > exit; > > } > > -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671

2 1

[OpenSER-Users] Insert in location table very slow
by Gregorio 15 Oct '07

15 Oct '07

Hi all. I am using openSer 1.3 with mySQL. My problem is that although registration is fast in SIP protocol terms, the contact binding is written in the 'location' table long time after this contact has received a 200 OK to his registration. So a user could think that he is registered before he is in the data base. Could it be a configuration issue? Can I solve this? Thanks! This is my config: # ----------- global configuration parameters ------------------------ debug=2 # debug level (cmd line: -dddddddddd) log_facility=LOG_LOCAL7 fork=yes log_stderror=no children=4 port=5060 listen=udp:192.168.1.206:5060 listen=tcp:192.168.1.206:5060 # ------------------ module loading ---------------------------------- #set module path mpath="/usr/local/lib/openser/modules/" loadmodule "mysql.so" loadmodule "xlog.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "mi_fifo.so" loadmodule "mi_xmlrpc.so" # ----------------- setting module-specific parameters --------------- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- modparam("usrloc", "db_mode", 2) # -- auth params -- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) #Agnadido para la Presencia modparam("presence|usrloc|pua", "db_url","mysql://openser:openserrw@127.0.0.1/openser") modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "clean_period", 100) modparam("presence", "max_expires", 3600) modparam("presence", "server_address", "sip:192.168.1.206:5060" ) #Presence_XML modparam("presence_xml", "db_url","mysql://openser:openserrw@127.0.0.1/openser") modparam("presence_xml", "xcap_table", "xcap") modparam("presence_xml", "force_active", 1) modparam("presence_xml", "integrated_xcap_server", 1) modparam("presence_xml", "pidf_manipulation", 1) modparam("presence_xml", "pidf_manipulation", 1) #Mi FIFO modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("mi_fifo", "fifo_mode", 0600) modparam("mi_fifo", "fifo_group", 0) modparam("mi_fifo", "fifo_group", "root") modparam("mi_fifo", "fifo_user", 0) modparam("mi_fifo", "fifo_user", "root") modparam("mi_fifo", "reply_dir", "/home/openser/tmp/") #Mi XMLRPC modparam("mi_xmlrpc", "port", 8080) modparam("mi_xmlrpc", "log_file", "/var/log/openser-xmlrpc.log") modparam("mi_xmlrpc", "reply_option", 0) modparam("mi_xmlrpc", "buffer_size", 8192) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops. Route filter"); exit; } if (msg:len >= max_len ) { sl_send_reply("513", "Message too big. Route filter"); exit; } if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (uri==myself) { route(1); }; if (uri==myself) { # presence handling if( is_method("PUBLISH|SUBSCRIBE")){ route(2); }; if (is_present_hf("MyHeader")) { t_relay("192.168.1.111"); exit; }; if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("openserDialcom", "subscriber")) { www_challenge("openserDialcom", "0"); exit; }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found in openSer"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } route[2] { # absorb retransmissions if (! t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); }; exit; }

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[OpenSER-Users] PUA_MI: Error in state assignment
by Schumann Sebastian 15 Oct '07

15 Oct '07

Hello all. Upgrading my OpenSER trunk version from Rel. 2404 to 2920 I encounter problems using the PUA and PUA_MI module. I control the PUA via management interface. In the old version, I could update my presence information and also the non-expiration worked finally. Furthermore, expiring states with Expires: 0 worked as well. After upgrading to the new SVN version, the update does not work anymore. When an user, who changed his state to online and does not want any expiration, wants his state to expire (sends msg. with Expire: 0) at some point, the following error occurs: Oct 15 15:26:39 [15815] DBG:pua_mi:mi_pua_publish: send publish Oct 15 15:26:39 [15815] DBG:pua:send_publish: pres_uri=sip:aastra@10.96.115.125 Oct 15 15:26:39 [15815] DBG:pua:search_htable: core_hash= 256 Oct 15 15:26:39 [15815] DBG:pua:search_htable: record not found Oct 15 15:26:39 [15815] DBG:pua:send_publish: insert type Oct 15 15:26:39 [15815] DBG:pua:send_publish: request for a publish with expires 0 and no record found OpenSER cannot assign the proper state in the database and remove it. Instead, the user stays online "forever". A manual removal from the database is necessary. Where there major changes in the modules? I checked my config against the documentation and cannot find any major modifications I didn't already apply. Any hints for solving this issue are appreciated! Thanks for your help. Sebastian

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[OpenSER-Users] Re: Openser-Perl module in onreply_route
by Murilo Lacerda Yoshida 15 Oct '07

15 Oct '07

Hi Mik, When I faced these problems with openser and DBI, I did a research, and if I remember it correctly, DBD::Sybase does not work correctly with output parameters. So one solution I guess would be you rewrite your procedure so that the parameter is only input and return the same parameter in a result set inside the procedure. That was one thing I had to do when I switched from DBD::ODBC to DBD::Sybase, as I had also some procedures with output parameters. After I rewrote the procedures and returned the output parameters as result sets I had no problems with this. HTH, Murilo -----Original Message----- From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of users-request(a)openser.org Sent: segunda-feira, 15 de outubro de 2007 08:04 To: users(a)openser.org Subject: Users Digest, Vol 29, Issue 47 Send Users mailing list submissions to users(a)openser.org To subscribe or unsubscribe via the World Wide Web, visit http://openser.org/cgi-bin/mailman/listinfo/users or, via email, send a message with subject or body 'help' to users-request(a)openser.org You can reach the person managing the list at users-owner(a)openser.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Users digest..." Today's Topics: 1. Re: Multidomain and in-dialog REFER auth issue (I?aki Baz Castillo) 2. Changing db table names (Dan-Cristian Bogos) 3. Re: Openser-Perl module in onreply_route (Bastian Friedrich) 4. Re: t_relay(), forward() and send(): retransmissions? (samuel) 5. Re: t_relay(), forward() and send(): retransmissions? (I?aki Baz Castillo) 6. Re: t_relay(), forward() and send(): retransmissions? (Klaus Darilion) 7. Re: Multidomain and in-dialog REFER auth issue (Klaus Darilion) 8. Re: Multidomain and in-dialog REFER auth issue (Klaus Darilion) 9. Re: Multidomain and in-dialog REFER auth issue (I?aki Baz Castillo) 10. Re: Multidomain and in-dialog REFER auth issue (I?aki Baz Castillo) 11. Re: Multidomain and in-dialog REFER auth issue (I?aki Baz Castillo) 12. change status code in reply (Cseke Tamas) ---------------------------------------------------------------------- Message: 1 Date: Mon, 15 Oct 2007 10:13:22 +0200 From: I?aki Baz Castillo <ibc(a)in.ilimit.es> Subject: Re: [OpenSER-Users] Multidomain and in-dialog REFER auth issue To: users(a)openser.org Message-ID: <200710151013.22807.ibc(a)in.ilimit.es> Content-Type: text/plain; charset="iso-8859-1" El Monday 15 October 2007 09:58:36 Iñaki Baz Castillo escribió: > How can my OpenSer know which domain this REFER goes? of course the "To" > header is not valid at all. > > Could be a solution a SQL query to "location" table looking for the URI > (sip:userB@80.98.123.23:5060) and getting the username and domain of this? Anyway this solution wouldn't be secure since userA(a)domainA.com could hack its "From" header in the REFER and appears as "@domainB.com". So I think I need to store dialog info in a table (in the 200-OK), with those info: - "From" header domain - RURI domain - fromtag - totag - "Call-ID" header and query this table in REFER amtching fromtag, totag and Call-ID, and just allow the REFER is "From" header and RURI domain are the same. "Dialog" module is not valid for me since it doesn't store RURI. Any suggestion about it? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es ------------------------------ Message: 2 Date: Mon, 15 Oct 2007 10:15:31 +0200 From: "Dan-Cristian Bogos" <dan.bogos(a)gmail.com> Subject: [OpenSER-Users] Changing db table names To: "users openser.org" <users(a)openser.org> Message-ID: <40e285320710150115k78aa907co56a4a7a1748bbfc2(a)mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 Hi All, I came to a situation where I need to use openser together with other apps on the same database, therefore I will need to change the table names. I know that most of the modules permit that on through config, but my concern is the table version, and the data which should be inside. Can anybody give me any hints on this issue? Ta, DanB ------------------------------ Message: 3 Date: Mon, 15 Oct 2007 10:19:20 +0200 From: Bastian Friedrich <bastian.friedrich(a)collax.com> Subject: Re: [OpenSER-Users] Openser-Perl module in onreply_route To: users(a)openser.org Message-ID: <200710151019.21008.bastian.friedrich(a)collax.com> Content-Type: text/plain; charset="iso-8859-1" Hi Mik, On Friday 12 October 2007, Mik Cheez wrote: > I had been using 'perl_exec' in my scripts with great success until I > decided to try using it in the onreply_route. The strangest thing now > happens; I use my perl script to write information to a database, and it > seems like the failure would have to be in DBI, but it only occurs in > Openser. a few weeks ago, someone else experienced problems when using DBI in OpenSER Perl scripts: http://openser.org/pipermail/users/2007-September/013388.html We currently do not have a proper solution for the problem, as the problems are created by DBI rather than by OpenSER. You might consider trying to establish some kind of OpenSER-internal locking mechanism for the DBI functions. Have you tried running your Perl script with only a single OpenSER instance (i.e. turning of forking)? Bastian -- Collax GmbH . Burkheimer Straße 3 . 79111 Freiburg . Germany p: +49 (0) 761-45684-24 f: +49 (0) 761-45684-10 www.collax.com Geschäftsführer: William K. Hite / Boris Nalbach AG München HRB 158898 . Ust.-IdNr: DE 814464942 \ Smash forehead on keyboard to continue. ------------------------------ Message: 4 Date: Mon, 15 Oct 2007 11:11:03 +0200 From: samuel <samu60(a)gmail.com> Subject: Re: [OpenSER-Users] t_relay(), forward() and send(): retransmissions? To: " I?aki Baz Castillo " <ibc(a)in.ilimit.es> Cc: users(a)openser.org Message-ID: <d18bd3a10710150211n5b0dd1e9oc75297296525c352(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" forward() and send() are the stateless functions, which basically means they just send the message and forget the transaction. t_relay() is the statefull routing function, so whenever you use this function openSER will take care of the transaction FSM. hope it helps, sam 2007/10/15, Iñaki Baz Castillo <ibc(a)in.ilimit.es>: > > Hi, how do forward() and send() route the messages? do they > retransmissions as > t_relay()? > > Thanks. > > > -- > Iñaki Baz Castillo > ibc(a)in.ilimit.es > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

1 0

[OpenSER-Users] Changing db table names
by Dan-Cristian Bogos 15 Oct '07

15 Oct '07

Hi All, I came to a situation where I need to use openser together with other apps on the same database, therefore I will need to change the table names. I know that most of the modules permit that on through config, but my concern is the table version, and the data which should be inside. Can anybody give me any hints on this issue? Ta, DanB

2 4

[OpenSER-Users] change status code in reply
by Cseke Tamas 15 Oct '07

15 Oct '07

Hello, I have the same problem what discussed in this thread: http://www.nabble.com/SF.net-SVN:-openser:--2835--trunk-modules-sl-t4533872… I'd like to change the status code, is it still inpossible with 1.2? I tried t_reply, sl_send_reply, replace functions but they don't work? Could you tell me any solution for this, if it is possible? Thanks any help, Tamas

1 0

[OpenSER-Users] t_relay(), forward() and send(): retransmissions?
by Iñaki Baz Castillo 15 Oct '07

15 Oct '07

Hi, how do forward() and send() route the messages? do they retransmissions as t_relay()? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

3 3

[OpenSER-Users] Openser-Perl module in onreply_route
by Mik Cheez 15 Oct '07

15 Oct '07

I had been using 'perl_exec' in my scripts with great success until I decided to try using it in the onreply_route. The strangest thing now happens; I use my perl script to write information to a database, and it seems like the failure would have to be in DBI, but it only occurs in Openser. I execute a stored procedure, and instead of doing this: exec procedurename @arg1 = '{sip status}', @arg2 = '{callid}' The procedure is called like this: declare @arg2 numeric({string length}) exec procedurename @arg1 = '{sip status}', @arg2 = '{callid}' output I have no idea where the thing is declaring @arg2 as an output variable. I can run my script manually forever and never get this message, but it happens about 50% of the time in Openser (i.e. half the time it works perfectly, without creating the output variable). I would try to run it somewhere else in my openser script, but the sip status isn't available in the request context (where I run similar scripts without any problems). The procedure is not supposed to return anything. Here's the function I created: ***** ***** use strict; use DBI; use DBI qw(:sql_types); use OpenSER qw ( log ); use OpenSER::Constants; use OpenSER::Message; sub proxy_onreply { my $m = shift; my $status; my $callid; my $sth; my $dbh; my $query; $status = $m->getStatus(); $callid = $m->pseudoVar("\$ci"); $dbh = DBI->connect("dbi:Sybase") || die "Database connection not made: $DBI::errstr"; $query = "exec procedurename \@arg1 = ?, \@arg1 = ?"; $sth = $dbh->prepare($query); $sth->bind_param(1, $status, SQL_VARCHAR); $sth->bind_param(2, $callid, SQL_VARCHAR); $sth->execute; $sth->finish; $dbh->disconnect(); } ***** ***** Any suggestions would be lovely. Best regards, Mik

2 1

[OpenSER-Users] Inbound SIP call to openser
by David Adade 15 Oct '07

15 Oct '07

Can somemone point me in the right direction. On a steep learning curve from asterisk to openser. I am tring to setup inbound PSTN to openser 1.2 I have registered nat clients. behind openser. How can i map DDI to nat clients I have already configure the nat clients in the subscriber table. Should i be looking at the Aliases table? Thanks in advance *********************************************************************** IMPORTANT NOTICE: The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. If you receive this in error, please contact the sender and delete the material from any computer. This e-mail is intended for the above named addressee only. Should you receive this e-mail in error please contact Customer Services on Tel: +44 (0)870 241 2439 ************************************************************************

2 1

[OpenSER-Users] "max_contacts", 1 and EI_R_TOO_MANY
by Jarkko Nevala 15 Oct '07

15 Oct '07

In OpenSER 1.2.0 EI_R_TOO_MANY has been defined by default as "Too many registered contacts" in modules/registrar/reply.c. This message is quite enough for me, but in which occations does UA get this message? I have modparam("registrar", "max_contacts", 1) in my configuration and every time I try to register another client with a username that allready has an AOR in locations table, I get response "503 Service Unavailable", not "503 Too many registered contacts". How should I modify reply.c to get "Too many registered contacts" when I have, well, too many registered contacts? I found this thread from the mailinglist http://www.mail-archive.com/users@openser.org/msg10796.html, but I do not think it really helps me, because the value of EI_R_TOO_MANY does not seem to effect the the response the second UA gets from the OpenSER. thank you JN

1 0

[OpenSER-Users] OpenSER Admin Training Course - Materials for donwload
by Bogdan-Andrei Iancu 15 Oct '07

15 Oct '07

Hi, I just uploaded the materials used during the OpenSER Admin Course in Rome (held on 29 September 2007). Check the even web page for the updates: http://www.openser.org/mos/view/OpenSER-Admin-Course---Rome-2007 and follow the "Materials" shortcut to get to the download web page. Regards, Bogdan

6 9

[OpenSER-Users] Usrloc lookup ad parallel forking
by Edoardo Serra 15 Oct '07

15 Oct '07

Hi all, I have OpenSER (1.2.2) acting as registrar server. If the same user hase more than one AOR I'd like to do parallel forking and send the INVITE to every contact I have for that user. At the moment my configuration file looks like: if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; exit; but it rely only to the AOR with the highest q value It should be a common situation but I cannot find any doc about it. Tnx in advance Regards Edoardo Serra

3 6

[OpenSER-Users] append_rpid_hf without authentication?
by Alex Massover 14 Oct '07

14 Oct '07

Hi! I want to add rpid before sending the INVITE to the gateway, because the gateway requires it, but the problem is that clients don't send REGISTER and I don't use any authentication (like www_authorize). Probably because of that my 'subscriber' table is empty all the time. Can I still use somehow append_rpid_hf() or I need to try to build it with append_hf() manually? BR, Alex.

2 1

[OpenSER-Users] UAC module parameters
by CB 14 Oct '07

14 Oct '07

I am trying to understand the relationship between the various module parameters for the uac module. Imagine I want to authenticate on a proxy user different credentials in different situations. The credential module parameter allows specification of the username, realm and password. I think (correct me if I'm wrong) that I can have multiple of these specified for various realms: modparam("uac","credential","user1:realm1:pw1") modparam("uac","credential","user2:realm2:pw2") According to the documentation, "This parameter is required if UAC authentication is used." What then is the purpose of the auth_realm_avp, auth_username_avp and auth_password_avp module parameters? Do they replace the credential parameter or are they for something different? Any help appreciated. Regards Cameron

2 1

[Serusers] uri contains no digits, only an IP
by Stephen Helfen 12 Oct '07

12 Oct '07

How am I supposed to route these? Essentially, I get an INVITE from an end-user. The uri is simply sip:XXX.XXX.XXX.XXX What is currently happening is that the IP address represented by Xs above is the IP of my proxy. The proxy sees this URI and forwards the call back to itself. This is obviously wrong, but I am not sure how to correct it. The routing data I need is in the to_uri, but I am not sure how to place this into the uri field. rewriteuri does not work on variables, only strings. How can I accomplish this? Currently on version 0.9.6. Thanks in advance, Steve _________________________________________________________________ Climb to the top of the charts! Play Star Shuffle: the word scramble challenge with star power. http://club.live.com/star_shuffle.aspx?icid=starshuffle_wlmailtextlink_oct

1 0

[OpenSER-Users] AVPs and PostgreSQL data types.
by Alex Balashov 12 Oct '07

12 Oct '07

Greetings, I have a problem passing AVPs to stored procedures (or in other SQL contexts) that require an integer argument. It is not a show-stopper, it just throws a warning: 0(6535) PG[get_columns]: Unhandled data type column (my_stored_proc) OID (869), defaulting to STRING This results from something like: avp_db_query("SELECT my_stored_proc($avp(S:my_avp))"); Now, $avp(S:my_avp) is a numerical value, so it does end up working. However, because AVPs are not natively "typed" as numerical or string values in a way that is represented at the Postgres layer of the avpops implementation, evidently, this situation arises. Is there any way to fix this? I have tried using CAST() in the SQL statement, but that did not appear to help. Thanks, -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671

2 2

[Serusers] MySql module load error
by Angela Gao 12 Oct '07

12 Oct '07

Hello, all, I have one question, if in ser.cfg, loadmodule "/home/linus/gao/usr/local/lib/ser/modules/mysql.so", does it means we must go to mysql website and download and install it? and then create db with name config in ser.cfg ? I got some error when start ser. 0(13676) ERROR: load_module: could not open module </home/linus/gao/usr/local/lib/ser/modules/mysql.so>: libmysqlclient.so.14: cannot open shared object file: No such file or directory 0(13676) parse error (81,13-14): failed to load module Thanks for any suggestion. -- Best regard, Angela

2 1

Re: Re: [OpenSER-Users] CPL extension
by Daniel 12 Oct '07

12 Oct '07

Hi Bogdan, I don't want to change CPL's standard element specified by RFC, but only add a new type of switch. But Internal parser and Interpreter are included within CPL-C Module of OpenSER, right? So, I have to say to OpenSER how it parses and manages this new element. Regards, Daniel Bogdan-Andrei Iancu wrote: Hi Daniel, CPL is specified by an IETF RFC, so changing it will break compatibility. anyhow, it is not sufficient to change the DTD - you need to change the internal parser (that converts the XML to a binary, easy to run format) and also the interpretor itself (using the binary format). regards, bogdan Daniel wrote: Hi All, I'd like to extend CPL functionality adding a new switch element. How can manage it ? I think I'd change the "cpl.dtd" file and "cpl-c" module, it's right? Thanks, Daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com <mailto:d.grotti@gmail.com> <d.grotti(a)gmail.com> ------------------------------------------------------------------------ _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

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Re: [OpenSER-Users] CPL extension
by Bogdan-Andrei Iancu 12 Oct '07

12 Oct '07

Hi Daniel, Daniel Grotti wrote: > Hi Bogdan, > I don't want to change CPL's standard element specified by RFC, but > only add a new type of switch. OK > But Internal parser and Interpreter are included within CPL-C Module > of OpenSER, right? yes, it is part of the cpl-c module. > So, I have to say to OpenSER how it parses and manages this new element. yes, you need to extend the parser and interpreter to recognize the new elements. regards, bogdan > > Regards, > Daniel > > > > Bogdan-Andrei Iancu wrote: >> Hi Daniel, >> >> CPL is specified by an IETF RFC, so changing it will break >> compatibility. >> >> anyhow, it is not sufficient to change the DTD - you need to change >> the internal parser (that converts the XML to a binary, easy to run >> format) and also the interpretor itself (using the binary format). >> >> regards, >> bogdan >> >> Daniel wrote: >>> Hi All, >>> I'd like to extend CPL functionality adding a new switch element. >>> How can manage it ? >>> I think I'd change the "cpl.dtd" file and "cpl-c" module, it's right? >>> >>> Thanks, >>> Daniel >>> >>> -- >>> Daniel Grotti >>> ________________________ >>> e-mail : d.grotti(a)gmail.com <mailto:d.grotti@gmail.com> >>> ------------------------------------------------------------------------ >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/users >> >> >

1 0

[Serusers] Fwd: need help implementing sip callback
by Rogelio Serrano 12 Oct '07

12 Oct '07

---------- Forwarded message ---------- From: Rogelio Serrano <rogelio.serrano(a)gmail.com> Date: Oct 11, 2007 5:18 PM Subject: Re: [Serusers] need help implementing sip callback To: Atle Samuelsen <clona(a)cyberhouse.no> On 10/11/07, Atle Samuelsen <clona(a)cyberhouse.no> wrote: > * Rogelio Serrano <rogelio.serrano(a)gmail.com> [071011 07:51]: > > any pointers? > > You could proberbly get some pointers if you would describe what you > wanted properly. If you want a function like *393939# to make it call > back the last caller tell us... I do not know.. > > - ATle > i want to ring phone a. then when phone a picks up i would play recording that phone a has n minutes for this call. then i ring phone b and when he picks up connect to phone a. so whats the best way to do this? when the two parties are pstn phones? the most obvious method is to use an rtp proxy. how is nat hairpin possible when i dont know which udp port both parties are going to use? > > > > -- > > Lay low and nourish in obscurity > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > -- Lay low and nourish in obscurity -- Lay low and nourish in obscurity

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[Serusers] I already fixed it//Fwd: Error to start ser
by Angela Gao 12 Oct '07

12 Oct '07

---------- Forwarded message ---------- From: Angela Gao <angela.gwh(a)gmail.com> Date: 2007-10-12 下午2:36 Subject: Error to start ser To: serusers(a)lists.iptel.org Hello, all, I download ser-2.0.0-rc1_src.tar.gz source file and compile it in my own os, and it is: Linux version 2.6.9-34.ELsmp ( bhcompile(a)hs20-bc1-7.build.redhat.com) (gcc version 3.4.5 20051201 (Red Hat 3.4.5-2)). The compile cmd is : make group_include="standard mysql" install I also get ser.buildsystem.latest.tar.gz and create my ser.cfg, no error during the compile. but error is thrown when run ser with cmd: ser -f /home/linus/gao/usr/local/etc/ser.cfg, error will be throued as below: [linus@hp380mf2 sbin]$ ser -f /home/linus/gao/usr/local/etc/ser.cfg ERROR: bad config file (1 errors) ser.cfg is attached in this mail why? Is there anyone can help me fix it? It would be very nice of you, thank you. -- Best regard, Angela -- Best regard, Angela

1 0

[OpenSER-Users] CPL extension
by Daniel 12 Oct '07

12 Oct '07

Hi All, I'd like to extend CPL functionality adding a new switch element. How can manage it ? I think I'd change the "cpl.dtd" file and "cpl-c" module, it's right? Thanks, Daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

2 1

Re: [Serusers] SER + Linksys SPA942 + Call transfer
by roayu 12 Oct '07

12 Oct '07

Hi Klaus! The UA does not received the REFER sent by SPA942. What should I need to configure to ensure that the UA receive the REFER from the transfer agent(SPA942)? Thanks. Cheers, Roa Yu -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of kgfleischmann(a)t-online.de Sent: Friday, October 12, 2007 2:22 PM To: serusers(a)lists.iptel.org Subject: Re: [Serusers] SER + Linksys SPA942 + Call transfer The NOTIFY is always sent from the UA, which received the REFER. SER only relays it. Did you check this? Cheers Klaus roayu wrote: > Hi there! > > After SPA942(transfer agent) sent REFER to SER, SER does not send a NOTIFY > to SPA942(transfer agent). What's causes this? Can anyone tell me. > > Thanks. > > Cheers, > Roa Yu :) > > -----Original Message----- > From: serusers-bounces(a)lists.iptel.org > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of roayu > Sent: Thursday, October 11, 2007 4:34 PM > To: Atle Samuelsen > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] SER + Linksys SPA942 + Call transfer > > > Hi Atle, > > Let me make my scenario clear on you. > > I have 2 softphone and 1 IP phone(SPA942). I would like to call from > Softphone A to SPA942, then use SPA942 transfer the call to Softphone B. > But, when I do so, the connection still remained on between SPA942 and > Softphone B. > > I managed to get Softphone A onHold and the SPA942 managed to call to > Softphone B. Once I press on the 'xfer' button after calling Softphone B, A > still onHold and no SIP signal to unHold A and to terminate B. > > If I would like to configure the ser.cfg file, which part that I need to > configure? And how do I do that? > > Thanks. > > Cheers, > Roa Yu > > > -----Original Message----- > From: Atle Samuelsen [mailto:clona@cyberhouse.no] > Sent: Thursday, October 11, 2007 3:19 PM > To: roayu > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] SER + Linksys SPA942 + Call transfer > > > Hi Roa,i > * roayu <roayu(a)ctisys.net> [071011 04:01]: >> Hi, there! >> >> I have some questions and help. >> 1) I would like to know can call transfer being done when the >> connection is P2P? When I tried to use relay (mediaproxy) to do the call >> transfer, it's able to transfer the call to the other party whereas when > the >> condition is P2P, it just can't pass the correct signal. > > I'm proberbly only tierd (9:15 am here) but P2P? Can you provide the > signalling and ser.cfg so we can understand what you are trying to > establish? > > I personally think you are trying to do a call transfer from one ua to a > other, where the original call went true mediaproxy, but you do not want > the "new" transferd call to go true it. > >> 2) Can SER support SPA942 ? Or is there some other settings that I need >> to configure on SER ? > > SPA942, SPA962, SPA2102, yea.. all Linksys SPA products work like a > charm with SER (Who has a 942 as his primary phone these days) > > > Best Regards > > ATle >> Thanks. >> >> Cheers, >> Roa Yu >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] Error to start ser
by Angela Gao 12 Oct '07

12 Oct '07

Hello, all, I download ser-2.0.0-rc1_src.tar.gz source file and compile it in my own os, and it is: Linux version 2.6.9-34.ELsmp ( bhcompile(a)hs20-bc1-7.build.redhat.com) (gcc version 3.4.5 20051201 (Red Hat 3.4.5-2)). The compile cmd is : make group_include="standard mysql" install I also get ser.buildsystem.latest.tar.gz and create my ser.cfg, no error during the compile. but error is thrown when run ser with cmd: ser -f /home/linus/gao/usr/local/etc/ser.cfg, error will be throued as below: [linus@hp380mf2 sbin]$ ser -f /home/linus/gao/usr/local/etc/ser.cfg ERROR: bad config file (1 errors) ser.cfg is attached in this mail why? Is there anyone can help me fix it? It would be very nice of you, thank you. -- Best regard, Angela

1 0

Re: [Serusers] SER + Linksys SPA942 + Call transfer
by roayu 12 Oct '07

12 Oct '07

Hi there! After SPA942(transfer agent) sent REFER to SER, SER does not send a NOTIFY to SPA942(transfer agent). What's causes this? Can anyone tell me. Thanks. Cheers, Roa Yu :) -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of roayu Sent: Thursday, October 11, 2007 4:34 PM To: Atle Samuelsen Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] SER + Linksys SPA942 + Call transfer Hi Atle, Let me make my scenario clear on you. I have 2 softphone and 1 IP phone(SPA942). I would like to call from Softphone A to SPA942, then use SPA942 transfer the call to Softphone B. But, when I do so, the connection still remained on between SPA942 and Softphone B. I managed to get Softphone A onHold and the SPA942 managed to call to Softphone B. Once I press on the 'xfer' button after calling Softphone B, A still onHold and no SIP signal to unHold A and to terminate B. If I would like to configure the ser.cfg file, which part that I need to configure? And how do I do that? Thanks. Cheers, Roa Yu -----Original Message----- From: Atle Samuelsen [mailto:clona@cyberhouse.no] Sent: Thursday, October 11, 2007 3:19 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] SER + Linksys SPA942 + Call transfer Hi Roa,i * roayu <roayu(a)ctisys.net> [071011 04:01]: > Hi, there! > > I have some questions and help. > 1) I would like to know can call transfer being done when the > connection is P2P? When I tried to use relay (mediaproxy) to do the call > transfer, it's able to transfer the call to the other party whereas when the > condition is P2P, it just can't pass the correct signal. I'm proberbly only tierd (9:15 am here) but P2P? Can you provide the signalling and ser.cfg so we can understand what you are trying to establish? I personally think you are trying to do a call transfer from one ua to a other, where the original call went true mediaproxy, but you do not want the "new" transferd call to go true it. > > 2) Can SER support SPA942 ? Or is there some other settings that I need > to configure on SER ? SPA942, SPA962, SPA2102, yea.. all Linksys SPA products work like a charm with SER (Who has a 942 as his primary phone these days) Best Regards ATle > > Thanks. > > Cheers, > Roa Yu > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

2 1

[OpenSER-Users] RE: openxcap Problem
by Schumann Sebastian 11 Oct '07

11 Oct '07

Hi Kevin. Unfortunately not. I read the tickets at openxcap.org but that didn't help me yet. I tried plenty of things, downgrade the libxml2 as recommended there, install the required stuff manually but neither worked out. You think the problem comes from xml schema checking from Proxy? I had actually similar problem on other compilation, but in this case I doubt as all xsd files for presence related AUIDs are brought by the openxcap package itself. I would also appreciate if sb. tried compilation and use on a "more current unstable" Debian system or similar (acc. library version differences) and can give additional hints. How did u proceed and so on - there must be some differences to the installation recommendation on openxcap.org. Best regards Sebastian -----Original Message----- From: kinnan2224(a)hotmail.com [mailto:kinnan2224@hotmail.com] Sent: Thu 10/11/2007 16:24 To: Schumann Sebastian Subject: openxcap Problem Hi, Sebastian. I met the same problem as yours. I am using Ubuntu 7.04 at the moment. Have you fixed that problem? I dont if the problem comes from xml schema checking via http with proxy. Can you give me some help? THANK U Regards, Kevin Schumann Sebastian wrote: > > Dear all > > I have a big problem with openxcap. I don't get it work :( As I saw on > their webpage, this mailing list can be posted for this concern as well. > > I am using version 0.9.3 and acc. the installation file on > http://www.openxcap.org all current dependencies from the Debian > unstable distribution. > > When I start /usr/bin/openxcap, the following error occurs in the > syslog: > > Sep 25 17:01:21 openxcap openxcap[11876]: [-] Log opened. > > Sep 25 17:01:21 openxcap openxcap[11876]: [-] Starting Open XCAP 0.9.3 > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] Traceback (most recent > call last): > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] File > "/usr/bin/openxcap", line 56, in ? > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap.server > import XCAPServer > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] File > "/usr/lib/python2.4/site-packages/xcap/server.py", line 21, in ? > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap import > authentication > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] File > "/usr/lib/python2.4/site-packages/xcap/authentication.py", line 21, in ? > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap.appusage > import getApplicationForURI > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] File > "/usr/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 466, > in ? > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] applications = > {'xcap-caps': XCAPCapabilitiesApplication(), > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] File > "/usr/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 64, > in __init__ > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] self.xml_schema = > etree.XMLSchema(xml_schema_doc) > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "xmlschema.pxi", > line 67, in etree.XMLSchema.__init__ > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] etree.XMLSchemaParseError: > Document is not valid XML Schema > > > > As I wrote, I followed all instructions from the openxcap webpage but > didn't find any hints how to make it work. > > > > Any help on that would be appreciated. > > > > Best regards > > Sebastian > > > -- > Sebastian Schumann > Diploma Student Architecture and Design Department > > Slovak Telekom, a. s. > T-Com, Operations Unit > Production and Service Division > Innovations and Enterprise Solution Subunit > > Address: Namestie Slobody 6, 817 62 Bratislava, Slovakia > Office: Room 449, Jarabinkova 1, 821 09 Bratislava, Slovakia > > +421 2 588 13332 (tel) > +421 910 643010 (mobile) > +49 175 1925928 (mobile) > > sebastian.schumann(a)t-com.sk > http://www.t-com.sk <http://www.t-com.sk/> > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > Quoted from: http://www.nabble.com/openxcap-Problem-tf4515559.html#a12879532

1 0

[OpenSER-Users] Dumb question.
by Alex Balashov 11 Oct '07

11 Oct '07

Is there a 'line extend' / 'line wrap' character that can be used inside the OpenSER config, a la Statement("with extensive parameters",\ "some continuation here"); I am sick of having to write obscenely long and hard-to-read lines in my config, which I have to do mainly because I issue SQL queries. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671

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[Serusers] ser response port are different with asterisk
by Jun Yin 11 Oct '07

11 Oct '07

I noticed a difference between ser and asterisk. When a phone send a register packet with src port 30001, dst port 5060 and "via:x.x.x.x:32001", ser will response a 200 response with "src port 5060, dst port 30001 and "via:x.x.x.x:32001", but asterisk will response a 200 response with "src port 5060, dst port 32001 and "via:x.x.x.x:32001" Don't know why they make this difference? which one more obey the sip standard? thanks. -- Rgds, Hans Yin Web: homeofhans.homeip.net Email: hansyin(a)gmail.com MSN: hansyin(a)hotmail.com Skype: hans_yin_vancouver

1 0

[OpenSER-Users] Presence BLF in OpenSer?
by Iñaki Baz Castillo 11 Oct '07

11 Oct '07

Hi, many phones as Thomson S2030 or Grandstream 3000 allow presence BLF. Unfortunatelly I think BLF (Busy Lamp Field) just works with Asterisk and it's not a RFC or a draft. Thomson S2030 allow subscribing to a extension state and it generates this SUBSCRIBE: ------------------------------------------------------------------------------- SUBSCRIBE sip:211@domain.org:5061;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.97:5061;branch=z9hG4bK4293581925314869375 From: <sip:210@domain.org:5050>;tag=c0a80101-167c2d To: <sip:211@domain.org:5061> Call-ID: 8b5c839-c0a80101-d-d5(a)192.168.1.97 CSeq: 1 SUBSCRIBE Max-Forwards: 70 Event: dialog Accept: application/dialog-info+xml Expires: 3600 Contact: <sip:210@192.168.1.97:5061;user=phone> User-Agent: THOMSON ST2030 hw5 fw1.52 00-14-7F-00-68-81 Content-Length: 0 ------------------------------------------------------------------------------- Is there any possibility to use this BLF feature with OpenSer in any way? -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[Serusers] Problem with PSTN calls from Asterisk towards SIP clients registered in SER
by Max Weltz 11 Oct '07

11 Oct '07

Hello, I am working on a project in which calls are placed from an Asterisk server towards the PSTN or SIP clients and then connected to another phone of the PSTN or another SIP client. Those calls are placed by a user (caller) through an interface. First a call is placed towards the caller from an extension in Asterisk and once the caller picks up the phone, the second call is placed and both are connected. In a first time all SIP clients were registered to Asterisk and there was no SER on the picture. Everything was working fine. But I have decided to use SER as SIP proxy now. So the SIP clients are registered with SER and Asterisk is still the originating point of all calls and provides the connection to the PSTN. Asterisk and SER are running on the same machine (Asterisk has port 5061 and SER port 5060) and Asterisk is registered to SER: ==sip.conf== [general] port=5061 bindport=5061 bindaddr=0.0.0.0 disallow=all ;allow=gsm allow=ulaw allow=alaw ;context=bogon-calls context=from-sip autocreatepeer=yes register=> asterisk:password@192.168.1.24:5060/maxAS [SERADDRESS] type=friend username=asterisk secret=password host=192.168.1.24 fromdomain=192.168.1.24 [maxAS] type=friend secret=password2 qualify=yes nat=no host=dynamic canreinvite=no context=from-sip Most things work fine: * a user can ask Asterisk to call a SIP client and connect it to another SIP client: Asterisk finds its way to SER and both calls are happily connected. * a user can also ask a call to be placed towards a SIP client and then this call is connected to a PSTN call, no problem either. The problem is when the user wants a call towards the PSTN to be connected to a SIP client. Once the call going towards the PSTN has been answered, Asterisk attempts to contact a SIP client which is registered in SER but SER sees the call coming from "sip:(PSTN phone number of the first call)@hostIP". There is no such client in the database and it is therefore not allowed to connect. In the following example Zap/1-1 is the interface towards the PSTN, 0761111111 the phone number dialled, and 6644 an extension registered with SER. Here is the Asterisk output. > Channel Zap/1-1 was answered. -- Executing Answer("Zap/1-1", "") in new stack -- Executing Goto("Zap/1-1", "from-sip|6644|1") in new stack -- Goto (from-sip,6644,1) -- Executing Answer("Zap/1-1", "") in new stack -- Executing Dial("Zap/1-1", "SIP/6644@SERADDRESS||r") in new stack -- Called 6644@SERADDRESS Oct 10 11:41:16 WARNING[4264]: chan_sip.c:7972 handle_response: Forbidden - wrong password on authentication for INVITE to '"0761111111" <sip:0761111111@192.168.1.24>;tag=as537d1ed7' -- SIP/SERADDRESS-50e7 is circuit-busy Here is the portion of the Asterisk Dialplan that is relevant and the ser.cfg (which comes from http://siprouter.onsip.org/doc/ gettingstarted/): ==extensions.conf== [context1] exten => 666,1,Answer() exten => 666,2,Goto(from-sip,6644,1) [from-sip] exten => _.,1,Answer exten => _.,2,Dial(SIP/${EXTEN}@SERADDRESS,,r) exten => _.,3,Hangup ==ser.cfg== debug=3 fork=no log_stderror=yes listen=192.168.1.24 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:password@192.168.1.26/ser" loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/uri_db.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql:// ser:password@192.168.1.26/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; }; if (method=="ACK") { route(1); break; } if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!www_authorize("192.168.1.24","subscriber")) { www_challenge("192.168.1.24","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("192.168.1.24","subscriber")) { proxy_challenge("192.168.1.24","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } I guess (but I am new to SER) that the idea is to make the call be placed as the SIP user "asterisk" and not as the SIP user "0761111111" but I don't know where to do that (inside asterisk before the SIP call is placed? inside ser.cfg?) or maybe just to skip the authentication for SIP URIs that look like a phone number. I also read in http://siprouter.onsip.org/doc/gettingstarted/ch09.html how to relay calls placed towards the PSTN from SER towards a gateway (Asterisk here) but I think it wouldn't do any good to send back the call to Asterisk. Asterisk would just end up placing another call towards the PSTN and that's not what I want. Any suggestion will be greatly appreciated, Thanks, //Max

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Re: [OpenSER-Users] Set up two SIP to PSTN calls and then connect them
by Andreas Granig 11 Oct '07

11 Oct '07

> To my knowledge, it’s not possible to use OpenSER alone to do this. > Options include WeSIP (beta software, restrictive licence) or Asterisk > (poor implementation of “peer” identification limits usability). Could > anyone advise what other alternatives there are? Is it possible to just > use OpenSER? What about SEMS? In the source tree, there is "examples/ctd.sh", which relies on the help of the involved UACs (using REFER). For just INVITE, SEMS is a good choice, and we plan to submit our click2dial plugin into the trunk within the next days. Andreas

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[OpenSER-Users] IMPORTANT: openser.org server down, Monday, Oct 15, 04:30 CET
by Daniel-Constantin Mierla 11 Oct '07

11 Oct '07

Hello, due to infrastructure maintenance of the place where the current openser.org server is located, Monday Oct 15, 04:30am Central European Time, the services will be down for about 2 hours. That include everything hosted in the server: - web pages (including documentation and dokuwiki) - mailing lists - downloads Hope everything will go according to the schedule and the services will be back online as planned. Cheers, Daniel

1 0

Re: [Serusers] SER + Linksys SPA942 + Call transfer
by roayu 11 Oct '07

11 Oct '07

Hi Atle, Let me make my scenario clear on you. I have 2 softphone and 1 IP phone(SPA942). I would like to call from Softphone A to SPA942, then use SPA942 transfer the call to Softphone B. But, when I do so, the connection still remained on between SPA942 and Softphone B. I managed to get Softphone A onHold and the SPA942 managed to call to Softphone B. Once I press on the 'xfer' button after calling Softphone B, A still onHold and no SIP signal to unHold A and to terminate B. If I would like to configure the ser.cfg file, which part that I need to configure? And how do I do that? Thanks. Cheers, Roa Yu -----Original Message----- From: Atle Samuelsen [mailto:clona@cyberhouse.no] Sent: Thursday, October 11, 2007 3:19 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] SER + Linksys SPA942 + Call transfer Hi Roa,i * roayu <roayu(a)ctisys.net> [071011 04:01]: > Hi, there! > > I have some questions and help. > 1) I would like to know can call transfer being done when the > connection is P2P? When I tried to use relay (mediaproxy) to do the call > transfer, it's able to transfer the call to the other party whereas when the > condition is P2P, it just can't pass the correct signal. I'm proberbly only tierd (9:15 am here) but P2P? Can you provide the signalling and ser.cfg so we can understand what you are trying to establish? I personally think you are trying to do a call transfer from one ua to a other, where the original call went true mediaproxy, but you do not want the "new" transferd call to go true it. > > 2) Can SER support SPA942 ? Or is there some other settings that I need > to configure on SER ? SPA942, SPA962, SPA2102, yea.. all Linksys SPA products work like a charm with SER (Who has a 942 as his primary phone these days) Best Regards ATle > > Thanks. > > Cheers, > Roa Yu > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

1 0

[OpenSER-Users] Questions about pike module
by Iñaki Baz Castillo 11 Oct '07

11 Oct '07

Hi, yesterday I tryed pike module: ------------------------------------------------------------------------------------------------------------ modparam("pike", "sampling_time_unit", 10) modparam("pike", "reqs_density_per_unit", 30) modparam("pike", "remove_latency", 130) route{ ### pike if (!pike_check_req()) { xlog("pike module has detected IP abuse. Terminating message.\n"); exit; }; # Sanity Check Section ... ... } ------------------------------------------------------------------------------------------------------------ I runned sipp and generate lot of messages from my laptop to my OpenSer server. After a while "pike_check_req()" returns FALSE and the message is terminated. Ok. But if during the sipp attack I do a call from my laptop softphone (same public IP then) most of the times the call is accepted, even if I see the xlog message (because sipp atack) and my IP is listed when doing: ~# openserctl fifo pike_list How is possible? And other question: what is exactly "remove_latency" parameter for? I read: "For how long the IP address will be kept in memory after the last request from that IP address. It's a sort of timeout value." - Is it seconds or miliseconds? - Does it mean the time that listed IP's will be "banned" (I mean the IP's appearing in "openserctl fifo pike_list")? I think is not this because I put: modparam("pike", "remove_latency", 9999999999999) and the IP dissapears of listed IP's after a few seconds (10 - 20). Thanks for any explanation. Regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] "exit;" instead of "break;" in pike module doc
by Iñaki Baz Castillo 11 Oct '07

11 Oct '07

Hi, in pike doc section 1.4.1 [*] there appears: Example 1-4. pike_check_req usage ... if (!pike_check_req()) { break; }; ... It should be "exit;", shouldn't it? [*] http://www.openser.org/docs/modules/devel/pike.html#AEN89 Regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 1

[Serusers] SER 2.0 - acc_syslog module report_ack and report_cancels parameters
by Tomasz Zieleniewski 11 Oct '07

11 Oct '07

Hi, Are the report_ack and report_cancels still supported in acc_syslog module? When I make modparam("acc_syslog", "report_ack", 1) I get that parameter was not found in the module. Cheers Tomasz

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Re: [Serusers] new_t: out of mem
by Jiri Kuthan 11 Oct '07

11 Oct '07

It could be. i'ld suggest upgrading SER so that you don't hunt historical errors and then following memory-problem-reporting guidelines (a bit tedious but provides the so needed info about what's wrong). For that, SER needs to be run for a while so that the syslog shows execsively frequentlya allcated fragments which haven't been freed. -jiri At 09:41 11/10/2007, Zappasodi Daniele wrote: >Hello, >I need help to interpret the syslog. >I have a problem with memory, maybe there is memory leak: after few hours and few cps, my ser goes out of memory >Feb 8 17:27:29 SAM-IP ser[2193]: ERROR: sip_msg_cloner: cannot allocate memory >Feb 8 17:27:29 SAM-IP ser[2193]: ERROR: new_t: out of mem: >Feb 8 17:27:29 SAM-IP ser[2193]: ERROR: t_newtran: new_t failed >If I stop the test and I wait, ser doesn't release memory and it continues to refuse every call with "out of mem". > >I have attached the end of the syslog (ser recompiled with DBG_QM_MALLOC options). >I don't understand if the messages in this trace contain info about a memory leak, but if I stop ser immediately after the start without any call, I don't see great differences in the syslog. >I'm using ser 0.9.6. >What else can I do to search more info about memory leak? > >Thanks, >Zappasodi Daniele > >********************************************************************** >The information in this message is confidential and may be legally >privileged. It is intended solely for the addressee. Access to this message >by anyone else is unauthorized. If you are not the intended recipient, any >disclosure, copying, or distribution of the message, or any action or >omission taken by you in reliance on it, is prohibited and may be unlawful. >Please immediately contact the sender if you have received this message inerror. > >********************************************************************** > >_______________________________________________ >Serusers mailing list >Serusers@lists.iptel.orghttp://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

1 0

[Serusers] new_t: out of mem
by Zappasodi Daniele 11 Oct '07

11 Oct '07

Hello, I need help to interpret the syslog. I have a problem with memory, maybe there is memory leak: after few hours and few cps, my ser goes out of memory Feb 8 17:27:29 SAM-IP ser[2193]: ERROR: sip_msg_cloner: cannot allocate memory Feb 8 17:27:29 SAM-IP ser[2193]: ERROR: new_t: out of mem: Feb 8 17:27:29 SAM-IP ser[2193]: ERROR: t_newtran: new_t failed If I stop the test and I wait, ser doesn't release memory and it continues to refuse every call with "out of mem". I have attached the end of the syslog (ser recompiled with DBG_QM_MALLOC options). I don't understand if the messages in this trace contain info about a memory leak, but if I stop ser immediately after the start without any call, I don't see great differences in the syslog. I'm using ser 0.9.6. What else can I do to search more info about memory leak? Thanks, Zappasodi Daniele ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message inerror. **********************************************************************

1 0

[Serusers] need help implementing sip callback
by Rogelio Serrano 11 Oct '07

11 Oct '07

any pointers? -- Lay low and nourish in obscurity

2 1

[Serusers] SER + Linksys SPA942 + Call transfer
by roayu 11 Oct '07

11 Oct '07

Hi, there! I have some questions and help. 1) I would like to know can call transfer being done when the connection is P2P? When I tried to use relay (mediaproxy) to do the call transfer, it's able to transfer the call to the other party whereas when the condition is P2P, it just can't pass the correct signal. 2) Can SER support SPA942 ? Or is there some other settings that I need to configure on SER ? Thanks. Cheers, Roa Yu

2 1

[OpenSER-Users] Set up two SIP to PSTN calls and then connect them
by CSB 11 Oct '07

11 Oct '07

I wish to set up two SIP to PSTN calls and then connect them similar to Jajah (is this called 3pcc?). The calls would be proxied by OpenSER and the PSTN interconnected handled by a third party provider with PSTN interconnect. To my knowledge, it's not possible to use OpenSER alone to do this. Options include WeSIP (beta software, restrictive licence) or Asterisk (poor implementation of "peer" identification limits usability). Could anyone advise what other alternatives there are? Is it possible to just use OpenSER? What about SEMS? Regards Cameron

1 0

[OpenSER-Users] Set up two SIP to PSTN calls and then connect them
by Cameron.Beattie 10 Oct '07

10 Oct '07

I wish to set up two SIP to PSTN calls and then connect them similar to Jajah (is this called 3pcc?). The calls would be proxied by OpenSER and the PSTN interconnected handled by a third party provider with PSTN interconnect. To my knowledge, it's not possible to use OpenSER alone to do this. Options include WeSIP (beta software, restrictive licence) or Asterisk (poor implementation of "peer" identification limits usability). Could anyone advise what other alternatives there are? Is it possible to just use OpenSER? What about SEMS? Regards Cameron

1 0

[OpenSER-Users] logging problem
by Klaus Darilion 10 Oct '07

10 Oct '07

Hi! I just compiled a fresh openser-1.2 (no installation - only "make all" and running it as non-privileged user from the source directory). Everything works fine except that I can not see L_DEBUG messages in syslog (debug=3 causes the same output in syslog as debug=4) and I'm stuck in finding the problem. When starting openser with -D -E -dd I see DEBUG messages too. thanks Klaus

2 5

AW: AW: [OpenSER-Users] MSILO: dump stored messages
by Cesar Troncoso 10 Oct '07

10 Oct '07

Hi Daniel, yes it works fine now, all messages are dumped in the order of receiving! Thanks for fixing it, now I just need to wait some time for openser 1.3 to be released ;-) Cheers, Cesar -----Ursprüngliche Nachricht----- Von: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] Gesendet: Donnerstag, 04. Oktober 2007 16:56 An: Cesar Troncoso Cc: users(a)openser.org Betreff: Re: AW: [OpenSER-Users] MSILO: dump stored messages Hello, can you test with latest trunk version. I added a constraint to order by auto incremented index. Let me know if works ok. Cheers, Daniel On 10/04/07 15:11, Cesar Troncoso wrote: > Hi, > > thanks for reply my email. I made the following test: I sent five different messages: > > one > two > three > four > five > > and I traced the SIP traffic passing through the server. They were stored in the DB in the correct sequence. So far so good. Now comes the problem, I saw the IM going out from my SIP Server and the sequence was wrong: > > three > five > two > four > one > > And exactly in that way appeared the IM in my client when the user went online. > > In my opinion the IM were not properly sent when they went out from OpenSER > > Cheers, > > Cesar > > -----Ursprüngliche Nachricht----- > Von: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > Gesendet: Donnerstag, 04. Oktober 2007 13:28 > An: Cesar Troncoso > Cc: users(a)openser.org > Betreff: Re: [OpenSER-Users] MSILO: dump stored messages > > Hello, > > they should be sent in order of receiving -- however I will double > check. Because they are sent in a very short time frame, maybe the get > in a different order to the user, because of network transmission. Can > you watch the sip traffic sent by proxy? > > Cheers, > Daniel > > > On 10/04/07 13:39, Cesar Troncoso wrote: > >> Hello, >> >> >> >> I have a problem by using the MSILO Module. The messages stored in the >> DB (table SILO) are not sent in the right sequence. Therefore when the >> user goes online again and receives the messages previously stored in >> the DB, he is not able to understand them. Any Ideas or suggestions >> how to deal this problem? >> >> >> >> Thanks, >> >> >> >> Cesar >> >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > >

1 0

Re: [Serusers] Get RealTime Online Users status
by roayu 10 Oct '07

10 Oct '07

Oh, I got it! Thanks Jiri! Btw, I have another questions. 1) I would like to know can call transfer being done when the connection is P2P? When I tried to use relay (mediaproxy) to do the call transfer, it's able to transfer the call to the other party whereas when the condition is P2P, it just can't pass the correct signal. 2) Can SER support SPA942 ? Or is there some settings that I need to configure on SER ? Thanks. Cheers, Roa Yu -----Original Message----- From: Jiri Kuthan [mailto:jiri@iptel.org] Sent: Wednesday, October 10, 2007 2:56 PM To: roayu; SIP Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Get RealTime Online Users status yes, set usrloc's database mode to 1. -jiri At 03:32 10/10/2007, roayu wrote: >Thanks for your reply. I found that it's only update after around 30 >seconds. Is there anyway that I can fasten the update on the database? > >Thanks. > >Cheers, >Roa Yu > >-----Original Message----- >From: SIP [mailto:sip@arcdiv.com] >Sent: Tuesday, October 09, 2007 9:14 PM >To: roayu >Cc: serusers(a)lists.iptel.org >Subject: Re: [Serusers] Get RealTime Online Users status > >roayu wrote: >> >> Hi there! >> >> Can anyone tell me how to get the RealTime Online user status other >> than using command '*serctl ul show*'? Or how can I store the realtime >> online user status to the MySQL db? >> >> Thanks. >> >> Cheers, >> >> Roa Yu J >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >The location table in the MySQL database stores what are, in effect, >currently online users. It is, however, only an approximation. When a UA >registers, it sends an expiration time on its registration and that gets >stored in the location table. If the user disconnects without cancelling >his registration (some UAs send am expire register message to 'log out' >and some don't), then the data may still be in the table until the >expire time occurs. Realistically, though, you can get a good idea of >currenly online users using the location table and while it may not be >100% accurate, it's close enough for government work, as it were. > >N. >_______________________________________________ >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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[Serusers] SER and OpenXCAP
by Tomasz Zieleniewski 10 Oct '07

10 Oct '07

Hi, is there any view about ser working with openXCAP serverI set up openXCAP and tried it but unfortunately there are some incompatibility issues.I got the following error in openXCAP: Invalid Document Selector xcap-root/pres-rules/users/pgr(a)voip.touk.pl/index (the Document Selector must be 'users' or 'global': 'pres-rules') my xcap server listen on port 8000 and my ser config is the following: modparam("xcap", "xcap_root","http://voip.touk.pl:8000/xcap-root") Cheers -tomasz

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