- sr-users - kamailio.org

Re: RE : [Serusers] Install from sources howto
by lakmal silva 15 Mar '05

15 Mar '05

Do a #make all --- Olivier Taylor <olivier.taylor(a)gmail.com> wrote: > Thx for the first part, what about compiling, wich > commands does I have to > issue, they are many possibilities in the > documentation. > > Regards, > > Olivier > > -----Message d'origine----- > De : lakmal silva [mailto:ruwan_lakmal@yahoo.fr] > Envoyé : dimanche 13 mars 2005 13:21 > À : Olivier Taylor; serusers(a)lists.iptel.org > Objet : Re: [Serusers] Install from sources howto > > > Hi Oliver, > > take a look at these links. It will give you a quick > intro to SER intallation and configuration. > > http://www.iptel.org/ser/doc/ser-howto/ser-Howto.html > http://cvs.berlios.de/cgi-bin/viewcvs.cgi/*checkout*/ser/sip_router/INSTALL? > rev=HEAD&content-type=text/plain > > > To untar the tarball, you can issue the following > command. > > #tar -xzvf ser-0.xx_src.tar.gz > > Alternatively you can download the source code from > the cvs tree. > > Regards, > > Lakmal > > > --- Olivier Taylor <olivier.taylor(a)gmail.com> wrote: > > Hi all, > > > > If I understand well, to use all the features of > > Ser, it's necessary to > > build Ser from sources and install it. > > I used rpms to install ser, I did configure it, > > everythings ok, but now I > > wanna do more now. > > > > Is there somewhere an installation guide from > > source, how to untar, to > > compile and so on? > > > > Thx, > > > > Olivier > > > > -----Message d'origine----- > > De : serusers-bounces(a)lists.iptel.org > [mailto:serusers-bounces@lists.iptel.org] De > > la part de Aldi Ramdhani > > Envoyé : dimanche 13 mars 2005 12:36 > > À : serusers(a)lists.iptel.org > > Objet : [Serusers] window messenger 5.0 > > > > > > Dear All, > > > > I'm running ser-0.8.14 on FreeBSD 5.3-RELEASE. I > > install it through ports > > collection with MySQL support. I try ro login > using > > Window Messenger 5.0. It > > works, but i see other user were offline even they > > are online. can somebody > > tell me what's wrong ??? > > > > _______________________________________________ > > 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 > > > > > > __________________________________ > Do you Yahoo!? > Yahoo! Small Business - Try our new resources site! > http://smallbusiness.yahoo.com/resources/ > > __________________________________ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/

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[Serusers] Why Was This INVITE Not Loose_Route()ed ??
by Java Rockx 15 Mar '05

15 Mar '05

Hi All. I just don't see why loose_route() returned FALSE for this INVITE message. Can anyone help? This INVITE was sent from my SIP phone to my SER-0.9 proxy. The IPs look funny because our SER proxy sits behind a Cisco 3600 and the ALG stuff has rewritten the IPs (ie, the 10.3.0.221 address). Regards, Paul U 2005/03/14 16:01:19.932196 69.200.205.122:5060 -> 10.3.0.221:5060 INVITE sip:9195531888@66.243.109.99:5060;user=phone SIP/2.0. Via: SIP/2.0/UDP 69.200.205.122:5060;branch=z9hG4bK2434592534. Route: <sip:10.3.0.221;ftag=10000000-0-1811969809;lr>. Route: <sip:216.229.127.60:5060;lr>. From: 3475626630 <sip:3475626630@66.243.109.99:5060;user=phone>;tag=550070175. To: <sip:9195531888@64.152.60.6;user=phone>;tag=10000000-0-1811969809. Call-ID: 348926-3319820023-453493(a)66.243.109.99. CSeq: 1 INVITE. Contact: <sip:3475626630@69.200.205.122:5060>. max-forwards: 70. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE. Content-Type: application/sdp. Content-Length: 171. . v=0. o=- 9528 3044 IN IP4 0.0.0.0. s=-. c=IN IP4 0.0.0.0. t=0 0. m=audio 13456 RTP/AVP 18 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendonly. a=ptime:20.

2 3

[Serusers] error in "make modules" with postgres
by Joao Pereira 15 Mar '05

15 Mar '05

Hello when I do "make modules", this error comes in the postgres module. Does someone know whats hapening? gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.0"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_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 -I/usr/local/include -I/usr/local/pgsql/include -I/usr/include/postgresql -c aug_alloc.c -o aug_alloc.o aug_alloc.c: In function `mem_bad': aug_alloc.c:118: warning: no return statement in function returning non-void aug_alloc.c: In function `mem_nomem': aug_alloc.c:161: warning: long unsigned int format, size_t arg (arg 4) aug_alloc.c: In function `mem_alloc': aug_alloc.c:228: warning: suggest parentheses around assignment used as truth value aug_alloc.c: In function `aug_set_nomem_func': aug_alloc.c:294: warning: left-hand operand of comma expression has no effect aug_alloc.c:294: warning: left-hand operand of comma expression has no effect aug_alloc.c:293: warning: statement with no effect aug_alloc.c: In function `aug_alloc_loc': aug_alloc.c:309: warning: left-hand operand of comma expression has no effect aug_alloc.c:309: warning: left-hand operand of comma expression has no effect aug_alloc.c:309: warning: left-hand operand of comma expression has no effect aug_alloc.c:309: warning: left-hand operand of comma expression has no effect aug_alloc.c:315: warning: left-hand operand of comma expression has no effect aug_alloc.c:315: warning: left-hand operand of comma expression has no effect aug_alloc.c:308: warning: statement with no effect aug_alloc.c:313: warning: statement with no effect aug_alloc.c: In function `aug_realloc_loc': aug_alloc.c:341: warning: left-hand operand of comma expression has no effect aug_alloc.c:341: warning: left-hand operand of comma expression has no effect aug_alloc.c:341: warning: left-hand operand of comma expression has no effect aug_alloc.c:341: warning: left-hand operand of comma expression has no effect aug_alloc.c:342: warning: left-hand operand of comma expression has no effect aug_alloc.c:371: warning: left-hand operand of comma expression has no effect aug_alloc.c:371: warning: left-hand operand of comma expression has no effect aug_alloc.c:339: warning: statement with no effect aug_alloc.c:342: warning: statement with no effect aug_alloc.c:370: warning: statement with no effect aug_alloc.c: In function `aug_free_loc': aug_alloc.c:384: warning: left-hand operand of comma expression has no effect aug_alloc.c:384: warning: left-hand operand of comma expression has no effect aug_alloc.c:384: warning: left-hand operand of comma expression has no effect aug_alloc.c:384: warning: statement with no effect aug_alloc.c: In function `aug_foster_loc': aug_alloc.c:415: warning: left-hand operand of comma expression has no effect aug_alloc.c:415: warning: left-hand operand of comma expression has no effect aug_alloc.c:415: warning: left-hand operand of comma expression has no effect aug_alloc.c:415: warning: left-hand operand of comma expression has no effect aug_alloc.c:436: warning: left-hand operand of comma expression has no effect aug_alloc.c:414: warning: statement with no effect aug_alloc.c:436: warning: statement with no effect aug_alloc.c:454: warning: statement with no effect aug_alloc.c:460: warning: statement with no effect aug_alloc.c:467: warning: statement with no effect aug_alloc.c: In function `aug_strdup_loc': aug_alloc.c:498: warning: left-hand operand of comma expression has no effect aug_alloc.c:498: warning: left-hand operand of comma expression has no effect aug_alloc.c:498: warning: left-hand operand of comma expression has no effect aug_alloc.c:504: warning: left-hand operand of comma expression has no effect aug_alloc.c:504: warning: left-hand operand of comma expression has no effect aug_alloc.c:498: warning: statement with no effect aug_alloc.c:502: warning: statement with no effect aug_alloc.c: In function `aug_vecdup_loc': aug_alloc.c:524: warning: left-hand operand of comma expression has no effect aug_alloc.c:524: warning: left-hand operand of comma expression has no effect aug_alloc.c:524: warning: statement with no effect gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.0"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_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 -I/usr/local/include -I/usr/local/pgsql/include -I/usr/include/postgresql -c aug_util.c -o aug_util.o aug_util.c: In function `aug_exit': aug_util.c:72: warning: left-hand operand of comma expression has no effect aug_util.c:72: warning: statement with no effect aug_util.c: In function `aug_abort_va': aug_util.c:80: warning: left-hand operand of comma expression has no effect aug_util.c:80: warning: left-hand operand of comma expression has no effect aug_util.c:80: warning: statement with no effect aug_util.c: In function `aug_setmodule': aug_util.c:114: warning: unused variable `len' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align ..... ..... -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -I/usr/local/include -I/usr/local/pgsql/include -I/usr/include/postgresql -c db_val.c -o db_val.o db_val.c: In function `str2valp': db_val.c:219: warning: passing arg 1 of `aug_strdup_loc' discards qualifiers from pointer target type db_val.c:226: warning: passing arg 1 of `memcpy' discards qualifiers from pointer target type db_val.c:251: warning: passing arg 1 of `memcpy' discards qualifiers from pointer target type gcc -shared -Wl,-O2 -Wl,-E aug_alloc.o aug_util.o dbase.o db_con.o db_mod.o db_res.o db_utils.o db_val.o -L/usr/local/lib -L/usr/local/pgsql/lib -L/usr/local/lib/pgsql -L/usr/pkg/lib -L/usr/pkg/lib/pgsql -lpq -o postgres.so

1 0

[Serusers] How to use SER with SIP Communicator
by John Varghese 15 Mar '05

15 Mar '05

Hi friends, I want to work with SIP on a LAN. For this today I have downloaded and installed SER on my local machine. I have managed to login. However I am not able to dial up another user who has logged-in. Both of us are on the same LAN. The error is "Remote part returned error response - 408- Request Timeout" I have kept the domain name as IP of my machine 10.0.0.101, where SER has been installed. The output of "select * from subscriber;" is as follows +----------------------------------+----------+------------+----------+------------+-----------+-------+-------------------+---------------------+---------------------+------------------------------------------+------+------------------+----------+----------------------------------+----------------------------------+-------+------------+----------+---------+ | phplib_id | username | domain | password | first_name | last_name | phone | email_address | datetime_created | datetime_modified | confirmation | flag | sendnotification | greeting | ha1 | ha1b | perms | allow_find | timezone | rpid | +----------------------------------+----------+------------+----------+------------+-----------+-------+-------------------+---------------------+---------------------+------------------------------------------+------+------------------+----------+----------------------------------+----------------------------------+-------+------------+----------+---------+ | ed679a94c2d06c2853e8eac7ccc12359 | john | 10.0.0.101 | john | | | | johnv(a)aftek.com | 2005-03-15 12:25:23 | 2005-03-15 12:26:35 | | o | | | ed679a94c2d06c2853e8eac7ccc12359 | b4bf9c3408f165f59af7eb72438a8e95 | NULL | 0 | NULL | john | | b8ca68a4e7eafeb5e4c7ec9dc3a9eaf2 | sandeep | 10.0.0.101 | sandeep | | | | sandeep(a)aftek.com | 2005-03-15 12:26:59 | 2005-03-15 12:27:18 | | o | | | b8ca68a4e7eafeb5e4c7ec9dc3a9eaf2 | e6705f3068a0bf4c45a0ec2689d55b21 | NULL | 0 | NULL | sandeep | I am pasting some lines from console of sip-communicator.sh Call-ID: 224990ce9dffd68b151bc394f811f687(a)10.0.0.101 CSeq: 1 INVITE From: "John Varghese" <sip:john@10.0.0.101:27882;transport=udp>;tag=22845412 To: <sip:sandeep@10.0.0.101>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-c802 Via: SIP/2.0/UDP 10.0.0.101:27882;branch=z9hG4bK0de890eafa356f841c05dc95564ea035 Server: Sip EXpress router (0.8.14 (i386/linux)) Warning: 392 10.0.0.101:5060 "Noisy feedback tells: pid=3784 req_src_ip=10.0.0.101 req_src_port=27882 in_uri=sip:sandeep@10.0.0.101 out_uri=*sip:61.11.16.124:5060*;transport=udp via_cnt==0" Content-Length: 0 Can anybody please help John

1 0

[Serusers] Where is the 'best practice' of 0.9.0 ser.cfg or above?
by Charles Wang 15 Mar '05

15 Mar '05

Dear ALL: As your discussions before, there is one or more ser.cfg version 0.9.0 for us to practice. Does anyone get it or any kind man can release a version for me to test? I usually test failed at call busy forward and call no answer forward. Help me, plz. Best Regard Charles

6 8

[Serusers] serctl 0.9.0 - cannot add user account
by matt morris 15 Mar '05

15 Mar '05

Hello list, Recently I compiled ser 0.9.0 with mysql acc, and ser seems working ok, except I get a warning message twice when I run SER: fix_socket_list: could not rev. resolve 192.168.0.100 (my SER box's internal ip) Output: Listening on udp: 192.168.0.100 [192.168.0.100]:5060 tdp: 192.168.0.100 [192.168.0.100]:5060 Aliases: *:(its WAN ip address):* *:ser1.mydomain.com:* etc And I port forwarded tcp and udp5060 to my SER box. Why do I get this warning?Is it simply because my SER box is behind NAT? Back to my main question: I got an error 400; check if you use aliases in SER message when I tried "serctl add newusername newpasswd email(a)mail.com". I exported my SIP_DOMAIN to be ser1.mydomain.com and also updated the serctl script with the appropriate db domain and access username and password. Here is the global definition section of my ser.cfg: debug=3 fork=yes listen=192.168.0.100 alias=ser1.mydomain.com alias=(my WAN ip) port=5060 children=4 fifo="/tmp/ser_fifo" fifo_db_url="mysql://username:pass@mysqldomain.com/ser" ... Thank you in advance. Best regards, Matt _________________________________________________________________ Scan and help eliminate destructive viruses from your inbound and outbound e-mail and attachments. http://join.msn.com/?pgmarket=en-ca&page=byoa/prem&xAPID=1994&DI=1034&SU=ht… Start enjoying all the benefits of MSN® Premium right now and get the first two months FREE*.

1 0

[Serusers] Questions about SER+SERWEB+SEMS deployment
by matt morris 15 Mar '05

15 Mar '05

Hello List, I'm a new user of SER. Just some questions about how I could deploy SER+SERWEB+SEMS, possibly on separate hosts, as I plan to have more than one SER proxy server with a centralized web server and media/voice mail server (1) Basically what features will I be missing if I decide to omit the fifo functions (if possible) in serweb? ( aliases, add new contact SIP address and who are online would be three features, right?) (2) I saw in one of the threads that there is a fifo_server.php file available for fifo-Internet relay, but not quite sure how to use it. Could someone give me some pointers? (3) Somewhere in the mailing list I saw the "fifo_db_url" global declaration, is it a feature only available in version 0.9.0? And is there a serweb version that will work with ser 0.9.0? (4) I have not tried SEMS yet, but since it also communicates with SER via fifo, I would assume it has to be in the same host as SER, right? Are there any existing modules/plugins that will allow SER and SEMS on different hosts? If not, would it be more scalable if I build an Asterisk box to handle voice mails? (5) If I have multiple SER proxy servers on different hosts, how could UA Alpha registered with SER proxy A be able to call UA beta which registered with SER proxy B? What setup/configuration would I need? I know I have some dumb questions - I'm new to voip and SER. Please try to help me out, I would really appreciate it. Thank you in advance. _________________________________________________________________ Take charge with a pop-up guard built on patented Microsoft® SmartScreen Technology. http://join.msn.com/?pgmarket=en-ca&page=byoa/prem&xAPID=1994&DI=1034&SU=ht… Start enjoying all the benefits of MSN® Premium right now and get the first two months FREE*.

2 3

[Serusers] Registration with digest authentication
by Tan, Chiang Kang 15 Mar '05

15 Mar '05

Hi all, May I know where to find some guidelines on how to implement a UA to do registration with digest authentication? Any links you are aware of will be useful. cheers Chiang

2 1

[Serusers] Intermittent Audio
by pat newham 15 Mar '05

15 Mar '05

Following on from my below email, I can now definately say the problem is not nat pings. Just to recap I am experiencing intermittent audio. It works when the phones have very recently registered, then sometimes theres one way audio and then sometimes no audio. Does anyone have any ideas what the problem could be or where I could begin to troubleshoot this? Hi, I have a strange problem. I have two grandstream budgetone clients on the same subnet behind nat registering with ser on a public address. Obviously their public addresses would be the same but they listen on different ports. When they initially register, I can the call,audio is transmitted and everything is successful. However sometimes theres only one way audio, other times theres no audio and then other times it works....I am guessing that this is because the nat router is forgetting the nat mapping so after a while when the nat mapping is "forgotten" and a packet arrives destined for a client, the router drops it.... Could someone verify this for me??...Am I on the right track?? I have the following settings in ser.cfg which I thought would keep the nat settings alive. modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT I also increased the nat keep alives "pings" sent in the configuration settings of the grandstream phone....Any further ideas?? Regards, Pat. Send instant messages to your online friends http://uk.messenger.yahoo.com

2 1

[Serusers] HELP!
by Choon Mun 15 Mar '05

15 Mar '05

Hi there everyone, I need help here. This is my ser.cfg file. Now i can use SJphone call to SJphone. However, when i try to call from SJphone to my PSTN no, it is connected but i'm facing problem to hear the voice from the receiver. Receiver also cant hear my voice. Is there anyone there who can give me a help. Thanks for everything. Regards, Lee # # $Id: nathelper.cfg,v 1.1.2.1 2003/11/24 14:47:18 janakj Exp $ # # simple quick-start config script including nathelper support # This default script includes nathelper support. To make it work # you will also have to install Maxim's RTP proxy. The proxy is enforced # if one of the parties is behind a NAT. # # If you have an endpoing in the public internet which is known to # support symmetric RTP (Cisco PSTN gateway or voicemail, for example), # then you don't have to force RTP proxy. If you don't want to enforce # RTP proxy for some destinations than simply use t_relay() instead of # route(1) # # Sections marked with !! Nathelper contain modifications for nathelper # # NOTE !! This config is EXPERIMENTAL ! # # ----------- global configuration parameters ------------------------ debug=7 # debug level (cmd line: -dddddddddd) fork=yes port=5060 log_stderror=yes # (cmd line: -E) fifo="/tmp/ser_fifo" #memlog=3 # Uncomment these lines to enter debugging mode /* debug=3 fork=yes port=5060 log_stderror=yes */ #check_via=no # (cmd. line: -v) #dns=no # (cmd. line: -r) #rev_dns=no # (cmd. line: -R) #children=4 #fifo="/tmp/ser_fifo" #alias="sipproxy.com" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/acc.so" #loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/print.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/exec.so" #loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/group_radius.so" #loadmodule "/usr/local/lib/ser/modules/uri.so" #loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/uri_radius.so" #loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" #loadmodule "/usr/local/lib/ser/modules/option.so" #loadmodule "/usr/local/lib/ser/modules/speeddial.so" # ----------------- setting module-specific parameters --------------- #modparam("acc|auth_db|group|uri_db|usrloc", "db_url", "mysql://root:mysqldb04@localhost/ser") #modparam("usrloc", "db_url", "mysql://root:mysqldb04@localhost/ser") #---------------------------- use_domain setting #modparam("auth_db|group|uri_db|usrloc", "use_domain", 1) #---------------------------- usrloc params #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) #modparam("usrloc", "timer_interval", 60) #modparam("usrloc", "desc_time_order", 1) #---------------------------- auth params modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("auth_radius", "service_type", 15) modparam("group_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("group_radius", "use_domain", 1) #modparam("auth_db", "user_column", "user_id") #modparam("auth_db", "calculate_ha1", yes) #modparam("auth_db", "password_column", "password") modparam("uri_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("uri_radius", "service_type", 11) #-----------------------------accounting params modparam("acc", "log_level", 1) modparam("acc", "radius_flag", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") #modparam("acc", "db_url", "mysql://ser:elanetw123@localhost/ser") #modparam("acc", "radius_log_flag", 1) modparam("acc", "radius_missed_flag", 2) #----------------------------- tm params modparam("tm", "fr_timer", 12) modparam("tm", "fr_inv_timer", 24) #----------------------------- rr params # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_double_rr", 0) modparam("rr", "enable_full_lr", 1) #----------------------------- registrar params #modparam("registrar", "nat_flag", 6) #modparam("registrar", "min_expires", 60) #modparam("registrar", "max_expires", 86400) #modparam("registrar", "default_expires", 3600) #modparam("registrar", "desc_time_order", 1) #modparam("registrar", "append_branches", 1) #modparam("registrar", "use_domain", 1) #----------------------------- nathelper params modparam("nathelper", "natping_interval", 10) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT #----------------------------- exec params modparam("exec", "setvars", 1) modparam("exec", "time_to_kill", 10) #----------------------------- domain params #modparam("domain", "db_mode", 1) #----------------------------- Xlog params modparam("xlog", "buf_size", 8192) # ------------------------- request routing logic ------------------- # main routing logic route{ # ------------------------------------------------------------ # Sanity checks -- messages # ------------------------------------------------------------ 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; }; # ------------------------------------------------------------ # Option Section # ------------------------------------------------------------ #if (search("^From: sip:sipsak@") && # (method=="OPTIONS") && (!uri=~:"sip:.*[@]+.*")) { # option_reply(); # break; #}; if (loose_route()) {t_relay(); break;}; if (method=="BYE" || method=="CANCEL" || method=="INVITE") { setflag(1); log(1, "Accounting Flag ON\n"); }; if (method=="INVITE") { setflag(2); log(1, "Missed Call Flag ON\n"); }; # !! Nathelper # ----------------------------------------------------------- # NAT test Section # ----------------------------------------------------------- if (nat_uac_test("3")) { if (method == "REGISTER" ||(!search("^Record-Route:"))) { log("LOG: Someone trying to register from private IP, rewriting\n"); fix_nated_contact(); # Rewrite contact with source IP of signalling # if (method == "REGISTER") { # fix_nated_register(); # } else if (method=="INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; if (!method=="REGISTER") record_route(); if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (uri=~"^sip:101*") { rewritehost("202.xxx.xx.xxx:5060"); # t_relay_to_udp("202.xxx.xx.xxx", "5060"); forward(uri:host, uri:port); # append_hf("p-hint:PSTN call\r\n"); # route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; if (uri==myself) { if (method=="REGISTER") { # log(1, "REGISTER: Authenticating user\n"); if (!radius_proxy_authorize("")) { # log(1, "REGISTER: challenging user\n"); proxy_challenge("", "1"); break; }; save("location"); break; }; #if (method=="INVITE") { # log(1, "INVITE\n"); # setflag(1); #rad_acc_request("Start"); #}; #if (method=="MESSAGE") { # log(1, "MESSAGE\n"); #setflag(1); #}; #if (method=="BYE" || method=="CANCEL") { # log (1, "BYE or CANCEL\N"); # setflag(1); #rad_acc_request("Stop"); #}; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sethostport("202.xxx.xx.xxx"); sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } # !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); } else if (nat_uac_test("1")) { fix_nated_contact(); }; }

1 0

[Serusers] digest authentication problem
by sip test 15 Mar '05

15 Mar '05

Hi all, I am new to ser. I am having the common problem where WM(4.7.2009) cannot be authenticated after the digest authentication is turn on. I have search for the solution posted in this forum and tried it but still cant solved it. I am running ser-0.8.12 on CENTOS. Please advice. Regards mo here is my ser.cfg # ----------- global configuration parameters ------------------------ # debug=3 # debug level (cmd line: -dddddddddd) # fork=yes # log_stderror=no # (cmd line: -E) listen=192.168.10.75 alias=server.com /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) # port=5060 # children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database 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" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "sql://ser:heslo@localhost/ser") # -- auth params -- modparam("auth_db", "db_url", "sql://ser:heslo@localhost/ser") # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # 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) # ------------------------- 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"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("server.com", "subscriber")) { www_challenge("server.com", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } --------------------------------- Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. Learn more.

1 0

RE: [Serusers] SER presence support
by kiko.vives＠ua.es 15 Mar '05

15 Mar '05

I will try it. Thanks! ---- Mensaje Original ---- > IN ser, pass SUBSCIRBE/NOTIFY to the destination UA. Pretty much the same > logic as INVITE/BYE. > > In WM, add other UA SIP addresses in the buddy list. Once you add, it sends > a SUBSCRIBE to the other UAs. When the other UAs change status, it sends > back NOTIFY. > > Richard > > > > -----Original Message----- > > From: kiko.vives(a)ua.es [mailto:kiko.vives@ua.es] > > Sent: Wednesday, March 09, 2005 8:11 AM > > To: richard(a)o-matrix.org; Chiang.Tan(a)thalesgroup.com; serusers(a)lists.iptel.org > > Subject: RE: [Serusers] SER presence support > > > > Hello Richard. > > > > How can I configure WM 5.0 to work in P2P mode as you say in your > > post ? > > > > Thanks ! > > > > ---- Mensaje Original ---- > > > > > Presence support has two ways, peer to peer and presence agent. > > > > > > In the p2p mode, SIP UAs exchange SUBSCRIBE/NOTIFY directly > > between each > > > other. Ser only forwards the sip packet. I got WM 5.0/5.1 working > > in this > > > mode. This mode heavily relies on UAs to support the feature. > > There are > > > still compatibility issues between UA implementations. > > > > > > In the presence agent mode, SIP UAs register with ser which stores > > the state > > > of UAs. Whenever there is a state change, ser will notify the > > interested > > > UAs. Ser has a PA module which is supposed to work this way. > > However I > > > haven't been able to make it work. As a side note, xten wrote a > > big patch > > > for the PA module a while ago, but unfortunately not in cvs yet. > > > > > > Xten has a product called 'eyebeam' which supports both modes. It > > is not > > > free. I would recommend waiting for a while before it becomes more > > stable. > > > It crashed Windows twice during our demo. > > > > > > Richard > > > > > > > > > > > > > -----Original Message----- > > > > From: serusers-bounces(a)lists.iptel.org [mailto:serusers- > > bounces(a)iptel.org] On > > > > Behalf Of Tan, Chiang Kang > > > > Sent: Wednesday, March 09, 2005 6:30 AM > > > > To: serusers(a)lists.iptel.org > > > > Subject: [Serusers] SER presence support > > > > > > > > Hi all, > > > > What's the current status of presence support in SER? May I > > know > > > > what are the clients, similar to windows messenger, that will > > interwork > > > > with > > > > SER? As I understand that windows messenger above version 5 does > > not work > > > > with ser anymore. > > > > > > > > cheers > > > > Chiang > > > > > > > > _______________________________________________ > > > > 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

[Serusers] FreeBSD
by Barry Murphy 15 Mar '05

15 Mar '05

I'm trying to offer a Iptel service within NZ with servers located in NZ to save on costly intl bandwidth. I want to install all modules with MySQL support so downloaded the source to install from, however I get this error: root@max:/home/icepick/ser-0.8.14# make modules "Makefile.sources", line 17: Missing dependency operator "Makefile.sources", line 19: Need an operator "Makefile", line 52: Missing dependency operator Error expanding embedded variable. So I tried installing from the /freebsd/ folder seeing as it looked like a port. root@max:/home/icepick/ser-0.8.14/freebsd# make Makefile error: there is no COMMENT variable defined for this port. Please, rectify this. *** Error code 1 Stop in /usr/home/icepick/ser-0.8.14/freebsd. So I took a look at the Makefile where it said .if freebsd >5 or whatever then use the COMMENTt tag, so I uncommented that and now get ... root@max:/home/icepick/ser-0.8.14/freebsd# make ===> Vulnerability check disabled => ser-0.8.12_src.tar.gz is not in /usr/home/icepick/ser-0.8.14/freebsd/distinfo. => Either /usr/home/icepick/ser-0.8.14/freebsd/distinfo is out of date, or => ser-0.8.12_src.tar.gz is spelled incorrectly. *** Error code 1 Stop in /usr/home/icepick/ser-0.8.14/freebsd. So I tried putting ser-0.8.14_src.tar.gz in every possible place, but still get the above error: Next up has anyone got example files to do the following: 1./ Allow users to signup using serweb (which I couldn't get working properly) 2./ Force users to signup with 64 (NZ) prefix infront of the number (I managed to edit this regex) 3./ Call eachother freely, so some sort of acl to say 64xxxxx calls can be made 4./ Linking to SipPhone 5./ Linking to FWD 6./ I'll link to many more, so one example would be good. If someone would like to help with this or possibly provide configs and modified source, It'd be much appreciated. Thanks Barry

2 2

Re: [Serusers] error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy
by Java Rockx 15 Mar '05

15 Mar '05

Charles, you can read through the README that is included with mediaproxy (the server) and the README that ins included in the ser mediaproxy module. Basically you need to first start the mediaproxy Python server. Once that is running you can start ser. rtpgenerator.py is just a stress testing tool. dnssrv is not required if you're just using one mediaproxy instance. If you ever get to multi-host deployments, then you may want to use dns srv records for load balancing. P On Tue, 22 Feb 2005 00:30:40 +0800, Charles Wang <lazy.charles(a)gmail.com> wrote: > Dear Paul : > > If I want to run a mediaproxy mode, what do I have to run? > serctl and mediaproxy.py and what? > Is it necessary about rtpgenerator.py? > And is it is necessary to setup a SRV service on DNS? > > After your explain, and I download the last Python and run mediaproxy well. > > But my call busy forward is still failed. > > On Mon, 21 Feb 2005 08:02:03 -0500, Java Rockx <javarockx(a)gmail.com> wrote: > > Are you saying that you have to NATed clients that can communicate? > > > > If so, the reason is that they are being the same firewall and thus > > can directly reach each other. > > > > Regards, > > Paul > > > > > > On Mon, 21 Feb 2005 20:54:33 +0800, Charles Wang <lazy.charles(a)gmail.com> wrote: > > > Dear Paul: > > > > > > I guess that you are right. > > > But I can not understand why I can make a call from two different NATs? > > > Don't they communicate via my ser? Or just signal channel via ser? > > > > > > Thank you for your explain. > > > > > > Charles > > > > > > On Mon, 21 Feb 2005 07:41:58 -0500, Java Rockx <javarockx(a)gmail.com> wrote: > > > > MediaProxy is a two-part project. SER only includes the interface to > > > > the actual mediaproxy. You must install the actual mediaproxy server, > > > > which is written in Python. > > > > > > > > http://www.ag-projects.com/MediaProxy.html > > > > > > > > Regards, > > > > Paul > > > > > > > > > > > > On Mon, 21 Feb 2005 20:26:19 +0800, Charles Wang <lazy.charles(a)gmail.com> wrote: > > > > > Dear Paul: > > > > > > > > > > So surprised it is. I never see any executable file or script named > > > > > "mediaproxy.py" in my source files. > > > > > I guess that the mediaproxy is only a module including in > > > > > "sip_router/modules/mediaproxy/mediaproxy.so". > > > > > When I can get a executable version of mediaproxy or where the mediaproxy.py is? > > > > > > > > > > And also I miss the "proxydispatcher". I can find a directory named > > > > > "dispatcher" under my sip_router/modules. > > > > > > > > > > I usually start my ser with "/usr/local/sbin/serctl start". > > > > > > > > > > Please let me know where I can get these files what you said. > > > > > > > > > > On Mon, 21 Feb 2005 07:04:48 -0500, Java Rockx <javarockx(a)gmail.com> wrote: > > > > > > Charles, > > > > > > > > > > > > How are you starting mediaproxy? Below is the start script I use. > > > > > > > > > > > > Regards, > > > > > > Paul > > > > > > > > > > > > #!/bin/sh > > > > > > # > > > > > > # chkconfig: 2345 90 20 > > > > > > # description: VoIP RTP Proxy Server > > > > > > # > > > > > > # processname: mediaproxy > > > > > > # pidfile: /var/run/mediaproxy.pid > > > > > > > > > > > > # source function library > > > > > > . /etc/rc.d/init.d/functions > > > > > > > > > > > > PATH=/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin > > > > > > > > > > > > INSTALL_DIR="/usr/local" > > > > > > RUNTIME_DIR="/var/run" > > > > > > > > > > > > PROXY="$INSTALL_DIR/mediaproxy/mediaproxy.py" > > > > > > DISPATCHER="$INSTALL_DIR/mediaproxy/proxydispatcher.py" > > > > > > PROXY_PID="$RUNTIME_DIR/mediaproxy.pid" > > > > > > DISPATCHER_PID="$RUNTIME_DIR/proxydispatcher.pid" > > > > > > > > > > > > # Options for mediaproxy and dispatcher. Do not include --pid <pidfile> > > > > > > # --pid <pidfile> will be added automatically if needed. > > > > > > PROXY_OPTIONS="--ip=24.48.42.20 --listen=127.0.0.1" > > > > > > DISPATCHER_OPTIONS="domain://sip.mycompany.com" > > > > > > > > > > > > NAME="mediaproxy" > > > > > > DESC="SER MediaProxy server" > > > > > > > > > > > > test -f $PROXY || exit 0 > > > > > > test -f $DISPATCHER || exit 0 > > > > > > > > > > > > if [ "$PROXY_PID" != "/var/run/mediaproxy.pid" ]; then > > > > > > PROXY_OPTIONS="--pid $PROXY_PID $PROXY_OPTIONS" > > > > > > fi > > > > > > if [ "$DISPATCHER_PID" != "/var/run/proxydispatcher.pid" ]; then > > > > > > DISPATCHER_OPTIONS="--pid $DISPATCHER_PID $DISPATCHER_OPTIONS" > > > > > > fi > > > > > > > > > > > > start() { > > > > > > echo -n "Starting $DESC: $NAME" > > > > > > $PROXY $PROXY_OPTIONS > > > > > > $DISPATCHER $DISPATCHER_OPTIONS > > > > > > echo "." > > > > > > } > > > > > > > > > > > > stop () { > > > > > > echo -n "Stopping $DESC: $NAME" > > > > > > kill `cat $PROXY_PID` > > > > > > kill `cat $DISPATCHER_PID` > > > > > > echo "." > > > > > > } > > > > > > > > > > > > case "$1" in > > > > > > start) > > > > > > start > > > > > > ;; > > > > > > stop) > > > > > > stop > > > > > > ;; > > > > > > restart|force-reload) > > > > > > stop > > > > > > #sleep 1 > > > > > > start > > > > > > ;; > > > > > > *) > > > > > > echo "Usage: /etc/init.d/$NAME {start|stop|restart|force-reload}" >&2 > > > > > > exit 1 > > > > > > ;; > > > > > > esac > > > > > > > > > > > > exit 0 > > > > > > > > > > > > > > > > > > On Mon, 21 Feb 2005 13:39:10 +0800, Charles Wang <lazy.charles(a)gmail.com> wrote: > > > > > > > Dear ALL: > > > > > > > > > > > > > > I make the UA 1033 as busy status(pick up the phone and do nothing). > > > > > > > > > > > > > > Then I try to make a call from 1011 to 1033. It should be redirect to > > > > > > > a PSTN phone number( sip:0939749xxx@ser.xxx.net.tw ). > > > > > > > I find it call failure and jump to failure_route[1], then swicth to > > > > > > > route[3](for PSTN). > > > > > > > But when I dump the package using ngrep. I can't find it try to > > > > > > > connect PSTN trunk( xxx.xxx.190.243 is a CISCO 5300). > > > > > > > I can find "SER: Connecting to PSTN....." message list at log file. > > > > > > > Then the log display "SER: SIP Call On-Net section route(2)" message. > > > > > > > It means that the call return to route[2] then failed. > > > > > > > I guess that it stops or jump out route[3] after rewritehost(xxx.xxx.190.243). > > > > > > > > > > > > > > Can anyone help me to trace the bug in the route[3]? > > > > > > > > > > > > > > Best Regard > > > > > > > Charles > > > > > > > > > > > > > > Subset of ser.cfg about route[3: > > > > > > > ------------------------------------------------------------------------------------------------------- > > > > > > > route[3] { > > > > > > > log(1, "SER: Demestic Call Off-Net section route(3)\n"); > > > > > > > > > > > > > > # All Domestic Calls Go To CISCO 5300 > > > > > > > if (method=="INVITE") { > > > > > > > if (!proxy_authorize("", "subscriber")) { > > > > > > > proxy_challenge("", "0"); > > > > > > > break; > > > > > > > } else if (!check_from()) { > > > > > > > log(1, "Spoofed SIP call attempt"); > > > > > > > sl_send_reply("403", "Use From=ID"); > > > > > > > break; > > > > > > > } else if (!(is_from_local() || is_uri_host_local())) { > > > > > > > sl_send_reply("403", "Please register to use our service"); > > > > > > > break; > > > > > > > }; > > > > > > > # enable caller id blocking for PSTN calls > > > > > > > if (isflagset(25)) { > > > > > > > append_rpid_hf(); > > > > > > > }; > > > > > > > }; > > > > > > > # SIP->PSTN calls get 45 seconds to timeout > > > > > > > log(1, "SER: Connecting to PSTN.....\n"); > > > > > > > avp_write("i:45", "inv_timeout"); > > > > > > > rewritehost("61.220.190.243"); > > > > > > > if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && > > > > > > > !search("^Route:")){ > > > > > > > sl_send_reply("479", "We don't forward to private IP addresses"); > > > > > > > break; > > > > > > > }; > > > > > > > if (method=="INVITE" || method=="ACK") { > > > > > > > use_media_proxy(); > > > > > > > }; > > > > > > > if (isflagset(31)) { # is voice mail? > > > > > > > t_on_failure("1"); > > > > > > > }; > > > > > > > t_on_reply("1"); > > > > > > > if (!t_relay()) { > > > > > > > if (method=="INVITE" || method=="ACK") { > > > > > > > end_media_session(); > > > > > > > }; > > > > > > > sl_reply_error(); > > > > > > > }; > > > > > > > } > > > > > > > > > > > > > > Dump using ngrep -d eth0 -W byline port 5060 > > > > > > > ------------------------------------------------------------------ > > > > > > > Notes: xxx.xxx.190.248 : is SER sip proxy > > > > > > > xxx.xxx.13.49 : is NAT > > > > > > > sip:1033@xxx.xxx.13.49:33536 : location of 1033 > > > > > > > sip:1011@xxx.xxx.13.49:35700 : location of 1011 > > > > > > > > > > > > > > interface: eth0 (xxx.xxx.190.240/255.255.255.240) > > > > > > > filter: ip and ( port 5060 ) > > > > > > > # > > > > > > > U xxx.xxx.13.49:35700 -> xxx.xxx.190.248:5060 > > > > > > > INVITE sip:1033@ser.xxx.net.tw SIP/2.0. > > > > > > > Via: SIP/2.0/UDP > > > > > > > 10.18.1.70:5060;rport;branch=z9hG4bK2754265110CD4289ACDA9EA0769C1A8F. > > > > > > > From: 1011 <sip:1011@ser.xxx.net.tw>;tag=3281269171. > > > > > > > To: <sip:1033@ser.xxx.net.tw>. > > > > > > > Contact: <sip:1011@10.18.1.70:5060>. > > > > > > > Call-ID: B6E73E25-87F9-40BB-A895-561E4ADE8AC8(a)10.18.1.70. > > > > > > > CSeq: 26231 INVITE. > > > > > > > Max-Forwards: 70. > > > > > > > Content-Type: application/sdp. > > > > > > > User-Agent: X-PRO build 1082. > > > > > > > Content-Length: 264. > > > > > > > . > > > > > > > v=0. > > > > > > > o=1011 46237906 46237906 IN IP4 10.18.1.70. > > > > > > > s=X-PRO. > > > > > > > c=IN IP4 10.18.1.70. > > > > > > > t=0 0. > > > > > > > m=audio 8000 RTP/AVP 8 3 98 97 101. > > > > > > > a=rtpmap:8 pcma/8000. > > > > > > > a=rtpmap:3 gsm/8000. > > > > > > > a=rtpmap:98 iLBC/8000. > > > > > > > a=rtpmap:97 speex/8000. > > > > > > > a=rtpmap:101 telephone-event/8000. > > > > > > > a=fmtp:101 0-15. > > > > > > > > > > > > > > # > > > > > > > U xxx.xxx.190.248:5060 -> xxx.xxx.13.49:35700 > > > > > > > SIP/2.0 407 Proxy Authentication Required. > > > > > > > Via: SIP/2.0/UDP > > > > > > > 10.18.1.70:5060;rport=35700;branch=z9hG4bK2754265110CD4289ACDA9EA0769C1A8F;received=xxx.xxx.13.49. > > > > > > > From: 1011 <sip:1011@ser.xxx.net.tw>;tag=3281269171. > > > > > > > To: <sip:1033@ser.xxx.net.tw>;tag=67771a809cdfb71129a699a517fbb1f0.7608. > > > > > > > Call-ID: B6E73E25-87F9-40BB-A895-561E4ADE8AC8(a)10.18.1.70. > > > > > > > CSeq: 26231 INVITE. > > > > > > > Proxy-Authenticate: Digest realm="ser.xxx.net.tw", > > > > > > > nonce="42196d5483841de0d6b79c7dfb4156e2c932e4cb". > > > > > > > Server: Sip EXpress router (0.10.99-dev0 (i386/linux)). > > > > > > > Content-Length: 0. > > > > > > > Warning: 392 xxx.xxx.190.248:5060 "Noisy feedback tells: pid=4199 > > > > > > > req_src_ip=xxx.xxx.13.49 req_src_port=35700 > > > > > > > in_uri=sip:1033@ser.xxx.net.tw out_uri=sip:1033@xxx.xxx.13.49:33536 > > > > > > > via_cnt==1". > > > > > > > . > > > > > > > > > > > > > > # > > > > > > > U xxx.xxx.13.49:35700 -> xxx.xxx.190.248:5060 > > > > > > > ACK sip:1033@ser.xxx.net.tw SIP/2.0. > > > > > > > Via: SIP/2.0/UDP > > > > > > > 10.18.1.70:5060;rport;branch=z9hG4bK2754265110CD4289ACDA9EA0769C1A8F. > > > > > > > From: 1011 <sip:1011@ser.xxx.net.tw>;tag=3281269171. > > > > > > > To: <sip:1033@ser.xxx.net.tw>;tag=67771a809cdfb71129a699a517fbb1f0.7608. > > > > > > > Contact: <sip:1011@10.18.1.70:5060>. > > > > > > > Call-ID: B6E73E25-87F9-40BB-A895-561E4ADE8AC8(a)10.18.1.70. > > > > > > > CSeq: 26231 ACK. > > > > > > > Max-Forwards: 70. > > > > > > > Content-Length: 0. > > > > > > > . > > > > > > > > > > > > > > # > > > > > > > U xxx.xxx.13.49:35700 -> xxx.xxx.190.248:5060 > > > > > > > INVITE sip:1033@ser.xxx.net.tw SIP/2.0. > > > > > > > Via: SIP/2.0/UDP > > > > > > > 10.18.1.70:5060;rport;branch=z9hG4bK1BDE5BF2809C4223A13FC5B1A9AC21CC. > > > > > > > From: 1011 <sip:1011@ser.xxx.net.tw>;tag=3281269171. > > > > > > > To: <sip:1033@ser.xxx.net.tw>. > > > > > > > Contact: <sip:1011@10.18.1.70:5060>. > > > > > > > Call-ID: B6E73E25-87F9-40BB-A895-561E4ADE8AC8(a)10.18.1.70. > > > > > > > CSeq: 26232 INVITE. > > > > > > > Proxy-Authorization: Digest > > > > > > > username="1011",realm="ser.xxx.net.tw",nonce="42196d5483841de0d6b79c7dfb4156e2c932e4cb",response="7004beab12b3ac6874c5cd86e5659930",uri="sip:1033@ser.xxx.net.tw". > > > > > > > Max-Forwards: 70. > > > > > > > Content-Type: application/sdp. > > > > > > > User-Agent: X-PRO build 1082. > > > > > > > Content-Length: 264. > > > > > > > . > > > > > > > v=0. > > > > > > > o=1011 46238507 46238507 IN IP4 10.18.1.70. > > > > > > > s=X-PRO. > > > > > > > c=IN IP4 10.18.1.70. > > > > > > > t=0 0. > > > > > > > m=audio 8000 RTP/AVP 8 3 98 97 101. > > > > > > > a=rtpmap:8 pcma/8000. > > > > > > > a=rtpmap:3 gsm/8000. > > > > > > > a=rtpmap:98 iLBC/8000. > > > > > > > a=rtpmap:97 speex/8000. > > > > > > > a=rtpmap:101 telephone-event/8000. > > > > > > > a=fmtp:101 0-15. > > > > > > > > > > > > > > # > > > > > > > U xxx.xxx.190.248:5060 -> xxx.xxx.13.49:35700 > > > > > > > SIP/2.0 100 trying -- your call is important to us. > > > > > > > Via: SIP/2.0/UDP > > > > > > > 10.18.1.70:5060;rport=35700;branch=z9hG4bK1BDE5BF2809C4223A13FC5B1A9AC21CC;received=xxx.xxx.13.49. > > > > > > > From: 1011 <sip:1011@ser.xxx.net.tw>;tag=3281269171. > > > > > > > To: <sip:1033@ser.xxx.net.tw>. > > > > > > > Call-ID: B6E73E25-87F9-40BB-A895-561E4ADE8AC8(a)10.18.1.70. > > > > > > > CSeq: 26232 INVITE. > > > > > > > Server: Sip EXpress router (0.10.99-dev0 (i386/linux)). > > > > > > > Content-Length: 0. > > > > > > > Warning: 392 xxx.xxx.190.248:5060 "Noisy feedback tells: pid=4201 > > > > > > > req_src_ip=xxx.xxx.13.49 req_src_port=35700 > > > > > > > in_uri=sip:1033@ser.xxx.net.tw out_uri=sip:1033@xxx.xxx.13.49:33536 > > > > > > > via_cnt==1". > > > > > > > . > > > > > > > > > > > > > > # > > > > > > > U xxx.xxx.190.248:5060 -> xxx.xxx.13.49:33536 > > > > > > > INVITE sip:1033@xxx.xxx.13.49:33536 SIP/2.0. > > > > > > > Record-Route: <sip:xxx.xxx.190.248;ftag=3281269171;lr=on>. > > > > > > > Via: SIP/2.0/UDP xxx.xxx.190.248;branch=z9hG4bKa7ac.b59d9107.0. > > > > > > > Via: SIP/2.0/UDP > > > > > > > 10.18.1.70:5060;received=xxx.xxx.13.49;rport=35700;branch=z9hG4bK1BDE5BF2809C4223A13FC5B1A9AC21CC. > > > > > > > From: 1011 <sip:1011@ser.xxx.net.tw>;tag=3281269171. > > > > > > > To: <sip:1033@ser.xxx.net.tw>. > > > > > > > Contact: <sip:1011@xxx.xxx.13.49:35700>. > > > > > > > Call-ID: B6E73E25-87F9-40BB-A895-561E4ADE8AC8(a)10.18.1.70. > > > > > > > CSeq: 26232 INVITE. > > > > > > > Proxy-Authorization: Digest > > > > > > > username="1011",realm="ser.xxx.net.tw",nonce="42196d5483841de0d6b79c7dfb4156e2c932e4cb",response="7004beab12b3ac6874c5cd86e5659930",uri="sip:1033@ser.xxx.net.tw". > > > > > > > Max-Forwards: 16. > > > > > > > Content-Type: application/sdp. > > > > > > > User-Agent: X-PRO build 1082. > > > > > > > Content-Length: 264. > > > > > > > . > > > > > > > v=0. > > > > > > > o=1011 46238507 46238507 IN IP4 10.18.1.70. > > > > > > > s=X-PRO. > > > > > > > c=IN IP4 10.18.1.70. > > > > > > > t=0 0. > > > > > > > m=audio 8000 RTP/AVP 8 3 98 97 101. > > > > > > > a=rtpmap:8 pcma/8000. > > > > > > > a=rtpmap:3 gsm/8000. > > > > > > > a=rtpmap:98 iLBC/8000. > > > > > > > a=rtpmap:97 speex/8000. > > > > > > > a=rtpmap:101 telephone-event/8000. > > > > > > > a=fmtp:101 0-15. > > > > > > > > > > > > > > # > > > > > > > U xxx.xxx.13.49:33536 -> xxx.xxx.190.248:5060 > > > > > > > SIP/2.0 486 Busy Here. > > > > > > > Via: SIP/2.0/UDP xxx.xxx.190.248;branch=z9hG4bKa7ac.b59d9107.0. > > > > > > > Via: SIP/2.0/UDP > > > > > > > 10.18.1.70:5060;received=xxx.xxx.13.49;rport=35700;branch=z9hG4bK1BDE5BF2809C4223A13FC5B1A9AC21CC. > > > > > > > Record-Route: <sip:xxx.xxx.190.248;ftag=3281269171;lr=on>. > > > > > > > Call-ID: B6E73E25-87F9-40BB-A895-561E4ADE8AC8(a)10.18.1.70. > > > > > > > CSeq: 26232 INVITE. > > > > > > > From: 1011 <sip:1011@ser.xxx.net.tw>;tag=3281269171. > > > > > > > To: <sip:1033@ser.xxx.net.tw>;tag=A8VnZRzqB2nyNgYQ. > > > > > > > Contact: <sip:1033@10.18.1.102:1718>. > > > > > > > Content-Length: 0. > > > > > > > . > > > > > > > > > > > > > > # > > > > > > > U xxx.xxx.190.248:5060 -> xxx.xxx.13.49:33536 > > > > > > > ACK sip:1033@xxx.xxx.13.49:33536 SIP/2.0. > > > > > > > Via: SIP/2.0/UDP xxx.xxx.190.248;branch=z9hG4bKa7ac.b59d9107.0. > > > > > > > From: 1011 <sip:1011@ser.xxx.net.tw>;tag=3281269171. > > > > > > > Call-ID: B6E73E25-87F9-40BB-A895-561E4ADE8AC8(a)10.18.1.70. > > > > > > > To: <sip:1033@ser.xxx.net.tw>;tag=A8VnZRzqB2nyNgYQ. > > > > > > > CSeq: 26232 ACK. > > > > > > > User-Agent: Sip EXpress router(0.10.99-dev0 (i386/linux)). > > > > > > > Content-Length: 0. > > > > > > > . > > > > > > > > > > > > > > # > > > > > > > U xxx.xxx.13.49:35700 -> xxx.xxx.190.248:5060 > > > > > > > . > > > > > > > > > > > > > > # > > > > > > > U xxx.xxx.13.49:35700 -> xxx.xxx.190.248:5060 > > > > > > > . > > > > > > > > > > > > > > # > > > > > > > U xxx.xxx.190.248:5060 -> xxx.xxx.13.49:35700 > > > > > > > SIP/2.0 408 Request Timeout. > > > > > > > Via: SIP/2.0/UDP > > > > > > > 10.18.1.70:5060;rport=35700;branch=z9hG4bK1BDE5BF2809C4223A13FC5B1A9AC21CC;received=xxx.xxx.13.49. > > > > > > > From: 1011 <sip:1011@ser.xxx.net.tw>;tag=3281269171. > > > > > > > To: <sip:1033@ser.xxx.net.tw>;tag=5f573bbafd260ada15def89f1b1724a2-6b7e. > > > > > > > Call-ID: B6E73E25-87F9-40BB-A895-561E4ADE8AC8(a)10.18.1.70. > > > > > > > CSeq: 26232 INVITE. > > > > > > > Server: Sip EXpress router (0.10.99-dev0 (i386/linux)). > > > > > > > Content-Length: 0. > > > > > > > Warning: 392 xxx.xxx.190.248:5060 "Noisy feedback tells: pid=4203 > > > > > > > req_src_ip=xxx.xxx.13.49 req_src_port=35700 > > > > > > > in_uri=sip:1033@ser.xxx.net.tw out_uri=sip:1033@xxx.xxx.13.49:33536 > > > > > > > via_cnt==0". > > > > > > > . > > > > > > > > > > > > > > # > > > > > > > U xxx.xxx.13.49:35700 -> xxx.xxx.190.248:5060 > > > > > > > ACK sip:1033@ser.xxx.net.tw SIP/2.0. > > > > > > > Via: SIP/2.0/UDP > > > > > > > 10.18.1.70:5060;rport;branch=z9hG4bK1BDE5BF2809C4223A13FC5B1A9AC21CC. > > > > > > > From: 1011 <sip:1011@ser.xxx.net.tw>;tag=3281269171. > > > > > > > To: <sip:1033@ser.xxx.net.tw>;tag=5f573bbafd260ada15def89f1b1724a2-6b7e. > > > > > > > Contact: <sip:1011@10.18.1.70:5060>. > > > > > > > Call-ID: B6E73E25-87F9-40BB-A895-561E4ADE8AC8(a)10.18.1.70. > > > > > > > CSeq: 26232 ACK. > > > > > > > Max-Forwards: 70. > > > > > > > Content-Length: 0. > > > > > > > . > > > > > > > > > > > > > > # > > > > > > > # > > > > > > > U xxx.xxx.190.248:5060 -> xxx.xxx.13.49:35700 > > > > > > > .... > > > > > > > # > > > > > > > U xxx.xxx.190.248:5060 -> xxx.xxx.13.49:33536 > > > > > > > .... > > > > > > > # > > > > > > > U xxx.xxx.13.49:33536 -> xxx.xxx.190.248:5060 > > > > > > > ................ > > > > > > > # > > > > > > > U xxx.xxx.13.49:35700 -> xxx.xxx.190.248:5060 > > > > > > > . > > > > > > > > > > > > > > _______________________________________________ > > > > > > > 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 > > > > > >

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[Serusers] I can't append a head field in the tm module
by szj 15 Mar '05

15 Mar '05

Hi: I want to insert a header field without the script. Instead I insert it in the tm module's t_relay() function. Next is my code. The first time I use the pkg_malloc(), but ser report that: BUG: qm_*: prev. fragm. tail overwritten(c0c0c000, abcdefed) so I changed the pkg_malloc to the shm_malloc(); this time the error happend again in the different place. typedef int(*local_fp)(struct sip_msg *msg, str *txt, int flags); str *str1; char *tmp = "P-Security-Nobody: this is a test header for \r\n"; local_fp append_to_reply = NULL; str1 = (str*)shm_malloc(sizeof(str)); if(str1) { str1->len = strlen(tmp); str1->s = (char*)shm_malloc(str1->len); strcpy(str1->s, tmp); append_to_reply = (local_fp)find_export("append_hf", 1, 0); if(append_to_reply){ DBG("DEBUG: SUNZJ find apppend_to_reply\n"); append_to_reply(p_msg, str1, NULL); } } the next is gdb's output. Program received signal SIGABRT, Aborted. [Switching to Thread -1208486208 (LWP 6439)] 0x00ad77a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 (gdb) (gdb) where #0 0x00ad77a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x00b1b955 in raise () from /lib/tls/libc.so.6 #2 0x00b1d319 in abort () from /lib/tls/libc.so.6 #3 0x0808cb9d in qm_debug_frag (qm=0xb5f7e000, f=0x6) at mem/q_malloc.c:156 #4 0x0808d1ab in qm_malloc (qm=0xb5f7e000, size=4107, file=0x13fb49 "h_table.c", func=0x13fb1e "build_cell", line=231) at mem/q_malloc.c:379 #5 0x0011b654 in build_cell (p_msg=0x814bec0) at shm_mem.h:135 #6 0x0012c23e in t_newtran (p_msg=0x814bec0) at t_lookup.c:988 #7 0x00123f4e in t_relay_to (p_msg=0x814bec0, proxy=0x0, proto=0, replicate=0) at t_funcs.c:241 #8 0x0012f450 in w_t_relay (p_msg=0x814bec0, _foo=0x0, _bar=0x0) at tm.c:893 #9 0x0804fe41 in do_action (a=0x814a4ac, msg=0x814bec0) at action.c:611 #10 0x08050bad in run_actions (a=0x814a4ac, msg=0x814bec0) at action.c:719 #11 0x08074fe8 in eval_elem (e=0x814a4fc, msg=0x814bec0) at route.c:605 #12 0x08075482 in eval_expr (e=0x814a4fc, msg=0x814bec0) at route.c:654 #13 0x080754be in eval_expr (e=0x814a540, msg=0x814bec0) at route.c:670 #14 0x0804edd5 in do_action (a=0x814a684, msg=0x814bec0) at action.c:587 #15 0x08050bad in run_actions (a=0x814a684, msg=0x814bec0) at action.c:719 #16 0x0805019f in do_action (a=0x814a3ac, msg=0x814bec0) at action.c:376 #17 0x08050bad in run_actions (a=0x814a3ac, msg=0x814bec0) at action.c:719 #18 0x0806de90 in receive_msg ( buf=0x80db480 "INVITE sip:862222@sip.suntest.com SIP/2.0\r\nVia: SIP/2.0/UDP 192.168.0.19:5500;branch=z9hG4bK958088909\r\nFrom: <sip:861111@sip.suntest.com>;tag=1137899792;tag=1613933050\r\nTo: <sip:862222@sip.suntest.com"..., len=1018, rcv_info=0xbffe1590) at receive.c:162 #19 0x080858dc in udp_rcv_loop () at udp_server.c:458 #20 0x0805d77f in main_loop () at main.c:1032 #21 0x0805ece5 in main (argc=2, argv=0xbffe1754) at main.c:1569 Anyone who tell me what the problem is will be appreciated. Thanks for your kind help Best Regards. Sun Zongjun

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Re: [Serusers] how can I look up the username in the location table?
by Klaus Darilion 14 Mar '05

14 Mar '05

Hi Dana! Also take a look at the following paramaters and fucntions from the registrar module: desc_time_order (integer) append_branches (integer) registered(domain) regards, Klaus PS: Please always send emails to the mailing list Dana Olson wrote: > Thanks for your reply. I wasn't sure about the URI and the To/From > thing. I looked at the packet and when you register, the username is > in both To and From. I guess I mistook the URI for that or something. > Thanks for your clarification on that. > > I'm trying to limit the number of people using each username to a > single person at a time. I read the mailing list of people trying to > do this before me, and everyone, including Jiri, didn't understand why > someone would want this. > > Well, I have a bunch inbound agents logging into my SER proxy with > softphones, receiving calls from an AudioCodes Mediant 2000 (T1 <-> > SIP) that is getting calls from an Avaya PBX. These inbound calls have > to go to whoever is logged in as that agent name, and only to them. We > can't have multiple people logged into the same name, because each > name corresponds to a certain phone number on the Avaya PBX. I hope > this makes sense. It's not really a normal SIP application in this > sense, which is why I need to remove the ability of a normal SIP > application. > > I'm using the latest stable release of SER, and can't be putting > anything less into production. I know that the next release has the > max_contacts option, but until it goes stable, I don't have a green > light to use it. > > I'll look into the avpops module, as I haven't really heard of it until now. > > -- > Dana > > > > On Fri, 11 Mar 2005 09:00:36 +0100, Klaus Darilion > <klaus.mailinglists(a)pernau.at> wrote: > >>Hi! >> >>Lookup takes the request URI und looks up the location table. In >>REGISTER messages, the address-of-record is in the To: header. >>Therefore, lookup() will fail always. >> >>Maybe you can overcome this by copying the To: URI into an AVP and then >>copying this AVP into the request URI. (avpops module). >> >>Btw: Why do you need this feature? >> >>regards, >>klaus >> >>Dana Olson wrote: >> >>>I want to look up the username in the location table during the >>>method==register if block. >>> >>>The reason I wish to do this is to see if there is another location >>>currently registered in the database for this particular username. How >>>can I do this? Do I have to use exec_msg() or exec_dset(), or does >>>lookup() work? >>> >>>I've tried lookup("location") but it failed to work, it seems. >>> >>>Can anyone help me out? Here is the block of code that I'm trying to use: >>> >>> if (method=="REGISTER") { >>> if (!search("^Expires: 0")) { >>> if (lookup("location")) { >>> sl_send_reply("200","Another registration."); >>> } else { >>> sl_send_reply("200", "First registration."); >>> } >>> } else { >>> sl_send_reply("200", "Unregistration."); >>> } >>> save_noreply("location"); >>> break; >>> } >>> >>>Note that it is sending the proper replies for "First registrations" >>>and "Unregistrations," but not for "Another registrations." >>> >>>Thanks in advance, >>> >>>Dana >>> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >> >> > >

2 1

[Serusers] udp_send failed
by Tina 14 Mar '05

14 Mar '05

Hello, I started SER performance test with sipp traffic generator on FreeBSD 5.3. After starting 300 calls per second I got the following error: Mar 14 16:35:09 devel ser[1506]: msg_send: ERROR: udp_send failed Mar 14 16:35:09 devel ser[1506]: ERROR: udp_send: sendto(sock,0x2a258ae0,568,0,0x286be26c,16): No buffer space available(55) SER send "503" the service not available. I don't use mediaproxy/rtpproxy to exclude problems there. Could someone help me with this? Thank you, Tina __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] RE: SER and radius client not talking
by AJ Grinnell 14 Mar '05

14 Mar '05

I think I have my problem solved. I am using a Cisco CSS11800 to load balance to two SER servers. This seems to cause issues with digest authentication. Anyone have experience with load balancing SER and using digest auth at the same time?

1 0

[Serusers] Please don't click Reply to post a new message
by Mohammad Khan 14 Mar '05

14 Mar '05

Why people prefer to click "Reply" button and change the subject line before submitting a new post in this mailing list. Thanks, Mohammad

1 0

[Serusers] SER and radius client not talking
by AJ Grinnell 14 Mar '05

14 Mar '05

I have had SER working with radius authentication on a previous server with no problems. Now that I am trying to build up a few more servers, I can not get SER to talk to radiusclient. I can run radiusclient successfully from the command line, but the last things I see when I try to authenticate against SER is: 0(17829) qm_free(0x810aa80, 0x8167bc0), called from sterman.c: radius_authorize_sterman(101) 0(17829) qm_free: freeing frag. 0x8167ba8 alloc'ed from sterman.c: radius_authorize_sterman(87) Any ideas? Im guessing it might be a library problem, but nothing is standing out to me.

1 0

RE: [Serusers] SER+Radius Accounting Not Working
by Lucas Aimaretto 14 Mar '05

14 Mar '05

Ray, I solved it but using B2BUA from vovida.org I'm not doing accounting with ser itself. Some say it introduces some delay in between the calls. At least, it do works for me. B2BUA does manage properly the ACCT-RADIUS messages. Regards, Lucas > -----Mensaje original----- > De: Raymond Chen [mailto:rchen@broadz.com] > Enviado el: Lunes, 14 de Marzo de 2005 03:21 p.m. > Para: 'Lucas Aimaretto'; 'Bruno Lopes F. Cabral'; serusers(a)lists.iptel.org > Asunto: RE: [Serusers] SER+Radius Accounting Not Working > > > Guys, > > I am having the same problem as you described. Did you get it solved? > > Ray > > -----Original Message----- > From: serusers-bounces(a)lists.iptel.org > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Lucas Aimaretto > Sent: Friday, December 31, 2004 6:00 AM > To: 'Bruno Lopes F. Cabral'; serusers(a)lists.iptel.org > Subject: RE: [Serusers] SER+Radius Accounting Not Working > > Bruno, > > Thanx for answering ... > > > can you please post me a > > grep Bye /etc/radiusclient/dictionary > > you're using? ... > > [root@radius root]# grep Bye /etc/radiusclient/dictionary > VALUE Sip-Method Bye 4 > VALUE Sip-Response-Code Bye 4 > VALUE Sip-Method Bye 4 > VALUE Sip-Response-Code Bye 4 > > > I managed to get double Starts and single > > Stops, but not much more. > > That's what I want to get (at least) ... but it makes no > sense to me. I receive 2 Starts (but the first corresponds to > an INVITE message and the 2nd to a BYE message :-s ). And > when any of the users hang up, I receive nothing. > > > Cheers > > Regards, > > Lucas > > > P.S. BTW, are you using 0.8.14 or other SER version? > > version: 0.8.14 (i386/linux) > flags: STATS:Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, > DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, > FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, > MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE > 1024, BUF_SIZE 65535 > @(#) $Id: main.c,v 1.168.4.3 2004/06/28 15:41:21 andrei Exp $ > main.c compiled on 16:02:31 Nov 10 2004 with gcc 2.96 > > > > > Lucas Aimaretto wrote: > > > Hi list. I have this problem. There's something that's > not working, > > > because, when ever I set up a call, I receive both messages > > INVITE and > > > BYE instantly. I think the correct way would it be to send Start > > > messages (Acct-Status-Type = Start) upon INVITEs and Stop > messages > > > (Acct-Status-Type = Stop) upon BYEs. But this is not > > happening. Look > > > ... > > > > > > radrecv: Accounting Request from host c0a801fd code=4, id=205, > > > length=270 > > > Acct-Status-Type = Start > > > Service-Type = Sip-Session > > > Sip-Response-Code = 200 > > > Sip-Method = Invite > > > User-Name = "1992001(a)192.168.1.253" > > > Calling-Station-Id = > "sip:1992001@192.168.1.253:5060;user=phone" > > > Called-Station-Id = > "sip:1992005@192.168.1.253:5060;user=phone" > > > Sip-Translated-Request-URI = "sip:1992005@192.168.1.178:11005" > > > Acct-Session-Id = "6dfb6640367b1ab7(a)192.168.1.113" > > > Sip-To-Tag = "54061171" > > > Sip-From-Tag = "39eeac148f846cb9" > > > Sip-CSeq = "1544" > > > NAS-IP-Address = 192.168.1.253 > > > NAS-Port-Id = 5060 > > > Acct-Delay-Time = 0 > > > Sending Accounting Ack of id 205 to c0a801fd (nas linux) > > > > > > radrecv: Accounting Request from host c0a801fd code=4, id=206, > > > length=270 > > > Acct-Status-Type = Start > > > Service-Type = Sip-Session > > > Sip-Response-Code = 200 > > > Sip-Method = Bye > > > User-Name = "1992001(a)192.168.1.253" > > > Calling-Station-Id = > "sip:1992001@192.168.1.253:5060;user=phone" > > > Called-Station-Id = > "sip:1992005@192.168.1.253:5060;user=phone" > > > Sip-Translated-Request-URI = "sip:1992005@192.168.1.178:11005" > > > Acct-Session-Id = "6dfb6640367b1ab7(a)192.168.1.113" > > > Sip-To-Tag = "54061171" > > > Sip-From-Tag = "39eeac148f846cb9" > > > Sip-CSeq = "1544" > > > NAS-IP-Address = 192.168.1.253 > > > NAS-Port-Id = 5060 > > > Acct-Delay-Time = 0 > > > Sending Accounting Ack of id 206 to c0a801fd (nas linux) > > > > > > Do you see? ...Both outputs show an Acct-Status-Type = Start, but > > > first output refers to an INVITE and second output refers to BYE > > > message. And the strange thing is that both messages arrive > > instantly, > > > as soon as the peer answers the phone. Why??? I don't > think this is > > > correct, isn't it? > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers > > > > -- > > No virus found in this incoming message. > > Checked by AVG Anti-Virus. > > Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: > 28/12/2004 > > > > > > -- > No virus found in this outgoing message. > Checked by AVG Anti-Virus. > Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 28/12/2004 > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers > > > -- > No virus found in this incoming message. > Checked by AVG Anti-Virus. > Version: 7.0.308 / Virus Database: 266.7.1 - Release Date: 09/03/2005 > > -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.7.1 - Release Date: 09/03/2005

1 0

[Serusers] calls between UA´s behind same NAT using nathelper/rtpproxy
by Rafael J. Risco G.V. 14 Mar '05

14 Mar '05

Hi I would like to know if its possible to comunicate SIP Clients betwen them when they are behind the same nat, I am using Nathelper/RTPProxy... thanks Rafael -- rrgv

1 0

[Serusers] SER+Radius Accounting Not Working
by Lucas Aimaretto 14 Mar '05

14 Mar '05

Hi list. I have this problem. There's something that's not working, because, when ever I set up a call, I receive both messages INVITE and BYE instantly. I think the correct way would it be to send Start messages (Acct-Status-Type = Start) upon INVITEs and Stop messages (Acct-Status-Type = Stop) upon BYEs. But this is not happening. Look ... radrecv: Accounting Request from host c0a801fd code=4, id=205, length=270 Acct-Status-Type = Start Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = Invite User-Name = "1992001(a)192.168.1.253" Calling-Station-Id = "sip:1992001@192.168.1.253:5060;user=phone" Called-Station-Id = "sip:1992005@192.168.1.253:5060;user=phone" Sip-Translated-Request-URI = "sip:1992005@192.168.1.178:11005" Acct-Session-Id = "6dfb6640367b1ab7(a)192.168.1.113" Sip-To-Tag = "54061171" Sip-From-Tag = "39eeac148f846cb9" Sip-CSeq = "1544" NAS-IP-Address = 192.168.1.253 NAS-Port-Id = 5060 Acct-Delay-Time = 0 Sending Accounting Ack of id 205 to c0a801fd (nas linux) radrecv: Accounting Request from host c0a801fd code=4, id=206, length=270 Acct-Status-Type = Start Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = Bye User-Name = "1992001(a)192.168.1.253" Calling-Station-Id = "sip:1992001@192.168.1.253:5060;user=phone" Called-Station-Id = "sip:1992005@192.168.1.253:5060;user=phone" Sip-Translated-Request-URI = "sip:1992005@192.168.1.178:11005" Acct-Session-Id = "6dfb6640367b1ab7(a)192.168.1.113" Sip-To-Tag = "54061171" Sip-From-Tag = "39eeac148f846cb9" Sip-CSeq = "1544" NAS-IP-Address = 192.168.1.253 NAS-Port-Id = 5060 Acct-Delay-Time = 0 Sending Accounting Ack of id 206 to c0a801fd (nas linux) Do you see ? ...Both outputs show an Acct-Status-Type = Start, but first output refers to an INVITE and second output refers to BYE message. And the strange thing is that both messages arrive instantly, as soon as the peer answers the phone. Why ??? I don't think this is correct, isn't it ? Any hint ? here goes my ser.cfg ... [...] # -- RADIUS ACC -- modparam("acc", "radius_config", "/etc/radiusclient.conf") modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) modparam("acc", "log_level", 1) modparam("acc", "service_type", 15) [...] if (loose_route()) { t_relay(); break; }; if (uri==myself) { if (method=="REGISTER") { # Lo siguiente es para auth con RADIUS if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; save("location"); break; }; if (method=="INVITE") { setflag(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; t_relay(); [...] Regards, Lucas -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 28/12/2004

3 3

[Serusers] cpl module with 8.0.14
by Aisling O'Driscoll 14 Mar '05

14 Mar '05

Hi all, Just to confirm the cpl module should be compatible with stable ser version 0.8.14,shouldnt it?? I have installed the cpl module according to the following howto: http://www.kom.tu-darmstadt.de/~jschmitt/HowToSER_CPL.htm However when i try ot load the cpl-c module in ser.cfg,I get the following error. Any ideas??: ERROR: module version mismatch for /usr/lib/ser/modules/cpl-c.so; core: 0.8.14; module: 0.8.14-5 Thanks, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

1 0

[Serusers] Help: mediaproxy error? or domain issue?
by Charles Wang 14 Mar '05

14 Mar '05

Dear ALL: I install my SER on two machines, one named "SER_A" and the other named "SER_B". Two UACs register to each server. One named "UAC_A" the other named "UAC_B". Server Domain Prefix UAC UAC's ID ------------------------------------------------------------------------- SER_A A.com 852 UAC_A 1011 SER_B B.com 886 UAC_B 2022 ------------------------------------------------------------------------- Prefix_Domain table: --------------------------------- domain | prefix ----------------|----------------- A.com | 852 B.com | 886 ----------------|----------------- My senario is below: UAC_A ======> SER_A ====> SER_B ====> UAC_B 1011 || || 2022 (dial a number || || (Ringing) 8862022) || Accept 2022(a)B.com || and forward to UAC_B change 8862022(a)A.com to 2022(a)B.com I install SER+MediaProxy on my systems and using pdt modules to do the prefix convert. I find these error behaviors on my SER_A's log file. And the call from 1011 to 2022. The call seems be process twice. You can find "Prefix To Domain Checking" twice at the beginning. Can anybody give me some tips to solve it? ---------------------- My log file ----------------------------------------------- ser /usr/local/sbin/ser[23458]: SER: Prefix To Domain Checking... ser /usr/local/sbin/ser[23458]: SER: SIP Call On-Net section route(2) ser /usr/local/sbin/ser[23456]: SER: Prefix To Domain Checking... ser /usr/local/sbin/ser[23456]: SER: SIP Call On-Net section route(2) ser proxydispatcher[23541]: command request D24BF98F-101E-43F4-9765-4DBB863A999D(a)192.168.11.5 192.168.11.5:8000:audio xxx.xxx.133.108 A.com local B.com remote X-PRO=20build=201082 info=from:1011@A.com,to:8862022@A.com,fromtag:3667188260,totag: ser /usr/local/sbin/ser[23463]: SER: Prefix To Domain Checking... ser /usr/local/sbin/ser[23463]: SER: SIP Call On-Net section route(2) ser proxydispatcher[23542]: command request D24BF98F-101E-43F4-9765-4DBB863A999D(a)192.168.11.5 192.168.11.5:8000:audio xxx.xxx.133.108 A.com local B.com remote X-PRO=20build=201082 info=from:1011@A.com,to:8862022@A.com,fromtag:3667188260,totag: ser proxydispatcher[23541]: domain A.com doesn't define any mediaproxy. ser proxydispatcher[23541]: will use default mediaproxy for this call. ser mediaproxy[18756]: command request D24BF98F-101E-43F4-9765-4DBB863A999D(a)192.168.11.5 192.168.11.5:8000:audio xxx.xxx.133.108 A.com local B.com remote X-PRO=20build=201082 info=from:1011@A.com,to:8862022@A.com,fromtag:3667188260,totag:,dispatcher ser mediaproxy[18756]: session D24BF98F-101E-43F4-9765-4DBB863A999D(a)192.168.11.5: started. listening on xxx.xxx.190.248:35020 ser mediaproxy[18756]: command execution time: 10.30 ms ser proxydispatcher[23541]: forwarding to mediaproxy on /var/run/mediaproxy.sock: got: 'xxx.xxx.190.248 35020' ser proxydispatcher[23541]: command execution time: 3365.77 ms ser proxydispatcher[23542]: domain A.com doesn't define any mediaproxy. ser proxydispatcher[23542]: will use default mediaproxy for this call. ser mediaproxy[18756]: command request D24BF98F-101E-43F4-9765-4DBB863A999D(a)192.168.11.5 192.168.11.5:8000:audio xxx.xxx.133.108 A.com local B.com remote X-PRO=20build=201082 info=from:1011@A.com,to:8862022@A.com,fromtag:3667188260,totag:,dispatcher ser mediaproxy[18756]: command execution time: 0.39 ms ser proxydispatcher[23542]: forwarding to mediaproxy on /var/run/mediaproxy.sock: got: 'xxx.xxx.190.248 35020' ser proxydispatcher[23542]: command execution time: 4101.53 ms ser /usr/local/sbin/ser[23463]: SER: Failure Route section failure_route(1) ser /usr/local/sbin/ser[23456]: SER: Prefix To Domain Checking... ser /usr/local/sbin/ser[23456]: SER: SIP Call On-Net section route(2) --------------------- My ser.cfg ----------------------------------------------------------- # ----------- global configuration parameters ------------------------ debug=10 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ alias=A.com alias=xxx.xxx.190.248 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo_mode=0666 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/speeddial.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/pdt.so" loadmodule "/usr/local/lib/ser/modules/options.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" # ----------------- setting module-specific parameters --------------- # -- mediaproxy params -- modparam("mediaproxy", "natping_interval", 30) modparam("mediaproxy", "sip_asymmetrics", "/usr/local/etc/ser/sip-asymmetric-clients") modparam("mediaproxy", "rtp_asymmetrics", "/usr/local/etc/ser/rtp-asymmetric-clients") # -- usrloc params -- modparam("usrloc", "db_mode", 1) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "desc_time_order", 1) # -- 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) # -- db_url params -- modparam("acc|auth_db|domain|group|permissions|speeddial|uri_db|usrloc|pdt", "db_url", "mysql://ser:heslo@localhost/ser") # -- use_domain params -- modparam("auth_db|group|registrar|speeddial|uri_db|usrloc", "use_domain", 0) # -- permissions params -- modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") # -- accounting params -- modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 1) modparam("acc", "log_fmt", "cdfimorstup") modparam("acc", "log_level", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "report_cancels", 1) modparam("acc", "report_ack", 0) # -- domain params -- modparam("domain", "db_mode", 1) # ------------- exec parameters modparam("exec", "setvars", 1) modparam("exec", "time_to_kill", 10) # -- registration params -- modparam("registrar", "nat_flag", 2) modparam("registrar", "min_expires", 60) modparam("registrar", "max_expires", 86400) modparam("registrar", "default_expires", 3600) modparam("registrar", "append_branches", 1) # -- avp params -- modparam("avpops", "avp_url", "mysql://ser:heslo@localhost/ser") modparam("avpops", "avp_table", "usr_preferences") #modparam("avpops", "use_domain", "1") modparam("avpops", "uuid_column", "uuid") modparam("avpops", "username_column", "username") modparam("avpops", "domain_column", "domain") modparam("avpops", "attribute_column", "attribute") modparam("avpops", "value_column", "value") modparam("avpops", "type_column", "type") modparam("avpops", "avp_aliases", "voicemail=i:500;calltype=i:700;fwd_no_answer_type=i:701;fwd_busy_type=i:702") # To use more than one tables example #modparam("avpops", "db_scheme", "scheme1:table=subscriber;uuid_column=uuid;value_column=first_name") # -- tm params -- modparam("tm", "fr_timer", 30) modparam("tm", "fr_inv_timer", 30) modparam("tm", "wt_timer", 5) modparam("tm", "fr_inv_timer_avp", "inv_timeout") # -- pdt params -- modparam("pdt", "db_table", "prefix_domain") modparam("pdt", "prefix", "") modparam("pdt", "hsize_2pow", 2) modparam("pdt", "sync_time", 300) modparam("pdt", "clean_time", 600) # -- logging params modparam("xlog", "buf_size", 8192) # -- group params -- modparam("group", "table", "grp") modparam("group", "user_column", "username") modparam("group", "domain_column", "domain") modparam("group", "group_column", "grp") # ------------------------- request routing logic ------------------- # main routing logic 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; }; # ------------------------------------------------------------------------ # NOTIFY Keep-Alive Section # ------------------------------------------------------------------------ if (method=="NOTIFY") { sl_send_reply("200", "OK"); break; }; # ------------------------------------------------------------------------ # NAT Test Section #1 # ------------------------------------------------------------------------ if (method=="REGISTER" && client_nat_test("3")) { fix_contact(); force_rport(); setflag(2); }; # ------------------------------------------------------------------------ # Registration Section # ------------------------------------------------------------------------ if (method=="REGISTER") { # allow all requests from user 700 - the Click2Dial controller if (!isflagset(14)) { if (!is_from_local()) { sl_send_reply("403", "Unknown Domain"); break; }; if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; # To - Use To username and (optionally) domain to check if (is_user_in("To", "demo-disabled")) { sl_send_reply("403", "Your evaluation period has expired"); break; }; # To - Use To username and (optionally) domain to check if (is_user_in("To", "disabled")) { sl_send_reply("403", "Your account has been disabled"); break; }; }; # snom sip phones use this header to start their # keep-alive mechanism for NAT bindings append_to_reply("P-NAT-Refresh: 15\r\n"); if (!save("location")) { sl_reply_error(); }; break; }; # ------------------------------------------------------------------------ # Accounting Section # 1 # ------------------------------------------------------------------------ if ((method=="INVITE" || method=="BYE") && !isflagset(14)) { setflag(1); }; # ------------------------------------------------------------------------ # NAT Tear-Down Section # ------------------------------------------------------------------------ if ((method == "BYE" || method == "CANCEL")) { end_media_session(); }; # ------------------------------------------------------------------------ # Record Route Section # ------------------------------------------------------------------------ if (!method=="REGISTER") { record_route(); }; # ------------------------------------------------------------------------ # Loose Route Section # ------------------------------------------------------------------------ if (loose_route()) { route(2); break; }; # ------------------------------------------------------------------------ # Prefix To Domain Section # ------------------------------------------------------------------------ log(1, "SER: Prefix To Domain Checking...\n"); prefix2domain(); # ------------------------------------------------------------------------ # NAT Test Section #1 # ------------------------------------------------------------------------ if (client_nat_test("3") && !search("^Record-Route:")) { force_rport(); fix_contact(); }; # ------------------------------------------------------------------------ # PSTN Section # ------------------------------------------------------------------------ if (method=="INVITE") { ## Deny PSTN to 0204 and 095 on Tawan if ((uri=~"^sip:0204[0-9]*@") || (uri=~"^sip:095[0-9]*@")) { sl_send_reply("503", "Service Unavailable"); break; }; }; # ------------------------------------------------------------------------ # Alias Routing Section # ------------------------------------------------------------------------ lookup("aliases"); if (!uri==myself) { route(2); break; }; # ------------------------------------------------------------------------ # Load ACL Section # ------------------------------------------------------------------------ if (method=="INVITE" && !isflagset(14)) { xlog("L_ERR", "Time:[%Tf] Method:<%rm> r-uri:<%ru>\n"); xlog("L_ERR", "IP:<%is> From:<%fu> To:<%tu> %ct\n"); if (is_from_local() || is_uri_host_local()) { # Check user from grp table if (is_user_in("From", "int")) { log(1, "SER: a INT user\n"); setflag(29); }; if (is_user_in("From", "free-pstn")) { log(1, "SER: a FREE-PSTN user\n"); setflag(28); }; if (avp_db_load("$from/username", "s:callidblock")) { if (avp_check("s:callidblock", "eq/y/i")) { setflag(25); }; }; }; if (avp_db_load("$ruri/username", "s:anoncallrej")) { if (avp_check("s:anoncallrej", "eq/y/i")) { log(1, "SER: a ANON-CALL-REJ user\n"); setflag(24); }; }; }; # ------------------------------------------------------------------------ # Anonymous Call Rejection Section # 24 # ------------------------------------------------------------------------ if (isflagset(24) && (method=="INVITE") && search("^(f|F)rom:.*(a|A)nonymous")) { route(8); break; }; # ------------------------------------------------------------------------ # 002 International Call Section # 29 # ------------------------------------------------------------------------ if (method=="INVITE" && uri=~"^sip:002[0-9]*@") { if (isflagset(29)) { log(1, "SER: an International Call route(6)\n"); route(6); } else { sl_send_reply("503", "Service Unavailable"); }; break; }; # ------------------------------------------------------------------------ # 0XXXXXXXXX Domestic Call Section # 29 # ------------------------------------------------------------------------ if (method=="INVITE" && uri=~"^sip:0[0-9]{9}@") { if (isflagset(29)) { log(1, "SER: a Domestic Call route(6)\n"); route(6); } else { sl_send_reply("503", "Service Unavailable"); }; break; }; # ------------------------------------------------------------------------ # Blind Call Forwarding Section # ------------------------------------------------------------------------ if (method=="INVITE") { log(1, "SER: BLIND CALL FORWARDING\n"); # here we must store the current (aka original) R-URI because if # we set call forwarding and the forwarded number is busy then we # need to use this original R-URI to determine which voicemail # box we should go to if (isflagset(31)) { avp_write("$ruri", "$voicemail"); }; if (avp_db_load("$ruri/username", "s:callfwd")) { avp_pushto("$ruri", "s:callfwd"); # lookup the call fowarding number to see if it is a served # sip number or a PSTN number # check forwarding number rules log(1, "SER: Check Forwarding Number Rules\n"); route(1); # test for domestic PSTN gateway if (uri=~"^sip:0[1-9]{9}@") { #if (avp_check("$calltype", "eq/dom/i")) { log(1, "SER: Start a domestic PSTN call route(3)\n"); route(3); break; }; # test for international PSTN gateway if (uri=~"^sip:002[1-9]*@") { # if (avp_check("$calltype", "eq/int/i")) { log(1, "SER: Start an international PSTN call route(6)\n"); route(6); break; }; # test for H323 gatekeeper if (uri=~"^sip:888251[0-9]*@") { # if (avp_check("$calltype", "eq/int/i")) { log(1, "SER: Start a H323 call route(9)\n"); route(9); break; }; }; }; # ------------------------------------------------------------------------ # Call Routing Section # ------------------------------------------------------------------------ if (!lookup("location")) { # if flag 31 (ie voicemail) is set and we made it here this means # the user's phone is not registered anywhere. We'll forward to # voicemail after this block because we need to check the call # forward settings first if (isflagset(31)) { # flag 19 means the user has voicemail but is not online # so we need to remember to send to voicemail if call # forwarding is not enabled setflag(19); }; if (method=="INVITE") { if (does_uri_exist()) { # subscriber record found, but they're offline log(1, "SER: Temporarily Unavailable\n"); sl_send_reply("480", "Temporarily Unavailable"); break; }; if (uri=~"^sip:0[0-9]{9}@") { # Send to PSTN Gateway if (isflagset(28) || (is_user_in("From", "int")) || (is_user_in("From", "free-pstn"))) { log(1, "SER: Start a PSTN call\n"); route(3); } else { log(1, "SER: 503 Service Unavailable 2\n"); sl_send_reply("503", "Service Unavailable"); }; break; }; if (uri=~"^sip:888251[0-9]*@") { # Send to H323 Gatekeeper if (isflagset(28) || (is_user_in("From", "int")) || (is_user_in("From", "free-pstn"))) { log(1, "SER: Start a H323 call\n"); route(9); } else { log(1, "SER: 503 Service Unavailable 2\n"); sl_send_reply("503", "Service Unavailable"); }; break; }; sl_send_reply("404", "User Not Found"); break; }; }; # ------------------------------------------------------------------------ # Call Forwarding Section # ------------------------------------------------------------------------ if (method=="INVITE") { # save R-URI in a temp AVP for later use avp_write("$ruri", "i:99"); # only load the forward no answer option if voice mail is not enabled if (!isflagset(31)) { if (avp_db_load("$ruri/username", "s:fwdnoanswer")) { route(1); setflag(27); }; }; if (avp_db_load("$ruri/username", "s:fwdbusy")) { route(1); setflag(26); }; avp_pushto("$ruri", "i:99"); }; if (isflagset(19)) { # send to voicemail route(5); } else { route(2); }; } route[1] { ----skip here------ } route[2] { log(1, "SER: SIP Call On-Net section route(2)\n"); if ((method=="INVITE") && !allow_trusted()) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); break; } else if (!check_from()) { log(1, "Spoofed SIP call attempt"); sl_send_reply("403", "Use From=ID"); break; } else if (!(is_from_local() || is_uri_host_local())) { sl_send_reply("403", "Please register to use our service"); break; }; }; if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; if (isflagset(25)) { replace("^From:(.*)>" , "From: \"Anonymous\" <sip:someone@anonymous.invalid>"); }; if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; t_on_failure("1"); t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[3] { ----skip here------ } route[4] { ----skip here------ } route[5] { ----skip here------ } route[6] { ----skip here------ } route[7] { ----skip here------ } route[8] { ----skip here------ } route[9] { ----skip here------ } onreply_route[1] { # Not all 2xx messages have a content body so here we # make sure our Content-Length > 0 to avoid a parse error if (status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:\ 0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_contact(); }; } failure_route[1] { log(1, "SER: Failure Route section failure_route(1)\n"); # if caller hung up then don't sent to voicemail if (t_check_status("487")) { break; }; if (isflagset(26) && t_check_status("486")) { # forward busy is flag 26 if (avp_pushto("$ruri", "s:fwdbusy")) { log(1, "SER: fork to fwdbusy\n"); avp_delete("s:fwdbusy"); append_branch(); resetflag(26); # test for domestic PSTN gateway if (uri=~"^sip:0[0-9]{9}@") { # if (avp_check("$fwd_busy_type", "eq/dom/i")) { # test for domestic PSTN gateway log(1, "SER: Busy Failure and Jump to route(6)\n"); route(6); } else if (uri=~"^sip:002[1-9][0-9]*@") { # } else if (avp_check("$fwd_busy_type", "eq/int/i")) { # test for international PSTN gateway log(1, "SER: Busy Failure and Jump to route(3)\n"); route(3); } else { # default to sip call log(1, "SER: Busy Failure and Jump to route(2)\n"); route(2); }; break; }; }; # here we can have either voicemail __OR__ forward no answer if (isflagset(27) && t_check_status("408")) { # forward no answer is flag 27 if (avp_db_load("$ruri/username", "s:fwdnoanswer")) { avp_pushto("$ruri", "s:fwdnoanswer"); log(1, "SER: fork to fwdnoanswer\n"); avp_delete("s:fwdnoanswer"); append_branch(); resetflag(27); if (uri=~"^sip:0[0-9]{9}@") { # if (avp_check("$fwd_no_answer_type", "eq/dom/i")) { # test for domestic PSTN gateway log(1, "SER: No Answer Failure and Jump to route(3)\n"); route(3); } else if (uri=~"^sip:002[1-9][0-9]*@") { # } else if (avp_check("$fwd_no_answer_type", "eq/int/i")) { # test for international PSTN gateway log(1, "SER: No Answer Failure and Jump to route(6)\n"); route(6); } else { # default to sip call log(1, "SER: No Answer Failure and Jump to route(2)\n"); route(2); }; break; }; } else if (isflagset(31) && avp_pushto("$ruri", "$voicemail")) { avp_delete("$voicemail"); log(1, "SER: No Answer Failure and Jump to route(4)\n"); route(4); break; }; }

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