- sr-users - kamailio.org

[Serusers] Implementing Codec Translator with Asterisk
by Frank Kostin 03 Aug '05

03 Aug '05

Hi everybody, Looking to implement Codec Translator with Asterisk - loading codec modules API in Asterisk to support transcoding. Has anyone experienced this issue, and does anyone have any suggestions or hint, simple scripts, whatever ? Thanks in advance and kind Regards, Frank --------------------------------- Yahoo! Mail for Mobile Take Yahoo! Mail with you! Check email on your mobile phone.

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[Users] Radius or SQL
by Jayesh Nambiar 03 Aug '05

03 Aug '05

Hi Bogdan, Thanx for the information about using allow_trusted. It works fine. I just wanted to clarify one thing, is using SQL more efficient or Radius is better. I wanted to know performance wise. What would be the overhead in both cases. one more thing, whenever we add a new IP address to the trusted table openser needs to be restarted. Is there any alternative to this. Which type of calls would be affected when we restart openser. please let me know some information about this. Thanx a lot. Jayesh

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[Users] more TCP problems
by Erich Nahum 03 Aug '05

03 Aug '05

Thanks to Bogdan, who pointed out that my UAC was not behaving properly. Having solved that problem in sipp, I now have a new issue. The UAC is opening a TCP connection to the proxy, but the proxy then forwards the INVITE to the UAS over UDP. I'm using the stock openser.cfg file. Any suggestions? Thanks, -Erich -- Erich M. Nahum IBM T.J. Watson Research Center Research Staff Member P.O. Box 704 nahum(a)watson.ibm.com Yorktown Heights NY 10598

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[Serusers] 407 Proxy Authentication Required
by Yiannis Martis 03 Aug '05

03 Aug '05

Hello there, I have installed ser 0.9.3 with vovida stun server and I use Linksys PAP2 UAs. This is what I get with ngrep when I try to make an IP-to-IP call to another UA: Please Help me... U 2005/08/03 13:55:57.380953 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 INVITE sip:2810318308@XXXXXXXXXX.com SIP/2.0. Record-Route: <sip:XX.XX.XX.XX;ftag=7ab32f2bbf2689feo0;lr=on>. Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK9925.01a26607.0. Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK-ae62fa6a;… [View More]rport=5060. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. To: <sip:2810318308@XXXXXXXXXX.com>. Call-ID: b521746f-5119658a(a)192.168.1.2. CSeq: 102 INVITE. Max-Forwards: 16. Contact: <sip:2810300308@XX.XX.XX.XX:5060>. Expires: 240. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 420. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 13486 13486 IN IP4 XX.XX.XX.XX. s=-. c=IN IP4 XX.XX.XX.XX. t=0 0. m=audio 16420 RTP/AVP 18 0 2 4 8 96 97 98 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. # U 2005/08/03 13:55:57.381212 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK9925.01a26607.0. Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK-ae62fa6a;rport=5060. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. To: <sip:2810318308@XXXXXXXXXX.com>;tag=5f52b14eb2ca58a7910364a1cdbf6fd5.a179. Call-ID: b521746f-5119658a(a)192.168.1.2. CSeq: 102 INVITE. Proxy-Authenticate: Digest realm="XXXXXXXXXX.com", nonce="42f0a3e9ea677d46038bdbeccaf8d95c18bce58e". Server: Sip EXpress router (0.9.3 (i386/linux)). Content-Length: 0. Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=4360 req_src_ip=XX.XX.XX.XX req_src_port=5060 in_uri=sip:2810318308@XXXXXXXXXX.com out_uri=sip:2810318308@XXXXXXXXXX.com via_cnt==2". . # U 2005/08/03 13:55:57.381353 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 ACK sip:2810318308@XXXXXXXXXX.com SIP/2.0. Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK9925.01a26607.0. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. Call-ID: b521746f-5119658a(a)192.168.1.2. To: <sip:2810318308@XXXXXXXXXX.com>;tag=5f52b14eb2ca58a7910364a1cdbf6fd5.a179. CSeq: 102 ACK. User-Agent: Sip EXpress router(0.9.3 (i386/linux)). Content-Length: 0. . # U 2005/08/03 13:55:57.664069 XX.XX.XX.XX:5060 -> XX.XX.XX.XX:5060 ACK sip:2810318308@XXXXXXXXXX.com SIP/2.0. Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK-ae62fa6a;rport. From: <sip:2810300308@XXXXXXXXXX.com>;tag=7ab32f2bbf2689feo0. To: <sip:2810318308@XXXXXXXXXX.com>;tag=5f52b14eb2ca58a7910364a1cdbf6fd5.a179. Call-ID: b521746f-5119658a(a)192.168.1.2. CSeq: 102 ACK. Max-Forwards: 70. Proxy-Authorization: Digest username="2810300308",realm="XXXXXXXXXX.com",nonce="42f0a3e8f78926d700140a7c490f06dd965ea748",uri="sip:2810318308@XXXXXXXXXX.com",algorithm=MD5,response="ac73134a0de530d527cfbe4fe1903707". Contact: <sip:2810300308@XX.XX.XX.XX:5060>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ [View Less]

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[Users] ovpops for serial forking and its cures
by Francesco Bottà 03 Aug '05

03 Aug '05

Hi all, I'm playing around serial forking with avpops module. My goal is to have one PSTN number mapped to N voip number; so, when the first voip number is busy, OPENSER catch 486 reply and then try with the second AOR and so on. But here raise the problem: the OPENSER forward the reply upstream with a code 407 Proxy Auth Required and the PSTN GW send his ACK and all stop here. Below ther's a snippet of mi openser.cfg: modparam("avpops", "avp_aliases", "serial_fork=i:665") [..skipping..] #… [View More]

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RE: [Serusers] is SER Dependable for prepaid System ? Anyone using itfor prepaid system?
by Sam Lee 03 Aug '05

03 Aug '05

Jim, Your prepaid system concept is a little different from mine. For mine, the SER is just a middleman. I route all my calls to Cisco CallManager which will then route accordingly. So I just get all my accounting info from the Callmanager CDR. I then run ASP scripts to check the users balances and if it ever finds a low-in-credit user, it will go into the mysql db by odbc and insert the user into a grp (acl is preconfigured to block this grp from accessing the PSTN). For your case, probably … [View More]the only thing u can do is to match 2 things, the dest. Country code and the different local charges for each country. Do comment about my set up. Regards, Sam -----Original Message----- From: Sanjim Sanjim [mailto:sanjim007@gmail.com] Sent: Wednesday, August 03, 2005 5:33 PM To: Sam Lee Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] is SER Dependable for prepaid System ? Anyone using itfor prepaid system? Sure Sam, I am currently developing a script which will check the credit balance of the user from mysql and accordingly rewrite the uri. If the balance is low... it will forward to SEMS announcement. Like that when SER receives a BYE, the Script should check the destination and total mins and calculate the amount and reduce it from the user credits. Currently the difficult part is the destination country code. I have stored all the codes in the mysql prefix table. But each country have sub codes for states. For example UK proper is only .02 cents, but mobile is .40 cents We have to differentiate the code properly. If you have any other suggestions kindly let me know. Thanks, JiM On 8/3/05, Sam Lee <sam.lee(a)super.net.sg> wrote: > I have been working for quite some time on deploying SER as a prepaid > system. Currently it is not done yet as there's still quite a number > of things to be done. > Probably we can share a few ideas down the road. > > Regards, > Sam > > -----Original Message----- > From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] > On Behalf Of Sanjim Sanjim > Sent: Tuesday, August 02, 2005 7:02 PM > To: serusers(a)lists.iptel.org > Subject: [Serusers] is SER Dependable for prepaid System ? Anyone > using itfor prepaid system? > > is SER Dependable for prepaid System ? Anyone using it for prepaid > system? > > I very much like the SER performance in the test environment. Only > drawback(maybe advantage if we provide unlimited plan) ) is stateless > operation. > > But I want to know anyone using it in realtime prepaid business? > > > > Thanks, > JiM > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > [View Less]

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[Serusers] Can I know Call ID of UAC from Proxy server?
by Abhijit A.Mahajani 03 Aug '05

03 Aug '05

Hi, Can I know the Call ID of a UAC / UAS who has been registered on th eproxy server? I need to send the MESSAGE to the Particular "UAC" but I do not know the Call ID ? There is no INVITE nor Call Established.But I need to send IM to that particular UAC. As Far As I Know, Call id is mandatory , Hence I need to know the Call id from the prosy server. How should I proceed? Thanks and Regards. Abhijit

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RE: [Users] Radius or SQL
by Jayesh Nambiar 03 Aug '05

03 Aug '05

hi bogdan, Thanx for the help. I am sorry to ask again but how do i use the "trusted_reload" command in the script. I tried different techniques but i could not get the results. please help!! Thanx again

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[Serusers] SER got error during the start due to MySQL connection 2
by ser＠ipcb.net 03 Aug '05

03 Aug '05

Hi all. I have configured ProxySER for my test clients and connected they via SER, but... I can't configure SER - MySQL interoperability. I've read "SIP.edu Cookbook" by ProxySER afterward I made the changes into ser.cfg and db configuration. After SER startup I've got the next log: Aug 2 19:21:21 fwd ser[11846]: Maxfwd module- initializing Aug 2 19:21:21 fwd ser[11849]: new_connection(): Access denied for user 'serro'@'localhost' (using password: YES) Aug 2 19:21:21 fwd ser[11849]: … [View More]db_init(): Could not create a connection Aug 2 19:21:21 fwd ser[11849]: ERROR: auth_db_bind: unable to connect to the database Aug 2 19:21:21 fwd ser[11849]: init_mod_child(): Error while initializing module auth_db Aug 2 19:21:21 fwd ser[11849]: ERROR: open_uac_fifo: init_child failed Aug 2 19:21:21 fwd ser[11849]: starting fifo server failed I've had the next config: # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ # debug=7 # debug level (cmd line: -dddddddddd) # fork=yes # log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode 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 "//lib/ser/modules/mysql.so" loadmodule "//lib/ser/modules/sl.so" loadmodule "//lib/ser/modules/tm.so" loadmodule "//lib/ser/modules/rr.so" loadmodule "//lib/ser/modules/maxfwd.so" loadmodule "//lib/ser/modules/usrloc.so" loadmodule "//lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "//lib/ser/modules/auth.so" loadmodule "//lib/ser/modules/auth_db.so" loadmodule "//lib/ser/modules/acc.so" loadmodule "//lib/ser/modules/exec.so" # loadmodule "//lib/ser/modules/group.so" # loadmodule "//lib/ser/modules/msilo.so" loadmodule "//lib/ser/modules/print.so" loadmodule "//lib/ser/modules/textops.so" loadmodule "//lib/ser/modules/uri.so" # loadmodule "//lib/ser/modules/uri_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", "mysql://ser:mypass@fwd.serout.com/myserdb") modparam("auth", "secret", "myphrase") # -- auth params -- # 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 ------------------- skipped... Where I can dig for solving this trouble? Thank you in advance. Alexander Naumov. [View Less]

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[Serusers] Ask Billing Server for SER
by Asep Saifudin 03 Aug '05

03 Aug '05

Hi All, Does anyone knows Billing Server for SIP based on SER ? whether for prepaid or postpaid system. I would be glad if there's Opensource SER Billing system but Licensed Billing also ok for me. Thanks & Regards, Asep

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RE: [Serusers] lookup 2 uri
by Samuel Osorio Calvo 03 Aug '05

03 Aug '05

Just one remark: only one of the two URI will be translated to the contact. I'll try to explain... imagine the incoming Req-URI is 1001(a)test.com. When it passes the first lookup, lookup("aliases"); SER will retrieve the two aliases and put one of them as the new req-uri and the other as branch. Req-URI: 1002(a)test.com branch: 1003(a)test.com Doing the second lookup, lookup("location"); will put the contact of 1002(a)test.com as Req-URI (if more than one … [View More]contact is present, the subsequent bindings will be added as branches) and the 1003(a)test.com will be left as branch untouched. Req-URI: 1002(a)10.1.1.102 branch: 1003(a)test.com This happens because lookup() only takes as input the req-URI. The result is that SER makes a DNS query for test.com, to send the request to 1003(a)test.com, instead of the end-point. If the DNS is properly configured, test.com should point to SER itself, making SER sending the message to itself via the network. Depending on the config file, it can work but you should check the logs to see what is exactly happening. User 1002 will get the message without any problem. NOTE: depending on the order in the database, 1003 and 1002 can exchange roles in upper explanation. As SER is now coded, above can not be changed (as far as I know).... Hope it helps, Samuel. Unclassified. >>> "Zeus Ng" <zeus.ng(a)isquare.com.au> 08/03/05 01:51AM >>> You would need two steps here, looup("aliases"), then lookup("location"). Zeus > -----Original Message----- > From: serusers-bounces(a)lists.iptel.org > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Richard Z > Sent: Wednesday, 3 August 2005 3:33 AM > To: Atle Samuelsen > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] lookup 2 uri > > > Hi Atle, > > Alias doesn't replace uri with its real location. I'd like to use > lookup("location") to find out its real location. For > example, if 102(a)test.com is at 102(a)10.1.1.102 and > 103(a)test.com is at 103(a)10.1.1.103. I'd like to replace all > uris with 102(a)10.1.1.102 and 103(a)10.1.1.103 in a single ser > transaction logic. Would it be possible? > > Thanks, > Richard > > > On 8/1/05, Atle Samuelsen <clona(a)camaro.no> wrote: > > > > Hi, > > > > you could do something like : > > > > serctl alias add 101 sip:102@test.com > > serctl alias add 101 sip:103@test.com > > > > This way the invite would be forked to both 102 and 103, > > > > just remember to have both lookup("aliases"); and > save("aliases"); in > > your config at the right spot :-) > > > > -Atle > > > > > > > > * Richard Z <rzheng(a)gmail.com> [050802 04:34]: > > > Hi, > > > > > > I am trying to implement a feature to ring two ip phones > > > simultaneously. For example, when I call 101(a)test.com, it > rings both > > > 101(a)test.com and 102(a)test.com. In this case, the ruri is > > > 101(a)test.com. I can use append_branch 102(a)test.com, but > is there a > > > way to lookup the locations of both 101 and 102? > > > > > > Thanks, > > > Richard > > > > > > _______________________________________________ > > > 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 [View Less]

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[Users] Looking for openser's gentoo ebuild
by Cheng Zhang 03 Aug '05

03 Aug '05

Hi, there In INSTALL doc [http://www.openser.org/docs/INSTALL] section 3-C, it seems to me that a gentoo's ebuild is already available in the official portage tree for OpenSER. Unfortunately I couldn't find it in my daily-rsync'ed Gentoo box, nor I can find it in http://packages.gentoo.org. Searching in both users and devel mailing lists and Google search couldn't find anything useful either. Am I missing something obvious here? Could someone give me a hint please? Thanks. -- Cheng

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[Serusers] How to rewrite source IP -- SIP packet forwarding.
by Ray Van Dolson 03 Aug '05

03 Aug '05

I'm using nathelper with SER. Here's a quickie diagram of my setup: [SIP ATA] <-------------> [SER] <------------------> [Asterisk] 10.0.201.5 eth1 - 10.0.201.7 eth0 - <public_ip1> eth0 - <public_ip0> I'm using the basic nathelper example configuration, except for one modification. I removed the call to t_relay and replaced it with: t_relay_to("<public_ip1>", "5060"); Basically I want a SIP packet to come in to … [View More]

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[Serusers] How to catch re-invite and signal rtpproxy ??
by jimmy 03 Aug '05

03 Aug '05

Hi all: I am using ser 0.8.14 with rtpproxy and kphone 0.4.1.1, When kphone holds the call, then unholds later, sometimes kphone will change the rtp port in SDP differ from previously used. Therefore, the rtp packet can't be forward correctly via rtpproxy if ser doesn't signal rtpproxy to change the forwarding port mapping I have asked before about this problem, and Zeus Ng have give me some information that the key factor is to catch the re-invite, but i don't know how … [View More]

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[Serusers] Re: Problem on pthread and dtmf
by Nicky 03 Aug '05

03 Aug '05

Please help..... ----- Original Message ----- From: Nicky To: sems(a)lists.iptel.org ; serusers(a)lists.iptel.org Sent: Tuesday, July 05, 2005 6:56 PM Subject: Problem on pthread and dtmf Dear all, Anyone encountered the problem below,please help. Thank you. ----- Original Message ----- From: Nicky To: sems(a)lists.iptel.org ; serusers(a)lists.iptel.org Sent: Thursday, June 16, 2005 3:51 PM Subject: Problem on pthread and dtmf Dear all, 1.… [View More]

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RE: [Serusers] Start and fnish call
by Sam Lee 03 Aug '05

03 Aug '05

Method==Bye ? Try a ngrep during a call transaction to see what are the header sent. -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Anderson Alves de Albuquerque Sent: Wednesday, August 03, 2005 12:20 AM To: serusers(a)lists.iptel.org Subject: [Serusers] Start and fnish call With SER how can I know when the client SIP finish a call? I need that SER can start a script when the client SIP finish a call. … [View More]

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RE: [Serusers] serctl add user error
by Sam Lee 03 Aug '05

03 Aug '05

Try adding an alias in it. Eg. Alias=192.168.2.2 -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of han yibing Sent: Wednesday, August 03, 2005 10:08 AM To: serusers(a)lists.iptel.org Subject: [Serusers] serctl add user error hi serctl add 8800 8800 8800(a)192.168.2.2 using above command to add user ,but get following error ERROR:400;check if you use aliases in SER,in fact ,i don't use alisses in ser.cfg file. please help … [View More]

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[Serusers] serctl add user error
by han yibing 03 Aug '05

03 Aug '05

hi serctl add 8800 8800 8800(a)192.168.2.2 using above command to add user ,but get following error ERROR:400;check if you use aliases in SER,in fact ,i don't use alisses in ser.cfg file. please help me. thanks ___________________________________________________________ 雅虎邮箱超强增值服务－2G超大空间、pop3收信、无限量邮件提醒 http://cn.mail.yahoo.com/mail_alert/promo1.html

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[Serusers] rtpproxy problem.
by Ray Van Dolson 03 Aug '05

03 Aug '05

I'm running into a problem with SER & rtpproxy trying to get a NAT scenario to work correctly. I'm still learning about SER, so bear with me... :) Here is my network layout in a nutshell: SIP ATA - 10.0.201.196 (private only, no NAT to internet) SER - eth0: 208.208.53.5/24 - eth1: 10.0.201.5/24 Asterisk - eth0: 65.234.34.5/24 *Everything* is configured to proxy through SER to Asterisk currently. My configuration file is here: http://webdev.digitalpath.net/~rayvd/voice/ser.cfg … [View More]

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[Serusers] failure_route problem
by davor jovanovic 03 Aug '05

03 Aug '05

Hi all, I have a problem with sending a call to PSTN gateway when user is offline. I always get "404 Not found" what is normal because user is not in location database, but I don't understand why failure_route[1] doesn't work. Below is my ser.cfg so if someone could help. I'm using ser 0.9.3. Any help would be appreciated. Thanks __________________________________ route{ # initial sanity checks -- messages with # max_forwards==0, or excessively … [View More]

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[Serusers] lookup 2 uri
by Richard Z 02 Aug '05

02 Aug '05

Hi, I am trying to implement a feature to ring two ip phones simultaneously. For example, when I call 101(a)test.com, it rings both 101(a)test.com and 102(a)test.com. In this case, the ruri is 101(a)test.com. I can use append_branch 102(a)test.com, but is there a way to lookup the locations of both 101 and 102? Thanks, Richard

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[Users] avp questions
by Klaus Darilion 02 Aug '05

02 Aug '05

Hi! I'm playing with AVPs stored in a database and wonder about the following example found in the doc: avp_db_load("$from","i:678"); This loads all AVPs with attribute==678, regardless of the from URI. Is there a usecase for this? If not, why not use the specifier /uri as default if no specifier is selected. Also, in avp_db_load the source may be an avp-alias. Thus, how do you differ between the From: header URI "$from" and the avp-alias "$from"? regards, Klaus PS: Maybe I'm a little … [View More]

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[Users] about the next release
by Daniel-Constantin Mierla 02 Aug '05

02 Aug '05

The features we proposed (and many other) for next release are almost done. As you can see in the roadmap (http://openser.org/roadmap.php), mandatory items were achieved, a few optional are still to be worked on and other that would be good to have in next release popped up on the mailing lists. The major item which is still pending is the authentication via uac module. Another thing that should be there is the clean up of serctl + support for the other database types. Other topics … [View More]

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[Serusers] Start and fnish call
by Anderson Alves de Albuquerque 02 Aug '05

02 Aug '05

With SER how can I know when the client SIP finish a call? I need that SER can start a script when the client SIP finish a call.

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[Serusers] URI rewrite utils
by Pavol Segeč 02 Aug '05

02 Aug '05

Hi, Ser has a few cmd which allows rewrite request uri or some part of it. Does SER have some cmd which allow rewrite caller uri or some part of it? For example, my uri is 1767(a)mydomain.com but when I'm calling outside of my VoIP world to PSTN (for example through Cisco call manager) I would like to identify myself by E.164 address format, i.e. +4214151341767(a)mydomain.com. It should be helpful for me change my URI. Thanks, palo

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[Serusers] MARK RODDY/LAKE is out of the office.
by mark.roddy＠lakecommunications.com 02 Aug '05

02 Aug '05

I will be out of the office starting 08/02/2005 and will not return until 08/18/2005. I will not be checking my mail. For Daidalos related issues please contact Stephen Butler. *********************************************************************************** This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify postmaster(a)… [View More]

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[Serusers] is SER Dependable for prepaid System ? Anyone using it for prepaid system?
by Sanjim Sanjim 02 Aug '05

02 Aug '05

is SER Dependable for prepaid System ? Anyone using it for prepaid system? I very much like the SER performance in the test environment. Only drawback(maybe advantage if we provide unlimited plan) ) is stateless operation. But I want to know anyone using it in realtime prepaid business? Thanks, JiM

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[Users] via header not added when transport=TCP?
by Erich Nahum 02 Aug '05

02 Aug '05

Howdy, I'm playing with OPENSER 0.9.5 with sipp as the workload generator on a bunch of linux boxes. When the transport is UDP, openser adds a via header like it is supposed to as a proxy. But when I use TCP no via header is added. This is a UAC->proxy->UAS scenario. Here's the header as seen at the UAS: | UDP message received [633] bytes : | | INVITE sip:service@10.0.1.41:5060 SIP/2.0 | Record-Route: <sip:10.0.1.42;ftag=1;lr=on> | Via: SIP/2.0/UDP 10.0.1.42;branch=… [View More]

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[Users] lcr from_gw in reply_route
by Klaus Darilion 02 Aug '05

02 Aug '05

Hi! Currently it is not possible to use the from_gw function in a reply_route. I'm no code expert, but to me it looks like it would be safe also to use it in a reply_route. Can please someone review this? regards, klaus

3 4

[Serusers] Please help me with NAT...
by Yiannis Martis 02 Aug '05

02 Aug '05

1 0

[Users] current calls
by Ozan Blotter 02 Aug '05

02 Aug '05

dear list, i'm still working on the billing system... there'r a few things left like active calls and active registers... from openser database, in "location" table i can read current registered users, and it's easy to make a PHP from there to show "active registers" from a web page. but in addition to it, i wan to see active calls, is there a specific way to do it from ser.cfg or database ? debug=3 fork=yes log_stderror=no listen=212.XXX.104.XXX port=5060 children=4 alias=212.XXX.104.XXX … [View More]dns=no rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/acc.so" modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", 1) modparam("domain", "db_mode", 1) modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 180) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/openser/sip-asymmetric-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/openser/rtp-asymmetric-clients") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) modparam("registrar", "default_expires", 60) modparam("registrar", "min_expires", 30) modparam("registrar", "nat_flag", 6) modparam("registrar", "use_domain", 1) modparam("rr", "enable_full_lr", 1) modparam("acc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("acc", "failed_transactions", 1) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section and Acc section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { record_route_preset("212.XXX.104.XXX:5060;nat=yes"); } else if (method!="REGISTER") { if!(uri=~"^sip:833[0-9]*@") { record_route(); setflag(1); } }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; }; if (uri==myself) { if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (uri=~"^sip:0[0-9]*@*") { rewritehost("195.XXX.122.XXX"); # this is our PSTN gateway route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; if (method=="CANCEL") { route(1); break; }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ +0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } [View Less]

3 2

[Users] Problems with digest authentication
by aimable 02 Aug '05

02 Aug '05

I have been using various versions of SER from last year without any problem but recently I made a new installation of OpenSER 0.9.5. Since then I am having problems with digest authentication from some of the phones. I have a bunch of 186 ATAs and Cisco 7940 phones but they cannot register to the server, while all the soft phones can register successfully. The server says Credentials with given realm not found. I tried to change the realm to localhost and to the the IP address of the server, … [View More]

2 3

[Serusers] Xlite not working with SER+SEMS
by yu wang 02 Aug '05

02 Aug '05

Hi, I've been working on conference with Xlite but it is not work. The problem is SEMS starts sending media (default wav) back to Xlite once SEMS receives the INVITE/SDP from Xlite. It seems a bug on Xlite not listening on the media port after sending the INVITE thus it replys with ICMP port unreachable. But I don't see anyone complain this on this list. Is something wrong with my configuration? Please please help me.... Thanks, Pat

3 3

Re: [Serusers] URI rewrite utils
by Klaus Darilion 02 Aug '05

02 Aug '05

Pavol Segeč wrote: > Hi, > > But as I see in rfc3261 such header as Remote-Party-ID is not there, or I'm > wrong? What this headre means? In which rfc it is described? I suppose that It never reached the RFC state, it was only a draft, but is supported by Cisco, Asterisk, ser, Inalp, ... http://www.ietf.org/proceedings/02mar/I-D/draft-ietf-sip-privacy-04.txt > for wished functionality should we need rewrite From and Contact headers Do not rewrite From: as it is forbidden (… [View More]

1 0

[Serusers] Radius Authentication
by Ryan Pagquil 02 Aug '05

02 Aug '05

Guys, I'm testing SER to authenticate with radius. But when I start authenticating I see this on the radius log: rad_recv: Access-Request packet from host 127.0.0.1:1131, id=222, length=262 User-Name = "rpagquil@server4all" Digest-Attributes = "\n\nrpagquil" Digest-Attributes = "\001\014server4all" Digest-Attributes = "\002*42ef48a123c4e75c2d998852eaa5d4fb14bc9917" Digest-Attributes = "\004\020sip:server4all" Digest-Attributes = "\003\… [View More]nREGISTER" Digest-Response = "1df283adcf333605c0007d8a86a2e332" Service-Type = Sip-Session Sip-URI-User = "rpagquil" Cisco-AVPair = "call-id=CE373D63037311DABFB500E04CAB4AB4@server4all" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 modcall: entering group authorize for request 150 modcall[authorize]: module "preprocess" returns ok for request 150 modcall[authorize]: module "chap" returns noop for request 150 modcall[authorize]: module "eap" returns noop for request 150 rlm_digest: Converting Digest-Attributes to something sane... Digest-User-Name = "rpagquil" Digest-Realm = "server4all" Digest-Nonce = "42ef48a123c4e75c2d998852eaa5d4fb14bc9917" Digest-URI = "sip:server4all" Digest-Method = "REGISTER" rlm_digest: Adding Auth-Type = DIGEST modcall[authorize]: module "digest" returns ok for request 150 rlm_realm: Looking up realm "server4all" for User-Name = "rpagquil@server4all" rlm_realm: No such realm "server4all" modcall[authorize]: module "suffix" returns noop for request 150 users: Matched rpagquil@server4all at 138 modcall[authorize]: module "files" returns ok for request 150 modcall[authorize]: module "mschap" returns noop for request 150 modcall: group authorize returns ok for request 150 rad_check_password: Found Auth-Type Digest auth: type "Digest" modcall: entering group Auth-Type for request 150 A1 = rpagquil:server4all:test A2 = REGISTER:sip:server4all KD = 94b43b69398cb3ca2eef355e9875d36f:42ef48a123c4e75c2d998852eaa5d4fb14bc9917:33f62c1688cd77a13c84b07b3877bb1c *rlm_digest: FAILED authentication modcall[authenticate]: module "digest" returns reject for request 150 modcall: group Auth-Type returns reject for request 150 auth: Failed to validate the user. Login incorrect: [rpagquil@server4all/<no User-Password attribute>] (from client me2 port 5060)* It says that there is no User-Password attribute contained in my authentication request. What could be the problem? Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com [View Less]

2 1

[Users] How to rewrite source IP -- SIP packet forwarding.
by Ray Van Dolson 02 Aug '05

02 Aug '05

I'm using nathelper with OpenSER 0.10.x. Here's a quickie diagram of my setup: [SIP ATA] <-------------> [OpenSER] <--------------> [Asterisk] 10.0.201.5 eth1 - 10.0.201.7 eth0 - <public_ip1> eth0 - <public_ip0> I'm using the basic nathelper example configuration, except for one modification. I removed the call to t_relay and replaced it with: t_relay_to_udp("<public_ip1>", "5060"); Basically I want a SIP packet … [View More]

2 1

[Serusers] SER <==> SEMS fifo permisions problem
by Pavol Segeč 02 Aug '05

02 Aug '05

Hi All ! I'm implementing SER, SEMS server, I've configured fifo as communication method between them (folowing install manual). In ser.cfg I've enabled conference part in INVITE message route handling as follows : if (uri=~"sip:100.*@") { if (!t_newtran()){ sl_send_reply("500","Could not create transaction"); break; }; t_reply("100","Trying - just wait a minute !"); # assumes that Sems configuration parameter 'socket_name=' … [View More]

2 4

[Serusers] can not open dictionary from freeradiusd!
by zhu lizhong 02 Aug '05

02 Aug '05

hello, guys: I installed freeradiusd and SER. now, i want to connect the two together. the problme is that when i include dictionary.ser in raddb/dictionary, and run radiusd -X, then the messages came out as follow: ------------------------------------- read_config_files: reading dictionary Errors reading dictionary: dict_init: /usr/local/etc/raddb/dictionary[14]: Couldn't open dictionary " /usr/local/share/freeradius/dictionary": No such file or directory Errors reading radiusd.conf ---------… [View More]

1 0

[Serusers] Problem with digest authentication
by aimable 02 Aug '05

02 Aug '05

Hi serusers, I have been using various versions of SER from last year without any problem but recently I made a new installation of OpenSER 0.9.5. Since then I am having problems with digest authentication from some of the phones. I have a bunch of 186 ATAs and Cisco 7940 phones but they cannot register to the server, while all the soft phones can register successfully. The server says Credentials with given realm not found. I tried to change the realm to localhost, the IP address, with no luck.… [View More]

2 1

[Serusers] No Audio
by Ashutosh Kumar 02 Aug '05

02 Aug '05

Hi, I am using the following ser.cfg with ser and mediaproxy, but I am getting the audio only in one direction. This does not seem a problem of NAT, as the nathelper is correclt rewriting the sdp messages, and the aor shows all clients bound with public IP. Earlier t_on_reply call in route[1] was causing an error sl_reply_error used: I'm terribly sorry, server error occured (1/SL) when one calls from one phone to another. Earlier I was getting proper bi-directional audio when I was using … [View More]STUN instead of mediaproxy. Is there something wrong with my ser.cfg #-------------------------------------------------- # # $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) # 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 ---------------------------------- loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/acc.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/domain.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_url", "sql://ser:heslo@localhost/ser") modparam("usrloc", "db_mode", 2) modparam("uri", "db_url", "sql://ser:heslo@localhost/ser") modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") modparam("auth_radius", "service_type", 15) modparam("acc", "db_url", "sql://ser:heslo@localhost/ser") modparam("acc", "db_missed_flag", 3) modparam("acc", "db_flag", 1) modparam("acc", "log_fmt", "cdfimorstup") modparam("acc", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") modparam("acc", "service_type", 15) modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 1) modparam("acc", "log_level", 2) #modparam("acc", "radius_missed_flag", 3) modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") # -- rr params -- modparam("rr", "enable_full_lr", 1) modparam("msilo", "registrar", "sip:registrar@mydomainxyz123.com") modparam("msilo", "db_url", "sql://ser:heslo@mydomainxyz123.com/ser") modparam("auth_db", "db_url", "sql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("uri", "subscriber_table", "subscriber") modparam("uri", "uri_table", "uri") modparam("uri", "uri_user_column", "username") modparam("uri", "uri_domain_column", "domain") modparam("uri", "uri_uriuser_column", "uri_user") modparam("uri", "subscriber_user_column", "username") modparam("uri", "subscriber_domain_column", "domain") # !! Nathelper 1 #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 # !! NAthelper 1 ends # ------------------------- 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 too big");break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- # 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 if (method=="INVITE"){ # && client_nat_test("3")) { record_route_preset("mydomainxyz123.com:5060;nat=yes"); route(3); } else if (method!="REGISTER") {record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- # if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- lookup("aliases"); if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) {setflag(6); #use_media_proxy(); }; ; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- # 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 (!method=="BYE") { force_rport(); fix_nated_contact(); }; if (uri!=myself) {route(1);break;}; if (uri==myself) { if (method=="CANCEL") {route(3); break;}; if (method=="REGISTER") {route(2); break;}; if (method=="MESSAGE") {route(4); break;}; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { if ((method=="INVITE" || method=="ACK") && t_newtran() ){ route(3);break; }; sl_send_reply("404", "Not Foundbusip1"); break; } else {setflag(1);t_relay();break;}; # if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- # t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { #end_media_session(); ; }; # sl_send_reply("404", "balleballeUser Not Found"); # sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); #if (!search("^Contact:\ +\*") && client_nat_test("7")) { # setflag(6); #fix_nated_register(); # force_rport(); #}; #if (!radius_www_authorize("")) { #if (!www_authorize("","subscriber")) { # www_challenge("","0"); # break; #}; #if (!check_to()) { # sl_send_reply("401", "Unauthorized"); # break; #}; consume_credentials(); save("location"); m_dump(); acc_rad_request("Some comment"); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- #if (client_nat_test("3")) { # setflag(7); # force_rport(); # fix_nated_contact(); #}; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); acc_db_request("404 missed call", "missed_calls"); acc_rad_request("404 missed call"); setflag(3); break; }; if (method=="CANCEL") { route(1); break; }; #if (!proxy_authorize("","subscriber")) { # proxy_challenge("","0"); # break; #} else if (!check_from()) { # sl_send_reply("403", "Use From=ID"); # break; #}; consume_credentials(); if (isflagset(6) || isflagset(7)) { # use_media_proxy(); ; }; route(1); } route[4] { if (!t_newtran()) { sl_reply_error(); break; }; if (m_store("0")) { t_reply("202", "Accepted for Later Delivery"); break; }; t_reply("503", "Service Unavailable"); break; } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { #if (!search("^Content-Length:\ +0")) { use_media_proxy(); }; ; }; if (client_nat_test("1")) { fix_nated_contact(); }; ; } Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com [View Less]

2 1

[Serusers] Non eixsting File in mediaproxy modparam
by Ashutosh Kumar 02 Aug '05

02 Aug '05

Hi, In my ser.cfg being used with mediaproxy for NAT traversal, there are two lines reuiqred by this package. modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") But these two files do not exist (*anywhere*) on my machine. I suppose that they are a part of the mediaproxy package.Am I missing them or is the package outdated. Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin … [View More]

2 1

[Serusers] Xlite not working with SER+SEMS
by yu wang 02 Aug '05

02 Aug '05

Hi, I've been working on conference with Xlite but it is not work. The problem is SEMS starts sending media (default wav) back to Xlite once SEMS receives the INVITE/SDP from Xlite. It seems a bug on Xlite not listening on the media port after sending the INVITE thus it replys with ICMP port unreachable. But I don't see anyone complain this on this list. Is something wrong with my configuration? Please please help me.... Thanks, Pat

1 0

[Serusers] serctll add error
by chris 02 Aug '05

02 Aug '05

hello i tried running serctl add 1001 1001 chris_cleofe2yahoo.com but i got this error: awk: syntax error near line 1 awk: bailing out near line 1 awk: syntax error near line 1 awk: bailing out near line 1 HA1 calculation failed pls advise. thnks.

4 10

[Serusers] ser and database
by Sebastian Kühner 01 Aug '05

01 Aug '05

Hi! Has anyone of you used a database to make routing? For example: If the caller marks 001 -> Rewritehost("xxx.xxx.xxx.111"); If the caller marks 0049 -> Rewritehost("xxx.xxx.xxx.222"); The two host names are in my postgres database... Once I used a command like this to stop calls of users that don't have credit: if (!exec_msg(' POS1=`expr index "$SIP_HF_FROM" sip: + 4`; LENGTH=`expr index "$SIP_HF_FROM" @ - $POS1`; … [View More]

1 0

[Serusers] gw-pstn.3.07.cfg with Call-Transfer support
by Bastian Schern 01 Aug '05

01 Aug '05

Hello, I tried out the SER config "gw-pstn.3.07.cfg" from www.onsip.org. Nearly everything works fine but "Call-Transfer" is not working. Is it possible to integrate "Call-Transfer" support into the "gw-pstn.3.07.cfg"? Does anybody integrates this already? Regards Bastian debug=3 fork=yes log_stderror=no listen=192.0.2.13 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/… [View More]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/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("192.0.2.13:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(5); route(1); break; }; if (uri==myself) { if (method=="ACK") { route(6); break; } else if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (uri=~"^sip:011[0-9]*@") { # International PSTN route(4); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { # Domestic PSTN route(4); break; }; sl_send_reply("404", "User Not Found"); break; }; if (method=="CANCEL") { route(1); break; }; route(5); route(1); } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("192.0.2.245"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); route(5); route(1); } route[5] { # ----------------------------------------------------------------- # RTP Proxy Enabler # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; } route[6] { # ------------------------------------------------------------------------ # ACK Handler # ------------------------------------------------------------------------ # ------------------------------------------------------------------------ # Aliases Section # ------------------------------------------------------------------------ lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; lookup("location"); route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ +0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } [View Less]

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[Serusers] codec question
by Innocent Evil 01 Aug '05

01 Aug '05

Scenario: IP Phone --> SIP Server --> Asterisk --> (Voice Mailbox) (G723.1, G729) -> (GSM) -> (GSM, G729) GSM coded file Is ipphone with codec g723.1 and g729 cabable to retrived voice mail from the above picture? If not, please explain a little more. Thanks a lot.

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[Serusers] UAC module status of CVS HEAD version
by Innocent Evil 01 Aug '05

01 Aug '05

Hello, Anybody know about the status of UAC module of CVS HEAD version. Does it work? Thanks,

4 4

[Serusers] NAT-flag and multiple branches
by Klaus Darilion 01 Aug '05

01 Aug '05

Hi! I wonder how the NAT flag will be set during lookup("location") in case of multiple branches and some of the contacts are behind NAT and some not. regards, klaus

2 1

AW: [Serusers] dbtext: table not loaded. example desired
by Müller Thorsten 01 Aug '05

01 Aug '05

Hi all, I'm still at dbtext: I have a line in ser.cfg which is now: modparam("auth_db", "db_url","dbtext:///usr/local/etc/ser/dbtext") where I have a file produced with the script from cesc called "subscriber". Given by DEBUG-mode from dbt_load_file the module "dbtext" searches this file in a directory called "/usr/local/etc/ser" and NOT in the dir given by the "modparam"-line Can somebody give me a hint, why this occurs? Another problem I had with the strncpy/strncat of database and … [View More]table-name. I had to change a line in db_file.c at line 82 from strncpy(path+dbn->len+1,tbn->s,tbn->len) to strncat(path,tbn->s,tbn->len) Is this a bug or is something wrong with my ser.cfg? Thanks Thorsten ser-version: 0.8.14 mt> -----Ursprüngliche Nachricht----- mt> Von: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] mt> Gesendet: Freitag, 29. Juli 2005 10:55 mt> An: Müller Thorsten mt> Cc: serusers(a)lists.iptel.org mt> Betreff: Re: [Serusers] dbtext: table not loaded. example desired mt> mt> mt> Hello, mt> mt> On 07/29/05 11:16, Müller Thorsten wrote: mt> mt> > Hi all, mt> > mt> > I try to set up ser with dbtext as authentication basis. mt> I run in some mt> > problems because the database is not found: mt> > DBT:db_query: table not loaded. mt> > mt> > After some googling a found that Cesc mt> [cesc.santa(a)gmail.com] had also mt> > done some work with dbtext and mailed that to "serusers" mt> or "serdev" mt> > mailinglists mt> > mt> > In a mail from May and June 05 he told that he has mt> written a "serctl" mt> > for dbtext and a script to produce the necessary files. mt> > mt> > Has somebody got these things and can mail me these mt> files! I could not mt> > found them in the mailing history. mt> > mt> > Has somebody also dealed with dbtext and can give me some example mt> > files for ser.cfg and dbtext files for mt> "authentication/subscriber" and mt> > "location". mt> > mt> rhe readme file of dbtext module has some examples of table mt> structures mt> as well as a simple config file. mt> mt> http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/mod mt> ules/dbtext/README?rev=HEAD&content-type=text/vnd.viewcvs-markup mt> mt> Daniel mt> mt> > Thanks in advance mt> > mt> > Thorsten Mueller mt> > mt> > ser 0.8.14 cross-compiled for arm mt> > My ser.cfg: mt> > #################################################### mt> > ... mt> > # Added by mt for authentication with dbtext mt> > loadmodule "/usr/local/lib/ser/modules/auth.so" mt> > loadmodule "/usr/local/lib/ser/modules/auth_db.so" mt> > loadmodule "/usr/local/lib/ser/modules/dbtext.so" mt> > mt> > modparam("auth_db", "db_url","dbtext:///var/dbtext/ser") mt> > mt> > # -- auth params -- mt> > modparam("auth_db", "calculate_ha1", 1) mt> > modparam("auth_db", "password_column", "password") mt> > modparam("auth_db", "user_column", "username") mt> > modparam("auth_db", "domain_column", "domain") mt> > ... mt> > mt> > if (uri==myself) { mt> > mt> > if (method=="REGISTER") { mt> > mt> > if (!www_authorize("domain.com", "subscriber")) { mt> > www_challenge("domain.com", "0"); mt> > break; mt> > }; mt> > save("location"); mt> > break; mt> > } mt> > ... mt> > ##################################################### mt> > mt> > I have a file in /var/dbtext/ser which is called "subscriber": mt> > ##################################################### mt> > username(str) password(str) ha1(str) domain(str) ha1b(str) mt> > suser:supasswd:xxx:domain.com:xxx mt> > ##################################################### mt> > mt> >----------------------------------------------------------- mt> ------------- mt> > mt> >_______________________________________________ mt> >Serusers mailing list mt> >serusers(a)lists.iptel.org mt> >http://lists.iptel.org/mailman/listinfo/serusers mt> > mt> > mt> [View Less]

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[Serusers] Problem with digest authentication
by aimable 01 Aug '05

01 Aug '05

Hi serusers, I have been using various versions of SER from last year without any problem but recently I made a new installation of OpenSER 0.9.5. Since then I am having problems with digest authentication from some of the phones. I have a bunch of 186 ATAs and Cisco 7940 phones but they cannot register to the server, while all the soft phones can register successfully. The server says Credentials with given realm not found. I tried to change the realm to localhost, the IP address, with no luck.… [View More]

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[Serusers] Adding Asterisk Support for Voice Messages
by ntandoh＠termilink.net 01 Aug '05

01 Aug '05

Does anyone have info on adding Asterisk Support for Voice Messages, that page is missing from the SIP - Getting Started - Issue 04a.doc www.ONsip.org ?

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[Serusers] sip_method
by Szabolcs Szasz 01 Aug '05

01 Aug '05

Hi my friends! I have been tested my SER config for a few weeks (ser-0.8.14). It's OK, but in the ser's MySQL database in the acc table appears a few row with sip_method=PRACK and sip_method=INFO (there are > 1100 rows with INVITE,ACK and BYE). What is this meaning? I cannot regenerate this row. can anybody help me? Thanks Szabolcs

1 0

[Serusers] Existing Radius Authentication
by Ryan Pagquil 01 Aug '05

01 Aug '05

Hi, Can I use my existing radius server as my login authentication for ser? The existing radius uses the system to read the user accounts, but explained on the radius howto i must create the user accounts on users file of the freeradius. Please help. Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

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RE: [Serusers] 1 way voice issue
by Sam Lee 01 Aug '05

01 Aug '05

Could the 183 session progress have anything to do with the one way voice ? ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Monday, August 01, 2005 4:54 PM To: Sam Lee; serusers(a)lists.iptel.org Subject: Re: [Serusers] 1 way voice issue Very hard to say without a complete ngrep trace. If both parties are on public IPs, there is probably something with the signalling, not the SDP payload. The 183 Session Progress is a so-called provisional reponse. … [View More]

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[Users] Using permissions
by jayesh nambiar 01 Aug '05

01 Aug '05

hi there, Can you please tell me if I can use the permissions module to trust the requests coming towards my SIP server using source IP. But my problem is I want to use flat files to store the source IPs I trust. Is it possible And if the IPs are trusted I need to forward them without any authentication. I need to store the trusted IPs in flat files since I dont need the overhead of querying DB. Please help me out. Thanx --------------------------------- Start your day with Yahoo! - make … [View More]

2 1

[Serusers] Any known problems with running SER behind NAT?
by Alan 01 Aug '05

01 Aug '05

Hi List, I wonder if you could share your experience with me of running SER behind NAT. Are there any known problems with doing it this way? - Server side NAT and have SER fix NAT'ed clients and (Media)Proxying them where needed. My specific setup is as follows:- SER(0.9.13) with a static private IP Firewall performing static on-to-one NAT Users will be on public IP's/(behind NAT) Does this cause any problems with the SIP/SDP packets? Also it is the intention to run Asterisk in the … [View More]

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[Serusers] 1 way voice issue
by Sam Lee 01 Aug '05

01 Aug '05

Hi, I've recently encountered some problem with my SIP service whereby i call out to a specific number and i encounter a one way voice. If i'm the initiator, i cannot hear the other party but he can hear me. At first i thought it was a return route issue (as i'm going thru NAT) , so i switch my SIP to a public IP but i still face the same problem. Its really only that specific PSTN number that i have dialed facing this problem. The only difference that i can think of is that PSTN number is on … [View More]

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[Serusers] STUN for NAT traversal...
by Roger Lewau 01 Aug '05

01 Aug '05

I'm trying to use STUN for NAT traversal for all clients not using 1 to 1 NATing, and mediaproxy for the rare few who does.But it seems that all traffic is proxied anyway, dont really understand why. I do call the routeblock for mediaproxy before relaying the call, but since STUN is used it should not detect that NAT is in place. Or am I missing some logic? Any Idea on how to accomplish this? Kind regards Roger - Getting one step closer to a really god service everyday ;)

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[Serusers] Mediaproxy Load Balacing
by smy＠addm.com 01 Aug '05

01 Aug '05

Hi all, I've tried playing with Mediaproxy for sometime now, and thou with some clarification from Greger, I'm still having some problems with getting it up and running. I'm trying to have 1 SER with multiple Mediaproxy, and here's my configuration. I'm desperate since I've experienced no improvement with the system after over a week playing with it! :< Any help is really appreciated. --------- ser.cfg --------- loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/… [View More]

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[Serusers] NAT Problem (Mediaproxy)... Pls help
by Yan Yu Lim 01 Aug '05

01 Aug '05

Hi Guys, I have a NAT problem, whereby I cannot get the RTP packets from a NATed user. Hence, I am trying to implement the Mediaproxy solution from ONSIP. 1. Mediaproxy and Ser are installed on the same PC, say 202.123.45.85. I've set SER to listen at 202.123.45.85, port 5060. How should I configure the ini file for Mediaproxy? Can someone please provide me with a simple example of how the mediaproxy ini file should be configured? 2. Assuming that I set mediaproxy to listen at 202.123.45.85, … [View More]

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[Serusers] Mediaproxy -- pls help!
by yy lim 01 Aug '05

01 Aug '05

Hi guys, I'm trying to get my mediaproxy working without much success. Both mediaproxy and ser are installed on the same machine, i.e. 201.23.13.145 for example. SER is listening on port 5060, and mediaproxy is set up as below. Please kindly advise. Rgrds, Zan ----- Mediaproxy.ini ------ [MediaProxy] start = yes socket = /var/run/mediaproxy.sock group = ser listen = 201.23.13.145:5062 allow = any proxyIP = 201.23.13.145 portRange = 5062:5064 TOS = 0xb8 idleTimeout = 60 holdTimeout = 3600 … [View More]forceClose = 0 accounting = off ------- ser.cfg ------- debug=3 fork=yes listen=201.23.13.145 log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:XXXXX@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/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/acc.so" # -- mediaproxy NAT modules --- loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" # ----------------- setting module-specific parameters --------------- # -- auth_radius params -- modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:XXXXXX@localhost/ser") modparam("acc", "db_url", "mysql://ser:XXXXXX@localhost/ser") # -- usrloc params -- # store user location in memory, not using database modparam("usrloc", "db_mode", 0) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", 1) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # mysql accounting # -- acc params -- modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 1) modparam("acc", "failed_transactions", 1) # -- tm params -- # set time for which ser will be waiting for a final response; # fr_inv_timer sets value for INVITE transactions, # fr_timer for all others modparam("tm","fr_inv_timer",15) modparam("tm","fr_timer",10) modparam("tm","fr_inv_timer_avp", "inv_timeout") #modparam("permissions", "db_mode", 1) #modparam("permissions", "trusted_table", "trusted") # -- Mediaproxy params -- modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("registrar", "nat_flag", 6) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # main routing logic route{ # ----------------------------- # Sanity check # ----------------------------- # 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; }; # ----------------------------- # Record Route Section # ----------------------------- # perform NAT check if (method=="INVITE" && client_nat_test("3")) { record_route_preset("201.23.13.145:5060;nat=yes"); log(1,"Invite + NAT test- Record Route\n"); } # 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 else if(method!="REGISTER") { record_route(); }; # ------------------------------- # Call Tear Down Section # ------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); log(1,"Tear Down Section\n"); }; # ------------------------- # loose-route processing # ------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; log(1,"Loose route processing\n"); }; route(1); break; }; setflag(1); # ----------------------------- # Call Type Processing Section # ----------------------------- if(uri != myself) { route(5); route(1); break; }; if (uri==myself) { if(method == "ACK") { route(6); break; } else if (method =="CANCEL") { route(3); break; } else if(method=="INVITE") { route(3); t_on_failure("1"); break; } else if(method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; route(1); } # ------------------------------- # Route Processing Section # ------------------------------- # ------------------------------ # Default Message Handler # ------------------------------ route[1] { t_on_reply("1"); if(!t_relay()) { if(method=="INVITE" || method=="ACK") { end_media_session(); log(1,"Route 1 default message handler\n"); }; sl_reply_error(); }; } # REGISTER Message Handler # ---------------------------------------------------------------- route[2] { sl_send_reply("100", "Trying"); if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); log(1,"Nated register\n"); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } # ----------------------------------------------------------------- # CANCEL & INVITE Message Handler # ----------------------------------------------------------------- route[3] { if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if(method=="INVITE" && !(src_ip==10.10.10.4)) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { # attempt handoff to PSTN if (uri=~"^sip:8[0-9]*@") { log(1, "Forwarding to PSTN\n"); route(4); break; }; # retrieve voicemail if (uri=~"^sip:2[0-9]*@") { log(1, "Retrieving voicemail\n"); route(7); break; }; sl_send_reply("404", "User Not Found!"); break; }; if(method=="CANCEL") { route(1); break; }; route(5); route(1); } # ----------------------------- # PSTN Handler # ----------------------------- route[4] { rewritehostport("10.10.10.4:5060"); avp_write("i:45", "inv_timeout"); # ------------------------------- # RTP Proxy Enabler # ------------------------------- route[5] { if(isflagset(6)||isflagset(7)) { use_media_proxy(); log(1,"Using rtp proxy\n"); }; } # ------------------------------- # ACK # ------------------------------- route[6] { if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found!"); break; }; route(5); route(1); } route(5); route(1); } # ------------------------------ # Asterisk voicemail # ------------------------------ route[7] { #redirect now! rewritehostport("201.23.13.145:5061"); append_branch(); t_relay_to_udp("201.23.13.145","5061"); } failure_route[1] { if(!(method=="BYE") && !(method=="CANCEL")) { log(1,"Activating voicemail!!\n"); }; #revert_uri(); #redirect now! rewritehostport("201.23.13.145:5061"); append_branch(); t_relay_to_udp("201.23.13.145","5061"); } onreply_route[1] { log(1,"On reply route/n"); if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ +0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com [View Less]

1 0

Re: [Serusers] serctll add error
by chris 01 Aug '05

01 Aug '05

hi kofi, were you able to run and test my config file? any developments? :) much thanks. chris ----- Original Message ----- From: chris To: Kofi Obiri-Yeboah ; serusers(a)lists.iptel.org Sent: Wednesday, July 27, 2005 11:46 AM Subject: Re: [Serusers] serctll add error hi kofi, i did commented out the listen=IP line. but i gives me the same error. ERROR: bad config file (1 errors) by the way, when i tried the hello_world.cfg from onsip i did not encountered any … [View More]problem and ser was able to run. thnks kofi, i will wait for you reply next week. chris. ----- Original Message ----- From: Kofi Obiri-Yeboah To: chris ; serusers(a)lists.iptel.org Sent: Wednesday, July 27, 2005 11:19 AM Subject: RE: [Serusers] serctll add error Chris, I will run your config file over the weekend and will get back to you. Clearly that fact that you are receiving a bad config file error is an indication that there is a problem. Could you please comment out the listen=IP address line and rerun ?????? KOFI -----Original Message----- From: chris [mailto:chris@fivestartel.com] Sent: Tuesday, July 26, 2005 10:19 AM To: Kofi Obiri-Yeboah; serusers(a)lists.iptel.org Subject: Re: [Serusers] serctll add error hi kofi, attached is my ser.cfg i got this sample script from onsip.org getting started doc. this is their auth-mysql.3.05.cfg sample script, i just change the IP address on the listen variable. running using this cfg file leads me to an error. ERROR: bad config file (1 errors) maybe u can help me fix this. i have no idea how i can give you more details on the error cause that is the only line the system gives me. thnks. debug=3 fork=yes log_stderror=no listen=203.115.128.41 # put your server IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" 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/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; }; if (uri==myself) { if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); }; } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } ----- Original Message ----- From: Kofi Obiri-Yeboah To: chris ; serusers(a)lists.iptel.org Sent: Tuesday, July 26, 2005 11:39 PM Subject: RE: [Serusers] serctll add error please send your config file -----Original Message----- From: chris [mailto:chris@fivestartel.com] Sent: Tuesday, July 26, 2005 6:34 AM To: Kofi Obiri-Yeboah; serusers(a)lists.iptel.org Subject: Re: [Serusers] serctll add error hi, i still got the same error bash-2.05# serctl add 1001 1001 chris(a)yahoo.com awk: syntax error near line 1 awk: bailing out near line 1 awk: syntax error near line 1 awk: bailing out near line 1 HA1 calculation failed bash-2.05# pls. advise on how to fix this. thnks. ----- Original Message ----- From: Kofi Obiri-Yeboah To: chris ; serusers(a)lists.iptel.org Sent: Tuesday, July 26, 2005 8:06 PM Subject: RE: [Serusers] serctll add error syntax is <username> <password> <email address> you have a typo in the email address use "@" instead of "2" use shift-2 before yahoo.com HTH -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org]On Behalf Of chris Sent: Tuesday, July 26, 2005 12:58 AM To: serusers(a)lists.iptel.org Subject: [Serusers] serctll add error hello i tried running serctl add 1001 1001 chris_cleofe2yahoo.com but i got this error: awk: syntax error near line 1 awk: bailing out near line 1 awk: syntax error near line 1 awk: bailing out near line 1 HA1 calculation failed pls advise. thnks. 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Re: [Serusers] Problem : Can SER process the reINVITE messages properly?
by Greger V. Teigre 01 Aug '05

01 Aug '05

BTW, I sketched out a few other solutions, where I believe at least a couple should be valid attempts. Did you try those and what were the results? g-) Ricardo Martinez wrote: > Hello Greger and SER users. > About this issue here are some test that i made. I added the > lookup("location") as you recommend me : > > # > > > > ----------------------------------------------------------------- # > Loose Route Section # > -----------------------------------… [View More]------------------------------ if > (loose_route()) { > > if (has_totag() && (method=="INVITE" || > method=="ACK")) { lookup("location"); > if (isflagset(6) || client_nat_test("3") || > search("^Route:.*;nat=yes")){ > setflag(6); > use_media_proxy(); > }; > }; > > route(1); > break; > }; > > > Unfortunately, i'm still unable to have voice in both ways, but this > modification seems to handle better the reINVITE message. And this > is why. For the normal approach (i mean the initial onsip_pstn > configuration) the second INVITE was not using the mediaproxym as > Greger pointed. Now, as you can see in the debugs that i'm > attaching, the second invite is using the mediaproxy, and therefore > the "c" parameter in the SDP message is being correctly modified by > SER. So, i don't understand why this time the call is failing. Could > it be that mediaproxy is not capable to handle the reINVITEs?. For > example in the debug from mediaproxy (i'm also attaching this one), i > see a warning message like this : warning: Received packet from a > third party: 200.0.0.7:16432. It seems like mediaproxy is not > accepting "voice packets" from the second IP (the one indicated by > the reINVITE). Why? > Also, when the call is established this is the output from the > "session" command : > > Caller Via Called > Status Duration Codec Type Traffic > ---------------------------------------------------------------------------- > -------------------------------------- > 200.0.0.6:17480 - 200.0.0.5:35094 - 200.0.0.4:46115 inactive > 0'24" G729 Audio 0/16.75k/16.75k > > The call is established but between the first IP (Asterisk box) and > the endpoint public IP. > So, could this be a problem with mediaproxy?. I read the changelog > from the version 1.3.1 and this is what i found : > > Changes from version 1.2.1 to 1.3.0 > ----------------------------------- > > - Only create a new session if a request belongs to the first INVITE > (no longer creates new sessions on re-INVITEs) > > could it be related?. > > I also have a general question about the reINVITE. As you can see > the first INVITE has in his URI the domain name for the number : > 5555848114(a)sipvoiss.desarrollo.redvoiss.net (Frame 1 from the debug). > The reINVITE sent by Asterisk has the URI : 5555848114(a)200.0.0.4. Is > this ok?, despite of this the reINVITE is correctly redirected to the > endpoint. (You can check this in the frame 13 from the debug). > > Thanks again. > > Regards, > > Ricardo Martinez.- > > > >> -----Mensaje original----- >> De: Greger V. Teigre [mailto:greger@teigre.com] >> Enviado el: Miércoles, 20 de Julio de 2005 3:06 >> Para: Ricardo Martinez; serusers(a)lists.iptel.org >> Asunto: Re: [Serusers] Problem : Can SER process the reINVITE >> messages properly? >> >> >> Hm. This was a tricky one. I would like to ask a second >> opinion from you >> NAT gurus on my musings below. What is the most appropriate >> response to >> Ricardo's scenario? WARNING to others: This is getting fairly >> technical on >> NAT/SDP... ;-) >> >> The reINVITE from Asterisk has a valid IP address and a valid >> (but new) SDP >> media ip (C=IN IP4 200.0.0.7). nat=yes is correctly not >> present in the >> reINVITE from Asterisk (it is not NATed). Thus >> use_media_proxy is correctly >> not used in the reINVITE, i.e. client_nat_test("3") and test >> on nat=yes will >> not trigger under the has_totag if statement (loose route >> processing): >> >> if (loose_route()) { >> >> if (has_totag() && (method=="INVITE" || method=="ACK")) { >> >> if (client_nat_test("3") || search("^Route:.*;nat=yes")) { >> setflag(6); >> use_media_proxy(); >> }; >> }; >> >> route(1); >> break; >> }; >> >> In the OK from the Linksys: >> U 200.0.0.5:5060 -> 200.0.0.6:5060 >> SIP/2.0 200 OK. >> To: >> <sip:5555848114@sipvoiss.desarrollo.redvoiss.net>;tag=64d7492d >> b018b13ai0. From: "5565625858196" >> <sip:5565625858196@sipvoiss.desarrollo.redvoiss.net>;tag=as5d352c9a. >> Call-ID: >> 0848ef78297116742562204e2c04cbcb(a)sipvoiss.desarrollo.redvoiss.net. >> CSeq: 104 INVITE. >> Via: SIP/2.0/UDP 200.0.0.6:5060;branch=z9hG4bK518d870e. >> Record-Route: <sip:200.0.0.5;ftag=as5d352c9a;lr=on>. >> Contact: 5555848114 <sip:5555848114@200.0.0.4:46081>. >> Server: Linksys/PAP2-2.0.10(LSb). >> Content-Length: 237. >> Content-Type: application/sdp. >> . >> v=0. >> o=- 33068 33068 IN IP4 192.168.15.200. >> s=-. >> c=IN IP4 192.168.15.200. >> t=0 0. >> m=audio 16424 RTP/AVP 18 100 101. >> a=rtpmap:18 G729/8000. >> a=rtpmap:100 NSE/8000. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-15. >> a=ptime:30. >> a=sendrecv. >> >> ...you have 192.168.15.200, a private c= address in SDP, but >> the Contact has >> correctly been fixed (200.0.0.4:46081). So it seems the OK is >> processed >> correctly in onreply even though it does not show up in the >> debug. Flags 6 >> and 7 are not set, so use_media_proxy is correctly not run. >> The problem is >> that flag 6 should have been set (the Linksys is NATed), but >> as the reINVITE >> is not processed using lookup("location") the flag will not >> be picked up >> from usrloc. >> ---------------------------------------- >> >> However, now that Asterisk in the reINVITE is requesting >> another (valid) IP >> as media destination, the question is: what is propor behavior? >> >> The most natural would be to add a lookup("location") after: >> if (has_totag() && (method=="INVITE" || method=="ACK")) >> and then add isflagset(6) to the or test: >> if (isflagset(6) || client_nat_test("3") || >> search("^Route:.*;nat=yes")) { >> setflag(6); >> use_media_proxy(); >> }; >> >> However, I'm not sure the lookup will work, because the R-URI >> is likely to >> be the direct address (5555848114@200.0.0.4:46081) and not >> the registered >> URI (@domain.com). It would be tempting to try to see if we >> somehow could >> make nat=yes set in the early INVITE. We would have to split the >> record_route section in two. One to handle !="REGISTER && >> !="INVITE" with a >> regular record_route() and wait until AFTER >> lookup("location") (maybe in >> route[1], the default handler) to add the following: >> if(method=="INVITE")){ >> if(isflagset(6)) { >> # INSERT YOUR IP ADDRESS HERE >> record_route_preset("192.0.2.13:5060;nat=yes"); >> } else { >> record_route(); >> } >> } >> >> This would make sure that not only dialogs where caller is >> nated, but also >> where callee is nated, will be tagged as nated. This again >> would cause >> nat=yes to be set and rtp proxy forced, further flag 6 is >> set, and thus the >> OK also processed with use_media_proxy(). >> >> BUT: Would this nat=yes cause any problems in other >> scenarios??? I'm not >> sure, but I don't think so. If that is correct, this is >> maybe the right >> solution. >> --------------------------------------- >> OTHER (SIMPLER?) SOLUTIONS: >> ---------- >> Instead you could add ||client_nat_test("1") together with >> isflagset 6 and 7 >> in your reply route: >> if ((isflagset(6) || isflagset(7) || client_nat_test("1")) && >> (status=~"(180)|(183)|2[0-9][0-9]")) >> >> but it will not work because the reINVITE did not get >> use_media_proxy(). >> >> Or, you could instead add fix_nated_sdp("2"); in your onreply route: >> if (client_nat_test("1")) { >> fix_nated_contact(); >> fix_nated_sdp("2"); >> }; >> WARNING! You will have to add else in between this test and >> the previous to >> avoid messing up use_media_proxy(). >> >> This will fix the SDP (c=IN IP4), but it will probably not >> help, because the >> port is still unknown. You could use "3" instead in >> fix_nated_sdp, this >> will add direction=active to the OK. However, another thread >> stated that >> Asterisk does not support direction=active. If it does, it >> will probably >> work and you get a non-proxied stream between Asterisk and >> the Linksys. >> >> Or finally, the third option would be to add the (hopefully) >> fixed IP in the >> loose_route test: >> if (src_ip==200.0.0.6 || client_nat_test("3") || >> search("^Route:.*;nat=yes")) { >> setflag(6); >> use_media_proxy(); >> }; >> >> This would cause the OK automatically to get use_media_proxy. >> >> Ricardo, could you please test these various solutions and >> see if they >> really work or if (where...) I go wrong :-) >> g-) >> >> Ricardo Martinez wrote: >>> Hello Greger. >>> Thanks for your answer on this topic. Now i'm attaching more debug >>> information (the /var/log/messages from mediaproxy, the ngrep output >>> and some xlog statements in the ser.cfg file) beside some >> comments in >>> the file reINVITE_debug_problem.txt. >>> >>>> - Is your session really set up initially (before the reINVITE)? >>>> (mediaproxy reports 0/0/0 bytes) >>> >>> At least i have ringback tone. Then, when the call is answered i >>> have an OK and a ACK message coming to my SER box, then inmediatly >>> the reINVITE message arrives Asterisk box. >>> >>> >>> For what i can see from the debug the "nat=yes" is never reached >>> because the caller has a "valid ip" and therefore the first "if" in >>> the statament >>> >>> if (method=="INVITE" && client_nat_test("3")) { >>> # INSERT YOUR IP ADDRESS HERE >>> record_route_preset("64.76.148.246:5060;nat=yes"); >>> xlog("L_INFO", "time [%Tf] RECORD ROUTE SECTION : >>> invite & client_nat_test(3) TRUE ,record_route_preset [%rm]\n"); >>> } else if (method!="REGISTER") { >>> xlog("L_INFO", "time [%Tf] RECORD ROUTE SECTION : >>> record_route [%rm]\n"); >>> record_route(); >>> }; >>> >>> from the RECORD ROUTE SECTION is FALSE. >>> >>> Also, i don't understand why the second OK (the one from the >>> reINVITE) is not procesed in the ONREPLY ROUTE, or at least i don't >>> see any statement from the "xlog" in the debug. Is this normal? >>> >>> Thanks.! >>> Regards, >>> >>> Ricardo Martinez.- >>> >>> >>>> -----Mensaje original----- >>>> De: Greger V. Teigre [mailto:greger@teigre.com] >>>> Enviado el: Lunes, 18 de Julio de 2005 2:45 >>>> Para: Ricardo Martinez; serusers(a)lists.iptel.org >>>> Asunto: Re: [Serusers] Problem : Can SER process the reINVITE >>>> messages properly? >>>> >>>> >>>> Hi Ricardo, >>>> Thanks for a detailed analysis. Some questions: >>>> - Is your session really set up initially (before the reINVITE)? >>>> (mediaproxy reports 0/0/0 bytes) >>>> - You didn't show the ngrep trace. The script uses nat=yes in >>>> the Route >>>> header of the INVITE to detect a nat'ed client. Can you >>>> verify that the >>>> reINVITE has the nat=yes? >>>> - You haven't showed the mediaproxy log (it will show the >>>> callers reporting >>>> in etc). That could help (default /var/log/messages) >>>> - You can put a log statement in the loose_route section >>>> after the test for >>>> nat=yes to see if use_media_proxy was called >>>> >>>> g-) >>>> >>>> Ricardo Martinez wrote: >>>>> Hello. >>>>> I'm having problems trying to make SER, NAT'd endpoints >>>> and reINVITE >>>>> work together. >>>>> I was using the "gw-pstn3.07.cfg" file from onsip.org to do some >>>>> tests, and this is what i have. In one side i have an >> Asterisk with >>>>> an endpoint registered in it (let's call it A). In the >> other side i >>>>> have a PAP2 under NAT (let's call it B). >>>>> >>>>> >>>>> A ---------- Asterisk ----------- SER ----------- B (NAT'd) >>>>> 200.0.0.7 200.0.0.6 200.0.0.5 >>>>> 10.0.0.4 >>>>> >>>>> When i make a call from "A" to "B" this is what i see (in terms of >>>>> SDP). Looking from SER. >>>>> >>>>> A --------- Asterisk ------------ SER ------------ B (NAT'd) >>>>> Public: >>>>> 200.0.0.4 >>>>> 200.0.0.7 200.0.0.6 200.0.0.5 >>>> Inside: >>>>> 10.0.0.1 >>>>> >>>>> INVITE >>>>> c:200.0.0.6:19996 >>>>> -------------------> >>>>> INVITE >>>>> c:200.0.0.5:35010 >>>>> ----------------> >>>>> >>>>> >>>>> Caller Via Called Status >>>>> Duration Codec Type Traffic >>>>> >>>> -------------------------------------------------------------- >>>> ------------ >>>>> 200.0.0.6:19996 - 200.0.0.5:35010 - ?.?.?.?:? inactive 0'04" >>>>> Unknown Audio 0/0/0 >>>>> >>>>> Total traffic: 0bps/0bps/0bps (in1/in2/out) >>>>> Session count: 1 >>>>> >>>>> So far is ok..........and the phone is answered >>>>> OK >>>>> c:10.0.0.1:16440 >>>>> <---------------- (the phone is >>>>> answered) >>>>> OK >>>>> c:200.0.0.5:35010 >>>>> <--------------------- >>>>> >>>>> reINVITE >>>>> c:200.0.0.7:19996 >>>>> ---------------------> >>>>> reINVITE >>>>> c:200.0.0.7:19996 >>>>> ----------------> >>>>> >>>>> OK >>>>> c:10.0.0.1:16440 >>>>> <---------------- >>>>> OK >>>>> c:10.0.0.1:16440 >>>>> <--------------------- >>>>> >>>>> Finally according to the "session" information : >>>>> >>>>> Caller Via Called >>>>> Status Duration Codec Type Traffic >>>>> >>>> -------------------------------------------------------------- >>>> -------------- >>>>> ---------- >>>>> 200.0.0.6:19996 - 200.0.0.5:35010 - 200.0.0.7:16420 inactive >>>>> 0'26" G729 Audio 0/11.48k/11.48k >>>>> >>>>> Total traffic: 0bps/0bps/0bps (in1/in2/out) >>>>> Session count: 1 >>>>> And the audio is only in one way. :( >>>>> >>>>> So. you can see the reINVITE message apparently is not being >>>>> processed as a call to a NAT'd endpoint and therefore is not using >>>>> the mediaproxy, you can see the second "OK" messsage has >> the invalid >>>>> IP from the NAT'd user is in his sdp information. >>>>> As i said it before i am using the gw-pstn configuration >>>> file from the >>>>> onsip.org and as far as i can remember this configuration >> can handle >>>>> the reINVITE? isn't >>>>> I'm also using the last version of the mediaproxy (1.3.1). >>>>> Can someone tell me what i'm doing wrong? >>>>> >>>>> Hope someone could help me here. >>>>> Thanks in advance. >>>>> Regards... >>>>> >>>>> Ricardo Martinez.- >>>>> >>>>> _______________________________________________ >>>>> Serusers mailing list >>>>> serusers(a)lists.iptel.org >>>>> http://lists.iptel.org/mailman/listinfo/serusers [View Less]

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Re: [Serusers] Problem : Can SER process the reINVITE messages properly?
by Greger V. Teigre 01 Aug '05

01 Aug '05

Ricardo, I think you are correct in your analysis of mediaproxy not accepting a third party. Also, I saw the post from somebody on the newest mediaproxy version. Have you tried it? Version 1.2.1 created a new session for reINVITEs, that's why the nat=yes is used in the ONsip.org configs. Version 1.3.0 added support for detecting that a reINVITE came in and would reuse the session (which I btw don't think is good for your scenario as you change the IP). 1.4.2 seems to address your problem,… [View More] so it would be interesting to hear if you have any success. As for the reINVITE with the IP address instead of domain, that's ok. g-) Ricardo Martinez wrote: > Hello Greger and SER users. > About this issue here are some test that i made. I added the > lookup("location") as you recommend me : > > # > > > > ----------------------------------------------------------------- # > Loose Route Section # > ----------------------------------------------------------------- if > (loose_route()) { > > if (has_totag() && (method=="INVITE" || > method=="ACK")) { lookup("location"); > if (isflagset(6) || client_nat_test("3") || > search("^Route:.*;nat=yes")){ > setflag(6); > use_media_proxy(); > }; > }; > > route(1); > break; > }; > > > Unfortunately, i'm still unable to have voice in both ways, but this > modification seems to handle better the reINVITE message. And this > is why. For the normal approach (i mean the initial onsip_pstn > configuration) the second INVITE was not using the mediaproxym as > Greger pointed. Now, as you can see in the debugs that i'm > attaching, the second invite is using the mediaproxy, and therefore > the "c" parameter in the SDP message is being correctly modified by > SER. So, i don't understand why this time the call is failing. Could > it be that mediaproxy is not capable to handle the reINVITEs?. For > example in the debug from mediaproxy (i'm also attaching this one), i > see a warning message like this : warning: Received packet from a > third party: 200.0.0.7:16432. It seems like mediaproxy is not > accepting "voice packets" from the second IP (the one indicated by > the reINVITE). Why? > Also, when the call is established this is the output from the > "session" command : > > Caller Via Called > Status Duration Codec Type Traffic > ---------------------------------------------------------------------------- > -------------------------------------- > 200.0.0.6:17480 - 200.0.0.5:35094 - 200.0.0.4:46115 inactive > 0'24" G729 Audio 0/16.75k/16.75k > > The call is established but between the first IP (Asterisk box) and > the endpoint public IP. > So, could this be a problem with mediaproxy?. I read the changelog > from the version 1.3.1 and this is what i found : > > Changes from version 1.2.1 to 1.3.0 > ----------------------------------- > > - Only create a new session if a request belongs to the first INVITE > (no longer creates new sessions on re-INVITEs) > > could it be related?. > > I also have a general question about the reINVITE. As you can see > the first INVITE has in his URI the domain name for the number : > 5555848114(a)sipvoiss.desarrollo.redvoiss.net (Frame 1 from the debug). > The reINVITE sent by Asterisk has the URI : 5555848114(a)200.0.0.4. Is > this ok?, despite of this the reINVITE is correctly redirected to the > endpoint. (You can check this in the frame 13 from the debug). > > Thanks again. > > Regards, > > Ricardo Martinez.- > > > >> -----Mensaje original----- >> De: Greger V. Teigre [mailto:greger@teigre.com] >> Enviado el: Miércoles, 20 de Julio de 2005 3:06 >> Para: Ricardo Martinez; serusers(a)lists.iptel.org >> Asunto: Re: [Serusers] Problem : Can SER process the reINVITE >> messages properly? >> >> >> Hm. This was a tricky one. I would like to ask a second >> opinion from you >> NAT gurus on my musings below. What is the most appropriate >> response to >> Ricardo's scenario? WARNING to others: This is getting fairly >> technical on >> NAT/SDP... ;-) >> >> The reINVITE from Asterisk has a valid IP address and a valid >> (but new) SDP >> media ip (C=IN IP4 200.0.0.7). nat=yes is correctly not >> present in the >> reINVITE from Asterisk (it is not NATed). Thus >> use_media_proxy is correctly >> not used in the reINVITE, i.e. client_nat_test("3") and test >> on nat=yes will >> not trigger under the has_totag if statement (loose route >> processing): >> >> if (loose_route()) { >> >> if (has_totag() && (method=="INVITE" || method=="ACK")) { >> >> if (client_nat_test("3") || search("^Route:.*;nat=yes")) { >> setflag(6); >> use_media_proxy(); >> }; >> }; >> >> route(1); >> break; >> }; >> >> In the OK from the Linksys: >> U 200.0.0.5:5060 -> 200.0.0.6:5060 >> SIP/2.0 200 OK. >> To: >> <sip:5555848114@sipvoiss.desarrollo.redvoiss.net>;tag=64d7492d >> b018b13ai0. From: "5565625858196" >> <sip:5565625858196@sipvoiss.desarrollo.redvoiss.net>;tag=as5d352c9a. >> Call-ID: >> 0848ef78297116742562204e2c04cbcb(a)sipvoiss.desarrollo.redvoiss.net. >> CSeq: 104 INVITE. >> Via: SIP/2.0/UDP 200.0.0.6:5060;branch=z9hG4bK518d870e. >> Record-Route: <sip:200.0.0.5;ftag=as5d352c9a;lr=on>. >> Contact: 5555848114 <sip:5555848114@200.0.0.4:46081>. >> Server: Linksys/PAP2-2.0.10(LSb). >> Content-Length: 237. >> Content-Type: application/sdp. >> . >> v=0. >> o=- 33068 33068 IN IP4 192.168.15.200. >> s=-. >> c=IN IP4 192.168.15.200. >> t=0 0. >> m=audio 16424 RTP/AVP 18 100 101. >> a=rtpmap:18 G729/8000. >> a=rtpmap:100 NSE/8000. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-15. >> a=ptime:30. >> a=sendrecv. >> >> ...you have 192.168.15.200, a private c= address in SDP, but >> the Contact has >> correctly been fixed (200.0.0.4:46081). So it seems the OK is >> processed >> correctly in onreply even though it does not show up in the >> debug. Flags 6 >> and 7 are not set, so use_media_proxy is correctly not run. >> The problem is >> that flag 6 should have been set (the Linksys is NATed), but >> as the reINVITE >> is not processed using lookup("location") the flag will not >> be picked up >> from usrloc. >> ---------------------------------------- >> >> However, now that Asterisk in the reINVITE is requesting >> another (valid) IP >> as media destination, the question is: what is propor behavior? >> >> The most natural would be to add a lookup("location") after: >> if (has_totag() && (method=="INVITE" || method=="ACK")) >> and then add isflagset(6) to the or test: >> if (isflagset(6) || client_nat_test("3") || >> search("^Route:.*;nat=yes")) { >> setflag(6); >> use_media_proxy(); >> }; >> >> However, I'm not sure the lookup will work, because the R-URI >> is likely to >> be the direct address (5555848114@200.0.0.4:46081) and not >> the registered >> URI (@domain.com). It would be tempting to try to see if we >> somehow could >> make nat=yes set in the early INVITE. We would have to split the >> record_route section in two. One to handle !="REGISTER && >> !="INVITE" with a >> regular record_route() and wait until AFTER >> lookup("location") (maybe in >> route[1], the default handler) to add the following: >> if(method=="INVITE")){ >> if(isflagset(6)) { >> # INSERT YOUR IP ADDRESS HERE >> record_route_preset("192.0.2.13:5060;nat=yes"); >> } else { >> record_route(); >> } >> } >> >> This would make sure that not only dialogs where caller is >> nated, but also >> where callee is nated, will be tagged as nated. This again >> would cause >> nat=yes to be set and rtp proxy forced, further flag 6 is >> set, and thus the >> OK also processed with use_media_proxy(). >> >> BUT: Would this nat=yes cause any problems in other >> scenarios??? I'm not >> sure, but I don't think so. If that is correct, this is >> maybe the right >> solution. >> --------------------------------------- >> OTHER (SIMPLER?) SOLUTIONS: >> ---------- >> Instead you could add ||client_nat_test("1") together with >> isflagset 6 and 7 >> in your reply route: >> if ((isflagset(6) || isflagset(7) || client_nat_test("1")) && >> (status=~"(180)|(183)|2[0-9][0-9]")) >> >> but it will not work because the reINVITE did not get >> use_media_proxy(). >> >> Or, you could instead add fix_nated_sdp("2"); in your onreply route: >> if (client_nat_test("1")) { >> fix_nated_contact(); >> fix_nated_sdp("2"); >> }; >> WARNING! You will have to add else in between this test and >> the previous to >> avoid messing up use_media_proxy(). >> >> This will fix the SDP (c=IN IP4), but it will probably not >> help, because the >> port is still unknown. You could use "3" instead in >> fix_nated_sdp, this >> will add direction=active to the OK. However, another thread >> stated that >> Asterisk does not support direction=active. If it does, it >> will probably >> work and you get a non-proxied stream between Asterisk and >> the Linksys. >> >> Or finally, the third option would be to add the (hopefully) >> fixed IP in the >> loose_route test: >> if (src_ip==200.0.0.6 || client_nat_test("3") || >> search("^Route:.*;nat=yes")) { >> setflag(6); >> use_media_proxy(); >> }; >> >> This would cause the OK automatically to get use_media_proxy. >> >> Ricardo, could you please test these various solutions and >> see if they >> really work or if (where...) I go wrong :-) >> g-) >> >> Ricardo Martinez wrote: >>> Hello Greger. >>> Thanks for your answer on this topic. Now i'm attaching more debug >>> information (the /var/log/messages from mediaproxy, the ngrep output >>> and some xlog statements in the ser.cfg file) beside some >> comments in >>> the file reINVITE_debug_problem.txt. >>> >>>> - Is your session really set up initially (before the reINVITE)? >>>> (mediaproxy reports 0/0/0 bytes) >>> >>> At least i have ringback tone. Then, when the call is answered i >>> have an OK and a ACK message coming to my SER box, then inmediatly >>> the reINVITE message arrives Asterisk box. >>> >>> >>> For what i can see from the debug the "nat=yes" is never reached >>> because the caller has a "valid ip" and therefore the first "if" in >>> the statament >>> >>> if (method=="INVITE" && client_nat_test("3")) { >>> # INSERT YOUR IP ADDRESS HERE >>> record_route_preset("64.76.148.246:5060;nat=yes"); >>> xlog("L_INFO", "time [%Tf] RECORD ROUTE SECTION : >>> invite & client_nat_test(3) TRUE ,record_route_preset [%rm]\n"); >>> } else if (method!="REGISTER") { >>> xlog("L_INFO", "time [%Tf] RECORD ROUTE SECTION : >>> record_route [%rm]\n"); >>> record_route(); >>> }; >>> >>> from the RECORD ROUTE SECTION is FALSE. >>> >>> Also, i don't understand why the second OK (the one from the >>> reINVITE) is not procesed in the ONREPLY ROUTE, or at least i don't >>> see any statement from the "xlog" in the debug. Is this normal? >>> >>> Thanks.! >>> Regards, >>> >>> Ricardo Martinez.- >>> >>> >>>> -----Mensaje original----- >>>> De: Greger V. Teigre [mailto:greger@teigre.com] >>>> Enviado el: Lunes, 18 de Julio de 2005 2:45 >>>> Para: Ricardo Martinez; serusers(a)lists.iptel.org >>>> Asunto: Re: [Serusers] Problem : Can SER process the reINVITE >>>> messages properly? >>>> >>>> >>>> Hi Ricardo, >>>> Thanks for a detailed analysis. Some questions: >>>> - Is your session really set up initially (before the reINVITE)? >>>> (mediaproxy reports 0/0/0 bytes) >>>> - You didn't show the ngrep trace. The script uses nat=yes in >>>> the Route >>>> header of the INVITE to detect a nat'ed client. Can you >>>> verify that the >>>> reINVITE has the nat=yes? >>>> - You haven't showed the mediaproxy log (it will show the >>>> callers reporting >>>> in etc). That could help (default /var/log/messages) >>>> - You can put a log statement in the loose_route section >>>> after the test for >>>> nat=yes to see if use_media_proxy was called >>>> >>>> g-) >>>> >>>> Ricardo Martinez wrote: >>>>> Hello. >>>>> I'm having problems trying to make SER, NAT'd endpoints >>>> and reINVITE >>>>> work together. >>>>> I was using the "gw-pstn3.07.cfg" file from onsip.org to do some >>>>> tests, and this is what i have. In one side i have an >> Asterisk with >>>>> an endpoint registered in it (let's call it A). In the >> other side i >>>>> have a PAP2 under NAT (let's call it B). >>>>> >>>>> >>>>> A ---------- Asterisk ----------- SER ----------- B (NAT'd) >>>>> 200.0.0.7 200.0.0.6 200.0.0.5 >>>>> 10.0.0.4 >>>>> >>>>> When i make a call from "A" to "B" this is what i see (in terms of >>>>> SDP). Looking from SER. >>>>> >>>>> A --------- Asterisk ------------ SER ------------ B (NAT'd) >>>>> Public: >>>>> 200.0.0.4 >>>>> 200.0.0.7 200.0.0.6 200.0.0.5 >>>> Inside: >>>>> 10.0.0.1 >>>>> >>>>> INVITE >>>>> c:200.0.0.6:19996 >>>>> -------------------> >>>>> INVITE >>>>> c:200.0.0.5:35010 >>>>> ----------------> >>>>> >>>>> >>>>> Caller Via Called Status >>>>> Duration Codec Type Traffic >>>>> >>>> -------------------------------------------------------------- >>>> ------------ >>>>> 200.0.0.6:19996 - 200.0.0.5:35010 - ?.?.?.?:? inactive 0'04" >>>>> Unknown Audio 0/0/0 >>>>> >>>>> Total traffic: 0bps/0bps/0bps (in1/in2/out) >>>>> Session count: 1 >>>>> >>>>> So far is ok..........and the phone is answered >>>>> OK >>>>> c:10.0.0.1:16440 >>>>> <---------------- (the phone is >>>>> answered) >>>>> OK >>>>> c:200.0.0.5:35010 >>>>> <--------------------- >>>>> >>>>> reINVITE >>>>> c:200.0.0.7:19996 >>>>> ---------------------> >>>>> reINVITE >>>>> c:200.0.0.7:19996 >>>>> ----------------> >>>>> >>>>> OK >>>>> c:10.0.0.1:16440 >>>>> <---------------- >>>>> OK >>>>> c:10.0.0.1:16440 >>>>> <--------------------- >>>>> >>>>> Finally according to the "session" information : >>>>> >>>>> Caller Via Called >>>>> Status Duration Codec Type Traffic >>>>> >>>> -------------------------------------------------------------- >>>> -------------- >>>>> ---------- >>>>> 200.0.0.6:19996 - 200.0.0.5:35010 - 200.0.0.7:16420 inactive >>>>> 0'26" G729 Audio 0/11.48k/11.48k >>>>> >>>>> Total traffic: 0bps/0bps/0bps (in1/in2/out) >>>>> Session count: 1 >>>>> And the audio is only in one way. :( >>>>> >>>>> So. you can see the reINVITE message apparently is not being >>>>> processed as a call to a NAT'd endpoint and therefore is not using >>>>> the mediaproxy, you can see the second "OK" messsage has >> the invalid >>>>> IP from the NAT'd user is in his sdp information. >>>>> As i said it before i am using the gw-pstn configuration >>>> file from the >>>>> onsip.org and as far as i can remember this configuration >> can handle >>>>> the reINVITE? isn't >>>>> I'm also using the last version of the mediaproxy (1.3.1). >>>>> Can someone tell me what i'm doing wrong? >>>>> >>>>> Hope someone could help me here. >>>>> Thanks in advance. >>>>> Regards... >>>>> >>>>> Ricardo Martinez.- >>>>> >>>>> _______________________________________________ >>>>> Serusers mailing list >>>>> serusers(a)lists.iptel.org >>>>> http://lists.iptel.org/mailman/listinfo/serusers [View Less]

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RE: [Serusers] LCR on onsip.org - No DB tables created?
by Dave 30 Jul '05

30 Jul '05

I used exactly that script. But It did not create any LCR related tables.!!! --- Ricardo Martinez <rmartinez(a)redvoiss.net> wrote: > Hello Dave. > If i'm not wrong, there is a shell script included > in the last SER > version to create all the DB tables related to SER. > I think there is in : > /sip_router/scripts/ and is called "ser_mysql.sh" > You can try that!. > > Regards, > Ricardo Martinez > > > > > -----Mensaje original----- &… [View More]

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RE: [Serusers] LCR on onsip.org - No DB tables created?
by Dave 30 Jul '05

30 Jul '05

Thanks Ricardo - Do I need to add these to "versions" table in ser database? If yes, what is the version number for these tables? Dave --- Ricardo Martinez <rmartinez(a)redvoiss.net> wrote: > Hello Dave. > I was almost sure that the tables for LCR were > defined there!, my mystake. > But, i have the table definitions (i can't rememeber > where i found it). You > can check with Juha if there is a mistake in the > tables. > > # > # Table structure for table … [View More]'gw' > # > > CREATE TABLE gw ( > gw_name VARCHAR(128) NOT NULL, > ip_addr INT UNSIGNED NOT NULL, > port SMALLINT UNSIGNED, > grp_id INT UNSIGNED NOT NULL, > PRIMARY KEY (gw_name), > KEY (grp_id) > ); > > # > # Table structure for table 'gw_grp' > # > > CREATE TABLE gw_grp ( > grp_id INT UNSIGNED NOT NULL AUTO_INCREMENT > PRIMARY KEY, > grp_name VARCHAR(64) NOT NULL > ); > > # > # Table structure for table 'lcr' > # > > CREATE TABLE lcr ( > prefix varchar(16) NOT NULL, > from_uri varchar(128) NOT NULL DEFAULT '%', > grp_id INT UNSIGNED NOT NULL, > priority TINYINT UNSIGNED NOT NULL, > KEY (prefix), > KEY (from_uri), > KEY (grp_id) > ); > > > Regards > Ricardo.- > > > -----Mensaje original----- > > De: Dave [mailto:ddx66@yahoo.com] > > Enviado el: Viernes, 29 de Julio de 2005 11:30 > > Para: Ricardo Martinez; serusers(a)lists.iptel.org > > Asunto: RE: [Serusers] LCR on onsip.org - No DB > tables created? > > > > > > I used exactly that script. But It did not create > any > > LCR related tables.!!! > > > > > > --- Ricardo Martinez <rmartinez(a)redvoiss.net> > wrote: > > > > > Hello Dave. > > > If i'm not wrong, there is a shell script > included > > > in the last SER > > > version to create all the DB tables related to > SER. > > > I think there is in : > > > /sip_router/scripts/ and is called > "ser_mysql.sh" > > > You can try that!. > > > > > > Regards, > > > Ricardo Martinez > > > > > > > > > > > > > -----Mensaje original----- > > > > De: Dave [mailto:ddx66@yahoo.com] > > > > Enviado el: Viernes, 29 de Julio de 2005 11:16 > > > > Para: serusers(a)lists.iptel.org > > > > Asunto: [Serusers] LCR on onsip.org - No DB > tables > > > created? > > > > > > > > > > > > Hi, I downloaded and installed the ser-0.9.3 > from > > > > onsip.org with the intention of using LCR. > > > Although, > > > > the LCR module was complied, I do not see the > > > tables > > > > lcr, gw in the DB? How do I create them? > > > > Thanks > > > > > > > > > __________________________________________________ > > > > Do You Yahoo!? > > > > Tired of spam? Yahoo! Mail has the best spam > > > protection around > > > > http://mail.yahoo.com > > > > > > > > > _______________________________________________ > > > > Serusers mailing list > > > > serusers(a)lists.iptel.org > > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > __________________________________________________ > > Do You Yahoo!? > > Tired of spam? Yahoo! Mail has the best spam > protection around > > http://mail.yahoo.com > > > ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs [View Less]

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[Serusers] Parrallel forking using SER and Mediaproxy
by Ramu Yadav 30 Jul '05

30 Jul '05

Hi all, I am experiencing some problems while configuring SER in order to fork between different contacts (a SIP UA and a SIP Gateway) and passing through a mediaproxy: The scenario is the following: (3 calls) UA1 -> SER : INVITE sip:ramu at xyz.com SER forks: SER -> UA2 : INVITE sip:ramu@[local-ip] {SDP is changed by mediaproxy module} SER -> GW: INVITE sip:PSTN-NUMBER@[gateway-ip] {SDP is changed by mediaproxy module} SER -> GW: INVITE sip:PSTN-NUMBER@[gateway-ip] {… [View More]

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[Users] feature request
by Klaus Darilion 30 Jul '05

30 Jul '05

Hi! In many situations it would be useful to verify the destination IP address of a message before sending the message. E.g. I want to make sure that accounting and authentication was applied before sending a request to a gateway. loose_route and DNS resolving make it impossible to handle all scenarios in the config script. Thinking about the problem I got the following idea: The tm module should, after all DNS lookups, verify the dest. IP against a blacklist (e.g. the GW table from … [View More]

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[Serusers] jabber to simple transport
by David Everly 30 Jul '05

30 Jul '05

I may not have the subject right, but I'm looking for how to do the following: I have a jabber client and a jabber server on localhost. Everyone else in the company uses sip/simple by connecting Messenger to olcs.company.com. I would like to make it so that my jabber client can browse to or discover a transport that will behave as if it is a Messenger client to olcs.company.com. Unfortunately, I don't know the correct vocabulary to even use to express this idea what I have read doesn't seem … [View More]

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[Users] debian package conflict with ser
by Klaus Darilion 29 Jul '05

29 Jul '05

Hi! ser and openser debian packages can't be installed on the same machine as the gen_ha1 utility conflicts. What about renaming it to gen_ha1_openser? regards, klaus

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[Serusers] Getting Started Workshop - Phone Help
by Mer lot 29 Jul '05

29 Jul '05

Were in the early stage of building up an initial SER configuration and find that even with the abundant information on onsip.orgs site and posts, a lot of things that can go wrong. We want to build up the full authenticated/NATed/ system specified in the onsip.orgs excellent 0.9.3 documentation and wed like to cooperate with several other people wanting to do the same in a sort of small virtual workshop. The goal is to help each other and accelerate our mutual progress and the emphasis … [View More]

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RE: [Serusers] ser.pif file not found
by Sam Lee 29 Jul '05

29 Jul '05

What did u run it as ? Root ? If not , then probably when u start ser , it cant write to /var/run/ser.pid , which is the default path and requires root permission. Sam ________________________________ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Colin Jordan Sent: Friday, July 29, 2005 5:28 PM To: serusers(a)lists.iptel.org Subject: [Serusers] ser.pif file not found I cannot control SER via serctl stop/start due to the fact that when it runs … [View More]

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RE: [Serusers] LCR on onsip.org - No DB tables created?
by Ricardo Martinez 29 Jul '05

29 Jul '05

Hello Dave. I was almost sure that the tables for LCR were defined there!, my mystake. But, i have the table definitions (i can't rememeber where i found it). You can check with Juha if there is a mistake in the tables. # # Table structure for table 'gw' # CREATE TABLE gw ( gw_name VARCHAR(128) NOT NULL, ip_addr INT UNSIGNED NOT NULL, port SMALLINT UNSIGNED, grp_id INT UNSIGNED NOT NULL, PRIMARY KEY (gw_name), KEY (grp_id) ); # # Table structure for table 'gw_grp' # CREATE … [View More]TABLE gw_grp ( grp_id INT UNSIGNED NOT NULL AUTO_INCREMENT PRIMARY KEY, grp_name VARCHAR(64) NOT NULL ); # # Table structure for table 'lcr' # CREATE TABLE lcr ( prefix varchar(16) NOT NULL, from_uri varchar(128) NOT NULL DEFAULT '%', grp_id INT UNSIGNED NOT NULL, priority TINYINT UNSIGNED NOT NULL, KEY (prefix), KEY (from_uri), KEY (grp_id) ); Regards Ricardo.- > -----Mensaje original----- > De: Dave [mailto:ddx66@yahoo.com] > Enviado el: Viernes, 29 de Julio de 2005 11:30 > Para: Ricardo Martinez; serusers(a)lists.iptel.org > Asunto: RE: [Serusers] LCR on onsip.org - No DB tables created? > > > I used exactly that script. But It did not create any > LCR related tables.!!! > > > --- Ricardo Martinez <rmartinez(a)redvoiss.net> wrote: > > > Hello Dave. > > If i'm not wrong, there is a shell script included > > in the last SER > > version to create all the DB tables related to SER. > > I think there is in : > > /sip_router/scripts/ and is called "ser_mysql.sh" > > You can try that!. > > > > Regards, > > Ricardo Martinez > > > > > > > > > -----Mensaje original----- > > > De: Dave [mailto:ddx66@yahoo.com] > > > Enviado el: Viernes, 29 de Julio de 2005 11:16 > > > Para: serusers(a)lists.iptel.org > > > Asunto: [Serusers] LCR on onsip.org - No DB tables > > created? > > > > > > > > > Hi, I downloaded and installed the ser-0.9.3 from > > > onsip.org with the intention of using LCR. > > Although, > > > the LCR module was complied, I do not see the > > tables > > > lcr, gw in the DB? How do I create them? > > > Thanks > > > > > > __________________________________________________ > > > Do You Yahoo!? > > > Tired of spam? Yahoo! Mail has the best spam > > protection around > > > http://mail.yahoo.com > > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com > [View Less]

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[Serusers] SER + PSTN problem
by Kuti Joe 29 Jul '05

29 Jul '05

Hi to all! I'm a new seruser, and I have a problem with SER and a PSTN gw what I can't solve. I have got 2 UAs, 1 SER server, 1 Cisco PSTN gw (by the side of the voip provider). All of them have public IP. Phone1 -------+ | SER --- PSTN gw | Phone2 -------+ When I call from Phone1 to Phone2 directly, it is working properly. But: If I call from Phone1 to Phone2 via the PSTN gw, then it's ringing out, and … [View More]

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RE: [Serusers] ser.pif file not found
by Sam Lee 29 Jul '05

29 Jul '05

Alternatively u could do a 'touch /var/run/ser.pid ' before doing serctl start -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Gary Reuter Sent: Friday, July 29, 2005 10:21 PM To: Colin Jordan Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] ser.pif file not found If you're using serctl to start ser, it should be setting the pid location properly... but the init.d ser script does not make ser create the pid file. The … [View More]

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RE: [Serusers] exec_msg problem
by Sam Lee 29 Jul '05

29 Jul '05

The problem is solved. Already post a reply earlier on with the solution. It was becos the function don't seem to like my syntax. Anyway, thanks for the response! -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Thomas Britis Sent: Friday, July 29, 2005 8:10 PM To: serusers(a)lists.iptel.org Subject: Re: [Serusers] exec_msg problem Do you run ser as root ? $ l /var/log/syslog -rw-r----- 1 root root 115568 2005-07-28 18:50 … [View More]/var/log/syslog If you're not root, so you can't write to /var/log/syslog. Sam Lee wrote: > Tried. It didn't work. :< > Still getting the same error. The result still came out, its just that it is followed by the error message. > > 101 INVITE : 3902 <sip:3902@x.x.x.x>;tag=5086b56866af3edo0 to > <sip:967258002@x.x.x.x> Jul 29 16:26:16 abc > /usr/local/sbin/ser[10577]: ERROR: exec_msg: cmd echo "$SIP_HF_CSEQ : > $SIP_HF_FROM to $SIP_HF_ TO" >> /var/log/syslog failed. > exit_status=-1, errno=32: Broken pipe > > > > -----Original Message----- > From: Xavier TRENTIN [mailto:xtrentin@softiphone.net] > Sent: Friday, July 29, 2005 4:11 PM > To: Sam Lee > Subject: Re: [Serusers] exec_msg problem > > Try to quote the expression you want to echo like this: > > exec_msg('echo "$SIP_HF_CSEQ : $SIP_HF_FROM to $SIP_HF_TO">> > /var/log/syslog'); > > > Xavier. > > Sam Lee a écrit: > > >>*********************** >>Pas de virus detecté dans l'attachement no filename Pas de virus >>detecté dans l'attachement no filename Pas de virus detecté dans >>l'attachement no filename >> >>Votre Mail a été vérifié par InterScan. >>***********-*********** >> >> >> >> >>---------------------------------------------------------------------- >>-- >> >>Sujet: >>[Serusers] exec_msg problem >>Expéditeur: >>"Sam Lee" <sam.lee(a)super.net.sg> >>Date: >>Fri, 29 Jul 2005 11:35:53 +0800 >>Destinataire: >><serusers(a)lists.iptel.org> >> >> >>Hi, >> >>Have anyone seen this kind of error message before : >> >>ERROR: exec_msg: cmd echo $SIP_HF_CSEQ : $SIP_HF_FROM to $SIP_HF_TO >> >>/var/log/syslog failed. exit_status=-1, errno=32: Broken pipe >> >>This command is actually in my ser.cfg where i capture this few >>environmental variables and push them into my /var/log/syslog. It >>effectively went in, but i still see this error message. >>Here is the original code in ser.cfg : >>exec_msg("echo $SIP_HF_CSEQ : $SIP_HF_FROM to $SIP_HF_TO >> >>/var/log/syslog"); >> >>Please help! >> >>Regards, >>Sam >> >>---------------------------------------------------------------------- >>-- >> >>_______________________________________________ >>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 > > -- Thomas Storino Britis TCNet Inf e Telecom LTDA _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers [View Less]

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RE: [Serusers] LCR on onsip.org - No DB tables created?
by Ricardo Martinez 29 Jul '05

29 Jul '05

Hello Dave. If i'm not wrong, there is a shell script included in the last SER version to create all the DB tables related to SER. I think there is in : /sip_router/scripts/ and is called "ser_mysql.sh" You can try that!. Regards, Ricardo Martinez > -----Mensaje original----- > De: Dave [mailto:ddx66@yahoo.com] > Enviado el: Viernes, 29 de Julio de 2005 11:16 > Para: serusers(a)lists.iptel.org > Asunto: [Serusers] LCR on onsip.org - No DB tables created? > > > Hi,… [View More]

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[Serusers] LCR on onsip.org - No DB tables created?
by Dave 29 Jul '05

29 Jul '05

Hi, I downloaded and installed the ser-0.9.3 from onsip.org with the intention of using LCR. Although, the LCR module was complied, I do not see the tables lcr, gw in the DB? How do I create them? Thanks __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] ser.pif file not found
by Colin Jordan 29 Jul '05

29 Jul '05

I cannot control SER via serctl stop/start due to the fact that when it runs via this command it is not finding a ser.pid. I have searched for ser.pid on my system and it is not there (was just making sure that serctl was not looking in the wrong spot for the PID file). Any suggestions?? DISCLAIMER: The information in this message is confidential and is legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized.If receiving in error … [View More]

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AW: [Serusers] dbtext: table not loaded. example desired
by Müller Thorsten 29 Jul '05

29 Jul '05

Hi, Thanks for your fast response! I read this readme file, but I'm stucked with it. Which file or files do I need for dbtext authentication? Is there a special name necessary? When I'm right I give the name in the ser.cfg with the following line if (!www_authorize("domain.com", "subscriber")) { In my case "subscriber". And the directory where to find the file with the following line modparam("auth_db", "db_url","dbtext:///var/dbtext/ser") in my case /var/dbtext/ser How can I use … [View More]not plain password but encrypted ones and finally are three "x" enough for ha1 and ha1b or do I need to hash some values and write them down here? Why does ser give me the reply "table not loaded", when these entries are correct meaning written after the README. How can I fill the "subscriber" file with users and passwords? Texteditor I think, because serctl is only for mysql. Thanks Thorsten Mueller mt> -----Ursprüngliche Nachricht----- mt> Von: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] mt> Gesendet: Freitag, 29. Juli 2005 10:55 mt> An: Müller Thorsten mt> Cc: serusers(a)lists.iptel.org mt> Betreff: Re: [Serusers] dbtext: table not loaded. example desired mt> mt> mt> Hello, mt> mt> On 07/29/05 11:16, Müller Thorsten wrote: mt> mt> > Hi all, mt> > mt> > I try to set up ser with dbtext as authentication basis. mt> I run in some mt> > problems because the database is not found: mt> > DBT:db_query: table not loaded. mt> > mt> > After some googling a found that Cesc mt> [cesc.santa(a)gmail.com] had also mt> > done some work with dbtext and mailed that to "serusers" mt> or "serdev" mt> > mailinglists mt> > mt> > In a mail from May and June 05 he told that he has mt> written a "serctl" mt> > for dbtext and a script to produce the necessary files. mt> > mt> > Has somebody got these things and can mail me these mt> files! I could not mt> > found them in the mailing history. mt> > mt> > Has somebody also dealed with dbtext and can give me some example mt> > files for ser.cfg and dbtext files for mt> "authentication/subscriber" and mt> > "location". mt> > mt> rhe readme file of dbtext module has some examples of table mt> structures mt> as well as a simple config file. mt> mt> http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/mod mt> ules/dbtext/README?rev=HEAD&content-type=text/vnd.viewcvs-markup mt> mt> Daniel mt> mt> > Thanks in advance mt> > mt> > Thorsten Mueller mt> > mt> > ser 0.8.14 cross-compiled for arm mt> > My ser.cfg: mt> > #################################################### mt> > ... mt> > # Added by mt for authentication with dbtext mt> > loadmodule "/usr/local/lib/ser/modules/auth.so" mt> > loadmodule "/usr/local/lib/ser/modules/auth_db.so" mt> > loadmodule "/usr/local/lib/ser/modules/dbtext.so" mt> > mt> > modparam("auth_db", "db_url","dbtext:///var/dbtext/ser") mt> > mt> > # -- auth params -- mt> > modparam("auth_db", "calculate_ha1", 1) mt> > modparam("auth_db", "password_column", "password") mt> > modparam("auth_db", "user_column", "username") mt> > modparam("auth_db", "domain_column", "domain") mt> > ... mt> > mt> > if (uri==myself) { mt> > mt> > if (method=="REGISTER") { mt> > mt> > if (!www_authorize("domain.com", "subscriber")) { mt> > www_challenge("domain.com", "0"); mt> > break; mt> > }; mt> > save("location"); mt> > break; mt> > } mt> > ... mt> > ##################################################### mt> > mt> > I have a file in /var/dbtext/ser which is called "subscriber": mt> > ##################################################### mt> > username(str) password(str) ha1(str) domain(str) ha1b(str) mt> > suser:supasswd:xxx:domain.com:xxx mt> > ##################################################### mt> > mt> >----------------------------------------------------------- mt> ------------- mt> > mt> >_______________________________________________ mt> >Serusers mailing list mt> >serusers(a)lists.iptel.org mt> >http://lists.iptel.org/mailman/listinfo/serusers mt> > mt> > mt> [View Less]

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RE: [Serusers] exec_msg problem
by Sam Lee 29 Jul '05

29 Jul '05

Tried. It didn't work. :< Still getting the same error. The result still came out, its just that it is followed by the error message. 101 INVITE : 3902 <sip:3902@x.x.x.x>;tag=5086b56866af3edo0 to <sip:967258002@x.x.x.x> Jul 29 16:26:16 abc /usr/local/sbin/ser[10577]: ERROR: exec_msg: cmd echo "$SIP_HF_CSEQ : $SIP_HF_FROM to $SIP_HF_ TO" >> /var/log/syslog failed. exit_status=-1, errno=32: Broken pipe -----Original Message----- From: Xavier TRENTIN [mailto:xtrentin@… [View More]

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[Serusers] dbtext: table not loaded. example desired
by Müller Thorsten 29 Jul '05

29 Jul '05

Hi all, I try to set up ser with dbtext as authentication basis. I run in some problems because the database is not found: DBT:db_query: table not loaded. After some googling a found that Cesc [cesc.santa(a)gmail.com] had also done some work with dbtext and mailed that to "serusers" or "serdev" mailinglists In a mail from May and June 05 he told that he has written a "serctl" for dbtext and a script to produce the necessary files. Has somebody got these things and can mail me these files! I … [View More]

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RE: [Serusers] exec_msg problem
by Sam Lee 29 Jul '05

29 Jul '05

Solved IT! Stupid error. Old : exec_msg("echo $SIP_HF_CSEQ : $SIP_HF_FROM to $SIP_HF_TO>> /var/log/syslog"); New : exec_msg("(echo $SIP_HF_CSEQ : $SIP_HF_FROM to $SIP_HF_TO)>> /var/log/syslog); That solves it! It seems like the exec_msg is not happy about the syntax that I gave it. :> Thank for all the help! Regards, Sam -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Sam Lee Sent: Friday, July 29, 2005 4:… [View More]30 PM To: Xavier TRENTIN; serusers(a)lists.iptel.org Subject: RE: [Serusers] exec_msg problem Tried. It didn't work. :< Still getting the same error. The result still came out, its just that it is followed by the error message. 101 INVITE : 3902 <sip:3902@x.x.x.x>;tag=5086b56866af3edo0 to <sip:967258002@x.x.x.x> Jul 29 16:26:16 abc /usr/local/sbin/ser[10577]: ERROR: exec_msg: cmd echo "$SIP_HF_CSEQ : $SIP_HF_FROM to $SIP_HF_ TO" >> /var/log/syslog failed. exit_status=-1, errno=32: Broken pipe -----Original Message----- From: Xavier TRENTIN [mailto:xtrentin@softiphone.net] Sent: Friday, July 29, 2005 4:11 PM To: Sam Lee Subject: Re: [Serusers] exec_msg problem Try to quote the expression you want to echo like this: exec_msg('echo "$SIP_HF_CSEQ : $SIP_HF_FROM to $SIP_HF_TO">> /var/log/syslog'); Xavier. Sam Lee a écrit: >*********************** >Pas de virus detecté dans l'attachement no filename Pas de virus >detecté dans l'attachement no filename Pas de virus detecté dans >l'attachement no filename > >Votre Mail a été vérifié par InterScan. >***********-*********** > > > > > ---------------------------------------------------------------------- > -- > > Sujet: > [Serusers] exec_msg problem > Expéditeur: > "Sam Lee" <sam.lee(a)super.net.sg> > Date: > Fri, 29 Jul 2005 11:35:53 +0800 > Destinataire: > <serusers(a)lists.iptel.org> > > > Hi, > > Have anyone seen this kind of error message before : > > ERROR: exec_msg: cmd echo $SIP_HF_CSEQ : $SIP_HF_FROM to $SIP_HF_TO >> > /var/log/syslog failed. exit_status=-1, errno=32: Broken pipe > > This command is actually in my ser.cfg where i capture this few > environmental variables and push them into my /var/log/syslog. It > effectively went in, but i still see this error message. > Here is the original code in ser.cfg : > exec_msg("echo $SIP_HF_CSEQ : $SIP_HF_FROM to $SIP_HF_TO >> > /var/log/syslog"); > > Please help! > > Regards, > Sam > >----------------------------------------------------------------------- >- > >_______________________________________________ >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 [View Less]

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[Serusers] BYE
by Sebastian Kühner 29 Jul '05

29 Jul '05

Hi! I'm connecting to a Cisco PSTN Gateway for international calls. I'm using a Port Restricted Cone NAT. I can make calls and make a hangup with MY phone, but if the callee at the other side of the gateway hangs up, I do not get the BYE-Message. That could be the problem of the Restricted Cone NAT... or someone has another idea? Thanks! Sebastian

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[Serusers] exec_msg problem
by Sam Lee 29 Jul '05

29 Jul '05

Hi, Have anyone seen this kind of error message before : ERROR: exec_msg: cmd echo $SIP_HF_CSEQ : $SIP_HF_FROM to $SIP_HF_TO >> /var/log/syslog failed. exit_status=-1, errno=32: Broken pipe This command is actually in my ser.cfg where i capture this few environmental variables and push them into my /var/log/syslog. It effectively went in, but i still see this error message. Here is the original code in ser.cfg : exec_msg("echo $SIP_HF_CSEQ : $SIP_HF_FROM to $SIP_HF_TO >> /var/… [View More]

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RE: [Serusers] Problem : Can SER process the reINVITE messages pr operly?
by Ricardo Martinez 29 Jul '05

29 Jul '05

Thanks again Alberto. Are you using the ser.cfg from onsip.org? I'm using it, and i can't make it work with reinvite to a nat'd endpoints. Can you give me some hints here?. For example what version of mediaproxy are you using?. If you're using the onsip.org configuration, Did you modified this configuration to make it work for reINVITE's? Hope you can help me, Thanks again Regards, Ricardo Martinez.- -----Mensaje original----- De: Alberto Cruz [mailto:acruz@tekbrain.com] Enviado … [View More]el: Miércoles, 27 de Julio de 2005 16:50 Para: Ricardo Martinez CC: 'serusers(a)lists.iptel.org' Asunto: Re: [Serusers] Problem : Can SER process the reINVITE messages pr operly? Importancia: Alta Yes it is working fine. Regards Ricardo Martinez wrote: Alberto. Thanks for your answer. You mean that your platform works well with the reINVITE's to a NAT'd endpoint?. Regards, Ricardo.- -----Mensaje original----- De: Alberto Cruz [ mailto:acruz@tekbrain.com <mailto:acruz@tekbrain.com> ] Enviado el: Miércoles, 27 de Julio de 2005 16:12 Para: Ricardo Martinez CC: 'Greger V. Teigre'; ' serusers(a)lists.iptel.org <mailto:serusers@lists.iptel.org> ' Asunto: Re: [Serusers] Problem : Can SER process the reINVITE messages pr operly? Importancia: Alta Ricardo this block is working fine to me: #------------------------------------------------- # subsequent messages withing a dialog should take the # path determined by record-routing #------------------------------------------------- if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); # The following lines are added due media proxy if (has_totag()&&(method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { # Mark as NAT'ed setflag(6); use_media_proxy(); }; }; if ((method=="BYE") && ((client_nat_test("3") || search("^Route:.*;nat=yes")))) { fix_nated_contact(); force_rport(); }; # end media session for BYE and CANCEL is done above # before entering the loose route. no need to call it here route(1); break; }; Maybe you have something wrong at your mediaproxy.ini file. Regards Alberto Cruz [View Less]

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RE: [Serusers] i can't start ser could somebody help~~
by Sam Lee 29 Jul '05

29 Jul '05

It would help if you check either /var/log/messages or /var/log/syslog for error message pertaining to SER. Regards, Sam ________________________________ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of ? ? Sent: Friday, July 29, 2005 2:25 PM To: serusers(a)lists.iptel.org Subject: [Serusers] i can't start ser could somebody help~~ hi the list! i have installed ser 0.8.12 and mysql,mysql is fine,but when restart ser then it appears 4 errors without … [View More]saying why.i just did only a few modifies to ser.cfg ,this is the processing statements.could somebody tell me there the problem is.thanks very much and forgive my terrible english:) ot@localhost root]# ps -A PID TTY TIME CMD 1 ? 00:00:04 init 2 ? 00:00:00 ksoftirqd/0 3 ? 00:00:00 events/0 4 ? 00:00:00 kblockd/0 6 ? 00:00:00 khelper 5 ? 00:00:00 khubd 7 ? 00:00:00 pdflush 8 ? 00:00:00 pdflush 10 ? 00:00:00 aio/0 9 ? 00:00:00 kswapd0 117 ? 00:00:00 kseriod 156 ? 00:00:00 kjournald 1205 ? 00:00:00 kjournald 1206 ? 00:00:00 kjournald 1578 ? 00:00:00 syslogd 1582 ? 00:00:00 klogd 1603 ? 00:00:00 portmap 1623 ? 00:00:00 rpc.statd 1650 ? 00:00:00 rpc.idmapd 1744 ? 00:00:00 smartd 1754 ? 00:00:00 acpid 1766 ? 00:00:00 cupsd 1954 ? 00:00:00 sshd &n bsp;1969 ? 00:00:00 xinetd 1988 ? 00:00:00 sendmail 1997 ? 00:00:00 sendmail 2008 ? 00:00:00 gpm 2030 ? 00:00:00 crond 2079 ? 00:00:00 xfs 2098 ? 00:00:00 atd 2117 ? 00:00:00 dbus-daemon-1 2133 ? 00:00:00 mdadm 2159 tty1 00:00:00 mingetty 2165 tty2 00:00:00 mingetty 2171 tty3 00:00:00 mingetty 2177 tty4 00:00:00 mingetty 2183 tty5 00:00:00 mingetty 2189 tty6 00:00:00 mingetty 2195 ? 00:00:00 gdm-binary 2382 ? 00:00:00 gdm-binary 2393 ? 00:03:22 X 2423 ? 00:00:00 gnome-session 2485 ? 00:00:00 ssh-agent 2495 ? 00:00:01 gconfd-2 2498 ? 00:00:00 gnome-keyring-d 2500 ? 00:00:00 bonobo-activati 2502 ? 00:00:00 gnome-settings- 2507 ? 00:00:00 fam 2599 ? 00:00:01 metacity 2603 ? 00:00:01 gnome-panel ;2605 ? 00:00:02 nautilus 2607 ? 00:00:01 magicdev 2609 ? 00:00:00 eggcups 2611 ? 00:00:00 pam-panel-icon 2613 ? 00:00:11 rhn-applet-gui 2614 ? 00:00:00 pam_timestamp_c 2617 ? 00:00:00 gnome-vfs-daemo 2620 ? 00:00:00 clock-applet 2648 ? 00:00:00 mapping-daemon 2650 ? 00:00:00 notification-ar 2652 ? 00:00:00 mixer_applet2 2659 ? 00:00:00 wnck-applet 2661 ? 00:00:59 gnome-terminal 2662 ? 00:00:00 gnome-pty-helpe 2663 pts/1 00:00:01 bash 4506 pts/1 00:00:00 vsftpd 8355 pts/1 00:00:01 find 8871 pts/1 00:00:00 safe_mysqld 8895 pts/1 00:00:00 mysqld 9017 pts/2 00:00:00 bash 12474 pts/1 00:00:00 ps __________________________________________________ 赶快注册雅虎超大容量免费邮箱? http://cn.mail.yahoo.com [View Less]

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[Serusers] i can't start ser could somebody help~~
by 皮面 29 Jul '05

29 Jul '05

hi the list! i have installed ser 0.8.12 and mysql,mysql is fine,but when restart ser then it appears 4 errors without saying why.i just did only a few modifies to ser.cfg ,this is the processing statements.could somebody tell me there the problem is.thanks very much and forgive my terrible english:) ot@localhost root]# ps -A PID TTY TIME CMD 1 ? 00:00:04 init 2 ? 00:00:00 ksoftirqd/0 3 ? 00:00:00 events/0 4 ? 00:00:00 kblockd/0 6 ? … [View More] 00:00:00 khelper 5 ? 00:00:00 khubd 7 ? 00:00:00 pdflush 8 ? 00:00:00 pdflush 10 ? 00:00:00 aio/0 9 ? 00:00:00 kswapd0 117 ? 00:00:00 kseriod 156 ? 00:00:00 kjournald 1205 ? 00:00:00 kjournald 1206 ? 00:00:00 kjournald 1578 ? 00:00:00 syslogd 1582 ? 00:00:00 klogd 1603 ? 00:00:00 portmap 1623 ? 00:00:00 rpc.statd 1650 ? 00:00:00 rpc.idmapd 1744 ? 00:00:00 smartd 1754 ? 00:00:00 acpid 1766 ? 00:00:00 cupsd 1954 ? 00:00:00 sshd 1969 ? 00:00:00 xinetd 1988 ? 00:00:00 sendmail 1997 ? 00:00:00 sendmail 2008 ? 00:00:00 gpm 2030 ? 00:00:00 crond 2079 ? 00:00:00 xfs 2098 ? 00:00:00 atd 2117 ? 00:00:00 dbus-daemon-1 2133 ? 00:00:00 mdadm 2159 tty1 00:00:00 mingetty 2165 tty2 00:00:00 mingetty 2171 tty3 00:00:00 mingetty 2177 tty4 00:00:00 mingetty 2183 tty5 00:00:00 mingetty 2189 tty6 00:00:00 mingetty 2195 ? 00:00:00 gdm-binary 2382 ? 00:00:00 gdm-binary 2393 ? 00:03:22 X 2423 ? 00:00:00 gnome-session 2485 ? 00:00:00 ssh-agent 2495 ? 00:00:01 gconfd-2 2498 ? 00:00:00 gnome-keyring-d 2500 ? 00:00:00 bonobo-activati 2502 ? 00:00:00 gnome-settings- 2507 ? 00:00:00 fam 2599 ? 00:00:01 metacity 2603 ? 00:00:01 gnome-panel 2605 ? 00:00:02 nautilus 2607 ? 00:00:01 magicdev 2609 ? 00:00:00 eggcups 2611 ? 00:00:00 pam-panel-icon 2613 ? 00:00:11 rhn-applet-gui 2614 ? 00:00:00 pam_timestamp_c 2617 ? 00:00:00 gnome-vfs-daemo 2620 ? 00:00:00 clock-applet 2648 ? 00:00:00 mapping-daemon 2650 ? 00:00:00 notification-ar 2652 ? 00:00:00 mixer_applet2 2659 ? 00:00:00 wnck-applet 2661 ? 00:00:59 gnome-terminal 2662 ? 00:00:00 gnome-pty-helpe 2663 pts/1 00:00:01 bash 4506 pts/1 00:00:00 vsftpd 8355 pts/1 00:00:01 find 8871 pts/1 00:00:00 safe_mysqld 8895 pts/1 00:00:00 mysqld 9017 pts/2 00:00:00 bash 12474 pts/1 00:00:00 ps __________________________________________________ 赶快注册雅虎超大容量免费邮箱? http://cn.mail.yahoo.com [View Less]

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[Serusers] SER + SERNAT + MySQL
by Truong Chi Thanh 29 Jul '05

29 Jul '05

Hi all, My name is chithanh. I'm working at Vietnam. Now I'm using with SER. I have a case with SER. | ser database | | | ============================ || || | SER (5060) | | SERNAT + nathelper + rtpproxy (5082) | || || | user1 | | user2 | … [View More] Explain : user1 registered SER ===> SER save("location") for user1 into table "location" of "ser" database. user2 registered SERNAT ===> SER save("location") for user2 into table "location" of "ser" database. My purpose : user1 can call for user2. user can call for user1. Real when I run with this case. User1 can't call for User2. (although i had setted db_mode=1 for both) because : + User1 INVITE User2 ==> SER lookup("location") into SER's memory. Not lookup in ser database. + Use2 INVITE User1 ==> SERNAT lookup("location") into SERNAT's memory. Not lookup in ser database. I want SER and SENAT when lookup("location"), they must lookup in table "location" of ser database. How do u to solve for this problem ? Please help me. =================================== My config of SERNAT : # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=yes # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5082 listen=abc.com alias=abc.com children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database 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/mysql.so" loadmodule "/usr/local//lib/ser/modules/nathelper.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local//lib/ser/modules/auth.so" loadmodule "/usr/local//lib/ser/modules/auth_db.so" #loadmodule "/usr/local//lib/ser/modules/auth_radius.so" loadmodule "/usr/local//lib/ser/modules/acc.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", 1) modparam("usrloc", "db_url", "sql://ser:heslo@localhost/ser") # -- auth params -- # 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) # Accounting modparam("acc", "log_level", 2) #modparam("acc", "log_flag", 3) #modparam("acc", "log_missed_flag", 3) modparam("acc", "db_flag", 3) modparam("acc", "db_missed_flag", 3) # ------------------------- 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("chinhnam.com","subscriber")) { www_challenge("chinhnam.com", "0"); break; }; fix_nated_contact(); save("location"); break; }; #End if(method=="REGISTER") if (method=="INVITE") { if (proxy_authorize("chinhnam.com","subscriber")) { if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }else { proxy_challenge("chinhnam.com", "0"); break; }; #End if (proxy_authorize("chinhnam.com", "subcriber")) }; #End if(method=="INVITE") fix_nated_contact(); force_rtp_proxy(); t_on_reply("1"); #setflag(3); }; #End if(uri=="myself") # 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(); }; } #End main routing logic onreply_route[1] { fix_nated_contact(); if (status=~"[12][0-9][0-9]") { force_rtp_proxy(); } } ============================================================================ My config of SER : # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=yes # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 listen=chinhnam.com alias=chinhnam.com children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database 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/mysql.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local//lib/ser/modules/auth.so" loadmodule "/usr/local//lib/ser/modules/auth_db.so" #loadmodule "/usr/local//lib/ser/modules/auth_radius.so" loadmodule "/usr/local//lib/ser/modules/acc.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", 1) modparam("usrloc", "db_url", "sql://ser:heslo@localhost/ser") # -- auth params -- # 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) # Accounting modparam("acc", "log_level", 2) #modparam("acc", "log_flag", 3) #modparam("acc", "log_missed_flag", 3) modparam("acc", "db_flag", 3) modparam("acc", "db_missed_flag", 3) # ------------------------- 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("chinhnam.com","subscriber")) { www_challenge("chinhnam.com", "0"); break; }; save("location"); break; }; #End if(method=="REGISTER") if (method=="INVITE") { if (proxy_authorize("chinhnam.com","subscriber")) { if(uri=~"sip:63[0-9]*@") { rewritehostport("192.168.200.248:5060"); }else if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }else { proxy_challenge("chinhnam.com", "0"); break; }; #End if (proxy_authorize("chinhnam.com", "subcriber")) }; #End if(method=="INVITE") #setflag(3); }; #End if(uri=="myself") # 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(); }; } #End main routing logic ============================================================================= Note: SER (5060) only signaling SIP for my user. Not use media proxy. SER (5082) : signaling SIP + RTP path for my user. Use nathelper + rtpproxy Thanks very much everything you had done for me. Truong Chi Thanh --------------------------------- Start your day with Yahoo! - make it your home page [View Less]

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[Serusers] SIP Service Providers supporting Windows Messenger 4.6
by Michael Kendall 28 Jul '05

28 Jul '05

Hi, Reading the SERusers digest for a few months has allowed me to read about a lot of very complex and specialized problems. It's a bit intimidating. Although I might some day want to learn to administer a Linux system and run my own SER server, at this point I would rather just become a paying customer on someone else's SER server. Many of the list participants are clearly professionals operating large SIP service providers. But none of these professionals have identified their … [View More]organizations clearly enough to let me become their customer. Perhaps because the traditions of the list prohibit users from advertising their commercial services. Can someone please give me a recommendation of a SIP service provider which has good service and reasonable pricing. I need a service which will allow me to call between two computers, each running Windows Messenger 4.6/4.7 . I am aware that the latest version, Windows Messenger 5.1 does not allow connecting to third party service providers, and that is the reason that I am using the older versions. The Microsoft site does not tell me how to configure version 5.1 for STUN or for symmetric NAT. Every link that I have found for downloading version 4.6 points me back to the download page at microsoft.com, and microsoft has actually removed the file so all I get from them is a polite apology. Fair enough. So here is an interesting legal question that I suspect a lot of you SER wizards will know the answer to: Is it illegal for third parties sites to publicly offer the download of version 4.6 after microsoft has chosen to stop offering it? It was after all being given away free by Microsoft a short time ago. I am aware that freeworlddialup.com (FWD) is a good no-cost SIP service provider. Does FWD use SER as their server software? The latest version of FWD offers several options for User Agent software, but none of these UAs include a whiteboard feature, which I need for my tutoring service. Therefore, I need a SIP Service Provider which supports Windows Messenger 4.6/4.7 and has some configuration instructions for this particular UA. Hopefully the config instructions will tell me how to set Messenger up for STUN when I have a standard NAT, and for mediaproxy when I encounter a symmetric NAT. Googling for "SIP Service Provider" and "Windows Messenger 4.6" has led me to a lot of dead ends and out-of-date pages, such as the "old FWD site" at pulver.com which has detailed instructions for configuring Messenger 4.6, but the instructions no longer work. "Service provider temporarily unavailable". (permanently) Can anybody offer a Service Provider recommendation or volunteer to take me on as a customer? thanks Michael Kendall [View Less]

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[Serusers] Dest IP
by Sebastian Kühner 28 Jul '05

28 Jul '05

Hello, I need the destination IP of the user of the first INVITE message (user -> ser)... but dst_ip == xxx.xxx.xxx.xxx gives me the IP of ser. The problem is that I only want to force the rtp_proxy if the users are behind a Port restricted cone NAT (the scr-ips of them are known) Here is my code: # Force rtpproxy for users that are behind a Port Restricted Cone NAT if ((src_ip == xxx.xxx.xxx.xxx | dst_ip == xxx.xxx.xxx.xxx) && !(uri=~"sip:[0-9]{9,30}@.*")) { … [View More]

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[Users] new features in development version
by Daniel-Constantin Mierla 28 Jul '05

28 Jul '05

Hello, the new features of OpenSER development version are detailed a bit in: http://openser.org/features-0.10.x.php If you find something missing or you want to add more text, please send a mail. Hope it will convince some new users to start testing it. Daniel

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[Users] typos in documentation and use_domain question
by Klaus Darilion 28 Jul '05

28 Jul '05

Hi! In the uri_db documentation (all parameters): modparam("db_url", ".... ^^^^^^ shouldn't this be uri_db? In the group_radius documentation: modparam("group", ... ^^^^^^ shouldn't this be group_radius? I also wonder about the use_domain paramter in the radius modules. group_radius has this parameter, whereas all other radius modules (uri_radius, avp_radius, auth_radius) do not have this parameter? Is there a reason for this? regards, klaus

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[Serusers] Over-long fields can crash SER with PostgreSQL
by Karl A. Krueger 28 Jul '05

28 Jul '05

We are using SER with the PostgreSQL module, so that SER and our Asterisk PSTN gateway and voice mail system can use the same back-end database for user records. (Anyone who's interested in how we did this, let me know -- it's still a little shaky, but it pretty much works.) Recently we have encountered a problem: it seems that if any database error occurs, SER hangs or crashes. We have so far been able to mostly work around these conditions by being Really Careful with the database, but it … [View More]

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Re: [Serusers] asterisk and ser combine, 407 proxy authentucation required !
by Iqbal 28 Jul '05

28 Jul '05

tell ser that all packets from asterisk IP dont need to be authenticated, use trusted_ip setup for it, that should do the trick Iqbal On 7/28/2005, "zhaomin" <zhaomin(a)kingdream.com> wrote: >hello! > I want to realized asterisk+ser combine working,ser is working fine and asterisk working well I think,I configure sip.conf and extensions.conf voicemail.conf as the asterisk handbook ,but when I registered 2000 to dial 2001 but I get error :407 proxy authentucation required, &… [View More]gt; where is problem in asterisk or ser ? >blow is the log of x-lite diagnostic,Very appreciation anybody help! >SEND TIME: 201758109 >SEND >> 221.202.129.167:5060 >INVITE sip:2001@netvoip.cn SIP/2.0 >Via: SIP/2.0/UDP 10.1.16.34:5060;rport;branch=z9hG4bK1DD16D11800A4C6686A65DD7C5662FFF >From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 >To: <sip:2001@netvoip.cn> >Contact: <sip:2000@10.1.16.34:5060> >Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 >CSeq: 53903 INVITE >Max-Forwards: 70 >Content-Type: application/sdp >User-Agent: X-Lite release 1103m >Content-Length: 244 > >v=0 >o=2000 201757640 201758109 IN IP4 10.1.16.34 >s=X-Lite >c=IN IP4 10.1.16.34 >t=0 0 >m=audio 8000 RTP/AVP 0 98 97 101 >a=rtpmap:0 pcmu/8000 >a=rtpmap:98 iLBC/8000 >a=rtpmap:97 speex/8000 >a=rtpmap:101 telephone-event/8000 >a=fmtp:101 0-15 > >RECEIVE TIME: 201758859 >RECEIVE << 221.202.129.167:5060 >SIP/2.0 407 Proxy Authentication Required >Via: SIP/2.0/UDP 10.1.16.34:5060;rport=5339;branch=z9hG4bK1DD16D11800A4C6686A65DD7C5662FFF >From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 >To: <sip:2001@netvoip.cn>;tag=b83deba332858fe2da009af570355a70.3a3b >Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 >CSeq: 53903 INVITE >Proxy-Authenticate: Digest realm="netvoip.cn", nonce="42e896c750edabc9057e78d345379a4991b02378" >Server: Sip EXpress router (0.10.99-dev7 (i386/linux)) >Content-Length: 0 >Warning: 392 221.202.129.167:5060 "Noisy feedback tells: pid=28404 req_src_ip=61.136.210.7 req_src_port=5339 in_uri=sip:2001@netvoip.cn out_uri=sip:2001@netvoip.cn via_cnt==1" > > >SEND TIME: 201758859 >SEND >> 221.202.129.167:5060 >ACK sip:2001@netvoip.cn SIP/2.0 >Via: SIP/2.0/UDP 10.1.16.34:5060;rport;branch=z9hG4bK1DD16D11800A4C6686A65DD7C5662FFF >From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 >To: <sip:2001@netvoip.cn>;tag=b83deba332858fe2da009af570355a70.3a3b >Contact: <sip:2000@10.1.16.34:5060> >Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 >CSeq: 53903 ACK >Max-Forwards: 70 >Content-Length: 0 > > >SEND TIME: 201758906 >SEND >> 221.202.129.167:5060 >INVITE sip:2001@netvoip.cn SIP/2.0 >Via: SIP/2.0/UDP 10.1.16.34:5060;rport;branch=z9hG4bK20B4A3F981D6492C9CA601C9D241F095 >From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 >To: <sip:2001@netvoip.cn> >Contact: <sip:2000@10.1.16.34:5060> >Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 >CSeq: 53904 INVITE >Proxy-Authorization: Digest username="2000",realm="netvoip.cn",nonce="42e896c750edabc9057e78d345379a4991b02378",response="337cb9deeceb1bf01e9768b6a5452568",uri="sip:2001@netvoip.cn" >Max-Forwards: 70 >Content-Type: application/sdp >User-Agent: X-Lite release 1103m >Content-Length: 244 > >v=0 >o=2000 201757640 201758109 IN IP4 10.1.16.34 >s=X-Lite >c=IN IP4 10.1.16.34 >t=0 0 >m=audio 8000 RTP/AVP 0 98 97 101 >a=rtpmap:0 pcmu/8000 >a=rtpmap:98 iLBC/8000 >a=rtpmap:97 speex/8000 >a=rtpmap:101 telephone-event/8000 >a=fmtp:101 0-15 > >RECEIVE TIME: 201759640 >RECEIVE << 221.202.129.167:5060 >SIP/2.0 100 trying -- your call is important to us >Via: SIP/2.0/UDP 10.1.16.34:5060;rport=5339;branch=z9hG4bK20B4A3F981D6492C9CA601C9D241F095 >From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 >To: <sip:2001@netvoip.cn> >Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 >CSeq: 53904 INVITE >Server: Sip EXpress router (0.10.99-dev7 (i386/linux)) >Content-Length: 0 >Warning: 392 221.202.129.167:5060 "Noisy feedback tells: pid=28398 req_src_ip=61.136.210.7 req_src_port=5339 in_uri=sip:2001@netvoip.cn out_uri=sip:2001@221.202.129.167:5061 via_cnt==1" > > >RECEIVE TIME: 201759671 >RECEIVE << 221.202.129.167:5060 >SIP/2.0 407 Proxy Authentication Required >Via: SIP/2.0/UDP 10.1.16.34:5060;branch=z9hG4bK20B4A3F981D6492C9CA601C9D241F095 >From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 >To: <sip:2001@netvoip.cn>;tag=as6e0c8cea >Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 >CSeq: 53904 INVITE >User-Agent: Asterisk PBX >Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER >Contact: <sip:2001@221.202.129.167:5061> >Proxy-Authenticate: Digest realm="asterisk", nonce="43fb4632" >Content-Length: 0 > > >SEND TIME: 201759671 >SEND >> 221.202.129.167:5060 >ACK sip:2001@netvoip.cn SIP/2.0 >Via: SIP/2.0/UDP 10.1.16.34:5060;rport;branch=z9hG4bK20B4A3F981D6492C9CA601C9D241F095 >From: 2000 <sip:2000@netvoip.cn>;tag=3731060067 >To: <sip:2001@netvoip.cn>;tag=as6e0c8cea >Contact: <sip:2000@10.1.16.34:5060> >Call-ID: E3160ED2-7745-4740-8E69-52E6710AF5A2(a)10.1.16.34 >CSeq: 53904 ACK >Max-Forwards: 70 >Content-Length: 0 > > [View Less]

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[Serusers] SER Asterisk Voicemail - Message Playback
by Aisling 28 Jul '05

28 Jul '05

Hi Everyone, Unfortunately when I was upgrading my version of SER to 0.9.0 I deleted my config file which contained code for forwarding to asterisk voicemail.Therefore I am attempting to set up all over again (Doh! I know!) :-( Anyhow SER seems to be correctly forwarding to Asterisk, the user can leave a message if the phone isn't picked up after 3 rings and then the message is sent to the users email account. I want the user to also be able to listen to their voicemails from the end user … [View More]device. It is my understanding from what I have coded, that user should be able to dial 9999 and when SER receives this it sends it to Asterisk' VoiceMailMain where the user will be asked for an account name and password. However there are problems when the user dials 9999. The user can hear nothing on the phone. Errors appear on both the SER screen and Asterisk screen so I am not sure what the problem is: On SER it says: Warning: sl_send_reply: I won't send a reply for ACK!! Forwarding to asterisk14(27355) contact_parser(): Empty body Parse_contact(): Error while parsing Get_contact_uri: Error while parsing Contact body On Asterisk it says: -- Executing VoiceMailMain("SIP/2092-dd89", "2092") in new stack -- Playing 'vm-login' (language 'en') WARNING: app_voicemail.c:3333 vm_execmain: Couldn't read username Spawn extension (test, 9999, 1) exited non-zero on 'SIP/2092-c660' The relevant parts of my ser.cfg and asterisk config files are shown below. Many thanks, Aisling. ------------------------------------------------------------------------ ----------------------------------- ser.cfg ------------------------------------------------------------------------ ---------------------------------- #Call Type Processing Section if(uri==myself){ if(uri=~"^sip:9[0-9]*@serveraddress"){ log(1, "forwarding to asterisk"); rewritehostport("serveraddress:port"); append_branch(); t_relay_to_udp("serveraddress", "port"); break; } if (method == "INVITE"){ t_on_failure("1"); route(3); break; } failure_route[1]{ revert_uri(); rewritehostport("serveraddress:port"); append_branch(); t_relay_to_udp("serveraddress", "port"); break(); } ------------------------------------------------------------------------ -------------------------------------- sip.conf ------------------------------------------------------------------------ ------------------------------------- [general] context=test port=5062 bindaddr=0.0.0.0 srvlookup=yes [2092] type=friend username=2092 canreinvite=no context=test mailbox=2092 host=dynamic nat=no dtmfmode=info disallow=all allow=alaw allow=ulaw [314] type=friend username=314 canreinvite=no context=test mailbox=314 host=dynamic nat=no dtmfmode=info disallow=all allow=alaw allow=ulaw ------------------------------------------------------------------------ ----------------------------------- extensions.conf ------------------------------------------------------------------------ ---------------------------------- [test] ;leave voice messages exten => 2092, 1, Voicemail(u2092) exten => 2092, 2, Hangup exten => 314, 1, Voicemail(u314) exten => 314, 2, Hangup ;play voice messages exten => 9999, 1, VoiceMailMain, 2092 exten => 9999, 2, Hangup exten => 9999, 1, VoiceMailMain, 314 exten => 9999, 2, Hangup ------------------------------------------------------------------------ ------------------------------------- voicemail.conf ------------------------------------------------------------------------ -------------------------------------- [default] 2092 => 1234, 2092, emailaddress 314 => 1234, 314, emailaddress -------------------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. [View Less]

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RE : RE : [Serusers] forwarding and accounting
by Olivier Taylor 28 Jul '05

28 Jul '05

Sorry guy, u think I don't contribute, that's ok, but I try helping as much as I can. Respect for the job done by the Ser team, but sorry, I and many of us don't understand why we have now two Ser...(an open one and an open one...) If u really want to contribute, don't fork, just contribute, otherwise it will be another software. If you don't want people use your work, go and join the Microsoft team and forget free software. That's only the poor opinion of a guy using unfairly free software. … [View More]

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[Serusers] forwarding and accounting
by Olivier Taylor 28 Jul '05

28 Jul '05

hi all, what about the accounting when forwarding a call. A-->B--ForwardTo-->C Forwarding is OK, even to Pstn, but... B as to pay the forwarding, but i don't see any record on acc neither radacct concerning this call for B. Any Idea? Olivier

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[Users] timer question
by Klaus Darilion 28 Jul '05

28 Jul '05

Hi! What should happen if the invite_timer expires and there is no failure route defined? In my case happens nothing, the phone is still ringing. Shouldn't openser (CVS) cancel the call and responds with 408? If I define a failure_route, everything works fine. regards, klaus debuglog: 9(21752) DEBUG: reply relayed. buf=0x81148d0: SIP/2.0 1..., shmem=0x40606568: SIP/2.0 1 9(21752) reply_received: FR_INV_TIMER = 10 9(21752) DEBUG: add_to_tail_of_timer[1]: 0x40604d7c 9(21752) … [View More]

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[Serusers] Presence and messaging: UA with SER
by Cameron Beattie 28 Jul '05

28 Jul '05

I want to be able to offer the following functionality (in addition to calling) to my users namely: notification when a user is on/offline, directory search, and chat. These simple requirements are met with ease by Skype. And yet there doesn't seem to be any combination of SER and a freely-available Windows user agent that offers this. I have been a little bit out of touch with this recently so things may have moved on since I last checked. The main problem seems to be on the UA side. I'… [View More]

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