- sr-users - kamailio.org

[Serusers] Re: Sip Proxy behind Nat.
by ravi reddy 07 Sep '06

07 Sep '06

Hi Greger, I found the exact problem which have been paining me. 1) the nated Sip-proxy; its sending the route headers as if it forgets it staying behind the Nat.andsending the call as some thing like this invite 99106883(a)81.21.33.35 ----this is my SER - server record<192.168.1.100>------- this is nated sip-proxy address contact<192.168.1.100>-----contact to this address only SER-server is frowarding the call to pstn Gateway as it recieves and just rewriting the host name. and the other problem(I think) 2) PSTN Gateway. even when call connects it is adding the strict route headers as it recieves from SER-server and when it sends 200 0k after connecting the call it is not getting back or SER-server is also not getting back the "ACK" from the nated proxy because the nated proxy itself add the contact header as (private-ip ) ofcourse it is following strictly the SIP-PROTOCOL so,these are the two problems i found , Some providers are providing calls to this nated sip-proxy -------> How ???? Why the SER-server is also not following the path where "INVITE" message came ????? How to stop the Nated Sip-proxy to stop sending the headers in "INVITE"message not including the record route or contact headers in the message --- ?????? Or is there any way to do in SER-server itself ( majorly prefered) not to contact private ip-address and to follow the next address which is NAT-IP ---??????? (I think its foolish question )---> Or is there any way to get registered another sip-proxy in location database.----??????? Please suggest me some thing to get out of this problem. Thank You. Regards, Ravi.

2 1

[Serusers] About the failure route problem
by AR. 07 Sep '06

07 Sep '06

Hi all I have a problme about the failure route My scenario is that: If voice gateway 1 is busy then forward the call to the voice gateway 2 My config is : route[5] { rewritehostport("voice_gateway_1:5060"); forward(uri:host, uri:port); t_on_failure("2"); if(isflagset(1)) append_branch(); setflag(1); t_relay(); break; } failure_route[2] { route(6); } route[6] { rewritehostport("voice_gateway_2:5060"); forward(uri:host, uri:port); break; } But it still doesn't work anyone help? Best Regards AR.

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[Serusers] Cancel 183 progress (result of forked invite)?
by Vamsi Pottangi 07 Sep '06

07 Sep '06

Hi, In my test setup, I have SER as proxy and SEMS as announcement server. In SER config I'm sending forking invite to SEMS using append_branch for every call made. For example when User1 dials User2, SER send INVITES to both SEMS and User2, while the User2 phone rings, announcement is played from SEMS to User1 using the early media 183 session progress message. Once the User2 answers the call, a CANCEL message is sent to SEMS and normal conversation happens between User1 and User2 which is expected. The problem which I'm facing is that when User2 doesn't answer the call or if rejects the call, SER never sends CANCEL to SEMS, so the annoucement is being played continously to User1 and User1 is unaware of the remote party status. How can I configure the SER script to handle such unanswered or call reject cases and send CANCEL to SEMS and inform User1 about the unanswered or cancel status? Does it cancel 183 progress only in case of 200 OK from other invite session? how can we handle others like 408, 486 etc? I'm not pasting my ser.cfg here because it is the basic sample script with just append_branch("SEMS") in case of INVITE. Did anybody tried out such scenarios? Any help would be appreciated. Thanks, ~Vamsi

2 1

[Serusers] SER as a system service
by vijay tiwari 07 Sep '06

07 Sep '06

hi all i want to start SER as the system Service Vijay Tiwari Network Admin Sachitel Communications Noida(U.P.) M- 09871633711 --------------------------------- Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail.

2 1

[Serusers] CPU %
by Teran Jesus 07 Sep '06

07 Sep '06

Hello i am using the ser 0.8.12, it works fine, right now i run the linux comand ps -aux , to see the list of proces that are runing and i see that i have SER proces like the next, can you said me if it is correct or i have a problem with my ser or is a problem with the version of the SER, because sometimes the SER needs to reset because don't respond, if you see the last SER reset tht i need to made was August 25. USER PID %CPU %MEM VSZ RSS TTY STAT START TIME COMMAND root 24416 32.4 0.7 38668 7456 ? R Aug25 5547:04 /usr/sbin/ser root 24421 32.4 0.7 38684 7468 ? R Aug25 5537:54 /usr/sbin/ser root 24426 32.4 0.7 38664 7484 ? R Aug25 5541:14 /usr/sbin/ser root 24431 32.4 0.7 38652 7468 ? R Aug25 5543:54 /usr/sbin/ser root 24437 0.0 0.8 38676 9056 ? S Aug25 5:56 /usr/sbin/ser root 24439 0.0 0.8 38756 9064 ? S Aug25 6:10 /usr/sbin/ser root 24444 32.4 0.7 38684 7456 ? R Aug25 5551:00 /usr/sbin/ser root 24450 32.4 0.7 38664 7472 ? R Aug25 5539:39 /usr/sbin/ser I ran the vmstat 1 10 too and i believe is not normall the processing, can you help me?. r b swpd free buff cache si so bi bo in cs us sy id wa 6 0 27440 17824 168972 649488 0 0 1 2 0 2 0 0 1 0 6 0 27440 17844 168972 649488 0 0 0 40 130 480364 12 87 1 0 6 0 27440 17844 168972 649488 0 0 0 0 118 475900 11 89 0 0 6 0 27440 17840 168972 649488 0 0 0 0 120 481403 11 87 2 0 6 0 27440 17840 168972 649488 0 0 0 0 114 483715 12 86 2 0 6 0 27440 17836 168972 649488 0 0 0 36 134 476488 12 87 1 0 6 0 27440 17840 168972 649488 0 0 0 64 137 482903 14 84 1 0 6 0 27440 17840 168972 649488 0 0 0 68 148 473984 14 86 0 0 6 0 27440 17840 168972 649488 0 0 0 64 143 476887 12 88 0 0 6 0 27440 17840 168972 649488 0 0 0 52 147 476060 12 88 0 0

2 1

[Serusers] 415 unsopported document format
by Javier Ramirez 07 Sep '06

07 Sep '06

Hi, I have a problem, when I send a PUBLISH with personal status, if have accent, the ser send me 415 unsopported document format. How to encode for ser accept my PUBLISH ? best regards Javier

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[Serusers] alias_db module exists?
by Ricardo Carvalho 07 Sep '06

07 Sep '06

Is Ser capable to implement the same offered by alias_db module of OpenSer? Thanks, Ricardo.

2 1

[Users] acc database support
by Tracy Lofton 07 Sep '06

07 Sep '06

Hi all, I am new to OpenSER and I am trying to save acc data in mysql database but I keep getting an error whenever I dial the 976 prefix number. What do I need to do, to configure the database as there is a table for acc, and i am not quite sure of what to do now. Here is a snippet of openser.cfg file relating to the area in question: if (uri=~"^sip:919(00|76)[0-9]{7}@") { acc_db_request("403 - 900/976 Disabled", "acc"); xlog("L_INFO", "Cancel attempt to call 9xx number (not 911), db recorded\n"); exit; }; Here is the .cfg error displayed on the terminal when calling 9-1 and then the 976 prefix: 2(12906) ERROR:acc:w_acc_db_request: DB support not configured 2(12906) Cancel attempt to call 9xx number (not 911), db recorded 3(12908) ERROR:acc:w_acc_db_request: DB support not configured Any help greatly appreciated. Thanks, Tracy _________________________________________________________________ Get real-time traffic reports with Windows Live Local Search http://local.live.com/default.aspx?v=2&cp=42.336065~-109.392273&style=r&lvl…

2 1

[Serusers] acc database support
by Tracy Lofton 06 Sep '06

06 Sep '06

Hi all, I am new to OpenSER/SER and I am trying to save acc data in mysql database but I keep getting an error whenever I dial the 976 prefix number. What do I need to do, to configure the database as there is a table for acc, and i am not quite sure of what to do now. Here is a snippet of my openser.cfg file relating to the area in question: if (uri=~"^sip:919(00|76)[0-9]{7}@") { acc_db_request("403 - 900/976 Disabled", "acc"); xlog("L_INFO", "Cancel attempt to call 9xx number (not 911), db recorded\n"); exit; }; Here is the .cfg error displayed on the terminal when calling 9-1 and then the 976 prefix: 2(12906) ERROR:acc:w_acc_db_request: DB support not configured 2(12906) Cancel attempt to call 9xx number (not 911), db recorded 3(12908) ERROR:acc:w_acc_db_request: DB support not configured Any help greatly appreciated. Thanks, Tracy _________________________________________________________________ Got something to buy, sell or swap? Try Windows Live Expo ttp://clk.atdmt.com/MSN/go/msnnkwex0010000001msn/direct/01/?href=http://exp…

1 2

[Serusers] how to stop SER
by Henry Duan 06 Sep '06

06 Sep '06

Hi, Sorry for my ignorance. I tried to stop SER, which is installed at RedHat Linux 9, however there is no ser.pid file(in /proc/ or /var/run/ directories). Can anybody suggest me a good way to stop SER so that I don't need to restart the machine to stop it? thanks!

2 1

[Users] Cancel 183 progress branch (result of forked invite)
by Vamsi Pottangi 06 Sep '06

06 Sep '06

Hi, In my test setup, I have Openser as proxy and SEMS as announcement server. In Openser config I'm sending forking invite to SEMS using append_branch for every call made. For example when User1 dials User2, Openser send INVITES to both SEMS and User2, while the User2 phone rings, announcement is played from SEMS to User1 using the early media 183 session progress message. Once the User2 answers the call, a CANCEL message is sent to SEMS and normal conversation happens between User1 and User2 which is expected. The problem which I'm facing is that when User2 doesn't answer the call or if rejects the call, Openser never sends CANCEL to SEMS, so the annoucement is being played continously to User1 and User1 is unaware of the remote party status. How can I configure the Openser script to handle such unanswered or call reject cases and send CANCEL to SEMS and inform User1 about the unanswered or cancel status? Does it cancel 183 progress only in case of 200 OK from other invite session? how can we handle others like 408, 486 etc? I'm not pasting my Openser.cfg here because it is the basic sample script with just append_branch("SEMS") in case of INVITE. Did anybody try out such scenarios? Any help would be appreciated. Thanks, ~Vamsi

3 2

[Serusers] CPU %
by Teran Jesus 06 Sep '06

06 Sep '06

> Hello i am using the ser 0.8.12, it works fine, right now i run the linux > comand ps -aux , to see the list of proces that are runing and i see that > i have SER proces like the next, can you said me if it is correct or i > have a problem with my ser or is a problem with the version of the SER, > because sometimes the SER needs to reset because don't respond, if you see > the last SER reset tht i need to made was August 25. > > > USER PID %CPU %MEM VSZ RSS TTY STAT START > TIME COMMAND > root 24416 32.4 0.7 38668 7456 ? R Aug25 5547:04 > /usr/sbin/ser > root 24421 32.4 0.7 38684 7468 ? R Aug25 5537:54 > /usr/sbin/ser > root 24426 32.4 0.7 38664 7484 ? R Aug25 5541:14 > /usr/sbin/ser > root 24431 32.4 0.7 38652 7468 ? R Aug25 5543:54 > /usr/sbin/ser > root 24437 0.0 0.8 38676 9056 ? S Aug25 5:56 > /usr/sbin/ser > root 24439 0.0 0.8 38756 9064 ? S Aug25 6:10 > /usr/sbin/ser > root 24444 32.4 0.7 38684 7456 ? R Aug25 5551:00 > /usr/sbin/ser > root 24450 32.4 0.7 38664 7472 ? R Aug25 5539:39 > /usr/sbin/ser > > I ran the vmstat 1 10 too and i believe is not normall the processing, > can you help me?. > > r b swpd free buff cache si so bi bo in cs us sy > id wa > 6 0 27440 17824 168972 649488 0 0 1 2 0 2 0 0 > 1 0 > 6 0 27440 17844 168972 649488 0 0 0 40 130 480364 12 87 > 1 0 > 6 0 27440 17844 168972 649488 0 0 0 0 118 475900 11 89 > 0 0 > 6 0 27440 17840 168972 649488 0 0 0 0 120 481403 11 87 > 2 0 > 6 0 27440 17840 168972 649488 0 0 0 0 114 483715 12 86 > 2 0 > 6 0 27440 17836 168972 649488 0 0 0 36 134 476488 12 87 > 1 0 > 6 0 27440 17840 168972 649488 0 0 0 64 137 482903 14 84 > 1 0 > 6 0 27440 17840 168972 649488 0 0 0 68 148 473984 14 86 > 0 0 > 6 0 27440 17840 168972 649488 0 0 0 64 143 476887 12 88 > 0 0 > 6 0 27440 17840 168972 649488 0 0 0 52 147 476060 12 88 > 0 0 > >

1 0

[Users] Aggregating SIP devices
by Kristian Kielhofner 06 Sep '06

06 Sep '06

Hello everyone, I have multiple Asterisk systems registered to an OpenSER server. Each of them have several different SIP accessible extensions. Using alias_db (or something), I need to be able to create contacts that are reachable from the OpenSER server to individual extensions on each Asterisk system. For instance: SIP Phone -> Asterisk System -> OpenSER The SIP phone is registered to the Asterisk system and has an extension (100). The Asterisk System is registered to OpenSER and has a usrloc entry (asterisk). I would like to setup an alias like this: asterisk-100 This alias should be able to dynamically get the location of the Asterisk system from usrloc and reach that extension directly, basically forming a SIP uri like this: sip://100@[asterisk server ip address - loaded dynamically from usrloc] That way, if I were to do this: External SIP Phone -> OpenSER -> Asterisk System -> SIP Phone (ext 100) I could "dial" asterisk-100 from that External SIP phone and get connected to the Asterisk System at extension 100. This would be similar to the following (in Asterisk land): [sip-in] exten => asterisk-100,1,Dial(SIP/asterisk/100,20) exten => asterisk-100,2,Congestion(10) The most obvious way to do this is create multiple subscriber entries for each Asterisk system and one register => line for each entry in Asterisk. I would rather not have to do that!!! Is this clear enough? What do you think? Thanks! -- Kristian Kielhofner

2 1

[Users] prevent a dialled username without dialling the domain to ring a phone registered with that username
by Ricardo Carvalho 06 Sep '06

06 Sep '06

How can I prevent a dialled username (without dialling the domain) to ring a phone registered with that username? I pretend to use a multi-domain architecture where apart from aliases of users, only dialling with username@domain will ring registered phones. Thanks, Ricardo.

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[Users] Automatic INVITE when fone goes off-hook
by Mahesh Paolini-Subramanya 06 Sep '06

06 Sep '06

I know, not *directly* openser related, but hey, it is tangentially related I've got a scenario where some of my users want to 'hard-wire' their fones so that when their fone goes off-hook, it automatically sends an INVITE to a different (pre-specified) fone. And yes, they don't want to have to press a key on their fone (don't ask. i *know* its dumb. but hey, its what they want...) Has anyone seen *anything* along these lines? Any fones that do something similar? cheers -- ******************************************* Mahesh Paolini-Subramanya (703) 386-1500 x9100 CTO mahesh(a)aptela.com Aptela, Inc. http://www.aptela.com "Aptela: How Business Answers The Call" *******************************************

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Re: [Users] OpenSER Summit
by Rodrigo P. Telles 06 Sep '06

06 Sep '06

Hi Bogdan, I think its a very good idea, there are many people using OpenSER for their SIP solutions around the world including us here in Brazil. Unfortunately won't be possible for me to get VoN Berlin but I'm going to VON Boston on next 9-11 (uhhhh), perhaps I could meet the OpenSER members that are visiting or passing by Boston there or in another place. Does some one is going to VoN Boston and wants to have a beer with others OpenSER members? Best regards, -- ============================================ Rodrigo P. Telles <telles(a)devel.it> IT Manager Devel-IT - http://www.devel.it IVOZ # 1029 Ourinhos: +55 14 3324-1200 São Paulo: +55 11 2124-7474 ============================================ > > Hello everybody, > > It's more than one year since the OpenSER project started. A large community of users and developers has been formed around OpenSER. It's a very active community and one of the most important goals of the OpenSER project is to keep the users community as synchronized as possible with the developers and management groups. > > During past time we met a lot of people that were happy to see us or we were happy to see them. So, we came to the idea of organizing a OpenSER Summit where OpenSER fans (all types of fans) should meet to share their experience, to discuss about future ways of developing the project or just to personally meet people from the mailing lists. The idea is to have people talking about how they are using OpenSER in different contexts and which were/might be their needs, development people talking about possible solutions and about their ideas of enhancing OpenSER, etc... > > We were thinking of taking advantage of the upcoming VoN in Berlin in the beginning of November and to organize the Summit within the same time frame - it will be more convenient for people to travel for two events at the same time. > > We would like to have some feedback from the community - how many people are interested in such a meeting, what will be their primary interests. > > Any other OpenSER-related ideas are most welcome! > > Best regards, > Bogdan

2 1

[Users] DNS Server
by Tracy Lofton 06 Sep '06

06 Sep '06

Hello All, I am new to OpenSER/Ser and have learned a lot and received a lot of assistance from all of you. Anyway, I would like to express my gratitude by extending Free Dynamic DNS services for anyone interested in utilizing a dynamic IP address in a static capacity while learning and using your OpenSER/SER sip router and/or Asterisk server. If interested go to: http://www.t4tm.net. While visiting the site review the details regarding the FREE Dynamic DNS Service, signup, add my name servers where you registered your domain name, download the dynamic IP updater client and that's it. I hope you all enjoy this service. Tracy _________________________________________________________________ Get the new Windows Live Messenger! http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=…

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[Serusers] SIP client with video???
by Joao Pereira 06 Sep '06

06 Sep '06

Hello to all can someone recommend me a nice SIP client with video for windows?? I tried X-Lite 3.0 but it's a lousy piece of software..... Does someone knows about a better software? Regards Joao Pereira

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[Serusers] log ( history of action for one user)
by Legos Engineering 06 Sep '06

06 Sep '06

HI. I wonder, if there is some log (like messages or another) which permit to know the history of one user ( his last calls, when he was called ...). I want to know that in order to add, an history of each user in the webcare. Therefore, if someone can say me, if this log exist and where it is. Thank you.

2 1

[Users] How to redirect based on from_uri and uri
by Darren DeLitizia 06 Sep '06

06 Sep '06

Here is what I want to do. Domestic routing defining interlata and intralata based off of ani and dnis (from uri and uri). Now let's not dive too much into the whole domestic routing thing or we'll be here for the next century. Bottom line is, this what I want to do with openser. 1. Look at ani (from_uri) (NPA - first 3 digits) 2. Look at dnis (uri) - (NPA - first 3 digits) 3. If the NPA (first 3 digits) in ani is in same as the DNIS NPA then redirect 555dnis@ip_address 4. If ANI NPA is in same as DNIS NPA then redirect 999dnbis@ip_address I am redirecting so I can be and stay stateless I am able to redirect but unable to figure out correct syntax for it to look at ani and dnis then redirect appropriately. Thanks all

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[Users] crash
by Helge Waastad 06 Sep '06

06 Sep '06

Hi, in my 1.1.0 version I syddenly got a core dump. Does any one know why? (gdb) bt #0 0x000000304448660b in strftime_l () from /lib64/tls/libc.so.6 #1 0x0000002a95b29de4 in time2mysql (_time=Variable "_time" is not available. ) at utils.c:54 #2 0x0000002a95b2a731 in val2str (_c=Variable "_c" is not available. ) at val.c:147 #3 0x0000002a95b26b13 in print_values (_c=0x643648, _b=0x2a95db5269 "'SIP/2.0 100 Trying\\r\\nVia: SIP/2.0/UDP 217.17.222.4;branch=z9hG4bK1957.c42a1af6.0\\r\\nVia: SIP/2.0/UDP 10.1.93.186:10001;rport=44323;received=217.17.222.6;branch=z9hG4bK2eb74daf34599095\\r\\nFrom: \\\"Har"..., _l=65431, _v=0x7fbfffe990, _n=10) at dbase.c:144 #4 0x0000002a95b27aa4 in db_insert (_h=0x643e08, _k=Variable "_k" is not available. ) at dbase.c:495 #5 0x0000002a97c26bdc in trace_sl_onreply_out (req=Variable "req" is not available. ) at siptrace.c:1078 #6 0x0000002a956b30ad in run_sl_callbacks (req=0x643f28, buffer=Variable "buffer" is not available. ) at sl_cb.c:88 #7 0x0000002a956b3608 in sl_send_reply (msg=0x643f28, code=100, text=0x608ee8 "Trying") at sl_funcs.c:149 #8 0x0000000000409c38 in do_action (a=dwarf2_read_address: Corrupted DWARF expression. ) at action.c:701 #9 0x000000000040a87e in do_action (a=dwarf2_read_address: Corrupted DWARF expression. ) at action.c:89 #10 0x000000000040b43d in do_action (a=dwarf2_read_address: Corrupted DWARF expression. ) at action.c:89 #11 0x000000000040b496 in do_action (a=dwarf2_read_address: Corrupted DWARF expression. ) at action.c:89 #12 0x000000000040b762 in run_top_route (a=Variable "a" is not available. ) at action.c:89 #13 0x000000000043188c in receive_msg (buf=0x2a97e4d2f8 "\026", len=822, rcv_info=0x7fbffff810) at receive.c:155 #14 0x000000000044fc74 in udp_rcv_loop () at udp_server.c:465 #15 0x000000000041f7f4 in main_loop () at main.c:925 #16 0x0000000000420446 in main (argc=Variable "argc" is not available. ) at main.c:1477 -- Helge Waastad Senior Engineer Systemavdelingen Smartnet

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[Users] aliases in multi-domain
by Ricardo Carvalho 06 Sep '06

06 Sep '06

I'm using multi-domain and I can only call users by their aliases if I'm dialling from a phone registered with the same domain of the called user or if I'm dialling alias@domain_of_the_callee from a phone registered with other domain. How can I make Ser allow that every user can contact other users even if they are from other domain by just dialling their alias (without dialling also their domain when the callee is from other domain than the caller)? Regards, Ricardo.

2 3

[Users] "BYE" messages change path
by ravi reddy 06 Sep '06

06 Sep '06

Hi users, This is ngrep report regarding my problem; -----------------<when a sip proxy behind NAT and the user registerd to that sip-proxy calls>------------------------- U 82.102.69.105:39871 -> 81.21.33.35:5060 INVITE sip:99106883@81.21.33.35:5060 SIP/2.0. To: "99106883"<sip:99106883@81.21.33.35:5060>. From: "12345"<sip:12345@81.21.33.35:5060>;tag=c86b66ad8b9187c8. Via: SIP/2.0/UDP 192.168.1.100:5060 ;branch=z9hG4bK-d87543-bcf89635ebeba2e78782465686dfaf52-1--d87543-;rport. Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKf638e18b56022ea3. Call-ID: a78d5c993a9dd6b4(a)192.168.1.102. CSeq: 47344 INVITE. Record-Route: <sip:192.168.1.100:5060>. Contact: <sip:192.168.1.100:5060>. Max-Forwards: 69. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE. Content-Type: application/sdp. Supported: replaces. User-Agent: Grandstream BT110 1.0.8.23. Content-Length: 361. . v=0. o=line2 8000 8000 IN IP4 192.168.1.102. s=SIP Call. c=IN IP4 82.102.69.105.---------------------------------------------> this is the NAT for the sip-proxy , t=0 0. m=audio 5004 RTP/AVP 18 4 2 97 9 0 101. a=fmtp:97 mode=20. a=fmtp:101 0-11. a=ptime:20. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:97 iLBC/8000. a=rtpmap:9 G722/16000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. -----------------------------<when my SIP-SERVER standing on public ip recieved the "INV" message from the above"----------------------------------- U 81.21.33.35:5060 -> pstngw:5060 INVITE sip:99106883@pstngw:5060 SIP/2.0. Record-Route: <sip:99106883@81.21.33.35 :5060;nat=yes;ftag=c86b66ad8b9187c8;lr=on>. To: "99106883"<sip:99106883@81.21.33.35:5060>. From: "12345"<sip:12345@81.21.33.35:5060>;tag=c86b66ad8b9187c8. Via: SIP/2.0/UDP 81.21.33.35;branch=z9hG4bK0ab.9522bc25.0. Via: SIP/2.0/UDP 192.168.1.100:5060;received=82.102.69.105 ;branch=z9hG4bK-d87543-bcf89635ebeba2e78782465686dfaf52-1--d87543-;rport=39871. Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKf638e18b56022ea3. Call-ID: a78d5c993a9dd6b4(a)192.168.1.102. CSeq: 47344 INVITE. Record-Route: <sip:192.168.1.100:5060>. Contact: <sip:82.102.69.105:39871>. Max-Forwards: 16. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE. Content-Type: application/sdp. Supported: replaces. User-Agent: Grandstream BT110 1.0.8.23. Content-Length: 360. . v=0. o=line2 8000 8000 IN IP4 192.168.1.102. s=SIP Call. c=IN IP4 81.21.33.35.--------------------------------------------->This is my SIP-SERVER standing on public ip (and is sending to pstngw) t=0 0. m=audio 60516 RTP/AVP 18 4 2 97 9 0 101. a=fmtp:97 mode=20. a=fmtp:101 0-11. a=ptime:20. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:97 iLBC/8000. a=rtpmap:9 G722/16000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. ---------------------------------------------<and the pstngw is making call>----------------------------------------------- U 81.21.33.35:5060 -> pstngw:39871 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 192.168.1.100:5060;received=82.102.69.105;branch=z9hG4bK-d87543-ac3b034487ba16641188e4c9e5ad0664-1--d87543-;rport=39871,SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK01c64e769ba6c176. From: "12345"<sip:12345@81.21.33.35:5060>;tag=9ca5c1fe3b43aad6. To: "99106883"<sip:99106883@81.21.33.35:5060>;tag=7E6010EC-1764. Date: Wed, 06 Sep 2006 10:27:40 GMT. Call-ID: 72d0e4e8adca2ab2(a)192.168.1.102. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 32352 INVITE. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 237. . v=0. o=CiscoSystemsSIP-GW-UserAgent 54 9902 IN IP4 pstngw s=SIP Call. c=IN IP4 81.21.33.35.-------------------------------------<in pstngw it noticed the contact header of SIP-SERVER>--------------------- t=0 0. m=audio 60518 RTP/AVP 18 100. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. a=rtpmap:100 X-NSE/8000. a=fmtp:100 192-194. a=ptime:20. ---------------------------------------------<after talking some time if any of UA hung the phone (here i am showing from pstn hung up) the result goes like this---------------- U pstngw:52991 -> 81.21.33.35:5060 BYE sip:99106883@81.21.33.35:5060;nat=yes;ftag=1bab08cdaad1d6e8;lr=on SIP/2.0. Via: SIP/2.0/UDP 81.21.38.15:5060. From: "99106883"<sip:99106883@81.21.33.35:5060>;tag=7E66E3D8-1292. To: "12345"<sip:12345@81.21.33.35:5060>;tag=1bab08cdaad1d6e8. Date: Wed, 06 Sep 2006 10:35:07 GMT. Call-ID: c2dd3fb9554ef6e4(a)192.168.1.102. User-Agent: Cisco-SIPGateway/IOS-12.x. Max-Forwards: 6. Route: <sip:192.168.1.100:5060>, <sip:82.102.69.105:39871>.------------------------> What is happening here ??????? Timestamp: 1157538919. CSeq: 101 BYE. Content-Length: 0. . # U 81.21.33.35:5060 -> 192.168.1.100:5060--------------------------------------------------->here unexpected problem arises for me it have to use 82.102.69.105 BYE sip:192.168.1.100:5060 SIP/2.0. Record-Route: <sip:81.21.33.35;ftag=7E66E3D8-1292;lr=on>. Via: SIP/2.0/UDP 81.21.33.35;branch=z9hG4bKb0f1.3fe32691.0. Via: SIP/2.0/UDP pstngw:5060. From: "99106883"<sip:99106883@81.21.33.35:5060>;tag=7E66E3D8-1292. To: "12345"<sip:12345@81.21.33.35:5060>;tag=1bab08cdaad1d6e8. Date: Wed, 06 Sep 2006 10:35:07 GMT. Call-ID: c2dd3fb9554ef6e4(a)192.168.1.102. User-Agent: Cisco-SIPGateway/IOS-12.x. Max-Forwards: 5. Route: <sip:82.102.69.105:39871>. Timestamp: 1157538919. CSeq: 101 BYE. Content-Length: 0. P-hint: rr-enforced. So the call never ends up from SIP SERVER to other party Make some comments on above and assist me where am I going wrong: and i am using www.openser.org pstn default script as ser.cfg Hope to get some help Thanks, Ravi.

1 0

[Serusers] SER & Asterisk (IVR problem)
by Roberto Lopes 06 Sep '06

06 Sep '06

Hi to all! I've a big problem with SER & Asterisk I've a network where i put them working together. Asterisk is the media gateway with PSTN as well as being the server which is making all the services available (IVR, voicemail, conference calls, etc). SER just routes calls either to the softphones, VoIP phones or to Asterisk who will just forward the calls to PSTN (via Asterisk). Till now everything works well except the service IVR which is working on Asterisk. Details: 192.168.226.13 - IP SER 192.168.226.45 - IP Asterisk The problem is only after the user (roberto(a)192.168.226.13 - authenticated and authorized in RADIUS) calls IVR. Till then the user calls, IVR gives some information to the user and everything goes well. Problem is when he presses a number which will have to redirect the call to another user. And why is there a problem. Because the initially roberto(a)192.168.226.13 to be able to call IVR had to rewrite host & port 192.168.226.45 so that the call would reach Asterisk who would then call to the IVR. That rewriting turned roberto(a)192.168.226.13 into roberto(a)192.168.226.45. That is, I can say, that it became a "new user". While it's on the IVR there's no problem, but when call is diverted it has to give its credentials... Credentials that don't exist in RADIUS cause it only accepts usernames that end in @192.168.226.13 (IP of SER) Here is the following code, both of extensions file in Asterisk and configure file of SER I hope somebody knows how to solve it or at least find a way how to. extensions.conf - Asterisk [sip] exten => 74001,1,Answer exten => 74001,2,SetMusicOnHold(default) exten => 74001,3,Set(TIMEOUT(digit)=5) exten => 74001,4,Set(TIMEOUT(response)=10) exten => 74001,5,Background(menu) exten => 1,1,Dial(SIP/roberto@192.168.226.13:5060,,r) ... exten => 7,1,Goto(sip,74001,1) exten => 8,1,Hangup ser.cfg - SER ... if ((uri=~"^sip:7[0-9]{4}@.*")) { route(5); break; }; ... route[5] { # Redirecting to Asterisk # MsgLog xlog("L_ALERT","(WARNING) Sending to Asterisk request of %fu to %tu\n"); rewritehostport("192.168.226.45:5060"); route(1); } Thanks for your help, Roberto Lopes

2 1

[Users] "BYE" messages change path
by ravi reddy 06 Sep '06

06 Sep '06

Hi Users, I am using SER-0.9.6 on public domain . and I am using another Sip-proxy behind NAT (ip-address=192.168.1.100:5060) i have some clients registered to that nated sip-proxy and here that nated sip-proxy forwards PSTN calls to SER on public domain. when call make from client (registered to nated sip-proxy) the session establishing suscessfully , but when the client hung the phone the SER on public domain is taking the BYE message and is not sending the BYE message to the other end and it is sending directly to the Sip-proxy which is a private-ip . So, here SER is not getting ACK and its keep on sending the BYE requests to the privateip instead of sending to the NAT address . In some docs which jiri wrote on record route he mentioned the same problem below is the link ; but he dint mention how to solve that record route problem, If SER send messages back as it recieving path (from NAT)there is no problem , but its changing the route to send BYE , For the request "CANCEL","ACK","INVITE","200 OK" every thing is happening in the same Route but it is not happening only for the "BYE" request . Hope some body can help me. Thank You. here is the link which i told;-http://216.239.59.104/search?q=cache:A32Unl9P0hkJ:voip.internet2.edu/…<http://216.239.59.104/search?q=cache:A32Unl9P0hkJ:voip.internet2.edu/meetin…> Regards, Ravi.

1 0

[Serusers] aliases in multi-domain
by Ricardo Carvalho 06 Sep '06

06 Sep '06

I'm using multi-domain and I can only call users by their aliases if I'm dialling from a phone registered with the same domain of the called user or if I'm dialling alias@domain_of_the_callee from a phone registered with other domain. How can I make Ser allow that every user can contact other users even if they are from other domain by just dialling their alias (without dialling also their domain when the callee is from other domain than the caller)? Regards, Ricardo.

2 3

Re: [Users] more on accounting
by tele 06 Sep '06

06 Sep '06

Glenn Dalgliesh wrote: >This is from previous list > >"i Glenn, > >For this feature to work, you need to be sure that you do record_route() on >the INVITE and that "append_fromtag" RR module param is not disable. > >regards, >bogdan " > > Yes i have read that thread. i always record_route all messages before do things. if (!method=="REGISTER") record_route(); by the sample openser.cfg and for be sure i have also forced: modparam("rr","append_fromtag",1) >-----Original Message----- >From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf >Of tele >Sent: Wednesday, June 28, 2006 11:05 AM >To: users(a)openser.org >Subject: [Users] more on accounting > >Hi all, > >I'm using OpenSER v1.0.1 and i have done an accounting compatible with >my billing system. >Now i have problem with detection of flow direction for billing reason. > >i'm talking about openser 1.0.x: > >the problem is when the BYE come from callee. >i know the problem is solved in openser1.1.x with the > >modparam("acc", "detect_direction", 1) > >i have read in modules documentation of 1.0.x that i can use >is_direction() of RR module for detect call flow direction but it says >that this must be called after loose_route() > >so for the accounting i must call setflag before loose_route() but what >can i do if i want detect call flow >before of exec setflag? > >Hmmm i think that i must install OpenSER 1.1.x and test with it > >:tele > > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >

5 21

[Serusers] Media Proxy and 2 interfaces (Private and Public)
by Kyriakos 06 Sep '06

06 Sep '06

Hello, Is there a way to have a machine that hosts media proxy have its one interface in a private network and the other on a public? >From what I read rtpproxy supports bridging which is basically what I want to do but I have not found anything for mediaproxy. Thanks KM

2 1

[Serusers] Sip Proxy behind Nat.
by ravi reddy 06 Sep '06

06 Sep '06

Hi Users, I am Using SER-0.9.6 on a public ip and using another sip-proxy(not SER) behind the NAT. and when the users who registerd to Sip-proxy which is behind NAT,, calls to the users who registered to public domain SER.. call is being established . but when hunging the phone the SER is not sending "BYE" messages to the NAT device it is sending to the Sip-proxy (not SER) that is for a private IP . so even when i hung the call from the Nated Sip proxy ,SER is getting messages through NAT but again SER is not sending "BYE" messages to other party its sending back to the nated sip-proxy having private IP , so the messages are going in a wrong direction and is being wasted . It looks like this is a BUG in SER , i would take this oppurtunity to report this. If this is not a bug please send me the suggestions ..... To solve my problem. thank you. regards, Ravi.

2 1

[Serusers] server error message " 500 I'm terribly sorry, server error occurred (1/SL)".
by ravi reddy 06 Sep '06

06 Sep '06

Hi users , I am proceeding in this way i have a SER-0.9.6 running fine :-) and I had given a username and password to a call-shop and this callshop owner with his username and password he connects to another 6 phones, Actually he bills to his 6 phones and I will bill him ."o.k overall Scenario is well and good". I made a little changes in ser.cfg and when the call made from the call shop up to call connecting is O.K but when we hung the phone the SER is not generating "BYE" messages to other party , so the call is on.. and i am not getting "Acct Stop" packet also SO How I can solve my problem :-( any suggestions will be appreciated . below is the message i am getting from SER when I hung the phone on one side -------------------<this message is came from callshop Nat address>---------<"it sends bye to my SER "------------------------------------------------------ U 82.102.69.105:32768 -> 81.21.33.35:5060 BYE sip:99106883@81.21.33.35:5060 SIP/2.0. To: "99106883"<sip:99106883@81.21.33.35:5060>;tag=78F9ECC4-166C. From: "12345"<sip:12345@81.21.33.35:5060>;tag=837e5b2ff0b4cf4a. Via: SIP/2.0/UDP 192.168.1.100:5060 ;branch=z9hG4bK-d87543-5983e94f152d226c82a4e76799fe58e5-1--d87543-;rport. Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK9bf920205fe3bef9. Call-ID: f1b0fe2b6cdf1456(a)192.168.1.102. CSeq: 9533 BYE. Route: <sip:99106883@81.21.33.35:5060;lr;nat=yes;ftag=837e5b2ff0b4cf4a>. Route: <sip:192.168.1.100:5060>. Record-Route: <sip:192.168.1.100:5060>. Contact: <sip:192.168.1.100:5060>. Max-Forwards: 69. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE. Proxy-Authorization: Digest username="12345", realm="81.21.33.35", algorithm=MD5, uri="sip:192.168.1.100:5060", nonce="44fd415a6754afa39f2668ce2ad11da6bfc65cde", response="4001812db780010a557b40683efd2f9e". Supported: replaces. User-Agent: Grandstream BT110 1.0.8.23. Content-Length: 0. . ----------------<here it is as soon as SER recieve Bye message it has to send Bye to other party >------<But it is sending to the hung up phone itself>--------- # U 81.21.33.35:5060 -> 192.168.1.100:5060 BYE sip:192.168.1.100:5060 SIP/2.0. Record-Route: <sip:81.21.33.35;ftag=837e5b2ff0b4cf4a;lr=on>. To: "99106883"<sip:99106883@81.21.33.35:5060>;tag=78F9ECC4-166C. From: "12345"<sip:12345@81.21.33.35:5060>;tag=837e5b2ff0b4cf4a. Via: SIP/2.0/UDP 81.21.33.35;branch=z9hG4bK936f.7ff7572.0. Via: SIP/2.0/UDP 192.168.1.100:5060;received=82.102.69.105 ;branch=z9hG4bK-d87543-5983e94f152d226c82a4e76799fe58e5-1--d87543-;rport=32768. Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK9bf920205fe3bef9. Call-ID: f1b0fe2b6cdf1456(a)192.168.1.102. CSeq: 9533 BYE. Record-Route: <sip:192.168.1.100:5060>. Contact: <sip:192.168.1.100:5060>. Max-Forwards: 16. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE. Proxy-Authorization: Digest username="12345", realm="81.21.33.35", algorithm=MD5, uri="sip:192.168.1.100:5060", nonce="44fd415a6754afa39f2668ce2ad11da6bfc65cde", response="4001812db780010a557b40683efd2f9e". Supported: replaces. User-Agent: Grandstream BT110 1.0.8.23. Content-Length: 0. . -----------------<And here I go iam getting this message and the call is not being stopped >---------------------------------- # U 81.21.33.35:5060 -> 82.102.69.105:32768 SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL). To: "99106883"<sip:99106883@81.21.33.35:5060>;tag=78F9ECC4-166C. From: "12345"<sip:12345@81.21.33.35:5060>;tag=837e5b2ff0b4cf4a. Via: SIP/2.0/UDP 192.168.1.100:5060 ;branch=z9hG4bK-d87543-5983e94f152d226c82a4e76799fe58e5-1--d87543-;rport=32768;received= 82.102.69.105. Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK9bf920205fe3bef9. Call-ID: f1b0fe2b6cdf1456(a)192.168.1.102. CSeq: 9533 BYE. Content-Length: 0. Warning: 392 81.21.33.35:5060 "Noisy feedback tells: pid=27773 req_src_ip= 82.102.69.105 req_src_port=32768 in_uri=sip:99106883@81.21.33.35:5060 out_uri=sip:192.168.1.100:5060 via_cnt==2". . any suggestions will be appreciated: Thank You.

2 1

[Serusers] Record Route problem :-(
by ravi reddy 06 Sep '06

06 Sep '06

Hi SER Users, I am using SER-0.9.6 on public domain . and I am using another Sip-proxy behind NAT (ip-address=192.168.1.100:5060) i have some clients registered to that nated sip-proxy and here that nated sip-proxy forwards PSTN calls to SER on public domain. when call make from client (registered to nated sip-proxy) the session establishing suscessfully , but when the client hung the phone the SER on public domain is taking the BYE message and is not sending the BYE message to the other end and it is sending directly to the Sip-proxy which is a private-ip . So, here SER is not getting ACK and its keep on sending the BYE requests to the privateip instead of sending to the NAT address . In some docs which jiri wrote on record route he mentioned the same problem below is the link ; but he dint mention how to solve that record route problem, If SER send messages back as it recieving path there is no problem , but its changing the route to send BYE , For the request "CANCEL","ACK","INVITE","200 OK" every thing is happening in the same Route but it is not happening only for the "BYE" request . Hope some body can help me. Thank You. here is the link which i told;- http://216.239.59.104/search?q=cache:A32Unl9P0hkJ:voip.internet2.edu/meetin… Regards, Ravi.

1 0

[Users] t_forward() functionality with t_relay()?
by Greg Fausak 06 Sep '06

06 Sep '06

I've got a dumb question. Can I get t_relay() functionality (transactions and whatnot) without getting the Via and potentially Record-Route tags in the header??? I warned you it was dumb.... -g -- Greg Fausak greg(a)thursday.com

3 4

[Users] Error in installing Ruby...Please help...Please
by Ferianto siregar 06 Sep '06

06 Sep '06

Dear all, Thank you very much for your help. A few days ago, I have sent a message that to ask about openser admin installation. Thank you very much for your help. Thanks Now, I have tried to install it again. But, When I compile the rubygerms-0.9.0.tar.gz, I got error message. This is the task that I have done before: 1. Untar the rubygerms packet # tar -zxvf rubygems-0.9.0.tar.gz # cd rubygems-0.9.0 # ruby setup.rb # gem install --include-dependencies When I type this command, "gem install --include-dependencies", I got error message. This is the error message: Attempting local installation of 'rails' Local gem file not found: rails*.gem Attempting remote installation of 'rails' Updating Gem source index for: http://gems.rubyforge.org ERROR: While executing gem ... (Gem::RemoteSourceException) Error fetching remote gem cache: getaddrinfo: Temporary failure in name esolution What`s wrong? I do hope anybody can help me to solve this problem. Please help me...Please help because of my lack skills..Please Note: 1. I do not have connection through the internet, so for rails packet, I have downloaded the rails-1.1.6.tar.gz. I have untar it in root directory(same as rubygems) and type command rails # script/server But, Nothing happens. Why? Please... Thank you very much Regards, Ferianto --------------------------------- Do you Yahoo!? Next-gen email? Have it all with the all-new Yahoo! Mail.

2 1

[Serusers] NAT Solution
by Asep Saifudin 06 Sep '06

06 Sep '06

Hi, Does anyone can help me to choose best NAT solution for SER/OpenSER ?NatHelper or Mediaproxy ? And if they compare with STUN, TURN, ICE technology Nathelper or Mediaproxy better than them ? My target is I want my SIP Server can pass all NAT/Firewall just like SkyPE which hard to be block by firewall and can pass all NAT system. ~Asep

2 1

[Serusers] SER not responding
by Henry Duan 06 Sep '06

06 Sep '06

hi, I'm new to SER. I downloaded SER0.9.6 source to my local Fedora Core 5. I run make all and make prefix=/ install, and then set SIPDOMAIN, and run ser. However it didn't respond with incomging Registration(it showed incoming Registration from Ethereal). (I used default ser.cfg). Can anybody tell me what's the problem, or how to debug? Thanks so much! Henry

3 2

[Users] OSS on openSER and Redriect .
by raviprakash sunkara 06 Sep '06

06 Sep '06

Hi Users, Can any one tell about OSS ( Operation Support System) on openSER, Why we need ? But i think to reduce the load balance on openSER server, Suppose is I use 3 OSS's on SIP server. Sip users should also insert in to 3 OSS sip. And other doubt is ? Suppose if we using the redirect , that forwording into other openser system, When openSER redirect to other openSER, the user should also insert, But i think that .... OSS and redirect on openSER are the same. Please for me in English, I know its naughty questioin, please excuse me ! -- -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com +91-9985077535

1 0

[Serusers] Free DNS Service
by Tracy Lofton 06 Sep '06

06 Sep '06

Hi All, I am new to OpenSER/SER but have been receiving quite a bit of help and would like to express my gratitude by letting you all know of the FREE DNS Service that I am offering to the Internet Community. Go to http://www.t4tm.net to find out more, signup and enjoy the benefits. Tracy _________________________________________________________________ Get the new Windows Live Messenger! http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=…

1 0

[Serusers] Preregistering a sip address?
by Ignacio Mas Ivars 06 Sep '06

06 Sep '06

Hi all! I have just started using the SIP Express Router and I am looking for a way to have a user preregistered when the router starts. Is there any way to configure that user1 should be registered by default at an IP address without needing to send the registration? I guess it will have to do with messing a bit with the routing engine, but I don't dare to touch it so far and I don't know how to do it. Any help would be very much appreciated! Thanks very much for any help! /Nacho --- ,,, (o o) --ooO--(_)---Ooo----------- Three may keep a secret, if two of them are dead.- --oo0--------0oo-----------

2 1

[Serusers] Free DNS Service
by Tracy Lofton 06 Sep '06

06 Sep '06

Hi All, I am new to SER/OpenSER and I have received much help and in return, I am offering free Dynamic DNS services to whoever is interested. To find out more go to http://www.t4tm.net, signup and enjoy the benefits. Tracy _________________________________________________________________ Got something to buy, sell or swap? Try Windows Live Expo ttp://clk.atdmt.com/MSN/go/msnnkwex0010000001msn/direct/01/?href=http://exp…

1 0

[Serusers] delay to write things in DB when using serctl
by Ricardo Carvalho 06 Sep '06

06 Sep '06

When I add with serctl some alias to the DB, only after some time it appears written in the aliases table of DB. How can I force that to be simultaneously written in DB? Any parameter I have to tune in ser.cfg? Regards, Ricardo.

2 1

[Users] Re: Request for discussion: t_relay() internal error processing
by Mark Kent 05 Sep '06

05 Sep '06

>> 2) t_relay will return several error codes: >> -1 = no transaction created -> need to use sl_* functions >> -2 = transaction created, relay failed -> may destroy If I understand this correctly, the difference between these two return codes really has to do with who it is that cares about it... -2 relates to this exact call, at this exact moment, chances are the very next call the openser box handles will be OK. It looks to me like -1 means serious trouble, for the entire call switching platform. It sounds like it could happen because of a resource shortage (memory), possibly because of a bug in openser, or a bug in the openser.cfg. If you get -1 once, then the chances that you'll get the same code on the next call are very high. Let's say we've run out of memory. Then, what tools are available at the openser.cfg script level that can handle this? Is there a way to force some internal garbage collection? Or to do a warm-restart (e.g., trash all transactions, but not registrations)? My point here is this: if I'm right about -1 meaning serious trouble that is likely to mean serious problems from now on, then this is a more general problem than just t_relay, and much bigger than just handling this one call or a call to one particular neighbour. It's true that the application may halt and catch fire, but there are plenty of errors that are catchable by openser and the larger question seems to be: what do we do then? If there were script-level tools that could be used to try to manage the failure, possibly clean up (in a large way) and try to keep going, then maybe we should have openser_failure route blocks that the user can register with openser to handle various errors. Thanks, -mark

2 1

[Users] Request for discussion: t_relay() internal error processing
by Bogdan-Andrei Iancu 05 Sep '06

05 Sep '06

Hi all, there are some topics largely discussed lately on the mailing list; in order to fix them problems first we need to find a solution for logical point of view - and here I think that several opinions are better than one :) In order to have failover based on serial forking, it is mandatory to be able to detect from script if the initial t_relay() fails. The problem is that now t_relay() masks any error by internally generating an error SIP reply. Of course, it can be internal errors (like lack of memory), msg errors (parse errors) or transport errors (DNS failed, send/connect failed, etc). Now, disabling the auto replying in case of error is not a problem. The problem is how to handle the errors from the script; two make distinction between the error classes a listed above will not be the purpose of this email ( I will open another one for this ;) ). What is important here is to be able to properly resume from the point were the error popped up. What I mean is: the t_relay() function builds the transaction and sends the requests. If the error occurred during transaction creation, from script, you will had to use only stateless functions to send err. replies. If the transaction was already created and the error popped up during relaying stage, you will need from the script to take care of the transaction and either destroy it, either reply it. Based on this, here is what I suggest: 1) have a new param for disabling the auto reply in t_relay() 2) t_relay will return several error codes: -1 = no transaction created -> need to use sl_* functions -2 = transaction created, relay failed -> may destroy transaction, reply via TM with error, trigger failover. any comments on this ? regards, bogdan

4 4

[Users] mediaproxy error
by Richard Z 05 Sep '06

05 Sep '06

Hi, I tried the lateset mediaproxy. When starting, got these errors and dispatcher won't start, /var/log/syslog:Sep 5 10:45:57 sip-1 proxydispatcher[4227]: File "/usr/local/mediaproxy/proxydispatcher.py", line 65, in ? /var/log/syslog:Sep 5 10:45:57 sip-1 proxydispatcher[4227]: File "/usr/local/mediaproxy/modules/dispatcher.py", line 39, in ? /var/log/syslog:Sep 5 10:45:57 sip-1 proxydispatcher[4227]: File "/usr/local/mediaproxy/modules/accounting.py", line 14, in ? /var/log/syslog:Sep 5 10:45:57 sip-1 proxydispatcher[4227]: File "/usr/local/mediaproxy/modules/mysql.py", line 8, in ? #ls /usr/local/mediaproxy/modules/ accounting.py daemon.py dispatcher.py mysql.py request.py serialize.py version.pyc accounting.pyc daemon.pyc dispatcher.pyc mysql.pyc request.pyc utilities.py configuration.py datatypes.py external options.py rtphandler.py utilities.pyc configuration.pyc datatypes.pyc formatting.py options.pyc rtphandler.pyc version.py Any suggestion? Thanks, Richard

1 0

[Serusers] No calls shown in serweb under accounting tab
by Chahn John Kim 05 Sep '06

05 Sep '06

Sorry for double posting. I forgot to put subject line in previous posting. Hi, all. I just installed serweb and it is working however there is are no calls shown under "accounting" tab, saying "no calls". Mysql table has call records with matching flag set in ser.cfg. Records seem to be correct for numerous test calls made. It just that serweb doesn't seem to pull those records out of mysql at all. Any idea where I should look at? Thanks in advance. -- Chahn John Kim

1 0

[Serusers] (no subject)
by Chahn John Kim 05 Sep '06

05 Sep '06

Hi, all. I just installed serweb and it is working however there is are no calls shown under "accounting" tab, saying "no calls". Mysql table has call records with matching flag set in ser.cfg. Records seem to be correct for numerous test calls made. It just that serweb doesn't seem to pull those records out of mysql at all. Any idea where I should look at? Thanks in advance. -- Chahn John Kim

1 0

[Serusers] prevent a dialled username without dialling the domain to ring a phone registered with that username
by Ricardo Carvalho 05 Sep '06

05 Sep '06

How can I prevent a dialled username (without dialling the domain) to ring a phone registered with that username? I pretend to use a multi-domain architecture where apart from aliases of users, only dialling with username@domain will ring registered phones. Thanks, Ricardo.

2 1

[Users] Proxy to Proxy Communication
by shiva sankar padhy 05 Sep '06

05 Sep '06

Dear Members! I am very new to This VOIP Technology. I have just installed the Openser. I am now able to Do the Communication between the users who are resisterd in the proxy but how do I establish the calls to another proxy.I mean Lets consider I have two Proxies and I want a user of Proxy 1 to coomunicate with a user of Proxy2.How will I edit the configuration files so that I can achieve that. And One more thing.How do I achieve the Nat Traversal in openser.Let us consider I am running my Server in 24.10.65.90(Public and static).and I have a Lan connection In the client side, When the two phones from the same LAN want to talk with each other by using the proxy at 24.10.65.90, Its not possible.I got Disconnected from the internet connection and also when I tried this.If I am calling to a Phone other than this LAN using proxy at 24.10.65.90, I am able to establish.What may be the problem to the previous. I hope I have explained my problems in a user Understandable way. I am expecting a sooner reply. Thanking you In Advance, Shiva

2 1

[Users] Freeradius dictionary additions..
by nick 05 Sep '06

05 Sep '06

When I add my dictionary.openser to Freradius v 1.0.1-3 on Centos 4.3, and after following all the steps outlined on the webpage: http://www.openser.org/docs/openser-radius-1.0.x.html When I get to testing the configuration, radiusd -X i get this: read_config_files: reading dictionary Errors reading dictionary: dict_init: No ATTRIBUTE "Sip-Method" defined for VALUE "Other" Errors reading radiusd.conf Unfortunately freeradius.org is now a parked domain (seems like a rather large oversight on their part...) so I can't ask them.. Has anyone here ever seen an issue like this? BTW, I'm using openser 1.1.0-notls compiled from source. Not that seems to apply here though.

6 10

[Serusers] DNS SRV Query's
by Ricardo Martinez 05 Sep '06

05 Sep '06

Hello list. I was doing some debugs on my SER machine and i found this these DNS query's: 9.381051 10.10.10.35 -> 10.10.10.22 DNS Standard query SRV _sip._udp.sipm1.mydomain.net 9.390312 10.10.10.22 -> 10.10.10.35 DNS Standard query response, No such name 9.390559 10.10.10.35 -> 10.10.10.22 DNS Standard query SRV _sip._udp.sipm1.mydomain.net 9.399772 10.10.10.22 -> 10.10.10.35 DNS Standard query response, No such name 11.337343 10.10.10.35 -> 10.10.10.22 DNS Standard query SRV _mediaproxy._tcp.sipproxy.mydomain.net 11.339614 10.10.10.22 -> 10.10.10.35 DNS Standard query response, No such name 11.345067 10.10.10.35 -> 10.10.10.22 DNS Standard query SRV _sip._udp.sipm1.mydomain.net 11.349127 10.10.10.22 -> 10.10.10.35 DNS Standard query response, No such name 11.349584 10.10.10.35 -> 10.10.10.22 DNS Standard query SRV _sip._udp.sipm1.mydomain.net 11.358696 10.10.10.22 -> 10.10.10.35 DNS Standard query response, No such name 17.247771 10.10.10.35 -> 10.10.10.22 DNS Standard query SRV _sip._udp.sipm1.mydomain.net 17.253948 10.10.10.22 -> 10.10.10.35 DNS Standard query response, No such name 17.254042 10.10.10.35 -> 10.10.10.22 DNS Standard query SRV _sip._udp.sipm1.mydomain.net 17.263298 10.10.10.22 -> 10.10.10.35 DNS Standard query response, No such name 18.988320 10.10.10.35 -> 10.10.10.22 DNS Standard query SRV _sip._udp.sipm1.mydomain.net 18.998219 10.10.10.22 -> 10.10.10.35 DNS Standard query response, No such name 18.998329 10.10.10.35 -> 10.10.10.22 DNS Standard query SRV _sip._udp.sipm1.mydomain.net 19.008071 10.10.10.22 -> 10.10.10.35 DNS Standard query response, No such name What modules is generating this querys' and why ? Thanks Ricardo

2 1

[Users] Mediaproxy with t38
by Leonid Fainshtein 05 Sep '06

05 Sep '06

Pavel, I think you have to check if your VoIP gateways can send UDPTL "no-signal" messages. Sometimes the caller gateway doesn't send any messages until it receives CED from callee. Because of that neither MediaProxy nor RTPProxy can't complete the ip/port binding for the media session. Also, check which ports are used for UDPTL session. If they are differ from the ports that were used during the previous voice (RTP) session than you probably have to re-establish mediaproxy session. At least it is true for RTP Proxy that I use. For example, I call unforce_rtp_proxy() and then force_rtp_proxy() on each re-INVITE. To be honest, I have never tried the MediaProxy but I suppose that it works in the similar way like the RTP Proxy. Best regards, Leonid Fainshtein Product Manager IP Gear Ltd., a division of Qualmax "Get your VoIP network "In Gear" with Claro gateways from IP Gear" e-mail: LFAIN(a)ipgear.net Tel: +972-4-9907-525 Fax: +972-4-9990-334 www.ipgear.net

1 0

[Users] Not Hung uping...
by raviprakash sunkara 05 Sep '06

05 Sep '06

Good Morning, In local network openser is working fine...... In Behind the NAt the voice is also fine.... But problem is Not hung. upping.... I think problem in Record router header.... Actually it should reply as " Record Router < sip: public.ip:...... > " , But for me it reply as " Record Router <sip: private.ip.... > " That why i think Bye request is not reaching to openser server and its not hung upping.... and my guess is router(firewall) causing this problem , But voice is fine.... Below one is log file.... sip_domain= public_ip It s not hung upping....... openser server system ip 192.168.2.75 And UAC are outside the NAT The media is fine..... but not hung upping at a same time..... I think the record router is not correct... which is red color marked one...... 59.93.74.129:41010 -> 192.168.2.75:5060 INVITE sip:9001@public_ip SIP/2.0..Via: SIP/2.0/UDP 59.93.74.129:5060 ;rport;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara < sip:9002@6 1.17.248.68>;tag=587704647..To: <sip:9001@public_ip>..Contact: < sip:9002@59.93.74.129:5060>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E(a)192.168.2. 34..CSeq: 35636 INVITE..Max-Forwards: 70..Content-Type: application/sdp..User-Agent: X-Lite release 1105x..Content-Length: 305....v=0..o=9002 35180894 3 5180906 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=00..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a =rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 59.93.74.129:5060 ;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=587704647..To: <sip:9001@public_ip>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E(a)192.168.2.34 ..CSeq: 35636 INV ITE..Server: OpenSer (1.1.0-notls (i386/linux))..Content-Length: 0..Warning: 392 192.168.2.75:5060 "Noisy feedback tells: pid=4700 req_src_ip=59.93.74. 129 req_src_port=41010 in_uri=sip:9001@public_ip out_uri= sip:9001@59.93.74.129:5060 via_cnt==1".... # U 192.168.2.75:5060 -> 59.93.74.129:5060 INVITE sip:9001@59.93.74.129:5060 SIP/2.0..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b 487.0..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public _ip>;tag=58770464 7..To: <sip:9001@public_ip>..Contact: <sip:9002@59.93.74.129:5060>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..M ax-Forwards: 69..Content-Type: application/sdp..User-Agent: X-Lite release 1105x..Content-Length: 305....v=0..o=9002 35180894 35180906 IN IP4 59.93.74.1 29..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtp map:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;br anch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara < sip:9002@public_ip>;tag=587704647..To: < sip:9001@public _ip>;tag=2115499902..Contact: <sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E(a)192.168.2.34 ..CSeq: 3 5636 INVITE..Server: X-Lite release 1105x..Content-Length: 0.... # U 59.93.74.129:41010 -> 192.168.2.75:5060 .. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;b ranch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara < sip:9002@public_ip>;tag=587704647..To: < sip:9001@public _ip>;tag=2115499902..Contact: <sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75;lr=on;ftag=587704647> ..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Server: X-Lite release 1105x..Content-Length: 0.... # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip >;tag=587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>..Record-Route: <sip: 192.168.2.75 ;lr=on;ftag=587704647 >..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Server: X-Lite release 1105x..Content-Length: 0.... # U 59.93.74.129:5060 -> 192.168.2.75:5060 .. # U 59.93.74.129:41010 -> 192.168.2.75:5060 .. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch =z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara < sip:9002@public_ip>;tag=587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: <sip :9001@59.93.74.129:5060>..Record-Route: <sip: 192.168.2.75 ;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Length: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite ..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/ 8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 59.93.74.129:5060 ;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag =587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>.. Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Ca ll-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Lengt h: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pc mu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a =sendrecv.. # U 59.93.74.129:5060 -> 192.168.2.75:5060 .. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch =z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: <sip :9001@59.93.74.129 :5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Length: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite ..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/ 8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag =587704647..To: < sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75;lr=on;ftag=587704647> ..Ca ll-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Lengt h: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pc mu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a =sendrecv.. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 192.168.2.75 ;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch =z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=587704647..To: < sip:9001@public_ip>;tag=2115499902..Contact: <sip :9001@59.93.74.129:5060>..Record-Route: <sip: 192.168.2.75 ;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Length: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite ..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/ 8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 59.93.74.129:5060 ;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag =587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>. .Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Ca ll-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Lengt h: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pc mu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a =sendrecv.. # U 59.93.74.129:41010 -> 192.168.2.75:5060 .. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch =z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: <sip :9001@59.93.74.129 :5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Length: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite ..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/ 8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag =587704647..To: < sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75;lr=on;ftag=587704647> ..Ca ll-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Lengt h: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pc mu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a =sendrecv.. -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com +91-9985077535

1 0

[Users] 415 unsopported document format.
by Javier Ramirez 04 Sep '06

04 Sep '06

Hi, I have a problem, when I send a PUBLISH with personal status, if have accent, the openser send me 415 unsopported document format. How to encode for openser accept my PUBLISH ? best regards Javier

2 1

[Users] How to configure multi-domain?
by Ricardo Carvalho 04 Sep '06

04 Sep '06

Besides from adding the following lines in configuration file and adding 2 diferent domains to the domain table, what else should I do to implement multi-domain? modparam("usrloc", "use_domain", 1) modparam("registrar", "use_domain", 1) My phones still register, but none of them can call any other, even if I call someone only by username or by username@domain. Regards, Ricardo.

2 1

[Users] run_failure_handlers warning message
by Federico Giannici 04 Sep '06

04 Sep '06

Since we upgraded to 1.1.0 from time to time the following warning is logged: WARNING:tm:run_failure_handlers: no UAC or CANCEL support (1, 2) Is it "normal" or does it indicate some problem some were? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

3 3

[Users] redirect and proxy functionality
by Mark Kent 04 Sep '06

04 Sep '06

I have a clash of concepts in my head, and it's messing with my openser config. 1) On one hand I think that if a phone is configured to use an outbound proxy then it should use it. Period. 2) On the other hand, I now do a "300 Redirect" and the phone sends back, to the same box, the same Invite but with the redirected r-uri. This is consistent with 1), but in this case I don't want this behaviour... If things are supposed to work like 2) then why wouldn't I just rewritehostport() and t_relay it without bothering with the redirect? In any case, I don't want to do that (rewritehostport), I want the redirect to be "try over there, and don't bug me again." I don't want to relay the dialog. So, in the real world, what should I hope to get out of a redirect (3xx) code? Thanks, -mark

2 1

[Users] uac_redirect out of memory?
by Greg Fausak 04 Sep '06

04 Sep '06

We've been doing a little bit of stress testing on the auc_redirect module. Throwing 60 calls per second at it. After quite a while I start getting error messages like: Sep 1 15:33:43 www openser[74528]: ERROR: sip_msg_cloner: cannot allocate memory Sep 1 15:33:43 www openser[74499]: redirect.g2.sbsqa2.com-LOOSE: time_t=1157142823 ^LINE=47 ^FILE=/var/local/ser/ROUTLET/config.pre ^call_id=5723-28564(a)10.88.129.253 ^cseq=3 ^contact=<sip:18001002132@10.88.129.253> ^from=sip:+18001002132@10.88.129.253 ^fromtag=18001002132 ^to=sip:+16661012126@10.88.128.116 ^totag=5724 ^method=BYE ^ruri=sip:+116661 012126@10.88.129.251:5060 ^messageid=263458^remark=Doing it Sep 1 15:33:43 www openser[74528]: ERROR: new_t: out of mem: Sep 1 15:33:43 www openser[74592]: redirect.g2.sbsqa2.com-LOOSE: time_t=1157142823 ^LINE=47 ^FILE=/var/local/ser/ROUTLET/config.pre ^call_id=5724-28564(a)10.88.129.253 ^cseq=3 ^contact=<sip:18001002133@10.88.129.253> ^from=sip:+18001002133@10.88.129.253 ^fromtag=18001002133 ^to=sip:+16661012127@10.88.128.116 ^totag=5725 ^method=BYE ^ruri=sip:+116661 012127@10.88.129.251:5060 ^messageid=263192^remark=Doing it Sep 1 15:33:43 www openser[74528]: ERROR: t_newtran: new_t failed Sep 1 15:33:43 www openser[74499]: ERROR: sip_msg_cloner: cannot allocate memory Sep 1 15:33:43 www openser[74592]: ERROR: new_t: out of mem: Sep 1 15:33:43 www openser[74499]: ERROR: new_t: out of mem: Sep 1 15:33:43 www openser[74592]: ERROR: t_newtran: new_t failed in the log file. I have seen similar problems with regular t_relay when we approach 200 calls per second. Is this a shared memory problem? How much shared memory to I need to set up? Have a great weekend! ---greg -- Greg Fausak greg(a)thursday.com

3 3

[Users] Not Hung uping
by raviprakash sunkara 04 Sep '06

04 Sep '06

Hi.... please forgive me in egnlish...... Below one is log file.... sip_domain= public_ip It s not hung upping....... openser server system ip 192.168.2.75 And UAC are outside the NAT The media is fine..... but not hung upping at a same time..... I think the record router is not correct... which is red color marked one...... 59.93.74.129:41010 -> 192.168.2.75:5060 INVITE sip:9001@public_ip SIP/2.0..Via: SIP/2.0/UDP 59.93.74.129:5060;rport;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@6 1.17.248.68>;tag=587704647..To: <sip:9001@public_ip>..Contact: < sip:9002@59.93.74.129:5060>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E(a)192.168.2. 34..CSeq: 35636 INVITE..Max-Forwards: 70..Content-Type: application/sdp..User-Agent: X-Lite release 1105x..Content-Length: 305....v=0..o=9002 35180894 3 5180906 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=00..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a =rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 59.93.74.129:5060 ;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=587704647..To: <sip:9001@public_ip>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INV ITE..Server: OpenSer (1.1.0-notls (i386/linux))..Content-Length: 0..Warning: 392 192.168.2.75:5060 "Noisy feedback tells: pid=4700 req_src_ip=59.93.74. 129 req_src_port=41010 in_uri=sip:9001@public_ip out_uri= sip:9001@59.93.74.129:5060 via_cnt==1".... # U 192.168.2.75:5060 -> 59.93.74.129:5060 INVITE sip:9001@59.93.74.129:5060 SIP/2.0..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b 487.0..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=58770464 7..To: <sip:9001@public_ip>..Contact: <sip:9002@59.93.74.129:5060>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..M ax-Forwards: 69..Content-Type: application/sdp..User-Agent: X-Lite release 1105x..Content-Length: 305....v=0..o=9002 35180894 35180906 IN IP4 59.93.74.1 29..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtp map:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;br anch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara < sip:9002@public_ip>;tag=587704647..To: <sip:9001@public _ip>;tag=2115499902..Contact: <sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 3 5636 INVITE..Server: X-Lite release 1105x..Content-Length: 0.... # U 59.93.74.129:41010 -> 192.168.2.75:5060 .. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;b ranch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara < sip:9002@public_ip>;tag=587704647..To: <sip:9001@public _ip>;tag=2115499902..Contact: <sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Server: X-Lite release 1105x..Content-Length: 0.... # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip >;tag=587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647 >..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Server: X-Lite release 1105x..Content-Length: 0.... # U 59.93.74.129:5060 -> 192.168.2.75:5060 .. # U 59.93.74.129:41010 -> 192.168.2.75:5060 .. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch =z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: <sip :9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Length: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite ..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/ 8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag =587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Ca ll-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Lengt h: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pc mu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a =sendrecv.. # U 59.93.74.129:5060 -> 192.168.2.75:5060 .. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch =z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: <sip :9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Length: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite ..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/ 8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag =587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Ca ll-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Lengt h: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pc mu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a =sendrecv.. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch =z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: <sip :9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Length: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite ..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/ 8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag =587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Ca ll-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Lengt h: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pc mu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a =sendrecv.. # U 59.93.74.129:41010 -> 192.168.2.75:5060 .. # U 59.93.74.129:5060 -> 192.168.2.75:5060 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 192.168.2.75;branch= z9hG4bKf13d.ace6b487.0;received=public_ip..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch =z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag=587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: <sip :9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Call-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Length: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite ..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/ 8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 192.168.2.75:5060 -> 59.93.74.129:41010 SIP/2.0 200 Ok..Via: SIP/2.0/UDP 59.93.74.129:5060;rport=41010;branch=z9hG4bKBEFDBAE75F0B4E22BC1C19062F5552F8..From: sunkara <sip:9002@public_ip>;tag =587704647..To: <sip:9001@public_ip>;tag=2115499902..Contact: < sip:9001@59.93.74.129:5060>..Record-Route: <sip:192.168.2.75 ;lr=on;ftag=587704647>..Ca ll-ID: 05566ACF-9B85-4F75-ABAC-64118C1C988E@192.168.2.34..CSeq: 35636 INVITE..Content-Type: application/sdp..Server: X-Lite release 1105x..Content-Lengt h: 305....v=0..o=9001 13517410 13530387 IN IP4 59.93.74.129..s=X-Lite..c=IN IP4 59.93.74.129..t=0 0..m=audio 8000 RTP/AVP 1 8 3 98 97 101..a=rtpmap:1 pc mu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a =sendrecv.. -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com +91-9985077535

1 0

[Users] another permissions documentation issue
by Juha Heinanen 04 Sep '06

04 Sep '06

http://www.openser.org/docs/modules/1.1.x/permissions.html does not include allow_refer_to(), but does include allow_uri(), which is not part of release 1.1.0. -- juha

2 1

[Users] ENUM
by Andrew And 04 Sep '06

04 Sep '06

I have problem with ENUM, below I am showing my DNS configuration (Bind version 9): $ORIGIN 1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@server1.com!" . $ORIGIN 8.8.5.2.1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@server2.com!" . When I use DIG "1.8.8.5.2.1.2.3.3.e164.arpa" or " 1.8.8.5.3.1.2.3.3.e164.arpa", I don´t have problem. But When I try "1.8.8.3.2.1.2.3.3.e164.arpa" DIG don´t obtain resolution. I see that I need to add in DNS the tree between 1.2.3.3 and 8.5.2.1.2.3.3: $ORIGIN 2.1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@server1.com!" . $ORIGIN 5.2.1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@server1.com!" . $ORIGIN 8.5.2.1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@server1.com!" . I have a lot of servers and this is not scalable, then I can´t use the tree between 1.2.3.3 and 8.5.2.1.2.3.3. Is there another configuration that I don´t need to use the tree between 1.2.3.3 and 8.5.2.1.2.3.3?

2 1

[Users] Record router problem in NAT RTPproxy.
by raviprakash sunkara 04 Sep '06

04 Sep '06

Hi Users , Good Morning, In local network openser is working fine...... In Behind the NAt the voice is also fine.... But problem is Not hung. upping.... I think problem in Record router header.... Actually it should reply as " Record Router < sip: public.ip:...... > " , But for me it reply as " Record Router <sip:private.ip.... > " That why i think Bye request is not reaching to openser server and its not hung upping.... and my guess is router(firewall) causing this problem , But voice is fine.... -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com +91-9985077535

1 0

[Serusers] SER ,asterisk registration
by Siqhamo Sifo 04 Sep '06

04 Sep '06

How do I let clients registered on SER make calls through asterisk and virse versa

2 1

[Users] Mediaproxy with t38
by Pavel D. Kuzin 04 Sep '06

04 Sep '06

Hello! Somebody using mediaproxy with t38? Can you provide me some configuration examples? Thank you. -- Pavel D.Kuzin System Administrator Nodex ISP St. Petersburg, Russia pk(a)nodex.ru http://nodex.ru

5 7

[Serusers] UA, SER, RTPPROXY in local net, some external clients with public ip
by Rustam Yusupov 04 Sep '06

04 Sep '06

Hello, serusers. I am newbie for SIP. I installed ser, rtpproxy and read manual very many times :-), but... Ser server and some clients in local network : ser - 192.168.100.1 UA1 - 192.168.100.2 UA2 - 192.168.100.3 Some clients have public ip address, not behind firewall... All work fine between local clients, but not work with external clients. I think that local clients register with local ip in Contact field from local network and cannot be fix with fix_nated_ function, and so can't access from external... Simple config or config for rtpproxy from gettingstart not help me... Can anybody help me with simple config for this network configuration? -- Rustam mailto:ru@aspec.ru

2 1

[Serusers] Final ACK problem
by Abdul Qadir 04 Sep '06

04 Sep '06

Hi I am trying to setup SER with nathelper module and portaone proxy. All other messages are sent on the nated IP of both user agents, except final ACK. Which is sent on private IP of called party. Here is the sequence .... I have also pasted my ser.cfg file My ser is running on public IP and both user agents are behind firewall. UA1 SER UA2 INVITE ===============> INVITE ===============> TRYING <=============== Ringing <=============== Ringing <=============== OK <=============== OK <=============== ACK ===============> ACK (sent on UA2's Private IP) ===============> debug=7 fork=yes log_stderror=yes listen=some_public_ip port=5060 check_via=no dns=no rev_dns=no server_signature=no sip_warning=no fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/xlog.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/exec.so" loadmodule "/usr/local/lib/ser/modules/print.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" # ----------------- setting module-specific parameters --------------- modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) modparam("nathelper","rtpproxy_sock", "udp:some_public_ip:port") modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 0) modparam("tm", "fr_timer", 12) modparam("tm", "fr_inv_timer", 30) modparam("tm", "fr_inv_timer_avp", "inv_timeout") # ------------------------- request routing logic ------------------- # main routing logic route { if(!mf_process_maxfwd_header("10")) { log(1, "too many hops\n"); sl_send_reply("483","Too Many Hops"); break; }; if(msg:len >= max_len ) { log(1, "message too big\n"); sl_send_reply("513", "Message too big"); break; }; if(loose_route()) { t_relay(); break; }; if(uri==myself) { if(method=="REGISTER") { fix_nated_register(); force_rport(); if(!save("location")) { log(1, "* * * * * < < < could not save location > > > * * * * *\n"); sl_reply_error(); }; break; }; if(method=="INVITE") { fix_nated_contact(); record_route(); if(!lookup("location")) { sl_send_reply("404", "subscriber is offline. Pls try later"); break; } force_rport(); force_rtp_proxy("l"); fix_nated_sdp("1"); } }; if(!t_relay()) { log(1, "relay error\n"); sl_reply_error(); break; }; } --------------------------------- Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail.

2 1

Re: [Serusers] Bandwidth parameter modification (b:) from ser.cfg
by Greger V. Teigre 04 Sep '06

04 Sep '06

Ah! Try: http://www.ietf.org/rfc/rfc3556.txt?number=3556 for description of b-parameter in SDP. g-) M. Victoria Peláez wrote: > Hi! > I will re-try with nathelper, mediaproxy and mangle modules ...... But > till now I haven't found anything about b parameter. About bandwidth > types I also don't now what that mean, I have written it down because > I have read it in some articles ... > > My problem is that I have one application that exchange a b parameter > (b=CT 140) during INVITE and isn't able to work with any other SIP > based application. And the only suitable explanation we have found is > that the b parameter is disturbing our communication. > > So thank you very much! We'll try something else! > > Best regards, > Victoria > > Greger V. Teigre escribió: >> Victoria, >> SIP and SER do not really look at the SDP payload (ideal world). >> However, nathelper, mediaproxy and the mangle (have a look at it) >> modules allow you to do tricks with SDP fooling the user agents >> involved in the dialog in some way (for example by making both talk >> to a rtp proxy server). >> As for bandwidth types (not sure what you mean here?), maybe somebody >> on the list knows of something, but it's slightly off topic on this >> list. >> g-) >> >> M. Victoria Peláez wrote: >>> Hi! >>> >>> I am a beginner with SIP/SER systems. I have been having a look into >>> ser.cfg, in order to modify some SDP parameters, for instance, >>> fix_nated_sdp to change connection IP, and I have found enough >>> information to do it. But, now, I am traying to change b parameter, >>> but cannot find where to do it (Which module manage it?). I am also >>> looking for a document with a description of all bandwidth types >>> (CT, AS). Could you tell me something about this stuff, please? >>> >>> Thank you very much in advance for your help. >>> >>> I am very sorry if this is a very silly question! >>> >>> Best regards, >>> Victoria >>> _______________________________________________ >>> Serusers mailing list >>> Serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >> >

1 0

[Serusers] RTP Proxy errors
by sip 04 Sep '06

04 Sep '06

So... I followed the instructions on OnSip on how to set up RTPProxy. However, I made a modification because I'm running it on a different machine... changing the modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpptoxy.sock") to modparam("nathelper", "rtpproxy_sock", "udp:XX.XX.XX.XX:22222") I've run RTPproxy on the remote machine as such: ./rtpproxy -f -l XX.XX.XX.XX -s udp:* 22222 And it's running. I start the ser server, and I see a slew of commands hit the RTPProxy console: received command "26742_0 V" sending reply "26742_0 20040107 " received command "26745_0 V" sending reply "26745_0 20040107 " received command "26791_0 V" sending reply "26791_0 20040107 " received command "26797_0 V" sending reply "26797_0 20040107 " received command "26748_0 V" sending reply "26748_0 20040107 " received command "26750_0 V" sending reply "26750_0 20040107 " received command "26751_0 V" sending reply "26751_0 20040107 " received command "26752_0 V" sending reply "26752_0 20040107 " received command "26784_0 V" sending reply "26784_0 20040107 etc, etc ad infinitum After a moment, the SER console spits back: ug 30 14:07:48 death ser[26742]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:48 death ser[26742]: WARNING: rtpp_test: can't get version of the RTP proxy Aug 30 14:07:48 death ser[26742]: WARNING: rtpp_test: support for RTP proxy has been disabled temporarily Aug 30 14:07:48 death ser[26745]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:48 death ser[26745]: WARNING: rtpp_test: can't get version of the RTP proxy Aug 30 14:07:48 death ser[26745]: WARNING: rtpp_test: support for RTP proxy has been disabled temporarily Aug 30 14:07:49 death ser[26791]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:49 death ser[26791]: WARNING: rtpp_test: can't get version of the RTP proxy Aug 30 14:07:49 death ser[26791]: WARNING: rtpp_test: support for RTP proxy has been disabled temporarily Aug 30 14:07:49 death ser[26797]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:49 death ser[26797]: WARNING: rtpp_test: can't get version of the RTP proxy Aug 30 14:07:49 death ser[26797]: WARNING: rtpp_test: support for RTP proxy has been disabled temporarily Aug 30 14:07:49 death ser[26748]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:49 death ser[26748]: WARNING: rtpp_test: can't get version of the RTP proxy Aug 30 14:07:49 death ser[26748]: WARNING: rtpp_test: support for RTP proxy has been disabled temporarily Aug 30 14:07:49 death ser[26750]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:49 death ser[26751]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:49 death ser[26752]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:49 death ser[26784]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:49 death ser[26789]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy Aug 30 14:07:49 death ser[26792]: ERROR: send_rtpp_command: timeout waiting reply from a RTP proxy etc, etc, ad infinitum I've done a search on this on Google and found a couple of people with the same questions and NO people responding with answers. Any ideas what's going on? N.

4 12

Re: [Users] Error when installing openseradmin.Please help..Please
by Ferianto siregar 04 Sep '06

04 Sep '06

Dear Nick, Thank you very much for your kind-hearted to help me. Thank you. Nick, If I can make a wish for some help to you, would you mind help me in these questions: 1. What do you mean by installing mysql? because when I run # rpm -qa|grep mysql, I got the version of mysql are : mysql-3.23.54a-11 mysql-server-3.23.54a-11 php-mysql-4.2.2-17 Do I need install mysql again? If I have to, How? Please 2. Is there any configuration in websever? or I just have to restart the webserver by using "# /etc/rc.d/init.d/httpd restart"? If I have to configure the webserver, Would you mind share the file..Please.. Please Help me Nick..Please...because my skill is too lack. Please... I am so sorry for bothered and waste your time Nick. I am Sorry. Thank you very much Regards, Ferianto nick <nick(a)mobilia.it> wrote: Ferianto siregar wrote: > Dear all, > > Thank you very much to all of you for your kind-hearted to reply my > questions. > Thank you very much. > > All, Now I try to build the openser admin in my openser server. I have > tried to build it for many times, but never successfully. I know that my > skill is too low. So I try to ask it to you. Please help me..Please. > > I have tried to build the openser admin packet by using README .But, I > am confused. Maybe it is because of my lack skill to understand it. > Here is the task that I have done: > 1. I have download Ruby and Rails packet. The versions that I have got > were rubygems-0.9.0.tar.gz and rails-1.1.6.tar.gz > > a. Untar the Rubygems packet > # tar -zxvf rubygems-0.9.0.tar.gz > # cd rubygems > # ruby setup.rb No errors on the install? > b. Untar the Rails packet > # tar -zxvf rails-1.1.6.tar.gz > # cd rails > # ruby script/server If you have gems installed you can just gem install rails --include-dependencies > > c. Untar the openseradmin packet > # tar -xjvf openseradmin-0.3.tar.bz2 > # cd openseradmin-0.3 > That's correct, but without rails installed it probably won't work.. > > d.Logging into the database server > #mysql -u root -p > > Connect openser; > > GRANT ALL ON openser.* TO 'openser'@'localhost' IDENTIFIED > BY 'openserrw'; > > USE openser; > > SOURCE /root/openseradmin-0.3/db/database_tables.sql; > > EXIT > > Here is the confusing problem that I got : > 1. When I try to run "gem install mysql", I got the error message. So, I > can not execute it. What`s wrong? Try gem install mysql --with-mysql-config=/path/to/mysql/lib > 2. When I try to access it by using web browser by typing > "http://localhost:3000", I get nothing. What`s wrong? You need to start the web server, but without mysql, it won't work.. > 3. Would you mind tell me if all the task that I have done correct or > not? I need some suggestion or adviseable. Please.... Install rails, then install mysql, then start the server, it should work fine... > > > I do hope anybody can help me. Please... > > > Regards, > > > > Ferianto > > ------------------------------------------------------------------------ > Yahoo! Messenger with Voice. Make PC-to-Phone Calls > > to the US (and 30+ countries) for 2¢/min or less. > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users --------------------------------- Stay in the know. Pulse on the new Yahoo.com. Check it out. --------------------------------- How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.

2 1

[Serusers] Help for sn2400 call
by pokerino76 04 Sep '06

04 Sep '06

The mediagateway it is installed in a webfarm of BT (British Telecom) , while the sn2400 it is a sipgateway. I do not succeed to carry out calls of the net outside, while I succeed to receive them. What I must make? thanks. ___________________________________________________________________ Prestiti da 15 istituti. Tan dal 5,90% per richieste entro il 31 Agosto. Servizio gratuito. http://click.libero.it/mutuionline1

1 0

[Serusers] Setting Accounting Flags:-(
by ravi reddy 04 Sep '06

04 Sep '06

Hi users , I am using ser.cfg from onsip.org and because i am not a programmer (means i dont understand the language)I am using the config file as i downloaded , before i gave a lot messages regarding this getting of multiple records in radius database as well as in SER mysql(ac) database also ; i am attaching the ser.cfg below please let me know where to set flag so that i can get single record per call ;-) thank you. debug=4 fork=yes log_stderror=yes listen=My Ip address # INSERT YOUR IP ADDRESS HERE port=5060 children=5 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_mode=766 fifo_db_url="mysql://ser:heslo@localhost/ser" sip_warning=yes server_signature=no 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/domain.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/xlog.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/acc.so" modparam("auth_db|permissions|domain|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("xlog", "buf_size", 8192) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("avpops", "avp_url", "mysql://ser:heslo@localhost/ser") modparam("avpops", "avp_table", "usr_preferences") # -- acc params -- # set the reporting log level modparam("acc", "log_level", 1) # number of flag, which will be used for accounting; if a message is # labeled with this flag, its completion status will be reported modparam("acc", "radius_flag", 11 ) modparam("acc", "radius_missed_flag",12) modparam("acc", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "service_type", 15) modparam("acc", "db_flag",13) modparam("acc", "db_table_acc", "acc") 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("81.21.33.35:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- setflag(11); if (loose_route() ){ ------------------------------------------------------>>>>>>>>>>here i set flag for radius accounting ; # mark routing logic in request #append_hf("P-hint: rr-enforced\r\n"); t_relay(); route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403", "Forbidden"); }; break; }; if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP 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()) { xlog("L_WARN", "[%Tf]: possible uri spoofing attempt!; method=%rm; IP=%is; from=%fu; to=%tu\n"); sl_send_reply("403", "Forbidden"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ---------------------------------------------------------- # INVITE Message Handler # ---------------------------------------------------------- if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (uri=~"^sip:1[0-9]{8}@") { strip(1); }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (uri=~"^sip:00[0-9]*@") { route(4); route(5); break; }; if (avp_db_load("$ruri/username", "s:callfwd")) { setflag(22); avp_pushto("$ruri", "s:callfwd"); route(6); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{8}@") { route(4); route(5); break; }; sl_send_reply("404", "User Not Found"); break; }; if (avp_db_load("$ruri/username", "s:fwdbusy")) { if (!avp_check("s:fwdbusy", "eq/$ruri/i")) { setflag(26); }; }; if (avp_db_load("$ruri/username", "s:fwdnoanswer")) { if (!avp_check("s:fwdnoanswer", "eq/$ruri/i")) { setflag(27); }; }; t_on_failure("1"); route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("pstn gateway ip"); avp_write("i:30", "inv_timeout"); t_on_failure("1"); route(4); route(1); } route[6] { # ---------------------------------------------------------- # Call Forwarding Handler # # This must be done as a route block because sl_send_reply() cannot be # called from the failure_route block # ---------------------------------------------------------- if (uri=~"^sip:1[0-9]{8}@") { strip(1); }; lookup("aliases"); if (uri!=myself) { if (!isflagset(22)) { append_branch(); }; route(4); route(1); break; }; if (uri=~"^sip:00[0-9]*@") { route(4); route(5); break; } if (uri=~"^sip:+[0-9]{3}[0-9]*@") { route(4); route(5); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{8}@") { #local pstn route(4); route(1); break; }; sl_send_reply("404", "User Not Found"); }; route(4); 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(); }; } .........................)

1 0

[Users] Error when installing openseradmin.Please help..Please
by Ferianto siregar 04 Sep '06

04 Sep '06

Dear all, Thank you very much to all of you for your kind-hearted to reply my questions. Thank you very much. All, Now I try to build the openser admin in my openser server. I have tried to build it for many times, but never successfully. I know that my skill is too low. So I try to ask it to you. Please help me..Please. I have tried to build the openser admin packet by using README .But, I am confused. Maybe it is because of my lack skill to understand it. Here is the task that I have done: 1. I have download Ruby and Rails packet. The versions that I have got were rubygems-0.9.0.tar.gz and rails-1.1.6.tar.gz a. Untar the Rubygems packet # tar -zxvf rubygems-0.9.0.tar.gz # cd rubygems # ruby setup.rb b. Untar the Rails packet # tar -zxvf rails-1.1.6.tar.gz # cd rails # ruby script/server c. Untar the openseradmin packet # tar -xjvf openseradmin-0.3.tar.bz2 # cd openseradmin-0.3 d.Logging into the database server #mysql -u root -p > Connect openser; > GRANT ALL ON openser.* TO 'openser'@'localhost' IDENTIFIED BY 'openserrw'; > USE openser; > SOURCE /root/openseradmin-0.3/db/database_tables.sql; > EXIT Here is the confusing problem that I got : 1. When I try to run "gem install mysql", I got the error message. So, I can not execute it. What`s wrong? 2. When I try to access it by using web browser by typing "http://localhost:3000", I get nothing. What`s wrong? 3. Would you mind tell me if all the task that I have done correct or not? I need some suggestion or adviseable. Please.... I do hope anybody can help me. Please... Regards, Ferianto --------------------------------- Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.

2 1

[Serusers] Proxy and Presence Agent support
by Henry Duan 04 Sep '06

04 Sep '06

hi, I'm new to SER. I need to use SER as a proxy and a PA as well. I'm wondering if I could just download the package unstable version of SER with presence<http://ftp.iptel.org/pub/ser/presence/> from iptel.org website. BTW, I'm using Linux Fedora 5, not debian. I'm wondering if there is any problem for this. Thanks a lot! henry

3 2

[Users] Problem with ENUM
by Andrew And 03 Sep '06

03 Sep '06

Below I m showing my DNS configuration (Bind version 9): $ORIGIN 1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@server1.com!" . $ORIGIN 8.8.5.2.1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@server2.com!" . When I use DIG "1.8.8.5.2.1.2.3.3.e164.arpa" or " 1.8.8.5.3.1.2.3.3.e164.arpa", I don´t have problem. But When I try "1.8.8.3.2.1.2.3.3.e164.arpa" DIG don´t obtain resolution. I see that I need to add in DNS the tree between 1.2.3.3 and 8.5.2.1.2.3.3: $ORIGIN 2.1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@server1.com!" . $ORIGIN 5.2.1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@ server1.com!" . $ORIGIN 8.5.2.1.2.3.3.e164.arpa. * NAPTR 1 1 "u" "E2U+sip" "!(^.*$)!sip:\\1@server1.com!" . I have a lot of servers and this is not scalable, then I can´t use the tree between 1.2.3.3 and 8.5.2.1.2.3.3. Is there another configuration that I don´t need to use the tree between 1.2.3.3 and 8.5.2.1.2.3.3?

1 0

[Serusers] SER+Asterisk
by Siqhamo Sifo 03 Sep '06

03 Sep '06

Where can I find docs on asterisk and ser integration.

1 0

[Users] Need help for openser admin installation
by Ferianto siregar 03 Sep '06

03 Sep '06

Dear all, Thanks to every body in this forum. Thank you very much I tried to build openser admin in my openser server. But, I am confused when i try to read the installation note. I do hope anybody can help me to understand it. These are my questions about it: 1. As I read, the openser admin need ruby packet. For installation of ruby, after we untar the packet, does we just type "gem mysql install" command?Because as i read from the installation openser admin, there are only this command. 2. When I try to build the openser admin, i got the error message. Here is the message : ERROR 1064 (42000): You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'VIEW usr_preferences_view AS SELECT id, username, domain, attribute, type AS att' at line 1 As I read from the message in this mailinglist before, this is because of the mysql version. I use mysql 3. Do i need use mysql 5? Where I can get the suitable varsion? That all my question. I do hope any body can help me. Thank you very much. Regards, Ferianto --------------------------------- Stay in the know. Pulse on the new Yahoo.com. Check it out.

1 0

[Serusers] Help for sn2400 call
by pokerino76 02 Sep '06

02 Sep '06

I have shaped the sn2400 for the ser, succeed to make telephone calls of the net inside ,the calls phone inside working, but I do not succeed to calls phone from the outside, which it must be the configuration of ser.cfg or the other? . Thaks for help.

2 1

[Serusers] SER+Asterisk
by Siqhamo Sifo 02 Sep '06

02 Sep '06

I have ser sitting on my iptables nat box and my asterisk box on the lan . Ser does forwarding so that any requests (register,invite,ack,...) to the nat box at 5060 r sent to my asterisk box on the lan .I can register from outside to my asterisk box but there is only one way audio , reason being that when the asterisk box send a sip packet whith session description the sdp part of the sip packet is not natted . I have tried : fix_nated_contact(); fix_nated_sdp("3");

3 2

[Serusers] Presence modules and LCS
by harrygaillac-sip＠yahoo.fr 02 Sep '06

02 Sep '06

Hello, Does ser presence modules are able to work with LCS micosoft ? http://www.microsoft.com/office/livecomm/prodinfo/compare.mspx#ECD anybody is experienced with LCS to answer ? Regards Harry p4.vert.ukl.yahoo.com uncompressed/chunked Fri Sep 1 12:13:36 GMT 2006 ___________________________________________________________________________ Découvrez un nouveau moyen de poser toutes vos questions quelque soit le sujet ! Yahoo! Questions/Réponses pour partager vos connaissances, vos opinions et vos expériences. http://fr.answers.yahoo.com

1 0

[Users] RTPProxy non working?
by Federico Giannici 02 Sep '06

02 Sep '06

I'm trying to install rtpproxy. It usually works, but there are some cases that don't work. For example, when a Cisco PSTN gateway is called, RTP traffic is proxied in one direction only. I think the problem is in the handling of the reply received from the gateway: it have two "c" lines but only one is modified. Here it is the body of the original reply: v=0. o=CiscoSystemsSIP-GW-UserAgent 2199 2489 IN IP4 83.211.2.216. s=SIP Call. c=IN IP4 83.211.2.217. t=0 0. m=audio 18850 RTP/AVP 8 13 101. c=IN IP4 83.211.2.217. a=rtpmap:8 PCMA/8000. a=rtpmap:13 CN/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11. a=direction:passive. And here it is the body modified by rtpproxy: v=0. o=CiscoSystemsSIP-GW-UserAgent 2199 2489 IN IP4 83.211.2.216. s=SIP Call. c=IN IP4 83.211.2.217. t=0 0. m=audio 35086 RTP/AVP 8 13 101. c=IN IP4 195.120.250.49. a=rtpmap:8 PCMA/8000. a=rtpmap:13 CN/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11. a=direction:passive. a=nortpproxy:yes. Only the secondo "c" line contain the IP of our rtpproxy. I don't know RTP enough to understand if the problem is in rtpproxy or in the gateway. Thanks for any info. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

3 4

[Users] openser admin question..
by nick 01 Sep '06

01 Sep '06

Is an error like this normal when doing the db install? ERROR 1064 (42000): You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'VIEW usr_preferences_view AS SELECT id, username, domain, attribute, type AS att' at line 1 By the way, while ruby on rails might be a great development platform, it's a right pain in the behind to install.

2 5

[Users] codecs problem...
by raviprakash sunkara 01 Sep '06

01 Sep '06

Hi Users...... A doubt For me in Codec.... I'm Using openSER and RTPProxy openser didn't take the Media tranmmision , it transmmit to RTPproxy or mediaproxy. Media(voice) taking place in RTP proxy... UA1 using the G729 codec -------> openser server -------------------> UA2 using G711u for above case it will not generate the porper handshanking............ But RTP Proxy may has transmiits the media , because it has codecs in ( media proxy or rtpproxy) For a realtime producttion, UAs may having the differnet codecs. plz resolve my doubt..... -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com +91-9985077535

1 0

[Users] Integration of Openser with CDRTool: Installation errors
by Alberto Pacifico 01 Sep '06

01 Sep '06

Hi all, I'm trying to install CDRTool (http://www.ag-projects.com/CDRTool.html) and some errors happened. When I try to access the CDRTool webpage, I'm able log in the page, but when I try to enter in the "call search page" some warnings appears: "Warning: main(SOAP/Client.php): failed to open stream: Arquivo ou diretÃ³rio nÃ£o encontrado in /var/www/CDRTool/rating_lib.phtml on line 4" "Fatal error: main(): Failed opening required 'SOAP/Client.php' (include_path='.:/opt/lampp/lib/php') in /var/www/CDRTool/rating_lib.phtml on line 4" In the file "rating_lib.phtml", the line 4 is: "require_once("SOAP/Client.php");" The file Client.php does not exist... But a file named "client.php" exists..and into it, theres an instruction calling "Client.php": "include("SOAP/Client.php");" An additional information is that I'm using the XAMPP Server Apache. Please, Can you help me? Thanks in advance! Alberto __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Users] openser+asterisk+voicemail password
by ram 01 Sep '06

01 Sep '06

Hi iam integrating with asterisk voice mail with Openser when we are creating openser views with asterisk the secret going to be Binary(0);, how can i change to numeric ? why i need is, take example If the user Phone logged in and the user have person voicemessage box he only can access his voice messages, when he dial secret If not any one can come and access voice message by dialing access number. with the config seen in the examples, person can dial *98 and hear all the messages and delete too. so for the security reason, how can i change the secret, iam trying to alter the table, but its say;s in view mode. any suggestions. Ra,

1 0

[Users] Question about voip-info.org's OpenSER/RTPProxy configuration example
by Jeremie Le Hen 01 Sep '06

01 Sep '06

Hi list, I was exploring the example configuration file in the following URL: http://voip-info.org/wiki/view/OpenSER+And+RTPProxy The following block bewilders me: % failure_route[1] { % if (isflagset(6) || isflagset(5)) { % unforce_rtp_proxy(); % } % } How a flag happens to be turned on there ? 6 is the registrar's nat_flag but no lookup() is performed. Furthermore, the tm module doesn't even depend on the registrar module. Flag 5 is set in route[2] which is obviously not called from failure_route[1]. Is there some black magic around here ? Thank you. Regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

2 3

[Users] RTPProxy failover... load-balancing ?
by Jeremie Le Hen 01 Sep '06

01 Sep '06

Hi list, I was wondering whether I could use multiple rtpproxy daemons along one OpenSER instance, in order to both provide a failover mechanism as well as load-balacing. Google immediately pointed me out the following thread: http://www.openser.org/pipermail/users/2005-October/001071.html It happens that openser/modules/nathelper/nathelper.c contains the following comment: % 2005-02-24 Added support for using more than one rtp proxy, in which % case traffic will be distributed evenly among them. In addition, % each such proxy can be assigned a weight, which will specify % which share of the traffic should be placed to this particular % proxy. % % Introduce failover mechanism, so that if SER detects that one % of many proxies is no longer available it temporarily decreases % its weight to 0, so that no traffic will be assigned to it. % Such "disabled" proxies are periodically checked to see if they % are back to normal in which case respective weight is restored % resulting in traffic being sent to that proxy again. % % Those features can be enabled by specifying more than one "URI" % in the rtpproxy_sock parameter, optionally followed by the weight, % which if absent is assumed to be 1, for example: % % rtpproxy_sock="unix:/foo/bar=4 udp:1.2.3.4:3456=3 udp:5.6.7.8:5432=1" At first glance, I would say the aforementioned comment states that nathelper achieves both failover and weighted load-balancing. Can anyone confirm this ? Ultimately, I would like to do use the following arch : ________+----------+ /________| RTPProxy | +---------+ // +----------+ | OpenSER |_____// +---------+ /\_________+----------+ / \ ______| RTPProxy | +---------+ / \/ +----------+ | OpenSER |__/_____/\ +---------+ \ \_____+----------+ \____________| RTPProxy | +----------+ If I understand correctly how it works, this should be possible to use N OpenSER instances along with M RTProxy instances. Am I right ? Thank you. Best regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

1 0

[Users] Simple configuration
by Vladimir Orehov 01 Sep '06

01 Sep '06

Hello, I'm new to this server, and I ask for help! I don't undertand for now how to configure openSER. Can someone send me simple configuration file "openser.cfg" wich included: -- accounting loggoing to a specific file ( like "/home/agb/acc.log") -- authentication from simple text file (if possible), or using SQL -- routing numbers started with 00 to specific gateway with ip (123.123.123.123) Another question is: how to enable debugs of this server? Please reply ASAP and best regards, Vladimir Orehov SIA Velkoms Support engineer

2 1

[Users] Question about voip-info.org's OpenSER/RTPProxy configuration example
by Jeremie Le Hen 01 Sep '06

01 Sep '06

Hi list, I was exploring the example configuration file in the following URL: http://voip-info.org/wiki/view/OpenSER+And+RTPProxy The following block bewilders me: % failure_route[1] { % if (isflagset(6) || isflagset(5)) { % unforce_rtp_proxy(); % } % } How a flag happens to be turned on there ? 6 is the registrar's nat_flag but no lookup() is performed. Furthermore, the tm module doesn't even depend on the registrar module. Flag 5 is set in route[2] which is obviously not called from failure_route[1]. Is there some black magic around here ? Thank you. Regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

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[Users] Error in running serweb. Please help me..Please
by Ferianto siregar 01 Sep '06

01 Sep '06

Dear all, First of all, I would like to say thanks to all of you for your help and suggestion to solve my problem untill now. Thank you very much. Now, I try to install serweb_2004-07-27 in my openser server. I have read the installation note that I got from the serweb packet carefully. So, I can build the serweb in my openser server successfully. It means that it can display a login screen in the browser. But, I have a problem when I try to login by using username "admin" with password "openserrw". (This is based on the installation note that I have read before) I got error message. This is the error message : Database error: Invalid SQL: select username, perms from subscriber where phplib_id = '721153ca8942ea1a08900f5b61b69a6c' MySQL Error: 1054 (Unknown column 'perms' in 'field list') Session halted. As I try to understand from the error message, it says that there is no "perms" column. So I try to check it in the subscriber table in mysql.But, I did not find "perms" column in the subscriber table. I just see these columns in the subscriber table: -phplib_id -last_name - sendnotification - rpid -username - phone - greeting -domn -domain - email_address - ha1 -uuid -password - datetime_created - hdb1 -first_name - datetime_modified - allow_find -confirmation - flag - timezone What should i do to solve this problem? What should I do if there is no "perms" column in the subscriber table? I do hope anybody can give me a suggestion because this is my first time to build the serweb and mysql. Please help me....Please... Note: I user openser1.1.0-tls and serweb_2004-07-27 As I see from the error message that the phplib_id is exactly matched with phplib_id where username "admin" with password "openserrw" (from the subscriber table). Thank you very much Regards, Ferianto --------------------------------- Get your email and more, right on the new Yahoo.com

3 2

[Users] Migrate from SER to open SER
by ram 01 Sep '06

01 Sep '06

Hi all I have VoIP CDrouter running But i have seen lot of limitation How can i migrate from SER to Open Ser with out lossing any data of users information and CDR Help will be appriciated Ram

2 1

[Users] Incorrect Route header
by Jeremie Le Hen 01 Sep '06

01 Sep '06

Hi list, I am running a NATing proxy. Everything works fine, except that some of my neighbour proxies send me BYE requests with an incorrect Route header containing their own IP address. The result is naturally that my proxy sends back the request and the call is never ended. I know this is a wrong behaviour and I have contacted the remote administrator to have him fix his proxy. But in the meantime I would like to find a workaround for my users. I tried to do the following, before calling loose_routing(). I am not sure if it's relevant or not nevertheless: % if (method == "BYE") { % # XXX Remove buggy Route header from BYE packets. % xlog("L_ALERT", "Removing buggy Route header from BYE request"); % remove_hf("Route"); % lookup("location"); % } However, I raised the verbosity level and found that the Route header is read as soon as the packet is received, therefore the above block is not really useful. Thank you for your ideas. Regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

2 2

[Users] VIA Parameter?
by Ricardo Martinez 01 Sep '06

01 Sep '06

Hello list. Sorry to ask this question here, but i can't find the answer in the RFC3261 Is this "VIA" permited according to the RFC? Via: SIP/2.0/UDP 69.20.6.205:15060, SIP/2.0/UDP 69.1.229.69:4081;branch=z9hG4bK-58e9ccfcef0f4c99915b8007cb9713e1-1156793814 It seems to be two VIA's in ONE line... Thanks Regards, Ricardo.-

3 2

[Users] SIP Redirection in Openser
by Padmaja RV 01 Sep '06

01 Sep '06

Hi all, I have tried implementing SIP redirection as mentioned in the example script below: route{ # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { log("REGISTER"); sl_send_reply("200", "ok"); break; }; # rewrite current URI, which is always part of destination ser rewriteuri("sip:parallel@iptel.org:9"); # append one more URI to the destination ser append_branch("sip:redirect@iptel.org:9"); # redirect now sl_send_reply("300", "Redirect"); } But that does redirection only for that account as the account is included in the code. I need the code to be able to redirect any call. Can you tell me how I can generalize the code so that redirection works for any account I choose to redirect? Thanks in advance, Padmaja

2 1

[Users] Authentication in Trunk-Mode
by Martin Klisch 01 Sep '06

01 Sep '06

Hey, are there working ways for authentication with sip trunks? i've read cisco gateways (PGW) and cisco call manager can do this. anyone knows if openser+asterisk support it too?

3 4

[Users] About 'location' table
by Michel Zenone 01 Sep '06

01 Sep '06

Hi! I have registered, thanks to openserctl, an user :'sip:testeur@192.168.0.102'. But when I tyr to reach him, the transaction isn't considered as local processed one. In location table there is: +username + domain + contact + |testeur | | sip:testeur@192.168.0.120:5080 | +---------+--------+--------------------------------+ -192.168.0.120 is the IP address of the user's phone. -192.168.0.102 is the openser IP address. The openser.cfg is as the initial openser.cfg. Thanks, Michel

2 1

[Users] OpenSER AstLinux build posted for testing
by Kristian Kielhofner 01 Sep '06

01 Sep '06

Hello OpenSer Users, It seems that many of you are also Asterisk users or at least subscribe to asterisk-users. Some of you might recognize AstLinux... Just in case you don't: http://www.astlinux.org ---and--- http://www.sf.net/projects/astlinux [in transition] Anyways, I have long hoped to build a version of AstLinux that includes OpenSer. Basically, what I have done is taken astlinux-trunk, add openser, add rtpproxy, de-select the Asterisk packages, and build! What I got was 11mb (gzipped) disk images ready to be written to CF cards/hard disks. As usual, I have created two images: *net4801.img.gz is for the Soekris net4801 SBC *i586.img.gz is for just about any i586 and higher machine There is also an ISO image ready to boot on or install to most i586 or higher machines. You guessed it, it is the one with .iso in the filename. It includes both the net4801 and i586 images. These are 48mb gzipped disk images, with GRUB bootloader and ext2 root filesystem. If you want to write them to disk, try this: gunzip [image.gz] dd if=[output file.img] of=/dev/sdb # <--- VERIFY THIS DEVICE Where /dev/sdb is where your USB CF flash adapter is. If you want access to the ext2 filesystem, try this: gunzip [image.gz] mkdir /mnt/image mount -t ext2 -o loop,offset=32256 [image.gz] /mnt/image If you are curious about how these images were created, take a look at SVN on the astlinux project page. The OpenSer version used for these builds was openser-1.1.0-notls. I built every module that did not depend on an external library that I don't have yet (sorry, no mysql, jabber, or radius yet). I need help!!! For the time being, I don't have mysql. Also, because AstLinux is targeted for compact flash, mysql isn't a good idea anyways. I was not able to get a working configuration in the time alloted using DBTEXT. Does anyone have a sample config that I can put in this build? As of now, openser, rtpproxy, openser modules, and all of the astlinux-style init scripts are done. All I need is a good sample config file to make it "plug and play". I also need testing! My current to-do list: - add mysql (and the openser modules, of course) - add radius support (freeradius?) - jabber/xmpp (depending on size, complexity, etc) - fix astshape (iproute2-based traffic shaper) RTP TOS values to be the same that rtpproxy uses, and standardize this across all astlinux builds - COME UP WITH A NAME. I have no idea what to call this thing. Any suggestions? SERlinux (or something like it) is too obvious and boring! You can grab the goodies here: http://www.krisk.org/openser/ Regardless, I hope that someone finds this interesting or useful. Let me know what you think! -- Kristian Kielhofner

3 2

[Users] icmp port unreachable
by Helge Waastad 01 Sep '06

01 Sep '06

Hi, (just a little bit on the side-bar) I've had some discussion with the developer of twinkle (which is, BTW, a very nice linux softphone) regarding handeling icmp port unreachable and switching to a lower priority SIP router (by DNS SRV). It seems like the dist I'm using (Dapper, kernel 2.6.15-26) will not easily allow the applications to receive the icmp on an unconnected udp socket, hence failover will not work. However on an 2.6.11.4-21.11 kernel it works. Does anyone know if there are any kernel opts which should be added for having this kind of support? Just my thoughts this morning......

2 1

[Users] Openser + Asterisk for P2P calls
by Lucio Maciel 01 Sep '06

01 Sep '06

Hi, Hi, im using an Openser integrated with Asterisk to route PSTN calls, this is working great, but i also want to route P2P calls to Asterisk like this UA1 -> Openser -> Asterisk (calls DIAL(SIP/xxxxx@openser) -> Openser -> UA2, to use its mediaproxy, and account all calls. When UA1 calls UA2, everything goes fine, UA2 receive the INVITE and sends the OK to UA1, the only thing is that UA2 waits for an ACK from the OK that never comes. Have someone tried to do this? How can i configure openser and asterisk for this? Thanks Lucio Maciel

2 1

[Users] IP authentication
by Brian Fertig 01 Sep '06

01 Sep '06

greetings, I have digest and basic auth setup. I am trying to figure out how to setup IP auth. I have read a lot of documentation but I can't seem to find anything to help me understand how to accomplish this. Can someone point me in a direction to some information on how to set this up? Brian

6 5

[Users] 1.1.x/permissions.html is not up to date
by Juha Heinanen 01 Sep '06

01 Sep '06

for some reason web page http://www.openser.org/docs/modules/1.1.x/permissions.html is not up to date. for example, 1.3.12. tag_col (string) 1.3.13. peer_tag_avp (string) are missing. they do exist in rel_1_1_0 source. -- juha

2 1

[Serusers] extra things confusing auth_radius?
by Natambu Obleton 01 Sep '06

01 Sep '06

Ok. I have two clients setup. One is X-Lite and the other is a polycom 601. I am doing radius authentication, but for some reason it doesn't work with the polycom 601 phone. In the debug the only thing that is different about them is something in the uri. Aug 30 09:16:51 laplata ser[25968]: DEBUG: get_hdr_field: <To> [72]; uri=[sip:311@voip.fasttrackcomm.net] Works, but.. Aug 30 09:17:53 laplata ser[25970]: DEBUG: get_hdr_field: <To> [45]; uri=[sip:311@voip.fasttrackcomm.net;user=phone] <mailto:%5bsip:311@voip.fasttrackcomm.net;user=phone%5d> Doesn't. Like it doesn't even send an authentication request to my radius server. I think it is the ';user=phone' part that is confusing it? So what I want to do is remove that part from the URI as a test. I just added a subst to remove it, but still isn't working with radius. It acts like the realm is no longer local. Here is my config file: http://www.natambu.com/serconf.html Natambu Obleton Network Engineer FastTrack Communications nobleton(a)fasttrackcomm.net (970) 247-3366 office (970) 247-2426 fax

2 3

[Serusers] SIP Contact header field issue incase of NAT
by Abdul Qadir 01 Sep '06

01 Sep '06

Hi All, I have problem connecting Nokia Phone with SER. The problem is When SER receives the registration request from the Nokia phone. The registration request contain the phone's private IP (192.xxx.xx.xx), since we use wifi for connection. Our proxy detects that since this is a private IP handles the NAT translation and determines the phone public IP address (e.g., 202.xxx.xx.xx, port xx). Our SIP proxy informs the phone that registration is successful but the contact information is changed from userid(a)192.xxx.xxx.xx to userid(a)202.xxx.xxx.xx, port xx. On recieving this changed contact info the Phone rejects 200 Ok. We tried to contact nokia people and they said that its part of RFC3261 to reject such contacts. After this I have also gone through RFC 3261 section "10.2.4" and it seems that phone is doing rite. Can any body tell me why SER is changing the contact address in case of NAT. It would be higly appreciated if you can give me some reference from some RFC or other document which I can show to nokia people. Best Regards, Abdul Qadir --------------------------------- How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.

3 2

[Serusers] Bandwidth parameter modification (b:) from ser.cfg
by "M. Victoria Peláez" 01 Sep '06

01 Sep '06

Hi! I am a beginner with SIP/SER systems. I have been having a look into ser.cfg, in order to modify some SDP parameters, for instance, fix_nated_sdp to change connection IP, and I have found enough information to do it. But, now, I am traying to change b parameter, but cannot find where to do it (Which module manage it?). I am also looking for a document with a description of all bandwidth types (CT, AS). Could you tell me something about this stuff, please? Thank you very much in advance for your help. I am very sorry if this is a very silly question! Best regards, Victoria

2 1

[Serusers] mediaproxy
by Javier Ramirez 01 Sep '06

01 Sep '06

Hi, I would like to know how to configure the cfg script, to configure the media proxy so that it can nat the calls between users that don't have public IP, but not when they have public IP. best Regards Javier

2 1

[Users] openser routing lookup question
by Dave Logan 01 Sep '06

01 Sep '06

Hey all, I have what may or may not be a simple situation. I've read through dozens of forum entries, dozens more list mails, and a handful of "tutorials" for openser, and I'm still missing something I think. I'm not afraid of the documentation, but neither the "core cookbook" nor the modules documentation had enough detail that I could catch on to what's going on. Let me lay out my scenario, and maybe somebody can point to a particular document or solution I can follow up on. I have multiple asterisk boxes: four as media gateways with TDM trunks, one feature server (VM), and three registrars. So needless to say, I have a lot of dialplan in there just for calls that stay in my network. What I'd like is to introduce a single openser box in there, and for any call that any of those asterisk machines gets, it sends it to openser, which then just calls a simple script (which I've already more or less written) to know what to do. So for example, I own the 555-1000(-1099) DID block DID 555-1000 is registered to machine1 DID 555-1001 is registered to machine2 In openser, I'm envisioning a call like such (php style notation): my $number = 5551000 in openser: $newdestination = exec('lookupdestination($number)'); now $newdestination is 5551000@machine1 or my $number = 5551001 in openser: $newdestination = exec('lookupdestination($number)'); now $newdestination is 5551001@machine2 Then I can just route it in the normal openser way. So there, my lookupdestination script knows all the numbers in my central database and where they're located, and on an unknown number assumes it's outside our network so just sends it to the gateway as is (5552000@gw1) or so. Thanks for any pointers at all, dave -- Dave Logan http://www.digitalcoven.com/ "Do let's pretend that I'm a hungry hyena, and you're a bone!" - Alice

2 1

[Users] Help in dailplan in asterisk
by raviprakash sunkara 01 Sep '06

01 Sep '06

Hi Users, I'm new to Asterisk, and I'm working with openSER , For Call Routing I'm using the OpenSER and for PBX, Voicemail and Conferencing I"m using the Asterisk. Now i'm planing for Voicemail, openser listen on 192.168.2.75:5060 Asterisk listen on 192.168.2.76:5060. in Extension.conf [from-sip[ exten => 9001,1,Ringing exten => 9001,2,Voicemail(9001) Just I wnat to leave the voicemail in voice mail box, Can anybody help me......... below message What it is ?................ Executing Ringing("SIP/9001-0984c1e8", "") in new stack -- Executing VoiceMail("SIP/9001-0984c1e8", "9001") in new stack Aug 31 19:31:10 ERROR[7869]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on (err 2002). Check debug for more info. Aug 31 19:31:10 WARNING[7869]: app_voicemail.c:2412 leave_voicemail: No entry in voicemail config file for '9001' Aug 31 19:31:10 WARNING[7869]: pbx.c:1700 pbx_extension_helper: No application 'Hungup' for extension (from-sip, 9001, 3) == Spawn extension (from-sip, 9001, 3) exited non-zero on 'SIP/9001-0984c1e8' -- Incoming call: Got SIP response 479 "Regretfully, we were not able to process the URI (479/SL)" back from 192.168.2.75 -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology Kondapur, Hi-tech city, Hyderabad. www.hyperion-tech.com +91-9985077535

1 0

[Users] avp_pushto() changed???
by Greg Fausak 31 Aug '06

31 Aug '06

This doesn't work for me any more: avp_pushto("$P-Asserted-Identity", "$avp(s:tmp_uri)"); I get this: Aug 31 13:42:00 www openser[21792]: ERROR:avpops:fixup_pushto_avp: unsupported destination "$P-Asserted-Identity"; expected $ru,$du,$br Has anything changed? -g -- Greg Fausak greg(a)thursday.com

3 2

[Users] mediaproxy
by Javier Ramirez 31 Aug '06

31 Aug '06

Hi, I would like to know how to configure the cfg script, to configure the media proxy so that it can nat the calls between users that don't have public IP, but not when they have public IP. best Regards Javier

1 0

[Users] mediaproxy
by Javier Ramirez 31 Aug '06

31 Aug '06

Hi, I would like to know how to configure the cfg script, to configure the media proxy so that it can nat the calls between users that don't have public IP, but not when they have public IP. best Regards Javier

1 0

[Serusers] mediaproxy
by Javier Ramirez 31 Aug '06

31 Aug '06

Hi, I would like to know how to configure the cfg script, to configure the media proxy so that it can nat the calls between users that don't have public IP, but not when they have public IP. best Regards Javier

1 0

[Users] OpenSER Summit
by Bogdan-Andrei Iancu 31 Aug '06

31 Aug '06

Hello everybody, It's more than one year since the OpenSER project started. A large community of users and developers has been formed around OpenSER. It's a very active community and one of the most important goals of the OpenSER project is to keep the users community as synchronized as possible with the developers and management groups. During past time we met a lot of people that were happy to see us or we were happy to see them. So, we came to the idea of organizing a OpenSER Summit where OpenSER fans (all types of fans) should meet to share their experience, to discuss about future ways of developing the project or just to personally meet people from the mailing lists. The idea is to have people talking about how they are using OpenSER in different contexts and which were/might be their needs, development people talking about possible solutions and about their ideas of enhancing OpenSER, etc... We were thinking of taking advantage of the upcoming VoN in Berlin in the beginning of November and to organize the Summit within the same time frame - it will be more convenient for people to travel for two events at the same time. We would like to have some feedback from the community - how many people are interested in such a meeting, what will be their primary interests. Any other OpenSER-related ideas are most welcome! Best regards, Bogdan

2 1

[Serusers] pre_auth(): Credentials with given realm not found
by ram 31 Aug '06

31 Aug '06

Hi all iam testing Voicemessages with Asterisk in the log i get frequently pre_auth(): Invalid nonce value received, very suspicious ! when i debug, i get the following message i have posted my config couple of day back 5(2213) get_hdr_field: cseq <CSeq>: <6> <REGISTER> 5(2213) DEBUG: is_maxfwd_present: value = 70 5(2213) parse_headers: flags=256 5(2213) DEBUG: get_hdr_body : content_length=0 5(2213) found end of header 5(2213) find_first_route(): No Route headers found 5(2213) loose_route(): There is no Route HF 5(2213) check_self - checking if host==us: 14==13 && [mydomain.com] == [myipaddress] 5(2213) check_self - checking if port 5060 matches port 5060 5(2213) parse_headers: flags=-1 5(2213) parse_headers: flags=-1 5(2213) check_via_address(some ip address, some ip address, 0) 5(2213) check_self - checking if host==us: 14==13 && [mydomain.com] == [myipaddress] 5(2213) check_self - checking if port 5060 matches port 5060 5(2213) parse_headers: flags=4096 5(2213) pre_auth(): Credentials with given realm not found 5(2213) build_auth_hf(): 'WWW-Authenticate: Digest realm="mydomain.com", nonce="44f6ee623e15edb81a22dcec25ebe8c58b6e056d" '

3 3

RE: [Users] openser admin question..
by peter.3.edwards＠bt.com 31 Aug '06

31 Aug '06

> mysql-4.1.20-1 > > There's no 5 built for centos yet, but this should be fairly > up to date... Well, there's always the option to compile from source ... which isn't quite as painful as it sounds. > The program works.. but I'm not sure what that error might > cause in way > of bugs, or whatever. me either ... sorry! > I had no end of problems installing mysql-ruby in the end, I ended up > running the extern.rb with some command line options > --with-mysql-config > and that got the makefile built. Yeah, same here ... it was quite tedious trying to get it to compile correctly. Ended up having to reinstall MySQL to get the libmysqlclient.so library which isn't bundled with the Solaris package. *sigh* > BTW, what version of mysql-ruby should I be running, I'm > using 2.5.1 at > the moment, 2.6 and 2.7 were also available I went for 2.7 - again, the latest version available - when I just built my Solaris 9 OpenSER Administrator installation. HTH, Peter. -- Peter Edwards tel +44 (0) 1473 606492 This Personal email contains my own opinions, sent In Confidence and does not represent the views of British Telecommunications plc. __________________________________________________________________ British Telecommunications plc Registered office: 81 Newgate Street London EC1A 7AJ Registered in England no. 1800000 This electronic message contains information from British Telecommunications plc which may be privileged and confidential. The information is intended to be for the use of the individual(s) or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this electronic message in error, please notify us by telephone or email (to the number or address above) immediately. Activity and use of the British Telecommunications plc email system is monitored to secure its effective operation and for other lawful business purposes. Communications using this system will also be monitored and may be recorded to secure effective operation and for other lawful business purposes.

1 0

[Serusers] LCR module documentation
by Graham Lewis 31 Aug '06

31 Aug '06

2 1

[Serusers] caller=callee, Not giving me Busy
by ram 31 Aug '06

31 Aug '06

Hi iam using old SER 0.814, since its deployed already in my Office we are in process of migrationg to openser soon mean time i need to fix some problems with existing server only that is the reason iam trying to modify the code and achive the following things 1. Voice mail 2. Caller =callee 3. if the user not in voicemail group ( get busy), if the user off line, Not found I have modified some code after reading all news group here iam attaching the config file which iam trying iam also getting the some errors in Log Aug 30 17:58:20 router ser[609]: ERROR: sl_reply_error used: I'm terribly sorry, server error occured (1/SL) Aug 30 17:58:27 router ser[604]: ERROR: sl_reply_error used: I'm terribly sorry, server error occured (1/SL) Aug 30 18:00:05 router ser[1564]: ERROR: tcp_init: bind(6, 0x80bd7c0, 16) on 127.0.0.1: Address already in use Aug 30 18:00:06 router ser[1571]: ERROR: tcp_init: bind(6, 0x80bd7c0, 16) on 127.0.0.1: Address already in use iam running a cron job to check the services #!/bin/sh mysqlcheck --password=cdrouter --auto-repair --all-databases sleep 4 ser sleep 4 chmod 777 /tmp/ser_fifo help will be appriciated Ram

2 1

[Serusers] How to Integrate SER Server with Jabber IM.
by srinivas sriram 31 Aug '06

31 Aug '06

Hello, I am look to send Missed Call Alert Instant Message(IM) to Jabber Client from SER Server. 1) I have Installed and Configured SER on IP(xyz.xyz.xyz.xx), and Added two(2) user in mysql i.e 1000, 2000 and also Configured SER Client(Kphone) with SER Server. 2) I have Installed and Configured Jabber Server i.e WildFire on IP(xyz.xyz.xyz.yy), and Added two(2) user, i.e 1000, 2000 ans also configured Jabber Client(Exodus) with Jabber Server. 3)My aim is to send Missed call alert, when a call is made from SER Client(K phone) User name:1000 to SER Client(Kphone) User name:2000, and User 2000 is not replyed. SER Server should send Instant Message(IM) to Jabber Client( User name: 2000) saying one missed call from user 1000. 4)Ser Server is running on xyz.xyz.xyz.xx Jabber Server is running on xyz.xyz.xyz.yy Came you please send me ser.cfg with give me this feature ( SER integration with Jabber IM ). And step to Integrate with Jabber IM. My configuration is attached as ser.cfg. Thanks & Regards ************************************ Sriram Srinivas ************************************

1 0

[Serusers] How to Integrate SER Server with Jabber IM.
by srinivas sriram 31 Aug '06

31 Aug '06

Hello, I am look to send Missed Call Alert Instant Message(IM) to Jabber Client from SER Server. 1) I have Installed and Configured SER on IP(xyz.xyz.xyz.xx), and Added two(2) user in mysql i.e 1000, 2000 and also Configured SER Client(Kphone) with SER Server. 2) I have Installed and Configured Jabber Server i.e WildFire on IP(xyz.xyz.xyz.yy), and Added two(2) user, i.e 1000, 2000 ans also configured Jabber Client(Exodus) with Jabber Server. 3)My aim is to send Missed call alert, when a call is made from SER Client(K phone) User name:1000 to SER Client(Kphone) User name:2000, and User 2000 is not replyed. SER Server should send Instant Message(IM) to Jabber Client( User name: 2000) saying one missed call from user 1000. 4)Ser Server is running on xyz.xyz.xyz.xx Jabber Server is running on xyz.xyz.xyz.yy Came you please send me ser.cfg with give me this feature ( SER integration with Jabber IM ). And step to Integrate with Jabber IM. My configuration is attached as ser.cfg. Thanks & Regards ************************************ Sriram Srinivas ************************************

1 0

RE: [Users] openser admin question..
by peter.3.edwards＠bt.com 31 Aug '06

31 Aug '06

> Is an error like this normal when doing the db install? > > ERROR 1064 (42000): You have an error in your SQL syntax; check the > manual that corresponds to your MySQL server version for the right > syntax to use near 'VIEW usr_preferences_view AS SELECT id, username, > domain, attribute, type AS att' at line 1 Hi, Nick. Not that it helps you much - but, no, I don't think that's normal - I didn't get this when installing on MySQL 5.0.24 (latest stable version), compiled from source. You might get compatibility issues if your version of MySQL is out of date ... what are you running? I'm sure Mike Williams will be along to help you debug it ... > By the way, while ruby on rails might be a great development > platform, > it's a right pain in the behind to install. Completely agree - especially when you're in a network which requires the use of a proxy to reach the web ... Bleugh. Still, once it's up and running it should be easy to maintain, or so I'm led to believe! Peter. -- Peter Edwards tel +44 (0) 1473 606492 This Personal email contains my own opinions, sent In Confidence and does not represent the views of British Telecommunications plc. __________________________________________________________________ British Telecommunications plc Registered office: 81 Newgate Street London EC1A 7AJ Registered in England no. 1800000 This electronic message contains information from British Telecommunications plc which may be privileged and confidential. The information is intended to be for the use of the individual(s) or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this electronic message in error, please notify us by telephone or email (to the number or address above) immediately. Activity and use of the British Telecommunications plc email system is monitored to secure its effective operation and for other lawful business purposes. Communications using this system will also be monitored and may be recorded to secure effective operation and for other lawful business purposes.

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[Serusers] Granstream AtA g729 codec --- help please
by ravi reddy 31 Aug '06

31 Aug '06

---------- Forwarded message ---------- From: ravi reddy <mravikreddy(a)gmail.com> Date: Aug 29, 2006 9:56 AM Subject: Re: [Serusers] Granstream AtA g729 codec --- help please To: "Greger V. Teigre" <greger(a)teigre.com> Thanks Greger for asking me, I want a small information regarding billing of sip calls (even though it is not related SER) please make a note to me :-) here in radacct i am getting more than one record for every call ; some folks told that i need some perl script to format all the database entries and write a fresh copy so that i can get one record for one call which is easy for billing. So, what i need a suggestion from you is 1) Do i need to learn perl language and write the script ?. 2) or is there any other way to control the overflow of start stop messages in to SER ...?. If you know please tell me . Thank You. Regards, Ravi. On 8/28/06, Greger V. Teigre <greger(a)teigre.com> wrote: > > Good to hear! > g-) > PS! No need to "sir"-me. Keep it informal, I'm Greger ;-) > > > ravi reddy wrote: > > Dear Sir, > > Thanks for your reply. i found the problem :-) > > the problem is itself in my pstn gateway that they have to configure to > allow my SER sending g729 codecs , > > i came to know from the folling grep message. > > v=0. > o=32331001 8000 8001 IN IP4 192.168.0.74. > s=SIP Call. > c=IN IP4 81.21.33.35. > t=0 0. > m=audio 60040 RTP/AVP 18 4 99 2. > a=sendrecv. > a=rtpmap:18 G729/8000. > a=rtpmap:4 G723/8000. > a=rtpmap:99 iLBC/8000. > a=fmtp:99 mode=20. > a=rtpmap:2 G726-32/8000. > a=ptime:20. > > > # > U 81.21.33.35:5060 -> 81.21.34.34:5068 > SIP/2.0 501 Not Implemented. > Via: SIP/2.0/UDP 192.168.0.74:5068;rport=5068;received= 81.21.34.34 > ;branch=z9hG4bK8bcaffffe66dffff. > From: "ravi" <sip:32331001@81.21.33.35>;tag=502effff5ddeffff. > To: <sip:99106883@81.21.33.35>;tag=E067A27C-3ED. > Date: Mon, 28 Aug 2006 08:24:22 GMT. > Call-ID: 61d40000ddc8ffff(a)192.168.0.74. > Server: Cisco-SIPGateway/IOS-12.x. > CSeq: 47822 INVITE. > Allow-Events: telephone-event. > Content-Length: 0. > > > because this 501 is internal server or gateway error so iam working on > that now. > > Thankyou. > > Regards, > Ravi. > > > > > On 8/28/06, Greger V. Teigre <greger(a)teigre.com > wrote: > > > > My gut feeling would tell me that the codec has nothing to do with it > > unless you find error messages in your ser log or var/log/messages. Sure > > nothing else has changed? > > g-) > > > > ravi reddy wrote: > > Hi SER users, > > > > Iam using SER-0.9.6 with mediaproxy-0.5 and every think > > works fine except with the codecs . G711 a & g711 ulaw works fine but when i > > tuned grandstream settings to use g729 codec for pstn calls the call is not > > done by the Mediaproxy server . > > > > but when i > > created a g729 fake rtp generator it looks fine in sessions.py . so in > > order to forward the g729 phone call in to the pstn world what i have to do > > ?. > > > > Some folks told that "integrate SER with Asterisk Works!" is it really > > works for me? > > > > my pstn provider supports all codecs . ofcourse iam working in that > > company itself. > > > > please suggest some thing , so that I can bye pass this problem. > > > > > > Thank You. > > > > > > Regards, > > Ravi. > > > > ------------------------------ > > > > _______________________________________________ > > Serusers mailing list > > > > > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > >

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[Serusers] Any plans to implement UAC module?
by Andy Thomas 31 Aug '06

31 Aug '06

Probably one for Greger- Does anyone know if there are plans to implement the UAC module (as seen in Openser) within the next version of Ser? I am currently running 0.9.6, and when I dial out to the PSTN, the CLI is shown as the Ser username, not the DID number All incoming DID numbers are specified as aliases, so when the PSTN dials in, it gets forward to the correct user. However I need to rewrite the outgoing SIP message, changing the the from: URI so that the alias is displayed, not the ser username. The UAC module is probably the easiest way changing the from: URI, unless anyone knows of a better way??

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Re: [Users] openser admin
by Mike Williams 30 Aug '06

30 Aug '06

On Monday 28 August 2006 03:15, raviprakash sunkara wrote: > Hi Mike , > Thanks for replying the mail , > openser administrator is wrriten in ruby on rails .. which is MVC structure Correct. > in database interface in web interface.. > If I commited in the subscriber in ur list... > In grp table also have to change from id to phplib_id.. No, that shouldn't be the case unless I'm mistaken. In the subscriber table, there was originally a phplib_id. I made the mistake of replacing it with id, instead of using both. The grp table never had a phplib_id, so id wasn't replacing anything. > > Tell me is openser administrator in fully stable and compitable with > openser. I see no reason why it should not be. > > But i think it may be unstable or stable version .. > i'm sure openser administrator 0.2 is unstable version > I need to know about the 0.3 .... > please for give me , if any mistakens in my secenrio .... Hopefully, they'll all be stable versions. However, it is still a very new project, and I cannot say, that with so few regular users of it, all the bugs are out. Mike Williams > > On 8/26/06, Mike Williams <mike(a)mikebwilliams.com> wrote: > > Peter & Sunkara, > > > > Thanks for the good idea, I hadn't even thought of uninstall or upgrade > > situations. > > > > The phplib_id problem can be solved by readding that column. There is a > > commented-out SQL statement in the SQL script that you can use. It is > > listed > > below as well. > > > > ALTER TABLE subscriber ADD COLUMN phplib_id varchar(32) NOT NULL default > > '' > > FIRST; > > > > I will work on these things today. Also, please let me know of any > > feature requests you have, otherwise I will have nothing to work on. > > > > Thanks, > > > > Mike Williams > > > > On Friday 25 August 2006 06:58, peter.3.edwards(a)bt.com wrote: > > > Mike, > > > > > > What's the upgrade procedure to go from 0.2 -> 0.3? > > > > > > Presumably readd the phplib_id column as per your mail on 21 Aug ... > > > and, what - just replace the old openseradmin-0.2 directory with this > > > one (including creating a new database.yml file)? > > > > > > Also, is there a way to uninstall OpenSER Administrator without > > > effecting OpenSER ... Presumably, just delete the openseradmin-x > > > directory ... But is there an equivalent SQL script to uninstall it's > > > database component? > > > > > > I think it would be good if you could include upgrade and uninstall > > > instructions within the release notes / on the website. > > > > > > Cheers, > > > > > > Peter. > > > > > > > -----Original Message----- > > > > From: users-bounces(a)openser.org > > > > [mailto:users-bounces@openser.org] On Behalf Of Mike Williams > > > > Sent: 24 August 2006 21:02 > > > > To: users(a)openser.org > > > > Subject: [Users] openser admin > > > > > > > > All, > > > > > > > > OpenSER Admin version 0.3 has been released. This fixes the bug where > > > > compatibility with openserctl was broken, and adds the > > > > ability to add other > > > > management accounts to the interface. > > > > > > > > Let me know what you think. > > > > > > > > Thanks, > > > > > > > > Mike Williams > > > > > > > > _______________________________________________ > > > > Users mailing list > > > > Users(a)openser.org > > > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > > > Users mailing list > > > Users(a)openser.org > > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users

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RE: [Users] openser admin
by peter.3.edwards＠bt.com 30 Aug '06

30 Aug '06

Hi, Mike. I've noticed that you've added Upgrade and Uninstall options to the OpenSER Administrator website - great stuff. Although, I did notice that your upgrade instructions don't mention anything about upgrading the database tables - I presume if you need to do that at a later date, you'll supply an equivalent SQL script (like the database_tables.sql and delete_database_tables.sql scripts for install and uninstall, respectively)? Might be worth mentioning that. As for your request for ideas for new features ... I think the documentation within OpenSER admin could do with a bit more content - I'm quite new to OpenSER and OpenSER Administrator so I'm not entirely clear what VoIP Accounts, Domains, Groups, Group Membership etc. specifically refer to (well, I get accounts and domains but it gets a bit fuzzier after that). Perhaps a little paragraph at the top of each page with a brief but clear description? On top of that, I guess I come less from a user-centric world and lean more towards the administrator side of things so I'm less interested in letting users login to change their profiles and more interested in seeing options that will allow easier adminstration - such as a web interface to configuring parts of the openser.cfg file - being able to toggle the debug level at will, stop/start/restart OpenSER from the web, change the port OpenSER listens on, enable/disable modules on startup - and, generally, anything that can be web-ified from the .cfg file. Obviously, I'm not talking about a replacement to the .cfg file - but something that can modify some of the various options on the fly would be useful ... Anyway, just thought I'd make some suggestions. I finally got the logging feature working today (oh, yes, when the log file doesn't exist, OpenSER doesn't handle the error very gracefully - so perhaps some work required on the error handling?) so I think I can safely say everything's working ... Many thanks, Peter. > -----Original Message----- > From: users-bounces(a)openser.org > [mailto:users-bounces@openser.org] On Behalf Of Mike Williams > Sent: 26 August 2006 15:07 > To: users(a)openser.org > Subject: Re: [Users] openser admin > > > Peter & Sunkara, > > Thanks for the good idea, I hadn't even thought of uninstall > or upgrade > situations. > > The phplib_id problem can be solved by readding that column. > There is a > commented-out SQL statement in the SQL script that you can > use. It is listed > below as well. > > ALTER TABLE subscriber ADD COLUMN phplib_id varchar(32) NOT > NULL default '' > FIRST; > > I will work on these things today. Also, please let me know > of any feature > requests you have, otherwise I will have nothing to work on. > > Thanks, > > Mike Williams -- Peter Edwards This Personal email contains my own opinions, sent In Confidence and does not represent the views of British Telecommunications plc. __________________________________________________________________ British Telecommunications plc Registered office: 81 Newgate Street London EC1A 7AJ Registered in England no. 1800000 This electronic message contains information from British Telecommunications plc which may be privileged and confidential. The information is intended to be for the use of the individual(s) or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this electronic message in error, please notify us by telephone or email (to the number or address above) immediately. Activity and use of the British Telecommunications plc email system is monitored to secure its effective operation and for other lawful business purposes. Communications using this system will also be monitored and may be recorded to secure effective operation and for other lawful business purposes.

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Re: [Serusers] Serweb: HTTP 403 Forbidden error
by Roberto Lopes 30 Aug '06

30 Aug '06

Hi to all, My apache errror log just showed me this: [Tue Aug 29 15:45:57 2006] [error] [client 193.137.239.11] (13)Permission denied: access to /serweb/ denied Roberto > > On 8/29/06, Karel Kozlik <karel(a)iptel.org> wrote: > > > > hi, > > what your apache error log say? > > I think there are no read rights for apache on serweb files. > > > > Karel > > > > Roberto Lopes napsal(a): > > > > > > Dear SER Users, > > > > > > > > > I've a problem with Serweb and I hope somebody could help me out > > > I know this problem was already talked about but i never saw a > > solution > > > to it, so I'll post it again hoping somebody will know how to solve > > this > > > mistake. > > > So the problem is that when i try to put serweb working in my firefox > > > browser the following error is shown: > > > > > > > > > Forbidden > > > > > > You don't have permission to access /serweb/index.html on this server. > > > > > > Additionally, a 403 Forbidden error was encountered while trying to > > use > > > an ErrorDocument to handle the request. > > > > > > > > ------------------------------------------------------------------------ > > > Apache/2.2.0 (Fedora) Server at xxx.xxx.xxx.xxx Port 80 > > > > > > > > > I think i installed everything correctly. > > > > > > I'll show some of the configurations I've made > > > > > > #1 - In "/etc/php.ini" I changed the memory limit from 8 to 12M > > because > > > of the Serweb extension classkit.so > > > … > > > memory_limit = 12M > > > > > > #2 - I used pear to install classkit.so > > > > > > pear install <path to classkit-0.4.tgz> > > > > > > #3 - In "php.ini" > > > … > > > ;;;;;;;;;;;;;;;;;;;;;; > > > ; Dynamic Extensions ; > > > ;;;;;;;;;;;;;;;;;;;;;; > > > … > > > extension=classkit.so > > > > > > #4 - In "/serweb-0.4/config/config_data_layer.php": > > > … > > > $config->data_sql->host[$i]['host']="<IP MySQL Server>"; > > > … > > > $config->data_sql->host[$i][ > > > 'user']="<DB User>"; > > > $config->data_sql->host[$i]['pass']="<User password>"; > > > .. > > > > > > #5 - In "/root/instalacao/serweb-0.9.4/config/config_paths.php" > > > > > > $config->root_path="/root/instalacao/serweb- 0.9.4/html/"; > > > > > > #6 - I created a log file > > > touch /var/log/serweb > > > chown apache /var/log/serweb > > > > > > #7 - I created a folder for SIP domain > > > mkdir /root/installation/serweb-0.4/html/domains/xxx.xxx.xxx.xxx(IP of > > SER) > > > > > > #8 - I copied the files and folder from the default folder > > > ("/root/installation/serweb-0.4/html/domains/_default/") to the one i > > > created. > > > > > > #9 - I specified the SIP domain in the file > > > "/root/installation/serweb-0.4/config/set_domain.php": > > > … > > > $config->domain = "xxx.xxx.xxx.xxx(IP of SER)"; > > > … > > > > > > #10 - I enabled logging in file > > > "/root/installation/serweb-0.4/config/config.php": > > > … > > > $config->enable_loging = true; > > > $config->log_file = "/var/log/serweb; > > > > > > #11 - In /etc/httpd/conf/httpd.conf I added an alias > > > … > > > > > > <Directory "/var/www/html"> > > > Options Indexes FollowSymLinks > > > AllowOverride None > > > Order deny,allow > > > Allow from all > > > </Directory> > > > > > > <IfModule mod_userdir.c> > > > UserDir disable root > > > #UserDir public_html > > > </IfModule> > > > > > > Alias /serweb/ "/root/instalacao/serweb-0.9.4/html" > > > <Directory "/root/instalacao/serweb-0.9.4/html"> > > > Options Indexes MultiViews > > > Options +FollowSymlinks > > > AllowOverride None > > > Order Deny,Allow > > > Allow from all > > > </Directory> > > > ... > > > > > > #12 - I then put some commands for Apache to begin when computer would > > > > > start > > > > > > chkconfig httpd on > > > service httpd start > > > > > > #13 - I introduced the following URL > > > > > > http://<IP of SER>/<alias i wrote in httpd.conf>/ > > > http://xxx.xxx.xxx.xxx/serweb/ > > > > > > > > > And then that HTTP 403 error shows up > > > > > > I really hope somebody can help me because I'm doing a VoIP project > > for > > > my school and I'd really like to have everything done in time. > > > > > > Thank you for your attention, > > > Roberto Lopes > > > > > > > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Serusers mailing list > > > Serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > >

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[Serusers] FW: [Serdev] variable header field value comparation
by Luis Silva 30 Aug '06

30 Aug '06

>From: "Luis Silva" <lfs12(a)hotmail.com> >To: serdev(a)iptel.org >Subject: [Serdev] variable header field value comparation >Date: Fri, 25 Aug 2006 13:44:53 +0000 > >Hi all!! > >I need to compare an header field value with an avp. I tried to use the >hf_value_exists("header_name","$value") but the header that I need to >compare isn't static. I replaced the header_name with an avp but it didn't >work. then I tried to use avp_check("$value", "eq/$hdr($header_name)")) >but again, it didn't work. am I doing something wrong? is there another >solution? > >regards, >Luís Silva > > >_______________________________________________ >Serdev mailing list >Serdev(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serdev

3 3

[Serusers] Error with credentials
by Daniel Mirrha Santos 30 Aug '06

30 Aug '06

Does anybody know what is this error about "is_user_in(): No authorized credentials found (error in scripts)"? Thanks, Daniel

3 9

[Serusers] SIP soft client supporting G.728
by VIDOVIC Mato 30 Aug '06

30 Aug '06

Hi, I am looking for a Windows SIP soft client supporting G.728 codec for narrowband interworking with a PSTN gateway. Can someone help me? Best regards, Mato

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[Serusers] caller=callee, Not giving me Busy
by ram 29 Aug '06

29 Aug '06

Hi all I have added the Follwing Line in my ser.cfg if (!uri==myself) { t_relay(); break; }; But iam not getting Busy Tone Ram

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[Serusers] t_was_cancelled() equivalent function for Ser?
by Ricardo Carvalho 29 Aug '06

29 Aug '06

Hi, I'm using Ser and I think I need to use the same functionality that the function t_was_cancelled() from openser does. As much as I know, this functions still doesn't exist in Ser. Is there any other that does the same? (I need it to be able to skip sending a call to Asterisk voicemail every time that a call is hangup by the caller when it is still ringing on the callee. If I don't such kind of function, if the caller hungs up when still ringing the callee UA, Ser sends the call to voicemail erroneously, which sends a voicemail e-mail to the ringed person) Thanks, Ricardo.

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[Serusers] Via Format?
by Ricardo Martinez 29 Aug '06

29 Aug '06

Hello list. Sorry to ask this question here, but i can't find the answer in the RFC3261 Is this "VIA" permited according to the RFC? Via: SIP/2.0/UDP 69.20.6.205:15060, SIP/2.0/UDP 69.1.229.69:4081;branch=z9hG4bK-58e9ccfcef0f4c99915b8007cb9713e1-1156793814 It seems to be two VIA's in ONE line... Thanks Regards, Ricardo.-

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RE: [Users] Please help me in Openser Admin
by peter.3.edwards＠bt.com 29 Aug '06

29 Aug '06

> 1. Is openser adminstrator a web application? I mean, does it have > same function like "serweb" in ser? "OpenSER Administrator: Mission The mission is to provide a simple interface to OpenSER that can be quickly and easily modified and upgraded. It is not designed to provide an interface for the customer, but is to be used only for administration duties." http://openseradmin.sourceforge.net/ > 2. How to install the openser administrator in openser server? > I use openser1.1.0-tls and what packets does it need? OpenSER Administrator: Installation http://openseradmin.sourceforge.net/docs.php > 3.Is the openser administrator just used by the administrator to manage > his/her server? or The openser administrator can be used like a > serweb?(the client also can use it) because as I read from the reply > message in this forum, the serweb can not use for openser1.1.0-tls See answer to question 1. > 4. If it is not like serweb, can anybody tell me what is the suitable > web application for openser ? As far as I'm aware from reading the website and mailing list archives, the only web application designed for use with OpenSER is OpenSER Administrator. It sounds like it might be possible to modify SERweb but that would require going "under the hood" ... Ferianto, all of this information is easily found from the website - rather copying your mail to three mailing lists which each have a slightly different remit, you may find you'd get a better response if you do some initial research first and then direct any further questions you still have, to the most appropriate list instead of sending to all of them at once. Even Microsoft know ... How to ask a question http://support.microsoft.com/?kbid=555375 HTH, Peter. -- Peter Edwards This Personal email contains my own opinions, sent In Confidence and does not represent the views of British Telecommunications plc. __________________________________________________________________ British Telecommunications plc Registered office: 81 Newgate Street London EC1A 7AJ Registered in England no. 1800000 This electronic message contains information from British Telecommunications plc which may be privileged and confidential. The information is intended to be for the use of the individual(s) or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this electronic message in error, please notify us by telephone or email (to the number or address above) immediately. Activity and use of the British Telecommunications plc email system is monitored to secure its effective operation and for other lawful business purposes. Communications using this system will also be monitored and may be recorded to secure effective operation and for other lawful business purposes.

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[Serusers] verify if an avp is null
by Joana Smith 29 Aug '06

29 Aug '06

Hi everyone!! How can I verify if an avp is null?? I tried if( $avp_name == '') {} but this doesn't work. any suggestions? tks a lot ps - I know that is possible to use is_sattr_set but I think avp module is deprecated in favor of avpops _______________________________________________________________________________________ Sabe em quanto tempo damos resposta para ter 4000Euros? Saber mais em http://www.iol.pt/correio/rodape.php?dst=0607071

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[Users] Voice-System SRL OpenSER Syslog Tutorial note suggestion
by peter.3.edwards＠bt.com 29 Aug '06

29 Aug '06

Hi, I've modified the Basic OpenSER Logging tutorial, to include a Solaris compatible entry for /etc/syslog.conf (note included below): http://www.openser.org/dokuwiki/doku.php?id=basic_openser_syslog_tutoria l The Voice-System SRL OpenSER Syslog Tutorial, could also do with having the same note added to this page: http://www.voice-system.ro/docs/ser-syslog/ar01s06.html Daniel, I've noted that you post here and that you're also the author of the above doc ... I thought you might want to look it over to see if it's worth including ... Cheers, Peter. "Note: that on Solaris 9 (and maybe other systems), the asterisk (*) is NOT a Priority Level ONLY a Source Facility - which means it can only be used on the left, and not the right of the period (.). In such a case, this equivalent line will also cause all messages to be logged ("debug" and higher priorities, "debug" is the lowest priority): local7.debug -/var/log/openser Source: Source Facilities VS. Priority Levels, page 267; Darrell Ambro - Solaris System Administration" -- Peter Edwards tel +44 (0) 1473 606492 This Personal email contains my own opinions, sent In Confidence and does not represent the views of British Telecommunications plc. __________________________________________________________________ British Telecommunications plc Registered office: 81 Newgate Street London EC1A 7AJ Registered in England no. 1800000 This electronic message contains information from British Telecommunications plc which may be privileged and confidential. The information is intended to be for the use of the individual(s) or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this electronic message in error, please notify us by telephone or email (to the number or address above) immediately. Activity and use of the British Telecommunications plc email system is monitored to secure its effective operation and for other lawful business purposes. Communications using this system will also be monitored and may be recorded to secure effective operation and for other lawful business purposes.

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[Serusers] Ser x Postgres
by Daniel Mirrha Santos 29 Aug '06

29 Aug '06

Its possible to use postgres to authenticate ser users? What I have to do to make it work with postgres? Thanks, Daniel

2 2

[Serusers] Build problem
by Greg Martin 29 Aug '06

29 Aug '06

I'm trying to build ser-0.9.6 on a sparc machine. 180> uname -rvmap SunOS tc90dv03 5.8 Generic_108528-13 sun4u sparc SUNW,UltraAX-i2 The errors I'm getting are: 172> make INSTALL=install include_modules="mysql" Makefile.rules:80: action.d: No such file or directory Makefile.rules:80: crc.d: No such file or directory Makefile.rules:80: daemonize.d: No such file or directory Makefile.rules:80: data_lump.d: No such file or directory Makefile.rules:80: data_lump_rpl.d: No such file or directory Makefile.rules:80: dprint.d: No such file or directory Makefile.rules:80: dset.d: No such file or directory Makefile.rules:80: error.d: No such file or directory Makefile.rules:80: fifo_server.d: No such file or directory Makefile.rules:80: flags.d: No such file or directory Makefile.rules:80: forward.d: No such file or directory Makefile.rules:80: hash_func.d: No such file or directory Makefile.rules:80: ip_addr.d: No such file or directory Makefile.rules:80: main.d: No such file or directory Makefile.rules:80: md5.d: No such file or directory Makefile.rules:80: md5utils.d: No such file or directory Makefile.rules:80: modparam.d: No such file or directory Makefile.rules:80: msg_translator.d: No such file or directory Makefile.rules:80: pass_fd.d: No such file or directory Makefile.rules:80: proxy.d: No such file or directory Makefile.rules:80: qvalue.d: No such file or directory Makefile.rules:80: re.d: No such file or directory Makefile.rules:80: receive.d: No such file or directory Makefile.rules:80: resolve.d: No such file or directory Makefile.rules:80: route.d: No such file or directory Makefile.rules:80: route_struct.d: No such file or directory Makefile.rules:80: script_cb.d: No such file or directory Makefile.rules:80: socket_info.d: No such file or directory Makefile.rules:80: sr_module.d: No such file or directory Makefile.rules:80: stats.d: No such file or directory Makefile.rules:80: tcp_main.d: No such file or directory Makefile.rules:80: tcp_read.d: No such file or directory Makefile.rules:80: timer.d: No such file or directory Makefile.rules:80: tsend.d: No such file or directory Makefile.rules:80: udp_server.d: No such file or directory Makefile.rules:80: unixsock_server.d: No such file or directory Makefile.rules:80: usr_avp.d: No such file or directory Makefile.rules:80: mem/f_malloc.d: No such file or directory Makefile.rules:80: mem/mem.d: No such file or directory Makefile.rules:80: mem/memtest.d: No such file or directory Makefile.rules:80: mem/q_malloc.d: No such file or directory Makefile.rules:80: mem/shm_mem.d: No such file or directory Makefile.rules:80: mem/vq_malloc.d: No such file or directory Makefile.rules:80: parser/hf.d: No such file or directory Makefile.rules:80: parser/msg_parser.d: No such file or directory Makefile.rules:80: parser/parse_allow.d: No such file or directory Makefile.rules:80: parser/parse_content.d: No such file or directory Makefile.rules:80: parser/parse_cseq.d: No such file or directory Makefile.rules:80: parser/parse_disposition.d: No such file or directory Makefile.rules:80: parser/parse_diversion.d: No such file or directory Makefile.rules:80: parser/parse_event.d: No such file or directory Makefile.rules:80: parser/parse_expires.d: No such file or directory Makefile.rules:80: parser/parse_fline.d: No such file or directory Makefile.rules:80: parser/parse_from.d: No such file or directory Makefile.rules:80: parser/parse_hname2.d: No such file or directory Makefile.rules:80: parser/parse_hostport.d: No such file or directory Makefile.rules:80: parser/parse_methods.d: No such file or directory Makefile.rules:80: parser/parse_nameaddr.d: No such file or directory Makefile.rules:80: parser/parse_param.d: No such file or directory Makefile.rules:80: parser/parse_rpid.d: No such file or directory Makefile.rules:80: parser/parse_rr.d: No such file or directory Makefile.rules:80: parser/parse_sipifmatch.d: No such file or directory Makefile.rules:80: parser/parse_to.d: No such file or directory Makefile.rules:80: parser/parse_uri.d: No such file or directory Makefile.rules:80: parser/parse_via.d: No such file or directory Makefile.rules:80: parser/parser_f.d: No such file or directory Makefile.rules:80: parser/digest/digest.d: No such file or directory Makefile.rules:80: parser/digest/digest_parser.d: No such file or directory Makefile.rules:80: parser/digest/param_parser.d: No such file or directory Makefile.rules:80: parser/contact/contact.d: No such file or directory Makefile.rules:80: parser/contact/parse_contact.d: No such file or directory Makefile.rules:80: db/db.d: No such file or directory Makefile.rules:80: db/db_fifo.d: No such file or directory Makefile.rules:80: lex.yy.d: No such file or directory Makefile.rules:80: cfg.tab.d: No such file or directory yacc -d -b cfg cfg.y conflicts: 1 shift/reduce make: *** [cfg.tab.c] Broken Pipe Any help would be appreciated, thanks. Greg Martin CTO Consumer Application Architecture TELUS Communications Company Phone: (780) 493-2786 Cell: (780) 718-4139

2 1

[Users] Question about GFLAGS Module
by Michel Zenone 29 Aug '06

29 Aug '06

Hi! I wonder if there is only a unique bitmap of flags or if there is a bitmap of flags for each SIP dialog? Thanks! Michel

2 1

[Users] Using subst() of Textops module
by Leonid Fainshtein 29 Aug '06

29 Aug '06

Hi, In the textops module doc (http://openser.org/docs/modules/1.0.x/textops.html#AEN124) there is an example that demonstrates how to replace URI in the header "To" with the message URI: # replace the uri in to: with the message uri (just an example) if ( subst('/^To:(.*)sip:[^@]*@[a-zA-Z0-9.]+(.*)$/t:\1\u\2/ig') ) {}; After I've fixed a small mistake (instead of /t: I put /To:) it worked. But I don't understand how does it work. It looks like "u" is substituted by URI from R-URI but I didn't find such core variable. Could somebody explain me? Thank you in advance, Leonid Fainshtein

3 2

[Serusers] Serweb: HTTP 403 Forbidden error
by Roberto Lopes 29 Aug '06

29 Aug '06

Dear SER Users, I've a problem with Serweb and I hope somebody could help me out I know this problem was already talked about but i never saw a solution to it, so I'll post it again hoping somebody will know how to solve this mistake. So the problem is that when i try to put serweb working in my firefox browser the following error is shown: Forbidden You don't have permission to access /serweb/index.html on this server. Additionally, a 403 Forbidden error was encountered while trying to use an ErrorDocument to handle the request. ------------------------------ Apache/2.2.0 (Fedora) Server at xxx.xxx.xxx.xxx Port 80 I think i installed everything correctly. I'll show some of the configurations I've made #1 - In "/etc/php.ini" I changed the memory limit from 8 to 12M because of the Serweb extension classkit.so … memory_limit = 12M #2 - I used pear to install classkit.so pear install <path to classkit-0.4.tgz> #3 - In "php.ini" … ;;;;;;;;;;;;;;;;;;;;;; ; Dynamic Extensions ; ;;;;;;;;;;;;;;;;;;;;;; … extension=classkit.so #4 - In "/serweb-0.4/config/config_data_layer.php": … $config->data_sql->host[$i]['host']="<IP MySQL Server>"; … $config->data_sql->host[$i]['user']="<DB User>"; $config->data_sql->host[$i]['pass']="<User password>"; .. #5 - In "/root/instalacao/serweb-0.9.4/config/config_paths.php" $config->root_path="/root/instalacao/serweb- 0.9.4/html/"; #6 - I created a log file touch /var/log/serweb chown apache /var/log/serweb #7 - I created a folder for SIP domain mkdir /root/installation/serweb-0.4/html/domains/xxx.xxx.xxx.xxx(IP of SER) #8 - I copied the files and folder from the default folder ("/root/installation/serweb-0.4/html/domains/_default/") to the one i created. #9 - I specified the SIP domain in the file "/root/installation/serweb-0.4/config/set_domain.php": … $config->domain = "xxx.xxx.xxx.xxx(IP of SER)"; … #10 - I enabled logging in file "/root/installation/serweb-0.4 /config/config.php": … $config->enable_loging = true; $config->log_file = "/var/log/serweb; #11 - In /etc/httpd/conf/httpd.conf I added an alias … <Directory "/var/www/html"> Options Indexes FollowSymLinks AllowOverride None Order deny,allow Allow from all </Directory> <IfModule mod_userdir.c> UserDir disable root #UserDir public_html </IfModule> Alias /serweb/ "/root/instalacao/serweb-0.9.4/html" <Directory "/root/instalacao/serweb-0.9.4/html"> Options Indexes MultiViews Options +FollowSymlinks AllowOverride None Order Deny,Allow Allow from all </Directory> ... #12 - I then put some commands for Apache to begin when computer would start chkconfig httpd on service httpd start #13 - I introduced the following URL http://<IP of SER>/<alias i wrote in httpd.conf>/ http://xxx.xxx.xxx.xxx/serweb/ And then that HTTP 403 error shows up I really hope somebody can help me because I'm doing a VoIP project for my school and I'd really like to have everything done in time. Thank you for your attention, Roberto Lopes

2 1

Re: [Serusers] Proxy Authentication
by Andrey Kouprianov 29 Aug '06

29 Aug '06

Hi, Why are you selecting 313814? The problem is with 313812 isnt it? Your trace log shows the conflict between 313812 and 313813 (not 313814)... :) Here's the examle (from RFC document) of authorizationm response from the client back to the server with a 2nd INVITE... Authorization: Digest username="bob", realm="biloxi.com", nonce="dcd98b7102dd2f0e8b11d0f600bfb0c093", uri="sip:bob@biloxi.com", qop=auth, nc=00000001, cnonce="0a4f113b", response="6629fae49393a05397450978507c4ef1", opaque="5ccc069c403ebaf9f0171e9517f40e41" As you can see uri is "sip:bob@bioxy.com" as well as the username is "bob" as well (i.e. they are same)... That gives me an impression there's something wrong with ur client-side setup.. Regards, Andrey. On 8/29/06, Javier Oviedo <joviedo(a)plcendesa.com> wrote: > Andrey Kouprianov escribió: > > Hi, > > > > First of all it is GREAT that you've sent your logs and ser.cfg!! I > > wish everyone would do it (however, this is rarely the case). > > > > You SJPhone cannot call, because for some reason your URI and username > > differ. Take a closer look at this part: > > > > Proxy-Authorization: Digest > > username="313813",realm="localhost",nonce="44f307373cf9fb29da3028442e23f976513fc91d",uri="sip:313812@sipserver.es",response="bfd77c8cdc1066487b170b19e1e0fe61",algorithm=MD5 > > > > > > The username is 313813, but the URI is sip:313812@sipserver.es. Notice > > any difference? Plz, take a look at your account setup in SJPhone > > closely. I think the problem should be there. > > > > Regards, > > > > Andrey. > > > > On 8/28/06, Javier Oviedo <joviedo(a)plcendesa.com> wrote: > >> > >> Hi all > >> > >> In my network schema I have a Sip Express Router to authenticate my > >> outbound > >> calls to a PSTN. I add a new equipment to a ser with "serctl add user > >> pass > >> e-mail" command. My softphone is registered in proxy but when I make > >> a call > >> to a PSTN I obtain the following error: Call rejected: 407 Proxy > >> Authentication Rquired : > >> > >> I use the SJPhone software. > >> > >> Any idea about where is the problem > >> > >> Thanks in advance > >> Joe > >> > >> My ser configuration is: > >> > >> debug=3 > >> fork=no > >> log_stderror=yes > >> > >> 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 (method=="ACK") { > >> route(1); > >> break; > >> } if (method=="INVITE") { > >> route(3); > >> break; > >> } else if (method=="REGISTER") { > >> route(2); > >> break; > >> }; > >> > >> lookup("aliases"); > >> > >> if (uri!=myself) { > >> route(1); > >> break; > >> }; > >> > >> if (!lookup("location")) { > >> sl_send_reply("404", "User Not Found"); > >> break; > >> }; > >> > >> route(1); > >> } > >> > >> route[1] { > >> > >> # > >> ----------------------------------------------------------------- > >> # Default Message Handler > >> # > >> ----------------------------------------------------------------- > >> if (!t_relay()) { > >> sl_reply_error(); > >> }; > >> } > >> > >> route[2] { > >> > >> # > >> ----------------------------------------------------------------- > >> # REGISTER Message Handler > >> # > >> ---------------------------------------------------------------- > >> sl_send_reply("100", "Trying"); > >> > >> if (!www_authorize("localhost","subscriber")) { > >> www_challenge("localhost","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("localhost","subscriber")) { > >> proxy_challenge("localhost","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); > >> } > >> > >> > >> > >> SoftPhone LOG: > >> > >> 17:04:43.1 > >> SENDING TO: 10.110.0.127:5060 > >> INVITE sip:313812@sipserver.es SIP/2.0 > >> To: <sip:313812@sipserver.es> > >> From: Joe<sip:313813@sipserver.es>;tag=52710612 > >> Via: SIP/2.0/UDP > >> 172.25.97.138:9528;branch=z9hG4bK-d87543-366872507-1--d87543-;rport > >> Call-ID: fc7b726c6545d126 > >> CSeq: 1 INVITE > >> Contact: <sip:313813@172.25.97.138:9528> > >> Max-Forwards: 70 > >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > >> SUBSCRIBE, > >> INFO > >> Content-Type: application/sdp > >> User-Agent: eyeBeam release 3004t stamp 16741 > >> Content-Length: 274 > >> > >> v=0 > >> o=- 32759240 32759303 IN IP4 172.25.97.138 > >> s=eyeBeam > >> c=IN IP4 172.25.97.138 > >> t=0 0 > >> m=audio 7262 RTP/AVP 100 6 0 8 3 18 5 101 > >> a=alt:1 1 : 8013E837 DD8642E8 172.25.97.138 7262 > >> a=fmtp:101 0-15 > >> a=rtpmap:100 speex/16000 > >> a=rtpmap:101 telephone-event/8000 > >> a=sendrecv > >> > >> 17:04:43.1 > >> RECEIVING FROM: 10.110.0.127:5060 > >> SIP/2.0 407 Proxy Authentication Required > >> To: > >> <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.3909 > >> From: Joe<sip:313813@sipserver.es>;tag=52710612 > >> Via: SIP/2.0/UDP > >> 172.25.97.138:9528;branch=z9hG4bK-d87543-366872507-1--d87543-;rport=9528 > >> Call-ID: fc7b726c6545d126 > >> CSeq: 1 INVITE > >> Proxy-Authenticate: Digest realm="localhost", > >> nonce="44f307373cf9fb29da3028442e23f976513fc91d" > >> Server: Sip EXpress router (0.9.6 (i386/linux)) > >> Content-Length: 0 > >> Warning: 392 10.110.0.127:5060 "Noisy feedback tells: pid=7865 > >> req_src_ip=172.25.97.138 req_src_port=9528 > >> in_uri=sip:313812@sipserver.es > >> out_uri=sip:313812@sipserver.es via_cnt==1" > >> > >> > >> 17:04:43.1 > >> SENDING TO: 10.110.0.127:5060 > >> ACK sip:313812@sipserver.es SIP/2.0 > >> To: > >> <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.3909 > >> From: Joe<sip:313813@sipserver.es>;tag=52710612 > >> Via: SIP/2.0/UDP > >> 172.25.97.138:9528;branch=z9hG4bK-d87543-366872507-1--d87543-;rport > >> Call-ID: fc7b726c6545d126 > >> CSeq: 1 ACK > >> Content-Length: 0 > >> > >> > >> 17:04:43.1 > >> SENDING TO: 10.110.0.127:5060 > >> INVITE sip:313812@sipserver.es SIP/2.0 > >> To: <sip:313812@sipserver.es> > >> From: Joe<sip:313813@sipserver.es>;tag=52710612 > >> Via: SIP/2.0/UDP > >> 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport > >> Call-ID: fc7b726c6545d126 > >> CSeq: 2 INVITE > >> Contact: <sip:313813@172.25.97.138:9528> > >> Max-Forwards: 70 > >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > >> SUBSCRIBE, > >> INFO > >> Content-Type: application/sdp > >> Proxy-Authorization: Digest > >> username="313813",realm="localhost",nonce="44f307373cf9fb29da3028442e23f976513fc91d",uri="sip:313812@sipserver.es",response="bfd77c8cdc1066487b170b19e1e0fe61",algorithm=MD5 > >> > >> User-Agent: eyeBeam release 3004t stamp 16741 > >> Content-Length: 274 > >> > >> v=0 > >> o=- 32759240 32759303 IN IP4 172.25.97.138 > >> s=eyeBeam > >> c=IN IP4 172.25.97.138 > >> t=0 0 > >> m=audio 7262 RTP/AVP 100 6 0 8 3 18 5 101 > >> a=alt:1 1 : 8013E837 DD8642E8 172.25.97.138 7262 > >> a=fmtp:101 0-15 > >> a=rtpmap:100 speex/16000 > >> a=rtpmap:101 telephone-event/8000 > >> a=sendrecv > >> > >> 17:04:43.1 > >> RECEIVING FROM: 10.110.0.127:5060 > >> SIP/2.0 100 trying -- your call is important to us > >> To: <sip:313812@sipserver.es> > >> From: Joe<sip:313813@sipserver.es>;tag=52710612 > >> Via: SIP/2.0/UDP > >> 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport=9528 > >> Call-ID: fc7b726c6545d126 > >> CSeq: 2 INVITE > >> Server: Sip EXpress router (0.9.6 (i386/linux)) > >> Content-Length: 0 > >> Warning: 392 10.110.0.127:5060 "Noisy feedback tells: pid=7865 > >> req_src_ip=172.25.97.138 req_src_port=9528 > >> in_uri=sip:313812@sipserver.es > >> out_uri=sip:313812@sipserver.es via_cnt==1" > >> > >> > >> 17:04:43.1 > >> RECEIVING FROM: 10.110.0.127:5060 > >> SIP/2.0 407 Proxy Authentication Required > >> To: > >> <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.fcd1 > >> From: Joe<sip:313813@sipserver.es>;tag=52710612 > >> Via: SIP/2.0/UDP > >> 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport=9528 > >> Call-ID: fc7b726c6545d126 > >> CSeq: 2 INVITE > >> Proxy-Authenticate: Digest realm="localhost", > >> nonce="44f307373cf9fb29da3028442e23f976513fc91d" > >> Server: Sip EXpress router (0.9.6 (i386/linux)) > >> Content-Length: 0 > >> Warning: 392 10.110.0.127:5060 "Noisy feedback tells: pid=7865 > >> req_src_ip=10.110.0.127 req_src_port=5060 in_uri=sip:313812@sipserver.es > >> out_uri=sip:313812@sipserver.es via_cnt==2" > >> > >> > >> 17:04:43.1 > >> SENDING TO: 10.110.0.127:5060 > >> ACK sip:313812@sipserver.es SIP/2.0 > >> To: > >> <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.fcd1 > >> From: Joe<sip:313813@sipserver.es>;tag=52710612 > >> Via: SIP/2.0/UDP > >> 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport > >> Call-ID: fc7b726c6545d126 > >> CSeq: 2 ACK > >> Content-Length: 0 > >> > >> > >> 17:04:43.2 Call (l:'Joe' r:'sip:313812@sipserver.es') - Call being > >> terminated. Reasons: "Proxy Authentication Required", (code: 407) > >> > >> > >> _______________________________________________ > >> 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 > > > > > Hi Andrey > > I think tha tihs part of log is correct: > > Proxy-Authorization: Digest > username="313813",realm="localhost",nonce="44f307373cf9fb29da3028442e23f976513fc91d",uri="sip:313812@sipserver.es",response="bfd77c8cdc1066487b170b19e1e0fe61",algorithm=MD5 > > > The username is the caller id and the uri is de destination number. My > SJphone configuration is simply: The user, pass and account name are > 313813. I attach my mysql subscriber table: > > > mysql> select * from subscriber where username=313814; > +----------------------------------+----------+----------------------+----------+------------+-----------+-------+-------------------+---------------------+---------------------+--------------+------+------------------+----------+----------------------------------+----------------------------------+------------+----------+------+------+------+ > | phplib_id | username | domain | > password | first_name | last_name | phone | email_address | > datetime_created | datetime_modified | confirmation | flag | > sendnotification | greeting | ha1 | > ha1b | allow_find | timezone | rpid | domn | > uuid | > +----------------------------------+----------+----------------------+----------+------------+-----------+-------+-------------------+---------------------+---------------------+--------------+------+------------------+----------+----------------------------------+----------------------------------+------------+----------+------+------+------+ > | 3fbf3938ba07fa0f280ffe44926e4cd3 | 313814 | sipserver.es | 313814 > | | | | joe(a)plcendesa.com | 2006-08-28 > 18:23:22 | 0000-00-00 00:00:00 | | o | > | | 3fbf3938ba07fa0f280ffe44926e4cd3 | > 147098ce3cd77298d89982cd37b5813a | 0 | NULL | NULL | NULL | > NULL | > +----------------------------------+----------+----------------------+----------+------------+-----------+-------+-------------------+---------------------+---------------------+--------------+------+------------------+----------+----------------------------------+----------------------------------+------------+----------+------+------+------+ > 1 row in set (0.00 sec) > > > and my uri table: > > mysql> select * from uri; > +----------+----------------------+----------+---------------------+ > | username | domain | uri_user | last_modified | > +----------+----------------------+----------+---------------------+ > | 313813 | sipserver.createc.es | 313813 | 2006-08-28 17:52:48 | > | 313814 | sipserver.createc.es | 313814 | 2006-08-28 18:23:22 | > +----------+----------------------+----------+---------------------+ > 2 rows in set (0.00 sec) > > > With the X-Lite softphone the same thing happens. > > Thanks in advace! > Regards > Joe >

1 0

[Users] Understanding the Call Flow
by Max Clark 29 Aug '06

29 Aug '06

Hi all, I am trying to get a better understanding of the call flow - specifically for distant users. In our current configuration we have all of the UAs registering to our OpenSER box which manages the authentication, and passes the call forward if allowed. Like so: UA -> SER -> PSTN GW -> PSTN I have been experimenting with using a mediaproxy/rtp proxy to improve the quality of service to users beyond the network edge - effectively: UA -> Proxy -> SER -> PSTN GW -> PSTN While this works great, we would like to place some PSTN Gateways closer to the edge of the network (for more than one obvious reason). In this case we would still have a central SER instance for authentication and accounting, but I would like for the edge proxies to be able to send the traffic to the closest applicable PSTN GW. How does one go about deploying something like this? What does the call flow look like? What software components do I need in what places? What is the glue that holds all of this together? Thanks, Max -- Max Clark http://www.clarksys.com

2 1

[Users] Please help me in openser administrator...Please
by Ferianto siregar 29 Aug '06

29 Aug '06

Dear all, Thanks to all of you for your kind-hearted to give me solution in solving my problem. Thank you very much. Thank you very much to Andrey Kouprianov for replying my message. Thanks I have some questions about the openser admin. I do anybody can help me to understand what the openser admin use for and how to install it. These are my question: 1. Is openser adminstrator a web application? I mean, does it have same function like "serweb" in ser? 2. How to install the openser administrator in openser server? I use openser1.1.0-tls and what packets does it need? 3.Is the openser administrator just used by the administrator to manage his/her server? or The openser administrator can be used like a serweb?(the client also can use it) because as I read from the reply message in this forum, the serweb can not use for openser1.1.0-tls 4. If it is not like serweb, can anybody tell me what is the suitable web application for openser ? so the client can register to in communication session by using a web application like serweb That is all my questions. I do hope anybody can help me. Please.... Thank you very much Regards, Ferianto --------------------------------- Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.

1 0

R: R: [Users] using AVPs to write into database
by D'Addelfio Davide 28 Aug '06

28 Aug '06

Thanks, I've looked into but it works only with 1.1.x version...!!! What's the different with avp_db_store command, may I use it to store something in my SQL DB? I'll try to do but I have some problems, he doesn't register... I have another question: I need to work over sdp messages, is possible to do a parsing, always using the scripts language...? Any thought? Thanks Davide -----Messaggio originale----- Da: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] Inviato: mercoledì 5 aprile 2006 19.15 A: D'Addelfio Davide Cc: Jayesh Nambiar; Bogdan-Andrei Iancu; openser Oggetto: Re: R: [Users] using AVPs to write into database Hello, On 04/05/06 18:01, D'Addelfio Davide wrote: > > Hi guys, I have a similar problem, maybe easier. I'm new in openser so > I'd like to have some help. > > I'd like openser create a table in MySQL db when receive an INVITE, a > row with from uri, method and something else, I've look into modules > docs but I've not understanding at all...could you help me, please? > if you are using openser development version, then take a look at avp_db_query() in avpops module http://openser.org/docs/modules/1.1.x/avpops.html#AEN239 Cheers, Daniel > Thanks for reply > > Davide > > ------------------------------------------------------------------------ > > *Da:* users-bounces(a)openser.org [mailto:users-bounces@openser.org] > *Per conto di *Jayesh Nambiar > *Inviato:* mercoledì 5 aprile 2006 15.53 > *A:* Bogdan-Andrei Iancu > *Cc:* openser > *Oggetto:* Re: [Users] using AVPs to write into database > > Hi Bogdan, > > Thanks for the reply and sorry for sending the earlier mail 4 times as > yahoo was not responding peoperly. > > When I do an avp_write(), why does the avp name get stored in the uuid > column? > > I am not using any uuid as of now. I also tried to add the username of > the person who is calling but it again creates two rows, one with the > value as > > username and other row with the value as forwarded number. > > I thought if there is a way to insert the username also, it wud be > easier to check, if the incoming call for that user is to be forwarded > or not. > > I get the following in my usr_preferences table after I dial 86 > followed by any number. > > +-----------+----------+--------+-----------+------+-----------------------------+---------------------+ > | uuid | username | domain | attribute | type | value | modified | > +-----------+----------+--------+-----------+------+-----------------------------+---------------------+ > | s:callfwd | | | callfwd | 0 | sip:5515551478@202.80.61.10 | > 2006-04-05 19:10:49 | > +-----------+----------+--------+-----------+------+-----------------------------+---------------------+ > > The script snippet is as follows: > > if(uri=~"^sip:86[0-9]*@") { > > > strip(2); > avp_write("$ruri", "s:callfwd"); > #avp_write("$from/username", "i:999"); > #avp_db_store("i:999", "i:/usr_preferences"); > avp_db_store("s:callfwd", "s:callfwd/usr_preferences"); > sl_send_reply("200", "OK"); > exit; > }; > > > */Bogdan-Andrei Iancu <bogdan(a)voice-system.ro>/* wrote: > > Hi, > > you need two steps: > 1) write the ruri (after strip) into an AVP - use avp_write() > 2) write the avp into the db - use avp_db_store() > > regards, > bogdan > > Jayesh Nambiar wrote: > > > Hi all, > > I am trying to implement the call forwarding feature in openser. The > > forwarding part is fine. But I have the follwing scenario: > > Whenever a user needs to change his forwarding number, he should be > > able to do so from his end device only. > > For eg: He first presses 86 and then the 10 digit number to be > > forwarded. Here the 86 has to be stripped and the 10 digit number > > should be inserted into the call forward value column of the > > preferences table. > > Also if he wants to remove the forwarded number, he can do so by > > pressing say for eg: 87. on receiving this number openser should > > delete that value from the > > table. > > Is this possible. I tried to do the following, but somehow it > does not > > change the > > value. > > if(uri=~"^sip:86[0-9]*@") { > > if(avp_db_load("$from/username", "s:callfwd")) { #check > > if call-fwd feature is enabled for the user > > log(1,"AVP condition returned true"); > > strip(2); > > avp_write("$ruri", "s:callfwd"); > > avp_print(); > > log(1,"AVP written"); > > sl_send_reply("200", "OK"); > > exit; > > }; > > }; > > Is avp_write the proper method or I guess avp_db_store can also help > > me. The avp_print() function also does not show me anything in the > > log. Are there any > > logical mistakes or I have mis-interpreted the syntax of avpops > functions. > > Please help me in thsi regard. > > Thanks a lot in advance. > > Jayesh. > > > > ------------------------------------------------------------------------ > > Jiyo cricket on Yahoo! India cricket > > > > Yahoo! Messenger Mobile > > > > Stay in touch with your buddies all the time. > > > >------------------------------------------------------------------------ > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org > >http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > ------------------------------------------------------------------------ > > Jiyo cricket on Yahoo! India cricket > <http://us.rd.yahoo.com/mail/in/mailcricket/*http:/in.sports.yahoo.com/crick…> > Yahoo! Messenger Mobile > <http://us.rd.yahoo.com/mail/in/mailmobilemessenger/*http:/in.mobile.yahoo.c…> > Stay in touch with your buddies all the time. > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

2 1

[Serusers] SER with VoiceMail Problem
by ram 28 Aug '06

28 Aug '06

Hi i am new to ser, iam going through the list, i found some of the similar queries http://lists.iptel.org/pipermail/serusers/2003-October/003132.html I have used the config file changed according to my setup and iam able to pick up messages from Asterisk also But i have some problem here is 1. i have registered user 8888 in my SER, when i dial 8888 its going to voice mail, how can i set the caller==calee Busy even i have added this if (!uri==myself) { t_relay(); break; }; But still no use. 2. when i dial to PSTN number, when the PSTN also go to voice mail after certain rings I suppose to get the voice Menu of other Side Number, but the voice go Blank ( instead of going to VoiceMenu) when checked in my Log, i get this error in my syslog Aug 28 18:33:55 router ser[757]: no location or no user Aug 28 18:34:01 router ser[751]: 10 digit exp match w/leading 1 Aug 28 18:34:01 router ser[751]: route[2]:SIP-to-PSTN call routed Aug 28 18:34:10 router ser[757]: no location or no user Aug 28 18:34:25 router ser[758]: no location or no user Aug 28 18:34:28 router ser[791]: ACC: call missed: from=88888 < sip:88888@mydomain.com>;tag=1674409670, i-uri= sip:1xxxxx80369@mydomain.com, method=INVITE, o-uri=sip:1xxxxx80369@myprovider:5060, code=408 Request Timeout Aug 28 18:34:28 router ser[791]: failure_route[5] Aug 28 18:34:28 router ser[791]: route[5]: re-relay to PSTN Aug 28 18:34:28 router ser[751]: ERROR: t_should_relay: status rewrite by UAS: stored: 408, received: 487 Aug 28 18:34:37 router ser[758]: 10 digit exp match w/leading 1 Aug 28 18:34:37 router ser[758]: route[2]:SIP-to-PSTN call routed Aug 28 18:34:37 router ser[758]: 10 digit exp match w/leading 1 Aug 28 18:34:37 router ser[758]: route[2]:SIP-to-PSTN call routed Aug 28 18:34:40 router ser[751]: no location or no user any suggestions Ram

2 1

[Serusers] Proxy Authentication
by Javier Oviedo 28 Aug '06

28 Aug '06

Hi all In my network schema I have a Sip Express Router to authenticate my outbound calls to a PSTN. I add a new equipment to a ser with "serctl add user pass e-mail" command. My softphone is registered in proxy but when I make a call to a PSTN I obtain the following error: *Call rejected: 407 Proxy Authentication Rquired *: I use the SJPhone software. Any idea about where is the problem Thanks in advance Joe My ser configuration is: debug=3 fork=no log_stderror=yes 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 (method=="ACK") { route(1); break; } if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!www_authorize("localhost","subscriber")) { www_challenge("localhost","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("localhost","subscriber")) { proxy_challenge("localhost","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); } SoftPhone LOG: 17:04:43.1 SENDING TO: 10.110.0.127:5060 INVITE sip:313812@sipserver.es SIP/2.0 To: <sip:313812@sipserver.es> From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-366872507-1--d87543-;rport Call-ID: fc7b726c6545d126 CSeq: 1 INVITE Contact: <sip:313813@172.25.97.138:9528> Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3004t stamp 16741 Content-Length: 274 v=0 o=- 32759240 32759303 IN IP4 172.25.97.138 s=eyeBeam c=IN IP4 172.25.97.138 t=0 0 m=audio 7262 RTP/AVP 100 6 0 8 3 18 5 101 a=alt:1 1 : 8013E837 DD8642E8 172.25.97.138 7262 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv 17:04:43.1 RECEIVING FROM: 10.110.0.127:5060 SIP/2.0 407 Proxy Authentication Required To: <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.3909 From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-366872507-1--d87543-;rport=9528 Call-ID: fc7b726c6545d126 CSeq: 1 INVITE Proxy-Authenticate: Digest realm="localhost", nonce="44f307373cf9fb29da3028442e23f976513fc91d" Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 10.110.0.127:5060 "Noisy feedback tells: pid=7865 req_src_ip=172.25.97.138 req_src_port=9528 in_uri=sip:313812@sipserver.es out_uri=sip:313812@sipserver.es via_cnt==1" 17:04:43.1 SENDING TO: 10.110.0.127:5060 ACK sip:313812@sipserver.es SIP/2.0 To: <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.3909 From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-366872507-1--d87543-;rport Call-ID: fc7b726c6545d126 CSeq: 1 ACK Content-Length: 0 17:04:43.1 SENDING TO: 10.110.0.127:5060 INVITE sip:313812@sipserver.es SIP/2.0 To: <sip:313812@sipserver.es> From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport Call-ID: fc7b726c6545d126 CSeq: 2 INVITE Contact: <sip:313813@172.25.97.138:9528> Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="313813",realm="localhost",nonce="44f307373cf9fb29da3028442e23f976513fc91d",uri="sip:313812@sipserver.es",response="bfd77c8cdc1066487b170b19e1e0fe61",algorithm=MD5 User-Agent: eyeBeam release 3004t stamp 16741 Content-Length: 274 v=0 o=- 32759240 32759303 IN IP4 172.25.97.138 s=eyeBeam c=IN IP4 172.25.97.138 t=0 0 m=audio 7262 RTP/AVP 100 6 0 8 3 18 5 101 a=alt:1 1 : 8013E837 DD8642E8 172.25.97.138 7262 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv 17:04:43.1 RECEIVING FROM: 10.110.0.127:5060 SIP/2.0 100 trying -- your call is important to us To: <sip:313812@sipserver.es> From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport=9528 Call-ID: fc7b726c6545d126 CSeq: 2 INVITE Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 10.110.0.127:5060 "Noisy feedback tells: pid=7865 req_src_ip=172.25.97.138 req_src_port=9528 in_uri=sip:313812@sipserver.es out_uri=sip:313812@sipserver.es via_cnt==1" 17:04:43.1 RECEIVING FROM: 10.110.0.127:5060 SIP/2.0 407 Proxy Authentication Required To: <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.fcd1 From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport=9528 Call-ID: fc7b726c6545d126 CSeq: 2 INVITE Proxy-Authenticate: Digest realm="localhost", nonce="44f307373cf9fb29da3028442e23f976513fc91d" Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 10.110.0.127:5060 "Noisy feedback tells: pid=7865 req_src_ip=10.110.0.127 req_src_port=5060 in_uri=sip:313812@sipserver.es out_uri=sip:313812@sipserver.es via_cnt==2" 17:04:43.1 SENDING TO: 10.110.0.127:5060 ACK sip:313812@sipserver.es SIP/2.0 To: <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.fcd1 From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport Call-ID: fc7b726c6545d126 CSeq: 2 ACK Content-Length: 0 17:04:43.2 Call (l:'Joe' r:'sip:313812@sipserver.es') - Call being terminated. Reasons: "Proxy Authentication Required", (code: 407)

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Re: [Serusers] rewrite the From part of SIP INVITE messages
by rcarvalho＠iric.up.pt 28 Aug '06

28 Aug '06

I'm sceptic about the use of functions that replace parts of SIP messages, because that way, will Ser still keep record of transactions? Wont those replacements affect Ser to recognise to whom route subsequent messages? Regards, Ricardo. Alexandr Dubovikov wrote: > On Fri, Aug 25, 2006 at 05:33:52PM +0100, Ricardo Carvalho wrote: >> I need to rewrite the From part of SIP INVITE messages before sending >> them to my IP Telco, or else they wont accept my calls! >> Anyone knows how can I do that? > > replace or subst > > the sample you can find in README file of textops module. > > >> Thanks, >> >> Ricardo. >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >

4 4

[Users] An error with exec_msg
by Zenone 28 Aug '06

28 Aug '06

Hi! I have an error when I try to run this in my openser.cfg: exec_msg("TAG=$(echo $SIP_HF_FROM|cut -f2 -d=);FROM=$(echo $SIP_HF_FROM|cut -f2 -d\<|cut -f1 -d\>|cut -f1-2 -d:);/usr/local/openser-1.1.0-tls/uri-folder/record $FROM $SIP_USER $TAG"); The obtained error is: Aug 28 15:11:21 localhost /usr/local/sbin/openser[32748]: ERROR:xl_parse_spec: bad pvar name "echo $SIP_HF_FROM|cut -f2 -d=);FROM=$(echo $SIP_HF_FROM|cut -f2 -d\<|cut -f1 -d\>|cut -f1-2 -d:);/usr/local/openser-1.1.0-tls/uri-folder/record $FROM $SIP_USER $TAG" Thanks in advance for your help! Michel ________________________________________________ Message sent using UebiMiau 2.7.8

3 4

Re: [Serusers] serweb gives a blank page
by Andy Thomas 28 Aug '06

28 Aug '06

Andrey, Thanks for the pointer - I turned on PHP logging and found the problem- PHP Fatal error: Smarty error: unable to write to $compile_dir '/tmp/serweb'. Be sure $compile_dir is writable by the web server user. in /home/serweb/smarty/Smarty.class.php on line 1102 Down to permissions on the /tmp/serweb directory - all now working, thanks!!

2 1

[Users] Please help me in Openser Admin
by Ferianto siregar 28 Aug '06

28 Aug '06

Dear all, Thanks to all of you for your kind-hearted to give me solution in solving my problem. Thank you very much. Thanks to openser developer team, who has released Openser Administrator in sourceforge.net I have some questions about the openser admin. I do anybody can help me to undestand what the openser admin use for and how to install it. These are my question: 1. Is openser adminstrator a web application? I mean, does it have same function like "serweb" in ser? 2. How to install the openser administrator in openser server? I use openser1.1.0-tls and what packets does it need? 3.Is the openser administrator just used by the administrator to manage his/her server? or The openser administrator can be used like a serweb?(the client also can use it) because as I read from the reply message in this forum, the serweb can not use for openser1.1.0-tls 4. If it is not like serweb, can anybody tell me what is the suitable web application for openser ? That is all my questions. I do hope anybody can help me. Thanks Regards, Ferianto --------------------------------- How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.

2 1

[Serusers] REGISTRATION high pace & build_contact() error
by Kostas Marneris 28 Aug '06

28 Aug '06

Hi all, using sipsak in usrloc mode to test the SER sip REGISTRAR with the following process : loop: ./sipsak -U -p host.foo.com -x 3600 -a passwd -s sip:user@foo.com goto loop I get the following mesg in LOG file after 2 mins : build_contact(): No memory left ... and after a while : ERROR: build_res_buf_from_sip_req: out of memory ; needs 28132 send_reply(): Error while sending 200 OK ... CRITICAL ERROR : Can't save Contact to MySQL Location DB (this is a log mesg in my ser.cfg in case : if (!save("location")) ) ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) although it seems that SER keeps going (?) to write in Location DB both SER's & MySQL (this seems strange). I have IBM eServer 336 x 2 : - 2 dual core Intel Xeon 3.6 GHz with 2MB Cache, EMT64 - 4 GB RAM - 2x72GB Ultra 320 Scsi disks # ser.cfg children=5 Could this be a memory leak problem in SER ? Is there any conf fine-tuning I can do in order to avoid that ? When I increase the 'children' up to 20, the problem happens again but within 5 mins. thanks for any help, Kostas

2 2

[Serusers] serweb gives a blank page
by Andy Thomas 28 Aug '06

28 Aug '06

I have installed Serweb, but when I run it I just get a blank page, with just the serweb title at the top. Nothing else appears, no login box or anything! I have set PEAR LOG to debug mode and the log only shows this- Aug 28 12:04:37 serweb [info] Useing file from default domain for filename: separator.html Aug 28 12:04:40 serweb [info] Useing file from default domain for filename: styles.css Aug 28 12:04:40 serweb [info] Useing file from default domain for filename: prolog.html Aug 28 12:04:40 serweb [info] Useing file from default domain for filename: separator.html Aug 28 12:04:41 serweb [info] Useing file from default domain for filename: styles.css Aug 28 12:04:41 serweb [info] Useing file from default domain for filename: prolog.html Aug 28 12:04:41 serweb [info] Useing file from default domain for filename: separator.html Aug 28 12:04:42 serweb [info] Useing file from default domain for filename: styles.css Aug 28 12:04:42 serweb [info] Useing file from default domain for filename: prolog.html Aug 28 12:04:42 serweb [info] Useing file from default domain for filename: separator.html The HTML generated is as follows- <!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Strict//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-strict.dtd"> <html> <head> <title>VoIPgb.net - web interface</title> <meta http-equiv="Content-Type" content="text/html; charset=utf-8"> <meta name="Author" content="Karel Kozlik <karel at iptel dot org>"> <meta http-equiv="PRAGMA" content="no-cache"> <meta http-equiv="Cache-control" content="no-cache"> <meta http-equiv="Expires" content="Mon, 28 Aug 2006 11:04:42 GMT"> <LINK REL="StyleSheet" HREF="/serweb/domains/_default/styles.css" TYPE="text/css"> </head> <script language="JavaScript" src="/serweb/js/login_completion.js.php"></script><body><h1> voipgb.net admin interface</h1><hr> <div class="swMain"> For some reason, the rest of the page isn't generated! Any ideas ??????

2 1

[Users] Problem with SQL queries in 1.1.0?
by Federico Giannici 28 Aug '06

28 Aug '06

We recently upgraded from version 1.0.0 to 1.1.0 from CVS of a few days ago. After a couple of days of uptime, some SQL errors began to appear. First there was this one: Aug 21 10:07:07 eowyn OpenSER[19895]: ERROR: warning_builder: buffer size exceeded Aug 21 10:07:07 eowyn OpenSER[19895]: WARNING: warning skipped -- too big Aug 21 10:07:07 eowyn OpenSER[19895]: submit_query: Aug 21 10:07:07 eowyn OpenSER[19895]: db_insert: Error while submitting query Aug 21 10:07:07 eowyn OpenSER[19895]: ERROR:acc:acc_db_request: Error while inserting to database Then, some errors like this: Aug 21 15:41:28 eowyn OpenSER[19895]: submit_query: Aug 21 15:41:28 eowyn OpenSER[19895]: db_insert: Error while submitting query Aug 21 15:41:28 eowyn OpenSER[19895]: ERROR:acc:acc_db_request: Error while inserting to database Or the following: Aug 21 19:02:10 eowyn OpenSER[20604]: submit_query: Aug 21 19:02:10 eowyn OpenSER[20604]: db_query: Error while submitting query Aug 21 19:02:10 eowyn OpenSER[20604]: NM ERROR: nm_hash_load_domains(): error with dbf.query() The nm_hash_load_domains() function is a module of mine, that make an SQL query. No problem with version 1.0.0. Is there any SQL known problem in the 1.1.0 CVS version? Maybe the first error caused some memory garbage? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

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[Serusers] Multi process RTP Proxy
by Ramu Yadav 28 Aug '06

28 Aug '06

Hi All, With single process RTP Proxy we are facing CPU issues. When we made around 80 or above odd calls, rtpproxy is taking CPU(50%), and flow of RTP packets delivering some what late to the UA. Because of this we are facing some sound quality issues. To get rid of this CPU usage, we are planning to write rtpproxy as a multi process. Is this a good idea to write rtpproxy as multi process or is there any work around to improve the performance of the rtpproxy. Help and Ideas are highly appreciated. -- Ramu Yadav

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[Serusers] Granstream AtA g729 codec --- help please
by ravi reddy 28 Aug '06

28 Aug '06

Hi SER users, Iam using SER-0.9.6 with mediaproxy-0.5 and every think works fine except with the codecs . G711 a & g711 ulaw works fine but when i tuned grandstream settings to use g729 codec for pstn calls the call is not done by the Mediaproxy server . but when i created a g729 fake rtp generator it looks fine in sessions.py . so in order to forward the g729 phone call in to the pstn world what i have to do ?. Some folks told that "integrate SER with Asterisk Works!" is it really works for me? my pstn provider supports all codecs . ofcourse iam working in that company itself. please suggest some thing , so that I can bye pass this problem. Thank You. Regards, Ravi.

2 1

[Serusers] ivr play when the Account Disabled
by ram 27 Aug '06

27 Aug '06

Hi all I am using SER, and testing in my setup i have few issues , and looking forward for the suggestion iam running over night script, check the balance if the balance go to Nagetive, iam deactivating the user Like removing the Grants of LD and INT they can make local call. How can i run IVR for the negative balance user, when they dial to Long Distance Balance not there message. and how can i setup IVR program to user to check the balance of the account. any suggestions. ram

3 2

[Serusers] multidomain with loose routing
by Weiter Leiter 27 Aug '06

27 Aug '06

Some UAs, when not explicitly configured with an outbound proxy (some even in this), follow rfc's recommendation of sending a request with a preloaded route set containing one entry, that of what they think it is their outbound proxy, determined through various means. This is a problem when ser as a serving proxy for multiple domains does the loose_route'ing: if the route entry in the message is an ip address, ser will realise that it is him that must serve the request and everything goes smooth. BUT if the client places a virtual domain in the route set, ser is fooled to believe that it should just proxy forward the request. Forwarding might be ok, as ser will forward to itself (suboptimal), but in this case how can i prevent illegal relaying? How should this be elegantly dealt with? The most obvious answer is to add "alias"es, but i want to handle the domains through domain module only. Shouldn't the domain module have some functionality to load the domains as aliases (at least on demand by module parameter)? Thanks.

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[Serusers] forward_tcp reliability
by Nhadie 27 Aug '06

27 Aug '06

Hi All, How reliable is forward_tcp? I have 3 SIP servers, 1 acting as the database of the three, I used forward tcp to replicate registration, unfortunately there are some instance that a local extension cannot contact another extension. Here's a part of my config if ((src_ip == 10.10.10.149) || (src_ip == 10.10.10.151)) { save_memory("location"); }; if (!www_authorize("sipdomain.com", "subscriber")) { www_challenge("sipdomain.com", "0"); break; }; if (!save("location")) { sl_reply_error(); } else { if ((!src_ip == 10.10.10.149) && (!src_ip == 10.10.10.151)) { forward_tcp("10.10.10.149", 5060); forward_tcp("10.10.10.151", 5060); }; break; should I use savce_memory? becuase I'm seeing some config that uses save_noreply, should I use this one? also should I changed the db_url parameter of usrloc and point it to the database of the main server? e.g. modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://ser:heslo@10.10.10.146/ser") Hope you guys can help me TIA Regards, Nhadie ________________________________________________ Message sent using UebiMiau 2.7

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RE: [Serusers] How to fix IP address stored in location db
by İlker Aktuna (Koç.net) 27 Aug '06

27 Aug '06

Hi, Thanks for your email. I get a "page not found" error when I try http://iptel.org/ser/gettingstarted <http://iptel.org/ser/gettingstarted> Actually at which URL is the example configuration script for using fix_nated_register() ? Also, which script should I edit to be able to use ser_ctl ? When I edit the serctl.conf file no change happens... Regards, ilker ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Friday, August 25, 2006 5:28 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] How to fix IP address stored in location db http://iptel.org/ser/gettingstarted Download the document and scripts for how to handle NATed clients. If you change default settings, you need to edit the script. g-) İlker Aktuna (Koç.net) wrote: Hi, Thanks for your reply. However, I couldn't understand it. What does exactly fix_nat_register do ? I have a client which does not have STUN feature and thus I can't make a call to this client. What should I do to make a call to this client ? Btw, I get the following errors when I use ser_ctl command: ser_ctl: (111, 'Connection refused'): error And WARNING: ListMethods on 'http://localhost:5060/' failed: socket.error My ser is running on 5061 port. Thanks, ilker -----Original Message----- From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Friday, August 25, 2006 12:10 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] How to fix IP address stored in location db The stored contact should be as received. You need to have the correct request URI when forwarding to a UA, even if the ip address is a private one. If not, the UA will most likely respond with 404 User not found. fix_nated_register() will put received ip:port in an avp and save will store this avp as received (you will see it with serctl ul show) g-) İlker Aktuna (Koç.net) wrote: Hi, I am trying to change stored uri in the location db. I assume it could be done with the help of nathelper module and I use fix_nated_register() function just before save_noreply("location") command. However I still see the internal IP address of the client in the location db. What is the easiest way to change the stored contact address ? Thanks, ilker __________ _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

2 1

[Serusers] (no subject)
by rcarvalho＠iric.up.pt 26 Aug '06

26 Aug '06

I'm sceptic about the use of functions that replace parts of SIP messages, because that way, will Ser still keep record of transactions? Wont those replacements affect Ser to recognise to whom route subsequent messages? Regards, Ricardo. Alexandr Dubovikov wrote: > On Fri, Aug 25, 2006 at 05:33:52PM +0100, Ricardo Carvalho wrote: >> I need to rewrite the From part of SIP INVITE messages before sending >> them to my IP Telco, or else they wont accept my calls! >> Anyone knows how can I do that? > > replace or subst > > the sample you can find in README file of textops module. > > >> Thanks, >> >> Ricardo. >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >

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RE: [Users] openser admin
by peter.3.edwards＠bt.com 26 Aug '06

26 Aug '06

Mike, What's the upgrade procedure to go from 0.2 -> 0.3? Presumably readd the phplib_id column as per your mail on 21 Aug ... and, what - just replace the old openseradmin-0.2 directory with this one (including creating a new database.yml file)? Also, is there a way to uninstall OpenSER Administrator without effecting OpenSER ... Presumably, just delete the openseradmin-x directory ... But is there an equivalent SQL script to uninstall it's database component? I think it would be good if you could include upgrade and uninstall instructions within the release notes / on the website. Cheers, Peter. > -----Original Message----- > From: users-bounces(a)openser.org > [mailto:users-bounces@openser.org] On Behalf Of Mike Williams > Sent: 24 August 2006 21:02 > To: users(a)openser.org > Subject: [Users] openser admin > > All, > > OpenSER Admin version 0.3 has been released. This fixes the bug where > compatibility with openserctl was broken, and adds the > ability to add other > management accounts to the interface. > > Let me know what you think. > > Thanks, > > Mike Williams > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

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[Users] Error in running serweb. Please help me..Please
by Ferianto siregar 26 Aug '06

26 Aug '06

Dear all, First of all, I would like to say thanks to all of you for your help and suggestion to solve my problem untill now. Thank you very much. Now, I try to install serweb_2004-07-27 in my openser server. I have read the installation note that I got from the serweb packet carefully. So, I can build the serweb in my openser server successfully. It means that it can display a login screen in the browser. But, I have a problem when I try to login by using username "admin" with password "openserrw". (This is based on the installation note that I have read before) I got error message. This is the error message : Database error: Invalid SQL: select username, perms from subscriber where phplib_id = '721153ca8942ea1a08900f5b61b69a6c' MySQL Error: 1054 (Unknown column 'perms' in 'field list') Session halted. As I try to understand from the error message, it says that there is no "perms" column. So I try to check it in the subscriber table in mysql.But, I did not find "perms" column in the subscriber table. I just see these columns in the subscriber table: -phplib_id -last_name - sendnotification - rpid -username - phone - greeting -domn -domain - email_address - ha1 -uuid -password - datetime_created - hdb1 -first_name - datetime_modified - allow_find -confirmation - flag - timezone What should i do to solve this problem? What should I do if there is no "perms" column in the subscriber table? I do hope anybody can give me a suggestion because this is my first time to build the serweb and mysql. Please help me....Please... Note: I user openser1.1.0-tls and serweb_2004-07-27 As I see from the error message that the phplib_id is exactly matched with phplib_id where username "admin" with password "openserrw" (from the subscriber table). Thank you very much Regards, Ferianto --------------------------------- Get your email and more, right on the new Yahoo.com

1 0

[Serusers] mysql modules load problems
by ram 26 Aug '06

26 Aug '06

Hi all iam install in new server RHEL 4.0 with Mysql 5.0 ser-0.97-pre compiled perfectly and installed perfectly when i enable mysql modules in ser.cfg and try to run [root@vmail ser]# ser -c -f ser.cfg 0(10560) ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: /usr/local/lib/ser/modules/mysql.so: undefined symbol: log 0(10560) parse error (28,13-14): failed to load module ERROR: bad config file (1 errors) what is the problem here Ram

2 1

[Serusers] rewrite the From part of SIP INVITE messages
by Ricardo Carvalho 25 Aug '06

25 Aug '06

I need to rewrite the From part of SIP INVITE messages before sending them to my IP Telco, or else they wont accept my calls! Anyone knows how can I do that? Thanks, Ricardo.

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RE: [Serusers] How to fix IP address stored in location db
by İlker Aktuna (Koç.net) 25 Aug '06

25 Aug '06

Hi, Thanks for your reply. However, I couldn't understand it. What does exactly fix_nat_register do ? I have a client which does not have STUN feature and thus I can't make a call to this client. What should I do to make a call to this client ? Btw, I get the following errors when I use ser_ctl command: ser_ctl: (111, 'Connection refused'): error And WARNING: ListMethods on 'http://localhost:5060/' failed: socket.error My ser is running on 5061 port. Thanks, ilker -----Original Message----- From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Friday, August 25, 2006 12:10 PM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] How to fix IP address stored in location db The stored contact should be as received. You need to have the correct request URI when forwarding to a UA, even if the ip address is a private one. If not, the UA will most likely respond with 404 User not found. fix_nated_register() will put received ip:port in an avp and save will store this avp as received (you will see it with serctl ul show) g-) İlker Aktuna (Koç.net) wrote: > Hi, > > I am trying to change stored uri in the location db. > I assume it could be done with the help of nathelper module and I use fix_nated_register() function just before save_noreply("location") command. > However I still see the internal IP address of the client in the location db. > > What is the easiest way to change the stored contact address ? > > Thanks, > ilker > __________ _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

2 1

[Users] Openser + Radius + Mysql: Acconting - AcctStopTime doesn't appear in the table radacct (radius DB)
by Alberto Pacifico 25 Aug '06

25 Aug '06

Hi all, I have problem with Accounting in radius database. The parameter AcctStopTime of the table named radacct isn't filled by Openser. The parameter AcctStartTime is okay. In my code of Openser I'm using: ... modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) # set the reporting log level modparam("acc", "log_level", 1) # number of the flag which will be used to mark messages for accounting modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) ... # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark the BYEs -- they are requests within a dialog if(method=="BYE") setflag(1); t_relay(); exit; }; # process the INVITEs if(method=="INVITE") { # mark the INVITEs for accounting setflag(1); # enforce record-routing so the BYEs will come through this server record_route(); }; .... Please, can you help me? Thanks! Alberto __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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Re: [Users] force_rtp_proxy() and m=video
by Atle Samuelsen 25 Aug '06

25 Aug '06

HI Adam, The reason for this is 2 reasons.. the nathelper module does not understand the port/1 parameter (wich says "use 2 ports, port 5010 and 5011) and it does also not understand the RTP/SAVPF. If you fix these 2 things, this should work :-) (Is this a Ojo?) -A In-Reply-To: <029301c6c82a$68a8c370$1601a8c0@oralnetws2> * Adam Linford <adam.linford(a)oralnet.co.uk> [060825 11:40]: > I've been happily using the force_rtp_proxy() function, exported from the > nathelper module for some time now. Recently, I've added some videophone > UA's to our system, and we are finding that rtpproxy is only relaying the > audio for sessions between video phones. > > Having ran some SIP traces I've noticed that the SDP is not being correctly > rewritten. Whilst the contact header for the audio is being allocated a > port from rtpproxy's range correctly, the corresponding m= field for the > video is not being rewritten at all. I've included an excerpt of the SDP I > am seeing below. > > v=0 > o=1000 16724 16724 IN IP4 192.168.1.66 > s=videophone > c=IN IP4 openser.ip.address.here > t=0 0 > a=sendrecv > m=audio 35030 RTP/AVP 98 0 8 > a=ptime:20 > a=rtpmap:98 iLBC/8000 > a=fmtp:98 mode=20 > m=video 5010/1 RTP/SAVPF 99 > b=AS:110 > a=rtpmap:99 H264/90000 > a=fmtp:99 > packetization-mode=1;parameter-add=0;sprop-parameter-sets=J0KgHpWgsTk=,KM4Ec > Q== > a=key-mgmt:key CnNnTdBxKycXpOh/Z6VPdeF1/u3+cVcQzAJAN4dWuGM > a=direction:active > > RTPproxy's UDP port range is 35000-65000, and as you can see, the m=audio > field is rewritten by calling force_rtp_proxy(), with the port value of > 35030, but the m=video field retains the port that the UA defines as its > starting UDP port for media. > > This is the first time I've dabbled with video and SIP, so was hoping > someone with more experience might already have hit this problem, before I > start delving into the code > > I'm running rtpproxy 0.3, and have also tried various other versions of > rtpproxy from cvs to no avail. > > Any help appreciated. > > Cheers, > Adam > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

1 0

[Serusers] About Cisco 7960 an option Call Forwarding All
by Javier Oviedo 25 Aug '06

25 Aug '06

Hi all In my SIP environment I have several Cisco 7960. These telephones have the option "Call Forwarding All", when I use this option with ser, the telephone generete a 302 SIP Code response, but the sip proxy does not forward the call to the new destiny. Any Idea to solve this problems Thanks in advance Joe

1 0

[Users] need to find out OUTGOING IF / SOCKET
by Gerry 25 Aug '06

25 Aug '06

Dear openser-users, I am trying to set-up a configuration where openser + RTPproxy are relaying messages between disconnected networks (i.e. openser + RTPproxy having more than one network-interface). With this method, I could additionally bridge IPv4-only and IPv6-only clients. Therefore I need to find out, on which interface (or socket, respectively) openser will send the message out because RTPproxy needs this info and is too dumb to find out by itself. I currently have a _very_ complicated solution which utilizes AVPs and mysql and it fails with NATed clients (detail: because the "contact" changes, I cannot look it up in the DB again). So, if you have any solutions or hints I would be VERY happy to "hear" from you. Greetings, Gerry. P.S.: Of course, I would be happy to share my config, but I don't want to SPAM this list. Therefore, if you are interested, just ask! But remember: my solution ONLY makes sense, if you're relaying between disconnected networks!

2 2

[Users] force_rtp_proxy() and m=video
by Adam Linford 25 Aug '06

25 Aug '06

I've been happily using the force_rtp_proxy() function, exported from the nathelper module for some time now. Recently, I've added some videophone UA's to our system, and we are finding that rtpproxy is only relaying the audio for sessions between video phones. Having ran some SIP traces I've noticed that the SDP is not being correctly rewritten. Whilst the contact header for the audio is being allocated a port from rtpproxy's range correctly, the corresponding m= field for the video is not being rewritten at all. I've included an excerpt of the SDP I am seeing below. v=0 o=1000 16724 16724 IN IP4 192.168.1.66 s=videophone c=IN IP4 openser.ip.address.here t=0 0 a=sendrecv m=audio 35030 RTP/AVP 98 0 8 a=ptime:20 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 m=video 5010/1 RTP/SAVPF 99 b=AS:110 a=rtpmap:99 H264/90000 a=fmtp:99 packetization-mode=1;parameter-add=0;sprop-parameter-sets=J0KgHpWgsTk=,KM4Ec Q== a=key-mgmt:key CnNnTdBxKycXpOh/Z6VPdeF1/u3+cVcQzAJAN4dWuGM a=direction:active RTPproxy's UDP port range is 35000-65000, and as you can see, the m=audio field is rewritten by calling force_rtp_proxy(), with the port value of 35030, but the m=video field retains the port that the UA defines as its starting UDP port for media. This is the first time I've dabbled with video and SIP, so was hoping someone with more experience might already have hit this problem, before I start delving into the code I'm running rtpproxy 0.3, and have also tried various other versions of rtpproxy from cvs to no avail. Any help appreciated. Cheers, Adam

1 0

[Serusers] Asterisk t38passthrough
by Ricardo Carvalho 25 Aug '06

25 Aug '06

Hi, I've installed Asterisk t38passthrough branch and I'm using one Grandstream ATA to connect Asterisk to a Fax machine. Every time I send a fax, it gets sent using codec G711, and never T.38. I added the following parameters in the [general] section as well as in device configurations: t38pt_udptl = yes t38pt_rtp = yes t38pt_tcp = yes I think that's the only thing that is needed to do to enable T.38 pass through... Why does Asterisk keeps sending in G711? Any help? Regards, Ricardo.

2 2

[Serusers] How to fix IP address stored in location db
by İlker Aktuna (Koç.net) 25 Aug '06

25 Aug '06

Hi, I am trying to change stored uri in the location db. I assume it could be done with the help of nathelper module and I use fix_nated_register() function just before save_noreply("location") command. However I still see the internal IP address of the client in the location db. What is the easiest way to change the stored contact address ? Thanks, ilker _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

2 1

Re: [Serusers] Voice Mail problem
by Greger V. Teigre 25 Aug '06

25 Aug '06

Please copy serusers. It's your failure route you should look at! g-) ram wrote: > Hi > > iam following the same document only to make my voice mail work > below is my config, its works if the user is registered locally > and if not available it will moving to voicemail > > But when iam dialing PSTN still its going to voicemail > > Aug 25 11:05:19 cdrouter ser[9337]: **************** vm start - begin > ****************** > Aug 25 11:05:19 cdrouter Sems[802]: Error: 404 voicemail: no email > address for user <1XXXXXX360> > Aug 25 11:05:19 cdrouter ser[9348]: ACC: call missed: from=99999 < > sip:99999@domain.com <mailto:sip:99999@domain.com>>;tag=2858293102, > i-uri=sip:1XXXXXXX360@domain.com <mailto:sip:1XXXXXXX360@domain.com>, > method=INVITE, o-uri= sip:1XXXXXX360@domain.com > <mailto:sip:1XXXXXX360@domain.com>, code=404 Not Found > Aug 25 11:05:19 cdrouter ser[9337]: **************** vm start - end > ****************** > > > ------------ > if (!lookup("location")) { > > # Voicemail specific configuration - begin > > if(method=="ACK" || method=="INVITE" || > method=="BYE"){ > > setflag(2); > if(t_newtran()){ > > t_reply("100","Trying -- just wait a > minute !"); > > if(method=="INVITE"){ > log(1,"**************** vm > start - begin ******************\n"); > > if(!vm("/tmp/am_fifo","voicemail")){ > log("could not contact the > answer machine\n"); > t_reply("500","could not > contact the answer machine"); > }; > log(1,"**************** vm start - end > ******************\n"); > break; > }; > > if(method=="BYE"){ > log(1,"**************** vm end - begin > ******************\n"); > if(!vm("/tmp/am_fifo","bye")){ > log("could not contact the > answer machine\n"); > t_reply("500","could not > contact the answer machine"); > }; > log(1,"**************** vm end - end > ******************\n"); > break; > }; > } > else { > log("could not create new transaction\n"); > sl_send_reply("500","could not create new > transaction"); > }; > > > /* Not found*/ > sl_send_reply("404","Not Found"); > }; > # Voicemail specific configuration - end > }; > > > > > On 8/24/06, *Greger V. Teigre* <greger(a)teigre.com > <mailto:greger@teigre.com>> wrote: > > Look at the features-callfwd.cfg example from onsip.org > <http://onsip.org/>, now available here: > http://www.iptel.org/ser/doc/gettingstarted > g-) > > ram wrote: > Hi > > I have existing SER Running > > i have local users ( extensions) and DID ( all other countries) > > iam able to make calls in and out , there is no problem > > but when the user not available, it has to send to Voice mail, > but its only sending the users > who are locally start with 9*, > > how can i send the user start with 9 or any DID not availble > send to voicemail > > here is my config > > > if (!lookup("location") && > ( uri=~"^sip:9.*@domain.com <mailto:9.*@domain.com>" ) > ) { > > # Voicemail specific configuration - begin > > if(method=="ACK" || method=="INVITE" || > method=="BYE"){ > > setflag(2); > if(t_newtran()){ > > t_reply("100","Trying -- just wait > a minute !"); > > if(method=="INVITE"){ > log(1,"**************** vm > start - begin ******************\n"); > > if(!vm("/tmp/am_fifo","voicemail")){ > log("could not contact the > answer machine\n"); > t_reply("500","could not > contact the answer machine"); > }; > log(1,"**************** vm start - > end ******************\n"); > break; > }; > > if(method=="BYE"){ > log(1,"**************** vm end - > begin ******************\n"); > if(!vm("/tmp/am_fifo","bye")){ > log("could not contact the > answer machine\n"); > t_reply("500","could not > contact the answer machine"); > }; > log(1,"**************** vm end - > end ******************\n"); > break; > }; > } > else { > log("could not create new transaction\n"); > sl_send_reply("500","could not create new > transaction"); > }; > > break; > }; > break; > }; > > > > any help will be great > > > > Ram > > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> > http://lists.iptel.org/mailman/listinfo/serusers > > >

1 0

[Users] Internet Providers changing the SIP headers
by Bruno Machado 25 Aug '06

25 Aug '06

Hi fellows. We are in trouble with this "deceitful" error. I mean, errors. Look at RURI. Our proxy receive this "thing". We have a number there! Sometimes, this number is shown within the word REGISTER, like that: "REGI:34846:STER". We can rewrite the URI, but Openser doesnt understand the method "REGI:34846:STER". Second error: Via header. The REGISTER is showing the NAT_DEVICE_IP instead UA_IP. And this VoIP isnt behind any VoIP device! So, "who" are changing it? "Who" can does it? Is possible that some "device" between UA and our proxy change the headers intentionaly? This kind of problem had been happening some months ago and began at the same day to VoIPs were different regions of the country (Brazil). We would like to know with someone already has face it. Thank you all. Bruno Machado U NAT_DEVICE_IP:50764 -> PROXY_IP:5060 REGISTER :50764sip:proxy.com.br SIP/2.0 Via: SIP/2.0/UDP NAT_DEVICE_IP:50765;branch=z9hG4bK427600003ca00000 From: <sip:7011423@proxy.com.br>;tag=ab510000bf39ffff To: <sip:7011423@proxy.com.br> Contact: * Call-ID: 499effff9ff20000(a)192.168.1.5 CSeq: 100 REGISTER Expires: 0 User-Agent: Grandstream HT386 1.0.3.17 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 U PROXY_IP:5060 -> NAT_DEVICE_IP:50765 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP NAT_DEVICE_IP:50765;branch=z9hG4bK427600003ca00000 From: <sip:7011423@proxy.com.br>;tag=ab510000bf39ffff To: <sip:7011423@proxy.com.br>;tag=22aae43fbf528d3860ded8429db379db.546f..Call-ID: 499effff9ff20000(a)192.168.1.5 CSeq: 100 REGISTER Proxy-Authenticate: Digest realm="proxy.com.br", nonce="44edfeda1fe71fa609e7599bf2513e999e3741dd" Content-Length: 0 --------------------------------- Novidade no Yahoo! Mail: receba alertas de novas mensagens no seu celular. Registre seu aparelho agora!

2 1

[Users] How to configure openser as a redirect server
by Padmaja RV 25 Aug '06

25 Aug '06

Hi all, I need to implement Openser as a redirect server for all the 3XX class of responses. I have checked the redirect.cfg example in the cvs and it includes only for the 300 response. The other two like 302 moved temporarily or 301 moved permanently are not included anywhere. Can anybody send me the script for the same? Thank You, Padmaja

2 1

[Serusers] maximum number of routes?
by Jean-Michel Hiver 25 Aug '06

25 Aug '06

Hi, I have found that ser yields an error when using up to 20 routes (i.e. route[20] block) and openser 40 routes. Is there an option to increase the number of routes ser can handle? Cheers, Jean-Michel.

2 2

[Serusers] avpops & usr_preferences -- multiple username/domain/attr sets
by O'Shaughnessy Evans 25 Aug '06

25 Aug '06

Hello. I'm trying to implement serial forking with the avpops module, but I've run into a small stumbling point. If I want to store more than 1 forwarding contact in the usr_preferences table, I can't because the table as originally defined from the ser codebase sets up (username, domain, attribute) as a primary key. To get around this, is it normal to use a different table that doesn't require that tuple to be unique, or is it recommended to change the definition of the usr_preferences table? In that case, what should the unique key be, if any? Thanks. -- = o'shaughnessy_evans -=- sys_admin @ (aloha|ilhawaii|myworld|turquoise).net = Pacific LightNet, Inc.

2 1

[Users] NAT traversal with mediaproxy on natted private address
by Simon Morvan 25 Aug '06

25 Aug '06

+----------+ +-----+ +------+ |OpenSER | +------+ |Phone| |Router| |MediaProxy| |Router| +-----+ +------+ +----------+ +------+ | | | | | | --------------------- -------------------------- -------------- Internet Private LAN A Private LAN B Here is the situation : SIP phones are in a private LAN A. They can speak with the server hosting openser and mediaproxy which is located on private LAN B. This server can speak with the internet through a router that nat (1:1) its private IP on a public one. To be able to interconnect inside sip phone with outside ones, openser must advertise the private address of the mediaproxy to the inside sip phones and its public (nated) address to the outside sip phones. But afaik , the media proxy module doesn't let me dynamically choose the IP which replace the phone IP in the SDP. How do you think I can get through this ? -- Simon Morvan.

3 3

[Users] Call a stored procedure
by Miroslaw Nawrot 24 Aug '06

24 Aug '06

Hi, Does someone have an axample of using avp_db_query() with a mysql stored procedure? I keep getting an error "submit_query: PROCEDURE openser.sp_getrouteinfo can't return a result set in the given context" whenever I pass a stored procedure call to that functions. Thanks. Mirek N Globalive Communications Corp. The Future of Voice T:416-640-1088 F:416-640-1089 60 Adelaide St E, 6th Floor Toronto, ON, M5C 3E4 www.globalive.com

1 0

[Users] openser admin
by Mike Williams 24 Aug '06

24 Aug '06

All, OpenSER Admin version 0.3 has been released. This fixes the bug where compatibility with openserctl was broken, and adds the ability to add other management accounts to the interface. Let me know what you think. Thanks, Mike Williams

1 0

[Users] How to change domain when REGISTERing UA?
by Dmitry Lyubimkov 24 Aug '06

24 Aug '06

Hi! Description of the problem: The some UA can send only IP addresses as domains. Some subscribers incorrectly configure their UA and as a result as the domain IP of SIP Proxy comes. In result in the table location the part of subscribers has the symbolical domain (voapp.ru) and part - IP address of SIP Proxy (62.33.22.14) It leads to to the certain complexities in switching incming calls. The problem disappears if to tell: modparam("usrloc", "use_domain", 0) modparam("registrar", "use_domain", 0) And use only names without domains. But in our case it does not approach since we have some independent domains. I tried to rewrite fields From and To route(8); save("location"); route[8] { #correct From and To in REGISTER messages with IP address instead of domain name if (search("From:.*@[0-9]+\.[0-9]+\.[0-9]+\.[0-9]+.*") or search("To:.*@[0-9+\.[0-9]+\.[0-9]+\.[0-9]+.*")) { subst("/^From:(.*)sip:(.*)@([0-9]+)\.([0-9]+)\.([0-9]+)\.([0-9]+)(.*)/Fr om:\1sip:\2@voapp.ru\7/i"); subst("/^To:(.*)sip:(.*)@([0-9]+)\.([0-9]+)\.([0-9]+)\.([0-9]+)(.*)/To:\ 1sip:\2@voapp.ru\7/i"); xlog("L_ERR", "REGISTER From To done\n"); return(-1); }; return(1); } Fields varied but function save took their not changed values from initial message. After that I have tried to send REGISTER message once again to myself for processing: if (!route(8)) { t_relay(); } else { save("location"); }; In this case in the location table there was a record with the corrected name of the domain. But in Contact field there was an address of SIP Proxy server instead of address of UA. Can anybody prompt idea how to to change domain in From To fields at reception of REGISTER message? Tnx. How to change domain when REGISTERing UA? Hi! Description of the problem: The some UA can send only IP addresses as domains. Some subscribers incorrectly configure their UA and as a result as the domain IP of SIP Proxy comes. In result in the table location the part of subscribers has the symbolical domain (voapp.ru) and part - IP address of SIP Proxy (62.33.22.14) It leads to to the certain complexities in switching incming calls. The problem disappears if to tell: modparam("usrloc", "use_domain", 0) modparam("registrar", "use_domain", 0) And use only names without domains. But in our case it does not approach since we have some independent domains. I tried to rewrite fields From and To route(8); save("location"); route[8] { #correct From and To in REGISTER messages with IP address instead of domain name if (search("From:.*@[0-9]+\.[0-9]+\.[0-9]+\.[0-9]+.*") or search("To:.*@[0-9+\.[0-9]+\.[0-9]+\.[0-9]+.*")) { subst("/^From:(.*)sip:(.*)@([0-9]+)\.([0-9]+)\.([0-9]+)\.([0-9]+)(.*)/Fr om:\1sip:\2@voapp.ru\7/i"); subst("/^To:(.*)sip:(.*)@([0-9]+)\.([0-9]+)\.([0-9]+)\.([0-9]+)(.*)/To:\ 1sip:\2@voapp.ru\7/i"); xlog("L_ERR", "REGISTER From To done\n"); return(-1); }; return(1); } Fields varied but function save took their not changed values from initial message. After that I have tried to send REGISTER message once again to myself for processing: if (!route(8)) { t_relay(); } else { save("location"); }; In this case in the location table there was a record with the corrected name of the domain. But in Contact field there was an address of SIP Proxy server instead of address of UA. Can anybody prompt idea how to to change domain in From To fields at reception of REGISTER message? Tnx. Dmitry

1 0

RE: [Serusers] Alternate port for rewritehost
by Stephen Helfen 24 Aug '06

24 Aug '06

Thanks...found it 2 seconds after sending the email! :PDate: Thu, 24 Aug 2006 21:21:10 +0200From: greger(a)teigre.comTo: steve_helfen(a)hotmail.comCC: serusers(a)iptel.orgSubject: Re: [Serusers] Alternate port for rewritehost Use rewritehostport("domain:5060") g-) Stephen Helfen wrote: All, I am receiving SIP traffic on a non-5060 port but need to change the port of the rewritehost to 5060 as the carrier is dropping my INVITE messages with the non-5060 port. Any suggestions? Thanks, Steve Search from any Web page with powerful protection. Get the FREE Windows Live Toolbar Today! Try it now! _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers _________________________________________________________________ Check the weather nationwide with MSN Search: Try it now! http://search.msn.com/results.aspx?q=weather&FORM=WLMTAG

1 0

[Serusers] SER and RTPproxy?!
by Caxton 24 Aug '06

24 Aug '06

Hi all: The document metion that "RTPproxy will not be installed on the SER server, but on a remote machine." If I install SER on "192.168.1.3", and install RTPproxy on "192.168.1.5". How do they work? Now, I install they on a same machine, and I run SER first, then usr command "rtpproxy -l xxx.xxx.xxx.xxx" to force rtpproxy. The SIP client can register behind NAT, but there are some problem with RTP stream. I found that RTP strem is able to relay to this machine, but this machine do not transmit those data to destination!! PC(A) Server(SER+RTPproxy) PC(B) RTP Stream(A) RTP Stream(B) 1. ----------------------------> <---------------------------- 2. Receive RTP stream A and B 3. Server should transmit RTP stream(A) to PC(B) and RTP stream(B) to PC(A), but this step do not work. (I am not sure that server does not forward the RTP packets or Clients do not receive RTP packet.) What is the problem? How should I operate if RTPproxy install on another machine? Because the command "rtpproxy" will not be used on the machine which have installed SER. Please help me, thank you. Regards, Caxton

3 2

[Serusers] Alternate port for rewritehost
by Stephen Helfen 24 Aug '06

24 Aug '06

All,I am receiving SIP traffic on a non-5060 port but need to change the port of the rewritehost to 5060 as the carrier is dropping my INVITE messages with the non-5060 port. Any suggestions?Thanks,Steve _________________________________________________________________ Search from any Web page with powerful protection. Get the FREE Windows Live Toolbar Today! http://get.live.com/toolbar/overview

2 1

[Users] Please help for guiding serweb installation in openser server
by Ferianto siregar 24 Aug '06

24 Aug '06

Dear all, Thank you very much to all of you for help me to answer my question about serweb before. So, based on your suggestion, I tried to install serweb 0.9.3 for my openser server. But, when I tried to install it, I am confused. I don`t understand about the installation note that I have got from untar the serweb 0.9.3 packet. These are some questions that I need to ask: 1. Where should we untar the serweb 0.9.3 packet? Should we untar it in root or anywhere we like? Because when we untar it, there will be a serweb folder in the location where we untar it (for example, when we untar the packet in root there will be a serweb directory in root) . This serweb folder also contain html directory and phplib directory. As I know, There are also html directory in /root/var/www. So, What directory should I use? Should I delete the html directory and phplib directory in /root/var/www/? 2. Is there any relationship between the html directory in /root/var/www and html directory from /root/serweb? Which one should I use? 3. In the installation note I got this message : "install it, unpack the distibution on your webserver so that the directory html is in document root and phplib is not accessible via web" Can anybody help me to explain what this message means? and give me an example? 4. If I can make a wish for all of you, I do hope anybody can share the serweb configuration that she/he has configured to me. So, it can become the guidelines for me. That `s all the question that I got when I tried to install the serweb. I am confused. I do hope anybody can help me because I am new in this system. Thank you very much for your help. Thanks Regards, Ferianto --------------------------------- Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.

2 1

[Users] NAT (again)
by Mark Kent 24 Aug '06

24 Aug '06

I'm testing my rtpproxy config with a linksys phone and a grandstream phone, either not NAT'ed, or behind a cisco (full cone), or behind a Netgear (symmetric). Things generally work, but I have a problem with the grandstream behind the symmetric nat. This is the register: U NAT-IP:3284 -> PROXY-IP:5060 REGISTER sip:sip.example.com SIP/2.0 Via: SIP/2.0/UDP NAT-IP:3280;branch=z9hG4bK0103d0ff79cc9c02 From: "Grandstream" <sip:frick@sip.example.com;user=phone>;tag=743335557afcade4 To: <sip:frick@sip.example.com;user=phone> Contact: <sip:frick@NAT-IP:3280;user=phone> Authorization: Digest username="frick", realm="sip.example.com", algorithm=MD5, uri="sip:sip.example.com", nonce="foo", response="bar" Call-ID: 11f6defb0857db17(a)192.168.0.100 CSeq: 116 REGISTER Expires: 60 User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 This is the resulting contact: ...Record(0x2864b680).. domain: 'location' aor : 'frick(a)sip.example.com' ~~~Contact(0x2864d4e0)~~~ domain : 'location' aor : 'frick(a)sip.example.com' Contact : 'sip:frick@NAT-IP:3280;user=phone' Expires : 34 q : Call-ID : '11f6defb0857db17(a)192.168.0.100' CSeq : 119 User-Agent: 'Grandstream BT100 1.0.6.7' received : 'sip:NAT-IP:3284' Path : '' State : CS_SYNC Flags : 1 Sock : PROXY-IP:5060 (0x8122f50) Methods : 5695 next : 0x0 prev : 0x0 ~~~/Contact~~~~ .../Record.. Now, what I don't understand is the two ports used above: 3280 and 3284. It looks like the phone thinks it is being natted to port 3280, but it's really getting port 3284 When I make a call I get U NAT-IP:3284 -> PROXY-IP:5060 INVITE sip:16509415674@sip.example.com;user=phone SIP/2.0 Via: SIP/2.0/UDP NAT-IP:3280;branch=z9hG4bK4a4c039fbcb5bef8 From: "Grandstream" <sip:frick@sip.example.com;user=phone>;tag=2dc2fc3c94e93951 To: <sip:16509415674@sip.example.com;user=phone> Contact: <sip:frick@NAT-IP:3280;user=phone> Supported: replaces Call-ID: 1f04f9c6ec474f70(a)192.168.0.100 CSeq: 26381 INVITE User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 419 v=0 o=frick 8000 8002 IN IP4 NAT-IP s=SIP Call c=IN IP4 NAT-IP t=0 0 m=audio 3295 RTP/AVP 0 8 4 18 2 15 97 9 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/16000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 and my side sends back the Proxy Authentication Required, but it sends it to port 3280. Now, is that based on the Via: header or the Contact: header, or (unlikely) the stored contact info from the REGISTER? The phone acts like it never got it (and I'm guessing it didn't) and sends the INVITE again without any auth data. The thing that bugs me is that this is an outbound call, from the phone out through the NAT box, I figured that would be doable. The Linksys942 works, so I'm hoping this is a Grandstream bug, or something I can fix in my openser config. Thanks, -mark

3 3

[Users] Limiting Call Volume and CPS
by Brian Jones 24 Aug '06

24 Aug '06

Hi, I was wondering if anyone had any suggestions on how I could use OpenSER to limit call volume and call per second on a peer basis. Thanks! Brian.

3 4

[Users] Weird forwarding with Route header
by Jeremie Le Hen 24 Aug '06

24 Aug '06

Hi list, my proxy (yy.yy.yy.yy) receives a message like this: BYE sip:12345678@xx.xx.xx.xx SIP/2.0 Record-Route: <sip:xx.xx.xx.xx;ftag=875188d2;lr=on> Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bK07cb.466d8587.0 Via: SIP/2.0/UDP ww.ww.ww.ww:5060;branch=z9hG4bK875196bf Route: <sip:xx.xx.xx.xx;ftag=1056999653;lr=on> My ruleset is something like this (with the REGISTER logic pulled out) : %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; force_rport(); if (!method == "REGISTER") { record_route(); }; if (loose_route()) { route(1); }; loopkup("location"); route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% I would expect the packet to leave my router, back to xx.xx.xx.xx : BYE sip:12345678@xx.xx.xx.xx SIP/2.0 Record-Route: <sip:yy.yy.yy.yy;lr;ftag=2873864> Record-Route: <sip:xx.xx.xx.xx;ftag=875188d2;lr=on> Via: SIP/2.0/UDP yy.yy.yy.yy;branch=zu83fdhDB8d.67235676.0 Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bK07cb.466d8587.0 Via: SIP/2.0/UDP ww.ww.ww.ww:5060;branch=z9hG4bK875196bf Route: <sip:xx.xx.xx.xx;ftag=1056999653;lr=on> But the outgoing packet is : BYE sip:xx.xx.xx.xx;ftag=1056999653;lr=on SIP/2.0 Record-Route: <sip:xx.xx.xx.xx;ftag=875188d2;lr=on> Via: SIP/2.0/UDP yy.yy.yy.yy;branch=z9hG4bK07cb.02005244.0 Via: SIP/2.0/UDP xx.xx.xx.xx;rport=5060;branch=z9hG4bK07cb.466d8587.0 Via: SIP/2.0/UDP ww.ww.ww.ww:5060;branch=z9hG4bK875196bf Do you have any idea about what is going on ? Thank you. Best regards, -- Jeremie Le Hen < jeremie at le-hen dot org >< ttz at chchile dot org >

2 2

[Users] dropping a 200
by Simon Morvan 24 Aug '06

24 Aug '06

Just to be sure : there's no way to call drop() in an onreply_route block when it's not a provisionnal reply ? -- Simon.

2 1

[Users] how to correctly handle NAT on inbound call from another SIP-proxy?
by Gerd Feiner 24 Aug '06

24 Aug '06

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 hi list, we have some trouble handling NAT correctly when receiving a call from another SIP-proxy. the mediaproxy-module does not seem to rewrite SDP when it finds another proxy's record-route header, though it invokes mediaproxy and sends uncorrected SDP to the other SIP-server resulting in one-way-audio as the provided c=-field includes the RFC1918-address of the client. when receiving calls through our Cisco AS5350 everything is fine ('cause it does not add the record-route). the addition of the header cannot be circumvented bacause the other provider wants to bill calls (obviously) and therefor needs to be in the SIP-dialog. is there a neat solution to this problem or do we have to get a session-border-controller or some B2BUA-config for handling the received calls? and by the way, how do i distinguish external INVITEs destined for one of our clients (uri==myself) from the INVITEs being sent out to the callee upon an external INVITE? if there was a way we could invoke mediaproxy from there? how is the recommended setup? brgds, g.feiner -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE7aVOqvDCGdeEb8oRArDjAJ9ssHEHBAIGmSB34oHQxuuT0ke15wCgmKvB chyomyJgqU9NVMep7QV1nFI= =r5Xb -----END PGP SIGNATURE-----

1 0

[Users] Q:ABOUT:MODULE:auth_radius
by Alejandro Sanchez 24 Aug '06

24 Aug '06

Hello. openser 1.1 radiusclient-ng 5 freeradius 1.1.1 I need to enable the modules *_radius for my openser instalation a follow the guide for enable radius then when i try to start openser i get the following error. 0(0) ERROR: auth_radius: can't get code for the Digest-Response attribute Anyone knows what is the cause for this error? Thank's Alex. ___________________________________________________________ Do You Yahoo!? La mejor conexión a Internet y <b >2GB</b> extra a tu correo por $100 al mes. http://net.yahoo.com.mx

2 1

[Serusers] multiple domains in Ser
by Ricardo Carvalho 24 Aug '06

24 Aug '06

Hi, Is there a way to coexist multiple domains in Ser, distinguishing users even if they have the same username in subscriber table but different domains? I'll explain better, I have subscriber table with: | username | domain | e-mail | ------------------------------------------------ | user_A | domain_A | user_A@domain_A | ------------------------------------------------ | user_A | domain_B | user_A@domain_B | and in domain table there are both domains domain_A and domain_B. But when someone calls user_A, (with or without writing their domain), both phones ring (if both users have their phones registered), when I want only one to ring because they are different users!! Is there a way in Ser to do this right? Regards, Ricardo.

2 2

Re: [Serusers] multiple domains in Ser
by Ricardo Carvalho 24 Aug '06

24 Aug '06

It's true Asterisk can use different contexts, but that doesn't solve my problem because I use Ser + Asterisk and Ser is used for routing SIP messages between SIP UAs, not Asterisk. In my architecture, Asterisk is only used for advanced services like voicemail or call conferencing and for gateway, not to route messages between UAs. Ricardo. openser(a)comcast.net wrote: > The way to do that is intergrate Ser with Asterisk and in Asterisk you > can build context and keep the same numbers for different context like > a pbx. > > > -------------- Original message -------------- > From: Ricardo Carvalho <rcarvalho(a)iric.up.pt> > > > Hi, > > > > Is there a way to coexist multiple domains in Ser, > distinguishing users > > even if they have the same username in subscriber table but > different > > domains? > > > > I'll explain better, I have subscriber table with: > > > > | username | domain | e-mail | > > ------------------------------------------------ > > | user_A | domain_A | user_A@domain_A | > > ------------------------------------------------ > > | user_A | domain_B | user_A@domain_B | > > > > and in domain table there are both domains domain_A and domain_B. > > > > But when someone calls user_A, (with or without writing their > domain), both > > phones ring (if both users have their phones > > registered), when I w! ant onl y one to ring because they are > different users!! > > Is there a way in Ser to do this right? > > > > Regards, > > > > Ricardo. > > > > > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] Voice Mail problem
by ram 24 Aug '06

24 Aug '06

Hi I have existing SER Running i have local users ( extensions) and DID ( all other countries) iam able to make calls in and out , there is no problem but when the user not available, it has to send to Voice mail, but its only sending the users who are locally start with 9*, how can i send the user start with 9 or any DID not availble send to voicemail here is my config if (!lookup("location") && ( uri=~"^sip:9.*@domain.com" ) ) { # Voicemail specific configuration - begin if(method=="ACK" || method=="INVITE" || method=="BYE"){ setflag(2); if(t_newtran()){ t_reply("100","Trying -- just wait a minute !"); if(method=="INVITE"){ log(1,"**************** vm start - begin ******************\n"); if(!vm("/tmp/am_fifo","voicemail")){ log("could not contact the answer machine\n"); t_reply("500","could not contact the answer machine"); }; log(1,"**************** vm start - end ******************\n"); break; }; if(method=="BYE"){ log(1,"**************** vm end - begin ******************\n"); if(!vm("/tmp/am_fifo","bye")){ log("could not contact the answer machine\n"); t_reply("500","could not contact the answer machine"); }; log(1,"**************** vm end - end ******************\n"); break; }; } else { log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); }; break; }; break; }; any help will be great Ram

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[Serusers] Broken ser.cfg
by Jean-Michel Hiver 23 Aug '06

23 Aug '06

Can somebody please take a look at the routing logic below... I am trying to use exec_dset along with another program to do some LCR with SER but it doesn't work properly (here is a capture of what it produces: http://pastebin.ca/146771 Any ideas of what's doing wrong? Code: # -------------------------------------- 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 >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; if (method=="REGISTER") { sl_send_reply("501", "Not Implemented"); break; }; if (method=="OPTIONS") { sl_send_reply("501", "Not Implemented"); break; }; record_route(); setflag(1); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; if (method=="INVITE") { route(2); break(); }; route(1); break(); } route[1] { if (!t_relay()) { sl_reply_error(); }; } route[2] { # replaces the INVITE URI with the right host exec_dset ("/usr/local/Asterisk-LCR/bin/ser-lcr"); if (uri=~"^sip:STOP@127") { sl_send_reply("503", "Service Unavailable"); break; }; t_on_failure("2"); if(isflagset(1)) { append_branch(); }; setflag(1); t_relay(); } # /usr/local/Asterisk-LCR/bin/ser-lcr will give us the next uri to try # in case of failure, or send sip:STOP@127.0.0.1 if no more routes failure_route[2] { route(2); } # --------------------------------------

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[Serusers] Need config help for PSTN call routing blocks to service provider.
by Chahn John Kim 23 Aug '06

23 Aug '06

Hi, I am new to SER and need experts' help in PSTN call routing block to call termination service provider in a scenario below: We have our own sip realm/server that will handle all VoIP calls among our own subscribers; sip.ours.com. This will forward all PSTN calls, authenticated and authorized by it, to sip.theirs.com, which will authorize usning our IP address and forward to pstngateway.theirs.com. What I have been given by SP is: sip.theirs.com, IP, myuserID and password. What they are given by me is our IP and hostaname. sip.ours.com and IP. My first try was simply; rewritehostport(sip.theirs.com:5060); t_relay(); ,which fails. Can someone tell me proper approach in ser commands based on the scenario above? Thank you so much in advance. John K

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Re: [Users] options for rtpproxy.
by Bogdan-Andrei Iancu 23 Aug '06

23 Aug '06

Hi Bill, Bill Zhang wrote: >Anyone can explain following options: > >1. -2fv > > v - print the version f - stay in foreground, don't become a daemon 2 - duplicate mode >2. -s path > > command socket path >3. -L nfiles > > maximum file descriptor number that can be opened regards, bogdan >The rest of them seems to be straight forward. This rtproxy documentation >is HARD to find:(. > >usage: rtpproxy [-2fv] [-l addr1[/addr2]] [-6 addr1[/addr2]] [-s path] [-t >tos] [-r rdir [-S sdir]] [-T ttl] [-L nfiles] > >Thanks in advance. > >Best Regards, >Bill > >

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Re: [Users] rtpproxy or RTPProxy, where to download or checkout--found it:). Re: Users Digest, Vol 15, Issue 61
by Mark Kent 23 Aug '06

23 Aug '06

In the RTPproxy source code, if you look at rtpp_log.h you really get the impression that syslog() functionality can be dropped in with ease. I still believe that, and believe it can be elegant, despite the fact that my solution is fairly crappy. But, it's done... so here it is (it relies on dprint.[ch] from the openser1.1 distribution): After ./configure, do this: In Makefile, change CFLAGS line like this: CFLAGS = -g -O2 -DUSESYSLOG -I../openser-1.1.0-tls (point to wherever you build openser) In rtpp_log.h, bracket the existing macros with this: #ifdef USESYSLOG #include "dprint.h" #define rtpp_log_open(app, call_id, flag) (0) #define rtpp_log_write(level, handle, format, args...) LOG(level, format, ##args) #define rtpp_log_ewrite(level, handle, format, args...) \ do { \ LOG(level, format, ##args); \ LOG(level, ": %s\n", strerror(errno)); \ } while(0); #define rtpp_log_close(handle) while (0) {} #else /* USESYSLOG */ {existing macros starting with rtpp_log_open and including rtpp_log_close} #endif /* USESYSLOG */ and stuff these lines up near the top of main.c, around line 64: #ifdef USESYSLOG int debug = 1; int log_stderr = 0; int log_facility = LOG_LOCAL7; char* log_name = "foobar"; int pt = 0, process_no = 0; #endif The link will fail when you do make, but just copy-and-paste the failed link command back into the shell and add ../openser-1.1.0-tls/dprint.o to the end of the line and it'll link, and run, and log. Also, some OS (Solaris) will require you to open and close the syslog channel. So, if you need to do that, then you need to define the rtpp_log_open and rtpp_log_close macros too. I'm on freebsd, and it seems I don't need to do that. -mark P.S. If someone has an elegant solution to this then please post it.

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[Serusers] SER / Missing call records
by Darren Sessions 23 Aug '06

23 Aug '06

All, We're experiencing a problem with missing call records. There is nothing in the log files that indicate a problem either with SER or our database server and likewise there is nothing in the database server's logs to indicate a problem. System loads are within reason on both servers, network connectivity has been tested and verified, and a through hardware shakedown on both boxes has been completed without any errors. We first noticed this problem when our underlying vendor's bill (based on minutes of course) did not correspond with our own. Upon doing a CDR comparison (based on what was in both the acc and calls tables), we were able to verify that the calls in dispute were indeed ours (based on the sip call-id's IP addresses). We're processing roughly 1.4 million calls per day on this SER server in question. Out of those calls, 508 calls (give or take a couple hundred) with a total duration of over 5,300 minutes - can not be accounted for in any of the SER database tables. We're running 0.9.5-pre1 with Suse Enterprise Server 9.0 (2.6.11 Kernel). Server is an HP DL380 with dual 3.2GHz processors with 6GB physical ram (only 1.4GB in use system wide); disk space is not an issue as well (only 10% in use). We've scoured the mailing lists and change logs for any hint of an issue similar to ours without any results. The closest we've been able to come that's anywhere near along the lines of our issue in any way that made any sense were these posts: http://lists.iptel.org/pipermail/serdev/2003-August/000602.html http://lists.iptel.org/pipermail/serusers/2005-February/015102.html At this point, we're not sure if this issue has something to do with the FIFO and SER making bulk dump uploads with call data to the database, or some other issue along those lines - or if this is a new issue all together. Any help would be greatly appreciated. Thanks, - Darren Fifo settings: fifo="/tmp/ser_fifo" fifo_db_url="mysql://xxx:xxx@xxx/serdb" Here are our module specific parameters: # -- timers -- modparam("tm", "wt_timer", 10) modparam("tm", "retr_timer1p1", 2) # -- usrloc params -- modparam("usrloc", "db_mode", 2) # -- database URL -- modparam("usrloc|acc|auth_db|group","db_url","mysql://xxx:xxx@xxx/serdb" ) # -- auth params -- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth","secret","xxx") # -- rr params -- modparam("rr", "enable_full_lr", 1) # -- Nathelper -- modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 10) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_disable", 1) # -- acc params -- modparam("acc", "db_flag", 3)

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[Serusers] Numbering Plan
by Daniel Mirrha Santos 23 Aug '06

23 Aug '06

Does anybody knows if I can create a numbering plan using a database? How is the easiest mode to do a numbering plan on ser? Thanks, Daniel

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Re: [Users] 403 Forbidden-----onsip configs testing. user not able to register
by Ferianto siregar 23 Aug '06

23 Aug '06

Hi, Ram its Not looking up in aliases table , it look ups only on location table. the each request and responses . So, You have to add your domain in the domain table by using : insert into domain values('urs_domain', ' date) ; in openser databases tables. I do hope it can help you. Thanks. On 8/22/06, ram wrote: > > Hi all > > iam trying to read the getting started document from onsip > when iam configuring pstn-gw.cfg > with the below config > > when the X-lite user try to loging , SIP/2.0 403 Forbidden, in the Log of > X-lite > > what is wrong , even i have changed AVP after reading module configs in > openser. > > any one suggest me what is wrong iam doing here > > Ram > > ----------------- > openser.cfg > # > # $Id: openser.cfg,v 1.6 2006/02/15 18:23:46 bogdan_iancu Exp $ > # > # simple quick-start config script > # > > # ----------- global configuration parameters ------------------------ > > debug=9 # debug level (cmd line: -dddddddddd) > fork=yes > log_stderror=no # (cmd line: -E) > > /* Uncomment these lines to enter debugging mode > #fork=no > #log_stderror=yes > */ > > check_via=no # (cmd. line: -v) > dns=no # (cmd. line: -r) > rev_dns=no # (cmd. line: -R) > listen=myip > port=5060 > children=4 > fifo="/tmp/openser_fifo" > fifo_db_url="mysql://openser:openserrw@localhost /openser" > > > # ------------------ module loading > ---------------------------------- > > > 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/textops.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/domain.so" > loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" > loadmodule "/usr/local/lib/openser/modules/nathelper.so" > loadmodule "/usr/local/lib/openser/modules/avpops.so" > loadmodule "/usr/local/lib/openser/modules/permissions.so" > > modparam("auth_db|permissions|uri_db|usrloc","db_url", > "mysql://openser:openserrw@localhost/openser") > 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("auth_db", "calculate_ha1", yes) > modparam("auth_db", "password_column", "password") > > modparam("registrar", "nat_flag", 6) > > modparam("rr", "enable_full_lr", 1) > > modparam("tm", "fr_timer_avp", "i:25") > modparam("tm", "fr_inv_timer_avp", "i:25") > > 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"); > return; > }; > > if (msg:len > max_len) { > sl_send_reply("513", "Message Overflow"); > return; > }; > > # > ----------------------------------------------------------------- > # Record Route Section > # > ----------------------------------------------------------------- > if (method=="INVITE" && client_nat_test("3")) { > # INSERT YOUR IP ADDRESS HERE > > record_route_preset("myproxyip: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); > return; > }; > > # > ----------------------------------------------------------------- > # Call Type Processing Section > # > ----------------------------------------------------------------- > if (!is_uri_host_local()) { > if (is_from_local() || allow_trusted()) { > route(4); > route(1); > } else { > sl_send_reply("403", "Forbidden"); > }; > return; > }; > > if (method=="CANCEL") { > route(1); > return; > } else if (method=="INVITE") { > route(3); > return; > } else if (method=="REGISTER") { > route(2); > return; > }; > > lookup("aliases"); > if (uri!=myself) { > route(4); > route(1); > return; > }; > > if (!lookup("location")) { > sl_send_reply("404", "User Not Found"); > return; > }; > > 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"); > return; > }; > > if (!check_to()) { > sl_send_reply("401", "Unauthorized"); > return; > }; > > consume_credentials(); > > if (!save("location")) { > sl_reply_error(); > }; > } > > route[3] { > > # > ----------------------------------------------------------------- > # INVITE Message Handler > # > ----------------------------------------------------------------- > > if (!allow_trusted()) { > > if (!proxy_authorize("","subscriber")) { > proxy_challenge("","0"); > return; > } else if (!check_from()) { > sl_send_reply("403", "Use From=ID"); > return; > }; > > consume_credentials(); > }; > > if (client_nat_test("3")) { > setflag(7); > force_rport(); > fix_nated_contact(); > }; > > if (uri=~"^sip:1[0-9]{10}@") { > strip(1); > }; > > lookup("aliases"); > if (uri!=myself) { > route(4); > route(1); > return; === message truncated === --------------------------------- Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.

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[Users] Problem with forking
by Claudio Miceli 23 Aug '06

23 Aug '06

I am in a big trouble. I am using openser 1.0.1 and I am still having some problems with redirect. I get some numbers in myredirect and I try to pass it with t_relay(), but at this moment everything gets confused. The calls do not happen sequentially or in any logical order. Is there another way of doing the forking ? Am i doing something terribly wrong ? And sometimes when I make a call my openser sends messages to just one client and not to the other, or sends to both but with a great time between them. What could it be ? Do someone know it ? Thanks for helping. Claudio Miceli de Farias

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RE: [Users] registered("location")
by Helge Waastad 23 Aug '06

23 Aug '06

Hi again, I have: modparam("group|auth_db|usrloc|uri_db|speeddial|registrar|avpops|alias_d b", "use_domain", 1) In location table: | username | domain | | | | | 123456 | smartnet.no | | 654321 | smartnet.no | If I set usrloc|registrar use_domain=0, the only thing that changes is that the domain column is empty. Br hw -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: 22. august 2006 18:06 To: Helge Waastad Cc: Users(a)openser.org Subject: Re: [Users] registered("location") strange.... do you have the use_domain param enabled or disabled? could you check if the location table has for "user" column the correct format (all entries): username@domain (if use_domain is on) username (if use_domain is off) regards, bogdan Helge Waastad wrote: >Hi, >Ive just stopped my openser server, >deleted all entries in the location table and restarted the OpenSER >server. > >Still i get the same: > >BUG:usrloc:db_load_urecord: AOR has no @ while use_domain is enabled > >message. > >br hw > >tir, 22,.08.2006 kl. 16.36 +0300, skrev Bogdan-Andrei Iancu: > > >>Hi, >> >>my guess is you had usrloc with use_domain and with contacts into the >>location table you switched use_domain on. >>BUG:usrloc:db_load_urecord: AOR has no @ while use_domain is enabled >>please remove the entire contact of the location table and try again. >> >>regards, >>bogdan >> >>Helge Waastad wrote: >> >> >> >>>Hi, >>> >>>I receive a >>>BUG:usrloc:db_load_urecord: AOR has no @ while use_domain is enabled >>> >>>in my syslog. Could this be the reason??? >>> >>>br hw >>> >>> >>> >>>tir, 22,.08.2006 kl. 15.28 +0300, skrev Bogdan-Andrei Iancu: >>> >>> >>> >>> >>>>Hi Helge, >>>> >>>>it should work. Can you check if any db query is performed by the function? >>>> >>>>regards, >>>>bogdan >>>> >>>>Helge Waastad wrote: >>>> >>>> >>>> >>>> >>>> >>>>>Hi, >>>>>I'm just wondering if the 'registered("location")' works when using >>>>>db only mode for usrloc? >>>>> >>>>>br hw >>>>> >>>>> >>>>>_______________________________________________ >>>>>Users mailing list >>>>>Users(a)openser.org >>>>>http://openser.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>

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[Users] Please help me for SMS module (SIP to GSM)
by ade darmawan 23 Aug '06

23 Aug '06

Please help me mr... i have read this module...but i don't understand with 4 question : 1. I found that in the SMS Module (SIP to GSM) don't explained the software which used by the client/user agent or where we should to do testing in client? 2. I wanna know the other facilities should be used, beside openser-1.0.0.tar.gz,mysql,DNS? 3. For the final result, is this module could send SMS by 2 directions? 4. Could you please the step to do the testing? Thank's for all --------------------------------- Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail.

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[Users] Byes not accounted
by Jayesh Nambiar 23 Aug '06

23 Aug '06

Hi all, I have got a strange problem with my accounting. This is my scenario: I have a registered user dialling a pstn number. The call is connected and then the registered UA switches off the IP device and there is no BYE from the registered UA. Now the pstn gateway sends a BYE after certain period and Openser tries to send that BYE to the registered UA, but the registered UA is powered off and it never replies. So Openser retransmits the BYE messages and finally generates a 408 response towards the gateway. But Openser does not account that BYE in the acc table. According to me, the Openser should acount that BYE for proper accouning, Please correct me if I am wrong. Also suggest some method so that I can make the unacknowledged BYEs account in the acc table. In my config I have: if(method=="BYE") { setflag(1); }; if(method!="REGISTER") { record_route();}; setflag(1); if(loose_route()) { ..... #the rest logic} an so on. I have flag 1 as db_flag. thanks in advance, w/regards, jayesh. --------------------------------- Here's a new way to find what you're looking for - Yahoo! Answers Send FREE SMS to your friend's mobile from Yahoo! Messenger Version 8. Get it NOW

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