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39 participants
37513 discussions

[Users] Whixh Router is Best for security...
by raviprakash sunkara 10 Oct '06

10 Oct '06

Hello Users, If I Keep the Router , My openser is Not working fine. But If place it Directly to modem, it works Kool and Fine. Actually My router is Firewall router which does have the NAT configuring , It was NetGear.. Can tell Which Router I need to Use for My Sip Server...... -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

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[Users] Openser ALG setup
by Lindsey Hans Joseph Lao 10 Oct '06

10 Oct '06

Hello All, I am setting up Openser in this configuration: Client ---------> (private interface) Router (public interface) --------> Registrar The router is configured with Openser having mediaproxy module support. What I want to achieve is for nat traversal to happen at the router side such that when the sip messages reach the registrar, it would already contain the public IP address in its contact header field. I am using Openser here more like an application level gateway. Openser is listening on the public interface only. Currently, my setup is not working with the private address of the client registering in the registrar. Is this a possible setup? Can someone show me the way? Thamk you very much. Hans

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[serusers] Connecting Another SER
by John Shellhouse 10 Oct '06

10 Oct '06

Hi Everyone, I currently have SER and Asterisk on one machine, and recently launched another SER on another machine and wanted to know how do I connect SER-1 to SER-2. Thanks in advance. John Shellhouse

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[Serusers] Need help proper way to see avpops value
by reza shah 10 Oct '06

10 Oct '06

Hi, I just try avpops to get a value from usr_preferences. I have followed the onsip.org article, and some of previous email in this list. Still have trouble to see the value inside avpops variable. May be i missed something. i`m using ser 0.9.6 with freebsd. Did i do the right way? my codes are: loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" modparam("avpops","avp_url","mysql://ser:heslo@localhost/ser") modparam("avpops","avp_table","usr_preferences") modparam("avpops","username_column","username") modparam("avpops","domain_column","domain") modparam("avpops","value_column","value") modparam("avpops","type_column","type") modparam("avpops","attribute_column","attribute") #inside register method avp_db_load("$from/domain","s:mapnumber"); xlog("L_NOTICE","value = $avp(s:mapnumber)\n"); __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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RE: [Serusers] DNS and IP
by Rosa De Santis 09 Oct '06

09 Oct '06

Hi again, Weiter. I already have "alias=the.domain.name" in my config, but its not working :( The complete scenario is like this: I have two SIP servers: server A and server B. I have clients registered in each server, lets call them UA1 UA2 registered on server A, and UA3 UA4 on server B. When I use ip to register the clients then I can make calls in all ways. UA1 to UA2, UA1 to UA3, and viceversa, etc... But if I use dns to register then users can call on the same server: UA1<->UA2, UA3<->UA4, but users from server A can't call to users on server B and viceversa. I have the dns registered on both servers. Please, any idea??? Thanks, Rosa. > ----------------------------------------------- > > Date: Sun, 8 Oct 2006 12:02:24 +0200 > > From: bp4mls(a)googlemail.com > > To: rosadesantis(a)hotmail.com > > Subject: Re: [Serusers] DNS and IP > > CC: serusers(a)iptel.org > > > > Make sure the box where SER is running can also resolve the name to it's own IP. > > You can also use "alias=the.domain.name" in you config. > > On 10/6/06, Rosa De Santis <rosadesantis(a)hotmail.com> wrote: > > Hello all. > > Please, I have a problem using DNS or IP on the client side. > > When I use IP all calls are fine, but if I use the DNS of the server that point to the same IP some calls can't be done. > > Is there any place to configure this on ser.cfg ??? > > Please help. > > Thanks. > > _________________________________________________________________ > > Busca desde cualquier página Web con una protección excepcional. Consigue la Barra de herramientas de Windows Live hoy mismo GRATUITAMENTE. > > http://toolbar.live.com_______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > _________________________________________________________________ > Llama a tus amigos de PC a PC: ¡Es GRATIS! > http://get.live.com/messenger/overview_____________________________________… > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers _________________________________________________________________ Consigue el nuevo Windows Live Messenger http://get.live.com/messenger/overview

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[Users] openser.org outage
by Daniel-Constantin Mierla 09 Oct '06

09 Oct '06

Hello, just to inform you, today the collocation site hosting the OpenSER site experienced lot of outages (http://www.hetzner-status.de/). We apologizes for the situation and we kindly ask you to re-summit any message which didn't get it to the mailing list. We look forward to implement a mirroring system to be able to handle smoothly such situations. Cheers, Daniel

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[Users] 404 not found retransmission
by Cseke Tamas 09 Oct '06

09 Oct '06

Hi, I have a retransmission problem, i recive an ack for an 404 reply, however opener 1.1.0 retransmit it: U 213.xxx.xxx.92:5060 -> 213.xxx.xxx.32:5060 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 192.168.1.106;rport=5060;received=2 17.xxx.xxx.22;branch=z9hG4bKf0dec68a3f9e217a..From: <sip:241@callshop.peakte lecom.me.uk>;tag=25dde6813cae8747..To: <sip:1@callshop.>;t ag=D3DB214-25D2..Date: Mon, 09 Oct 2006 11:37:18 GMT..Call-ID: 3c47a6403890 c2f2@192.168.1.106..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 42170 INVITE.. Allow-Events: telephone-event..Content-Length: 0.... # U 213.xxx.xxx.32:5060 -> 213.xxx.xxx.92:5060 ACK sip:1@callshop. SIP/2.0..Via: SIP/2.0/UDP 192.168.1.10 6;branch=z9hG4bKf0dec68a3f9e217a..From: <sip:241@callshop. >;tag=25dde6813cae8747..To: <sip:1@callshop.>;tag=D3DB214- 25D2..Contact: <sip:241@192.168.1.106>..Authorization: Digest username="241 ", realm="callshop.", algorithm=MD5, uri="sip:1@callshop ", nonce="452a37e35171ad890ccd763950d954c45090f992", respon se="f68ecea0d5196e8e88abeac6a68"..Call-ID: 3c47a6403890c2f2(a)192.168.1. 106..CSeq: 42170 ACK..User-Agent: Grandstream BT110 1.0.8.23..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Cont ent-Length: 0.... # U 213.xxx.xxx.92:5060 -> 213.xxx.xxx.32:5060 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 192.168.1.106;rport=5060;received=2 17.xxx.xxx.22;branch=z9hG4bKf0dec68a3f9e217a..From: <sip:241@callshop.peakte lecom.me.uk>;tag=25dde6813cae8747..To: <sip:1@callshop.>;t ag=D3DB214-25D2..Date: Mon, 09 Oct 2006 11:37:18 GMT..Call-ID: 3c47a6403890 c2f2@192.168.1.106..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 42170 INVITE.. Allow-Events: telephone-event..Content-Length: 0.... # U 213.xxx.xxx.92:5060 -> 213.xxx.xxx.32:5060 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 192.168.1.106;rport=5060;received=2 17.xxx.xxx.22;branch=z9hG4bKf0dec68a3f9e217a..From: <sip:241@callshop.peakte lecom.me.uk>;tag=25dde6813cae8747..To: <sip:1@callshop.>;t ag=D3DB214-25D2..Date: Mon, 09 Oct 2006 11:37:18 GMT..Call-ID: 3c47a6403890 c2f2@192.168.1.106..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 42170 INVITE.. Allow-Events: telephone-event..Content-Length: 0.... 0(15109) DEBUG: to body [<sip:1@callshop>] 0(15109) get_hdr_field: cseq <CSeq>: <42170> <INVITE> 0(15109) parse_headers: flags=8 0(15109) DEBUG: t_reply_matching: hash 39491 label 455964109 branch 2 0(15109) DEBUG:tm:REF_UNSAFE: after is 1 0(15109) DEBUG: t_reply_matching: reply matched (T=0xb3df7618)! 0(15109) DEBUG: t_check: msg id=35 global id=35 T end=0xb3df7618 0(15109) DEBUG:tm:reply_received: org. status uas=100, uac[2]=100 local=0 is_in vite=1) 0(15109) parse_headers: flags=8 0(15109) DEBUG:tm:build_local: using FROM=<From: <sip:241@callshop me.uk>;tag=25dde6813cae8747 >, TO=<To: <sip:1@callshop>;tag=D3DB214-25D2 >, CSEQ_N=<CSeq: 42170> 0(15109) parse_headers: flags=80 0(15109) DEBUG: get_hdr_body : content_length=0 0(15109) found end of header 0(15109) DEBUG:tm:t_should_relay_response: T_code=100, new_code=404 0(15109) DEBUG:tm:t_pick_branch: picked branch 2, code 404 0(15109) DEBUG:tm:relay_reply: branch=2, save=0, relay=2 0(15109) old size: 482, new size: 427 0(15109) build_res_from_sip_res: copied size: orig:83, new: 28, rest: 399 msg= SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.106;rport=5060;received=217.xxx.xxx.22;branch=z9hG4bKf 0dec68a3f9e217a From: <sip:241@callshop>;tag=25dde6813cae8747 To: <sip:1@callshop>;tag=D3DB214-25D2 Date: Mon, 09 Oct 2006 11:37:18 GMT Call-ID: 3c47a6403890c2f2(a)192.168.1.106 Server: Cisco-SIPGateway/IOS-12.x CSeq: 42170 INVITE Allow-Events: telephone-event Content-Length: 0 0(15109) DEBUG: add_to_tail_of_timer[4]: 0xb3df76c8 0(15109) DEBUG: add_to_tail_of_timer[0]: 0xb3df76d8 0(15109) DEBUG:tm:relay_reply: sent buf=0x812e928: SIP/2.0 4..., shmem=0xb3df8f 50: SIP/2.0 4 0(15109) DEBUG: cleanup_uac_timers: RETR/FR timers reset 0(15109) DEBUG:tm:UNREF_UNSAFE: after is 0 0(15109) DEBUG:destroy_avp_list: destroying list (nil) 0(15109) receive_msg: cleaning up 0(15109) SIP Request: 0(15109) method: <ACK> 0(15109) uri: <sip:1@callshop> 0(15109) version: <SIP/2.0> 0(15109) parse_headers: flags=2 0(15109) Found param type 232, <branch> = <z9hG4bKf0dec68a3f9e217a>; state=16 0(15109) end of header reached, state=5 0(15109) parse_headers: Via found, flags=2 0(15109) parse_headers: this is the first via 0(15109) After parse_msg... 0(15109) preparing to run routing scripts... 0(15109) parse_headers: flags=8 0(15109) DEBUG: add_param: tag=D3DB214-25D2 0(15109) DEBUG:parse_to:end of header reached, state=29 0(15109) DBUG:parse_to: display={}, ruri={sip:1@callshop} 0(15109) DEBUG: get_hdr_field: <To> [53]; uri=[sip:1@callshop ] 0(15109) DEBUG: to body [<sip:1@callshop>] 0(15109) parse_headers: flags=100 0(15109) get_hdr_field: cseq <CSeq>: <42170> <ACK> 0(15109) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(15109) DEBUG: add_param: tag=25dde6813cae8747 0(15109) DEBUG:parse_to:end of header reached, state=29 0(15109) DBUG:parse_to: display={}, ruri={sip:241@callshop} 0(15109) DEBUG: t_check: msg id=36 global id=35 T start=0xffffffff 0(15109) parse_headers: flags=ffffffffffffffff 0(15109) DEBUG: get_hdr_body : content_length=0 0(15109) found end of header 0(15109) parse_headers: flags=78 0(15109) t_lookup_request: start searching: hash=39491, isACK=1 0(15109) DEBUG: RFC3261 transaction matched, tid=f0dec68a3f9e217a 0(15109) DEBUG:tm:REF_UNSAFE: after is 1 0(15109) DEBUG: t_lookup_request: transaction found (T=0xb3df7618) 0(15109) DEBUG: t_check: msg id=36 global id=36 T end=0xb3df7618 0(15109) DEBUG:t_check_status: checked status is <404> 0(15109) parse_headers: flags=58 0(15109) parse_headers: flags=200 0(15109) find_first_route: No Route headers found 0(15109) loose_route: There is no Route HF 0(15109) grep_sock_info - checking if host==us: 26==12 && [callshop. m.me.uk] == [213.166.8.92] 0(15109) grep_sock_info - checking if port 5060 matches port 5060 0(15109) grep_sock_info - checking if host==us: 26==12 && [callshop. m.me.uk] == [213.166.8.92] 0(15109) grep_sock_info - checking if port 5060 matches port 5060 0(15109) DEBUG: t_newtran: msg id=36 , global msg id=36 , T on entrance=0xb3df7 618 * 0(15109) ERROR: t_newtran: transaction already in process 0xb3df7618* 0(15109) DEBUG:tm:UNREF_UNSAFE: after is 0 0(15109) DEBUG:destroy_avp_list: destroying list (nil) 0(15109) receive_msg: cleaning up 1(15110) DEBUG: timer routine:1,tl=0xb3df78ec next=0xb3df7818 1(15110) DEBUG: timer routine:1,tl=0xb3df7818 next=0xb3df7744 1(15110) DEBUG: timer routine:1,tl=0xb3df7744 next=(nil) 1(15110) DEBUG: timer routine:4,tl=0xb3df76c8 next=0xb3df78dc 1(15110) DEBUG: retransmission_handler : reply resending (t=0xb3df7618, SIP/2.0 4 ... ) 1(15110) DEBUG:tm:t_retransmit_reply: buf=0xb7db8c60: SIP/2.0 4..., shmem=0xb3d f8f50: SIP/2.0 4 1(15110) DEBUG: add_to_tail_of_timer[5]: 0xb3df76c8 1(15110) DEBUG: retransmission_handler : done 1(15110) DEBUG: timer routine:4,tl=0xb3df78dc next=0xb3df7808 1(15110) DEBUG: timer routine:4,tl=0xb3df7808 next=0xb3df7734 1(15110) DEBUG: timer routine:4,tl=0xb3df7734 next=(nil) 1(15110) DEBUG: timer routine:5,tl=0xb3df76c8 next=(nil) 1(15110) DEBUG: retransmission_handler : reply resending (t=0xb3df7618, SIP/2.0 4 ... ) 1(15110) DEBUG:tm:t_retransmit_reply: buf=0xb7db8c60: SIP/2.0 4..., shmem=0xb3d f8f50: SIP/2.0 4 1(15110) DEBUG: add_to_tail_of_timer[6]: 0xb3df76c8 1(15110) DEBUG: retransmission_handler : done 1(15110) DEBUG: timer routine:6,tl=0xb3df76c8 next=(nil) 1(15110) DEBUG: retransmission_handler : reply resending (t=0xb3df7618, SIP/2.0 4 ... ) 1(15110) DEBUG:tm:t_retransmit_reply: buf=0xb7db8c60: SIP/2.0 4..., shmem=0xb3d f8f50: SIP/2.0 4 1(15110) DEBUG: add_to_tail_of_timer[7]: 0xb3df76c8 1(15110) DEBUG: retransmission_handler : done 1(15110) DEBUG: timer routine:7,tl=0xb3df76c8 next=(nil) 1(15110) DEBUG: retransmission_handler : reply resending (t=0xb3df7618, SIP/2.0 4 ... ) 1(15110) DEBUG:tm:t_retransmit_reply: buf=0xb7db8c60: SIP/2.0 4..., shmem=0xb3d f8f50: SIP/2.0 4 1(15110) DEBUG: add_to_tail_of_timer[7]: 0xb3df76c8 1(15110) DEBUG: retransmission_handler : done 1(15110) INFO: signal 2 received 1(15110) Memory status (pkg): 1(15110) fm_status (0x811fd80): 1(15110) heap size= 1048576 1(15110) used= 34976, used+overhead=55056, free=993520 1(15110) max used (+overhead)= 55880 1(15110) dumping free list: 1(15110) hash = 7 fragments no.: 1, unused: 0 bucket size: 56 - 56 (first 56) 1(15110) hash = 16 fragments no.: 6, unused: 0 bucket size: 128 - 128 (first 128) 1(15110) hash = 255 fragments no.: 1, unused: 0 bucket size: 2040 - 2040 (first 2040) 1(15110) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 990656) 1(15110) TOTAL: 9 free fragments = 993520 free bytes 1(15110) ----------------------------- thanks any help, Tamas

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RE: [Serusers] DNS and IP
by Rosa De Santis 09 Oct '06

09 Oct '06

Hi, Weiter. Thanks for your answer. I already have "alias=the.domain.name" in my config, but its not working :( Any ideas??? Thanks again. Rosa. ----------------------------------------------- > Date: Sun, 8 Oct 2006 12:02:24 +0200 > From: bp4mls(a)googlemail.com > To: rosadesantis(a)hotmail.com > Subject: Re: [Serusers] DNS and IP > CC: serusers(a)iptel.org > > Make sure the box where SER is running can also resolve the name to it's own IP. > You can also use "alias=the.domain.name" in you config. > On 10/6/06, Rosa De Santis <rosadesantis(a)hotmail.com> wrote: > Hello all. > Please, I have a problem using DNS or IP on the client side. > When I use IP all calls are fine, but if I use the DNS of the server that point to the same IP some calls can't be done. > Is there any place to configure this on ser.cfg ??? > Please help. > Thanks. > _________________________________________________________________ > Busca desde cualquier página Web con una protección excepcional. Consigue la Barra de herramientas de Windows Live hoy mismo GRATUITAMENTE. > http://toolbar.live.com_______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers _________________________________________________________________ Llama a tus amigos de PC a PC: ¡Es GRATIS! http://get.live.com/messenger/overview

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Re: [Users] Multi Call-Legs accounting configuration
by Ricardo Carvalho 09 Oct '06

09 Oct '06

I've done like Rafael told in the e-mail shown bellow for 1.1.x version, although I was expecting to see in acc table to be written A B and C contacts of forwarded calls (when A calls B and B forwards to C), but there is never written B contact, the one who should be billed for that call... the entries in my acc table for forwarded calls are like this: | sip_from | sip_to | i_uri | o_uri | from_uri | to_uri | src_leg | dst_leg | |---------------------------------------------------------------------------| | A | C | C | C | A | C | A | C | As seen, my acc table doesn't show B contact!! How should I edit my openser.cfg file so that B contact be shown in acc table for forwarded calls? Thanks, Ricardo. Rafael J. Risco G.V. wrote: > hi ricardo, > for 1.1.x version, try this: > modparam("acc", "multi_leg_enabled", 1) > modparam("acc", "src_leg_avp_id", 110) > modparam("acc", "dst_leg_avp_id", 111) > ... > route { > > ## populate the avps (for Multi call leg accounting): > if (!method=="REGISTER" && !method=="NOTIFY") { > avp_write("$from/username", "$avp(i:110)"); > avp_write("$ruri/username", "$avp(i:111)"); > }; > ..... > > for 1.2 or cvs : > # for mysql-based accouting, use the names of the columns > modparam("acc", "multi_leg_info", > "leg_src=$avp(i:110);leg_dst=$avp(i:111)") > > Do not forget to add src_leg and dst_leg columns in acc table > (leg_src, leg_dst for 1.2) > > rafael > > > On 10/6/06, Ricardo Carvalho <rcarvalho(a)iric.up.pt> wrote: >> I want to use Multi Call-Legs accounting to properly account forwarded >> calls made "in the phones". For example user A calls user B which >> forwards the call to user C. There is only one SIP call but with 2 legs >> ( A to B and B to C). Accounting the legs of a call is required for >> proper billing of the calls. >> >> As can be seen in acc module documentation, I've added the following 3 >> lines to my openser.cfg script: >> >> modparam("acc", "multi_leg_enabled", 1) >> modparam("acc", "src_leg_avp_id", 110) >> modparam("acc", "dst_leg_avp_id", 110) >> >> Although, after testing some forwarded calls, nothing has been written >> to src_leg or dst_leg columns of acc table!! >> What have I done wrong? How may I implement this feature correctly? >> >> Regards, >> Ricardo. >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >

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[Users] openser.org down?
by Brandon Price 09 Oct '06

09 Oct '06

Can't access it from NYC, or Sanfran.

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