sr-users August 2004

sr-users@lists.kamailio.org

181 participants
424 discussions

RE: [Serusers] Asterisk inside a NAT, client inside ANOTHER NAT

by C.K

Hi Zeus, Yes, my SER is running behind netscreen firewall + NAT and has a public IP address (202.129.171.223). I have enable all outoing port and limited incoming port (5004 and 5005). Which is running in the company I work for. The UAs is running at home which is behind my NETGEAR router, if I do a port fowarding on my NETGEAR router to forward 5004 to my grandstream (129.168.0.3) then I can hear the voice. But I can't hear the echo. I have try to call home from the office (X-lite to grandstream), but only one direction can hear, the UA at home can hear the voice but UA in my office can't the the voice. Like you said, properly is the voice is block by the NAT. But I give you another sinareo, if I turn on the port forwarding, I call talk to another UA which has public IP address (connected to another ADSL line). Please advice, C.K --- Zeus Ng <zeus.ng(a)isquare.com.au> wrote: > C.K., > > I notice that your SER is behind a NAT (I assume), > not just a firewall. From > my experience, SER will only function properly if it > has a public IP. PAT or > NAT for SER will not work. I don't know how you get > around it. Are you > running it in your lab? Forgive me but I would say > that it will not work in > a real world environment. > > Regards, > > Zeus > > > > -----Original Message----- > > From: C.K [mailto:ckng128@yahoo.com] > > Sent: Saturday, 21 August 2004 9:17 AM > > To: Zeus Ng > > Subject: RE: [Serusers] Asterisk inside a NAT, > client inside > > ANOTHER NAT > > > > > > Hi Zeus, Attach is my ngrep.log, please give me > some > > guideline. > > > > Many thanks > > > >

19 years, 10 months

RE:[Serusers] phones both in a LAN cannot get through when ser ison Internet

by gg

Klaus Darilion Thank you vary much! You see . The OS is Red Hat Linux 7.2. So I have to install ser-0.8.10. Can it be upgraded on this OS? If it can , how to upgrade it and which version should I choose? Can I install ser-0.8.14 or other version directly without doing anything before I do this ? Thank you once again! 　　　　　　　　gg 　　　　　　　　gdut200104024532(a)163.com 　　　　　　　　　　2004-08-23

19 years, 10 months

RE: [Serusers] NAT ping and consumer router

by Dave Bath

Hey there, At home I have Netgear ADSL router (1 x ADSL, 4 x eth) and have not had any problems with NAT devices dropping off. Mainly use Zyxel Wireless 2000 handsets. Dave -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Jesus Rodriguez Sent: 22 August 2004 19:59 To: Richard Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] NAT ping and consumer router On Sun, 22 Aug 2004, Richard wrote: > Does anyone know and use a consumer router which works with NAT ping? > > I did some research recently and can't find any. Basically if a phone is > behind NAT, we need to keep the NAT binding in the router active even if > there is no activity from the phone. NAT ping from ser (either rtpproxy or > mediaproxy) tries to ping the phone with an empty SIP packet. However this > inbound UDP packet can't always keep the NAT binding active because some NAT > firewalls ONLY refresh the timer based on outbound packets. So the result is > that the binding expires after a certain time even if NAT ping is enabled. > > For example, Dlink falls in this category. It appears the timeout is 3 > minutes and NAT ping won't make it active. > > If anyone has a good experience with any consumer router, can you please let > share with us? Use an UA that supports it (Sipura or Cisco for example). Saludos JesusR. ------------------------------- Jesus Rodriguez VozTelecom Sistemas, S.L. jesusr(a)voztele.com http://www.voztele.com Tel. 902360305 ------------------------------- _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

19 years, 10 months

[Serusers] ApplicationAgent with ser

by ser die

hello friends, i have seen about application agent in the iptel.org ,but i could not get any hint where can we download that and how can we incorparate that into ser. Is it a module or else different entity. with regards serdiehard. __________________________________ Do you Yahoo!? Take Yahoo! Mail with you! Get it on your mobile phone. http://mobile.yahoo.com/maildemo

19 years, 10 months

[Serusers] invite timer & calls to PSTN

by O'Shaughnessy Evans

Hello, all. I have a problem I'm trying to work out: when a stateful call goes out from the proxy to the PSTN, the fr_inv_timer will eventually trigger a failure. The thing is, I don't want calls going outside my SIP proxy to have any timer associated with them. I need it for calls to local users so they can fall back to voicemail, but I don't need it for anything else. E.g. if one of my SIP users calls out to a PSTN number and, for whatever reason, wants that call to ring 300 times, the timer in the SIP proxy shouldn't inhibit that ability. I can get around this by using forward(), but if the SIP user is coming from a NAT'd device, I have to use a t_relay() instead and then I'm stuck with the timer running. Is there a way around this? Thanks for your help. -- = o'shaughnessy evans = = sys admin @ (aloha|kona|myworld).net =

19 years, 10 months

Re: [Serusers] phones both in a LAN cannot get through when ser ison Internet

by mean

19 years, 10 months

Re: [Serusers] phones both in a LAN cannot get through when ser ison Internet

by g

Klaus Darilion Thank you vary much ! You see. The OS is Red Hat Linux 7.2. So I have to install ser-0.8.10. can it be upgraded on this OS? If it can, how to upgrade it and which version should I choose? Can I install ser-0.8.14 or other version directly without doing anything before I do this? thank you once again ! >You have a problem with the NAT traversal! > >Install a newer ser (0.8.14) and use the nathelper module to rewrite the >SIP messages. Search the archive for "NAT". > >regards, >klaus > >gg wrote: >> Hi everyone ! >> I have installed ser-0.8.10 and ser-mysql-0.8.10 on Red Hat Linux 7.2. After I >> configured the script (even did not add anything). The SIP server can worked well in LAN . >> there are Two telephones both in the LAN can get though. But when I put ser on Internet >> and the two phones still in a LAN which links to Internet, they cann't get through. >> I have no idea what I can do . >> anyone help me please! >> Thank you! >> >> 　　　　　　　　gg >> 　　　　　　　　gdut200104024532(a)163.com >> 　　　　　　　　　　2004-08-20 >> >> >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)iptel.org >> http://mail.iptel.org/mailman/listinfo/serusers >> >> > = = = = = = = = = = = = = = = = = = = = 　　　　　　　　致礼！　　　　　　　　g 　　　　　　　　gdut200104024532(a)163.com 　　　　　　　　　　2004-08-22

19 years, 10 months

[Serusers] fifo + cpl

by ser die

hello friends , when i try this command in the command prompt serctl fifo LOAD_CPL ::214103@xxx.xxx.xxx.in ::cpl214103.xml in debug i have seen this *********************************************** 9(8689) DEBUG:cpl_load: user@host=::214103@xxx.xxx.xxx.in 9(8689) ERROR:cpl-c:cpl_load: unable to read cpl_file name from FIFO command 9(8689) ERROR: fifo_server: command (LOAD_CPL) processing failed *************************************************** so can any body say the correct way with regards serdiehard __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage! http://promotions.yahoo.com/new_mail

19 years, 11 months

[Serusers] Voicemail and sems help

by Rafael J. Risco G.V

Hi , I am trying to add Voicemail services... the problem is that when the call is missed (busy, not answered or not connected) I can see some debug in ser´s second instance for users in "voicemail group" but the call is closed and a busy tone return to the caller... you can see my 3 config files in the following url and details from both ser debugs, please someone send me some advice. Config files in: http://rrisco.pub.millicom.com.pe/ser_config_files.txt - ser.cfg for Main SER running on 5060 - voicemail.cfg for second instance on port 5090 - sems.conf for Media Server (sems) DEBUG from 1st instamce (port 5060) --------------------------------------- 4(31737) DEBUG: relay_reply: branch=1, save=0, relay=1 4(31737) old size: 669, new size: 607 4(31737) build_res_from_sip_res: copied size: orig:108, new: 46, rest: 561 msg= SIP/2.0 404 not reponsible for host in r-uri Via: SIP/2.0/UDP 10.0.0.236:5060;branch=z9hG4bKa200bcf5a4114 From: <sip:6603000@call.millicom.com.pe>;tag=a200bcf5a4 To: <sip:6604000@call.millicom.com.pe>;tag=3749ec7003921b5c92fe06c5dc660395.5093 Call-ID: a2609600-48ac-bcf1-81f5-0002a40055b2(a)10.0.0.236 CSeq: 114 INVITE Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 200.110.2.131:5090 "Noisy feedback tells: pid=31707 req_src_ip=200.110.2.131 req_src_port=5060 in_uri=sip:6604000@call.millicom.com.pe:5090 out_uri=sip:6604000@call.millicom.com.pe:5090 via_cnt==2" 4(31737) DEBUG: reply relayed. buf=0x80dec48: SIP/2.0 4..., shmem=0x422be8a8: SIP/2.0 4 4(31737) DBG: callback type 7, id 1 entered 4(31737) DEBUG: cleanup_uacs: RETR/FR timers reset 4(31737) DEBUG: add_to_tail_of_timer[4]: 0x422c287c 4(31737) DEBUG: add_to_tail_of_timer[0]: 0x422c2890 4(31737) DEBUG:destroy_avp_list: destroing list (nil) 4(31737) receive_msg: cleaning up [root@gkproxy01 admin]# [root@gkproxy01 admin]# 2(31735) SIP Request: 2(31735) method: <ACK> 2(31735) uri: <sip:6604000@call.millicom.com.pe> 2(31735) version: <SIP/2.0> 2(31735) parse_headers: flags=1 2(31735) Found param type 232, <branch> = <z9hG4bKa200bcf5a4114>; state=16 2(31735) end of header reached, state=5 2(31735) parse_headers: Via found, flags=1 2(31735) parse_headers: this is the first via 2(31735) After parse_msg... 2(31735) preparing to run routing scripts... 2(31735) DEBUG : sl_filter_ACK: to late to be a local ACK! 2(31735) DEBUG : is_maxfwd_present: searching for max_forwards header 2(31735) parse_headers: flags=128 2(31735) DEBUG: add_param: tag=3749ec7003921b5c92fe06c5dc660395.5093 2(31735) end of header reached, state=29 2(31735) DEBUG: get_hdr_field: <To> [78]; uri=[sip:6604000@call.millicom.com.pe] 2(31735) DEBUG: to body [<sip:6604000@call.millicom.com.pe>] 2(31735) get_hdr_field: cseq <CSeq>: <114> <ACK> 2(31735) DEBUG: get_hdr_body : content_length=0 2(31735) DEBUG: is_maxfwd_present: value = 70 2(31735) DEBUG: add_param: tag=a200bcf5a4 2(31735) end of header reached, state=29 2(31735) parse_headers: flags=256 2(31735) found end of header 2(31735) find_first_route(): No Route headers found 2(31735) loose_route(): There is no Route HF 2(31735) check_self - checking if host==us: 20==13 && [call.millicom.com.pe] == [200.110.2.131] 2(31735) check_self - checking if port 5060 matches port 5060 2(31735) check_self - checking if host==us: 20==9 && [call.millicom.com.pe] == [127.0.0.1] 2(31735) check_self - checking if port 5060 matches port 5060 2(31735) query="select grp from grp where username='6604000' AND grp='voicemail'" 2(31735) is_user_in(): User is in group 'voicemail' 2(31735) rwrite(): Rewriting Request-URI with 'sip:6604000@200.110.6.58' 2(31735) DEBUG: t_addifnew: msg id=11467 , global msg id=11434 , T on entrance=0xffffffff 2(31735) parse_headers: flags=-1 2(31735) parse_headers: flags=60 2(31735) t_lookup_request: start searching: hash=31284, isACK=1 2(31735) DEBUG: RFC3261 transaction matched, tid=a200bcf5a4114 2(31735) DEBUG: t_lookup_request: transaction found (T=0x422c27c8) 2(31735) DEBUG: cleanup_uacs: RETR/FR timers reset 2(31735) DEBUG: add_to_tail_of_timer[2]: 0x422c2810 2(31735) DEBUG:destroy_avp_list: destroing list (nil) 2(31735) receive_msg: cleaning up DEBUG FROM 2nd INSTANCE (PORT 5090) -------------------------------------------------- ] 7(31709) get_hdr_field: cseq <CSeq>: <113> <INVITE> 7(31709) DEBUG: get_hdr_body : content_length=180 7(31709) DEBUG: is_maxfwd_present: value = 69 7(31709) check_self - checking if host==us: 20==9 && [call.millicom.com.pe] == [127.0.0.1] 7(31709) check_self - checking if port 5090 matches port 5090 7(31709) check_self - checking if host==us: 20==13 && [call.millicom.com.pe] == [200.110.2.131] 7(31709) check_self - checking if port 5090 matches port 5090 7(31709) check_self: host != me 7(31709) parse_headers: flags=-1 7(31709) found end of header 7(31709) check_via_address(200.110.2.131, 200.110.2.131, 0) 7(31709) DEBUG:destroy_avp_list: destroing list (nil) 7(31709) receive_msg: cleaning up 5(31705) SIP Request: 5(31705) method: <ACK> 5(31705) uri: <sip:6605454@call.millicom.com.pe:5090> 5(31705) version: <SIP/2.0> 5(31705) parse_headers: flags=1 5(31705) Found param type 232, <branch> = <z9hG4bKd517.2ea1e793.0>; state=16 5(31705) end of header reached, state=5 5(31705) parse_headers: Via found, flags=1 5(31705) parse_headers: this is the first via 5(31705) After parse_msg... 5(31705) preparing to run routing scripts... 5(31705) parse_headers: flags=4 5(31705) DEBUG: add_param: tag=3749ec7003921b5c92fe06c5dc660395.01b8 5(31705) end of header reached, state=29 5(31705) DEBUG: get_hdr_field: <To> [78]; uri=[sip:6605454@call.millicom.com.pe] 5(31705) DEBUG: to body [<sip:6605454@call.millicom.com.pe>] 5(31705) DEBUG: sl_filter_ACK : local ACK found -> dropping it! 5(31705) DEBUG:destroy_avp_list: destroing list (nil) 5(31705) receive_msg: cleaning up 6(31707) SIP Request: 6(31707) method: <INVITE> 6(31707) uri: <sip:6604000@call.millicom.com.pe:5090> 6(31707) version: <SIP/2.0> 6(31707) parse_headers: flags=1 6(31707) Found param type 232, <branch> = <z9hG4bK43a7.d99b0dc7.1>; state=16 6(31707) end of header reached, state=5 6(31707) parse_headers: Via found, flags=1 6(31707) parse_headers: this is the first via 6(31707) After parse_msg... 6(31707) preparing to run routing scripts... 6(31707) DEBUG : is_maxfwd_present: searching for max_forwards header 6(31707) parse_headers: flags=128 6(31707) Found param type 232, <branch> = <z9hG4bKa200bcf5a4114>; state=16 6(31707) end of header reached, state=5 6(31707) parse_headers: Via found, flags=128 6(31707) parse_headers: this is the second via 6(31707) end of header reached, state=9 6(31707) DEBUG: get_hdr_field: <To> [36]; uri=[sip:6604000@call.millicom.com.pe] 6(31707) DEBUG: to body [<sip:6604000@call.millicom.com.pe> ] 6(31707) get_hdr_field: cseq <CSeq>: <114> <INVITE> 6(31707) DEBUG: get_hdr_body : content_length=180 6(31707) DEBUG: is_maxfwd_present: value = 69 6(31707) check_self - checking if host==us: 20==9 && [call.millicom.com.pe] == [127.0.0.1] 6(31707) check_self - checking if port 5090 matches port 5090 6(31707) check_self - checking if host==us: 20==13 && [call.millicom.com.pe] == [200.110.2.131] 6(31707) check_self - checking if port 5090 matches port 5090 6(31707) check_self: host != me 6(31707) parse_headers: flags=-1 6(31707) found end of header 6(31707) check_via_address(200.110.2.131, 200.110.2.131, 0) 6(31707) DEBUG:destroy_avp_list: destroing list (nil) 6(31707) receive_msg: cleaning up 8(31711) SIP Request: 8(31711) method: <ACK> 8(31711) uri: <sip:6604000@call.millicom.com.pe:5090> 8(31711) version: <SIP/2.0> 8(31711) parse_headers: flags=1 8(31711) Found param type 232, <branch> = <z9hG4bK43a7.d99b0dc7.1>; state=16 8(31711) end of header reached, state=5 8(31711) parse_headers: Via found, flags=1 8(31711) parse_headers: this is the first via 8(31711) After parse_msg... 8(31711) preparing to run routing scripts... 8(31711) parse_headers: flags=4 8(31711) DEBUG: add_param: tag=3749ec7003921b5c92fe06c5dc660395.5093 8(31711) end of header reached, state=29 8(31711) DEBUG: get_hdr_field: <To> [78]; uri=[sip:6604000@call.millicom.com.pe] 8(31711) DEBUG: to body [<sip:6604000@call.millicom.com.pe>] 8(31711) DEBUG: sl_filter_ACK : local ACK found -> dropping it! 8(31711) DEBUG:destroy_avp_list: destroing list (nil) 8(31711) receive_msg: cleaning up Thank You Rafael --------------------------------- Do You Yahoo!? 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19 years, 11 months

[Serusers] Ports and SER.

by C.K

Hello all, Please guide me on ports needed to be open for the incoming audio to SER, the environment is as follow: SER <-> Netscreen FIREwall <-> INTERNET <-> NAT <-> UA1 and UA2. I could not pass the audio from UA1 to UA2. Many thanks.

19 years, 11 months

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sr-users August 2004