sr-users December 2003

sr-users@lists.kamailio.org

108 participants
257 discussions

[Serusers] RE: Serusers Digest, Vol 8, Issue 16

by Darren Sessions

Situation: Endpoint is called Party (Sipura SPA2000) Calling Party is routed through from a TDM call and VoIP'd by Cisco 3640 router then sent to SER. Endpoint receives call - everything works. When the Sipura hangs up the call, it sends a BYE to ser - but ser does not send the BYE to the router to disconnect the TDM channel. All other situations result in a completed call with channels released on the router just fine. After further inspection - we noticed that when the Sipura hangs up and send the BYE to SER, SER responds with this: SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 11.11.111.155:5060;branch=z9hG4bK-b76fe29f From: <sip:8646783182@22.22.222.154>;tag=bc5a80577e3d5642 To: <sip:8644679887@33.33.333.157>;tag=322BCA98-25A3 Call-ID: F429FDCE-30B011D8-BA0EDB9C-52307829(a)33.33.333.157 CSeq: 101 BYE Contact: <sip:8644679887@44.44.444.148> Server: Sip EXpress router (0.8.12 (i386/linux)) Content-Length: 0 Warning: 392 44.44.444.148:5060 "Noisy feedback tells: pid=20033 req_src_ip=22.22.222.154 req_src_port=5060 in_uri=sip:8644679887@44.44.444.148:5060 out_uri=sip:8644679887@44.44.444.148:5060 via_cnt==0" Any ideas? Thanks, - Darren

20 years, 11 months

[Serusers] Voicemail errors and other issues

by Gavin Bensom

Hello All, First, I created subtle problems in SER by having an alias entry for the host ip in the ser.cfg file. I don't recommend doing the following [where 10.10.10.49 is the ip of the host where SER is running]. alias=10.10.10.49 Next, I've had similar problems to a user who posted recently about SEMS being unstable. Occassionally it just stops playing the greeting, however, even if the greeting doesn't play, SEMS will still record audio. Overall, I've had a lot of frustration with SEMS. Attached is my ser.cfg file and some error messages from syslog. Here is a scenario which is causing problems with voicemail. ------------------------ User sip:info@mydomain.com is not logged in anywhere, however that user has calls being forwarded to sip:mvestal@mydomain.com and sip:esavelle@mydomain.com. The location table shows these entries correctly. In addition, mvestal and esavelle are also in the location table at specific ip addresses. When a call come in for sip:info@mydomain.com, it gets forked propery. However, if no one answers the call before the fr_inv_timer expires, SER tries to hand the call off to VMSER [second instance of SER which is running on port 5090 and is handling voicemail] but the error messages in the attached logs show up. The calling party typically hears ringing continue after SER had timed out, and then gets a busy tone a few seconds later. I'm also going to try Asterisk. There is a specific feature of Asterisk which I think would benefit SER/SEMS a lot. We will set up Asterisk so that we do and append_branch(); in SER as soon as a call is initiated to an internal extension. Asterisk can then be set to wait a certain amount of time before it answers the call. This way, the fr_inv_time feature of SER isn't necessary for getting calls into voicemail [SEMS or Asterisk.] Also, it is very nice to be able to use a phone to check voicemail. No all users are necessarily sitting on their email inboxes all day long. Any help on the bug would be greatly appreciated. Noah --------------------------------- Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing Dec 16 16:14:02 jiffypop /usr/local/sbin/ser[24298]: ACC: call missed: from=<sip:4083709675@10.10.10.5>;tag=4979781C-1816, i-uri= sip:esavelle@mydomain.com, method=INVITE, o-uri=sip:esavelle@10.10.10.167:5060, code=408 Request Timeout Dec 16 16:14:02 jiffypop /usr/local/sbin/ser[24298]: failure_route[4] Dec 16 16:14:02 jiffypop /usr/local/sbin/ser[24298]: route[4]:call failed: forward to voicemail Dec 16 16:14:02 jiffypop /usr/local/sbin/ser[24298]: ACC: call missed: from=<sip:4083709675@10.10.10.5>;tag=4979781C-1816, i-uri= sip:mvestal@mydomain.com, method=INVITE, o-uri=sip:mvestal@10.10.10.98:5060, code=408 Request Timeout Dec 16 16:14:02 jiffypop /usr/local/sbin/ser[24298]: failure_route[4] Dec 16 16:14:02 jiffypop /usr/local/sbin/ser[24242]: ERROR: t_should_relay: status rewrite by UAS: stored: 408, received: 487 Dec 16 16:14:02 jiffypop /usr/local/sbin/ser[24229]: ACC: request acknowledged: from=<sip:4083709675@10.10.10.5>;tag=4979781C-181 6, i-uri=sip:esavelle@mydomain.com, method=ACK, o-uri=sip:esavelle@10.10.10.167:5060, code=180 Dec 16 16:14:02 jiffypop /usr/local/sbin/ser[24298]: route[4]:call failed: forward to voicemail Dec 16 16:14:02 jiffypop /usr/local/sbin/ser[24298]: ACC: call missed: from=<sip:4083709675@10.10.10.5>;tag=4979781C-1816, i-uri= sip:6600@10.10.10.49:5060, method=INVITE, o-uri=sip:mvestal@mydomain.com, code=408 Request Timeout Dec 16 16:14:02 jiffypop /usr/local/sbin/vmser[29524]: t_newtran created Dec 16 16:14:02 jiffypop /usr/local/sbin/vmser[29524]: **************** vm start - begin ****************** Dec 16 16:14:02 jiffypop /usr/local/sbin/vmser[29523]: t_newtran created Dec 16 16:14:02 jiffypop /usr/local/sbin/vmser[29523]: **************** vm start - begin ****************** Dec 16 16:14:02 jiffypop /usr/local/sbin/vmser[29524]: **************** vm start - end ****************** Dec 16 16:14:02 jiffypop /usr/local/sbin/vmser[29523]: **************** vm start - end ****************** Dec 16 16:14:02 jiffypop Sems[19578]: Error: 482 Loop detected Dec 16 16:14:41 jiffypop /usr/local/sbin/ser[24240]: ACC: transaction answered: from="bkern" <sip:bkern@mydomain.com>;tag=9f5e9 214-b2f7-404c-a109-e20502c672c7, i-uri=sip:fchuang@mydomain.com, method=SUBSCRIBE, o-uri=sip:10.10.10.192:13584, code=200 Dec 16 16:14:49 jiffypop /usr/local/sbin/ser[24298]: ACC: call missed: from=<sip:4083709725@10.10.10.5>;tag=497A2F34-544, i-uri=s ip:mvestal@mydomain.com, method=INVITE, o-uri=sip:mvestal@10.10.10.98:5060, code=408 Request Timeout Dec 16 16:14:49 jiffypop /usr/local/sbin/ser[24298]: failure_route[4] Dec 16 16:14:49 jiffypop /usr/local/sbin/ser[24298]: route[4]:call failed: forward to voicemail Dec 16 16:14:49 jiffypop /usr/local/sbin/ser[24298]: ACC: call missed: from=<sip:4083709725@10.10.10.5>;tag=497A2F34-544, i-uri=s ip:esavelle@mydomain.com, method=INVITE, o-uri=sip:esavelle@10.10.10.167:5060, code=408 Request Timeout Dec 16 16:14:49 jiffypop /usr/local/sbin/ser[24298]: failure_route[4] Dec 16 16:14:49 jiffypop /usr/local/sbin/ser[24231]: ERROR: t_should_relay: status rewrite by UAS: stored: 408, received: 487 Dec 16 16:14:49 jiffypop /usr/local/sbin/ser[24229]: ACC: request acknowledged: from=<sip:4083709725@10.10.10.5>;tag=497A2F34-544 , i-uri=sip:mvestal@mydomain.com, method=ACK, o-uri=sip:mvestal@10.10.10.98:5060, code=180 Dec 16 16:14:49 jiffypop /usr/local/sbin/ser[24298]: route[4]:call failed: forward to voicemail Dec 16 16:14:49 jiffypop /usr/local/sbin/ser[24298]: ACC: call missed: from=<sip:4083709725@10.10.10.5>;tag=497A2F34-544, i-uri=s ip:6600@10.10.10.49:5060, method=INVITE, o-uri=sip:mvestal@mydomain.com, code=408 Request Timeout Dec 16 16:14:49 jiffypop /usr/local/sbin/vmser[29524]: t_newtran created Dec 16 16:14:49 jiffypop /usr/local/sbin/vmser[29524]: **************** vm start - begin ****************** Dec 16 16:14:49 jiffypop /usr/local/sbin/vmser[29523]: t_newtran created Dec 16 16:14:49 jiffypop /usr/local/sbin/vmser[29523]: **************** vm start - begin ****************** Dec 16 16:14:49 jiffypop /usr/local/sbin/vmser[29524]: **************** vm start - end ****************** Dec 16 16:14:49 jiffypop /usr/local/sbin/vmser[29523]: **************** vm start - end ****************** Dec 16 16:14:49 jiffypop Sems[19578]: Error: 482 Loop detected Dec 16 16:15:10 jiffypop /usr/local/sbin/ser[24298]: ACC: call missed: from=<sip:5107392224@10.10.10.5>;tag=497A8154-2635, i-uri= sip:6611@10.10.10.49:5060, method=INVITE, o-uri=sip:service@10.10.10.242:5060, code=408 Request Timeout Dec 16 16:15:10 jiffypop /usr/local/sbin/ser[24298]: failure_route[4] Dec 16 16:15:10 jiffypop /usr/local/sbin/ser[24298]: route[4]:call failed: forward to voicemail Dec 16 16:15:10 jiffypop /usr/local/sbin/ser[24233]: ERROR: t_should_relay: status rewrite by UAS: stored: 408, received: 487 Dec 16 16:15:10 jiffypop /usr/local/sbin/vmser[29524]: t_newtran created Dec 16 16:15:10 jiffypop /usr/local/sbin/vmser[29524]: **************** vm start - begin ****************** Dec 16 16:15:10 jiffypop /usr/local/sbin/vmser[29524]: **************** vm start - end ****************** Dec 16 16:15:18 jiffypop /usr/local/sbin/vmser[29523]: t_newtran created Dec 16 16:15:18 jiffypop /usr/local/sbin/vmser[29523]: **************** vm end - begin ****************** Dec 16 16:15:18 jiffypop /usr/local/sbin/vmser[29523]: **************** vm end - end ****************** Dec 16 16:15:28 jiffypop /usr/local/sbin/ser[24298]: ACC: call missed: from=<sip:4083709675@10.10.10.5>;tag=497AC670-18B, i-uri=s ip:mvestal@mydomain.com, method=INVITE, o-uri=sip:mvestal@10.10.10.98:5060, code=408 Request Timeout Dec 16 16:15:28 jiffypop /usr/local/sbin/ser[24298]: failure_route[4] Dec 16 16:15:28 jiffypop /usr/local/sbin/ser[24298]: route[4]:call failed: forward to voicemail Dec 16 16:15:28 jiffypop /usr/local/sbin/ser[24298]: ACC: call missed: from=<sip:4083709675@10.10.10.5>;tag=497AC670-18B, i-uri=s ip:esavelle@mydomain.com, method=INVITE, o-uri=sip:esavelle@10.10.10.167:5060, code=408 Request Timeout Dec 16 16:15:28 jiffypop /usr/local/sbin/ser[24298]: failure_route[4] Dec 16 16:15:28 jiffypop /usr/local/sbin/ser[24240]: ERROR: t_should_relay: status rewrite by UAS: stored: 408, received: 487 Dec 16 16:15:28 jiffypop /usr/local/sbin/ser[24229]: ACC: request acknowledged: from=<sip:4083709675@10.10.10.5>;tag=497AC670-18B , i-uri=sip:mvestal@mydomain.com, method=ACK, o-uri=sip:mvestal@10.10.10.98:5060, code=180 Dec 16 16:15:28 jiffypop /usr/local/sbin/ser[24298]: route[4]:call failed: forward to voicemail Dec 16 16:15:28 jiffypop /usr/local/sbin/ser[24298]: ACC: call missed: from=<sip:4083709675@10.10.10.5>;tag=497AC670-18B, i-uri=s ip:6600@10.10.10.49:5060, method=INVITE, o-uri=sip:mvestal@mydomain.com, code=408 Request Timeout Dec 16 16:15:28 jiffypop /usr/local/sbin/vmser[29524]: t_newtran created Dec 16 16:15:28 jiffypop /usr/local/sbin/vmser[29524]: **************** vm start - begin ****************** Dec 16 16:15:28 jiffypop /usr/local/sbin/vmser[29525]: t_newtran created Dec 16 16:15:28 jiffypop /usr/local/sbin/vmser[29525]: **************** vm start - begin ****************** Dec 16 16:15:28 jiffypop /usr/local/sbin/vmser[29524]: **************** vm start - end ****************** Dec 16 16:15:28 jiffypop /usr/local/sbin/vmser[29525]: **************** vm start - end ****************** Dec 16 16:15:28 jiffypop Sems[19578]: Error: 482 Loop detected Dec 16 16:15:31 jiffypop Sems[19578]: Error: while getting return code from Ser

20 years, 11 months

[Serusers] Using asterisk as a voicemail server with SER

by Samy Touati

Hi, Can I use asterisk as a voicemail server, where unanswered calls get forwarded to asterisk acting as a voicemail server. I like to fact that asterisk voicemail can be accessed through a phone (IVR). Is this setup feasable, has anyone tried it ? Thanks. Samy.

20 years, 11 months

RE: [Serusers] BYE Message Problem

by Darren Sessions

Sipura Firmware is 1.0.20 We are already in talks with Sipura about this issue as we found a bug for them yesterday. It appears though based on the Etheral traces and ngrep that the Sipura is sending the BYE messages - but SER is not forwarding them along. We are using Ser 8.12 on a Sun Ultra 60 - and as I said before, all other instances work flawlessly. Thanks, - Darren -----Original Message----- From: Andres [mailto:andres@telesip.net] Sent: Thursday, December 18, 2003 2:58 PM To: Darren Sessions; serusers(a)lists.iptel.org Subject: Re: [Serusers] BYE Message Problem On Thursday 18 December 2003 14:50, Darren Sessions wrote: > We also have a problem with BYE's not being sent from an on-net call > (sipura to sipura). What firmware version are you using? We have been testing the SPA2000 for about 3 weeks now and have not seen this issue. > > -----Original Message----- > From: Darren Sessions > Sent: Thursday, December 18, 2003 2:32 PM > To: 'serusers(a)lists.iptel.org' > Subject: RE: Serusers Digest, Vol 8, Issue 16 > Importance: High > > > Situation: > > Endpoint is called Party (Sipura SPA2000) > > Calling Party is routed through from a TDM call and VoIP'd by Cisco 3640 > router then sent to SER. > > Endpoint receives call - everything works. When the Sipura hangs up the > call, it sends a BYE to ser - but ser does not send the BYE to the router > to disconnect the TDM channel. > > All other situations result in a completed call with channels released on > the router just fine. > > After further inspection - we noticed that when the Sipura hangs up and > send the BYE to SER, SER responds with this: > > > SIP/2.0 481 Call/Transaction Does Not Exist > Via: SIP/2.0/UDP 11.11.111.155:5060;branch=z9hG4bK-b76fe29f > From: <sip:8646783182@22.22.222.154>;tag=bc5a80577e3d5642 > To: <sip:8644679887@33.33.333.157>;tag=322BCA98-25A3 > Call-ID: F429FDCE-30B011D8-BA0EDB9C-52307829(a)33.33.333.157 > CSeq: 101 BYE > Contact: <sip:8644679887@44.44.444.148> > Server: Sip EXpress router (0.8.12 (i386/linux)) > Content-Length: 0 > Warning: 392 44.44.444.148:5060 "Noisy feedback tells: pid=20033 > req_src_ip=22.22.222.154 req_src_port=5060 > in_uri=sip:8644679887@44.44.444.148:5060 > out_uri=sip:8644679887@44.44.444.148:5060 via_cnt==0" > > > Any ideas? > > Thanks, > > - Darren > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

20 years, 11 months

[Serusers] BYE Message Problem

by Darren Sessions

We also have a problem with BYE's not being sent from an on-net call (sipura to sipura). -----Original Message----- From: Darren Sessions Sent: Thursday, December 18, 2003 2:32 PM To: 'serusers(a)lists.iptel.org' Subject: RE: Serusers Digest, Vol 8, Issue 16 Importance: High Situation: Endpoint is called Party (Sipura SPA2000) Calling Party is routed through from a TDM call and VoIP'd by Cisco 3640 router then sent to SER. Endpoint receives call - everything works. When the Sipura hangs up the call, it sends a BYE to ser - but ser does not send the BYE to the router to disconnect the TDM channel. All other situations result in a completed call with channels released on the router just fine. After further inspection - we noticed that when the Sipura hangs up and send the BYE to SER, SER responds with this: SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 11.11.111.155:5060;branch=z9hG4bK-b76fe29f From: <sip:8646783182@22.22.222.154>;tag=bc5a80577e3d5642 To: <sip:8644679887@33.33.333.157>;tag=322BCA98-25A3 Call-ID: F429FDCE-30B011D8-BA0EDB9C-52307829(a)33.33.333.157 CSeq: 101 BYE Contact: <sip:8644679887@44.44.444.148> Server: Sip EXpress router (0.8.12 (i386/linux)) Content-Length: 0 Warning: 392 44.44.444.148:5060 "Noisy feedback tells: pid=20033 req_src_ip=22.22.222.154 req_src_port=5060 in_uri=sip:8644679887@44.44.444.148:5060 out_uri=sip:8644679887@44.44.444.148:5060 via_cnt==0" Any ideas? Thanks, - Darren

20 years, 11 months

RE: [Serusers] Re: Xten-RTPProxy choppy audio

by Klaus Darilion

I can reproduce the choppy audio setup: budgetone-100 <----> ser+rtpproxy(version from today) on redhat 9 (or 8, not sure) <------>x-lite build 1088 on win xp the choppy sound occours every 5 seconds, in both directions. RTP-analysis: RTP stream from budgetone to x-lite (via rtpproxy) is fine (no jitter, no loss) RTP stream from x-lite to budgetone (via rtpproxy): very strange: x-lite switches to a new SSRC (and seq-nr. start at 0 again) every 5 seconds. so, in my opinion: rtpproxy works fine, x-lite is guilty. but why does this happen only if the rtp proxy is involved? ...further investigations are planned... Klaus > -----Original Message----- > From: Jan Janak [mailto:janak@fokus.fraunhofer.de] > Sent: Wednesday, December 10, 2003 4:09 PM > To: Adrian Georgescu > Cc: serusers(a)lists.iptel.org; Ricardo Villa > Subject: Re: [Serusers] Re: Xten-RTPProxy choppy audio > > > Well, we can generate the traces locally, but I haven't encounter the > problem you describe so it makes no sense. > > You wrote you are able to reproduce the problem, in that case I would > like to ask you to generate the traces that show the problem > so we could > analyze and fix it. > > Also please tell us on what OS does this happen ? (I mean the > OS the proxy > is running on). > > Another question, what is the license of your RTP proxy ? I > didn't find > any licensing info in the sources. Will the sources be available ? > > Jan. > > On 10-12 16:04, Adrian Georgescu wrote: > > On Wednesday, Dec 10, 2003, at 15:48 Europe/Amsterdam, > Ricardo Villa > > wrote: > > > > >Adrian, > > > > > >Do you have an Etheral trace trace of such a call (using G711)? > > > > I guess ethereal traces can be generated with ethereal program and > > decoded locally on your servers if you want to isolate this. > > > > > I can > > >decode it and produce an audio file for all to examine. > This way we > > >can get > > >to the bottom of this. > > >Thanks, > > >Ricardo > > > > _______________________________________________ > > 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 > >

20 years, 11 months

[Serusers] test

by Jan Janak

ignore

20 years, 11 months

Re: [Serusers] Using asterisk as a voicemail server with SER

by Kannaiyan Natesan

This is only the forwarding code. You need to merge with the default configuration code which came with asterisk. Kannaiyan ----- Original Message ----- From: "Samy Touati" <samy(a)tunix.com> To: "Kannaiyan Natesan" <nkans(a)lycos.co.uk> Sent: Thursday, December 18, 2003 5:15 PM Subject: Re: [Serusers] Using asterisk as a voicemail server with SER > Hi, > > Sorry to bother on this, but I'm trying to understand how all of his > work. > In the code snippet, you sent me ser is redirecting the call to > asterisk. Are the phones registering in SIP to asterisk or to ser ? > I want the phones to register with ser but have mailboxes on asterisk. > Is this supported in asterisk, are you doing this? > > Thanks again. > > Samy. > > > ----- Original Message ----- > From: "Kannaiyan Natesan" <nkans(a)lycos.co.uk> > Date: Monday, December 15, 2003 2:54 pm > Subject: Re: [Serusers] Using asterisk as a voicemail server with SER > > > Samy, > > > > Hope you find this useful. > > here is the snippet > > > > # ------------------ module loading ------------------------------- > > --- > > > > ... your other modules here > > > > loadmodule "modules/sl/sl.so" > > loadmodule "modules/tm/tm.so" > > > > # ----------------- module parameters --------------- > > > > # -- tm params -- > > # fr_inv_timer sets value for INVITE transactions > > # fr_timer for all others > > > > modparam("tm", "fr_inv_timer", 15 ) > > modparam("tm", "fr_timer", 10 ) > > > > #------------------------------------------------------- > > > > route{ > > > > ......... > > > > append_branch("sip:asterisk@asthost:portno"); # tell your > > asterisk URI here > > # forward if the call fails > > t_on_failure("1"); > > # start forwarding all calls now > > t_relay(); > > > > } > > > > failure_route[1] { > > log(1,"asterisk voicemail\n"); > > t_relay(); > > } > > > > Kannaiyan > > > > ----- Original Message ----- > > From: "Samy Touati" <samy(a)tunix.com> > > To: "Kannaiyan Natesan" <nkans(a)lycos.co.uk> > > Cc: <serusers(a)lists.iptel.org> > > Sent: Monday, December 15, 2003 7:45 PM > > Subject: Re: [Serusers] Using asterisk as a voicemail server with SER > > > > > > > Only one user on asterisk will collect all voice mails. > > > I was thinking that I will need to create all the extensions in > > > asterisk so when the user cannot be reached at his phone, then ser > > > will rewrite the request to asterisk keeping the original uri. > > > Is this how to do it ? > > > > > > Thanks. > > > Samy. > > > > > > > > > > > > ----- Original Message ----- > > > From: "Kannaiyan Natesan" <nkans(a)lycos.co.uk> > > > Date: Monday, December 15, 2003 2:41 pm > > > Subject: Re: [Serusers] Using asterisk as a voicemail server > > with SER > > > > > > > Samy, > > > > > > > > That is feasible. > > > > You can create a sip user in asterisk and forward the call > > on busy. > > > > > > > > Kannaiyan > > > > > > > > > > > > ----- Original Message ----- > > > > From: "Samy Touati" <samy(a)tunix.com> > > > > To: <serusers(a)lists.iptel.org> > > > > Sent: Monday, December 15, 2003 7:35 PM > > > > Subject: [Serusers] Using asterisk as a voicemail server with SER > > > > > > > > > > > > > Hi, > > > > > > > > > > Can I use asterisk as a voicemail server, where unanswered > calls > > > > get > > > > > forwarded to asterisk acting as a voicemail server. > > > > > I like to fact that asterisk voicemail can be accessed > > through a > > > > phone > > > > > (IVR). > > > > > Is this setup feasable, has anyone tried it ? > > > > > > > > > > Thanks. > > > > > > > > > > Samy. > > > > > > > > > > > > > > > _______________________________________________ > > > > > Serusers mailing list > > > > > serusers(a)lists.iptel.org > > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > >

20 years, 11 months

[Serusers] Дорое утро!

by Waterfowl V. Saut

ПРЕДЛОЖЕНИЯ ПО ВЫДЕЛЕННЫМ СЕРВЕРАМ (дата-центры заубежа и России). icq 510642, e-mail: contact(a)emailacc.com AMD Duron™ 800MHZ (95$/месяц) AMD Duron™ 800MHZ Memory: 256MB PC133 SDRAM Hard Drive: 30GB IDE или 9GB SCSI Трафик: 100GB/MONTH IP Addresses: 5 Port: 10/100MBPS SWITCHED VLAN Management: FULLY MANAGED $30.00 установка $ 95.00 ежемесячно аку ------------------------------------------------------------- Intel Celeron™ 1.0GHZ (105$/месяц) Intel Celeron™ 1.0GHZ Memory: 256MB PC133 SDRAM Hard Drive: 30GB IDE or 9GB SCSI Трафик: 100GB/MONTH IP Addresses: 5 Port: 10/100MBPS SWITCHED VLAN Management: BASIC MANAGED $30.00 установка $105.00 ежемесяно Юслшека ------------------------------------------------------------- AMD Athlon™ XP 1700 1.47GHZ (130$/месяц) AMD Athlon™ XP 1700 1.47GHZ Memory: 256MB PC133 SDRAM Hard Drive: 40GB IDE or 18GB SCSI Трафик: 100GB/MONTH IP Addresses: 5 Port: 10/100MBPS SWITCHED VLAN Management: BASIC MANAGED $30.00 установка $130.00 ежемесячно Кеера ------------------------------------------------------------- Intel Pentium™ 4 1.5GHZ (160$/месяц) Intel Pentium™ 4 1.5GHZ Memory: 256MB PC133 SDRAM Трафик: 40GB IDE or 18GB SCSI Bandwidth: 1000GB/MONTH IP Addresses: 5 Port: 10/100MBPS SWITCHED VLAN Management: BASIC MANAGED $30.00 установка $160.00 ежемесячно Шифуку ------------------------------------------------------------- Intel Pentium™ 4 1.7GHZ (170$/месяц) Intel Pentium 4™ 1.7GHZ Memory: 512MB DDR RAM Hard Drive: 60GB IDE or 9GB SCSI Трафик: 1000GB/MONTH IP Addresses: 5 Port: 10/100MBPS SWITCHED VLAN Management: FULLY MANAGED $100.00 установка $170.00 ежемесячно С<rndline[random1.txt]>дез ------------------------------------------------------------- Intel Celeron™ 2.4GHZ (170$/месяц) Intel Celeron™ 2.4GHZ Memory: 512MB DDR RAM Hard Drive: 60GB IDE or 9GB SCSI Трафик: 1000GB/MONTH IP Addresses: 5 Port: 10/100MBPS SWITCHED VLAN Management: FULLY MANAGED $100.00 установка $170.00 ежемесячно Феекари ------------------------------------------------------------- Intel Pentium 4™ 2.4MHZ (215$/месяц) Intel Pentium 4™ 2.4GHZ Memory: 512MB DDR RAM Hard Drive: 80GB IDE or 36GB SCSI Трафик: 1000GB/MONTH IP Addresses: 5 Port: 10/100MBPS SWITCHED VLAN Management: FULLY MANAGED $200.00 установка $215.00 ежемесячно Рапуриши ------------------------------------------------------------- DUAL Intel Pentium 3™ 800MHZ (200$/месяц) Intel Pentium 3™ DUAL 800MH Memory: 256MB PC133 SDRAM Hard Drive: 40GB IDE or 18GB SCSI Трафик: 1000GB/MONTH IP Addresses: 5 Port: 10/100MBPS SWITCHED VLAN Management: BASIC MANAGED $100.00 установка $200.00 ежемесячно Енлвик :::::::CЕРВРЕА В РОССИИ, не ограниченный* траффик::::::::::::: ================================================ Бипкенда Pentium IV Celeron 1.7GHz (179$/месяц) Pentium IV Celeron 1.7GHz Memory: 512Mb RAM Hard Drive: 20Gb IDE Трафик: неограничен* $100.00 установка $179.00 ежемесячно Ысаник ------------------------------------------------------------- Pentium IV 2.4GHz (249$/месяц) Pentium IV 2.4GHz Memory: 1Gb RAM Hard Drive: 40Gb IDE Трафик: неограничен* $150.00 установка $249.00 ежемесячно Ферит ------------------------------------------------------------- Dual Pentium III 1.26GHz (310$/месяц) Dual Pentium III 1.26GHz Memory: 1Gb RAM Hard Drive: 36Gb UW SCSI Трафик: неограничен* $150.00 установка $310.00 ежемесячно Эличтила ------------------------------------------------------------- места, скрылся за парчовой завесой и вернулся с доказательством. Под

20 years, 11 months

[Serusers] test

by Jan Janak

ignore

20 years, 11 months

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sr-users December 2003