sr-users December 2003

sr-users@lists.kamailio.org

108 participants
257 discussions

[Serusers] RE: Serusers Digest, Vol 8, Issue 16
by Darren Sessions 18 Dec '03

18 Dec '03

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

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[Serusers] Voicemail errors and other issues
by Gavin Bensom 18 Dec '03

18 Dec '03

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

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[Serusers] Using asterisk as a voicemail server with SER
by Samy Touati 18 Dec '03

18 Dec '03

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.

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RE: [Serusers] BYE Message Problem
by Darren Sessions 18 Dec '03

18 Dec '03

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

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[Serusers] BYE Message Problem
by Darren Sessions 18 Dec '03

18 Dec '03

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

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RE: [Serusers] Re: Xten-RTPProxy choppy audio
by Klaus Darilion 18 Dec '03

18 Dec '03

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 > >

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[Serusers] test
by Jan Janak 18 Dec '03

18 Dec '03

ignore

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Re: [Serusers] Using asterisk as a voicemail server with SER
by Kannaiyan Natesan 18 Dec '03

18 Dec '03

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 > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > >

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[Serusers] Дорое утро!
by Waterfowl V. Saut 18 Dec '03

18 Dec '03

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

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[Serusers] test
by Jan Janak 18 Dec '03

18 Dec '03

ignore

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sr-users December 2003