sr-users November 2010

sr-users@lists.kamailio.org

75 participants
112 discussions

Problem with Kamailio not routing ACK to a 200 OK
by Morten Isaksen 15 Nov '10

15 Nov '10

Hi, I have a strange problem with Kamailio 3.0.2. When one of our end users makes a call Kamailio does not route the ACK (in response to the 200 OK). For all other end users it works fine. For me it looks the the has_totag() checks for some reason fails and then t_check_trans() thinks it is a ACK to a local transactions and then terminates the script. Otherwise there should be more lines in the log file. The conf is pretty standard. route{ xlog("L_WARN", "New request - M=$rm RURI=$… [View More]ru F=$fu T=$tu IP=$si ID=$ci\n"); xlog("L_WARN", "ua=$ua"); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } if (has_totag()) { xlog("L_WARN", "has_totag start - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { xlog("L_WARN", "loose_route - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests #route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { xlog("L_WARN", "ACK t_check_trans - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { xlog("Ignoring ACK - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } xlog("L_WARN", "has_totag end - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } setflag(4); t_check_trans(); ... The log files show: Oct 28 10:51:13 sip-core-1 /usr/local/sbin/kamailio[10503]: WARNING: <script>: New request - M=ACK RURI=sip:1105@178.21.248.56:5060 F=sip:87776688@sip.uni-tel.dk T=sip:869 87106(a)sip.uni-tel.dk IP=178.21.248.20 ID=AI231CA9BD0A4A1C53(a)10.0.0.150 Oct 28 10:51:13 sip-core-1 /usr/local/sbin/kamailio[10503]: WARNING: <script>: ua=Aastra Intelligate The trace U 2010/10/28 10:51:02.616337 178.21.248.7:5060 -> 178.21.248.56:5060 INVITE sip:86987106@178.21.248.56 SIP/2.0. Record-Route: <sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>. Record-Route: <sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>. Via: SIP/2.0/UDP 178.21.248.7;branch=z9hG4bK690c.9cfea506.0. Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0. Via: SIP/2.0/UDP 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592. To: <sip:86987106@sip.uni-tel.dk>. From: <sip:87776688@sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F. Call-ID: AI231CA9BD0A4A1C53(a)10.0.0.150. CSeq: 2 INVITE. Max-Forwards: 68. Contact: <sip:87776688@87.104.233.108:5060;line=AI7EFC34995E724DD7>. Accept: application/sdp. Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER. P-Preferred-Identity: <sip:87776688@sip.uni-tel.dk>. Privacy: none. User-Agent: Aastra Intelligate. Content-Type: application/sdp. Content-Length: 280. X-trunktype: IC. . v=0. o=intelligate 1194032777 1194032777 IN IP4 87.104.233.106. s=call. c=IN IP4 178.21.248.22. t=0 0. m=audio 60984 RTP/AVP 8 0 18 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=ptime:20. # U 2010/10/28 10:51:02.636854 178.21.248.56:5060 -> 178.21.248.7:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 178.21.248.7;branch=z9hG4bK690c.9cfea506.0. Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0. Via: SIP/2.0/UDP 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592. From: <sip:87776688@sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F. To: <sip:86987106@sip.uni-tel.dk>;tag=1c2073920452. Call-ID: AI231CA9BD0A4A1C53(a)10.0.0.150. CSeq: 2 INVITE. Supported: em,timer,replaces,path,early-session,resource-priority. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002. Content-Length: 0. . ##### U 2010/10/28 10:51:04.134445 178.21.248.56:5060 -> 178.21.248.7:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 178.21.248.7;branch=z9hG4bK690c.9cfea506.0. Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0. Via: SIP/2.0/UDP 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592. From: <sip:87776688@sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F. To: <sip:86987106@sip.uni-tel.dk>;tag=1c2073920452. Call-ID: AI231CA9BD0A4A1C53(a)10.0.0.150. CSeq: 2 INVITE. Contact: <sip:1105@178.21.248.56:5060>. Record-Route: <sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>,<sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>. Supported: em,timer,replaces,path,early-session,resource-priority. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002. Content-Type: application/sdp. Content-Length: 259. . v=0. o=AudiocodesGW 2073965634 2073965290 IN IP4 178.21.248.56. s=Phone-Call. c=IN IP4 178.21.248.56. t=0 0. m=audio 6050 RTP/AVP 8 101. c=IN IP4 178.21.248.56. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. # U 2010/10/28 10:51:04.134748 178.21.248.7:5060 -> 178.21.248.20:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0. Via: SIP/2.0/UDP 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592. From: <sip:87776688@sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F. To: <sip:86987106@sip.uni-tel.dk>;tag=1c2073920452. Call-ID: AI231CA9BD0A4A1C53(a)10.0.0.150. CSeq: 2 INVITE. Contact: <sip:1105@178.21.248.56:5060>. Record-Route: <sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>,<sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>. Supported: em,timer,replaces,path,early-session,resource-priority. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002. Content-Type: application/sdp. Content-Length: 259. . v=0. o=AudiocodesGW 2073965634 2073965290 IN IP4 178.21.248.56. s=Phone-Call. c=IN IP4 178.21.248.56. t=0 0. m=audio 6050 RTP/AVP 8 101. c=IN IP4 178.21.248.56. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. # U 2010/10/28 10:51:04.136586 178.21.248.56:5060 -> 178.21.248.7:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 178.21.248.7;branch=z9hG4bK690c.9cfea506.0. Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0. Via: SIP/2.0/UDP 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592. From: <sip:87776688@sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F. To: <sip:86987106@sip.uni-tel.dk>;tag=1c2073920452. Call-ID: AI231CA9BD0A4A1C53(a)10.0.0.150. CSeq: 2 INVITE. Contact: <sip:1105@178.21.248.56:5060>. Record-Route: <sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>,<sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>. Supported: em,timer,replaces,path,early-session,resource-priority. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002. Content-Type: application/sdp. Content-Length: 259. . v=0. o=AudiocodesGW 2073965634 2073965290 IN IP4 178.21.248.56. s=Phone-Call. c=IN IP4 178.21.248.56. t=0 0. m=audio 6050 RTP/AVP 8 101. c=IN IP4 178.21.248.56. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. # U 2010/10/28 10:51:04.136837 178.21.248.7:5060 -> 178.21.248.20:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0. Via: SIP/2.0/UDP 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592. From: <sip:87776688@sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F. To: <sip:86987106@sip.uni-tel.dk>;tag=1c2073920452. Call-ID: AI231CA9BD0A4A1C53(a)10.0.0.150. CSeq: 2 INVITE. Contact: <sip:1105@178.21.248.56:5060>. Record-Route: <sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>,<sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>. Supported: em,timer,replaces,path,early-session,resource-priority. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002. Content-Type: application/sdp. Content-Length: 259. . v=0. o=AudiocodesGW 2073965634 2073965290 IN IP4 178.21.248.56. s=Phone-Call. c=IN IP4 178.21.248.56. t=0 0. m=audio 6050 RTP/AVP 8 101. c=IN IP4 178.21.248.56. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. ############################## U 2010/10/28 10:51:12.881179 178.21.248.56:5060 -> 178.21.248.7:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 178.21.248.7;branch=z9hG4bK690c.9cfea506.0. Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0. Via: SIP/2.0/UDP 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592. From: <sip:87776688@sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F. To: <sip:86987106@sip.uni-tel.dk>;tag=1c2073920452. Call-ID: AI231CA9BD0A4A1C53(a)10.0.0.150. CSeq: 2 INVITE. Contact: <sip:1105@178.21.248.56:5060>. Record-Route: <sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>,<sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>. Supported: em,timer,replaces,path,early-session,resource-priority. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002. Content-Type: application/sdp. Content-Length: 259. . v=0. o=AudiocodesGW 2073965634 2073965290 IN IP4 178.21.248.56. s=Phone-Call. c=IN IP4 178.21.248.56. t=0 0. m=audio 6050 RTP/AVP 8 101. c=IN IP4 178.21.248.56. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. # U 2010/10/28 10:51:12.882388 178.21.248.7:5060 -> 178.21.248.20:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0. Via: SIP/2.0/UDP 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592. From: <sip:87776688@sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F. To: <sip:86987106@sip.uni-tel.dk>;tag=1c2073920452. Call-ID: AI231CA9BD0A4A1C53(a)10.0.0.150. CSeq: 2 INVITE. Contact: <sip:1105@178.21.248.56:5060>. Record-Route: <sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>,<sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>. Supported: em,timer,replaces,path,early-session,resource-priority. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002. Content-Type: application/sdp. Content-Length: 259. . v=0. o=AudiocodesGW 2073965634 2073965290 IN IP4 178.21.248.56. s=Phone-Call. c=IN IP4 178.21.248.56. t=0 0. m=audio 6050 RTP/AVP 8 101. c=IN IP4 178.21.248.56. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. ##### <<<<<<<<<<<<<<<<<<<<<< This is the problem packet >>>>>>>>>>>>>>>>>>>>> U 2010/10/28 10:51:13.267863 178.21.248.20:5060 -> 178.21.248.7:5060 ACK sip:1105@178.21.248.56:5060 SIP/2.0. Record-Route: <sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>. Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.2. Via: SIP/2.0/UDP 87.104.233.108:5060;rport=5060;branch=z9hG4bK07103fe69c8af048b9b8216eb2f7233f. To: <sip:86987106@sip.uni-tel.dk>;tag=1c2073920452. From: <sip:87776688@sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F. Call-ID: AI231CA9BD0A4A1C53(a)10.0.0.150. CSeq: 2 ACK. Max-Forwards: 69. Route: <sip:178.21.248.7;lr=on;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>. User-Agent: Aastra Intelligate. Content-Length: 0. -- Morten Isaksen [View Less]

2 8

append_branch
by Rizwan Qureshi 15 Nov '10

15 Nov '10

Hi All, I just want to know how I can use append_branch() function for register requests. What I want to do is register all users on all asterisk servers by forwarding the register requests (to all asterisk servers) and then if the sip proxy throws call to any of the server the asterisk server, the call should be successful. If its possible then please let me know. Cheers -- Best Regards Rizwan Qureshi

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Re: [SR-Users] advertised_address and RR
by Ovidiu Sas 15 Nov '10

15 Nov '10

On Mon, Nov 15, 2010 at 8:04 AM, Torbjörn Abrahamsson <torbjorn.abrahamsson(a)gmail.com> wrote: > Hello. > > > We are trying to use Kamailio 3.0.3 behind a NAT. After reading the cookbook > we added the advertised_address parameter set to the public IP in our cfg. > This works for the via-header but the record-route-header still uses the > internal IP. Shouldn’t this be changed as well? The documentation says that > advertised_address should apply to RR. … [View More]

2 1

advertised_address and RR
by Torbjörn Abrahamsson 15 Nov '10

15 Nov '10

Hello. We are trying to use Kamailio 3.0.3 behind a NAT. After reading the cookbook we added the advertised_address parameter set to the public IP in our cfg. This works for the via-header but the record-route-header still uses the internal IP. Shouldnt this be changed as well? The documentation says that advertised_address should apply to RR. We have also tried to use record_route_preset, but still ends up with the internal IP in the record-route-header. We found some discussions … [View More]

1 0

siremis 2.0.0 CDRS link issues !
by X- x -X 15 Nov '10

15 Nov '10

Hi all, I was wondering if Im the only one getting this error (in siremis 2.0.0 with kamailio 3.1). i.e. I have activated the accounting modules & all but still when I click the "CDR list" in "Accounting Services" it errors on me saying that CDRS table is missing, which ofcourse is missing from the "openser" mysql DB. Why is CDRS table missing ? Is it okay (advisable) to create a CDRS table mentioned in siremis 1.0 docs (http://siremis.asipto.com/install-accounting/ ) Cheers --- X [… [View More]2010-11-12 14:01:08 (GMT)] An exception occurred while executing this script: Error message: #0, Error in query:: SELECT T0.cdr_id, T0.src_username, T0.src_domain, T0.dst_username, T0.dst_domain, T0.call_start_time, T0.duration, T0.sip_call_id, T0.sip_from_tag, T0.sip_to_tag, T0.src_ip, T0.cost, T0.created FROM `cdrs` T0 LIMIT 10. SQLSTATE[42S02]: Base table or view not found: 1146 Table 'openser.cdrs' doesn't exist Script name and line number of error: /var/www/siremis-2.0.0/openbiz/bin/data/BizDataObj_Lite.php:447 function: _run_search ( Array(2) ) @ /var/www/siremis-2.0.0/openbiz/bin/data/BizDataObj_Lite.php 243 function: fetch ( ) @ /var/www/siremis-2.0.0/openbiz/bin/easy/EasyForm.php 690 function: fetchDataSet ( ) @ /var/www/siremis-2.0.0/openbiz/bin/easy/FormRenderer.php 69 function: renderSmarty ( Object(EasyForm), "/var/www/siremis-2.0.0/siremis/modules/ser/template/grid.tpl" ) @ /var/www/siremis-2.0.0/openbiz/bin/easy/FormRenderer.php 45 function: render ( Object(EasyForm) ) @ /var/www/siremis-2.0.0/openbiz/bin/easy/EasyForm.php 1648 function: renderHTML ( ) @ /var/www/siremis-2.0.0/openbiz/bin/easy/EasyForm.php 1538 function: render ( ) @ /var/www/siremis-2.0.0/openbiz/bin/easy/ViewRenderer.php 83 function: renderSmarty ( Object(EasyView), "/var/www/siremis-2.0.0/siremis/modules/ser/template/view.tpl" ) @ /var/www/siremis-2.0.0/openbiz/bin/easy/ViewRenderer.php 40 function: render ( Object(EasyView) ) @ /var/www/siremis-2.0.0/openbiz/bin/easy/EasyView.php 274 function: _render ( ) @ /var/www/siremis-2.0.0/openbiz/bin/easy/EasyView.php 237 function: render ( ) @ /var/www/siremis-2.0.0/openbiz/bin/BizController.php 221 function: renderView ( "ser.view.CdrsListView", "", "", Null, "" ) @ /var/www/siremis-2.0.0/openbiz/bin/BizController.php 107 function: dispatchRequest ( ) @ /var/www/siremis-2.0.0/openbiz/bin/BizController.php 32 function: include_once ( "/var/www/siremis-2.0.0/openbiz/bin/BizController.php" ) @ /var/www/siremis-2.0.0/siremis/bin/controller.php 6 function: include ( "/var/www/siremis-2.0.0/siremis/bin/controller.php" ) @ /var/www/siremis-2.0.0/siremis/bin/_forward.php 90 function: include ( "/var/www/siremis-2.0.0/siremis/bin/_forward.php" ) @ /var/www/siremis-2.0.0/siremis/index.php 3 Please ask system administrator for help... [View Less]

2 1

Siremis 2.0: error in LCR table editing??
by Klaus Feichtinger 15 Nov '10

15 Nov '10

Hello list, has anybody else tried using SIREMIS 2.0 for administrating LCR tables in kamailio 3.1.0? I am not happy with the form / input boxes that are presented in the web browser, because some fields are not as expected..... Following points are - from my point of view - incorrect: 1) table "LCR_Gateway_List": the fields 'URI-scheme', 'Transport' and 'Strip' are represented by checkboxes. However, in reality these fields are defined with smallint(3) types and e.g. transport is … [View More]

5 6

Call from web
by Sergey Babichev 12 Nov '10

12 Nov '10

2 2

Re: [SR-Users] Question about the Q-value.
by Kosilov Fedor 12 Nov '10

12 Nov '10

OK, and how do you usually deal with this? I was thinking of putting the Q-value into the "Display Name" field, so when the device registers, I could get it using $fn (reference to display name of 'From' header) and put it directly to the location table in the database...

4 6

help in kamailio and tunnel server
by hala alramli 12 Nov '10

12 Nov '10

i want to instal tunnel server in kamailio . i want to ask if there is any opensource voip tunnel server . and if there is an tutorial to configure kamailio with tunnel server thanks for help hala

2 1

Re: [SR-Users] noack and t_relay erros errors
by JR Richardson 11 Nov '10

11 Nov '10

> Hi All, > > I'm still getting these errors and I'm struggling to resolve the > problem. I think I'm missing an append_branch or something simple in > my config, a little guiedance will be appriciated. > > The error: > > Nov 11 10:23:26 sip-router1 /usr/local/sbin/kamailio[23739]: ERROR: tm > [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding > Nov 11 10:23:26 sip-router1 /usr/local/sbin/kamailio[23739]: ERROR: tm > [tm.c:1300]: ERROR: … [View More]

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sr-users November 2010