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[Users] t_replicate: Invalid nonce balue received
by Mark Price 05 Dec '06

05 Dec '06

Hi, I'm attempting to use t_replicate in order to load balance the registration proxy. However, when the REGISTER packet goes to proxy2, it says "401 Unauthorized". Both openser installations are working off of the same mysql database. The first proxy has a srv record for the domain being served, the second does not. The first openser.cfg looks like this: if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("voipcall.com", "subscriber")) { www_challenge("voipcall.com", "0"); exit; }; save("location"); t_replicate('proxy2'); exit; }; the second looks like this: if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("voipcall.com", "subscriber")) { www_challenge("voipcall.com", "0"); exit; }; save("location"); exit; }; Both openser instances have aliases for the domain, and as I said before, are operating off of the same database backend. Furthermore, if I set the sip client's sip domain so that it will register directly with proxy2, the phone registers with now problem. However, when the t_replicate executes, syslog says this: Dec 4 23:26:01 /usr/sbin/openser[15365]: check_nonce(): comparing [4574af56734dbd1d5c2c81242a74ba5cb5703c79] and [4574af56989148c2191ea66c5c3c1d75726da244] Dec 4 23:26:01 /usr/sbin/openser[15365]: pre_auth(): Invalid nonce value received Thanks, Mark

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[Users] BYE from call recipient not passing the firewall
by Mark Price 05 Dec '06

05 Dec '06

Hi, I am experiencing the following: openser 1.1 and asterisk are on public IP addresses, with openser acting as a sip proxy in front of asterisk. sip clients A and B live behind the same firewall. Let A be a TCP UAC. Let B be a UAC using TCP or UDP. If A calls B and B hangs up, A never sees the BYE packet, and never hangs up. If A calls B and A hangs up, the BYE packet is transmitted just fine, and B hangs up as it should. tcpdump at the sip proxy shows that when B hangs up, the SIP hangup dialog is as expected between B<->openser<->asterisk Then, asterisk sends a BYE through openser which is addressed to A, but A never responds. To be more exact, A never responds because the BYE message is not passing through the firewall. Perhaps someone has seen such a problem? I have included a collection of facts below about the situation. If there are others I can provide you with, let me know. If A and B are both TCP, then clients A and B each have exactly one tcp stream between themselves and openser. I.e. tcpdump shows precisely 2 each of syn and syn/ack packets. The source and destination ports of the final BYE packet are the same as the destination and source ports of the original INVITE packet. Furthermore, the time between phone registration and the sending of the last BYE packet is on the order of 20 seconds, and the behavior is consistent, so it is unlikely that the TCP stream timed out. The following four packets were cut and pasted from tcpdump. Would more information be helpful? The following packets represent: 1. invite from asterisk to openser 2. invite from openser to client B 3. bye from asterisk to openser 4. bye from openser to client B Recall that INVITE(tcp.src, tcp.dst) == BYE (tcp.dst,tcp.src). INVITE sip:9043067733@joinuneta.com:5065 SIP/2.0\r\n From: "Mark Price2" <sip:9043060000@66.129.95.24>;tag=as50a4e684\r\n To: <sip:9043067733@joinuneta.com:5065>\r\n Contact: <sip:9043060000@66.129.95.24>\r\n Call-ID: 6e9c8160194ea4eb59afa0da0c22633d(a)66.129.95.24\r\n CSeq: 102 INVITE\r\n INVITE sip:9043067733@66.177.61.238:17522;transport=TLS;rinstance=58c12cb6506504ae SIP/2.0\r\n From: "Mark Price2" <sip:9043060000@66.129.95.24>;tag=as50a4e684\r\n To: <sip:9043067733@joinuneta.com:5065>\r\n Contact: <sip:9043060000@66.129.95.24>\r\n Call-ID: 6e9c8160194ea4eb59afa0da0c22633d(a)66.129.95.24\r\n CSeq: 102 INVITE\r\n BYE sip:9043060000@66.177.61.238:17559 SIP/2.0\r\n From: "9043067733 (Softphone)"<sip:9043067733@joinuneta.com >;tag=as3f8914c4\r\n To: "Mark Price"<sip:9043060000@joinuneta.com>;tag=3c60f44e\r\n Contact: <sip:9043067733@66.129.95.24>\r\n Call-ID: ZDFkMjBlMWJlOGZlZWE4NmZlMzQ2NWE0OWNiOGYzYzU.\r\n CSeq: 102 BYE\r\n BYE sip:9043060000@66.177.61.238:17559 SIP/2.0\r\n From: "9043067733 (Softphone)"<sip:9043067733@joinuneta.com >;tag=as3f8914c4\r\n To: "Mark Price"<sip:9043060000@joinuneta.com>;tag=3c60f44e\r\n Contact: <sip:9043067733@66.129.95.24>\r\n Call-ID: ZDFkMjBlMWJlOGZlZWE4NmZlMzQ2NWE0OWNiOGYzYzU.\r\n Thanks, Mark Price

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[Users] forward_reply error
by johnk＠koreanet.us 05 Dec '06

05 Dec '06

Hi all. After UA makes a PSTN call, if remote PSTN peer hangs up first, server correctly receives BYE from gateway, hit BYE section in config and CDR records accordingly through radius. But if UA hangs up first, BYE section in config never gets hit, instead server logs "ERROR: forward_reply: no 2nd via found in reply" right after " ACC: transaction answered:...", which was logged when the call was answered by called PSTN. CDR accounting only records without STOP time and duration being zero. Any comments will be appreciated.

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[Users] regexp and performance...
by Mahesh Paolini-Subramanya 05 Dec '06

05 Dec '06

Which is more efficient? if (uri=~"^sip:foo.*") { or if (avp_check("$rU", "re/^foo/i")) { And, in the long run, given Amdahl's law, does it even matter? Enquiring minds want to know -- ******************************************* Mahesh Paolini-Subramanya (703) 386-1500 x9100 CTO mahesh(a)aptela.com Aptela, Inc. http://www.aptela.com "Aptela: How Business Answers The Call" *******************************************

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[Users] 302 routing?
by John Peters 05 Dec '06

05 Dec '06

Not sure why this is happening,

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[Users] radius auth change attribute
by tele＠plexia.com 05 Dec '06

05 Dec '06

Hello all, this was already asked on the SER mailinglist without response. is there a way to rewrite radius auth request information packet before sending to radius? i need to change User-Name attribute of the authentication request. reading the source of auth_radius module i think this is not possible right now. the User-Name is filled by the digest credential username + realm. could be usefull to have the possibility of customize this. thank you, :tele

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RE: [Serusers] Retry count setting for invites
by Paul Antinori (pantinor) 04 Dec '06

04 Dec '06

no that was the TM module docs I used. It was an sgml file. ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Monday, December 04, 2006 3:34 PM To: Paul Antinori (pantinor) Cc: serusers(a)iptel.org Subject: Re: [Serusers] Retry count setting for invites ?? Was that a question? Did you read the documentation I referred to? g-) Paul Antinori (pantinor) wrote: I am using the timers as below: modparam("tm","fr_timer",5) modparam("tm","fr_inv_timer",5) There do not seem to be any max count parameters to set according to the TM module docs. Thanks, Paul ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Monday, December 04, 2006 6:51 AM To: Paul Antinori (pantinor) Cc: serusers(a)iptel.org Subject: Re: [Serusers] Retry count setting for invites See the FAQs on iptel.org for docs on 0.9.x timers and What's new in 0.10.x for docs on the new timers in SER Ottendorf. g-) Paul Antinori (pantinor) wrote: Can the number of times that an INVITE is sent with UDP be configured with a modparam type setting in the cfg file? For example, a max invite retry count setting? I saw modparam "tm" settings for timers, but nothing for the retry count. Thank you, Paul ________________________________ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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RE: [Serusers] Retry count setting for invites
by Paul Antinori (pantinor) 04 Dec '06

04 Dec '06

I am using the timers as below: modparam("tm","fr_timer",5) modparam("tm","fr_inv_timer",5) There do not seem to be any max count parameters to set according to the TM module docs. Thanks, Paul ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Monday, December 04, 2006 6:51 AM To: Paul Antinori (pantinor) Cc: serusers(a)iptel.org Subject: Re: [Serusers] Retry count setting for invites See the FAQs on iptel.org for docs on 0.9.x timers and What's new in 0.10.x for docs on the new timers in SER Ottendorf. g-) Paul Antinori (pantinor) wrote: Can the number of times that an INVITE is sent with UDP be configured with a modparam type setting in the cfg file? For example, a max invite retry count setting? I saw modparam "tm" settings for timers, but nothing for the retry count. Thank you, Paul ________________________________ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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RE: [Serusers] SRV failover question
by Paul Antinori (pantinor) 04 Dec '06

04 Dec '06

Hi Steve, Yes I am trying to get SER to resend the INVITE to priority 2 after priority 1 timed out due to connection down scenario. Thank you, Paul -----Original Message----- From: Steve Blair [mailto:blairs@isc.upenn.edu] Sent: Monday, December 04, 2006 9:37 AM To: Paul Antinori (pantinor) Cc: Greger V. Teigre; serusers(a)iptel.org Subject: Re: [Serusers] SRV failover question Paul: Are you trying to get the proxy server to re-send the invite based on the SRV list or the phone? _Steve Paul Antinori (pantinor) wrote: > Greger and all, > > The URL does not really help. You will see my cfg script below is > already using rewritehostport with no port and the SRV name. > I also am not using t_on_failure as it should not be necessary > according the post by Klaus. > > Heres my SRV table with the ranking priorities: > > _sip._udp.vxml.pats.cisco.com SRV service location: > priority = 1 > weight = 1 > port = 5060 > svr hostname = vxml-1.pats.cisco.com > _sip._udp.vxml.pats.cisco.com SRV service location: > priority = 2 > weight = 1 > port = 5060 > svr hostname = vxml-2.pats.cisco.com > _sip._udp.vxml.pats.cisco.com SRV service location: > priority = 3 > weight = 1 > port = 5060 > svr hostname = vxml-3.pats.cisco.com > pats.cisco.com nameserver = ns1.pats.cisco.com > vxml-1.pats.cisco.com internet address = 10.86.129.x > vxml-2.pats.cisco.com internet address = 10.86.129.x > vxml-3.pats.cisco.com internet address = 10.86.129.x > ns1.pats.cisco.com internet address = 10.86.129.x > > > Thanks anyone for any help, > > Paul > > > > ---------------------------------------------------------------------- > -- > *From:* Greger V. Teigre [mailto:greger@teigre.com] > *Sent:* Monday, December 04, 2006 6:41 AM > *To:* Paul Antinori (pantinor) > *Cc:* serusers(a)iptel.org > *Subject:* Re: [Serusers] SRV failover question > > http://lists.iptel.org/pipermail/serusers/2006-October/030813.html > > Paul Antinori (pantinor) wrote: >> >> Hi, >> >> Regarding the SRV query addition in (ser-0.9.7-pre7), when the call >> fails on the first priority destination, it never seems to attempt to >> send the INVITE to the second priority destination in the SRV >> response list. Is there anything else to script in the ser.cfg file >> to get this to work? >> >> Thanks for help, >> >> Paul >> >> [root@iptel-sip-proxy /]# /usr/sbin/tethereal | grep SIP Capturing on >> eth0 >> 10.100556 10.86.142.144 -> 10.86.129.17 SIP Request: INVITE >> sip:77710107773365@10.86.129.17:5060;transport=udp >> 10.101774 10.86.129.17 -> 10.86.142.144 SIP Status: 100 trying -- >> your call is important to us >> 14.233596 10.86.129.17 -> 10.86.142.144 SIP Status: 408 Request >> Timeout 14.235930 10.86.142.144 -> 10.86.129.17 SIP Request: ACK >> sip:77710107773365@10.86.129.17:5060;transport=udp >> 2139 packets captured >> >> [root@iptel-sip-proxy /]# /usr/sbin/tethereal | grep DNS Capturing on >> eth0 >> 4.279486 10.86.129.17 -> 10.86.129.16 DNS Standard query SRV >> _sip._udp.vxml.pats.cisco.com >> 4.279897 10.86.129.16 -> 10.86.129.17 DNS Standard query response >> SRV 1 1 5060 vxml-1.pats.cisco.com SRV 2 1 5060 vxml-2.pats.cisco.com >> SRV 3 1 5060 vxml-3.pats.cisco.com >> >> 4.280045 10.86.129.17 -> 10.86.129.16 DNS Standard query A >> vxml-1.pats.cisco.com >> 4.280231 10.86.129.16 -> 10.86.129.17 DNS Standard query response A >> 10.86.129.2 <--this is unplugged from the network >> >> >> debug=3 >> fork=yes >> log_stderror=yes >> rev_dns=no >> port=5060 >> children=4 >> check_via=no >> sip_warning=yes >> fifo="/tmp/ser_fifo" >> uid="nobody" >> gid="nobody" >> >> #aliases for this proxy server >> #ie hostnames/domains that it routes for alias=sox.cisco.com >> alias=pats.cisco.com >> >> loadmodule "/usr/local/lib/ser/modules/sl.so" >> loadmodule "/usr/local/lib/ser/modules/tm.so" >> loadmodule "/usr/local/lib/ser/modules/rr.so" >> loadmodule "/usr/local/lib/ser/modules/maxfwd.so" >> loadmodule "/usr/local/lib/ser/modules/usrloc.so" >> loadmodule "/usr/local/lib/ser/modules/registrar.so" >> loadmodule "/usr/local/lib/ser/modules/textops.so" >> >> modparam("usrloc", "db_mode", 0) >> modparam("rr", "enable_full_lr", 1) >> modparam("tm","fr_timer",5) >> modparam("tm","fr_inv_timer",5) >> >> route{ >> >> >> if (!mf_process_maxfwd_header("10")) { >> sl_send_reply("483","Too Many Hops"); >> break; >> }; >> >> if (uri=~"^sip:1") { >> log(1, "SER found 1*\n"); >> rewritehostport("ccm.pats.cisco.com");route(1);break; >> } >> if (uri=~"^sip:2") { >> log(1, "SER Found 2*\n"); >> rewritehostport("ccm.pats.cisco.com");route(1);break; >> } >> if (uri=~"^sip:7") { >> log(1, "SER Found 7*\n"); >> rewritehostport("vxml.pats.cisco.com");route(1);break; >> } >> if (uri=~"^sip:8") { >> log(1, "SER Found 8*\n"); >> rewritehostport("SER.pats.cisco.com");route(1);break; >> } >> if (uri=~"^sip:9") { >> log(1, "SER Found 9*\n"); >> >> rewritehostport("ringtone.pats.cisco.com");route(1);break; >> } >> >> log(1,"Could Not Match DN to Route\n"); >> route(1); >> } >> >> route[1] >> { >> if (!t_relay()) { >> sl_reply_error(); >> }; >> } >> >> --------------------------------------------------------------------- >> --- >> >> _______________________________________________ >> 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] need help with AVP module (print_sattr())
by Kostas Marneris 04 Dec '06

04 Dec '06

Hello, I have the following config : loadmodule "/opt/ser-0.9.6/lib/ser/modules/avp.so" and in main route block, after the first Sanity Check Section : uri2attr("ruri"); print_sattr("ruri"); # if (is_sattr_set("ruri")) { xlog("L_NOTICE", "SET"); print_sattr("ruri"); } else { xlog("L_NOTICE", "NOT SET"); }; It seems that 'print_sattr()' simply does not work. On first 'print_sattr("ruri")' command I have nothing at my syslog. On 'if' block, the function 'is_sattr_set()' works ok, I get the "SET" message @ syslog, but not the 'ruri' attribute. Am I miss something ? thanks for any help, Kostas -- e-mail: K.Marneris(a)otenet.gr

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