- sr-users - kamailio.org

new features: write access to branch attributes
by Daniel-Constantin Mierla 22 Apr '10

22 Apr '10

Hello, yesterday I committed to git new feature that allow write access to branch attributes. It is about branches kept in core before TM takes over and creates the branches for transaction. These branches are filled when you do location lookup, alias db lookup or append_branch in route block. You can update the attributes: - r-uri - outbound proxy address (dst uri) - path - q value - local socket to be used for sending - branch flags More at: http://sip-router.org/wiki/cookbooks/pseudo-variables/devel#branch_attribut… The access can be done with index, 0 is the first. Negative value of index starts counting from the last, therefore index -1 is the last branch. If the index is missing, then first branch is used. Adding a new branch can be done with km_append_branch(...) of kex module or append_branch() functions. Dropping a branch is by assigning $null to uri attribute of branch. Among benefits: - drop invalid branches before creating the transaction - you can lookup the aor from branches in location table, avoiding a loopback of a branch (e.g., for alias db case) Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla

2 1

Catching internally generated BYE event & other
by Alex Balashov 21 Apr '10

21 Apr '10

Hello, I am using the dialog module to time out excessively long calls: modparam("dialog", "default_timeout", 21600) # 6 hours modparam("dialog", "dlg_flag", 2) ... route { ... ... setflag(2); $dlg_ctx(timeout_bye) = 2; ... ... if(!t_relay()) sl_reply_error(); } I have two unrelated questions: 1) Why does $dlg_ctx(timeout_bye) work here? My understanding is that $dlg_ctx(...) is a PV namespace container that exposes some aspect of the allocated dialog tracking structure. The dialog structure is created via a TM->dialog callback triggered by t_relay(); why is it possible to access it prior to that point? 2) The spoofed BYEs that are generated by the proxy with this setting do not show up as sequential requests, nor are catchable in script at all. As a result, BYE events are not written to CDRs with 'acc' and I have no way of knowing when the call actually closed. What is the most efficient and canonical solution to this problem? I can think of two: (a) Define a $dlg_ctx(timeout_route) and have it write a BYE event artificially using acc_db_request(). (b) Loop the BYE request back to the proxy with some sort of differentiating URI parameter or similar that causes it to be dropped instead of forwarded. But this is really nasty. What is your suggestion? Thanks, -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

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Appended headers are not added in serial forking
by Iñaki Baz Castillo 21 Apr '10

21 Apr '10

Hi, I receive a request, append some custom headers and use "load_gws()" (from LCR module). If the first gateway fails I do "next_gw()" and this second request to the second gateway doesn't contain the previously appended custom headers. Is it the expected behavior? -- Iñaki Baz Castillo <ibc(a)aliax.net>

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SQLOps question
by Maciej Bylica 21 Apr '10

21 Apr '10

Hello, I am trying to use SQLOps Module in my kamailio 1.5 configuration. According to your explanations every result of sql_query should be freed just after using that data. So it means that there will be better to fetch for THE SAME data three times instead of just one. Am I right? But nevertheless I've been trying to query just one and free the result of the sql_query at the end of the file. Could sb tell me if the result of sql query executed in the main part of the script is remembered and could be used on failure route as well? Thanks, Maciej.

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Re: [SR-Users] Unregister when tcp connection timeout
by Daniel-Constantin Mierla 20 Apr '10

20 Apr '10

please keep the mailing list cc-ed, others may benefit of this discussion, participate with ideas or solutions. On 4/20/10 9:18 PM, Bruno Bresciani wrote: > Thank you for your help Daniel... I was reading the source code to > understand the parameter "connection_timeout" in the TLS module 3.0.x > version and check if it help me too. When timer "connection_timeout" > expire, the function that die idle tls connection cold generate a user > unregister regarding this connection. That event happens in a different place than location service and a location record does not keep reference to the connection, therefore you need to do a bit of coding. Cheers, Daniel > > > 2010/4/20 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > > > On 4/20/10 8:29 PM, Daniel-Constantin Mierla wrote: > > > > On 4/20/10 7:30 PM, Iñaki Baz Castillo wrote: > > 2010/4/20 Bruno Bresciani<bruno.bresciani(a)gmail.com > <mailto:bruno.bresciani@gmail.com>>: > > Hi people, this is my first time that I writing to > this list and I nedd some > help... When kamailio lost connection with a user > registred over TLS > (timeout for examlple), I want this user be unregister > in kamailio too. How > can I make this unregister? > > It's a commendable feature request, but it's not > implemented right now. > > > it might now be easy to catch connection lost (forward or wait > state case). > > > ^^^ I meant: it might NOT be easy ... > > E.g., for forward case, detection is after config processing, when > trying to relay. If there are many branches, it is not even > reported if one branch succeeds or sends another negative reply code. > > > > Also, unregistering a contact is not really following the rfc, > anyhow, this is not something complex to add if you have the > contact details. > > What you can do right now is to avoid trying to send if the > connection is not longer open, see: > http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.0.x#set_forward_n… > > > Contact is in memory, but no penalty for performances. Very > useful for NAT cases or when UA is supposed to keep the > connection open. > > Cheers, > Daniel > > > -- > Daniel-Constantin Mierla * http://www.asipto.com/ * > http://twitter.com/miconda * > http://www.linkedin.com/in/danielconstantinmierla > > -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla

1 0

Unregister when tcp connection timeout
by Bruno Bresciani 20 Apr '10

20 Apr '10

Hi people, this is my first time that I writing to this list and I nedd some help... When kamailio lost connection with a user registred over TLS (timeout for examlple), I want this user be unregister in kamailio too. How can I make this unregister? Thank you...

3 3

PVs in Regular Expression Transformations
by jh＠tutpro.com 20 Apr '10

20 Apr '10

i tried to embed a PV in regex transformation, but didn't get desired result. is it possible or must regex only contain strings. here is an example, where the idea is to strip string "456" from the end of string "123456": $var(test) = "123456"; $var(rest) = "456"; $var(test) = $(var(test){re.subst,/^(.*)$var(rest)$/\1/}); xlog("result is $var(test)\n"); unfortunately, the result is 0 rather than "123". i also tried $var(test) = $(var(test){re.subst,/^(.*)\$var(rest)$/\1/}); and $var(test) = $(var(test){re.subst,/^(.*)$(var(rest))$/\1/}); but still 0. -- juha

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[Kamailio-Users] new module: embedded lua
by Daniel-Constantin Mierla 20 Apr '10

20 Apr '10

Hello, few weeks ago I uploaded a new module named app_lua, providing functionality to execute Lua (http://www.lua.org) scripts in your configuration file. Lua is small and fast, therefore fits very well in a sip proxy environment. It has a lot of extensions, so when you cannot do something with existing cfg functions, check this one as well. The readme and the api are available at: http://sip-router.org/wiki/api/lua/devel http://sip-router.org/docbook/sip-router/branch/master/modules/app_lua/app_… The API will be enhanced during this development cycle, your feedback will help a lot in this direction. By now you get access to pseudo-variables, sip headers, logging functions and sl module. For new comers in community, if you are a Perl fan, then you can use the perl module in a similar way. Also, another module was uploaded recently, app_python, allowing to write python extensions. Module and API for Python are not yet documented, but I think this is not a barrier for a true Python fan, actually I hope will be a motivation to contribute the docs :-) Cheers, Daniel -- Daniel-Constantin Mierla SIP Server Professional Solutions * http://www.asipto.com/

1 1

Striping Character from $rU
by dotnetdub 19 Apr '10

19 Apr '10

Hi, Running 1.53 and also 3.01 I know there are operators to strip() and strip_tail() to remove characters from the front and end of a uri - what would people recommend to make sure that the uri only ever contains numerical value? If I wanted to for example strip the - character what would be the best way? TIA. Stephen.

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[Fwd: Re: [Kamailio-Users] Possible timing/latency relatedproblem ?]
by Jeff Brower 19 Apr '10

19 Apr '10

Juha- Vikram had posted a while back on the thread where he explained a bit more about his resolution and the fact the issue was occurring with VoipSwitch but not asterisk. Let us know if you want a link to that. Thanks for following up. -Jeff -------------------- Original Message ----------------- Subject: Re: [SR-Users] [Kamailio-Users] Possible timing/latency relatedproblem ? From: "Vikram Ragukumar" <vragukumar(a)signalogic.com> Date: Mon, April 19, 2010 11:15 am To: "Juha Heinanen" <jh(a)tutpro.com> Cc: users(a)lists.kamailio.org ------------------------------------------------------- Juha, > what was the conclusion regarding this? did the problem go > away when you called nat traversal functions both on 180 > ringing and 183 session progress? Yes. The problem did go away once we started forcing rtpproxy on 180 Ringing also. Thanks and Regards, Vikram.

1 0

[Kamailio-Users] Possible timing/latency related problem ?
by Vikram Ragukumar 19 Apr '10

19 Apr '10

Hello, Scenario 1 ---------- --------- -------- -------- -------- -------- |Cellphone|--|Wireless|--|Internet|--|Kamailio|--|Asterisk| | | |Router | | | |rtpproxy| | | --------- -------- -------- -------- -------- |<-Lowlatency->| Scenario 2 ---------- --------- -------- -------- -------- -------- |Cellphone|--|Wireless|---|Internet|--|Kamailio|--|Asterisk| | | |Router | | | |rtpproxy| | | --------- -------- -------- -------- -------- |<-Highlatency->| Both scenario 1 and scenario 2 are similar in setup, excepting that in scenario 2 there is a greater latency involved in SIP and RTP transmission between the wireless router and Kamailio+rtpproxy server. In scenario 1 calls from the cellphone using a SIP softphone app go through 100% of the time with both endpoints of the call being audible. However in scenario 2, calls from the cellphone go through with both endpoints of the call being audible only sometimes, and during all other attempts, call goes through with no voice from the cellphone. In scenario 2 the SIP softphone can always register, the SDP connection address is always rewritten to use rtpproxy so it does not appear to be a NAT issue, rtpproxy's logs show a session being set up and torn down, with RTP stats for caller being 0. In both scenarios, the equipment and their configuration are identical. (Both cellphone's are Nokia e51, router's are Netgear wgr614) What might be probable causes for the behavior seen in scenario 2 ? Could there be a timing/latency related issue ? Any pointers on how to approach this problem will be appreciated. Thanks and Regards, Vikram.

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loadbalancer
by Hector Muñoz 19 Apr '10

19 Apr '10

Hi all, I have a kamailio sip server with two trunk connections to two asterisk servers. The register server is kamailio. I'd like to be able to configure a load balancing with kamailio between two asterisk servers. How can i do that? Do you know any good tutorial for configure load balancing with dispatcher module in kamalio? Thanks in advance. Regards. -- jabber: triptik(a)12jabber.com blog: http://impresionesdeunloco.wordpress.com

1 0

sr-trunk with mysql support
by Meftah Tayeb 17 Apr '10

17 Apr '10

hi dear all, i'm trying to use latest trunk with mysql i have down compiling kamailio now i don't know ghow do i enable mysql for kamailio (not for kamdbctl) the db is created successfully but kamailio is unable to connect to the db default user/pass, i don't change anything how do i use this define stufs? thanks

3 3

load balancing redirect
by Benjamin Lawetz 17 Apr '10

17 Apr '10

Hi, I'm trying to setup Kamailio to act as a load-balancer in front of asterisk servers. I want it to do the following: - The UA sends an invite to Kamailio - Kamailio sends a 30x redirect to the UA (dispatcher) - The UA sends it's invite to the asterisk server I've seen a couple of examples, but doesn't seem to work for me. Here's the relevant config: route{ if (method=="REGISTER") { [...clip...] }; if (method=="INVITE") { ds_select_domain("2","8"); #ds_select_dst("2","8"); sl_send_reply("300","Redirect"); exit; }; } What seems to happen is that Kamailio does send the redirect, but with no indication of where to go. ds_select_domain (or ds_select_dst ... I having trouble understanding the difference like many people it seems) I'm guessing selects an appropriate uri, but doesn't put it the contact for the redirect. Is there anyway to see what was set by the ds_select_dst? what pv would have this value? What do I need to set prior to sending a redirect reply for it to be properly forwarded? BTW: using Kamailio 3.0.1 Thank you, -- Benjamin

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wiki: upgrade from 3.0.x to 3.1.x
by Daniel-Constantin Mierla 17 Apr '10

17 Apr '10

Hello, I started the page that should collect notes about how to upgrade from version 3.0.x to 3.1.0: http://sip-router.org/wiki/install/3.0.x-to-3.1.x 3.1.0 is not yet released, target date is September 2010, right now is the code on master branch in GIT repository. However, this page will help people that want to test it or use some modules from it, making as well the life easier before release. Please add to the document as you change (or discover) something breaking compatibility with 3.0.x - the wiki is open. Thanks, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla

1 0

reply when drop() in branch route
by jh＠tutpro.com 17 Apr '10

17 Apr '10

i call drop() in branch route and it is the only branch left. the branch gets correctly dropped, i.e., the request is not send out, but the reply to UAC is strange: U 2010/04/14 21:03:43.750712 192.98.102.10:5060 -> 192.98.103.2:5074 SIP/2.0 477 Unfortunately error on sending to next hop occurred (477/TM). Via: SIP/2.0/UDP 192.98.103.2:5074;rport=5074;branch=z9hG4bKaoxzewgj. To: <sip:0407058055@192.98.102.10>;tag=c02670ad1171fe45d9ff9a27d6c2cb82-e03b. From: <sip:+35816234567@foo.bar>;tag=bclsw. Call-ID: doibilvynrtqxpk@localhost. CSeq: 134 INVITE. Server: SIP Proxy (3.0.99-dev1 (i386/linux)). Content-Length: 0. why this kind of reply, because the proxy didn't even try to send the request to next ho? is it possible to somehow choose in the script, which reply to send? in this example, i'm using lcr to choose gws and none of them was suitable, so something like '503 service not available' would be more appropriate. -- juha

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tm response issue
by Klaus Darilion 16 Apr '10

16 Apr '10

Hi! Kamailio does not forward the received 503 response but instead generates a new 500 response. Failure routes are not activated. U 2010/04/14 15:08:26.201865 83.136.22.111:5060 -> 83.136.32.159:5060 SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 83.136.32.159;branch=z9hG4bK2a5a.0d5ea987.0;received=83.136.32.159 Via: SIP/2.0/UDP 10.10.0.51:5086;received=83.136.33.3;rport=5086;branch=z9hG4bK2901533893 From: <sip:klaus.darilion@labs.nic.at>;tag=1939498352 To: <sip:01888801505641636@labs.nic.at>;tag=as1b74528e Call-ID: 3387977514 CSeq: 21 INVITE Server: Asterisk PBX 1.6.2.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 X-Asterisk-HangupCause: Unallocated (unassigned) number X-Asterisk-HangupCauseCode: 1 U 2010/04/14 15:08:26.201865 83.136.32.159:5060 -> 83.136.22.111:5060 ACK sip:01888801505641636@labs.nic.at SIP/2.0 Via: SIP/2.0/UDP 83.136.32.159;branch=z9hG4bK2a5a.0d5ea987.0 From: <sip:klaus.darilion@labs.nic.at>;tag=1939498352 To: <sip:01888801505641636@labs.nic.at>;tag=as1b74528e Call-ID: 3387977514 CSeq: 21 ACK Max-Forwards: 69 Content-Length: 0 # U 2010/04/14 15:08:26.201865 83.136.32.159:5060 -> 83.136.33.3:5086 SIP/2.0 500 Service Unavailable Via: SIP/2.0/UDP 10.10.0.51:5086;rport=5086;branch=z9hG4bK2901533893;received=83.136.33.3 From: <sip:klaus.darilion@labs.nic.at>;tag=1939498352 To: <sip:01888801505641636@labs.nic.at>;tag=c3829fdbb3bfc7416671e56cae8e633d-a9b5 Call-ID: 3387977514 CSeq: 21 INVITE X-Info: (83.136.32.159:1035) Enforcing NAT traversal for local caller X-Info: (83.136.32.159:1035) Enforcing symmetric response routing Server: kamailio (3.0.1 (i386/linux)) Content-Length: 0 Is this a feature or a bug? thanks klaus

2 1

Re: [SR-Users] load balancing redirect
by Iñaki Baz Castillo 16 Apr '10

16 Apr '10

2010/4/16 Benjamin Lawetz <benjaminlawetz(a)gmail.com>: >> I strongly suggest not to do that. Instead: >> >> - The UA sends an invite to Kamailio >> - Kamailio uses LCR module to randomly select asterik-1 or asterisk-2. >> - Kamailio sends the request to the selected asterisk. >> - If there is a failure (to inspect in failure_route) Kamailio uses >> next_ge() (LCR module) and routes the INVITE to the other asterisk. >> >> This will work really better. > > Good idea, hadn't thought of a retry for the invite (was thinking of > removing the server from the dispatch if it failed, but your idea is > better). > I'm guessing this doesn't add too much load to Kamailio. > For learning/understanding the dispatcher... was what I wanted to do possible? Please, keep the mail within the maillist thread. No, it doesn't add too much load to Kamailio. This is one of the major purposes of a transaction stateful proxy (the serial forking in order to implement failover). Take a look to the LCR module. It gives you load balancing (with priorities if you want) and failover features. It's all you need in your scenario. -- Iñaki Baz Castillo <ibc(a)aliax.net>

1 0

call not being accounted when rejected by global blacklist
by Panagiotis Skoulikaritis 15 Apr '10

15 Apr '10

Dear list members I'm using kamailio 3 and the problem that I'm facing is that when a call is rejected by global blacklist, kamailio does not generate an accounting record. Is this normal behavior or I need to configure something somewhere ? Below are parts of the config file # ----- acc params ----- modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 0) modparam("acc", "failed_transaction_flag", 3) #!ifdef WITH_ACCDB modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "db_url", "mysql://username:password@localhost/openser") modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif ---------------------- if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } ------------------------- if (is_method("INVITE")) { setflag(1); # do accounting setflag(2); # log missed calls setflag(3); # log failed calls setflag(4); # do siptrace setflag(5); # log dialog ------------- route[name] { # --- Remove # or %23 from rU --- bm_start_timer("hash_remove"); if($rU =~ "^.*%+") { xlog("------- The number contains %23 ---- "); $rU = $(rU{re.subst,/^(.*)%23(.*)/\1\2/}); xlog(" ------- The perl $rU ------- "); } if($rU =~ "^.*#+") { xlog("------- The number contains # ----"); $rU = $(rU{re.subst,/^(.*)#(.*)/\1\2/}); xlog("------- The perl $rU ------- "); } bm_log_timer("hash_remove"); # --- PDT section ------ get service prefix to be appended before is send out bm_start_timer("pdt_lookup"); if(!prefix2domain("2", "0")) { sl_send_reply("404", "Prefix not found"); exit; } $var(dstgrp) = 0; $var(dstgrp) = $(rd{s.int}); if($var(dstgrp)==0) { sl_send_reply("404", "Service not found"); exit; } $avp(s:service_prefix) = $var(dstgrp); xlog("--$rU----------- $avp(s:service_prefix) --- PDT ---------\n"); bm_log_timer("pdt_lookup"); # --- DialPlan section --- get the routing group id xlog("----- The rU is $rU before DP -----\n"); bm_start_timer("dialplan_lookup"); if(dp_translate("1", "$rU/$rU")) { $avp(s:dstgrp) = $(var(dpattrs){s.int}); xlog("----- The destination Drouting GroupID is $avp(s:dstgrp) -----\n "); } bm_log_timer("dialplan_lookup"); xlog("----- The rU is $rU after DP -----\n"); # --- Global Blacklist section --- bm_start_timer("blacklist_lookup"); if(!check_blacklist("globalblacklist")){ xlog("----- The number is blacklisted globaly! -----"); sl_send_reply("403", "Forbidden"); exit; } bm_log_timer("blacklist_lookup"); # --- Dynamic routing section --- route the call based on the routing group id bm_start_timer("drouting_lookup"); if (!do_routing("$avp(s:dstgrp)")) { xlog("----- DRouting don't match dstgrp $avp(s:dstgrp) -----\n"); sl_send_reply("404","Route Not Found -$rU-"); exit; } bm_log_timer("drouting_lookup"); xlog("-----DRouting with $rU and $avp(s:dstgrp) -and $avp(s:service_prefix)----\n"); # Append service prefix in front of number $ru = "sip:" + $avp(s:service_prefix) + $rU + "@" + $rd; xlog(" ------ $ru ----$rU- $avp(s:user)---\n"); t_on_failure("FAIL_ONE"); route(RELAY); exit; return; } Regards Panagiotis

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[Kamailio-Users] PKG memory issues
by Iñaki Baz Castillo 15 Apr '10

15 Apr '10

Hi, I've experimented PKG out of memory in a production server. Of course I'm already increasing such value but I would like to know if the current settings and traffic could run into PKG memory issues: - Dual core INTEL XEON 3.00 GHz server. - Kamailio PKG memory = 4 MB. - Kamailio Shared memory = 64 MB. - Kamailio just listens in a single UDP port (8 childrens). - Just INVITE method is handled (no registration, no subscription). The script does the following for each request: - 'permissions' module to match source IP (just ~20 entries in 'address' table). - 2 custom SQL queries (returning a simple value). - 'dialog' memory (just in memory). - 'uac' module to change From header. - There are 10 AVP's set per transaction. - 'lcr' module for routing to two gateways (just 2 entries in 'lcr' table). - 'rtpproxy' is forced for every call. The server has been working properly for months but these days the traffic has been duplicated, having ~400 simultaneous calls in peak hours. Also note that such calls come from callcenters so they are ver "fast". With this environment, is it normal to get into PKG memory issues (4 MB)? I understand that it makes sense, but I would like to hear some opinions. Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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minutes IRC meeting 2010-04-14
by Henning Westerholt 15 Apr '10

15 Apr '10

Hello, the minutes of the developer IRC meeting from yesterday are now available in our wiki: http://sip-router.org/wiki/devel/irc-meetings/2010-04-14-minutes Best regards, Henning -- Henning Westerholt - Development Consumer Products / Consumer Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

1 0

Hostname in VIA on Forward
by Eric Hiller 15 Apr '10

15 Apr '10

Hello all, I need to have kamailio put its hostname on its via append instead of its IP address. How can I do this or is it possible? Thank you,Eric _________________________________________________________________ Hotmail has tools for the New Busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:…

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Re: [SR-Users] Failure to load module db_mysql.so due to undefined symbol: log
by Pratab Ali 14 Apr '10

14 Apr '10

Hello Marius, Yes, I can confirm that MySQL-devel-community-5.1.43-1.sles10.i586.rpm was also installed. I think the log_10_int is included by something that's linked in and has nothing to do with MySQL. A colleague and friend has installed on a Debian system with success. I am about to install Debian over my SuSE installation. Thanks Pratab. -----Original Message----- From: marius zbihlei [mailto:marius.zbihlei@1and1.ro] Sent: 14 April 2010 14:34 To: Pratab Ali Subject: Re: [SR-Users] Failure to load module db_mysql.so due to undefined symbol: log Pratab Ali wrote: > Hello > > My version of libmysqlclient was installed from this file > MySQL-client-community-5.1.43-1.sles10.i586.rpm. So, that'd make it > version 5.1.43.1 I'd guess. > > Hello, I presume you have also installed MySQL-devel-community-5.1.43-1.sles10.i586.rpm <http://rpm.pbone.net/index.php3/stat/4/idpl/13531085/dir/other/com/MySQ L-devel-community-5.1.43-1.sles10.i586.rpm.html> for the development headers and libraries ... Can you check if this is also installed . I have tested on Debian with libmysqlclient-dev 5.1.45 and it works. I still wonder about the log_10_int symbol... Marius > Unfortunately, it is the config file parser that is detecting the > unresolved symbol and so refusing to run kamailio at all. > > Thanks > Pratab > >

3 7

reminder: irc devel meeting, apr 14, 15:00utc
by Daniel-Constantin Mierla 14 Apr '10

14 Apr '10

Hello, short reminders about tomorrow's IRC devel meeting on #sip-router / irc.freenode.net. Add your topics to wiki if you want something to be discussed: http://sip-router.org/wiki/devel/irc-meetings/next Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla

1 2

syntax error with kamailio
by alexis heron 14 Apr '10

14 Apr '10

Hi, I have a problem to routing with kamailio. When I restart kamailio I have this error message : 0(710) : <core> [cfg.y:3328]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 640, column 2-3: syntax error 0(710) : <core> [cfg.y:3328]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 640, column 2-3: ERROR: bad config file (2 errors) here is my line 640 : if (uri=~"sip:9[0-9][0-9]@.*") { //line 640 log(1, "Matched Cisco Call Manager"); route(4); }; Can you help me please thank you

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[NAT] BYE not received to caller
by Pascal Maugeri 14 Apr '10

14 Apr '10

Hi I need some "guidelines" to troubleshoot the following issue: a) A is behind NAT b) B is not behind NAT c) A calls B, SIP INVITE is sent over TCP d) A's firewall does NAT and changes the source port to let's say p1 e) B releases the call and sends BYE over UDP f) Kamailio sends the BYE to A, over UDP, to the NATed source port p1 ?! 2 comments: - This scenario works perfectly when A is the one who disconnects the call. and of course when no NAT is involved everything works ok - if works when I comment the line "fix_nated_contact()" in the route[NAT] block: route[NAT]{ #!ifdef WITH_NAT force_rport(); #if (nat_uac_test("19")) { if (nat_uac_test("3")) { if (method=="REGISTER") { setbflag(10); fix_nated_register(); } else { #fix_nated_contact(); [...] Any hint is very welcome. Cheers Pascal

3 7

Failure to load module db_mysql.so due to undefined symbol: log
by Pratab Ali 14 Apr '10

14 Apr '10

Hello, I have installed from source version 3.0.1 of Kamailio on an OpenSuSE 10 Linux system. Currently I am attempting to enable persistence in my Kamailio with the help of MySQL. In the config file kamailio.cfg I have enabled MySQL as follows #!define WITH_MYSQL In the config file db_mysql.so is the first module to be loaded by calling loadmodule, but the module fails to load. By turning, Kamailio, debugging on I get the following explanation. 0(9858) DEBUG: <core> [sr_module.c:382]: load_module: trying to load </usr/local/lib/kamailio/modules/db_mysql.so> 0(9858) ERROR: <core> [sr_module.c:390]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules/db_mysql.so>: /usr/local/lib/kamailio/modules/db_mysql.so: undefined symbol: log Using the unix/linux tool 'nm' I see the following. mindy:~/tmp # nm /usr/local/lib/kamailio/modules/db_mysql.so | grep log U getlogin@@GLIBC_2.0 U log 001a6e40 R log_10 001e5d00 D log_10_int U log_level_info U log_stderr U syslog@@GLIBC_2.0 And for sure the capital 'U' means undefined. I have also searched the source code that came with the kamailio package, for definitions of 'log'. And it also looks like I'll be in trouble later for log_level_info, log_level_stderr and syslog are undefined too. However, log_level_info I have just found in source files in the kamailio package. So, why isn't it linking in? Unfortunately, searching on the internet I have not yet come across a forum post that tackles the above issue. As, I suspect everyone else has got the build right. Have I built db_mysql.so incorrectly? Though I built it using the make files that came with the package. Is it a case of linking in the appropriate libraries to resolve these undefined symbols? Finally, I have a colleague who has used the same version of kamailio but installed on a Debian lenny distro. He has been successful in getting db_mysql loaded and persistence enabled. However, he has an older version of mysql (by only one minor-version number) though. Having looked that the changes' history for the two mysql versions, I see no obvious reasons that could be the cause of my failure. By comparing notes, he did nothing too different than I did. We both followed the instructions in the INSTALL and README files. There is no particular reason for me using OpenSuSE 10 other than it was already installed and the computer was not being used by anyone. I am considering installing Debian Lenny on my computer to match my colleague, would this be prudent or a waste of time? i.e I may face my same problem because I'm missing a stage? Regards and thank you, in advance, for any help given. Pratab

3 5

[Kamailio-Users] Relaying NOTIFY UDP messages over TCP
by Pascal Maugeri 14 Apr '10

14 Apr '10

Hi We need to force some NOTIFY messages going through our kamailio instance over UDP to be sent over TCP. For that purpose we inserted the following relaying rule at the beginning of the routing script: route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(msg:len>1300 && is_method("NOTIFY") && proto==UDP) { $du = "sip:" + $(ruri{uri.host}) + ":" + $(ruri{uri.port}) + ";transport=tcp"; t_relay(); exit; } [...] This works well if the destination of the NOTIFY message is not behind NAT. We're looking for a solution for the NAT case. We did some tests using lookup("location") before changing $du but we did not get any success. Do you have any idea on how to implement this relaying properly ? Is it acceptable to change the $du and then execute the t_relay as we did ? Cheers Pascal

3 7

hox to a SIPtrunk with kamailio 3.0.0
by alexis heron 14 Apr '10

14 Apr '10

Hi, I would like to make a SIPTrunk with kamailio to redirect to a 9xx numbers in Cisco Call manager, can you help me please. Here's what I did but it does not work. if (uri=~"sip:9[0-9][0-9]@ccm-ipv6.crihan.fr") { log(1, "Matched Cisco Call Manager"); route(4); }; route(4) { log(1, "In route[4]"); rewritehostport("195.221.29.145.fr:5060"); append_hf("P-hint: Forwarded to CCM\r\n"); xlog("L_INFO", "$rm from $fu to $tu"); t_on_failure("2"); if (!t_relay()) { sl_reply_error(); }; exit; }

4 6

when mySQL is unreachable ...
by Alex rsm 14 Apr '10

14 Apr '10

Hi, My Kamailio server uses an external mysql database with perl script to proxy traffic. When mysql server is unreachable, Kamailio do not respond with 100 Trying and instead responses with 484 (Address Incomplete) to the INVITE message. Is this a normal behavior? I need to response with 503 when mysql is unreachable. Can I do it in kamailio.cfg or I need to implement it in the perl script? Thanks, A R _________________________________________________________________ The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multiaccount&ocid=PID28…

3 2

RTP destination address uncorrect
by xiangfeng_006 14 Apr '10

14 Apr '10

  Hi all: I am trying to setup follow IMS test environment：                          |----------------|                          |   kamailo      |-----------SIP--------->SIP server （10.100.1.12） 8700                  |                   |                                            |                                            7700 UA1-SIP/RTP-->|                   |                                            |--------------SIP/RTP---------->UA （10.100.1.51） 10.100.1.41        |                   |                                            |                          | RTP proxy   |------------RTP------->Media Gateway（10.100.1.15）                          |----------------|                             PC 10.100.1.10 SIP signaling： 1. UA1 send INVITE to PC message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.41 session name: A conversation connection information: IN IP4 10.100.1.41 2. PC send INVITE to SIP server message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.41 connection information: IN IP4 10.100.1.10 .............100/183-------------------- 3. SIP server send 200 OK to PC message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.15 connection information: IN IP4 10.100.1.15 4. PC send 200 OK to UA1 message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.15 connection information: IN IP4 10.100.1.10 after ACK, UA1 and UA2 are in calling, but no any voice I found when PC transfer the RTP packages（from UA1） to Media Gateway ，its destination address is 10.100.1.12. and not is 10.100.1.15 How to set PC (RTP proxy) transfer RTP data packages to 10.100.1.15( Media Gateway address) ? thanks !

1 0

set kamailo with RTP proxy as redirect server
by xiangfeng_006 14 Apr '10

14 Apr '10

Hi all: I am trying to setup follow IMS test environment：                          |----------------|                          |   kamailo      |-----------SIP--------->SIP server （10.100.1.12） 8700                  |                   |                                            |                                            7700 UA1-SIP/RTP-->|                   |                                            |--------------SIP/RTP---------->UA （10.100.1.51） 10.100.1.41        |                   |                                            |                          | RTP proxy   |------------RTP------->Media Gateway（10.100.1.15）                          |----------------|                             PC 10.100.1.10 SIP signaling： 1. UA1 send INVITE to PC message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.41 session name: A conversation connection information: IN IP4 10.100.1.41 2. PC send INVITE to SIP server message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.41 connection information: IN IP4 10.100.1.10 .............100/183-------------------- 3. SIP server send 200 OK to PC message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.15 connection information: IN IP4 10.100.1.15 4. PC send 200 OK to UA1 message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.15 connection information: IN IP4 10.100.1.10 after ACK, UA1 and UA2 are in calling, but no any voice I found when PC transfer the RTP packages（from UA1） to Media Gateway ，its destination address is 10.100.1.12. and not is 10.100.1.15 How to set PC (RTP proxy) transfer RTP data packages to 10.100.1.15( Media Gateway address) ? thanks !

1 0

set kamailo with RTP proxy as redirect server
by xiangfeng_006 14 Apr '10

14 Apr '10

Hi all: I am trying to setup follow IMS test environment：                          |----------------|                          |   kamailo      |-----------SIP--------->SIP server （10.100.1.12） 8700                  |                   |                                            |                                            7700 UA1-SIP/RTP-->|                   |                                            |--------------SIP/RTP---------->UA （10.100.1.51） 10.100.1.41        |                   |                                            |                          | RTP proxy   |------------RTP------->Media Gateway（10.100.1.15）                          |----------------|                             PC 10.100.1.10 SIP signaling： 1. UA1 send INVITE to PC message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.41 session name: A conversation connection information: IN IP4 10.100.1.41 2. PC send INVITE to SIP server message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.41 connection information: IN IP4 10.100.1.10 .............100/183-------------------- 3. SIP server send 200 OK to PC message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.15 connection information: IN IP4 10.100.1.15 4. PC send 200 OK to UA1 message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.15 connection information: IN IP4 10.100.1.10 after ACK, UA1 and UA2 are in calling, but no any voice I found when PC transfer the RTP packages（from UA1） to Media Gateway ，its destination address is 10.100.1.12. and not is 10.100.1.15 How to set PC (RTP proxy) transfer RTP data packages to 10.100.1.15( Media Gateway address) ? thanks !

1 0

set kamailo with RTP proxy as redirect server
by xiangfeng_006 14 Apr '10

14 Apr '10

Hi all: I am trying to setup follow IMS test environment： |----------------| | kamailo |-----------SIP--------->SIP server （10.100.1.12） 8700 | | | | 7700 UA1----SIP/RTP----> | | | |--------------SIP/RTP---------->UA （10.100.1.51） 10.100.1.41 | | | | | RTP proxy |------------RTP------->Media Gateway（10.100.1.15） |----------------| PC 10.100.1.10 SIP signaling： 1. UA1 send INVITE to PC message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.41 session name: A conversation connection information: IN IP4 10.100.1.41 2. PC send INVITE to SIP server message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.41 connection information: IN IP4 10.100.1.10 .............100/183-------------------- 3. SIP server send 200 OK to PC message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.15 connection information: IN IP4 10.100.1.15 4. PC send 200 OK to UA1 message bady ： Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.15 connection information: IN IP4 10.100.1.10 after ACK, UA1 and UA2 are in calling, but no any voice I found when PC transfer the RTP packages（from UA1） to Media Gateway ，its destination address is 10.100.1.12. and not is 10.100.1.15 How to set PC (RTP proxy) transfer RTP data packages to 10.100.1.15( Media Gateway address) ? thanks !

1 0

set kamailo with RTP proxy as redirect server
by xiangfeng_006 14 Apr '10

14 Apr '10

 Hi all:   I am trying to setup follow IMS test environment：                                     |----------------|                                     |   kamailo      |-----------SIP--------->SIP server （10.100.1.12）    8700                          |        |           |                                       |                                           7700     UA1----SIP/RTP----> |        |           |                                       |--------------SIP/RTP---------->UA （10.100.1.51）     10.100.1.41               |        |           |                                       |                                     |   RTP proxy |------------RTP------->Media Gateway（10.100.1.15）                                     |----------------|                                                                                PC 10.100.1.10                                        SIP signaling：  1.  UA1 send INVITE  to PC          message bady ：                          Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.41                          session name: A conversation                          connection  information: IN IP4 10.100.1.41  2.  PC send INVITE to SIP server          message bady ：                          Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.41                           connection  information: IN IP4 10.100.1.10 .............100/183-------------------- 3.   SIP server  send  200 OK  to PC                  message bady ：                          Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.15                                                  connection  information: IN IP4 10.100.1.15 4. PC send 200 OK to UA1                  message bady ：                          Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.15                          connection  information: IN IP4 10.100.1.10   after ACK,  UA1 and UA2 are in calling, but no any voice   I found  when  PC transfer the RTP packages（from UA1） to Media Gateway ，its destination address is 10.100.1.12. and not is 10.100.1.15            How to set PC (RTP proxy)  transfer RTP data packages  to 10.100.1.15( Media Gateway address) ?    thanks !

1 0

set kamailo with RTP proxy as redirect server
by xiangfeng_006 14 Apr '10

14 Apr '10

 Hi all:   I am trying to setup follow IMS test environment：                                     |----------------|                                     |   kamailo      |-----------SIP--------->SIP server （10.100.1.12）    8700                          |        |           |                                       |                                           7700     UA1----SIP/RTP----> |        |           |                                       |--------------SIP/RTP---------->UA （10.100.1.51）     10.100.1.41               |        |           |                                       |                                     |   RTP proxy |------------RTP------->Media Gateway（10.100.1.15）                                     |----------------|                                                                                PC 10.100.1.10                                        SIP signaling：  1.  UA1 send INVITE  to PC          message bady ：                          Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.41                          session name: A conversation                          connection  information: IN IP4 10.100.1.41  2.  PC send INVITE to SIP server          message bady ：                          Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.41                           connection  information: IN IP4 10.100.1.10 .............100/183-------------------- 3.   SIP server  send  200 OK  to PC                  message bady ：                          Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.15                                                  connection  information: IN IP4 10.100.1.15 4. PC send 200 OK to UA1                  message bady ：                          Ower/Creator ,session id:8700 123456 654321 IN IP4 10.100.1.15                          connection  information: IN IP4 10.100.1.10   after ACK,  UA1 and UA2 are in calling, but no any voice   I found  when  PC transfer the RTP packages（from UA1） to Media Gateway ，its destination address is 10.100.1.12. and not is 10.100.1.15            How to set PC (RTP proxy)  transfer RTP data packages  to 10.100.1.15( Media Gateway address) ?    thanks !

1 0

when mySQL is unreachable ...
by Robert R 13 Apr '10

13 Apr '10

Hi, My Kamailio server uses an external mysql database with perl script to proxy traffic. When mysql server is unreachable, Kamailio do not respond with 100 Trying and instead responses with 484 (Address Incomplete) to the INVITE message. Is this a normal behavior? I need to response with 503 when mysql is unreachable. Can I do it in kamailio.cfg or I need to implement it in the perl script? Thanks, R

1 0

Compiling and installing sip-router with some SER modules
by Steven C. Blair 13 Apr '10

13 Apr '10

I'd like to build the latest version of sip-router from source but include server modules from the modules_s directory. I thought the command 'make include_modules="modules_s/permissions modules_s/pike modules_s/ratelimit modules_s/sms modules_s/sanity" ' would do the trick but it doesn't. Instead all modules from the modules_s are built. Is there a way to compile and install the core + modules_s/(permissions,pike,ratelimit,sms,sanity) ? Thanks --- ISC Networking & Telecommunications 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 215-573-8396 215-898-9348 (fax)

3 2

[OT] Nagios SIP Plugin
by Iñaki Baz Castillo 13 Apr '10

13 Apr '10

Hi, I've coded a SIP plugin for Nagios. There is some other similar plugin but it's less customizable and less powerful. Homepage: http://dev.sipdoc.net/projects/sip-stuff/wiki/NagiosSIPplugin Features: - SIP UDP and TCP transport protocols. - Customizable parameters (From URI, Request URI, server address, local UDP port, timeout internal). - The replied status code can be matched against an expected value to be considered as valid. Regards. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 2

Calling number Routing possibility in 1.5rel.
by Maciej Bylica 13 Apr '10

13 Apr '10

Dear ALL, I am playing around with kamailio 1.5. Could somebody explain me what module best fit to route the call on the basis of number A (calling number). What is more all rules should be able to be modified directly from mysql db. I've run through 3.0 modules and figured out that there is drouting module which is really great, but is it present in 1.5 rel? If negative, what other modules can be used. Thx in advance, Maciej.

2 7

[Serusers] Problems Using iptel accounts
by Nahum Nir 12 Apr '10

12 Apr '10

Hi all, I have made two accounts at iptel: nahum1 and nahum2. I have two computers running X-Lite, both of them registered succesfully. But when I try to invite I get 404 Not Found. I'm dialing nahum2(a)iptel.org. What an I doing worng? Thanks, Nahum

2 1

My wish
by Alex Balashov 12 Apr '10

12 Apr '10

Allow cfg_get parameters to be used as modparams, too, so I can define one set of global database properties and pass them as db_urls to every module that uses one and also to 'sqlops' as an 'sqlcon', and similar things. -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

3 16

Close TLS connection and Registrar Server (Kamailio V1.4.x)
by edson.gomes.leme 11 Apr '10

11 Apr '10

Hello, We are using Kamailio V1.4 with TCP/TLS. The Kamailio is configured the TCP lifetime to the expire value of the REGISTER by using the tcp_persistent_flag parameter of Kamailio's registrar module. Note: The SIP proxy should not close the TCP connection as it is not capable of opening a new one. Kamailio tries to open a TLS connection to the client. The Kamailio should use the existing TLS/TCP connection that was established by the client during registration. We are applying NAT traversal techniques to instruct Kamailio to reuse the established TCP/TLS connection. Example: for each register call fix_nated_register(). For every other request and response call fix_nated_contact(). I have got one SIP phone registered via TLS to kamailio-1.4.X version with Expires = 3600. phone1: connected via ip1:port1 and Expires = 3600. Example: After 15 minutes TLS connection is closed. The SIP Phone opens a new TLS connection and registers via TLS to Kamailio. Thus, we have the following records on the Registrar Server (Kamailio): phone1: connected via ip1:port1 and Expires = 2700 [3600 – (15 min * 60 s) = 2700]. phone1: connected via ip1:port2 and Expires = 3600. Is this the desirable behavior? Shouldn’t the Kamailio (Registrar Server) delete the below registry? phone1: connected via ip1:port1 and Expires = 2700 [3600 – (15 min * 60 s) = 2700]. Best regards! Edson G. Leme

2 1

Close TLS connection and Registrar Server (Kamailio V1.4.x)
by edson.gomes.leme 11 Apr '10

11 Apr '10

1 0

users@kamailio and sr-users@sip-router merged
by Daniel-Constantin Mierla 10 Apr '10

10 Apr '10

Hello, the process of merging users(a)list.kamailio.org and sr-users(a)lists.sip-router.org is completed, offering a single place for community discussions via email related to stable versions and generic topics, also removing confusion for new comers which to use for best feedback. Because sr-users was a rather new mailing list, I imported the archive from users. It will help to continue the topics and have the entire knowledge base built over years in the same place, for web searches. The old Users mailing list archive still exists separately, link at: http://lists.kamailio.org/pipermail/users/ No new posts will get there, subscription is disabled, messages to users list are directed to sr-users (therefore replying to existing threads on either of the mailing lists work just fine). No matter you were a Users or SR-Users subscriber, updates to your profile can be done at: http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users I case your password is no longer valid, you can retrieve it via password reminder at: http://lists.sip-router.org/cgi-bin/mailman/options/sr-users Thanks, Daniel PS. Some of you may have received an unintentional "unsubscribed from sr-users-new mailing list" message, just ignore it - happened when I simulated synchronization of members on temporary name sr-users-new, no unsubscribe action was actually done -- sorry for inconvenience. -- Daniel-Constantin Mierla

1 0

[Kamailio-Users] merging users mailing list
by Daniel-Constantin Mierla 10 Apr '10

10 Apr '10

Hello, I am starting the merging process of users mailing list. Please ignore any notifications you get, if you get them is by mistake in my config -- none of subscribed users will be affected, the work is done for testing on a temporary new list, therefore everyone is safe. Sorry for any inconvenience! I will announce when it is finished. Daniel

1 0

[Kamailio-Users] dispatcher has problems
by Vic Jolin 09 Apr '10

09 Apr '10

I just noticed that if I have 4 servers set like this # group sip addresses of your * units 1 sip:XXX.XXX.XXX.XXX:5060 1 sip:XXX.XXX.XXX.XXX:5060 1 sip:XXX.XXX.XXX.XXX:5060 1 sip:XXX.XXX.XXX.XXX:5060 the registration does not work and I get 401 unauthorized. My setup is like this Openser as proxy sends all SIP messages to load balanced asterisk servers So sip phones should be able to register to the asterisk server The routing part in kamailio.cfg looks like this if ( method=="REGISTER" || method=="NOTIFY" || method=="OPTIONS" || method=="ACK" || method=="MESSAGE") { fix_nated_contact(); fix_nated_register(); ds_select_dst("1","4"); forward();#uri:host, uri:port); exit(); } if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } else if (method=="INVITE"){ log("VCTOR: Got an invite\n"); fix_nated_contact(); force_rtp_proxy(); ds_select_dst("1","4"); forward();#uri:host, uri:port); t_on_failure("1"); }; search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); if (!is_method("INVITE|REGISTER|MESSAGE|BYE|NOTIFY|ACK|CANCEL|REFER|SUBSCRIBE|OPTIONS|INFO")) { xlog("L_INFO", "=== [$rm] Method Not Implement ===========\n"); sl_send_reply("501", "Not implemented here"); return; }; any thoughts?

2 2

[Kamailio-Users] lcr module creating extra branch
by Jon Bonilla 09 Apr '10

09 Apr '10

Hi I am using Kamailio 3.0.1 and experiencing a strange behaviour of the lcr module. In short: Using next_gw() function makes my server to send two INVITEs to different destinations! The INVITE Kamailio receives is like: R=sip:4644235465675757@77.77.77.77 F=sip:manwe2@77.77.77.77 T=sip:4644235465675757@77.77.77.77 To try this I have set this in my gw table: sercmd lcr.dump_gws { lcr_id: 1 grp_id: 17 ip_addr: 91.121.117.27 hostname: sindominio.net port: 5080 scheme: sip transport: UDP strip: 0 tag: weight: 1 flags: 37 defunct_until: 268313248 } sercmd lcr.dump_lcrs { lcr_id: 1 prefix: from_uri: grp_id: 17 priority: 1 } Hostname and ip_addr don't match (sindominio.net points to 82.144.4.26). What I capture leaving Kamailio is: U 2010/04/08 17:51:09.614055 77.77.77.77:5060 -> 91.121.117.27:5080 INVITE sip:+4644235465675757@77.77.77.77 SIP/2.0' Record-Route: <sip:77.77.77.77;lr=on;ftag=hgtcn;nat=yes;vsf=czFwOgs5JCAEJEYoJGk3ajxNYmBrPBYC>' Via: SIP/2.0/UDP 77.77.77.77;branch=z9hG4bK90e8.cd3135e1.0' Via: SIP/2.0/UDP 10.0.0.92;received=91.115.172.34;rport=53658;branch=z9hG4bKiwibmejv' Max-Forwards: 69' To: <sip:4644235465675757@77.77.77.77>' From: "1102" <sip:1102@77.77.77.77>;tag=hgtcn' Call-ID: qxvyddwlnmurqvi@multivac' CSeq: 571 INVITE' Contact: <sip:manwe2@91.115.172.34:53658>' Content-Type: application/sdp' Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE' Supported: replaces,norefersub,100rel' User-Agent: Twinkle/1.4.2' Content-Length: 226' ' v=0' o=twinkle 1296734594 70729915 IN IP4 77.77.77.77' s=-' c=IN IP4 77.77.77.77' t=0 0' m=audio 38232 RTP/AVP 8 101' a=rtpmap:8 PCMA/8000' a=rtpmap:101 telephone-event/8000' a=fmtp:101 0-15' a=ptime:20' a=nortpproxy:yes' # U 2010/04/08 17:51:09.614074 77.77.77.77:5060 -> 82.144.4.26:5060 INVITE sip:+4644235465675757@sindominio.net SIP/2.0' Record-Route: <sip:77.77.77.77;lr=on;ftag=hgtcn;nat=yes;vsf=czFwOgs5JCAEJEYoJGk3ajxNYmBrPBYC>' Via: SIP/2.0/UDP 77.77.77.77;branch=z9hG4bK90e8.cd3135e1.1' Via: SIP/2.0/UDP 10.0.0.92;received=91.115.172.34;rport=53658;branch=z9hG4bKiwibmejv' Max-Forwards: 69' To: <sip:4644235465675757@77.77.77.77>' From: "1102" <sip:1102@77.77.77.77>;tag=hgtcn' Call-ID: qxvyddwlnmurqvi@multivac' CSeq: 571 INVITE' Contact: <sip:manwe2@91.115.172.34:53658>' Content-Type: application/sdp' Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE' Supported: replaces,norefersub,100rel' User-Agent: Twinkle/1.4.2' Content-Length: 226' ' v=0' o=twinkle 1296734594 70729915 IN IP4 77.77.77.77' s=-' c=IN IP4 77.77.77.77' t=0 0' m=audio 38232 RTP/AVP 8 101' a=rtpmap:8 PCMA/8000' a=rtpmap:101 telephone-event/8000' a=fmtp:101 0-15' a=ptime:20' a=nortpproxy:yes' My kamailio.cfg script does something like this: if(!load_gws("1", "$ru")) ... if(!next_gw()) ... t_on_branch("BRANCH_ROUTE_CLI_RTP"); if(!t_relay_to("0x01")) ... Branch route_cli_rtp: do rtpproxy stuff No append_branch is executed at all. I have two INVITES IMHO because next_gw() function does create the second one. Any idea about this behaviour?

3 10

[Kamailio-Users] sr-docs mailing list
by Daniel-Constantin Mierla 09 Apr '10

09 Apr '10

Hello, I renamed kamailio-docs(a)lists.kamailio.org to sr-docs(a)lists.sip-router.org. Info about it at: http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-docs Lately, several discussions touched the documentation aspects, in a need for a better structuring and updates for the new context of sip router project. The old mailing list was active for a while, but with the start of sip router project was ignored because of more focus on integration work. I hope some of you will join, help to build new documentation and spot mistakes in existing versions. Volunteers that want to lead various documentation efforts are more than welcome! Just say what you need... Thanks, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla

1 0

[Kamailio-Users] Kamailio Blocking
by dotnetdub 09 Apr '10

09 Apr '10

Hello List, I have been running a very stable Kamailio 1.4 install for over a year now with no downtime. From time to time I get a message from the OS telling me that task kamailio:xxxx blocked for more than 120 seconds and a dump into syslog. I recently setup Kamailio 3.01 and it's been running for around 3 weeks now. Purely testing at the moment, there is only 1 SIP registration and almost no activity most of the time. I was checking dmesg and: [269844.208011] INFO: task kamailio:28504 blocked for more than 120 seconds. [269844.208093] "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. [269844.208178] kamailio D f6d77bf8 0 28504 28484 [269844.208180] f771e140 00000082 00000002 f6d77bf8 f7091390 f771e2cc c4019020 00000000 [269844.208184] 00000000 011615c9 f7091398 00000282 00000000 00000000 00000000 ffffffff [269844.208188] f5cba384 f5cba38c f5cba388 f771e140 c02c91ec f6969d68 f5cba38c f771e140 [269844.208191] Call Trace: [269844.208203] [<c02c91ec>] __mutex_lock_slowpath+0x50/0x7b [269844.208208] [<c02c9082>] mutex_lock+0xa/0xb [269844.208210] [<c02b7d4d>] unix_dgram_recvmsg+0x3e/0x231 [269844.208214] [<c0161beb>] get_page_from_freelist+0xc1/0x3e9 [269844.208218] [<c0161045>] __rmqueue_smallest+0x83/0xe3 [269844.208221] [<c025f771>] sock_recvmsg+0xde/0xf9 [269844.208226] [<c01387b0>] autoremove_wake_function+0x0/0x2d [269844.208231] [<c01620e9>] __alloc_pages_internal+0xb5/0x34e [269844.208236] [<c0260595>] sys_recvfrom+0xb4/0x116 [269844.208239] [<c011af13>] do_page_fault+0x0/0x8f9 [269844.208242] [<c0108540>] do_notify_resume+0x5f8/0x649 [269844.208247] [<c011b3c5>] do_page_fault+0x4b2/0x8f9 [269844.208250] [<c0260a48>] sys_socketcall+0x135/0x19e [269844.208254] [<c0108853>] sysenter_past_esp+0x78/0xb1 [269844.208257] [<c02c0000>] xenfb_probe+0xd1/0x35b [269844.208261] ======================= [269844.208262] INFO: task kamailio:28506 blocked for more than 120 seconds. [269844.208314] "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. [269844.208395] kamailio D d0e00947 0 28506 28484 [269844.208397] f77f8e60 00000082 00000002 d0e00947 00004292 f77f8fec c4024020 00000001 [269844.208401] 00000000 00000001 00000002 00000000 f7091398 00000001 00000001 c01211f2 [269844.208404] f5cba384 f5cba38c f5cba388 f77f8e60 c02c91ec f5cba38c f60add68 f77f8e60 [269844.208408] Call Trace: [269844.208410] [<c01211f2>] __wake_up_sync+0x2a/0x3e [269844.208413] [<c02c91ec>] __mutex_lock_slowpath+0x50/0x7b [269844.208416] [<c02c9082>] mutex_lock+0xa/0xb [269844.208418] [<c02b7d4d>] unix_dgram_recvmsg+0x3e/0x231 [269844.208421] [<c0161beb>] get_page_from_freelist+0xc1/0x3e9 [269844.208424] [<c0161045>] __rmqueue_smallest+0x83/0xe3 [269844.208427] [<c025f771>] sock_recvmsg+0xde/0xf9 [269844.208431] [<c01387b0>] autoremove_wake_function+0x0/0x2d [269844.208435] [<c01620e9>] __alloc_pages_internal+0xb5/0x34e [269844.208440] [<c0260595>] sys_recvfrom+0xb4/0x116 [269844.208444] [<c0180d0d>] cp_new_stat64+0xfc/0x10e [269844.208449] [<c011b3c5>] do_page_fault+0x4b2/0x8f9 [269844.208452] [<c0260a48>] sys_socketcall+0x135/0x19e [269844.208456] [<c0108853>] sysenter_past_esp+0x78/0xb1 [269844.208459] [<c02c0000>] xenfb_probe+0xd1/0x35b [269844.208462] ======================= [274198.229285] INFO: task kamailio:28504 blocked for more than 120 seconds. [274198.229340] "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. [274198.229422] kamailio D f6d77bf8 0 28504 28484 [274198.229425] f771e140 00000082 00000002 f6d77bf8 f7091390 f771e2cc c4019020 00000000 [274198.229429] 00000000 011615c9 f7091398 00000282 00000000 00000000 00000000 ffffffff [274198.229432] f5cba384 f5cba38c f5cba388 f771e140 c02c91ec f6969d68 f5cba38c f771e140 [274198.229436] Call Trace: [274198.229447] [<c02c91ec>] __mutex_lock_slowpath+0x50/0x7b [274198.229453] [<c02c9082>] mutex_lock+0xa/0xb [274198.229455] [<c02b7d4d>] unix_dgram_recvmsg+0x3e/0x231 [274198.229459] [<c0161beb>] get_page_from_freelist+0xc1/0x3e9 [274198.229463] [<c0161045>] __rmqueue_smallest+0x83/0xe3 [274198.229466] [<c025f771>] sock_recvmsg+0xde/0xf9 [274198.229471] [<c01387b0>] autoremove_wake_function+0x0/0x2d [274198.229476] [<c01620e9>] __alloc_pages_internal+0xb5/0x34e [274198.229481] [<c0260595>] sys_recvfrom+0xb4/0x116 [274198.229484] [<c011af13>] do_page_fault+0x0/0x8f9 [274198.229487] [<c0108540>] do_notify_resume+0x5f8/0x649 [274198.229492] [<c011b3c5>] do_page_fault+0x4b2/0x8f9 [274198.229495] [<c0260a48>] sys_socketcall+0x135/0x19e [274198.229499] [<c0108853>] sysenter_past_esp+0x78/0xb1 [274198.229502] [<c02c0000>] xenfb_probe+0xd1/0x35b [274198.229506] ======================= [274198.229507] INFO: task kamailio:28506 blocked for more than 120 seconds. [274198.229559] "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. [274198.229641] kamailio D d0e00947 0 28506 28484 [274198.229643] f77f8e60 00000082 00000002 d0e00947 00004292 f77f8fec c4024020 00000001 [274198.229646] 00000000 00000001 00000002 00000000 f7091398 00000001 00000001 c01211f2 [274198.229650] f5cba384 f5cba38c f5cba388 f77f8e60 c02c91ec f5cba38c f60add68 f77f8e60 [274198.229653] Call Trace: [274198.229656] [<c01211f2>] __wake_up_sync+0x2a/0x3e [274198.229659] [<c02c91ec>] __mutex_lock_slowpath+0x50/0x7b [274198.229662] [<c02c9082>] mutex_lock+0xa/0xb [274198.229664] [<c02b7d4d>] unix_dgram_recvmsg+0x3e/0x231 [274198.229667] [<c0161beb>] get_page_from_freelist+0xc1/0x3e9 [274198.229670] [<c0161045>] __rmqueue_smallest+0x83/0xe3 [274198.229673] [<c025f771>] sock_recvmsg+0xde/0xf9 [274198.229677] [<c01387b0>] autoremove_wake_function+0x0/0x2d [274198.229681] [<c01620e9>] __alloc_pages_internal+0xb5/0x34e [274198.229686] [<c0260595>] sys_recvfrom+0xb4/0x116 [274198.229690] [<c0180d0d>] cp_new_stat64+0xfc/0x10e [274198.229696] [<c011b3c5>] do_page_fault+0x4b2/0x8f9 [274198.229699] [<c0260a48>] sys_socketcall+0x135/0x19e [274198.229703] [<c0108853>] sysenter_past_esp+0x78/0xb1 [274198.229706] [<c02c0000>] xenfb_probe+0xd1/0x35b There was no activity when this happened. ON the 1.4 box there could be about 10 sessions setup when it happens. kamctl ps: Process:: ID=14 PID=28504 Type=MI DATAGRAM Process:: ID=15 PID=28505 Type=MI DATAGRAM Process:: ID=16 PID=28506 Type=MI DATAGRAM Even on the old install this doesn't seem to cause any problem and same here on 3.01 but would like to try and solve it. Any idea? Regards, Stephen

4 5

[Kamailio-Users] fr_inv_timer in kam-3.0
by Andreas Granig 09 Apr '10

09 Apr '10

Hi, In the docs at http://kamailio.org/docs/modules/3.0.x/modules/tm.html#fr_inv_timer it says that fr_inv_timer_avp should be defined like this, without $ or $avp: modparam("tm", "fr_inv_timer_avp", "my_fr_inv_timer") In kam <= 1.5 I did it like that: modparam("tm", "fr_inv_timer_avp", "$avp(s:callee_fr_inv_timer)") which doesn't give me an error on startup with kam-3.0 either, but the timer doesn't get fired (I use seconds for that as noted in the docs). If I change it to modparam("tm", "fr_inv_timer_avp", "callee_fr_inv_timer") then I get the error "malformed or non AVP callee_fr_inv_timer AVP definition", same with setting it to "s:callee_fr_inv_timer". Anyhow, it seems to be deprecated anyways, so I'm looking to get t_set_fr() working. I'm just curious how I can use a var or AVP loaded from DB to set the value on-the-fly? t_set_fr(...) seems to allow only constants to be set. Couldn't find anything in the docs regarding that one. Thanks, Andreas

5 14

[Kamailio-Users] drop in branch/onsend-route in kam-3.0
by Andreas Granig 08 Apr '10

08 Apr '10

Hi, Looks like behavior of drop has changed in Kamailio 3.0. Before, I was able to drop() a specific branch in branch_route depending on various checks. http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.0.x#drop still says this is possible, http://kamailio.org/docs/modules/3.0.x/modules/tm.html#t_on_branch says it's not possible. So calling drop() in branch_route has no effect, and I have no idea how to mark a branch in branch_route to be dropped in onsend_route, because checking a bflag doesn't work there, and neither does checking $rU. Any ideas on how to accomplish dropping a specific branch? Andreas

5 7

[Kamailio-Users] Understanding memory-leaks by inspecting PKG/SHM memory status at shutdown
by Iñaki Baz Castillo 08 Apr '10

08 Apr '10

Hi, according to the wiki page: http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:memory In order to detect a possible memory leak (in PGK or SHM) a good way is enabling memory debugging and shuting down the server, so: "the memory manager will dump the memory status. Normally most of the memory is cleaned during shutdown. If there is memory leak, it should be visible as not-freed memory." But I would like to know how exactly to detect if there is a memoty leak by inspecting the SHM/PKG status, something like this: ----------------------------------------------------------------------------------------------------------------------------- [32715]: Memory status (pkg): [32715]: fm_status (0x701a40): [32715]: heap size= 16777216 [32715]: used= 189696, used+overhead=247392, free=16529824 [32715]: max used (+overhead)= 247392 [32715]: dumping free list: [32715]: hash = 1 fragments no.: 1, unused: 0#012#011#011 bucket size: 8 - 8 (first 8) [32715]: hash = 2058 fragments no.: 1, unused: 0#012#011#011 bucket size: 8388608 - 16777216 (first 16530136) [32715]: TOTAL: 2 free fragments = 16530144 free bytes [32678]: Memory status (shm): [32678]: fm_status (0x7fa57921c000): [32678]: heap size= 67108864 [32678]: used= 16, used+overhead=35872, free=67072992 [32678]: max used (+overhead)= 2821064 [32678]: dumping free list: [32678]: hash = 1 fragments no.: 45, unused: 0#012#011#011 bucket size: 8 - 8 (first 8) [32678]: hash = 2 fragments no.: 4, unused: 0#012#011#011 bucket size: 16 - 16 (first 16) [32678]: hash = 3 fragments no.: 16, unused: 0#012#011#011 bucket size: 24 - 24 (first 24) [32678]: hash = 4 fragments no.: 2, unused: 0#012#011#011 bucket size: 32 - 32 (first 32) [32678]: hash = 5 fragments no.: 1, unused: 0#012#011#011 bucket size: 40 - 40 (first 40) [32678]: hash = 6 fragments no.: 1, unused: 0#012#011#011 bucket size: 48 - 48 (first 48) [32678]: hash = 7 fragments no.: 39, unused: 0#012#011#011 bucket size: 56 - 56 (first 56) [32678]: hash = 8 fragments no.: 1, unused: 0#012#011#011 bucket size: 64 - 64 (first 64) [32678]: hash = 9 fragments no.: 1, unused: 0#012#011#011 bucket size: 72 - 72 (first 72) [32678]: hash = 10 fragments no.: 2, unused: 0#012#011#011 bucket size: 80 - 80 (first 80) [32678]: hash = 15 fragments no.: 1, unused: 0#012#011#011 bucket size: 120 - 120 (first 120) [32678]: hash = 25 fragments no.: 1, unused: 0#012#011#011 bucket size: 200 - 200 (first 200) [32678]: hash = 40 fragments no.: 5, unused: 0#012#011#011 bucket size: 320 - 320 (first 320) [32678]: hash = 112 fragments no.: 1, unused: 0#012#011#011 bucket size: 896 - 896 (first 896) [32678]: hash = 128 fragments no.: 4, unused: 0#012#011#011 bucket size: 1024 - 1024 (first 1024) [32678]: hash = 129 fragments no.: 2, unused: 0#012#011#011 bucket size: 1032 - 1032 (first 1032) [32678]: hash = 224 fragments no.: 1, unused: 0#012#011#011 bucket size: 1792 - 1792 (first 1792) [32678]: hash = 258 fragments no.: 3, unused: 0#012#011#011 bucket size: 2064 - 2064 (first 2064) [32678]: hash = 473 fragments no.: 1, unused: 0#012#011#011 bucket size: 3784 - 3784 (first 3784) [32678]: hash = 1026 fragments no.: 1, unused: 0#012#011#011 bucket size: 8208 - 8208 (first 8208) [32678]: hash = 2049 fragments no.: 2, unused: 0#012#011#011 bucket size: 16384 - 32768 (first 16512) [32678]: hash = 2051 fragments no.: 1, unused: 0#012#011#011 bucket size: 65536 - 131072 (first 98336) [32678]: hash = 2056 fragments no.: 1, unused: 0#012#011#011 bucket size: 2097152 - 4194304 (first 2621440) [32678]: hash = 2060 fragments no.: 1, unused: 0#012#011#011 bucket size: 33554432 - 67108864 (first 64287800) [32678]: TOTAL: 137 free fragments = 67072992 free bytes ----------------------------------------------------------------------------------------------------------------------------- This is, how should look this SHM/PKG status example (after shutdown) in case there is a memory leak? how to detect not-freed memory? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 5

[Kamailio-Users] irc devel meeting
by Daniel-Constantin Mierla 08 Apr '10

08 Apr '10

Hello, I know some of devels are still in Easter vacation, but I hope we can get together soon on irc to sketch the plan for 3.1 release. I propose next week, Wednesday, April 14, 15:00UTC, on #sip-router channel hosted by irc.freenode.net If you are a developer and want to participate, please announce. If it is not possible to attend for major contributors, we can look for another date (eventually start a pool to see the day were most of us are available). Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla

4 4

[Serusers] ReTransmission of CANCELs on 0.9.7
by rupert.organ＠bt.com 08 Apr '10

08 Apr '10

Hi All, Has anyone come up against an issue on 0.9.6 / 0.9.7 whereby a SIP App Server, delivers a CANCEL to the SER. The SER proxies the CANCELs to all the forked endpoints. At the same time it sends "200 Cancelling" back to the SIP App server...(this suppresses CANCEL retransmission from the SIP App server). If one of the CANCELs from the SER to the SIP UA gets lost....the phone keeps ringing. There is no CANCEL retransmission on the SER !!! There is CANCEL retransmission on the SIP App, but when it sends SER replies "no pending branches" Ideally want the SER do CANCEL retransmissions. I understand the timers were rewritten in 0.10.x but this is not hardened version like 0.9.6....any suggestions? Thanks in advance Rupert

4 3

[Kamailio-Users] merging users mailing lists
by Daniel-Constantin Mierla 08 Apr '10

08 Apr '10

Hello, I propose to merge the users mailing lists, most of the traffic these days is about 3.0 and even there are 2 stables branches now, they are sync'ed, so same code more or less. For 3.1 will be one stable branch, falvor selection will be only a matter of make command. Lately common useful topics are discussed on those different mailing lists, notifications and knowledge base building require cross-posting, lot of overhead imo. Any other opinion? Like with devel mailing lists, existing email addresses for users ML can still be used, just that end on same ML. Natural choice will be to have users@kamailio and serusers@iptel to be directed to sr-users(a)lists.sip-router.org Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

6 8

[Kamailio-Users] mc (Midnight Commander) syntax highlighting scheme
by Константин 07 Apr '10

07 Apr '10

Hi all! For easy editing kamailio configuration files I developed a syntax highlighting scheme for mc. Preview The scheme includes basic syntactic structures of Kamailio 1.5, all constants, variables, operators and standart core functions. To install it, download the file kamailio.syntax from attachment and copy it to /usr/share/mc/syntax (for Debian Etch) Next, add to the file /usr/share/mc/syntax/Syntax including: file ..\*\\.cfg$ Kamailio\sconfig include kamailio.syntax Thats all, I hope its helps someone ;) -- Shpinev Konstantin mailto:kks_mail@inbox.ru

3 2

[Kamailio-Users] Version 3
by dotnetdub 07 Apr '10

07 Apr '10

Hi List, Anybody using this in production yet? If so what kind of volume and how is reliability? Looking to move to this platform, looks very good, interested to hear some experiences. Thanks, Stephen

5 6

Re: [Kamailio-Users] Odd Behaviour with Kamailio + TLS
by Hemanshu Patel 02 Apr '10

02 Apr '10

Yes, i had installed self signed certificates via IE. Bria shows it and validates them during registration process, its the call initiation where it creates problem. Kamailio starts without fix_nated_register. With this function it gives configuration file error. and i had load nathelper.so and that is been confirmed via logs -- Regards, Hemanshu Patel SIS,Ahmedabad. M: 09601295238 > Just checking... > > Some time ago, I made some tests with Bria and self signed certs... It was a little tricky to find that I had to install the cert through IE. Did you install them this way? Can you check Bria's log to ensure that the cert is accepted? > > Edson. > > Henning Westerholt escreveu: >> On Thursday 01 April 2010, Hemanshu Patel wrote: >>> yes i did, and log shows that nathelper.so load without any problem. kamailio loads and work perfectly with grandstream devices with TLS and >>> SRTp. it gives problem with Bria softphones only. >>> I am using self signed certificates which works perfectly with grandstream, can there be any chances that due to self sign ness of the >>> certificates, it doesnt work well with bria devices. >> Hi Hemanshu, >> ok, so the error you reported in your previous message is solved. It could be >> possible that the Bria softphone implement a more strict policy with regards >> to the check of the certificates. But you should be able to find something in >> the UA documentation about this, you could ask the vendor or other users. As >> the last resort just test with certificates that are not self signed. ;-) >> Cheers, >> Henning >> _______________________________________________ >> Kamailio (OpenSER) - Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >

2 2

[Kamailio-Users] Odd Behaviour with Kamailio + TLS
by Hemanshu Patel 01 Apr '10

01 Apr '10

Hi all, I am not sure it is strange behavior or its the desirable one. i have got two phones registered via TLS to kamailio-1.5.X version. phone1: connected via ip1:port1 and having contact value ip1:port2 phone2: connected via ip2:port3 and having contact value ip2:port4 when i have just this two phones i can see 2 TCP connected to my kamailio from both phones from ip1:port1 and ip2:port3. now when i call from phone1 to phone2, rather then using the already established and TLS authenticated connection kamailio creates two new connection from kamailio to ip1:port2 and ip2:port4. mean kamailio creates new connection to the contact value stored in location table of openser DB. Is this the desirable behavior? shouldn't kamailio use the alreay established connection? -- Regards, Hemanshu Patel SIS,Ahmedabad. M: 09601295238

4 6

SEMS 1.2.0 released
by Stefan Sayer 01 Apr '10

01 Apr '10

Hello, it is my pleasure to announce that SEMS 1.2.0 has been released. This release again brings many improvements to this fine, free SIP Media and Application server. Get the source at http://ftp.iptel.org/pub/sems/sems-1.2.0.tar.gz Some debian packages at http://ftp.iptel.org/pub/sems/1.2/1.2.0/packages/ Changelog: http://ftp.iptel.org/pub/sems/doc/current/changelog.html Documentation got some new getting started tutorials: http://ftp.iptel.org/pub/sems/doc/current/index.html And there is now an experimental lenny32 vmware image available, configured as webconference server (but usable for all applications with minimal reconfiguration): https://developer.berlios.de/project/shownotes.php?release_id=17393 For more information see the home page at http://iptel.org/sems . Among the most interesting news I would like to emphasize that its powerful state machine scripting engine DSM has matured a lot between 1.1 and 1.2 versions. Thanks to its growing number of modules (including MySQL DB, conferencing, Python and AWS), live script reloading and monitoring, its interfacing to external applications, and its extended script language it has become a good service development and deployment platform. Other interesting additions is a CMake based build system, an in-memory DB 'monitoring' for administrative and applications use, a twitter app, and numerous fixes and improvements all around. Respect and many Thanks go to all contributors and sponsors of the SEMS project. With Best Regards Stefan Sayer P.S. Apologies if you have received multiple copies of this message. -- Stefan Sayer VoIP Services Consulting and Development Warschauer Str. 24 10243 Berlin tel:+491621366449 sip:sayer@iptel.org email/xmpp:stefan.sayer@gmail.com

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[Serusers] Registration process : 401 then 403, is it possible ?
by inge 01 Apr '10

01 Apr '10

Hi, I'm looking for a workarround to avoid a buggy CPE. The CPE doesn't stop to try to register after he got a 401 Unauthorized. So this produce a flooding. That's why I'm looking for something to send a first 401 and then a 403 if the registration keep failing. Thanks in advance for your idea. Regards, Adrien

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Re: [Kamailio-Users] Users Digest, Vol 58, Issue 76
by Rothe, Marcus, D22-WHV 31 Mar '10

31 Mar '10

"users-request(a)lists.kamailio.org" <users-request(a)lists.kamailio.org> schrieb: Send Users mailing list submissions to users(a)lists.kamailio.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.kamailio.org/cgi-bin/mailman/listinfo/users or, via email, send a message with subject or body 'help' to users-request(a)lists.kamailio.org You can reach the person managing the list at users-owner(a)lists.kamailio.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Users digest..." Today's Topics: 1. Re: Odd Behaviour with Kamailio + TLS (Klaus Darilion) 2. Re: Registrar: Save modified contact / set the q value (Daniel-Constantin Mierla) 3. Re: more type conversion wonders (Daniel-Constantin Mierla) ---------------------------------------------------------------------- Message: 1 Date: Wed, 31 Mar 2010 11:34:45 +0200 From: Klaus Darilion <klaus.mailinglists(a)pernau.at> Subject: Re: [Kamailio-Users] Odd Behaviour with Kamailio + TLS To: Hemanshu Patel <hemanshu.patel(a)saicare.com> Cc: users(a)lists.kamailio.org Message-ID: <4BB31735.6030500(a)pernau.at> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Am 31.03.2010 08:56, schrieb Hemanshu Patel: > > Hi all, > > I am not sure it is strange behavior or its the desirable one. > > i have got two phones registered via TLS to kamailio-1.5.X version. > > phone1: connected via ip1:port1 and having contact value ip1:port2 > phone2: connected via ip2:port3 and having contact value ip2:port4 > > when i have just this two phones i can see 2 TCP connected to my kamailio > from both phones from ip1:port1 and ip2:port3. > > now when i call from phone1 to phone2, rather then using the already > established and TLS authenticated connection kamailio creates two new > connection from kamailio to ip1:port2 and ip2:port4. > mean kamailio creates new connection to the contact value stored in > location table of openser DB. > > Is this the desirable behavior? shouldn't kamailio use the alreay > established connection? This is the standard Kamailio behavior. Thus, you should apply NAT traversal techniques to instruct Kamailio to reuse the established connection. In short: for each register call fix_nated_register(). For every other request and response call fix_nated_contact(). Thus, if you use the default configuration and apply NAT traversal it should work. regards klaus ------------------------------ Message: 2 Date: Wed, 31 Mar 2010 11:37:27 +0200 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [Kamailio-Users] Registrar: Save modified contact / set the q value To: Ovidiu Sas <osas(a)voipembedded.com> Cc: NeoTel Lists <MailingLists(a)neotel.at>, users(a)lists.kamailio.org Message-ID: <4BB317D7.4030102(a)gmail.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 3/30/10 8:29 PM, Ovidiu Sas wrote: > If you need this in 1.5, try to fix the contact header and then > forward the REGISTER back to yourself. > The loopback REGISTER should have the fixed Contact header and now you > can call save(). > You will need to tweak a a little bit the config to get everything > right into the usrloc (the loopback REGISTER will come from the server > IP) and it might work. > it works for no nat scenario, otherwise make sure you will get the looped message on the same socket and you add before looping a header with source ip and port (corresponding nat box pinhole) that you store in received avp. Cheers, Daniel > Regards, > Ovidiu Sas > > On Tue, Mar 30, 2010 at 11:36 AM, NeoTel Lists<MailingLists(a)neotel.at> wrote: > >> Hello Everybody! >> >> Is there any way to save(domain, 0x02) the contact after it has been changed >> somehow in Kamailio<= 1.5? >> Do I have to upgrade to 3.0 and use msg_apply_changes() before save()? >> >> Or: Can I set the to be saved q value somehow? >> >> br >> Walter >> >> _______________________________________________ >> Kamailio (OpenSER) - Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> >> > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ------------------------------ Message: 3 Date: Wed, 31 Mar 2010 11:41:29 +0200 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [Kamailio-Users] more type conversion wonders To: Juha Heinanen <jh(a)tutpro.com> Cc: users(a)lists.sip-router.org Message-ID: <4BB318C9.1080302(a)gmail.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 3/31/10 8:40 AM, Juha Heinanen wrote: > i played a bit more with selects and found that statement > > if ($sht(auth=>foo::count)> @cfg_get.local.gw_timeout) { > xlog("foo"); > } > I guess selects have types, while pseudo-variables are kind of type agnostic. Any PV has a string representation of the value, comparison is done as integer if both PVs are integers, otherwise is done as string. The safe side is to use PV representation of selects when comparing with another PV. PV and selects is another point for the future coherence. Cheers, Daniel > produces error > > 0(9213) :<core> [cfg.y:3379]: parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 507, column 28-55: bad expression: type mismatch: str instead of int at (507,55) > > but the error goes away if i either make explicit conversion > > if ($sht(auth=>foo::count)> (int)(a)cfg_get.local.gw_timeout) { > xlog("foo"); > } > > or use $sel > > if ($sht(auth=>foo::count)> $sel(cfg_get.local.gw_timeout)) { > xlog("foo"); > } > > this is thus exactly opposite than in my t_set_fr ordeal where i got > conversion error when i used $sel instead of @. > > in my opinion, no (int) conversion should be needed in the firs case, > because the value of the cfg variable is int. > > in summary, writing statements that include selects is very difficult > and error prone. > > -- juha > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ------------------------------ _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users End of Users Digest, Vol 58, Issue 76 *************************************

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Re: [Kamailio-Users] Users Digest, Vol 58, Issue 76
by Rothe, Marcus, D22-WHV 31 Mar '10

31 Mar '10

"users-request(a)lists.kamailio.org" <users-request(a)lists.kamailio.org> schrieb: Send Users mailing list submissions to users(a)lists.kamailio.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.kamailio.org/cgi-bin/mailman/listinfo/users or, via email, send a message with subject or body 'help' to users-request(a)lists.kamailio.org You can reach the person managing the list at users-owner(a)lists.kamailio.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Users digest..." Today's Topics: 1. Re: Odd Behaviour with Kamailio + TLS (Klaus Darilion) 2. Re: Registrar: Save modified contact / set the q value (Daniel-Constantin Mierla) 3. Re: more type conversion wonders (Daniel-Constantin Mierla) ---------------------------------------------------------------------- Message: 1 Date: Wed, 31 Mar 2010 11:34:45 +0200 From: Klaus Darilion <klaus.mailinglists(a)pernau.at> Subject: Re: [Kamailio-Users] Odd Behaviour with Kamailio + TLS To: Hemanshu Patel <hemanshu.patel(a)saicare.com> Cc: users(a)lists.kamailio.org Message-ID: <4BB31735.6030500(a)pernau.at> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Am 31.03.2010 08:56, schrieb Hemanshu Patel: > > Hi all, > > I am not sure it is strange behavior or its the desirable one. > > i have got two phones registered via TLS to kamailio-1.5.X version. > > phone1: connected via ip1:port1 and having contact value ip1:port2 > phone2: connected via ip2:port3 and having contact value ip2:port4 > > when i have just this two phones i can see 2 TCP connected to my kamailio > from both phones from ip1:port1 and ip2:port3. > > now when i call from phone1 to phone2, rather then using the already > established and TLS authenticated connection kamailio creates two new > connection from kamailio to ip1:port2 and ip2:port4. > mean kamailio creates new connection to the contact value stored in > location table of openser DB. > > Is this the desirable behavior? shouldn't kamailio use the alreay > established connection? This is the standard Kamailio behavior. Thus, you should apply NAT traversal techniques to instruct Kamailio to reuse the established connection. In short: for each register call fix_nated_register(). For every other request and response call fix_nated_contact(). Thus, if you use the default configuration and apply NAT traversal it should work. regards klaus ------------------------------ Message: 2 Date: Wed, 31 Mar 2010 11:37:27 +0200 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [Kamailio-Users] Registrar: Save modified contact / set the q value To: Ovidiu Sas <osas(a)voipembedded.com> Cc: NeoTel Lists <MailingLists(a)neotel.at>, users(a)lists.kamailio.org Message-ID: <4BB317D7.4030102(a)gmail.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 3/30/10 8:29 PM, Ovidiu Sas wrote: > If you need this in 1.5, try to fix the contact header and then > forward the REGISTER back to yourself. > The loopback REGISTER should have the fixed Contact header and now you > can call save(). > You will need to tweak a a little bit the config to get everything > right into the usrloc (the loopback REGISTER will come from the server > IP) and it might work. > it works for no nat scenario, otherwise make sure you will get the looped message on the same socket and you add before looping a header with source ip and port (corresponding nat box pinhole) that you store in received avp. Cheers, Daniel > Regards, > Ovidiu Sas > > On Tue, Mar 30, 2010 at 11:36 AM, NeoTel Lists<MailingLists(a)neotel.at> wrote: > >> Hello Everybody! >> >> Is there any way to save(domain, 0x02) the contact after it has been changed >> somehow in Kamailio<= 1.5? >> Do I have to upgrade to 3.0 and use msg_apply_changes() before save()? >> >> Or: Can I set the to be saved q value somehow? >> >> br >> Walter >> >> _______________________________________________ >> Kamailio (OpenSER) - Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> >> > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ------------------------------ Message: 3 Date: Wed, 31 Mar 2010 11:41:29 +0200 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [Kamailio-Users] more type conversion wonders To: Juha Heinanen <jh(a)tutpro.com> Cc: users(a)lists.sip-router.org Message-ID: <4BB318C9.1080302(a)gmail.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 3/31/10 8:40 AM, Juha Heinanen wrote: > i played a bit more with selects and found that statement > > if ($sht(auth=>foo::count)> @cfg_get.local.gw_timeout) { > xlog("foo"); > } > I guess selects have types, while pseudo-variables are kind of type agnostic. Any PV has a string representation of the value, comparison is done as integer if both PVs are integers, otherwise is done as string. The safe side is to use PV representation of selects when comparing with another PV. PV and selects is another point for the future coherence. Cheers, Daniel > produces error > > 0(9213) :<core> [cfg.y:3379]: parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 507, column 28-55: bad expression: type mismatch: str instead of int at (507,55) > > but the error goes away if i either make explicit conversion > > if ($sht(auth=>foo::count)> (int)(a)cfg_get.local.gw_timeout) { > xlog("foo"); > } > > or use $sel > > if ($sht(auth=>foo::count)> $sel(cfg_get.local.gw_timeout)) { > xlog("foo"); > } > > this is thus exactly opposite than in my t_set_fr ordeal where i got > conversion error when i used $sel instead of @. > > in my opinion, no (int) conversion should be needed in the firs case, > because the value of the cfg variable is int. > > in summary, writing statements that include selects is very difficult > and error prone. > > -- juha > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ------------------------------ _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users End of Users Digest, Vol 58, Issue 76 *************************************

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[Kamailio-Users] more type conversion wonders
by jh＠tutpro.com 31 Mar '10

31 Mar '10

i played a bit more with selects and found that statement if ($sht(auth=>foo::count) > @cfg_get.local.gw_timeout) { xlog("foo"); } produces error 0(9213) : <core> [cfg.y:3379]: parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 507, column 28-55: bad expression: type mismatch: str instead of int at (507,55) but the error goes away if i either make explicit conversion if ($sht(auth=>foo::count) > (int)(a)cfg_get.local.gw_timeout) { xlog("foo"); } or use $sel if ($sht(auth=>foo::count) > $sel(cfg_get.local.gw_timeout)) { xlog("foo"); } this is thus exactly opposite than in my t_set_fr ordeal where i got conversion error when i used $sel instead of @. in my opinion, no (int) conversion should be needed in the firs case, because the value of the cfg variable is int. in summary, writing statements that include selects is very difficult and error prone. -- juha

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[Kamailio-Users] Registrar: Save modified contact / set the q value
by NeoTel Lists 31 Mar '10

31 Mar '10

Hello Everybody! Is there any way to save(domain, 0x02) the contact after it has been changed somehow in Kamailio <= 1.5? Do I have to upgrade to 3.0 and use msg_apply_changes() before save()? Or: Can I set the to be saved q value somehow? br Walter

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Re: [Kamailio-Users] Registrar: Save modified contact / set the qvalue
by NeoTel Lists 30 Mar '10

30 Mar '10

Again sorry, too much trial&error makes a lot of noise ... if (avp_subst("$(avp(contact))", "/;phone-context=q([01]\.[0-9]+)(.*>)?([^>]*)/\2\3;q=\1/i")) { # just to have it handy int that var(Q) $var(Q) = $(avp(contact){s.select,1,>}{param.value,q}); remove_hf("Contact"); append_hf("Contact: $(avp(contact))\r\n"); # must not add q= twice } -> Same result: q=-1 in Database. -----Ursprüngliche Nachricht----- Von: users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] Im Auftrag von NeoTel Lists Gesendet: Dienstag, 30. März 2010 18:56 An: Alex Balashov; users(a)lists.kamailio.org Betreff: Re: [Kamailio-Users] Registrar: Save modified contact / set the qvalue Sorry for being unclear: I meant: Modify the contact in the script, then do save(). E.g. $(avp(contact)) = $ct; # Those Patton can't send the q value ... Misuse their ;phone-context param if (avp_subst("$(avp(contact))", "/;phone-context=q([01]\.[0-9])(.*>)?([^>]*)/\2\3;q=\1/i")) { # just to have it handy int that var(Q) $var(Q) = $(avp(contact){s.select,1,>}{param.value,q}); remove_hf("Contact"); append_hf("Contact: $(avp(contact));q=$var(Q)r\n"); # subst() does the same, also not save()ed } ... setbflag(1); fix_nated_register(); save("contactimpl_nat", "0x02"); Br Walter -----Ursprüngliche Nachricht----- Von: users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] Im Auftrag von Alex Balashov Gesendet: Dienstag, 30. März 2010 17:58 An: users(a)lists.kamailio.org Betreff: Re: [Kamailio-Users] Registrar: Save modified contact / set the q value On 03/30/2010 11:36 AM, NeoTel Lists wrote: > Hello Everybody! > Is there any way to save(domain, 0x02) the contact after it has been > changed somehow in Kamailio <= 1.5? If it were not possible, how would nathelper:fix_nated_contact() work? -- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.com/ _______________________________________________ Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users _______________________________________________ Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

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Re: [Kamailio-Users] Registrar: Save modified contact / set the q value
by NeoTel Lists 30 Mar '10

30 Mar '10

Sorry for being unclear: I meant: Modify the contact in the script, then do save(). E.g. $(avp(contact)) = $ct; # Those Patton can't send the q value ... Misuse their ;phone-context param if (avp_subst("$(avp(contact))", "/;phone-context=q([01]\.[0-9])(.*>)?([^>]*)/\2\3;q=\1/i")) { # just to have it handy int that var(Q) $var(Q) = $(avp(contact){s.select,1,>}{param.value,q}); remove_hf("Contact"); append_hf("Contact: $(avp(contact));q=$var(Q)r\n"); # subst() does the same, also not save()ed } ... setbflag(1); fix_nated_register(); save("contactimpl_nat", "0x02"); Br Walter -----Ursprüngliche Nachricht----- Von: users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] Im Auftrag von Alex Balashov Gesendet: Dienstag, 30. März 2010 17:58 An: users(a)lists.kamailio.org Betreff: Re: [Kamailio-Users] Registrar: Save modified contact / set the q value On 03/30/2010 11:36 AM, NeoTel Lists wrote: > Hello Everybody! > Is there any way to save(domain, 0x02) the contact after it has been > changed somehow in Kamailio <= 1.5? If it were not possible, how would nathelper:fix_nated_contact() work? -- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.com/ _______________________________________________ Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

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[Kamailio-Users] Quick REGEX to match Private IP
by Uriel Rozenbaum 30 Mar '10

30 Mar '10

Hi guys, Does anyone have a REGEX syntax to match a private IP on the 192.168.x.x range? I'm trying with: if($avp(s:ip_origen)=~"192.168(\.([1]?\d{1,2}|2[0-4]{1}\d{1}|25[0-5]{1})){2}" ) But all IPs pass as private, even public ones. Thanks! Uriel

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[Kamailio-Users] Fast lock loop with arm
by Zappasodi Daniele 30 Mar '10

30 Mar '10

Hi, I have a big problem with fast lock mutex on arm CPU: sometimes I find one or more children that go in an infinite loop, in the while loop of get_lock function. They remain in Run for long time (sometimes hours) and openser keeps 100% CPU. Now I have changed the functions get_lock and tsl in order to obtain more info and I observe that (if and) when the process finish the cycle, it have done a big amount of cycles. This the log with my added info: Mar 26 20:29:08 SAM-IP openser[1645]: NOTICE:tm:_lock: ret 1 (cycle) Mar 26 20:29:51 SAM-IP openser[1645]: NOTICE:tm:_lock: ret 1 (cycle) Mar 26 20:29:55 SAM-IP openser[1645]: NOTICE:tm:_lock: ret 1 (cycle) Mar 26 20:29:55 SAM-IP openser[1645]: NOTICE:tm:t_retransmit_reply: MYTM lock [process in loop] Mar 26 20:33:46 SAM-IP openser[1645]: NOTICE:tm:_lock: ret 59374917 (cycle) [after 4 minutes and 59374917 cycles, this is an example with a "short" loop] Mar 26 20:33:46 SAM-IP openser[1645]: NOTICE:tm:t_retransmit_reply: MYTM lock done I'm not able to recognize the call flow that generates the loop (I know only that it always happens when t_retransmit_reply calls LOCK_REPLIES), but I urgently need a work around to escape from the loop. This is the get_lock function with my counter j: inline static int get_lock(fl_lock_t* lock) { #ifdef ADAPTIVE_WAIT int i=ADAPTIVE_WAIT_LOOPS; int j=1; /***** my change ****/ #endif while(tsl(lock)){ #ifdef BUSY_WAIT #elif defined ADAPTIVE_WAIT j++; /***** my change ****/ if (i>0) i--; else sched_yield(); #else sched_yield(); #endif } return(j); } Can I break the lock when my counter reaches a big value? Should I call the unlock function after the break? which value can be considered too big? Thanks, Daniele ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message in error. **********************************************************************

2 7

[Kamailio-Users] Kammailio as proxy and clustered asterisk as registrar
by Vic Jolin 30 Mar '10

30 Mar '10

Hi, I would like to seek help from the list regarding my setup. I have kamailio setup to send any sip messages to load balance asterisk clusters. The registration part is successful. The calling part is not, calls get dropped, I believe it's rtp issue or maybe nat. I'm very new to kamailio and would like to ask for help on this. Tia!

3 2

[Kamailio-Users] kam 1.5 sqlops sql_query timeout value
by Robert McGilvray 30 Mar '10

30 Mar '10

Hello, I'm using the sqlops module for some custom queries against mysql. Sometimes during a cluster failure the mysqld nodes will hang there waiting for the backend to finish up whatever it's doing, it still accepts the connection and the query but doesn't return results. I have a pair of F5 load balancers in front of the two sql nodes, so there is redundancy as long as the cluster is operational. I'd prefer to keep the cross-site failover in kamailio. I looked through the docs on sqlops and I can't find any reference to a timeout value. I'd like to implement a failover in the script to my other database cluster but if kam waits for a long time before returning a negative it may not work very well. Consider this code in my script for 911 services in my US offices: (I rewrite the rpid/pai and ruri based on IP address then send it to my provider(s)) modparam("sqlops","sqlcon","gokam=>mysql://*****:*****@172.20.180.21/sip _gokam") if (!sql_query("gokam", "select location,cidname,cidnum,ciddomain,e911number,inet_ntoa(network) as network,inet_ntoa(netmask) \ as netmask from e911 where (inet_aton('$si') & netmask) = network", "result")) { sl_send_reply("500", "Database error"); exit; } What I'd like to do is: modparam("sqlops","sqlcon","gokam=>mysql://****:****@172.20.180.21/sip_g okam") modparam("sqlops","sqlcon","gokam_site2=>mysql://*****:*****@172.23.180. 21/sip_gokam") if (!sql_query("gokam", "select location,cidname,cidnum,ciddomain,e911number,inet_ntoa(network) as network,inet_ntoa(netmask) \ as netmask from e911 where (inet_aton('$si') & netmask) = network", "result")) { xlog("L_CRIT", "Primary database failure, using alternate\n"); if (!sql_query("gokam_site2", "select location,cidname,cidnum,ciddomain,e911number,inet_ntoa(network) as network,inet_ntoa(netmask) \ as netmask from e911 where (inet_aton('$si') & netmask) = network", "result")) { sl_send_reply("500", Database error"); exit; } What is the default timeout for sql_query before it returns a negative, is it configurable? Thanks! Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

3 2

[Kamailio-Users] get warning when sip client is not online
by s.maertens＠telenet.be 30 Mar '10

30 Mar '10

Hello, I have a kamailio setup running for several SIP devices. I would like to be warned whenever one of those devices is not reacheable anymore. That means when they go offline or are not anymore registered to the kamailio server. (timeout, unregister, ...). I am using the location table so we would be able to write a script that just at specific times checks the difference between the subscriber table and the location table but maybe there is a much better and more realtime solution. So basically the question is : how to execute a script whenever a host gets removed fom the location table. I am running Kamailio 3.0.1 All hints are very much appreciated :)

3 2

[Kamailio-Users] GSoC 2010 Student Registration is Open
by Daniel-Constantin Mierla 30 Mar '10

30 Mar '10

Hello, if you are a student (or know one) interested in working with SIP and Presence over the summer, earn some money and experience, you have the choice of Kamailio - SIP Router project within SIP Communicator organization: http://www.kamailio.org/w/gsoc-2010/ Registration is now open, up to April 9, more details in above link. If you have further questions, drop an email. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla

1 0

ser avp question
by jh＠tutpro.com 29 Mar '10

29 Mar '10

modules_s/domain domain_lookup() function can be used to assign domain attributes to avps. for example, lookup_domain("$fd", "foo.bar"), assigns attributes of domain foo.bar to avps $fd.x, where x is name of attribute. my question is, how can i index these avps if there is more than one with the same name? statement $var(attr) = $fd.attr[]; does not give syntax errors, but $var(attr) = $fd.attr[$var(i)]; produces 0(7368) : <core> [cfg.y:3379]: parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 3174, column 24-30: syntax error 0(7368) : <core> [cfg.y:3379]: parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 3174, column 24-30: bad command 0(7368) : <core> [cfg.y:3382]: parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 3174, column 31: bad command 0(7368) : <core> [cfg.y:3382]: parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 3174, column 31: bad command -- juha

3 5

weird dp_translate() behavior
by jh＠tutpro.com 29 Mar '10

29 Mar '10

while playing with dp_translate(), i noticed the following weird behavior. i have a rule with dpid=11, where match_exp='^\+' and subst_exp and repl_exp are empty. my test goes like this: $rU = "+35892345670"; if (dp_translate("11", $rU/$rU")) { xlog("ruri user is '$rU'\n"); }; $var(test) = "+35892345670"; if (dp_translate("11", "$var(test)/$var(test)")) { xlog("var(test) is '$var(test)'\n"); }; and this is produced to syslog: Mar 29 20:01:39 localhost /usr/sbin/sip-proxy[20533]: ERROR: ruri user is '+35892345670' Mar 29 20:01:39 localhost /usr/sbin/sip-proxy[20533]: ERROR: var(test) is '' that is, $rU is not translated to empty string, but $var(test) is. can someone provide an explanation for this? what should the expected behavior be, when subst_exp and repl_exp are empty? original author of dialplan module is Andreea-Ancuta Onofrei, but i don't know if he is still around. -- juha

1 1

@msg.%s syntax
by jh＠tutpro.com 29 Mar '10

29 Mar '10

first section of page http://sip-router.org/wiki/cookbooks/selects/devel had example @msg.["P-anyheader-youwant"]) that, however, produced syntax error. i removed the dot and got rid of the error. i then went and fixed the wiki page too. however, later on the page there still is @msg.%s and many more like that. are they generated automatically? if so, then fixing them on the wiki page would not help. also, is there somewhere a summary of pv syntax of s and k flavors? -- juha

2 5

Re: [Kamailio-Users] Problem with secure TLS call
by Hemanshu Patel 29 Mar '10

29 Mar '10

With TLS it still is not working Mar 27 11:39:16 [4421] INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 255 kb Mar 27 11:39:16 [4425] WARNING:usrloc:dbrow2info: non-local socket <tcp:172.16.16.218:5091>...ignoring Mar 27 11:39:18 [4431] INFO:core:tls_accept: client did not present a certificate Mar 27 11:39:55 [4432] INFO:core:tls_accept: client did not present a certificate Mar 27 11:40:01 [4431] ERROR:rr:w_record_route: Double attempt to record-route Mar 27 11:40:01 [4426] ERROR:core:tls_connect: SSL_ERROR_SYSCALL err=Connection reset by peer(104) Mar 27 11:40:01 [4426] ERROR:core:tls_connect: something wrong in SSL: 5 (ret=-1) err=Connection reset by peer(104) Mar 27 11:40:01 [4426] ERROR:core:tcp_send: failed to send Mar 27 11:40:01 [4426] ERROR:core:msg_send: tcp_send failed Mar 27 11:40:01 [4425] WARNING:core:run_actions: null action list (rec_level=1) my doubt is that if somethings wrong with SSL certificates that i created myself (i.e my own root CA, and self signed certificates), then how come registration is working like charm? if there is some problem with Certificates then registration should also not work. am i right? ---- Regards, Hemanshu Patel Sr. Software Engg SIS,Ahmedabad Mo:09601295238 On Sat 27/03/10 9:50 AM , "Hemanshu Patel" <hemanshu.patel(a)saicare.com> wrote: > I havent tested over TCP, let me check it > but hardphone, i mean hardware based phones from grandstream gvx3140 > works > fine with same implementation on TLS. > -- > Regards, > Hemanshu Patel > M: 09601295238 > > Does eyebeam with SIP over TCP is working? > > > > Am 26.03.2010 13:43, schrieb Hemanshu Patel: > >> > >> i am still having this problem. > >> when i use two grandstream phone everything works fine, > >> i can make calls on TLS and users can talk to each other. > >> > >> But when i use two eyebream phone, it doesnt work, gives > following error > >> > >> :33 [2875] WARNING:core:init_ssl_ctx_behavior: server > verification NOT > >> activated. Weaker security. > >> [ panreg-tls]$ > >> [ panreg-tls]$ > >> [ panreg-tls]$ Mar 26 18:11:59 [2889] > >> ERROR:rr:w_record_route: Double attempt to record-route > >> Mar 26 18:12:09 [2884] ERROR:core:tcp_blocking_connect: timeout > 10 s > >> elapsed from 10 s > >> Mar 26 18:12:09 [2884] ERROR:core:tcpconn_connect: > tcp_blocking_connect > >> failed > >> Mar 26 18:12:09 [2884] ERROR:core:tcp_send: connect failed > >> Mar 26 18:12:09 [2884] ERROR:core:msg_send: tcp_send failed > >> Mar 26 18:12:10 [2883] ERROR:core:tcp_blocking_connect: timeout > 10 s > >> elapsed from 10 s > >> Mar 26 18:12:10 [2883] ERROR:core:tcpconn_connect: > tcp_blocking_connect > >> failed > >> Mar 26 18:12:10 [2883] ERROR:core:tcp_send: connect failed > >> Mar 26 18:12:10 [2883] ERROR:core:msg_send: tcp_send failed > >> Mar 26 18:12:10 [2883] WARNING:core:run_actions: null action list > >> (rec_level=1) > >> Mar 26 18:12:11 [2881] ERROR:core:tcp_blocking_connect: timeout > 10 s > >> elapsed from 10 s > >> Mar 26 18:12:11 [2881] ERROR:core:tcpconn_connect: > tcp_blocking_connect > >> failed > >> Mar 26 18:12:11 [2881] ERROR:core:tcp_send: connect failed > >> Mar 26 18:12:11 [2881] ERROR:core:msg_send: tcp_send failed > >> Mar 26 18:12:13 [2882] ERROR:core:tcp_blocking_connect: timeout > 10 s > >> elapsed from 10 s > >> Mar 26 18:12:13 [2882] ERROR:core:tcpconn_connect: > tcp_blocking_connect > >> failed > >> Mar 26 18:12:13 [2882] ERROR:core:tcp_send: connect failed > >> Mar 26 18:12:13 [2882] ERROR:core:msg_send: tcp_send failed > >> Mar 26 18:12:19 [2884] ERROR:core:tcp_blocking_connect: timeout > 10 s > >> elapsed from 10 s > >> Mar 26 18:12:19 [2884] ERROR:core:tcpconn_connect: > tcp_blocking_connect > >> failed > >> Mar 26 18:12:19 [2884] ERROR:core:tcp_send: connect failed > >> Mar 26 18:12:19 [2884] ERROR:core:msg_send: tcp_send failed > >> > >> > >> > >> Any idea? Same configuration works with grandstream, while they > cant in > >> softphone? and gives 477 errot code. > >> > >> my asterisk log is as below: > >> > >> 26 18:08:09] WARNING[2833]: chan_sip.c:1648 setup_crypto: > Hemanshu: > >> local_key641 jhbAJ7jFE1p/Ngn4kVxy7qTTmkJpeBiN6W98+gmM len 40 > >> -- Called kamailio/1003 > >> -- Got SIP response 477 "Send failed (477/SL)" back from > >> 172.16.16.218 > >> -- No one is available to answer at this time (1:0/0/0) > >> -- Executing [_default:4] Hangup("SIP/5091-8c001430", "") > >> in > >> new stack > >> == Spawn extension (hemu_default, 1003, 4) exited non-zero on > >> 'SIP/5091-8c001430' > >> > ::Disconnected form Oracle, trying to connect again.. > >> > ::Tried a lot, not getting connected.. > >> -- Got SIP response 477 "Send failed (477/SL)" back from > >> 172.16.16.218 > >> > >> > >> Any idea what could be the problem? > >> > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > ---- SIS Blade Server, Desktop & Thin Client is now available on DGS & D rate contract. For more detail kindly visit our website http://www.saicare.com

2 4

Re: [Kamailio-Users] Problem with secure TLS call
by Hemanshu Patel 29 Mar '10

29 Mar '10

Here is the complete call log for One call from two clients eyebream and bria running on single PC. Mar 27 12:36:36 [5492] DBG:core:destroy_avp_list: destroying list (nil) Mar 27 12:36:36 [5492] DBG:core:receive_msg: cleaning up Mar 27 12:36:42 [5492] DBG:core:tcp_receive_timeout: 0x7fd335f04798 expired (50, 51) lt=0 Mar 27 12:36:42 [5492] DBG:core:io_watch_del: io_watch_del (0x74efe0, 18, -1, 0x10) fd_no=2 called Mar 27 12:36:42 [5492] DBG:core:release_tcpconn: releasing con 0x7fd335f04798, state 0, fd=18, id=3 Mar 27 12:36:42 [5492] DBG:core:release_tcpconn: extra_data 0x7fd335f017b8 Mar 27 12:36:42 [5499] DBG:core:handle_tcp_child: reader response= 7fd335f04798, 0 from 0 Mar 27 12:36:42 [5499] DBG:core:io_watch_add: io_watch_add(0x74ee80, 24, 2, 0x7fd335f04798), fd_no=17 Mar 27 12:36:42 [5499] DBG:core:handle_tcp_child: cmd CONN_RELEASE 0x7fd335f04798 refcnt= 0 Mar 27 12:36:45 [5499] DBG:core:handle_tcpconn_ev: data available on 0x7fd335eddc18 23 Mar 27 12:36:45 [5499] DBG:core:io_watch_del: io_watch_del (0x74ee80, 23, -1, 0x0) fd_no=18 called Mar 27 12:36:45 [5499] DBG:core:send2child: to tcp child 0 0(5492), 0x7fd335eddc18 Mar 27 12:36:45 [5492] DBG:core:handle_io: received n=8 con=0x7fd335eddc18, fd=18 Mar 27 12:36:45 [5492] DBG:core:io_watch_add: io_watch_add(0x74efe0, 18, 2, 0x7fd335eddc18), fd_no=1 Mar 27 12:36:45 [5492] DBG:core:tls_update_fd: New fd is 18 Mar 27 12:36:45 [5492] DBG:core:_tls_read: 1109 bytes read Mar 27 12:36:45 [5492] DBG:core:tcp_read_req: content-length= 512 Mar 27 12:36:45 [5492] DBG:core:parse_msg: SIP Request: Mar 27 12:36:45 [5492] DBG:core:parse_msg: method: <INVITE> Mar 27 12:36:45 [5492] DBG:core:parse_msg: uri: <sip:1002@172.16.16.218:5091;transport=tls> Mar 27 12:36:45 [5492] DBG:core:parse_msg: version: <SIP/2.0> Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=2 Mar 27 12:36:45 [5492] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d8754z-e333132001727d6d-1---d8754z->; state=6 Mar 27 12:36:45 [5492] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Mar 27 12:36:45 [5492] DBG:core:parse_via: end of header reached, state=5 Mar 27 12:36:45 [5492] DBG:core:parse_headers: via found, flags=2 Mar 27 12:36:45 [5492] DBG:core:parse_headers: this is the first via Mar 27 12:36:45 [5492] DBG:core:receive_msg: After parse_msg... Mar 27 12:36:45 [5492] DBG:core:receive_msg: preparing to run routing scripts... Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=100 Mar 27 12:36:45 [5492] DBG:maxfwd:is_maxfwd_present: value = 70 Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=8 Mar 27 12:36:45 [5492] DBG:core:parse_to: end of header reached, state=10 Mar 27 12:36:45 [5492] DBG:core:parse_to: display={}, ruri={sip:1002@172.16.16.218:5091} Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: <To> [31]; uri=[sip:1002@172.16.16.218:5091] Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: to body [<sip:1002@172.16.16.218:5091> ] Mar 27 12:36:45 [5492] DBG:siputils:has_totag: no totag Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=78 Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: cseq <CSeq>: <1> <INVITE> Mar 27 12:36:45 [5492] DBG:tm:t_lookup_request: start searching: hash=44320, isACK=0 Mar 27 12:36:45 [5492] DBG:tm:matching_3261: RFC3261 transaction matching failed Mar 27 12:36:45 [5492] DBG:tm:t_lookup_request: no transaction found Mar 27 12:36:45 [5492] DBG:core:parse_to_param: tag=9f574228 Mar 27 12:36:45 [5492] DBG:core:parse_to: end of header reached, state=29 Mar 27 12:36:45 [5492] DBG:core:parse_to: display={"1003"}, ruri={sip:1003@172.16.16.218:5091} Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: checking if host==us: 13==13 && [172.16.16.218] == [172.16.16.218] Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: checking if port 5090 matches port 5091 Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: checking if host==us: 13==13 && [172.16.16.218] == [172.16.16.218] Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: checking if port 5092 matches port 5091 Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: checking if host==us: 13==13 && [172.16.16.218] == [172.16.16.218] Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: checking if port 5091 matches port 5091 Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: match found for: [172.16.16.218:5091] Mar 27 12:36:45 [5492] DBG:core:check_self: host == me Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=10000 Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: content_length=512 Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: found end of header Mar 27 12:36:45 [5492] DBG:auth:pre_auth: credentials with given realm not found Mar 27 12:36:45 [5492] DBG:auth:reserve_nonce_index: second= 22, sec_monit= -1, index= 3 Mar 27 12:36:45 [5492] DBG:auth:build_auth_hf: nonce index= 3 Mar 27 12:36:45 [5492] DBG:auth:build_auth_hf: 'Proxy-Authenticate: Digest realm="172.16.16.218", nonce="4badaea30000000316e854bb0194116d66618a39dcd9340c" ' Mar 27 12:36:45 [5492] DBG:sl:send_reply: reply in stateless mode (sl) Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=ffffffffffffffff Mar 27 12:36:45 [5492] DBG:core:check_via_address: params 172.16.17.85, 172.16.17.85, 0 Mar 27 12:36:45 [5492] DBG:core:tcp_send: tcp connection found (0x7fd335eddc18), acquiring fd Mar 27 12:36:45 [5492] DBG:core:tcp_send: c= 0x7fd335eddc18, n=16 Mar 27 12:36:45 [5499] DBG:core:handle_ser_child: read response= 7fd335eddc18, 1, fd -1 from 7 (5492) Mar 27 12:36:45 [5492] DBG:core:tcp_send: after receive_fd: c= 0x7fd335eddc18 n=8 fd=19 Mar 27 12:36:45 [5492] DBG:core:tcp_send: sending... Mar 27 12:36:45 [5492] DBG:core:tls_update_fd: New fd is 19 Mar 27 12:36:45 [5492] DBG:core:tls_write: write was successful (519 bytes) Mar 27 12:36:45 [5492] DBG:core:tcp_send: after write: c= 0x7fd335eddc18 n=519 fd=19 Mar 27 12:36:45 [5492] DBG:core:tcp_send: buf= SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/TLS 172.16.17.85:4403;branch=z9hG4bK-d8754z-e333132001727d6d-1---d8754z-;rport=1301 To: <sip:1002@172.16.16.218:5091>;tag=2cf7f9a22b6374f465182a893bba7dec.5636 From: "1003"<sip:1003@172.16.16.218:5091>;tag=9f574228 Call-ID: M2Y2OThmYmYwZjM4ODhhYzk4Mjc0MGFhMDQyNDUzZWU. CSeq: 1 INVITE Proxy-Authenticate: Digest realm="172.16.16.218", nonce="4badaea30000000316e854bb0194116d66618a39dcd9340c" Server: Kamailio (1.5.4-tls (x86_64/linux)) Content-Length: 0 Mar 27 12:36:45 [5492] DBG:core:destroy_avp_list: destroying list (nil) Mar 27 12:36:45 [5492] DBG:core:receive_msg: cleaning up Mar 27 12:36:45 [5492] DBG:core:tls_update_fd: New fd is 18 Mar 27 12:36:45 [5492] DBG:core:_tls_read: 370 bytes read Mar 27 12:36:45 [5492] DBG:core:tcp_read_req: content-length= 0 Mar 27 12:36:45 [5492] DBG:core:parse_msg: SIP Request: Mar 27 12:36:45 [5492] DBG:core:parse_msg: method: <ACK> Mar 27 12:36:45 [5492] DBG:core:parse_msg: uri: <sip:1002@172.16.16.218:5091;transport=tls> Mar 27 12:36:45 [5492] DBG:core:parse_msg: version: <SIP/2.0> Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=2 Mar 27 12:36:45 [5492] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d8754z-e333132001727d6d-1---d8754z->; state=6 Mar 27 12:36:45 [5492] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Mar 27 12:36:45 [5492] DBG:core:parse_via: end of header reached, state=5 Mar 27 12:36:45 [5492] DBG:core:parse_headers: via found, flags=2 Mar 27 12:36:45 [5492] DBG:core:parse_headers: this is the first via Mar 27 12:36:45 [5492] DBG:core:receive_msg: After parse_msg... Mar 27 12:36:45 [5492] DBG:core:receive_msg: preparing to run routing scripts... Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=8 Mar 27 12:36:45 [5492] DBG:core:parse_to_param: tag=2cf7f9a22b6374f465182a893bba7dec.5636 Mar 27 12:36:45 [5492] DBG:core:parse_to: end of header reached, state=29 Mar 27 12:36:45 [5492] DBG:core:parse_to: display={}, ruri={sip:1002@172.16.16.218:5091} Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: <To> [73]; uri=[sip:1002@172.16.16.218:5091] Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: to body [<sip:1002@172.16.16.218:5091>] Mar 27 12:36:45 [5492] DBG:sl:sl_filter_ACK: local ACK found -> dropping it! Mar 27 12:36:45 [5492] DBG:core:destroy_avp_list: destroying list (nil) Mar 27 12:36:45 [5492] DBG:core:receive_msg: cleaning up Mar 27 12:36:45 [5492] DBG:core:tls_update_fd: New fd is 18 Mar 27 12:36:45 [5492] DBG:core:_tls_read: 1339 bytes read Mar 27 12:36:45 [5492] DBG:core:tcp_read_req: content-length= 512 Mar 27 12:36:45 [5492] DBG:core:parse_msg: SIP Request: Mar 27 12:36:45 [5492] DBG:core:parse_msg: method: <INVITE> Mar 27 12:36:45 [5492] DBG:core:parse_msg: uri: <sip:1002@172.16.16.218:5091;transport=tls> Mar 27 12:36:45 [5492] DBG:core:parse_msg: version: <SIP/2.0> Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=2 Mar 27 12:36:45 [5492] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d8754z-863b295f961d253f-1---d8754z->; state=6 Mar 27 12:36:45 [5492] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Mar 27 12:36:45 [5492] DBG:core:parse_via: end of header reached, state=5 Mar 27 12:36:45 [5492] DBG:core:parse_headers: via found, flags=2 Mar 27 12:36:45 [5492] DBG:core:parse_headers: this is the first via Mar 27 12:36:45 [5492] DBG:core:receive_msg: After parse_msg... Mar 27 12:36:45 [5492] DBG:core:receive_msg: preparing to run routing scripts... Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=100 Mar 27 12:36:45 [5492] DBG:maxfwd:is_maxfwd_present: value = 70 Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=8 Mar 27 12:36:45 [5492] DBG:core:parse_to: end of header reached, state=10 Mar 27 12:36:45 [5492] DBG:core:parse_to: display={}, ruri={sip:1002@172.16.16.218:5091} Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: <To> [31]; uri=[sip:1002@172.16.16.218:5091] Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: to body [<sip:1002@172.16.16.218:5091> ] Mar 27 12:36:45 [5492] DBG:siputils:has_totag: no totag Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=78 Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: cseq <CSeq>: <2> <INVITE> Mar 27 12:36:45 [5492] DBG:tm:t_lookup_request: start searching: hash=44317, isACK=0 Mar 27 12:36:45 [5492] DBG:tm:matching_3261: RFC3261 transaction matching failed Mar 27 12:36:45 [5492] DBG:tm:t_lookup_request: no transaction found Mar 27 12:36:45 [5492] DBG:core:parse_to_param: tag=9f574228 Mar 27 12:36:45 [5492] DBG:core:parse_to: end of header reached, state=29 Mar 27 12:36:45 [5492] DBG:core:parse_to: display={"1003"}, ruri={sip:1003@172.16.16.218:5091} Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: checking if host==us: 13==13 && [172.16.16.218] == [172.16.16.218] Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: checking if port 5090 matches port 5091 Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: checking if host==us: 13==13 && [172.16.16.218] == [172.16.16.218] Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: checking if port 5092 matches port 5091 Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: checking if host==us: 13==13 && [172.16.16.218] == [172.16.16.218] Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: checking if port 5091 matches port 5091 Mar 27 12:36:45 [5492] DBG:core:grep_sock_info: match found for: [172.16.16.218:5091] Mar 27 12:36:45 [5492] DBG:core:check_self: host == me Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=10000 Mar 27 12:36:45 [5492] DBG:auth:check_nonce: comparing [4badaea30000000316e854bb0194116d66618a39dcd9340c] and [4badaea30000000316e854bb0194116d66618a39dcd9340c] Mar 27 12:36:45 [5492] DBG:core:db_new_result: allocate 48 bytes for result set at 0x78d6c8 Mar 27 12:36:45 [5492] DBG:db_mysql:db_mysql_get_columns: 1 columns returned from the query Mar 27 12:36:45 [5492] DBG:core:db_allocate_columns: allocate 8 bytes for result names at 0x78aab8 Mar 27 12:36:45 [5492] DBG:core:db_allocate_columns: allocate 4 bytes for result types at 0x79ea58 Mar 27 12:36:45 [5492] DBG:db_mysql:db_mysql_get_columns: allocate 16 bytes for RES_NAMES[0] at 0x78d010 Mar 27 12:36:45 [5492] DBG:db_mysql:db_mysql_get_columns: RES_NAMES(0x78d010)[0]=[password] Mar 27 12:36:45 [5492] DBG:db_mysql:db_mysql_get_columns: use DB_STRING result type Mar 27 12:36:45 [5492] DBG:core:db_allocate_rows: allocate 16 bytes for rows at 0x79ea38 Mar 27 12:36:45 [5492] DBG:core:db_allocate_row: allocate 32 bytes for row values at 0x79ea78 Mar 27 12:36:45 [5492] DBG:core:db_str2val: converting STRING [1234] Mar 27 12:36:45 [5492] DBG:auth_db:get_ha1: HA1 string calculated: 6650a5aab1d658b465ee304e21b8db36 Mar 27 12:36:45 [5492] DBG:auth:check_response: our result = '19406efa15546eb60e105c164a1feb3d' Mar 27 12:36:45 [5492] DBG:auth:check_response: authorization is OK Mar 27 12:36:45 [5492] DBG:auth:post_auth: nonce index= 3 Mar 27 12:36:45 [5492] DBG:core:db_free_columns: freeing 1 columns Mar 27 12:36:45 [5492] DBG:core:db_free_columns: freeing RES_NAMES[0] at 0x78d010 Mar 27 12:36:45 [5492] DBG:core:db_free_columns: freeing result names at 0x78aab8 Mar 27 12:36:45 [5492] DBG:core:db_free_columns: freeing result types at 0x79ea58 Mar 27 12:36:45 [5492] DBG:core:db_free_rows: freeing 1 rows Mar 27 12:36:45 [5492] DBG:core:db_free_row: freeing row values at 0x79ea78 Mar 27 12:36:45 [5492] DBG:core:db_free_rows: freeing rows at 0x79ea38 Mar 27 12:36:45 [5492] DBG:core:db_free_result: freeing result set at 0x78d6c8 Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=14000 Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: content_length=512 Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: found end of header Mar 27 12:36:45 [5492] DBG:core:comp_scriptvar: str 29 : 1003 ---- Invite ---- Mar 27 12:36:45 [5492] DBG:core:comp_scriptvar: str 29 : 1003 Mar 27 12:36:45 [5492] ERROR:rr:w_record_route: Double attempt to record-route Mar 27 12:36:45 [5492] DBG:dispatcher:ds_select_dst: set [1] Mar 27 12:36:45 [5492] DBG:dispatcher:ds_select_dst: alg hash [789753065] Mar 27 12:36:45 [5492] DBG:dispatcher:ds_select_dst: selected [0-1/0] <sip:172.16.16.218:5070> Mar 27 12:36:45 [5492] DBG:tm:t_newtran: transaction on entrance=(nil) Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=ffffffffffffffff Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=78 Mar 27 12:36:45 [5492] DBG:tm:t_lookup_request: start searching: hash=44317, isACK=0 Mar 27 12:36:45 [5492] DBG:tm:matching_3261: RFC3261 transaction matching failed Mar 27 12:36:45 [5492] DBG:tm:t_lookup_request: no transaction found Mar 27 12:36:45 [5492] DBG:tm:run_reqin_callbacks: trans=0x7fd335eee6b0, callback type 1, id 0 entered Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=ffffffffffffffff Mar 27 12:36:45 [5492] DBG:core:check_via_address: params 172.16.17.85, 172.16.17.85, 0 Mar 27 12:36:45 [5492] DBG:core:_shm_resize: resize(0) called Mar 27 12:36:45 [5492] DBG:core:tcp_send: tcp connection found (0x7fd335eddc18), acquiring fd Mar 27 12:36:45 [5492] DBG:core:tcp_send: c= 0x7fd335eddc18, n=16 Mar 27 12:36:45 [5499] DBG:core:handle_ser_child: read response= 7fd335eddc18, 1, fd -1 from 7 (5492) Mar 27 12:36:45 [5492] DBG:core:tcp_send: after receive_fd: c= 0x7fd335eddc18 n=8 fd=19 Mar 27 12:36:45 [5492] DBG:core:tcp_send: sending... Mar 27 12:36:45 [5492] DBG:core:tls_update_fd: New fd is 19 Mar 27 12:36:45 [5492] DBG:core:tls_write: write was successful (352 bytes) Mar 27 12:36:45 [5492] DBG:core:tcp_send: after write: c= 0x7fd335eddc18 n=352 fd=19 Mar 27 12:36:45 [5492] DBG:core:tcp_send: buf= SIP/2.0 100 Giving a try Via: SIP/2.0/TLS 172.16.17.85:4403;branch=z9hG4bK-d8754z-863b295f961d253f-1---d8754z-;rport=1301 To: <sip:1002@172.16.16.218:5091> From: "1003"<sip:1003@172.16.16.218:5091>;tag=9f574228 Call-ID: M2Y2OThmYmYwZjM4ODhhYzk4Mjc0MGFhMDQyNDUzZWU. CSeq: 2 INVITE Server: Kamailio (1.5.4-tls (x86_64/linux)) Content-Length: 0 Mar 27 12:36:45 [5492] DBG:tm:_reply_light: reply sent out. buf=0x79f5e8: SIP/2.0 1..., shmem=0x7fd335f007a8: SIP/2.0 1 Mar 27 12:36:45 [5492] DBG:tm:_reply_light: finished Mar 27 12:36:45 [5492] DBG:core:mk_proxy: doing DNS lookup... Mar 27 12:36:45 [5492] DBG:core:build_req_buf_from_sip_req: id added: <;i=1>, rcv proto=3 Mar 27 12:36:45 [5492] DBG:tm:set_timer: relative timeout is 500000 Mar 27 12:36:45 [5492] DBG:tm:insert_timer_unsafe: [4]: 0x7fd335eee8d0 (54200000) Mar 27 12:36:45 [5492] DBG:tm:set_timer: relative timeout is 30 Mar 27 12:36:45 [5492] DBG:tm:insert_timer_unsafe: [0]: 0x7fd335eee900 (83) Mar 27 12:36:45 [5492] DBG:tm:t_relay_to: new transaction fwd'ed Mar 27 12:36:45 [5492] DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Mar 27 12:36:45 [5492] DBG:core:destroy_avp_list: destroying list (nil) Mar 27 12:36:45 [5492] DBG:core:receive_msg: cleaning up Mar 27 12:36:45 [5487] DBG:core:parse_msg: SIP Reply (status): Mar 27 12:36:45 [5487] DBG:core:parse_msg: version: <SIP/2.0> Mar 27 12:36:45 [5487] DBG:core:parse_msg: status: <100> Mar 27 12:36:45 [5487] DBG:core:parse_msg: reason: <Trying> Mar 27 12:36:45 [5487] DBG:core:parse_headers: flags=2 Mar 27 12:36:45 [5487] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bKd1da.1a6f9b73.0>; state=6 Mar 27 12:36:45 [5487] DBG:core:parse_via_param: found param type 236, <i> = <1>; state=6 Mar 27 12:36:45 [5487] DBG:core:parse_via_param: found param type 234, <received> = <172.16.16.218>; state=16 Mar 27 12:36:45 [5487] DBG:core:parse_via: end of header reached, state=5 Mar 27 12:36:45 [5487] DBG:core:parse_headers: via found, flags=2 Mar 27 12:36:45 [5487] DBG:core:parse_headers: this is the first via Mar 27 12:36:45 [5487] DBG:core:receive_msg: After parse_msg... Mar 27 12:36:45 [5487] DBG:core:forward_reply: found module tm, passing reply to it Mar 27 12:36:45 [5487] DBG:tm:t_check: start=0xffffffffffffffff Mar 27 12:36:45 [5487] DBG:core:parse_headers: flags=22 Mar 27 12:36:45 [5487] DBG:core:parse_via_param: found param type 234, <received> = <172.16.17.85>; state=6 Mar 27 12:36:45 [5487] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d8754z-863b295f961d253f-1---d8754z->; state=6 Mar 27 12:36:45 [5487] DBG:core:parse_via_param: found param type 235, <rport> = <1301>; state=16 Mar 27 12:36:45 [5487] DBG:core:parse_via: end of header reached, state=5 Mar 27 12:36:45 [5487] DBG:core:parse_headers: via found, flags=22 Mar 27 12:36:45 [5487] DBG:core:parse_headers: parse_headers: this is the second via Mar 27 12:36:45 [5487] DBG:core:parse_to: end of header reached, state=10 Mar 27 12:36:45 [5487] DBG:core:parse_to: display={}, ruri={sip:1002@172.16.16.218:5091} Mar 27 12:36:45 [5487] DBG:core:get_hdr_field: <To> [31]; uri=[sip:1002@172.16.16.218:5091] Mar 27 12:36:45 [5487] DBG:core:get_hdr_field: to body [<sip:1002@172.16.16.218:5091> ] Mar 27 12:36:45 [5487] DBG:core:get_hdr_field: cseq <CSeq>: <2> <INVITE> Mar 27 12:36:45 [5487] DBG:core:parse_headers: flags=8 Mar 27 12:36:45 [5487] DBG:tm:t_reply_matching: hash 44317 label 934934177 branch 0 Mar 27 12:36:45 [5487] DBG:tm:t_reply_matching: REF_UNSAFE: after is 1 Mar 27 12:36:45 [5487] DBG:tm:t_reply_matching: reply matched (T=0x7fd335eee6b0)! Mar 27 12:36:45 [5487] DBG:tm:t_check: end=0x7fd335eee6b0 Mar 27 12:36:45 [5487] DBG:tm:reply_received: org. status uas=100, uac[0]=0 local=0 is_invite=1) Mar 27 12:36:45 [5487] DBG:tm:t_should_relay_response: T_code=100, new_code=100 Mar 27 12:36:45 [5487] DBG:tm:relay_reply: branch=0, save=0, relay=-1 Mar 27 12:36:45 [5487] DBG:tm:set_timer: relative timeout is 120 Mar 27 12:36:45 [5487] DBG:tm:insert_timer_unsafe: [1]: 0x7fd335eee900 (173) Mar 27 12:36:45 [5487] DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Mar 27 12:36:45 [5487] DBG:core:destroy_avp_list: destroying list (nil) Mar 27 12:36:45 [5487] DBG:core:receive_msg: cleaning up Mar 27 12:36:45 [5489] DBG:core:parse_msg: SIP Request: Mar 27 12:36:45 [5489] DBG:core:parse_msg: method: <INVITE> Mar 27 12:36:45 [5489] DBG:core:parse_msg: uri: <sip:1002@172.16.16.218:5090> Mar 27 12:36:45 [5489] DBG:core:parse_msg: version: <SIP/2.0> Mar 27 12:36:45 [5489] DBG:core:parse_headers: flags=2 Mar 27 12:36:45 [5489] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK7f71d414>; state=6 Mar 27 12:36:45 [5489] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Mar 27 12:36:45 [5489] DBG:core:parse_via: end of header reached, state=5 Mar 27 12:36:45 [5489] DBG:core:parse_headers: via found, flags=2 Mar 27 12:36:45 [5489] DBG:core:parse_headers: this is the first via Mar 27 12:36:45 [5489] DBG:core:receive_msg: After parse_msg... Mar 27 12:36:45 [5489] DBG:core:receive_msg: preparing to run routing scripts... Mar 27 12:36:45 [5489] DBG:core:parse_headers: flags=100 Mar 27 12:36:45 [5489] DBG:core:parse_to: end of header reached, state=10 Mar 27 12:36:45 [5489] DBG:core:parse_to: display={}, ruri={sip:1002@172.16.16.218:5090} Mar 27 12:36:45 [5489] DBG:core:get_hdr_field: <To> [31]; uri=[sip:1002@172.16.16.218:5090] Mar 27 12:36:45 [5489] DBG:core:get_hdr_field: to body [<sip:1002@172.16.16.218:5090> ] Mar 27 12:36:45 [5489] DBG:core:get_hdr_field: cseq <CSeq>: <102> <INVITE> Mar 27 12:36:45 [5489] DBG:maxfwd:is_maxfwd_present: value = 70 ---- From SC/MS/IC ---- Mar 27 12:36:45 [5489] DBG:core:rewrite_uri: rewriting Request-URI with 'sip:1002@172.16.17.85:25923;rinstance=133b23266b1f2e83;transport=TLS' Mar 27 12:36:45 [5489] DBG:core:mk_proxy: doing DNS lookup... Mar 27 12:36:45 [5489] DBG:core:parse_headers: flags=2000 Mar 27 12:36:45 [5489] DBG:core:get_hdr_field: content_length=394 Mar 27 12:36:45 [5489] DBG:core:forward_request: sending: INVITE sip:1002@172.16.17.85:25923;rinstance=133b23266b1f2e83;transport=TLS SIP/2.0 Record-Route: <sip:172.16.16.218:5091;transport=tls;r2=on;lr=on> Record-Route: <sip:172.16.16.218:5090;r2=on;lr=on> Via: SIP/2.0/TLS 172.16.16.218:5091;branch=z9hG4bK7f71d414 Via: SIP/2.0/UDP 172.16.16.218:5070;received=172.16.16.218;branch=z9hG4bK7f71d414;rport=5070 From: "1003" <sip:1003@172.16.16.218:5070>;tag=as7351a581 To: <sip:1002@172.16.16.218:5090> Contact: <sip:1003@172.16.16.218:5070> Call-ID: 7c10eaf0305d3dff6dac07db404d5a60(a)172.16.16.218 CSeq: 102 INVITE User-Agent: PANTHER-SC Max-Forwards: 69 Date: Sat, 27 Mar 2010 07:06:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 394 v=0 o=root 2434 2434 IN IP4 172.16.17.85 s=session c=IN IP4 172.16.17.85 t=0 0 m=audio 54168 RTP/SAVP 0 18 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv AES_CM_128_HMAC_SHA1_80 inline:qGi2zArgAgHDiUoqc0wbBofpBISpZINZxrqDoa7W . Mar 27 12:36:45 [5489] DBG:core:forward_request: orig. len=935, new_len=1182, proto=3 Mar 27 12:36:45 [5489] DBG:core:tcp_send: no open tcp connection found, opening new one Mar 27 12:36:45 [5489] DBG:core:print_ip: tcpconn_new: new tcp connection to: 172.16.17.85 Mar 27 12:36:45 [5489] DBG:core:tcpconn_new: on port 25923, type 3 Mar 27 12:36:45 [5489] DBG:core:tls_tcpconn_init: entered: Creating a whole new ssl connection Mar 27 12:36:45 [5489] DBG:core:tls_tcpconn_init: name based TLS client domains are disabled Mar 27 12:36:45 [5489] DBG:core:tls_tcpconn_init: no TLS client domain AVP set, looking for socket based TLS client domain Mar 27 12:36:45 [5489] DBG:core:tls_find_client_domain: virtual TLS client domain not found, Using default TLS client domain settings Mar 27 12:36:45 [5489] DBG:core:tls_tcpconn_init: found socket based TLS client domain [0.0.0.0:0] Mar 27 12:36:45 [5489] DBG:core:tls_tcpconn_init: Setting in CONNECT mode (client) Mar 27 12:36:45 [5489] DBG:core:tls_tcpconn_init: TLS server extension is not activated Mar 27 12:36:45 [5489] DBG:core:tls_tcpconn_init: TLS server name is not set, sending ClientHello without server_name Mar 27 12:36:45 [5489] DBG:core:tcp_send: sending... Mar 27 12:36:45 [5489] DBG:core:tls_update_fd: New fd is 12 Mar 27 12:36:45 [5499] DBG:core:handle_ser_child: read response= 7fd335f23098, 2, fd 25 from 4 (5489) Mar 27 12:36:45 [5499] DBG:core:tcpconn_add: hashes: 495, 4 Mar 27 12:36:45 [5499] DBG:core:io_watch_add: io_watch_add(0x74ee80, 25, 2, 0x7fd335f23098), fd_no=17 Mar 27 12:36:45 [5499] DBG:core:handle_tcpconn_ev: data available on 0x7fd335f23098 25 Mar 27 12:36:45 [5489] ERROR:core:tls_connect: SSL_ERROR_SYSCALL err=Connection reset by peer(104) Mar 27 12:36:45 [5499] DBG:core:io_watch_del: io_watch_del (0x74ee80, 25, -1, 0x0) fd_no=18 called Mar 27 12:36:45 [5489] ERROR:core:tls_connect: something wrong in SSL: 5 (ret=-1) err=Connection reset by peer(104) Mar 27 12:36:45 [5499] DBG:core:send2child: to tcp child 1 0(5494), 0x7fd335f23098 Mar 27 12:36:45 [5489] DBG:core:tcp_send: after write: c= 0x7fd335f23098 n=-1 fd=12 Mar 27 12:36:45 [5489] DBG:core:tcp_send: buf= INVITE sip:1002@172.16.17.85:25923;rinstance=133b23266b1f2e83;transport=TLS SIP/2.0 Record-Route: <sip:172.16.16.218:5091;transport=tls;r2=on;lr=on> Record-Route: <sip:172.16.16.218:5090;r2=on;lr=on> Via: SIP/2.0/TLS 172.16.16.218:5091;branch=z9hG4bK7f71d414 Via: SIP/2.0/UDP 172.16.16.218:5070;received=172.16.16.218;branch=z9hG4bK7f71d414;rport=5070 From: "1003" <sip:1003@172.16.16.218:5070>;tag=as7351a581 To: <sip:1002@172.16.16.218:5090> Contact: <sip:1003@172.16.16.218:5070> Call-ID: 7c10eaf0305d3dff6dac07db404d5a60(a)172.16.16.218 CSeq: 102 INVITE User-Agent: PANTHER-SC Max-Forwards: 69 Date: Sat, 27 Mar 2010 07:06:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 394 v=0 o=root 2434 2434 IN IP4 172.16.17.85 s=session c=IN IP4 172.16.17.85 t=0 0 m=audio 54168 RTP/SAVP 0 18 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv AES_CM_128_HMAC_SHA1_80 inline:qGi2zArgAgHDiUoqc0wbBofpBISpZINZxrqDoa7W Mar 27 12:36:45 [5494] DBG:core:handle_io: received n=8 con=0x7fd335f23098, fd=20 Mar 27 12:36:45 [5489] ERROR:core:tcp_send: failed to send Mar 27 12:36:45 [5494] DBG:core:io_watch_add: io_watch_add(0x74efe0, 20, 2, 0x7fd335f23098), fd_no=1 Mar 27 12:36:45 [5489] ERROR:core:msg_send: tcp_send failed Mar 27 12:36:45 [5494] DBG:core:io_watch_del: io_watch_del (0x74efe0, 20, -1, 0x10) fd_no=2 called Mar 27 12:36:45 [5494] DBG:core:release_tcpconn: releasing con 0x7fd335f23098, state -2, fd=20, id=4 Mar 27 12:36:45 [5494] DBG:core:release_tcpconn: extra_data 0x7fd335f00ef8 Mar 27 12:36:45 [5489] DBG:sl:sl_reply_error: error text is Send failed (477/SL) Mar 27 12:36:45 [5499] DBG:core:handle_tcp_child: reader response= 7fd335f23098, -2 from 1 Mar 27 12:36:45 [5499] DBG:core:tcpconn_destroy: delaying (0x7fd335f23098, flags 0002) ... Mar 27 12:36:45 [5489] DBG:core:parse_headers: flags=ffffffffffffffff Mar 27 12:36:45 [5489] DBG:core:get_hdr_field: found end of header Mar 27 12:36:45 [5499] DBG:core:handle_ser_child: read response= 7fd335f23098, -2, fd -1 from 4 (5489) Mar 27 12:36:45 [5499] DBG:core:tcpconn_destroy: destroying connection 0x7fd335f23098, flags 0002 Mar 27 12:36:45 [5489] DBG:core:check_via_address: params 172.16.16.218, 172.16.16.218, 0 Mar 27 12:36:45 [5499] DBG:core:tls_close: closing SSL connection Mar 27 12:36:45 [5499] DBG:core:tls_update_fd: New fd is 25 Mar 27 12:36:45 [5499] DBG:core:tls_shutdown: shutdown successful Mar 27 12:36:45 [5499] DBG:core:tls_tcpconn_clean: Cleanup function entered Mar 27 12:36:45 [5486] DBG:core:parse_msg: SIP Request: Mar 27 12:36:45 [5486] DBG:core:parse_msg: method: <ACK> Mar 27 12:36:45 [5486] DBG:core:parse_msg: uri: <sip:1002@172.16.16.218:5090> Mar 27 12:36:45 [5486] DBG:core:parse_msg: version: <SIP/2.0> Mar 27 12:36:45 [5486] DBG:core:parse_headers: flags=2 Mar 27 12:36:45 [5486] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK7f71d414>; state=6 Mar 27 12:36:45 [5486] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Mar 27 12:36:45 [5486] DBG:core:parse_via: end of header reached, state=5 Mar 27 12:36:45 [5486] DBG:core:parse_headers: via found, flags=2 Mar 27 12:36:45 [5486] DBG:core:parse_headers: this is the first via Mar 27 12:36:45 [5486] DBG:core:receive_msg: After parse_msg... Mar 27 12:36:45 [5486] DBG:core:receive_msg: preparing to run routing scripts... Mar 27 12:36:45 [5486] DBG:core:parse_headers: flags=8 Mar 27 12:36:45 [5486] DBG:core:parse_to_param: tag=2cf7f9a22b6374f465182a893bba7dec.adb6 Mar 27 12:36:45 [5486] DBG:core:parse_to: end of header reached, state=29 Mar 27 12:36:45 [5486] DBG:core:parse_to: display={}, ruri={sip:1002@172.16.16.218:5090} Mar 27 12:36:45 [5486] DBG:core:get_hdr_field: <To> [73]; uri=[sip:1002@172.16.16.218:5090] Mar 27 12:36:45 [5486] DBG:core:get_hdr_field: to body [<sip:1002@172.16.16.218:5090>] Mar 27 12:36:45 [5486] DBG:sl:sl_filter_ACK: local ACK found -> dropping it! Mar 27 12:36:45 [5486] DBG:core:destroy_avp_list: destroying list (nil) Mar 27 12:36:45 [5486] DBG:core:receive_msg: cleaning up Mar 27 12:36:45 [5489] DBG:core:destroy_avp_list: destroying list (nil) Mar 27 12:36:45 [5489] DBG:core:receive_msg: cleaning up Mar 27 12:36:45 [5485] DBG:core:parse_msg: SIP Reply (status): Mar 27 12:36:45 [5485] DBG:core:parse_msg: version: <SIP/2.0> Mar 27 12:36:45 [5485] DBG:core:parse_msg: status: <404> Mar 27 12:36:45 [5485] DBG:core:parse_msg: reason: <Not Found> Mar 27 12:36:45 [5485] DBG:core:parse_headers: flags=2 Mar 27 12:36:45 [5485] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bKd1da.1a6f9b73.0>; state=6 Mar 27 12:36:45 [5485] DBG:core:parse_via_param: found param type 236, <i> = <1>; state=6 Mar 27 12:36:45 [5485] DBG:core:parse_via_param: found param type 234, <received> = <172.16.16.218>; state=16 Mar 27 12:36:45 [5485] DBG:core:parse_via: end of header reached, state=5 Mar 27 12:36:45 [5485] DBG:core:parse_headers: via found, flags=2 Mar 27 12:36:45 [5485] DBG:core:parse_headers: this is the first via Mar 27 12:36:45 [5485] DBG:core:receive_msg: After parse_msg... Mar 27 12:36:45 [5485] DBG:core:forward_reply: found module tm, passing reply to it Mar 27 12:36:45 [5485] DBG:tm:t_check: start=0xffffffffffffffff Mar 27 12:36:45 [5485] DBG:core:parse_headers: flags=22 Mar 27 12:36:45 [5485] DBG:core:parse_via_param: found param type 234, <received> = <172.16.17.85>; state=6 Mar 27 12:36:45 [5485] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d8754z-863b295f961d253f-1---d8754z->; state=6 Mar 27 12:36:45 [5485] DBG:core:parse_via_param: found param type 235, <rport> = <1301>; state=16 Mar 27 12:36:45 [5485] DBG:core:parse_via: end of header reached, state=5 Mar 27 12:36:45 [5485] DBG:core:parse_headers: via found, flags=22 Mar 27 12:36:45 [5485] DBG:core:parse_headers: parse_headers: this is the second via Mar 27 12:36:45 [5485] DBG:core:parse_to_param: tag=as06cb485b Mar 27 12:36:45 [5485] DBG:core:parse_to: end of header reached, state=29 Mar 27 12:36:45 [5485] DBG:core:parse_to: display={}, ruri={sip:1002@172.16.16.218:5091} Mar 27 12:36:45 [5485] DBG:core:get_hdr_field: <To> [46]; uri=[sip:1002@172.16.16.218:5091] Mar 27 12:36:45 [5485] DBG:core:get_hdr_field: to body [<sip:1002@172.16.16.218:5091>] Mar 27 12:36:45 [5485] DBG:core:get_hdr_field: cseq <CSeq>: <2> <INVITE> Mar 27 12:36:45 [5485] DBG:core:parse_headers: flags=8 Mar 27 12:36:45 [5485] DBG:tm:t_reply_matching: hash 44317 label 934934177 branch 0 Mar 27 12:36:45 [5485] DBG:tm:t_reply_matching: REF_UNSAFE: after is 1 Mar 27 12:36:45 [5485] DBG:tm:t_reply_matching: reply matched (T=0x7fd335eee6b0)! Mar 27 12:36:45 [5485] DBG:tm:t_check: end=0x7fd335eee6b0 Mar 27 12:36:45 [5485] DBG:tm:reply_received: org. status uas=100, uac[0]=100 local=0 is_invite=1) Mar 27 12:36:45 [5485] DBG:core:parse_headers: flags=8 Mar 27 12:36:45 [5485] DBG:tm:build_local: using FROM=<From: "1003"<sip:1003@172.16.16.218:5091>;tag=9f574228 >, TO=<To: <sip:1002@172.16.16.218:5091>;tag=as06cb485b >, CSEQ_N=<CSeq: 2> Mar 27 12:36:45 [5485] DBG:tm:extract_parsed_hdrs: ----parsing the buf req - first line Mar 27 12:36:45 [5485] DBG:core:parse_headers: flags=ffffffffffffffff Mar 27 12:36:45 [5485] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bKd1da.1a6f9b73.0>; state=6 Mar 27 12:36:45 [5485] DBG:core:parse_via_param: found param type 236, <i> = <1>; state=16 Mar 27 12:36:45 [5485] DBG:core:parse_via: end of header reached, state=5 Mar 27 12:36:45 [5485] DBG:core:parse_headers: via found, flags=ffffffffffffffff Mar 27 12:36:45 [5485] DBG:core:parse_headers: this is the first via Mar 27 12:36:45 [5485] DBG:core:parse_via_param: found param type 234, <received> = <172.16.17.85>; state=6 Mar 27 12:36:45 [5485] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d8754z-863b295f961d253f-1---d8754z->; state=6 Mar 27 12:36:45 [5485] DBG:core:parse_via_param: found param type 235, <rport> = <1301>; state=16 Mar 27 12:36:45 [5485] DBG:core:parse_via: end of header reached, state=5 Mar 27 12:36:45 [5485] DBG:core:parse_headers: via found, flags=ffffffffffffffff Mar 27 12:36:45 [5485] DBG:core:parse_headers: parse_headers: this is the second via Mar 27 12:36:45 [5485] DBG:core:parse_to: end of header reached, state=10 Mar 27 12:36:45 [5485] DBG:core:parse_to: display={}, ruri={sip:1002@172.16.16.218:5091} Mar 27 12:36:45 [5485] DBG:core:get_hdr_field: <To> [31]; uri=[sip:1002@172.16.16.218:5091] Mar 27 12:36:45 [5485] DBG:core:get_hdr_field: to body [<sip:1002@172.16.16.218:5091> ] Mar 27 12:36:45 [5485] DBG:core:get_hdr_field: cseq <CSeq>: <2> <INVITE> Mar 27 12:36:45 [5485] DBG:core:get_hdr_field: content_length=512 Mar 27 12:36:45 [5485] DBG:core:get_hdr_field: found end of header Mar 27 12:36:45 [5485] DBG:tm:t_should_relay_response: T_code=100, new_code=404 Mar 27 12:36:45 [5485] DBG:tm:t_pick_branch: picked branch 0, code 404 Mar 27 12:36:45 [5485] DBG:tm:is_3263_failure: dns-failover test: branch=0, last_recv=404, flags=2 Mar 27 12:36:45 [5485] WARNING:core:run_actions: null action list (rec_level=1) Mar 27 12:36:45 [5485] DBG:tm:relay_reply: branch=0, save=0, relay=0 Mar 27 12:36:45 [5485] DBG:core:parse_headers: flags=2000 Mar 27 12:36:45 [5485] DBG:core:get_hdr_field: content_length=0 Mar 27 12:36:45 [5485] DBG:core:build_res_buf_from_sip_res: old size: 548, new size: 454 Mar 27 12:36:45 [5485] DBG:core:build_res_buf_from_sip_res: copied size: orig:117, new: 23, rest: 431 msg= SIP/2.0 404 Not Found Via: SIP/2.0/TLS 172.16.17.85:4403;received=172.16.17.85;branch=z9hG4bK-d8754z-863b295f961d253f-1---d8754z-;rport=1301 From: "1003"<sip:1003@172.16.16.218:5091>;tag=9f574228 To: <sip:1002@172.16.16.218:5091>;tag=as06cb485b Call-ID: M2Y2OThmYmYwZjM4ODhhYzk4Mjc0MGFhMDQyNDUzZWU. CSeq: 2 INVITE User-Agent: PANTHER-SC Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 Mar 27 12:36:45 [5485] DBG:tm:set_timer: relative timeout is 30 Mar 27 12:36:45 [5485] DBG:tm:insert_timer_unsafe: [0]: 0x7fd335eee828 (83) Mar 27 12:36:45 [5485] DBG:core:tcp_send: tcp connection found (0x7fd335eddc18), acquiring fd Mar 27 12:36:45 [5485] DBG:core:tcp_send: c= 0x7fd335eddc18, n=16 Mar 27 12:36:45 [5499] DBG:core:handle_ser_child: read response= 7fd335eddc18, 1, fd -1 from 1 (5485) Mar 27 12:36:45 [5485] DBG:core:tcp_send: after receive_fd: c= 0x7fd335eddc18 n=8 fd=9 Mar 27 12:36:45 [5485] DBG:core:tcp_send: sending... Mar 27 12:36:45 [5485] DBG:core:tls_update_fd: New fd is 9 Mar 27 12:36:45 [5485] DBG:core:tls_write: write was successful (454 bytes) Mar 27 12:36:45 [5485] DBG:core:tcp_send: after write: c= 0x7fd335eddc18 n=454 fd=9 Mar 27 12:36:45 [5485] DBG:core:tcp_send: buf= SIP/2.0 404 Not Found Via: SIP/2.0/TLS 172.16.17.85:4403;received=172.16.17.85;branch=z9hG4bK-d8754z-863b295f961d253f-1---d8754z-;rport=1301 From: "1003"<sip:1003@172.16.16.218:5091>;tag=9f574228 To: <sip:1002@172.16.16.218:5091>;tag=as06cb485b Call-ID: M2Y2OThmYmYwZjM4ODhhYzk4Mjc0MGFhMDQyNDUzZWU. CSeq: 2 INVITE User-Agent: PANTHER-SC Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 Mar 27 12:36:45 [5485] DBG:tm:relay_reply: sent buf=0x7916f0: SIP/2.0 4..., shmem=0x7fd335f007a8: SIP/2.0 4 Mar 27 12:36:45 [5485] DBG:tm:cleanup_uac_timers: RETR/FR timers reset Mar 27 12:36:45 [5485] DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Mar 27 12:36:45 [5485] DBG:core:destroy_avp_list: destroying list (nil) Mar 27 12:36:45 [5485] DBG:core:receive_msg: cleaning up Mar 27 12:36:45 [5492] DBG:core:tls_update_fd: New fd is 18 Mar 27 12:36:45 [5492] DBG:core:_tls_read: 343 bytes read Mar 27 12:36:45 [5492] DBG:core:tcp_read_req: content-length= 0 Mar 27 12:36:45 [5492] DBG:core:parse_msg: SIP Request: Mar 27 12:36:45 [5492] DBG:core:parse_msg: method: <ACK> Mar 27 12:36:45 [5492] DBG:core:parse_msg: uri: <sip:1002@172.16.16.218:5091;transport=tls> Mar 27 12:36:45 [5492] DBG:core:parse_msg: version: <SIP/2.0> Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=2 Mar 27 12:36:45 [5492] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d8754z-863b295f961d253f-1---d8754z->; state=6 Mar 27 12:36:45 [5492] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Mar 27 12:36:45 [5492] DBG:core:parse_via: end of header reached, state=5 Mar 27 12:36:45 [5492] DBG:core:parse_headers: via found, flags=2 Mar 27 12:36:45 [5492] DBG:core:parse_headers: this is the first via Mar 27 12:36:45 [5492] DBG:core:receive_msg: After parse_msg... Mar 27 12:36:45 [5492] DBG:core:receive_msg: preparing to run routing scripts... Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=8 Mar 27 12:36:45 [5492] DBG:core:parse_to_param: tag=as06cb485b Mar 27 12:36:45 [5492] DBG:core:parse_to: end of header reached, state=29 Mar 27 12:36:45 [5492] DBG:core:parse_to: display={}, ruri={sip:1002@172.16.16.218:5091} Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: <To> [46]; uri=[sip:1002@172.16.16.218:5091] Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: to body [<sip:1002@172.16.16.218:5091>] Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=100 Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: cseq <CSeq>: <2> <ACK> Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: content_length=0 Mar 27 12:36:45 [5492] DBG:core:get_hdr_field: found end of header Mar 27 12:36:45 [5492] DBG:maxfwd:is_maxfwd_present: max_forwards header not found! Mar 27 12:36:45 [5492] DBG:siputils:has_totag: totag found Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=200 Mar 27 12:36:45 [5492] DBG:rr:find_first_route: No Route headers found Mar 27 12:36:45 [5492] DBG:rr:loose_route: There is no Route HF Mar 27 12:36:45 [5492] DBG:core:parse_headers: flags=78 Mar 27 12:36:45 [5492] DBG:tm:t_lookup_request: start searching: hash=44317, isACK=1 Mar 27 12:36:45 [5492] DBG:tm:matching_3261: RFC3261 transaction matched, tid=-d8754z-863b295f961d253f-1---d8754z- Mar 27 12:36:45 [5492] DBG:tm:t_lookup_request: REF_UNSAFE: after is 1 Mar 27 12:36:45 [5492] DBG:tm:t_lookup_request: transaction found (T=0x7fd335eee6b0) Mar 27 12:36:45 [5492] DBG:tm:cleanup_uac_timers: RETR/FR timers reset Mar 27 12:36:45 [5492] DBG:tm:insert_timer_unsafe: [2]: 0x7fd335eee730 (58) Mar 27 12:36:45 [5492] DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Mar 27 12:36:45 [5492] DBG:core:destroy_avp_list: destroying list (nil) Mar 27 12:36:45 [5492] DBG:core:receive_msg: cleaning up Mar 27 12:36:45 [5492] DBG:core:tls_update_fd: New fd is 18 Mar 27 12:36:45 [5492] DBG:core:_tls_read: 4 bytes read Mar 27 12:36:45 [5492] DBG:core:tcp_read_req: content-length= 0 Mar 27 12:36:45 [5492] DBG:core:tcp_send: tcp connection found (0x7fd335eddc18), acquiring fd Mar 27 12:36:45 [5492] DBG:core:tcp_send: c= 0x7fd335eddc18, n=16 Mar 27 12:36:45 [5499] DBG:core:handle_ser_child: read response= 7fd335eddc18, 1, fd -1 from 7 (5492) Mar 27 12:36:45 [5492] DBG:core:tcp_send: after receive_fd: c= 0x7fd335eddc18 n=8 fd=19 Mar 27 12:36:45 [5492] DBG:core:tcp_send: sending... Mar 27 12:36:45 [5492] DBG:core:tls_update_fd: New fd is 19 Mar 27 12:36:45 [5492] DBG:core:tls_write: write was successful (2 bytes) Mar 27 12:36:45 [5492] DBG:core:tcp_send: after write: c= 0x7fd335eddc18 n=2 fd=19 Mar 27 12:36:45 [5492] DBG:core:tcp_send: buf= one more thing, i have got two clients from one PC, still look at netstat result hemanshu@localhost user]$ netstat -antp | grep kamailio (Not all processes could be identified, non-owned process info will not be shown, you would have to be root to see it all.) tcp 0 0 172.16.16.218:5091 0.0.0.0:* LISTEN 4947/kamailio tcp 0 0 172.16.16.218:5092 0.0.0.0:* LISTEN 4947/kamailio tcp 0 0 172.16.16.218:5091 172.16.17.85:1295 ESTABLISHED 4964/kamailio tcp 0 0 172.16.16.218:5091 172.16.17.85:1296 ESTABLISHED 4964/kamailio tcp 0 0 172.16.16.218:5091 172.16.17.85:1289 ESTABLISHED 4964/kamailio tcp 0 0 172.16.16.218:5091 172.16.17.85:1293 ESTABLISHED 4964/kamailio tcp 0 0 172.16.16.218:5091 172.16.17.85:1291 ESTABLISHED 4964/kamailio tcp 0 0 172.16.16.218:5091 172.16.17.85:1286 ESTABLISHED 4964/kamailio tcp 0 0 172.16.16.218:5091 172.16.17.85:1285 ESTABLISHED 4964/kamailio Both clients keep creating new connection, and not reusing the already connected connection. i this this si the problem ---- Regards, Hemanshu Patel Sr. Software Engg Mo:09601295238 On Sat 27/03/10 11:37 AM , Hemanshu Patel <hemanshu.patel(a)saicare.com> wrote: > With TLS it still is not working > Mar 27 11:39:16 [4421] INFO:core:probe_max_receive_buffer: using a > UDP receive buffer of 255 kb > Mar 27 11:39:16 [4425] WARNING:usrloc:dbrow2info: non-local socket > ...ignoring > Mar 27 11:39:18 [4431] INFO:core:tls_accept: client did not present > a certificate > Mar 27 11:39:55 [4432] INFO:core:tls_accept: client did not present > a certificate > Mar 27 11:40:01 [4431] ERROR:rr:w_record_route: Double attempt to > record-route > Mar 27 11:40:01 [4426] ERROR:core:tls_connect: SSL_ERROR_SYSCALL > err=Connection reset by peer(104) > Mar 27 11:40:01 [4426] ERROR:core:tls_connect: something wrong in > SSL: 5 (ret=-1) err=Connection reset by peer(104) > Mar 27 11:40:01 [4426] ERROR:core:tcp_send: failed to send > Mar 27 11:40:01 [4426] ERROR:core:msg_send: tcp_send failed > Mar 27 11:40:01 [4425] WARNING:core:run_actions: null action list > (rec_level=1) > my doubt is that if somethings wrong with SSL certificates that i > created myself (i.e my own root CA, and self signed certificates), > then how come registration is working like charm? > if there is some problem with Certificates then registration should > also not work. > am i right? > ---- > Regards, > Hemanshu Patel > Sr. Software Engg > SIS,Ahmedabad > Mo:09601295238 > On Sat 27/03/10 9:50 AM , "Hemanshu Patel" wrote: > > I havent tested over TCP, let me check it > > but hardphone, i mean hardware based phones from grandstream > gvx3140 > > works > > fine with same implementation on TLS. > > -- > > Regards, > > Hemanshu Patel > > M: 09601295238 > > > Does eyebeam with SIP over TCP is working? > > > > > > Am 26.03.2010 13:43, schrieb Hemanshu Patel: > > >> > > >> i am still having this problem. > > >> when i use two grandstream phone everything works fine, > > >> i can make calls on TLS and users can talk to each other. > > >> > > >> But when i use two eyebream phone, it doesnt work, gives > > following error > > >> > > >> :33 [2875] WARNING:core:init_ssl_ctx_behavior: server > > verification NOT > > >> activated. Weaker security. > > >> [ panreg-tls]$ > > >> [ panreg-tls]$ > > >> [ panreg-tls]$ Mar 26 18:11:59 [2889] > > >> ERROR:rr:w_record_route: Double attempt to record-route > > >> Mar 26 18:12:09 [2884] ERROR:core:tcp_blocking_connect: timeout > > 10 s > > >> elapsed from 10 s > > >> Mar 26 18:12:09 [2884] ERROR:core:tcpconn_connect: > > tcp_blocking_connect > > >> failed > > >> Mar 26 18:12:09 [2884] ERROR:core:tcp_send: connect failed > > >> Mar 26 18:12:09 [2884] ERROR:core:msg_send: tcp_send failed > > >> Mar 26 18:12:10 [2883] ERROR:core:tcp_blocking_connect: timeout > > 10 s > > >> elapsed from 10 s > > >> Mar 26 18:12:10 [2883] ERROR:core:tcpconn_connect: > > tcp_blocking_connect > > >> failed > > >> Mar 26 18:12:10 [2883] ERROR:core:tcp_send: connect failed > > >> Mar 26 18:12:10 [2883] ERROR:core:msg_send: tcp_send failed > > >> Mar 26 18:12:10 [2883] WARNING:core:run_actions: null action > list > > >> (rec_level=1) > > >> Mar 26 18:12:11 [2881] ERROR:core:tcp_blocking_connect: timeout > > 10 s > > >> elapsed from 10 s > > >> Mar 26 18:12:11 [2881] ERROR:core:tcpconn_connect: > > tcp_blocking_connect > > >> failed > > >> Mar 26 18:12:11 [2881] ERROR:core:tcp_send: connect failed > > >> Mar 26 18:12:11 [2881] ERROR:core:msg_send: tcp_send failed > > >> Mar 26 18:12:13 [2882] ERROR:core:tcp_blocking_connect: timeout > > 10 s > > >> elapsed from 10 s > > >> Mar 26 18:12:13 [2882] ERROR:core:tcpconn_connect: > > tcp_blocking_connect > > >> failed > > >> Mar 26 18:12:13 [2882] ERROR:core:tcp_send: connect failed > > >> Mar 26 18:12:13 [2882] ERROR:core:msg_send: tcp_send failed > > >> Mar 26 18:12:19 [2884] ERROR:core:tcp_blocking_connect: timeout > > 10 s > > >> elapsed from 10 s > > >> Mar 26 18:12:19 [2884] ERROR:core:tcpconn_connect: > > tcp_blocking_connect > > >> failed > > >> Mar 26 18:12:19 [2884] ERROR:core:tcp_send: connect failed > > >> Mar 26 18:12:19 [2884] ERROR:core:msg_send: tcp_send failed > > >> > > >> > > >> > > >> Any idea? Same configuration works with grandstream, while they > > cant in > > >> softphone? and gives 477 errot code. > > >> > > >> my asterisk log is as below: > > >> > > >> 26 18:08:09] WARNING[2833]: chan_sip.c:1648 setup_crypto: > > Hemanshu: > > >> local_key641 jhbAJ7jFE1p/Ngn4kVxy7qTTmkJpeBiN6W98+gmM len 40 > > >> -- Called kamailio/1003 > > >> -- Got SIP response 477 "Send failed (477/SL)" back from > > >> 172.16.16.218 > > >> -- No one is available to answer at this time (1:0/0/0) > > >> -- Executing [_default:4] Hangup("SIP/5091-8c001430", "") > > >> in > > >> new stack > > >> == Spawn extension (hemu_default, 1003, 4) exited non-zero > on > > >> 'SIP/5091-8c001430' > > >> > ::Disconnected form Oracle, trying to connect > again.. > > >> > ::Tried a lot, not getting connected.. > > >> -- Got SIP response 477 "Send failed (477/SL)" back from > > >> 172.16.16.218 > > >> > > >> > > >> Any idea what could be the problem? > > >> > > > > > _______________________________________________ > > Kamailio (OpenSER) - Users mailing list > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > > > > ---- SIS Blade Server, Desktop & Thin Client is now available on DGS > & D rate contract. For more detail kindly visit our website > http://www.saicare.com > > ---- SIS Blade Server, Desktop & Thin Client is now available on DGS & D rate contract. For more detail kindly visit our website http://www.saicare.com

2 1

INVITE timers
by Alex Hermann 27 Mar '10

27 Mar '10

Hello, I'm trying to create a scenarion which was very simple with Kamailio 1.x, but seems impossible with sr3.0. I need to do failover for INVITE's and also limit the call setup time per-branch, so I defined the problem in 2 parts that need to work separately, but also combined into 1 proxy. 1) Fast failover. After t_relay, the destination should be tried for max 2 sec per host. No maximum for the whole set. Thus, if a destination has 4 SRV records, and each host fails after 2 secs, a 408 will be sent back after a total of 8 secs. If a (proivisional) response arrives within 2 sec, a 408 MUST NOT be sent. The exact value of the timer should be configurable per t_relay call. In kamailio this is accomplished by setting the fr_timer(_avp) to 2000ms. In sr3.0 I have yet to find a combination of timer settings that work. 2) Call setup duration. If a destination host returns a response, I want to limit the attempt to 80 secs (default, should be configurable per t_relay call) of ringing/progress/whatever. After that a failure route will take care to route to a different destination, at this time this timer should be restarted. At no other time after the _first_ provisional reponse after the t_relay call should the timer be restarted. In Kamailio this was accomplished by just setting fr_inv_timer(_avp) to the desired value before t_relay(). In sr3.0 this timer will be restarted after additional provisional responses and will not be resetted after a new t_relay call (from failure_route). Any suggestions on how to get the desired behaviours are welcomed (at the moment scenario 1 is more important than 2). -- Greetings, Alex Hermann

3 6

Re: [Kamailio-Users] Problem with secure TLS call
by Hemanshu Patel 27 Mar '10

27 Mar '10

Yups, it works fine with TCP. All packets are forwarded to other party perfectly and everything work smooth. only with TLS, on softphone it doesnt work. any more suggestions ---- Regards, Hemanshu Patel Sr. Software Engg SIS,Ahmedabad Mo:09601295238 On Sat 27/03/10 9:50 AM , "Hemanshu Patel" <hemanshu.patel(a)saicare.com> wrote: > I havent tested over TCP, let me check it > but hardphone, i mean hardware based phones from grandstream gvx3140 > works > fine with same implementation on TLS. > -- > Regards, > Hemanshu Patel > M: 09601295238 > > Does eyebeam with SIP over TCP is working? > > > > Am 26.03.2010 13:43, schrieb Hemanshu Patel: > >> > >> i am still having this problem. > >> when i use two grandstream phone everything works fine, > >> i can make calls on TLS and users can talk to each other. > >> > >> But when i use two eyebream phone, it doesnt work, gives > following error > >> > >> :33 [2875] WARNING:core:init_ssl_ctx_behavior: server > verification NOT > >> activated. Weaker security. > >> [ panreg-tls]$ > >> [ panreg-tls]$ > >> [ panreg-tls]$ Mar 26 18:11:59 [2889] > >> ERROR:rr:w_record_route: Double attempt to record-route > >> Mar 26 18:12:09 [2884] ERROR:core:tcp_blocking_connect: timeout > 10 s > >> elapsed from 10 s > >> Mar 26 18:12:09 [2884] ERROR:core:tcpconn_connect: > tcp_blocking_connect > >> failed > >> Mar 26 18:12:09 [2884] ERROR:core:tcp_send: connect failed > >> Mar 26 18:12:09 [2884] ERROR:core:msg_send: tcp_send failed > >> Mar 26 18:12:10 [2883] ERROR:core:tcp_blocking_connect: timeout > 10 s > >> elapsed from 10 s > >> Mar 26 18:12:10 [2883] ERROR:core:tcpconn_connect: > tcp_blocking_connect > >> failed > >> Mar 26 18:12:10 [2883] ERROR:core:tcp_send: connect failed > >> Mar 26 18:12:10 [2883] ERROR:core:msg_send: tcp_send failed > >> Mar 26 18:12:10 [2883] WARNING:core:run_actions: null action list > >> (rec_level=1) > >> Mar 26 18:12:11 [2881] ERROR:core:tcp_blocking_connect: timeout > 10 s > >> elapsed from 10 s > >> Mar 26 18:12:11 [2881] ERROR:core:tcpconn_connect: > tcp_blocking_connect > >> failed > >> Mar 26 18:12:11 [2881] ERROR:core:tcp_send: connect failed > >> Mar 26 18:12:11 [2881] ERROR:core:msg_send: tcp_send failed > >> Mar 26 18:12:13 [2882] ERROR:core:tcp_blocking_connect: timeout > 10 s > >> elapsed from 10 s > >> Mar 26 18:12:13 [2882] ERROR:core:tcpconn_connect: > tcp_blocking_connect > >> failed > >> Mar 26 18:12:13 [2882] ERROR:core:tcp_send: connect failed > >> Mar 26 18:12:13 [2882] ERROR:core:msg_send: tcp_send failed > >> Mar 26 18:12:19 [2884] ERROR:core:tcp_blocking_connect: timeout > 10 s > >> elapsed from 10 s > >> Mar 26 18:12:19 [2884] ERROR:core:tcpconn_connect: > tcp_blocking_connect > >> failed > >> Mar 26 18:12:19 [2884] ERROR:core:tcp_send: connect failed > >> Mar 26 18:12:19 [2884] ERROR:core:msg_send: tcp_send failed > >> > >> > >> > >> Any idea? Same configuration works with grandstream, while they > cant in > >> softphone? and gives 477 errot code. > >> > >> my asterisk log is as below: > >> > >> 26 18:08:09] WARNING[2833]: chan_sip.c:1648 setup_crypto: > Hemanshu: > >> local_key641 jhbAJ7jFE1p/Ngn4kVxy7qTTmkJpeBiN6W98+gmM len 40 > >> -- Called kamailio/1003 > >> -- Got SIP response 477 "Send failed (477/SL)" back from > >> 172.16.16.218 > >> -- No one is available to answer at this time (1:0/0/0) > >> -- Executing [_default:4] Hangup("SIP/5091-8c001430", "") > >> in > >> new stack > >> == Spawn extension (hemu_default, 1003, 4) exited non-zero on > >> 'SIP/5091-8c001430' > >> > ::Disconnected form Oracle, trying to connect again.. > >> > ::Tried a lot, not getting connected.. > >> -- Got SIP response 477 "Send failed (477/SL)" back from > >> 172.16.16.218 > >> > >> > >> Any idea what could be the problem? > >> > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > ---- SIS Blade Server, Desktop & Thin Client is now available on DGS & D rate contract. For more detail kindly visit our website http://www.saicare.com

1 0

[Kamailio-Users] Problem with secure TLS call
by Detlef Pilzecker 27 Mar '10

27 Mar '10

Hi, I am using kamailio 3.0.1 (x86_64/linux) 0822a9 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. and I am using the client "PhonerLite" (http://www.phonerlite.de). The phone is registered on my server with TLS. If I call someone or myself I get an error message. What is wrong with my server? (TLS calls with PhonerLite with the provider antisip.com are possible without problems!) With UDP calling myself I get "486:Busy Here" on the phone, this is OK and normal. With TLS I get "477:Unfortunately error on sending to next hop occurred(477/SL)" on the phone and some ERROR messages like the following on the kamailio.log (you can see the rest of the log in the attached zip): Mar 21 20:24:00 vs208140 /usr/local/sbin/kamailio[20013]: DEBUG: <core> [msg_translator.c:200]: check_via_address(95.90.205.74, 95.90.205.74, 0) Mar 21 20:24:00 vs208140 /usr/local/sbin/kamailio[20013]: DEBUG: <core> [tcp_main.c:1786]: tcp_send: no open tcp connection found, opening new one Mar 21 20:24:10 vs208140 /usr/local/sbin/kamailio[20013]: ERROR: <core> [tcp_main.c:618]: connect 95.90.205.74:5061 failed (timeout) Mar 21 20:24:10 vs208140 /usr/local/sbin/kamailio[20013]: ERROR: <core> [tcp_main.c:621]: ERROR: tcp_blocking_connect 95.90.205.74:5061: timeout 10 s elapsed from 10 s Mar 21 20:24:10 vs208140 /usr/local/sbin/kamailio[20013]: ERROR: <core> [tcp_main.c:1168]: ERROR: tcp_do_connect: tcp_blocking_connect 95.90.205.74:5061 failed Mar 21 20:24:10 vs208140 /usr/local/sbin/kamailio[20013]: ERROR: <core> [tcp_main.c:1237]: ERROR: tcp_do_connect 95.90.205.74:5061: failed (115) Operation now in progress Mar 21 20:24:10 vs208140 /usr/local/sbin/kamailio[20013]: ERROR: <core> [tcp_main.c:1955]: ERROR: tcp_send 95.90.205.74:5061: connect failed Mar 21 20:24:10 vs208140 /usr/local/sbin/kamailio[20013]: ERROR: tm [../../forward.h:191]: msg_send: ERROR: tcp_send failed The kamailio.cfg file is also in the zip attached. The PC is connected with the router (I opened the TCP ports 5060-5062) that is connected to the Internet. The problem also happens if I disable the firewall of the server and of the PC. Can someone give me help? Thanks in advance! Regards Detlef Pilzecker

3 8

topoh
by dotnetdub 26 Mar '10

26 Mar '10

Hi List, We are using Kamailo 3.01 and testing topoh for starters: Kamailio3 = IP of Kamailio Server and Asterisk=IP of Asterisk server Observations: U 2010/03/26 12:05:56.665583 Kamailio3:5060 -> Asterisk:5060 SIP/2.0 100 trying -- your call is important to us^@ Via: SIP/2.0/UDP Kamailio3:5060;branch=z9hG4bK08fffea4;rport=5060^@ From: "xxxxxxxxx" <sip:xxxxx@asterisk>;tag=as7ac63338^@ To: <sip:Xxxxxxx@Kamailio3>^@ Call-ID: 2f7a1c040f4920d5591695956ec8c42c@Kamailio3^@ CSeq: 103 INVITE^@ Server: kamailio (3.0.1 (i386/linux))^@ Content-Length: 0^@ Warning: 392 Kamailio:5060 "Noisy feedback tells: pid=23782 req_src_ip=Asterisk req_src_port=5060 in_uri=sip:xxxxxx@Kamailioout_uri=sip:Translated@ *UPSTREAMPROVIDERSIP* Provider via_cnt==1"^@ The noisy feedback contains everything that is hidden by topoh in the contact: Contact: <sip:10.1.1.10;line=sr-N6IAzBVAWxFAzx3lzxclMxylPxcsOBVlWGZ6WJZfWlq-W.y6My**>^ Also: U 2010/03/26 12:06:00.125828 Kamailio3:5060 -> Asterisk:5060 SIP/2.0 180 Ringing^@ From: "xxxxxxx"<sip:xxxxxxxx@Kamailio3>;tag=as7ac63338^@ To: <sip:xxxxxx@Kamailio3 >;tag=fccd978-b342ea59-13c4-50022-82e974-27d2fa70-82e974^@ Call-ID: 2f7a1c040f4920d5591695956ec8c42c@Kamailio3^@ CSeq: 103 INVITE^@ Via: SIP/2.0/UDP Asterisk:5060;branch=z9hG4bK08fffea4;rport=5060^@ Record-Route: <sip:10.1.1.10;line=sr-N6IAzBclOBF4WLZfWBF7M.VLz6fLHRQ7z6srpxusg9MXgSIqHReEgJZwgSKEgSjLMc**>^@ Record-Route: <sip:Kamailio3;lr=on;did=bff.60851b12>^@ Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY^@ Contact: <sip:10.1.1.10;line=sr-N6IAzBVAWxFAzx3lzxclMxylPxcsOBVlWGZ6WJZfWlq-W.y6My**>^@ Content-Type: application/sdp^@ Content-Length: 228^@ ^@ v=0^@ *o=Upstream Switch Name 160624700 160624700 IN IP4 Upstream Providers IP^@* *s=Upstream Switch Name^@* * * The upstream info appears here also. I am using engage_media_proxy and using topoh with defaul parameters. Am I missing something ? Thanks, Stephen.

3 6

[Kamailio-Users] Recording RTP session
by Hector.Ortiz＠swisscom.com 26 Mar '10

26 Mar '10

Hi everybody I've configured a SIP server (OpenSER 1.3.2) to work with rtpproxy and so far everything is fine, but now I want to be able to record the audio from a conversation. I've read that this is possible by using the function start_recording from the module NATHelper but I'm not able to get it working RTPproxy 1.2.1 is running with the following options: root@openser:/opt/rtpproxy-1.2.1/bin# ps uax | grep rtpproxy rtpproxy 30827 0.0 0.0 26968 964 ? Ssl 14:49 0:00 /opt/rtpproxy-1.2.1/bin/rtpproxy -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -l 192.168.34.1 -s udp:127.0.0.1 7890 -r /etc/openser/stored_conversations -S /tmp -a -d DBUG -P root 31547 0.0 0.0 7524 892 pts/1 R+ 15:59 0:00 grep rtpproxy Output from netstat root@openser:/opt/rtpproxy-1.2.1/bin# netstat -tupna | grep rtpproxy udp 0 0 127.0.0.1:7890 0.0.0.0:* 30827/rtpproxy Relevant parts of the configuration file ... # ------ nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@openser.org") modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7890") modparam("nathelper", "force_socket", "udp:localhost:7890") ... route[6] { xlog("L_INFO", "ROUTE 6"); # NAT handling # Set reply routing block, to which control is passed after a transaction # completed with a negative result but before sending a final reply t_on_failure("1"); # Check if we're NATed if (isflagset(5) || isbflagset(6)) { # Use rtpproxy force_rtp_proxy(); if ( start_recording() ) { xlog("L_INFO", "RECORDING..."); }; } # Set reply routing block, to which control is passed each time a reply # (provisional or final) for the transaction is received t_on_reply("1"); } ... onreply_route[1] { xlog("L_INFO", "ONREPLY_ROUTE"); if ((isflagset(5) || isbflagset(6)) && status=~"(180)|(183)|(2[0-9][0-9])") { xlog("L_INFO", "NAT'd TRANSACTION"); fix_nated_contact(); force_rtp_proxy(); if ( start_recording() ) { xlog("L_INFO", "RECORDING..."); }; } else if (nat_uac_test("1")) { fix_nated_contact(); } exit; } When I start OpenSER I see the following in syslog Mar 19 16:08:45 openser /usr/sbin/openser[31630]: NOTICE:core:main: version: openser 1.3.2-notls (x86_64/linux) Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:core:main: using 256 Mb shared memory Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:core:main: using 1 Mb private memory per process Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:sl:mod_init: Initializing StateLess engine Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:tm:mod_init: TM - initializing... Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:maxfwd:mod_init: initializing... Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:usrloc:ul_init_locks: locks array size 512 Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:registrar:mod_init: initializing... Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:textops:mod_init: initializing... Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:xlog:mod_init: initializing... Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:avpops:avpops_init: initializing... Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:acc:mod_init: initializing... Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:dialog:mod_init: Dialog module - initializing Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 255 kb Mar 19 16:08:45 openser /usr/sbin/openser[31630]: INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 255 kb Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31632_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31632_0 20040107 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31632_1 VF 20050322" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31632_1 1 " Mar 19 16:08:45 openser /usr/sbin/openser[31632]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31631_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31631_0 20040107 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31633_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31633_0 20040107 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31634_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31634_0 20040107 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31635_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31635_0 20040107 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31636_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31636_0 20040107 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31637_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31637_0 20040107 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31638_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31638_0 20040107 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31637_1 VF 20050322" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31637_1 1 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31631_1 VF 20050322" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31631_1 1 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31634_1 VF 20050322" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31634_1 1 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31636_1 VF 20050322" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31636_1 1 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31633_1 VF 20050322" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31633_1 1 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31635_1 VF 20050322" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31635_1 1 " Mar 19 16:08:45 openser /usr/sbin/openser[31631]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled Mar 19 16:08:45 openser /usr/sbin/openser[31634]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled Mar 19 16:08:45 openser /usr/sbin/openser[31637]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled Mar 19 16:08:45 openser /usr/sbin/openser[31633]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled Mar 19 16:08:45 openser /usr/sbin/openser[31636]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled Mar 19 16:08:45 openser /usr/sbin/openser[31635]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled Mar 19 16:08:45 openser /usr/sbin/openser[31638]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31639_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31639_0 20040107 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31638_1 VF 20050322" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31638_1 1 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31640_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31640_0 20040107 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31642_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31642_0 20040107 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31643_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31643_0 20040107 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31644_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31644_0 20040107 " Mar 19 16:08:45 openser /usr/sbin/openser[31642]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled Mar 19 16:08:45 openser /usr/sbin/openser[31643]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31645_0 V" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31645_0 20040107 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31639_1 VF 20050322" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31639_1 1 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31642_1 VF 20050322" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31642_1 1 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31644_1 VF 20050322" Mar 19 16:08:45 openser /usr/sbin/openser[31644]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31644_1 1 " Mar 19 16:08:45 openser /usr/sbin/openser[31640]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled Mar 19 16:08:45 openser /usr/sbin/openser[31639]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31640_1 VF 20050322" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31640_1 1 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31643_1 VF 20050322" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31643_1 1 " Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:handle_command: received command "31645_1 VF 20050322" Mar 19 16:08:45 openser rtpproxy[30827]: DBUG:doreply: sending reply "31645_1 1 " Mar 19 16:08:45 openser /usr/sbin/openser[31645]: INFO:nathelper:rtpp_test: rtp proxy <udp:127.0.0.1:7890> found, support for it enabled But when I place the call no debug info from rtpproxy is being generated nor the RTP session is being saved to file. Any idea what the problem can be? Thanks in advance Héctor

3 18

[Kamailio-Users] Kamailio as proxy and Asterisk as registrar
by Vic Jolin 26 Mar '10

26 Mar '10

Hi all! Kamailio newbie here. Well my setup is like this Kamailio as load balancer and proxy -> Multiple clustered asterisk as registrars What I wanted to do was have all clients registered to asterisk so that most of the pbx stuff will be done thru it and I'm using Asterisk Realtime as well and doing A2billing at the same time so I was thinking if my setup will actually work. I have already configured kamailio to do load balancing and send to my asterisk servers and my xlite is able to register to my asterisk server, the problem I have right now is when I try to call, the call gets dropped. I believe it's something about asterisk not being able to send the correct INVITE or not being able to receive the ACK or something to that effect. Please help me, as I've said I'm just a newbie with kamailio. TIA!

1 0

[Kamailio-Users] Fwd: Kamailio as a proxy between Trunk and PBX.
by Espen Berg 26 Mar '10

26 Mar '10

We are planning a new voip system. The trunk is provided to us on a private 10.x.x.x/29 range, and therefore not available direct to our end customers. We plan to have some kind of SIP/RTP proxy in front of the trunk, and a PBX in front of this. The trunk doesn't have any authentication, the end customers should connect and authenticate to the PBX in front, and the RTP streams should terminate to the proxy. Is this possible to achieve by using Kamailio and RTPProxy? Trunk | | 10.x.x.x/29 | SIP Proxy | PBX Asterisk etc. | Customer I have tried many different configurations and browsed every bundled example and web, but still I am not able to do what I want. This have already given me some grey hears, so I guess it's time to check if this is even possible. If not, is there any other approach that is more recommendable? If someone has done this before I would appreciate some example configuration that I could work with. Regards Espen, Norway

2 2

[Serusers] change in mailing list management
by Pavel Kasparek 26 Mar '10

26 Mar '10

Hello all, since now it is possible to use only web interface to manage mailing lists hosted on lists.iptel.org server. We had to disable the email commands interface as part of fight against spam (it was causing too many messages being sent from us, recognized by some blacklists as spam). This means that for all actions like subscribing, un-subscribing, changing your preferences etc. you have to use the web interface. You can find a link to it at end of every message sent to the mailing list. Sorry if it causes any inconvenience to you. Pavel Kasparek

1 0

[Kamailio-Users] Kamailio 3.0.1
by dotnetdub 26 Mar '10

26 Mar '10

Hi List, I have Kamailio 3.0.1 running from GIT with my old 1.5.3 config file. Everything seems fine except the RPID. I'm not getting any RPID sent in the SIP message and if I put debug=3 I see: DEBUG: auth [rpid.c:177]: no rpid AVP My config: # auth modparam("auth", "rpid_avp", "$avp(s:rpid)") modparam("auth", "rpid_prefix", "<sip:") modparam("auth", "rpid_suffix","@asterisk.ie ;party=calling;id-type=subscriber;privacy=off;screen=yes>") # auth_db modparam("auth_db", "db_url", "mysql://openser:XSXXXX@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "$avp(s:rpid)=rpid") in my route: append_rpid_hf(); Probably missing something very simple. Any help would be appreciated. Thanks, Stephen.

2 13

route() call with pv argument
by jh＠tutpro.com 25 Mar '10

25 Mar '10

i just made a test and noticed that route() accepts a pv argument. at least i didn't get anhy syntax error from route("$var(test)"); this is good if it works. Core functions section of http://sip-router.org/wiki/cookbooks/core-cookbook/devel#route does not list route() function call at all. -- juha

2 6

[Kamailio-Users] fr_inv_timer not honored (fr_inv_timer/2 is used in fact)
by Iñaki Baz Castillo 25 Mar '10

25 Mar '10

Hi, I set fr_inv_timer=150 and use lcr module to balance (and failover) between two gateways. In the failure route I set (simplified): failure_route[FAILURE_ROUTE_OUT] { if (t_local_replied("last")) { next_gw(); t_relay(); } } This means that if a INVITE routed to gateway-1 replies a provisional response but no final response within 150 seconds, then Kamailio should generate a local 408 and failure route would relaty to gateway-2. But what I see is that such "timeout" time is 75 seconds (the half of 150 !). I remember a similar "issue" when handling with SRV DNS domains returning more than one destinations, but in my case I just use single IP's in lcr module. But the fact is that I use exactly 2 gateways in lcr (150/2 = 75). But anyhow it doesn't make sense. Do I miss something? Thanks. PD: Important, could occur the same with fr_timer? -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 11

route is not called and no error is produced
by jh＠tutpro.com 25 Mar '10

25 Mar '10

i tested if route() with pv arg would work. i created a route block: route [test] { xlog("here we are\n"); return; } and then as the very first thing in the main route block i have: $var(test) = "test"; route("$var(test)"); then i send the proxy a sip request and don't get anything related to this to syslog, i.e., no "here we are" line nor any error message. this looks like i bug to me, since if route() accepts a pv, i should see "here we are" syslog entry or if it doesn't, i should see some kind of error message. comments? -- juha

1 0

[Kamailio-Users] Establish Call but no audio transferred on different network
by RAJNIKANT VANZA 25 Mar '10

25 Mar '10

Hello all, I am using kamailio-1.5.0 + mediaproxy. it works properly when connection is in between two clients between same lan. But when I connecting between two client who are on differet proxy and behind different NAT, call has been established but no audio get transferred. What are the changed i suppose to do, or any parameters should get enabled while compilation, Thanks in advance. -- Best Regards, Rajnikant Vanza

2 1

[Kamailio-Users] Radius
by dotnetdub 25 Mar '10

25 Mar '10

Hi, I'm using Kamailio - posted this to the opensips list as I thought I may get support from the ag guys but nothing.. Has anyone else experienced this or have an idea as to what it may be? Having a very small but annoying issue with Radius Accounting. Everything is getting written fine except for 'Source-Port' This appears in radius.log everytime a record is written/updated. - Error: rlm_radutmp: NAS OpenSER port 5060 unknown packet type 15 It's writing this into the text files and value to the radacct table: Source-Port = "\304\023" All other information is being written correctly. CDRTool version: 7.0 Freeradius version: 1.1.7-4 mediaproxy version: 2.3.10 Kamailio Version: 1.5.3 Using the dictionary files that ship with CDRTool >From accounting: modparam("acc", "db_url", "mysql://openser:XXXXXXXXXXXX@localhost/openser") modparam("acc", "db_flag", 2) modparam("acc", "log_flag", 2) modparam("acc", "db_missed_flag", 3) modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "radius_flag", 2) modparam("acc", "radius_missed_flag", 3) modparam("acc", "radius_extra", "User-Name=$Au; \ Calling-Station-Id=$from; \ Called-Station-Id=$to; \ Sip-Translated-Request-URI=$avp(s:translated_uri); \ Sip-RPid=$avp(s:rpid); \ Source-IP=$si; \ Source-Port=$sp; \ Canonical-URI=$avp(s:can_uri); \ Billing-Party=$avp(s:billing_party); \ Divert-Reason=$avp(s:divert_reason); \ X-RTP-Stat=$hdr(X-RTP-Stat); \ Contact=$hdr(contact); \ Event=$hdr(event); \ SIP-Proxy-IP=$avp(s:sip_proxy_ip); \ ENUM-TLD=$avp(s:enum_tld)") Obviously it's not a show stopper but it's just bugging the hell out of me :) If any more info is required please let me know and I will provide ASAP. TIA, Stephen

2 2

[Kamailio-Users] Kamdbctl not installing and Install error
by Nathaniel L Keeling 24 Mar '10

24 Mar '10

I have installed Kamailio 3.0 but without db support. When I tried to add support for postgres, the install errors and the kamdbctl does not install in order to create the database tables. I am installing on Solaris 10 and have included the error from the install: Makefile.defs defs skipped gmake[1]: Entering directory `/usr/local/src/kamailio-3.0.0/modules_k/xlog' touch /usr/local/kamailio-3.0.0/lib/kamailio/modules_k/xlog.so ginstall -m 755 xlog.so /usr/local/kamailio-3.0.0/lib/kamailio/modules_k gmake[1]: Leaving directory `/usr/local/src/kamailio-3.0.0/modules_k/xlog' # other configs /bin/sh: syntax error at line 1: `;' unexpected gmake: *** [install-cfg] Error 2 Thanks Nathaniel

4 7

why is while loop not terminated?
by jh＠tutpro.com 24 Mar '10

24 Mar '10

i'm in the process of enhancing enum mudule enum_query, so that suffic param can be given in a pseudo variable. before committing to git, i tested if the new version is working correctly, and for that purpose wrote this kind of route: route [PERFORM_ENUM_QUERIES] { xlog("L_INFO", "suffix_list = <$var(suffix_list)>\n"); $var(i) = 0; $var(domain) = $(var(domain_list){s.select,0,,}); while ($var(domain) != "") { xlog("L_INFO", "Query $var(i) for $var(domain)\n"); if (enum_query("$var(domain)")) { xlog("L_INFO", "Return success!\n"); return(1); }; $var(i) = $var(i) + 1; $var(domain) = $(var(domain_list){s.select,$var(i),,}); }; return(-1); } looks like the enhanced enum_query works as expected, but the while loop is not terminating by the return(1) statement (see below). i'm puzzled. have i done some stupid mistake that i cannot see? -- juha Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: suffix_list = <e164.test.fi.> Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: enum [enum.c:724]: Got suffix <e164.test.fi.> Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Return success! Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: INFO: Query 0 for e164.test.fi. Mar 24 18:29:56 localhost /usr/sbin/sip-proxy[24589]: ERROR: <core> [action.c:1144]: ERROR: runaway while (1342, 25): more then 100 loops

1 1

what is $name?
by jh＠tutpro.com 24 Mar '10

24 Mar '10

i made a typo in script and wrote $domain instead of $var(domain). then i started sip router and didn't get any error messages. what does $name stand for when $name is not any of the ones listed in SIP Router Pseudo-Variables document? -- juha

2 2

Re: [Kamailio-Users] Establish Call on different network.
by john david 24 Mar '10

24 Mar '10

Dear all, started rtpproxy, enabled nat in cfg file. but still there is only connection between two client from different network, unable to get audio if used kamailio server. something more that i should add, except rtpproxy and enabling nathelper from cfg file. or else is there any way to debug where it get stuck. Thanks in advance.

2 1

[Kamailio-Users] need help on xilte
by false 24 Mar '10

24 Mar '10

Hi all i use kamailio sip proxy, its version is: [root@localhost kamailio]# /opt/sbin/kamctl fifo version Server:: kamailio (3.0.1 (i386/linux)) i use 2 xlite softphones to register to sip proxy and register successfull and use xlite make call to another xlite softphone and get 488 Not Accept Here all sip proxy, xlite softphones in the same LAN please help to fix in the sip proxy that doesnt check the SDP Thank you Ha` and the debug log: DEBUG: tm [t_reply.c:1059]: ->>>>>>>>> T_code=100, new_code=488 DEBUG: tm [t_lookup.c:1080]: DEBUG: t_check_msg: msg id=2 global id=2 T start=0xb60ae9f0 DEBUG: tm [t_lookup.c:1152]: DEBUG: t_check_msg: T already found! DEBUG: tm [t_reply.c:1549]: DEBUG: relay_reply: branch=0, save=0, relay=0 DEBUG: <core> [msg_translator.c:1876]: old size: 500, new size: 440 DEBUG: <core> [msg_translator.c:1894]: build_res_from_sip_res: copied siz ontent-Length: 0ull "SDP: Incompatible media format: no common codec."19-1--d87543-;rport=51020 "Tốt hơn, thoáng gọn hơn, nhanh hơn -Trải nghiệm Yahoo! Mail mới hôm nay! http://vn.mail.yahoo.com"

2 1

[Kamailio-Users] need help on ACC modules
by false 24 Mar '10

24 Mar '10

Hi all i use the kamailio version 3.0.1 and try to understand ACC module and found the tutorial http://www.voice-system.ro/docs/uac/ does it still work with kamailio 3.0.1, could you please present a working config with kamailio 3.0.1 How to work with the uac-create.sql, the sip account from VOIP provider username: x.x.x.x password: x.x.x.x outbound proxy: x.x.x.x domain name: sbc.net expired time : 3600s what info should i insert into database l_uuid: l_username: l_domain: r_username: r_domain: realm: auth_username: auth_password: auth_proxy: expires: when i start the sip server with acc module and got error and the config parameter: modparam("uac", "reg_db_url","mysql://openser:openserrw@localhost/openser") DEBUG: <core> [sr_module.c:617]: find_param_export: parameter <reg_db_url> not found in module <uac> ERROR: <core> [modparam.c:143]: set_mod_param_regex: parameter <reg_db_url> not found in module <uac> ERROR: <core> [modparam.c:143]: set_mod_param_regex: parameter <reg_db_url> not found in module <uac> Thank you Phần mềm chống virus và adware miễn phí đã có cùng Yahoo! Thanh Công Cụ. Tải về ngay! Tải ngay. http://vn.toolbar.yahoo.com/

2 1

[Kamailio-Users] Kamailio+asterisk
by Marcus Kvarsell 24 Mar '10

24 Mar '10

Hello everyone! If i want to have Kamailio as front-end and 3 asterisk as back-end. Sip-registration should be done by Kamailio, and usage of meet-me rooms by asterisk-servers which should be load-balanced by Kamailio. is it possible of creating outgoing call with Kamailio and when call is answered, realtime route this call dynamically based on some information coming from unixODBC source, to a meet-me room? any help appreciated!! Regards / Marcus

3 2

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