- sr-users - kamailio.org

PUA_XMPP: Notify in a non existing dialog
by Dan-Cristian Bogos 10 Mar '12

10 Mar '12

Guys, After my previous discovered issue (Kamailio not happy to talk with Openfire for subscriptions), trying to find work arounds, thought about using pua_mi to enforce subscriptions coming from XMPP side. The following events happen: * I subscribe over mi interface successfully. * Presence sends NOTIFY out. * PUA_XMPP rejects the NOTIFY saying that it does not match an existing dialog. I was wondering where does PUA_XMPP check the existing dialogs and if there is other way to fix my issues? Bellow you can find commands, ngrep as well as kamailio error to syslog. Ta for any kind of tip, DanB """ ### Console ### kamctl fifo pua_subscribe sip:dan@hiddendom.com sip:user01@hiddendom.com presence 3600 ### Trace ### U 2012/03/02 18:17:07.318753 127.0.0.1:5060 -> 127.0.0.1:5060 SUBSCRIBE sip:dan@hiddendom.com SIP/2.0. Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK5ddb.d2ee4046.0. To: sip:dan@hiddendom.com. From: sip:user01@hiddendom.com;tag=7bcd2f0283a17a6a3eb34b70a7dbcaec-55da. CSeq: 10 SUBSCRIBE. Call-ID: 77d7b19267fdcae8. Content-Length: 0. User-Agent: kamailio (3.2.0 (x86_64/linux)). Max-Forwards: 70. Event: presence. Contact: <sip:user01@hiddendom.com>. Expires: 3610. . # T 2012/03/02 18:17:07.319501 127.0.0.1:35067 -> 127.0.0.1:3306 [AP] .....select status,reason from watchers where presentity_uri='sip:dan@hiddendom.com' AND watcher_username='user01' AND watcher_domain='hiddendom.com' AND event='presence' # T 2012/03/02 18:17:07.319795 127.0.0.1:3306 -> 127.0.0.1:35067 [AP] .....7....def.prxdb.watchers.watchers.status.status.?...........7....def.prxdb.watchers.watchers.reason.reason...@........................... ## T 2012/03/02 18:17:07.319885 127.0.0.1:35067 -> 127.0.0.1:3306 [AP] .....insert into watchers (presentity_uri,watcher_username,watcher_domain,event,status,inserted_time ) values ('sip:dan@hiddendom.com','user01','hiddendom.com','presence',1,1330712227) # T 2012/03/02 18:17:07.320079 127.0.0.1:3306 -> 127.0.0.1:35067 [AP] ........... # U 2012/03/02 18:17:07.320250 127.0.0.1:5060 -> 127.0.0.1:5060 SIP/2.0 202 OK. Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK5ddb.d2ee4046.0;rport=5060. To: sip:dan@hiddendom.com;tag=c87c23e595340de8a004872f7dfab287-19b1. From: sip:user01@hiddendom.com;tag=7bcd2f0283a17a6a3eb34b70a7dbcaec-55da. CSeq: 10 SUBSCRIBE. Call-ID: 77d7b19267fdcae8. Expires: 3600. Contact: <sip:127.0.0.1>. Server: kamailio (3.2.0 (x86_64/linux)). Content-Length: 0. . # T 2012/03/02 18:17:07.320317 127.0.0.1:35067 -> 127.0.0.1:3306 [AP] .....select body,expires,etag,sender from presentity where domain='hiddendom.com' AND username='dan' AND event='presence' order by received_time # T 2012/03/02 18:17:07.320541 127.0.0.1:3306 -> 127.0.0.1:35067 [AP] .....7....def.prxdb presentity presentity.body.body.?...........=....def.prxdb presentity presentity.expires.expires.?...........7....def.prxdb presentity presentity.etag.etag...@.....P...;....def.prxdb presentity presentity.sender.sender...........................Y.<?xml version="1.0"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns:c="urn:ietf:params:xml:ns:pidf:cipid" entity="dan(a)hiddendom.com"> <tuple id="0x7f7e550f0d70"> <status> <basic>open</basic> </status> </tuple> </presence> 1330712820.a.1330712207.16317.1.1.......... # U 2012/03/02 18:17:07.320996 127.0.0.1:5060 -> 127.0.0.1:5060 NOTIFY sip:user01@hiddendom.com SIP/2.0. Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKae68.a038faa7.0. To: sip:user01@hiddendom.com;tag=7bcd2f0283a17a6a3eb34b70a7dbcaec-55da. From: sip:dan@hiddendom.com;tag=c87c23e595340de8a004872f7dfab287-19b1. CSeq: 2 NOTIFY. Call-ID: 77d7b19267fdcae8. Content-Length: 345. User-Agent: kamailio (3.2.0 (x86_64/linux)). Max-Forwards: 70. Event: presence. Contact: <sip:127.0.0.1>. Subscription-State: active;expires=3670. Content-Type: application/pidf+xml. . <?xml version="1.0"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns:c="urn:ietf:params:xml:ns:pidf:cipid" entity="dan(a)hiddendom.com"> <tuple id="0x7f7e550f0d70"> <status> <basic>open</basic> </status> </tuple> </presence> ############# kamailio.log: ############# Mar 2 18:17:07 prx3 /usr/sbin/kamailio[16319]: INFO: presence [notify.c:1593]: NOTIFY sip:user01@hiddendom.com via sip:user01@hiddendom.com on behalf of sip:dan@hiddendom.com for event presence Mar 2 18:17:07 prx3 /usr/sbin/kamailio[16317]: ERROR: pua_xmpp [simple2xmpp.c:198]: Notify in a non existing dialog """

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Kamailio and MySQL/Galera Replication...
by Krishna Kurapati 10 Mar '12

10 Mar '12

Hi, I am trying to setup Kamailio in an Active-Active configuration behind a TCP load balancer. On the database side, looking at MySQL cluster as independent node set vs. Kamailio/MySQL server with Galera synchronous replication. The first configuration requires 3 or 5 servers. 2 for Kamailio and 3 for MySQL. The second probably just requires 2 servers. I am leaning towards Kamailio/MySQL/Galera. Could you share setup/performance issue... Thanks Krish Kura

1 0

insert_hf in CANCEL message
by Reda Aouad 10 Mar '12

10 Mar '12

Hi, insert_hf doesn't work for CANCEL messages. It doesn't work either on 487 replies in failure_route.. Why? Is it because they are re-generated on every hop? Is there a workaround? Reda

2 1

SIP problems using a comrex Access
by Andy Bosley 10 Mar '12

10 Mar '12

Hi Support, We have a comrex access unit that we have previously successfully used for connections using your SIP service. For some reason now though, we can no longer get the unit to work. We know the username (LCR.ACCESS1) is ok as we can access the account through the website. However, entering the same details in to the comrex fails. The comrex status displays "OFFLINE (Registering)" and does not get any further. Is it possible for you to check that the account is ok and that the unit has successfully been authenticated. regards Andy __________________________ Andy Bosley Supervisor, London Control Room Technology Operations Technology Division, BBC Room 3045, Broadcasting House Tel External: +44 02077 654976 Tel Internal: 54976 http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this.

2 1

XHTTP or XMLRPC modules
by Uri Shacked 10 Mar '12

10 Mar '12

hi, i need to start some kind of proccess in my kamailio server (manualy, not by timer). the best way i can think of is to send a HTTP request. i can do it with the XMLRPC module, but i get an error because i dont realy send XML. i try the XHTTP module. but the event_rout[xhttp_request] does not run. i get : 19(2796) ERROR: <core> [tcp_read.c:965]: ERROR: tcp_read_req: bad request, state=7, error=4 buf: POST / HTTP/1.0 User-Agent: Fiddler Host: 10.2.0.55:5060 parsed: POST / HTTP/1.0 User-Agent: Fiddler Host: 10.2.0.55:5060 any ideas?

2 1

Backing up htable to SQL table ...
by Krishna Kurapati 09 Mar '12

09 Mar '12

Hi, This is my first time attempt using htable module. How can I back up the htable content in the database incase the server restarts. I saw a htable in the MySQL database. My observation has been that htable module is not automatically backing up to database even though I added modparams to point to the database. Do I need to explicitly backup htable content to database using sqlops or some other mechanism? Thanks Krish Kura

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Advice on placement of t_replicate (REGISTER)
by Asgaroth 09 Mar '12

09 Mar '12

Hi Guru's, I'd just like some advice on t_replace and its placement in the routing script. A lot of examples show the following setup: save("location"); t_replicate("backup_server"); Would it not be wiser to have something like: if (save("location")) { t_replcate("backup_server"); } Would there be any scenario where you would want to t_replicate if save("location") fails for some reason? The configuration I refer to is if the location database is in memory. I suspect you may want to replicate all if location is going to db core in case of external failure. But would it matter on the placement if the location was in memory? Thanks

2 1

kamailio with flowroute
by romon.zaman 09 Mar '12

09 Mar '12

hello room, i was trying to add sip provider(like. flowroute,vitelity) with user-pass authentication in kamailio to accept inbound calls. any help? thanks

3 2

no hangup when there is no network or when the UAC on the other side is power off
by Rabary 08 Mar '12

08 Mar '12

Hi everybody, We are using kamailio+callcontrol+cdrtool+freeradius+mysql to build a voip prepaid based system: . 2 SIP accounts created and one of them is prepaid . we can debit balance when we call with the SIP prepaid account, and when the balance is 0, the call hang up . if we hangup a call session from the prepaid SIP and the balance is not still 0, the call hangup and the system debit the appropriate cost . but when we call from the prepaid SIP account, and we disconnect the ethernet cable or the alimentation of the other UAC, the call don't hangup until the balance is set to 0 . if the phone is still ringing too, when we disconnect the other UAC, the caller UAC still ringing . the 2 uacs and kamailio are in the same subnet network the question is, how can we tell to kamailio to detect if the other side is down and then send bye immediately so thecalculating of the credit will be correct? Thank you. --

1 0

Rewrite $tU
by Lucas Alvarez 08 Mar '12

08 Mar '12

I want rewrite $tU but I'm not being able, I'm doing the following: remove_hf("To"); insert_hf("To: sip:$rU@$rd\r\n", "From");

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Limiting registrations to known users without auth
by Pedro Antonio Vico Solano 08 Mar '12

08 Mar '12

3 4

default IP to forward TCP request
by Laurent Schweizer 07 Mar '12

07 Mar '12

Hello all, I have a linux server with different IP. eth0 -> 10.2.2.4 eth0:1 -> 10.2.2.5 I have kamailio listening on IP: listen=udp:10.2.2.5:5060 listen=tcp:10.2.2.5:5060 and in the script I do : rewritehostport("10.2.2.10:5060;transport=tcp"); if (!t_relay()) { sl_reply_error(); } exit; the problem is that t_relay is not using the IP 10.2.2.5 to forward this SIP request but the eth0 IP ( 10.2.2.4) I see that they is a function force_send_socket , I need to use this function to set the correct IP used to forward the request ? As I'm using only one IP 10.2.2.5 for openser, why he cannot always use this ip as default Ip ? Best regards Laurent

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Re: [SR-Users] sr-users Digest, Vol 82, Issue 21
by Javier Gallart 07 Mar '12

07 Mar '12

Hello this is what I normally use: if (defined $var(x)) {...} Regards Javi > ------------------------------ > > Message: 4 > Date: Tue, 6 Mar 2012 20:05:32 -0600 > From: Krishna Kurapati <kkurapat(a)gmail.com> > Subject: [SR-Users] Improving Kamailo configuration parser ... > To: kamailio users <sr-users(a)lists.sip-router.org> > Message-ID: > <CAOzCjAfcaRQ+HT3r+mwckPyaVJH+EVFWNa_Hj6tAhGyD2DaMJA(a)mail.gmail.com > > > Content-Type: text/plain; charset="iso-8859-1" > > I used > > if ($var(x) != null) { ... } > > in one of the routes, and tried to run Kamailio server. > The server exited with an error. I did not know why. > > Once I changed it to if ($var(x) != $null) {} the error went away. > I missed $ infront null. I think it is hard to trace such problems once a > large chunk > of logic is written since many of us come from C++ and scripting background > where $null > is not a common practice. > > Also if I tried to use if (!$var(x)) { ... } Kamailio tries to convert the > value of $var(x) to integer. > prints WARNINGS in the log when the $var(x) consists of a string. > > Krish Kura >

1 0

Improving Kamailo configuration parser ...
by Krishna Kurapati 07 Mar '12

07 Mar '12

I used if ($var(x) != null) { ... } in one of the routes, and tried to run Kamailio server. The server exited with an error. I did not know why. Once I changed it to if ($var(x) != $null) {} the error went away. I missed $ infront null. I think it is hard to trace such problems once a large chunk of logic is written since many of us come from C++ and scripting background where $null is not a common practice. Also if I tried to use if (!$var(x)) { ... } Kamailio tries to convert the value of $var(x) to integer. prints WARNINGS in the log when the $var(x) consists of a string. Krish Kura

2 1

msilo is failing to in db connection...
by Krishna Kurapati 06 Mar '12

06 Mar '12

Hi all, I am trying to use msilo module from modules_k. Unfortunately it if failing during DB initialization. Other modules are initializing fine. I am using Kamailio 3.2.2. Not sure why the connection is not found. I tried to increase the MAX mysql connections to 250. I see the same error after I increase the connections. The config file is pretty standard: loadmodule "msilo.so" modparam("msilo", "add_date", 0); modparam("msilo", "db_url", DBURL); # The DBURL works for other modules... Here is the Log output: 0(14115) DEBUG: <core> [cfg.y:1684]: loading module msilo.so 0(14115) DEBUG: <core> [sr_module.c:557]: load_module: trying to load </usr/local/kamailio-3.2.2/lib64/kamailio/modules_k/msilo.so> ..... 0(14117) DEBUG: <core> [sr_module.c:928]: DEBUG: init_mod: msilo 0(14117) DEBUG: <core> [sr_module.c:667]: find_export_record: found <db_bind_api> in module db_mysql [/usr/local/kamailio-3.2.2/lib64/kamailio/modules/db_mysql.so] 0(14117) DEBUG: <core> [db.c:212]: using db bind api for db_mysql 0(14117) DEBUG: <core> [db.c:299]: connection 0x7fc20d1ace60 not found in pool 0(14117) DEBUG: db_mysql [km_my_con.c:94]: opening connection: mysql://xxxx:xxxx@localhost/openser 0(14117) ERROR: db_mysql [km_my_con.c:109]: driver error: Access denied for user 'openser'@'localhost' (using password: YES) 0(14117) ERROR: <core> [db.c:303]: could not add connection to the pool 0(14117) ERROR: msilo [msilo.c:319]: failed to connect to the database 0(14117) ERROR: <core> [sr_module.c:932]: init_mod(): Error while initializing module msilo (/usr/local/kamailio-3.2.2/lib64/kamailio/modules_k/msilo.so) ERROR: error while initializing modules Any help? Thanks

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Re: [SR-Users] msilo is failing to in db connection...
by Daniel-Constantine Mierla 06 Mar '12

06 Mar '12

fwd to the list - reply all failed in first place ... > Hello, > > the module connects in mod init to check the table version, otherwise it does not do any message storage related operations with the database. It should work fine at runtime even if mod init does not connect to db. > > But it is strange of mod init fails but child init goes on. Do you have a firewall with policies related to active connections (rate of new connections)? > > Cheers, > Daniel > > On Sat, Mar 3, 2012 at 6:16 PM, Krishna Kurapati <kkurapat(a)gmail.com> wrote: > Further investigation into the issue revealed that the db_conn in the module_init is failing... When I commented out the code, > the module is initialized ok. Not sure if this is will fine. > > #if 0 > db_con = msilo_dbf.init(&ms_db_url); > if (!db_con) > { > LM_ERR("failed to connect to the database\n"); > return -1; > } > > if(db_check_table_version(&msilo_dbf, db_con, &ms_db_table, S_TABLE_VERSION) < 0) { > LM_ERR("error during table version check.\n"); > return -1; > } > if(db_con) > msilo_dbf.close(db_con); > db_con = NULL; > #endif > > Is the module not suppose to connect to database in the module init? > > Thanks > > > On Sat, Mar 3, 2012 at 10:13 AM, Krishna Kurapati <kkurapat(a)gmail.com> wrote: > Hi, > > Thanks for the response. I looked at the DBURL and it looked just fine. I wanted to test and see if the module is reading the config > param or not, I deliberately changed it to > > modparam("msilo", "db_url", "mysql://root:xxxxx@localhost/openser"); > > When I ran it, the logs shows the user as 'openser' not the 'root' that I gave in config param. May be somewhere in the code, > db_url is being overwritten by DEFAULT_DB_URL. > > 0(28806) DEBUG: db_mysql [km_my_con.c:94]: opening connection: mysql://xxxx:xxxx@localhost/openser > 0(28806) ERROR: db_mysql [km_my_con.c:109]: driver error: Access denied for user 'openser'@'localhost' (using password: YES) > 0(28806) ERROR: <core> [db.c:312]: could not add connection to the pool 0(28806) ERROR: msilo [msilo.c:365]: failed to connect to the database > > Thanks > Krish Kura > > On Sat, Mar 3, 2012 at 5:14 AM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > Hello, > > the error says the username, password or IP address are invalid to connect to mysql -- access denied. There does not seem like being other connections using same db_url, check for typos in db_url parameter. > > Cheers, > Daniel > > > > On 3/3/12 12:39 AM, Krishna Kurapati wrote: >> Hi all, >> >> I am trying to use msilo module from modules_k. Unfortunately it if failing during DB initialization. >> Other modules are initializing fine. I am using Kamailio 3.2.2. >> >> Not sure why the connection is not found. I tried to increase the MAX mysql connections to 250. I see the >> same error after I increase the connections. >> >> The config file is pretty standard: >> >> loadmodule "msilo.so" >> modparam("msilo", "add_date", 0); >> modparam("msilo", "db_url", DBURL); # The DBURL works for other modules... >> >> Here is the Log output: >> >> 0(14115) DEBUG: <core> [cfg.y:1684]: loading module msilo.so >> 0(14115) DEBUG: <core> [sr_module.c:557]: load_module: trying to load </usr/local/kamailio-3.2.2/lib64/kamailio/modules_k/msilo.so> >> >> ..... >> >> 0(14117) DEBUG: <core> [sr_module.c:928]: DEBUG: init_mod: msilo >> 0(14117) DEBUG: <core> [sr_module.c:667]: find_export_record: found <db_bind_api> in module db_mysql [/usr/local/kamailio-3.2.2/lib64/kamailio/modules/db_mysql.so] >> 0(14117) DEBUG: <core> [db.c:212]: using db bind api for db_mysql >> 0(14117) DEBUG: <core> [db.c:299]: connection 0x7fc20d1ace60 not found in pool >> 0(14117) DEBUG: db_mysql [km_my_con.c:94]: opening connection: mysql://xxxx:xxxx@localhost/openser >> 0(14117) ERROR: db_mysql [km_my_con.c:109]: driver error: Access denied for user 'openser'@'localhost' (using password: YES) >> 0(14117) ERROR: <core> [db.c:303]: could not add connection to the pool 0(14117) ERROR: msilo [msilo.c:319]: failed to connect to the database >> 0(14117) ERROR: <core> [sr_module.c:932]: init_mod(): Error while initializing module msilo (/usr/local/kamailio-3.2.2/lib64/kamailio/modules_k/msilo.so) >> ERROR: error while initializing modules >> >> Any help? >> >> Thanks >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla -- http://www.asipto.com > http://linkedin.com/in/miconda -- http://twitter.com/miconda > > > > > > -- > Daniel-Constantin Mierla > http://www.asipto.com

1 0

disptacher failover
by Klaus Darilion 06 Mar '12

06 Mar '12

Hi! Imagine a dispatcher list with 3 entries. I wonder what happens if one of the targets fail. Is the traffic moved evenly to the remaining 2 nodes or is the whole traffic of the broken node moved to one of the nodes? I took a quick look at the code and it seems that the whole traffic is moved to the next node in the list (i=i+1): while (ds_skip_dst(idx->dlist[i].flags)) { if(ds_use_default!=0 && idx->nr!=1) i = (i+1)%(idx->nr-1); else i = (i+1)%idx->nr; Is my assumption correct? Thanks Klaus

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Question regarding... LCR Module
by kamailio Padawan 06 Mar '12

06 Mar '12

Hello, I have attempted to ask questions on IRC with no response. I really appreciate any help from anybody! I am using Kamailio 3.2.2. I am unable to get anything loaded in $avp(i:709). I am attempting to route between 2 media handling asterisk boxes. I am trying to load balance between them. I have the following entries in my tables... *lcr_gw* * id,lcr_id,gw_name,ip_addr,hostname,port,params,uri_scheme,transport,strip,prefix,tag,flags,defunct * 1, 1, media1, *1.1.3.123*, *1.1.3.123*, 5060, , 1, 1, 0, 0, 0 2, 1, media2, *1.1.4.123*, *1.1.4.123*, 5060, , 1, 1, 0, 0, 0 *lcr_rule* *id,lcr_id,prefix,from_uri,stopper,enabled* * * 1, 1, *[1][0-9][0-9][0-9][0-9]*, 0, 1 *lcr_rule_target* *id,lcr_id,rule_id,gw_id,priority,weight* 1, 1, 1, *1*, 1, 5 2, 1, 1, *2*, 1, 5 I was under the impression that, if my from URI was 15555, and I have 2 gateways, that I can call *load_gws("1","$fU")* and return the gateways that match the *from_uri* rule of 15555 , then look the rule_id up in * lcr_rule_target* and return the *2* gws gw_id[1] and gw_id[2], which are the 2 entries in the* lcr_gw* table..... and it would be weighted 5 and 5 so that it would load balance. Right now..... I get nothing loaded in $avp(i:709). I thought this would show the IP/Hostname of the media1 or media2 gws. The variable is null far before the call to *load_gws*. I'm loading it at the top in a modparam like *modparam("lcr", "gw_uri_avp", "$avp(i:709)")* *modparam("lcr", "ruri_user_avp", "$avp(i:500)")* When i get the result of calling the *load_gws*() function into a variable. The variable is *2*. Is that the number of rows returned? Why is that parameter [avp(i:709)] always empty? How can i get next_gw() to complete and not fail? Thank you so much.... I am very new to Kamailio and am very interested... Kamailio Padawan Learner

2 1

TOS value in configuration
by Olle E. Johansson 04 Mar '12

04 Mar '12

I've set tos=IPTOS_RELIABILITY in my configuration, which follows https://www.kamailio.org/wiki/cookbooks/3.2.x/core#tos which has the following examples: tos=IPTOS_LOWDELAY tos=0x10 tos=IPTOS_RELIABILITY but I get a configuration error: 0(45338) : <core> [cfg.y:3501]: parse error in config file /usr/local/etc/kamailio/kam-global.inc, line 29, column 5-21: syntax error 0(45338) : <core> [cfg.y:3501]: parse error in config file /usr/local/etc/kamailio/kam-global.inc, line 29, column 5-21: number expected Has this option been deprecated or is it a small bug? /O

2 1

Facebook page for Kamailio
by Olle E. Johansson 04 Mar '12

04 Mar '12

Just to make sure we're in the flow, I've started a Facebook page for Kamailio. http://www.facebook.com/kamailio Feel free to like it if you're a Facebook user. There's also a Google+ page. If you have news to share, you're welcome to add it on these pages. /O

2 1

Building Up a Kamailio Server
by Faisal Rehman 03 Mar '12

03 Mar '12

Hi Everyone, I am totally new to Kamailio & going to build up Kamailio server as an XMPP server with the following wanted functionalities: -Have XMPP client to register to my kamailio server - The XMPP clients should all be able to see each others presence - The XMPP clients should be able to IM other XMPP clients - The registered sip client will be able to just dial any other sip client. No need for any gateway or trunk for now I have a good previous background with SIP but its my first time to work with XMPP & Kamailio & that's why I am in need of a bit help & assistance. I have installed Kamailio on my CentOS box & now going to start to configure it in order to complete the above requirements, so any links or any refferals will be highly appreciated. Regards, Faisal Rehman

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XMPP module not passing subscribes coming from xmpp side
by Dan-Cristian Bogos 02 Mar '12

02 Mar '12

Hey Guys, I was wondering if anyone having experience with using xmpp module as presence gateway between SIP and XMPP. I started evaluating it and till now got some weird results, not sure if it is just my setup which does not match the one widely used or the module is incomplete. Regarding my setup, I use OpenFIRE 3.7.1 on one side and Kamailio 3.2.0 out of debian packages on the other. The issue I got right now is that the subscribe coming from XMPP side produces no action/reply on SIP side. Bellow you can see the packet coming from OpenFIRE: # T 2012/03/02 09:49:57.743750 127.0.0.1:5275 -> 127.0.0.1:49965 [AP] <presence id="4h8F6-19" to="dan4(a)gw.mydomain.com" type="subscribe" from="dan(a)mydomain.com"><c xmlns="http://jabber.org/protocol/caps" ext="voice-v1 video-v1 camera-v1 " hash="sha-1" node="http://jitsi.org" ver="ohjz8WKq0ZQRiNWRNcapX4BBHLk="/></presence> """ Anyone here got this working somehow? Ta, DanB

2 1

iOS push notification using Kamailio
by Krishna Kurapati 02 Mar '12

02 Mar '12

Hi, I have been looking at the usrloc module's source code to see how the uuid that is sent in +sip.instance parameter in the Contact header. Apparently, the module ensures that only one contact per uuid exists. However the uuid is not stored in the database. The reason I need the uuid in the database is to access it when the iOS device is offline inorder to send a push notification. I am using pjsip which supports sending the uuid of the device ( w/ rfc 5626 support) in the REGISTER message. Has anyone implemented push notification using this or other methods using Kamailio? Thanks Krish Kura

2 1

automatically switch to TCP
by Klaus Darilion 01 Mar '12

01 Mar '12

Hi! Is there a feature somewhere (core, tm) to automatically switch from UDP to TCP? It may be useful with bodies > 64K. regards Klaus

2 1

Kamailio error code 500 (6/SL)
by Pavel Klochan 01 Mar '12

01 Mar '12

Hi. I'm a newbie to Kamailio (so do not judge strictly). I have experience only with Asterisk and now I'm trying to configure Kamailio. With default config all is working as expected, but I'm trying to configure it from zero with minimal functions, but using parts from default config (those parts that I figured out). With my present config local calls are working great, but I can't make outbound calls. I've attached config file, sip-trace and debug log. I think there is a route loop somewhere but I can't figure out where. I would be grateful if you pointed out to me where exactly my mistake or just point me in the right direction =) Thanks in advance

2 2

loose_route() with pre-loaded Route in ACK
by Andreas Granig 01 Mar '12

01 Mar '12

Hi, I ran into an interesting scenario when experimenting with kamailio 3.1 as a stateless proxy (no tm module used): There is an initial INVITE without to-tag and with pre-loaded Route header pointing to the external address of the lb. When I call loose_route for that, it returns false and the Route header is removed. That's what I expected, and it's documented in http://www.kamailio.org/docs/modules/3.1.x/modules_k/rr.html#loose-route-id. However if a 407 is relayed back from the registrar via the lb to the UA, it sends an ACK, again with pre-loaded Route header like in the initial INVITE, this time with to-tag. In that case, loose_route() returns true, but the $du is $null and the R-URI is still the same. So the Route is consumed, but the URI is not used. Since it's a stateless proxy, this hop-by-hop ACK needs to be routed the same way as the initial INVITE, so what I'm doing is something like that: if(loose_route()) { if($du == $null && $ru == $ou) { # skip loose-routing for such a request } } Is this how it's supposed to work? And is there some documentation of how loose_route() detects whether Route points to itself or not? I'm not using auto_aliases, nor do I set any aliases manually. Does it determine internally whether the Route points to its local ip and port? If not, how come the $du is not set although loose_route returns true? Andreas

2 9

DRouting cause Kamailio 3.2 to stop
by Fatima Chahrour~Vanrise Support 01 Mar '12

01 Mar '12

Hi, I am newly experiencing kamailio 3.2. I have installed kamailio and MySQL and tried the test call from 1 PC(x-lite) to another PC(x-lite) and ended successfully. Now moving further, I have configured dynamic routing module and applied needed parameters and scripts to finally be able to start kamailio without any issue. Also I set a testing scenario in the related database tables BUT my problem is that whenever am trying to call the predefined prefix (961) from x-lite am getting a 'calling' message on it and then request timeout where I find that kamailio has stopped running. I kamctrl start again and call 961 then get calling so I wait and kamailio stps again. Below is the script used in the routing logic and database record. Any help to achieve a successful call is highly appreciated. dr_gatways: 3, 1, '192.168.111.15', 0, '', '', 2, 'PC1' dr_groups: 2, '.*', '.*', 0, 'Anyone' dr_gw_lists: 2, '3', 'Test Call' dr_rules: 2, '0', '961', '', 0, '1', '#2', 'Lebanon Fix' ####### Routing Logic ######## # main request routing logic route{ if (is_method("INVITE")) { setflag(1); # do accounting route(4); # Call our dynamic route. } } .. .. .. route[4] { if (!do_routing("0")) { send_reply("503", "No Rules matching the URI"); exit; } if (is_method("INVITE")) { t_on_failure("4"); } route(RELAY); } .. .. .. failure_route[4] { if (use_next_gw()) { t_relay(); exit; } else { t_reply ("503", "Service not available"); exit; } } Thanks in Advance, F.Chahrour

4 8

memory allocation failure
by Robert R 29 Feb '12

29 Feb '12

Hi, The openser service died on my servers after "memory allocation failure". This issue has been occurred a couple of times on two different servers within the last two months. Feb 29 08:31:59 /usr/sbin/openser[7115]: ERROR:core:do_action: memory allocation failure Feb 29 08:31:59 /usr/sbin/openser[7115]: ERROR:core:pv_set_ruri_user: do action failed Feb 29 08:31:59 /usr/sbin/openser[7115]: ERROR:core:do_assign: setting PV failed .... Feb 29 08:31:59 /usr/sbin/openser[7115]: ERROR:core:insert_cond_lump_after: out of pkg memory Feb 29 08:31:59 /usr/sbin/openser[7115]: ERROR:rr:build_rr: failed to insert lumps Feb 29 08:31:59 /usr/sbin/openser[7115]: ERROR:rr:record_route: failed to insert inbound Record-Route Feb 29 08:31:59 kernel: [8806798.263792] openser[7115] general protection rip:7f6e17eae69c rsp:7fff21dac5b0 error:0 Feb 29 08:31:59 /usr/sbin/openser[7164]: CRITICAL:core:receive_fd: EOF on 22 It seems increasing memory size in /etc/default/openser to 1024 didn't fix the issue. # Amount of memory to allocate for the running OpenSER server (in Mb) MEMORY=1024 Any advice would be appreciated. Thanks, R

2 1

Presence between different domains
by Manuel Perez 29 Feb '12

29 Feb '12

Hello, I've used kamailio 3.2.2, i have rls, simple presence and xcap support working. I've replicated this server into another one in a different domain, but presence between domains is not working. I'm using for presence presence.so, presence_xml.so and presence_mwi.so. In route[WITHINDLG] i have the following lines: ... if (is_method("SUBSCRIBE|NOTIFY") && uri==myself) route(PRESENCE); ... When NOTIFY from user B at domain B arrives to the server at domain A, the message is not forwarded to the user A, How can i achieve this? Thank you.

2 3

dialog doesn't clear calls that were re-INVITEd shortly after ACK
by Andrew Pogrebennyk 29 Feb '12

29 Feb '12

Hi, I noticed dialog module showing calls that have already been terminated with state=4. The kamailio version is 3.1.5, I'm aware of certain limitations of dialog module in this version (must use stateless replies, create dialog after 407), but this is something rather new. In kamailio config I'm using dlg_manage() for initial INVITEs and BYE/CANCEL and in-dialog requests are loose-routed, so nothing finicky. The call scenario is: A calls B, B answers the call, A immediately after sending ACK for the 200 OK issues re-INVITE. After some seconds A hangs up, and the dialog stays in memory in state=4, e.g.: dialog:: hash=3304:623200556 state:: 4 ref_count:: 2 timestart:: 1329938667 timeout:: 78590167 callid:: MGExMjRiYzMzN2Q5YWU4YjRiMjFjYTYwYWZlMGFlMzY. from_uri:: sip:sipwise-user1@67.202.62.202 from_tag:: 4207e429 caller_contact:: sip:sipwise-user1@80.108.64.151:46420 caller_cseq:: 3 caller_route_set:: <sip:127.0.0.1:5060;lr;r2=on>,<sip:67.202.62.202:5060;lr;r2=on> caller_bind_addr:: udp:127.0.0.1:5062 callee_bind_addr:: udp:127.0.0.1:5062 to_uri:: sip:43991002@67.202.62.202 to_tag:: 6C8F89E9-4F4540EA0001F575-AC557700 callee_contact:: sip:sipwise-user2@127.0.0.1:5080 callee_cseq:: 11 callee_route_set:: When I set dlg_match_mode=1 the problem goes away. I would like however to understand why the dialog fails to clear in default mode (DID_ONLY), because the dialog cookie seems to be preserved by caller in all in-dialog requests properly. There is nothing in the debug log and trace attached that catches my eye. Do you see anything wrong there? In trace file the proxy address is 127.0.0.1:5062. You may notice that the actual signaling flow is a little bit more complicated, I will gladly answer any questions about it. For this test I've used two lines in the eyeBeam softphone, when one line answers the other is automatically put on hold which is what I want. Then I un-hold the callee and clear the call. The customer is reporting multiple stale calls when re-INVITE is sent by Cirpack PSTN gw immediately after call connect, which is basically the same call flow I am talking about, it is just not practical for me to run kamailio with debug=3 there all the time. I am going to try theredlg_match_mode=1 though and see if the problem goes away. Any ideas?

1 1

Re: [SR-Users] sr-users Digest, Vol 81, Issue 95
by Yufei Tao 29 Feb '12

29 Feb '12

Hi Thanks for your reply! I already set the accounting flag like you say. And all normal BYEs are inserted into the acc table fine. Just internally generated BYEs don't get inserted in acc table. I've done a bit googling and added event_route: event_route[tm:local-request] { if (is_method("BYE")) { xlog("L_INFO", "event_route: BYE: callid=$ci, cseq=$cs - internally generated\n"); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } } When I have two clients in a call, I disconnect the network for both clients, so no RTP either direction, which triggered mediaproxy to send dlg_bye_dlg to Kamailio, which sent BYE to both clients. I can see from the log that event_route did get executed, and BYE sent to both clients, but still BYE isn't inserted to acc table. The code inside event_route() above is copied from my route[WITHINDLG], which for client-generated BYEs operates correctly (inserts BYEs into acc table). But in event_route it didn't work. I also tried to use sql_query to insert BYE into acc table in event_route and it causes Kamailio to crash. Using sql_query to select * from acc also causes Kamailio to crash. Any idea where I've done wrong? I'm very much lost. Thank you! Yufei > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 27 Feb 2012 13:21:50 -0500 > From: Alex Balashov <abalashov(a)evaristesys.com> > Subject: Re: [SR-Users] How to do accounting for Kamailio-generated > BYEs? > To: sr-users(a)lists.sip-router.org > Message-ID: <4F4BC9BE.8000704(a)evaristesys.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > On 02/27/2012 12:58 PM, Yufei Tao wrote: >> Hi >> >> I've got a mediaproxy that detects no media, and sends dlg_end_dlg to >> Kamailio via mi_datagram socket. Then Kamailio sends BYE to both calling >> parties. It all works fine. But my question is, how to do accounting for >> these Kamailio-generated BYEs? They don't appear in the acc table. > In the loose_route() section of the config script, set the accounting > flag for BYEs, e.g. > > if(is_method("BYE")) > setflag(xx); > > -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

1 2

Re: [SR-Users] CallControl and MediaProxy
by Reda Aouad 29 Feb '12

29 Feb '12

Ok thanks Daniel. I'll do what you suggested and we'll see how to proceed. Thanks again Reda On 27 févr. 2012, at 17:46, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Hello, On 2/26/12 11:01 AM, Reda Aouad wrote: Daniel, I sent you the log file by private email, for privacy concerns. ok, no time yet to look over them, but as you say it does not call mediaproxy explicitely from config and callcontrol, maybe you should look inside mediaproxy module, might be automatically executed on dialog or tm module callbacks -- then the patch has to be done for mediaproxy module. I took a quick look at the log files, and didn't see any call to mediaproxy module. Though, the mediaproxy log file indicates a media session engaged, and CDRTool shows media information for the established session. The mediaproxy module is loaded, but no function from the module is called at all. It can be via some internal callbacks registered to some modules such as tm and dialog, or even script execution callbacks... so look over the code of mediaproxy module. You can run kamailio with debug=3 and from the logs you should get some hints about what is the way media proxy is executed. Cheers, Daniel Thanks Reda On Sat, Feb 25, 2012 at 18:58, Reda Aouad <reda.aouad(a)gmail.com> wrote: > I looked in detail in the source of the call control module and didn't > find anything related to the mediaproxy module neither.. > I'll send you the output of cfgtrace soon. > > Reda > > > > On Fri, Feb 24, 2012 at 23:46, Daniel-Constantin Mierla <miconda(a)gmail.com > > wrote: > >> A quick grep over the callcontrol sources didn't reveal a point where >> mediaproxy is engaged. >> >> Can you load debugger module and enable cfgtrace, then run such a >> scenario and send out the output from cfgtrace to see all the config >> actions executed? >> >> Cheers, >> Daniel >> >> >> On 2/24/12 7:56 PM, Reda Aouad wrote: >> >> It would be great Daniel if you could do it. >> I will be more than happy to test it. >> A function parameter would be more flexible than a module parameter. >> I expect it shouldn't affect the behavior of the external application >> CallControl. >> >> Thanks in advance :) >> >> Reda >> >> >> >> On Fri, Feb 24, 2012 at 08:09, Daniel-Constantin Mierla < >> miconda(a)gmail.com> wrote: >> >>> Hello, >>> >>> I am not using mediaproxy at all (but nathelper/rtpproxy), neither the >>> call control module, but making an option (module parameter or function >>> parameter) for call control to bind to another module like media proxy, >>> should not be big deal if it is all you are looking for -- I can look over >>> it and send a patch if you are going to help testing it. I cannot do it >>> these days, though, being out of the office. >>> >>> Cheers, >>> Daniel >>> >>> >>> On 2/23/12 8:59 PM, Reda Aouad wrote: >>> >>> First, I am posting about the wrong behavior of CallControl (or most >>> probably Kamailio modules) which leaves no option. I should be the only >>> one deciding about how to handle timeouts. If I decide to take some risk, >>> no module should oblige me to do otherwise. >>> >>> Mediaproxy detects ONLY RTP timeouts from BOTH parties, because linux >>> conntrack rules it uses are bi-directional. If a single party stops sending >>> RTP for whatever reason (connection lost, codec with silence detection >>> used, ....), mediaproxy doesn't care and doesn't act upon it. This is a >>> feature, and a wanted one, to mainly support voice-detecting codecs. Think >>> also about conferences for example, in which only a single person talks for >>> a long time while others are silent and don't send RTP. >>> >>> Single-side RTP timeout because of a real problem (loosing network >>> connection for example) should be handled with other methods, such as SIP >>> session timers. >>> >>> MY POINT IS : I don't see it practical to handle RTP flows for EVERY >>> call to handle the least probable scenario: an RTP timeout from both (or >>> all) parties. >>> >>> If I understood well, mediaproxy updates the CDR when it detects an >>> RTP timeout from both parties. CallControl can look in the CDR to debit the >>> correct balance, instead of attaching itself to the dialog module to detect >>> dialog termination. >>> >>> This is an extract from the call_control module : >>> >>> Even when mediaproxy is unable to end the dialog because it was not >>> started with engage_media_proxy(), the callcantrol application is still >>> able to detect calls that did timeout sending media, by looking in the >>> radius accounting records for entries recorded by mediaproxy for calls that >>> did timeout. These calls will also be ended gracefully by the callcontrol >>> application itself. >>> >>> >>> Unless there is something I miss.. >>> >>> I also opened a bug about the issue because call_control doesn't have >>> the same behavior with OpenSips. It doesn't force mediaproxy. >>> >>> Reda >>> >>> >>> >>> On Thu, Feb 23, 2012 at 20:00, Jeff Brower <jbrower(a)signalogic.com>wrote: >>> >>>> Reda- >>>> >>>> > It's clear but not necessary. It can look at radius records fixed by >>>> > mediaproxy on RTP timeout to debit the correct balance as well. And >>>> why >>>> > also force it on postpaid calls which it doesn't control at all ? >>>> >>>> I don't understand how you plan to tear down Kamailio calls that >>>> suffer RTP time-out? >>>> >>>> -Jeff >>>> >>>> > What happens is cost and performance issues for additional calls >>>> passing >>>> > through my mediaproxy server, which I didn't plan for at first. No >>>> audio >>>> > issue at all. >>>> > >>>> > Reda >>>> > >>>> > >>>> > >>>> > On Thu, Feb 23, 2012 at 11:58, Sammy Govind <govoiper(a)gmail.com> >>>> wrote: >>>> > >>>> >> Reading from the module docs its clear why it needs to engage >>>> media/rtp >>>> >> proxy to start,stop billing or timer of a call. so what happens when >>>> it >>>> >> engages mediaproxy on unwanted calls !? audio-issues? >>>> >> >>>> >> >>>> >> On Thu, Feb 23, 2012 at 1:21 PM, Reda Aouad <reda.aouad(a)gmail.com> >>>> wrote: >>>> >> >>>> >>> Thanks Sammy. I didn't get any reply yet. >>>> >>> >>>> >>> CallControl is an application used with CDRTool for prepaid calls. >>>> It >>>> >>> calculates the maximum call duration based on the user's balance. >>>> Once the >>>> >>> call's duration limit is reached, it sends BYE to both calling >>>> parties, >>>> >>> terminating the call. At the end of a prepaid call, terminated >>>> either by >>>> >>> the user or by CallControl, it debits the user's balance according >>>> to the >>>> >>> call's duration. >>>> >>> >>>> >>> The Call_Control module interfaces with this external application. >>>> >>> >>>> >>> call_control function is called in Kamailio's cfg to check if the >>>> user >>>> >>> has prepaid or postpaid account, and get the max call duration for >>>> prepaid >>>> >>> users. CallControl controls only prepaid calls, not postpaid ones. >>>> >>> >>>> >>> So call control and NAT traversal using mediaproxy are two >>>> differents >>>> >>> things which i can't link, since I don't want mediaproxy for every >>>> call. >>>> >>> And since the function call_control is called on every invite before >>>> >>> knowing if the user has a prepaid account or not, it engages >>>> mediaproxy for >>>> >>> every call. >>>> >>> >>>> >>> CallControl relies on mediaproxy to detect RTP timeouts and debit >>>> the >>>> >>> correct balance from a prepaid account based on the last instant the >>>> >>> mediaproxy saw an RTP packet. >>>> >>> >>>> >>> But why to force using mediaproxy with no choice? And why to force >>>> it for >>>> >>> every call, whether it falls under CallControl's control or not? >>>> >>> >>>> >>> I am using Kamailio 3.2. >>>> >>> >>>> >>> >>>> >>> Reda >>>> >>> >>>> >>> On 23 févr. 2012, at 07:21, Sammy Govind <govoiper(a)gmail.com> >>>> wrote: >>>> >>> >>>> >>> Hi, >>>> >>> I can see you posting multiple times on both proxies listings so >>>> I'm sure >>>> >>> you havent heard back from anyone.I am not at all familiar with your >>>> >>> functions in email but could it be possible for you to determine on >>>> which >>>> >>> calls you need to engage mediaproxy and on which not to, then on >>>> the base >>>> >>> of that flag use the call_control function ! >>>> >>> your problem is complicated for me atleast. I hope somebody could >>>> answer >>>> >>> you accurately and precisely. >>>> >>> >>>> >>> btw, what are you using in real? opensips or kamailio, which >>>> version? and >>>> >>> in what context you need to use the call_control function? >>>> >>> >>>> >>> Thanks, >>>> >>> Sammy >>>> >>> >>>> >>> On Thu, Feb 23, 2012 at 12:45 AM, Reda Aouad <reda.aouad(a)gmail.com >>>> >wrote: >>>> >>> >>>> >>>> Hi, >>>> >>>> >>>> >>>> When I use the function call_control( ) of the call_control >>>> module, it >>>> >>>> automatically engages mediaproxy if it finds the mediaproxy module >>>> loaded. >>>> >>>> If the mediaproxy module is not loaded, call_control doesn't even >>>> try to >>>> >>>> engage it. >>>> >>>> >>>> >>>> I need mediaproxy for NAT traversal in some cases, but don't want >>>> it to >>>> >>>> be engaged on every call. >>>> >>>> >>>> >>>> How can I disable this behavior? >>>> >>>> >>>> >>>> Thanks >>>> >>>> Reda >>>> >>>> >>>> >>>> _______________________________________________ >>>> >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>>> list >>>> >>>> sr-users(a)lists.sip-router.org >>>> >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>> >>>> >>>> >>>> >>> _______________________________________________ >>>> >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>>> list >>>> >>> sr-users(a)lists.sip-router.org >>>> >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >>> >>>> >>> >>>> >>> _______________________________________________ >>>> >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>>> list >>>> >>> sr-users(a)lists.sip-router.org >>>> >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >>> >>>> >>> >>>> >> >>>> >> _______________________________________________ >>>> >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>>> list >>>> >> sr-users(a)lists.sip-router.org >>>> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >> >>>> >> >>>> > _______________________________________________ >>>> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>>> list >>>> > sr-users(a)lists.sip-router.org >>>> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> > >>>> >>>> >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >>> -- >>> Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda >>> >>> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> -- >> Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda >> >> > _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda

2 4

obtaining htable array index
by Asgaroth 28 Feb '12

28 Feb '12

Hi All, Is it possible to asertain the array index of an htable entry based on its value? For example, say I have an htable called foo with the following entries: Entry:: 11 testing[0]:: hello-2 testing[1]:: 1 testing::size:: 2 is it possible to: [a] extract the array index based on a regex of the value, for example, something like value count: shtcv(foo=>hello-*) [b] if [a] is possible, how would I extract that value (from the associated array index returned in [a] above) from the htable using sht()? Is the above possible, or am I using the wrong tool for the job? I did have a look at the htable module docs, but i didnt see anything that stood out to me showing that i could achieve the above. I saw an shtval exported function, but i dont see it documented in the pseudo-var wiki page. I then tried it in the routing script but got an error message to the affect that it couldnt find the PV. I assume its not exported. All comments welcome and appreciated? Thanks

2 4

Dropping 183 and sending 180
by Fabian Borot 27 Feb '12

27 Feb '12

Hello We need to implement this scenario: client -> proxy --> provider The provider sends 183 with SDP and we need to drop it and send 180 instead. I managed to do it like this: 1- on_reply block, I catch the 183 if it comes, call a route block named "DROP_180" then drop it: if ($rs == "183") { route(DROP_180); drop(); } 2- on the new route block "DROP_180" I use the command send_reply(180, "Ringing") route[DROP_180] { xlog("L_INFO","mylog: On Route DROP_180."); send_reply(180,"Ringing"); } I had to do it on a separate block because the send_reply command could not be used on the same on_reply block, also I had to send the 180 first before the 183 is dropped because the drop() apparently stops the script execution and the 180 was not sent The idea works fine, BUT the To tag on the locally generated 180 is not the same as the one on the original 183 message. After the progress (183 message), it comes the 200 OK, which we forward downstream unaltered. Now, the client seems to ignore the 200 OK, (there is no ACK), we think that since the To tag on the original 183 and the 200 OK does not match the one on the 180 that is why the customer is processing the 200 OK. Any ideas here? txs a lot

3 4

XMLOPS xpath issues
by Olle E. Johansson 27 Feb '12

27 Feb '12

Friends, I'm having issues with the xpath support. If I run Daniel's example in the XMLOPS, xpath works as documented. But if I take the body of a PUBLISH from the and run xpath, I don't get expected results. Now, I'm no XPATH guru so I may be totally off the markup here... Here's the test script: $xml(pub=>doc) = $rb; $var(contact)=$xml(pub=>xpath:registration/contact@event); xlog("--- Publish contact $var(contact) \n"); xlog("--- Publish aor $xml(pub=>xpath:registration@aor) \n"); xlog("--- Publish aor: $xml(pub=>xpath://@aor) \n"); xlog("--- Publish reg state: $xml(pub=>xpath://reginfo@state) \n"); xlog("--- Publish uri: $xml(pub=>xpath:contact/uri/text()) \n"); I've tested with various paths, orginating from /reginfo or just picking an attribute. Here's the XML from Carsten's PUA_REGINFO module: <?xml version="1.0"?> <reginfo xmlns="urn:ietf:params:xml:ns:reginfo" version="0" state="full"> <registration aor="sip:oej@testnamn.se" id="0x80590a590" state="terminated"/> </reginfo> I wonder if it's something with using the request body that doesn't parse properly in the XMLOPS module? Error messages are a series of "XPath error : Invalid expression" /O

4 6

Using Voicemail strange behavior - git head 3.3
by David 27 Feb '12

27 Feb '12

I was testing the head version in GIT; I ran across a strange issue. The new config includes a "WITH_VOICEMAIL" definition I am not sure exactly how it is designed to work, but by default if the "usrloc" fails it creates a branch and goes to voicemail That part work correctly. It seems however if a phone is registered and the you get a busy, the failure route creates a branch but also retries the original RURI but I dont see a fork being created for the new branch to the voicemail server. Seems to just retry the original RURI. I am using the default config from the kamailio. repo. Any ideas. Thanks.

1 0

change DBURL without restarting kamailio?
by Uri Shacked 27 Feb '12

27 Feb '12

Hi, Is there a way to set a DBURL as a parametes that i can change in run time without restarting kamailio? It is usfull if i have a backup data base and i dont want to use mysql clustering... Cheers, Uri

2 1

CSEQ problem with SPA504G
by David 27 Feb '12

27 Feb '12

Hello, I am trying to connect to a kamailio server using a Cisco SPA504G phone using TCP instead of TCP. When in UDP, it works fine, but in TCP I get errors relating to CSEQ. I included the trace taking with ngrep after the email. I don't understand the CSeq error. I had a look at RFC3261, it says that the CSEQ number has to increase and that's what it has done. Can anyone tell me where the errorlies ? I tried changeing the check sanity to not check that CSeq is valid, but it screws up authentication as it mis detects the method type. Please advise, David interface: eth0 (SERVERIP/255.255.255.128) filter: (ip or ip6) and ( port 5060 and host 206.248.130.233 #### T 206.248.130.233:5075 -> SERVERIP:5060 [AP] REGISTER sip:SERVERNAME.TLD SIP/2.0. Via: SIP/2.0/TCP 192.168.1.225:5075;branch=z9hG4bK-b03863b5;rport. From: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>;tag=3fa2fa4d354b4261o0. To: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>. Call-ID: 4c3756fb-9b03d67a(a)192.168.1.225. CSeq: 50446 REGISTER. Max-Forwards: 70. Contact: "104 Zone T" <sip:testzone.104@192.168.1.225:5075;transport=tcp>;expires=3600. Warning: 399 spa "STUN Server Not Reachable". User-Agent: Cisco/SPA504G-7.4.9c. Content-Length: 0. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE. Supported: replaces. . ## T SERVERIP:5060 -> 206.248.130.233:5075 [AP] SIP/2.0 401 Unauthorized. Via: SIP/2.0/TCP 192.168.1.225:5075;received=206.248.130.233;branch=z9hG4bK-b03863b5;rport=5075. From: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>;tag=3fa2fa4d354b4261o0. To: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>;tag=a889ade7c63805944f01d11bb71a916b.6a29. Call-ID: 4c3756fb-9b03d67a(a)192.168.1.225. CSeq: 50446 REGISTER. WWW-Authenticate: Digest realm="SERVERNAME.TLD", nonce="4f4b87c31ea308c5fdf54c90cf2b30a400fa97a7". Server: OmniVigil 5.0a. Content-Length: 0. . ## T 206.248.130.233:5075 -> SERVERIP:5060 [AP] REGISTER sip:SERVERNAME.TLD SIP/2.0. Via: SIP/2.0/TCP 192.168.1.225:5075;branch=z9hG4bK-dcfc09e;rport. From: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>;tag=3fa2fa4d354b4261o0. To: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>. Call-ID: 4c3756fb-9b03d67a(a)192.168.1.225. CSeq: 50447 REGISTER. Max-Forwards: 70. Authorization: Digest username="testzone.104",realm="SERVERNAME.TLD",nonce="4f4b87c31ea308c5fdf54c90cf2b30a400fa97a7",uri="sip:SERVERNAME.TLD",algorithm=MD5,response="77ddf2a1baafd33ed0c675fdc074bc03". Contact: "104 Zone T" <sip:testzone.104@206.248.130.233:5075;transport=tcp>;expires=3600. Warning: 399 spa "STUN Server Not Reachable". User-Agent: Cisco/SPA504G-7.4.9c. Content-Length: 0. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE. Supported: replaces. . ## T SERVERIP:5060 -> 206.248.130.233:5075 [AP] SIP/2.0 400 CSeq method does not match request method. Via: SIP/2.0/TCP 192.168.1.225:5075;branch=z9hG4bK-dcfc09e;rport=5075;received=206.248.130.233. From: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>;tag=3fa2fa4d354b4261o0. To: "104 Zone T" <sip:testzone.104@SERVERNAME.TLD>;tag=1feca66ab82b1f32a901c4cd8a4b7a05.e8e6. Call-ID: 4c3756fb-9b03d67a(a)192.168.1.225. CSeq: 50447 REGISTER. Server: kamailio (3.1.5 (x86_64/linux)). Content-Length: 0. .

2 1

How to do accounting for Kamailio-generated BYEs?
by Yufei Tao 27 Feb '12

27 Feb '12

Hi I've got a mediaproxy that detects no media, and sends dlg_end_dlg to Kamailio via mi_datagram socket. Then Kamailio sends BYE to both calling parties. It all works fine. But my question is, how to do accounting for these Kamailio-generated BYEs? They don't appear in the acc table. I'm using Kamailio 3.1.5. Thank you very much! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

2 1

CallControl and MediaProxy
by Reda Aouad 27 Feb '12

27 Feb '12

Hi, When I use the function call_control( ) of the call_control module, it automatically engages mediaproxy if it finds the mediaproxy module loaded. If the mediaproxy module is not loaded, call_control doesn't even try to engage it. I need mediaproxy for NAT traversal in some cases, but don't want it to be engaged on every call. How can I disable this behavior? Thanks Reda

4 12

Social networking event, London, March 6, 2012
by Daniel-Constantin Mierla 27 Feb '12

27 Feb '12

Hello, we are already several folks planing to go out after the first day of UC Expo and continue discussions about latest news of Kamailio/SER project and VoIP/Telecomunications. If you want to join, drop me a message to send you the details of the location. Looking forward to meeting many of you next week, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Re: [SR-Users] How to do request URI rewrites using database tables
by Javier Gallart 27 Feb '12

27 Feb '12

Hello Ben dialplan module should help you: http://kamailio.org/docs/modules/3.2.x/modules/dialplan.html Regards Javi > ------------------------------ > > Message: 5 > Date: Fri, 17 Feb 2012 12:49:04 +1300 > From: Ben WIlliams <benwilliams(a)joobworld.com> > Subject: Re: [SR-Users] How to do request URI rewrites using database > tables? > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <CANfYGQkeK3i=vBbdox+ekGeeKmjWhhgarUJcAo3oV=rrO=Wwig(a)mail.gmail.com > > > Content-Type: text/plain; charset=ISO-8859-1 > > I've managed to get dbaliases and lcr to do most of this. But the lcr > prefix does not allow regular expressions. Is there any other module > that allows you to store in the database a regular expression rewrite > rule? > > On Thu, Feb 16, 2012 at 1:21 PM, Ben WIlliams <benwilliams(a)joobworld.com> > wrote: > > Hi, can someone please recommend the most appropriate modules to > > rewrite R-URIs based on a database lookup table? I've read the > > documentation for lcr and carrierroute but not sure if they can do > > this. > > > > In most cases it will be a simple R-URI rewrite but I also need to > > rewrite based on From user. > > > > ie > > > > R-URI match ? ? ? ? ? ? From match ? ? ? ? ? ? ? ? ?new R-URI > > =========== ? ? ? ? ? ? ========== ? ? ? ? ? ? ? ? ?========= > > *97(a)example.com ? ? ? ? 6[0-9](a)example.com ? ? ? ? ?*97(a)10.0.0.1 > > *97(a)example.com ? ? ? ? 7[0-9](a)example.com ? ? ? ? ?*97(a)10.0.0.2 > > 6[0-9](a)example.com ? ? ?match any ? ? ? ? ? ? ? ? ? rewrite domain to > 10.0.0.1 > > abc(a)example.com ? ? ? ? match any ? ? ? ? ? ? ? ? ? def(a)example.com > > > > Thanks > > Ben > > > >

2 1

Have-my-domain! - host your own SIP domain at iptel.org
by SuperCel.mx.tc 26 Feb '12

26 Feb '12

Sirs: We cannot get our domain to configure to the iptel SIP Service (Have-my-domain! - host your own SIP domain at iptel.org). I have attached a copy of how we've tried setting the DNS, but we cannot find anywhere that gives us info on the "SIP" and/or "protocol UDP". Can you assist us so that we can use our own domain (http://supercel.tel) so that we can send SIP calls across this domain? Sincerely, Frank, CTO SuperCel.tel

1 0

Have-my-domain! - host your own SIP domain at iptel.org
by SuperCel.mx.tc 26 Feb '12

26 Feb '12

Sirs: We cannot get our domain to configure to the iptel SIP Service. I have attached a copy of how we've tried setting the DNS, but we cannot find anywhere that gives us info on the "SIP" and/or "protocol UDP". Can you assist us so that we can use our own domain (http://supercel.tel) so that we can send SIP calls across this domain? Sincerely, Frank Riley, CTO SuperCel.tel

1 0

Kamailio segmentation fault
by Fabian Borot 24 Feb '12

24 Feb '12

HelloOur kamailio proxy (version 1.5.2 ) crashed today due to a segmentation fault:Feb 22 16:52:10 AW-PROXY-01 kernel: kamailio[3667]: segfault at 000000000000000b rip 00002ab2b47bf93f rsp 00007ffff62dd270 error 6Apparently due to lack of private memory:[root@AW-PROXY-01 log]# cat proxy.log | grep ERROR | grep dbFeb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:db_unixodbc:db_unixodbc_convert_rows: no private memory left Feb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:db_unixodbc:db_unixodbc_convert_result: converting rows failed Feb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:db_unixodbc:db_unixodbc_store_result: failed to convert result Feb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:core:db_do_query: error while storing result Feb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:db_unixodbc:db_unixodbc_fetch_result: no private memory left Feb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:htable:ht_db_load_table: Error while fetching result I was hoping to see a core file somewhere but I haven't found one yet. I did not have the line "disable_core_dump=no" set on the config file because supposedly is set to no by default: >From the online guide for my version: //////////////////// Disable_core_dumpCan be 'yes' or 'no'. By default core dump limits are set to unlimited or a high enough value. Set this config variable to 'yes' to disable core dump-ing (will set core limits to 0).Default value is 'no'. ///////////////////How can I make sure that a core file is created after a situation like this? Where will it be created? How can I check the status of the private memory? I use the "kamctl fifo get_statistics all" but I don't know which one relates to "private memory": Example:shmem:total_size = 2147483648 shmem:used_size = 16860928 shmem:real_used_size = 17033968 shmem:max_used_size = 19174528 shmem:free_size = 2130449680 shmem:fragments = 1850I know that with so little information is hard to diagnose, but if anybody has an idea as to what else I can do I greatly appreciate it. Thank you in advancefborot

3 7

avp_delete problem [kamailio 1.5]
by Sven Knoblich 24 Feb '12

24 Feb '12

Hello all, i am currently confused by using the avpops function avp_delete. When i run avp_delete without the flag-value \g only the last value will be unset (like an undo). Is this the wanted behaviour? EXAMPLE: $avp(test) = "test" if( $avp(test)){xlog("L_NOTICE","result1:$avp(test)\n");} $avp(test) = "nooo"; if( $avp(test)){xlog("L_NOTICE","result2:$avp(test)\n");} avp_delete("$avp(test)"); if( $avp(test)){xlog("L_NOTICE","result3:$avp(test)\n");} RESULT: result1:test result2:nooo result3:test Could anybody helps me to understand that? thanks in advance, Sven

4 5

Adding parameter to headers
by Uri Shacked 24 Feb '12

24 Feb '12

Hi, I need to add a parameter to the contact header. suppose to look like: Contact: <sip:anonymous@domain:port;param1=XXX;param2=YYY> How do I add the parameters? Should i concatenate the $ct with the parameters? Or is there a better way? BR, Uri

2 1

fix_nated_sdp issue
by Ric Marques 24 Feb '12

24 Feb '12

Greetings, I'm not sure if I found a bug, or if I just have something completely misconfigured... I'm a total newb with Kamailio, working on a proof of concept design. Here's my configuration: provider -> nat firewall -> kamailio/rtpproxy -> asterisk For outbound calls from a phone registered to asterisk via kamailio, I'm trying to use fix_nated_sdp("2", "10.50.50.8") to rewrite the media ip address to resolve my audio issues, where 10.50.50.8 is the address outside my firewall. What I'm running into is the 'c=' line doesn't get re-written properly... it inserts the specified address in front of the existing address, and I end up with the following line in my INVITE: c=IN IP4 10.50.50.810.0.10.10 I have the fix_nated_sdp command under route[sipout], because I only want to use it on calls being sent outside the nat firewall. Here's the sip invite without the 'fix_nated_sdp' command: -------------------------------------------------------------------------------------------------------------- INVITE sip:19165551212@xxx.xxx.xxx.xxx SIP/2.0 Record-Route: <sip:10.0.10.10;lr=on;ftag=as5498b77e;nat=yes> Via: SIP/2.0/UDP 10.50.50.8.;branch=z9hG4bK4b3a.960f6466.0 Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bK145db73e;rport=5060 Max-Forwards: 69 From: "1009" <sip:1009@10.0.10.11>;tag=as5498b77e To: <sip:19165551212@xxx.xxx.xxx.xxx> Contact: <sip:1009@10.0.10.11:5060> Call-ID: 06b8bb1b7dd7801d7b3b9c917fcb9b12@10.0.10.11:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r356107 Date: Wed, 22 Feb 2012 03:06:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 309 P-hint: outbound v=0 o=root 604360056 604360056 IN IP4 10.0.10.10 s=Asterisk PBX SVN-branch-1.8-r356107 c=IN IP4 10.0.10.10 t=0 0 m=audio 9702 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=nortpproxy:yes -------------------------------------------------------------------------------------------------------------- Here's the sip invite with the 'fix_nated_sdp' command: -------------------------------------------------------------------------------------------------------------- INVITE sip:19167828326@xxx.xxx.xxx.xxx SIP/2.0 Record-Route: <sip:10.0.10.10;lr=on;ftag=as49e00c81;nat=yes> Via: SIP/2.0/UDP 10.50.50.8.;branch=z9hG4bK1eab.800c4724.0 Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bK20d28324;rport=5060 Max-Forwards: 69 From: "1009" <sip:1009@10.0.10.11>;tag=as49e00c81 To: <sip:19167828326@xxx.xxx.xxx.xxx> Contact: <sip:1009@10.0.10.11:5060> Call-ID: 4def5539675b6f644b99bb300e8ec8d6@10.0.10.11:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r356107 Date: Wed, 22 Feb 2012 03:18:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 347 P-hint: outbound v=0 o=root 1009117068 1009117068 IN IP4 10.0.10.10 s=Asterisk PBX SVN-branch-1.8-r356107 c=IN IP4 10.50.50.8.10.0.10.10 t=0 0 m=audio 13540 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=oldmediaip:10.0.10.11 a=nortpproxy:yes -------------------------------------------------------------------------------------------------------------- Is this a bug, or is it likely I have something else screwed up? Thank you in advance for your assistance - this list is an incredible resource! -Ric

2 3

Require Certificate
by Bruno Bresciani 22 Feb '12

22 Feb '12

Hi All, Does it possible on tls module require certificates only some hosts? My kamailio comunicate with several UA clients (Softphones and IP phones) and one SIP gateway. I want require certificate only gateway SIP, it's possible? The parameter *modparam("tls", "require_certificate", 1)* require certificate of all clients... Best Regards

2 4

Topoh hiding prefixes
by Efelin Novak 22 Feb '12

22 Feb '12

Hi folks, I have a topology like this: UA -> kamailio -> PSTN GW. When I'm placing calls to a PSTN I have to append prefixes so the PSTN GW knows how to route the call. So the request-uri is <prefix><number>@<gateway_ip> e.g. 999123456(a)192.168.1.1. When the UA calls to the GW a "200 OK" message contains a contact header with 999123456(a)192.168.1.1. This is a problem because I don't want my customers to see gateway prefixes. The Topoh module only hides IP addresses or the domain part. Is there any way to remove prefixes. This document (http://www.kamailio.org/events/2011-Cluecon/DCM-kamailio-security.pdf) stated "encoding IP and prefixes can be set via parameters". Can I remove prefixes from the contact in any other way than some contact header substitutions? Thanks Efelin

3 9

CallControl and MediaProxy
by Reda Aouad 22 Feb '12

22 Feb '12

Hi, I noticed that calling call_control( ) function forces engage_media_proxy( ). Is there a way to disable this behavior ? Thanks Reda

1 0

Kamailio v3.2 problem with dialog module.
by Phillman25 Kyriacou 21 Feb '12

21 Feb '12

Dear List After upgrading from V3.1.5 to V3.2. Kamailio crashes with the below error messages: Feb 20 16:43:59 sipproxy /usr/local/sbin/kamailio[11705]: ERROR: <core> [db.c:392]: invalid number of rows received: 4, dialog_vars Feb 20 16:43:59 sipproxy /usr/local/sbin/kamailio[11705]: ERROR: <core> [db.c:419]: querying version for table dialog_vars Feb 20 16:43:59 sipproxy /usr/local/sbin/kamailio[11705]: ERROR: dialog [dlg_db_handler.c:153]: error during dialog-vars version check. Feb 20 16:43:59 sipproxy /usr/local/sbin/kamailio[11705]: ERROR: dialog [dialog.c:646]: failed to initialize the DB support Feb 20 16:43:59 sipproxy /usr/local/sbin/kamailio[11705]: ERROR: <core> [sr_module.c:932]: init_mod(): Error while initializing module dialog (/usr/local/lib/kamailio/modules_k/dialog.so) My dialog paramaters are as follows: # ----------------- DIALOG MODULE PARAMETERS----------------------------------# modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 4096) modparam("dialog", "rr_param", "did") modparam("dialog", "dlg_flag", 4) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "default_timeout", 21600) modparam("dialog", "dlg_extra_hdrs", "NULL") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "detect_spirals", 1) modparam("dialog", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("dialog", "db_mode", 1) modparam("dialog", "db_update_period", 60) modparam("dialog", "db_fetch_rows", 500) modparam("dialog", "table_name", "dialog") modparam("dialog", "from_uri_column", "from_uri") modparam("dialog", "from_tag_column", "from_tag") modparam("dialog", "to_uri_column", "to_uri") modparam("dialog", "to_tag_column", "to_tag") modparam("dialog", "h_id_column", "hash_id") modparam("dialog", "h_entry_column", "hash_entry") modparam("dialog", "state_column", "state") modparam("dialog", "start_time_column", "start_time") modparam("dialog", "timeout_column", "timeout") modparam("dialog", "sflags_column", "sflags") modparam("dialog", "bridge_controller", "sip:controller@kamailio.org") modparam("dialog", "default_timeout", 7200) i realised that when i set the "modparam("dialog", "db_mode", 0)" kamailio starts fine, seems to be a problem connecting to the database. Does anyone know what is wrong? Thanking you in advance Phillip

2 3

UAC canceling a specific early dialog with BYE
by Andrew Pogrebennyk 21 Feb '12

21 Feb '12

Hello, we have a Kamailio proxy which gets the call from PSTN gw and does some call forwarding (serial forking) to several destinations through our sbc The call flow I am looking at is: - Kamailio sends INVITE to branch_1. - branch_1 sends 180 with to-tag*, proxy relays it to the gw * 180 meets the requirements for dialog creating 18x responses in sections 12.1, 12.1.1 because it contains to-tag, contact and mirrored record-route. - After some seconds Kamailio sends a CANCEL to branch_1. - And sends the INVITE to branch_2. - branch_1 replies 200 for the CANCEL and 487 for the INVITE. - branch_2 replies 180 and 200 for the INVITE. - When PSTN gw receives that it sees it still needs to cancel other early dialog established by 180 from branch_1. - The PSTN gw sends a BYE with to-tag of branch_1 to cancel this specific early dialog. SIP allows early dialogs to individually released while other dialogs continue, as written in RFC, section 15: "The BYE request is used to terminate a specific session or attempted session. In this case, the specific session is the one with the peer UA on the other side of the dialog. (...). The caller’s UA MAY send a BYE for either confirmed or early dialogs, and the callee’s UA MAY send a BYE on confirmed dialogs, but MUST NOT send a BYE on early dialogs." The BYE follows the loose routing path, proxy gets 481 from the sbc and forwards that response back to PSTN gw, which somehow breaks it. AFAICS it's not specified in RFC what should the behavior look like when getting both a 200 and error-class response for the INVITE (quotes are most welcome!). IMO it would be more correct to absorb BYE in proxy but I see a big problem here: branch_2 can even ring for 5 minutes and it's not feasible for proxy to have a wt-timer that long. Also it's not possible to inform the gw that early dialog has cleared as soon as we receive 200/487 from branch_1. So I'm not sure which party is at fault and if we can workaround that somehow in the Kamailio. Any thoughts?

1 1

Re: [SR-Users] Kamailio ims / rtpproxy
by trojak＠st.fri.uniza.sk 21 Feb '12

21 Feb '12

Hi All! We did litttle research: - both behind the NAT = all OK = straeming over the rtpproxy - caller behind NAT (other public) = all BAD = no change in SDP (Connection Address) - rtp stream sending on private addres - calle behind NAT (other public) = all BAD = SDP has changed but only private IP address... public IP unchanged = so rtpproxy do not ralay one way stream Can somebody (especialy IMS) please help us Thanks Best regards, Lubo > Hi Daniel, > > thank you for answer. I think that parts of kamailio default config > will not working with pcscf. (e.g. there is on pcscf no database - > it has hss) > > It partialy works with pcscf config -> it rewrites Connecition > Infromation IP in SDP: > - sometimes with IP of rtpproxy (more often when both clietnts are > behind NAT) > - sometimes with public IP of NATed client (we are not sure why) > - or unchanged (when one client is behind NAT and other is public) > > but this works only with Monster client, with Boghe IP is unchanged > (shouldn't it be client independent ?). > > Maybe someone from IMS help us. > > Thanks all for any advice. > > Best regards, > > Lubo > > Daniel-Constantin Mierla <miconda(a)gmail.com>: > >> Hello, >> >> On 1/23/12 10:43 AM, trojak(a)st.fri.uniza.sk wrote: >>> >>> Hi, >>> >>> I am trying to make kamailio PCSCF working with rtpproxy (for NAT >>> traversal). >>> I would like to ask, if there is any tutorial for this, or if you >>> can give me any advice. >> Maybe IMS guys can give other hints, what is on public so far was >> summarized at: >> * http://www.kamailio.org/w/2011/01/ims-extensions-available-for-testing/ >> >>> I also have installed rtpproxy package; #!define WITH_NAT, and >>> defined rtpproxy port in config file... >>> But this is not working. I noticed, that in default config file of >>> pcscf is nothing in routing logic with rtpproxy (or it is >>> unnecessary ?), so I try to copy route[RELAY] route[NAT] >>> route[RTPPROXY] modules from kamailio (no pcscf) config file... >>> but also with no success. >>> >>> Can you help me please ? What I am doing wrong ? Do you have any >>> configuration working with rtpproxy ? >> Be sure you can the nat handling routes for INVITES behind nat. You >> can follow the logic in the default config file and adapt it for >> you pcscf config. In summary should be like: >> - invite comes in, test if it is from caller behind nat >> - lookup location returns a branch flag if the calee is behind nat >> (you have to set that branch flag for register request coming from >> users) >> - if caller or callee are behind nat, engage rtpproxy for that call >> and set a onreply route to update sdp for 18x/200 replies >> - default config adds a Route header parameter for nated calls to >> detect easily the natted situation for re-invites or bye >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla -- http://www.asipto.com >> http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Dialogs not removed from memory, and occasionally persistent in DB as well
by Øyvind Kolbu 20 Feb '12

20 Feb '12

Hi, we use the dialog-module to keep track of concurrent calls, and then set a treshold for simultanious calls. After we upgraded from 1.5 to 3.2, we've been having a lot of dialogs which are stuck in both the database and memory. kamailio -V version: kamailio 3.2.1 (i386/linux) 918035-dirty flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_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, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 918035 -dirty compiled on 10:00:10 Dec 20 2011 with gcc 4.1.2 Our configuration: modparam("dialog", "dlg_flag", 3) modparam("dialog", "default_timeout", 21600) modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "db_url", VOIP_DATA4) modparam("dialog", "db_mode", 1) #!ifdef VOIP1 modparam("dialog", "table_name", "dialog1") modparam("dialog", "vars_table_name", "dialog_vars1") #!endif #!ifdef VOIP2 modparam("dialog", "table_name", "dialog2") modparam("dialog", "vars_table_name", "dialog_vars2") #!endif modparam("dialog", "profiles_with_value", "busy") To Kamailio instances, voip1 and voip2, share the same config, but define VOIP1 and VOIP2, respectively. Then in the main route: if !allow_trusted() { route(AUTH); } if (is_method("INVITE")) { setflag(3); dlg_manage(); if ($avp(s:f_uid) != $null) { set_dlg_profile("busy","$avp(s:f_uid)"); get_profile_size("busy", "$avp(s:f_uid)", "$var(dlg_busy)"); xlog("L_INFO", "BUSY: f_uid: $avp(s:f_uid), dlg_busy: $var(dlg_busy)\n"); } } $avp(s:f_uid) is set in route(AUTH), so it will catch calls originating from one of our users. Then later in the callforward route: # Testing dialog $var(dlg_busy) = 0; $var(dlg_busy1) = 0; $var(dlg_busy2) = 0; #!ifdef VOIP1 get_profile_size("busy", "$rU", "$var(dlg_busy1)"); sql_query("data4","select id from dialog2 where caller_contact like 'sip:$rU@%' or callee_contact like 'sip:$rU@%'", "dialog"); $var(dlg_busy2) = $dbr(dialog=>rows); #!endif #!ifdef VOIP2 get_profile_size("busy", "$rU", "$var(dlg_busy2)"); sql_query("data4","select id from dialog1 where caller_contact like 'sip:$rU@%' or callee_contact like 'sip:$rU@%'", "dialog"); $var(dlg_busy1) = $dbr(dialog=>rows); #!endif sql_result_free("dialog"); $var(dlg_busy) = $var(dlg_busy1) + $var(dlg_busy2); And then check if $var(dlg_busy) is above a users treshold. Finally when routing to a local user we mark it as busy: set_dlg_profile("busy","$rU"); As of writing I have 5 entries in the dialog1-table, all valid, but: # kamctl fifo dlg_list | grep -c "^dialog" 18 # kamctl fifo dlg_list | grep state | sort | uniq -c 13 state:: 1 5 state:: 4 on the other hand in the busy profile, which is the one we use: # kamctl fifo profile_list_dlgs busy | grep -c "^dialog" 11 # kamctl fifo profile_list_dlgs busy | grep state | sort | uniq -c 6 state:: 1 5 state:: 4 It seems as this happens when, for some reason, kamailio is trying to parse duplicate INVITEs. From the logs: # grep "^Jan 17" /var/log/messages | grep BUSY | grep -v "dlg_busy: 1" | cut -d\ -f8- |sort | uniq -c 1 BUSY: f_uid: 2367449, dlg_busy: 2 3 BUSY: f_uid: 2367453, dlg_busy: 2 1 BUSY: f_uid: 2367453, dlg_busy: 3 1 BUSY: f_uid: 2367453, dlg_busy: 4 1 BUSY: f_uid: 2574912, dlg_busy: 2 1 BUSY: f_uid: 2582270, dlg_busy: 2 1 BUSY: f_uid: ahmedma, dlg_busy: 2 1 BUSY: f_uid: josephim, dlg_busy: 2 1 BUSY: f_uid: perla, dlg_busy: 2 1 BUSY: f_uid: rosalinm, dlg_busy: 2 1 BUSY: f_uid: suresha, dlg_busy: 2 1 BUSY: f_uid: sveinb, dlg_busy: 2 I've full debugging logs available, if needed. -- Øyvind Kolbu

5 13

How to deal with diversion loops.
by Uri Shacked 20 Feb '12

20 Feb '12

Hi, the maxfwd will not help me prevent the loop on the first round, it would only stop it a little sooner the X times. no? the dlg_var is interesting, but i didn't understand how to use it exactly. how do i match two dialogs? is the spiral detection on dialog module helps? thanks, Uri >Hi Uri, >for that you can easily count the forwards by using the maxfwd module >(mf_process_maxfwd_deader <your max>) or you can store all your spiraled >calls into a dialog-variable like >$dlg_var(forward-chain)=<B>;<C>|<C>;<B> and check whether the next >number is already stored or not. Bye Sven Am 20.02.2012 13:20, schrieb Uri Shacked: >* *>* Hi, *>* *>* The service i built in kamailio is simple - a caller dials an Access *>* number, my gateway send it to kamailio. *>* *>* kamailio finds the shadow number for this Access number and send an *>* invite to the same gateway with the shadow number as destination *>* (state full proxy). *>* *>* In kamailio i change only the RURI and the “To” and “From” stays *>* untouched. *>* *>* Now, when a client performs “diversion”, I can see that I can find *>* myself in an unwanted long loop…. *>* *>* Example : *>* *>* A calls B *>* *>* B diverted the call unconditionally to C. *>* *>* C is an Access Number that its shadow number is D. *>* *>* D diverted the call to B……. *>* *>* Well, that is a bad loop to have! *>* *>* It can be longer or shorter, but the idea I understood (I think). *>* *>* I thought about diversion check, counters, dialog “To” and “From” *>* check, etc…. *>* *>* Anyone here dealt with it? *>* *>* What will be the best way to check and see if the situation accrues? *>* *>* Thanks, *>* *>* Uri *>* *

1 0

integration kamailio-sems
by Uri Shacked 20 Feb '12

20 Feb '12

Thanks, did you tried to develop an IVR system based on SEMS? will it be better? harder? more flexible than Asterisk?

1 0

Kamailio as SBC
by Martin Schiøtz 20 Feb '12

20 Feb '12

Hi I'm new to be working with SIP. We have a SIP server and it handles SIP well but it does not handle far-end-nat. Can the Kamailio server be used as a SBC (Session Border Controller) to handle far-end-nat? Just forward all SIP packages to a SIP server and fix the far-end-nat problem? Cheers! Martin

2 2

How to deal with diversion loops.
by Uri Shacked 20 Feb '12

20 Feb '12

Hi, The service i built in kamailio is simple - a caller dials an Access number, my gateway send it to kamailio. kamailio finds the shadow number for this Access number and send an invite to the same gateway with the shadow number as destination (state full proxy). In kamailio i change only the RURI and the “To” and “From” stays untouched. Now, when a client performs “diversion”, I can see that I can find myself in an unwanted long loop…. Example : A calls B B diverted the call unconditionally to C. C is an Access Number that its shadow number is D. D diverted the call to B……. Well, that is a bad loop to have! It can be longer or shorter, but the idea I understood (I think). I thought about diversion check, counters, dialog “To” and “From” check, etc…. Anyone here dealt with it? What will be the best way to check and see if the situation accrues? Thanks, Uri

2 1

integration kamailio-sems
by Uri Shacked 20 Feb '12

20 Feb '12

Hi, I started reading a little on SEMS. Tried to look for the way Kamailio works has the frontend sip server and SEMS will be the media server (IVR, announcments etc). I didn't understand which kamailio module should i work with in order to open sockets with SEMS. Or do i get it all wrong? Uri

2 1

$time(year) gives 1900
by Uri Shacked 20 Feb '12

20 Feb '12

Hi, I am using the $time(name) pv. On my lab machine (centos 5.7) i get the right value (2012). But, on my other machine (centos 6.2) i get 1900. what am i missing here?

2 2

Mtree VS Htable getting huge amount of data from DB not on local machine
by Uri Shacked 20 Feb '12

20 Feb '12

> > Hi, > > untill today i used the mtree module to get some data from a remote DB and > load it into memory. > the table in the data base has 5 million rows. prefix is a number and > value is some string. > > i moved to htable because i need the prefix exact match and not longest.... > > now, after 5-10 sec, i get the following error (that i never got in mtree) > - "lost connection to mysql during query" and kamailio does not start. > > any ideas how to solve it? > > cheers, > > Uri >

2 2

Registration replication
by Spencer Thomason 19 Feb '12

19 Feb '12

Hello, I'm currently working on a HA setup with geographically separated proxies. Each proxy is in a different data center with an MPLS link between them. The proxies access a locally connected PostgreSQL database that is replicated from a master in the primary data center. Note that the replicated dbs are read only. Everything works great for read only queries but registrations seem to pose a problem as they must use the primary db directly to deal with the writes. If I set userloc to db mode 2 the secondary proxies wipe out the entries in the database. If I use db mode 3 everything works great but all of the secondary proxies must use the primary database directly and a network link failure will cause a service disruption. Is there any way I can use separate read and write databases without p_userloc? Is there a better way to replicate registrations maintaining availability and scalability? And advice is appreciated. Thanks, Spencer Connected by DROID on Verizon Wireless

2 1

Segmentation Fault encountered in pv.so
by Simpson Chua 19 Feb '12

19 Feb '12

Hi, Since I am getting errors in from the usrloc module, I am trying another way to parse out the information I need using pv.so. I'm encountering similar segfaults issues. Here are snippets of my configs: modparam("sipcapture", "db_url", "mysql://xxx:xxx@localhost/test_db") modparam("sipcapture", "capture_on", 1) modparam("sipcapture", "table_name", "sip_capture") modparam("sipcapture", "raw_socket_listen", "127.0.0.1:5060-9000") modparam("sipcapture", "raw_interface", "eth2") modparam("sipcapture", "raw_sock_children", 4) modparam("sipcapture", "raw_moni_capture_on", 1) modparam("sipcapture", "promiscious_on", 1) route { if (method == "REGISTER") { if (avp_check("$si","eq/10.xxx.xxx.xxx")) { #save("location", "0x04"); # This currently results in segfault and is being addressed by Alexandr sip_capture(); xlog("The received IP is $Ri\n"); # Just to test if I am able to parse the received IP from the Via headers correctly drop; } drop; } } Feb 17 13:41:02 ubuntu kernel: [251382.760104] kamailio[9414]: segfault at 31 ip 00007fb4ef077611 sp 00007fff0c8a6468 error 4 in pv.so[7fb4ef06a000+39000] Core was generated by `kamailio -w /home/core'. Program terminated with signal 11, Segmentation fault. #0 pv_get_rcvip (msg=0x7fb4efdd7598, param=0x7fb4efdcef88, res=0x7fff0c8a64b0) at pv_core.c:647 647 || msg->rcv.bind_address->address_str.s==NULL) (gdb) bt #0 pv_get_rcvip (msg=0x7fb4efdd7598, param=0x7fb4efdcef88, res=0x7fff0c8a64b0) at pv_core.c:647 #1 0x000000000049b5ad in pv_get_spec_value (msg=0x7fb4efdd7598, sp=0x7fb4efdcef70, value=0x7fff0c8a64b0) at pvapi.c:1180 #2 0x000000000049b71d in pv_printf (msg=0x7fb4efdd7598, list=<optimized out>, buf=<optimized out>, len=0x7fff0c8a6558) at pvapi.c:1239 #3 0x00007fb4ed0d442e in xlog_helper (msg=<optimized out>, xm=0x7fb4efdceef0, level=-1, line=0) at xlog.c:168 #4 0x000000000041babb in do_action (h=0x7fff0c8a73b0, a=0x7fb4efdd02a0, msg=0x7fb4efdd7598) at action.c:1122 #5 0x000000000041abe2 in run_actions (h=0x7fff0c8a73b0, a=0x7fb4efdd0080, msg=0x7fb4efdd7598) at action.c:1610 #6 0x000000000041c05b in do_action (h=0x7fff0c8a73b0, a=0x7fb4efdd10d0, msg=0x7fb4efdd7598) at action.c:1107 #7 0x000000000041abe2 in run_actions (h=0x7fff0c8a73b0, a=0x7fb4efdd10d0, msg=0x7fb4efdd7598) at action.c:1610 #8 0x000000000041c05b in do_action (h=0x7fff0c8a73b0, a=0x7fb4efdd1350, msg=0x7fb4efdd7598) at action.c:1107 #9 0x000000000041abe2 in run_actions (h=0x7fff0c8a73b0, a=0x7fb4efdd1350, msg=0x7fb4efdd7598) at action.c:1610 #10 0x0000000000422fa2 in run_top_route (a=0x7fb4efdd1350, msg=0x7fb4efdd7598, c=<optimized out>) at action.c:1683 #11 0x00000000004b061e in receive_msg (buf=<optimized out>, len=<optimized out>, rcv_info=<optimized out>) at receive.c:207 #12 0x00007fb4edb538b0 in raw_capture_rcv_loop (rsock=5, port1=5060, port2=8933, ipip=0) at sipcapture.c:1518 #13 0x00007fb4edb53e1f in init_rawsock_children () at sipcapture.c:572 #14 0x00007fb4edb54033 in child_init (rank=<optimized out>) at sipcapture.c:542 #15 0x0000000000507a0f in init_mod_child (m=0x7fb4efdcb2b0, rank=0) at sr_module.c:886 #16 0x0000000000507974 in init_mod_child (m=0x7fb4efdcb668, rank=0) at sr_module.c:883 #17 0x0000000000507974 in init_mod_child (m=0x7fb4efdcc178, rank=0) at sr_module.c:883 #18 0x0000000000507974 in init_mod_child (m=0x7fb4efdcc4b0, rank=0) at sr_module.c:883 #19 0x0000000000507974 in init_mod_child (m=0x7fb4efdccaf0, rank=0) at sr_module.c:883 #20 0x0000000000507974 in init_mod_child (m=0x7fb4efdcd030, rank=0) at sr_module.c:883 #21 0x0000000000476149 in main_loop () at main.c:1661 #22 0x000000000041a940 in main (argc=<optimized out>, argv=0x7fff0c8a7be8) at main.c:2475 (gdb) p *msg $1 = {id = 82, pid = 9418, fwd_send_flags = {f = 0 '\000', blst_imask = 0 '\000'}, rpl_send_flags = {f = 0 '\000', blst_imask = 0 '\000'}, first_line = { type = 1, len = 43, u = {request = {method = { s = 0x7fb4edd57d2a "REGISTER sip:abc.def.com SIP/2.0\r\nFrom: \"9995551234\" <sip:9995551234@abc.def.com;rp=SandboxTestingRegistrations>;tag=F9EEDA6C-E203EB89\r\nTo: <sip:9995551234@abc.def.com>\r\nCal"..., len = 8}, uri = { s = 0x7fb4edd57d33 "sip:abc.def.com SIP/2.0\r\nFrom: \"9995551234\" <sip:9995551234@abc.def.com;rp=SandboxTestingRegistrations>;tag=F9EEDA6C-E203EB89\r\nTo: <sip:9995551234@abc.def.com>\r\nCall-ID: 225"..., len = 24}, version = { s = 0x7fb4edd57d4c "SIP/2.0\r\nFrom: \"9995551234\" <sip:9995551234@abc.def.com;rp=SandboxTestingRegistrations>;tag=F9EEDA6C-E203EB89\r\nTo: <sip:9995551234@abc.def.com>\r\nCall-ID: 2257e150-b49dfd8d-c8d924b2@1"..., len = 7}, method_value = 32}, reply = {version = { s = 0x7fb4edd57d2a "REGISTER sip:abc.def.com SIP/2.0\r\nFrom: \"9995551234\" <sip:9995551234@abc.def.com;rp=SandboxTestingRegistrations>;tag=F9EEDA6C-E203EB89\r\nTo: <sip:9995551234@abc.def.com>\r\nCal"..., len = 8}, status = { s = 0x7fb4edd57d33 "sip:abc.def.com SIP/2.0\r\nFrom: \"9995551234\" <sip:9995551234@abc.def.com;rp=SandboxTestingRegistrations>;tag=F9EEDA6C-E203EB89\r\nTo: <sip:9995551234@abc.def.com>\r\nCall-ID: 225"..., len = 24}, reason = { s = 0x7fb4edd57d4c "SIP/2.0\r\nFrom: \"9995551234\" <sip:9995551234@abc.def.com;rp=SandboxTestingRegistrations>;tag=F9EEDA6C-E203EB89\r\nTo: <sip:9995551234@abc.def.com>\r\nCall-ID: 2257e150-b49dfd8d-c8d924b2@1"..., len = 7}, statuscode = 32}}}, via1 = 0x7fb4efdd7ef0, via2 = 0x7fb4efdd8218, headers = 0x7fb4efdd7e50, last_header = 0x7fb4efdd87c0, parsed_flag = 18446744073709551615, h_via1 = 0x7fb4efdd80d8, h_via2 = 0x7fb4efdd8178, callid = 0x7fb4efdd19b0, to = 0x7fb4efdd1910, cseq = 0x7fb4efdd7c80, from = 0x7fb4efdd7e50, contact = 0x7fb4efdd84a0, maxforwards = 0x7fb4efdd8680, route = 0x0, record_route = 0x0, content_type = 0x0, content_length = 0x7fb4efdd87c0, authorization = 0x0, expires = 0x7fb4efdd8720, proxy_auth = 0x0, supported = 0x0, require = 0x0, proxy_require = 0x0, unsupported = 0x0, allow = 0x0, event = 0x0, accept = 0x0, accept_language = 0x7fb4efdd85e0, organization = 0x0, priority = 0x0, subject = 0x0, user_agent = 0x7fb4efdd8540, server = 0x0, content_disposition = 0x0, accept_disposition = 0x0, diversion = 0x0, rpid = 0x0, refer_to = 0x0, session_expires = 0x0, min_se = 0x0, sipifmatch = 0x0, subscription_state = 0x0, date = 0x0, identity = 0x0, identity_info = 0x0, pai = 0x0, ppi = 0x0, path = 0x0, privacy = 0x0, body = 0x0, eoh = 0x7fb4edd58036 "12963462\", realm=\"proxyApp\", nonce=\"proxyAppXgyrmj82aTujp3sdBW\", uri=\"sip:abc.def.com\", response=\"039891b3f5999b90a645d3e808c2147c\", algorithm=MD5, cnonce=\"l7/u88HtGJCOAmM\", qop=auth, nc="..., unparsed = 0x7fb4edd58036 "12963462\", realm=\"proxyApp\", nonce=\"proxyAppXgyrmj82aTujp3sdBW\", uri=\"sip:abc.def.com\", response=\"039891b3f5999b90a645d3e808c2147c\", algorithm=MD5, cnonce=\"l7/u88HtGJCOAmM\", qop=auth, nc="..., rcv = {src_ip = {af = 2, len = 4, u = {addrl = {140411255757578, 4294967291}, addr32 = {184916746, 32692, 4294967291, 0}, addr16 = {39690, 2821, 32692, 0, 65531, 65535, 0, 0}, addr = "\n\233\005\v\264\177\000\000\373\377\377\377\000\000\000"}}, dst_ip = {af = 2, len = 4, u = {addrl = {140414021193994, 5273972}, addr32 = { 2950353162, 32692, 5273972, 0}, addr16 = {53514, 45018, 32692, 0, 31092, 80, 0, 0}, addr = "\n\321Ú¯\264\177\000\000tyP\000\000\000\000"}}, src_port = 5060, dst_port = 5060, proto_reserved1 = 32692, proto_reserved2 = -266587208, src_su = {s = {sa_family = 2, sa_data = "\023\304\n\233\005\v\000\000\000\000\000\000\000"}, sin = {sin_family = 2, sin_port = 50195, sin_addr = {s_addr = 184916746}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 2, sin6_port = 50195, sin6_flowinfo = 184916746, sin6_addr = {__in6_u = { __u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, bind_address = 0x1, proto = 1 '\001'}, buf = 0x7fb4edd57d2a "REGISTER sip:abc.def.com SIP/2.0\r\nFrom: \"9995551234\" <sip:9995551234@abc.def.com;rp=SandboxTestingRegistrations>;tag=F9EEDA6C-E203EB89\r\nTo: <sip:9995551234@abc.def.com>\r\nCal"..., len = 782, new_uri = {s = 0x0, len = 0}, dst_uri = {s = 0x0, len = 0}, parsed_uri_ok = 1, parsed_uri = {user = {s = 0x0, len = 0}, passwd = {s = 0x0, len = 0}, host = { s = 0x7fb4edd57d37 "abc.def.com SIP/2.0\r\nFrom: \"9995551234\" <sip:9995551234@abc.def.com;rp=SandboxTestingRegistrations>;tag=F9EEDA6C-E203EB89\r\nTo: <sip:9995551234@abc.def.com>\r\nCall-ID: 2257e15"..., len = 20}, port = {s = 0x0, len = 0}, params = {s = 0x0, len = 0}, sip_params = {s = 0x0, len = 0}, headers = {s = 0x0, len = 0}, port_no = 0, proto = 0, type = SIP_URI_T, flags = 0, transport = {s = 0x0, len = 0}, ttl = { s = 0x0, len = 0}, user_param = {s = 0x0, len = 0}, maddr = {s = 0x0, len = 0}, method = {s = 0x0, len = 0}, lr = {s = 0x0, len = 0}, r2 = {s = 0x0, len = 0}, transport_val = {s = 0x0, len = 0}, ttl_val = {s = 0x0, len = 0}, user_param_val = {s = 0x0, len = 0}, maddr_val = {s = 0x0, len = 0}, method_val = {s = 0x0, len = 0}, lr_val = {s = 0x0, len = 0}, r2_val = {s = 0x0, len = 0}}, parsed_orig_ruri_ok = 0, parsed_orig_ruri = {user = {s = 0x0, len = 0}, passwd = {s = 0x0, len = 0}, host = {s = 0x0, len = 0}, port = {s = 0x0, len = 0}, params = {s = 0x0, len = 0}, sip_params = {s = 0x0, len = 0}, headers = {s = 0x0, len = 0}, port_no = 0, proto = 0, type = ERROR_URI_T, flags = 0, transport = {s = 0x0, len = 0}, ttl = {s = 0x0, len = 0}, user_param = {s = 0x0, len = 0}, maddr = {s = 0x0, len = 0}, method = {s = 0x0, len = 0}, lr = {s = 0x0, len = 0}, r2 = {s = 0x0, len = 0}, transport_val = {s = 0x0, len = 0}, ttl_val = {s = 0x0, len = 0}, user_param_val = {s = 0x0, len = 0}, maddr_val = {s = 0x0, len = 0}, method_val = { s = 0x0, len = 0}, lr_val = {s = 0x0, len = 0}, r2_val = {s = 0x0, len = 0}}, add_rm = 0x0, body_lumps = 0x0, reply_lump = 0x0, add_to_branch_s = '\000' <repeats 57 times>, add_to_branch_len = 0, hash_index = 0, msg_flags = 0, flags = 0, set_global_address = {s = 0x0, len = 0}, set_global_port = {s = 0x0, len = 0}, force_send_socket = 0x0, path_vec = {s = 0x0, len = 0}} (gdb) p *msg->rcv.bind_address Cannot access memory at address 0x1 (gdb) Thanks, Simpson ________________________________ From: "sr-users-request(a)lists.sip-router.org" <sr-users-request(a)lists.sip-router.org> To: sr-users(a)lists.sip-router.org Sent: Friday, February 17, 2012 7:34 AM Subject: sr-users Digest, Vol 81, Issue 52 Send sr-users mailing list submissions to sr-users(a)lists.sip-router.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.sip-router.org You can reach the person managing the list at sr-users-owner(a)lists.sip-router.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Re: ndb_redis module fails after a while (Daniel-Constantin Mierla) 2. Re: Kamailio Exits With: Segmentation Fault Error 4 in usrloc.so (Alexandr Dubovikov) ---------------------------------------------------------------------- Message: 1 Date: Fri, 17 Feb 2012 14:30:44 +0100 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [SR-Users] ndb_redis module fails after a while To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Cc: Javier Gallart <jgallartm(a)gmail.com> Message-ID: <4F3E5684.3090203(a)gmail.com> Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" Hello, right, very dummy paste position for resetting the redis context, use the attached patch or exchange lines 228 and 229. in redis_client.c Thanks, Daniel On 2/17/12 1:13 PM, Javier Gallart wrote: > Hi Daniel, Andrew > > I've just tested the patch and kamailio crashes when the redis server > is stopped. This is what I could get: > -From the logs: > Feb 17 06:33:32 r-gate-test ./kamailio[23812]: ALERT: <core> > [main.c:751]: child process 23819 exited by a signal 11 > Feb 17 06:33:32 r-gate-test ./kamailio[23812]: ALERT: <core> > [main.c:754]: core was generated > Feb 17 06:33:32 r-gate-test ./kamailio[23812]: INFO: <core> > [main.c:766]: INFO: terminating due to SIGCHLD > > -An the backtrace: > (...) > Core was generated by `./kamailio -f ../etc/kamailio/kamailio.cfg'. > Program terminated with signal 11, Segmentation fault. > #0 redisFree (c=0x0) at hiredis.c:817 > 817 if (c->fd > 0) > (gdb) bt > #0 redisFree (c=0x0) at hiredis.c:817 > #1 0x00007f726f035461 in redisc_reconnect_server > (rsrv=0x7f7271c93ac0) at redis_client.c:229 > #2 0x00007f726f037240 in redisc_exec (srv=<value optimized out>, > cmd=0x7fffe98c6090, argv1=<value optimized out>, argv2=<value > optimized out>, argv3=<value optimized out>, > res=<value optimized out>) at redis_client.c:298 > #3 0x00007f726f034f7d in w_redis_cmd3 (msg=0x7f7271d7b018, > ssrv=<value optimized out>, scmd=<value optimized out>, > sres=0x7f7271d74b58 "h\341\313qr\177") at ndb_redis_mod.c:156 > #4 0x0000000000417025 in do_action (h=0x7fffe98c6570, > a=0x7f7271cc5248, msg=<value optimized out>) at action.c:1134 > #5 0x000000000041e58b in run_actions (h=<value optimized out>, > a=<value optimized out>, msg=<value optimized out>) at action.c:1610 > #6 0x000000000041e8d4 in run_actions_safe (h=0x7fffe98c7610, > a=0x7f7271c93781, msg=0x7f7271c93780) at action.c:1662 > #7 0x00000000004b731d in rval_get_int (h=0x7fffe98c7610, msg=0x0, > i=0x7fffe98c6bd8, rv=0x3, cache=0x4) at rvalue.c:920 > #8 0x00000000004bb87c in rval_expr_eval_int (h=0x7fffe98c7610, > msg=0x7f7271d7b018, res=0x7fffe98c6bd8, rve=0x7f7271cc6768) at > rvalue.c:1914 > #9 0x0000000000417c7c in do_action (h=0x7fffe98c7610, > a=0x7f7271ccb9d0, msg=<value optimized out>) at action.c:1092 > #10 0x000000000041e58b in run_actions (h=<value optimized out>, > a=<value optimized out>, msg=<value optimized out>) at action.c:1610 > #11 0x0000000000417cd7 in do_action (h=0x7fffe98c7610, > a=0x7f7271ccbb10, msg=<value optimized out>) at action.c:1111 > #12 0x000000000041e58b in run_actions (h=<value optimized out>, > a=<value optimized out>, msg=<value optimized out>) at action.c:1610 > #13 0x000000000041795e in do_action (h=0x7fffe98c7610, a=<value > optimized out>, msg=<value optimized out>) at action.c:732 > #14 0x000000000041e58b in run_actions (h=<value optimized out>, > a=<value optimized out>, msg=<value optimized out>) at action.c:1610 > #15 0x000000000041e862 in run_top_route (a=0x7f7271c94888, > msg=0x7f7271d7b018, c=<value optimized out>) at action.c:1683 > #16 0x0000000000498f36 in receive_msg ( > buf=0x8bb100 "INVITE sip:34661574758@79.170.68.215:5060 > <http://sip:34661574758@79.170.68.215:5060> SIP/2.0\r\nVia: > SIP/2.0/UDP 79.170.68.214:5060;branch=z9hG4bK-9451-1-0\r\nFrom: > 34661574758 <sip:34661574758@79.170.68.214:5060 > <http://sip:34661574758@79.170.68.214:5060>>;tag=9451SIPpTag001\r\nTo: > sut <sip:"..., len=<value optimized out>, rcv_info=0x7fffe98c7910) at > receive.c:207 > #17 0x0000000000525987 in udp_rcv_loop () at udp_server.c:544 > #18 0x00000000004635f4 in main_loop () at main.c:1585 > #19 0x0000000000465e62 in main (argc=3, argv=0x7fffe98c7c08) at > main.c:2475 > > > Regards > > Javi > > On Fri, Feb 17, 2012 at 11:39 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > thanks for testing, indeed it was an extra declaration left over. > Can you try with the new patch attached? > > Cheers, > Daniel > > > On 2/17/12 11:10 AM, Andrew Pogrebennyk wrote: > > Hi Daniel, > > On 02/17/2012 10:47 AM, Daniel-Constantin Mierla wrote: > > I made a patch for server reconnect -- I had no access to > a computer > with redis lib installed for the moment, hopefully it > compiles. If you > can try and tell the result, it would be great, I can > commit then. > > I may be able to test this patch as well. Currently > compilations bails > out on attempt to redeclare redisc_reconnect_server function > parameter: > > CC (gcc) [M ndb_redis.so] ndb_redis_mod.o > CC (gcc) [M ndb_redis.so] redis_client.o > redis_client.c: In function 'redisc_reconnect_server': > redis_client.c:206:19: error: 'rsrv' redeclared as different > kind of symbol > redis_client.c:202:46: note: previous definition of 'rsrv' was > here > make[1]: *** [redis_client.o] Error 1 > make: *** [modules] Error 1 > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users > mailing list > sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierla -- http://www.asipto.com > http://linkedin.com/in/miconda -- http://twitter.com/miconda > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 1

avp_db_query - closing the connection
by graham 19 Feb '12

19 Feb '12

Hi all, I have been using more and more avps for per-call queries populated by custom mysql queries. I would like to know how the connections of mysql queries are being closed. I tried to locate a command to do such, but unable to locate. Do they close on their own, after a certain timeout period? I am running kamailio 1.5.5, notls. The proxies are getting busier and I want to keep all the resources in check ... Thanks, -graham

2 1

question on getting RTP stats
by Nelson Pereira 19 Feb '12

19 Feb '12

Is there a way to get asterisk to publish RTP stats in the bye message? I would like to parse theses to DB and get stats from them. If so, how? Thanks Np

1 0

Kamailio or VOS3000
by Abid Naseem 17 Feb '12

17 Feb '12

How kamailio is different from VOS3000 or Nextone or PORTASwitch when we are talking about wholesale routing? Sent from my iPhone

2 1

Re: [SR-Users] Need example of Kamailio.cfg for a port mirror setup
by Nelson Pereira 17 Feb '12

17 Feb '12

I have libpcap, it's in /usr/lib64 but it's libpcap0.9.8 >:/usr/local/src/captagent # find / |grep libpcap /usr/share/doc/packages/libpcap-devel /usr/share/doc/packages/libpcap-devel/pcap.xml /usr/share/doc/packages/libpcap-devel/pcap.html /usr/share/doc/packages/libpcap-devel/pcap.txt /usr/share/doc/packages/libpcap0 /usr/share/doc/packages/libpcap0/LICENSE /usr/share/doc/packages/libpcap0/TODO /usr/share/doc/packages/libpcap0/CREDITS /usr/share/doc/packages/libpcap0/README.linux /usr/share/doc/packages/libpcap0/README /usr/share/doc/packages/libpcap0/CHANGES /usr/lib64/libpcap.a /usr/lib64/libpcap.so.0.9.8 /usr/lib64/libpcap.so /usr/lib64/libpcap.so.0 On Fri, Feb 17, 2012 at 10:38 AM, Max Mühlbronner <mm(a)42com.com> wrote: > This is a multi-part message in MIME format. > --------------070003040807050803020008 > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > Content-Transfer-Encoding: 8bit > > Hi, > > you are missing libpcap0.8 which is a dependency for the captagent. > > > BR > > Max M. > > Am 17.02.2012 16:31, schrieb Nelson Pereira: > > ok, tried to do the make install, getting the bellow: > > aurast:/usr/local/src/captagent # /usr/local/bin/captagent > > /usr/local/bin/captagent: error while loading shared libraries: > > libpcap.so.0.8: cannot open shared object file: No such file or directory > > > > Do i need to compile this? > > > > Also just tried running the captagent: > > aurast:/usr/local/src/captagent # ./captagent > > ./captagent: error while loading shared libraries: libpcap.so.0.8: > > cannot open shared object file: No such file or directory > > > > What am I doing wrong? > > > > This is a SUSE SLES11 64bit server running asterisk 1.8.6 > > > > Thanks > > > > > > > > On Fri, Feb 17, 2012 at 10:24 AM, Nelson Pereira <kitkat0981(a)gmail.com > > <mailto:kitkat0981@gmail.com>> wrote: > > > > from the atttached kamailio.cfg file, what should be changed to > > get captagent receiver working in kamailio instead of port mirroring? > > > > Thanks > > > > > > On Fri, Feb 17, 2012 at 7:29 AM, Alexandr Dubovikov > > <voip(a)start4.info <mailto:voip@start4.info>> wrote: > > > > > > The link has been not changed, you know it already: > > > > http://code.google.com/p/homer/wiki/HOWTO > > > > Homer Capture Agent (for all other unsupported system) > > > > if your SIP system (PBX, proxy) doesn't have a built-in > > capture agent with HEP or IPIP encapsulation support, you can > > use Homer Capture Agent (captagent). This is an external > > application (daemon) that collect SIP messages thru PCAPlib > > and send them to your capture server. The captagent support > > IPv4 and IPv6. > > > > preventing your next question: the captagent your can find in > > the git. > > > > http://code.google.com/p/homer/source/browse/ > > > > > > Wbr, > > Alexandr > > > > > > > > 17.02.2012 13:15, Nelson Pereira wrote: > >> Can captagent be installed on a asterisk server? If so, is > >> there a howto? > >> > >> Sent from my iPhone4 on IOS4 > >> > >> On Feb 17, 2012, at 7:07 AM, Alexandr Dubovikov > >> <voip(a)start4.info <mailto:voip@start4.info>> wrote: > >> > >>> Hi, > >>> > >>> start two captagents instance, each on own interface and > >>> duplicate all traffic to your kamailio node. (HEP protocol) > >>> > >>> Wbr, > >>> Alexandr > >>> > >>> 16.02.2012 17:51, Nelson Pereira wrote: > >>>> I would like to know if someone can show me an example of > >>>> there setup configuration in kamailio.cfg for capturing > >>>> packets comming in from a mirror port. > >>>> The reason I ask is that if I do a TCPDUMP on the server, I > >>>> see all traffic to and from different sources on my Voice > >>>> network. > >>>> > >>>> But when I use Homer SIPCAPTURE which uses Kamailio in the > >>>> background, I only see traffic sent FROM my PBX but not > >>>> seing the traffic comming back to it. > >>>> > >>>> I mean I only see one side of the call, yet TCPDUMP > >>>> confirmed I see everything.... > >>>> > >>>> What am I missing ? > >>>> > >>>> Thanks > >>>> > >>>> NP > >>>> > >>>> > >>>> _______________________________________________ > >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users > mailing list > >>>> sr-users(a)lists.sip-router.org <mailto: > sr-users(a)lists.sip-router.org> > >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >>> > >>> _______________________________________________ > >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users > >>> mailing list > >>> sr-users(a)lists.sip-router.org > >>> <mailto:sr-users@lists.sip-router.org> > >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >> > >> > >> _______________________________________________ > >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users > mailing list > >> sr-users(a)lists.sip-router.org <mailto: > sr-users(a)lists.sip-router.org> > >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users > > mailing list > > sr-users(a)lists.sip-router.org > > <mailto:sr-users@lists.sip-router.org> > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > > > > > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > > sr-users(a)lists.sip-router.org > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > 42com Telecommunication GmbH > Straße der Pariser Kommune 12-16 / D-10243 Berlin > > E-mail: mm(a)42com.com > Homepage: www.42com.com > > Firmenangaben/Company information: > > Handelsregister/Commercial register: Amtsgericht Berlin HRB 99071 B > Umsatzsteuer-ID/VAT-ID: DE223812306, Geschäftsführer/CEO: Thomas Reinig, > Alexander Reinig > > > Diese E-Mail enthält Informationen von 42com Telecommunication GmbH. Diese > sind möglicherweise vertraulich und ausschließlich für den Adressaten > bestimmt. Sollten Sie diese elektronische Nachricht irrtümlicherweise > erhalten haben, so informieren Sie uns bitte unverzüglich telefonisch oder > per E-Mail. > > This message is intended only for the use of the individual or entity to > which it is addressed. If you have received this message in error, please > delete the message and notify us immediately. > > > --------------070003040807050803020008 > Content-Type: text/html; charset=ISO-8859-1 > Content-Transfer-Encoding: 7bit > > <html> > <head> > <meta content="text/html; charset=ISO-8859-1" > http-equiv="Content-Type"> > </head> > <body bgcolor="#FFFFFF" text="#000000"> > Hi,<br> > <br> > you are missing libpcap0.8 which is a dependency for the captagent.<br> > <br> > <br> > BR<br> > <br> > Max M.<br> > <br> > Am 17.02.2012 16:31, schrieb Nelson Pereira: > <blockquote > cite=" > mid:CAC6cfSgZgOJQ0-X5ANsHEQnmt+AvV96404dU_nr53dTNdbcUoA@mail.gmail.com" > type="cite">ok, tried to do the make install, getting the bellow: > <div> > <div>aurast:/usr/local/src/captagent # > /usr/local/bin/captagent</div> > <div>/usr/local/bin/captagent: error while loading shared > libraries: libpcap.so.0.8: cannot open shared object file: No > such file or directory</div> > <div><br> > </div> > <div>Do i need to compile this?</div> > <div><br> > </div> > <div>Also just tried running the captagent:</div> > <div> > <div>aurast:/usr/local/src/captagent # ./captagent</div> > <div>./captagent: error while loading shared libraries: > libpcap.so.0.8: cannot open shared object file: No such file > or directory</div> > <div><br> > </div> > </div> > <div>What am I doing wrong?</div> > <div><br> > </div> > <div>This is a SUSE SLES11 64bit server running asterisk 1.8.6</div> > <div><br> > </div> > <div>Thanks</div> > <div><br> > </div> > <div><br> > </div> > <br> > <div class="gmail_quote"> > On Fri, Feb 17, 2012 at 10:24 AM, Nelson Pereira <span > dir="ltr"><<a moz-do-not-send="true" > href="mailto:kitkat0981@gmail.com">kitkat0981(a)gmail.com > </a>></span> > wrote:<br> > <blockquote class="gmail_quote" style="margin:0 0 0 > .8ex;border-left:1px #ccc solid;padding-left:1ex"> > from the atttached kamailio.cfg file, what should be changed > to get captagent receiver working in kamailio instead of > port mirroring? > <div><br> > </div> > <div>Thanks</div> > <div class="HOEnZb"> > <div class="h5"> > <div><br> > <br> > <div class="gmail_quote"> > On Fri, Feb 17, 2012 at 7:29 AM, Alexandr Dubovikov > <span dir="ltr"><<a moz-do-not-send="true" > href="mailto:voip@start4.info" target="_blank"> > voip(a)start4.info</a>></span> > wrote:<br> > <blockquote class="gmail_quote" style="margin:0 0 0 > .8ex;border-left:1px #ccc solid;padding-left:1ex"> > <div bgcolor="#FFFFFF" text="#000000"> <br> > The link has been not changed, you know it > already:<br> > <br> > <a moz-do-not-send="true" > href="http://code.google.com/p/homer/wiki/HOWTO" > target="_blank"> > http://code.google.com/p/homer/wiki/HOWTO</a><br> > <br> > Homer Capture Agent (for all other unsupported > system)<br> > <br> > if your SIP system (PBX, proxy) doesn't have a > built-in capture agent with HEP or IPIP > encapsulation support, you can use Homer Capture > Agent (captagent). This is an external > application (daemon) that collect SIP messages > thru PCAPlib and send them to your capture > server. The captagent support IPv4 and IPv6.<br> > <br> > preventing your next question: the captagent > your can find in the git.<br> > <br> > <a moz-do-not-send="true" > href=" > http://code.google.com/p/homer/source/browse/" > target="_blank"> > http://code.google.com/p/homer/source/browse/</a><br> > <br> > <br> > Wbr,<br> > Alexandr > <div> > <div><br> > <br> > <br> > 17.02.2012 13:15, Nelson Pereira wrote: > <blockquote type="cite"> > <div>Can captagent be installed on a > asterisk server? If so, is there a > howto?<br> > <br> > Sent from my iPhone4 on IOS4</div> > <div><br> > On Feb 17, 2012, at 7:07 AM, Alexandr > Dubovikov <<a moz-do-not-send="true" > href="mailto:voip@start4.info" > target="_blank">voip(a)start4.info</a>> > > wrote:<br> > <br> > </div> > <blockquote type="cite"> > <div> Hi,<br> > <br> > start two captagents instance, each on > own interface and duplicate all > traffic to your kamailio node. (HEP > protocol)<br> > <br> > Wbr,<br> > Alexandr<br> > <br> > 16.02.2012 17:51, Nelson Pereira > wrote: > <blockquote type="cite">I would like > to know if someone can show me an > example of there setup configuration > in kamailio.cfg for capturing > packets comming in from a mirror > port. > <div>The reason I ask is that if I > do a TCPDUMP on the server, I see > all traffic to and from different > sources on my Voice network.</div> > <div><br> > </div> > <div>But when I use Homer SIPCAPTURE > which uses Kamailio in the > background, I only see traffic > sent FROM my PBX but not seing the > traffic comming back to > it. </div> > <div><br> > </div> > <div>I mean I only see one side of > the call, yet TCPDUMP confirmed I > see everything....</div> > <div><br> > </div> > <div>What am I missing ?</div> > <div><br> > </div> > <div>Thanks</div> > <div><br> > </div> > <div>NP</div> > <br> > <fieldset></fieldset> > <br> > > <pre>_______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > <a moz-do-not-send="true" href="mailto:sr-users@lists.sip-router.org" > target="_blank">sr-users(a)lists.sip-router.org</a> > <a moz-do-not-send="true" href=" > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" > target="_blank"> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a> > </pre> > </blockquote> > <br> > </div> > </blockquote> > <blockquote type="cite"> > > <div><span>_______________________________________________</span><br> > <span>SIP Express Router (SER) and > Kamailio (OpenSER) - sr-users > mailing list</span><br> > <span><a moz-do-not-send="true" > href="mailto: > sr-users(a)lists.sip-router.org" > target="_blank"> > sr-users(a)lists.sip-router.org</a></span><br> > <span><a moz-do-not-send="true" > href=" > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" > target="_blank"> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > </a></span><br> > </div> > </blockquote> > <br> > <fieldset></fieldset> > <br> > > <pre>_______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > <a moz-do-not-send="true" href="mailto:sr-users@lists.sip-router.org" > target="_blank">sr-users(a)lists.sip-router.org</a> > <a moz-do-not-send="true" href=" > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" > target="_blank"> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a> > </pre> > </blockquote> > <br> > </div> > </div> > </div> > <br> > _______________________________________________<br> > SIP Express Router (SER) and Kamailio (OpenSER) - > sr-users mailing list<br> > <a moz-do-not-send="true" > href="mailto:sr-users@lists.sip-router.org" > target="_blank">sr-users(a)lists.sip-router.org > </a><br> > <a moz-do-not-send="true" > href=" > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" > target="_blank"> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br> > <br> > </blockquote> > </div> > <br> > </div> > </div> > </div> > </blockquote> > </div> > <br> > </div> > <br> > <fieldset class="mimeAttachmentHeader"></fieldset> > <br> > <pre wrap="">_______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > <a class="moz-txt-link-abbreviated" href="mailto: > sr-users(a)lists.sip-router.org">sr-users(a)lists.sip-router.org</a> > <a class="moz-txt-link-freetext" href=" > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users"> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a> > </pre> > </blockquote> > <br> > <br> > <pre class="moz-signature" cols="72">-- > 42com Telecommunication GmbH > Straße der Pariser Kommune 12-16 / D-10243 Berlin > > E-mail: <a class="moz-txt-link-abbreviated" href="mailto:mm@42com.com"> > mm(a)42com.com</a> > Homepage: <a class="moz-txt-link-abbreviated" href="http://www.42com.com"> > www.42com.com</a> > > Firmenangaben/Company information: > > Handelsregister/Commercial register: Amtsgericht Berlin HRB 99071 B > Umsatzsteuer-ID/VAT-ID: DE223812306, Geschäftsführer/CEO: Thomas > Reinig, Alexander Reinig > > > Diese E-Mail enthält Informationen von 42com Telecommunication GmbH. > Diese sind möglicherweise vertraulich und ausschließlich > für den Adressaten bestimmt. Sollten Sie diese elektronische Nachricht > irrtümlicherweise erhalten haben, so informieren Sie uns bitte > unverzüglich telefonisch oder per E-Mail. > > This message is intended only for the use of the individual or entity to > which it is addressed. If you have received this message in error, please > delete the message and notify us immediately.</pre> > </body> > </html> > > --------------070003040807050803020008-- > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >

1 1

Need example of Kamailio.cfg for a port mirror setup
by Nelson Pereira 17 Feb '12

17 Feb '12

I would like to know if someone can show me an example of there setup configuration in kamailio.cfg for capturing packets comming in from a mirror port. The reason I ask is that if I do a TCPDUMP on the server, I see all traffic to and from different sources on my Voice network. But when I use Homer SIPCAPTURE which uses Kamailio in the background, I only see traffic sent FROM my PBX but not seing the traffic comming back to it. I mean I only see one side of the call, yet TCPDUMP confirmed I see everything.... What am I missing ? Thanks NP

2 5

Re: [SR-Users] how to match exact string value in mtree (Uri Shacked)
by Javier Gallart 17 Feb '12

17 Feb '12

Hello Uri I had similar needs and I found the ndb_redis module more suited for that type of task. Instead of a tree you have a hash like this: tname tprefix tvalue. If you do a hget nts $avp(DID) and you get a not null value you have found your exact match. It works very well for me and the time it takes for that "query" is barely noticeable. Of course redis does not address items like persistence, etc the same way a rdbms does. Hope it helps. Regards Javi On Thu, Feb 16, 2012 at 12:00 PM, <sr-users-request(a)lists.sip-router.org>wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. how to match exact string value in mtree (Uri Shacked) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 16 Feb 2012 12:54:50 +0200 > From: Uri Shacked <ushacked(a)gmail.com> > Subject: [SR-Users] how to match exact string value in mtree > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <CAMMbDhTFNXAE-K88=AeMjO7AnA_QJV3Ajj3AH-AxemXN3ze6HQ(a)mail.gmail.com > > > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > > I am using Mtree to match prefix numbers, some of them starts with 0 or > characters like D for example. > so, the mtree param is like this: > > modparam("mtree", "db_url", CFGDB) > modparam("mtree", "mtree", "name=nts;dbtable=service_numbers_view;type=0;") > modpmodparam("mtree", "char_list", "0123456789*+#YMDabcdefgh") > modparam("mtree", "pv_value", "$avp(mtval)") > modparam("mtree", "pv_values", "$avp(mtvals)") > > The thing is, that i tried all: > > if(!mt_match("nts", "$avp(DID)","1")) > if(!mt_match("nts", "$avp(DID)","2")) > if(!mt_match("nts", "$avp(DID)","0")) > > and lets say i have the both prefix in the nts mtree: > 09555 > 09555333 > > And the prefix i search for is $avp(DID)=09555444 > > I allwasy get the 09555 because it is the longest match. > I need exact match..... > > how do i do that? >

3 7

ndb_redis module fails after a while
by Javier Gallart 17 Feb '12

17 Feb '12

Hi all I have started making some tests with the ndb_redis module. So far we have not stressed the module (no more than 5 HGET commands/second at maximum). It works well, but with at some point it starts failing. The failures are easily found because the logs always show this: INFO: <core> [main.c:811]: INFO: signal 13 received After that the redis value is always null. If I restart kamailio it starts working again. I've run kamailio with debug=4 but I haven't seen more useful information. On the redis side, I could find nothing in the logs either, the number of clientes connected is alway much less than the configured maximum, Any idea? On the other hand, if I restart redis we need to restart kamailio to restore the connections. Is the reconnection to redis on the roadmap? Thanks in advance Regards Javi

3 12

Kamailio Exits With: Segmentation Fault Error 4 in usrloc.so
by Simpson Chua 17 Feb '12

17 Feb '12

Hi, Kamailio is exiting with a segmentation fault when trying to do a save("location"). Here is some information I gathered. Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25721]: DEBUG: <core> [parser/msg_parser.c:103]: found end of header Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25721]: ERROR: <core> [db.c:435]: invalid parameter value Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25721]: ERROR: usrloc [udomain.c:564]: failed to use table location Feb 16 15:30:19 ubuntu kernel: [171540.056663] kamailio[25721]: segfault at 89 ip 00007fa1f9a41452 sp 00007fff971a3af0 error 4 in usrloc.so[7fa1f9a37000+1f000] Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25708]: ALERT: <core> [main.c:751]: child process 25720 exited by a signal 11 Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25708]: ALERT: <core> [main.c:754]: core was generated Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25708]: INFO: <core> [main.c:766]: INFO: terminating due to SIGCHLD Core was generated by `kamailio -w /home/core'. Program terminated with signal 11, Segmentation fault. #0 db_insert_ucontact (_c=0x7f5f8f737eb0) at ucontact.c:474 474 vals[11].val.str_val = _c->sock->sock_str; (gdb) bt #0 db_insert_ucontact (_c=0x7f5f8f737eb0) at ucontact.c:474 #1 0x00007f5f916d9649 in insert_ucontact (_r=<optimized out>, _contact=<optimized out>, _ci=<optimized out>, _c=0x7fff52bbeb88) at urecord.c:473 #2 0x00007f5f914b785f in insert_contacts (_m=0x7f5f93d7f458, _d=0x7f5f8f72f680, _a=0x7fff52bbec00) at save.c:428 #3 0x00007f5f914b8d10 in add_contacts (_mode=1, _a=0x7fff52bbec00, _d=0x7f5f8f72f680, _m=0x7f5f93d7f458) at save.c:737 #4 save (_m=0x7f5f93d7f458, _d=0x7f5f8f72f680, _cflags=4) at save.c:790 #5 0x000000000041ba87 in do_action (h=0x7fff52bbf5d0, a=0x7f5f93d7f238, msg=0x7f5f93d7f458) at action.c:1128 #6 0x000000000041abe2 in run_actions (h=0x7fff52bbf5d0, a=0x7f5f93d7f238, msg=0x7f5f93d7f458) at action.c:1610 #7 0x000000000041c05b in do_action (h=0x7fff52bbf5d0, a=0x7f5f93d7fe08, msg=0x7f5f93d7f458) at action.c:1107 #8 0x000000000041abe2 in run_actions (h=0x7fff52bbf5d0, a=0x7f5f93d7fe08, msg=0x7f5f93d7f458) at action.c:1610 #9 0x0000000000422fa2 in run_top_route (a=0x7f5f93d7fe08, msg=0x7f5f93d7f458, c=<optimized out>) at action.c:1683 #10 0x00000000004b061e in receive_msg (buf=<optimized out>, len=<optimized out>, rcv_info=<optimized out>) at receive.c:207 #11 0x00007f5f91b038b0 in raw_capture_rcv_loop (rsock=5, port1=5060, port2=8933, ipip=0) at sipcapture.c:1518 #12 0x00007f5f91b03e1f in init_rawsock_children () at sipcapture.c:572 #13 0x00007f5f91b04033 in child_init (rank=<optimized out>) at sipcapture.c:542 #14 0x0000000000507a0f in init_mod_child (m=0x7f5f93d7b2b0, rank=0) at sr_module.c:886 #15 0x0000000000507974 in init_mod_child (m=0x7f5f93d7b668, rank=0) at sr_module.c:883 #16 0x0000000000507974 in init_mod_child (m=0x7f5f93d7c178, rank=0) at sr_module.c:883 #17 0x0000000000507974 in init_mod_child (m=0x7f5f93d7c4b0, rank=0) at sr_module.c:883 #18 0x0000000000476149 in main_loop () at main.c:1661 #19 0x000000000041a940 in main (argc=<optimized out>, argv=0x7fff52bbfd68) at main.c:2475 Any idea why this is happening? Is there something in the REGISTER message that is causing this? Thanks, Simpson

2 2

Re: [SR-Users] how to match exact string value in mtree (Uri Shacked)
by Daniel-Constantin Mierla 17 Feb '12

17 Feb '12

Hello, On 2/17/12 1:04 PM, Javier Gallart wrote: > Hi Daniel, > > no, in order to find the best match we do a manual trick using > the s.prefixes transformation (I think you suggested that). could be, at least is the way I do prefix matching with mysql. Postgres, on the other hand, afaik, has an index plugin to match the longest prefix, might not be in default distro, but can be found on the net for sure. > We start with the full $rU and stop when we find a non-null value. > This kamailio is serving many thousands of calls per hour, and the > response time is pretty much the same as when we used the mtree > module. For us , using redis has the advantage of being able to load > the tree data faster than when we used postgres without the need to > perform a mtree reload. Loading huge amount of records from mysql takes time indeed. Maybe prefix matching will be added in redis in the future, or scripted server side in lua, as I understood it has such feature. Another option would be to add a custom command in kamailio like redis_cmd_pmatch() to do internally these prefix-based commands until a match, could simplify the config if it becomes something very popular. Also, I forgot about the other solution in kamailio - pdb module written by 1&1 for number portability: * http://kamailio.org/docs/modules/stable/modules/pdb.html Just mentioning it here for let it be known, if redis is working fine for you, makes no sense to look for other solutions. Cheers, Daniel > > Regards > > Javi > > > On Fri, Feb 17, 2012 at 10:25 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello Javier, > > one question about redis, can it do longest prefix matching or all > prefixes match? > > Cheers, > Daniel > > > On 2/16/12 2:51 PM, Javier Gallart wrote: >> Hello Uri >> >> I had similar needs and I found the ndb_redis module more suited >> for that type of task. Instead of a tree you have a hash like >> this: tname tprefix tvalue. If you do a hget nts $avp(DID) and >> you get a not null value you have found your exact match. It >> works very well for me and the time it takes for that "query" is >> barely noticeable. Of course redis does not address items like >> persistence, etc the same way a rdbms does. >> Hope it helps. >> >> Regards >> >> Javi >> >> On Thu, Feb 16, 2012 at 12:00 PM, >> <sr-users-request(a)lists.sip-router.org >> <mailto:sr-users-request@lists.sip-router.org>> wrote: >> >> Send sr-users mailing list submissions to >> sr-users(a)lists.sip-router.org >> <mailto:sr-users@lists.sip-router.org> >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> or, via email, send a message with subject or body 'help' to >> sr-users-request(a)lists.sip-router.org >> <mailto:sr-users-request@lists.sip-router.org> >> >> You can reach the person managing the list at >> sr-users-owner(a)lists.sip-router.org >> <mailto:sr-users-owner@lists.sip-router.org> >> >> When replying, please edit your Subject line so it is more >> specific >> than "Re: Contents of sr-users digest..." >> >> >> Today's Topics: >> >> 1. how to match exact string value in mtree (Uri Shacked) >> >> >> ---------------------------------------------------------------------- >> >> Message: 1 >> Date: Thu, 16 Feb 2012 12:54:50 +0200 >> From: Uri Shacked <ushacked(a)gmail.com >> <mailto:ushacked@gmail.com>> >> Subject: [SR-Users] how to match exact string value in mtree >> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router >> (SER) - >> Users Mailing List" <sr-users(a)lists.sip-router.org >> <mailto:sr-users@lists.sip-router.org>> >> Message-ID: >> <CAMMbDhTFNXAE-K88=AeMjO7AnA_QJV3Ajj3AH-AxemXN3ze6HQ(a)mail.gmail.com >> <mailto:AeMjO7AnA_QJV3Ajj3AH-AxemXN3ze6HQ@mail.gmail.com>> >> Content-Type: text/plain; charset="iso-8859-1" >> >> Hi, >> >> I am using Mtree to match prefix numbers, some of them starts >> with 0 or >> characters like D for example. >> so, the mtree param is like this: >> >> modparam("mtree", "db_url", CFGDB) >> modparam("mtree", "mtree", >> "name=nts;dbtable=service_numbers_view;type=0;") >> modpmodparam("mtree", "char_list", "0123456789*+#YMDabcdefgh") >> modparam("mtree", "pv_value", "$avp(mtval)") >> modparam("mtree", "pv_values", "$avp(mtvals)") >> >> The thing is, that i tried all: >> >> if(!mt_match("nts", "$avp(DID)","1")) >> if(!mt_match("nts", "$avp(DID)","2")) >> if(!mt_match("nts", "$avp(DID)","0")) >> >> and lets say i have the both prefix in the nts mtree: >> 09555 >> 09555333 >> >> And the prefix i search for is $avp(DID)=09555444 >> >> I allwasy get the 09555 because it is the longest match. >> I need exact match..... >> >> how do i do that? >>

2 1

get myself ip address on start
by Uri Shacked 17 Feb '12

17 Feb '12

Hi, i would like to get the kamailio server ip address on start up and save it in an $shv(). i tried to use "myself" but did manage to get the ip from it. any ideas? Uri

1 0

Re: [SR-Users] Kamailio Exits With: Segmentation Fault Error 4 in usrloc.so
by Daniel-Constantin Mierla 17 Feb '12

17 Feb '12

Hello, On 2/17/12 1:11 PM, Alexandr Dubovikov wrote: > Hi Daniel, > > he use raw_socket in the sipcapture module and save all REGISTER > messages to location. (something like fraud controling for > registration). I think in this case he have to ignore socket listener. not sure I understood -- it is on the server that saves to homer's database? It gets the traffic over a raw socket and then executes the config file as usual with functions from other modules? Cheers, Daniel > > Wbr, > Alexandr > > > > > 17.02.2012 09:19, Daniel-Constantin Mierla wrote: >> Hello, >> >> looks like an invalid listen socket structure, quite strange... what >> version of kamailio do you have? >> >> Also, in gdb, frame 0, can you send the output of: >> >> p *_c >> p *_c->sock >> >> You can replace sensitive data (like IP), if you wish. >> >> Cheers, >> Daniel >> >> On 2/16/12 10:52 PM, Simpson Chua wrote: >>> Hi, >>> >>> Kamailio is exiting with a segmentation fault when trying to do a >>> save("location"). Here is some information I gathered. >>> >>> Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25721]: DEBUG: >>> <core> [parser/msg_parser.c:103]: found end of header >>> Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25721]: ERROR: >>> <core> [db.c:435]: invalid parameter value >>> Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25721]: ERROR: >>> usrloc [udomain.c:564]: failed to use table location >>> Feb 16 15:30:19 ubuntu kernel: [171540.056663] kamailio[25721]: >>> segfault at 89 ip 00007fa1f9a41452 sp 00007fff971a3af0 error 4 in >>> usrloc.so[7fa1f9a37000+1f000] >>> Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25708]: ALERT: >>> <core> [main.c:751]: child process 25720 exited by a signal 11 >>> Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25708]: ALERT: >>> <core> [main.c:754]: core was generated >>> Feb 16 15:30:19 ubuntu /usr/local/sbin/kamailio[25708]: INFO: <core> >>> [main.c:766]: INFO: terminating due to SIGCHLD >>> >>> Core was generated by `kamailio -w /home/core'. >>> Program terminated with signal 11, Segmentation fault. >>> #0 db_insert_ucontact (_c=0x7f5f8f737eb0) at ucontact.c:474 >>> 474 vals[11].val.str_val = _c->sock->sock_str; >>> (gdb) bt >>> #0 db_insert_ucontact (_c=0x7f5f8f737eb0) at ucontact.c:474 >>> #1 0x00007f5f916d9649 in insert_ucontact (_r=<optimized out>, >>> _contact=<optimized out>, _ci=<optimized out>, _c=0x7fff52bbeb88) at >>> urecord.c:473 >>> #2 0x00007f5f914b785f in insert_contacts (_m=0x7f5f93d7f458, >>> _d=0x7f5f8f72f680, _a=0x7fff52bbec00) at save.c:428 >>> #3 0x00007f5f914b8d10 in add_contacts (_mode=1, _a=0x7fff52bbec00, >>> _d=0x7f5f8f72f680, _m=0x7f5f93d7f458) at save.c:737 >>> #4 save (_m=0x7f5f93d7f458, _d=0x7f5f8f72f680, _cflags=4) at save.c:790 >>> #5 0x000000000041ba87 in do_action (h=0x7fff52bbf5d0, >>> a=0x7f5f93d7f238, msg=0x7f5f93d7f458) at action.c:1128 >>> #6 0x000000000041abe2 in run_actions (h=0x7fff52bbf5d0, >>> a=0x7f5f93d7f238, msg=0x7f5f93d7f458) at action.c:1610 >>> #7 0x000000000041c05b in do_action (h=0x7fff52bbf5d0, >>> a=0x7f5f93d7fe08, msg=0x7f5f93d7f458) at action.c:1107 >>> #8 0x000000000041abe2 in run_actions (h=0x7fff52bbf5d0, >>> a=0x7f5f93d7fe08, msg=0x7f5f93d7f458) at action.c:1610 >>> #9 0x0000000000422fa2 in run_top_route (a=0x7f5f93d7fe08, >>> msg=0x7f5f93d7f458, c=<optimized out>) at action.c:1683 >>> #10 0x00000000004b061e in receive_msg (buf=<optimized out>, >>> len=<optimized out>, rcv_info=<optimized out>) at receive.c:207 >>> #11 0x00007f5f91b038b0 in raw_capture_rcv_loop (rsock=5, port1=5060, >>> port2=8933, ipip=0) at sipcapture.c:1518 >>> #12 0x00007f5f91b03e1f in init_rawsock_children () at sipcapture.c:572 >>> #13 0x00007f5f91b04033 in child_init (rank=<optimized out>) at >>> sipcapture.c:542 >>> #14 0x0000000000507a0f in init_mod_child (m=0x7f5f93d7b2b0, rank=0) >>> at sr_module.c:886 >>> #15 0x0000000000507974 in init_mod_child (m=0x7f5f93d7b668, rank=0) >>> at sr_module.c:883 >>> #16 0x0000000000507974 in init_mod_child (m=0x7f5f93d7c178, rank=0) >>> at sr_module.c:883 >>> #17 0x0000000000507974 in init_mod_child (m=0x7f5f93d7c4b0, rank=0) >>> at sr_module.c:883 >>> #18 0x0000000000476149 in main_loop () at main.c:1661 >>> #19 0x000000000041a940 in main (argc=<optimized out>, >>> argv=0x7fff52bbfd68) at main.c:2475 >>> >>> Any idea why this is happening? Is there something in the REGISTER >>> message that is causing this? >>> >>> Thanks, >>> Simpson >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierla --http://www.asipto.com >> http://linkedin.com/in/miconda -- http://twitter.com/miconda >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 3

Question on ("sipcapture", "raw_interface", "")
by Nelson Pereira 17 Feb '12

17 Feb '12

Is it possible to monitor 2 interfaces? I have 2 Asterisk servers and they are on 2 different switches so I need to capture ETH0 and ETH3 which is where the mirror ports are connected to from each switch. Thanks

3 3

CAcert root in Lumicall
by Daniel Pocock 17 Feb '12

17 Feb '12

I've been contemplating Daniel's earlier question about using the CAcert certificates with Lumicall sip5060.net should already accept mutual authentication from other Kamailio instances running with a CAcert certificate However, the Lumicall dialer itself will only connect to servers that are using a cert signed by a root CA trusted within Android. This applies to both the SIP and STUN over TLS support. It may be possible for me to embed the CACert root in the Lumicall package and create a custom trust manager - allowing Lumicall to work with any server using CAcert TLS certificates. Does anyone have any strong feelings about not doing something like that?

3 9

how to match exact string value in mtree
by Uri Shacked 17 Feb '12

17 Feb '12

Hi, I am using Mtree to match prefix numbers, some of them starts with 0 or characters like D for example. so, the mtree param is like this: modparam("mtree", "db_url", CFGDB) modparam("mtree", "mtree", "name=nts;dbtable=service_numbers_view;type=0;") modpmodparam("mtree", "char_list", "0123456789*+#YMDabcdefgh") modparam("mtree", "pv_value", "$avp(mtval)") modparam("mtree", "pv_values", "$avp(mtvals)") The thing is, that i tried all: if(!mt_match("nts", "$avp(DID)","1")) if(!mt_match("nts", "$avp(DID)","2")) if(!mt_match("nts", "$avp(DID)","0")) and lets say i have the both prefix in the nts mtree: 09555 09555333 And the prefix i search for is $avp(DID)=09555444 I allwasy get the 09555 because it is the longest match. I need exact match..... how do i do that?

2 1

image missing in xmlrpc module doc
by Andrew Pogrebennyk 17 Feb '12

17 Feb '12

Hi, Was reading through the xmlrpc docs, could anybody with an access check why Figure "RPC Example" is missing from http://kamailio.org/docs/modules/devel/modules/xmlrpc.html#fig.rpc_example as well as 3.1 and 3.2 module docs? Thanks, Andrew

1 0

Kamailio in VUC session, next Friday, Feb 17
by Daniel-Constantin Mierla 17 Feb '12

17 Feb '12

Hello, there will be a VoIP Users Conference (VUC) session dedicated to Kamailio SIP Server next week, Friday, February 17, starting with 17:00GMT. VUC is an online weekly audio conferencing event, you can connect via SIP and IRC to interact with the other participants. More details (including the ways to connect) are available at: * http://www.kamailio.org/w/2012/02/kamailio-at-voipuserconference-fri-feb-17… Join the event to learn about latest developments and future plans for the project, ask questions about your interested topics. Spreading the word about the event is also appreciated. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 2

mi_xmlrpc not compiling on Solaris
by Nathaniel L Keeling 17 Feb '12

17 Feb '12

Hello, I am trying to compile mi_xmlrpc on Solaris 10 64bit OS and I am getting this error: CC (gcc) [M mi_xmlrpc.so] abyss_xmlrpc_server.o abyss_xmlrpc_server.c:948:1: error: conflicting types for 'shutdownAbyss' abyss_xmlrpc_server.c:945:34: note: previous declaration of 'shutdownAbyss' was here abyss_xmlrpc_server.c: In function 'normalLevelAbyssRun': abyss_xmlrpc_server.c:1005:42: warning: passing argument 2 of 'xmlrpc_registry_set_shutdown' from incompatible pointer type [enabled by default] /opt/csw/include/xmlrpc-c/server.h:114:1: note: expected 'void (* const)(struct xmlrpc_env * const, void * const, const char * const, void * const)' but argument is of type 'void (*)(struct xmlrpc_env * const, void * const, const char * const)' gmake[1]: *** [abyss_xmlrpc_server.o] Error 1 gmake: *** [modules_k] Error 1 Thanks Nathaniel L Keeling

3 2

How to do request URI rewrites using database tables?
by Ben WIlliams 16 Feb '12

16 Feb '12

Hi, can someone please recommend the most appropriate modules to rewrite R-URIs based on a database lookup table? I've read the documentation for lcr and carrierroute but not sure if they can do this. In most cases it will be a simple R-URI rewrite but I also need to rewrite based on From user. ie R-URI match From match new R-URI =========== ========== ========= *97(a)example.com 6[0-9](a)example.com *97(a)10.0.0.1 *97(a)example.com 7[0-9](a)example.com *97(a)10.0.0.2 6[0-9](a)example.com match any rewrite domain to 10.0.0.1 abc(a)example.com match any def(a)example.com Thanks Ben

1 1

Problems with TCP
by Mino Haluz 16 Feb '12

16 Feb '12

Hi, our customers are using mostly UDP but some of them want to use TCP. The problem is, I get various TCP errors in kamailio log and I do not understand what they mean 1) ERROR: <core> [tcp_main.c:4130]: connect <customerIP:port> failed This means kamailio is trying to connect with TCP to the customer ?? 2) WARNING: tcp_do_connect <customerIP:port> : could not find corresponding listening socket for IPaddress , using default... ( where IPaddress is kamailio physical IP address) Kamailio has one virtual IP address (heartbeat IP address) and one physical. It listens on the virtual IP address but I do not understand why it is trying to use the physical IP address ? I was investigating this thing because the BYE message is not being relayed if the TCP is used (everytime the 2. error message is printed), so that's why I have to understand what it really means... And the last question, 3) If kamailio is relaying the traffic to the PSTN gateway, and the customer is using TCP, kamailio uses TCP as well for connecting to the PSTN gw. Is this behaviour normal ? I would expect that customer -- TCP --> kamailio --- UDP --> GW, and not customer -- TCP --> kamailio --- TCP --> GW, Thanks, Mino

3 3

Kamailio Asterisk Followup
by Greg Mannie 16 Feb '12

16 Feb '12

Hello Sorry for another newbie question, but eventually with your "greatly appreciated" help I will get proficient in this application. After reading much RFC reading and docs for Kamailio I see where the benefits of using the sip proxy for registering devices while using asterisk for voicemail or ivr etc. has great benefit. I am not finding much on my end user interaction. If I use realtime integration and have multi domain use on kamailio, how do I allow the end user to configure their own IVR? Is it possible to use modules like the call flow control from freepbx and allow users to configure this themselves? Regards, Greg

4 4

Current Vacancy
by apache＠iptel.org 16 Feb '12

16 Feb '12

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Job Category: Customer Support/Client Care Reference Code: HJR/EU/551-188 Job Status: Full Time, Temporary/Contract/Project Great company represented in 32 countries worldwide is seeking a Customer Service Representative. This is a temporary position with the great opportunity of going permanent. The responsibilities are as follows: assist to work with existing clients, arrange and assure preparation, final inspection, timely implementation of orders as well as all related documentation and paperwork. [...] Content analysis details: (20.7 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 1.3 RCVD_IN_BL_SPAMCOP_NET RBL: Received via a relay in bl.spamcop.net [Blocked - see <http://www.spamcop.net/bl.shtml?89.252.8.22>] 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL [89.252.8.22 listed in zen.spamhaus.org] 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL 0.8 RCVD_IN_SORBS_WEB RBL: SORBS: sender is an abusable web server [89.252.8.22 listed in dnsbl.sorbs.net] 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [89.252.8.22 listed in bb.barracudacentral.org] 1.7 URIBL_DBL_SPAM Contains an URL listed in the DBL blocklist [URIs: eurojoboard.com] 1.7 URIBL_BLACK Contains an URL listed in the URIBL blacklist [URIs: eurojoboard.com] 3.6 HELO_DYNAMIC_IPADDR2 Relay HELO'd using suspicious hostname (IP addr 2) 3.5 HELO_DYNAMIC_SPLIT_IP Relay HELO'd using suspicious hostname (Split IP) 0.0 TVD_RCVD_IP TVD_RCVD_IP 1.3 DATE_IN_PAST_24_48 Date: is 24 to 48 hours before Received: date 0.8 BAYES_50 BODY: Bayes spam probability is 40 to 60% [score: 0.5553] 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS

1 0

strage result in math with $vars in 3.2.2
by Uri Shacked 16 Feb '12

16 Feb '12

Hi, i am using some math to send to my NOC the memory usage of kamailio. this is what i do: $var(xxx)=$stat(real_used_size)*100/$stat(total_size) before the 3.2.2 upgrade, i usualy got regular results - 3,5,20 etc.... now, i get results like -8, -32 and so... any ideas why? here are the $stat numbers that gets me the -8 result: $stat(real_used_size)=31899516 $stat(total_size)=134217728

2 3

How to reload dispatcher destinations
by Mino Haluz 16 Feb '12

16 Feb '12

Hi, is there any way how to reload dispatcher destinations (located in db) without restarting the kamailio when algorithm 10 (call load balancing) is used ? I read in documentation, that this command is disabled when this algorithm is used. Thanks, Mino

2 2

some questions
by 393319152 16 Feb '12

16 Feb '12

dear friends: I'm using the kamailio as my sip server and it has a public ip. There are two nac: A and B, A has a privite ip, B has a privite ip that is seted with VPN . And now, A can register successfully, but B can't . I want A to connect B and trans information. I don't know why , I hope you can help me.

2 1

Kamailio v3.2.2 Released
by Daniel-Constantin Mierla 15 Feb '12

15 Feb '12

Hello, Kamailio SIP Server v3.2.2 stable release is out. This is a maintenance release of the latest stable branch, 3.2, that includes fixes since release of v3.2.1. There is no change to database schema or configuration language structure that you have to do on installations of v3.2.0 or v3.2.1. Deployments running previous v3.2.x versions are strongly recommended to be upgraded to v3.2.2. For more details about version 3.2.2 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2012/01/kamailio-v3-2-2-released/ RPM packages are currently on the build service, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

3 3

Reuse TLS connections
by Bruno Bresciani 15 Feb '12

15 Feb '12

Hi all, There is the possibility to kamailio reuse the TLS connection created by other SIP server? When kamailio use t_relay function to send a SIP request message to other server, kamailio verify if already exist some connection established with the destiny and use it even if this connection was created by the other SIP server. In short, I want to keep only one connection between kamailio and SIP server. Sometimes kamailio will be a client and others a server. Best Regards

2 3

struct cmd_export_t
by Bruno Bresciani 15 Feb '12

15 Feb '12

Hi all, I'm writing a module to kamailio and can't find detail information about the parameter *"fixup_function fixup"* present at struct *cmd_export_t*. I can't understand which functionality this parameter... Someone have any information or documentation about this subject. Best Regards

2 2

AVP obtain with regular expression
by Anto 14 Feb '12

14 Feb '12

Hello I'm using AVP, but I can not finish assimilating the concept. I try to input messages, according to their DID. Once it enters an INVITE message treatment with alias_db for its user, but I want some AVP for each DID. I can think of three cases. The first I have no very clear if possible and if it were, I think is not optimal. 1) Given the configuration of the module: modparam("avpops","avp_url","mysql://user:passwd@host/database") Had thought if I can use regular expressions to get the "prs" in the database (DID_IN, DID_DEST, etc). USR_PREFERENCES table: | id | uuid | username | domain | attribute | type | value | modified | | 1 | XXX | user1 | dmn.tld | 571XXX_IN | 1 | XXX | | | 2 | XXX | user2 | dmn.tld | 571XXX_DEST | 1 | XXX | | 2) Maybe better to use the module configuration schemes: modparam("avpops","db_scheme","scheme1:uuid_col=uid;value_col=job;value_type=string;table=emp") modparam("avpops","db_scheme","scheme2:username_col=user;domain_col=domain;value_col=email;table=users") 3) Or use avp_db_query() function. Can anyone guide me? I'm pretty lost :-S. The first case, I know if possible and possibly more difficult to maintain. Thanks very much. Best regards Anto

1 0

problems since upgrading mysql and php
by Nelson Pereira 14 Feb '12

14 Feb '12

Getting the bellow error after upgrading php and mysql as requested: Access forbidden! You don't have permission to access the requested directory. There is either no index document or the directory is read-protected. If you think this is a server error, please contact the webmaster <mailto:root@cbnhmr01plc> . Error 403 172.20.25.149 <http://172.20.25.149/> Tue Feb 14 13:37:07 2012 Apache/2.2.10 (Linux/SUSE) Regards, Nelson Pereira Sr. Network Administrator, Information Systems Canadian Bank Note Company 18 Auriga Drive Ottawa, Ontario K2E 7T9 Phone: (613) 225-6566 Ext 2347 Cell: (613) 410-7608

2 1

Record-route and high avaibility
by Anto 14 Feb '12

14 Feb '12

Hello I have a proxy that makes the function of balancer and I have a doubt. This balancing between two proxies that do accounting, if any falls during the course of a call, you lose accounting (record-route). Is that correct? Is it advisable to have at each proxy server as a slave and balance the ip? Thanks Regards

2 2

creating DB in MySQL for Homer Capture server gives errors
by Nelson Pereira 14 Feb '12

14 Feb '12

Hi all, Trying to create the Homer SIP capture (Kamailio) DB and getting an error: Error SQL query: CREATE TABLE `sip_capture` ( `id` bigint(20) unsigned NOT NULL AUTO_INCREMENT, `date` timestamp NOT NULL DEFAULT '0000-00-00 00:00:00', `micro_ts` bigint(18) NOT NULL DEFAULT '0', `method` varchar(50) NOT NULL DEFAULT '', `reply_reason` varchar(100) NOT NULL, `ruri` varchar(200) NOT NULL DEFAULT '', `ruri_user` varchar(100) NOT NULL DEFAULT '', `from_user` varchar(100) NOT NULL DEFAULT '', `from_tag` varchar(64) NOT NULL DEFAULT '', `to_user` varchar(100) NOT NULL DEFAULT '', `to_tag` varchar(64) NOT NULL, `pid_user` varchar(100) NOT NULL DEFAULT '', `contact_user` varchar(120) NOT NULL, `auth_user` varchar(120) NOT NULL, `callid` varchar(100) NOT NULL DEFAULT '', `callid_aleg` varchar(100) NOT NULL DEFAULT '', `via_1` varchar(256) NOT NULL, `via_1_branch` varchar(80) NOT NULL, `cseq` varchar(25) NOT NULL, `diversion` varchar(256) NOT NULL, `reason` varchar(200) NOT NULL, `content_type` varchar(256) NOT NULL, `authorization` varchar(256) N[...] MySQL said: <http://172.20.25.149/phpMyAdmin/url.php?url=http%3A%2F%2Fdev.mysql.com%2Fdo c%2Frefman%2F5.0%2Fen%2Ferror-messages-server.html&token=1d6b9d77bce4c6f1648 83e3c6dd6bc09> Description: Documentation #1064 - You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'PARTITION BY RANGE ( UNIX_TIMESTAMP(`date`) ) ( PARTITION pmax VALUES LESS THAN ' at line 50 System Information: . Apache/2.2.10 (Linux/SUSE) . MySQL client version: 5.0.67 . PHP extension: mysqli Server is SLES 11 SP1

2 1

get server parameters from DB when kamailio starts
by Uri Shacked 14 Feb '12

14 Feb '12

Hi again... (i do find myself ask many questions these days.... :-)) well, now here is one more: i have 2 kamailio servers that i use in 2 locations. both use the same tables and DBURL. each of my server turns to a number portability (NP) server at its same location. for example, if server 1 has the call, it will turn to NP server 1 with the query, and if the query fails, it will then turn to NP server 2. the kamailio server 2 does the same. but, turns to NP server 2 first and if it fails, to NP server 1. once this is understood.... here is my issue - i want to have 1 generic cfg file for my 2 servers. today, i use the same cfg file, but in server 1 i set it in the cfg to its group, and on server 2 the same but a different group.... i tried to understand how do i get each server to "pull" its group from the DB. something like "what is my group when my IP is X.Y.Z.Q ? i raed the cfg_db module but didn't understand how to set it on my cfg file and on the DB. what is the best way to do what i want? hope i was clear.... BR, thanks to all the guys that help me so far, i appriciate it a lot! Uri

3 8

London Meetup in March, at UC Expo
by Daniel-Constantin Mierla 14 Feb '12

14 Feb '12

Hello, it is very likely I will be in London and visit UC Expo show (http://www.ucexpo.co.uk/) at least the first day. Therefore I am thinking of organizing a Kamailio/SIP/VoIP social networking event (perhaps a dinner or pub meeting), on the evening of the 5th or 6th of March. Are people from community interested in joining such event? If yes, write me your preferred date -- based on feedback in the next days, it will be decided whether it is going to happen and when. Also, if many people would be interested in a project developments and updates session (like the one did in 2010: http://www.kamailio.org/w/present-and-future-of-sip-routing-2010-london/), I can try to get a location for 2-3 hours on the 5th of March for such meetup. Let me know as well your interest in such event. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

get myself ip address on start
by Uri Shacked 14 Feb '12

14 Feb '12

Hi, i would like to get the kamailio server ip address on start up and save it in an $shv(). i tried to use "myself" but did manage to get the ip from it. any ideas? Uri

1 0

Is there something like "silence detect hangup"
by Stoyan Mihaylov 13 Feb '12

13 Feb '12

Problem - if connection drop, call can persist. In Asterisk there is silencedetecthangup - which should cause hangup, if there is full silence for desired period of time. Unfortunately it does not hangup. I mean: SIP client 1 ->Kamailio -> Asterisk ->Kamailio -> SIP client 2 If I drop connection for one of SIP clients, I expect call should be automatically hangup after a time I set (20 sec). But call persists. In worst case - if connection drops for both clients, call will persist until Asterisk is restarted. I will continue to look how to solve problem with Asterisk, but I am curious if this can be done from Kamailio also. If I can cancel call from both places - it will be great. I need to ensure that if something wrong happens, call will be dropped within 30 sec maximum. Stoyan

2 2

Is exec module ignoring time_to_kill parameter?
by Miguel Baptista 13 Feb '12

13 Feb '12

Hello, I am running kamailio 3.1.3 and I am making some tests with the exec module. I am running a "dummy" script and it worked like a charm. Now, I am testing what happens when the script takes "too much time" to return a result. According to exec module documentation the/ time_to_kill /parameter should take care of it. *time_to_kill* (INTEGER) /Specifies the longest time a program is allowed to execute. If the time is exceeded, the program is killed. / BTW ... I am assuming /time /is in milliseconds. I think the documentation should refer the time units. Apparently kamailio ignores this parameter (or it doesn't work as I am expecting). No matter which value I assign to /time_to_kill/, kamailio doesn't kill the script and waits for the script to finish. In the meanwhile some SIP timers are triggered ... resulting into a failed call. The test I am doing is quite simple. For incoming INVITEs, use a external script to check the status of a specific test user, if it returns the keyword "busy" sends it to voicemail otherwise route the INVITE to the local user. The test script returns a result after about 8 seconds. Shouldn't the /time_to_kill/ be triggered and stop the script? Here goes some relevant parts of my configuration. In attachment I am sending the kamailio log with a debug value of 4. /# ----- exec params -----/ /modparam("exec", "time_to_kill", 10)/ /# --- exec params end -----/ /#route just to check the exec module route[TESTEXEC] { xlog("L_INFO", "TESTEXEC: Entered ($rU)\n"); if ($rU == "miguel") { $var(calendarTMP)="/root/" + $rU + "_test.pl"; exec_avp("$var(calendarTMP)", "$avp(s:test)"); xlog("L_INFO", "TESTEXEC: Returned value is $avp(s:test)\n"); if ($avp(s:test) == "busy") { $ru = "sip:userbusy@" + ASTSERVIPPORT; # Voicemail route(RELAY); } } xlog("L_INFO", "TESTEXEC: Exiting ($rU)\n"); return; } / Regards, Miguel Baptista

2 2

TCP keepalive on server and client
by Yufei Tao 13 Feb '12

13 Feb '12

Hi I want to use TCP keepalive to detect dead connections between Kamailio and my SIP clients. I've got some questions about this: 1. If I enable it on Kamailio by setting for example: tcp_crlf_ping = yes tcp_keepalive = yes tcp_keepidle = 60 tcp_keepintvl = 10 tcp_keepcnt = 3 >From what I read by gooling, that means Kamailio only responds to the crlfcrlf sent by the client, but can't be configured to send the crlfcrlf. Is this right? 2. On my client, if I use setsockopt to set the corresponding values: SO_KEEPALIVE TCP_KEEPCNT TCP_KEEPIDLE TCP_KEEPINTVL would it make the client send crlfcrlf? The reason I'm confused is, with these settings, if Kamailio can only respond to crlfcrlf, but not send, then would the client only respond too, as in 1. and 2., they seem to configure the same set of parameters? Please help me clarify this. Thanks very much! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

3 2

Kamailio to Skype/Gtalk/Yahoo
by Abid Naseem 13 Feb '12

13 Feb '12

Hi, Just wanted to know if it is possible for kamailio to route calls to user's skype/gtalk/yahoo messenger? -- Regards, *Abid Naseem* +44 (793) 741 5043 +1 (415) 562 6333

2 1

Howto configure to use SQLITE
by Perazim 13 Feb '12

13 Feb '12

I am a new user and am trying to configure kamailio to use SQLITE instead of MYSQL. I see that SQLITE option is rather new and maybe not as documented as MYSQL. I am running on fedora 14 and have installed MYSQL3 from rpms. Kamailio I built from sources because I need the SQLITE support. What kamailio.cfg and kamctlrc parameters do I need to change to do this? I have already set these in kamctlrc: ... ## database type: MYSQL, PGSQL, ORACLE, DB_BERKELEY, DBTEXT, or SQLITE # by default none is loaded # # If you want to setup a database with kamdbctl, you must at least specify # this parameter DBENGINE=SQLITE ## database host DBHOST=localhost ## database name (for ORACLE this is TNS name) # DBNAME=openser ... and ran: kamdbctl create openser and kamctl add user1 password1 successfully. Now, it appears that I need to change kamailio.cfg for SQLITE. Can I simply edit this file substituting SQLITE for MYSQL and loading the SQLITE module? How about this part: ... ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openser:openserrw@localhost/openser" #!endif ... Will this URL work with SQLITE? Thanks in advance, Perazim

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OT: Sipwise raises Series A round for Kamailio-based soft-switch
by Andreas Granig 11 Feb '12

11 Feb '12

Hey guys, We're thrilled to announce that we closed our Series A financing round for our Kamailio-based soft-switch: http://www.sipwise.com/news/sipwise_series_a/ We're desperately looking for Voip engineers with good skills in SIP, Kamailio and/or Sems, so if you're interested in improving our open-source solution based on Kamailio, Sems, Asterisk etc, please contact me! Best regards, Andreas

1 0

Detection of broken TCP connections to clients
by Yufei Tao 10 Feb '12

10 Feb '12

Hi When clients register to Kamailio over TCP/TLS, if I set tcp_connection_lifetime to be quite small, like 30 seconds, and let clients send refreshes every 20 seconds, for example. When client crashes or have network problems, the TCP connection with Kamailio will be closed by the server after 30 seconds of being idle. My question is, is there a way of letting Kamailio tell which user the broken TCP connection belongs to, and then remove the user from the location table and update presence information etc? The reason I want this is, if a user publishes its status of being online with expiring time of 1 hour, but then crashed, then the location table entry and presence information of this user would be wrong for the rest of the hour. How do I get around this? I assume server can send periodic OPTION messages but that would be a bit expensive. Thank you very much! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

3 3

Re: [SR-Users] dialog vars issue
by Daniel-Constantin Mierla 09 Feb '12

09 Feb '12

Hello, back on mailing list ... I think I got it -- the acc record is written when 200 ok for invite is sent out, dialog is processing it before. Therefore dialog structure is updated with to-tag from the reply, but the building of acc record is executed using the INVITE request taken from transaction, where is no to-tag (acc internal design). This results in failing getting the dialog, because of to-tag mismatch. I will send a patch soon. Cheers, Daniel On 2/8/12 2:39 PM, Jon Bergli Heier wrote: > Hi, this is a gzipped log from a test call using kamailio 3.2.1. There's some noise in the log from REGISTERs and so on. > The dlg_var I'm setting is pzzid, printed as zid=<value>. > > Thanks, > Jon -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 2

Howto configure to use SQLITE
by Perazim 09 Feb '12

09 Feb '12

I am a new user and am trying to configure kamailio to use SQLITE instead of MYSQL. I see that SQLITE option is rather new and maybe not as documented as MYSQL. I am running on fedora 14 and have installed MYSQL3 from rpms. Kamailio I built from sources because I need the SQLITE support. What kamailio.cfg and kamctlrc parameters do I need to change to do this? I have already set these in kamctlrc: ... ## database type: MYSQL, PGSQL, ORACLE, DB_BERKELEY, DBTEXT, or SQLITE # by default none is loaded # # If you want to setup a database with kamdbctl, you must at least specify # this parameter DBENGINE=SQLITE ## database host DBHOST=localhost ## database name (for ORACLE this is TNS name) # DBNAME=openser ... and ran: kamdbctl create openser and kamctl add user1 password1 successfully. Now, it appears that I need to change kamailio.cfg for SQLITE. Can I simply edit this file substituting SQLITE for MYSQL and loading the SQLITE module? How about this part: ... ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openser:openserrw@localhost/openser" #!endif ... Will this URL work with SQLITE? Thanks in advance, Perazim

1 0

Kamailio Socket
by Reda Aouad 09 Feb '12

09 Feb '12

Is this normal on restarting Kamailio ? ERROR: ctl [ctl.c:379]: ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1) RA

2 1

get server parameters from DB when kamailio starts
by Uri Shacked 08 Feb '12

08 Feb '12

Hi again... (i do find myself ask many questions these days.... :-)) well, now here is one more: i have 2 kamailio servers that i use in 2 locations. both use the same tables and DBURL. each of my server turns to a number portability (NP) server at its same location. for example, if server 1 has the call, it will turn to NP server 1 with the query, and if the query fails, it will then turn to NP server 2. the kamailio server 2 does the same. but, turns to NP server 2 first and if it fails, to NP server 1. once this is understood.... here is my issue - i want to have 1 generic cfg file for my 2 servers. today, i use the same cfg file, but in server 1 i set it in the cfg to its group, and on server 2 the same but a different group.... i tried to understand how do i get each server to "pull" its group from the DB. something like "what is my group when my IP is X.Y.Z.Q ? i raed the cfg_db module but didn't understand how to set it on my cfg file and on the DB. what is the best way to do what i want? hope i was clear.... BR, thanks to all the guys that help me so far, i appriciate it a lot! Uri

1 0

finally, an open alternative to Viber
by Daniel - Lumicall.org 08 Feb '12

08 Feb '12

The rise of Viber has, for some people, been a case of `skype, not again!' until now... Lumicall is now in the Android market - and it fully interacts with other SIP products using ENUM and SRV records. Any feedback on this is welcome. Interconnect (Kamailio calling to/from Lumicall users) is based on the idea of federated SIP, it is explained at http://www.lumicall.org for those who want to connect up to it. Please bear in mind: Lumicall supports ICE (RFC 5245) for NAT traversal, it is using the ice4j implementation from the Jitsi community. This makes the SIP packets bigger and often they are too big for the MTU of a UDP packet. When using ICE, it seems essential to use TLS, to avoid the MTU problems and also to avoid routers mangling the SIP headers (ICE doesn't need help from routers, they only confuse the algorithm)

2 2

What is the valid values for flags field in dispatcher table?
by Gary Chen 08 Feb '12

08 Feb '12

Trying to figure out the values for flags field. where can I find the meaning of flags for each value like: flags 1 = ? flags 2 = ? flags 3 = ? flags 4 = ? flags 5 = ? flags 6 = ? flags 7 = ? flags 8 = ? Gary

2 1

Re: [SR-Users] "kamctl cr show" when carrierroute table has another name
by Uri Shacked 08 Feb '12

08 Feb '12

OK i got it. on the kamctl, the show is a SQL and the reload is $CTLCMD this is the difference... thanks On Wed, Feb 8, 2012 at 7:45 PM, Henning Westerholt <hw(a)kamailio.org> wrote: > On Wednesday 08 February 2012, Uri Shacked wrote: > > Can u explain why does the reload takes the table name from cfg and > > the show from the kamctlrc ? > > Hi Uri, > > the reload is executed in the server, which already uses the correct table. > The show is executed from the script (its just a database operation, you > could > also use a mysql client), which uses the wrong table. > > Viele Grüße/ best regards, > > Henning Westerholt >

1 0

Re: [SR-Users] "kamctl cr show" when carrierroute table has another name
by Uri Shacked 08 Feb '12

08 Feb '12

> > > anyone for this? > > Hi, > > after resolving the field type issue :-), i changed the params in kamctlrc > to point to the right DB. > now, i notice that when i use the command "kamctl cr show" iget an error > that it does not find the table "carrierroute". > in the cfg file it is all set and works fine, cr dump is working as > expected as well. > > in the kamctl file i can find the param - CARRIERROUTE_TABLE. but where is > it set from? > anyone knows where do i make tha change in order for the kamctl to "know" > the names of the tables i use? > > thanks, > Uri >

2 3

(no subject)
by Jon Bergli Heier 08 Feb '12

08 Feb '12

Hi, since kamailio 3.2.1 I've had problems with using dlg_var to store extra accounting fields with the acc module (using db_extra and mysql). When I set a dlg_var in the request route, the first acc row (INVITE) appears as if the dlg_var was empty, but looks fine for the second (BYE) row. In 3.2.0 both rows got the value I assigned. missed_calls doesn't appear to have this problem. Any ideas? The problem also seems to exist in kamailio 3.2.2. I've done a git bisect and the problem seems to appear since commit 5bdee1ffe820d3e9e75898bb252d1660b4bbad60. Thanks, Jon

2 3

best way to start kamailio at boot
by Uri Shacked 08 Feb '12

08 Feb '12

Hi, any help will be graet.... * hi, *>* *>* when i moved the /etc/init.d/kamailio file. kamailio did not start at boot. *>* now, i set the debug level to 4 and run kamailio in both ways. *>* *>* 1. "/etc/init.d/kamailio start" *>* * the resault was : *Feb 1 19:59:04 net-ivr-KamIN-J1 kamailio: DEBUG: <core> [mem/q_malloc.c:202]: qm_malloc_init: QM_OPTIMIZE=16384, /ROUNDTO=2048 Feb 1 19:59:04 net-ivr-KamIN-J1 kamailio: DEBUG: <core> [mem/q_malloc.c:204]: qm_malloc_init: QM_HASH_SIZE=2099, qm_block size=218344 Feb 1 19:59:04 net-ivr-KamIN-J1 kamailio: DEBUG: <core> [mem/q_malloc.c:205]: qm_malloc_init(0x7f1cf585b000, 2147483648), start=0x7f1cf585b000 Feb 1 19:59:04 net-ivr-KamIN-J1 kamailio: DEBUG: <core> [mem/q_malloc.c:213]: qm_malloc_init: size= 2147483648, init_overhead=218440 Feb 1 19:59:04 net-ivr-KamIN-J1 kamailio: DEBUG: <core> [mem/shm_mem.c:203]: shm_mem_init: success* >* *>* you can see the shm_mem size is 2048Mb. *>* * 2."kamailio" the resault was : *Feb 1 19:56:50 net-ivr-KamIN-J1 kamailio: DEBUG: <core> [mem/q_malloc.c:202]: qm_malloc_init: QM_OPTIMIZE=16384, /ROUNDTO=2048 Feb 1 19:56:50 net-ivr-KamIN-J1 kamailio: DEBUG: <core> [mem/q_malloc.c:204]: qm_malloc_init: QM_HASH_SIZE=2099, qm_block size=218344 Feb 1 19:56:50 net-ivr-KamIN-J1 kamailio: DEBUG: <core> [mem/q_malloc.c:205]: qm_malloc_init(0x7f8618310000, 33554432), start=0x7f8618310000 Feb 1 19:56:50 net-ivr-KamIN-J1 kamailio: DEBUG: <core> [mem/q_malloc.c:213]: qm_malloc_init: size= 33554432, init_overhead=218440 Feb 1 19:56:50 net-ivr-KamIN-J1 kamailio: DEBUG: <core> [mem/shm_mem.c:203]: shm_mem_init: success* ** the resault is 32Mb. any ideas? >* *>* On Wed, Feb 1, 2012 at 5:01 PM, Jason Penton <jason.penton at smilecoms.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >wrote: *>* *>>* I am not familiar with Centos but I suspect there is another script being *>>* run from somewhere. What I would do is move the /etc/init.d/kamailio script *>>* somewhere safe (your home directory for eg). Then reboot and see if *>>* kamailio still starts - if it does you need to find out whihc script is *>>* being called and where it is. *>>* *>>* If you are not running X by default you are prob in runlevel 3. If you *>>* are running X, then it will be 5. check where you runlevel scripts are *>>* pointing. e.g. *>>* *>>* ls -al /etc/rc3.d/SXXkamailio (X = [0-9]) *>>* *>>* Cheers *>>* Jason *>>* *>>* On Wed, Feb 1, 2012 at 4:26 PM, Uri Shacked <uri.shacked at gmail.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >wrote: *>>* *>>>* Hi, *>>>* *>>>* when i reboot the machine, it starts with 32Mb and i do not understand *>>>* why... *>>>* so, does it use a startup file that i am not familiar with? *>>>* what is the appropriate runlevel? how do i set it and where? *>>>* *>>>* thanks, *>>>* Uri *>>>* *>>>* *>>>* >Hi Sammy, *>>>* > *>>>* >When you use the init script, it is starting Kamailio with the correct *>>>* >arguments to get the environment you wish (-m 2048 probably). If *>>>* however, *>>>* >you just call kamailio these args are not there. Im not sure why you are *>>>* >running with kamailio command directly. Why don't you add your init *>>>* script *>>>* >to the appropriate runlevel for auto startup? *>>>* > *>>>* >Cheers *>>>* >Jason *>>>* *>>>* On Wed, Feb 1, 2012 at 3:06 PM, Uri Shacked <ushacked at gmail.com< http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> *>>>* wrote: *>>>* *>>>* >* Hi again.... *>>>* *>* *>>>* *>* i installed kamailio 3.2.2 today. *>>>* *>* i used the chkconfig tool to get kamailio run at boot. *>>>* *>* i edited both /etc/init.d/kamailio and /etc/default/kamailio files *>>>* to run *>>>* *>* with 2048Mg of memory. *>>>* *>* now, when i run /etc/init.d/kamailio start, it starts with the right *>>>* *>* amount of memory. *>>>* *>* when i rum "kamailio" only, it starts with 32Mb.... *>>>* *>* *>>>* *>* i use centos 6.2 64bit. *>>>* *>* any ideas? *>>>* *>* *>>>* *>* *>>>* *>* *>>>* *>* *>>>* *>* *>>>* *>* On Tue, Oct 25, 2011 at 1:18 PM, Uri Shacked <ushacked at gmail.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> *>>>* wrote: *>>>* *>* *>>>* *>>* Hi. *>>>* *>>* I already checked it all. The kamailio start and stop command did *>>>* not *>>>* *>>* work. The init script i used was for debian. Now i use another *>>>* script, *>>>* *>>* for centos. It works. I add the kamailio proccess with chkconfig *>>>* tool. *>>>* *>>* Thanks. *>>>* *>>* *>>>* *>>* On 10/25/11, Sammy Govind <govoiper at gmail.com< http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> *>>>* wrote: *>>>* *>>* > Thanks Alex, *>>>* *>>* > *>>>* *>>* > Hey Uri, can you check if you've /etc/init.d/kamailio file *>>>* exists - open *>>>* *>>* > that file if exists and cross verify few parameters. *>>>* *>>* > Few of these are as follows *>>>* *>>* > *>>>* *>>* > PATH=/sbin:/bin:/usr/sbin:/usr/bin *>>>* *>>* > DAEMON=*/usr/local/sbin/kamailio* *>>>* *>>* > NAME=kamailio *>>>* *>>* > DESC=kamailio *>>>* *>>* > HOMEDIR=/var/run/ *>>>* *>>* > *PIDFILE*=$HOMEDIR/$NAME.pid *>>>* *>>* > DEFAULTS=*/etc/default/kamailio.default* *>>>* *>>* > RUN_KAMAILIO=yes *>>>* *>>* > *>>>* *>>* > Make sure /etc/default/kamailio.default file exits. here are the *>>>* *>>* contents of *>>>* *>>* > my file. *>>>* *>>* > *>>>* *>>* > RUN_KAMAILIO=yes *>>>* *>>* > USER=kamailio *>>>* *>>* > GROUP=kamailio *>>>* *>>* > MEMORY=64 *>>>* *>>* > DUMP_CORE=no *>>>* *>>* > *>>>* *>>* > On Tue, Oct 25, 2011 at 3:38 AM, Klaus Darilion < *>>>* *>>* > klaus.mailinglists at pernau.at< http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> *>>>* wrote: *>>>* *>>* > *>>>* *>>* >> Test the startupscript if they work: *>>>* *>>* >> *>>>* *>>* >> /etc/init.d/kamailio start *>>>* *>>* >> /etc/init.d/kamailio stop *>>>* *>>* >> *>>>* *>>* >> If this works, then make sure to have proper symlinks from the *>>>* *>>* >> corresponding runlevel to the startup script. Do it manually or *>>>* use the *>>>* *>>* >> chkconfig tool. *>>>* *>>* >> *>>>* *>>* >> klaus *>>>* *>>* >> *>>>* *>>* >> *>>>* *>>* >> On 24.10.2011 08:00, Uri Shacked wrote: *>>>* *>>* >> *>>>* *>>* >>> hi again, *>>>* *>>* >>> i found out that on my centos 5.4 the command *>>>* start-stop-daemon that *>>>* *>>* the *>>>* *>>* >>> /etc/init.d/kamailio uses is not recognized. *>>>* *>>* >>> it is kamailio3.2 any ideas? *>>>* *>>* >>> *>>>* *>>* >>> *>>>* *>>* >>> On Mon, Oct 24, 2011 at 3:03 PM, Uri Shacked <ushacked at *>>>* gmail.com< http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> *>>>* *>>* >>> <mailto:ushacked at gmail.com< http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>>> *>>>* wrote: *>>>* *>>* >>> *>>>* *>>* >>> Hi, *>>>* *>>* >>> what is the best way to start kamailio at boot? *>>>* *>>* >>> the example in the install is not working for me.... *>>>* *>>* >>> *>>>* *>>* >>> *>>>* * *>>>* *>>* *>>* This email is subject to the disclaimer of Smile Communications (PTY) Ltd. at http://www.smilecoms.com/disclaimer *>>* *>>* *>* *

2 2

sipp issues with dialog module
by Uri Shacked 08 Feb '12

08 Feb '12

Hi, anyone knows why the dialog module leaves active dialogs even after a bye was recieved when using sipp? once i had issues with the record route and i forced sipp to deal with that, is there something to set on the bye sent by sipp? BR, Uri

2 2

Accessing indexed header PV from app_lua module in Kamailio
by Chris Austin 08 Feb '12

08 Feb '12

Hi all, I have have been attempting to implement Remote-Party-ID support using the app_lua module in Kamailio. Since more than one Remote-Party-ID header may exist, this requires accessing the header pseudo-variable with an index. For example: $(hdr(Remote-Party-ID)[2]) Doing this causes Kamailio to output an error to the log when it fails to parse the PV: 2012-02-02 16:38:58|E| 6(26940) ERROR: app_lua [app_lua_sr.c:892]: invalid pv [$(hdr(remote-party-id)[0])] (-1/26) This was triggered using sr.pv.get("$(hdr(remote-party-id)[0])") from within Lua. I have also tried sr.pv.get("$hdr(remote-party-id)[0]") which also failed: 2012-02-02 18:26:28|E|15(26956) ERROR: app_lua [app_lua_sr.c:892]: invalid pv [$hdr(remote-party-id)[0]] (21/24) Am I doing something wrong? If not, any ideas for a workaround? Thanks for your time.

2 1

Store auth username in registrations
by Spencer Thomason 08 Feb '12

08 Feb '12

Hello, I have clients which register to Kamailio using an auth username. The from username is used for caller id on an INVITE. This causes a problem if the client does not use the auth username in the from for the REGISTER. The caller id number is then saved as the AOR username which doesn't correspond to a valid account. Is there a way I can store the authorized username instead of the From username in the AOR? Thanks, Spencer

2 5

debuger, debug level and log
by Uri Shacked 08 Feb '12

08 Feb '12

hi, i didn't fully understand the differences between the debug level set at the Global Parameters section and the debugger module. i would like to start kamailio with debug level 0, and to be able to raise or lower the level in command line. the log is set to be writen to /var/log/kamailio, and i can trace it using "tail". so, anyideas how to use the debug options with the option to change level at runtime? BR, Uri

2 1

High Availability for ongoing calls
by Mariana Arduini 07 Feb '12

07 Feb '12

Hi guys! I'm facing some problems with dialog information while trying to implement a HA environment. We have 2 clustered SIP servers working as a Active/Standby failover and the use of a Virtual IP address. We want to make sure that if proxy #1 goes down during a call, proxy #2 would be able to handle the sequential requests for that call, since both proxies are using the same database for dialog module information and we have set the db_mode dialog parameter to 1 (REALTIME - any dialog information changes will be reflected into the database immediately). Instead, what happened is proxy #2 was not aware of the dialog and gave us a "500 Server error occurred" message after receiving the BYE. Here are the log errors: /usr/local/sbin/opensips[12617]: New request: BYE /usr/local/sbin/opensips[12617]: The request has TO tag /usr/local/sbin/opensips[12617]: cannot match request to a dialog /usr/local/sbin/opensips[12617]: ERROR:dialog:w_validate_dialog: null dialog /usr/local/sbin/opensips[12617]: in-dialog bogus request /usr/local/sbin/opensips[12617]: validate_dialog return: -4 /usr/local/sbin/opensips[12617]: loose_route() performed, rr headers were decoded /usr/local/sbin/opensips[12617]: ERROR:core:parse_uri: uri too short: <0> (1) /usr/local/sbin/opensips[12617]: ERROR:core:parse_sip_msg_uri: bad uri <0> /usr/local/sbin/opensips[12617]: ERROR:tm:new_t: uri invalid /usr/local/sbin/opensips[12617]: ERROR:tm:t_newtran: new_t failed /usr/local/sbin/opensips[12617]: Could not execute t_relay() Virtual IP address is working, a new call goes through with no problem. Apparently, the dialog module is not saving all dialogs information to the database right away. Does anyone know about any other info we should be storing in the database, using some sort of module parameter or exported function from config file? In other words, what do I need to do in order to send an initial INVITE to one server and, when that server goes down, send the BYE to another server with the same call dialog? Thanks in advance. Mariana.

2 2

FreeBSD comiling problem
by Serhat AKCA 07 Feb '12

07 Feb '12

Hello I want to run Kamailio under FreeBSD. I compiled with the makefiles under freebsd packages. But It gives alotof errors. I want to use 3.2.2 version but I cannot compile. Could you help me To compile in FreeBSD ? Another question, which is the best OS for Kamailio ? In the forums, they are saying FreeBSD is more stable and More performans. Regards Serhat

4 3

Proxy topology question
by Greg Mannie 06 Feb '12

06 Feb '12

Hello After much reading I have come to the realization that after years of using Asterisk I know very little about Sip. I have my Kamailio box up, I have Asterisk 1.8.x running with realtime working. I thought it would be just a case of registering SIP trunks from my provider to the kamailio and registering our internal asterisk servers to the kamailio. Much of what I read talks about using Asterisk as the PSTN interface, but that interface is through a sip trunk purchased from a provider. Won't Kamailio be the PSTN gateway? The idea here is to pool all the sip trunks from the various hosted asterisk solutions (VM running asterisk) and point them all to a proxy to facilitate the aggregation of traffic. I have been reading SIP tutorials and the mailing list archives. If anyone has a sample config and perhaps a little direction it would be highly appreciated. Thank you Greg

2 5

loading carrierroute failes using another DB
by Uri Shacked 06 Feb '12

06 Feb '12

Hi, I changed the location of the DB kamailio works with. i copied the relevalt tables and changed the relevant names in the cfg file. while trying to load kamailio i realized that it misses the "veresion" table. first - can someone explain the version table usage? second, after creating and copying that data from the version table. i tried to load kamailio again and got the follwing: " DEBUG: carrierroute [cr_db.c:345]: loading, cycle 0 0(24689) ERROR: <core> [daemonize.c:307]: Main process exited before writing to pipe" any help? thanks, Uri

2 3

"kamctl cr show" when carrierroute table has another name
by Uri Shacked 06 Feb '12

06 Feb '12

Hi, after resolving the field type issue :-), i changed the params in kamctlrc to point to the right DB. now, i notice that when i use the command "kamctl cr show" iget an error that it does not find the table "carrierroute". in the cfg file it is all set and works fine, cr dump is working as expected as well. in the kamctl file i can find the param - CARRIERROUTE_TABLE. but where is it set from? anyone knows where do i make tha change in order for the kamctl to "know" the names of the tables i use? thanks, Uri

1 0

Kamailio 3.1.5 core dump from the snmpstats module
by Klaus Feichtinger 06 Feb '12

06 Feb '12

Hello list, I`ve found a negative phenomenon during startup of kamailio 3.1.x (orgininally found in version 3.1.5) that results in a core dump, triggered by the snmpstats module. During startup of the SIP proxy server machine the linux service "heartbeat" is responsible for starting required ressources like mysql, kamailio and snmpd. These services are monitored by the tool "monit". Monit is supervising the state of the named services and if kamailio is not up and running, monit is trying to restart the service. During startup the mysql database takes some time until it is ready to use and therefore kamailio can`t connect to the database. Monit is detecting this and restarting kamailio. This is done so often until kamailio is up and running. Finally all services are up and running and the machine is working fine. However, I do not like the core dump that was created during this "restart phase". As soon as the module "snmpstats" is not used, no core dump is created any more. Information about the core dump looks as follows: (1) output of kamailio log files: <prefix> kamailio: INFO: <core> [tcp_main.c:4726]: init_tcp: using epoll_lt as the io watch method (auto detected) <prefix> /usr/sbin/kamailio[3027]: ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) <prefix> /usr/sbin/kamailio[3027]: ERROR: <core> [db.c:289]: could not add connection to the pool <prefix> /usr/sbin/kamailio[3027]: ERROR: lcr [lcr_mod.c:359]: unable to connect to the database <prefix> /usr/sbin/kamailio[3027]: ERROR: lcr [lcr_mod.c:571]: unable to open database connection <prefix> /usr/sbin/kamailio[3027]: ERROR: <core> [sr_module.c:875]: init_mod(): Error while initializing module lcr (/usr/lib/kamailio/modules/lcr.so) <prefix> /usr/sbin/kamailio[3027]: INFO: snmpstats [snmpstats.c:387]: The SNMPStats module got the kill signal <prefix> kamailio: ERROR: <core> [daemonize.c:307]: Main process exited before writing to pipe <prefix> kamailio: INFO: <core> [tcp_main.c:4726]: init_tcp: using epoll_lt as the io watch method (auto detected) <prefix> kamailio: WARNING: <core> [daemonize.c:352]: pid file contains old pid, replacing pid <prefix> /usr/sbin/kamailio[3322]: INFO: usrloc [hslot.c:53]: locks array size 512 <prefix> /usr/sbin/kamailio[3322]: INFO: auth [auth_mod.c:312]: auth: qop set, but nonce-count (nc_enabled) support disabled <prefix> /usr/sbin/kamailio[3322]: INFO: pua [pua.c:360]: the query returned no result <prefix> /usr/sbin/kamailio[3322]: INFO: pike [ip_tree.c:88]: probing 256 set size <prefix> /usr/sbin/kamailio[3322]: INFO: <core> [udp_server.c:178]: INFO: udp_init: SO_RCVBUF is initially 111616 <prefix> /usr/sbin/kamailio[3322]: INFO: <core> [udp_server.c:229]: INFO: udp_init: SO_RCVBUF is finally 262142 <prefix> /usr/sbin/kamailio[3329]: INFO: mi_datagram [mi_datagram.c:326]: a new child 0/3329 (2) output of GDB / core file: Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 1024 -u kamailio -g kama'. Program terminated with signal 11, Segmentation fault. [New process 3027] #0 0xb775539c in freeInterprocessBuffer () at interprocess_buffer.c:354 354 interprocess_buffer.c: No such file or directory. in interprocess_buffer.c (gdb) I know that it is not a fine method in killing kamailio during initialisation. But I have not found a better way yet. The other modules like LCR have no problem with the kill signal during initialisation. My main question is now: how could these core dumps being avoided? Did anybody have the same experience as me? Thanks in advance, Klaus

2 2

Kamailio, Freeradius and CDRTool
by Reda Aouad 05 Feb '12

05 Feb '12

Hi, I am trying to get Kamailio to work with CDRTool. In the step of installing freeradius and adding kamailio's dictionary dictionary.kamailio to freeradius, I get the following error when starting freeradius -X, which fails to start: Error: Errors reading dictionary: dict_init: No ATTRIBUTE "Sip-Method" defined for VALUE "Other" Replacing Kamailio's dictionary with OpenSips' one, freeradius starts correctly. I assume then that there's something missing in Kamailio's dictionary. I am using Kamailio v3.2.0 and it's dictionary file. Any help ? Thanks :) RA

1 1

3.2.2
by dotnetdub 03 Feb '12

03 Feb '12

Hi All, I've just installed 3.2.2. I'm using a config that was working fine with 3.1 and all seems ok. I get one error that I don't get on 3.1 at startup. kamailio: ERROR: <core> [ppcfg.c:79]: ### added subst expression: /DB/mysql This is repeated for all SUBST statements in the config. I get these errors also: [1293000.484026] INFO: task kamailio:19912 blocked for more than 120 seconds. [1293000.484081] "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. [1293000.484169] kamailio D c4608e58 0 19912 19899 0x00000000 [1293000.484173] f6b5d980 00000082 40827cc6 c4608e58 000006cf c1441e20 c1441e20 000000fd [1293000.484177] f6b5db3c c4608e20 00000000 4085cc46 00049753 ffe3a684 ffffffff f7077bac [1293000.484180] f6b5d9ac f6b5db3c f6b5d980 c102b2b7 00000001 c4608e20 00000004 00000005 [1293000.484184] Call Trace: [1293000.484191] [<c102b2b7>] ? check_preempt_wakeup+0x18c/0x202 [1293000.484195] [<c127f55f>] ? __mutex_lock_common+0xe8/0x13b [1293000.484198] [<c127f5c1>] ? __mutex_lock_slowpath+0xf/0x11 [1293000.484200] [<c127f652>] ? mutex_lock+0x17/0x24 [1293000.484202] [<c127f652>] ? mutex_lock+0x17/0x24 [1293000.484205] [<c123f68b>] ? unix_dgram_recvmsg+0x49/0x1e4 [1293000.484207] [<c10268bd>] ? __wake_up_common+0x34/0x59 [1293000.484210] [<c102a97c>] ? __wake_up_sync_key+0x33/0x49 [1293000.484213] [<c11dd3b0>] ? __sock_recvmsg+0x50/0x58 [1293000.484215] [<c11dd8e7>] ? sock_recvmsg+0x9d/0xb6 [1293000.484218] [<c104a652>] ? autoremove_wake_function+0x0/0x2d [1293000.484221] [<c1092950>] ? __alloc_pages_nodemask+0xf3/0x4d9 [1293000.484224] [<c108e16b>] ? find_lock_page+0x1b/0x36 [1293000.484227] [<c1025769>] ? kmap_atomic_prot+0xd7/0xfc [1293000.484230] [<c113cd18>] ? cpumask_any_but+0x20/0x2b [1293000.484232] [<c11de830>] ? sys_recvfrom+0xc0/0x120 [1293000.484235] [<c1143b63>] ? copy_from_user+0x27/0x10e [1293000.484237] [<c1025769>] ? kmap_atomic_prot+0xd7/0xfc [1293000.484240] [<c10a2deb>] ? handle_mm_fault+0x982/0xa22 [1293000.484243] [<c104a652>] ? autoremove_wake_function+0x0/0x2d [1293000.484246] [<c10baa1d>] ? fsnotify_access+0x5a/0x61 [1293000.484248] [<c11decee>] ? sys_socketcall+0x123/0x1aa [1293000.484251] [<c100813b>] ? sysenter_do_call+0x12/0x28 19912 is MI datagram. I have been getting these errors since 1.x and what I normally do is echo 0 > /proc/sys/kernel/hung_task_timeout_secs to get rid of it. It must be a module that I am loading. Is this possible to debug? I can again get rid of the messages but would love to know whats causing it. BR Shena

1 1

question about NAT being behind of a Load Balancer
by Albert Petit 03 Feb '12

03 Feb '12

Hi , Sorry for previous email. Hit send too soon . Good morning, On my system i've a structure similar to: SIP Clients SIP Platform [-------------------] [---------------------------------] S1 @IP2 S2 @IP3 SIP UACs SIP LB ... [NATted and @IP1 non Nattted] SN @IPN SIP LB is based on Kamailio dispatcher and is responsible of distributing the load between different Servers SIP UAS. S1, S2,... SN Servers have a Kamailio Server too which will forward received balanced traffic to appropiate service (Presence , registrar,...) Considering SIP LB and all SIP UAS have Public IP, how NAT from the UACs could be handled on this scenario? Initially I thought NAT should be handled on SIP LB however I would like the most simple possible load balancer with no need to read user locations . Responsible of location users should be S1,S2,..:SN Then, would it be possible to handle NAT of the SIP UACs *from the SIP UAS *with NAT Helper module? What i'm afraid is: If all traffic to S1,S2,..SN comes from SIP LB then nat_helper could think all clients are NATTed because top most via will be SIP LB. What I need, I think, is that NAT Helper decides if there is NAT looking the Via Header added originally by UAC , and not the top most via (which would be SIP LB). Is that possible? If possible how should i configure Nat helper for that? BR Albert

2 3

Re: [SR-Users] sr-users Digest, Vol 81, Issue 7
by Yufei Tao 03 Feb '12

03 Feb '12

Hi I think I got the tcp_crlf_ping and the tcp_keepalive parameters mixed up. So for tcp_crlf_ping, Kamailio will only respond. While for tcp_keepalive parameters, if enabled on both server and client, both will be sending the keepalives, and the receiver will respond. In order to use tcp_crlf_ping, client's app level rather than TCP level would have to send the crlfcrlf (pings) and process the crlf (pongs) in order to detect if Kamailio is up. Is this right? Yufei On 03/02/12 11:00, sr-users-request(a)lists.sip-router.org wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. TCP keepalive on server and client (Yufei Tao) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 3 Feb 2012 10:33:11 +0000 > From: Yufei Tao <yufei.tao(a)redembedded.com> > Subject: [SR-Users] TCP keepalive on server and client > To: <sr-users(a)lists.sip-router.org> > Message-ID: <4F2BB7E7.2090601(a)redembedded.com> > Content-Type: text/plain; charset="ISO-8859-1" > > Hi > > I want to use TCP keepalive to detect dead connections between Kamailio > and my SIP clients. I've got some questions about this: > > 1. If I enable it on Kamailio by setting for example: > > tcp_crlf_ping = yes > tcp_keepalive = yes > tcp_keepidle = 60 > tcp_keepintvl = 10 > tcp_keepcnt = 3 > > >From what I read by gooling, that means Kamailio only responds to the > crlfcrlf sent by the client, but can't be configured to send the > crlfcrlf. Is this right? > > 2. On my client, if I use setsockopt to set the corresponding values: > > SO_KEEPALIVE > TCP_KEEPCNT > TCP_KEEPIDLE > TCP_KEEPINTVL > > would it make the client send crlfcrlf? > > The reason I'm confused is, with these settings, if Kamailio can only > respond to crlfcrlf, but not send, then would the client only respond > too, as in 1. and 2., they seem to configure the same set of parameters? > Please help me clarify this. Thanks very much! > > Yufei > > -- > Yufei Tao > Red Embedded > > This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. > > You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. > > Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ > > > > ------------------------------ > > _______________________________________________ > sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > End of sr-users Digest, Vol 81, Issue 7 > *************************************** -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

2 1

(no subject)
by Albert Petit 03 Feb '12

03 Feb '12

Hi, SIP UAS1 SIP LB <--------------------->

2 1

Kamailio ims / rtpproxy
by trojak＠st.fri.uniza.sk 02 Feb '12

02 Feb '12

Hi, I am trying to make kamailio PCSCF working with rtpproxy (for NAT traversal). I would like to ask, if there is any tutorial for this, or if you can give me any advice. I also have installed rtpproxy package; #!define WITH_NAT, and defined rtpproxy port in config file... But this is not working. I noticed, that in default config file of pcscf is nothing in routing logic with rtpproxy (or it is unnecessary ?), so I try to copy route[RELAY] route[NAT] route[RTPPROXY] modules from kamailio (no pcscf) config file... but also with no success. Can you help me please ? What I am doing wrong ? Do you have any configuration working with rtpproxy ? Best wishes Lubo

2 3

RPMs for v3.2.2
by Daniel-Constantin Mierla 02 Feb '12

02 Feb '12

Hello, for the records, the RPMs for v3.2.2 are available (32b and 64b: CentOS 5 and 6, RedHat 5.0 and 6.0, OpenSUSE 11.4 and 12.1, Fedora 15 and 16): * http://www.kamailio.org/wiki/packages/rpms Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Re: [SR-Users] best way to start kamailio at boot
by Jason Penton 01 Feb '12

01 Feb '12

I am not familiar with Centos but I suspect there is another script being run from somewhere. What I would do is move the /etc/init.d/kamailio script somewhere safe (your home directory for eg). Then reboot and see if kamailio still starts - if it does you need to find out whihc script is being called and where it is. If you are not running X by default you are prob in runlevel 3. If you are running X, then it will be 5. check where you runlevel scripts are pointing. e.g. ls -al /etc/rc3.d/SXXkamailio (X = [0-9]) Cheers Jason On Wed, Feb 1, 2012 at 4:26 PM, Uri Shacked <uri.shacked(a)gmail.com> wrote: > Hi, > > when i reboot the machine, it starts with 32Mb and i do not understand > why... > so, does it use a startup file that i am not familiar with? > what is the appropriate runlevel? how do i set it and where? > > thanks, > Uri > > > >Hi Sammy, > > > >When you use the init script, it is starting Kamailio with the correct > >arguments to get the environment you wish (-m 2048 probably). If however, > >you just call kamailio these args are not there. Im not sure why you are > >running with kamailio command directly. Why don't you add your init script > >to the appropriate runlevel for auto startup? > > > >Cheers > >Jason > > On Wed, Feb 1, 2012 at 3:06 PM, Uri Shacked <ushacked at gmail.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> > wrote: > > >* Hi again.... > *>* > *>* i installed kamailio 3.2.2 today. > *>* i used the chkconfig tool to get kamailio run at boot. > *>* i edited both /etc/init.d/kamailio and /etc/default/kamailio files to > run > *>* with 2048Mg of memory. > *>* now, when i run /etc/init.d/kamailio start, it starts with the right > *>* amount of memory. > *>* when i rum "kamailio" only, it starts with 32Mb.... > *>* > *>* i use centos 6.2 64bit. > *>* any ideas? > *>* > *>* > *>* > *>* > *>* > *>* On Tue, Oct 25, 2011 at 1:18 PM, Uri Shacked <ushacked at gmail.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> > wrote: > *>* > *>>* Hi. > *>>* I already checked it all. The kamailio start and stop command did not > *>>* work. The init script i used was for debian. Now i use another > script, > *>>* for centos. It works. I add the kamailio proccess with chkconfig > tool. > *>>* Thanks. > *>>* > *>>* On 10/25/11, Sammy Govind <govoiper at gmail.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> > wrote: > *>>* > Thanks Alex, > *>>* > > *>>* > Hey Uri, can you check if you've /etc/init.d/kamailio file exists > - open > *>>* > that file if exists and cross verify few parameters. > *>>* > Few of these are as follows > *>>* > > *>>* > PATH=/sbin:/bin:/usr/sbin:/usr/bin > *>>* > DAEMON=*/usr/local/sbin/kamailio* > *>>* > NAME=kamailio > *>>* > DESC=kamailio > *>>* > HOMEDIR=/var/run/ > *>>* > *PIDFILE*=$HOMEDIR/$NAME.pid > *>>* > DEFAULTS=*/etc/default/kamailio.default* > *>>* > RUN_KAMAILIO=yes > *>>* > > *>>* > Make sure /etc/default/kamailio.default file exits. here are the > *>>* contents of > *>>* > my file. > *>>* > > *>>* > RUN_KAMAILIO=yes > *>>* > USER=kamailio > *>>* > GROUP=kamailio > *>>* > MEMORY=64 > *>>* > DUMP_CORE=no > *>>* > > *>>* > On Tue, Oct 25, 2011 at 3:38 AM, Klaus Darilion < > *>>* > klaus.mailinglists at pernau.at<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> > wrote: > *>>* > > *>>* >> Test the startupscript if they work: > *>>* >> > *>>* >> /etc/init.d/kamailio start > *>>* >> /etc/init.d/kamailio stop > *>>* >> > *>>* >> If this works, then make sure to have proper symlinks from the > *>>* >> corresponding runlevel to the startup script. Do it manually or > use the > *>>* >> chkconfig tool. > *>>* >> > *>>* >> klaus > *>>* >> > *>>* >> > *>>* >> On 24.10.2011 08:00, Uri Shacked wrote: > *>>* >> > *>>* >>> hi again, > *>>* >>> i found out that on my centos 5.4 the command start-stop-daemon > that > *>>* the > *>>* >>> /etc/init.d/kamailio uses is not recognized. > *>>* >>> it is kamailio3.2 any ideas? > *>>* >>> > *>>* >>> > *>>* >>> On Mon, Oct 24, 2011 at 3:03 PM, Uri Shacked <ushacked at > gmail.com <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> > *>>* >>> <mailto:ushacked at gmail.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>>> > wrote: > *>>* >>> > *>>* >>> Hi, > *>>* >>> what is the best way to start kamailio at boot? > *>>* >>> the example in the install is not working for me.... > *>>* >>> > *>>* >>> > * > This email is subject to the disclaimer of Smile Communications (PTY) Ltd. at http://www.smilecoms.com/disclaimer

2 2

Re: [SR-Users] best way to start kamailio at boot
by Uri Shacked 01 Feb '12

01 Feb '12

On Wed, Feb 1, 2012 at 4:26 PM, Uri Shacked <uri.shacked(a)gmail.com> wrote: > Hi, > > when i reboot the machine, it starts with 32Mb and i do not understand > why... > so, does it use a startup file that i am not familiar with? > what is the appropriate runlevel? how do i set it and where? > > thanks, > Uri > > > >Hi Sammy, > > > >When you use the init script, it is starting Kamailio with the correct > >arguments to get the environment you wish (-m 2048 probably). If however, > >you just call kamailio these args are not there. Im not sure why you are > >running with kamailio command directly. Why don't you add your init script > >to the appropriate runlevel for auto startup? > > > >Cheers > >Jason > > On Wed, Feb 1, 2012 at 3:06 PM, Uri Shacked <ushacked at gmail.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> > wrote: > > >* Hi again.... > *>* > *>* i installed kamailio 3.2.2 today. > *>* i used the chkconfig tool to get kamailio run at boot. > *>* i edited both /etc/init.d/kamailio and /etc/default/kamailio files to > run > *>* with 2048Mg of memory. > *>* now, when i run /etc/init.d/kamailio start, it starts with the right > *>* amount of memory. > *>* when i rum "kamailio" only, it starts with 32Mb.... > *>* > *>* i use centos 6.2 64bit. > *>* any ideas? > *>* > *>* > *>* > *>* > *>* > *>* On Tue, Oct 25, 2011 at 1:18 PM, Uri Shacked <ushacked at gmail.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> > wrote: > *>* > *>>* Hi. > *>>* I already checked it all. The kamailio start and stop command did not > *>>* work. The init script i used was for debian. Now i use another > script, > *>>* for centos. It works. I add the kamailio proccess with chkconfig > tool. > *>>* Thanks. > *>>* > *>>* On 10/25/11, Sammy Govind <govoiper at gmail.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> > wrote: > *>>* > Thanks Alex, > *>>* > > *>>* > Hey Uri, can you check if you've /etc/init.d/kamailio file exists > - open > *>>* > that file if exists and cross verify few parameters. > *>>* > Few of these are as follows > *>>* > > *>>* > PATH=/sbin:/bin:/usr/sbin:/usr/bin > *>>* > DAEMON=*/usr/local/sbin/kamailio* > *>>* > NAME=kamailio > *>>* > DESC=kamailio > *>>* > HOMEDIR=/var/run/ > *>>* > *PIDFILE*=$HOMEDIR/$NAME.pid > *>>* > DEFAULTS=*/etc/default/kamailio.default* > *>>* > RUN_KAMAILIO=yes > *>>* > > *>>* > Make sure /etc/default/kamailio.default file exits. here are the > *>>* contents of > *>>* > my file. > *>>* > > *>>* > RUN_KAMAILIO=yes > *>>* > USER=kamailio > *>>* > GROUP=kamailio > *>>* > MEMORY=64 > *>>* > DUMP_CORE=no > *>>* > > *>>* > On Tue, Oct 25, 2011 at 3:38 AM, Klaus Darilion < > *>>* > klaus.mailinglists at pernau.at<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> > wrote: > *>>* > > *>>* >> Test the startupscript if they work: > *>>* >> > *>>* >> /etc/init.d/kamailio start > *>>* >> /etc/init.d/kamailio stop > *>>* >> > *>>* >> If this works, then make sure to have proper symlinks from the > *>>* >> corresponding runlevel to the startup script. Do it manually or > use the > *>>* >> chkconfig tool. > *>>* >> > *>>* >> klaus > *>>* >> > *>>* >> > *>>* >> On 24.10.2011 08:00, Uri Shacked wrote: > *>>* >> > *>>* >>> hi again, > *>>* >>> i found out that on my centos 5.4 the command start-stop-daemon > that > *>>* the > *>>* >>> /etc/init.d/kamailio uses is not recognized. > *>>* >>> it is kamailio3.2 any ideas? > *>>* >>> > *>>* >>> > *>>* >>> On Mon, Oct 24, 2011 at 3:03 PM, Uri Shacked <ushacked at > gmail.com <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> > *>>* >>> <mailto:ushacked at gmail.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>>> > wrote: > *>>* >>> > *>>* >>> Hi, > *>>* >>> what is the best way to start kamailio at boot? > *>>* >>> the example in the install is not working for me.... > *>>* >>> > *>>* >>> > * >

1 0

best way to start kamailio at boot
by Uri Shacked 01 Feb '12

01 Feb '12

Hi, what is the best way to start kamailio at boot? the example in the install is not working for me....

5 9

subscriber specific dialplan
by Bayram Karagoz 01 Feb '12

01 Feb '12

Hi, I'm trying to make when each subscriber dials 7 digits add subscriber based specific prefix to dialed number defined in subscriber or dialplan table. I know that it's possible with avp but I'm not professional about avp. could you send me a successful scenario and config example please?

1 1

redirect client to keep original ruri
by Andrew Pogrebennyk 01 Feb '12

01 Feb '12

Hello, assuming that I want to use contact from 302 response as outbound proxy but keep the original Request-URI, what should I do? Calling the revert_uri() after get_redirects() in failure_route doesn't do the trick.

3 7

save a Q850 cause reason from a reply
by Uri Shacked 01 Feb '12

01 Feb '12

Hi guys..... the header searching and saving the Q850 code didn't work. here is what i wrote in the cfg, in the failure route: *# manage failure routing cases failure_route[MANAGE_FAILURE] { xlog("L_CRIT", "failure \n"); if($avp(CAR)=="CALLOUT"){ xlog("L_CRIT", "failure in callout \n"); ##if(is_present_hf("Reason")) { xlog("L_CRIT", "failure in reason\n"); $avp(Q850)=$(hdr(Reason){param.value,cause}{s.int}); xlog("L_CRIT", "failure in reason $avp(Q850)\n"); ##}* here is the log print...... * 3(3433) CRITICAL: <script>: failure 3(3433) CRITICAL: <script>: failure in callout 3(3433) CRITICAL: <script>: failure in reason 3(3433) ERROR: <core> [lvalue.c:347]: non existing right pvar 3(3433) ERROR: <core> [lvalue.c:407]: assignment failed at pos: (524,15-524,54) 3(3433) CRITICAL: <script>: failure in reason <null>* the command *is_present_hf("Reason")) * is under ## because of tests... when it was in use, the script didn't even went in the "if", the command didn't recognize the header. ** here is the sip trace for the 486, the header is there... * Message Header Call-ID: 6994099017399497785-1328029200-13379 From: <sip:36264529;ss7-cpc=10;cpc=ordinary@172.16.10.3 ;user=phone>;tag=6994099017399497785 To: <sip:39955011@172.16.60.161;user=phone>;tag=9299942026613191738 CSeq: 1 INVITE Via: SIP/2.0/UDP 172.16.60.161;received=172.16.60.161;branch=z9hG4bK08d6.c653f346.2,SIP/2.0/UDP 172.16.10.3:5060;branch=z9hG4bK-6110080000017439-ac100a03-1 Supported: timer,100rel Reason: Q.850;cause=17;text="UserBusy" Reason Protocols: Q.850 Cause: 17(0x11)[User busy] Content-Length: 0*

2 1

Re: [SR-Users] Kamailio v3.2.2 Released
by Daniel-Constantin Mierla 01 Feb '12

01 Feb '12

On 2/1/12 11:58 AM, Uri Shacked wrote: > worked OK, thanks. > [...] ok, keep cc-ing the list for topics discussed there, so people reading the archive later know what was the final solution. Cheers, Daniel > > On Wed, Feb 1, 2012 at 11:40 AM, Uri Shacked <ushacked(a)gmail.com > <mailto:ushacked@gmail.com>> wrote: > > i will try again. > i did edit the modules.lst file with all the modules i use.... > will let you know ASAP. > > On Wed, Feb 1, 2012 at 11:33 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > you haven't compiled and installed db_mysql from 3.2.2, so it > is installed the one from 3.2.1. Be sure you include db_mysql > in the list of modules to be compiled and installed, one > option is: > > make FLAVOUR=kamailio include_modules="db_mysql" cfg > make all > make install > > Cheers, > Daniel > > > On 2/1/12 10:30 AM, Uri Shacked wrote: >> >> made install and got the following error: >> "kamailio >> loading modules under >> /usr/local/lib64/kamailio/modules_k/:/usr/local/lib64/kamailio/modules/ >> 0(21428) ERROR: <core> [sr_module.c:415]: ERROR: module >> version mismatch for >> /usr/local/lib64/kamailio/modules/db_mysql.so; core: >> kamailio 3.2.2 (x86_64/linux); module: kamailio 3.2.1 >> (x86_64/linux)" >> >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla --http://www.asipto.com <http://www.asipto.com/> > http://linkedin.com/in/miconda -- http://twitter.com/miconda > > > -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Re: [SR-Users] Kamailio v3.2.2 Released
by Uri Shacked 01 Feb '12

01 Feb '12

> made install and got the following error: > "kamailio > loading modules under > /usr/local/lib64/kamailio/modules_k/:/usr/local/lib64/kamailio/modules/ > 0(21428) ERROR: <core> [sr_module.c:415]: ERROR: module version mismatch > for /usr/local/lib64/kamailio/modules/db_mysql.so; core: kamailio 3.2.2 > (x86_64/linux); module: kamailio 3.2.1 (x86_64/linux)" >

3 2

Alternative Debian APT repository
by Daniel-Constantin Mierla 01 Feb '12

01 Feb '12

Hello, besides existing deb.kamailio.org, I added builds for Debian 5.0, 6.0 and Ubuntu 10.04 to the OpenSUSE Build Service (OBS). These repositories are built only for stable releases in 3.2.x series. Main target for them is to be sure the releases compile properly, across main Linux distributions, OBS providing an easy interface to keep updating and building until everything is ok. Anyhow, these alternative repositories may be handy to use for some of you. Details about setting APT sources are available at: http://www.kamailio.org/wiki/packages/debs#using_obs_repository Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Re: [SR-Users] [newbie] questions
by Fred Posner 01 Feb '12

01 Feb '12

On Jan 31, 2012, at 6:32 PM, Me wrote: > Hi, Fred, > > >> For the sake of argument, let's say it was a smart-ass answer-- if your post was to "be pointed in the right direction where I could get (further) help" then that next help should either be: >> >> a) the wiki >> b) a training class >> c) a consultant >> > >> I'd say it again though - this thread is for me to seek answers/advice/help/guidance - if you, the previous poster, or anyone else for that matter is unwilling or unable to provide one, then just move along - there is nothing to see here. Smart-arse comments like "get a consultant" isn't what I am looking for, nor is the reason for starting this thread on this mailing list. >> > Can you read? > >> On forums I notice that those who get a better response usually use their real name instead of "Me" with a private/alias email address. >> > If I tell you that my real name is Josh Oliveira 'Capo' De Souza would that make you feel any better? Would you like to know my ssn as well? How about my date of birth? > > If I tell you where I live would that make you go away and prevent you from posting off-topic drivel on the above thread or ask silly questions you know you won't get answers to? TA! > > Josh Josh, I'm sorry-- my response wasn't insulting whatsoever. I'm not sure what your problem is, but to me it looks like you're looking for a fight. I erred on the side of thinking you were above that-- my mistake. I bid you a good day. --fred

1 0

[newbie] questions
by Me 31 Jan '12

31 Jan '12

Apologies if any of the questions below are a bit dumb - I don't pretend to be an expert in SIP/VOIP - I am just an ordinary user looking for answers. Our current setup involves processing a small number of internal sip accounts (up to 10, no more than that) and one "public" one (with a separate registrar) in the following way: On our server we have three interfaces: eth0, eth1 and tun0. eth0 is our entry point to the public internet, eth1 faces our internal network and tun0 is a private vpn, which connects all our smartphones to the internal network (via Wifi, EDGE/2G/3G etc). This gives us the mobility. Up until now, we have been routing voip calls via a commercial (closed source), very limited, terribly outdated (Pentium code base!) and rather buggy sip proxy. I had to employ a lot of hacks on our server in order to route calls as this proxy can only listen on a single interface. It was also a nightmare to maintain. Unsurprisingly, I decided that enough is enough and I am now determined to replace it. We route calls in the following way: all machines (PCs are all Linux based) & smartphones have their own sip/voip client installed on them (also using bluetooth). Internal calls are routed via the proxy between ourselves either on the internal net (eth1), or between the vpn and eth1 (eth1<->tun0). External calls (going out, i.e. outbound) are routed externally to our registrar, using a single separate voip account, via eth1<->eth0 or tun0<->eth0. As I am now looking to replace our proxy, I looked at Kamailio, but was soon completely overwhelmed by it (no offence intended, it was just too much to take at first). I would appreciate if any of you could give me a hand, or at least point me in the right direction, with the following issues: I presume I could configure Kamailio to listen on more than one interface and act as a proxy. How do I do that, so that it listens on all 3 interfaces and proxies requests in the following way: - calls made to <userX>@ourdomain.net to be routed internally via eth1 (internal net) or tun0 (private vpn); - calls made to anybody else to be routed externally via eth0 (public) using the separate "public" sip account with our external registrar; - calls made to the public sip account (from outside - the "public") need to be routed to a "nominated" internall account (say <user0>@ourdomain.net); - all other (internal) calls need to be routed depending on which interface this account has been registered/logged in - either the internal net (eth1) or the private vpn (tun0 - the smartphones). Obviously, calls need to be received (and routed properly) from all 3 interfaces. Is all of this possible with Kamailio? I want to avoid unnecessary complexities of the setup (as I already mentioned above - I am just a user and by no means an expert in sip/voip) and do not want to deploy something I do not need - I need to keep the memory footprint to a bare minimum, possibly without sacrificing performance. Once this is done, I would then move on to the next phase and use IM & ENUM, but this is once the above works. I looked at other alternatives, but I got very confused there as well - I couldn't figure out what exactly is the difference between, say, OpenSER, Kamailio, OpenSIPS and SIP-Router even? What is the best software to use in order to achieve the above setup? One last thing - I am a developer by trade and I am not afraid of "tweaking" things when needed. I was successful in compiling Kamailio from source (I use Fedora on all our machines) and I was pleased that I could exclude from the RPM .spec file the modules I think I did not need. I also made some modification of my own to make the database modules (mysql, postgresql and unixodbc) configurable in the same way the rest of the modules are. I could submit patches, if needed, so that these are incorporated into future releases - how do I do that? I could not do the same with OpenSIPS, however (which I also tried - out of curiosity!) - everything there seems to be lumped and compiled together regardless of whether it is needed or not. Any help as to helping me with the above issues is greatly appreciated, many thanks in advance for taking the time!

7 13

Kamailio Proxy and Asterisk
by Greg Mannie 31 Jan '12

31 Jan '12

Hello, We have been using Asterisk for sometime and over the last year have started hosting instances for our clients on a vmware platform. These virtual pbx are located on public ip addresses and each customer has their own SIP trunk arrangements with various providers. We have decided to pool our resources and would like to start aggregating traffic. From reading online I have installed Kamailio 3.1.x with Asterisk 1.6.x. On top of this I have installed Siremis 3.2 in hopes of using it as a graphical front end. I am wanting to use Kamailio as a proxy to the SIP providers and allow the Asterisk to register only with Kamailio. Is there a link to some example configs with a tutorial pertaining to this type of deployment? If this is a good solution, it is my intention to recommend a consultant be hired to assist with a production server. I just need to learn allot more, before I can recommend Kamailio for the job. Do you think it's a good fit? Thank you for your kind responses, Greg

3 2

Enable application/pidf+xml
by Vogt, Jonathan 31 Jan '12

31 Jan '12

Hello all, I'm trying to set up kamailio as a sip server with presence support for Bravis. So far I got the call processing running on either openser-1.3.2-3build1 or 3.2.0+lucid1 running. Problem is the notification is not using Content-Type: application/pidf+xml My openser config is at http://pastebin.ca/2106401 My kamailio config is pretty much a stock configuration. It is at http://pastebin.ca/2106404 I enabled mysql and presence. I also tried using x-lite but there is no presence information either. modparam("presence_xml", "force_active", 1) did not seem to make a difference. (I cleaned my watchers table) I can see the database has the information but the notify packets are just like this: NOTIFY sip:2001@10.0.76.0:56239 SIP/2.0 Via: SIP/2.0/UDP 10.0.10.23;branch=z9hG4bKcb76.b2dab2d4.0 To: sip:2001@sip.intern.domain;tag=90cd05fe2b0619108127da16a6e721e1 From: sip:2002@sip.intern.domain;tag=10.785.1327490724.1 CSeq: 22 NOTIFY Call-ID: 90cd05fe-2b06-1910-9660-da16a6e721e1 Content-Length: 0 User-Agent: OpenSER (1.3.2-notls (i386/linux)) Max-Forwards: 70 Event: presence Contact: <sip:10.0.10.23:5060> Subscription-State: active;expires=300 Where is my mistake. As I'm new to kamailio please bear with me, if I just didn't find the right switch. Thanks Jonathan Vogt

4 13

IRC devel meeting
by Daniel-Constantin Mierla 31 Jan '12

31 Jan '12

Hello, I would like to schedule an IRC meeting to discuss about the development of the project in 2012. The topics coming in my mind at this moment: - roadmap to next major release v3.3.0 - outstanding issues at this time, if any - packaging - ways to automate the process better - documentation - suggestions to improve it and attract more contributions in this field - ideas for new features - administration - web site, mailing lists, etc ... do we miss some good resource out there that should be added in the eco-system? Please add your suggestions for agenda to the wiki page: * http://www.kamailio.org/wiki/devel/irc-meetings/2012a I proposed two dates (Jan 26 and Jan 31, 15:00GMT) and made a Doodle poll for it: * http://www.doodle.com/qzteweb62sqax5r6 If you wish other options for dates, let me know. Based on results, we will announce the final date. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 3

Upgrade Kamailio from V3.1.5 to V3.2
by Phillman25 Kyriacou 31 Jan '12

31 Jan '12

Dear List I have Kamailio v3.1.5 currently installed and working, as this is currently in production, could someone please tell me what would be the safest way to upgrade to v3.2 without losing any data? Thanking you in advance for your support Phillip

1 0

new git commiter: Claudio Furrer
by Henning Westerholt 31 Jan '12

31 Jan '12

Hello, Claudio Furrer <elcaio(a)gmail.com> has now git commit access for his work on gentoo packaging. He create several patches and extensions for our gentoo package files in the last months and also maintain an own unofficial portage overlay tree at http://gpo.zugaina.org/net-misc/sip-router since some time. He just updated the gentoo packages in git master today, so if you're an gentoo user why not just give it a try - he is looking forward to your feedback. Welcome Claudio! Best regards, Henning

1 0

panning for v3.2.2
by Daniel-Constantin Mierla 31 Jan '12

31 Jan '12

Hello, I would like to package 3.2.2 on the same day with IRC devel meeting announced in the previous email (if is is happening this month) -- it is getting close to 2 months since v3.2.1. If you are aware of issues, report them on bug tracker to see what can be done before 3.2.1. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 2

xmlops/xpath access of multiple nodes
by Alex Balashov 30 Jan '12

30 Jan '12

Hi, I guess this is more of an xpath question than a Kamailio one, so forgive me. All the same, how does one deal with a document like this using $xml(...): <?xml version="1.0" encoding="utf-8" ?> <sheep> <lamb> ... </lamb> <lamb> ... </lamb> </sheep> Specifically, how do I access the second, third, Nth element with the same tag, but without knowledge of any unique identifying attributes? Is there a useful way to iterate through them in Kamailio route script? I would have thought something like this could work: $var(i) = 0; while(defined $xml(x=>xpath:/sheep/lamb[$var(i)])) { ... $var(i) = $var(i) + 1; } However, obviously, that's not it. So what's the right way to do it? My desire is to buffer data from these nodes into an AVP array. As you might guess, it's a route set over which I need to serially fork. Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 1

Re: [SR-Users] save a Q850 cause reason from a reply
by Uri Shacked 30 Jan '12

30 Jan '12

On Mon, Jan 30, 2012 at 7:39 PM, Uri Shacked <uri.shacked(a)gmail.com> wrote: > Hi, > thanks for the options. i will test it soon. > looks like a perfect solution. > > BR, > Uri >

1 0

Kamailio Crashing
by Ali Jawad 30 Jan '12

30 Jan '12

Hi I am running Kamailio and at times it just crashes, sometimes it happens once a day sometimes a few times a day, in debug mode it shows the following when it crashes http://pastebin.com/ZcQRnXmM GDB shows http://pastebin.com/02wVK0eZ GDB BT Shows http://pastebin.com/3TBAuQAp Please advice -- *Ali Jawad * *Information Systems Manager* *Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +9611373725/ext 116 FAX: +9611375554*

2 4

alias problem
by Stoyan Mihaylov 30 Jan '12

30 Jan '12

My whole configuration is: [Sip clients] < = > Kamailio 3.2 <=> Asterisk servers (behind Kamailio) Asterisk servers have only local IP addresses, and I use t_relay instead of forward. Kamailio runs on same server as rtpproxy. Everything is fine if clients connect to Kamailio with its IP address - global, or if they are behind Kamailio with local address. When clients connect to Kamailio using sip.ourcompany.com, then call (video also) is OK, but ACK and BYE do not work. BYE receives not here (404), and ACK die somewhere. I forward BYE and ACK in case when src_ip==$td to Asterisk server. If one of clients use IP - then calls initiated from it are OK (BYE/ACK - are going correctly - to Asterisk and to other client also). But calls from other client have problems with BYE and ACK. To use sip.ourcompany.com - I put: alias=sip.ourcompany.com route[ACKBYE] { #!ifdef WITH_PSTN if (is_method("BYE|ACK")) { xlog("L_ALERT","AB $rm $sht(forw=>$ft) $td"); if(src_ip==$td){ #I have to rewrite du - messages loop in Kamailio, I store in $sht(forw=>$ft) $du which I use during INVITE. $du=$sht(forw=>$ft); route(RELAY); exit; } xlog("L_ALERT","ACK,Bye Not me"); } #!endif return; }

2 2

Open Source billing
by Abid Naseem 30 Jan '12

30 Jan '12

Looking for open source billing solution for Kamailio: 1) rating 2) client interface for cdr 3) admin interface Help needed. Thanks and keep inspiring! -abid Sent from my iPhone

3 4

exec_avp size limitation
by Gustavo Garcia Bernardo 30 Jan '12

30 Jan '12

Hi folks, I'm having problems with exec_avp because is truncating the output of the script executed. Looking at the code I see exec_avp is imposing a limit of MAX_URI_SIZE (1024) but I don't understand why. The limit should be the maximum size for an avp variable in my opinion. Regards, G. Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra política de envío y recepción de correo electrónico en el enlace situado más abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at http://www.tid.es/ES/PAGINAS/disclaimer.aspx

2 1

memory leak fix on xcap_ auth.c
by laura testi 30 Jan '12

30 Jan '12

Hi Daniel, I'm sending you some memory leak fix we have found and fixed on modules/utils/xcap_ auth.c. The fixed memory leak is in pres_watcher_allowed (), and it is about the "xmlFreeDoc" that it's never called for the "xcap_tree" variable. We downloaded the source file from master branch and then we fixed it. You can find here the diff from original and new code and in attach both the original and the new files. diff -u a/modules/utils/xcap_auth.c b/modules/utils/xcap_auth.c --- a/modules/utils/xcap_auth.c +++ b/modules/utils/xcap_auth.c @@ -280,13 +280,15 @@ } node= get_rule_node(subs, xcap_tree); - if (node== NULL) + if (node== NULL){ + xmlFreeDoc(xcap_tree); return 0; - + } /* process actions */ actions_node = xmlNodeGetChildByName(node, "actions"); if (actions_node == NULL) { LM_DBG("actions_node NULL\n"); + xmlFreeDoc(xcap_tree); return 0; } LM_DBG("actions_node->name= %s\n", actions_node->name); @@ -294,6 +296,7 @@ sub_handling_node = xmlNodeGetChildByName(actions_node, "sub-handling"); if (sub_handling_node== NULL) { LM_DBG("sub_handling_node NULL\n"); + xmlFreeDoc(xcap_tree); return 0; } sub_handling = (char*)xmlNodeGetContent(sub_handling_node); @@ -302,6 +305,7 @@ if (sub_handling == NULL) { LM_ERR("Couldn't get sub-handling content\n"); + xmlFreeDoc(xcap_tree); return -1; } if (strncmp((char*)sub_handling, "block", 5) == 0) { @@ -325,10 +329,12 @@ else { LM_ERR("unknown subscription handling action\n"); xmlFree(sub_handling); + xmlFreeDoc(xcap_tree); return -1; } xmlFree(sub_handling); + xmlFreeDoc(xcap_tree); return 0; Kind Regards, laura

2 1

Kamailio TLS with intermediate CA certificates
by Daniel Pocock 30 Jan '12

30 Jan '12

I found that my TLS client was not happy because my server cert is signed by an intermediate root. A quick search in Google found other people mentioning the same problem, but no solution or documentation. I've had a quick look in the Kamailio source and I notice it is using the call: SSL_CTX_use_certificate_chain_file to load the certificate specified in tls.cfg with certificate=myserver.pem In practice, this means the intermediate certificates can be appended to myserver.pem and Kamailio will present them to the TLS client: Example: Trust heirarchy: trusted root - inter 1 - inter 2 - server.example.com.pem Construct the PEM file in this exact order: cat server.example.com.pem > chain-server.example.com.pem cat inter2.pem >> chain-server.example.com.pem cat inter1.pem >> chain-server.example.com.pem and then, in tls.cfg: certificate=chain-server.example.com.pem

4 7

Kamailio for NATted users with ICE and TURN
by Daniel Pocock 30 Jan '12

30 Jan '12

There are now a number of TURN implementations available: http://www.resiprocate.org/ReTurn_Overview http://turnserver.sourceforge.net/ and for many purposes TURN relays and ICE are much better than relying on NAT helper and rtpproxy When clients use ICE: - the client gets upset if Kamailio mangles the SDP to use rtpproxy - it is actually not necessary for the SDP to have addresses tweaked - because ICE deals with it anyway In practice, it still appears necessary for the registration server to deal with NATted contact addresses though Can anyone clarify how Kamailio should be configured for ICE clients? In other words, is it possible to have Kamailio dealing with NATted contact addresses, but not touching the SDP? To limit the scope of the question: - assume non-ICE clients do not need to be supported at all (just ignore them, although a nice cfg would scan all SDP and send a helpful error back if it sees a 192.168 address and no a=ice* attributes) - assume clients are connecting with TLS over TCP, so there is no risk that some intermediate device like a router is mangling the SIP headers or SDP

2 2

cr_dump_routes buffer too small
by Mino Haluz 30 Jan '12

30 Jan '12

Is there any way how to print it this way ? sercmd -s unixs:/tmp/kamailio_ctl mi_fifo cr_dump_routes error: 500 - Internal server error processing 's': buffer too small (overflow) (-2)

1 0

RPM packages - repositories update
by Daniel-Constantin Mierla 30 Jan '12

30 Jan '12

Hello, the repositories with build RPMs done via OpenSUSE Build Services (OBS) were updated. Fedora 14 is no longer offered by OBS, instead I added support for Fedora 15 and 16. Also, there are now dedicated builds for RedHat 6 and OpenSUSE 12.1. Links to repos are provided at: * http://www.kamailio.org/wiki/packages/rpms Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

save a Q850 cause reason from a reply
by Uri Shacked 30 Jan '12

30 Jan '12

Hi, the gateway i use for sending call to pstn has an option of sending a header with Q850 reason. it looks like this: " Reason: Q.850;cause=17;text="UserBusy" Reason Protocols: Q.850 Cause: 17(0x11)[User busy] " now, i need to save the reason for reports. the reason changes and i am trying to find a way to parse it and save the code only. usualy i use the "search" function to get text from headers that are the gateway's inventions :-) here i can find the text "Q850....." but how do i get the cause code? i guess i cab write something long with textops and parsing. but, is there a better and nicer way to perform it? thanks, Uri

3 2

Registrar ignoring socket
by Spencer Thomason 30 Jan '12

30 Jan '12

Hello, I'm working on building a HA setup with separate registrar and proxy servers. The registrar server writes to a master database that is replicated to a local database on each of the proxy servers. A load balancer then sits in front of the proxys. When routing to a registered user on the proxys I receive an error message that the socket is unknown and ignored (obviously because its on another machine). Is this just a warning message or are there any consequences to doing it this way? Is my topology flawed and would there be a better way? Thanks for your input, Spencer Connected by DROID on Verizon Wireless

3 3

Re: [SR-Users] Kamailio Redundancy examples/pointers required
by Javier Gallart 30 Jan '12

30 Jan '12

Hello the net.ipv4.ip_nonlocal_bind = 1 options works fine. We have a very basic redundancy level, but it fits our current needs; sometimes it's even harder to troubleshoot the redundancy failures than the actual failures. We have 2 identical servers running kamailio and listening to the same ip addresses/ports (you need to enable the sysctl option for that). A heartbeat process controls in what machine the actual ip addresses are active. The DB is in one of the servers but we update the shared memory in both. For us having the DB down for a while is not critical because all the data is in memory. When one servers goes down the other one takes over immediately. It's a very comfortable setup also for doing upgrades in the configuration in a controlled manner. Hope it helps Regards Javi > ------------------------------ > > Message: 6 > Date: Thu, 26 Jan 2012 12:13:18 +0500 > From: Sammy Govind <govoiper(a)gmail.com> > Subject: Re: [SR-Users] Kamailio Redundancy examples/pointers required > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <CAJUJwtjMGTuNe1Cu5+BDthgj0xrSJrBM03XzcjZhWJ8Ar7=onQ(a)mail.gmail.com > > > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > Thanks for taking out time for this. really appreciate your ideas. Here are > few things I've found useful so far. > > http://kamailio.org/events/2011-fosdem/p_usrloc.pdf > This looks like one thing that one could have to ensure more reliability > but we do need other application level and physical level failover > techniques additionally. > > For using any technique like Linux-HA, Heartbeat, keepalived, UCARP or any > floating/virtual IP techniques I really hope that the sysctl option > mentioned in b/m thread really works. > http://www.mail-archive.com/sr-users@lists.sip-router.org/msg03714.html > http://old.nabble.com/Kamilio-Failover-Options-td21947502.html > DNS SRV is one option. Using another Kamailio on top of two Kamailio is > again a SPOF so I think its of no use at all. > > I hope to get a very reliable and working solution out of these forums. > > Regards, > Sammy > > On Wed, Jan 25, 2012 at 9:49 PM, Andreas Granig <agranig(a)sipwise.com> > wrote: > > > Hi Sammy, > > > > On 01/25/2012 02:08 PM, Sammy Govind wrote: > > > I've been looking for ways to create a redundant Kamailio cluster. I've > > > googled alot but haven't got any concrete or final wording on any one > > > solution that'll just work perfectly. The basic requirement is that in > > > > HA is never really easy :) > > > > > case of Kamailio application failure or in case of physical machine > > > error the whole setup just starts working on secondary server. > > > > > > Please suggest anything whichever you guys are using. Any reference > URLs > > > would be very much appreciated. > > > > For 1+1 setups, you can use a master/master replication (people also use > > drbd to sync the whole table space, not sure how good that works) on db > > level, use write-back mode in kamailio to make sure your registrations > > are written to DB immediately, then use heartbeat to do the IP/process > > fail-overs on network/power failures. On top, you can do manual checks > > to trigger a fail-over in case of high-level failures. I guess this is > > the typical and most straight-forward scenario. Your mileage might vary, > > depending on specific environment variables (e.g. you might want to > > announce your active server via RIP/OSPF using quagga instead of the > > more common gratuitous arp if your servers are geographically split). > > > > For scaling larger, put a kamailio instance acting as load-balancer in > > front of your proxies and scale them by providing some sharding > > information to the lb. > > > > Also the new db_cassandra backend in master branch looks pretty > > promising when it comes to HA/scaling. Maybe someone can provide some > > best practices regarding that as well? > > > > Andreas > > > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > > sr-users(a)lists.sip-router.org > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > >

2 2

Using forward(); with PVs
by Spencer Thomason 28 Jan '12

28 Jan '12

Hello, I'm experiencing some difficulty using forward with PVs. If the IP and port are specified directly everything works but if I use PVs for the values startup fails with a configuration error. What I'm trying to do is statelessly forward an OPTIONS request to a B2BUA that sits behind Kamailio without hard coding the address into the configuration file. Is there a better way to accomplish this? Thanks, Spencer Connected by DROID on Verizon Wireless

2 2

Selects vs PVs
by Spencer Thomason 28 Jan '12

28 Jan '12

Hello all, I'm currently reworking our routing scripts in preparation for an upgrade to 3.2.2 from 3.1.5. I'm taking this time to perform a general cleanup an optimization. In regard to selects vs PVs, is there any reason to choose one or the other select over a PV when both exist for the needed data? Thanks, Spencer

2 1

AVPs vs Variables
by Spencer Thomason 27 Jan '12

27 Jan '12

Hello all, I'm a little confused as to the difference between avps and vars and when I would use one or the other. I'm currently using sqlops to populate a few avps. Would vars be more suitable? Thanks, Spencer Connected by DROID on Verizon Wireless

3 3

pua_usrloc, PUBLISH not send when contact is deleted or expires
by Giacomo Vacca 27 Jan '12

27 Jan '12

Hi all, I've made a clean installation of kamailio 3.2.0 from 'http://deb.kamailio.org/kamailio32 squeeze main' repository, on a squeeze debian machine (2.6.32-5-686). Installed also kamailio-presence-modules, same version. I'm just using the default kamailio.cfg that comes with the package, with these minimal additions: loadmodule "db_text.so" loadmodule "pua.so" loadmodule "pua_usrloc.so" ... modparam("pua", "db_url", "text:///usr/share/kamailio/dbtext/kamailio") modparam("pua_usrloc", "default_domain", "192.168.142.130") Inside the registration management logic I've added pua_set_publish() right before saving the location: route[REGISTRAR] { if (is_method("REGISTER")) { ... pua_set_publish(); if (!save("location")) sl_reply_error(); exit; } } When I first register (XLite 4.1 for Windows), pua_usrloc correctly gets the callback and is able to generate a PUBLISH request through pua. But when there's an un-registration, or when the contact expires, even if the callbacks are correctly fired, the PUBLISH (which I'd expect with status 'closed') is not generated. >From the logs (debug level 4), an un-registration: 8(7652) DEBUG: usrloc [ul_callback.h:89]: contact=0xb50ae7cc, callback type 4/4, id 3 entered 8(7652) DEBUG: pua_usrloc [ul_publish.c:216]:DELETE type 8(7652) DEBUG: pua_usrloc [ul_publish.c:255]: uri= sip:20701090@192.168.142.130 8(7652) DEBUG: pua_usrloc [ul_publish.c:66]: publ: 8(7652) DEBUG: pua_usrloc [ul_publish.c:67]: uri= sip:20701090@192.168.142.130 8(7652) DEBUG: pua_usrloc [ul_publish.c:68]: id=UL_PUBLISH.NmY5Zjc2YjcwNzEwNTk4YTY1NzA2Y2Y0MWQyMGRhOGY. 8(7652) DEBUG: pua_usrloc [ul_publish.c:69]: expires= 0 8(7652) DEBUG: pua [send_publish.c:403]: pres_uri=sip:20701090@192.168.142.130 8(7652) DEBUG: pua [hash.c:121]: core_hash= 444 8(7652) DEBUG: pua [hash.c:174]: record not found 8(7652) DEBUG: pua [send_publish.c:444]: insert type 8(7652) DEBUG: pua [send_publish.c:448]: UPDATE_TYPE and no record found 8(7652) DEBUG: pua [send_publish.c:454]: request for a publish with expires 0 and no record found 8(7652) DEBUG: sl [sl.c:278]: reply in stateless mode (sl) As an additional information, the pua table in /usr/share/kamailio/dbtext/kamailio/pua has not been populated. The AOR was correctly displayed with 'ul show' while the registration was valid. Any suggestion on what am I be doing wrong? Would it be worth testing the same scenario using mysql as DB? Thanks in advance for your time. Giacomo Truphone Limited is a limited liability company registered in England & Wales, registered office: 5 New Street Square, London EC4A 3TW. Registered No. 04187081. VAT No. GB 851 5278 19. Tru is a brand name of Truphone and is a Truphone Communications Service. Truphone is a trading name for a number of distinct legal entities that operate in combination. www.truphone.com<http://www.truphone.com>. This e-mail, and any attachment(s), may contain information which is confidential and/or privileged, and is intended for the addressee only. If you are not the intended recipient, you may not use, disclose, copy or distribute this information in any manner whatsoever. If you have received this e-mail in error, please contact the sender immediately and delete it.

2 11

rpid to from header.
by David 27 Jan '12

27 Jan '12

One of my upstream gateways does not support rpid, so I want to rewrite the from header from the rpid. is just a matter if setting $fU = $rpid? Thanks.

3 3

dbcassandra - 483
by David 27 Jan '12

27 Jan '12

Anca, for some strange reason I am getting "SIP/2.0 483 Too Many Hops" back when I try to register. I am running kamaikio -EEEEEE and I see we get authenticated correctly but then lots of other info is spit out very quickly and I am having a hard time trying to understand it. I think it may have to do with the location table. But I could be wrong. Any ideas how to debug? Thank you.

2 3

Radius Generates Core Dump
by Omar Labs 188 Gmail 27 Jan '12

27 Jan '12

Hello: I was testing the Server Kamailio 3.2.1 with Accounting and Radius, it crash generating a Big Core Dump, and freezing up, If I remove the Radius module from configuration, accounting via the database works fine. Did anybody else have the same problem. Appears a Memory leak. Thanks

2 1

Kamailio for Wholesale Termination
by Abid Naseem 27 Jan '12

27 Jan '12

Dear *All*, I am new to voip world and interested in knowing if I we can use *KAMAILIO + * *SIREMIS *for wholesale voip termination? We buy direct routes and give it to resellers..currently using asterisk. appreciate your support! keep inspiring. -- Regards, *Abid Naseem* +44 (793) 741 5043 +1 (415) 562 6333

2 1

Kamailio with Private IP - ACK for 200 OK problem
by Krishna Kurapati 26 Jan '12

26 Jan '12

Hi, I am having trouble with setting Record Route when Kamailio is behind a Load balancer. The Load balancer acts as a static NAT where the public IP of the load balancer is mapped to private IP of the Kamailio. This is creating issues with ACK for 200 OK since Kamailio is stuffing the 200 OK with private-ip in the Record-Route. The UAC looks like trying to send ACK back to this private IP which is not reachable. Is there a configuration option to let kamailio use Public IP when setting record-route in 200 OK? Thanks for any help... Krish Kura

2 1

Memory usage problem in xmpp modules
by laura testi 26 Jan '12

26 Jan '12

Hi, we are using the XMPP gateway modules: • PUA • PUA_XMPP • XMPP The behavior is ok but since the modules are started, the memory usage is getting increased and now also the swap memory area is used. I send to you the following information: 1. Numer of records from PUA and xmpp_presentity tables, they are quite stable: SQL> select count(*) from pua; COUNT(*) ---------- 10218 SQL> select count(*) from xmpp_presentity; COUNT(*) ---------- 754 2. list of PKG fragments with gdb : $1 = {size = 40, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $2 = {size = 32, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $3 = {size = 40, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $4 = {size = 40, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $5 = {size = 40, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $6 = {size = 104, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $7 = {size = 32, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $8 = {size = 32, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $9 = {size = 56, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $10 = {size = 96, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $11 = {size = 56, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $12 = {size = 112, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $13 = {size = 24, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $14 = {size = 24, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $15 = {size = 40, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $16 = {size = 40, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160} $17 = {size = 40, u = {nxt_free = 0x0, is_free = 0}, file = 0x2b657aa12c00 "presence: utils_func.h", func = 0x2b657aa1aed0 "uandd_to_uri", line = 66, check = 4042322160}…………. 3. Output of sercmd cfg.set_now_int core mem_dump_pkg : ……………….. 10969. N address=0xb1b320 frag=0xb1b2f0 size=104 used=1 qm_status: alloc'd from presence: utils_func.h: uandd_to_uri(66) qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed qm_status: 10970. N address=0xb1b3e8 frag=0xb1b3b8 size=32 used=1 qm_status: alloc'd from presence: utils_func.h: uandd_to_uri(66) qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed qm_status: 10971. N address=0xb1b468 frag=0xb1b438 size=32 used=1 qm_status: alloc'd from presence: utils_func.h: uandd_to_uri(66) qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed qm_status: dumping free list stats : qm_status: hash= 1. fragments no.: 4, unused: 0 bucket size: 8 - 8 (first 8) qm_status: hash= 2. fragments no.: 8, unused: 0 bucket size: 16 - 16 (first 16) qm_status: hash= 3. fragments no.: 6, unused: 0 bucket size: 24 - 24 (first 24) qm_status: hash= 4. fragments no.: 1, unused: 0 bucket size: 32 - 32 (first 32) qm_status: hash= 6. fragments no.: 3, unused: 0 bucket size: 48 - 48 (first 48) qm_status: hash= 8. fragments no.: 8, unused: 0 bucket size: 64 - 64 (first 64) qm_status: hash= 9. fragments no.: 6, unused: 0 bucket size: 72 - 72 (first 72) qm_status: hash= 12. fragments no.: 5, unused: 0 bucket size: 96 - 96 (first 96) qm_status: hash= 14. fragments no.: 6, unused: 0 bucket size: 112 - 112 (first 112) qm_status: hash= 17. fragments no.: 32, unused: 0 bucket size: 136 - 136 (first 136) qm_status: hash= 19. fragments no.: 1, unused: 0 bucket size: 152 - 152 (first 152) qm_status: hash= 21. fragments no.: 1, unused: 0 bucket size: 168 - 168 (first 168) qm_status: hash= 29. fragments no.: 1, unused: 0 bucket size: 232 - 232 (first 232) qm_status: hash= 31. fragments no.: 2, unused: 0 bucket size: 248 - 248 (first 248) qm_status: hash= 33. fragments no.: 2, unused: 0 bucket size: 264 - 264 (first 264) qm_status: hash= 36. fragments no.: 1, unused: 0 bucket size: 288 - 288 (first 288) qm_status: hash= 57. fragments no.: 1, unused: 0 bucket size: 456 - 456 (first 456) qm_status: hash= 60. fragments no.: 1, unused: 0 bucket size: 480 - 480 (first 480) qm_status: hash= 76. fragments no.: 709, unused: 0 bucket size: 608 - 608 (first 608) qm_status: hash= 88. fragments no.: 1, unused: 0 bucket size: 704 - 704 (first 704) qm_status: hash= 116. fragments no.: 1, unused: 0 bucket size: 928 - 928 (first 928) qm_status: hash= 145. fragments no.: 1, unused: 0 bucket size: 1160 - 1160 (first 1160) qm_status: hash= 216. fragments no.: 2, unused: 0 bucket size: 1728 - 1728 (first 1728) qm_status: hash= 2049. fragments no.: 1, unused: 0 bucket size: 16384 - 32768 (first 22032) qm_status: hash= 2057. fragments no.: 1, unused: 0 bucket size: 4194304 - 8388608 (first 5982824) Is it a memory leak in one of these modules? Thanks in advance and kind regards, laura

5 6

get the kamailio up time and other params from the cfg file.
by Uri Shacked 26 Jan '12

26 Jan '12

Hi, is there a way to get the total up time of kamailio from the cfg file? like $stat(used_size) gives me the stat of the shared memory used size..... and if i am in it... is there a way to get the dispatcher status as well? i am trying to find a way to export these params with out the need to pull them using kamctl..... any ideas? thanks, Uri

4 3

Re: [SR-Users] detect the direction of the BYE
by Uri Shacked 26 Jan '12

26 Jan '12

On Thu, Jan 26, 2012 at 9:24 AM, Uri Shacked <uri.shacked(a)gmail.com> wrote: > Hi, > > i would like to detct the direction of the BYE. was it drom the caller or > the callee? > in the ACC module i can do that if i analyze the to and from tags... > how do i do it (in a nice way) in the cfg file? > > cheers. >

2 1

Call Control Behavior
by Reda Aouad 26 Jan '12

26 Jan '12

Hi, My question concerns Call Control. When a SIP call's timer ends, Call Control uses dlg_end_dlg MI command of the dialog module to send a BYE to the UA. Can I intercept the BYE message, drop it, change it and send instead an INVITE or a REDIRECT using (maybe) the UAC module? The INVITE/REDIRECT would connect the UA to a media gateway to announce a message when the session ends. Any ideas? This also leads to thoughts on accounting and rating. In the case of sending and INVITE, the rating of the call becomes inaccurate since there's no BYE when the call ends, but only after the media gateway's announcement ends. In the case of a REDIRECT, the UA sends a BYE to the established call before the INVITE to the media gateway, and thus rating remains totally accurate. What do you think? Thanks RA

1 0

Kamailio Redundancy examples/pointers required
by Sammy Govind 26 Jan '12

26 Jan '12

Hello list, I've been looking for ways to create a redundant Kamailio cluster. I've googled alot but haven't got any concrete or final wording on any one solution that'll just work perfectly. The basic requirement is that in case of Kamailio application failure or in case of physical machine error the whole setup just starts working on secondary server. Please suggest anything whichever you guys are using. Any reference URLs would be very much appreciated. Thanks and Regards, Sammy

2 2

Gentoo ebuild pkg v3.2.1
by caio 25 Jan '12

25 Jan '12

Hello, I want to announce the availability of an ebuild for version 3.2.1 on Gentoo Portage Overlays. Here the git link: http://goo.gl/JXLQ4 The unofficial overlays portage website: * http://gpo.zugaina.org/net-misc/sip-router Remember this is not an official ebuild, and neither it's not at official gentoo portage tree. Henning, can you add/update it at sip-router git repository (pkg/ser/gentoo)? Thank you and best regards, Claudio

1 0

access to the "Available Functions" in the modules & the way to bridge a call based on username and domain
by Xi Liu 25 Jan '12

25 Jan '12

Hi, I'm new to kamailio, have some questions here hope anyone can help! Question 1, I'm trying to use some of the functions in usrloc module, but for some reason if I use those from "Available Functions" part, the kamailio won't start and give an error of "unknown command <ul_get_ucontact>, missing loadmodule?" The usrloc module is definitely loaded, and it will start as long as I avoid using those "Available Functions". I tried some other modules it will lead to same error. Question 2, What is the simplest way to bridge a call based on username and domain? I was trying to make a simple experiment I used to do in Freeswitch, ========================== Have 1000,1001,1002 registered, When 1000 calls 9999, the calls will be bridged to 1001 or 1002 based on different time period. ========================== While in Freeswitch, it would be as simple as "bridge xxxx(a)domain.com", and then freeswitch will lookup contact information and forward the call. But in kamailio, I only succeed in forwarding calls by change the $du to the exact contact uri, which is "sip:xxxx@ip-of-the-phone:port", the call r_relay(). In order to do this, I need call a perl script to lookup the location database to find a match… I'm 99% sure I'm doing wrong here… there must be a way way easier than that, but so far I can't find it…. Please help… Thanks, Xi

2 2

UAC module + customize user_agent
by Douglas Adami 25 Jan '12

25 Jan '12

Hello everyone, I'm new here and I go to test my settings, but I need to authenticate a remote server, I was reading about the UAC module, one has configuration examples? I can also customize the user_agent authentication, because the service only allows specific user_agents? Thanks in advance! Regards, Doug

2 1

Re: [SR-Users] Radius and kamailio: avp unknown attribute
by DucDung 25 Jan '12

25 Jan '12

Hi man, I have the same problem like you posted in sip-router.org. The problem is that : rc_avpair_new: unknown attribute 5 I see that u have already resolved that problem before. So can u help me to fix it. Thank you very much. Please contact me follow these emails : ducdung0909(a)yahoo.com or ducdung0909(a)gmail.com .

2 1

Images missing
by sjs205.linux＠googlemail.com 25 Jan '12

25 Jan '12

Hello all, Just reading through some of the docs on the website; is it possible to get the images in this file: http://sip-router.org/docbook/sip-router/branch/master/sip/sip_introduction… An excellent introduction into SIP :D Kind Regards, Steven

2 1

Re: [SR-Users] Homer SIP Capture Setup Question
by JR Richardson 24 Jan '12

24 Jan '12

> On 01/16/2012 09:11 PM, JR Richardson wrote: > > > So my question; what is the difference between a drop; and exit; > > within the on_reply route > > Remember that an onreply_route is a callback that allows you to > intercept a reply, but the reply is forwarded by default afterward (if > using stateful forwarding). This is different from a request route, > where it's up to you to marshal everything, and to explicitly forward > the message. > > Thus, exiting an onreply_route has null effect. It doesn't do > anything to stop the reply from being forwarded, because forwarding is > the behaviour--the onreply_route is just a hook. In contrast, 'drop' > explicitly drops the message and prevents further forwarding of the > reply from taking place. > > This is different than in a request route, where exiting precludes any > forwarding action, because forwarding action must be taken explicitly > -- e.g. t_relay(). > > -- > Alex Balashov - Principal Aaaah, concise explanation, once again, thanks for turning on the light bulb Alex. JR

2 1

Is t_on_failure with uac_req_send possible?
by Nathan Angelacos 24 Jan '12

24 Jan '12

I'm trying to arm the failure route for a message sent via uac_req_send - similar to: { $uac_req(method)="OPTIONS"; $uac_req(ruri)="sip:kamailio.org"; $uac_req(furi)="sip:kamailio.org"; $uac_req(turi)="sip:kamailio.org"; t_on_failure("UAC_FAIL"); uac_req_send(); } failure_route("UAC_FAIL") { xlog ...... } If there is no response from the remote server, kamailio resends until the timers expire; but the failure route is not run. Similarly, if a 404 is sent back from the remote server, the failure route is not run. Am I missing something obvious? Thanks

3 3

(no subject)
by Florian Meister 24 Jan '12

24 Jan '12

unsubscribe

2 1

unable to find dialog for ACK with route param
by Uri Shacked 24 Jan '12

24 Jan '12

Hi, i use the dialog module. i create the dialog using dlg_manage() right after the invite is recieved. in a case where the final destination rejects the call (486 busy), i get the following error: " WARNING: dialog [dlg_handlers.c:1054]: unable to find dialog for ACK with route param....." i guess the dialog module deletes the dialog on the 486 reply and gives me this erro when an ACK for the 486 is recieved from the caller... no a bug issue, but, is there a way to overcome it? thanks, Uri

2 2

Problem with www_challenge
by Stoyan Mihaylov 23 Jan '12

23 Jan '12

Hi, If I am using IP address, I have no problems. If I use domain name (pointing to same IP address) then phones do not register. I have couple of hardware and software SIP phones and when I put name of domain (stribogkonsult.com) - then no registration, if I put IP - no problems. I put IP, and names in domain table. Effect is - when I use domain name, there are bulk of messages in syslog - like messages reenter kamailio again and again. xlog("Auth attemps $au\n"); - I receive many times if I use domain name, if I use IP - once only. Problem can be somewhere else of course. But everything else work as I expect. route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER")) { if (!www_authorize("$td", "subscriber")) { xlog("RegisterChalange $au $td\n"); www_challenge("$td", "1"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } xlog("Auth attemps $au\n"); } else { #!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif # authenticate if from local subscriber if (from_uri==myself) { xlog("Register MySelf\n"); if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$fU || $au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } if ($au!=$rU) { sl_send_reply("403","Forbidden R-URI"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; }

2 2

db_cassandra
by David 23 Jan '12

23 Jan '12

Using thrift/0.8.0/thrift-0.8.0.tar.gz I get this when trying to build db_cassandra. In file included from /usr/local/include/thrift/protocol/TProtocol.h:23, from /usr/local/include/thrift/protocol/TBinaryProtocol.h:23, from dbcassa_base.cpp:36: /usr/local/include/thrift/transport/TTransport.h:34: error: expected constructor, destructor, or type conversion before ‘readAll’ /usr/local/include/thrift/transport/TTransport.h:107: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TTransport.h:111: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TTransport.h:124: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TTransport.h:128: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TTransport.h:139: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TTransport.h:156: error: ISO C++ forbids declaration of ‘uint8_t’ with no type /usr/local/include/thrift/transport/TTransport.h:156: error: expected ‘,’ or ‘...’ before ‘*’ token /usr/local/include/thrift/transport/TTransport.h:160: error: ISO C++ forbids declaration of ‘uint8_t’ with no type /usr/local/include/thrift/transport/TTransport.h:160: error: expected ‘,’ or ‘...’ before ‘*’ token /usr/local/include/thrift/transport/TTransport.h:172: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TTransport.h:214: error: ISO C++ forbids declaration of ‘uint8_t’ with no type /usr/local/include/thrift/transport/TTransport.h:214: error: expected ‘;’ before ‘*’ token /usr/local/include/thrift/transport/TTransport.h:218: error: expected ‘;’ before ‘virtual’ /usr/local/include/thrift/transport/TTransport.h:218: error: ISO C++ forbids declaration of ‘uint8_t’ with no type /usr/local/include/thrift/transport/TTransport.h:218: error: ‘uint8_t’ declared as a ‘virtual’ field /usr/local/include/thrift/transport/TTransport.h:218: error: expected ‘;’ before ‘*’ token /usr/local/include/thrift/transport/TTransport.h:231: error: expected ‘;’ before ‘void’ /usr/local/include/thrift/transport/TTransport.h:231: error: ‘uint32_t’ has not been declared /usr/local/include/thrift/transport/TTransport.h:235: error: ‘uint32_t’ has not been declared /usr/local/include/thrift/transport/TTransport.h: In member function ‘void apache::thrift::transport::TTransport::write(int)’: /usr/local/include/thrift/transport/TTransport.h:158: error: ‘buf’ was not declared in this scope /usr/local/include/thrift/transport/TTransport.h:158: error: ‘len’ was not declared in this scope In file included from /usr/local/include/thrift/protocol/TBinaryProtocol.h:23, from dbcassa_base.cpp:36: /usr/local/include/thrift/protocol/TProtocol.h: At global scope: /usr/local/include/thrift/protocol/TProtocol.h:184: error: expected constructor, destructor, or type conversion before ‘skip’ /usr/local/include/thrift/protocol/TProtocol.h:305: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:309: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:312: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:314: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:316: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:320: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:322: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:324: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:328: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:330: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:333: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:335: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:338: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:340: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:342: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:344: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:346: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:348: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:350: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:352: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:354: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:356: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:363: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:369: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:374: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:379: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:386: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:391: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:396: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:403: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:408: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:413: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:418: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:423: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:428: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:433: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:438: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:443: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:448: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:453: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:458: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:463: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:472: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:476: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:478: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:480: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:482: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:486: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:488: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:492: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:494: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:497: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:499: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:502: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:504: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:506: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:508: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:510: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:512: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:514: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:516: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:518: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:520: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:522: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:529: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:534: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:539: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:544: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:551: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:556: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:561: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:566: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:571: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:576: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:581: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:586: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:591: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:596: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:601: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:606: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:611: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:616: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:621: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:631: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:639: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TProtocol.h:643: error: ‘uint32_t’ does not name a type In file included from /usr/local/include/thrift/protocol/TBinaryProtocol.h:24, from dbcassa_base.cpp:36: /usr/local/include/thrift/protocol/TVirtualProtocol.h:42: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:52: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:57: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:63: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:68: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:78: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:83: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:91: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:96: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:103: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:108: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:115: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:120: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:126: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:132: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:138: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:144: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:150: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:156: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:162: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:168: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:174: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:184: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:190: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:196: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:201: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:211: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:216: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:221: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:231: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:236: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:243: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:248: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:255: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:260: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:266: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:272: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:278: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:284: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:290: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:296: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:302: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:308: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:329: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:336: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:341: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:345: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:349: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:356: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:360: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:364: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:370: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:374: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:379: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:383: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:388: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:392: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:396: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:400: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:404: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:408: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:412: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:416: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:420: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:428: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:435: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:439: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:443: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:447: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:454: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:458: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:464: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:468: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:473: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:477: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:482: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:486: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:490: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:494: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:498: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:502: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:506: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:510: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:514: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:518: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:522: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:536: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TVirtualProtocol.h:548: error: ‘uint32_t’ does not name a type In file included from dbcassa_base.cpp:36: /usr/local/include/thrift/protocol/TBinaryProtocol.h:92: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:96: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:99: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:101: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:103: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:107: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:109: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:111: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:115: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:117: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:119: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:121: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:123: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:125: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:127: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:129: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:131: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:133: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:135: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:137: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:139: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:146: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:150: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:152: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:154: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:156: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:160: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:162: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:166: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:168: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:170: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:172: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:174: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:176: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:180: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:182: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:184: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:186: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:188: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:190: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:192: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:195: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/protocol/TBinaryProtocol.h:208: error: ISO C++ forbids declaration of ‘uint8_t’ with no type /usr/local/include/thrift/protocol/TBinaryProtocol.h:208: error: expected ‘;’ before ‘*’ token /usr/local/include/thrift/protocol/TBinaryProtocol.h: In constructor ‘apache::thrift::protocol::TBinaryProtocolT<Transport_>::TBinaryProtocolT(boost::shared_ptr<X>)’: /usr/local/include/thrift/protocol/TBinaryProtocol.h:51: error: class ‘apache::thrift::protocol::TBinaryProtocolT<Transport_>’ does not have any field named ‘string_buf_’ /usr/local/include/thrift/protocol/TBinaryProtocol.h: In constructor ‘apache::thrift::protocol::TBinaryProtocolT<Transport_>::TBinaryProtocolT(boost::shared_ptr<X>, int32_t, int32_t, bool, bool)’: /usr/local/include/thrift/protocol/TBinaryProtocol.h:65: error: class ‘apache::thrift::protocol::TBinaryProtocolT<Transport_>’ does not have any field named ‘string_buf_’ /usr/local/include/thrift/protocol/TBinaryProtocol.h: In destructor ‘apache::thrift::protocol::TBinaryProtocolT<Transport_>::~TBinaryProtocolT()’: /usr/local/include/thrift/protocol/TBinaryProtocol.h:69: error: ‘string_buf_’ was not declared in this scope In file included from /usr/local/include/thrift/protocol/TBinaryProtocol.h:24, from dbcassa_base.cpp:36: /usr/local/include/thrift/protocol/TVirtualProtocol.h: At global scope: /usr/local/include/thrift/protocol/TVirtualProtocol.h: In instantiation of ‘apache::thrift::protocol::TVirtualProtocol<apache::thrift::protocol::TBinaryProtocolT<apache::thrift::transport::TTransport>, apache::thrift::protocol::TProtocolDefaults>’: /usr/local/include/thrift/protocol/TBinaryProtocol.h:37: instantiated from ‘apache::thrift::protocol::TBinaryProtocolT<apache::thrift::transport::TTransport>’ /usr/local/include/thrift/protocol/TBinaryProtocol.h:259: instantiated from here /usr/local/include/thrift/protocol/TVirtualProtocol.h:554: error: no members matching ‘apache::thrift::protocol::TProtocolDefaults::readBool’ in ‘class apache::thrift::protocol::TProtocolDefaults’ In file included from dbcassa_base.cpp:36: /usr/local/include/thrift/protocol/TBinaryProtocol.h: In instantiation of ‘apache::thrift::protocol::TBinaryProtocolT<apache::thrift::transport::TTransport>’: /usr/local/include/thrift/protocol/TBinaryProtocol.h:259: instantiated from here /usr/local/include/thrift/protocol/TBinaryProtocol.h:178: error: no members matching ‘apache::thrift::protocol::TVirtualProtocol<apache::thrift::protocol::TBinaryProtocolT<apache::thrift::transport::TTransport>, apache::thrift::protocol::TProtocolDefaults>::readBool’ in ‘class apache::thrift::protocol::TVirtualProtocol<apache::thrift::protocol::TBinaryProtocolT<apache::thrift::transport::TTransport>, apache::thrift::protocol::TProtocolDefaults>’ In file included from /usr/local/include/thrift/protocol/TBinaryProtocol.h:277, from dbcassa_base.cpp:36: /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:31: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:51: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:56: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:62: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:67: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:78: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:83: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:89: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:100: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:105: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:114: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:119: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:128: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:133: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:140: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:146: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:153: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:160: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:167: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:179: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:189: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:198: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ In file included from /usr/local/include/thrift/protocol/TBinaryProtocol.h:277, from dbcassa_base.cpp:36: /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:230: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:235: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:241: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:246: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:263: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:268: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:289: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:294: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:312: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:317: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:335: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:340: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:348: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:356: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:367: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:378: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:389: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:404: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:412: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ /usr/local/include/thrift/protocol/TBinaryProtocol.tcc:417: error: expected constructor, destructor, or type conversion before ‘TBinaryProtocolT’ In file included from /usr/local/include/thrift/transport/TSocket.h:33, from dbcassa_base.cpp:37: /usr/local/include/thrift/transport/TVirtualTransport.h:50: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TVirtualTransport.h:53: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TVirtualTransport.h:56: error: ISO C++ forbids declaration of ‘uint8_t’ with no type /usr/local/include/thrift/transport/TVirtualTransport.h:56: error: expected ‘,’ or ‘...’ before ‘*’ token /usr/local/include/thrift/transport/TVirtualTransport.h:59: error: ISO C++ forbids declaration of ‘uint8_t’ with no type /usr/local/include/thrift/transport/TVirtualTransport.h:59: error: expected ‘;’ before ‘*’ token /usr/local/include/thrift/transport/TVirtualTransport.h:62: error: expected ‘;’ before ‘void’ /usr/local/include/thrift/transport/TVirtualTransport.h:62: error: ‘uint32_t’ has not been declared /usr/local/include/thrift/transport/TVirtualTransport.h: In member function ‘void apache::thrift::transport::TTransportDefaults::write(int)’: /usr/local/include/thrift/transport/TVirtualTransport.h:57: error: ‘buf’ was not declared in this scope /usr/local/include/thrift/transport/TVirtualTransport.h:57: error: ‘len’ was not declared in this scope /usr/local/include/thrift/transport/TVirtualTransport.h: At global scope: /usr/local/include/thrift/transport/TVirtualTransport.h:94: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TVirtualTransport.h:98: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TVirtualTransport.h:102: error: ISO C++ forbids declaration of ‘uint8_t’ with no type /usr/local/include/thrift/transport/TVirtualTransport.h:102: error: expected ‘,’ or ‘...’ before ‘*’ token /usr/local/include/thrift/transport/TVirtualTransport.h:106: error: ISO C++ forbids declaration of ‘uint8_t’ with no type /usr/local/include/thrift/transport/TVirtualTransport.h:106: error: ‘uint8_t’ declared as a ‘virtual’ field /usr/local/include/thrift/transport/TVirtualTransport.h:106: error: expected ‘;’ before ‘*’ token /usr/local/include/thrift/transport/TVirtualTransport.h:110: error: expected ‘;’ before ‘virtual’ /usr/local/include/thrift/transport/TVirtualTransport.h:110: error: ‘uint32_t’ has not been declared /usr/local/include/thrift/transport/TVirtualTransport.h:124: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TVirtualTransport.h: In member function ‘virtual void apache::thrift::transport::TVirtualTransport<Transport_, Super_>::write_virt(int)’: /usr/local/include/thrift/transport/TVirtualTransport.h:103: error: ‘buf’ was not declared in this scope /usr/local/include/thrift/transport/TVirtualTransport.h:103: error: ‘len’ was not declared in this scope In file included from dbcassa_base.cpp:37: /usr/local/include/thrift/transport/TSocket.h: At global scope: /usr/local/include/thrift/transport/TSocket.h:100: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TSocket.h:105: error: ISO C++ forbids declaration of ‘uint8_t’ with no type /usr/local/include/thrift/transport/TSocket.h:105: error: expected ‘,’ or ‘...’ before ‘*’ token /usr/local/include/thrift/transport/TSocket.h:110: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TSocket.h:216: error: ISO C++ forbids declaration of ‘sockaddr’ with no type /usr/local/include/thrift/transport/TSocket.h:216: error: expected ‘;’ before ‘*’ token /usr/local/include/thrift/transport/TSocket.h:237: error: ISO C++ forbids declaration of ‘sockaddr’ with no type /usr/local/include/thrift/transport/TSocket.h:237: error: expected ‘,’ or ‘...’ before ‘*’ token /usr/local/include/thrift/transport/TSocket.h:290: error: ‘sockaddr_in’ does not name a type /usr/local/include/thrift/transport/TSocket.h:291: error: ‘sockaddr_in6’ does not name a type In file included from /usr/local/include/thrift/transport/TTransportUtils.h:29, from dbcassa_base.cpp:38: /usr/local/include/thrift/transport/TBufferTransports.h:62: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TBufferTransports.h:75: error: ‘uint32_t’ does not name a type /usr/local/include/thrift/transport/TBufferTransports.h:94: error: ISO C++ forbids declaration of ‘uint8_t’ with no type /usr/local/include/thrift/transport/TBufferTransports.h:94: error: expected ‘,’ or ‘...’ before ‘*’ token /usr/local/include/thrift/transport/TBufferTransports.h:107: error: ISO C++ forbids declaration of ‘uint8_t’ with no type /usr/local/include/thrift/transport/TBufferTransports.h:107: error: expected ‘;’ before ‘*’ token dbcassa_base.cpp:1031: error: expected ‘;’ at end of input dbcassa_base.cpp:1031: error: expected ‘}’ at end of input In file included from /usr/local/include/thrift/transport/TTransportUtils.h:29, from dbcassa_base.cpp:38: /usr/local/include/thrift/transport/TBufferTransports.h: In member function ‘void apache::thrift::transport::TBufferBase::write(int)’: /usr/local/include/thrift/transport/TBufferTransports.h:95: error: ‘new_wBase’ was not declared in this scope /usr/local/include/thrift/transport/TBufferTransports.h:95: error: ‘wBase_’ was not declared in this scope /usr/local/include/thrift/transport/TBufferTransports.h:95: error: ‘len’ was not declared in this scope /usr/local/include/thrift/transport/TBufferTransports.h:96: error: ‘wBound_’ was not declared in this scope /usr/local/include/thrift/transport/TBufferTransports.h:97: error: ‘buf’ was not declared in this scope /usr/local/include/thrift/transport/TBufferTransports.h:101: error: ‘buf’ was not declared in this scope /usr/local/include/thrift/transport/TBufferTransports.h:101: error: ‘writeSlow’ was not declared in this scope /usr/local/include/thrift/transport/TBufferTransports.h: At global scope: /usr/local/include/thrift/transport/TBufferTransports.h:102: error: expected unqualified-id at end of input /usr/local/include/thrift/transport/TBufferTransports.h:102: error: expected ‘}’ at end of input /usr/local/include/thrift/transport/TBufferTransports.h:102: error: expected ‘}’ at end of input /usr/local/include/thrift/transport/TBufferTransports.h:102: error: expected ‘}’ at end of input make[1]: *** [dbcassa_base.o] Error 1 make: *** [install-modules] Error 1

3 10

set acc flag to dlg_bye
by Uri Shacked 23 Jan '12

23 Jan '12

thanks..... i already did. no bye is inserted to the DB. 2 invites for the 183 and 200, 1 ack, and that is it... another idea? :-) " Hi, set acc flag in local_route, On Sat, Jan 21, 2012 at 1:08 PM, Uri Shacked <ushacked at gmail.com<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> wrote: " >* Hi, *>* *>* i wonder, anyone has a good idea of how to set a flag, that will insert *>* the bye kamailio created to end the dialog into the acc table? *>* i am checking "early media", "cancel", "ack" etc.... but when kamaiilo *>* creates the bye with the dlg_bye command, there is no acc raw for that.... *>* i can create a sqlops and insert with the call's params... but there must *>* be a nicer way, no? *>* *>* BR, *>* Uri *>* *

2 4

acc flag before droping the call
by Uri Shacked 22 Jan '12

22 Jan '12

Hi, I am doing an internal chaek right after the following: " ........... # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting ............. " sometimes this check does not find the conditions needed and creates a sl_reply (404). the scenario works OK. now, i need the acc tabale record.... should probably be "INVITE".....""....... ""..... "404" .... "not found".... well, there is no record there. i tried the ACCFAILED flag as well, still does not work. i guess it has something to do with the transaction stage.... so, any idea on how to get ACC record with the ACC module there? BR, Uri

2 1

set acc flag to dlg_bye
by Uri Shacked 21 Jan '12

21 Jan '12

Hi, i wonder, anyone has a good idea of how to set a flag, that will insert the bye kamailio created to end the dialog into the acc table? i am checking "early media", "cancel", "ack" etc.... but when kamaiilo creates the bye with the dlg_bye command, there is no acc raw for that.... i can create a sqlops and insert with the call's params... but there must be a nicer way, no? BR, Uri

2 1

SIP Service not Connecting to VoIP Provider
by IPAN Gov-Europe 20 Jan '12

20 Jan '12

Sirs: Our SIP service with you (pbx(a)iptel.org) is not connecting to the VoIP Provider we installed. It was working until today... What can we do to fix this? Frank Riley, CTO IPANGov www.IPANGov.edu.tc

3 2

onreply _route syntax error for homer capture server
by JR Richardson 20 Jan '12

20 Jan '12

Hi All, Tuning my homer capture server and ran into a syntax error in the onreply_route. The idea is to NOT to capture NOTIFY|SUBSCRIBE|OPTIONS. The route block compiles ok but I get the following errors for the lines in the onreply_route: 0(6443) : <core> [cfg.y:3504]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 48, column 46: syntax error 0(6443) : <core> [cfg.y:3504]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 49, column 15: bad expression 0(6443) : <core> [cfg.y:3504]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 49, column 16: ERROR: bad config file (3 errors) route { if (!(method =~ "^(NOTIFY|SUBSCRIBE|OPTIONS)")) { sip_capture(); } exit; } 47 onreply_route { 48 if (!($rm =~ "^(NOTIFY|SUBSCRIBE|OPTIONS)") { 49 sip_capture(); 50 } 51 drop; 52 } This example was used from the homer wiki http://code.google.com/p/homer/wiki/Kamailio root@homer:/etc/kamailio# kamailio -V version: kamailio 3.2.1 (x86_64/linux) bee094 Any guidance is appreciated. Thanks. JR -- JR Richardson Engineering for the Masses

2 2

Group-based routing (Carrierroute - DRouting)
by Carlo Dimaggio 20 Jan '12

20 Jan '12

Hi all, I need group-based routing, that is I want a routing table for each type of subscriber (residential, business,...). I have seen that Drouting does it throught groupID and I'm wondering about the same behaviour with carrierroute module (as LCR uses only flat costs). Can the "carrier" parameter be the right way to do it? I have no special requirement about the number of routes. However, Is the DRouting actively maintained? Regards,

3 2

I am unable to call other iptel users and my voice mail is not working
by praseeth k m 20 Jan '12

20 Jan '12

Hi Team, I am unable to call other iptel users and my voice mail is not working . Would you be able to help me on this ? my user id is praseeth.panicker Regards, Praseeth

2 1

ip address error with exec module
by Uri Shacked 19 Jan '12

19 Jan '12

Daniel hi, a little late but, here is the level 3 debug log.... Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2387]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2388]: DEBUG: tm [t_reply.c:1509]: DEBUG: cleanup_uac_timers: RETR/FR timers reset Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2387]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2388]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2387]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2388]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2387]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2388]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2387]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2388]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2387]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2388]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2387]: DEBUG: <core> [xavp.c:365]: destroying xavp list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2388]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2387]: DEBUG: <core> [receive.c:291]: receive_msg: cleaning up Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2388]: DEBUG: <core> [xavp.c:365]: destroying xavp list (nil) Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2388]: DEBUG: <core> [receive.c:291]: receive_msg: cleaning up Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2401]: ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=860 a=65 n=assign Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2401]: ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=861 a=26 n=exec_msg Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2401]: : exec [../../parser/../ip_addr.h:688]: BUG: ip_addr2sbuf: unknown address family 0 Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2401]: DEBUG: <core> [md5utils.c:67]: DEBUG: MD5 calculated: 64d3de5c2447acd6006b8539d1246e10 Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2401]: DEBUG: exec [exec_hf.c:267]: SIP_HF_CONTENT_LENGTH=0 Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2401]: DEBUG: exec [exec_hf.c:267]: SIP_HF_CSEQ=1 OPTIONS Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2401]: DEBUG: exec [exec_hf.c:267]: SIP_HF_CALLID=123 Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2401]: DEBUG: exec [exec_hf.c:267]: SIP_HF_TO=<you(a)kamailio.org> Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2401]: DEBUG: exec [exec_hf.c:267]: SIP_HF_FROM=<you(a)kamailio.org>;tag=123 Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2401]: DEBUG: exec [exec_hf.c:267]: SIP_HF_VIA=SIP/2.0/UDP 127.0.0.1 Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2401]: DEBUG: exec [exec_mod.c:175]: executing [nc -u -w0 10.2.0.54 514 <<< ",denied_calls-0,dos-green"] Jan 18 07:36:23 kamailiolabroute /usr/local/sbin/kamailio[2401]: DEBUG: <core> [main.c:852]: SIGCHLD received: we do not worry about grand-children Jan 18 07:36:25 kamailiolabroute /usr/local/sbin/kamailio[2390]: DEBUG: tm [t_reply.c:1168]: ->>>>>>>>> T_code=0, new_code=408 Jan 18 07:36:25 kamailiolabroute /usr/local/sbin/kamailio[2390]: DEBUG: tm [t_reply.c:1931]: DEBUG: local_reply: branch=0, save=0, winner=0 Jan 18 07:36:25 kamailiolabroute /usr/local/sbin/kamailio[2390]: DEBUG: tm [t_reply.c:1968]: DEBUG: local transaction completed Jan 18 07:36:25 kamailiolabroute /usr/local/sbin/kamailio[2390]: DEBUG: tm [t_hooks.c:288]: DBG: trans=0xb00a772c, callback type 256, id 0 entered Jan 18 07:36:25 kamailiolabroute /usr/local/sbin/kamailio[2390]: DEBUG: dispatcher [dispatch.c:2345]: OPTIONS-Request was finished with code 408 (to sip:10.10.21.26, group 47)

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Kamailio Keepalive
by Reda Aouad 18 Jan '12

18 Jan '12

Hi, I am using the nat_keepalive function of the nat_traversal module, using the OPTIONS message, and my code is the following : if (is_method("REGISTER")) nat_keepalive(); I see that Kamailio still sends OPTIONS packets even if the client unregisters itself. It stops later, but I don't know when exactly. Is that a normal behaviour? What is the keepalive timeout? Why doesn't Kamailio remove the client from its keepalive list if it unregisters and is no more in the user location? And is there a way to remove a contact from the user location if it doesn't respond to keepalive packets? Thanks Reda

2 1

re-route SIP response message
by Robert R 18 Jan '12

18 Jan '12

Hi, Can I re-route a SIP response message? In other words, after the SIP proxy receives a 503 from node A, I need to change the TO field and re-route it to node B. I tried to modify $rd or $rU and didn't work. Thanks, R

2 1

New feature: embedded MSRP relay
by Daniel-Constantin Mierla 18 Jan '12

18 Jan '12

Hello, a new module is now on development branch, named msrp, offering an embedded MSRP relay. Message Session Relay Protocol is specified by IETF ( core and relay part in RFC4975 and RFC4976), one of typical use cases is Instant Messaging sessions negotiated via INVITE-200ok-ACK, a different approach than SIP MESSAGE request. There is no external dependency, the transport layer from core is reused to manage TCP/TLS connections used for MSRP. A MSRP relay is pretty much an independent node, thus you can run Kamailio just to deal with MSRP traffic. But there is no problem to run Kamailio to handle SIP and MSRP traffic in the same instance, same or different sockets (it is working even when SIP/MSRP are sent over the same connection or different connections to same port). I couldn't find a reliable and trustable open source SIP phone to test with it so far (hints are welcome), the module was tested with network tools and chaining Kamailio instances. Help with testing and feedback is very appreciated -- I can provide guidelines to adjust the config file to fit tester's needs, just contact me off list or via IRC channel. For the moment, the relation (user,session) management is done in the config, using htable module for example (see the README), it may be added inside the module for the future to make it easier overall, although is less than 10 lines of config -- this is mainly for extra security reasons, to check if the session id matches the connection that was opened when the session was created, otherwise the user authentication functionality is done reusing existing functions from auth module. Another benefit is performances, storing local socket and connection IP/port saves some time to lookup the connection. Building MSRP relay on top of SIP server was done first for the benefit of reusing the transport layer from the core for IPv4/IPv6 and TCP/TLS, which is mature, scalable and offers asynchronous communication. Besides that, you have most of the config tools to route SIP requests available for routing MSRP (authentication, authorization, IP checking, accounting, a.s.o.). I made a news post with more details: * http://www.kamailio.org/w/2012/01/new-module-embedded-msrp-relay/ The readme of the new module is available at: * http://kamailio.org/docs/modules/devel/modules/msrp.html Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

4 6

sip-router and related projects at FOSDEM 2012 - social event
by Stefan Sayer 17 Jan '12

17 Jan '12

Hello everyone, as has become already a tradition, we would like to invite all VoIP people and everyone interested in sip-router and related projects (SER, Kamailio/OpenSER, OpenSIPS, SEMS etc) who is attending FOSDEM, Europe's biggest Free and Open Source conference, to a dinner on the evening of Saturday February 4th. This is the perfect opportunity to socialize among the developers, get to know the people behind email addresses and commits, do networking, find the people for the next project, and/or just hang out together with fine food and drinks. Participation at the event is free of charge, everybody pays for their own expenses. In order to do reservation in a restaurant please send me a short mail if you would like to attend. Please also indicate if you don't want to have your contact details communicated to the other social event attendees. If you are still wondering whether it's worth to go to FOSDEM - among all the other interesting talks this year in the telephony-devroom on Sunday, 5th, there will be a few directly related talks (the exact time may be subject to change): 11:30 Andreas Granig: From zero to VoIP provider in 15 minutes 12:55 Stefan Sayer: Session Border Control with SEMS 15:25 Daniel-Constantin Mierla: Secure SIP Communications with Kamailio 16:15 Saúl Ibarra Corretgé: SIP beyond VoIP http://fosdem.org/2012/schedule/track/telephony_and_communications_devroom Hope to see you all in Brussels! With Best Regards Stefan Sayer -- tel:+491621366449 sip:sayer@iptel.org mailto/xmpp:stefan.sayer@gmail.com

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