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tutorials:faq:main [2012/10/09 11:08]
miconda [DNS]
tutorials:faq:main [2021/08/27 08:11] (current)
miconda [SIP Message Processing]
Line 44: Line 44:
 !!! Yes. !!! Yes.
  
 +
 +??? Is it possible to reload Kamailio configuration file?
 +
 +!!! If you use native scripting language, you must restart after you update the configuration file.
 +
 +But note that many global parameters can be changed via RPC/MI commands without restart (e.g., TCP connecting timeout, debug level). Applying changes related to loaded modules or routing block require always a restart.
 +
 +If you use a KEMI scripting language (Lua, Python, JavaScript, Ruby, Squirrel), then you can reload the routing logic script without restarting Kamailio by issuing an RPC command (see KEMI interpreter modules documentation for more details: app_lua, app_python, app_python3, app_jsdt, app_ruby, app_sqlang).
 +
 +??? What is the license of Kamailio?
 +
 +!!! Kamailio is an open source application licensed under GNU Public License version 2 (aka GPLv2). It can be used for free "as in beer" on your infrastructure. Keep in mind that you need also distribute the source code of your changes, if you distribute it as a binary to your customer. For more information have a look to the [[https://www.gnu.org/licenses/old-licenses/gpl-2.0-faq.html.en|GPLv2 FAQ]].
 +
 +Starting with end of 2008, contributions to core and several modules are done under BSD license. That means parts of it can be extracted and used under BSD license terms. But over all, when used as one application, the use must follow the terms of GPLv2, because GPLv2 is viral. 
 +
 +??? Can it be installed on Windows?
 +
 +!!! No. There is no official development pushed in this direction, Linux/Unix being the de facto for running servers. But it might not be big effort to make it compile using the Linux-like environments for Windows (such as cygwin).
 +
 +
 +=?==== Transport Layers =====
  
 ??? What transport layer protocols are implemented in Kamailio? ??? What transport layer protocols are implemented in Kamailio?
Line 53: Line 74:
 Kamailio can be used to bridge between any combinations of these protocols, e.g., receiving SIP packets on UDP over IPv4 and sending out on TLS over IPv6. Kamailio can be used to bridge between any combinations of these protocols, e.g., receiving SIP packets on UDP over IPv4 and sending out on TLS over IPv6.
  
-??? Is it possible to reload Kamailio configuration file?+Starting with version 4.0.0, Kamailio supports WebSocket transport protocol, allowing modern web browsers to call between them directly through Kamailio.
  
-!!! No, you must restart after you update the configuration file.+??? Does Kamailio have support for WebSockets?
  
-But note that many global parameters can be changed via RPC/MI commands without restart (e.g.TCP connecting timeout, debug level). Applying changes related to loaded modules or routing block require always a restart.+!!! Yes, starting with version 4.0.0. It supports WebSockets over TCP (WSand over TLS (WSS), on both IPv4 and IPv6.
  
-??? What is the license of Kamailio?+??? Can Kamailio do gatewaying between transport layers?
  
-!!! Kamailio is an open source application licensed under GNU Public License version 2 (aka GPLv2)It can be used for free "as in beer".+!!! YesYou can use Kamailio to receive SIP message on one transport and send out on a different transport. For example, receive TLS on IPv6 and send out UDP on IPv4.
  
-Starting with end of 2008, contributions to core and several modules are done under BSD license. That means parts of it can be extracted and used under BSD license terms. But over all, when used as one application, the use must follow the terms of GPLv2, because GPLv2 is viral. +??? Can Kamailio be used to call from web browsers (webrtc) to classic SIP phones?
  
 +!!! Yes, from signaling point of view. Remember that Kamailio is SIP packet router, not handling media plane at all. Simply, Kamailio can do gatewaying from WebSocket to UDP, TCP, TLS or SCTP for all SIP messages.
  
 +However, note that media plane (RTP streams) requires special secure encoding of the RTP packets, which may not be supported by the old SIP phones.
 =?==== Modules ===== =?==== Modules =====
  
Line 76: Line 99:
 !!! Inside the source tree, there are three directories: !!! Inside the source tree, there are three directories:
   * modules/   * modules/
-  * modules_k/ +  * modules_k/ - no longer used since version 4.0.0 
-  * modules_s/+  * modules_s/ - no longer used since version 4.0.0
  
 Each module is located in its own sub-folder. The documentation for modules that are bundled in Kamailio packages is available at: Each module is located in its own sub-folder. The documentation for modules that are bundled in Kamailio packages is available at:
Line 84: Line 107:
 ??? Which modules are compiled by default? ??? Which modules are compiled by default?
  
-!!! The modules that have the same dependencies as the core of Kamailio. Those that are not compiled by default are kept in the variable 'exclude_modules' of Makefile.+!!! The modules that have the same dependencies as the core of Kamailio. Those that are not compiled by default are kept in the variable 'exclude_modules' of Makefile. To print the list of excluded modules: 
 + 
 +<code> 
 +make printvar v=exclude_modules 
 +</code>
  
 ??? How can I compile and install a module that is excluded by default? ??? How can I compile and install a module that is excluded by default?
Line 95: Line 122:
 make install make install
 </code> </code>
 +
 +??? I use 'make all', but I can't get the db_mysql.so. How can I get this?
 +
 +!!! The module is not compiled by default, because depends on libmysqlclient. Use following commands to get in compiled and installed:
 +
 +<code>
 +make modules include_modules="db_mysql"
 +
 +make install include_modules="db_mysql" 
 +</code>
 +
 +??? How to compile only one module?
 +
 +!!! First be sure that the core compiles fine with the command:
 +
 +<code>
 +make 
 +</code>
 +
 +To compile a single module, use:
 +
 +<code>
 +make modules modules=modules/modname
 +</code>
 +
 +Replace modname with the real name of the module you want to compile -- for example, compiling only tls module:
 +
 +<code>
 +make modules modules=modules/tls
 +</code>
 +
 +=?==== SIP Message Processing =====
 +
 +??? Why changes made to headers or body are not visible immediately?
 +
 +!!! Kamailio is working always with the received SIP message (request or reply). Changes done to headers or body are kept as a list of operations to be applied before writing to the network. If you want changes to be applied immediately, use **msg_apply_changes()** from textopsx module:
 +
 +  * http://kamailio.org/docs/modules/stable/modules/textopsx.html#textopsx.msg_apply_changes
 +
 +Be careful with it and don't use it after record_route(), because this function has to insert a header containing the IP address from the local socket that is going to be used for forwarding - the socket is not yet know during configuration file execution.
 +
 +For example, considering that X-Hdr does not exist in incoming message, here are some cases:
 +
 +  * without applying changes
 +
 +<code c>
 +append_hf("X-Hdr: xyz\r\n");
 +
 +if(is_present_hf("X-Hdr")) {
 +   # this part is not executed
 +} else {
 +   # this part is executed
 +}
 +</code>
 +
 +  * with applying changes
 +
 +<code c>
 +append_hf("X-Hdr: xyz\r\n");
 +
 +msg_apply_changes();
 +
 +if(is_present_hf("X-Hdr")) {
 +   # this part is executed
 +} else {
 +   # this part is not executed
 +}
 +</code>
 +
 +??? Why parts of From/To header appear many times?
 +
 +!!! After doing header management operations, it can result that parts of From/To header are duplicated or the result is the concatenation of many values. That is related to previous question, because the changes are not applied immediately and updates to parts of headers are not a simple set operation.
 +
 +For example, if From username is "alice" and the operations in the config file are:
 +
 +<code c>
 +$fU = "bob";
 +...
 +$fU = "carol";
 +</code>
 +
 +The result can be that From username is "bobcarol".
 +
 +One solution is to use **msg_apply_changes()** in between:
 +
 +<code c>
 +$fU = "bob";
 +msg_apply_changes();
 +...
 +$fU = "carol";
 +</code>
 +
 +Another solution is to keep the value in a variable (e.g., avp or xavp) and do the operation only once, like:
 +
 +<code c>
 +$xavp(fuser) = "bob";
 +...
 +$xavp(fuser[0]) = "carol";
 +...
 +$fU = $xavp(fuser);
 +</code>
 +
 +Updating From/To headers is recommended to be done in **branch_route**, specially if it is needed to have different value for outgoing branches.
 +
 +The examples above were with assignments to $fU (can be other vars as well, such as $fu, $tU, $tu, ..), but it is the same behaviour when using functions such as uac_replace_from() or uac_replace_to().
 +
 +??? Why body or other parts of the SIP message appear multiple times?
 +
 +!!! It is practically the same reason as for multiple From/To parts.
 +
 +The configuration file does an operation that changes parts in the headers or body many times. For example, if the body has two concatenated IP addresses for media stream, then likely the rtpproxy function is used twice, or, if the SDP appears two time, then the rtpengine function is used two times.
 +
 +If the update has to be done many times, use **msg_apply_changes()** in between, otherwise refactor the config to perform the operation only once (e.g., the RTP relaying functions should be used in a branch_route block).
 +
 +??? How to set different header values for each destination of a SIP request?
 +
 +!!! Set the value of the header inside a **branch_route**.
 +
 +All the operations done over a SIP message inside the **request_route** (including from the sub-routes executed from request_route) will be common to all outgoing branches. If you want to do updates only for specific destinations, use branch routes.
 +
 +Example:
 +
 +  * add X-VBox header only when sending to voicemail server
 +  * add X-Peer-ID header when sending somewhere else
 +
 +<code c>
 +request_route {
 +  ...
 +  if(is_method("INVITE")) {
 +    t_on_branch("SETHEADERS");
 +    t_on_failure("REROUTE");
 +  }
 +  t_relay();
 +  exit;
 +}
 +
 +branch_route[SETHEADERS] {
 +   if($rd=="voicemail.server.com") {
 +     append_hf("X-VBox: $fU\r\n");
 +   } else {
 +     append_hf("X-Peer-ID: abc\r\n");
 +   }
 +}
 +
 +failure_route[REROUTE] {
 +  if (t_is_canceled()) {
 +    exit;
 +  }
 +  if (t_check_status("486|408")) {
 +    $du = $null;
 +    $ru = "sip:voicemail.server.com";
 +    t_on_failure("REROUTE");
 +    exit;
 +  }
 +}
 +</code>
 +
 +Very important is also to be aware that doing same operation many times in request_route is not overwriting the previous value, but combines them. For example, if you do two times uac_replace_from(), the From header can become corrupt. Like:
 +
 +<code c>
 +request_route {
 +  ...
 +  uac_replace_from("sip:test1@kamailio.org");
 +  uac_replace_from("sip:test2@kamailio.org");
 +  ...
 +}
 +</code>
 +
 +Results in From header having the URI: **sip:test1@kamailio.orgsip:test2@kamailio.org**. Again, use branch_route to do the operations if you need different From header for outgoing branches.
 +
 +
 +??? How to remove a single header field when a header appears multiple times?
 +
 +!!! SIP allows that certain header fields may appear multiple times in a SIP message. This header fields (e.g. Via, Route, Record-Route, Contact) may be written either on a single line (with comma separated) or on multiple lines (see [[http://tools.ietf.org/html/rfc3261#section-7.3|RFC 3261 section 7.3]] for details).
 +
 +The remove_hf() function from textops module always removes all header fields with a certain name, thus it can not be used in this case.
 +
 +To address a certain header (regardless if headers are in a single line or in separate lines) use the @hf_value select. The trick is to load the textopsx module (this select used to be in ser's textops module). Note, header names must use '_' instead of '-'. To delete or manipulate a certain header also use the functions from the textopsx module, e.g: remove_hf_value(). Note, here '-' is used for headers with '-'
 +
 +For example, incoming message:
 +<code>
 +Record-Route: <sip:1.1.1.1;lr=on;nat=yes>
 +Record-Route: <sip:2.2.2.2:5060;lr;transport=udp>,
 +   <sip:3.3.3.3:22506;lr;transport=udp>
 +</code>
 +
 +Dump all headers:
 +<code>
 +xlog("$sel(@hf_value.Record_Route)");
 +</code>
 +
 +Dump the first header (<sip:1.1.1.1;lr=on;nat=yes>):
 +<code>
 +xlog("$sel(@hf_value.Record_Route[1])");
 +</code>
 +
 +Dump the second last header (<sip:2.2.2.2:5060;lr;transport=udp>):
 +<code>
 +xlog("$sel(@hf_value.Record_Route[-2])");
 +</code>
 +
 +Remove the last header (<sip:3.3.3.3:22506;lr;transport=udp>):
 +<code>
 +remove_hf_value("Record-Route[-1]");
 +#or in above case:
 +remove_hf_value("Record-Route[3]");
 +</code>
 +
 +??? Why the SIP requests are replied with "483 Too Many Hops" or "513 Message Too Large"?
 +
 +!!! In both cases, the reason is probably an error in request routing script which caused an infinite loop.
 +
 +You can easily verify whether this happens by watching SIP traffic on loopback interface, for example using ngrep:
 +
 +<code>
 +ngrep -d lo -qt -W byline port 5060
 +</code>
 +
 +
 +A typical reason for misrouting is a failure to match local domain correctly. If a server fails to recognize a request for itself, it will try to forward it to current URI in believe it would forward them to a foreign domain.
 +
 +Alas, it forwards the request to itself again. This continues to happen until value of the max_forwards header field reaches zero or the request grows too big.
 +
 +The solution: make sure that domain matching is correctly configured.
 +
 +A quick way to achieve that is to introduce a config option to kamailio.cfg:
 +
 +<code>
 +alias=domainname
 +</code>
 +
 +where domainname has to be replaced with name of domain, which you wish to serve by Kamailio and which appears in request-URIs.
 +
 +
 +??? I send SIP requests to Kamailio, but nothing happens, why?
 +
 +!!! Check if you have a firewall rule dropping traffic on SIP port (5060). Note that network sniffing tools have hooks in kernel before the firewall, so even if you see the SIP packets with ngrep or wireshark, they may be dropped by the firewall.
 +
 +If you have the pike module loaded, double check to see if you don't block valid trusted traffic with it.
 +
 +You can increase the debug parameter value to 3 in the configuration file, restart kamailio and watch syslog messages to see if there is any text printed by kamailio. For each SIP packet received on the networks, Kamailio is printing at least few debug messages.
 =?==== Media Streams ===== =?==== Media Streams =====
  
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 !!! No, //however// Kamailio can be configured to proxy media if needed.  !!! No, //however// Kamailio can be configured to proxy media if needed. 
-  * [[http://www.kamailio.org/docs/modules/stable/|See rtpproxy, iprtpproxyand mediaproxy]]+  * [[http://www.kamailio.org/docs/modules/stable/|See rtpengine, rtpproxy, lrkproxyor mediaproxy]]
  
 ??? What codecs are supported by Kamailio? ??? What codecs are supported by Kamailio?
Line 116: Line 384:
 ??? What are the minimum knowledge requirements before starting with Kamailio? ??? What are the minimum knowledge requirements before starting with Kamailio?
  
-!!! To be familiar with Linux and understand Session Initiation Protocol (SIP - RFC3261).+!!! To be familiar with Linux or UNIX (FreeBSD, OpenBSD, OS/X) and understand Session Initiation Protocol (SIP - RFC3261).
  
 ??? Any step-by-step guide to install Kamailio? ??? Any step-by-step guide to install Kamailio?
Line 155: Line 423:
  
 !!! Use the bug tracker available at: !!! Use the bug tracker available at:
-  * http://sip-router.org/tracker+  * https://github.com/kamailio/kamailio/issues
  
 ??? Where can I buy commercial support? ??? Where can I buy commercial support?
Line 173: Line 441:
 !!! Kamailio can do next DNS query types: A-Name, C-Name, AAAA (IPv6), SRV, NAPTR and TXT. !!! Kamailio can do next DNS query types: A-Name, C-Name, AAAA (IPv6), SRV, NAPTR and TXT.
  
-??? Can it do DSN-based (SRV) load balancing?+??? Can it do DNS-based (SRV) load balancing?
  
 !!! Yes. !!! Yes.
 +
 +
 +??? How to avoid reverse DNS for **src_ip** comparison?
 +
 +!!! Do not enclose the IP address between quotes. Use like:
 +
 +<code>
 +if(src_ip==1.2.3.4) { ...
 +</code>
 +
 +The next example results in reverse DNS operation, because src_ip is compared with a string:
 +
 +<code>
 +if(src_ip=="1.2.3.4") { ...
 +</code>
  
 =?==== Instant Messaging and Presence ===== =?==== Instant Messaging and Presence =====
Line 217: Line 500:
 The xcap_client module can be used when you want to interact with an external XCAP server (that could be another instance of Kamailio). The xcap_client module can be used when you want to interact with an external XCAP server (that could be another instance of Kamailio).
  
 +=?==== NAT Traversal =====
  
 +??? Can Kamailio be used with phones connecting from behind NAT?
  
 +!!! Yes, Kamailio supports the so called "far-end NAT traversal". One solution is to use nathelper for NAT traversal of SIP signaling, and rtpproxy modules with the rtpproxy (http://rtpproxy.org) application for relaying RTP packets.
 +
 +Read the comments at top of default configuration file shipped with Kamailio, to enable NAT traversal logic there, one has to enable the define WITH_NAT:
 +
 +<code>
 +#!define WITH_NAT
 +</code>
 +
 +??? Can Kamailio be installed behind NAT (e.g., like on Amazon EC2)?
 +
 +!!! Yes. You have to set the public IP address as advertised attribute to listen parameter:
 +
 +<code>
 +  listen=_localsocket_ advertise _publicaddr_
 +</code>
 +
 +Example:
 +
 +<code>
 +  listen=udp:10.10.10.10:5060 advertise 11.11.11.11:5060
 +</code>
 +
 +=?==== Configuration File Processing =====
 +
 +
 +??? Can I check the configuration files for syntax errors?
 +
 +!!! Yes. Use '-c' command line parameter, like:
 +
 +<code>
 +/usr/local/sbin/kamailio -c -f /usr/local/etc/kamailio/kamailio.cfg
 +</code>
 +
 +Note that even configuration file is reported to be ok, there might still be troubles starting kamailio with the respective configuration file. For example, the database URL can be syntactically correct, but the access itself is not working due to wrong username or password, causing kamailio to fail starting.
 +
 +??? Do I have to delete the AVPs added in the configuration file for a SIP message?
 +
 +!!! No. The AVPs are automatically deleted when associated SIP transaction is destroyed.
 +
 +If you want to do it on purpose, you can delete the AVP using assignment to $null, like:
 +
 +<code>
 +$avp(name) = $null;
 +</code>
 +
 +??? What is a pseudo-variable?
 +
 +!!! A pseudo-variable is a special token that is expanded at runtume with a specific value. It can be used inside the parameters of some functions or in configuration file expressions.
 +
 +A pseudo-variable can refer to the value of an avp, body of a header, part of a SIP message or other variables from system.
 +
 +For more see:
 +  * http://www.kamailio.org/wiki/cookbooks/devel/pseudovariables
 +
 +??? How to iterate through the items in a comma separated string?
 +
 +!!! If you have a variable holding a string like "a,b,c,d" and want to get each character separately, you have to use a WHILE loop with {s.count,separator} and {s.select,index,separator} transformations.
 +
 +Here is an example:
 +
 +<code c>
 +$var(x) = "a,b,c,d";
 +$var(i) = 0;
 +$var(n) = $(var(x){s.count,,});
 +
 +while( $var(i) <= $var(n) ) {
 +   xlog("token at position $var(i) is: $(var(x){s.select,$var(i),,})\n");
 +   $var(i) = $var(i) + 1;
 +}
 +</code>
 +
 +??? How is the function return code evaluated?
 +
 +!!! Configuration file interpreter evaluates the return code of a function as follow:
 +
 +  * <0 (negative value) - it is evaluated to false
 +  * >0 (positive value) - it is evaluated to true
 +  * =0 (zero) - it is evaluated as exit (stop execution of configuration file)
 +
 +Example:
 +
 +<code c>
 +
 +if(function_returns_one()) {
 +   # it goes here
 +} else {
 +   # it doesn't go here
 +}
 +...
 +if(function_returns_minus_one()) {
 +   # it doesn't go here
 +} else {
 +   # it goes here
 +}
 +...
 +if(function_returns_zero()) {
 +   # it doesn't go here
 +} else {
 +   # it doesn't go here
 +}
 +</code>
 +
 +Note that you can use $rc to get the return code value, like:
 +
 +<code c>
 +function();
 +xlog("returned code by function() is $rc\n");
 +</code>
 +
 +However, if return code is 0, the next action after function() is not executed. It can be used only of positive or negative response code.
 +
 +On the other hand, the pseudo-variables (including $rc) have to be compared as an integer, the rules for evaluating return code of the functions do not apply, for example:
 +
 +<code c>
 +$var(x) = 0;
 +if($var(x) == 0) {
 +  # do something
 +}
 +</code>
 +
 +To compare the return code of a function with a number is recommended to use the value stored in $rc, because the priority of the operators can convert function return code to internal-true or internal-false. Therefore the next approach should be used:
 +
 +<code c>
 +function_returns_four();
 +$var(rc4) = $rc;
 +if($var(rc4)==4) {
 +   # it goes here
 +} else {
 +   # it doesn't go here
 +}
 +...
 +function_returns_two();
 +$var(rc2) = $rc;
 +if($var(rc4)==$var(rc2)) {
 +   # it doesn't go here
 +} else {
 +   # it goes here
 +}
 +...
 +if (function_returns_four() && $rc ieq 6 ) {
 +   # it doesn't go here
 +} else if ($rc ieq 4) {
 +   # it goes here
 +} else {
 +   # it doesn't go here
 +}
 +</code>
 +
 +Note that $rc is overwritten by a new function call, therefore store its value in another variable (like $var(...)) if it is needed for longer time.
 +
 +For pseudo-variables, the non-zero value is evaluated to true and zero to false, for example:
 +
 +<code c>
 +$var(x) = 1;
 +if($var(x)) {
 +  # do something
 +}
 +</code>
 +
 +??? How is the SIP request retransmission handled?
 +
 +!!! The next snippet is detecting retransmissions:
 +
 +<code c>
 +    # handle retransmissions
 +      if (!is_method("ACK")) {
 +          if(t_precheck_trans()) {
 +              t_check_trans();
 +              exit;
 +          }
 +          t_check_trans();
 +      }
 +</code>
 +
 +The **ACK** request is skipped because it doesn't have a SIP response, therefore it doesn't create a transaction, being forwarded in stateless mode.
 +
 +The function **t_precheck_trans()** returns true if the same SIP request is processed at that time by another Kamailio process, so in this case it is a retransmission. The inner **t_check_trans()** has the role of detecting if the SIP transaction was created by the other process, and if yes, send again the last SIP response of the transaction, when one was already sent out (a requirement from SIP specs). The **exit** is then used to finish the execution of the config file for that request.
 +
 +If the function **t_precheck_trans()** returns false, the SIP request is not under processing by another Kamailio process, but it can be the case that the request processing was finished, request being sent out in stateful mode, therefore **t_check_trans()** is used to see if a transaction is found in shared memory corresponding to the same request. If such transaction is found in shared memory, then **t_check_trans()** triggers internally the **exit** for configuration file execution.
 +=?==== Troubleshooting =====
 +
 +??? I get 'out of memory' errors, what can be done?
 +
 +!!! Most probably you run out of memory allocated for Kamailio. The application does not use the entire system memory, but up to a configured size.
 +
 +There are two memory pools used by Kamailio:
 +  * shared memory - shm - common at application level, where most of the data needed for long time is stored (such as location records, least cost routing records, transactions, etc.). The default size is 32MB.
 +  * private memory - pkg - allocated per process, used for local storage and temporary operations. The default size is 4MB.
 +
 +To increase the sizes for memory pools you have to give following command line parameters:
 +  * **-m SIZE** - specify the shared memory size in MB
 +  * **-M SIZE** - specify the private memory size in MB
 +
 +For example, start Kamailio with 512MB of shared memory and 8MB of private memory:
 +
 +<code>
 +/usr/local/bin/kamailio -m 512 -M 8 ...
 +</code>
 +
 +=?==== Tools =====
 +
 +??? How to generate HA1 value in command line?
 +
 +!!! Use following command:
 +
 +<code>
 +echo -n 'user:realm:password' | md5sum | awk '{ print $1 }'
 +</code>
 +
 +??? How to generate HA1B value in command line?
 +
 +!!! Use following command:
 +
 +<code>
 +echo -n 'user@realm:realm:password' | md5sum | awk '{ print $1 }'
 +</code>
 +
 +??? Kamctl gives an error regarding opening FIFO file, why?
 +
 +!!! Kamctl couldn't connect to FIFO file for sending MI commands - default FIFO file is **/tmp/kamailio_fifo**.
 +
 +Check if your configuration loads the mi_fifo module and configures a fifo name.
 +
 +Example for the configuration syntax:
 +<code>
 +loadmodule "mi_fifo.so"
 +
 +modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")
 +</code>
 +
 +Check also the access privileges of the FIFO file in order to be sure that the user running kamctl can read and write to the file.
 +
 +??? Kamcmd sometime fails to execute RPC commands with number parameters, why?
 +
 +!!! There are situations when a **kamcmd** RPC command fails because the parameters that are provided are auto-converted to integer numbers. This happens when the implementation of the RPC commands expects a string parameter (e.g., username). 
 +
 +To prevent **kamcmd** to do the auto-conversion, you can prefix its parameter with **s:**.
 +
 +Example:
 +
 +<code>
 +# - next command fails
 +kamcmd uac.reg_refresh 1001
 +
 +# use instead
 +kamcmd uac.reg_refresh s:1001
 +</code>
tutorials/faq/main.1349773711.txt.gz · Last modified: 2012/10/09 11:08 by miconda