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Dialplan Module
by Nelson Migliaro 05 Sep '17

05 Sep '17

Hello, Do you know if its possible to

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Re: [SR-Users] Incorrect value when setting number like values in Htable using Kamcmd
by Tony - Telecomy 05 Sep '17

05 Sep '17

Prefixing the numbers with "s:" fixed the issue. Thank you for your help. Tony

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Incorrect value when setting number like values in Htable using Kamcmd
by Tony - Telecomy 05 Sep '17

05 Sep '17

Hello, I'm having an issue setting phone numbers values in htable using kamcmd when the phone number is above the integer range 2147483647. Initially I planned to set the values as strings but this is not working as expected as well. Please find below some examples: root@tst:~/tst/test# kamcmd htable.seti tst_sim integer_limit "2147483647" root@tst:~/tst/test# kamcmd htable.seti tst_sim integer_exceeded_limit "2147483648" root@tst:~/tst/test# kamcmd htable.sets tst_sim integer_as_string_valid "2147483647" root@tst:~/tst/test# kamcmd htable.sets tst_sim integer_as_string_invalid "2147483648" root@tst:~/tst/test# kamcmd htable.dump tst_sim | grep -A 1 integer name: integer_as_string_valid value: 2147483647 -- name: integer_limit value: 2147483647 -- name: integer_as_string_invalid value: 18446744071562067968 -- name: integer_exceeded_limit value: -2147483648 my ctl params are as follow: modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") modparam("ctl", "binrpc_max_body_size", 1024) modparam("ctl", "binrpc_struct_max_body_size", 1024) Thanks, Tony

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telcosuite3-crm sql
by Kordován Szabolcs 05 Sep '17

05 Sep '17

Hi, I would like to try the IMS according to the Kamailio World 2017: Kamailio For IMS And VoLTE presentation. Unfortunately I have problem with telcosuite, the sql structure is missing. Where can I find the whole create.sql? Thanks for help. Regards, Szabolcs

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Kamailio SIP protocol over WSS
by Washington Costa 05 Sep '17

05 Sep '17

Hi, I solved my problem. I was missing load some modules and routers. I saw this file as example: https://gist.github.com/jesusprubio/4066845 but is important to know that every module the begin "mi" they aren't supporte on Kamailio 5.0. You will need replace for the module relative the version 5.0. Washington L da Costa Analista de Sistemas (82) 9 9823-4782

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Question about fix_nated_contact.
by Володимир Іванець 04 Sep '17

04 Sep '17

Hello everyone, I have system setup where there are few Asterisk servers behind Kamailio that is behind the firewall. There is a problem with call initiation. When phone from outside sends INVITE to Kamailio, it then passes it to one of Asterisk servers. Asterisk server eventually replies to Kamailio with 200 OK. Contact header contains Asterisk's local IP address and port. Kamailio executes function *fix_nated_contact* before sending reply to the phone. However Contact header will still have Asterisk's local socket. *fix_nated_contact* is being executed with no doubt. I can see xlog output that I set before and after it. Can someone tell how exactly *fix_nated_contact* gets "*request's source address:port pair*" and if it can be printed to the log? Thank you!

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Homer confusion ....
by Mark Boyce 04 Sep '17

04 Sep '17

Hi All Trying to get a basic version of homer working on Kamailio V5. So far I have sipcapture throwing everything in to a single mysql table. So first question; Should I be using sipcapture or siptrace? I’m not sure I can see the difference. Second question; On what I have working so far using sipcapture, I’m seeing the invites, subscribes, cancels, act’s all end up in the database. What I’m missing is any responses from Kamailio back to the handset (AUTh Challenges etc) modparam("sipcapture", "db_url", "mysql://sipcapture:secretpassword@localhost/homer_data") modparam("sipcapture", "capture_on", 1) modparam("sipcapture", "hep_capture_on", 1) modparam("sipcapture", "insert_retries", 5) modparam("sipcapture", "insert_retry_timeout", 10) plus a sprinkling of "sip_capture(); “ in the “xxx_route {“ blocks What am I missing? Mark

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presence_xml mod_init(): Can't bind xcap_client
by Borislav Gergov 04 Sep '17

04 Sep '17

*Hi I have presence_xml parameter force_active set to 1 and still i get **error** presence_xml:mod_init: Can't bind xcap_client* *kamailio 5.1.0-dev6 (x86_64/linux) *

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MWI forwarding between Kamailio and Freeswitch
by Wang, Wallace 04 Sep '17

04 Sep '17

Hello We're using Kamailio & freeswitch as my company voice system. So far, everything is good, except the MWI. I tried to find out how to forward MWI between Kamailio and Freeswitch, or let the Presence_MWI module can work with the freeswitch behind Kamailio. If it was possiable, could anyone give me an idea or some links or demo configure for this. I really appreciate your help. Best Regards Wallace Wang

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uac +kamailio + dispatcher + asterisk
by Жан Базаров 04 Sep '17

04 Sep '17

Hello! provider trunks registration on kamailio UAC?! it's clear. but how does asterisk find out through which trunk the call should be made? I tried to set the header from asterisk dialplan. for example: if (is_method("INVITE")) { #record_route_preset("109.195.102.122"); route(DIRECTION); setflag(FLT_ACC); # do accounting } # ------ LOADBALANCE ROUTE ------------ # if(!ds_is_from_list()) { route(DISPATCH); } route[DISPATCH] { #round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(NATMANAGE); route(DRELAY); exit; } route[DIRECTION] { if ($hdr(x-trunk) != $null) { if (!is_method("BYE")){ $fu=""; t_on_failure("MANAGE_FAILURE"); $dlg_ctx(timeout_route) = "DIALOG_END"; $avp(i:10)=43200; $dlg_ctx(timeout_bye) = 0; sql_pvquery("ca", "select l_uuid, auth_username, auth_password, realm, l_domain, r_domain from uacreg where id='$hdr(x-trunk)'", "$avp(uuid), $avp(uname), $avp(passwd), $avp(realm), $avp(src_ipaddr), $avp(dst_ipaddr)"); t_on_failure("MANAGE_FAILURE"); $dlg_ctx(timeout_route) = "DIALOG_END"; $avp(i:10)=43200; $dlg_ctx(timeout_bye) = 0; $fu=""; uac_replace_from("sip:$avp(uname)@$avp(dst_ipaddr)"); $tu="sip:"+$tU+"@"+$avp(dst_ipaddr); $ru="sip:"+$tU+"@"+$avp(dst_ipaddr); remove_hf("Contact"); $var(contact)="sip:"+$avp(uname)+"@10.49.9.2:5060"; insert_hf("Contact: <$var(contact)>\r\n"); #insert_hf("Contact: "); msg_apply_changes(); fix_nated_register(); xlog("L_INFO","Contact header $var(contact) 111111111111111111111111111111111111111 is $ct {$ct}\n"); route(RELAY); } #### BYE } ### XTRUNK But in this configuration I do not come bye but when I register providers trunks on asterisk - problem with BYE not visible. but I can not register provider-trunks on all the asterisks, because incoming invite arrives at the link + address, and all the asteriscs ring. All my asterisk's behind nat

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