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42 participants
37519 discussions

set host as inactive and probing in dispatcher
by Uri Shacked 09 Dec '11

09 Dec '11

hi, i use "ds_select_domain()" to route a call. if the destination reply with an error i am using the next lines in a route that i called from the failure route: if($avp(OSPRPL)!=404) { ds_mark_dst("IP"); } after trying this line i do "kamctl fifo ds_list" and i see the host is still at "AX". and in the next call the dispatcher still tries the same destination. why? how do i set a destinatio in dispatcher to be inactive? this is my modparams: #!ifdef WITH_DISPATCHER #------ dispatcher params --------- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "attrs_avp", "$avp(dsattrs)") modparam("dispatcher", "ds_ping_from", "sip:user@host.com") modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=405") modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_hash_size", 8) modparam("dispatcher", "flags", 2) #!endif

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unset host domain in dispatcher module
by Uri Shacked 09 Dec '11

09 Dec '11

Hi, i am using the dispatcher module in order to send the call to a LNP server. after the 3xx reply from the LNP server, i would like to send the invite back to the proxy that initiate the call from the beginnig. how do i do that? it seems that kamailio dispatcher sets the host domain in the Internet Protocol, and the sip request line does not change so to be short.. two questions: 1. how do i "undo" the dispatcher? 2. how do i make the dispatcher change the requst lineas well? (i am using ds_select_dst or ds_select_domain and they do the same...) the carrierroute module, do it better, but i need the dispatcher for the "keep alive" usage. ideas will be great :-) Uri

2 3

How to i use record_route_advertised_address
by Gnaneshwar Gatla 09 Dec '11

09 Dec '11

Hello, I have Kamailio 3.1.0 behind a NAT. We need the Firewall for sip-security reasons. Currently I'm using record_route_preset("advertised_Address") for keeping the advertised address in the record-route header. This works for fine when I have all clients running on UDP protocol. I currently have some TLS clients added to our network, Kamailio does not insert the second record-route header with the advertised-IP whenever there is a call between a UDP and a TLS Client. Instead it inserts the internal/Local IP of the server. I had also noticed the record_route_preset() function does not include the transport parameter ("transport=UDP" or "transport=TLS"). I was looking through the source code of Kamailio and found record_route_advertised_address() in record.c . I would like to know how to use this function with the Kamailio config. Regards Gnani

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Presence Dialog not being created
by Gnaneshwar Gatla 08 Dec '11

08 Dec '11

Hello, Kamailio version: 3.1.5 I have been trying to use presence for Event:Dialog. I have used PUA_Dialoginfo module to accomplish this task. The PUA_dialoginfo module states that if "override_lifetime" is not used, the value of the expires is taken from dialog module. I have tried the module without the "override_lifetime" which did not create the dialog in the presentity. When used debug, I did see the xml being generated but finds the dialog "expires=0" and deletes the xml(please find the log below). But when "override_lifetime" is set, the dialog in the presentity table is set until. But this has a problem, modparam("pua_dialoginfo", "override_lifetime", 300) 1. if a call is still going on more than the override_lifetime the presentity is deleted. 2. the presentity information is available until the "override_lifetime" even after the call is hung-up. How do I get this module working without the "override_lifetime" being used. The Log: Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua_dialoginfo [dialog_publish.c:242]: new_body:#012<?xml version="1.0"?>#012<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:user@mydemo.com">#012 <dialog id="112014dff048e71e" call-id="112014dff048e71e" direction="initiator">#012 <state>Trying</state>#012 <remote>#012 <identity>sip:user01@mydemo.com</identity>#012 <target uri="sip:user01@mydemo.com"/>#012 </remote>#012 <local>#012 <identity>sip:user@mydemo.com</identity>#012 <target uri="sip:user@mydemo.com"/>#012 </local>#012 </dialog>#012</dialog-info>#012 Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: DEBUG: <core> [db_res.c:81]: freeing 1 columns Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua_dialoginfo [dialog_publish.c:290]: publish uri= sip:user@mydemo.com Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0x8314aa4 Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: <core> [mem/q_malloc.c:366]: qm_malloc(0x829bee0, 756) called from pua_dialoginfo: dialog_publish.c: dialog_publish(302) Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: <core> [mem/q_malloc.c:428]: qm_free(0x829bee0, 0x8314aa4), called from <core>: db_res.c: db_free_columns(86) Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: <core> [mem/q_malloc.c:406]: qm_malloc(0x829bee0, 756) returns address 0x82e84bc frag. 0x82e84a4 (size=900) on 1 -th hit Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x8314a8c alloc'ed from db_mysql: km_res.c: db_mysql_get_columns(78) Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua_dialoginfo [dialog_publish.c:51]: publ: Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: DEBUG: <core> [db_res.c:94]: freeing result names at 0x8318650 Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua_dialoginfo [dialog_publish.c:52]: uri= sip:user@mydemo.com Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: <core> [mem/q_malloc.c:428]: qm_free(0x829bee0, 0x8318650), called from <core>: db_res.c: db_free_columns(95) Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua_dialoginfo [dialog_publish.c:53]: id= DIALOG_PUBLISH.112014dff048e71e Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x8318638 alloc'ed from <core>: db_res.c: db_allocate_columns(148) Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua_dialoginfo [dialog_publish.c:54]: expires= 0 Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: DEBUG: <core> [db_res.c:99]: freeing result types at 0x8318684 Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua [send_publish.c:403]: pres_uri=sip:user@mydemo.com Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: <core> [mem/q_malloc.c:428]: qm_free(0x829bee0, 0x8318684), called from <core>: db_res.c: db_free_columns(100) Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua [hash.c:121]: core_hash= 504 Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x831866c alloc'ed from <core>: db_res.c: db_allocate_columns(157) Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua [hash.c:171]: record not found Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: DEBUG: <core> [db_res.c:54]: freeing 1 rows Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua [send_publish.c:444]: insert type Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: DEBUG: <core> [db_row.c:97]: freeing row values at 0x8314b14 Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua [send_publish.c:448]: UPDATE_TYPE and no record found Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: <core> [mem/q_malloc.c:428]: qm_free(0x829bee0, 0x8314b14), called from <core>: db_row.c: db_free_row(98) Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua [send_publish.c:454]: request for a publish with expires 0 and no record found Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x8314afc alloc'ed from <core>: db_row.c: db_allocate_row(114) Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: <core> [mem/q_malloc.c:428]: qm_free(0x829bee0, 0x82e84bc), called from pua_dialoginfo: dialog_publish.c: dialog_publish(357) Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: DEBUG: <core> [db_res.c:62]: freeing rows at 0x8314adc Regards Gnaneshwar

2 4

Problem to verify contact header
by Bruno Bresciani 08 Dec '11

08 Dec '11

Hi All, Kamailio generate a core at line below if(_msg->contact!= NULL || _msg->contact->body.s!= NULL){ _msg is a sip_msg struct that my module receive from kamailio. I want verify if on that request messagem have a contact header, but a core is being generated when contact header isn't present on message. Someone knows why this is happening? Cheers

3 4

How to test registration setting
by Efelin Novak 08 Dec '11

08 Dec '11

Hi folks, I would like to know what to use for the following scenario. I want users to be able to test their registration by sending REGISTER and Kamailio sends them 200 OK if credentials are OK. However I don't want the aor to be stored in location table. This REGISTER should be only used for testing the settings. After some time I would allow the registration to be stored and users will normally receive calls. I tried to implement this by not doing save(location) when the flag from load_credential is set. Unfortunately it is the function save who sends 200 OK. When it is not being called, a user agent keeps sending REGISTER until SIP triggers fail. So is there any other common practice how to test registers? Is there any flag that can prevent storing location? Can I send 200 OK to these user in any other way? Thanks Efelin

2 2

Re: [SR-Users] Remote Party ID
by Daniel-Constantin Mierla 08 Dec '11

08 Dec '11

Hello, On 12/7/11 10:03 AM, Pavel Segeč wrote: > > HI, > > Thank you. Just as I sent my previous mail I look into kamctl command > syntax and I found > > -- command 'rpid' - manage Remote-Party-ID (RPID) > > rpid add <username> <rpid> ......... add rpid for a user (*) > > rpid rm <username> ................. set rpid to NULL for a user (*) > > rpid show <username> ............... show rpid of a user > > How this work? > this is for adding/removing/showing the rpid column in subscriber table -- if you look in database, there is a column named rpid for subscriber table. Practically, setting it and loading with load_credentials is a way to use it in the config file. In config then you can use append_hf() add header as you need, or if it is rpid specific format, see siputils module for other options: http://kamailio.org/docs/modules/stable/modules_k/siputils.html Cheers, Daniel > pavel > > *From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* Tuesday, December 06, 2011 11:14 PM > *To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List > *Cc:* Pavel Segeč > *Subject:* Re: [SR-Users] Remote Party ID > > Hello, > > On 12/6/11 2:50 PM, Pavel Segeč wrote: > > Hi, > > I'm looking for some recommendations or guidlines. My leading IT > department which is providing PSTN connectivity requires to include > Remote Party ID in SIP Messages in a case where From URI is not in > telco format (I prefer email like style of addressing). How to simply > assign to an user its Remote Party ID which will be used when PSTN > calling will occure? On what should I focus on? To use user > preferences with AVP pairs? or something another? > > since it is about one attribute associated with the caller, a > recommended way is to add a new column in subscriber table, say it is > named 'rpid'. > > Then, in config file, set the load_credentials parameter of auth_db to > load the rpid value and store in an avp. After you authenticate the > caller, the avp will be set to the rpid value, see: > > http://kamailio.org/docs/modules/stable/modules_k/auth_db.html#id2528175 > > Using this option, practically you use same db query that loads the > user password for authentication to fetch the rpid value. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla --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

[OT] sip:provider CE v2.4 released
by Andreas Granig 07 Dec '11

07 Dec '11

Ladies and Gentlemen, I'm happy to announce the official release of sip provider CE v2.4, an easy-to-use, free and open-source soft-switch built on top of Kamailio and Sems. As always, we provide a dead-simple installer to start from scratch, upgrade scripts from v2.2 and Virtualbox and VMware images to test-drive a turn-key installation. http://www.sipwise.com/news/announcements/spce-v2_4-release/ The SPCE shows the capabilities of what can be created with flexible and powerful software like Kamailio and Sems, when the proper provisioning, billing, configuration and packaging glue is being put around that. The SPCE v2.4 is currently deployed around the world acting as an SBC in front of old legacy 3rd party switches to provide SIP service to mobile clients, as a Class4 peering concentrator to provide ENUM/PSTN access to other soft-switches, and as standard Class5 soft-switches for all kinds of access networks. It's ideal for new Kamailio/Sems users to have a head-start over beginning completely from scratch, because the SPCE comes with customer- and admin-web-interfaces, SOAP/XMLRPC-APIs, provisioning and rating engines etc. No need to tinker with all the various configuration files, unless you want to. :) You can find the full release announcement with the changelog highlights over here: http://www.sipwise.com/news/announcements/spce-v2_4-release/ Thanks, Andreas

3 3

[OT] job offer - System administrator middleware & acs
by Henning Westerholt 07 Dec '11

07 Dec '11

[OT] Job offer: Interested in working with one of the biggest Voice over IP networks in europe? We're hiring and have one additional IT operation position open in our team in Karlsruhe, Germany. You would be responsible in a team of motivated system administrators for the operation and maintenance of the high- available middleware cluster and CPE auto-configuration services for the DSL and mobile products at 1&1. Furthermore you would be support the on-call duty for this services in the team. A solid understanding of german is necessary for the job, for more informations (additional requirements, how to apply..) please refer to the opening on the job web page. Of course please feel free to contact me per mail as well. http://jobs.1und1.de/xml/JobsDetail?job_id=3777 Best regards, Henning -- Henning Westerholt - Head of IT Operations Internet Access & Communications 1&1 Internet AG, Brauerstraße 48, 76135 Karlsruhe, Germany

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Re: [SR-Users] Remote Party ID
by Daniel-Constantin Mierla 06 Dec '11

06 Dec '11

Hello, On 12/6/11 2:50 PM, Pavel Segeč wrote: > > Hi, > > I'm looking for some recommendations or guidlines. My leading IT > department which is providing PSTN connectivity requires to include > Remote Party ID in SIP Messages in a case where From URI is not in > telco format (I prefer email like style of addressing). How to simply > assign to an user its Remote Party ID which will be used when PSTN > calling will occure? On what should I focus on? To use user > preferences with AVP pairs? or something another? > since it is about one attribute associated with the caller, a recommended way is to add a new column in subscriber table, say it is named 'rpid'. Then, in config file, set the load_credentials parameter of auth_db to load the rpid value and store in an avp. After you authenticate the caller, the avp will be set to the rpid value, see: http://kamailio.org/docs/modules/stable/modules_k/auth_db.html#id2528175 Using this option, practically you use same db query that loads the user password for authentication to fetch the rpid value. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

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