- sr-users - kamailio.org

Make SIP session between two kamailio servers
by Cockhootec Yahrabee 21 Feb '16

21 Feb '16

Hi everyone, I am curious if there is a way to make SIP session between two kamailio server instances eg. if there is way to use kamailio as an SIP client and SIP server at the same time. I am thinking about the following configuration: kamailio main server ----- kamailio server B ----- SIP client B1 | | | kamailio server A ----- SIP client A2 | | | SIP client A1 I want to be able to make SIP session between: SIP client A1 : SIP client A2 kamailio server A : kamailio server B SIP client A1 : SIP client B1 (this would be great) If there is currently no way to do this do you think that it would be possible to create the module which will support this kind of functionality? Thanks in advance for the replies

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How long will be client online in kamailio server since client is forcely closed
by sgy 21 Feb '16

21 Feb '16

Hi All, If the client is forcely closed, the client will be shown as online state in kamailio when I run the command: "kamctl online". How long will be the client online since client is closed? Thank you. -- Best Regards, Shengy

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Re: [SR-Users] [SPAM]Re: Registration Failure - IMS
by Franz Edler 20 Feb '16

20 Feb '16

Hello Cristina, > In the attach you can find the trace of the registration. I have checked your tracefile and confirm: that is a problem which needs further debugging. Up to now I did not activate "WITH_REGINFO" in my installation, therefore I cannot compare. If I find time (this weekend) I will do that and come back with the findings. BR Franz

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rfc: kamailio.cfg polishing for v4.4
by Daniel-Constantin Mierla 20 Feb '16

20 Feb '16

Hello, wondering if people feel it is need to polish the current default kamailio.cfg, to prepare the version for next major release. It can be browsed online at: - https://github.com/kamailio/kamailio/blob/master/etc/kamailio.cfg It is mainly about very common scenarios that are encountered out there and we should deal with by default. Have in mind that there are the basic and advanced versions of kamailio.cfg in the etc folder. Also, I am looking for tunings for core parameters and module parameters that make the config more suitable for common use case. Speaking of those extra config files, I think the kamailio-advanced.cfg and kamailio-basic.cfg should be deployed in shared doc folder, not being active configs upon default installation. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

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BUG: qm_free: freeing already freed pointer
by Rene Montilva 19 Feb '16

19 Feb '16

Hi list i've got this bug report in kamailio 4.3 at syslog, then it frozen, i checked my config file looking for htable parameter, and i notice i have htable param without autexpire is this reason why kamailio crash? "/usr/sbin/kamailio[24113]: : <core> [mem/q_malloc.c:453]: qm_free(): BUG: qm_free: freeing already freed pointer (0x7f673dead5d8), called from tm: h_table.c: free_cell(133), first free tm: h_table.c: free_cell(133) - aborting " -- Ing. Rene Montilva *FOSS Developer and VoIP Engineer.*

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Re: [SR-Users] Error or Bug: [t_msgbuilder.c:490]: build_local_reparse(): ERROR: build_local_reparse: HDR_EOH_T was not found
by Sid Mason 19 Feb '16

19 Feb '16

Yes, there is a append_hf("Remote-Party-ID: below is the complete section: # Send to Asterisk route[TOASTERISK] { ds_mark_dst("P"); if(!ds_select_dst("1", "4")) { sl_send_reply("500", "Service Unavailable"); xlog("L_INFO","[$fU@$si:$sp]{$rm} No destinations available for $rd \n"); exit; } # Replace FROM header to Kamailio's address # http://www.kamailio.org/docs/modules/4.3.x/modules/uac.html#uac.f.uac_repla… uac_replace_from("","sip:$sel(cfg_get.kamailio.bind_external_domain):5060"); # rewrite Destinatin URI #$du= $au + "@10.1.4.22:5060"; # Add Remote-Party-ID to header append_hf("Remote-Party-ID: \"$au\" <sip:$au@ $sel(cfg_get.kamailio.bindip_external)>;party=calling;privacy=off;screen=no"); xlog("L_INFO","route[TOASTERISK]: [$fU@$si:$sp]{$rm} From Outside World to Asterisk Box $du\n"); rtpproxy_manage("cawei"); route(RELAY); exit; } Regards Subject: Re: [SR-Users] Error or Bug: [t_msgbuilder.c:490]: build_local_reparse(): ERROR: build_local_reparse: HDR_EOH_T was not found Message-ID: <56B79AFD.8010505(a)gmail.com> Content-Type: text/plain; charset="windows-1252" Hello, do you have an action like: append_hf("Remote-Party-ID: ..."); in kamailio.cfg? Can you paste all that action here? Cheers, Daniel

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Registration Failure
by Cristina Caridi 19 Feb '16

19 Feb '16

Hello, I'm a new starter of Kamailio (SER). I am using the version 4.4.0-dev6 in a Debian machine. I've configured all the IMS network elements, the FHoSS and also Boghe IMS/RCS client in a Windows machine. When an IMS user belonging to my network tries to register and IP SEC mechanism is enabled, the registration flow stops at 401 Unauthorized from S-CSCF -> I-CSCF -> P-CSCF -> client. The problem is that the HSS in MAA message does not send the SIP-Auth-Data-item that should be cointain the Confidentiality Key and the Integrity Key; so the client is not able to challenge the S-CSCF sending another REGISTER message. Do you have any suggestions about this? When IP SEC is disabled (by managing client options), the registration works well. Maybe is there any configuration at HSS level that i'm not considering? Thank you in advance for your help, Cristina

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ACC Module
by Alberto Sagredo 19 Feb '16

19 Feb '16

I have a strange behaviour when configuring ACC module to put CDRs It must be as simple as this: modparam("acc", "cdr_enable", 1) But got this error: feb 19 09:26:14 kamailio /usr/local/sbin/kamailio[23712]: ERROR: acc [acc_mod.c:604]: mod_init(): failed to init cdr generation feb 19 09:26:14 kamailio /usr/local/sbin/kamailio[23712]: ERROR: <core> [sr_module.c:962]: init_mod(): Error while initializing module acc (/usr/local/lib64/kamailio/modules/acc.so) I have read documentation and seems ok. Tables in databases are created and ALTER commands applied. IM getting actually ACC registers on acc table, but could not active cdrs for acc. What am i missing?

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dispatcher flag
by alexandre.anconetti＠orange.com 19 Feb '16

19 Feb '16

Hi, I'm using the dispatcher module with Kamailio 4.3.4, the keep alive mechanism and the routing are OK, but I would like to get in my request_route the flag(s) (active, inactive, probing ...) that the dispatcher give to each destination. I know that in command line I can run kamcmd or kamctl to get : NRSETS: 1 RECORDS: { SET: { ID: 1 TARGETS: { DEST: { URI: sip: xxx.xxx.xxx.xxx:xxxx FLAGS: IP PRIORITY: 0 } DEST: { URI: sip:xxx.xxx.xxx.xxx:xxxx FLAGS: AP PRIORITY: 0 How can I get that in the request_route process ? Any suggestion would be appreciated ;) Best Regards, Alexandre. _________________________________________________________________________________________________________________________ Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration, Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci. This message and its attachments may contain confidential or privileged information that may be protected by law; they should not be distributed, used or copied without authorisation. If you have received this email in error, please notify the sender and delete this message and its attachments. As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified. Thank you.

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LCR Module issue
by Alberto Sagredo 18 Feb '16

18 Feb '16

Hi I have LCR working but not able to add more gateways? It seems only first one is accepted What could be wrong? I show you database data and kamcmd commands output. mysql> select * from lcr_rule_target; +----+--------+---------+-------+----------+--------+ | id | lcr_id | rule_id | gw_id | priority | weight | +----+--------+---------+-------+----------+--------+ | 1 | 1 | 1 | 1 | 1 | 1 | | 2 | 1 | 2 | 1 | 1 | 1 | | 3 | 2 | 2 | 2 | 1 | 1 | +----+--------+---------+-------+----------+--------+ 3 rows in set (0.00 sec) mysql> select * from lcr_rule; +----+--------+-----------+----------+-------------+---------+---------+ | id | lcr_id | prefix | from_uri | request_uri | stopper | enabled | +----+--------+-----------+----------+-------------+---------+---------+ | 1 | 1 | 6 | NULL | NULL | 1 | 1 | | 2 | 1 | DIDNUMBER | NULL | NULL | 1 | 1 | | 3 | 2 | DIDNUMBER | NULL | NULL | 1 | 1 | +----+--------+-----------+----------+-------------+---------+---------+ 3 rows in set (0.00 sec) mysql> select * from lcr_gw; +----+--------+----------------+--------------+----------+------+--------+------------+-----------+-------+--------+------+-------+---------+ | id | lcr_id | gw_name | ip_addr | hostname | port | params | uri_scheme | transport | strip | prefix | tag | flags | defunct | +----+--------+----------------+--------------+----------+------+--------+------------+-----------+-------+--------+------+-------+---------+ | 1 | 1 | Carrier_Prueba | 172.x.x.x | NULL | 5060 | NULL | 1 | 1 | 0 | NULL | NULL | 1 | NULL | | 5 | 2 | Proveedor_1 | 172.x.x.x | NULL | 5060 | NULL | 1 | 1 | 0 | NULL | NULL | 0 | NULL | +----+--------+----------------+--------------+----------+------+--------+------------+-----------+-------+--------+------+-------+---------+ 2 rows in set (0.00 sec) mysql> root@kamailio:~# kamcmd lcr.reload root@kamailio:~# kamcmd lcr.dump_rules { lcr_id: 1 rule_id: 2 prefix: DIDNUMBER from_uri: request_uri: stopper: 1 } { gw_index: 1 priority: 1 weight: 1 } { lcr_id: 1 rule_id: 1 prefix: 6 from_uri: request_uri: stopper: 1 } { gw_index: 1 priority: 1 weight: 1 } 9 1 root@kamailio:~# kamcmd lcr.dump_gws { lcr_id: 1 gw_id: 1 gw_index: 1 gw_name: Carrier_Prueba scheme: sip: ip_addr: 172.x.x.x hostname: port: 5060 params: transport: ;transport=udp strip: 0 prefix: tag: flags: 1 state: 0 defunct_until: 0

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Questions
by francisco rodriguez 18 Feb '16

18 Feb '16

Hi; Today I'm trying to use my users and include unregistered, there is a problem, this was solved soon or do. I appreciate your attention and cooperation Francisco Rodriguez

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Re: [SR-Users] Kamailio IMS deployment
by sainath.ellendula＠wipro.com 17 Feb '16

17 Feb '16

Hi Franz, Thanks for your response. Even we observed in Wireshark traces, no username is included in REGISTER request. Any changes required? No, we are not using NATURAL in our platform. Observed there is no communication between PCSCF – ICSCF while registration and also attaching the tracefile(tcpdump) using command “tcpdump -i any -s 3000 -w /var/tmp/trace_OpenIMS.pcap not port 22 and not port 3306”. Please suggest me if I have missed anything. Regards, Sainath From: Franz Edler [mailto:franz.edler@technikum-wien.at] Sent: Tuesday, February 9, 2016 1:01 PM To: Sainath Ellendula (NEP) <sainath.ellendula(a)wipro.com> Cc: basitstar(a)hotmail.com; sr-users(a)lists.sip-router.org Subject: AW: RE: Re:[SR-Users] Kamailio IMS deployment The REGISTER request is OK. There is no username included in REGISTER request. Regarding the mismatch: do you use NATURAL on your platform? Next step should be a tracefile (tcpdump) and later a debug-log from the node which shows the problem. Von Samsung-Tablet gesendet -------- Ursprüngliche Nachricht -------- Von: sainath.ellendula(a)wipro.com<mailto:sainath.ellendula@wipro.com> Datum: 2016.02.09 08:05 (GMT+01:00) An: franz.edler(a)technikum-wien.at<mailto:franz.edler@technikum-wien.at> Cc: basitstar(a)hotmail.com<mailto:basitstar@hotmail.com>, sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org> Betreff: RE: Re:[SR-Users] Kamailio IMS deployment Hi Franz, Please find ping status and seems good. root@server:~# ping -c 1 hss.net1.test PING hss.net1.test (10.0.0.9) 56(84) bytes of data. 64 bytes from server.net1.test (10.0.0.9): icmp_seq=1 ttl=64 time=0.017 ms --- hss.net1.test ping statistics --- 1 packets transmitted, 1 received, 0% packet loss, time 0ms rtt min/avg/max/mdev = 0.017/0.017/0.017/0.000 ms root@server:~# ping -c 1 pcscf.net1.test PING pcscf.net1.test (10.0.0.10) 56(84) bytes of data. 64 bytes from server.net1.test (10.0.0.10): icmp_seq=1 ttl=64 time=0.020 ms --- pcscf.net1.test ping statistics --- 1 packets transmitted, 1 received, 0% packet loss, time 0ms rtt min/avg/max/mdev = 0.020/0.020/0.020/0.000 ms root@server:~# ping -c 1 icscf.net1.test PING icscf.net1.test (10.0.0.11) 56(84) bytes of data. 64 bytes from server.net1.test (10.0.0.11): icmp_seq=1 ttl=64 time=0.018 ms --- icscf.net1.test ping statistics --- 1 packets transmitted, 1 received, 0% packet loss, time 0ms rtt min/avg/max/mdev = 0.018/0.018/0.018/0.000 ms root@server:~# ping -c 1 scscf.net1.test PING scscf.net1.test (10.0.0.12) 56(84) bytes of data. 64 bytes from server.net1.test (10.0.0.12): icmp_seq=1 ttl=64 time=0.017 ms --- scscf.net1.test ping statistics --- 1 packets transmitted, 1 received, 0% packet loss, time 0ms rtt min/avg/max/mdev = 0.017/0.017/0.017/0.000 ms root@server:~# Seems Some progress happened: Tried register using Boghe IMS client and observed few errors: 0(1445) INFO: <script>: [REGISTER] from [sip:alice@net1.test] to [sip:alice@net1.test] 0(1445) ERROR: <script>: REGISTER (sip:alice@net1.test (10.0.0.8:55280) to sip:alice@net1.test, 2006cdca-0076-bacb-5010-bffa714b3cdb) 0(1445) DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <7994> <REGISTER> 0(1445) DEBUG: <core> [parser/msg_parser.c:187]: get_hdr_field(): DEBUG: get_hdr_body : content_length=0 0(1445) DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 70 0(1445) DEBUG: <core> [parser/msg_parser.c:89]: get_hdr_field(): found end of header 0(1445) DEBUG: sanity [mod_sanity.c:248]: w_sanity_check(): sanity checks result: 1 0(1445) DEBUG: siputils [checks.c:97]: has_totag(): no totag 0(1445) DEBUG: <core> [select.c:412]: run_select(): Calling SELECT 0x7fb548dd2400 0(1445) DEBUG: tm [t_lookup.c:1011]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=1 T start=0xffffffffffffffff 0(1445) DEBUG: tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=65064, isACK=0 0(1445) DEBUG: tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching failed 0(1445) DEBUG: tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found 0(1445) DEBUG: tm [t_lookup.c:1080]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=2 T end=(nil) 0(1445) DEBUG: sl [sl.c:280]: send_reply(): reply in stateless mode (sl) 0(1445) DEBUG: siputils [checks.c:97]: has_totag(): no totag 0(1445) DEBUG: tm [t_lookup.c:1011]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=2 T start=(nil) 0(1445) DEBUG: tm [t_lookup.c:1086]: t_check_msg(): DEBUG: t_check_msg: T previously sought and not found 0(1445) DEBUG: ims_registrar_pcscf [save.c:273]: save_pending(): contact requesting to expire in 600000 seconds 0(1445) DEBUG: ims_usrloc_pcscf [usrloc.c:106]: get_alias_host_from_contact(): no alias param 0(1445) DEBUG: ims_usrloc_pcscf [usrloc.c:155]: get_hash_slot(): using host for hash [10.0.0.8] 0(1445) DEBUG: ims_usrloc_pcscf [usrloc.c:159]: get_hash_slot(): Returning hash slot: [167] 0(1445) DEBUG: ims_usrloc_pcscf [udomain.c:465]: get_pcontact(): Searching for contact in P-CSCF usrloc [sip:alice@10.0.0.8:55280;transport=udp] 0(1445) DEBUG: ims_usrloc_pcscf [usrloc.c:106]: get_alias_host_from_contact(): no alias param 0(1445) DEBUG: ims_usrloc_pcscf [usrloc.c:183]: get_aor_hash(): using host for hash [10.0.0.8] 0(1445) DEBUG: ims_usrloc_pcscf [usrloc.c:187]: get_aor_hash(): Returning hash slot: [1662767783] 0(1445) DEBUG: ims_usrloc_pcscf [udomain.c:483]: get_pcontact(): Comparing needle user@host:port [alice@10.0.0.8:55280] and contact_user@contact_host:port [alice@10.0.0.8:51219] 0(1445) DEBUG: ims_usrloc_pcscf [udomain.c:494]: get_pcontact(): Searching for [sip:alice@10.0.0.8:55280;transport=udp] and comparing to [sip:alice@10.0.0.8:51219;transport=udp] 0(1445) DEBUG: ims_usrloc_pcscf [udomain.c:553]: get_pcontact(): Port don't match: 51219 (contact) 51219 (received) != 55280! 0(1445) DEBUG: ims_registrar_pcscf [save.c:310]: save_pending(): Adding pending pcontact: <sip:alice@10.0.0.8:55280;transport=udp> 0(1445) DEBUG: ims_usrloc_pcscf [pcontact.c:132]: new_pcontact(): New contact [<sip:alice@10.0.0.8:55280;transport=udp>] with 0 associated IMPUs in state: [registration pending] 0(1445) DEBUG: ims_registrar_pcscf [save.c:314]: save_pending(): registering for UL callback 0(1445) DEBUG: ims_usrloc_pcscf [usrloc.c:106]: get_alias_host_from_contact(): no alias param Please guide me for any more changes has to be done, As observed in Wireshark(Register request from 10.0.0.8(host)-PCSCF as REGISTER sip:net1.test SIP/2.0). I think the register request should sent as REGISTER sip:alice@net1.test SIP/2.0 Thanks in advance!! Regards, Sainath From: Franz Edler [mailto:franz.edler@technikum-wien.at] Sent: Tuesday, February 9, 2016 3:00 AM To: Sainath Ellendula (NEP) <sainath.ellendula(a)wipro.com<mailto:sainath.ellendula@wipro.com>> Cc: basitstar(a)hotmail.com<mailto:basitstar@hotmail.com>; sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org> Subject: RE: Re:[SR-Users] Kamailio IMS deployment Sainath, what was the result for ping on - icsct.net1.test - scscf.net1.test - hss.net1.test to increase debug level you have to modify the corresponding cfg-file, e.g. kamailio-icscf.cfg in line 35: debug=2 Just increase to debug=3 and then start the process again. BR Franz The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com<http://www.wipro.com> The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

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[TOPOH] Contact header for 302 Moved Temporarily
by SamyGo 17 Feb '16

17 Feb '16

Hi All, I've recently stumbled upon this little hitch while using kamailio with topoh module that the Contact header do not contains the User part for the 302 Moved temporarily packet. *My topology:* UserA<====\ ->Kamailio<===>FreeSwitch UserB<====/ The B party has set Call forwarding on their phone hence phone sends a 302 Moved Temporarily to Kamailio. Via: SIP/2.0/UDP TOPOH.KAMAILIO.IP:5060;branch=z9hG4bK0767.4e4e57deca89da5c5f5ae97228325f85.0 Via: SIP/2.0/UDP TOPOH.KAMAILIO.IP;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.y7MGZLMGZwM.1LgRWIC9gIgx4fMGZAOBVAOBNfzuaVHRaYpB1LNSQLpx4uMx3Az6eL3RsBCxu-zRrUWSeOgjeBk.IVm4ds34aONc** From: "+4319714111" <sip:+4319714111@FREESWITCH.IP.HERE>;tag=yUr5UZ7eF794K To: <sip:502@USER.B.IP.HERE:5060>;tag=105223296 Call-ID: cd811276-4b4d-1234-66ae-005056867dbc CSeq: 87257355 INVITE Contact: <sip:06606017597@FREESWITCH.IP.HERE:5060> User-Agent: Yealink SIP-T46G 28.80.0.70 Diversion: <sip:502@USER.B.IP.HERE:5060>;reason=unconditional Content-Length: 0 This is modified in Kamailio TOPOH and sent to FreeSwitch as following SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.0.20.71;received=10.0.20.71;rport=5060;branch=z9hG4bKeBcy9HKK9cDKr From: "+4319714111" <sip:+4319714111@FREESWITCH.IP.HERE>;tag=yUr5UZ7eF794K To: <sip:502@USER.B.IP.HERE:5060>;tag=105223296 Call-ID: cd811276-4b4d-1234-66ae-005056867dbc CSeq: 87257355 INVITE Contact: <*sip:TOPOH.KAMAILIO.IP* ;line=sr-N6IAzBy6WBy6MxFwW.qwPSW5ohWINhaYNLu4g9W4OhWI3wKLgRsIpUg5kGs7g9P-W.y6My**> User-Agent: Yealink SIP-T46G 28.80.0.70 Diversion: <sip:502@USER.B.IP.HERE:5060>;reason=unconditional Content-Length: 0 Which results in a Call Originate from FreeSwitch with RURI as this: INVITE *sip:TOPOH.KAMAILIO.IP*;line=sr-N6IAzBy6WBy6MxFwW.qwPSW5ohWINhaYNLu4g9W4OhWI3wKLgRsIpUg5kGs7g9P-W.y6My** SIP/2.0 I am thinking that in TOPOH Module some patch is required to atleast retain the $rU for 3XX replies , not sure if this will break some RFC or Kamailio stability etc ! Thanks, Sammy.

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siptrace module plain sip mode
by Grant Bagdasarian 17 Feb '16

17 Feb '16

Hello, The documentation of the siptrace module specifies duplication of sip messages to a capture server using the HEP protocol or plain SIP mode. Which params to I need to set (and with which values) in the siptrace and sipcapture to enable plain SIP mode? I tried several configurations but none of the worked. Seems like only HEP works. My test kamailio (sip capture node with sipcapture module) doesn't receive the duplicated SIP messages. Any ideas? Regards, Grant

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Installatioin issues
by Ed Todd 17 Feb '16

17 Feb '16

I am installing KAMAILIO under DEBIAN. I have got it to work fine (TLS) until I upgraded my Comcast to business service to get static IP’s. Now it no longer works. I am using the Cisco DPC3941 modem/router and am tring to connect an IPhone on WIFI to the PC SIP server on the LAN. Here are the log entries: Feb 16 19:14:59 SIP-Server kamailio[2010]: NOTICE: <core> [main.c:738]: handle_sigs(): Thank you for flying kamailio!!! Feb 16 19:14:59 SIP-Server kamailio[2019]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 16 20:01:25 SIP-Server kamailio: INFO: tls [tls_init.c:385]: init_tls_compression(): tls: init_tls: disabling compression... Feb 16 20:01:26 SIP-Server kamailio: INFO: <core> [tcp_main.c:4745]: init_tcp(): using epoll_lt as the io watch method (auto detected) Feb 16 20:01:26 SIP-Server kamailio[3876]: INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob Feb 16 20:01:26 SIP-Server kamailio[3876]: INFO: rr [rr_mod.c:160]: mod_init(): outbound module not available Feb 16 20:01:26 SIP-Server kamailio[3876]: INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 1024 Feb 16 20:01:26 SIP-Server kamailio[3876]: ERROR: ctl [init_socks.c:126]: init_unix_sock(): ERROR: init_unix_sock: bind: No such file or directory [2] Feb 16 20:01:26 SIP-Server kamailio[3876]: ERROR: ctl [ctl.c:283]: mod_init(): ERROR: ctl: mod_init: init ctrl. sockets failed Feb 16 20:01:26 SIP-Server kamailio[3876]: ERROR: <core> [sr_module.c:968]: init_mod(): Error while initializing module ctl (/usr/lib/x86_64-linux-gnu/kamailio/modules/ctl.so) I am informed that SIP-ALG is disabled. Any ideas?

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Remove_hf_is not working
by aman jeff 17 Feb '16

17 Feb '16

Hi , I am working kamailio 4.4.0 ,i want to remove my Contact header completely.So I used remove_hf..but its now working ..Let me know the solution Am very thankful to you Regards Jefff

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db_cluster.so module and kamctlrc DBHOST= field
by Hosted Services 16 Feb '16

16 Feb '16

Hi there, New user to Kamailio here. We currently have it up and running in a virtualized environment with 1 Kamailio sever, 1 Asterisk server and 1 MySQL server. I'm currently writing install scripts to make deploying new nodes/servers easy and to keep settings the same across the board. I've chosed to load the db_cluster.so module in kamailio.cfg, as we will have 2x MySQL servers in master-master replication which will contain the 'kamailio' and 'asterisk' tables. I've just hit a stumbling block - in `kamctlrc`, there is a field called `DBHOST=`. How can I reference my cluster here? In kamailio.cfg, I simply define DBURL as "cluster//<clustername>". What is the syntax for 'DBHOST=' in 'kamctlrc'? Can I reference the cluster? Can I have 2 separate DBHOST= lines? Looking for some guidance on this one. Thanks, Derek B.

2 1

Database Insert
by Daniel W. Graham 16 Feb '16

16 Feb '16

I see there is a new SQLops function sql_query_async that has been added in 4.2 (I believe?). Is there a good way to perform database insert using version 4.0? Using mysql Thanks! -dan

2 4

Need to validate a IP address with subnet mask
by Sunil More 16 Feb '16

16 Feb '16

Hi All, I want to validate a IP address in kamailio for requests coming from different IPs. I have browsed and found permissions module, however is there a way where i can directly compare a source IP address to an address stored in an avp and get a true or false value. something like this $si==192.168.2.0/24 Thanking You, Sunil More

3 7

Textops and Multipart Body - adding ISUP
by Sergey Basov 16 Feb '16

16 Feb '16

Hi All. I need to use SIP-I with my upstream to set A number as unknown. I was read about adding incapsulation ISUP using if(has_body("application/sdp")) { set_body_multipart(); msg_apply_changes() $var(acm) = "7e Od 04 55 75 69 20 4d 61 6b 65 43 61 6c 6c"; append_body_part("$var(acm)","application/isup;version=itu-t92+","signal; handling=optional"); msg_apply_changes() xlog("L_INFO", "ISUP Changes Applied Succesfully"); } It is converts body to multipart and inserts ISUP But it inserts is as text. Is there any possibility to insert basic ISUP message to update it after using sipt_destination($rU, 31, 4); and sipt_set_calling($fU, 4, 0, 3); Thank you. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com

3 10

kamailio 4 with asterisk
by Sébastien Brice 15 Feb '16

15 Feb '16

this one (written by Daniel) http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb maybe a bit outdated but still consistent ? the thing i am really stuck with (and concerning real-time) is that none of my extensions (from asterisk CLI) are online: ns3325046*CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime 102/102 (Unspecified) D Auto (No) No 0 Unmonitored Cached RT 103/103 (Unspecified) D Auto (No) No 0 Unmonitored Cached RT 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] that drives asterisk crazy ! and logger reports: app_dial.c:2411 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) every time i place a call. The tutorial written by daniel mention a channel configuration pretty minimal: INSERT INTO sipusers (name, defaultuser, host, sippasswd, fromuser, fromdomain, mailbox) VALUES ('102', '102', 'dynamic', '102', '102', 'yoursip.com', '102'); and since there's no context associated to the 102 extension i cant figure out where that channel enter the dialplan ? [public] [LocalSet] [default] ???? and a dialplan exten => _1XX,1,Dial(SIP/${EXTEN}) exten => _1XX,n,Voicemail(${EXTEN},u) exten => _1XX,n,Hangup exten => _1XX,101,Voicemail(${EXTEN},b) exten => _1XX,102,Hangup i am sorry to bother you with issues more asterisk oriented than kamailio. By the way i took a good start with kamailio as it seems to work flawlessly on my system. thx you. On Mon, Feb 15, 2016 at 12:26:06PM +0100, Sébastien Brice wrote: > Hi Everyone, i like the way this tutorial explain asterisk and kamailio integration. Which tutorial? > the only thing i missed is asterisk behaviors'r regarding sip registration ? That was a part of a tutorial I once saw. In essence asterisk uses the kamailio database, UA registers on kamailio and is stored there, asterisk sees the same data (realtime). Sébastien BRICE VoIP, Support et Intégration

2 1

no pkg memory left
by Gholamreza Sabery 15 Feb '16

15 Feb '16

Hello. I am seeing the following error repeating again and again in the log file: /sbin/kamailio[14510]: ERROR: registrar [reply.c:199]: build_contact(): no pkg memory left I am using 64MB of SHM memory and 8MB of PKG memory. What is the reason for this error? How to solve it?

2 2

kamailio 4 with asterisk
by Sébastien Brice 15 Feb '16

15 Feb '16

Hi Everyone, i like the way this tutorial explain asterisk and kamailio integration. the only thing i missed is asterisk behaviors'r regarding sip registration ? I tryed to place a call between 102 and 103 extensions and experimenting an issue Asterisk tells me that the subscriber is absent and I'm sent directly to voicemail ! -- Executing [103@public:1] Dial("SIP/102-00000001", "SIP/103") in new stack [Feb 14 21:00:15] WARNING[19444][C-00000001]: app_dial.c:2411 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) ns3325046*CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime 102/102 (Unspecified) D Auto (No) No 0 Unmonitored Cached RT 103/103 (Unspecified) D Auto (No) No 0 Unmonitored Cached RT 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] apart that my sip.conf and extensions.conf are very minimal: [LocalSets] exten => _1XX,1,Dial(SIP/${EXTEN}) exten => _1XX,n,Voicemail(${EXTEN},u) exten => _1XX,n,Hangup exten => _1XX,102,Voicemail(${EXTEN},b) exten => _1XX,103,Hangup [general] context=LocalSets ; Default context for incoming calls. Defaults to 'default' rtcachefriends=yes ; Cache realtime friends by adding them to the internal list i did INSERT TO the users in mysql tables (sipusers, sipregs and voicemail) and registering extensions from UA works ok (i am using jitsi) Whats wrong with my setup ? thank you

2 1

kamctl add VS kamctl ul add under SIPML
by kssyrr 15 Feb '16

15 Feb '16

2 3

uac module configuration
by Andrew Tan 15 Feb '16

15 Feb '16

Hello, Submit a ticket in GitHub apologize about that I should have sent my question here. I would like some help in configuring the uac module to route to an gateway without specifically setting the URI-domain from the UAC. I would like to set change it in the Kamailio side. Currently I use uac_auth on failure route. All is working if I set the URI from on the client to xxx(a)gatewayaddress.com. But if I just dial just the number as is which translate to xxx(a)kamailioserver.com the invite doesnt go anywhere. My failure route is like so failure_route[MANAGE_FAILURE] { xlog("DEBUG: MANAGE FAILURE ROUTE: "); if ($T_reply_code == "407") # there's also 401 { $avp(arealm) = GATEWAY; $avp(auser) = USERNAME; $avp(apass) = PASSWORD; #$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); #rewritehost("sip.ca.didlogic.net"); uac_auth(); rtpengine_offer("force trust-address replace-origin replace-session-connection ICE=remove RTP/AVP"); route(NATMANAGE); route(RELAY); exit; } } Thank you

2 1

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