- sr-users - kamailio.org

authcheck and fd variable
by Robert Lieback 10 Dec '19

10 Dec '19

Hello, i'm looking into kamalio (btw: thanks to asipto for the great book) and i hope someone can push me in the right direction. I have a question/problem with some lines in the default config (kamailio.cfg) around line 754: ... if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } ... auth_check() and auth_challenge doesn't work, but i've provided the correct credentials. When i replace "$fd" with domain "voip.XXXXXX.de" it works. This domain is also defined as alias in the config and all sip messages i send to the server have a correct from header with this domain. First step for me would be to look whats stored in $fd, but how can i output the content of $fd? Cause log(2, "$fd") or xlog("$fd") doesn't work and throws errors about unknown variables. Thanks, Robert Lieback

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Kamailio management group updates
by Daniel-Constantin Mierla 09 Dec '19

09 Dec '19

Hello, Jesus Rodrigues and Inaki Baz Castillo changed their work roles a while ago, so they no longer deal with Kamailio and they do not expect to do it in the near future. To keep the Management group with members connected as much as possible with the project, Fred Posner and Victor Seva were proposed to replace them. For the new members in the public lists: the management group of Kamailio is not involved in the daily development or administration of the project -- all those are done on public mailing lists or kadmin group (internal infrastructure administration) -- management team is supposed to take the decisions on sensitive situations for the evolution of the project. Should anyone have opinions about these changes, let's discuss on sr-users mailing list. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

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duplicated registrations
by Liu, Charles 08 Dec '19

08 Dec '19

Hi All, Does this look correct? I am using Zoiper client and did multiple registration/re-registration, but I saw duplicated contacts in the database for the same user/device(alice). I was expecting to see only one. Is this a bug? How do I prevent this happening? "select * from scscf.contact" id contact params path received user_agent expires callid 49 sip:bob@192.168.10.201:60944;rinstance=44b4024df17bd284;transport=UDP NULL <sip:term@pcscf.mnc001.mnc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-08 10:31:51 r7jCYp7JuVbA8QjocnouLw.. 64 sip:alice@192.168.10.241:60383;rinstance=e850d3ed0605bcd3;transport=UDP NULL <sip:term@pcscf.mnc001.mnc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-08 10:31:11 EHLOADkLUEe8d0CMpdLL9A.. 142 sip:alice@192.168.10.241:60383;rinstance=bc755d87cc438890;transport=UDP NULL <sip:term@pcscf.mnc001.mnc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-08 10:31:51 P1rBkHGUVdv7oqW3_UWQPg.. 143 sip:alice@192.168.10.241:60383;rinstance=146303cf7501a125;transport=UDP NULL <sip:term@pcscf.mnc001.mnc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-08 10:32:15 etUEoAGSM4q4e_39wFKa7w.. 144 sip:alice@192.168.10.241:60383;rinstance=e0a260b7b78ecf6b;transport=UDP NULL <sip:term@pcscf.mnc001.mnc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-08 10:32:17 WUtIYeGlBZmHD3TGklmKCQ.. 145 sip:alice@192.168.10.241:60383;rinstance=2681245431c4e4e4;transport=UDP NULL <sip:term@pcscf.mnc001.mnc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-08 10:32:19 HNZPyeO173jfscKYNOT0bw.. Thanks, --Charles CONFIDENTIALITY NOTICE: This email and any attachments are for the sole use of the intended recipient and may contain material that is proprietary, confidential, privileged or otherwise legally protected or restricted under applicable government laws. Any review, disclosure, distributing or other use without expressed permission of the sender is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies without reading, printing, or saving.

2 2

ul.dump empty
by Liu, Charles 07 Dec '19

07 Dec '19

Hi All, I am running Kamailio v5.2.4 with P/S/I-cscf instances on 3 different machines with default IMS example configurations. Kamcmd ul.dump show empty, but I can make calls... any idea what is the problem? root@pcscf:/# kamcmd ulpcscf.status { Domain: location Size: 512 AoRs: { Stats: { Records: 7 Max-Slots: 6 } } } root@pcscf:/# kamcmd ul.dump { Domains: { } } CONFIDENTIALITY NOTICE: This email and any attachments are for the sole use of the intended recipient and may contain material that is proprietary, confidential, privileged or otherwise legally protected or restricted under applicable government laws. Any review, disclosure, distributing or other use without expressed permission of the sender is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies without reading, printing, or saving.

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Pike module never release IP
by Andrew Chen 06 Dec '19

06 Dec '19

Hi all, Have anyone ever experienced pike module not releasing the IP after "remove_latency" period? We ran a load test and this is what happened to one of the two paired nodes and I had to restart kamailio process to release it. I'm running Kam 5.2.5 Thanks. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

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How can I make calls between the two domains?
by Yanko 06 Dec '19

06 Dec '19

Hello: I have two virtual machines with two kamailo domains VM1: Domain: 172.16.16.1 Users:2000, 2001, 2002... VM2: Domain: 172.16.16.2 Users:3000, 3001, 3002... In each domain I can make calls. For example, user 2000 can call 2001. User 3000 can call 3001. How can I make calls between the two domains? What modules do I need to load? How to configure the kamailio.cfg file? Thanks, YANKO.

2 1

Scripts still valid?
by David Villasmil 05 Dec '19

05 Dec '19

Hello guys, Is this still valid? https://github.com/kamailio/kamailio/blob/master/misc/examples/scripts/ctd.… Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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call disconnected between 15 to 45 minutes
by Liu, Charles 05 Dec '19

05 Dec '19

Hi all, I am still looking for some help to debug why S-CSCF send BYE to caller and callee after a call was connected. from wireshark, I just saw S-CSCF send BYE to both caller and callee. That matched the S-CSCF debug output. But the question is why S-CSCF delete contact and send BYE to a active call? There was no timer expire or something like that in the log... Anyone can help? Thanks, --Charles 19(24) DEBUG: <core> [core/mem/q_malloc.c:519]: qm_free(): freeing frag. 0x7fb09413e6d0 alloc'ed from cdp: diameter_msg.c: AAATranslateMessage(484) 19(24) DEBUG: tm [t_lookup.c:1618]: t_lookup_ident(): transaction found 19(24) DEBUG: <core> [core/mem/q_malloc.c:367]: qm_malloc(): qm_malloc(0x7fb093db6000, 6872) called from core: core/sip_msg_clone.c: sip_msg_shm_clone(496) 19(24) DEBUG: <core> [core/mem/q_malloc.c:412]: qm_malloc(): qm_malloc(0x7fb093db6000, 6872) returns address 0x7fb09414d450 frag. 0x7fb09414d418 (size=6872) on 1 -th hit 19(24) ERROR: *** cfgtrace:failure_route=[REG_SAR_REPLY] c=[/etc/kamailio/kamailio.cfg] l=767 a=26 n=xlog 19(24) DEBUG: <script>: saa_return code is 1 19(24) ERROR: *** cfgtrace:failure_route=[REG_SAR_REPLY] c=[/etc/kamailio/kamailio.cfg] l=769 a=26 n=xlog 19(24) DEBUG: <script>: saa_return code (for scscf_save on register) is 1 19(24) ERROR: *** cfgtrace:failure_route=[REG_SAR_REPLY] c=[/etc/kamailio/kamailio.cfg] l=783 a=21 n=switch 19(24) ERROR: *** cfgtrace:failure_route=[REG_SAR_REPLY] c=[/etc/kamailio/kamailio.cfg] l=772 a=26 n=xlog 19(24) DEBUG: <script>: SAR success - 200 response sent from module 19(24) ERROR: *** cfgtrace:failure_route=[REG_SAR_REPLY] c=[/etc/kamailio/kamailio.cfg] l=773 a=2 n=exit 19(24) DEBUG: <core> [core/mem/q_malloc.c:475]: qm_free(): qm_free(0x7fb093db6000, 0x7fb09414d450), called from tm: t_reply.c: free_faked_req(948) 19(24) DEBUG: <core> [core/mem/q_malloc.c:519]: qm_free(): freeing frag. 0x7fb09414d418 alloc'ed from core: core/sip_msg_clone.c: sip_msg_shm_clone(496) 19(24) DEBUG: <core> [core/mem/q_malloc.c:475]: qm_free(): qm_free(0x7fb093db6000, 0x7fb09413e978), called from ims_registrar_scscf: cxdx_sar.c: free_saved_transaction_data(94) 19(24) DEBUG: <core> [core/mem/q_malloc.c:519]: qm_free(): freeing frag. 0x7fb09413e940 alloc'ed from ims_registrar_scscf: save.c: save(1414) 19(24) DEBUG: <core> [core/mem/q_malloc.c:475]: qm_free(): qm_free(0x7fb093db6000, 0x7fb09413e0b0), called from ims_registrar_scscf: reply.c: free_contact_buf(830) 19(24) DEBUG: <core> [core/mem/q_malloc.c:519]: qm_free(): freeing frag. 0x7fb09413e078 alloc'ed from ims_registrar_scscf: reply.c: build_contact(443) 19(24) DEBUG: <core> [core/mem/q_malloc.c:475]: qm_free(): qm_free(0x7fb093db6000, 0x7fb094145218), called from ims_registrar_scscf: reply.c: free_contact_buf(835) 19(24) DEBUG: <core> [core/mem/q_malloc.c:519]: qm_free(): freeing frag. 0x7fb0941451e0 alloc'ed from ims_registrar_scscf: reply.c: build_contact(435) 19(24) DEBUG: <core> [core/mem/q_malloc.c:475]: qm_free(): qm_free(0x7fb093db6000, 0x7fb094147dc0), called from ims_registrar_scscf: cxdx_sar.c: free_saved_transaction_data(98) 19(24) DEBUG: <core> [core/mem/q_malloc.c:519]: qm_free(): freeing frag. 0x7fb094147d88 alloc'ed from ims_registrar_scscf: save.c: save(1396) 19(24) DEBUG: <core> [core/mem/q_malloc.c:475]: qm_free(): qm_free(0x7fb093db6000, 0x7fb09415b4c8), called from cdp: transaction.c: cdp_free_trans(182) 19(24) DEBUG: <core> [core/mem/q_malloc.c:519]: qm_free(): freeing frag. 0x7fb09415b490 alloc'ed from cdp: transaction.c: cdp_add_trans(110) 19(24) DEBUG: <core> [core/mem/q_malloc.c:475]: qm_free(): qm_free(0x7fb093db6000, 0x7fb0941442a0), called from cdp: transaction.c: cdp_free_trans(183) 19(24) DEBUG: <core> [core/mem/q_malloc.c:519]: qm_free(): freeing frag. 0x7fb094144268 alloc'ed from cdp: transaction.c: cdp_add_trans(105) 13(18) DEBUG: ims_usrloc_scscf [ims_usrloc_scscf_mod.c:481]: timer(): Syncing cache 13(18) DEBUG: ims_usrloc_scscf [udomain.c:291]: mem_timer_udomain(): *** mem_timer_udomain - checking contacts - START *** 13(18) DEBUG: ims_usrloc_scscf [udomain.c:301]: mem_timer_udomain(): We have a [3gpp=0] contact in the new contact list in slot 95 = [sip:bob@ims.mnc001.mnc001.3gppnetwork.org] (sip:bob@192.168.164.248:50448;rinstance=989f943bd5098a11;transport=UDP) which expires in -60.000000 seconds and has a ref count of 0 (state: Contact with delayed delete) 13(18) DEBUG: ims_usrloc_scscf [udomain.c:309]: mem_timer_udomain(): contact in state CONTACT_DELATED_DELETE is about to be deleted13(18) DEBUG: ims_usrloc_scscf [udomain.c:301]: mem_timer_udomain(): We have a [3gpp=0] contact in the new contact list in slot 464 = [sip:alice@ims.mnc001.mnc001.3gppnetwork.org] (sip:alice@192.168.164.203:44640;rinstance=e35e9704884c3208;transport=UDP) which expires in 50.000000 seconds and has a ref count of 1 (state: Contact valid) 13(18) DEBUG: ims_usrloc_scscf [udomain.c:301]: mem_timer_udomain(): We have a [3gpp=0] contact in the new contact list in slot 478 = [sip:bob@ims.mnc001.mnc001.3gppnetwork.org] (sip:bob@192.168.164.248:50448;rinstance=9e742618a5f6bcae;transport=UDP) which expires in 56.000000 seconds and has a ref count of 1 (state: Contact valid) 13(18) DEBUG: ims_usrloc_scscf [udomain.c:353]: mem_timer_udomain(): *** mem_timer_udomain - checking contacts - FINISHED *** 13(18) DEBUG: ims_usrloc_scscf [udomain.c:359]: mem_timer_udomain(): *** mem_timer_udomain - checking IMPUs - START *** 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:372]: process_impurecord(): Checking validity of IMPU: <sip:alice@ims.mnc001.mnc001.3gppnetwork.org> registration subscriptions 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:404]: process_impurecord(): Public Identity sip:alice@ims.mnc001.mnc001.3gppnetwork.org, Barred: [0], State: [registered], contacts [1], 3gppcontacts [0] 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:441]: process_impurecord(): Contact #0 - sip:alice@192.168.164.203:44640;rinstance=e35e9704884c3208;transport=UDP, Ref [1] (expires in 50 seconds) (State: 0) 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:372]: process_impurecord(): Checking validity of IMPU: <sip:bob@ims.mnc001.mnc001.3gppnetwork.org> registration subscriptions 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:404]: process_impurecord(): Public Identity sip:bob@ims.mnc001.mnc001.3gppnetwork.org, Barred: [0], State: [registered], contacts [1], 3gppcontacts [0] 3(18) DEBUG: ims_usrloc_scscf [impurecord.c:441]: process_impurecord(): Contact #0 - sip:bob@192.168.164.248:50448;rinstance=9e742618a5f6bcae;transport=UDP, Ref [1] (expires in 56 seconds) (State: 0) 3(18) DEBUG: ims_usrloc_scscf [udomain.c:384]: mem_timer_udomain(): *** mem_timer_udomain - checking IMPUs - FINISHED *** 3(18) DEBUG: ims_usrloc_scscf [udomain.c:406]: mem_timer_udomain(): deleting contact [sip:bob@ims.mnc001.mnc001.3gppnetwork.org] 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:598]: delete_scontact(): Deleting contact: [sip:bob@192.168.164.248:50448;rinstance=989f943bd5098a11;transport=UDP] 13(18) DEBUG: ims_usrloc_scscf [usrloc_db.c:370]: db_delete_ucontact(): Deleting ucontact [sip:bob@192.168.164.248:50448;rinstance=989f943bd5098a11;transport=UDP] 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:310]: mem_delete_ucontact(): Checking if dialog_data is there and needs to be torn down 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:314]: mem_delete_ucontact(): first dialog is not 0 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:317]: mem_delete_ucontact(): Going to tear down dialog with info h_entry [1694] h_id [3889] 13(18) DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7fb09414ce40 with 1 -> 3 13(18) DEBUG: ims_dialog [dlg_hash.c:889]: lookup_dlg(): dialog id=3889 found on entry 1694 13(18) DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fb09414ce40 with 1 -> 2 13(18) DEBUG: ims_dialog [dlg_req_within.c:379]: dlg_terminate(): terminating confirmed dialog 13(18) DEBUG: <core> [core/mem/q_malloc.c:367]: qm_malloc(): qm_malloc(0x7fb09b5ae010, 18) called from ims_dialog: dlg_req_within.c: build_extra_hdr(229) 13(18) DEBUG: <core> [core/mem/q_malloc.c:412]: qm_malloc(): qm_malloc(0x7fb09b5ae010, 24) returns address 0x7fb09b640c30 frag. 0x7fb09b640bf8 (size=64) on 1 -th hit 13(18) DEBUG: <core> [core/mem/q_malloc.c:367]: qm_malloc(): qm_malloc(0x7fb09b5ae010, 248) called from ims_dialog: dlg_req_within.c: build_dlg_t(81) 13(18) DEBUG: <core> [core/mem/q_malloc.c:412]: qm_malloc(): qm_malloc(0x7fb09b5ae010, 248) returns address 0x7fb09b74b100 frag. 0x7fb09b74b0c8 (size=248) on 1 -th hit 13(18) DEBUG: <core> [core/mem/q_malloc.c:367]: qm_malloc(): qm_malloc(0x7fb09b5ae010, 72) called from core: core/parser/parse_rr.c: do_parse_rr_body(68) 13(18) DEBUG: <core> [core/mem/q_malloc.c:412]: qm_malloc(): qm_malloc(0x7fb09b5ae010, 72) returns address 0x7fb09b74b048 frag. 0x7fb09b74b010 (size=80) on 1 -th hit 13(18) DEBUG: ims_dialog [dlg_req_within.c:272]: send_bye(): sending BYE to caller 13(18) DEBUG: ims_dialog [dlg_hash.c:1047]: ref_dlg(): ref dlg 0x7fb09414ce40 with 1 -> 3 CONFIDENTIALITY NOTICE: This email and any attachments are for the sole use of the intended recipient and may contain material that is proprietary, confidential, privileged or otherwise legally protected or restricted under applicable government laws. Any review, disclosure, distributing or other use without expressed permission of the sender is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies without reading, printing, or saving.

2 2

Question regarding priority in dispatcher module
by Floimair Florian 05 Dec '19

05 Dec '19

Dear all! After reading the documentation on dispatcher module I still have a question that isn't really documented. Regarding the priority value of an entry in a dispatcher list: Is a lower numerical priority value regarded as higher priority or is it the other way around? Hopefully someone can help me to clarify this. With best regards FLORIAN FLOIMAIR Software Development Commend International GmbH Saalachstrasse 51 5020 Salzburg, Austria commend.com LG Salzburg / FN 178618z

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Re: [SR-Users] RTPengine with rejected RE-INVITE when port is changing
by Daniel Tryba 05 Dec '19

05 Dec '19

On Thu, Dec 05, 2019 at 09:37:51AM +0000, Laurent Schweizer wrote: > Hello, > > I already see old post about this : > https://opensips.org/pipermail/users/2014-November/030451.html > > but I???m interested to know if now they is a solution ???? > > so the issue is a RE-INVITE rejected (415 Unsupported Media or 488 ) and the RTP port is changing. > Is that case the old RTP relay (I???m using RTPengine) is destroyed after 30sec. > Any way to restore them ? I'm confused: - opensips != kamailio - rtpproxy != rtpengine But RFC 3261 states (page 16): During the session, either Alice or Bob may decide to change the characteristics of the media session. This is accomplished by sending a re-INVITE containing a new media description. This re- INVITE references the existing dialog so that the other party knows that it is to modify an existing session instead of establishing a new session. The other party sends a 200 (OK) to accept the change. The requestor responds to the 200 (OK) with an ACK. If the other party does not accept the change, he sends an error response such as 488 (Not Acceptable Here), which also receives an ACK. However, the failure of the re-INVITE does not cause the existing call to fail - the session continues using the previously negotiated characteristics. Full details on session modification are in Section 14. So if this happens with kamailio/rtpengine that is a bug in (I presume) rtpengine.

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siputils: how does sip_p_charging_vector generate the header
by George Diamantopoulos 05 Dec '19

05 Dec '19

Hello, I've been looking at the sip_p_charging_vector() function of siputils. From the documentation it is not evident how the PCV header is generated. Are the exported $pcv pseudovariables writable, so that the function canl use those when generating the header? Thanks! BR, George

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anycast + dmq_usrloc
by Nuno Miguel Reis 05 Dec '19

05 Dec '19

Hi guys. I'm setting up an anycast scenario with several kamailios sharing an active state and I've noticed that the dmq_usrloc module is not replication the usrloc socket_info attribute when a register comes in through one of the kamailio nodes. Is there a reason behind this? I'm thinking on adding the replication for that attribute in the code. Another way of address this woud be by having a modparam to replicate the socket. What do you think? This is what I'm seeing: *anycast node that actually gets the SIP REGISTER:* $ kamcmd ul.dump { Domains: { Domain: { Domain: location Size: 32768 AoRs: { Info: { AoR: 309(a)nmreis.test HashID: -35263718 Contacts: { Contact: { Address: sip:309@192.168.150.158:55060;transport=udp;registering_acc=nmreis_test Expires: 585 Q: -1.000000 Call-ID: 6e7635ce4604e16400ee763b2ead55b2@0:0:0:0:0:0:0:0 CSeq: 4 User-Agent: Jitsi2.11.5602Linux Received: sip: 2.2.2.2:31171 Path: [not set] State: CS_SYNC Flags: 1 CFlags: 192 Socket: udp: 1.1.1.1:5060 Methods: -1 Ruid: uloc-5de8056d-22ab-f Instance: [not set] Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 1 Last-Keepalive: 1575487540 Last-Modified: 1575487540 } } } } Stats: { Records: 1 Max-Slots: 1 } } } } *anycast node that gets the USRLOC update via KDMQ:* $ kamcmd ul.dump { Domains: { Domain: { Domain: location Size: 32768 AoRs: { Info: { AoR: 309(a)nmreis.test HashID: -35263718 Contacts: { Contact: { Address: sip:309@192.168.150.158:55060;transport=udp;registering_acc=nmreis_test Expires: 574 Q: -1.000000 Call-ID: 6e7635ce4604e16400ee763b2ead55b2@0:0:0:0:0:0:0:0 CSeq: 4 User-Agent: Jitsi2.11.5602Linux Received: sip: 2.2.2.2:31171 Path: [not set] State: CS_SYNC Flags: 1 CFlags: 192 Socket: [not set] Methods: -1 Ruid: uloc-5de7ff7b-4617-3 Instance: [not set] Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 1 Last-Keepalive: 1575487540 Last-Modified: 1575487540 } } } } Stats: { Records: 1 Max-Slots: 1 } } } } *the actual KDMQ message:* *KDMQ sip:usrloc@10.100.100.21:5060 <http://sip:usrloc@10.100.100.21:5060> SIP/2.0 Via: SIP/2.0/UDP 10.100.100.22;branch=z9hG4bK6034.290e4186000000000000000000000000.0 To: <sip:usrloc@10.100.100.21:5060 <http://sip:usrloc@10.100.100.21:5060>> From: <sip:usrloc@10.100.100.22:5060 <http://sip:usrloc@10.100.100.22:5060>>;tag=3393f0703fb0ccaca74109ff37de39f5-1e4c092b CSeq: 10 KDMQ Call-ID: 0cfde4d32a31addc-8875(a)127.0.0.1 <0cfde4d32a31addc-8875(a)127.0.0.1> Content-Length: 433 Max-Forwards: 1 Content-Type: application/json {"action":1,"aor":"309(a)nmreis.test","ruid":"uloc-5de8056d-22ab-f","c":"sip:309@192.168.150.158:55060;transport=udp;registering_acc=nmreis_test","received":"sip:2.2.2.2:31171 <http://2.2.2.2:31171>","path":"","callid":"6e7635ce4604e16400ee763b2ead55b2@0:0:0:0:0:0:0:0","user_agent":"Jitsi2.11.5602Linux","instance":"","expires":1575487600,"cseq":2,"flags":1,"cflags":192,"q":-1,"last_modified":1575487000,"methods":4294967295,"reg_id":0,"server_id":0}* -- Nuno Miguel Reis Departamento de Engenharia Informática Faculdade de Ciências e Tecnologia Universidade de Coimbra mobile: +351 918519145 -- get my PGP key @ http://pgp.dei.uc.pt:11371/pks/lookup?op=get&search=0x750296EECEAD2B6D

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anycast + dmq_usrloc
by Nuno Miguel Reis 04 Dec '19

04 Dec '19

Hi guys. I'm setting up an anycast scenario with several kamailios sharing an active state and I've noticed that the dmq_usrloc module is not replicating the usrloc socket_info attribute when a register comes in through one of the kamailio nodes. Is there a reason behind this? I'm thinking on adding the replication for that attribute in the code. Another way of address this woud be by having a modparam to replicate the socket. What do you think? This is what I'm seeing: anycast node that actually gets the SIP REGISTER: $ kamcmd ul.dump { Domains: { Domain: { Domain: location Size: 32768 AoRs: { Info: { AoR: 309(a)nmreis.test HashID: -35263718 Contacts: { Contact: { Address: sip:309@192.168.150.158:55060;transport=udp;registering_acc=nmreis_test Expires: 585 Q: -1.000000 Call-ID: 6e7635ce4604e16400ee763b2ead55b2@0:0:0:0:0:0:0:0 CSeq: 4 User-Agent: Jitsi2.11.5602Linux Received: sip: 2.2.2.2:31171 [http://2.2.2.2:31171/] Path: [not set] State: CS_SYNC Flags: 1 CFlags: 192 Socket: udp: 1.1.1.1:5060 [http://1.1.1.1:5060/] Methods: -1 Ruid: uloc-5de8056d-22ab-f Instance: [not set] Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 1 Last-Keepalive: 1575487540 Last-Modified: 1575487540 } } } } Stats: { Records: 1 Max-Slots: 1 } } } } anycast node that gets the USRLOC update via KDMQ: $ kamcmd ul.dump { Domains: { Domain: { Domain: location Size: 32768 AoRs: { Info: { AoR: 309(a)nmreis.test HashID: -35263718 Contacts: { Contact: { Address: sip:309@192.168.150.158:55060;transport=udp;registering_acc=nmreis_test Expires: 574 Q: -1.000000 Call-ID: 6e7635ce4604e16400ee763b2ead55b2@0:0:0:0:0:0:0:0 CSeq: 4 User-Agent: Jitsi2.11.5602Linux Received: sip: 2.2.2.2:31171 [http://2.2.2.2:31171/] Path: [not set] State: CS_SYNC Flags: 1 CFlags: 192 Socket: [not set] Methods: -1 Ruid: uloc-5de7ff7b-4617-3 Instance: [not set] Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 1 Last-Keepalive: 1575487540 Last-Modified: 1575487540 } } } } Stats: { Records: 1 Max-Slots: 1 } } } } the actual KDMQ message: KDMQ sip:usrloc@10.100.100.21:5060 [http://sip:usrloc@10.100.100.21:5060/] SIP/2.0 Via: SIP/2.0/UDP 10.100.100.22;branch= z9hG4bK6034.290e4186000000000000000000000000.0 To: < sip:usrloc@10.100.100.21:5060 [http://sip:usrloc@10.100.100.21:5060/] > From: < sip:usrloc@10.100.100.22:5060 [http://sip:usrloc@10.100.100.22:5060/] >;tag= 3393f0703fb0ccaca74109ff37de39f5-1e4c092b CSeq: 10 KDMQ Call-ID: 0cfde4d32a31addc-8875(a)127.0.0.1 [mailto:0cfde4d32a31addc-8875@127.0.0.1] Content-Length: 433 Max-Forwards: 1 Content-Type: application/json {"action":1,"aor":"309(a)nmreis.test","ruid":"uloc-5de8056d-22ab-f","c":"sip:309@192.168.150.158:55060;transport=udp;registering_acc=nmreis_test","received":"sip: 2.2.2.2:31171 [http://2.2.2.2:31171/] ","path":"","callid":"6e7635ce4604e16400ee763b2ead55b2@0:0:0:0:0:0:0:0","user_agent":"Jitsi2.11.5602Linux","instance":"","expires":1575487600,"cseq":2,"flags":1,"cflags":192,"q":-1,"last_modified":1575487000,"methods":4294967295,"reg_id":0,"server_id":0} Looking forward to hear from you. BR, -- Nuno Miguel Reis Departamento de Engenharia Informática Faculdade de Ciências e Tecnologia Universidade de Coimbra

1 0

Vendor Specific RBT Call Flow
by Hamid Hashmi 04 Dec '19

04 Dec '19

Dear Users Can the following call flow be implemented, if yes, can you give some suggestions with modules information? [cid:d15f60e7-73ec-4cd8-8954-92f472a94ef7] Regards Hamid R. Hashmi

3 2

siremis help and usage index?
by PICCORO McKAY Lenz 04 Dec '19

04 Dec '19

it's there some help manual usage for siremis? the only KB source i have are asipto.. and only are install manuals!! Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

3 2

Re: [SR-Users] kamailio IMS supporting VOLTE
by Shubhendu Das 04 Dec '19

04 Dec '19

Hi , Does Kamailio has its own HSS supporting CX Interface ? BR, Shubhendu From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Mojtaba Sent: Saturday, November 23, 2019 4:37 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] kamailio IMS supporting VOLTE Hello, Running the IMS network for supporting VoLTE with Kamailio module and FHOSS is challenging. THe Kamailio works great in IMS network but You have to optimise your own database. Using FHOSS without any support or development is very risky. In general speaking, running VoLTE have three main section and the IMS network is one of them. You might have some challenges in EPC and RAN access. WIth Best Regards. Mojtaba On Thu, Nov 21, 2019 at 3:23 PM Shubhendu Das <shubhendu.das(a)azcom.in> wrote: Hi , I have a query regarding Kamailio open source SIP server. Does it have any release, where its supports the feature of IMS supporting VOLTE with real commercial mobile phones(IMSI based) ? If so, what is the release number ? Please also let me know about the supported mobile phones list. BR, Shubhendu Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- --Mojtaba Esfandiari.S -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

2 1

Fwd: app-ads.txt monetization proposal
by Joel Serrano 04 Dec '19

04 Dec '19

Not sure if this is BS or worth exploring, it's not my expertise area so I'm forwarding to you all just in case ---------- Forwarded message --------- From: Lorenzo from Instal <lorenzo.tacchio(a)instal.com> Date: Tue, Dec 3, 2019 at 8:56 AM Subject: app-ads.txt monetization proposal To: <joel(a)textplus.com> app-ads.txt monetization proposal Hello , I am Lorenzo, Senior Account Manager at Instal; we are currently buying your app's traffic via DSPs. We thought would be beneficial to work directly with you, *monetizing your inventory with no intermediate levels at all*. The key of our success is the *simple technical integration*. No SDK is required, we will add our ads.txt to your page and your app would be included in our media buying plan with budgets of *more than $1M per month*. Would you be interested to discover more in detail this opportunity? Please share your Skype ID or add me, *live:loresseintes*, to connect further! Kind Regards, Lorenzo Instal Srl, Via Ippolito Pindemonte, Florence, Italy Manage preferences <https://hs-3923849.s.hubspotemail.net/hs/manage-preferences/unsubscribe?d=V…>

1 1

secfilter usage
by Aymeric Moizard 03 Dec '19

03 Dec '19

Hi all, I'm testing and moving my kamailio script to use the newer secfilter module. In the past, I was replying "200 Ok" to "friendly scanners"... With the newer secfilter module, it looks like I can't send a reply with "sl_send_reply("200", "OK")" secf_check_ua(); if ($? == -2) { sl_send_reply("200", "OK"); exit; } I have read the code of the secfilter, but I was not able to see any specific code to silently discard the SIP request. I can see in the documentation about "secf_check_sqli_all", that the SIP message is supposed to be "dropped". I can see "w_check_sqli_all" returns 0 on detection and w_check_ua returns -2 upon detection. Are the message discarded because a negative value was returned? Would it be doable, using the secfilter, to still reply 200 Ok? Regards Aymeric -- Antisip - http://www.antisip.com

2 5

kamailio IMS supporting VOLTE
by Shubhendu Das 03 Dec '19

03 Dec '19

Hi , I have a query regarding Kamailio open source SIP server. Does it have any release, where its supports the feature of IMS supporting VOLTE with real commercial mobile phones(IMSI based) ? If so, what is the release number ? Please also let me know about the supported mobile phones list. BR, Shubhendu -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

2 2

Contact header requirement
by David Villasmil 02 Dec '19

02 Dec '19

Hello guys, I'm connecting to a provider who requires the contact contains the proxy's IP... I know this is weird and totally off-rfc... I setup the topoh module and used modparam("topoh", "mask_ip", "[PROXY_IP]:5060") but this makes the contact: Contact: <sip:[PROXY_IP]:5060;line=sr-Z213nshDqt98-2qiIoT8GFf. y8PWype.SRy0nRo3nCT*> I know this is how it works, but they don't like the additional ";line=...." Has anyone faced this kind of thing? is there any other way to set this to something like: Contact: <sip:[PROXY_IP]:5060> that won't mess everything? Thanks all, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 7

$null + 0 is not int
by Juha Heinanen 02 Dec '19

02 Dec '19

Why is it that $null + 0 is 0 but not int? What is it? -- Juha $avp(test) = $null + 0; if (is_int("$avp(test)")) INFO(test $avp(test) IS int); else INFO(test $avp(test) IS NOT int); Dec 1 02:46:01 salmon /usr/bin/sip-proxy[10138]: INFO: test 0 IS NOT int

2 5

Kamailio World 2020 - Call For Presentations
by Daniel-Constantin Mierla 02 Dec '19

02 Dec '19

Hello, I would like to announce that Call for Presentations at Kamailio World 2020 is now open. You can submit your proposal or see more details at: - https://www.kamailioworld.com/k08/call-for-speakers/ The 8th edition of the event takes place again in Berlin, Germany, during April 27-29, 2020. Expect over 150 participants, developers and community members as well as representatives of other popular open source VoIP projects such as Asterisk or FreeSwitch. Looking forward to meeting many of you there! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

1 0

Possible conflict between t_newtran and setflag for ACC
by Patrick Wakano 01 Dec '19

01 Dec '19

Hello list, Hope you all doing well! I am using the ACC module and using the setflag() function as done in several examples. It works fine. However, I've added the t_newtran() function almost in the begging of the INVITE handler to help the retransmission detection and after that I noticed the ACC was not saving anything in DB. So after debugging I discovered that if I call the t_newtran() before setting the ACC flags, the module will not save the calls in DB, but if I call it after setting the ACC flags, it works.... So my question is, is this a bug or it is a expected side effect so when one is using t_newtran you must be careful and set all your transaction flags before? (ACC are the only transaction flags I am using so can't tell if other modules have the same problem) This is happening in Kamailio 5.2.2. Thank you! Kind regards, Patrick Wakano

3 6

tcp limit, too many open files
by David Villasmil 01 Dec '19

01 Dec '19

Hello guys, I'm facing an issue where, if i start kamailio as root, all works properly, but if i start it as kamailio, i get hundreds of: handle_new_connect(): error while accepting connection(24): Too many open files My limits is like: kamailio soft core unlimited kamailio soft data unlimited kamailio soft fsize unlimited kamailio soft memlock unlimited kamailio soft nofile unlimited kamailio soft rss unlimited kamailio hard stack 240 kamailio soft cpu unlimited kamailio soft nproc unlimited kamailio soft as unlimited kamailio soft priority -11 kamailio soft locks unlimited kamailio soft sigpending unlimited kamailio soft msgqueue unlimited kamailio soft nice -11 kamailio soft nofile 65536 kamailio hard nofile 65536 root soft core unlimited root soft data unlimited root soft fsize unlimited root soft memlock unlimited root soft nofile unlimited root soft rss unlimited root hard stack 240 root soft cpu unlimited root soft nproc unlimited root soft as unlimited root soft priority -11 root soft locks unlimited root soft sigpending unlimited root soft msgqueue unlimited root soft nice -11 root soft nofile 65536 root hard nofile 65536 I can't figure out why this is happening... Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

7 17

Kamailio as SBC for Microsoft Teams
by Henning Westerholt 29 Nov '19

29 Nov '19

Hello, I have created a "How-To" blog post on using Kamailio as "session border controller" for Microsoft Teams Direct Routing: https://skalatan.de/de/blog/kamailio-sbc-teams You can this way use all the existing possibilities that Kamailio provide to interact with MS Teams as well. Best regards, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

7 10

Adding "q" value for the main branch
by Maxim M 28 Nov '19

28 Nov '19

Maxim Malygin <maxim.malygin(a)gmail.com> 17:08 (5 минут назад) кому: sr-users Hello, I need to form list of Contact headers with "q" parameters for "302 Moved Temporary". I do something like this: ... # $xavp(contacts) - contains list of contact URIs (uri) and Q values (q) $var(i) = 0; $var(num) = $cnt($xavp(contacts)); while($var(i) < $var(num)) { if($var(i) == 0) { # How to add q value to this branch? $ru = $xavp(contacts[$var(i)]=>uri); } else { append_branch("$xavp(contacts[$var(i)]=>uri)", "$xavp(contacts[$var(i)]=>q)"); } } send_reply("302", "Moved Temporarily"); ... Is it possible to set "q" value for the main branch? For additional branches I do it via append_branch("uri", q_value) but I didn't find any way to do it for the main branch pointed via $ru/$du. The pseudo-variable "$branch" also doesn't allow me to access the main branch. Thanks, Maxim

1 0

Get totally rid of H.264 SDP
by Kjeld Flarup 28 Nov '19

28 Nov '19

Hi I have a door telephone, with GSM fallback. Since telephone operators are going away from SS7, more and more traffic goes SIP. Thus we now have some numbers where we can see that doing an invite with H.264 causes them to fail. I have tried to remove coded 96 with sdp_remove_codecs_by_id, but it still leaves most of the video SDP in the invite. SDP received by kamailio Media Description, name and address (m): video 6100 RTP/AVP 96 Media Attribute (a): rtcp:6101 IN IP4 192.168.237.239 Media Attribute (a): rtpmap:96 H264/90000 Media Attribute (a): fmtp:96 profile-level-id=42000d;packetization-mode=1 Media Attribute (a): sendrecv After sdp_remove_codecs_by_id I send this. But without "96" Media Description, name and address (m): video 6534 RTP/AVP Media Attribute (a): sendrecv Media Attribute (a): rtcp:6535 Media Attribute (a): ice-ufrag:35ZngQdL Media Attribute (a): ice-pwd:DG6kUbcYwBFKNS6YzCG27wRlG4 Media Attribute (a): candidate:r9KO0wNPMTGktklS 1 UDP 2130706431 192.168.237.15 6534 typ host Media Attribute (a): candidate:r9KO0wNPMTGktklS 2 UDP 2130706430 192.168.237.15 6535 typ host My problem is that the door telephone does not do an ACK when it gets the 200 OK My suspicion is that it does not understand the video response Media Description, name and address (m): video 0 RTP/AVP 0 Media Attribute (a): ice-ufrag:35ZngQdL Media Attribute (a): ice-pwd:DG6kUbcYwBFKNS6YzCG27wRlG4 Media Attribute (a): sendrecv Is there a way to remove the whole video part and not just the coded id? It is not an option to disable video in the door telephone, as we need the video. It is just in the fallback is must be removed. -- --------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

1 0

Kamailio does not detect message
by Adesh Pandey 28 Nov '19

28 Nov '19

Hi, I have below scenario: User registers on Kamailio server, calls to a number which is forwarded to asterisk through dispatcher, asterisk server answers and sends a text message to the caller using sendText but first message is delivered to client with message "501 Not Implemented" but in the subsequent calls asterisk shows that the text was delivered but kamailio did not detect the message. Guys, any help would be appreciated, I am using the below mentioned config. https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg Kamailio version: 5.2

1 0

Kamailio developers meeting - follow up remarks
by Daniel-Constantin Mierla 27 Nov '19

27 Nov '19

Hello, last week we had the 2nd annual Kamailio Developers Meeting hosted by Sipgate in Dusseldorf, Germany: * https://www.kamailio.org/w/developers-meeting/ 16 people were at the event in various roles. Giacomo Vacca published on his blog a good summary of what happened there: * https://www.giacomovacca.com/2019/11/my-notes-on-kamailio-developer-meeting… This year we had a lot of discussions, as well as work done on multiple planes, not only Kamailio code. So I am trying to list here some of the conclusions for future development, the technical aspects of the meeting, so everyone is aware and can provide feedback. 1) Effective work was done on: * kamailio code * kamailio rpm packaging * kamailio tools (kamctl) * kamailio release process * kamailio project keys (to be used to sign the packages) 2) Documentation 2.a) Wiki * it was somehow a rough consensus to move the wiki content to github, along with changing the format from dokuwiki markdown to the standard/github markdown. This should enable people to make pull requests so developers or community members can review and aprove new content. It also makes it easier to contribute using existing github account, now the kamailio.org/wiki is requiring to make a dedicated account, which many prefer not to do it. * the presentation can be done either by using mkdocs to generate html files hosted on kamailio.org or using the github provided wiki portal. 2.b) Docs for variables and transformations * there was a proposal to move them in the documentation the modules that export them, there are pros and cons, needs more discussions. Now they are in the wiki, so this probably has to resume after deciding on 2.a). 3) Kamailio Modules 3.a) replication (dmq) - several participants discussed about negotiations between nodes to take active role on some cases (e.g., active dialogs) 3.b) api integration - quite some interest in JSON-based API routing, concluding in extending rtjson to cover more use cases 3.c) security - have options to restrict the use of TLS1.3 or newer 4) Kamailio Releases * v5.3.2 was released during the event, allowing to document the process * work to automatize the process is planned, then eventually assing teams for takeing cares of releases from specific branches 5) Kamailio Testing * existing docker-based testing framework should be extended and integrated in CD/CI pipeline 6) Kamailio packages 6.a) rpms * rpm.kamailio.org has been prepared and is expected to take over the opensuse build service for building rpms and hosting them. Expected to provide support for hosting many kamailio versions in the same release series so one can do downgrade to older releases. Also, there is work in progress to provide nightly builds. 6.b) debs * work is planned to offer many kamailio versions in the same release series 7) Kamailio tools * kamctl/kamdbctl should be obsoleted in favor of kamcli, which offers a better framework for input validation and output formatting, as well as better portability, no longer depending on shell interpreter 8) Various discussions * kemi exports from C point of view and how to combine the documentation for modules and their kemi exports * how to make kamaiio friendlier in virtualized environments (ended up in the need of making the use of advertised address a bit more dynamic) * project organizatoric topics - to be approached separately * next events - Fosdem - someone should submit a proposal to present about Kamailio 9) Long term goals We speak here more or less about Kamailio 6.0 ... * change the behaviour of the native config interpreter to be consistent with the other programming languages in terms of handling the response code (change what is now: the evaluation of negative value to false and positive value to true and the hidden return 0 to exit) * make the pool of processes more generic, so they can handle traffic from more sockets (being sip traffic or something else) -- this should make better use of resources, as some sockets might be less busy that others I hope I covered the important topics, if I remember something else, I will reply on this thread. Or maybe other participants can contribute missing topics. Should anyone have comments or suggestions on the above topics, or new ones, let's discuss on sr-users because it impacts the long term use of Kamailio (sr-dev is cc-ed now for notification purposes). Overall, there were 2 very intensive days, however in a friendly and relaxed environment offered by Sipgate. Extremely useful discussions, not only about Kamailio, but about RTC ecosystem. At the evening social network event event, a couple of external people joined, some of them presenting very interesting use cases of Kamailio. Many thanks to all participants that allocated time and resources to come to Dusseldorf, as well as to the companies that covered expenses for participants. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

2 2

Kamailio xavp_contact - Segmentation Fault
by Ilie Soltanici 27 Nov '19

27 Nov '19

Hello, Kamailio is crashing when i'm trying to set the parameter; modparam("tm|usrloc", "xavp_contact", "ulattrs") That's happening with Kamailio 5.3.1 and 5.2.5 too. Crash is happening when Im register 2 devices with the same extension. This is the core dump: [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib64/libthread_db.so.1". Core was generated by `/usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /usr/local/etc/ka'. Program terminated with signal 11, Segmentation fault. #0 0x000000000057cca2 in xavp_get_internal (name=0x8, list=0x0, idx=0, prv=0x0) at core/xavp.c:288 288 core/xavp.c: No such file or directory. (gdb) bt #0 0x000000000057cca2 in xavp_get_internal (name=0x8, list=0x0, idx=0, prv=0x0) at core/xavp.c:288 #1 0x0000000000581869 in xavp_insert (xavp=0x0, idx=0, list=0x0) at core/xavp.c:752 #2 0x00007f05be0ee3a1 in ki_t_next_contacts (msg=0x7f05cb76c218) at t_serial.c:552 #3 0x00007f05be0f09e8 in t_next_contacts (msg=0x7f05cb76c218, key=0x0, value=0x0) at t_serial.c:756 #4 0x0000000000434d41 in do_action (h=0x7ffe93c9cad0, a=0x7f05cb642250, msg=0x7f05cb76c218) at core/action.c:1067 #5 0x00000000004418e8 in run_actions (h=0x7ffe93c9cad0, a=0x7f05cb642250, msg=0x7f05cb76c218) at core/action.c:1572 #6 0x0000000000441fa9 in run_actions_safe (h=0x7ffe93c9de90, a=0x7f05cb642250, msg=0x7f05cb76c218) at core/action.c:1636 #7 0x000000000065734e in rval_get_int (h=0x7ffe93c9de90, msg=0x7f05cb76c218, i=0x7ffe93c9cf78, rv=0x7f05cb642570, cache=0x0) at core/rvalue.c:912 #8 0x000000000065b8fe in rval_expr_eval_int (h=0x7ffe93c9de90, msg=0x7f05cb76c218, res=0x7ffe93c9cf78, rve=0x7f05cb642568) at core/rvalue.c:1910 #9 0x000000000065bd51 in rval_expr_eval_int (h=0x7ffe93c9de90, msg=0x7f05cb76c218, res=0x7ffe93c9d42c, rve=0x7f05cb642c98) at core/rvalue.c:1918 #10 0x0000000000434807 in do_action (h=0x7ffe93c9de90, a=0x7f05cb645028, msg=0x7f05cb76c218) at core/action.c:1043 #11 0x00000000004418e8 in run_actions (h=0x7ffe93c9de90, a=0x7f05cb637e48, msg=0x7f05cb76c218) at core/action.c:1572 #12 0x0000000000431767 in do_action (h=0x7ffe93c9de90, a=0x7f05cb583858, msg=0x7f05cb76c218) at core/action.c:691 #13 0x00000000004418e8 in run_actions (h=0x7ffe93c9de90, a=0x7f05cb574750, msg=0x7f05cb76c218) at core/action.c:1572 #14 0x0000000000442071 in run_top_route (a=0x7f05cb574750, msg=0x7f05cb76c218, c=0x0) at core/action.c:1657 #15 0x00000000005874a9 in receive_msg (buf=0xa6ec00 <buf.6868> "OPTIONS sip:10@192.168.0.231:5060 SIP/2.0\r\nVia: SIP/2.0/UDP 192.168.0.231;branch=z9hG4bKd8a9.1b3abe656532065414a2f35a8916008d.1\r\nVia: SIP/2.0/UDP 192.168.0.131:5060;received=192.168.0.131;branch=z9hG"..., len=694, rcv_info=0x7ffe93c9e4d0) at core/receive.c:341 #16 0x000000000047bf81 in udp_rcv_loop () at core/udp_server.c:541 #17 0x0000000000424f9d in main_loop () at main.c:1669 #18 0x000000000042c688 in main (argc=13, argv=0x7ffe93c9ea18) at main.c:2710 Thank you.

2 1

Kamailio Parrallel Forking
by Ilie Soltanici 27 Nov '19

27 Nov '19

Hello, I'm trying to use parallel forking if one extension is registered from multiple devices. (using WebSockets) This is the related code from kamailio.cfg: ------------------------------------------------------------------------------------------------------------------------------------ if (!t_load_contacts()) { xlogl("L_WARN", "Error loading contacts for $rU\n"); sl_send_reply("500", "Server Internal Error (Code:$cfg(line))"); exit; } else { xlogl("L_INFO", "Contacts loaded for $rU\n"); } if (!t_next_contacts()) { xlogl("L_INFO", "t_next_contacts - Only one contact found for $rU, calling\n"); } else { xlogl("L_INFO", "t_next_contacts - Multiple contacts found, parallel forking\n"); } route(RELAY); ------------------------------------------------------------------------------------------------------------------------------------ I noticed that on Kamailio 5.2.5 - this code is working fine, and Kamailio is sending an INVITE to both Devices.But, on Kamailio 5.3.1 - that's not working anymore. For some reason, kamailio is sending just one INVITE to one device with wrong URI. The URI is set to the second device instead of the correct one. Was something changed in the t_next_contacts function related to this? Thank you

1 0

Kamailio 5.2.3 segfaulting
by Daniel Greenwald 27 Nov '19

27 Nov '19

I'm seeing this in the log and kamailio is crashing every hour or so: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 33/var/log/kamailio/kamailio.log Any ideas? Should I emergency upgrade to 5.3.1?

2 12

No Audio using RTP Proxy
by Sujit Roy 27 Nov '19

27 Nov '19

Hello I am facing a problem as below. Please suggest for the work around. My call flow is like this. SIP Gateway-1 (IP x.179) -> SIP Gateway-2 ( IP x.177) -> Kamalio+RTPProxy So when the call arrives at Kamalio+RTPProxy, i m getting below in log. Nov 26 23:25:31 rtpproxy[18508]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 1b7c870763616c6c15fff410@ 192.168.100.177, tag 1aa18fc201a68168;1 requested, type strong Nov 26 23:25:31 rtpproxy[18508]: INFO:1b7c870763616c6c15fff410@ 192.168.100.177:rtpp_command_ul_handle: new session on IPv4 port 15920 created, tag 1aa18fc201a68168;1 Nov 26 23:25:31 rtpproxy[18508]: INFO:1b7c870763616c6c15fff410@192.168.100.177:rtpp_stream_prefill_addr: pre-filling caller's RTP address with 192.168.100.177:27360 Nov 26 23:25:31 rtpproxy[18508]: INFO:1b7c870763616c6c15fff410@ 192.168.100.177:rtpp_stream_prefill_addr: pre-filling caller's RTCP address with 192.168.100.177:27361 But x.177 is working on signalling mode only ( Not routing Media ) . As a result, i m not getting any voice from IP x.179 What can be done to change the caller's RTP address to x.179 in RTPProxy ? Thanks -- Regards =================== Sujit Roy

2 1

Encode host ip in Contact header for semi-stateless setup
by Daniel Greenwald 27 Nov '19

27 Nov '19

At kamailio world 2017, Jan Lorenz of Pascom explained how they use kamailio in a semi-stateless manner and encode the asterisk's IP into the contact header on the way out so that it can be decoded and used to route properly on the way back without Record-Routing. I would like to create such a setup. I'm looking for pointers if there is a particular module/method that would be ideal to to use for this. Any other gotcha's to consider in such a setup. My plan is to keep the transaction on the same kamailio by not altering the VIA but allow the subsequent in-dialog transactions to potentially hit another kamailio by changing the contact to a shared SRV record (and encoding the freeswitch box's IP into it for later retrieval.) Thanks in advance. Daniel Greenwald

4 4

dialog module: Does dlg_manage() have to be called for all requests?
by George Diamantopoulos 27 Nov '19

27 Nov '19

Hello all, I've been reading through the dialog module docs again, and there seems to be a discrepancy in what's suggested in the docs. In the intro, it is stated that 'To create the dialog associated with an initial INVITE request, execute the function “dlg_manage()"'. Later on, in section "7.10" the following example code is provided: request_route { ... if(is_method("INVITE") && !has_totag()) { $dlg_ctx(timeout_route) = "DLGTIMEOUT"; $dlg_ctx(timeout_bye) = 1; } dlg_manage(); ... } In this example code, dlg_manage seems to be called for all requests, as it is not dependent on the conditional that only requests with no to-tag are to be handled. Previously, Daniel had suggested to me in person that one also run dlg_manage() for CANCELs, and in-dialog reINVITEs, BYEs and ACKs to mitigate some buggy corner cases. Is this still the case? What do you guys and gals do in production? Have you had to revise how you use dlg_manage in terms of calling it for initial INVITEs only vs all requests? Thanks! BR, George

5 6

Modify SIP Message Header and Message Body parameter
by Sujit Roy 26 Nov '19

26 Nov '19

Hello Please see below SIP Message SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.1681.101:5060;branch=z9hG4bK5a61798d7c850d39;rport=5060 From: <sip:summona@ 192.1681.101>;tag=51a0dc1e620de16d To: <sip:101880171234567@ 192.1681.102> Call-ID: 6fd9b9d563616c6c15f2d7fa@ 192.1681.101 CSeq: 21701 INVITE *Server: kamailio (5.1.9 (x86_64/linux))* Content-Length: 0 *1. I want to Change the Server Parameter . How to do it ? * SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.1681.101:5060;rport=5060;branch=z9hG4bK5a61798d7c850d39 Record-Route: <sip: 192.1681.102;lr> From: <sip:summona@ 192.1681.101>;tag=51a0dc1e620de16d To: <sip:101880171234567 @ 192.1681.102>;tag=as728725b3 Call-ID: 6fd9b9d563616c6c15f2d7fa@ 192.1681.101 CSeq: 21701 INVITE Server: VOS3000 V2.1.4.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 600;refresher=uas *Contact: <sip:1018801751481417@104.216.39.66:5060 <http://sip:1018801751481417@104.216.39.66:5060>>* Content-Type: application/sdp Require: timer Content-Length: 289 v=0 o=root 912643161 912643161 IN IP4 192.1681.104 s=VOS3000 V2.1.4.0 c=IN IP4 192.1681.104 t=0 0 m=audio 10784 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:230 a=sendrecv *a=nortpproxy:yes* 2. I Want to hide the Contact Parameter and a= nortpproxy:yes ==> How to do it as well ? Thanks in advance. -- Regards =================== Sujit Roy

2 1

Beginner Help: Relay vs Forward
by Michael Iedema 26 Nov '19

26 Nov '19

Hi all, I am trying to place Kamailio between several SIP services which are differentiated by port and by the type of message they handle. To be concrete, here are the services: Originator: - reachable on port 5060 - originates all traffic to the services Service A: - reachable on port 5061 - handles REGISTER Service B: - reachable on port 5062 - handles INVITE Service C: - reachable on port 5063 - handles MESSAGE The services, originator and Kamailio instance may all be run on the same IP or may be split onto different addresses. Previously, the originator needed to know the address and port of each service. Now I want to make Kamailio responsible for that task so it can be done more intelligently. To start though I simply want to get traffic flowing between the services and this is where I’m struggling to climb the learning curve. In my test setup the originator and all services are on the same IP with only Kamailio running on a different IP. Address 1 Address 2 ======================== Originator <--> Kamailio:6060 | | Service A <-------> | Service B <-------> | Service C <-------> | My first attempt at the logic looks like this: # HOST DEFINITIONS #!substdef "/HOST_ORIGINATOR/192.168.86.110/" #!substdef "/HOST_REGISTER/192.168.86.110/“ #!substdef "/HOST_INVITE/192.168.86.110/" #!substdef "/HOST_MESSAGE/192.168.86.110/" # PORT DEFINITIONS #!substdef “/PORT_ORIGINATOR/5060/" #!substdef "/PORT_REGISTER/5061/“ #!substdef "/PORT_INVITE/5062/" #!substdef "/PORT_MESSAGE/5063/" # ROUTE LOGIC route { # ORIGINATOR sent us something if ( $(ct{nameaddr.uri}{uri.port}) == HOST_ORIGINATOR ) { xlog("L_INFO", "ORIGINATOR: sent $rm\n”); if (is_method("REGISTER")) { $du = "sip: HOST_REGISTER:PORT_REGISTER"; xlog("L_INFO", "ORIGINATOR -> REGISTER: Destination URI is now $du\n"); t_relay(); exit; } } … #### END SNIPPET #### My very newbie questions are: - Am I even on the right track here? I understand that t_relay() establishes a stateful dialog handler between the originator and register services. - Can I accomplish this bi-directional path statelessly with forward()? - Do I need to add logic to handle the messages from the REGISTER service back to ORIGINATOR? This config currently fails because of "no corresponding socket found” when the relay triggers. I believe this is because I’m running the Kamailio instance in a Docker container so my first step today is to set up a native install. I also looked at the dispatcher module which seems very close to what I want to do. Maybe that’s a more natural way to go? Anyway, apologies for the lengthy explanation. I hope it is at least clear what I intend to do. I will gladly research and do trial and error to solve this but need to know I’m going down roughly the right path. I’m not sure what I should even be googling even of the time :-) Thanks as always for your time and the great software! Regards, -Michael

4 5

Re-use TCP connections on different transactions
by Joel Serrano 26 Nov '19

26 Nov '19

Hello, I'm setting up a Kamailio instance behind a TCP load balancer (with proxy protocol and NAT routing: meaning Kam stays in the flow all the time). I've managed to get working almost everything we need for our service, except for one thing, and that is for Kam to use existing connections for subsequent transactions: Following this example: [image: image.png] EXT & INT represent the external and internal interface of a LB between the UAC and Kamailio, using TLS on both legs and proxy protocol. Transaction 1: INVITE, 100, 180, 183, 200 OK UAC 1.1.1.1:1111 -> 2.2.2.2:443 (EXT) 3.3.3.3:3333 (INT) -> 7.7.7.7:5060 (Kamailio) Transaction 2: ACK UAC 1.1.1.1:1112 -> 2.2.2.2:443 (EXT) 4.4.4.4:4444 (INT) -> 7.7.7.7:5060 (Kamailio) Transaction 3: BYE Kam 7.7.7.7:5060 -> 3.3.3.3:3333 (INT) 2.2.2.2:443 (EXT) -> 1.1.1.1:1111 UAC My problem is with Transaction 3. In this case the BYE is originated by the callee, and Kam has to send it to the caller. As the TCP load balancer is between Kam and the UAC, Kam has to send it to the LB so then the LB can forward it back to the UAC. This works well for msgs that belong to the same transaction (INVITE, 100, 180, 183, 200 OK) but it fails when they don't belong to the same transaction. Thanks to the newly added $tcp(c_si) and $tcp(c_sp) pseudovars, I can save the internal IP:Port of the LB, so I can send stuff later to it, my problem is that Kam doesn't seem to allow this? On the original INVITE, I use the following to save where I have to reach the UAC: add_contact_alias("$tcp(c_si)", "$tcp(c_sp)", "tls"); Then, handle_ruri_alias() will take care of setting $du to the correct (internal LB) IP:Port so I can reach the UAC, this works. My problem is that Kamailio doesn't identify that there is a valid existing TLS connection still up (from the INVITE), and tries to create a new one (and this obviously doesn't gives all sorts of problems). So when I run handle_ruri_alias(), and $du is set to 3.3.3.3:3333 (from the example above), instead of using the existing connection, Kamailio tries to create a new one. I have a log statement right before with the result of tcp_conid_state(1) (the connid is 1 for this connection) and the $rc is 1 (Connection is OK), but when I tell Kamailio it has to use it I get this in the logs: DEBUG: {1 11726467 BYE gqR1qqNK8B} <core> [core\/tcp_main.c:2060]: tcp_send(): no open tcp connection found, opening new one"} And then the problems begin... I have tried playing around with: tcp_reuse_port tcp_connection_match But no luck..! I also thought it could be a problem of the connection being created on one worker, and a different worker handling BYE transaction, so tested with children=1 and tcp_children=1, but still same problem. A more detailed log: In blue my log statement checking for the status of conid "1", in red Kam not being able to find it, although it exists (as validated by tcp_conid_state(), and even in netstat I can see the connection established). In this log, 35.191.0.66:60271 would be the equivalent of 3.3.3.3:3333 and 104.175.176.242:28157 would be 1.1.1.1:1111 from the example above. ... "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/tcp_main.c:1657]: _tcpconn_find(): found connection by id: 1"} "message":" NOTICE: {1 11727734 BYE 5-LX4GdI9X} <script>: JOEL TEST New request - M=BYE TCP STATUS:1 ID=5-LX4GdI9X"} ... "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} nathelper [nathelper.c:1144]: handle_ruri_alias(): setting dst_uri to <sip:35.191.0.66:60271;transport=tls>"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} nathelper [nathelper.c:1166]: handle_ruri_alias(): rewriting r-uri to <sip:linphone@104.175.176.242:28157;transport=tls>"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_lookup.c:1328]: t_newtran(): msg (0x7f3c884259d0) id=534\/18664 global id=532\/18664 T start=0xffffffffffffffff"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_lookup.c:497]: t_lookup_request(): start searching: hash=63496, isACK=0"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_lookup.c:455]: matching_3261(): RFC3261 transaction matching failed - via branch [z9hG4bK808f.eee2444f92a02cb33e1b7a21f20bc6bb.0]"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_lookup.c:675]: t_lookup_request(): no transaction found"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_hooks.c:336]: run_reqin_callbacks_internal(): trans=0x7f3c83b8c598, callback type 1, id 0 entered"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/crypto\/md5utils.c:67]: MD5StringArray(): MD5 calculated: 3071029feb05962b26b53a9664a14210"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} siputils [checks.c:123]: has_totag(): totag found"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} rr [loose.c:1095]: check_route_param(): route params checking against [;r2=on;lr=on;ftag=Eb~TbdfTA;did=cab.01e2;nat=yes] (orig: [r2=on;lr=on;ftag=Eb~TbdfTA;did=cab.01e2;nat=yes])"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} rr [loose.c:1101]: check_route_param(): params are <;r2=on;lr=on;ftag=Eb~TbdfTA;did=cab.01e2;nat=yes>"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} siputils [checks.c:123]: has_totag(): totag found"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/msg_translator.c:161]: check_via_address(): (198.1.54.228, 198.1.54.228, 0)"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/tcp_main.c:2060]: tcp_send(): no open tcp connection found, opening new one"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/ip_addr.c:229]: print_ip(): tcpconn_new: new tcp connection: 35.191.0.66"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/tcp_main.c:1242]: tcpconn_new(): on port 60271, type 3"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/tcp_main.c:1561]: tcpconn_add(): hashes: 1446:2350:0, 5"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tls [tls_server.c:199]: tls_complete_init(): completing tls connection initialization"} ... So time to seek help from the community, any suggestions/ideas/comments? Sorry if all this sounds confusing, I've tried my best to put in text the whole scenario in and "understandable" way... Is this even doable? Thanks, Joel.

3 10

How to enforce REGISTER with AKAv1 (IMS)
by Moshe A 26 Nov '19

26 Nov '19

Hello, How can we enforce REGISTER with AKAv1? Here our setup 1. kamailio version 4.3.4 for IMS 2. Boghe IMS client v1.0.58.550 3. scscf.cfg is configure with: #!define REG_AUTH_DEFAULT_ALG "AKAv1-MD5" 4. Set Authentication Schemes in Private User Identity - IMPI for user bob to: Digest-AKAv1 (3GPP) results: First REGISTER fails with 401 Unauthorized Second REGISTER 200 OK, but the REGISTER is MD5 and not AKAv1 Attached is tcpdump. By the way: in which DB should we find Authentication Schemes? can't find it in hss_db, kamailio and not in scscf. Thank you, Moshe.

1 0

call not going to end point
by Gaurav Bmotra 26 Nov '19

26 Nov '19

hi i m using kamailio 5.4 dev Docker image when i try to call form calientA to clientB i got (Temporarily Unavailable (code: 480)) i m using freeswitch as media server so the flow and structure should be like this ClientA---->Kamailio------>FS---->Kamailio>---->ClientB but issue is (what i m getting ) ClientA---->Kamailio------>FS---->Kamailio>-------->FS-------->kamailio--------->FS IT CREATE THE CALL LOOP b/w KAMAILIO AND FREESWITCH *MY CONFIG FILE IS ATACHED * in logs i m getting -------------------------------------------------------- Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 38902: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:get_command: received command "18_4 Uc0,8,101 9857c00a-8947-1238-c6a7-12590b724429 3.82.96.70 22082 jKrKymKNg480p;1" Nov 24 10:25:36 ip-172-20-61-37 local0.info rtpproxy[6]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 9857c00a-8947-1238-c6a7-12590b724429, tag jKrKymKNg480p;1 requested, type strong Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 35888: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:get_command: received command "20_4 Uc0,8,101 98589959-8947-1238-c6a7-12590b724429 3.82.96.70 29846 m5a51aNvapN6D;1" Nov 24 10:25:36 ip-172-20-61-37 local0.info rtpproxy[6]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 98589959-8947-1238-c6a7-12590b724429, tag m5a51aNvapN6D;1 requested, type strong Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 37706: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:get_command: received command "13_4 Uc0,8,101 985962c1-8947-1238-c6a7-12590b724429 3.82.96.70 29432 pQXp50p3471BN;1" Nov 24 10:25:36 ip-172-20-61-37 local0.info rtpproxy[6]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 985962c1-8947-1238-c6a7-12590b724429, tag pQXp50p3471BN;1 requested, type strong Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 45078: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:get_command: received command "16_5 Uc0,8,101 985a3ce3-8947-1238-c6a7-12590b724429 3.82.96.70 25096 r9F88praZSeHc;1" Nov 24 10:25:36 ip-172-20-61-37 local0.info rtpproxy[6]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 985a3ce3-8947-1238-c6a7-12590b724429, tag r9F88praZSeHc;1 requested, type strong Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 58376: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" -------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- *here is my config file * #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ACCDB #!define WITH_NAT # # Kamailio (OpenSER) SIP Server v5.4 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.org> # # Refer to the Core CookBook at https://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To use RTPEngine (instead of RTPProxy) for nat traversal execute: # - define WITH_RTPENGINE # - install RTPEngine: https://github.com/sipwise/rtpengine # - start RTPEngine: # rtpengine --listen-ng=127.0.0.1:2223 ... # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### 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. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes for each UDP socket * - value inherited by tcp_children and sctp_children when not set explicitely */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* number of SIP routing processes for all TCP/TLS sockets */ # tcp_children=8 /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ # auto_aliases=no /* add local domain aliases */ # alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ # listen=udp:10.0.0.10:5060 listen=udp:LOCAL_IP:5091 advertise PUBLIC_IP:5091 /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 /* upper limit for TCP connections (it includes the TLS connections) */ tcp_max_connections=2048 #!ifdef WITH_TLS enable_tls=yes /* upper limit for TLS connections */ tls_max_connections=2048 #!endif ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" loadmodule "dispatcher.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" #!ifdef WITH_RTPENGINE loadmodule "rtpengine.so" #!else loadmodule "rtpproxy.so" #!endif #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- #-------------------dispatcher params------------- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/run/kamailio/kamailio_ctl") # ----- sanity params ----- modparam("sanity", "autodrop", 0) # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT #!ifdef WITH_RTPENGINE # ----- rtpengine params ----- modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") #!else # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") #!endif # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains #route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN #route(PSTN); route(DISPATCH); # user location service } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); $rcv_addr="sip:"+$si+":"+$sp; if(starts_with("$rcv_addr","$du")) { route(LOCATION); } else { route(RELAY); } exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { xnotice("RTF_DISPATCH: $rm $rU [$ci]"); if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("(^5)") or (t_branch_timeout() and !t_branch_replied())) { xnotice("RTF_DISPATCH: WARNING $du is broken and marked as inactive"); # ds_mark_dst("ip"); if(ds_next_dst()) { #xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # Per SIP request initial checks route[REQINIT] { # no connect for sending replies set_reply_no_connect(); #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if($ua =~ "friendly-scanner|sipcli|sipvicious|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP request from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) { sl_reply_error(); } exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; #!ifdef WITH_RTPENGINE if(nat_uac_test("8")) { rtpengine_manage("replace-origin replace-session-connection"); } else { rtpengine_manage("trust-address replace-origin replace-session-connection"); } #!else if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } #!endif if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } if(isbflagset(FLB_NATB)) { # no connect message in a dialog involving NAT traversal if (is_request()) { if(has_totag()) { set_forward_no_connect(); } } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies reply_route { if(!sanity_check("17604", "6")) { xlog("Malformed SIP response from $si:$sp\n"); drop; } } # Manage incoming replies in transaction context onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } -- *Regards:* Gaurav Kumar

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How to get the audio codec used for each established call.
by Abdoul Osséni 25 Nov '19

25 Nov '19

Hello all, I try to save in the CDR the audio codec used for each established call. Can you help me? Regards Abdoul.I

3 4

SDP Modification
by Sujit Roy 25 Nov '19

25 Nov '19

Hello How can i change c= and o= in SDP using RTPProxy ? Please advise Thanks -- Regards =================== Sujit Roy

2 1

Passing media thru RTP Proxy
by Sujit Roy 25 Nov '19

25 Nov '19

Hello I have installed and configured both Kamailio and RTP Proxy and calls are going. But i m unable to force media to go thru RTP Proxy. How can i enforce media to go thru RTP Proxy ? Thanks -- Regards =================== Sujit Roy

3 2

Kamailio stopped relaying SIP text message from asterisk
by Adesh Pandey 25 Nov '19

25 Nov '19

Hi, I have been working on kamailio with following scenario: 1. Webrtc SIP Users register to Kamailio 2. Users initiate outgoing calls to some other numbers 3. Kamailio dispatches outgoing calls to Asterisk 4. Asterisk answers the call and then send a text message to the caller using SendText, which was working fine a few days back but all of sudden it has stopped working 5. At Asterisk end it is showing that message was delivered successfully but when I check SIP user logs, there is no message event. Please review my config file at following URL: https://gist.github.com/adeshpandey/2c338db7f5d992267f7415de0325e0b4 Developers, I also want to know if there is any way to log *text message* in kamailio server.

3 3

Unable to connect to sip server via proxy
by Ashwini Kumar 24 Nov '19

24 Nov '19

Hi I have two network interfaces on my system in which one interface is connected to my office WAN/LAN and other is connected to SIP private network connected to PSTN. Now, I want to access sip server using the internal IP of our office LAN as proxy. *192.168.6.2 (Office LAN IP as proxy) -> 10.50.185.2 (SIP)* So on machine 192.168.6.2, I setup kamailio as SIP proxy server so that our office LAN machine can connect to SIP server directly via SIP proxy. I used config defined in - *https://github.com/kamailio/kamailio/blob/master/misc/examples/mixed/kamailio-minimal-proxy.cfg* <https://github.com/kamailio/kamailio/blob/master/misc/examples/mixed/kamail…> and I changed the local domain alias with SIP IP -* 10.50.185.2*. When I try to connect to SIP via proxy using *linphone* (UDP) I got below errors in* /var/log/syslog* and shows *connection in progress* every time. Nov 23 15:15:49 ubantudesk /usr/sbin/kamailio[25784]: WARNING: {1 20 REGISTER ek82LguQ6z} sanity [sanity.c:585]: check_cl(): content length header missing in request Nov 23 15:15:53 ubantudesk /usr/sbin/kamailio[25785]: WARNING: {1 20 REGISTER ek82LguQ6z} sanity [sanity.c:585]: check_cl(): content length header missing in request Nov 23 15:15:57 ubantudesk /usr/sbin/kamailio[25784]: WARNING: {1 20 REGISTER ek82LguQ6z} sanity [sanity.c:585]: check_cl(): content length header missing in request Nov 23 15:16:01 ubantudesk /usr/sbin/kamailio[25785]: WARNING: {1 20 REGISTER ek82LguQ6z} sanity [sanity.c:585]: check_cl(): content length header missing in request Nov 23 15:16:05 ubantudesk /usr/sbin/kamailio[25784]: WARNING: {1 20 REGISTER ek82LguQ6z} sanity [sanity.c:585]: check_cl(): content length header missing in request Can you help me what I should do? -- *Thanks & Regards,* *Ashwini Kumar | Technology |* www.industrybuying.com | Ghitorni, Delhi - 110030 | India | È*+91-8802137782* | * ashwinikumar(a)industrybuying.com |

2 1

Avoid full table scan for dispatcher table on db_redis using ds_reload() from config script
by Joel Serrano 23 Nov '19

23 Nov '19

Hello, I'm trying out redis as db backend, and right now I only have dispatcher records there... Every so often, I do a ds_reload() from within Kam config script, and I see in logs: Nov 22 20:36:35 test COPS[25531]: WARNING: db_redis [redis_dbase.c:1098]: db_redis_perform_query(): performing full table scan on table 'dispatcher' while performing query After enabling debug logs: Nov 22 21:15:35 test COPS[26661]: DEBUG: db_redis [redis_dbase.c:1761]: db_redis_query(): querying prefix (table) 'dispatcher' Nov 22 21:15:35 test COPS[26661]: DEBUG: db_redis [redis_dbase.c:1811]: db_redis_query(): no columns given to build query keys, falling back to full table scan Nov 22 21:15:35 test COPS[26661]: DEBUG: <core> [db_res.c:119]: db_new_result(): allocate 56 bytes for result set at 0x7f0be4273898 Nov 22 21:15:35 test COPS[26661]: DEBUG: <core> [db_res.c:156]: db_allocate_columns(): allocate 40 bytes for result names at 0x7f0be4273938 Nov 22 21:15:35 test COPS[26661]: DEBUG: <core> [db_res.c:167]: db_allocate_columns(): allocate 20 bytes for result types at 0x7f0be42739c8 Nov 22 21:15:35 test COPS[26661]: WARNING: db_redis [redis_dbase.c:1098]: db_redis_perform_query(): performing full table scan on table 'dispatcher' while performing query Given the msg... "no columns given to build query keys, falling back to full table scan", I assume I'm missing a keys modparam.. but I don't know how to build it, and the docs (to me) seem confusing? https://kamailio.org/docs/modules/devel/modules/db_redis.html#db_redis.p.ke… Can anyone help me understand how to build the keys modparam for dispatcher table? Or if the reason for this is different, what do I need to do to avoid the full table scan and the warning in the log? Thanks, Joel.

1 1

Question about ims_ipsec_pcscf
by Woscek, Martin W. 22 Nov '19

22 Nov '19

Hi all, We are looking for information as to what the ims_ipsec_pcscf module was written for and tested against. What was the target UE/EPC/PCRF that Tsvetomir Dimitrov wrote the module towards? Has an emulated EPC been tested with this module? If so which one(s)? Thanks in advance, Martin

1 0

Cannot read an indexed xavp from Kemi-Lua
by Michel Pelletier 22 Nov '19

22 Nov '19

Hi, I am having trouble reading an indexed xavp from Kemi-Lua. The xavp contains 3 entries but I can only access the first one. The xavp is the one filled by the dispatcher module after a call to KSR.dispatcher.ds_select(group, 4) with the flags parameter set to 2. I can get the number of entries using KSR.pvx.xavp_child_get("ds_ctx", "cnt") which returns 3. I can also get the contents of the first element using, for example, KSR.pvx.xavp_child_get("ds_dst", "attrs"). But all my attempts at adding an index value using KSR.pvx.xavp_get() or KSR.pvx.xavp_child_get() return the value 'nil'; Below is the output of KSR.pvx.pv_xavp_print(): +++++ start XAVP list: (nil) ((nil)) (level=0) ----- end XAVP list: (nil) (level=0) +++++ start XAVP list: 0x7f94ce01f1f8 ((nil)) (level=0) *** (l:0 - 0x7f94ce01f1f8) XAVP name: ds_ctx XAVP id: 1762531321 XAVP value type: 6 XAVP value: <xavp:0x7f94ce01f150> +++++ start XAVP list: 0x7f94ce01f150 (0x7f94ce01f218) (level=1) *** (l:1 - 0x7f94ce01f150) XAVP name: cnt XAVP id: 7279890 XAVP value type: 1 XAVP value (int): 3 *** (l:1 - 0x7f94ce01f150) end ----- end XAVP list: 0x7f94ce01f150 (level=1) *** (l:0 - 0x7f94ce01f1f8) end *** (l:0 - 0x7f94ce01eaf8) XAVP name: ds_dst XAVP id: 1762531862 XAVP value type: 6 XAVP value: <xavp:0x7f94ce01ea30> +++++ start XAVP list: 0x7f94ce01ea30 (0x7f94ce01eb18) (level=1) *** (l:1 - 0x7f94ce01ea30) XAVP name: attrs XAVP id: 1835799177 XAVP value type: 2 XAVP value (str): duid=newjersey3;maxload=200 *** (l:1 - 0x7f94ce01ea30) end *** (l:1 - 0x7f94ce01e988) XAVP name: grp XAVP id: 7056653 XAVP value type: 1 XAVP value (int): 1 *** (l:1 - 0x7f94ce01e988) end *** (l:1 - 0x7f94ce01e8c8) XAVP name: uri XAVP id: 8122237 XAVP value type: 2 XAVP value (str): sip:123.123.123.123:5060 *** (l:1 - 0x7f94ce01e8c8) end ----- end XAVP list: 0x7f94ce01ea30 (level=1) *** (l:0 - 0x7f94ce01eaf8) end *** (l:0 - 0x7f94ce01edd0) XAVP name: ds_dst XAVP id: 1762531862 XAVP value type: 6 XAVP value: <xavp:0x7f94ce01ed08> +++++ start XAVP list: 0x7f94ce01ed08 (0x7f94ce01edf0) (level=1) *** (l:1 - 0x7f94ce01ed08) XAVP name: attrs XAVP id: 1835799177 XAVP value type: 2 XAVP value (str): duid=newjersey2;maxload=200 *** (l:1 - 0x7f94ce01ed08) end *** (l:1 - 0x7f94ce01ec60) XAVP name: grp XAVP id: 7056653 XAVP value type: 1 XAVP value (int): 1 *** (l:1 - 0x7f94ce01ec60) end *** (l:1 - 0x7f94ce01eba0) XAVP name: uri XAVP id: 8122237 XAVP value type: 2 XAVP value (str): sip:123.123.123.123:5060 *** (l:1 - 0x7f94ce01eba0) end ----- end XAVP list: 0x7f94ce01ed08 (level=1) *** (l:0 - 0x7f94ce01edd0) end *** (l:0 - 0x7f94ce01f0a8) XAVP name: ds_dst XAVP id: 1762531862 XAVP value type: 6 XAVP value: <xavp:0x7f94ce01efe0> +++++ start XAVP list: 0x7f94ce01efe0 (0x7f94ce01f0c8) (level=1) *** (l:1 - 0x7f94ce01efe0) XAVP name: attrs XAVP id: 1835799177 XAVP value type: 2 XAVP value (str): duid=newjersey1;maxload=200 *** (l:1 - 0x7f94ce01efe0) end *** (l:1 - 0x7f94ce01ef38) XAVP name: grp XAVP id: 7056653 XAVP value type: 1 XAVP value (int): 1 *** (l:1 - 0x7f94ce01ef38) end *** (l:1 - 0x7f94ce01ee78) XAVP name: uri XAVP id: 8122237 XAVP value type: 2 XAVP value (str): sip:123.123.123.123:5060 *** (l:1 - 0x7f94ce01ee78) end ----- end XAVP list: 0x7f94ce01efe0 (level=1) *** (l:0 - 0x7f94ce01f0a8) end ----- end XAVP list: 0x7f94ce01f1f8 (level=0) Regards, Michel Pelletier

1 0

kamailio call loop issue B/W kamailio and freeswitch
by Gaurav Bmotra 22 Nov '19

22 Nov '19

hi im using kamailio 5.3 as register server and using dispatcher i m dendign the calls to freeswitch .. ibut the issue is then freeswitch send back call to kamailio after processing media ... it should go back to client but it again send call to freeswitch ,, which create the loop and hangup the call with (Temporarily Unavailable (code: 480)) *what i want *client1------>Kamailio------>FS-----Kamailio----->Client2 *but I am getting this * client1------>Kamailio------>FS-----Kamailio----->FS----->kamailio----->FS -- here is my config file data --- #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ACCDB #!define WITH_NAT # # Kamailio (OpenSER) SIP Server v5.2 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.org> # # Refer to the Core CookBook at https://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### 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. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes for each UDP socket * - value inherited by tcp_children and sctp_children when not set explicitely */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* number of SIP routing processes for all TCP/TLS sockets */ # tcp_children=8 /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ # auto_aliases=no /* add local domain aliases */ # alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ # listen=udp:10.0.0.10:5060 listen=udp:172.20.LOCAL_IP:5091 advertise 3.83.PUBLIC_IP:5091 /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 /* upper limit for TCP connections (it includes the TLS connections) */ tcp_max_connections=2048 #!ifdef WITH_TLS enable_tls=yes /* upper limit for TLS connections */ tls_max_connections=2048 #!endif ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" loadmodule "dispatcher.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- dispatcher params ----- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains #route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN #route(PSTN); route(DISPATCH); # user location service route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) { sl_reply_error(); } exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } *Regards:* Gaurav Kumar

1 0

Re: [SR-Users] disparcher making loop with call
by Gaurav Bmotra 22 Nov '19

22 Nov '19

hi thank you for reply i have check but same issue ,,, is the config file data plz check if you can help thank you ------------------------------------ #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ACCDB #!define WITH_NAT # # Kamailio (OpenSER) SIP Server v5.2 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.org> # # Refer to the Core CookBook at https://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### 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. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes for each UDP socket * - value inherited by tcp_children and sctp_children when not set explicitely */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* number of SIP routing processes for all TCP/TLS sockets */ # tcp_children=8 /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ # auto_aliases=no /* add local domain aliases */ # alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ # listen=udp:10.0.0.10:5060 listen=udp:172.20.LOCAL_IP:5091 advertise 3.83.PUBLIC_IP:5091 /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 /* upper limit for TCP connections (it includes the TLS connections) */ tcp_max_connections=2048 #!ifdef WITH_TLS enable_tls=yes /* upper limit for TLS connections */ tls_max_connections=2048 #!endif ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" loadmodule "dispatcher.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- dispatcher params ----- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains #route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN #route(PSTN); route(DISPATCH); # user location service route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) { sl_reply_error(); } exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } ------------------------------------------------------------------------- On Fri, Nov 22, 2019 at 4:57 PM Gaurav Bmotra <saigauravmehra91(a)gmail.com> wrote: > hi thank you for reply > i have check but same issue > i have attached my config file ... can you check if there is any issue > > thank you > > On Fri, Nov 22, 2019 at 4:41 PM David Villasmil < > david.villasmil.work(a)gmail.com> wrote: > >> Hello, >> >> You’re only pasting the dispatcher route. >> >> You need to follow the script logic. An invite comes in, how does it know >> the call is coming from the client and has to be sent to freeSWITCH? >> Conversely, how does it know it’s coming from freeswitch? >> >> Probably using the permissions module with >> >> https://kamailio.org/docs/modules/3.1.x/modules_k/permissions.html#id2528655 >> >> You can check if the $si is the freeswitch, then send to clientB >> >> On Fri, 22 Nov 2019 at 11:40, Daniel-Constantin Mierla <miconda(a)gmail.com> >> wrote: >> >>> Hello, >>> >>> there is something wrong in your config file, failing to detect the >>> traffic from freeswitch. Load debugger module and enable cfgtrace logging, >>> then see what actions are executed in the configuration file, maybe you can >>> spot what goes wrong there. >>> >>> Otherwise, to get more help from the mailing lists, the full config is >>> needed. >>> >>> Cheers, >>> Daniel >>> On 22.11.19 09:51, Gaurav Bmotra wrote: >>> >>> hi >>> i m using kamailio 5.3 as register server and using dispatcher i m >>> dendign the calls to freeswitch .. >>> >>> but the issue is then freeswitch send back call to kamailio after >>> processing media ... it should go back to client but it again send call to >>> freeswitch ,, which create the loop and hangup the call with (Temporarily >>> Unavailable (code: 480)) >>> >>> >>> *what i want *client1------>Kamailio------>FS-----Kamailio----->Client2 >>> >>> *but I am getting this * >>> >>> client1------>Kamailio------>FS-----Kamailio----->FS----->kamailio----->FS >>> >>> >>> *Routing Logic *------------------------------- >>> if(!ds_is_from_list()) { >>> >>> route(DISPATCH); >>> } >>> >>> -------- >>> # Dispatch requests >>> >>> >>> >>> route[DISPATCH] { >>> >>> >>> >>> # round robin dispatching on gateways group '1' >>> >>> >>> >>> if(!ds_select_dst("1", "4")) { >>> >>> >>> send_reply("404", "No destination"); >>> >>> >>> exit; >>> >>> >>> >>> } >>> >>> >>> >>> xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: >>> $xavp(_dsdst_=>attrs))\n"); >>> >>> >>> t_on_failure("RTF_DISPATCH"); >>> >>> >>> >>> route(RELAY); >>> >>> >>> >>> exit; >>> >>> >>> >>> } >>> >>> >>> >>> >>> >>> >>> >>> # Try next destionations in failure route >>> >>> >>> >>> failure_route[RTF_DISPATCH] { >>> >>> >>> >>> xnotice("RTF_DISPATCH: $rm $rU [$ci]"); >>> >>> >>> >>> if (t_is_canceled()) { >>> >>> >>> exit; >>> >>> >>> >>> } >>> >>> >>> >>> # next DST - only for 500 or local timeout >>> >>> >>> if (t_check_status("(^5)") >>> >>> >>> or (t_branch_timeout() and !t_branch_replied())) >>> { >>> >>> xnotice("RTF_DISPATCH: WARNING $du is broken and marked >>> as inactive"); >>> >>> >>> # ds_mark_dst("ip"); >>> >>> >>> >>> if(ds_next_dst()) { >>> >>> >>> >>> #xdbg("--- SCRIPT: retrying to <$ru> via <$du> >>> (attrs: $xavp(_dsdst_=>attrs))\n"); >>> >>> >>> t_on_failure("RTF_DISPATCH"); >>> >>> >>> >>> route(RELAY); >>> >>> >>> >>> exit; >>> >>> >>> >>> } >>> >>> >>> >>> } >>> >>> >>> >>> } >>> >>> >>> -- >>> thank you >>> >>> >>> >>> >>> >>> >>> >>> *Regards:* >>> Gaurav Kumar >>> >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >>> -- >>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>> Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> sr-users(a)lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >> -- >> Regards, >> >> David Villasmil >> email: david.villasmil.work(a)gmail.com >> phone: +34669448337 >> > > > -- > > > > > > > > > *Regards:* > Gaurav Kumar > > -- *Regards:* Gaurav Kumar

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