sr-users June 2020

sr-users@lists.kamailio.org

71 participants
105 discussions

by Sam Ware

I'm using Sanity Checks. Is an underscore (_) not allowed in the host portion of the RURI? I'm on version 5.3. I reviewed the pase_uri function in the core parse_uri.c but I am not super proficient in C. case URI_HOST_P: switch(*p) { check_host_end; default: if(!isalnum(*p) && (*p != '.') && (*p != '-') && !uri_host_char_allowed(*p)) { goto error_bad_host; } } break; I see it calls out periods (.) and dashes (-) but not underscores. I'm wondering why, being underscores are allowed in URLs. -- Sam D Ware

4 years, 4 months

Changing Debug Level on the fly

by Sam Ware

I am trying to change the debug level without restarting the service. My research says this is right but I get an error message. $ sudo -u kamailio kamcmd dbg.set_mod_level core 2 error: 500 - cannot store parameter -- Sam D Ware

4 years, 4 months

SIP OPTIONS between UE and P-CSCF

by BALL SUN

Hi We have setup UE with Kamailio IMS, and after the registration is successful. The 2nd SIP REGISTER is sent from UE to P-CSCF through IPSEC tunnel. But we found that the SIP OPTIONS (I believed it is a health check) from P-CSCF to UE is not through the IPSEC tunnel. our question is 1. Should SIP OPTIONS run inside the IPSEC tunnel or it should be outside? 2. if it can be configured to run inside the IPSEC tunnel, what configuration can we change to achieve it? - RBK

4 years, 4 months

How to add parameter ;user=phone in RURI

by Minh Phan

Hi List, I'm having trouble adding the parameter user=phone in the RURI. I have tried to use the function add_uri_param from siputils module but it doesn't work (see below). Here is the code I use to add user=phone parameter to FROM, TO and RURI uac_replace_from("", "sip:$var(normalized_number)@sip.domain.com ;user=phone"); uac_replace_to("", "sip:$var(normalized_number)@sip.domain.com;user=phone"); $rU = $var(normalized_number); *add_uri_param*("user=phone"); Thanks in advance for any hints, Best regards, Minh Phan

4 years, 4 months

RTP voice failover

by Evgeniy

Is that possible to implement RTP voice fail-over? In case RTP node goes down - the call should not be interrupted. This is kinda strange - but client want to have super redundant system. For example use 2 RTP nodes and simultaneously send 2 RTP stream over 2 nodes (yes - network traffic will increase) Or maybe is a way to recover UDP connections after a crash and replace node state. Maybe use a Kubernetes tools for that purpose ? Here what i found in mailing-list - https://lists.kamailio.org//pipermail/sr-users/2015-December/091006.html Many thanks for any hints to this topic.

4 years, 4 months

Kamailio IMS with ENUM

by BALL SUN

Hi Is there any information that can help to setup ENUM query in Kamailio IMS? what we wanted to achieve is to when A in domain A (the local domain) makes an INVITE to B, it will check the domain of B from ENUM backend. if it is not on the local domain, it will route to remote IMS core. Thanks - RBK

4 years, 4 months

Does kamailio sip domain makes sense if i defined some alias

by PICCORO McKAY Lenz

in kamctlrc i defined an DOMAIN but later that ip was updated.. so my question: it make sense inclusivelly if already defined more allias in kamailiorc.cfg file? in case of more complicated response,, anybody can explain me about that why? -- Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

4 years, 4 months

Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) -RESOLVED

by Martin W Woscek

Hello, Just wanted to update what fixed our issue. The "contact_user" which is "asterisk" that we were using for the Kamailio context for outgoing SIP trunk/channel in the pjsip.conf file of the Asterisk 16 server was not a valid user in the IMS HSS database (FHoSS implementation). However not being an expert in the IMS, IMHO, the IMS should have kicked out this INVITE as soon as credentials were checked at the front door of the IMS, but rather Kamailio sent it from I-CSCF to S-CSCF to P-CSCF, which is the correct flow, until the P-CSCF sent it back to I-CSCF which then resulted in a 604 HSS user unknown being sent thru the reverse direction back to Asterisk. Anyway, its always good to end Friday on a high note. Thanks, _Martin From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Martin W Woscek Sent: Wednesday, June 24, 2020 1:32 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>; miconda(a)gmail.com Subject: Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) Just to clarify what we are seeing, here is a packet from the INVITE from the S-CSCF (6060) to the P-CSCF(4060). I have eliminated the S-CSCF as this packet looks correct. The P-CSCF is not handling the request URI correctly, and simply sends it back to the I-CSCF where I-CSCF cannot find the user in the HSS based on the ruri updated by the S-CSCF with the local IP address of the UE (192.168.20.1). The P-CSCF should send this INVITE to the UE (703222888(a)192.168.20.1<mailto:703222888@192.168.20.1>) No. Arrival Time Source Destination Source Port Destination Port SIP to tag SIP from tag Method Record-Route Userinfo Request-URI Media Port Protocol Length Request-URI Host Port Owner Username Media Type Media Format Origin-Host Type slice_type SIP from address User Part SIP from address Host Part SIP to address User Part SIP to address Host Part Contact URI User Part Contact URI Host Part Request-URI User Part Request-URI Host Part Request-URI Host Port Info 1553 Jun 17, 2020 09:31:21.722451703 Eastern Daylight Time 192.168.20.128 192.168.20.128 6060 4060 d490a30c-42a8-4fc0-9fe9-b7a21537386d INVITE mt sip:7032228888@192.168.20.1:63871;transport=udp 10626,13370 SIP/SDP 2263 63871 - audio,video ITU-T G.711 PCMU,DynamicRTP-Type-101,0,101,101,DynamicRTP-Type-99,99,99 7033331111 asterisk.irisims.org 7032228888 ims522.irisims.org asterisk 192.168.20.131 7032228888 192.168.20.1 63871 Request: INVITE sip:7032228888@192.168.20.1:63871;transport=udp | Frame 1553: 2263 bytes on wire (18104 bits), 2263 bytes captured (18104 bits) on interface any, id 0 Linux cooked capture Internet Protocol Version 4, Src: 192.168.20.128, Dst: 192.168.20.128 User Datagram Protocol, Src Port: 6060, Dst Port: 4060 Session Initiation Protocol (INVITE) Request-Line: INVITE sip:7032228888@192.168.20.1:63871;transport=udp SIP/2.0 Method: INVITE Request-URI: sip:7032228888@192.168.20.1:63871;transport=udp Request-URI User Part: 7032228888 Request-URI Host Part: 192.168.20.1 Request-URI Host Port: 63871 [Resent Packet: False] Message Header Record-Route: <sip:mt@192.168.20.128:6060;lr=on;ftag=d490a30c-42a8-4fc0-9fe9-b7a21537386d;did=7d2.e0f1> Route: <sip:term@pcscf522.irisims.org;lr> Via: SIP/2.0/UDP 192.168.20.128:6060;branch=z9hG4bK285b.2a57abaad57cfd55bc18a23d4e9cefc8.0 Via: SIP/2.0/UDP 192.168.20.128;branch=z9hG4bK285b.793ba7dcf9fcd18ddef225b83896340e.1 Via: SIP/2.0/UDP 192.168.20.131:5060;received=192.168.20.131;rport=5060;branch=z9hG4bKPjef451437-0a4f-4732-ac54-318740956335 From: <sip:7033331111@asterisk.irisims.org>;tag=d490a30c-42a8-4fc0-9fe9-b7a21537386d To: <sip:7032228888@ims522.irisims.org> Contact: <sip:asterisk@192.168.20.131:5060> Call-ID: 39c15c42-b0b7-48b5-b4bf-ac7a99fbe660 [Generated Call-ID: 39c15c42-b0b7-48b5-b4bf-ac7a99fbe660] CSeq: 19314 INVITE Route: <sip:ims522.irisims.org:5060;lr> Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 Max-Forwards: 68 User-Agent: Asterisk PBX 16.6.0 Content-Type: application/sdp Content-Length: 1121 Message Body From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Martin W Woscek Sent: Wednesday, June 24, 2020 11:21 AM To: miconda(a)gmail.com<mailto:miconda@gmail.com>; Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Subject: Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) Hi, The issue is the P should be sending to the UE not back to the I like the failed flow: Asterisk16-[phone] --->ICSCF--->S-CSCF--->P-CSCF--->I-CSCF ---> [604 HSS user unknown] to Asterisk16[phone] The successful call should look like: Asterisk16-[phone] --->ICSCF--->S-CSCF--->P-CSCF--->Kamailio-[UE] Does anyone have a example P-cscf scripts including the mt.cfg or other script file that make this work? The reverse calls work just fine. Thanks, _Martin From: Daniel-Constantin Mierla <miconda(a)gmail.com<mailto:miconda@gmail.com>> Sent: Wednesday, June 24, 2020 4:53 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>; Martin W Woscek <mwoscek(a)mitre.org<mailto:mwoscek@mitre.org>> Subject: [EXT] Re: [SR-Users] Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) Hello, Kamailio is not generating the 200ok for INVITE (calls), it just sends out what it received from end point, after consuming own Via header (of course, other headers can be changed based on config rules, but Kamailio is not in charge of setting To-tag). Cheers, Daniel On 19.06.20 21:19, Martin W Woscek wrote: Hello all, SIP/UE (boghe or imsdroid) client registered to Kamailio makes call to an Asterisk registered SIP phone is successful: [UE]-Kamailio--->INVITE--->Asterisk16-[phone] [UE]-Kamailio--->200 OK with to_tag--->Asterisk16-[phone] But in reverse direction for a call, Kamailio does not return the SIP OK so no to_tag is sent so call fails to ring and complete: [phone]-Asterisk16 --->INVITE--->Kamailio Where the UE device never receives the INVITE, Asterisk never gets and 200 OK message with the to_tag from the I-CSCF, the call flow itself gets lost in Kamailio, where the P-CSCF sends final INVITE to I-CSCF and and ultimately a 604 HSS user unknown message is sent back to Asterisk from the I-CSCF. Basically Im using the default "sample" configs for both the P and the I-CSCF. Our sauce is in the S-CSCF for out going calls that originate by a registered UE. Any insight or sample Kamailio configuration that Im lacking? Has anyone done this and could share the asterisk and Kamailio script snippets that make it possible. Thanks, _Martin _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- www.asipto.com<http://www.asipto.com> www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda> Funding: https://www.paypal.me/dcmierla

4 years, 4 months

Question about textops

by Mojtaba

Hello, Some confusing things while developing in Kamailio: In is_method_f function in textops module, the code checks the HDR_CSEQ_F in msg also, and if the method name appears in CSEQ_F, The result is returned TRUE. These question will arise why in this function, the CSEQ field in message will check? if(parse_headers(msg, HDR_CSEQ_F, 0)!=0 || msg->cseq==NULL) { LM_ERR("cannot parse cseq header\n"); return -1; /* should it be 0 ?!?! */ } if(m->s==0) return (get_cseq(msg)->method_id&m->len)?1:-1; else return (get_cseq(msg)->method_id==METHOD_OTHER && get_cseq(msg)->method.len==m->len && (strncasecmp(get_cseq(msg)->method.s, m->s, m->len)==0))?1:-1; I think it would be nice to add new function like: is_cseq_method_f in module to avoid some problem understanding. -- --Mojtaba Esfandiari.S

4 years, 4 months

rtpengine inserted random line into sdp

by Andrew Chen

Hi forum, I'm starting my rtpengine project and I'm facing a strange problem with rtpengine. I am seeing this in the SDP part of the INVITE: a=rtcp:52021 a=rtcp-mux 2001:470:7:3A7:0:0:0:2 a=direction:active a=oldmediaip:54.153.25.234 As you can see there is a random insert of the local kamailio IP in there and I'm have tried the following to remove it but failed to succeed: - Remove in the offer the combination of ICE and replace-session-connection flags - Switched between rtpengine_offer and rtpengine_manage - Restarted ngcp-rtpengine-daemon process - Used sdpops function sdp_remove_line_by_preifx() Currently running Kamailio 5.1.2. I am running out of ideas. Any suggestions would be greatly appreciated. -- 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.*

4 years, 4 months

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sr-users June 2020