- sr-users - kamailio.org

Transport question
by Wilkins, Steve 18 May '18

18 May '18

Hi All, I was attempting to switch from UDP to TCP for relaying. I used t_relay_to_tcp() instead of t_relay() => if (!t_relay_to_tcp()) { sl_reply_error(); } The problem is that I am still seeing the SIP messages as UDP on the Server that the messages are relayed to. Any Ideas? Thank you, Steve

6 17

Hunting - Dynamic logistics
by KamDev Essa 18 May '18

18 May '18

So I finally got to the docs on hunting under the TM Modules and there was an awesome write up and it is making a lot more sense now. Basic logic is to append_branch with your extensions and and manage serial and/or parallel forking by setting the q values. The script is static. Is there a module that automatically loads registered contacts from a grouped list of extensions or is the best way to dip into the db and DIY? I would like to know how some of you have implemented hunting, Also is there a way to set the fr_inv_timer of each branch. Thanks in advance. KD

2 1

Alcatel-Lucent Enterprise acquires majority stake in Sipwise
by Andreas Granig 18 May '18

18 May '18

Dear community, I'd like to inform you about Alcatel-Lucent Enterprise (ALE) acquiring a majority stake in Sipwise. Over the last 10 years, we at Sipwise were committed to contributing significant efforts to the open source telecom environment, resulting amongst others in the free and open source Sipwise CE soft-switch appliance, the rtpengine media relay, as well as major contributions to the Kamailio SIP proxy. Under the ALE roof, Sipwise is continuing to operate as its own brand, and our commitment to open source and our community components will be as strong as ever. Please find the official announcement here: https://www.sipwise.com/ale-sipwise/ Best regards, Andreas

3 2

Re: [SR-Users] rtpproxy not forwarding rtp packets.
by Giovanni Tommasini - evosip 18 May '18

18 May '18

Hi Younas, when you have a call if you make a trace in your sip server with sngrep or tshark can you see the RTP packet ? you says "I have kamailio server behind nat with rptproxy.", so is it possible that the router in front your SIP server blocks the traffic? could you have a trace there? I mean the packets exit the rtpproxy but don't arrive to callee or just don't come out of the rtpproxy? Giovanni Tommasini | *evosip* <http://evosip.cloud> 2018-05-18 1:36 GMT+02:00 Aqs Younas <aqsyounas(a)gmail.com>: > Thanks for replying. Callee is actually a sip provider. I see no firewall > or iptables rules on server to prevent rtp following towards callee. > > Is there anything i can do to see which thing is blocking rtps. > > Best > > On Fri, 18 May 2018 at 3:59 AM, Mack Hendricks <ap(a)goflyball.com> wrote: > >> Is there something blocking RTP traffic from reaching the callee? Is the >> callee a carrier or the actually endpoint endpoint (aka SIP Phone)? >> >> *Mack Hendricks / Head of Support / dOpenSource* >> web: http://dopensource.com >> support: +888-907-2085 >> dSIPRouter <http://dsiprouter.org> - GUI focused on implementing >> Kamailio to provide SIP Trunking and PBX Hosting Services >> >> On May 18, 2018, at 12:50 AM, Aqs Younas <aqsyounas(a)gmail.com> wrote: >> >> Greetings list, >> >> >> I have kamailio server behind nat with rptproxy. But i am getting no >> voice on the call. After taking trace i could see that rtpproxy was getting >> rtp packets but not packets was being forwarded towards callee side. >> >> Though i see in rtpproxy logs, packets being relayed from caller side. >> >> Does rtpproxy wait to receive a single rtp frame from callee before >> fowarding that to callee? >> >> Since, i am getting no rtp from callee. So, might be rptrpoxy is waiting >> to learn source address of callee side rtp, that is why it is not >> forwarding rtp packets from caller towards callee. >> >> >> Any pointer/suggestion is much appreciated. >> >> Best Regards, >> >> Aqs Younas >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > >

4 4

Help on how to debug cfg_get variable having unexpected value?
by Malcolm O'Hare 18 May '18

18 May '18

I'm trying to debug an issue I've encountered where a variable that I've set using kamcmd cfg.set has an unexpected value when used in kamailio. The variable should be 0 or 1, but it looks like its getting set accidently somehow. 2018-05-11T22:02:58.344260+00:00 ip-172-31-129-45 kamailio[5253]: DEBUG: <core> [select.c:263]: resolve_select(): 'with_dynamodb_user_data' 2018-05-11T22:02:58.344264+00:00 ip-172-31-129-45 kamailio[5253]: DEBUG: <core> [cfg/cfg_select.c:174]: select_cfg_var(): DEBUG: select_cfg_var(): select fixup is postponed: features.with_dynamodb_user_data 2018-05-11T22:05:03.467110+00:00 ip-172-31-129-45 kamailio[5551]: INFO: <core> [cfg/cfg_ctx.c:608]: cfg_set_now(): INFO: cfg_set_now(): features.with_dynamodb_user_data has been changed to 1 2018-05-11T22:15:51.695118+00:00 ip-172-31-129-45 kamailio[5574]: [2c03c333-1705-41b1-aa26-6f10492a5a6f] DEBUG: with_dynamodb_user_data is 1157627905 I've tried to debug using gdb using breaks and set a watch on the memory address of the var, but it looks like it keeps using different memory locations. The value in the address pointed at by group.handle seems to change for each invocation of set_cfg_now, so when I try and put a watch on the address of *group.handle + var.offset it never gets hit. (gdb) print *group $36 = {num = 25, mapping = 0x7f56ee4474f0, vars = 0x0, add_var = 0x0, size = 164, meta_offset = 968, var_offset = 984, handle = 0x7f56edf92fd0, orig_handle = 0x0, dynamic = 1 '\001', next = 0x0, name_len = 8, name = "f"} (gdb) print *var $62 = {def = 0x7f56ee471668, name_len = 23, pos = 24, offset = 164, flag = 0} Any ideas? Malcolm

2 2

Auth module with shared "secret"
by Asgaroth 18 May '18

18 May '18

Hi All, I am testing a scenario where we have 3 proxies and I'm trying to ensure that if a messages comes in to 1 proxy, which challenges the ua, is able to be auth'd successfully on one of the remaining 2 proxies if, for some reason the ua decides to send the response to one of the remaining two proxies. I am testing this using a seperate kamailio proxy which loadbalances messages across the 3 proxies that I am testing with. I have 1 ua (zoiper) which I tell to subscribe for mwi, the initial subscribe is sent to proxy 1, which sends a 407 back, the new subscribe with auth, from the ua, is sent to proxy 2, which deems the nonce as expired and rechallenges, this continues in a loop across the 3 proxies and never authenticates. I think I'm missing something in my settings that I just cannot put my finger on, and/or, I'm mis-understanding the documentation and how these settings interact with each other. My auth module settings are as follows: # # ----- auth params ----- # modparam("auth", "qop", "auth") modparam("auth", "nonce_count", 1) modparam("auth", "one_time_nonce", 1) modparam("auth", "nid_pool_no", 4) modparam("auth", "nc_array_size", 4194304) modparam("auth", "otn_in_flight_no", 8388608) modparam("auth", "auth_checks_no_dlg", 9) modparam("auth", "auth_checks_in_dlg", 15) modparam("auth", "auth_checks_register", 11) modparam("auth", "secret", "secret_32_char_str") I tried with one_time_nonce enabled and disabled but the result is the same, continous rechallenge. The documentation mentions that if nonce_count and one_time_nonce are enable, and a ua sends a response with nonce and qop in the auth header then one_time_nonce will not be used, one_time nonce will be used as a fallback in the case the ua does not support qop. My auth block in the routing logic looks like so: route[AUTH_CHECK]{ xlog("L_DBG", "route[AUTH_CHECK] : $rm : Performing authentication checks for '$var(creds_key)'"); $var(challenge_flags) = 1; if ( ! pv_auth_check("$fd", "$sht(creds=>$var(creds_key)::Password)", "20", "1") ) { switch($retcode) { case -1: xlog("L_DBG", "route[AUTH_CHECK] : $rm : Generic error occurred, no reply sent out."); break; case -2: xlog("L_DBG", "route[AUTH_CHECK] : $rm : Invalid password supplied, re-challenging client and removing from hash table."); sht_rm_name_re("creds=>$var(creds_key)"); break; case -4: xlog("L_DBG", "route[AUTH_CHECK] : $rm : Nonce has expired, re-challenging client."); $var(challenge_flags) = $var(challenge_flags) + 16; break; case -5: xlog("L_DBG", "route[AUTH_CHECK] : $rm : No credentials supplied, challenging client."); break; case -6: xlog("L_DBG", "route[AUTH_CHECK] : $rm : Nonce has already been used to auth a previous request, challenging client."); break; case -8: xlog("L_DBG", "route[AUTH_CHECK] : $rm : Auth user is different to From/To user, challenging client."); break; } xlog("L_INFO", "route[AUTH_CHECK] : $rm : auth_challenge will be called with '$fd' and '$var(challenge_flags)'"); auth_challenge("$fd", "$var(challenge_flags)"); exit; } consume_credentials(); } When I look at the logs I always see the proxies seeing the nonce as expired: oUBGp0n6HRWu_mvPPxJQqg..: INFO: <script>: route[AUTH_CHECK] : SUBSCRIBE : auth_challenge will be called with 'example.com' and '17' oUBGp0n6HRWu_mvPPxJQqg..: INFO: <script>: route[AUTH_CHECK] : SUBSCRIBE : auth_challenge will be called with 'example.com' and '17' oUBGp0n6HRWu_mvPPxJQqg..: INFO: <script>: route[AUTH_CHECK] : SUBSCRIBE : auth_challenge will be called with 'example.com' and '17' oUBGp0n6HRWu_mvPPxJQqg..: INFO: <script>: route[AUTH_CHECK] : SUBSCRIBE : auth_challenge will be called with 'example.com' and '17' oUBGp0n6HRWu_mvPPxJQqg..: INFO: <script>: route[AUTH_CHECK] : SUBSCRIBE : auth_challenge will be called with 'example.com' and '17' oUBGp0n6HRWu_mvPPxJQqg..: INFO: <script>: route[AUTH_CHECK] : SUBSCRIBE : auth_challenge will be called with 'example.com' and '17' I have checked: [1] that the proxies are all syncing time, and they are not out of sync, [2] The source address of the client doesnt change (in case auth_checks_no_dlg is failing), [3] The r-uri of the subscription dialog is the same (in case auth_checks_no_dlg is failing) I am using kamailio version 5.0.6: kamailio -V version: kamailio 5.0.6 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 15:57:47 Mar 5 2018 with gcc 4.8.5 Any thoughts on what I can look at or if I'm missing something fundamental here? All suggestions/criticisms are appreciated. Thanks for reading.

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rtpproxy not forwarding rtp packets.
by Aqs Younas 17 May '18

17 May '18

Greetings list, I have kamailio server behind nat with rptproxy. But i am getting no voice on the call. After taking trace i could see that rtpproxy was getting rtp packets but not packets was being forwarded towards callee side. Though i see in rtpproxy logs, packets being relayed from caller side. Does rtpproxy wait to receive a single rtp frame from callee before fowarding that to callee? Since, i am getting no rtp from callee. So, might be rptrpoxy is waiting to learn source address of callee side rtp, that is why it is not forwarding rtp packets from caller towards callee. Any pointer/suggestion is much appreciated. Best Regards, Aqs Younas

2 2

Further question on transport issue.
by Wilkins, Steve 17 May '18

17 May '18

Hello, I am still struggling with getting TCP relay to work. Kamailio is now trying to send TCP but it is using the incorrect port. Kamailio is relaying to Asterisk on xx.xx.xx.xx:some-large-port but Asterisk is listening on xx.xx.xx.xx:5060. I'm not sure why this is happening. I would appreciate any help. Thank you,

3 4

cfg script 101
by KamDev Essa 17 May '18

17 May '18

I have decided to do Hunting DIY. I needed some help with the cfg script part and using a database dip. Say I do the db dip sql_xquery("ca", "call usp_GetHuntBranches( '$rU', '$rd');", "ra");that will return the uri/q pairs and I will add using append_branch dynamically. How do I do the looping in the script? I have 2 ways to deliver the list via the sp. - Rows with uri and q values. easiest - Single row with variable paired fields like say ur1,q1,ur2,q2 etc etc Questions : - Can the script handle sql_xquery with multiple rows. If so, can I get a code snippet or example that show me how to iterate through. Is there a "while" loop capability in the script? - Is there a "substring" or "split" function in the script that could allow me to get the data as a single text with uri and q separated by a \ or |. - Or is there a document that explains the various generic functions I can use the cfg script area. KD

2 2

NatHelper's Keep-alive timeout behind proxy
by Asgaroth 17 May '18

17 May '18

Hi All, We are looking to enable nathelpers keepalive timeout parameter where it will remove an aor's contact if it doesn't receive a reply from the ua. The problem we have is that our registrar's are seperated from our proxies, so this means nathelper is sending options messages via the proxy to the ua. In this case, if the us has lost connectivity, the kamailio proxy sends a 408 Request Timeout back to the registrar and the nathelper module sees that as a response to the initial options message it sends, so it never actually removes the contact from the location table. Is there a way I can stop the proxy from sending any responses for these specific nat ping/options messages, or, is there a way I can tell the nathelper modules that a 408 response should be treated as the ua being down and then after the timeout interval remove the contact from the location table? I've had a read of the nathelper, tm and sl modules but I didnt see anything that jumps out at me that I could achieve one or the other possible solutions I mentioned above. Does anyone have any suggestions or pointers on how we can achieve this with the registrars being seperate from the edge proxies. Any pointrs/tips etc would be greatly appreciated. Thanks

2 2

Updated NAT traversal article
by Alex Balashov 17 May '18

17 May '18

Hi, I have updated this article with some other topics: http://blog.csrpswitch.com/server-side-nat-traversal-with-kamailio-the-defi… "The definitive guide" is of course an ambitious and moving target, so there may be more evolution in the future. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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t_load_contacts() and t_next_contacts() guts
by KamDev Essa 16 May '18

16 May '18

I am tackling Hunting and see that there are two functions recommended to accomplish this. Before I jump into using them I wanted to know a bit about each regard to the call flow. >From where does t_load_contacts() load contacts and how does it order then in case of a serial fork flow, Is there a generic way of doing it say with a loop in the script. Docs on these two functions are a bit cryptic for me H

2 7

Contact Header
by KamDev Essa 16 May '18

16 May '18

My scenario is UA <====> Kamailio <====> Freeswitch UA calls a number. Kam delivers to FS and FS plays a wav file. My problem is that Kam relays the 200 OK from FS back to UA without change and UA ACKs back to FS because the FS IP is in the Contact Header. - Why does Kam not change the contact header since the ACK should come back to Kam - If this is normal what do I need to do to change the contact header for the 200K to the UA to make this work. KD

3 4

Whats is wrong in this call
by Sergey Safarov 15 May '18

15 May '18

Hello I have a call with not passed ACK by Kamailio. http://prntscr.com/jhtobo Same behaviour i looked before when SIP endpoint contact string is wrong In this call look as contact string correct is correct (packets 7 and 8) and i not understand why ACK (packet 9) not passed to callee. Could you help me understand this. PCAP file is create using homer tracer and TCP transport packets is stored as UDP. Sergey

3 4

Last negative 5XX reply not getting relayed
by Sergiu Pojoga 15 May '18

15 May '18

Not sure if it's a default Kamailio behavior or something I'm doing wrong... When receiving a negative reply for an INVITE (more like in the 5xx range), proxy doesn't relay the message to the originator but instead issues it's own "500 Service Unavailable". That doesn't happen, from observations so far, to "603 Declined" or "486 Busy Here" or "404 Not Found" or "200 OK" replies. *For example, here's the negative reply proxy receives from the callee:* 2018/05/14 10:52:37.716627 65.XX.XX.166:5060 -> 65.XX.XX.167:5060 SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 65.xx.xx.167;branch=z9hG4bK9223.00b4f3143d245a895c4bb6a107add258.0;received=65.xx.xx.167;rport=5060 Via: SIP/2.0/UDP 205.xx.xx.221;received=205.xx.xx.221;rport=5060;branch=z9hG4bKQ4XpXga0vgZ8B From: "514XXXXXXX" <sip: 514XXXXXXX @205.xx.xx.221>;tag=tXj0NHrje9jNS To: <sip:450XXXXXXX@65.xx.xx.167>;tag=as18e452b7 Call-ID: 48eac296-d229-1236-a685-005056a149a4 CSeq: 122818714 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas X-Asterisk-HangupCause: Subscriber absent X-Asterisk-HangupCauseCode: 20 Content-Length: 0 *Here's what the originator gets in the final message:* 2018/05/14 10:52:37.718143 65.xx.xx.167:5060 -> 205.xx.xx.221:5060 SIP/2.0 500 Service Unavailable Via: SIP/2.0/UDP 205.xx.xx.221;rport=5060;branch=z9hG4bKQ4XpXga0vgZ8B;received=205.xx.xx.221 From: "514XXXXXXX" <sip:514XXXXXXX@205.xx.xx.221>;tag=tXj0NHrje9jNS To: <sip:450XXXXXXX@65.xx.xx.167>;tag=31a2c69b5d434ef4a60796e26f568bf0-9343 Call-ID: 48eac296-d229-1236-a685-005056a149a4 CSeq: 122818714 INVITE Server: Kamailio 5.1 Content-Length: 0 I'd like any reply to be relayed back as is, unless I do some t_check_status() => t_reply() for specific reasons (which I have none at this point). Thanks Sergiu

3 5

ACC_CDRS table with src and dst data are inverted
by Emanuele Gambaro 15 May '18

15 May '18

Hi to all, I have some troubles with acc_db module and acc_cdrs table. First scenario: Alice calls Bob and after a while, when a call is established, Alice hangs up, sending a BYE. The acc_cdrs table is filled up with the right data: * src_user, src_domain, src_ip are from Alice account (the caller) * dst_user etc etc are from Bob account (the callee) In this scenario all works as expected: the src data are from who starts the call, and the dst data are from who receives the call. In the following scenario src and dst data are inverted. Alice calls Bob and after a while, when a call is established, Bob hangs up, sending a BYE. The acc_cdrs table now is: * src_user, src_domain, src_ip are from Bob account (the callee that sends the BYE message) * dst_user etc etc are from Alice account (the caller that receives the BYE) This is the intended behavior or I’m missing something? How I can record the CDR data with the right CALLER and CALLEE data ? Emanuele

2 1

UAC Module Registration Table
by Greg Adams 15 May '18

15 May '18

Hey there, I am currently using the UAC module to maintain remote registrations. I am just curious if it is possible to register new users immediately rather than waiting for reg_timer_interval? I am not seeing any functionality like that but just wondering if I am missing something or if it would be possible to add in. For example an RPC command that takes a key and instantly registers that user. Thanks, Greg

2 1

Parallel Forking - 1 Ring on extension to notify that Call Forward is in place
by KamDev Essa 14 May '18

14 May '18

I am trying to simulate a Broadsoft feature that blips the extension with a single ring that basically heralds the user that their extension has CF turned on. I know parallel forking can do that. But how do I set the fr timer on the call to the extension @ just 1 ring (5 seconds) and let the other one the default timeout set in fr_inv_timer. KD

2 1

Tagging a call with custom pseudo variables - cfg 101
by KamDev Essa 14 May '18

14 May '18

Being spoiled by Freeswitch profiles and dialplans, the stock 1 piece Kamailio cfg file is cool @ first to get certain call flows knocked out but quite a maze when you get in deeper. Right now, using the 1 piece stock cfg file, I am faced with the fact that my carrier outbound calls started failing after I enabled VM. Basically WITH_VOICEMAIL in route LOCATION started sending the carrier calls to TOVOICEMAIL and it basically dies there. I can see the route(TOVOICEMAIL) line and can condition it out with an if but I really don't now how to differentiate between a ext to ext call or a outbound call in the cfg file. However given SQL I could. Is there a way for me to place custom pseudo variables in the call right @ the start and use them in the script as needed. For example I could mark the call as ext to ext or carrier bound right @ the start and dodge VM in the LOCATION route. KD

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Question about using Kamailio and Asterisk and flow of an "INVITE"
by Wilkins, Steve 13 May '18

13 May '18

Hello All, I am trying to resolve, in my mind, the flow of a WebRTC<=>WebRTC call using Kamailio and Asterisk. Each WebRTC client is registered in Kamailio and when I call WebTRC Client1 from WebRTC Client2 what I see is -> The Invite is sent from Kamailio to Asterisk and then Asterisk is sending the Invite back to Kamailio. Also depending on The version of Asterisk, the INVITE will then get forwarded to the AOR that is registered in Kamailio for the called number. Does this seem correct? It seems like there is an extra hop in there. The reason I am now very curious now is because everything works fine if using Kamailio 5.0 and Asterisk 14.6, but I switch to Asterisk 15.3 I get the extra hop and call is dropped after 30 seconds. I would appreciate any thoughts on this. Thank you in advance.

3 6

"Sleeping Giant - No Snore" Kam?
by KamDev Essa 11 May '18

11 May '18

We have noticed pesky SIP hackers already knocking on our POC servers. Kam is holding the fort by sending out the proverbial 401/407 (snore). I can SQLOPS to do a dip to check destination domain or Source IP and nix the snore for alien requests, but wanted to know if there was a native way to keep silent when we have a request from foes. KD

3 6

Understanding port changing
by Social Boh 11 May '18

11 May '18

Hello list, I'm looking for to understand a problem have with Kamailio. The configuration is: user (behind NAT) -> Kamilio (Public IP) -> user (behind NAT) When I make a call from user A to user B (both registered) , User B receive the INVITE correctly but in the ringing SIP message change the port present on the Contact Header and same think on 200OK so when Kamailio try to send it ACK SIP message to this port fail because the Softphone listen on the port where has received the INVITE from Kamailio. This happens with 3 different Softphone: BRIA, Xlite, Linphone on the same computer. Thank you Regards -- --- I'm SoCIaL, MayBe

2 1

WebRTC to SIP gateway
by Pan Christensen 11 May '18

11 May '18

Hello! It's been several years since I've used Kamailio. My current employer wants to implement WebRTC, which is currently not supported in our SIP backend, and asked if I could set up a Kamailio server as a gateway. I've been able to make calls in all directions between SIP and WebRTC clients registered locally on Kamailio. When I tried to connect the server to the SIP backend, I ran into an issue. I'm able to register SIP clients in the backend via the gateway and make calls everywhere. However, the WebRTC client fails to register. Here are the messages between the Kamailio gateway and the SIP backend: U 2018/05/09 10:12:58.316643 GATEWAY:15060 -> DOMAIN:5060 REGISTER sip:DOMAIN SIP/2.0. Via: SIP/2.0/UDP GATEWAY:15060;branch=z9hG4bK4fc6.04d1730be5b78d595c69a3aa137987c1.0. Via: SIP/2.0/AUTO lr2l9s72ehhc.invalid;rport=61353;received=CLIENT;branch=z9hG4bK5927151. Max-Forwards: 68. To: <sip:4777519304@DOMAIN>. From: <sip:4777519304@DOMAIN>;tag=9qhqhrnj3s. Call-ID: j5h7830ivr5dfc2mn5sov1. CSeq: 4 REGISTER. Contact: <sip:c3qkm4fv@lr2l9s72ehhc.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:0ef2deac-2d56-465a-840b-543b9fd01af8>";expires=600. Expires: 600. Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO. Supported: path,gruu,outbound. User-Agent: JsSIP 3.2.9. Content-Length: 0. Path: <sip:GATEWAY:15060;lr>. . U 2018/05/09 10:12:58.368409 DOMAIN:5060 -> GATEWAY:15060 SIP/2.0 400 Wrong transport. Provided transport either invalid or not supported.. Via: SIP/2.0/UDP GATEWAY:15060;branch=z9hG4bK4fc6.04d1730be5b78d595c69a3aa137987c1.0. Via: SIP/2.0/AUTO lr2l9s72ehhc.invalid;rport=61353;received=CLIENT;branch=z9hG4bK5927151. To: <sip:4777519304@DOMAIN>;tag=91334f57. From: <sip:4777519304@DOMAIN>;tag=9qhqhrnj3s. Call-ID: j5h7830ivr5dfc2mn5sov1. CSeq: 4 REGISTER. Content-Length: 0. I believe that this error message is caused by ';transport=ws' in the Contact header. I'm not allowed to modify this header. In the backend database, I found that some other clients have ';transport=UDP' in their path headers, so I tried to add that. (Why can I not add parameters in path module without adding username?) I still got the same error. How do I best proceed? For your information: We have outsourced the development of the WebRTC client, so we are able to change it. We also have the option of paying the supplier of the backend for development there. With kind regards Pan B. Christensen Developer Phonect AS Brugata 19, PB 9156 Grønland, N-0133 Oslo, Norway E-mail: pan.christensen(a)phonect.no<mailto:pan.christensen@phonect.no> Mobile: +47 41 88 88 00 [cid:image007.png@01D3A0E8.376921D0] <http://www.phonect.no/> [facebook_2]<https://www.facebook.com/phonectno> [LinkedIn_logo_initials (1)] <https://www.linkedin.com/company/44983?trk=tyah&trkInfo=clickedVertical%3Ac…>

2 2

"No Answer" time out after number of rings
by KamDev Essa 11 May '18

11 May '18

Does Kamailio have any way to control how long it tries to connect per call? KD

2 5

Is Media Release (signaling only) possible in Kamailio?
by KamDev Essa 10 May '18

10 May '18

What cfg files changes do I need to make to get Kamailio to be a signally only server, yet manipulate the SDP part of the INVITE message to allow remote parties to send media to each other? In Freeswitch terms "bypassmedia". KD

2 14

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