- sr-users - kamailio.org

t_save_lumps() and failure_route
by Julien Chavanton 14 Jun '18

14 Jun '18

Hi, I have a scenario where I am using. t_save_lumps(), lookup() and then I can choose to drop some branches in branch routes. However, it is possible that I endup without any branch left, in this case t_relay() is returning and no failure route is called/created. At this point I need to recover the original message I believe it is not possible to restore/reset the message without passing by failure route ? Else, I could trigger a failure route using t_relay(), is there any "handy/clean" way to trigger failure_route() ? Regards

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Memory leak in tm with push notifications
by Ivaylo Markov 14 Jun '18

14 Jun '18

Hello, I am trying to set up Kamailio as a push notifications proxy, closely following the example in the "Kamailio in a Mobile World" presentation (https://www.slideshare.net/FedericoCabiddu/kamailioinamobileworld-51617342). I am running Debian 9 and Kamailio 5.1.3 from the official Debian repositories. I believe the main modules involved in the issue below are tm, tmx, and tsilo. Every call passing through the proxy leads to a small memory leak in the tm module - there is a large amount of "delayed free" memory cells from tm's internal hash table. At some point the shared memory runs out and Kamailio restarts. Using the "kamcmd corex.shm_summary" command I was able to see that the top users of shared memory are "tm: h_table.c: build_cell" and "core: core/sip_msg_clone.c: sip_msg_shm_clone" with the same allocation count. I experimented with removing different parts of the configuration and noticed that commenting out the "t_continue(...)" call in the "PUSHJOIN" route (see slide #22) prevents the leak from happening. Maybe something in that function is incrementing the reference counter to the hash table cell, but it is not decrementing the counter when done? I tried looking around the source code of the tm and tmx modules, but saw nothing suspicious. I also tried using gdb with a breakpoint in t_continue_helper (tm/t_suspend.c:166) hoping to see what else is accessing the htable cell, but was unable to find anything of use. Has someone encountered anything like this? Can you provide more directions on debuggin this? I can provide some bits of configuration, but an entire test setup would be rather difficult, unfortunately. Thank you for your time, Ivo

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integrate huawei HSS with kamailio
by eyas barhouk 14 Jun '18

14 Jun '18

hello dears i'm trying to integrate kamailio IMS with Huawei Hss instead of Fraunhofer HSS , so is there any one has an experiment with this or tested it before ??? if yes, could you please guide me how to do that ? or where to start ? thanks in advance

2 2

Forbidden - Private identity not found (Authorization: username)
by eyas barhouk 14 Jun '18

14 Jun '18

Hello dears when i was trying to build an IMS platform by using kamailio with sip digest authentication scheme and ims_auth module, i couldn't register and the result was as following : 1-the first register message passed to the s-cscf then S-cscf replaying with the message “401 unauthorized - challenging the ue” 2- the second register pass to the S-cscf with the authorization header then the S-CSCF replying with "Forbidden - Private identity not found (Authorization: username)" So based on that could any one tell me what is the reason of this error and how to fix it ?? Thanks in advance

2 5

Using a var as timeout value in dlg_set_timeout_by_profile function
by jenus＠cyberchaos.nl 13 Jun '18

13 Jun '18

I'm trying to use the dlg_set_timeout_by_profile in the xhhtp route to update the dialog timeout for active calls. But if i use a var as the timeout value the call disconnects immediately. This works fine (call disconnects after 80 sec): dlg_set_timeout_by_profile("subscriber_outbound", "$var(SipName)@$var(SipDomain)::$var(DialogCallid)", "80"); This fails (call disconnects immediately: $var(DialogTimeout)= 80; dlg_set_timeout_by_profile("subscriber_outbound", "$var(SipName)@$var(SipDomain)::$var(DialogCallid)", "$var(DialogTimeout)"); Are vars supported in the dlg_set_timeout_by_profile function? And is it possible to use it in any route as described in the docs? I'm running kamailio 5.1.3. Thanks, Jan

2 1

Registration of UACs i n a partially encrypted scenario
by mip FKF 13 Jun '18

13 Jun '18

Hello, we are pretty new to SIP and kamailio, we do have some questions regarding the following scenario: We have a number of UACs in a small network which are required to communicate without encryption because the are not able to consume certificates. We want to use kamailio (as a proxy?) to establish an encrypted connection to a backend UAS. 1. Is it possible to directly register the UACs with the UAS eventhough communication between kamailio and the UAS is encrypted ? 2. How do we need to configure kamailio in order to make this scenario work ? Thank you for your suppport

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Re: [SR-Users] Question on Contact Header on Kamailio 200 OK to PBX
by Floimair Florian 13 Jun '18

13 Jun '18

Yes it should, but only if “rewrite_contact” is set to “no” for the endpoint (chan_pjsip). With best regards Florian Floimair Innovation - Software-Development COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstraße 51 http://www.commend.com<http://www.commend.com/> Security and Communication by Commend FN 178618z | LG Salzburg From: sr-users <sr-users-bounces(a)lists.kamailio.org> on behalf of "Wilkins, Steve" <swwilkins(a)mitre.org> Reply-To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Date: Tuesday, 12. June 2018 at 16:27 To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: [SR-Users] Question on Contact Header on Kamailio 200 OK to PBX Hi All, Kamailio is sending a 200 OK back to an INVITE from Asterisk, and Asterisk is sending the ACK back but the AOR can’t be found in Kamailio 200 OK Kamailio to Asterisk with a contact of Contact: <sip:fgectrdv@9ot28m83bkur.invalid;alias=128.147.123.1~63486~6;alias=128.147.123.1~63486~6;transport=ws;ob> Asterisk ACK to Kamailio Request-Line: ACK sip:fgectrdv@192.33.11.108:5060;transport=TCP;alias=128.147.123.1~63486~6;alias=128.147.123.1~63486~6;ob SIP/2.0 Shouldn’t Asterisk be using fgectrdv@9ot28m83bkur in the ACK? Or did Kamailio send Incorrect Contact? Note: fgectrdv@9ot28m83bkur is the what Kamailio shows as AOR. Thank you, -Steve

1 0

Asterisk inviting Kamailio
by Wilkins, Steve 13 Jun '18

13 Jun '18

Hello All, I have noticed that sometimes when a call is made from one endpoint to another through Asterisk via Kamailio, Asterisk sends an INVITE to Kamailio even after the call has been established. Sometimes this does not happen. When it does happen, calls drop. Why would an INVITE be sent back to Kamailio? Also, Pan, I am working on the response you requested yesterday. Thanks you All, -Steve

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Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio
by Wilkins, Steve 12 Jun '18

12 Jun '18

Got it working. Thank you everyone. -----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Wilkins, Steve Sent: Sunday, June 10, 2018 3:06 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio >>>Email originates from a non-MITRE system. Use caution.<<< Alex, Pan, Daniel,... Could this group => group:BUNDLE audio video in Message Body have anything to do with my Kamailio Video issue. Thank you!! -Steve -----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Wilkins, Steve Sent: Sunday, June 10, 2018 12:14 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio Hi Alex, No, I'm not using rtpengine. It must definitely be some sort of Codec issue though, since I can seem to move the problem around and actually finally get video on the softphone. It was a little strange that I lost Audio on the WebRTC client though, since I only disabled VP8. Since Kamailio is only relaying I am so confused why introducing Kamailio is messing up Video. Thank you, -Steve -----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Alex Balashov Sent: Sunday, June 10, 2018 11:46 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio Unless you're using rtpengine, you're just moving the problem around. Kamailio does nothing with SDP unless by way of rtpengine or told in some other way, such as sdpops. On June 10, 2018 11:44:47 AM EDT, "Wilkins, Steve" <swwilkins(a)mitre.org> wrote: >I just did a test where I disabled VP8 in Kamailio using SDPOPS and I >now get Video on the softphone however, I lost two-way Audio. Kamailio >seems to be doing something with the codecs but still can't put my >finger on it. The WebRTC Client, who is the caller, needs VP8 for >Video, and apparently Audio. > >I'm not sure if I getting closer or just moving the problem around. > >Thanks All! > >From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf >Of Wilkins, Steve >Sent: Saturday, June 9, 2018 12:47 PM >To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> >Subject: [SR-Users] No Video between WebRTC Client and Softphone when >using Kamailio...works without Kamailio > >>>>Email originates from a non-MITRE system. Use caution.<<< >I am desperately trying to resolve a big issue I have when using >Kamailio 5.2.0 and Asterisk 15.3. > >As soon as I go through Kamailio, I get no Video on either side of the >call; I do get two-way audio and the call stays connected. If I simply >go right through Asterisk, I also get two-way Video. I am having a >tough time determining why Kamailio is messing with the Video portion >of the call. >I have included a full pcap file. As a side note when comparing pcap >traces of the Non Proxied vs Proxied call, the INVITE and 200 OK from >Asterisk and Softphone are exactly the same. > >Thank you, >-Steve -- Alex -- Sent via mobile, please forgive typos and brevity. _______________________________________________ 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

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Question on Contact Header on Kamailio 200 OK to PBX
by Wilkins, Steve 12 Jun '18

12 Jun '18

Hi All, Kamailio is sending a 200 OK back to an INVITE from Asterisk, and Asterisk is sending the ACK back but the AOR can't be found in Kamailio 200 OK Kamailio to Asterisk with a contact of Contact: <sip:fgectrdv@9ot28m83bkur.invalid;alias=128.147.123.1~63486~6;alias=128.147.123.1~63486~6;transport=ws;ob> Asterisk ACK to Kamailio Request-Line: ACK sip:fgectrdv@192.33.11.108:5060;transport=TCP;alias=128.147.123.1~63486~6;alias=128.147.123.1~63486~6;ob SIP/2.0 Shouldn't Asterisk be using fgectrdv@9ot28m83bkur in the ACK? Or did Kamailio send Incorrect Contact? Note: fgectrdv@9ot28m83bkur is the what Kamailio shows as AOR. Thank you, -Steve

1 0

Facing Problem Registration / Invite pass through one side voice
by Ankit vasava 12 Jun '18

12 Jun '18

Hello , I have installed opensips latest version and have free switch. All user is registered with free switch through opensips like registration pass though you can find out my config file into attachments also pcap log. My free switch IP is *192.168.1.147:5060 <http://192.168.1.147:5060>* Opensip IP is : *192.168.1.12:5069 <http://192.168.1.12:5069>* I m sending registration and invite request to public IP and that public IP is bind to open sips. *Problem:* when external user who is registered through public ip when he is dial any extension number who is registered with local ip at that time we are facing one way voice. I need voice(RTP) could be transfer both the side. -- *Regards,Ankit vasava* Sr. VoIP Developer *Phone: *+91-9879491525 *Skype: *ankit.vasava Http://in.linkedin.com/in/ankitvasava

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unable to Kamailio IMS 5.1.4?
by vinay kumar 12 Jun '18

12 Jun '18

Hi All, Im getting following error when i start kamailio.cfg of icscf. ERROR: ctl [init_socks.c:115]: init_unix_sock(): ERROR: init_unix_sock: bind: No such file or directory [2] ERROR: ctl [ctl.c:290]: mod_init(): ERROR: ctl: mod_init: init ctrl. sockets failed ERROR: <core> [core/sr_module.c:990]: init_mod(): Error while initializing module ctl (/usr/local/lib64/kamailio/modules/ctl.so) can anyone help me on this?? Regards, Vinay

1 0

unable to bring up kamailio 5.1.4
by vinay kumar 12 Jun '18

12 Jun '18

Hi, Im facing following issue when I tried to bring up kamailio IMS 5.1.4. ERROR: <core> [core/modparam.c:140]: set_mod_param_regex(): parameter <hashing_type> of type <2> not found in module <ims_usrloc_pcscf> CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/pcscf/kamailio.cfg, line 354, column 47: Can't set module parameter can anyone share complete installation steps for Kamailio IMS 5.1.4 for volte call?? Regards, Vinay

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Problem with TLS
by Arik Halperin 12 Jun '18

12 Jun '18

Hello, I’m using TLS After receiving 200OK from kamailio: r2voip.clear2voipdialer I/(NativeSdk_2_0) 1528174138320 PJSIP: (NativeSdk_2_0) 1528174138320 PJSIP:2018-05 07:48:58.319 pjsua_core.c RX 2203 bytes Response msg 200/INVITE/cseq=8107 (rdata0x7a2c56fb38) from TLS 70.36.25.65:443: SIP/2.0 200 OK Via: SIP/2.0/TLS 10.134.232.109:44097;received=109.253.173.146;rport=31373;branch=z9hG4bKPj4MV5llP9SW5ufk-OcFB-Qh78PmIQFrRk;alias Record-Route: <sips:10.168.10.227:5099;r2=on;lr=on;ftag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO;nat=yes> Record-Route: <sips:70.36.25.65:443;transport=tls;r2=on;lr=on;ftag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO;nat=yes> From: "number" <sips:972523391991@kamprod.telemessage.com<mailto:972523391991@kamprod.telemessage.com>>;tag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO To: <sips:1111111@kamprod.telemessage.com<mailto:1111111@kamprod.telemessage.com>>;tag=64H63g861ajHj Call-ID: Sq4jR85o3Caz2XTXo-71FKAdbJ1x9vz2 CSeq: 8107 INVITE Contact: <sip:1111111@10.168.10.200:5080;transport=tls> User-Agent: FreeSWITCH-mod_sofia/1.6.20+git~20180123T214909Z~987c9b9a2a~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Require: timer Supported: ti PJSIP responds with: Secure dialog requires SIPS scheme in Contact and Record-Route headers, ending the session What is the reason for this? How can I fix this issue? Thanks, Arik Halperin

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Disable basic authentication?
by Skip Morse 11 Jun '18

11 Jun '18

Hi All, I couldn't get a clear answer from the documentation. Based on the description of www_authenticate and proxy_authenticate: " The function verifies credentials according to RFC2617. If the credentials are verified successfully then the function will succeed and mark the credentials as authorized (marked credentials can be later used by some other functions). If the function was unable to verify the credentials for some reason then it will fail and the script should call www_challenge which will challenge the user again." The RFC outlines basic and digest auth. Is there a way to disable the ability for a UA to use basic auth? Or maybe these is disabled already? I'm likely understanding this wrong, it seems to me that it's allowed for a UA to auth with basic even if the challenge is for digest. Any clarification would be helpful. Thanks! -Skip

2 2

SIP/2.0 603 Failed to get local SDP for sip to webrtc (SIPML5)call
by vinod mn 11 Jun '18

11 Jun '18

Hi i configured sipml5 and kamailio , sipml5 clients registers to kamailio when i make a call to kamailio call lands on the sipml5 client but i am getting this error *"SIP/2.0 603 Failed to get local SDP" *on chrome/firefox/opera webbrowsers. kamilio configuration as per http://www.kamailio.org/docs/ modules/4.4.x/modules/websocket.html#idp53411716 -- Thanks and regards Vinod.M.N

2 1

sip digest authentication
by eyas barhouk 11 Jun '18

11 Jun '18

Hello dears I’m trying to setup Kamailio with sip digest authentication scheme by using the following configuration from s-cscf side : modparam("ims_auth", "registration_default_algorithm", "HSS-Selected") and the result was as following : 1. when the register message passed form the s-cscf to the hss and in the diameter trace I found the “SIP-AUTHENTICATION-SCHEME” in the MAR request as “Digest-MD5” 1. After that the HSS send MAA message with “DIAMETER-ERROR-AUTH-SCHEME-NOT-SUPPORTED” 2. S-cscf replay with the message “Forbidden – privet identity not found (Authorization : username)” So based on that could any one tell where was my mistake or how can I setup Kamailio with SIP-DIGEST scheme Thanks

2 4

JSON RPC Error: LUMP_RPL_BODY already added!
by Asgaroth 11 Jun '18

11 Jun '18

Hi All, We are trying to query the json rpc server on our registrar for the contact of an aor, but we are seeing the following error(s) being displayed in the log when we query the location table: ERROR: <core> [core/data_lump_rpl.c:83]: add_lump_rpl2(): LUMP_RPL_BODY already added! ERROR: xhttp [xhttp_mod.c:410]: xhttp_send_reply(): Error while adding reply lump We are getting the aor details back in response from the rpc call, but the error's in the log are a little concerning and I cannot see where we are doing something wrong. As an example, the following curl request returns the correct response, but the above error is displayed in the log, Am I doing something wrong here, I cannot see what the issue is: curl --header 'Content-Type: application/json' --data-binary '{ "jsonrpc": "2.0", "method": "ul.lookup", "params": [ "location", "user(a)example.com" ], "id": 3 }' http://registrar.example.com/rpc A secondary question is, is there an upper limit on the size of the "id" parameter? This kamailio instance is v5.0.6. Any thoughts as to what this error means? Thanks

2 5

(N)DB_FIREBASE module?
by Pan Christensen 11 Jun '18

11 Jun '18

Greetings all. We are currently developing web and mobile apps using firebase: https://firebase.google.com/products/realtime-database/ . We have been discussing the possibility of Kamailio querying this database. I assume that it will be possible to query it using HTTP and JSON modules, but it would probably be easier to use the DB API. Any plans to develop such a module? 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: 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…>

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Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio
by Wilkins, Steve 10 Jun '18

10 Jun '18

I just did a test where I disabled VP8 in Kamailio using SDPOPS and I now get Video on the softphone however, I lost two-way Audio. Kamailio seems to be doing something with the codecs but still can't put my finger on it. The WebRTC Client, who is the caller, needs VP8 for Video, and apparently Audio. I'm not sure if I getting closer or just moving the problem around. Thanks All! From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Wilkins, Steve Sent: Saturday, June 9, 2018 12:47 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio >>>Email originates from a non-MITRE system. Use caution.<<< I am desperately trying to resolve a big issue I have when using Kamailio 5.2.0 and Asterisk 15.3. As soon as I go through Kamailio, I get no Video on either side of the call; I do get two-way audio and the call stays connected. If I simply go right through Asterisk, I also get two-way Video. I am having a tough time determining why Kamailio is messing with the Video portion of the call. I have included a full pcap file. As a side note when comparing pcap traces of the Non Proxied vs Proxied call, the INVITE and 200 OK from Asterisk and Softphone are exactly the same. Thank you, -Steve

2 3

No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio
by Wilkins, Steve 09 Jun '18

09 Jun '18

I am desperately trying to resolve a big issue I have when using Kamailio 5.2.0 and Asterisk 15.3. As soon as I go through Kamailio, I get no Video on either side of the call; I do get two-way audio and the call stays connected. If I simply go right through Asterisk, I also get two-way Video. I am having a tough time determining why Kamailio is messing with the Video portion of the call. I have included a full pcap file. As a side note when comparing pcap traces of the Non Proxied vs Proxied call, the INVITE and 200 OK from Asterisk and Softphone are exactly the same. Thank you, -Steve

1 0

Combining lookup_branches() and t_(next|load)_contacts()
by Nathan Ward 09 Jun '18

09 Jun '18

Hi, About a year ago I posted this, but haven’t had much chance to follow up on it: https://lists.kamailio.org/pipermail/sr-users/2017-March/096337.html I’ve been doing some work on this issue (we worked around it until now), and have reduced the config right down to see where things are going wonky. The scenario I have, is inbound calls (to let’s say “A@domain") which I have aliased to two or more different numbers (let’s say “B@domain” and “C@domain”). Both B and C are registering, but are not always registered. I resolve the aliases with alias_db_lookup() (in my sample below, I use seturi() and append_branch() to avoid any confusion about where things may be going weird), then look them up in the location table with lookup_branches(). I then need to do t_load_contacts() and t_next_contacts(), as in some call scenarios I add a serial fork before or after the main call (pre-call announcements etc.). Then, I call t_relay(). This works great most of the time - however, if for example B is registered, but C is not registered, lookup_branches() notes that B is "Not found in usrloc”, and then t_load_contacts() seems to not realise that this isn’t usable, and it attempts to do a DNS lookup for the domain and so on, which I don’t really want it to be doing. Should lookup_branches() be removing a branch it can’t resolve? I would expect it to behave like that, anyway. Can I configure it to do that, or do I need to manually loop through them or something to make sure they resolved, and remove the ones that didn’t? I’ve tried swapping it around, so I call lookup_branches() after t_next_contacts() but the behaviour is the same. I imagine I can’t be the only person trying to do this - how have others made it work? Config sample: seturi('sip:a@domain'); append_branch('sip:b@domain'); xlog("L_DBG", "1 ds: [$ds]\n"); lookup_branches('location'); xlog("L_DBG", "2 ds: [$ds]\n"); t_load_contacts(); xlog("L_DBG", "3 ds: [$ds]\n"); t_next_contacts(); xlog("L_DBG", "4 ds: [$ds]\n"); -- Nathan Ward

1 1

Peculiar Kamailio Asterisk behavior on outbound calls
by Wilkins, Steve 09 Jun '18

09 Jun '18

Hi All, Issue: when a Call is made through Kamailio and Asterisk. Asterisk uses incorrect Video RTP Payload Type when sending Video packets. I have a situation where I make a call from Device A to Device B and Device A is Registered in Kamailio. When Device A Calls Device B, Kamailio sends an 'INVITE' to Asterisk, Asterisk then 'INVITES' Device B. I get two-way Audio, the call stays connected, however, when Video packets are sent to Device B, the RTP Payload Type is incorrect. The port is correct, but just not the Payload Type. Here is where I think Kamailio is involved. In the first Invite from Kamailio to Asterisk, one of the offered Video codecs is '100 H264'; interesting enough, Device B wants to use '115 H264' and when Asterisk sends out Video packets, it is using '100' instead of '115', and of course I have no Video. I don't know if this is just a coincidence but it sure seems like that is where the issue may lie. Has anyone ever seen this behavior? The Asterisk teams does not think it's an Asterisk issue. Thank you, -Steve

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Randomizing t_load_contacts() with same q value?
by Daniel Tryba 08 Jun '18

08 Jun '18

I'm looking for a way to randomize the contacts for serial forking if they have the same q value (so instead of doing parallel forking to those targets). Without taking the q value into account I could do something with the defined contacts_avp before calling t_next_contacts(). But I have no idea how to randomize and serialize the branches with the same q value after calling t_next_contacts(). Any ideas?

3 2

Monitoring Dispatcher
by Tim Bowyer 08 Jun '18

08 Jun '18

Hi All, Currently trying to work out the best way to keep track of how many calls are being sent to any given dispatcher destination. I remember being able to do this years ago with the load balancer module in OpenSIPS: opensipsctl fifo lb_list. Dispatching algorithm '10' seems to be the best fit for my requirements, as I'm essentially wanting to send calls to a large quantity of FreeSwitch docker containers acting as fax receiving nodes. Equal weight, just filling them in some balanced fashion. Any hints or ideas would be much appreciated! Kind regards, Tim Bowyer

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