Re: [SR-Users] Kamailio - passing registration to another sip server

Am 25.08.2016 um 16:19 schrieb E. Schmidbauer: Hi Matt, Check out this post and config: https://blog.voipxswitch.com/2016/08/11/kamailio-and-freeswitch-on-the-same-server-with-nsq-and-jansson-rpc/ It demonstrates how to handle NAT & forward a REGISTER to FreeSWITCH (on localhost) Thanks, Emm

Re: [SR-Users] A question to you, SIP providers: what is your policy on PBXes calling themselves

Am 07.06.2015 um 05:06 schrieb Antonio Gómez Soto: Hello, This is question on PBX behavior, what is the right thing to do, and how do PBX's generally behave. If a user on a phone, dials a number, which happens to be configured on the same phone system (for example another tenant), there are two

Re: [SR-Users] IPSec supporting open source SIP Server

Hi! I think he refers to IMS where IPsec is used dynamically. With the first REGISTER the client and the server exchange data which can be used (together with the password) to calculate the IPsec security associations. Then the client and the SIP server communicate with the OS to setup the IPs

[SR-Users] rtpproxy 2.0

Hi! I just found out that there is a new rtpproxy release: http://www.rtpproxy.org/post/v2release/ Has anybody tested it and want to share some experiences? Or have people turned to rtpengine meanwhile? (I am still using rtpproxy 1.2 as I do not need new features). regards Klaus ___

Re: [SR-Users] log_prefix not working in reply_route

On 18.05.2015 15:41, Daniel-Constantin Mierla wrote: > Hello, > > On 18/05/15 14:57, Klaus Darilion wrote: >> Hi! >> >> Should log_prefix also work for xlog in reply routes? In my case it >> doesn't (Kamailio 4.2.2) > it should work for received sip r

[SR-Users] log_prefix not working in reply_route

Hi! Should log_prefix also work for xlog in reply routes? In my case it doesn't (Kamailio 4.2.2) Thanks Klaus ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/

Re: [SR-Users] Dialog module generates error since upgrade to 4.2.2

On 23/01/15 12:40, Jan Hazenberg wrote: >>>> I indeed see the same spiral messages in the logs as well: >>>> >>>> Jan 22 14:50:33 sip /usr/sbin/kamailio[14084]: DEBUG: dialog >>>> [dlg_handlers.c:831]: dlg_new_dialog(): Callid >>>> '035a

Re: [SR-Users] Dialog module generates error since upgrade to 4.2.2

be a spiraled request. regards Klaus On 22.01.2015 21:02, Jan Hazenberg wrote: > Klaus, > > Yes, that solves the problem here as well. > > Thanks, > > Jan > > Klaus Darilion schreef op 2015-01-22 16:16: >> Hi Jan! >> >> I replaced dlg_manage with

Re: [SR-Users] fix_nated_contact and IPv6

Sounds like a bug. Anyway, as fix_nated_contact is not standard conform, it is better to use handle_ruri_alias() and add_contact_alias() (see the Kamailio default config file). Maybe they handle also IPv6 correct. regards Klaus On 22.01.2015 16:26, Sebastian Damm wrote: > Hi, > > I'm trying to

Re: [SR-Users] Dialog module generates error since upgrade to 4.2.2

Hi Jan! I replaced dlg_manage with setflag(dialog flag). This way the dialog is created only when a transaction is created. It solved my problems. regards Klaus On 22.01.2015 14:47, Klaus Darilion wrote: > Nice, I have the same problem after upgrading from 4.1.7 to 4.2.2. > .

Re: [SR-Users] Dialog module generates error since upgrade to 4.2.2

Nice, I have the same problem after upgrading from 4.1.7 to 4.2.2. . in my case it seems the dialog callbacks are not executed for responses ... I am still debugging . On 22.01.2015 13:25, Jan Hazenberg wrote: > Hi All, > > I'm running into a issue with the dialog module since the up

[SR-Users] RFC: best way to create dialogs

Hi! I use the dialog module to count and limit concurrent calls per user. It worked fine with 4.1.7 but fails with 4.2.2. My config basically looks like: route{ ... dlg_manage() ... authentication (stateless replies + exit) ... t_on_reply() t_on_branch() ... t_relay() exit; }

[SR-Users] tsilo question

Hi! Currently tsilo allows to add branches in failure route or request route (e.g. on REGISTER). would be be possible to allow this function also in reply routes? E.g. if one branch replies with 3xx, to add a new branch to the CFWD target but still have the other branches ringing. regards Klaus

Re: [SR-Users] reREGISTER problems: contact is inserted instead updated

, Klaus Darilion wrote: > Hi! > > Kamailio 4.1.6 with default settings for usrloc and registrar modul and > db_mode=1 (write through). > > Fritzbox sends a reREGISTER and Kamailio tries to insert the contact > into the database, but this fails as this contact for this AoR alread

[SR-Users] reREGISTER problems: contact is inserted instead updated

Hi! Kamailio 4.1.6 with default settings for usrloc and registrar modul and db_mode=1 (write through). Fritzbox sends a reREGISTER and Kamailio tries to insert the contact into the database, but this fails as this contact for this AoR already exists. So I think there are 2 scenarios: Kamailio fa

Re: [SR-Users] uac_from_replace does not restore when changing display name only

Further more, "auto" mode with "restore_dlg=1" does automatically change the From-uri for in-dialog requests, but not the From-display name. Is this also intended? Thanks Klaus On 10.10.2014 16:02, Klaus Darilion wrote: > Hi! > > It seems that uac_replace_from

[SR-Users] uac_from_replace does not restore when changing display name only

Hi! It seems that uac_replace_from does restore the original URI only if the >From URI is changed. Only changing the Display-name, eg: uac_replace_from("$avp(pad)",""); does not add the RR-cookie, thus the original will not be restored (e.g. on responses). Is this on purpose or a bug? re

Re: [SR-Users] [sr-dev] RFC: updating default values

On 02.10.2014 10:48, Juha Heinanen wrote: > Daniel-Constantin Mierla writes: > >> One more came in my mind: >> >> - failure_reply_mode in tm set to 3 (now is 0) > > i have been using 3 there. IMO this makes sense ___ SIP Express Router (SER) and Kam

Re: [SR-Users] Relaying ACK to Asterisk

Something is going wrong here: On 24.09.2014 18:41, Igor Potjevlesch wrote: > DEBUG: rr [loose.c:90]: is_preloaded(): is_preloaded: No That's correct. The ACK is not pre-loaded (with a route set). Checking first local URI (either alias= or listen= statement) > DEBUG: [socket_info.c:583]: grep_s

Re: [SR-Users] Relaying ACK to Asterisk

ion URI: "sip:1...@sip.fqdn.tld" > Algorithm: MD5 > Digest Authentication Response: > "55ca32ecf5ab7b32895fd2e545d7033c" > Contact: 22 :5060> > SIP Display info: 22 > Contact URI: sip:ABCDEFGHIJ@:5060 &g

Re: [SR-Users] Relaying ACK to Asterisk

dump the whole ACK packet received by Kamailio, and the packet looped by Kamailio regards Klaus On 20.08.2014 18:38, Igor Potjevlesch wrote: > Hello, > > > > I’m having trouble with this scenario (Kamailio and Asterisk are working > on the same server, Asterisk listens on 4060 instead of 5060

[SR-Users] upgrade question to Kamailio 4, how to handle ruid in location/aliases

Hi! I have to upgrade a production server from 3.x to 4.x. 4.x added a unique constraint on ruid. I have plenty of entries in aliases and usrloc table. How do I update these entries to get the ruid field populated? Can I do an upgrade reusing the entries stored in the usrloc table or do I have t

Re: [SR-Users] [sr-dev] new module db_kazoo

Kazoo seems to be a product name of 2600Hz. If the module is generic, then I would suggest to name it amqp, which better describe what it does. regards Klaus On 09.09.2014 14:44, Daniel-Constantin Mierla wrote: > Hello, > > I see there are some new functions prefixed with kazoo_, using a > termi

Re: [SR-Users] Location of Encryption Algorithm in TLS Module

The TLS module uses OpenSSL libraries. You can force a dedicated algorithm with the standard OpenSSL methods. If you want to change an algorithm, you have to change OpenSSL. regards Klaus On 10.09.2014 08:09, aawaise wrote: > Hello, > > I have downloaded and unzipped kamailio package available o

Re: [SR-Users] [sr-dev] How to uniquely identify SIP WS / WSS endpoint

On 03.09.2014 03:09, Muhammad Shahzad wrote: > Thank you so much for your informative response. > > Yes the "peer" may be correct term in this sense as i am trying to > identify "devices" (SIP UAs or Proxy) that are directly connected to > Kamailio via SIP signalling (i.e. there is no other inte

Re: [SR-Users] Support for TLS server_name extension (aka SNI=server name indication)

should be brought back if we lost it. Maybe you can adapt the old > patch if it not something that complex and you have time to look at it. > Otherwise, any further details about what you had to do in the past > would help to add support for it again. > > Daniel > > On

Re: [SR-Users] [sr-dev] How to uniquely identify SIP WS / WSS endpoint

Not sure what you trying to do, but the Via header is for transactions. It may be different for every transaction. Thus, if you need transaction matching (requests to responses) then you are fine and should use purely the branch id. If you want to match messages from one transaction to messages fr

Re: [SR-Users] Use Kamailio as (SIP-TLS) Gateway to connect to (non TLS) fritz.box for fixed-line calls

On 29.08.2014 11:12, Lukas Wygasch wrote: > Is there a Module i could use to do this Job? (configurable through > Siremis e.g. Dispatcher List?) Kamailio is a proxy, thus it can not authenticate itself to some other SIP server. (Actually it can using the uac module but this is not nice). > How

Re: [SR-Users] Support for TLS server_name extension (aka SNI=server name indication)

Indeed, currently Kamailio does not support SNI (was dropped with ser merge) Klaus On 29.08.2014 16:11, Daniel-Constantin Mierla wrote: > Hello, > > starting with 3.0 we got the implementation from SER at that time (being > more flexible with config and later getting asynchronous support). > >

Re: [SR-Users] Late parallel forking

On 21.07.2014 14:59, Daniel-Constantin Mierla wrote: > Hello, > > you may get similar results using t_cancel_callid((): > - http://kamailio.org/docs/modules/stable/modules/tmx.html#idm8272 > > For each call you have to store the call-id, cseq and the target user > somehow (e.g., using htable).

Re: [SR-Users] Strange avp_check behavior

Answering myself: This is a "feature". Assigning AVPs multiple times add an additional AVP with the same name. To overwrite an existing AVP the following syntax is needed: $(avp(pattern)[*]) = . regards Klaus On 07.07.2014 12:50, Klaus Darilion wrote: > Hi! > > K

[SR-Users] Strange avp_check behavior

Hi! Kamailio 4.1.4. When using avp_check with fnmatch and AVP the result is always TRUE if it is true for 1 time. It only happens with $avp(), but not with $var(). Is this a bug or some special behavior of AVPs? xlog("L_ERR", "=== with $$avp(...)"); $avp(rpid) = "+499539110

Re: [SR-Users] many-to-one network topology hide with TOPOH

Seems like you should an B2BUA/SBC (take a look at sems) Am 05.06.2014 20:53, schrieb Maciej Marczyn'ski: Hi Everyone, recently I'm playing around with the topoh module but I couldn't find the functionality I'm looking for. I'd like to hide my network topology before the end-user but not only

Re: [SR-Users] check for the number of open branches in Reply

It may work catching 200 OK in reply_route, but catching >=300 in failure route. The failure route is executed only once for all branches (it chooses the most important response code) regards Klaus Am 04.06.2014 11:49, schrieb Sebastian Damm: Hi, I have a scenario where I want to send a cust

Re: [SR-Users] Installing Kamailio on Ubuntu

you have to configure APT: http://www.kamailio.org/wiki/packages/debs On 19.03.2014 10:26, Swetha Raj wrote: Hi, I am trying to install Kamailio on Ubuntu 12.04,but hitting the below issue. root@test:/usr/local# apt-get install kamailio kamailio-mysql-modules kamailio-tls-modules Reading

Re: [SR-Users] how to get value of route param?

On 10.03.2014 06:18, Juha Heinanen wrote: rr module has check_route_param(re) function that can be used to check if local route header has a param that matches re. however, there does not exist a function to find out what is the value of a given rr param. for example, if route header has param

Re: [SR-Users] How to encode domain in username for Asterisk REGISTER forwarding (was: How to configure Kamailio + Asterisk (on same server) to route between several disjoint networks?)

AFAIK Asterisk supports multiple domains, but it seems that is not used for registration. On 09.03.2014 00:58, Alex Villací­s Lasso wrote: The above is the non-encoding version. What is the best way to modify it to do a domain encoding in the username? Or, if a better solution exists, what is

Re: [SR-Users] [sr-dev] New official Debian and Ubuntu repository

Thanks to all involved On 03.03.2014 14:16, Victor Seva wrote: The new build system for Debian and Ubuntu packages is now in place. This service is kindly sponsored by SipWise [0] thanks to Andreas Granig [1]. Sipwise is providing the hosting and man power to create and manage this new system.

Re: [SR-Users] How to configure Kamailio + Asterisk (on same server) to route between several disjoint networks?

Am 27.02.2014 23:43, schrieb Alex Villací­s Lasso: Is this setup recognizable as an already-solved problem (minus the localhost trick)? How is it done correctly? I think your setup is too complex. If I didn't missed your requirements I think you can do it this way: From routing point of view,

Re: [SR-Users] Replacing an ACME Packet Net-Net SBC

For sure I would use Kamailio as an SBC, but nevertheless I see these sortcoming in Kamailio (I don't know hov other SBCs handle this) - config changes require a restart: most of the time this goes fast, but sometimes processes may fail to start (ports not freed by the OS, ...). Further, state

Re: [SR-Users] prevent relaying to myself

Am 26.02.2014 22:48, schrieb Henry Fernandes: Is there a way to prevent relaying requests to myself in route[RELAY]? To relyable prevent this (eg. the destination IP address (your server IP address) may be hidden behind others domain) you have to check the destination in the branch route. r

Re: [SR-Users] How to configure Kamailio + Asterisk (on same server) to route between several disjoint networks?

Puh, too many questions in one email. First, you should describe what you want to achieve. Eg. is there routing between the networks done by the server? E.g. can a clinet on 10.1.0.0/24 ping a client on 192.168.0.0/16? If yes, there is no need for Kamailio/Asterisk to listen on multiple interf

Re: [SR-Users] kamailio as sip router and rtp proxy without reg on kamailio

Read the default configuration to understand how routing within Kamailio works. Regarding REGISTER: just remove all the REGISTER handling (and PUBLISH, SUBSCRIBE and NOTIFY) in the config, then the REGISTERs will be routed like all other messages. Then take a look at the PSTN gateway feature

Re: [SR-Users] NAT helper frustrations with multiple kamailios

On 05.02.2014 11:20, Daniel Tryba wrote: The problem I have with add_contact_alias/handle_ruri_alias is that any kamailio in the path will parse these hints whether or not it is actually ment for that kamailio. handle_ruri_alias should only be used by the last proxy in the chain, thus by the

Re: [SR-Users] Change of source ip address

I am a bit confused. I suspect with SBC you mean the Kamailio proxy? On 05.02.2014 14:54, Diego Alejandro Ozuna Escalada wrote: When a UA sends an INVITE to the SBC, the responses (back) to UA are being sent with the correct external source IP (X.X.248.194) but if there are retransmissions of t

Re: [SR-Users] websocket relay

> listen=tcp:10.2.3.4:5080 You only have a single socket: TCP. If you want to use websocket over TLS you need at least also: listen=tls:10.2.3.4:5090 (or whatever port you like, eg. 443) Further, if you talk to Asterisk with UDP, you also need a "udp" listen statement. regards Klaus __

Re: [SR-Users] Asterisk re-INVITE race condition, error 500.

Asterisk's transcation layer is quite buggy - so it may also be that the reINVITE with Cseq 103 is a retransmission of a previous transaction (which was not stopped correctly). regards Klaus On 04.02.2014 08:52, dotnetdub wrote: Hi Olle, Just a quick update.. I've gone through this in detai

Re: [SR-Users] Kamailio behind NAT

properly but no audio flows between phones. Now I am in the process of trying to locate where is the problem by comparing how both files handle the NAT support. Thank you - Original Message - From: Klaus Darilion Sent: 01/23/14 08:12 AM To: Kamailio (SER) - Users Mailing List Subject: Re

Re: [SR-Users] File transfering problem with RTPproxy

I have not used MSRP yet. But I guess you can detect MSRP by inspecting the SDP. USe the functions from textops module or sdpops module to check if the SDP is a normal call setup or an MSRP session. regards Klaus On 29.01.2014 13:13, Ravi wrote: Dear Klaus, Thank you for the reply, Ya i wil

Re: [SR-Users] kamailio with mediaproxy-ng, 488 Not Acceptable Here

iguration could you please send me an example of configuration that should work with websockets, rtpproxy-ng->mediaproxy-ng in order to remove one possible cause? Thank you. Best regards, Mihai M On Wed, Jan 29, 2014 at 1:31 PM, Klaus Darilion mailto:klaus.mailingli...@pernau.at>> w

Re: [SR-Users] File transfering problem with RTPproxy

For file transfer you need an MSRP relay. AFAIK rtpproxy can not handle MSRP. See the MRSP module http://kamailio.org/docs/modules/4.1.x/modules/msrp.html regards Klaus On 29.01.2014 09:44, Wingsravi R wrote: Dear Daniel & Kamailio'ns I am working on File transferring feature between two SI

Re: [SR-Users] kamailio with mediaproxy-ng, 488 Not Acceptable Here

Hi! The problem is different SDP formats between normal SIP clients/gateways, and WebRTC clients. You have to instruct mediaproxy-ng to rewrite the SDP and do the conversion (encryption, ...). So either the rtpproxy_ng calls lack the commands for the "gatewaying", or the webrtc clients use

Re: [SR-Users] Kamailio behind NAT

On 23.01.2014 10:29, John Smith wrote: Hello Klaus, I had already two sockets bound each to two independent physical interfaces. I have added the force_send_socket at each rtpproxy Just for clarity: force_send_socket is for near_end NAT traversal of the SIP signaling, whereas manage_rtppr

Re: [SR-Users] Kamailio behind NAT

Am 21.01.2014 17:33, schrieb John Smith: The next test has been to comment out the rtpproxy_manage at NATMANAGE function and to put it both at route[RELAY] and onreply(route) following your post in this list from January 2013:http://lists.sip-router.org/pipermail/sr-users/2013-January/076254.h

Re: [SR-Users] Kamailio behind NAT

t is the best way to proceed? To check if the IP is from the outside and then rewrite via rtpproxy_offer in the NATMANAGE block? Thank you - Original Message - From: Klaus Darilion Sent: 01/21/14 05:25 AM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio behind NAT

Re: [SR-Users] Kamailio behind NAT

Actually, it should work without any NAT traversal done in Asterisk, if Asterisk communicates never direct with the phones, but only via Kamailio and rtpproxy. In this case, Asterisk can use private IP addresses. All the near-end NAT traversal can be done in Kamailio. regards Klaus On 21.01.2

Re: [SR-Users] Kamailio behind NAT

On 21.01.2014 13:24, John Smith wrote: I might be making wrong assumptions regarding this traffic flow. Is that correct? That depends on your policy. It is up to you to define how RTP should be routed. There are basically 2 choices: a) RTP from clients is handled by rtpproxy: phone1 <-na

Re: [SR-Users] Kamailio behind NAT

On 21.01.2014 12:27, Fred Posner wrote: With a patched version of rtpproxy you can advertise your private ip. http://www.fredposner.com/voip/1457/kamailio-behind-nat/ Aha, nice. Haven't known of this one. I always specified the "adverstised IP address" when calling manage_rtpproxy(). That

Re: [SR-Users] SIP Update multi-homed

On 20.01.2014 17:48, Keith wrote: Hi, I have a kamailio server which is responding to SIP updates but sending from the wrong IP. I have multiple IPs in the same subnet on the same NIC. Is there anyway to say send update back out of the interface it was received on? I think that should be don

Re: [SR-Users] Replicate registration

On 19.01.2014 14:05, Volkan Oransoy wrote: I am trying to setup an active-active pair of sip proxies. I have connected two kamailio boxes to the same PostgreSQL database and my usrloc db_mode is 3. I can see the registration data in the database but "ul show" outputs of two device is not same.

Re: [SR-Users] question about nat_uac_test

fix_nated_register(). Do not use fix_nated_contact(). Further, apply NAT traversal only if the clients are directly "connected" to your proxy (not if there are other devices inbetween). On Tue, Jan 7, 2014 at 5:01 AM, Klaus Darilion mailto:klaus.mailingli...@pernau.at>>

Re: [SR-Users] question about nat_uac_test

On 03.01.2014 16:59, Brian Davis wrote: REGISTER sip:test1.test.com:5060 SIP/2.0 Via: SIP/2.0/UDP 96.xxx.xxx.xxx:33745;rport;branch=z9hG4bKf5s1p`n3TRv5TZx5RXy.RVv+JPz8Nat*UX!8KRx4SRx Via: SIP/2.0/UDP 192.xx.xxx.xxx:33745;branch=z9hG4bKeb263246c44095f072d8167dd0c7

Re: [SR-Users] SIP Security Architectural Question to Use RTP/Media Proxy or Not?

On 02.01.2014 17:00, Jr Richardson wrote: Would it be prudent to open UDP media ports from Internet to PBX's on a case-by-case basis, basically white listing media streams or is there any attack vulnerability with UDP in the media port range or should I open up media port range to all PBX's and

Re: [SR-Users] kamailio presence server becomes unresponsive

FYI: In master there is a nice way to get all the BTs: utils/kamctl: new command 'trap' - useful to get a full bt dump of all kamailio processes - handy in dead-lock investigatigations regards Klaus On 20.12.2013 19:06, Daniel-Constantin Mierla wrote: Hello, the bt is from custom timer proc

Re: [SR-Users] Caller not receiving RTP feed

If rtpproxy is behind NAT, you usually have to instruct Kamailio to write the public rtpproxy IP address into the SDP, instead of the local one (which is sent from rtpproxy to kamailio). e.g: rtpproxy_manage("co","your.public.ip.address"); regards Klaus On 20.12.2013 17:07, Benjamin Trent wro

Re: [SR-Users] force_send_socket

Try with a "static" assignment with force_send_socket(). If this works, try a static assignment with $fs. If this works, try the dynamic assignment with PVs. regards Klaus On 11.12.2013 11:32, Keith wrote: Thanks for the info guys, unfortunately it's not sending the from ip address properly n

Re: [SR-Users] IPv4, IPv6, RTPProxy and Kamailio

On 03.12.2013 14:23, Mark Zeman wrote: Hello all, The subject says most of it, I think. We set up our Kamailio and RTPProxy according to http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6 with the addition of an alias (siplab.ch), and the DNS to go with it, as well as TLS and SRTP. However

Re: [SR-Users] Kamailio for route traffic only

That's quite easy - that's a typical load-balancer setup. Just store the mapping for example in a DB and then use the sqlops module to query the DB and get the respective IP address of the user. But before you add this "routing logic" I would recommend to add Kamailio with a static forwarding

Re: [SR-Users] voice prompts / early media and kamailio

I think it is a conceptual question indeed. You "abuse" the 403 error in some table (actually "missed_calls" is for missed calls, not for rejected calls) to log/account a rejected call. Make it more explicit. If you want to track rejected calls, make a dedicated table and insert an record into

Re: [SR-Users] append_hf to CANCEL

A hack would be to loop the CANCEL to Kamailio again and forward it then stateless. When forwarding the CANCEL stateless, you can add headers. regards Klaus Am 20.11.2013 15:19, schrieb Grant Bagdasarian: I see, so there is no way to append a header to the CANCEL created by Kamailio? I tried t

Re: [SR-Users] Kamailio sending HTTP request to another app

On 18.11.2013 15:45, Alex Balashov wrote: http://kamailio.org/docs/modules/4.1.x/modules/tmx.html#idp15326008 Does it only suspend the transaction, but not the script processing? Is there somewhere a more complete example how to do some async stuff meanwhile and then resume the transaction

Re: [SR-Users] kamailio tls.reload core dump

patch was done on 4.0.3. Thanks, On 11/15/2013 3:59 AM, Klaus Darilion wrote: Hi Ding Ma! It would be great if you can provide the patch at the tracker. https://sip-router.org/tracker/ regards Klaus On 25.10.2013 01:54, Ding Ma wrote: Is this the right way to build without optimization? make

Re: [SR-Users] kamailio tls.reload core dump

ump hasn't happened even if we reload tls every 5 mins when there are some active TLS connections. Can we make these fixes into kamailio code base? What's the process to submit changes for review? Thanks, Ding On 10/24/2013 03:18 AM, Klaus Darilion wrote: You should build Kamailio w

Re: [SR-Users] #TM: timing problem in serial forking (INVITE vs. CANCEL)

I think it would be nice if the CANCELs are sent before the INVITE. But this will never ensure the order how they are received at the client side. E.g. there can be packet loss which drops the CANCEL but not the INVITE, or with load balancing the INVITE can overtake the CANCEL. And if the clien

Re: [SR-Users] media between 2 clients behind separate NAT

On 25.10.2013 14:19, Vassilis Radis wrote: Hello, When I have 2 clients using a kamailio proxy, and both of the clients are behind their own NAT, then my only options for relaying media between them is using some kind of intermediate rtp proxy or STUN etc? STUN is just a method for a client t

Re: [SR-Users] kamailio tls.reload core dump

work. The core dump hasn't happened even if we reload tls every 5 mins when there are some active TLS connections. Can we make these fixes into kamailio code base? What's the process to submit changes for review? Thanks, Ding On 10/24/2013 03:18 AM, Klaus Darilion wrote: You should bu

Re: [SR-Users] kamailio tls.reload core dump

You should build Kamailio without optimizations. "" does not bring much information. regards Klaus On 23.10.2013 21:48, Ding Ma wrote: Hi, all This is related to the previous tls.reload not safe email chain. Now we have a detailed gdb output that shows the stack trace of the core dump. Please

Re: [SR-Users] Monitoring events (presence)

On 22.10.2013 08:59, Juha Heinanen wrote: Klaus Darilion writes: I think a cool feature would be a 'wildcard' subscription, e.g.: SUBSCRIBE sip:*@mydomain.com to receive all events of mydomain.com, or SUBSCRIBE *@*. yes, cool feature for nsa. :-) They don't need this

Re: [SR-Users] AWS LOOP detected

On 21.10.2013 22:47, julian arsanches wrote: for now i reply with 488 wich tell me something is bad but i thought that kamailio will know when a message is send from itself to itself. Sending a message to itself is a legal use case (it is called "spiraling"), thus there is no such automatic

Re: [SR-Users] Monitoring events (presence)

OF course you could just SUBSCRIBE to get NOTIFYs. But then you would need to subscribe to all users (e.g. subscribe a user whenever there is a new registration). I think a cool feature would be a 'wildcard' subscription, e.g.: SUBSCRIBE sip:*@mydomain.com to receive all events of mydomain.com,

Re: [SR-Users] AWS LOOP detected

//www.kamailio.org/wiki/cookbooks/4.0.x/core#listen It simplifies the config by not needing to use alias parameter, advertised_address/set_addvertised_address() and record_route_preset().> Cheers, Daniel On 10/18/13 8:23 AM, Klaus Darilion wrote: / />/ />/ On 17.10.2013 17:31, julian arsanches wrote:

Re: [SR-Users] syntax error and invalid arguments when trying to start Kamailio

1. you should tell us which line exactly is line 356. 2. Maybe DBURL or MULTIDOMAIN are not defined. regards Klaus On 19.10.2013 23:27, Cory Sanders wrote: Sorry if this is a duplicate email. First one bounced. "I am having trouble starting Kamailio. Please see below for the error output an

Re: [SR-Users] why kamcmd tls.reload is not safe

I remember that long time ago there was an email discussing the problem in details. MAybe it was on one of the old mailing lists (ser, openser). IIRC the feature and the detailed discussion way by Jan Janak. Maybe this helps you to refine your Google search. regards Klaus On 19.10.2013 21:33

Re: [SR-Users] Record-route in dialog requests

Just as a note: If you need to record_route for in-dialog requests then the clients are buggy and you should report the issue. regards Klaus On 18.10.2013 19:46, Spencer Thomason wrote: Thanks Daniel, That did the trick! BR, Spencer On Oct 18, 2013, at 10:32 AM, Daniel-Constantin Mierla

Re: [SR-Users] Loop detected on aws setup

On 17.10.2013 17:31, julian arsanches wrote: Hi all, before hand thanks for all the support received on this channel. I have an issue with an installed server on a aws instance which is giving me routing loops, my setup is simple, i have alias set for both ips internal and external and the loo

Re: [SR-Users] Kamilio and AWS Route 53 latency regions

Hi Coy! On 16.10.2013 14:29, Coy Cardwell wrote: Thanks. By "as long as IP connectivity between the outbound proxies and registrars is not filtered", what exactly must not be filtered? The proxies and their 'local' registrars will be in the same private IP cloud. Then it should be fine. Are

Re: [SR-Users] Kamilio and AWS Route 53 latency regions

On 15.10.2013 13:21, Coy Cardwell wrote: Thanks. I am using DB only mode. There will be a number of servers in the end, so i will have to look further into the issue I guess, since assumptions were made about how Kamailio works. From what I can tell, I think I will have to implement the outbou

Re: [SR-Users] Kamilio and AWS Route 53 latency regions

Hi! First, if 2 servers share the same DB, then userloc must be switched to DB-only mode: http://kamailio.org/docs/modules/4.0.x/modules/usrloc.html#idp16939424 But this leads you to another problem. As Fred already mentioned, SIP clients (or the NAT of the user) often refuse messages which a

Re: [SR-Users] SIP Trunks

On 14.10.2013 14:57, Keith wrote: Hi, Klaus, thank you for pointing me in the right direction with SIP trunks, got it working so thanks! Basically I did exactly what you said: - Dialled number - Match that number to a registered user (had to create a new table for that) - Lookup user - Replac

Re: [SR-Users] SIP Trunks

Am 09.10.2013 17:56, schrieb Keith: Hi, Can anyone point me in the right direction for setting up SIP trunks? Whenever I send a call to a registered user on a trunk it just sends to destination s@x.x.x.x. Is there anyway to say "these extensions are location at this destination IP and port". F

Re: [SR-Users] Scripting - Adding custom code

To dynamically add routes you do not need to understand how the code works. Use one of the routing modules (lcr, drouting, prefix_route, carrierroute ). Ususally they are configured via a DB backend and perform a DB lookup for every call, or cache the routing table and the routing table can

[SR-Users] keep_hf("") meaning

Hi! keep_hf("") keeps all headers, as every header matches the regexp "". IMO an empty regexp should remove all headers (except the mandatory ones). Otherwise I have to use something like keep_hf("hope-this-header-never-exists") regards Klaus ___

Re: [SR-Users] Apply changes made to sip reply in onreply_route

Am 08.10.2013 12:14, schrieb Grant Bagdasarian: Hello, I've setup two Kamailio machines, one which does all the processing and the second one which always replies with a 500 Server Internal Error, to test my Dispatcher fail-over. When routing a call, the call is always routed to the second

Re: [SR-Users] append_hf to reply generated by kamailio

Use append_to_reply(txt) before sl_send_reply(): http://kamailio.org/docs/modules/4.1.x/modules/textops.html#idp17040608 regards Klaus On 03.10.2013 11:38, Grant Bagdasarian wrote: Hello, Is it possible to append a new header to a reply generated by Kamailio and also have it present when dupli

Re: [SR-Users] Replace header value in failure_route

Do you need the regex? Does remove/append_hf work? eg: remove_hf("X-Dispatcher"); append_hf("X-Dispatcher: $(avp(dsattrs){param.value,dispatcher})"); regards klaus On 02.10.2013 15:55, Grant Bagdasarian wrote: Hello, I’m trying to replace the value of a custom header in the failure_route, u

Re: [SR-Users] Websockets Keep-Alive

look at the "keepalive_.*" modparams for the websocket module. The TCP connection timeout should be set to something a little greater than the WebSocket ping interval is set to. Regards, Peter On 26 September 2013 12:37, Juha Heinanen mailto:j...@tutpro.com>> wrote: Klaus

[SR-Users] Websockets Keep-Alive

Hi! Question to the experts: Is keep-alive for the Websockets TCP connection automatically done by the Websockets Layer (client or server), or do I have to do it manually (nathelper pinging). Thanks Klaus ___ SIP Express Router (SER) and Kamailio (

Re: [SR-Users] TLS versions and ciphers questions.

Kamailio uses OpenSSL. Thus, it mainly supports what your OpenSSL version supports. This means, Kamailio can use all TLS versions and ciphers that your libssl supports. You can configure Kamailio to use certain ciphers, see: http://kamailio.org/docs/modules/4.0.x/modules/tls.html#cipher_list Y

Re: [SR-Users] kamailio configuration for pass-thru proxy and username rewrite

If you want to hide your upstream providers from your customers, you should use a B2BUA (sems, Asterisk). Probably the most simple setup would be: customers -- Kamailio -- Asterisk -- PSTN Provider Kamailio is used for Registrations, NAT traversal, Authentication ... Calls to the

Re: [SR-Users] Kamailio does not send reply to source port

On 19.08.2013 13:04, Alex Balashov wrote: On 08/19/2013 07:00 AM, Mino Haluz wrote: (palmface). I am always forgetting about this, it should be called where initial request is processed or also for sequential requests? Initial. Sequential requests and their replies are sent according to a

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