- sr-users - kamailio.org

Kamailio -PSTN Gateway
by Zaka Ul Isam 30 Sep '14

30 Sep '14

Hello Folks: Please help with above, I have browsed and tried various suggestions on this list without much luck! I think problem can be reduced to three questions ;) 1: Apart from DEFINE WITH PSTN directive, do I need certain modules to be compiled? (DIALPLAN, CARRIERROUTE, LCR aso) If yes, then we ought to put up a list of modules required for each task/ activity..... 2: How Kamailio core decides the call routing? i.e: based on dialed digits as specified in dialplan . # - update the … [View More]condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00355)[0][4-9]{3,20}$")) return; && #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "10.1.6" desc "PSTN GW Address" pstn.gw_port = "5080" desc "PSTN GW Port" #!endif 3: If Kamailio is listening on say wlan0 interface 10.42.0.1:5060 (Ubuntu HOT SPOT) can it route to PSTN Gateway/ Softswitch on P2P1 (Eth0) without Bridging? Put another way, does kernel route towards the specified interface or Kamailio is capable of routing based on active routing cache? Thanking you in advance, Zak ________________________________ Albtelecom & Eagle Mobile invite you to protect together the Environment. Please do not print the e-mail unless really necessary. Albtelecom & Eagle Mobile iu ftojnë të mbrojmë së bashku Mjedisin. Lutemi të mos e printoni këtë komunikim elektronik nëse nuk është vërtet i nevojshëm. ________________________________ This e-mail and any files transmitted with it are confidential and intended solely for the use of the addressee/s. If you are not the intended recipient you are hereby notified that any dissemination, forwarding, copying or use of any of the information is strictly prohibited. If you receive this e-mail in error, please notify the sender immediately and delete it! Albtelecom makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. The opinions expressed in this message may belong to sender alone and may not necessarily reflect the opinions of Albtelecom. Albtelecom shall bear no liability for any loss or damage caused by software or e-mail viruses. Ky mesazh dhe çdo informacion i transmetuar në përmbajtje te këtij mesazhi është konfidencial dhe është i destinuar vetëm për marrësin e destinuar. Nëse nuk jeni marrësi i destinuar, Ju bëjmë me dije se çdo përhapje, transmetim, kopjim apo përdorim i çdo informacioni është i ndaluar. Nëse e merrni këtë mesazh gabimisht, ju lutem kontaktoni urgjentisht nisësin e tij dhe fshijeni atë. Albtelecom nuk jep asnjë garanci për saktësinë apo plotësinë e informacionit në përmbajtje të këtij mesazhi dhe nuk mban asnjë përgjegjësi për informacionin e përmbajtur, transmetimin, marrjen, ruajtjen apo përdorimin e tij në çfarëdolloj mënyre. Mendimet e shprehura në këtë mesazh mund t'i përkasin vetë nisësit dhe nuk mund të reflektojnë domosdoshmërisht qëndrimet e Albtelecom. Albtelecom nuk do të mbaje asnjë përgjegjësi për humbje ose dëme te shkaktuara nga programet apo viruset. [View Less]

2 2

kamailio call limit
by Yuriy Gorlichenko 30 Sep '14

30 Sep '14

Hello. I try to test with SIPp my stak of kamailio->asterisk. I run SIPp with 200 calls/sec and see only 68 at maximum active calls at server. When I set 500 calls/sec with limit 1000 I see 68 active connections again. So when I try test SIPp to asterisk without Kam i see wright maximum of active channels/calls. My question - does Kamailio have limits ffor calls? If no- help me to understand please what is going on with my server. Thanks.

3 4

Re: [SR-Users] R: Re: R: Re: RTPPROXY & BRANCH
by Richard Fuchs 29 Sep '14

29 Sep '14

On 09/29/14 14:51, Marino Mileti wrote: > Without rtpproxy: > > - A offers port a1,a2 (audio video) in INVITE to B,C (in case of no natted > client so no needs of rtpproxy) > - B offers port b1,b2 (183) > - C offers port c1,c2 (182). > - A starts to send audio/video RTP to B on port b1,b2 > - A starts to send audio/video RTP to C on port c1,c2 > > With rtpproxy: > > - A offers port a1,a2 (audio video) in INVITE to Kamailio.... > - Kamailio contact … [View More]

1 0

Re: [SR-Users] R: Re: R: Re: RTPPROXY & BRANCH
by Richard Fuchs 29 Sep '14

29 Sep '14

On 09/29/14 14:23, Marino Mileti wrote: > The problem isn't on 183s but on the multiple INVITE that Kamailio sends to > clients behind rtpengine. Rtpengine open new ports for answer but on INVITE > the rtpengine ports are the same...This happens because for all these > clients the callid is still the same..so for rtpengine there's no > difference...for this reason I can see early media only on one of the > receivers I still don't understand what difference an RTP proxy is … [View More]

2 1

Re: [SR-Users] rtpproxy & extra_id_pv
by Richard Fuchs 29 Sep '14

29 Sep '14

On 09/29/14 14:29, Marino Mileti wrote: > Wow! Do you have an example of how to do that? How I have to modify my > kamailio.conf in order to instructs rtpproxy to user from-tag & to-tag in > this way? You don't have to do anything, tags are already included in all the messages. cheers

1 0

Re: [SR-Users] rtpproxy & extra_id_pv
by Richard Fuchs 29 Sep '14

29 Sep '14

On 09/29/14 14:08, Marino Mileti wrote: > But with from-tag and To-tag it's possible to instruct rtpengine to generate > new couple of ports for each branch of a call? In the source code of > rtpengine it seems that it check only the callid parameter Yes it will. The call-id is only a vague umbrella under which the from/to tags are collected. All media parameters, ports etc are associated with the tags, and a single call can contain any number of different from/to tags, all with their … [View More]

2 1

R: Re: R: Re: RTPPROXY & BRANCH
by Marino Mileti 29 Sep '14

29 Sep '14

No no. The video will be sent by the caller user to all the callees. I'l try to explain better. My scenario is: - A make a call to a group... B & C are group member...so Kamailio is able to call them in parallel using alias.. - B & C receive the INVITEs & replies with two separate 183 with SDP (in SDP they specified where they are able to receive audio&video) - A receives two 183...& starts to send its RTP video stream to B & C (early media) - … [View More]

3 7

Re: [SR-Users] rtpproxy & extra_id_pv
by Richard Fuchs 29 Sep '14

29 Sep '14

On 09/26/14 16:57, Marino Mileti wrote: > Hello, > >> On Friday 26 September 2014 16:44:44 Marino Mileti wrote: >>> Hi guys, >>> I've seen that setting the parameter extra_id_pv, every branch should >>> be a different callid.. >>> How can i set this parameter? I've tried with : >>> modparam("rtpproxy", "extra_id_pv", "$avp(extra_id)") >>> >>> but in the INVITE message the callid is still the same for all >>> … [View More]

2 1

R: Re: RTPPROXY & BRANCH
by Marino Mileti 29 Sep '14

29 Sep '14

Because I've more than 1 client behind NAT (1,2,3 mobile phones) and I would like to reach all of them in parallel mode. I can't use for all of them same ports because all mobile clients have early media (the receive video media before they answer) So at the moment this scenario is not possible? Is there any possible turnaround? ----Messaggio originale---- Da: rfuchs(a)sipwise.com Data: 24-set-2014 1.15 A: <sr-users(a)lists.sip-router.org> Ogg: Re: [SR-Users] RTPPROXY &amp; … [View More]

3 3

rtpproxy_offer and fix_nated_sdp in one route
by Sebastian Damm 29 Sep '14

29 Sep '14

Hi, I switched from rtpproxy module to the rtpproxy-ng module lately, and noticed a strange behavior. In my branch route to the device, I have two statements: fix_nated_sdp("1"); rtpproxy_offer(); The first command appends a line with "direction:active" to the SDP. The second one puts the RTP proxy in the stream. This worked all well with the old rtpproxy module. But with the new rtpproxy-ng module, I get an empty line after the SDP body, just before the "direction:active" line in the SDP, … [View More]

3 4

event_route[core:receive-parse-error] is not executed 4.1.6
by Julia Boudniatsky 29 Sep '14

29 Sep '14

Hello, I try to use an event_route[core:receive-parse-error] { xlog("L_WARN", "Event-parse-error: $rm from $avp(inc_carrier)/n$mb/n"); } corelog=1 debug=0 kamailio 4.1.6 from source Wrong header "From" is simulated by SIPP. In log I receive only ERROR:<core> [parser/parse_from.c:79]: parse_from_header() ...... Thank you, Julia

1 0

Kamailio with Jitsi - Presence trouble
by Gaurav Kumar 29 Sep '14

29 Sep '14

Hello all!I am trying to setup a secure videoconferencing setup for my small office. After a lot of convincing, my bosses have allowed me to create a setup and have given me a live IP. I have used it on a Ubuntu 12.04 setup and want to setup a SIP server for very few users (less than 10, at most) to use it through Jitsi. I am trying to follow the tutorials available at:http://kb.asipto.com/kamailio:skypelikeserviceinlessthanonehour&https://… am able to log into Jitsi on independent machines,… [View More]

1 0

event_route[core:receive-parse-error] is not executed 4.1.6
by Julia Boudniatsky 29 Sep '14

29 Sep '14

Hello, I try to use an event_route[core:receive-parse-error] { xlog("L_WARN", "Parse-error: $rm from $avp(inc_carrier)/n$mb/n"); }

1 0

Does kamailio reloads the modules per sip message ?
by Rahul MathuR 29 Sep '14

29 Sep '14

Hello, We have observed a strange behavior in corex module that it gets loaded at every sip packet which arrives to kamailio. We put a static variable and saw that it gets re-initialized to 0 everytime any sip packet comes to it. Could you please tell me how to stop it and load it just once. Thanks in advance ! -- Warm Regds. MathuRahul

2 1

DBTEXT Engine & ALIAS_DB
by Marino Mileti 29 Sep '14

29 Sep '14

Hi to all, I've setup FS with dbtext module, and now I'm trying to make a alias_db_lookup(dbaliases) but it doesn't find any alias.Same conf with mysql db works great. Is it possible to have alias with dbtext engine? Marino

2 1

tm.t_uac_wait documentation?
by Juha Heinanen 29 Sep '14

29 Sep '14

i could not find any documentation about tm.t_uac_wait rpc command. is it documented somewhere? -- juha

2 3

tm.t_uac_wait does not use advertise address
by Juha Heinanen 29 Sep '14

29 Sep '14

based on more tests, it appears that tm.t_uac_wait does not use advertise address (if given on listen line) when it substitutes !! (SUBST_CHARs) in request headers. i'll open a bug report on it. -- juha

2 1

cnxcc serial fork
by Kelvin Chua 29 Sep '14

29 Sep '14

hi carlos, i have a serial forking scenario to several providers as failover. and i used cnxcc_set_max_credit to control the session credit/time. just wanted to make sure it does what i think it should do so i made a little experiment. the logic goes this way. 1. cnxcc_set_max_credit -> $100, cost per second -> $1 2. call will fail, try the next provider 3. cnxcc_set_max_credit -> $200, cost per second -> $2 what i am expecting to see is, if i do kamctl kamcmd cnxcc.… [View More]

2 3

rtpproxy & extra_id_pv
by Marino Mileti 26 Sep '14

26 Sep '14

Hi guys, I've seen that setting the parameter extra_id_pv, every branch should be hae different callid.. How can i set this parameter? I've tried with : modparam("rtpproxy", "extra_id_pv", "$avp(extra_id)") but in the INVITE message the callid is still the same for all branches. Any suggest?

2 2

Kamailio server crashing
by aawaise 26 Sep '14

26 Sep '14

Hello, I installed kamailio and replaced kamailio.cfg with a customized kamailio.cfg and replaced kamctlrc file also. And restarted the kamailio server. Server gets started ok and after a minute or two, I get the error, " '/usr/local/kamailio/sbin' doesnot belong to any package ". I am facing this problem for quite some time now. Any solution to this will be very helpful. Regards, Aawaise. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-server-crashing-… [View More]

2 1

Bash Code Injection and 'exec' module
by Seudin Kasumovic 25 Sep '14

25 Sep '14

Hi kamailio users, we are witnesses of new discovered bug in bash: Bash Code Injection Vulnerability via Specially Crafted Environment Variables (CVE-2014-6271) https://access.redhat.com/node/1200223 As exec module exports all SIP headers in environment so it's was easy to push bash command. There is attached simple kamailio test config file. With sipp we sent header to output 123 into file /tmp/123 like this: User-Agent: () { :;}; echo 123 > /tmp/123 Debug output from kamailio is: 5(… [View More]

2 4

Re: [SR-Users] Kamailio - Separated Servers
by Bruno Emer 24 Sep '14

24 Sep '14

Hello Guys! Good news about my tests here! I found some configurations relative to Outbound proxy and I get this working. Now, I can have two outbound proxies, one registrar and one database. I can register one user in the Outbound Proxy1 and one user in Outbound Proxy 2 and place a call between these users. But now I'm facing another problem! The ACK messages are not being delivered, so the session is being dropped after a few seconds. I think that the problem is in the Contact field. Trying … [View More]

3 6

Group call
by marino.mileti＠alice.it 24 Sep '14

24 Sep '14

Hi guys, is possible to make a group call with kamailio? Example...i dial 1000 and kamailio make a parallel forking to 1001,1002,1003? In case of yes...every group member (1001,1002,1003) will be resolved by "location"? Because some member is natted so RTPPROXY should be engaged Is there any examples? BRMarino

4 3

Re: [SR-Users] Sip Event Filter
by Raúl Alexis Betancor Santana 24 Sep '14

24 Sep '14

Hi, you are facing the problem the wrong way. You could do that easy on the UAC side. For example, with Asterisk as UAC, you only need to pass a 10s timeout as paran to the dial app. A sip-proxy its not for what you are trying to acomplish. What you are trying to do it's a UAC or B2BUA task. Best regards. El 24/09/2014 09:00, Delifisek Tux <delifisek.tux(a)gmail.com> escribió: > > Hello, > > Very newbie here. Even I may not know what I'm talking about. Please be patience, … [View More]

1 0

Sip Event Filter
by Delifisek Tux 24 Sep '14

24 Sep '14

Hello, Very newbie here. Even I may not know what I'm talking about. Please be patience, thank you :) I had auto dialer and I have problem with unreachable numbers. So I wish if I call busy, unavailable, ringing but not answering numbers, system will hangup before the voice mail active. After some googling I found a sip proxy may solve my problem. I setup kamailio and manage to work with an example edge proxy setup. Then what ? My wish is, after the number dialed, start to filter every … [View More]

1 0

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users