sr-users October 2009

sr-users@lists.kamailio.org

63 participants
105 discussions

[Serusers] question regarding sip protocol

by Sabatella, Michael

Hi, I am new to the sip protocol. I have been doing packet traces to understand how the protocol works. I am using the ser sip proxy version 0.9.6. I notice that once my phone registers with a proxy I get a 200 ok. Every 20 seconds after that the client sends a packet to the proxy. It is an Request: OPTIONS packet. My proxy responds with a Status:404 Not found. I tried to do some research on this but could not find anything useful. Does this response of a 404 matter? How can you fix this? I have read that the client does this as an application level ping to make sure the proxy is up. If I have 100s of people registering that is a lot of extra traffic. Is there a way to stop this options query or have the proxy respond properly? Thanks Michael Network Services Engineer ________________________________ DISCLAIMER: This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unintended recipients are prohibited from taking action on the basis of information in this e-mail.E-mail messages may contain computer viruses or other defects, may not be accurately replicated on other systems, or may be intercepted, deleted or interfered with without the knowledge of the sender or the intended recipient. If you are not comfortable with the risks associated with e-mail messages, you may decide not to use e-mail to communicate with IPC. IPC reserves the right, to the extent and under circumstances permitted by applicable law, to retain, monitor and intercept e-mail messages to and from its systems.

14 years, 6 months

[Kamailio-Users] preparing Kamailio 1.5.3

by Daniel-Constantin Mierla

Hello, Kamailio 1.5.3 will be released today. Please commit anything you have for branch 1.5 on Sourceforge.net svn in the next two hours. Afterwards we will start preparing the release. Cheers, Daniel

14 years, 6 months

[Kamailio-Users] max channel capacity

by Jeff Brower

All- On a reasonably fast server, say quad-core, what is the approx maximum number of G711 IP calls when using Kamailio and rtpproxy? What if rtpproxy runs on a second server? No echo can, no packet concealment, etc... just G711. -Jeff

14 years, 6 months

Re: [Kamailio-Users] use case for settlement free peering

by Daniel-Constantin Mierla

welcome -- cc-ing mailing list is recommended all the time -- help people to get the conclusion of discussions. On 20.10.2009 20:04 Uhr, A G wrote: > Great. Thanks for the feedback! > > On Mon, Oct 19, 2009 at 3:47 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com> wrote: > >> Hello, >> >> On 19.10.2009 20:18 Uhr, A G wrote: >> >>> Greetings: >>> >>> I'm looking for advice on a project/proof of concept I'm working on. >>> I would like to create a settlement-free peering fabric for voice >>> traffic between and among some peer institutions in my area. Because >>> this is more of a side-project for cost cutting measure, I'm primarily >>> looking at open source software, though commercial product >>> recommendations would be helpful as well. >>> >>> The organizations I would like to connect have their own PBXs with >>> large blocks of numbers (whole NPA-NXXs), with no number portability >>> in or out. I imagine both at the individual PBXs and peering fabric, >>> the number routing would be static. To put another way, we would >>> manually configure which connections the block of telephone numbers is >>> reachable at. >>> >>> Here is the required ASCII art diagram :) >>> >>> >>> +-------+ >>> | PBX | >>> +-------+ >>> | >>> +-------+ >>> | SBC | >>> +-------+ >>> | >>> | >>> +---+ +---+ .--------. +---+ +---+ >>> | P | | S | / \ | S | | P | >>> | B |--| B |------ ( ???? )----------| B |--| B | >>> | X | | C | \ / | C | | X | >>> +---+ +---+ `---------' +---+ +---+ >>> | >>> | >>> +-------+ >>> | SBC | >>> +-------+ >>> | >>> +-------+ >>> | PBX | >>> +-------+ >>> >>> >>> >>> >>> For scalability reasons, a full mesh of connections between and among >>> the SBCs is not an attractive option. >>> >>> Here's what I think I need: >>> Basic SIP routing >>> TCP, TLS, and UDP support >>> >>> What would be nice to have: >>> IPv6 >>> CDR >>> >>> What is probably not needed: >>> User agent client registration, presence, IM, voice mail >>> >>> I see there are several different open source voice projects. >>> Do you think this is an appropriate use for Kamailio? >>> >>> >> Kamailio can be used in such scenario. It has a lot of features to help you >> routing calls -- see modules such as lcr, carrierroute, dialplan. >> >> If you do heavy tls/tcp, then upcoming Kamailio 3.0 will have lot of >> improvements in this areas. IPv6 is supported as well for core routing. You >> can generate CDRs using acc module and some stored procedures -- see for >> example: >> http://siremis.asipto.com/install-accounting/ >> >> Cheers, >> Daniel >> >> >>> I'm seeking comments on what you would use for this situation. >>> Are there any existing projects along these lines? >>> Is there one project that is better than another for this application? >>> >>> >>> >> -- >> Daniel-Constantin Mierla >> * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin >> * http://www.asipto.com/index.php/sip-router-masterclass/ >> >> >> -- Daniel-Constantin Mierla * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin * http://www.asipto.com/index.php/sip-router-masterclass/

14 years, 6 months

Re: [Kamailio-Users] Problem with LCR "ping"

by Alexandr Dubovikov

Hi Ricardo, you use ping_interval 10 seconds. For 126 gateways it's extrem short . Can you set this param up to 60 seconds and check again ? Wbr, Alexandr P.S. will be nice to see the result from "ngrep -W byline -t -l port 5060", if it possible :-) > > -----Original Message----- > From: users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] On Behalf Of Ricardo Martinez > Sent: Thursday, October 15, 2009 4:48 PM > To: users(a)lists.kamailio.org > Subject: [Kamailio-Users] Problem with LCR "ping" > > Hello list. > > I have a question/problem regarding the LCR module. I'm using Kamailio version 1.5.2. The LCR tables have : > > > > gw table : 126 records > > lrc tables : 120 records. > > > > I'm using the "ping" (ping = 1) feature in all my gateways, part of the cfg file is as follows : > > > > modparam("lcr","db_url","mysql://openserro:openserro@localhost/openser") > > modparam("tm", "fr_inv_timer_avp", "$avp(i:704)") > > modparam("lcr", "gw_uri_avp", "$avp(i:709)") > > modparam("^auth$|lcr", "rpid_avp", "$avp(i:302)") > > modparam("lcr", "ruri_user_avp", "$avp(i:500)") > > modparam("lcr", "flags_avp", "$avp(i:712)") > > modparam("lcr", "fetch_rows", 3000) > > modparam("lcr", "ping_interval", 10) > > modparam("lcr", "ping_from", "sip:pinger@mydomain.net") > > modparam("lcr", "ping_method", "OPTIONS") > > > > From time to time I can see 3 o 4 gateways marked as down (with the command kamctl lcr dump). I'm not manually monitoring the gateways every minute so is possible that in a particular moment the gateways could be unreachable, but when I run the command "kamctl lcr dump" the gateways are on-line, that's for sure. > > So I'm not understanding why LCR keep marking the gateways as down, even if they are answering to the OPTIONS request. Could this be a bug in the LCR module? Or maybe I'm missing something?. The silly part of all this is the gateways marked as down are always the same. Could this be related to a memory issue?? > > > > Can someone guide me here?. > > > > Thanks in advance, > > Regards, > > Ricardo Martinez.- > > > > -- Alexandr Dubovikov * baron@iRC RusNet * mailto:shurik@start4.info AD1-UANIC * ICQ: 122351182 * http://www.start4.info

14 years, 6 months

[Serusers] SER Staring Problem

by sourav.das1＠wipro.com

Hi, I am trying to install the SER through the following steps. Download and extract the package : * Download the source package containing mysql support for SER from: http://ftp.iptel.org/pub/ser/latest/ <http://ftp.iptel.org/pub/ser/latest/> * Save the downloaded package at /usr/src folder * Extract the package * Change directory to the extracted package Make and install Make the program and the modules and then install the program in the folder /usr/local. # make && make modules # make prefix=/usr/local install After installing the SER on the Linux system I am trying to start SER through the following command. /usr/local/sbin/serctl start While executing the above command I am getting the following error. Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed Please let me know how to resolve this or something is missing. Regards, Sourav Please do not print this email unless it is absolutely necessary. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

14 years, 6 months

[Kamailio-Users] kamailio 1.5.3 scheduled for next wednesday

by Henning Westerholt

Hi all, as it has been three month since the last minor release of the kamailio 1.5 branch i'd suggest that we do a 1.5.3 release soon. I want to propose next wednesday the 14.10.2009 as date for the release. So if you have any pending patches in the trunk or in your internal repositories that you want to like to see in this release, please commit them until 12:00 UTC this day. Thank you, Henning -- Henning Westerholt - Development Consumer Products / Consumer Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

14 years, 6 months

[Kamailio-Users] use case for settlement free peering

by A G

Greetings: I'm looking for advice on a project/proof of concept I'm working on. I would like to create a settlement-free peering fabric for voice traffic between and among some peer institutions in my area. Because this is more of a side-project for cost cutting measure, I'm primarily looking at open source software, though commercial product recommendations would be helpful as well. The organizations I would like to connect have their own PBXs with large blocks of numbers (whole NPA-NXXs), with no number portability in or out. I imagine both at the individual PBXs and peering fabric, the number routing would be static. To put another way, we would manually configure which connections the block of telephone numbers is reachable at. Here is the required ASCII art diagram :) +-------+ | PBX | +-------+ | +-------+ | SBC | +-------+ | | +---+ +---+ .--------. +---+ +---+ | P | | S | / \ | S | | P | | B |--| B |------ ( ???? )----------| B |--| B | | X | | C | \ / | C | | X | +---+ +---+ `---------' +---+ +---+ | | +-------+ | SBC | +-------+ | +-------+ | PBX | +-------+ For scalability reasons, a full mesh of connections between and among the SBCs is not an attractive option. Here's what I think I need: Basic SIP routing TCP, TLS, and UDP support What would be nice to have: IPv6 CDR What is probably not needed: User agent client registration, presence, IM, voice mail I see there are several different open source voice projects. Do you think this is an appropriate use for Kamailio? I'm seeking comments on what you would use for this situation. Are there any existing projects along these lines? Is there one project that is better than another for this application? Thank you

14 years, 6 months

Presence gateway between SIP and XMPP

by afshin afzali

Hi Guys, You stated : "*This module is a gateway for presence between SIP and XMPP. * * It translates one format into another and uses xmpp, pua and presence modules to manage the transmition of presence state information*" My first question is that does it enable exchange of presence data between SIP and XMPP in two ways ? If yes how I could use the pua_xmpp_notify and pua_xmpp_req_winfo APIs to do so ? Regards, -- Afshin

14 years, 6 months

[Kamailio-Users] Odd behavior svn rev 2:5901M

by brandon＠cryy.com

Hello, I am noticing very rarely however it is happening where kamailio will route to the completely wrong sip uri. In this scenario I see it routing to one of our voicemail applications, thus placing voicemails into the wrong mailbox identifier. What is odd in this particular scenario I was able to capture is that it attempts to deliver for the same call-id to the right location and wrong location within a very very very small time frame causing our voicemail application to pick the wrong branch. What is even odder is this is a recovery from a 487 request cancelled so it is coming from failure route, appending a branch, so there should only be a single branch to voicemail. I clearly see in the logs after setting the ruri that it is correct, I have no idea where it is getting this ghost mailbox uri identifier, it is almost as some how the call got mixed up with a completely different call. Just throwing this out here to see if anyone else has experienced this. I will attach supporting evidence to this email for this scenario shortly. The last thing to mention for now is that it is definitely random and not sure-proof reproducible. Thank you for your time and input ahead of time. Look forward to hearing back from you all! Have a great weekend. Sent from my Verizon Wireless BlackBerry

14 years, 7 months

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sr-users October 2009