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[Serusers] SER Staring Problem
by sourav.das1＠wipro.com 20 Oct '09

20 Oct '09

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

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[Kamailio-Users] kamailio 1.5.3 scheduled for next wednesday
by Henning Westerholt 20 Oct '09

20 Oct '09

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

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[Kamailio-Users] use case for settlement free peering
by A G 19 Oct '09

19 Oct '09

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

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Presence gateway between SIP and XMPP
by afshin afzali 18 Oct '09

18 Oct '09

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

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[Kamailio-Users] Odd behavior svn rev 2:5901M
by brandon＠cryy.com 16 Oct '09

16 Oct '09

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

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[Kamailio-Users] Problem with LCR "ping"
by Ricardo Martinez 16 Oct '09

16 Oct '09

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.-

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[Kamailio-Users] dispatcher failover between two groups
by Robert McGilvray 16 Oct '09

16 Oct '09

Kam 1.5 I'm using dialplan module to return a setid for the matching regex which is passed onto dispatcher to select the proper gateways. This works fine including failover to the next gateway in the group in failure_route. What I can't quite wrap my head around is how I can setup failover between different dispatcher groups. The scenario is as follows: 2 external PSTN gateways (sip trunks) in group 1 6 internal gateways with PSTN access via PRIs. Two gateways per group for a total of three separate setids in dispatcher. I'm doing some LCR across the 6 internal gateways which are all in different cities via regex matching in dialplan. What I'd like to do is try one of the internal groups first if it matches the local NPA/NXX, and if ALL gateways in that group fail move on to the external gateway group, and try all those gateways (and vice versa in my external gateways fail for non matching numbers). I thought of two possible solutions but I'm not sure if either will work how I expect. 1) Use the priority field in dispatcher to setup a large gateway group that has all the gateways I want to use. - I tested this out in kam 1.5 and the priority field seems to be ignored when hasing on call-id. - What is the intended usage of priority? Will setting two gateways to 0 and the rest to 10 cause the load balancing functions to only select the ones with 0? It seems to be ignoring priority and using all 8 gateways 2) Once in a failure_route can I move the processing into a request_route and call ds_select_dst again with a different setid and append it as a branch? Is there a cleaner solution? Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

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[Kamailio-Users] Codecs
by Anders 16 Oct '09

16 Oct '09

Hi, A very basis 101-question, which I couldn't find the answer to elsewhere: how/where do I see/set codecs allowed and disallowed in Kamailio? Thanks!

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Re: [Kamailio-Users] Having problems using RTPProxy to bridge internal/external networks
by Örn Arnarson 15 Oct '09

15 Oct '09

Hi, I have exactly the same scenario; no NAT involved. Just trying to bridge between these two interfaces. Now, to clarify better what I have done; As I understand it, force_rtp_proxy / rtpproxy_offer needs to be used to inform RTP proxy that it will have to relay the packets. So, that pretty much makes it necessary to use that call. As I understand it, you can't call both fix_nated_sdp and force_rtp_proxy, as the SDP will be mangled (two IP addresses, one after the other). However, I can't get the SDP rewritten properly without using fix_nated_sdp. No matter how I call force_rtp_proxy, I can't specify my own IP address. It works as expected when using fix_nated_sdp though. Here's how I call it: force_rtp_proxy("coe","10.252.1.8"); I've tried multiple variants of flags, c, co, o, coe, coi, i, no flags, etc etc, but no matter what I do, the public interface IP address does not change to the one supplied in the function, and remains the public IP address. So, to summarize; I can change all the headers to be correct with the fix_nated_sdp function. When using that function, RTPProxy does not relay the RTP streams. When I use force_rtp_proxy / rtpproxy_offer, RTPProxy does indeed relay the streams, but does not alter the SDP IP address information to the required specifications. Is there something that I'm missing in the usage of the force_rtp_proxy function, or is it not working as documented? Best regards, Örn 2009/10/14 Alex Balashov <abalashov(a)evaristesys.com>: > Örn Arnarson wrote: > >> Oh, I forgot one thing; >> >> I'm using fix_nated_sdp, as previously mentioned. I have a hunch that >> because I'm not using force_rtp_proxy/rtpproxy_offer it's not being >> relayed (a hunch, because I don't fully understand how rtpproxy >> works). Problem is though, I can't get the IP addresses correct by >> using rtpproxy_offer. > > On behalf of Joe (same company) I will add that there is no NAT scenario > here. Both the private and public network are directly routable from the > rtpproxy box; there is no IP address oversubscription, and no reason to > ignore ports or addresses in SDP and replace them with received parameters. > This is not a far-end NAT traversal solution. > > What is basically being asked here is how to bridge between two interfaces > (internal, external) without NAT using rtpproxy, and if anyone can provide a > canonical implementation or example. > > We've got it running in bridge mode, e.g. > > rtpproxy -F -s udp:10.x.x.x:5050 -l 10.x.x.x/89.y.y.y > > ... and basically, are just wondering how to get this to work on the > Kamailio side using rtpproxy_offer()/rtpproxy_answer(). > > The RTP proxy machine is a dedicated host; the SIP proxies are virtual. > > > -- > Alex Balashov - Principal > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 >

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[Kamailio-Users] SIP-T messages
by Luis Jimenez 15 Oct '09

15 Oct '09

Hi, can some one please guide me on how can i parse the received SIP-T messages from the SDP in Kamailio. Thanks in advance Luis Jimenez

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