May 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Better approach for multi-tenant SIP testing (part 1)
by Cal Leeming [Simplicity Media Ltd] 19 Jun '13

19 Jun '13

Hello all, After nearly a year of active development on building our own voice platform, we've adopted some cool tricks to speed up the workflow and make testing less tedious. Over the next few months I'll be posting more articles on the tricks we have used and the lessons we have learnt. Here is part 1 - a write up about a neat hack we used to virtualize phone testing; http://blog.simplicitymedialtd.co.uk/533/quick-and-easy-approach-for-multi-… For those that don't have time to read, here is a screenshot; http://i.imgur.com/e9kUdxg.jpg This trick, which we've used in development for approx 6 months, has saved countless hours in development overheads, with one-click start/stop functionality and all the necessary modules right there in one place. Of course, this should never be used as a full replacement as nothing beats a proper hardware test to catch those weird edge cases, but it does allow you to do the majority of your testing (95%) from a single screen rather than a desk full of phones. It also makes attaching wireshark a lot easier, without having to use span ports and the alike. Hope this helps someone else! Cal

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Anyone able to get freeswitch to register a gateway over TLS or TCP? I can't get it to not send UDP packets for reg.
by Mitch Capper 14 Jun '13

14 Jun '13

I am trying to get freeswitch to register with another freeswitch server (on TLS port 5061) however default the config settings I believe should set it, it always trys to register over UDP (after registration or with registration set to false however it does send TCP then for outbound calls). If anyone has a gateway config that actually registers over TCP or TLS that would be great. The client gateway/profile config at: http://pastebin.freeswitch.org/15383is what i believe should work but isn't I have tried setting contact-params on the gateway to tport=tls tport=tcp tport=tcp;transport=tls and transport=tls without any luck. Anyway even TCP would be helpful as it would allow me to trace better how the sip registration decides to use TCP or UDP. I tried digging into libsofia but its a bit hard to trace and understand with its generic event structure and tagging (plus the documentation isn't the most rebust on registration options). Thanks ~Mitch

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Kamailio+Freeswitch SBC
by Camila Troncoso 14 Jun '13

14 Jun '13

Hi all, I’m working for a while with Kamailio+Freeswitch as SBC. I have this structure: * * * * * * * * * * * * * * * * When I make a call, from one client coneccted to LCR, it is route to my SBC and afterwards to his destiny in the cloud passing thourgh a gateway. When the destiny is unreachable, the LCR reroutes the call to another gateway. Sometimes the LCR send this retry again to the SBC , becouse the second gateway is also in the cloud, but when this happens, Fresswitch answer with : “482 Request merged”. Is there a way to solve this? Regards, * * *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

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Gateway Call Limits
by Alex Ynema 12 Jun '13

12 Jun '13

Hi, I'm trying to figure out how to limit the number of calls a Gateway is allowed to use. Our Sip provider has provided up with 200 which I need to set within the system somehow. What's the best way to handle it for an outgoing only system. I've been trying to figure out how how to configure 'Rate limiting concurrent outgoing calls via a provider' which is mentioned in the wiki a bit but nothing specific on what to actually do. *Alex Ynema** *| IT Consultant alex(a)opensystems.net.au Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 Office: +61 8 9427 2500 Mobile: +61 404 796 894 <http://www.ynema.net/> <http://www.ynema.net/>IT Consultant for Open Systems Support www.opensystems.net.au

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pickup endpoint - Break the Simultaneous Ring
by Lloyd Aloysius 12 Jun '13

12 Jun '13

Hello All The following break the Simultaneous Ring.User registered two different devices. Only one Device ring after add the *,pickup/dave(a)alcan.mydomain.ca * *bridge_data: * {sip_invite_domain=alcan.mydomain.ca ,ignore_early_media=true,force_transfer_context=alcan.mydomain.ca }[leg_timeout=20]user/dave@alcan.mydomain.ca*,pickup/dave@alcan.mydomain.ca* Thanks LLoyd * <http://www.sunteltech.ca/blog/>*

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USSD support for mod_gsmopen
by Yuriy Ostapchuk 11 Jun '13

11 Jun '13

Hi, I've submitted a patch that implements USSD messaging support. (useful for checking and refilling account balance and other provider-specific usage) Here is the JIRA ticket: http://jira.freeswitch.org/browse/FS-5078 If this feature is considered useful, I would appreciate if it's integrated in git. Thanks, Yuriy

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Too many channels used in my scenario
by Mino Haluz 11 Jun '13

11 Jun '13

Hi, I am thinking of this topology: caller->freeswitch->proxy in proxy, there will be some routing module as carrierroute or so and then call is sent back callee<-freeswitch<-proxy thing is, if only one call is placed, 4 channels are used. If I used sangoma transcoding card, every channel has to be transcoded to the internal FS coded and back, so 8 transcoding channels in sangoma driver are used for just one call (!). How could I tell freeswitch to do not pass SDP information to the proxy side and let it use only 2 channels as in standard caller->callee call. Thanks, Mino

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How to add username in SIP packet OPTIONS
by "Гришин В.С." 11 Jun '13

11 Jun '13

My gataway profile <include> <gateway name="SIP_M200"> <param name="username" value="USERNAME_STRING"/> <param name="realm" value="172.30.0.2"/> <param name="register" value="false"/> <param name="caller-id-in-from" value="true"/> <param name="ping" value="5"/> </gateway> </include> SIP packet which is accepted by the remote UA Thu Nov 29 16:38:24 2012 RECEIVE 15:5060 <-- 172.30.0.1:5080 OPTIONS sip:172.30.0.2;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.30.0.1:5080;rport;branch=z9hG4bK6DKm54N817tUS Max-Forwards: 70 *From: <sip:172.30.0.2>;tag=BZN3H5ZBpm8gQ To: <sip:172.30.0.2>* Call-ID: 79818268-b4c4-1230-60ac-00d0b7ded66a CSeq: 36758947 OPTIONS User-Agent: FreeSWITCH-mod_sofia/1.2.3+git~20120920T220849Z~f718a5e8e6 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 but in reply arrives nothing... the developer points to that that in a SIP packet there is no field username in *From: <sip:172.30.0.2>;tag=BZN3H5ZBpm8gQ <----in this string no username To: <sip:172.30.0.2> <---- and in this string no username* it is necessary to make as From: "USERNAME_STRING" <sip:USERNAME_STRING@172.30.0.1>;tag=9DcB7gSNZZ41j What it is necessary to change or add in a gateway profile (or other place) for this purpose? Vladislav Grishin

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G729 ptime mismatch on ingress and egress legs
by Ashwin Rath 10 Jun '13

10 Jun '13

Hi I am trying to configure FS as a B2BUA which can do ptime conversions between the ingress and egress leg. The question is, if both legs have G729 as the codec but ptime 20 on one and 30 on another, would i still need to get the licensed mod_com_g729 or can it work with the passthrough mod_g729. I already tried it out, but it doesnt seem to work. Is there some configuration needed for this ? -- Thegrid

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JIRA contributions - 15/05/2013
by Cal Leeming [Simplicity Media Ltd] 10 Jun '13

10 Jun '13

Hello all, Last week I made some contributions on JIRA by triaging some tickets. FS-3964 - requested larger bounty offer, if no bite then reclassify as nice to have FS-4211 - requested further info, close after 2 weeks if no response. FS-4281 - requested more info, waiting on core devs FS-4891 - requested more info, updated wiki docs, close after 2 weeks if no response FS-5154 - requested OP create new ticket for seperate issues, close after 2 weeks if no response However, I was unable to action them due to lack of access on JIRA. I sent an email to Ken/Michael requesting access to close tickets and mark as resolved etc, but realised this was not the appropriate way as you shouldn't need to have extra privileges in order to do ticket triage. Therefore, I'd like to open up discussion about how we can improve JIRA to allow for public contributions and make ticket triaging easier. My suggestions are; * Add a new step in the workflow that switches it into "needs more client info", so it automatically closes the ticket after X days if no extra info is received. * Have a "maybe one day/needs contributions" status in the workflow, rather than leaving the ticket as open.. that way we can see really how many are waiting, and how many are waiting on contributions. * Have a "needs design decision" status in the workflow, indicating that a core dev needs to make the final call * Introduce an official 5-for-1 ticket system where users can triage 5 tickets and request that a core dev look at any ticket of their choice in return. I think we stand a much better chance of getting a lot more public contributions if we implement these changes. Probably a bit too short notice to discuss on todays conf call, but it would be good to get some ideas/feedback in time for next weeks call. Modifying JIRA and making workflow changes is no easy task, so it would need to be thoroughly discussed and understood. I'll be on the call today 20 minutes early today if anyone wants to discuss. Any thoughts/ideas? Thanks Cal

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