March 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to route ata question
by Charles Bujold 15 Mar '11

15 Mar '11

My question is I have an ATA FXO (Grandstream ht503) that currently sends calls to an extension, I want the calls to be filtered by the other Freeswitch settings and I am not certain how to proceed. I originally created a Main IVR (ext 5555) and sent the incoming FXO ATA calls to the IVR extension. Now I want to send the calls to some other extension or inbound route so that the calls are routed through all the Freeswitch settings such as holidays and time of day conditions before being sent to the IVR or some after hours message. I'm not certain how I should setup the FXO ATA to send the calls so Freeswitch processes them properly? Please advise. cjb

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How to stop SIP auth challenge (REGISTER) msgs?
by Mario G 15 Mar '11

15 Mar '11

I see this was introduced in a recent git. I know they are not an error but does anyone know if there is a way to stop these? Something I need to change on the phones? These are really a problem while trying to working on problems. The log is filled with these. Thanks. WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [200(a)100.200.1.7] from ip 100.200.1.12

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Question about proxy_media
by Boris Kovalenko 15 Mar '11

15 Mar '11

Hello! I have two sip profiles. One is for transit traffic with proxy_media enabled by default. Second is a SoftPBX for my clients with a features like a IVR, conference, attended call transfer and so on. So the question is - what happens when A leg is from transit profile and B leg is to SoftPBX profile. What mode of proxy_media will be set when call is connected? May my user on B leg use DTMF for call transfer for example? What mode will be used for FAX? -- С уважением, Борис Коваленко ЗАО "Тагнет" (3435) 494991

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IP Failover with Sofia Recover - not recovering?!
by Avi Marcus 15 Mar '11

15 Mar '11

I finally figured out how to get pacemaker to start profiles/trigger a sofia recover on moving the floating IP (but an FS crash it still can't handle) Anyway - I 1) set up odbc for the core, and mysql master-master replication, so I see calls & channels being logged in both 2) I set track calls in all the profiles So when I set it to transfer from one node to another, I'm kinda surprised I'm not getting a recovery of the calls. I do see that the DB has a hostname stored - "sip1" or "sip2" depending on where the call originated, but even if I change it manually, a sofia recover seems to ways say "no calls to recover" Am I missing something?! -Avi Marcus

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SIP profile and alias ip
by Boris Kovalenko 14 Mar '11

14 Mar '11

Hello! Reading documentation about sip profiles found a <param name="alias">. So I may use same profile with two IP's. Is there any disadvantages to use alias? If endpoint will be connected to alias IP what IP FS will use for outgoing RTP (alias ip or rtp-ip)? Also, if gateway (for outbound calls) is reachable (by host route table) via alias IP will it be used for RTP instead of primary rtp-ip? -- Regards, Boris

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trunk without authentication
by Borin 14 Mar '11

14 Mar '11

Hello FS users, Is it possible in FS to allow all calls from a particular IP address without authentication? like in asterisk with insecure port, invite. I tried to add a new carrier in /usr/local/freeswitch/conf/sip_profiles/external/new-carrier.xml <include> <gateway name="new-carrier"> <param name="username" value="anything"/> <param name="password" value="anything"/> <param name="realm" value="carrier.ip.address"/> <param name="register" value="false"/> <param name="caller-id-in-from" value="true"/> </gateway> </include> But after restart it stil sends 407 Proxy Authentication Required to the carrier. Is there any way to trust the carrier IP?

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Where to download zrtp/zfone - server offline?
by Peter Steinbach 14 Mar '11

14 Mar '11

I tried to install zrtp and to download Zfone, but I get http://www.zfone.com/server_problems.html "Sorry, but our download server is offline." (since 29-Jan-2011) Does anybody know of another download source? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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Multiple Registrations not working
by Yihui Li 14 Mar '11

14 Mar '11

Hi, I am trying to use Multiple Registrations feature in bridge, but found it not always working when there are more than 1 bridge targets like below: <action application="bridge" data="user/1015,user/1016" /> where 1015 is registered on 2 PC. To simplify my case, I have also tested it with CLI command. Test 1, originate user/1015 &parks both PC are ringing, Test 2, originate user/1015,user/1016 &parks only 1 PC is ringing in internal.xml, I added <param name="multiple-registrations" value="true"/> I am running git-head on ubuntu. Can anyone help with this issue? Thanks. Eric

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Can you use ext-ip.. autonat: with -nonat ?
by Mario G 14 Mar '11

14 Mar '11

Changed router now only 1 out of 5 calls come in. I previously had "ext-sip/rtp-ip:1.2.3.4" and -nonat on another router but having major issues on new one. Wiki does not answer this question: Can you use "exp-rtp/sip-ip"autonat:1.2.3.4" when freeswitch is started with -nonat ? Outbound is fine, inbound fails 80 percent if the time. I really wanted use ALG but when I did all worked fine except call dropped while conversing.

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multiple gateways and NAT
by Johan Helsingius 14 Mar '11

14 Mar '11

I am sure this question comes up all the time... I want to use several different SIP gateway providers (for different regions and services). Freeswitch seems to deal with NAT just great with one external gateway, but multiple gateways seems to be a rather thornier issue. So, do I need totally separate profiles for each gateway? And do they have to use different ports? And does that mean that I need to get my firewall to port map each of them separately (I would assume they all try to use the same 5060 port to reach my system in case of an incoming call)? Julf

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