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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 68, Issue 77
by virendra bhati 10 Feb '12

10 Feb '12

Michael, One of my friends from USA is able to loding with server at port 5060. It means it's not server issue at all. In India me at hyderabad and another one from Delhi is not able to register with port 5060. So it's not my ISP issue. Below is the sever logs for port 5060. I have Asterisk and FreeSwitch both thnigs on it. Asterisk is closed and freeSwitch is working now. [root@IS-15502 ~]# netstat -pln | grep 5060 tcp 0 0 78.129.163.44:5060 0.0.0.0:* LISTEN 3933/freeswitch tcp 0 0 ::1:5060 :::* LISTEN 3933/freeswitch udp 0 0 78.129.163.44:5060 0.0.0.0:* 3933/freeswitch udp 0 0 ::1:5060 :::* 3933/freeswitch [root@IS-15502 ~]# iptables -L Chain INPUT (policy ACCEPT) target prot opt source destination Chain FORWARD (policy ACCEPT) target prot opt source destination Chain OUTPUT (policy ACCEPT) target prot opt source destination On Fri, Feb 10, 2012 at 12:39 AM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: I am not able to reg with 5060 via softphone (Michael Collins) > 2. Re: Can someone help me with a complicated configuration? > (Michael Collins) > 3. Re: skipping vm and plays enter the pwd (Michael Collins) > > > ---------- Forwarded message ---------- > From: Michael Collins <msc(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 9 Feb 2012 11:02:02 -0800 > Subject: Re: [Freeswitch-users] I am not able to reg with 5060 via > softphone > You need to find out why you can't reg on port 5060. I see that you have > an external IP address listed there. Is that the only interface on your > server? One trick you can do is to swap the internal.xml and external.xml > sip profiles. Basically just rename the files and restart freeswitch and > see if you can register on 5060. If not you'll need to supply console logs > w/ siptraces and put them on pastebin.freeswitch.org so the gang here can > take a look. > > -MC > > On Wed, Feb 8, 2012 at 4:50 AM, virendra bhati <virbhati(a)gmail.com> wrote: > >> >> Hi list, >> >> I am new for FreeSwitch. I was working with asterisk since 4 years. Now i >> want to learn FreeSwitch. I have installed at my server. But when I try to >> reg with server .I am not able to register with it with port 5060. but if I >> will 5080 then reg with server but can't call to local extension like 1001 >> - 1019. >> >> Please guide me what is the issue ? >> >> *fs_CLI log:-* >> >> EXECUTE sofia/external/1006(a)78.129.163.44 bridge(user/1005(a)78.129.163.44) >> 2012-02-08 12:59:37.493777 [DEBUG] switch_ivr_originate.c:1885 variable >> string 0 = [presence_id=1005(a)78.129.163.44] >> 2012-02-08 12:59:37.493777 [ERR] switch_ivr_originate.c:2430 Cannot >> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] >> 2012-02-08 12:59:37.493777 [DEBUG] switch_ivr_originate.c:3228 Originate >> Resulted in Error Cause: 606 [USER_NOT_REGISTERED] >> 2012-02-08 12:59:37.493777 [ERR] switch_ivr_originate.c:2430 Cannot >> create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] >> 2012-02-08 12:59:37.493777 [DEBUG] switch_ivr_originate.c:3228 Originate >> Resulted in Error Cause: 606 [USER_NOT_REGISTERED] >> 2012-02-08 12:59:37.493777 [INFO] mod_dptools.c:2355 Originate Failed. >> Cause: USER_NOT_REGISTERED >> EXECUTE sofia/external/1006(a)78.129.163.44 answer() >> >> -- >> >> Thanks and regards >> >> Virendra Bhati >> +91-8885268942 >> Software Engineer >> E-mail-: virbhati(a)gmail.com >> Skype id:- virbhati2 >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > ---------- Forwarded message ---------- > From: Michael Collins <msc(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 9 Feb 2012 11:03:18 -0800 > Subject: Re: [Freeswitch-users] Can someone help me with a complicated > configuration? > What kind of devices will connect to the public IP addresses? How will > *they* handle a colo change that results in a new IP address? > > -MC > > On Wed, Feb 8, 2012 at 9:28 AM, Josue Diaz Cruz <jdiaz(a)coinfru.com> wrote: > >> ** >> I have a server with 4 ETH. Each one have a different IP >> In 2 cases i fixed the ip address, In the other 2 cases i have 2 cable >> connections in bridge mode so i receive the public ip in dhcp mode. >> >> eth0 local address 192.168.1.100 >> eth1 local address 192.168.2.100 >> eth2 public address dhcp (cable) >> eth3 public address dhcp (cable) >> >> I was thinking to build 4 different profiles for each eth port, but i >> came to know that if we change the place of colocation it will not work >> properly because the dhcp ip. It is any way to attach a profile to the eth >> port, not the ip? >> >> Or if you know a better way to manage this case. I would like to receive >> traffic from the 2 public ip's and route it both like if i have just one >> profile. >> >> * >> >> Josue Diaz Cruz >> >> Departamento Tecnico y Soporte >> >> jdiaz(a)coinfru.com >> >> >> >> C/ Balsicas 3 >> >> Alquerias | 30580 | Murcia >> >> www.coinfru.com >> >> >> >> >> >> * >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > ---------- Forwarded message ---------- > From: Michael Collins <msc(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 9 Feb 2012 11:09:20 -0800 > Subject: Re: [Freeswitch-users] skipping vm and plays enter the pwd > This isn't enough information to diagnose. Please supply a log of the call > from start to finish. Also, you probably should include the dialplan > extension that is handling this process. > > -MC > > On Wed, Feb 8, 2012 at 11:49 AM, Aakash <aakashviswam(a)gmail.com> wrote: > >> Hi All, >> >> I have configured has any inbound calls enter to my inbound number play an >> ivr and transfer to user ext (eg 1080)pressed by customer,if the user dint >> pick the call enter to their voicemail automatically.But sometimes i am >> getting please enter password followed by # by skipping voicemail >> message.I >> have attached my logs below >> >> http://pastebin.freeswitch.org/18321 >> >> >> Regards, >> Aakash.V >> >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/skipping-vm-and-plays-enter-t… >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbhati(a)gmail.com Skype id:- virbhati2

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I am not able to reg with 5060 via softphone
by virendra bhati 10 Feb '12

10 Feb '12

Hi list, I am new for FreeSwitch. I was working with asterisk since 4 years. Now i want to learn FreeSwitch. I have installed at my server. But when I try to reg with server .I am not able to register with it with port 5060. but if I will 5080 then reg with server but can't call to local extension like 1001 - 1019. Please guide me what is the issue ? *fs_CLI log:-* EXECUTE sofia/external/1006(a)78.129.163.44 bridge(user/1005(a)78.129.163.44) 2012-02-08 12:59:37.493777 [DEBUG] switch_ivr_originate.c:1885 variable string 0 = [presence_id=1005(a)78.129.163.44] 2012-02-08 12:59:37.493777 [ERR] switch_ivr_originate.c:2430 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2012-02-08 12:59:37.493777 [DEBUG] switch_ivr_originate.c:3228 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2012-02-08 12:59:37.493777 [ERR] switch_ivr_originate.c:2430 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] 2012-02-08 12:59:37.493777 [DEBUG] switch_ivr_originate.c:3228 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2012-02-08 12:59:37.493777 [INFO] mod_dptools.c:2355 Originate Failed. Cause: USER_NOT_REGISTERED EXECUTE sofia/external/1006(a)78.129.163.44 answer() -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbhati(a)gmail.com Skype id:- virbhati2

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Server with 2 NICS
by Luis Daniel Lucio Quiroz 10 Feb '12

10 Feb '12

Helo :) I'm new to FS but not to linux. I've sucessfully install FS into a centos 6.2. But I have an issue. Server has 2 nics: venet0: 10.9.19.5/16 eth0: <public IP> when starting fs, it listen SIP (5060/udp) using venet0. Phones can register and they talk each other. However, we are using an external GW and that fails. Doing a depth analisys I've realized that the reason is that because venet0 is being used, SIP payload has venet0 within but server IP is eth0's so this is tipical mitchach. I've sucessfully configure in vars.xml the local_ip4 (or whatever it writes) to use public IP. but after that no phones are registering. (before anyone says, yes phones points to public IP of course). Nmap reports that SIP (5060/udp) is listen by eth0 but when I got phone SIP, i only get Registering request without an answer. I really need to make this listen in eth0. What suggestion do you have. Please void telling me to turn of venet0 (not possible for politics). TIA

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FreeTDM: cannot set option
by Valery Kalinin 10 Feb '12

10 Feb '12

Hi! Work with FreeTDM (latest git) & wanpipe (3.5.24) freetdm.conf.xml: <analog_spans> <span name="FXO"> <param name="enable_callerid" value="false"/> <param name="wait_dialtone_timeout" value="0"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </analog_spans> but in log this options is not set: 2012-02-10 06:09:01.194586 [INFO] ftmod_wanpipe.c:401 [s1c2][1:4] Configured wanpipe device FD: 34, DTMF: softwa 2012-02-10 06:09:01.194646 [INFO] ftdm_io.c:4912 Configured 2 channel(s) 2012-02-10 06:09:01.194702 [DEBUG] mod_freetdm.c:3131 analog_spans var = enable_callerid 2012-02-10 06:09:01.194729 [DEBUG] mod_freetdm.c:3131 analog_spans var = wait_dialtone_timeout 2012-02-10 06:09:01.194748 [DEBUG] mod_freetdm.c:3131 analog_spans var = dialplan 2012-02-10 06:09:01.194757 [DEBUG] mod_freetdm.c:3131 analog_spans var = context 2012-02-10 06:09:01.195109 [INFO] ftdm_io.c:5002 Loading SIG from /usr/local/freeswitch/mod/ftmod_analog.so 2012-02-10 06:09:01.195139 [INFO] ftdm_io.c:5185 auto-loaded 'analog' 2012-02-10 06:09:01.195150 [DEBUG] ftmod_analog.c:195 Configuring span FXO for analog signaling ... 2012-02-10 06:09:01.195165 [DEBUG] ftmod_analog.c:210 Analog config var = tonemap 2012-02-10 06:09:01.195183 [DEBUG] ftmod_analog.c:210 Analog config var = digit_timeout 2012-02-10 06:09:01.195193 [DEBUG] ftmod_analog.c:210 Analog config var = max_dialstr 2012-02-10 06:09:01.195205 [DEBUG] ftmod_analog.c:210 Analog config var = hotline 2012-02-10 06:09:01.195223 [DEBUG] ftmod_analog.c:210 Analog config var = enable_callerid 2012-02-10 06:09:01.195232 [DEBUG] ftmod_analog.c:210 Analog config var = answer_polarity_reverse 2012-02-10 06:09:01.195241 [DEBUG] ftmod_analog.c:210 Analog config var = hangup_polarity_reverse 2012-02-10 06:09:01.195250 [DEBUG] ftmod_analog.c:210 Analog config var = polarity_callerid 2012-02-10 06:09:01.195267 [DEBUG] ftmod_analog.c:210 Analog config var = polarity_delay 2012-02-10 06:09:01.195294 [DEBUG] ftmod_analog.c:210 Analog config var = callwaiting 2012-02-10 06:09:01.195312 [DEBUG] ftmod_analog.c:210 Analog config var = wait_dialtone_timeout 2012-02-10 06:09:01.195322 [DEBUG] ftmod_analog.c:229 Wait dial tone ms = 5000 2012-02-10 06:09:01.195331 [DEBUG] ftmod_analog.c:302 [s1c1][1:3] Enabled call waiting 2012-02-10 06:09:01.195340 [DEBUG] ftmod_analog.c:302 [s1c2][1:4] Enabled call waiting See option: Wait dial tone ms = 5000 But in span I set this parameter to zero! What am I doing wrong?

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Hangup Cause in LUA Hangup Hook
by Gerald Weber 10 Feb '12

10 Feb '12

Hello, is there a way to get the HangupCause in a HangupHook in LUA ? following commands return nil: s:getVariable("hangup_cause") s:getVariable("hangup_cause_q850") s:hangupCause() returns "NONE" I know I can set api_hangup_hook and use the env object to get the channel vars, but i think that's a litte overkill to use 2 scripts for 1 Call. Thx gw

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Registration problem after attempting to install mod_opal and updating to latest git pull
by Phil Quesinberry 10 Feb '12

10 Feb '12

After a recent unsuccessful attempt to install mod_opal and updating to a recent git, I now get some odd reports of gateway registration failures, although everything appears to be working ok call-wise for the moment. Doing a 'make uninstall' from /root/opal and updating to the latest git tonight yielded no improvement. Here are some excerpts of failures reported in fs_cli: 2012-02-09 00:12:59.992099 [WARNING] sofia.c:4717 Unregister QS8002 2012-02-09 00:12:59.992099 [WARNING] sofia.c:4730 Ping failed QS8002 with code 900 - count -1/-1/1, state UP 2012-02-09 00:12:59.992099 [WARNING] sofia.c:4717 Unregister HL7612 2012-02-09 00:12:59.992099 [WARNING] sofia.c:4730 Ping failed HL7612 with code 900 - count -1/-1/1, state UP 2012-02-09 00:13:00.011988 [WARNING] sofia.c:4717 Unregister HL7611 2012-02-09 00:13:00.011988 [WARNING] sofia.c:4730 Ping failed HL7611 with code 900 - count -1/-1/1, state UP 2012-02-09 00:13:00.011988 [WARNING] sofia.c:4717 Unregister HL7519 2012-02-09 00:13:00.011988 [WARNING] sofia.c:4730 Ping failed HL7519 with code 900 - count -1/-1/1, state UP 2012-02-09 00:13:00.011988 [WARNING] sofia.c:4717 Unregister CallCentric_T.38 2012-02-09 00:13:00.011988 [WARNING] sofia.c:4730 Ping failed CallCentric_T.38 with code 900 - count -1/-1/1, state UP 2012-02-09 00:13:01.992050 [WARNING] sofia_reg.c:474 QS8002 Failed Registration [0], setting retry to 60 seconds. 2012-02-09 00:13:01.992050 [WARNING] sofia_reg.c:474 HL7612 Failed Registration [0], setting retry to 90 seconds. 2012-02-09 00:13:01.992050 [WARNING] sofia_reg.c:474 HL7611 Failed Registration [0], setting retry to 90 seconds. 2012-02-09 00:13:01.992050 [WARNING] sofia_reg.c:474 HL7519 Failed Registration [0], setting retry to 60 seconds. 2012-02-09 00:13:01.992050 [WARNING] sofia_reg.c:474 CallCentric_T.38 Failed Registration [0], setting retry to 60 seconds. To give a sense of the timing, the three groups of warnings below are contiguous, from a single fs_cli session: 2012-02-09 00:21:18.984384 [WARNING] sofia.c:4717 Unregister HL7611 2012-02-09 00:21:18.984384 [WARNING] sofia.c:4730 Ping failed HL7611 with code 900 - count -1/-1/1, state UP 2012-02-09 00:21:20.988379 [WARNING] sofia_reg.c:474 HL7611 Failed Registration [0], setting retry to 90 seconds. 2012-02-09 00:21:22.988382 [WARNING] sofia.c:4717 Unregister HL7519 2012-02-09 00:21:22.988382 [WARNING] sofia.c:4730 Ping failed HL7519 with code 900 - count -1/-1/1, state UP 2012-02-09 00:21:24.988318 [WARNING] sofia_reg.c:474 HL7519 Failed Registration [0], setting retry to 60 seconds. 2012-02-09 00:22:00.447778 [WARNING] sofia_reg.c:1422 SIP auth challenge (REGISTER) on sofia profile 'internal' for [226(a)qsystemseng.no-ip.org] from ip 74.93.222.182 2012-02-09 00:22:04.147688 [WARNING] sofia.c:4717 Unregister HL7612 2012-02-09 00:22:04.147688 [WARNING] sofia.c:4730 Ping failed HL7612 with code 900 - count -1/-1/1, state UP 2012-02-09 00:22:04.167699 [WARNING] sofia.c:4717 Unregister HL7611 2012-02-09 00:22:04.167699 [WARNING] sofia.c:4730 Ping failed HL7611 with code 900 - count -1/-1/1, state UP 2012-02-09 00:22:04.167699 [WARNING] sofia.c:4717 Unregister CallCentric_T.38 2012-02-09 00:22:04.167699 [WARNING] sofia.c:4730 Ping failed CallCentric_T.38 with code 900 - count -1/-1/1, state UP 2012-02-09 00:22:05.167744 [WARNING] sofia_reg.c:474 HL7611 Failed Registration [0], setting retry to 90 seconds. 2012-02-09 00:22:05.167744 [WARNING] sofia_reg.c:474 CallCentric_T.38 Failed Registration [0], setting retry to 60 seconds. 2012-02-09 00:22:05.167744 [WARNING] sofia.c:4717 Unregister QS8002 2012-02-09 00:22:05.167744 [WARNING] sofia.c:4730 Ping failed QS8002 with code 900 - count -1/-1/1, state UP 2012-02-09 00:22:07.171680 [WARNING] sofia_reg.c:474 QS8002 Failed Registration [0], setting retry to 60 seconds. 2012-02-09 00:22:07.171680 [WARNING] sofia.c:4717 Unregister HL7519 2012-02-09 00:22:07.171680 [WARNING] sofia.c:4730 Ping failed HL7519 with code 900 - count -1/-1/1, state UP One of the gateways is SIPBroker, and calls through that gate starting failing after the update. I realized that the gateway was configured to ping SIPBroker unnecessarily and restarting the gateway would restore service until the ping. I disabled the ping for that gateway and calls kept working after that. Any ideas as to what might be going on? There was no such problem before the FS update/attempt to use mod_opal. I can provide additional/more detailed info if needed. Thanks! Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com

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[newbie] questions
by Josh 09 Feb '12

09 Feb '12

I am considering using FreeSWITCH as a replacement of our (rather crappy, outdated and buggy) sip proxy system, and have the following system setup on the server: * eth0 faces the external Internet interface, *but* it does not have a public IP address (it has a private one given to it by my ISP's DHCP server); * eth1 faces our internal network (say 10.1.1.0/24); * tun0 serves all mobile smartphones and connects to the internal network (it has a different ip range, say 10.1.2.0/24) - they are all connected via the Internet using OpenVPN; I would like to configure FreeSWITCH as a local registrar and ask it to handle internal calls between ourselves (eth1<->tun0). I would also like to use one external VOIP provider to which FreeSWITCH registers on startup. This also needs to have all NAT-related issues dealt with as well (hopefully by FreeSWITCH, if possible). I would also need to emphasise that FreeSWITCH *cannot* just listen on any (public-facing) interface port in order to receive calls from the external provider, simply because my ISP won't allow it (i.e. I can't just "open" that port and hope that someone will contact me - the connection needs to be initiated by our server for incoming call to get routed through). The purpose of registering this external account is so that both the smart phones (tun0) and the internal net (eth1) users could use this account to make external calls (starting with "0"). Obviously, I need these calls to be routed properly via the external VOIP account. In addition to that, I would also need to receive calls from that external account to a nominated internal one (say on extension 20). Could this be done relatively easily in FreeSWITCH? A question about binding: in order to be able to use both tun0 and eth1 interfaces so that FreeSWITCH serves the calls from both eth1 and tun0 devices/networks, is it possible to instruct it to bind on different IP addresses (on tun0 and eth1 respectively) in order to serve calls to/from these networks? I do *not* wish to bind to 0.0.0.0 - just use the interfaces I am interested in - tun0 and eth1 in my case. If all of the above is possible, do I have to mess about and configure SNAT/DNAT separately and use sip kernel module "helpers" etc (I *do* have the routing set up properly though - that I am sure works 100%) or is FreeSWITCH perfectly capable of setting it all up for me? One final query: is there a decent distribution source package of FreeSWITCH, which I could take and then use it to build a binary package (I use Fedora on all of our machines). I need this, because all of our images are built using the rpm packaging system and I cannot just "fetch the latest git", compile and install FreeSWITCH - I need it packaged, so that the image builder picks it up and distributes it to the machine concerned. I checked the SUSE repository (as instructed on the FreeSWITCH wiki), but found 2 packages, both of which cannot be build due to errors! Many thanks in advance!

3 9

finally, an open alternative to Viber
by Daniel - Lumicall.org 09 Feb '12

09 Feb '12

The rise of Viber has, for some people, been a case of `skype, not again!' until now... Lumicall is now in the Android market - and it fully interacts with other SIP products using ENUM and SRV records. Any feedback on this is welcome. Interconnect (FreeSwitch calling to/from Lumicall users) is based on the idea of federated SIP, it is explained at http://www.lumicall.org for those who want to connect up to it. Please bear in mind: Lumicall supports ICE (RFC 5245) for NAT traversal, it is using the ice4j implementation from the Jitsi community. This makes the SIP packets bigger and often they are too big for the MTU of a UDP packet. When using ICE, it seems essential to use TLS, to avoid the MTU problems and also to avoid routers mangling the SIP headers (ICE doesn't need help from routers, they only confuse the algorithm) Various other issues are addressed in the release notes and FAQ http://www.lumicall.org/faq If there are specific issues with FreeSwitch compatibility (either for the Lumicall app, or for interconnect using ENUM/SRV/TLS) then I'm happy to start a dedicated page or wiki to collect the solutions in one place for FreeSwitch users.

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presence on latest GIT
by Vik Killa 09 Feb '12

09 Feb '12

I've noticed the past week that BLF was not working, I pulled latest GIT and tried to figure out what was going wrong. After trying many different things on the Polycom, i enabled presence debug in sofia. right awaythere were a lot of errors thrown. Is sofia presence broken in latest GIT? If so, any ETA on when it will be fixed? Thanks.

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http://www.uceprotect.net the craptastic RBL
by Ken Rice 09 Feb '12

09 Feb '12

So... It appears the FreeSwitch list server has been listed as a Spam source at http://www.uceprotect.net UCEProtects solution wait atleast a week then maybe they will drop the list server from their spam listing or pay them $115 and they¹ll delist it Oh! BTW we can prevent the server from being listed again for $115 for 2 years of white listing.... 1. Why would we want to do that? 2. Why don¹t they provide a way for obvious legit projects such as FreeSWITCH to get immediately Delisted? 3. Why do they allow anyone willing to pay them a few bucks to get themselves delisted? Sounds like they are just in it for the money... If you are using this RBL drop it, its not really blocking your spam K

4 4