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Re: [Freeswitch-users] [Freeswitch-dev] Cluecon 2007 is almost here don't forget to register. June 26th - 28th
by Brian West 01 Jun '07

01 Jun '07

Not yet but we can setup one ASAP. /b On Jun 1, 2007, at 11:53 AM, Peter Saint-Andre wrote: > Brian West wrote: > >> Cluecon Telephony Developer Conference 2007 (June 26th - 28th) > > Be there or be square! Is there a dedicated discussion list so that > participants can coordinate? > > Peter > > > _______________________________________________ > Freeswitch-dev mailing list > Freeswitch-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org

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Cluecon 2007 is almost here don't forget to register. June 26th - 28th
by Brian West 31 May '07

31 May '07

FreeSWITCHers, If you haven't please take a moment and look at http:// www.cluecon.com, This year is on track to be even better than our previous two years. If you can't make it please tell all your friends and co-workers that might be interested. Meet the people that make Open Source VoIP possible. Surj Patel and I will be working the conference together this year so please show up.. interact.. have fun. Cluecon Telephony Developer Conference 2007 (June 26th - 28th) will be held in: Best Western Inn of Chicago 162 E Ohio Street Chicago, Illinois 60611 Phone: (312) 787-3100 Toll Free: (800) 557-2378 US/Canada Call 877 742-CLUE for more information about the conference or to just chat with me. Thanks, Brian West PS: <subliminal-message>YOU MUST GO!!!</subliminal-message>

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Re: [Freeswitch-users] Strange Reg
by Anthony Minessale 28 May '07

28 May '07

you need to monitor both the sip and the rtp traffic. Run freeswitch with TPORT_LOG=1 and also run wireshark on the machine capturing udp traffic on the interface where the sip calls will take place. Also make sure the sofia.conf.xml profile is setup the same as the default example in the source tree apart from the options necessary to enable the directory auth. Make sure the box freeswitch is on has no firewall enabled while you are testing (we do not support getting things to work with a firewall that's your job) Also make sure the nat router the ata is behind does not also have a firewall that is intentionally blocking RTP traffic (getting a trace from there with wireshark also would be helpful) Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Dave <lists+freeswitch(a)bigfatdave.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, May 25, 2007 1:24:36 AM Subject: Re: [Freeswitch-users] Strange Reg Now that I've documented what works, let's move on to what doesn't :-) I can't get outgoing calls to work at all (from the ATA out), and incoming calls ring, but have no audio in either direction for about half a minute, before just dying completely. Before I go around posting tons of logs, can somebody just give me some pointers about what to look for? Thank you, - Dave On Mon, May 21, 2007 at 05:17:30PM -0700, Anthony Minessale wrote: > You need to make sure you setup the user directory correctly. > > http://wiki.freeswitch.org/wiki/FreeSwitch_XML_User_Directory_Guide > > > My earlier email provided a precise example and warnings about making sure the tags etc match up. > > When you setup the <domain> tag in the directory > you must make sure there is a domain tag that has the same name as the host portion of the to: header in the register. > > Using forensics from your traces here is the exact domain tag to use > in directory.xml with 1 caveat: > > ----------------------------------------------------------------------- > **replace the string 1234 with the password you specified in the ata** > ----------------------------------------------------------------------- > > *NOTE* > normally $${domain} would be ok here as the name > assuming that the domain variable is set in freeswitch.xml > but I put the exact domain name to avoid confusion. > > *ALSO NOTE* > You can comment the password param if you want it to > accept any password for this user. > > <domain name="freeswitch1.asnetinc.net"> > <user id="homebase"> > <params> > <param name="password" value="1234"/> > </params> > <variables> > <variable name="force-contact" value="nat-connectile-dysfunction"/> > </variables> > </user> > </domain> > > I ask that if you get this working you please document this on our wiki in exchange for this exhaustive description of what you need to do. > > > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale@hotmail.com > JABBER:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > iax:guest@conference.freeswitch.org/888 > googletalk:conf+888@conference.freeswitch.org > pstn:213-799-1400 > > > ----- Original Message ---- > From: Dave <lists+freeswitch(a)bigfatdave.com> > To: freeswitch-users(a)lists.freeswitch.org > Sent: Monday, May 21, 2007 6:49:00 PM > Subject: Re: [Freeswitch-users] Strange Reg > > Okay, I tried: > > recv 434 bytes from udp/[67.80.171.57]:48406 at 23:41:21.754015: > ------------------------------------------------------------------------ > REGISTER sip:freeswitch1.asnetinc.net SIP/2.0 > From: <sip:homebase@freeswitch1.asnetinc.net>;tag=a010033-13c4-108-40e02-46ec > To: <sip:homebase@freeswitch1.asnetinc.net> > Call-ID: a010033-13c4-108-40dfd-133e > CSeq: 1 REGISTER > Via: SIP/2.0/UDP 10.0.0.221:48406;branch=z9hG4bK-108-40e02-4ab5 > Max-Forwards: 70 > Supported: replaces > User-Agent: FXSO_GW > Contact: <sip:homebase@10.0.0.221:48406>;q=0.5 > Expires: 60 > Content-Length: 0 > > ------------------------------------------------------------------------ > 2007-05-21 23:41:21 [DEBUG] sofia_reg.c:387 sofia_reg_handle_register() Requesti > ng Registration from: [homebase(a)freeswitch1.asnetinc.net] > > ...and it responds with: > > send 640 bytes to udp/[67.80.171.57]:48406 at 23:41:21.840657: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 10.0.0.221:48406;branch=z9hG4bK-108-40e02-4ab5;received=67.8 > 0.171.57 > From: <sip:homebase@freeswitch1.asnetinc.net>;tag=a010033-13c4-108-40e02-46ec > To: <sip:homebase@freeswitch1.asnetinc.net>;tag=e1arjB5HrFcHF > Call-ID: a010033-13c4-108-40dfd-133e > CSeq: 1 REGISTER > User-Agent: FreeSWITCH(mod_sofia) > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, > REFER, UPDATE, REGISTER, INFO > Supported: 100rel, precondition > WWW-Authenticate: Digest realm="freeswitch1.asnetinc.net", nonce="119b66e2-04 > 43-49a9-980f-d5b150274ff6", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > > Any ideas as to what's wrong here? > > Thanks, > - Dave > > > On Mon, May 21, 2007 at 12:17:12PM -0700, Anthony Minessale wrote: > > ok how about this: > > > > try rev 5207 and do everything I described before only in the > > force-contact set it to nat-connectile-dysfunction > > > > <variable name="force-contact" value="nat-connectile-dysfunction"/> > > > > This will rewrite the host and ip of the contact to use the > > ip and port the req was received from. To benefit from this feature, you will probably need to set the ata to constantly register to maintain this dynamic reverse mapping created by the nat routers. Most of them will destroy any mappings with inactivity after a certain period of time so you must make your reg freq more often that that time period whatever it may be. > > > > > > > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale@hotmail.com > > JABBER:anthony.minessale@gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888@conference.freeswitch.org > > iax:guest@conference.freeswitch.org/888 > > googletalk:conf+888@conference.freeswitch.org > > pstn:213-799-1400 > > > > > > ----- Original Message ---- > > From: Dave <lists+freeswitch(a)bigfatdave.com> > > To: freeswitch-users(a)lists.freeswitch.org > > Sent: Monday, May 21, 2007 11:24:55 AM > > Subject: Re: [Freeswitch-users] Strange Reg > > > > Hi Anthony, > > > > I assume that means we can change the topolgy to something like this: > > ATA<->NAT<->Internet<->FS1 > > > > Basically, what we're doing here is telling FS1 how to find ATA. > > There are only two problems: > > 1) NAT is on DHCP. (I assume this can be worked around by using DynDNS > > instead of an IP.) > > 2) As I discovered while looking for the mystery 10.0.0.221 IP, NAT is > > actually two NATs, one behind the other, and I only have access to one > > of them. > > > > While you're at it, why don't you just make an option (<variable > > name='my-ata-sucks' value='true'/>) to totally break RFCs and mimic > > Asterisk's treatment? :-) > > > > Thanks a bunch for all your help, > > - Dave > > > > > > On Mon, May 21, 2007 at 07:25:01AM -0700, Anthony Minessale wrote: > > > Hi, > > > > > > You sound like a good candidate for my new experimental force contact feature. > > > > > > For this to work you need to use the directory for auth instead of the blind reg mode. > > > > > > *) Edit sofia.conf.xml and comment the blind reg option. > > > *) Edit directory.xml and add a <user> tag to the <domain name="$${domain}"> > > > section making sure $${domain} is also the name of the domain portion of > > > the sip's client registration. > > > > > > edit the file to look something like this and put in a absolute contact to use when this user registers regardless of what he advertises. This means that if you are using nat you should setup a permenant port forwarding for whatver uri you mention. > > > > > > e.g. > > > > > > Assuming your public ip is 66.2.2.2 and your phone is 10.0.0.2 > > > > > > add a rule to always port forward udp port 5555 of 66.2.2.2 to 10.0.0.2 5060 > > > then edit the following so the contact uri is user@66.2.2.2:5555 > > > > > > Whenever the phone registers with this user the specified contact will be added to the database instead of the real one this should make it possible for your phone to receive sip traffic even if it does not support STUN. > > > > > > <domain name="$${domain}"> > > > <user id="myuser"> > > > <params> > > > <param name="password" value="mypass"/> > > > </params> > > > <variables> > > > <variable name="force-contact" value="[TESTING] sip:myuser@66.2.2.2:5555"/> > > > </variables> > > > </user> > > > </domain> > > > > > > Anthony Minessale II > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > ClueCon http://www.cluecon.com/ > > > > > > AIM: anthm > > > MSN:anthony_minessale@hotmail.com > > > JABBER:anthony.minessale@gmail.com > > > IRC: irc.freenode.net #freeswitch > > > > > > FreeSWITCH Developer Conference > > > sip:888@conference.freeswitch.org > > > iax:guest@conference.freeswitch.org/888 > > > googletalk:conf+888@conference.freeswitch.org > > > pstn:213-799-1400 > > > > > > > > > ----- Original Message ---- > > > From: Dave <lists+freeswitch(a)bigfatdave.com> > > > To: FSwtch Usrs <freeswitch-users(a)lists.freeswitch.org> > > > Sent: Monday, May 21, 2007 2:20:53 AM > > > Subject: [Freeswitch-users] Strange Reg > > > > > > Hi guys, > > > > > > I've got a rather interesting situation, here. I've got an ATA > > > registering to a FreeSWITCH(TM) (FS2), both behind NAT. So far, so good. > > > Now, the FreeSWITCH(TM) registers to another FreeSWITCH(TM) outside the > > > NAT (FS2), and that's where the problems start. Here's the configuration: > > > > > > ATA<->FS2<->NAT<->Internet<->FS1 > > > > > > ATA: 10.1.0.51 (behind NAT) > > > FS2: 10.1.0.30 (behind NAT) > > > FS1: freeswitch1.asnetinc.net (64.118.94.19) > > > > > > Now, here's where it gets very interesting. This is what FS1 gets when > > > FS2 registers: > > > > From: [R352928(a)freeswitch1.asnetinc.net] > > > > Contact: ["user" <sip:R352928@10.0.0.221:39260>] > > > > Expires: [3600] > > > > nta_agent: tport: Bad message > > > > > > Now, where on Earth did 10.0.0.221 come from??? (I've actually hardcoded > > > both external_rtp_ip and server_name to the public IP of the NAT in the > > > FS2 config, but for some strange reason, FS2 apparently claims to be at > > > 10.0.0.221, which isn't even its local IP.) > > > > > > As if that's not interesting enough, I get this from FS1, too, when it > > > tries to connect to another SIP server on the public Internet: > > > > outbound(0xb4e1e1a0): detected NAT: freeswitch1.asnetinc.net != 64.118.94.19 > > > > > > Now, since a forward DNS resolution claims that the inequality is false, I > > > assume it attempted a reverse DNS lookup, which would fail (since my ISP > > > takes forever to update PTR records). > > > > > > As usual, any hints, tips, tricks, ideas, suggestions, flames, questions, > > > or (ideally) answers are greatly appreciated :-) > > > > > > Thank you, > > > - Dave > > > > > > _______________________________________________ > > > 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 > > > > > > > > > > > > > > > > > > > > > > > > > > > ____________________________________________________________________________________Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. > > > http://answers.yahoo.com/dir/?link=list&sid=396545433 > > > _______________________________________________ > > > 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 > > > > > > > > > > > > > > > > > > > > ____________________________________________________________________________________ > > Need Mail bonding? > > Go to the Yahoo! Mail Q&A for great tips from Yahoo! Answers users. > > http://answers.yahoo.com/dir/?link=list&sid=396546091 > > _______________________________________________ > > 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 > > > > > > > > > ____________________________________________________________________________________ > No need to miss a message. Get email on-the-go > with Yahoo! Mail for Mobile. Get started. > http://mobile.yahoo.com/mail > _______________________________________________ > 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 ____________________________________________________________________________________Need a vacation? Get great deals to amazing places on Yahoo! Travel. http://travel.yahoo.com/

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Re: [Freeswitch-users] Strange Reg
by Anthony Minessale 25 May '07

25 May '07

You need to make sure you setup the user directory correctly. http://wiki.freeswitch.org/wiki/FreeSwitch_XML_User_Directory_Guide My earlier email provided a precise example and warnings about making sure the tags etc match up. When you setup the <domain> tag in the directory you must make sure there is a domain tag that has the same name as the host portion of the to: header in the register. Using forensics from your traces here is the exact domain tag to use in directory.xml with 1 caveat: ----------------------------------------------------------------------- **replace the string 1234 with the password you specified in the ata** ----------------------------------------------------------------------- *NOTE* normally $${domain} would be ok here as the name assuming that the domain variable is set in freeswitch.xml but I put the exact domain name to avoid confusion. *ALSO NOTE* You can comment the password param if you want it to accept any password for this user. <domain name="freeswitch1.asnetinc.net"> <user id="homebase"> <params> <param name="password" value="1234"/> </params> <variables> <variable name="force-contact" value="nat-connectile-dysfunction"/> </variables> </user> </domain> I ask that if you get this working you please document this on our wiki in exchange for this exhaustive description of what you need to do. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Dave <lists+freeswitch(a)bigfatdave.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, May 21, 2007 6:49:00 PM Subject: Re: [Freeswitch-users] Strange Reg Okay, I tried: recv 434 bytes from udp/[67.80.171.57]:48406 at 23:41:21.754015: ------------------------------------------------------------------------ REGISTER sip:freeswitch1.asnetinc.net SIP/2.0 From: <sip:homebase@freeswitch1.asnetinc.net>;tag=a010033-13c4-108-40e02-46ec To: <sip:homebase@freeswitch1.asnetinc.net> Call-ID: a010033-13c4-108-40dfd-133e CSeq: 1 REGISTER Via: SIP/2.0/UDP 10.0.0.221:48406;branch=z9hG4bK-108-40e02-4ab5 Max-Forwards: 70 Supported: replaces User-Agent: FXSO_GW Contact: <sip:homebase@10.0.0.221:48406>;q=0.5 Expires: 60 Content-Length: 0 ------------------------------------------------------------------------ 2007-05-21 23:41:21 [DEBUG] sofia_reg.c:387 sofia_reg_handle_register() Requesti ng Registration from: [homebase(a)freeswitch1.asnetinc.net] ...and it responds with: send 640 bytes to udp/[67.80.171.57]:48406 at 23:41:21.840657: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.221:48406;branch=z9hG4bK-108-40e02-4ab5;received=67.8 0.171.57 From: <sip:homebase@freeswitch1.asnetinc.net>;tag=a010033-13c4-108-40e02-46ec To: <sip:homebase@freeswitch1.asnetinc.net>;tag=e1arjB5HrFcHF Call-ID: a010033-13c4-108-40dfd-133e CSeq: 1 REGISTER User-Agent: FreeSWITCH(mod_sofia) Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition WWW-Authenticate: Digest realm="freeswitch1.asnetinc.net", nonce="119b66e2-04 43-49a9-980f-d5b150274ff6", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ Any ideas as to what's wrong here? Thanks, - Dave On Mon, May 21, 2007 at 12:17:12PM -0700, Anthony Minessale wrote: > ok how about this: > > try rev 5207 and do everything I described before only in the > force-contact set it to nat-connectile-dysfunction > > <variable name="force-contact" value="nat-connectile-dysfunction"/> > > This will rewrite the host and ip of the contact to use the > ip and port the req was received from. To benefit from this feature, you will probably need to set the ata to constantly register to maintain this dynamic reverse mapping created by the nat routers. Most of them will destroy any mappings with inactivity after a certain period of time so you must make your reg freq more often that that time period whatever it may be. > > > > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale@hotmail.com > JABBER:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > iax:guest@conference.freeswitch.org/888 > googletalk:conf+888@conference.freeswitch.org > pstn:213-799-1400 > > > ----- Original Message ---- > From: Dave <lists+freeswitch(a)bigfatdave.com> > To: freeswitch-users(a)lists.freeswitch.org > Sent: Monday, May 21, 2007 11:24:55 AM > Subject: Re: [Freeswitch-users] Strange Reg > > Hi Anthony, > > I assume that means we can change the topolgy to something like this: > ATA<->NAT<->Internet<->FS1 > > Basically, what we're doing here is telling FS1 how to find ATA. > There are only two problems: > 1) NAT is on DHCP. (I assume this can be worked around by using DynDNS > instead of an IP.) > 2) As I discovered while looking for the mystery 10.0.0.221 IP, NAT is > actually two NATs, one behind the other, and I only have access to one > of them. > > While you're at it, why don't you just make an option (<variable > name='my-ata-sucks' value='true'/>) to totally break RFCs and mimic > Asterisk's treatment? :-) > > Thanks a bunch for all your help, > - Dave > > > On Mon, May 21, 2007 at 07:25:01AM -0700, Anthony Minessale wrote: > > Hi, > > > > You sound like a good candidate for my new experimental force contact feature. > > > > For this to work you need to use the directory for auth instead of the blind reg mode. > > > > *) Edit sofia.conf.xml and comment the blind reg option. > > *) Edit directory.xml and add a <user> tag to the <domain name="$${domain}"> > > section making sure $${domain} is also the name of the domain portion of > > the sip's client registration. > > > > edit the file to look something like this and put in a absolute contact to use when this user registers regardless of what he advertises. This means that if you are using nat you should setup a permenant port forwarding for whatver uri you mention. > > > > e.g. > > > > Assuming your public ip is 66.2.2.2 and your phone is 10.0.0.2 > > > > add a rule to always port forward udp port 5555 of 66.2.2.2 to 10.0.0.2 5060 > > then edit the following so the contact uri is user@66.2.2.2:5555 > > > > Whenever the phone registers with this user the specified contact will be added to the database instead of the real one this should make it possible for your phone to receive sip traffic even if it does not support STUN. > > > > <domain name="$${domain}"> > > <user id="myuser"> > > <params> > > <param name="password" value="mypass"/> > > </params> > > <variables> > > <variable name="force-contact" value="[TESTING] sip:myuser@66.2.2.2:5555"/> > > </variables> > > </user> > > </domain> > > > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale@hotmail.com > > JABBER:anthony.minessale@gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888@conference.freeswitch.org > > iax:guest@conference.freeswitch.org/888 > > googletalk:conf+888@conference.freeswitch.org > > pstn:213-799-1400 > > > > > > ----- Original Message ---- > > From: Dave <lists+freeswitch(a)bigfatdave.com> > > To: FSwtch Usrs <freeswitch-users(a)lists.freeswitch.org> > > Sent: Monday, May 21, 2007 2:20:53 AM > > Subject: [Freeswitch-users] Strange Reg > > > > Hi guys, > > > > I've got a rather interesting situation, here. I've got an ATA > > registering to a FreeSWITCH(TM) (FS2), both behind NAT. So far, so good. > > Now, the FreeSWITCH(TM) registers to another FreeSWITCH(TM) outside the > > NAT (FS2), and that's where the problems start. Here's the configuration: > > > > ATA<->FS2<->NAT<->Internet<->FS1 > > > > ATA: 10.1.0.51 (behind NAT) > > FS2: 10.1.0.30 (behind NAT) > > FS1: freeswitch1.asnetinc.net (64.118.94.19) > > > > Now, here's where it gets very interesting. This is what FS1 gets when > > FS2 registers: > > > From: [R352928(a)freeswitch1.asnetinc.net] > > > Contact: ["user" <sip:R352928@10.0.0.221:39260>] > > > Expires: [3600] > > > nta_agent: tport: Bad message > > > > Now, where on Earth did 10.0.0.221 come from??? (I've actually hardcoded > > both external_rtp_ip and server_name to the public IP of the NAT in the > > FS2 config, but for some strange reason, FS2 apparently claims to be at > > 10.0.0.221, which isn't even its local IP.) > > > > As if that's not interesting enough, I get this from FS1, too, when it > > tries to connect to another SIP server on the public Internet: > > > outbound(0xb4e1e1a0): detected NAT: freeswitch1.asnetinc.net != 64.118.94.19 > > > > Now, since a forward DNS resolution claims that the inequality is false, I > > assume it attempted a reverse DNS lookup, which would fail (since my ISP > > takes forever to update PTR records). > > > > As usual, any hints, tips, tricks, ideas, suggestions, flames, questions, > > or (ideally) answers are greatly appreciated :-) > > > > Thank you, > > - Dave > > > > _______________________________________________ > > 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 > > > > > > > > > > > > > > > > > > ____________________________________________________________________________________Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. > > http://answers.yahoo.com/dir/?link=list&sid=396545433 > > _______________________________________________ > > 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 > > > > > > > > > > ____________________________________________________________________________________ > Need Mail bonding? > Go to the Yahoo! Mail Q&A for great tips from Yahoo! Answers users. > http://answers.yahoo.com/dir/?link=list&sid=396546091 > _______________________________________________ > 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 ____________________________________________________________________________________ No need to miss a message. Get email on-the-go with Yahoo! Mail for Mobile. Get started. http://mobile.yahoo.com/mail

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pastebin
by Brian West 25 May '07

25 May '07

Can someone that is good with php provide me with a patched version of pastebin that has captcha support .. I have had to do some drastic things to stop spammers. I'm not that great with php but someone out there can whip up a patched version ? /b

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Re: [Freeswitch-users] Does FS handle remote NATed phones?
by Anthony Minessale 24 May '07

24 May '07

thanks =D Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Jason Garland <jgarland(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, May 24, 2007 12:05:26 PM Subject: Re: [Freeswitch-users] Does FS handle remote NATed phones? http://wiki.freeswitch.org/wiki/Talk:Event_Socket http://wiki.freeswitch.org/wiki/Event_Socket On 5/24/07, Anthony Minessale <anthmct(a)yahoo.com> wrote: > > We do have mod_event_socket's "socket" app which is similar to the network > based fast-agi (FAGI). This application makes an outbound tcp connection to > the specified ip:port and the other end can control the call in similar ways > to an agi only the dialect is much different and it supports async > operations e.g. telling the call to play foo.wav and getting an immediate > return to listen for dtmf as the file plays rather than block till the file > is over. > > there is an example perl script in tree: > > "scripts/socket/sock.pl" > > The data syntax from the dialplan is > <ip>:<port> [<keywords>] > > keywords: > > async: > > The "async" keyword indicates that all commands will return instantly making > it possible to monitor the socket for events etc while the > stack of commands are executing. > > If the async keyword is absent then all calls will block until the > command has finished. > > full: > > The "full" keyword indicates that the other end will have the full command > set for event_socket this is the same command set an inbound connection to > event_socket has so you can execute api commands, get global events etc. > > If the "full" keyword is absent the command set and events are limited > to that paticular call > > Here are examples of how to use it in the dialplan. > > <action application="socket" data="127.0.0.1:8084"/> > <action application="socket" data="127.0.0.1:8084 async"/> > <action application="socket" data="127.0.0.1:8084 full"/> > <action application="socket" data="127.0.0.1:8084 async full"/> > > > I think if wasim's angel want's to implement it a good idea > is to write it as a separate application and use this interface > to control the call then let the forked processes in the remote > application provide the AGI emulation. > > We have someone working on a C library to use as an event socket library > also. > but in the meantime there is also a crude C example with no client library > but just inline commands in "scripts/socket/socket2me" this small C program > demonstrates how you can use the interface to not only control the call but > to also request a 2 way media stream and process the data. This application > uses spandsp to implement a fax send/recv similar to asterisk's rxfax txfax > only over the loopback interface. > > > > /me hopes somone makes a wiki page out of this info =D > > > > > > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale@hotmail.com > JABBER:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > iax:guest@conference.freeswitch.org/888 > googletalk:conf+888@conference.freeswitch.org > pstn:213-799-1400 > > > ----- Original Message ---- > From: Wasim Baig <wasim(a)convergence.pk> > To: freeswitch-users(a)lists.freeswitch.org > Sent: Thursday, May 24, 2007 7:13:24 AM > Subject: Re: [Freeswitch-users] Does FS handle remote NATed phones? > > On 5/24/07, Fred <codecomplete(a)free.fr> wrote: > > > Hopefully, the AGI interface works the same, so that existing scripts will > > still work in FS. > > nope, not as yet, but i'm convincing an angel to write up a mod_agi that > would allow using most if not all of the AGI functions in freeswitch > (atleast SetVar and GetVar, if not Dial) > > imho thats one of the main bottlenecks for people who have existing > asterisk deployments on AGI from switching to FS, lets see if we can get > some good news on this shortly > > -- > wasim h. baig | principal consultant | convergence pk | +92 300 8508070 > _______________________________________________ > 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 > > > ________________________________ > Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. > _______________________________________________ > 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 ____________________________________________________________________________________ Don't pick lemons. See all the new 2007 cars at Yahoo! Autos. http://autos.yahoo.com/new_cars.html

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Re: [Freeswitch-users] Does FS handle remote NATed phones?
by Anthony Minessale 24 May '07

24 May '07

We do have mod_event_socket's "socket" app which is similar to the network based fast-agi (FAGI). This application makes an outbound tcp connection to the specified ip:port and the other end can control the call in similar ways to an agi only the dialect is much different and it supports async operations e.g. telling the call to play foo.wav and getting an immediate return to listen for dtmf as the file plays rather than block till the file is over. there is an example perl script in tree: "scripts/socket/sock.pl" The data syntax from the dialplan is <ip>:<port> [<keywords>] keywords: async: The "async" keyword indicates that all commands will return instantly making it possible to monitor the socket for events etc while the stack of commands are executing. If the async keyword is absent then all calls will block until the command has finished. full: The "full" keyword indicates that the other end will have the full command set for event_socket this is the same command set an inbound connection to event_socket has so you can execute api commands, get global events etc. If the "full" keyword is absent the command set and events are limited to that paticular call Here are examples of how to use it in the dialplan. <action application="socket" data="127.0.0.1:8084"/> <action application="socket" data="127.0.0.1:8084 async"/> <action application="socket" data="127.0.0.1:8084 full"/> <action application="socket" data="127.0.0.1:8084 async full"/> I think if wasim's angel want's to implement it a good idea is to write it as a separate application and use this interface to control the call then let the forked processes in the remote application provide the AGI emulation. We have someone working on a C library to use as an event socket library also. but in the meantime there is also a crude C example with no client library but just inline commands in "scripts/socket/socket2me" this small C program demonstrates how you can use the interface to not only control the call but to also request a 2 way media stream and process the data. This application uses spandsp to implement a fax send/recv similar to asterisk's rxfax txfax only over the loopback interface. /me hopes somone makes a wiki page out of this info =D Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Wasim Baig <wasim(a)convergence.pk> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, May 24, 2007 7:13:24 AM Subject: Re: [Freeswitch-users] Does FS handle remote NATed phones? On 5/24/07, Fred <codecomplete(a)free.fr> wrote: Hopefully, the AGI interface works the same, so that existing scripts will still work in FS. nope, not as yet, but i'm convincing an angel to write up a mod_agi that would allow using most if not all of the AGI functions in freeswitch (atleast SetVar and GetVar, if not Dial) imho thats one of the main bottlenecks for people who have existing asterisk deployments on AGI from switching to FS, lets see if we can get some good news on this shortly -- wasim h. baig | principal consultant | convergence pk | +92 300 8508070 _______________________________________________ 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 ____________________________________________________________________________________Get the Yahoo! toolbar and be alerted to new email wherever you're surfing. http://new.toolbar.yahoo.com/toolbar/features/mail/index.php

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Re: [Freeswitch-users] Does FS handle remote NATed phones?
by Fred 24 May '07

24 May '07

At 12:00 23/05/2007 -0400, Anthony Minessale wrote: >We are essentially in BETA right now for the first release so we would >welcome your input as a tester. I got Asterisk to work, but I'll look into Freeswitch when I have a bit of time, as Asterisk seems one big pile of stuff that could use some overhaul. Hopefully, the AGI interface works the same, so that existing scripts will still work in FS. Cheers Fred.

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Re: [Freeswitch-users] Does FS handle remote NATed phones?
by Anthony Minessale 23 May '07

23 May '07

We are essentially in BETA right now for the first release so we would welcome your input as a tester. FreeSWITCH is more than ready to move SIP traffic, all I would warn is that some of the configuration and behaviors therein have not been finalized and are hence subject to change between now and the final release but there probably aren't many. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Fred <codecomplete(a)free.fr> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, May 22, 2007 2:52:12 PM Subject: [Freeswitch-users] Does FS handle remote NATed phones? Hello It's been over a week now, and I still can't get voice RX/TX to work between a Linksys VoIP gateway and a remote IP phone with Asterisk as the PBX: http://codecomplete.free.fr/3102_nat/voip_no_sound_open_ports.jpg The phone rings, but no sound either way. I'm beginning to wonder if Asterisk can handle being behind a NAT, and bridge calls to remote phones that are themselves behind a NAT :-/ If not, is FreeSwitch the solution because of its support for STUN? Is it ready for production, or is it a bit early yet? Thank you Fred. _______________________________________________ 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 ____________________________________________________________________________________Fussy? Opinionated? Impossible to please? Perfect. Join Yahoo!'s user panel and lay it on us. http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7

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Does FS handle remote NATed phones?
by Fred 22 May '07

22 May '07

Hello It's been over a week now, and I still can't get voice RX/TX to work between a Linksys VoIP gateway and a remote IP phone with Asterisk as the PBX: http://codecomplete.free.fr/3102_nat/voip_no_sound_open_ports.jpg The phone rings, but no sound either way. I'm beginning to wonder if Asterisk can handle being behind a NAT, and bridge calls to remote phones that are themselves behind a NAT :-/ If not, is FreeSwitch the solution because of its support for STUN? Is it ready for production, or is it a bit early yet? Thank you Fred.

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