September 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Sporadic outbound call drops and dynamic IP address updates
by Tim Bock 18 Sep '13

18 Sep '13

Hi, Running freeswitch 1.2.12, built on Debian 7.0. My ext-rtp-ip and ext-sip-ip are set to the freeswitch stun server. The DSL modem has a dynamic IP address, and I'm not currently using a dynamic dns updater (switching to a static IP in a few weeks). I've been having a sporadic issue where outbound calls only last for 25 seconds or so, and are then dropped. The Nexvortex tech said that it was a problem with my FS pbx--that it was ignoring the Contact header and continuing to send to the proxy (66.23.129.253 in this example). I initially thought it might be some problem with the 1.3 version of freeswitch I was using, so I rebuilt using the latest stable (described above). The problem seemed to go away...for a day or so. It was back again today, and no changes were made in the interim. While I was stumbling through trying to figure out what was wrong, magically the outbound calls started behaving again. Again, I didn't touch the configuration...I was just making calls and trying to make sense of the debug info I was seeing (via wireshark and sip trace). I am fairly new to debugging SIP at this level. Below are some excerpts from the sip trace...this was a *successful* outbound call (no drop after 25 seconds). My question is: Is there any merit to what the tech said? I see in this sequence that NV sets the Contact header, but FS continues to send to the 66.23 address, though I also see in the trace that the Contact address is contained in the FS response. I also notice NV seems to be using Sonus, though I didn't see anything on the "RTP Issues" page which seemed relevant. Does anyone have any thoughts about why sometimes everything works ok and other times it doesn't? I have a second FS box (identical platform) registered with a different provider...zero problems. A second, but less pressing question: does FS automatically handle changes to a dynamic external IP address? As I'm using stun, it seems like it should, but the modem received a new address midday a few days ago, and I could no longer receive calls. A "sofia status" revealed that the external profile still showed the old external IP (and this was several hours later). I just restarted FS and everything was fine, but was wondering if I have some config parameter wrong somewhere that it didn't update on its own. Thanks for any help... Tim I fuzzed my phone #s, IPs, and some Proxy Auth info, but everything else should be as logged. recv 874 bytes from udp/[66.23.129.253]:5060 at 02:57:16.514012: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.0.2:5080;rport=5080;branch=z9hG4bKF9DB405pNct2S Record-Route: <sip:66.23.129.254:5060;lr>,<sip:15136852816@66.23.129.253:5060;nat=y es;ftag=r011rpa0Syg0S;lr=on> From: "Kitchen Angels" <sip:15054717780@px3.nexvortex.com>;tag=r011rpa0Syg0S To: <sip:15136852816@px3.nexvortex.com>;tag=gK07d8165c Call-ID: dcb948b2-9ab0-1231-c297-001cc0382911 CSeq: 49400454 INVITE Contact: <sip:72.15.219.140;did=27d.e851d0a3> Allow: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Content-Length: 235 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 17065 29458 IN IP4 208.79.54.210 s=SIP Media Capabilities c=IN IP4 208.79.54.212 t=0 0 m=audio 18928 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 recv 1034 bytes from udp/[66.23.129.253]:5060 at 02:57:21.804773: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.2:5080;rport=5080;branch=z9hG4bKF9DB405pNct2S Record-Route: <sip:66.23.129.254:5060;lr>,<sip:15136852816@66.23.129.253:5060;nat=y es;ftag=r011rpa0Syg0S;lr=on> From: "Bob" <sip:15136857780@px3.nexvortex.com>;tag=r011rpa0Syg0S To: <sip:15136852816@px3.nexvortex.com>;tag=gK07d8165c Call-ID: dcb948b2-9ab0-1231-c297-001cc0382911 CSeq: 49400454 INVITE Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay , multipart/mixed Contact: <sip:72.15.219.140;did=27d.e851d0a3> Allow: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Require: timer Supported: timer Session-Expires: 1800;refresher=uac Content-Length: 235 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 17065 29458 IN IP4 208.79.54.210 s=SIP Media Capabilities c=IN IP4 208.79.54.212 t=0 0 m=audio 18928 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 send 774 bytes to udp/[66.23.129.253]:5060 at 02:57:21.811091: ------------------------------------------------------------------------ ACK sip:72.15.219.140;did=27d.e851d0a3 SIP/2.0 Via: SIP/2.0/UDP 175.29.62.55:5080;rport;branch=z9hG4bKgj735UptjNgNN Route: <sip:15136852816@66.23.129.253:5060;nat=yes;ftag=r011rpa0Syg0S;lr=on> Route: <sip:66.23.129.254:5060;lr> Max-Forwards: 70 From: "Bob" <sip:15136857780@px3.nexvortex.com>;tag=r011rpa0Syg0S To: <sip:15136852816@px3.nexvortex.com>;tag=gK07d8165c Call-ID: dcb948b2-9ab0-1231-c297-001cc0382911 CSeq: 49400454 ACK Contact: <sip:gw+nexvortex-main@175.29.62.55:5080;transport=udp;gw=nexvortex-main> Proxy-Authorization: Digest username="dummy", realm="nexvortex.com", nonce="U jkXvFI5FpDt+8Hzk/viti", algorithm=MD5, uri="sip:15136852816@px3.nexvortex.co m", response="20200d7ccb42e873a3f4e16c8f9d9de9" Content-Length: 0 send 1033 bytes to udp/[66.23.129.253]:5060 at 02:58:13.951291: ------------------------------------------------------------------------ BYE sip:72.15.219.140;did=27d.e851d0a3 SIP/2.0 Via: SIP/2.0/UDP 175.29.62.55:5080;rport;branch=z9hG4bKHU0v7p7XFy67g Route: <sip:15136852816@66.23.129.253:5060;nat=yes;ftag=r011rpa0Syg0S;lr=on> Route: <sip:66.23.129.254:5060;lr> Max-Forwards: 70 From: "Bob" <sip:15136857780@px3.nexvortex.com>;tag=r011rpa0Syg0S To: <sip:15136852816@px3.nexvortex.com>;tag=gK07d8165c Call-ID: dcb948b2-9ab0-1231-c297-001cc0382911 CSeq: 49400455 BYE Contact: <sip:gw+nexvortex-main@175.29.62.55:5080;transport=udp;gw=nexvortex-main> User-Agent: FreeSWITCH-mod_sofia/1.2.12+git~20130826T225725Z~4990a541bf Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, N OTIFY Supported: timer, precondition, path, replaces Proxy-Authorization: Digest username="dummy", realm="nexvortex.com", nonce="U jkXvFI5FpDt+8Hzk/viti", algorithm=MD5, uri="sip:72.15.219.140;did=27d.e851d0 a3", response="3561a42f5bfa281fad3f39077534732d" Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 -- Tim Bock Synacktics, LLC www.synacktics.com

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inherit_codec failing with pre-media
by sangdrax8 17 Sep '13

17 Sep '13

I pulled the latest freeswitch from git this morning, and am still having issues using inherit_codec=true. It seems the only way I can get it to defer to the choice of the b-leg call is to remove the ringback, and thus the need for early media. My setup has 2 phones with the following codecs configured: phone1000: g722, ulaw phone1001: ulaw My goal is to have a call initiated by phone1000 (a-leg) to defer the codec choice until phone1001 (b-leg) chooses one. This would allow them to talk with no transcoding required by freeswitch. With the following dial plan, this fails to work. <extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="export" data="dialed_extension=$1"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="inherit_codec=true" /> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> </condition> </extension> It does log that it is going to defer: 2013-09-17 13:15:10.956535 [DEBUG] mod_sofia.c:4542 [zrtp_passthru] Setting a-leg inherit_codec=true 2013-09-17 13:15:10.956535 [DEBUG] mod_sofia.c:4545 [zrtp_passthru] Setting b-leg absolute_codec_string='G722@8000h@20i@64000b,PCMU@8000h@20i@64000b' then it tries to provide the ringback for the a-leg, and it negotiates to g722 since freeswitch supports that and it is the first match. Later when the b-leg is answered, it also negotiates, this time to ulaw as that is all b supports. My call completes, but freeswitch has to transcode between the two. In an attempt to see inherit_codec work, I then commented out the ringback line from the above dialplan. With only this one change, freeswitch now waits until the b-leg negotiation, and passes that back to the a-leg. The call completes and both handsets are using ulaw and no transcoding is required. The problem being that the calling party never gets a ring while waiting. I can't imagine that the only way to use inherit_codec=true is with all ringing turned off. Can someone tell me if there is an additional option I am missing? All documentation I read seems to indicate that just adding "inherit_codec=true" will fix things, but even the default.xml provided as an example uses ringback and therefore fails.

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how to encrypt the sip message?
by 彭勇 17 Sep '13

17 Sep '13

one of my sip gateway using a encrypted SIP message. char * sip_encrypt(char* str) { ... return encrypted_str; } i should send encrypted to the gateway. the gateway also send me a encrypted SIP message, and i should decrypt it. then how can i do it in Freeswitch? where can i code insert sip_encrypt()? thanks -- Peng Yong

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Re: [Freeswitch-users] NAT issue from client
by Lloyd Aloysius 17 Sep '13

17 Sep '13

Set the following in the user vars for the 1001 and reload and test https://wiki.freeswitch.org/wiki/NAT_Traversal sip_force_contact =NDLB-connectile-dysfunction NDLB-received-in-nat-reg-contact = true * * On Tue, Sep 17, 2013 at 12:29 PM, ThanhTruong <ngoc217thanh(a)yahoo.com>wrote: > This is log in FS_CLI when i get siptrace and log level. > http://pastebin.freeswitch.org/21412 > > my router is Smartel network, this is as another router and open for all. > I have tried with many sip server as asterisk, and it works fine. > > :( > > Thanks, > > ThanhTruong > > ------------------------------ > *From:* Lloyd Aloysius <lloyd.aloysius(a)gmail.com> > > *To:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > *Sent:* Tuesday, September 17, 2013 10:55 PM > > *Subject:* Re: [Freeswitch-users] NAT issue from client > > sofia global siptrace on and capture the log and post in paste bin. > > What is the made/model of the router? > * > * > > > On Tue, Sep 17, 2013 at 11:22 AM, ThanhTruong <ngoc217thanh(a)yahoo.com>wrote: > > Hi All, > > I have tried to build new server with GIT version, and current version of > FS. > > Then, I have disable SIP ALG in my router too. > > I have make a full test call from 1011 to 1012 to check with this issue, > used ngrep to capture all package send and receive. > > http://pastebin.freeswitch.org/21411 > > my FS ip is 123.123.123.123. > > The call drop after 30 seconds(to 32 seconds). > > I see that when FS get 200OK, it send ACK to 1011, but it keep get 200OK > from client. > > Hope some one can help me on this case. > > my dialplan : > > <extension name="public_extensions"> > <condition field="destination_number" expression="(\d+)$" > > <action application="set" data="hangup_after_bridge=true"/> > <action application="bridge" > data="sofia/internal/${destination_number}@${domain_name}"/> > </condition> > </extension> > > Thank all in advance. > > ThanhTruong > > ------------------------------ > *From:* Cal Leeming [Simplicity Media Ltd] < > cal.leeming(a)simplicitymedialtd.co.uk> > > *To:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > *Sent:* Tuesday, September 17, 2013 5:14 PM > > *Subject:* Re: [Freeswitch-users] NAT issue from client > > Many routers/phones have problems because of their loose interpretation of > SIP. > > So, here's what you need to do; > > * Perform a test call reproducing the problem from start to finish, > collect as many debug logs as possible, see [1], [2], [3] > * Ensure to enable SIP trace in [3] > * Attempt to find the problem yourself, if you're still having a problem > then pastebin the results [4] > > I cannot stress how often this problem comes up, and you are unlikely to > get a response unless you have clearly shown to be doing your homework. > > Please make sure you include (in detail) what you have attempted to do to > fix the problem, what your own thoughts are, and as much info as possible. > > Hope this helps > > Cal > > [1] https://wiki.freeswitch.org/wiki/Debugging_Freeswitch > [2] http://wiki.freeswitch.org/wiki/Troubleshooting_Freeswitch > [3] http://wiki.freeswitch.org/wiki/Sofia-SIP#Debugging_Sofia-SIP > [4] http://wiki.freeswitch.org/wiki/Reporting_Bugs#Using_The_Pastebin > > > On Tue, Sep 17, 2013 at 5:32 AM, ThanhTruong <ngoc217thanh(a)yahoo.com>wrote: > > HI all, > > In my case, my FS server is in public IP. and I see that I am getting nat > in client (sip phone) side. > I have tried with sturn server, but no luck, I have checked contact detail > in database and see that it can get external IP correct. > > I have read much, tried more than 2 weeks with this, but cant get it works > :( > > That is a bad situation for me. > :( > ThanhTruong > > ------------------------------ > *From:* Paul <pasha(a)prosperity4ever.com> > *To:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > *Cc:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > *Sent:* Tuesday, September 17, 2013 12:40 AM > *Subject:* Re: [Freeswitch-users] NAT issue from client > > Edit your public sip profile xml and make sure that your internal ip of > the server and external ip of the server (for both rtp and sip) are set > properly. That should fix it. > > Paul > > On Mon, 16 Sep, 2013 at 10:02 AM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming(a)simplicitymedialtd.co.uk> wrote: > > Hello, > > You'll see this has been raised many times already, a quick google search > shows; > > https://www.google.co.uk/search?q=freeswitch+30+seconds > > http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073842.html > http://jira.freeswitch.org/browse/FS-3496 > > http://freeswitch-users.2379917.n2.nabble.com/NAT-issues-Outbound-Call-drop… > > http://freeswitch-users.2379917.n2.nabble.com/Re-call-disconnects-after-32-… > > Please review all those answers and let us know if it resolves your > problem. > > If not then please update the ticket with some debug logs (remember to > paste into pastebin, not direct into the ticket) > https://wiki.freeswitch.org/wiki/Debugging_Freeswitch > http://wiki.freeswitch.org/wiki/Troubleshooting_Freeswitch > > Hope this helps > > Cal > > > On Mon, Sep 16, 2013 at 5:48 PM, ThanhTruong <ngoc217thanh(a)yahoo.com>wrote: > > Hi all, > > I have built basic FS server with default sip account 1011 & 1012. > I have registered 2 this accounts with xlite, portgo. them in NAT via my > router. > When I make call, call connect and drop after 30 seconds. > > my dialplan is: > > <extension name="public_extensions"> > <condition field="destination_number" expression="(\d+)$" > > <action application="set" data="hangup_after_bridge=true"/> > <action application="bridge" > data="sofia/internal/${destination_number}@${domain_name}"/> > </condition> > > > I have tried many configure but not success. > PLease help to fix this issue. > > I am veny new in this system and tried many cases to fix it. > > Thank all for suggesttion. > > ThanhTruong > > _________________________________________________________________________ > 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 > > > > _________________________________________________________________________ > 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 > > > > _________________________________________________________________________ > 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 > > > > _________________________________________________________________________ > 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 > > > > _________________________________________________________________________ > 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 > > > > _________________________________________________________________________ > 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 > > > > _________________________________________________________________________ > 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 > >

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Re: [Freeswitch-users] NAT issue from client
by Lloyd Aloysius 17 Sep '13

17 Sep '13

sofia global siptrace on and capture the log and post in paste bin. What is the made/model of the router? * * On Tue, Sep 17, 2013 at 11:22 AM, ThanhTruong <ngoc217thanh(a)yahoo.com>wrote: > Hi All, > > I have tried to build new server with GIT version, and current version of > FS. > > Then, I have disable SIP ALG in my router too. > > I have make a full test call from 1011 to 1012 to check with this issue, > used ngrep to capture all package send and receive. > > http://pastebin.freeswitch.org/21411 > > my FS ip is 123.123.123.123. > > The call drop after 30 seconds(to 32 seconds). > > I see that when FS get 200OK, it send ACK to 1011, but it keep get 200OK > from client. > > Hope some one can help me on this case. > > my dialplan : > > <extension name="public_extensions"> > <condition field="destination_number" expression="(\d+)$" > > <action application="set" data="hangup_after_bridge=true"/> > <action application="bridge" > data="sofia/internal/${destination_number}@${domain_name}"/> > </condition> > </extension> > > Thank all in advance. > > ThanhTruong > > ------------------------------ > *From:* Cal Leeming [Simplicity Media Ltd] < > cal.leeming(a)simplicitymedialtd.co.uk> > > *To:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > *Sent:* Tuesday, September 17, 2013 5:14 PM > > *Subject:* Re: [Freeswitch-users] NAT issue from client > > Many routers/phones have problems because of their loose interpretation of > SIP. > > So, here's what you need to do; > > * Perform a test call reproducing the problem from start to finish, > collect as many debug logs as possible, see [1], [2], [3] > * Ensure to enable SIP trace in [3] > * Attempt to find the problem yourself, if you're still having a problem > then pastebin the results [4] > > I cannot stress how often this problem comes up, and you are unlikely to > get a response unless you have clearly shown to be doing your homework. > > Please make sure you include (in detail) what you have attempted to do to > fix the problem, what your own thoughts are, and as much info as possible. > > Hope this helps > > Cal > > [1] https://wiki.freeswitch.org/wiki/Debugging_Freeswitch > [2] http://wiki.freeswitch.org/wiki/Troubleshooting_Freeswitch > [3] http://wiki.freeswitch.org/wiki/Sofia-SIP#Debugging_Sofia-SIP > [4] http://wiki.freeswitch.org/wiki/Reporting_Bugs#Using_The_Pastebin > > > On Tue, Sep 17, 2013 at 5:32 AM, ThanhTruong <ngoc217thanh(a)yahoo.com>wrote: > > HI all, > > In my case, my FS server is in public IP. and I see that I am getting nat > in client (sip phone) side. > I have tried with sturn server, but no luck, I have checked contact detail > in database and see that it can get external IP correct. > > I have read much, tried more than 2 weeks with this, but cant get it works > :( > > That is a bad situation for me. > :( > ThanhTruong > > ------------------------------ > *From:* Paul <pasha(a)prosperity4ever.com> > *To:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > *Cc:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > *Sent:* Tuesday, September 17, 2013 12:40 AM > *Subject:* Re: [Freeswitch-users] NAT issue from client > > Edit your public sip profile xml and make sure that your internal ip of > the server and external ip of the server (for both rtp and sip) are set > properly. That should fix it. > > Paul > > On Mon, 16 Sep, 2013 at 10:02 AM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming(a)simplicitymedialtd.co.uk> wrote: > > Hello, > > You'll see this has been raised many times already, a quick google search > shows; > > https://www.google.co.uk/search?q=freeswitch+30+seconds > > http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073842.html > http://jira.freeswitch.org/browse/FS-3496 > > http://freeswitch-users.2379917.n2.nabble.com/NAT-issues-Outbound-Call-drop… > > http://freeswitch-users.2379917.n2.nabble.com/Re-call-disconnects-after-32-… > > Please review all those answers and let us know if it resolves your > problem. > > If not then please update the ticket with some debug logs (remember to > paste into pastebin, not direct into the ticket) > https://wiki.freeswitch.org/wiki/Debugging_Freeswitch > http://wiki.freeswitch.org/wiki/Troubleshooting_Freeswitch > > Hope this helps > > Cal > > > On Mon, Sep 16, 2013 at 5:48 PM, ThanhTruong <ngoc217thanh(a)yahoo.com>wrote: > > Hi all, > > I have built basic FS server with default sip account 1011 & 1012. > I have registered 2 this accounts with xlite, portgo. them in NAT via my > router. > When I make call, call connect and drop after 30 seconds. > > my dialplan is: > > <extension name="public_extensions"> > <condition field="destination_number" expression="(\d+)$" > > <action application="set" data="hangup_after_bridge=true"/> > <action application="bridge" > data="sofia/internal/${destination_number}@${domain_name}"/> > </condition> > > > I have tried many configure but not success. > PLease help to fix this issue. > > I am veny new in this system and tried many cases to fix it. > > Thank all for suggesttion. > > ThanhTruong > > _________________________________________________________________________ > 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 > > > > _________________________________________________________________________ > 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 > > > > _________________________________________________________________________ > 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 > > > > _________________________________________________________________________ > 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 > > > > _________________________________________________________________________ > 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 > >

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Record file not save when call is disconnect in session:recordFile
by Sanath Prasanna 17 Sep '13

17 Sep '13

Hi, I used LUA for voicemail (vm) system. I want to save message & send message to end user about vm notification, even user disconnect call by himself when recording message. Currently if user disconnect within recording, file is not save & notification is not going. Pls help to solve that. AheS. Robert code: session:recordFile(vm_file_namCurrentle, vm_record_duration, vm_record_duration, 0); -- pressing # ends the recording

2 2

Index files failing to download while updating package repositories on my raspberry pi running Debian Wheezy
by Pranav Lal 17 Sep '13

17 Sep '13

Hi all, I am trying to download freeswitch on to my raspberry pi running debian Wheezy.. I am following the instructions for debian at http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Upgrading_an d_Re-installation Add the FreeSWITCH APT repository: (for Debian Wheezy) echo 'deb http://files.freeswitch.org/repo/deb/debian/ wheezy main' >> /etc/apt/sources.list.d/freeswitch.list PL] ok. Import the repo signing key curl http://files.freeswitch.org/repo/deb/debian/freeswitch_archive_g0.pub | apt-key add - PL] This process was flawless. However, when I did a apt-get update, I got the following error. Hit http://archive.raspberrypi.org wheezy Release.gpg Hit http://raspberrypi.collabora.com wheezy Release.gpg Get:1 http://files.freeswitch.org wheezy Release.gpg [198 B] Hit http://raspberrypi.collabora.com wheezy Release Hit http://archive.raspberrypi.org wheezy Release Get:2 http://mirrordirector.raspbian.org wheezy Release.gpg [490 B] Get:3 http://files.freeswitch.org wheezy Release [3,221 B] Get:4 http://mirrordirector.raspbian.org wheezy Release [14.4 kB] Hit http://raspberrypi.collabora.com wheezy/rpi armhf Packages Hit http://archive.raspberrypi.org wheezy/main armhf Packages Get:5 http://mirrordirector.raspbian.org wheezy/main armhf Packages [7,414 kB] Ign http://raspberrypi.collabora.com wheezy/rpi Translation-en_GB Ign http://archive.raspberrypi.org wheezy/main Translation-en_GB Ign http://raspberrypi.collabora.com wheezy/rpi Translation-en Ign http://archive.raspberrypi.org wheezy/main Translation-en Hit http://mirrordirector.raspbian.org wheezy/contrib armhf Packages Hit http://mirrordirector.raspbian.org wheezy/non-free armhf Packages Hit http://mirrordirector.raspbian.org wheezy/rpi armhf Packages Ign http://mirrordirector.raspbian.org wheezy/contrib Translation-en_GB Ign http://mirrordirector.raspbian.org wheezy/contrib Translation-en Ign http://mirrordirector.raspbian.org wheezy/main Translation-en_GB Ign http://mirrordirector.raspbian.org wheezy/main Translation-en Ign http://mirrordirector.raspbian.org wheezy/non-free Translation-en_GB Ign http://mirrordirector.raspbian.org wheezy/non-free Translation-en Ign http://mirrordirector.raspbian.org wheezy/rpi Translation-en_GB Ign http://mirrordirector.raspbian.org wheezy/rpi Translation-en Fetched 7,432 kB in 16min 49s (7,358 B/s) W: Failed to fetch http://files.freeswitch.org/repo/deb/debian/dists/wheezy/Release Unabl e to find expected entry 'main/binary-armhf/Packages' in Release file (Wrong sources.list entry or malformed file) E: Some index files failed to download. They have been ignored, or old ones used instead. root@raspberrypi:/usr/local# Should I continue with the installation or should I stick to building from source? I plan to use freeswitch as a PBX for my house where I will connect a number of mobile phones using SIP. Pranav

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Re: [Freeswitch-users] NAT issue from client
by Lloyd Aloysius 17 Sep '13

17 Sep '13

Thanh, What is the made and model of your router. My recommendation is capture the trace and find the reason for the problem. Your problem is related to the network and not freeswitch. If the router have SIP ALG. Disable SIP ALG and try. Lloyd On Tue, Sep 17, 2013 at 6:14 AM, Cal Leeming [Simplicity Media Ltd] < cal.leeming(a)simplicitymedialtd.co.uk> wrote: > Many routers/phones have problems because of their loose interpretation of > SIP. > > So, here's what you need to do; > > * Perform a test call reproducing the problem from start to finish, > collect as many debug logs as possible, see [1], [2], [3] > * Ensure to enable SIP trace in [3] > * Attempt to find the problem yourself, if you're still having a problem > then pastebin the results [4] > > I cannot stress how often this problem comes up, and you are unlikely to > get a response unless you have clearly shown to be doing your homework. > > Please make sure you include (in detail) what you have attempted to do to > fix the problem, what your own thoughts are, and as much info as possible. > > Hope this helps > > Cal > > [1] https://wiki.freeswitch.org/wiki/Debugging_Freeswitch > [2] http://wiki.freeswitch.org/wiki/Troubleshooting_Freeswitch > [3] http://wiki.freeswitch.org/wiki/Sofia-SIP#Debugging_Sofia-SIP > [4] http://wiki.freeswitch.org/wiki/Reporting_Bugs#Using_The_Pastebin > > > On Tue, Sep 17, 2013 at 5:32 AM, ThanhTruong <ngoc217thanh(a)yahoo.com>wrote: > >> HI all, >> >> In my case, my FS server is in public IP. and I see that I am getting nat >> in client (sip phone) side. >> I have tried with sturn server, but no luck, I have checked contact >> detail in database and see that it can get external IP correct. >> >> I have read much, tried more than 2 weeks with this, but cant get it >> works :( >> >> That is a bad situation for me. >> :( >> ThanhTruong >> >> ------------------------------ >> *From:* Paul <pasha(a)prosperity4ever.com> >> *To:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> *Cc:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> *Sent:* Tuesday, September 17, 2013 12:40 AM >> *Subject:* Re: [Freeswitch-users] NAT issue from client >> >> Edit your public sip profile xml and make sure that your internal ip of >> the server and external ip of the server (for both rtp and sip) are set >> properly. That should fix it. >> >> Paul >> >> On Mon, 16 Sep, 2013 at 10:02 AM, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming(a)simplicitymedialtd.co.uk> wrote: >> >> Hello, >> >> You'll see this has been raised many times already, a quick google search >> shows; >> >> https://www.google.co.uk/search?q=freeswitch+30+seconds >> >> http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073842.html >> http://jira.freeswitch.org/browse/FS-3496 >> >> http://freeswitch-users.2379917.n2.nabble.com/NAT-issues-Outbound-Call-drop… >> >> http://freeswitch-users.2379917.n2.nabble.com/Re-call-disconnects-after-32-… >> >> Please review all those answers and let us know if it resolves your >> problem. >> >> If not then please update the ticket with some debug logs (remember to >> paste into pastebin, not direct into the ticket) >> https://wiki.freeswitch.org/wiki/Debugging_Freeswitch >> http://wiki.freeswitch.org/wiki/Troubleshooting_Freeswitch >> >> Hope this helps >> >> Cal >> >> >> On Mon, Sep 16, 2013 at 5:48 PM, ThanhTruong <ngoc217thanh(a)yahoo.com>wrote: >> >> Hi all, >> >> I have built basic FS server with default sip account 1011 & 1012. >> I have registered 2 this accounts with xlite, portgo. them in NAT via my >> router. >> When I make call, call connect and drop after 30 seconds. >> >> my dialplan is: >> >> <extension name="public_extensions"> >> <condition field="destination_number" expression="(\d+)$" > >> <action application="set" data="hangup_after_bridge=true"/> >> <action application="bridge" >> data="sofia/internal/${destination_number}@${domain_name}"/> >> </condition> >> >> >> I have tried many configure but not success. >> PLease help to fix this issue. >> >> I am veny new in this system and tried many cases to fix it. >> >> Thank all for suggesttion. >> >> ThanhTruong >> >> _________________________________________________________________________ >> 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 >> >> >> >> _________________________________________________________________________ >> 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 >> >> >> >> _________________________________________________________________________ >> 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 >> >> > > _________________________________________________________________________ > 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 > >

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Re: [Freeswitch-users] OpenSIPs + Freeswitch + Registrations
by Chris B. Ware 17 Sep '13

17 Sep '13

Steven, you pointed me on the right direction to fix it: since I use xml_curl, there was a not closed tag and dsn was not present on internal profile. Thank you very much.

1 0

TLS + NAT
by Vincent Xia 17 Sep '13

17 Sep '13

hi guys, both TLS and NAT are working fine with my FS system, but put them together raises problem, e.g. i can have encrypted voice calls(TLS+SRTP) within the same LAN, and uncrypted voice calls over different LANs, but when trying TLS+SRTP over different LANs, i cannot get my softphone(here im using Linphone) register to FS, it reports timeout. i have a dedicated profile for NAT as below, could anyone figure out what the problem is? <profile name="tnat"> <settings> <param name="debug" value="0" /> <param name="sip-trace" value="no" /> <param name="rfc2833-pt" value="101" /> <param name="sip-port" value="5063" /> <param name="dialplan" value="XML" /> <param name="context" value="public" /> <param name="dtmf-duration" value="100" /> <param name="codec-prefs" value="$${outbound_codec_prefs}" /> <param name="use-rtp-timer" value="true" /> <param name="hold-music" value="$${hold_music}" /> <param name="rtp-timer-name" value="soft" /> <param name="manage-presence" value="false" /> <param name="aggressive-nat-detection" value="true" /> <param name="apply-nat-acl" value="rfc1918" /> <param name="inbound-codec-negotiation" value="generous" /> <param name="nonce-ttl" value="60" /> <param name="auth-calls" value="true" /> <param name="rtp-timeout-sec" value="1800" /> <param name="rtp-ip" value="$${local_ip_v4}" /> <param name="sip-ip" value="$${local_ip_v4}" /> <param name="ext-rtp-ip" value="111.111.111.111" /> <param name="ext-sip-ip" value="111.111.111.111" /> <param name="force-register-domain" value="111.111.111.111" /> <param name="stun-enabled" value="false" /> <param name="rtp-timeout-sec" value="300" /> <param name="rtp-hold-timeout-sec" value="1800" />  <param name="tls" value="true"/>  <param name="tls-only" value="false"/>  <param name="tls-bind-params" value="transport=tls"/>  <param name="tls-sip-port" value="5064"/>    <param name="tls-passphrase" value=""/>  <param name="tls-verify-date" value="true"/>   <param name="tls-verify-policy" value="none"/>  <param name="tls-verify-depth" value="2"/>  <param name="tls-verify-in-subjects" value=""/>  <param name="tls-version" value="$${sip_tls_version}"/>  <param name="inbound-late-negotiation" value="true"/>  <param name="inbound-zrtp-passthru" value="true"/> </settings> </profile>

1 0