October 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

make local mirror of yum repo?
by Денис 20 Oct '14

20 Oct '14

How make mirror of yum repo?

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Re: [Freeswitch-users] AASTRA Phones - NAT issue
by Sean Devoy 20 Oct '14

20 Oct '14

Thanks Moishe. I have definitely tried true and false for that. I even tried SIP ALG on the local router. Same result. I have it sitting next to a CISCO SPA504G that is working. The sofia global siptrace matches field for field on the 2 phones. Cisco registers, AASTRA keeps retrying with the same SEQ number. I appreciate the answers, but we are moving on under the assumption AASTRA 480i do not work behind NAT. I will ship one (continental US) if anyone thinks I have just screwed it up. Thanks, Sean From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Moishe Grunstein Sent: Saturday, October 18, 2014 10:50 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] AASTRA Phones - NAT issue This should do set on the sip profile phone is registering to, http://wiki.freeswitch.org/wiki/NDLB#NDLB-force-rport Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support(a)nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg@01C72F94.9EE45D60]<http://www.nysolutions.com/> Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Sean Devoy Sent: Friday, October 17, 2014 1:55 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] AASTRA Phones - NAT issue Thanks Loyd. If that is truly from an AASTRA 480i, please tell me what firmware version and where to get it if you know. Sean From: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Lloyd Aloysius Sent: Friday, October 17, 2014 7:04 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] AASTRA Phones - NAT issue It is under the network settings Advanced Network Settings DHCP Download Options [Any \/] LLDP [X]Enabled LLDP packet interval 30 NAT IP 0.0.0.0 NAT SIP Port 51620 NAT RTP Port 51720 STUN Server 0.0.0.0 STUN Port 3478 TURN Server 0.0.0.0 TURN Port 3479 TURN User ID [ ] TURN Password [ ] Rport (RFC 3581) [X]Enabled /Lloyd On Fri, Oct 17, 2014 at 12:55 AM, Sean Devoy <sdevoy(a)bizfocused.com<mailto:sdevoy@bizfocused.com>> wrote: Brian, Thanks for the prompt response. However, I have searched the web config pages, users guide and administrator’s guides. I cannot find any reference to “rport”. Could it be hiding under some other name I don’t know? Here is the SIP config page: Global SIP Settings Top of Form Basic SIP Authentication Settings Screen Name Phone Number Caller ID Authentication Name Password BLA Number Line Mode Basic SIP Network Settings Proxy Server Proxy Port Outbound Proxy Server Outbound Proxy Port Registrar Server Registrar Port Registration Period Advanced SIP Settings Explicit MWI Subscription Enabled Send MAC Address in REGISTER Message Enabled Send Line Number in REGISTER Message Enabled Session Timer T1 Timer T2 Timer Transaction Timer Transport Protocol Registration Retry Timer BLF Subsription Period RTP Settings RTP Port Basic Codecs (G.711 u-Law, G.711 a-Law, G.729) Enabled Force RFC2833 Out-of-Band DTMF Enabled Customized Codec Preference List DTMF Method Silence Suppression Enabled Bottom of Form Thanks again, Sean From: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Brian West Sent: Thursday, October 16, 2014 6:47 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] AASTRA Phones - NAT issue None of those will have anything to do with this issue, in fact most of those aren't even needed, enable rport in your device and the problems go away. None of those can actually apply during the challenge phase its a chicken egg scenario as the user isn't looked up at that point, its just a blind challenge. On Thu, Oct 16, 2014 at 5:23 PM, Chris Tunbridge <blasterjr(a)gmail.com<mailto:blasterjr@gmail.com>> wrote: i have a deployment with 10x480i's 20x6730i's and 8x6757i's i use <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> and i have aggressive-nat-detection enabled, as well as some other NDLB settings in my internal.xml Brian mentioned rport, i don't personally use it with any of my stuff, however i have seen it used in the past. Here's the settings i have set, not sure if all of them are needed, but i do have working 480i's with this configuration <param name="aggressive-nat-detection" value="true"/> <param name="NDLB-broken-auth-hash" value="true"/> <param name="NDLB-sendrecv-in-session" value="true"/> <param name="NDLB-received-in-nat-reg-contact" value="true"/> On Thu, Oct 16, 2014 at 4:16 PM, Brian West <brian(a)freeswitch.org<mailto:brian@freeswitch.org>> wrote: And by rport I mean in the Aasstra phone. On Thu, Oct 16, 2014 at 5:15 PM, Brian West <brian(a)freeswitch.org<mailto:brian@freeswitch.org>> wrote: Its because you don't have rport enabled, if you notice your register says in the contact 71.121.183.4:5067<http://71.121.183.4:5067> so thats where we send the 401, chances are your phone doesn't get it because we are going to the contact. In this case the phone never gets the 401 and we keep sending it. On Thu, Oct 16, 2014 at 4:57 PM, Sean Devoy <sdevoy(a)bizfocused.com<mailto:sdevoy@bizfocused.com>> wrote: Hi All, I am trying to get an AASTRA 480i working with FS. I have other phones working just fine (Cisco and Polycom). I have updated the firmware. I can see in the “sofia global siptrace” the REGISTER request, the Unauthorized response, but it does not appear to be making it back to the phone. That statement is based on my observation the CSEQ does not change. I have added and removed: <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> I have tried many different option through the phones web interface, but no luck yet. Anyone have a working configuration (phone and FS) they could share)? The key siptrace packets are here (these happen to be from the attempt with both UDP and TCP): recv 811 bytes from udp/[71.121.183.4]:5060 at 19:28:05.468752: ------------------------------------------------------------------------ REGISTER sip:fs_bfis.bizfocused.com:5060<http://fs_bfis.bizfocused.com:5060> SIP/2.0 Via: SIP/2.0/UDP 71.121.183.4:5067;branch=z9hG4bKba321b220 Max-Forwards: 70 Content-Length: 0 To: 224 <sip:224@fs_bfis.bizfocused.com:5060<http://sip:224@fs_bfis.bizfocused.com:5060>> From: 224 <sip:224@fs_bfis.bizfocused.com:5060<http://sip:224@fs_bfis.bizfocused.com:5060>>;tag=23c56ce0efc9c8d Call-ID: 9fda297c6de44793402fc2f778f24fb0(a)71.121.183.4<mailto:9fda297c6de44793402fc2f778f24fb0@71.121.183.4> CSeq: 7581459 REGISTER Contact: 224 <sip:224@71.121.183.4:5067;srcadr=10.10.40.49:5060;srcadr=10.10.40.49:5060<http://10.10.40.49:5060>>;expires=300 Contact: 224 <sip:224@71.121.183.4:5067;transport=tcp;srcadr=10.10.40.49:5060;srcadr=10.10.40.49:5060<http://10.10.40.49:5060>>;expires=300 Proxy-Require: com.nortelnetworks.firewall Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Aastra-Mac:00085D03237F Aastra-Line:1 User-Agent: Aastra 480i/1.4.0.1048 Brcm Callctrl/1.5.1.0<http://1.5.1.0> MxSF/v3.2.6.26 ------------------------------------------------------------------------ send 667 bytes to udp/[71.121.183.4]:5067 at 19:28:05.470167: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 71.121.183.4:5067;branch=z9hG4bKba321b220 From: 224 <sip:224@fs_bfis.bizfocused.com:5060<http://sip:224@fs_bfis.bizfocused.com:5060>>;tag=23c56ce0efc9c8d To: 224 <sip:224@fs_bfis.bizfocused.com:5060<http://sip:224@fs_bfis.bizfocused.com:5060>>;tag=eKgtr8tcS78QH Call-ID: 9fda297c6de44793402fc2f778f24fb0(a)71.121.183.4<mailto:9fda297c6de44793402fc2f778f24fb0@71.121.183.4> CSeq: 7581459 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.2.22+git~20140309T212137Z~65fed130e5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com<http://fs_bfis.bizfocused.com>", nonce="33a7d4af-e832-4d75-a284-b7002edadfc5", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 811 bytes from udp/[71.121.183.4]:5060 at 19:28:05.970923: ------------------------------------------------------------------------ REGISTER sip:fs_bfis.bizfocused.com:5060<http://fs_bfis.bizfocused.com:5060> SIP/2.0 Via: SIP/2.0/UDP 71.121.183.4:5067;branch=z9hG4bKba321b220 Max-Forwards: 70 Content-Length: 0 To: 224 <sip:224@fs_bfis.bizfocused.com:5060<http://sip:224@fs_bfis.bizfocused.com:5060>> From: 224 <sip:224@fs_bfis.bizfocused.com:5060<http://sip:224@fs_bfis.bizfocused.com:5060>>;tag=23c56ce0efc9c8d Call-ID: 9fda297c6de44793402fc2f778f24fb0(a)71.121.183.4<mailto:9fda297c6de44793402fc2f778f24fb0@71.121.183.4> CSeq: 7581459 REGISTER Contact: 224 <sip:224@71.121.183.4:5067;srcadr=10.10.40.49:5060;srcadr=10.10.40.49:5060<http://10.10.40.49:5060>>;expires=300 Contact: 224 <sip:224@71.121.183.4:5067;transport=tcp;srcadr=10.10.40.49:5060;srcadr=10.10.40.49:5060<http://10.10.40.49:5060>>;expires=300 Proxy-Require: com.nortelnetworks.firewall Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Aastra-Mac:00085D03237F Aastra-Line:1 User-Agent: Aastra 480i/1.4.0.1048 Brcm Callctrl/1.5.1.0<http://1.5.1.0> MxSF/v3.2.6.26 ------------------------------------------------------------------------ send 667 bytes to udp/[71.121.183.4]:5067 at 19:28:05.971138: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 71.121.183.4:5067;branch=z9hG4bKba321b220 From: 224 <sip:224@fs_bfis.bizfocused.com:5060<http://sip:224@fs_bfis.bizfocused.com:5060>>;tag=23c56ce0efc9c8d To: 224 <sip:224@fs_bfis.bizfocused.com:5060<http://sip:224@fs_bfis.bizfocused.com:5060>>;tag=eKgtr8tcS78QH Call-ID: 9fda297c6de44793402fc2f778f24fb0(a)71.121.183.4<mailto:9fda297c6de44793402fc2f778f24fb0@71.121.183.4> CSeq: 7581459 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.2.22+git~20140309T212137Z~65fed130e5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com<http://fs_bfis.bizfocused.com>", nonce="33a7d4af-e832-4d75-a284-b7002edadfc5", algorithm=MD5, qop="auth" Content-Length: 0 Thanks, Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian West brian(a)freeswitch.org<mailto:brian@freeswitch.org> [http://billing.freeswitch.org/templates/default/img/whmcslogo.png] Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001<tel:%2B19184209001> | F:+19184209002<tel:%2B19184209002> | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest -- Brian West brian(a)freeswitch.org<mailto:brian@freeswitch.org> [http://billing.freeswitch.org/templates/default/img/whmcslogo.png] Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001<tel:%2B19184209001> | F:+19184209002<tel:%2B19184209002> | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@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<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian West brian(a)freeswitch.org<mailto:brian@freeswitch.org> [http://billing.freeswitch.org/templates/default/img/whmcslogo.png] Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001<tel:%2B19184209001> | F:+19184209002<tel:%2B19184209002> | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@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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(no subject)
by Kamil Nigmatullin 20 Oct '14

20 Oct '14

Dear all, Today we had an attack. One of our clients lost password to his SIP account. So with this password attackers made calls on our client's behalf to very expensive destinations. We have Opensips as a border controller and Freeswitch as a Softswitch. This phone was confugured for 1 concurrent line using module limit of FS. Howerver they somehow managed to make several concurrent calls per one account. On CDR's we found that there was Attended transfer. Does anybody knows what kind of attack was that and how I can protect us against this? When I check if limit works whith a sipphone, I see that it workd 100%. Thanks in advance

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Peer Identity outdated after SIP transfer
by jcabezas＠inovax.com.br 18 Oct '14

18 Oct '14

Hi, When user A blind-transfers user B to user C (using standard REFER signaling) Freeswitch does the transfer and B then talks with C, that's OK. But B's UA still keeps displaying A as the peer and not C, as would be correct. I need that information to be displayed up-to-date in the transferee UA. How can that be achieved ? - A proxy-like way of dealing with transfer signaling can be enabled in FS ? - Any specific signaling for updating peer identity, between Freeswitch and transferee, can be enabled in FS? Thanks, Julio.

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process_cdr variable
by Vladimir Getmanshchuk 17 Oct '14

17 Oct '14

Hi, JFYI session:setVariable('process_cdr', ‘b_only') from lua works incorrect… (got a-leg too at mod_cdr-csv)... it works correct only in {process_cdr=b_only} before bridge. -- Yours sincerely, Vladimir Getmanshchuk

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In-call artifacts with PCMA/PCMU
by Iskren Hadzhinedev 17 Oct '14

17 Oct '14

Hi everyone, Some of the calls using PCMA/PCMU have artifacts (distorted sound, “robotic” voice, stutter) on what looks like a random basis. The other codecs I have tried don’t have issues (tried GSM, iLBC,G722, G729, SILK). Apart from FreeSWITCH, the only other thing running on the machine is a static webpage served by nginx and the load is almost always at 0. Any help or advise on how to chase down this issue is highly appreciated! System specs: Machine: Xen VM RAM: 1GB CPU: Dualcore 2.4 Ghz 64bit OS: CentOS 6.5 FreeSWITCH Version: 1.5.13b+git~20140621T003222Z~47891d5caa~64bit Max # simultaneous calls: 12 Calls per second: at most 3 Endpoints used for testing: Hardware phones: Yealink T42G, Yealink T48G, Gigaset DE900, Cisco 7975 (with SIP firmware) Softphones: CounterPath Bria for Android, CsipSimple (android), Linphone, Jitsi Mobile provider and a landline via gateway. Thanks, -- *Iskren Hadzhinedev*

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 17 Oct '14

17 Oct '14

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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Peer Identity outdated after SIP transfer
by jcabezas＠inovax.com.br 17 Oct '14

17 Oct '14

For now I am testing with soft-phones, X-Lite e.g. Anyway Wireshark running in transferee X-Lite host does not show any SIP messages during/after the transfer process, only in the call setup and call teardown. Thanks.

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Disabling autonat/external access
by Charles Colbourn 17 Oct '14

17 Oct '14

Hi, I'm currently trying to use a local instance of freeswitch as a test/dev environment for embedding voice & video in an application. It's version 1.5.12b 32bit running on windows 7. The trouble is, nothing I do seems to prevent it from opening access to the outside world, so I'm ending up with loads of attempts to register from 192.111.153.106. My config is currently in a hacked/broken state such that they can't authenticate (and I've changed the passwords anyway), but it's a damn nuisance. I've tried adding -nonat and -nonat -nonatmap to the startup parameters. I've commented out the <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> values in both external and interal sip profiles (and renamed the external one to a non xml extension so it doesn't get read). I've also tried setting those parameters to invalid values. I've searched over and over and dug through documentation. Nothing seems to work, this external connection still keeps trying to authenticate with various extension numbers. Please, am I missing something obvious here or something? It seems strange that it would be so difficult to lock the thing down for testing. thanks, Charles.

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RTP problem/NAT
by Miha 17 Oct '14

17 Oct '14

Hi, first my FS is on public ip. We have migrated from 1.2.x to Version 1.4.8+git~20140910T181155Z~9f72b11892~64bit (git 9f72b11 2014-09-10 18:11:55Z 64bit as we needed different server. I have noticed that a lot of costumers are complaining that a lot of phone call has been cut in about 1:30 min of the call. From what I can see from sip trace is that FS begins to send media to private ip. link to trace: http://pastebin.com/9M6X6SFp So, rtp is send first to right ip and media is ok but in about 1:30 min of the call the FS sends update and begin to send rtp to private ip. thx for help! miha

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