August 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

outbound calling fails - unallocated number
by Jeremiah Bailey 21 Aug '14

21 Aug '14

Hello, I have installed FS onto RHEL 7 running in AWS EC2 and configured my gateway(les.net). I have inbound calling to my DID working properly, however, outbound calling is not working. I submitted what I believe is the relevant information to pastebin.freeswitch.org. (there is a second post on pastebin because I accidentally posted my trunk info and forgot the SIP trace, I did change my trunk creds) http://pastebin.freeswitch.org/23125 Thanks, jbailey

2 1

No audio in several profiles
by S.Mohammad Emami Razavi 21 Aug '14

21 Aug '14

When i have two sofia profiles on two network interfaces and i register my phone on second profile, i have no audio. Everything is in local computer. Firewall does not exist. I have this warning: switch_core_session.c:1554 sofia/customer_access/1009@192.168.10.10:5060 using scheduler due to bypass media or media is not established. Any idea?

1 0

Media after hang-up???
by Stuart Mills 21 Aug '14

21 Aug '14

Hi, I’m experiencing a strange issue calling a destination within the UK, whereby media seems to still be connected after hang up. I only know this as the destination is a PBX, supplied by Unify (formerly Siemens), and that PBX on occasions records a voicemail, when a message is taken and after the call is hung up, there is a continuous tone heard on the end of the recording. The call logs show the right duration and the carrier has also confirmed that the call ended as the logs suggest. So, I’m wondering if anyone else has come across the same sort of issue using FreeSWITCH? The end user has told me they never used to experience this issue using a different provider, and I can’t reproduce this issue at all using other types of PBX, all calls end as they should and voicemail recordings end when the call does. This issue doesn’t seem to be a FreeSWITCH one, just thought it worth an ask to the group. Cheers, Stuart

4 6

ClueCon presentations
by Oleg Stolyar 21 Aug '14

21 Aug '14

Hi guys, are the presentations available online yet? If so, where? If not, when?

2 1

Re: [Freeswitch-users] WebRTC Calls via Kamailio rejecting with 488
by Kamrul Khan 20 Aug '14

20 Aug '14

No I didnt --Forwarded Message Attachment-- From: brian(a)freeswitch.org To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 20 Aug 2014 14:03:43 -0500 Subject: Re: [Freeswitch-users] WebRTC Calls via Kamailio rejecting with 488 Have you modified the default sofia profile codec preferences at all? On Wed, Aug 20, 2014 at 1:47 PM, Kamrul Khan <dodul(a)live.com> wrote: Hi, This is the continuation of my previous mail, "Freeswitch srtp rejecting SRTP with 488‏". I have changed the subject line as according to the feedback from this mailing list Im convinced its not a SRTP issue. Let me describe my scenario a bit more detail. I am generating call from a webclient using simpl5. The call then hits a Kamailio server and Kamailio here acts like a webRTC proxy that forwards the call to Freeswitch. Freeswitch receives the call on udp port 5060. (not over websocket) We tested the same thing with asterisk and thats working fine. but when we try it with Freeswitch it rejects the call with 488. And in console log it shows [CS_NEW] [INCOMPATIBLE_DESTINATION] Now, we assume (as per the feedback from this mailing list) the issue is not with SRTP but with codec negotiation. Therefore, for testing purpose we stripped off all the codec attributes from our INVITE leaving only rtpmap:0 PCMU/8000 and tested again. We were hoping to see something new in our console log this time but surprisingly it showed up the exact same log as before. The console log is given below. Please somebody give us some clue whats going wrong? Thanks in advanced. 2014-08-20 05:03:42.183856 [DEBUG] sofia.c:5847 IP 192.168.146.133 Rejected by acl "domains". Falling back to Digest auth.2014-08-20 05:03:42.253550 [DEBUG] sofia.c:5847 IP 192.168.146.133 Rejected by acl "domains". Falling back to Digest auth. 2014-08-20 05:03:42.269373 [NOTICE] switch_channel.c:669 New Channel sofia/internal/alice(a)192.168.146.133 [07526f12-2862-11e4-8afd-9de9b4eb98f8]2014-08-20 05:03:42.272360 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/alice(a)192.168.146.133) Running State Change CS_NEW 2014-08-20 05:03:42.272360 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/alice(a)192.168.146.133) State NEW2014-08-20 05:03:42.281094 [DEBUG] sofia.c:4153 Channel sofia/internal/alice(a)192.168.146.133 entering state [received][100] 2014-08-20 05:03:42.281094 [DEBUG] sofia.c:4164 Remote SDP:v=0o=Mozilla-SIPUA-31.0 20849 1 IN IP4 0.0.0.0s=Doubango Telecom - firefoxt=0 0a=ice-ufrag:ca607972 a=ice-pwd:dd5932ce75801e50fae435678cd6755ba=fingerprint:sha-256 8B:B0:38:A0:DA:3D:7C:DE:7F:76:F0:35:AF:1A:DE:E9:25:8C:16:E7:D4:3F:75:B2:64:56:79:21:2D:16:7D:05m=audio 57096 UDP/RTP 0c=IN IP4 184.69.59.132 a=ptime:20a=rtpmap:0 PCMU/8000a=fmtp:101 0-15a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-levela=setup:actpassa=candidate:0 1 UDP 2128609535 172.16.1.188 57096 typ host a=candidate:1 1 UDP 1692467199 184.69.59.132 57096 typ srflx raddr 172.16.1.188 rport 57096a=candidate:5 1 UDP 2128543999 192.168.56.1 57097 typ host a=candidate:10 1 UDP 2128478463 192.168.232.1 57098 typ hosta=candidate:15 1 UDP 2128412927 192.168.146.1 57099 typ host a=candidate:0 2 UDP 2128609534 172.16.1.188 57100 typ hosta=candidate:1 2 UDP 1692467198 184.69.59.132 57100 typ srflx raddr 172.16.1.188 rport 57100 a=candidate:5 2 UDP 2128543998 192.168.56.1 57101 typ hosta=candidate:10 2 UDP 2128478462 192.168.232.1 57102 typ host a=candidate:15 2 UDP 2128412926 192.168.146.1 57103 typ hosta=rtcp-mux 2014-08-20 05:03:42.281094 [NOTICE] sofia.c:4354 Hangup sofia/internal/alice(a)192.168.146.133 [CS_NEW] [INCOMPATIBLE_DESTINATION] 2014-08-20 05:03:42.281094 [DEBUG] switch_channel.c:2102 Send signal sofia/internal/alice(a)192.168.146.133 [KILL]2014-08-20 05:03:42.282146 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/alice(a)192.168.146.133 [BREAK] 2014-08-20 05:03:42.282146 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/alice(a)192.168.146.133) Running State Change CS_HANGUP2014-08-20 05:03:42.282146 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/alice(a)192.168.146.133) State HANGUP 2014-08-20 05:03:42.282146 [DEBUG] mod_sofia.c:414 Channel sofia/internal/alice(a)192.168.146.133 hanging up, cause: INCOMPATIBLE_DESTINATION2014-08-20 05:03:42.328550 [DEBUG] mod_sofia.c:476 Responding to INVITE with: 488 2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:46 sofia/internal/alice(a)192.168.146.133 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/alice(a)192.168.146.133) State HANGUP going to sleep2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/alice(a)192.168.146.133) State Change CS_HANGUP -> CS_REPORTING 2014-08-20 05:03:42.328550 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/alice(a)192.168.146.133 [BREAK]2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/alice(a)192.168.146.133) Running State Change CS_REPORTING 2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:590 (sofia/internal/alice(a)192.168.146.133) State REPORTING2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:53 sofia/internal/alice(a)192.168.146.133 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:590 (sofia/internal/alice(a)192.168.146.133) State REPORTING going to sleep2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/alice(a)192.168.146.133) State Change CS_REPORTING -> CS_DESTROY 2014-08-20 05:03:42.328550 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/alice(a)192.168.146.133 [BREAK]2014-08-20 05:03:42.328550 [DEBUG] switch_core_session.c:1164 Session 18 (sofia/internal/alice(a)192.168.146.133) Locked, Waiting on external entities 2014-08-20 05:03:42.328550 [NOTICE] switch_core_session.c:1182 Session 18 (sofia/internal/alice(a)192.168.146.133) Ended2014-08-20 05:03:42.328550 [NOTICE] switch_core_session.c:1184 Close Channel sofia/internal/alice(a)192.168.146.133 [CS_DESTROY] 2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:428 (sofia/internal/alice(a)192.168.146.133) Running State Change CS_DESTROY2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:439 (sofia/internal/alice(a)192.168.146.133) State DESTROY 2014-08-20 05:03:42.328550 [DEBUG] mod_sofia.c:341 sofia/internal/alice(a)192.168.146.133 SOFIA DESTROY2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:60 sofia/internal/alice(a)192.168.146.133 Standard DESTROY 2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:439 (sofia/internal/alice(a)192.168.146.133) State DESTROY going to sleep From: freeswitch-users-request(a)lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 98, Issue 138 To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 20 Aug 2014 22:18:01 +0400 Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." --Forwarded Message Attachment-- From: andretodd(a)verizon.net To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 20 Aug 2014 13:35:01 -0400 Subject: Re: [Freeswitch-users] high pdd Hi, I’m using the master. Should I try 1.4.7? I’ve done a dump after things started to go bad/ backed up. I see 8193 threads in _thread_cond_timedwait. Looks like its on line 81 of thread_cond.c res = WaitForSingleObject(cond->semaphore, timeout_ms); The value of timeout_ms is 2016850392. Not sure if this means anything. Thanks for the support guys. Andre From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, August 20, 2014 11:57 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] high pdd 1.2.X is EOL, 1.4.7 is the current release. We have releases and tags in git that no longer carry the 'stable' moniker starting with 1.4. On Sun, Aug 17, 2014 at 1:39 PM, Andre Demattia <andretodd(a)verizon.net> wrote: Thanks, I'm using the newest master and the last stable 1.2.23 We are willing to pay to get this fixed. Andre From: Anthony Minessale Sent: ‎8/‎17/‎2014 12:30 PM To: Freeswitch-users Subject: Re: [Freeswitch-users] high pdd We dont have a lot of tools to troubleshoot windows. You should mention windows in your subject to have a better chance to get the attention of our windows users. You should consider installing debian 7 on the same hardware and perform similar tests to help determine if your issue is o/s specific or not. You also have not indicated the revision of FS. If you have not worked with latest master, you should be also to rule out improvements. On Aug 17, 2014 9:54 AM, "Andre Demattia" <andretodd(a)verizon.net> wrote: Yes that's off and I believe all the databases are off too. No authentication or registration is enabled. The sizes of the database are 0 until I call the shutdown command then its about 90k. I have the DBS in Ra drive too. Windows seems pretty good until the 3000 sessions per min mark. Then I just get long pdd and the threads get backed up then no more sessions can be created. Funny thing is I has 10 profiles and the same results. I also tried two consoles with the same results. FreeSWITCH is great. I'd be grateful for help. I've worked on this for 2 years and hate to give up. If any one from consulting(a)freeswitch.org is willing to help I'd appreciate that. Thanks again, Andre From: Anthony Minessale Sent: ‎8/‎17/‎2014 10:13 AM To: Freeswitch-users Subject: Re: [Freeswitch-users] high pdd Did you comment out manage-presence ? Also we have no real idea how well FS performs in windows under extreme load. On Aug 17, 2014 5:42 AM, "Andre Demattia" <andretodd(a)verizon.net> wrote: Thanks Ken, I'll try the consulting again. Andre From: Ken Rice Sent: ‎8/‎17/‎2014 1:19 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] high pdd if the last post does not work contact consulting(a)freeswitch.org for pro help from one of the developers... thats about as far as the list will take you Ken Sent from my iPad On Aug 16, 2014, at 15:23, "Andre" <andretodd(a)verizon.net> wrote: Going with the idea it’s IO related I did a db_cache status. I just disabled limit in my code by removing it but limit should be HASH not a database however I stills how a database. Below is the results of the db cache status. I’m not real sure what its saying. I should have all registration and limit disabled and should not be using any sql database from my understanding. Anyone know what the results mean? I’ve only tested 2 calls on this status. [The entire original message is not included.] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [The entire original message is not included.] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)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 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 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest --Forwarded Message Attachment-- From: stuart.mills3(a)btopenworld.com To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 20 Aug 2014 19:17:34 +0100 Subject: Re: [Freeswitch-users] Media after hang-up??? Hi Mike, Thanks for the reply. I know for sure this isn’t FreeSWITCH causing the issue, I get the CDR posted at the right time and durations are correct, it’s almost as if the far end PBX hasn’t noticed the hang-up, even though the BYE does get sent. My carrier mentioned today (following a full trace from their end) that the PBX isn’t sending an ACK to the BYE, does that make any sense given the issue I am seeing? My main issue here is if it’s a PBX configuration issue, I am unable to look at and diagnose myself, so some sort of starting point would be really helpful . Cheers, Stuart From: Michael Jerris Sent: Wednesday, August 20, 2014 5:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Media after hang-up??? I have never seen an issue like this, a pcap might tell you more, but i seriously doubt we are still sending media after the call hangs up. I would suggest taking a look at a pcap and the debug logs and seeing that the call is actually being hung up on the leg facing the pbx and we really stop sending media. Mike On Aug 19, 2014, at 8:19 PM, Stuart Mills <stuart.mills3(a)btopenworld.com> wrote: Hi, I’m experiencing a strange issue calling a destination within the UK, whereby media seems to still be connected after hang up. I only know this as the destination is a PBX, supplied by Unify (formerly Siemens), and that PBX on occasions records a voicemail, when a message is taken and after the call is hung up, there is a continuous tone heard on the end of the recording. The call logs show the right duration and the carrier has also confirmed that the call ended as the logs suggest. So, I’m wondering if anyone else has come across the same sort of issue using FreeSWITCH? The end user has told me they never used to experience this issue using a different provider, and I can’t reproduce this issue at all using other types of PBX, all calls end as they should and voicemail recordings end when the call does. This issue doesn’t seem to be a FreeSWITCH one, just thought it worth an ask to the group. Cheers, Stuart _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)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 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 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 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 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest

2 1

limit concurrent call toward certain extension
by S.Mohammad Emami Razavi 20 Aug '14

20 Aug '14

How to limit concurrent inbound calls(IVRs) toward certain extension?! Is there a solution in xml configuration for this?!

2 1

Unable to establish a TLS call between two FSClients on Windows
by Szeto, Steven 20 Aug '14

20 Aug '14

I am trying to verify that SIP TLS calls work through FreeSwitch running on Windows 7. But the outgoing leg of the call does not get established by FreeSwitch. Setup: 1 PC running Windows 7 and FreeSwitch version 1.4.7 FSClient 1.2.3.5 is installed on two PCs and are configured & registered via TLS to the FreeSwitch server. When I make a call from one FSClient (1003) to the other (1005), I see that the call request gets to the FreeSwitch server: recv 1600 bytes from tls/[10.47.26.44]:14183 at 19:56:53.257366: ------------------------------------------------------------------------ INVITE sip:1005@sszetofs1a.design.mitel.com;transport=tls SIP/2.0 From: "x1003 tls" <sip:1003@sszetofs1a.design.mitel.com >;tag=cKD7B99Z9124m To: <sip:1005@sszetofs1a.design.mitel.com> ------------------------------------------------------------------------ send 353 bytes to tls/[10.47.26.44]:14183 at 19:56:53.257366: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/TLS 10.47.26.44:12347 ;branch=z9hG4bKpKHKUN4jFUerN;rport=14183 From: "x1003 tls" <sip:1003@sszetofs1a.design.mitel.com >;tag=cKD7B99Z9124m To: <sip:1005@sszetofs1a.design.mitel.com> The call gets challenged and authenticated: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/TLS 10.47.26.44:12347 ;branch=z9hG4bKpKHKUN4jFUerN;rport=14183 From: "x1003 tls" <sip:1003@sszetofs1a.design.mitel.com >;tag=cKD7B99Z9124m To: <sip:1005@sszetofs1a.design.mitel.com>;tag=6B6B175KHK8mN ACK sip:1005@sszetofs1a.design.mitel.com;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.47.26.44:12347;branch=z9hG4bKpKHKUN4jFUerN Max-Forwards: 70 From: "x1003 tls" <sip:1003@sszetofs1a.design.mitel.com >;tag=cKD7B99Z9124m To: <sip:1005@sszetofs1a.design.mitel.com>;tag=6B6B175KHK8mN I then see INVITE/100 Trying being exchanged between the caller and FreeSwitch: INVITE sip:1005@sszetofs1a.design.mitel.com;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.47.26.44:12347;branch=z9hG4bKQvacXgNpc44aH Max-Forwards: 70 From: "x1003 tls" <sip:1003@sszetofs1a.design.mitel.com >;tag=cKD7B99Z9124m To: <sip:1005@sszetofs1a.design.mitel.com> SIP/2.0 100 Trying Via: SIP/2.0/TLS 10.47.26.44:12347 ;branch=z9hG4bKQvacXgNpc44aH;rport=14183 From: "x1003 tls" <sip:1003@sszetofs1a.design.mitel.com >;tag=cKD7B99Z9124m To: <sip:1005@sszetofs1a.design.mitel.com> I see a new channel being created to try and reach the terminating party (1005): 2014-08-20 15:56:53.387379 [INFO] mod_dialplan_xml.c:558 Processing x1003 tls <1003>->1005 in context default 2014-08-20 15:56:53.467387 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/sip:gw+3@10.47.26.34:12347 [d416a13a-fb99-4eb4-8645-24104ad5ed97] But that's it ... no SIP messages are sent to 1005 Any thoughts? Are there any parameters I can set within FreeSwitch to make the outbound call leg work? -- *Regards,* *Steve Szeto* *MiContact Center IVR Team* *Software Designer* Tel.: 613-592-5660 Ext. 71698 Email: steven_szeto(a)mitel.com <steven_szeto(a)mitel.com_> 350 Legget Drive Kanata, ON Canada K2K 2W7 *www.mitel.com <http://www.mitel.com/_>* -- This e-mail (including any attachments) is for the sole use of the intended recipient(s) and may contain information that is confidential and/or protected by legal privilege. Any unauthorized review, use, copy, disclosure or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please notify Mitel immediately and destroy all copies of this e-mail. Mitel does not accept any liability for breach of security, error or virus that may result from the transmission of this message.

1 0

Re: [Freeswitch-users] Freeswitch srtp rejecting SRTP with 488
by Kamrul Khan 20 Aug '14

20 Aug '14

Hi,Thanks for your reply. In that case, how to know what exactly the issue is. The requested codecs in my INVITE is as follows:a=rtpmap:109 opus/48000/2a=ptime:20a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-levela=setup:actpassa=candidate:0 1 UDP 2128609535 172.16.1.188 56772 typ hosta=candidate:2 1 UDP 1692467199 184.69.59.132 56772 typ srflx raddr 172.16.1.188 rport 56772a=candidate:5 1 UDP 2128543999 192.168.56.1 56773 typ hosta=candidate:10 1 UDP 2128478463 192.168.232.1 56774 typ hosta=candidate:15 1 UDP 2128412927 192.168.146.1 56775 typ hosta=candidate:0 2 UDP 2128609534 172.16.1.188 56776 typ hosta=candidate:1 2 UDP 1692467198 184.69.59.132 56776 typ srflx raddr 172.16.1.188 rport 56776a=candidate:5 2 UDP 2128543998 192.168.56.1 56777 typ hosta=candidate:10 2 UDP 2128478462 192.168.232.1 56778 typ hosta=candidate:15 2 UDP 2128412926 192.168.146.1 56779 typ hosta=rtcp-muxThe codec installed in my freeswitch:show codecs type,name,ikey codec,ADPCM (IMA),mod_voipcodecs codec,AMR,mod_amr codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.722,mod_voipcodecs codec,G.723.1 6.3k,mod_g723_1 codec,G.726 16k,mod_voipcodecs codec,G.726 16k (AAL2),mod_voipcodecs codec,G.726 24k,mod_voipcodecs codec,G.726 24k (AAL2),mod_voipcodecs codec,G.726 32k,mod_voipcodecs codec,G.726 32k (AAL2),mod_voipcodecs codec,G.726 40k,mod_voipcodecs codec,G.726 40k (AAL2),mod_voipcodecs codec,G.729,mod_g729 codec,GSM,mod_voipcodecs codec,H.261 Video (passthru),mod_h26x codec,H.263 Video (passthru),mod_h26x codec,H.263+ Video (passthru),mod_h26x codec,H.263++ Video (passthru),mod_h26x codec,H.264 Video (passthru),mod_h26x codec,LPC-10,mod_voipcodecs codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,Polycom(R) G722.1/G722.1C,mod_siren codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,Speex,mod_speex codec,iLBC,mod_ilbc Is is rejecting just because I dont have opus installed in my freeswitch? But, I do have ulaw and alaw. The same configuration works fine with asterisk and I dont have OPUS installed there aswell.Asterisk codec list: ID TYPE NAME DESCRIPTION -------------------------------------------------------------------------------- --- 100001 audio g723 (G.723.1) 100002 audio gsm (GSM) 100003 audio ulaw (G.711 u-law) 100004 audio alaw (G.711 A-law) 100011 audio g726 (G.726 RFC3551) 100006 audio adpcm (ADPCM) 100019 audio slin (16 bit Signed Linear PCM) 100007 audio lpc10 (LPC10) 100008 audio g729 (G.729A) 100009 audio speex (SpeeX) 100016 audio speex16 (SpeeX 16khz) 100010 audio ilbc (iLBC) 100005 audio g726aal2 (G.726 AAL2) 100012 audio g722 (G722) 100021 audio slin16 (16 bit Signed Linear PCM (16kHz)) 300001 image jpeg (JPEG image) 300002 image png (PNG image) 200001 video h261 (H.261 Video) 200002 video h263 (H.263 Video) 200003 video h263p (H.263+ Video) 200004 video h264 (H.264 Video) 200005 video mpeg4 (MPEG4 Video) 400001 text red (T.140 Realtime Text with redundancy) 400002 text t140 (Passthrough T.140 Realtime Text) 100013 audio siren7 (ITU G.722.1 (Siren7, licensed from Polycom)) 100014 audio siren14 (ITU G.722.1 Annex C, (Siren14, licensed from Polycom)) 100017 audio testlaw (G.711 test-law) 100015 audio g719 (ITU G.719) 100028 audio speex32 (SpeeX 32khz) 100020 audio slin12 (16 bit Signed Linear PCM (12kHz)) 100022 audio slin24 (16 bit Signed Linear PCM (24kHz)) 100023 audio slin32 (16 bit Signed Linear PCM (32kHz)) 100024 audio slin44 (16 bit Signed Linear PCM (44kHz)) 100025 audio slin48 (16 bit Signed Linear PCM (48kHz)) 100026 audio slin96 (16 bit Signed Linear PCM (96kHz)) 100027 audio slin192 (16 bit Signed Linear PCM (192kHz)) 100018 audio silk8 (SILK Custom Format 8khz) 100018 audio silk12 (SILK Custom Format 12khz) 100018 audio silk16 (SILK Custom Format 16khz) 100018 audio silk24 (SILK Custom Format 24khz) --Forwarded Message Attachment-- From: sertys(a)gmail.com To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 20 Aug 2014 12:43:11 +0200 Subject: Re: [Freeswitch-users] Freeswitch srtp rejecting SRTP with 488 He's implying that something else apart from the SRTP part is giving you the 488. Like an inability to negotiate a proper codec for the call. On Wed, Aug 20, 2014 at 9:01 AM, <dodul(a)live.com> wrote: Hello, Thanks for your reply. I'm sorry I guess didn't get you. Are you saying the log I provided in my earlier email is incomplete? Or I have to check some other log? If that is so can you please clarify which log? (Eg log name) Or it is there in the log I provided, but I have ignored the part I needed to go through? If that's the case can you please indicate which part? Thanks in advanced :) Sent from my “contract free” BlackBerry® smartphone on the WIND network. -----Original Message----- From: freeswitch-users-request(a)lists.freeswitch.org Sender: freeswitch-users-bounces(a)lists.freeswitch.org Date: Wed, 20 Aug 2014 10:06:25 To: <freeswitch-users(a)lists.freeswitch.org> Reply-To: freeswitch-users(a)lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 98, Issue 128 Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org --Forwarded Message Attachment-- From: matt(a)inveroak.com To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 20 Aug 2014 11:59:35 +0100 Subject: Re: [Freeswitch-users] Internal IP showing in SDP Hi Brian, it seems that this was not the issue. Some more details: We have an opensips server using the load_balancer module to route the calls to the freeswitch box. (https://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS) When I route the calls directly to the freeswitch's external IP, the calls connect fine and have audio. However when I route the calls to the opensips server first, the call will connect but without audio. The strange thing is Opensips uses Freeswitch's internal address (from the load_balancer table) to forward the INVITE packet, yet when the ACK packet comes in from the provider opensips tries to forward it to the FS external IP address rather than the internal IP. I understand this is probably more an opensips question, so will understand if I should post this to their mailing list, but any tips would be appreciated. thanks Matt On 19/08/2014 17:25, Matt Broad wrote: thanks brian, will give it a go and see what happens. thanks Matt On 19/08/2014 17:06, Brian West wrote: Can you try prefixing the ext-rtp-ip and ext-sip-ip with autonat:0.0.0.0 and make sure local-network-acl is set to rfc1918.auto? On Tue, Aug 19, 2014 at 9:20 AM, Matt Broad <matt(a)inveroak.com> wrote: Hi, does it make any difference if my internal IP address is being sent in the SDP connection information? I have 2 sip providers where 1 I can bridge calls and receive audio and 1 that I can bridge calls but get no audio. Even with the below settings the SDP message body contains c=IN IP4 192.168.1.1. I have vars set as: <X-PRE-PROCESS cmd="set" data="external_rtp_ip=my.ip.goes.here"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=my.ip.goes.here"/> I have setup the external profile as follows: <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> running sofia status profile external gives (replace 0.0.0.0 with the real external IP declared in vars): ================================================================================================= Name external Domain Name N/A Auto-NAT false DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 192.168.1.1 Ext-RTP-IP 0.0.0.0 SIP-IP 192.168.1.1 Ext-SIP-IP 0.0.0.0 URL sip:mod_sofia@0.0.0.0:5080 BIND-URL sip:mod_sofia@0.0.0.0:5080;maddr=192.168.1.1;transport=udp,tcp WS-BIND-URL sip:mod_sofia@192.168.1.1:5086;transport=ws HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN G722,PCMU,PCMA,GSM CODECS OUT PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false CALLS-IN 8 FAILED-CALLS-IN 1 CALLS-OUT 8 FAILED-CALLS-OUT 1 REGISTRATIONS 0 thanks Matt _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)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 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 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)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 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 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

3 2

Re: [Freeswitch-users] Installing OPUS codec
by Kamrul Khan 20 Aug '14

20 Aug '14

I installed from source as I mentioned mod_opus is in my source directory. But the mod_opus.so is not in my freeswitch installation module directory. So, for some reason the module didnt get build during my installation. From: freeswitch-users-request(a)lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 98, Issue 134 To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 20 Aug 2014 19:54:30 +0400 Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." --Forwarded Message Attachment-- From: brian(a)freeswitch.org To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 20 Aug 2014 10:47:48 -0500 Subject: Re: [Freeswitch-users] Installing OPUS codec mod_opus is part of the default build, are you installing via packages or from source? On Tue, Aug 19, 2014 at 7:57 PM, Kamrul Khan <dodul(a)live.com> wrote: Hi, Can anyone tell me how to install OPUS module in FREESWITCH ? I have the source in my source directory. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)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 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 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest --Forwarded Message Attachment-- From: brian(a)freeswitch.org To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 20 Aug 2014 10:49:38 -0500 Subject: Re: [Freeswitch-users] Freeswitch srtp rejecting SRTP with 488 Show the logs on FreeSWITCH from right before the inbound invite arrives and after the call is rejected with 488, If you're doing WebRTC you have no reason to even be off on the SRTP wiki pages in the first place. On Wed, Aug 20, 2014 at 5:43 AM, Daniel Ivanov <sertys(a)gmail.com> wrote: He's implying that something else apart from the SRTP part is giving you the 488. Like an inability to negotiate a proper codec for the call. On Wed, Aug 20, 2014 at 9:01 AM, <dodul(a)live.com> wrote: Hello, Thanks for your reply. I'm sorry I guess didn't get you. Are you saying the log I provided in my earlier email is incomplete? Or I have to check some other log? If that is so can you please clarify which log? (Eg log name) Or it is there in the log I provided, but I have ignored the part I needed to go through? If that's the case can you please indicate which part? Thanks in advanced :) Sent from my “contract free” BlackBerry® smartphone on the WIND network. -----Original Message----- From: freeswitch-users-request(a)lists.freeswitch.org Sender: freeswitch-users-bounces(a)lists.freeswitch.org Date: Wed, 20 Aug 2014 10:06:25 To: <freeswitch-users(a)lists.freeswitch.org> Reply-To: freeswitch-users(a)lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 98, Issue 128 Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)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 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 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 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 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest --Forwarded Message Attachment-- From: andrew(a)cassidywebservices.co.uk To: freeswitch-users(a)lists.freeswitch.org Date: Wed, 20 Aug 2014 16:53:58 +0100 Subject: Re: [Freeswitch-users] Freeswitch dialplan issue secondly you may also need to set the actions as inline to use a variable you've just set. add inline="true" to you actions. On 20 August 2014 15:03, Steven Ayre <steveayre(a)gmail.com> wrote: Is the first condition matching? On 20 August 2014 13:20, Nick Zaitsev <nick.zaitsev(a)mail.ru> wrote: Good day to you. I have strange issue. I would like to set the ext in dialplan in that way: if there is local registered ext in freeswitch fs should route the call there,else route the call outside, i wrote this in dialplan: <extension name="fs3_outbound"> <condition field="destination_number" expression="^([123][012345]\d\d\d)$"> <action application="set" data="ext=$1"/> <action application="set"data="domain_name=${domain}"/> </condition> <condition field="${sofia_contact(*/${ext}@${domain_name})}" expression="error/user_not_registered"> <action application="bridge" data="sofia/internal/${ext}(a)10.160.1.21"/> <anti-action application="bridge" data="user/${ext}@${domain_name}"/> </condition> </extension> But it seems,that anti-action is not match at all( Thank you for your time,best regards. -- Nick Zaitsev _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)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 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 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Andrew Cassidy BSc (Hons) MBCS SSCAManaging Director T 03300 100 960 F 03300 100 961 E andrew(a)cassidywebservices.co.uk W www.cassidywebservices.co.uk

3 2

Fw: Freeswitch dialplan issue
by Nick Zaitsev 20 Aug '14

20 Aug '14

Yes,the first condition is mathcing. From: Steven Ayre Sent: Wednesday, August 20, 2014 6:03 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch dialplan issue Is the first condition matching? On 20 August 2014 13:20, Nick Zaitsev <nick.zaitsev(a)mail.ru> wrote: Good day to you. I have strange issue. I would like to set the ext in dialplan in that way: if there is local registered ext in freeswitch fs should route the call there,else route the call outside, i wrote this in dialplan: <extension name="fs3_outbound"> <condition field="destination_number" expression="^([123][012345]\d\d\d)$"> <action application="set" data="ext=$1"/> <action application="set"data="domain_name=${domain}"/> </condition> <condition field="${sofia_contact(*/${ext}@${domain_name})}" expression="error/user_not_registered"> <action application="bridge" data="sofia/internal/${mailto:ext%7D@10.160.1.21"/> <anti-action application="bridge" data="user/${ext}@${domain_name}"/> </condition> </extension> But it seems,that anti-action is not match at all( Thank you for your time,best regards. -- Nick Zaitsev _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

1 0