- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Proxy Media and Call Transfer
by Keith 10 Apr '15

10 Apr '15

Hi, I am currently using Freeswitch in combination with Kamailio to act as a SBC. Freeswitch is bridging the media and acting as the B2BUA. I have an issue that if I proxy media (required so that the PBX behind the SBC can play MOH etc) when I do a transfer the caller stays on hold and the callee hears nothing. The SIP messaging is good as when you hangup one side the other is disconnected. Interestingly, when I do not proxy the media the transfer process works, however when using hold Freeswitch, because it is involved in the media path, tries to play back MOH. This is not suitable for two reasons; it requires suitable codecs on the SBC and also it means you only have a single MOH file for all customers. It will also, I believe, increase load. I can't see any way around this but hopefully someone out there can help. Many thanks Keith

1 0

Cannot call OUT
by T Fred Farmington 10 Apr '15

10 Apr '15

To the list admin - please delete the previous 2 postings from me on this Subject as 'dead wood' I don't know how I posted the same info twice and, in so doing, broke the thread. Plus at this point the info contained is obsolete. Regardless... To the other gurus..... On the FreeSWITCH Confluence ( https://freeswitch.org/confluence/display/FREESWITCH/XML+Dialplan ) it says: Dialing Through A Gateway (SIP Provider) A gateway is a means for making outbound calls through a SIP provider. For example: sofia/gateway/mygateway.com/1234 This will dial through the gateway named "mygateway.com" to user 1234. Note how there is no need to append anything after the user "1234" On attempting to call to an 'outside' number I see the following In my FS log: bridge(sofia/gateway/velocity_outbound/nnnnnnnnnn) which, according to the FS Confluence info, should initiate the call through the defined Gateway -- velocity_outbound AND I should see the number going to to the defined SIP Gateway Proxy in the log file ( NOTE - the SIP Gateway Proxy IP is included in the acl_conf.xml file as "allow" ) But the log NEVER shows that this is happening. In fact I NEVER see any reference after that point to the SIP Gateway Proxy at all Why is this not working? Your assistance would be greatly appreciated. Thanks TF

2 3

Mod-skypeopen
by Ali Jibran 09 Apr '15

09 Apr '15

Hi. I'm using mod_skypeopen and everything's working fine. The call comes in executes ivr and on DTMF it is forwarded into a queue. But for some reasons the phone aren't ringing. I have another trunk attached as a gateway. When I call from that number the phones rings fine. Can someone tell me why that is? Thanks.

3 4

centos 6 and fs 1.4.18 rpms
by ik 09 Apr '15

09 Apr '15

Hello, I know that there is a new version of FS, but I cannot find the rpms for CentOS 6 in the yum repos. There is no documentation about the repos (there used to be one in confluence) My freeswitch.repo looks like this: [freeswitch] name=FreeSWITCH Packages for Enterprise Linux $releasever - $basearch baseurl=http://files.freeswitch.org/yum/$releasever/$basearch #mirrorlist= http://mirrors.freeswitch.org/mirrorlist?repo=freeswitch-5&arch=$basearch #failovermethod=priority enabled=1 gpgcheck=1 gpgkey=file:///etc/pki/rpm-gpg/RPM-GPG-KEY-FREESWITCH So what am I missing here? Thanks, Ido

4 6

Proxy Media and Call Transfer
by keithhubner 09 Apr '15

09 Apr '15

Hi, I am currently using Freeswitch in combination with Kamailio to act as a SBC. Freeswitch is bridging the media and acting as the B2BUA. I have an issue that if I proxy media (required so that the PBX behind the SBC can play MOH etc) when I do a transfer the caller stays on hold and the callee hears nothing. The SIP messaging is good as when you hangup one side the other is disconnected. Interestingly, when I do not proxy the media the transfer process works, however when using hold Freeswitch, because it is involved in the media path, tries to play back MOH. This is not suitable for two reasons; it requires suitable codecs on the SBC and also it means you only have a single MOH file for all customers. It will also, I believe, increase load. I can't see any way around this but hopefully someone out there can help. Many thanks Keith -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Proxy-Media-and-Call-Transfer… Sent from the freeswitch-users mailing list archive at Nabble.com.

2 1

Lookup effective_caller_id_name from directory
by Paul Atreides 09 Apr '15

09 Apr '15

Hi, when I make an internal call to a user then I just see "Outbound call" and the called number in my the display. The callee sees the value set in effective_caller_id_name in the display from my extension. Is it possible to look up the called number from the directory without using mod_cidlookup. I just want to use the directory from freeswitch. It would be really great if i could see the group name too when I call one. Thanks

2 2

Intermittent audio issue with Chrome 40-42 and WebRTC
by Oleg Stolyar 09 Apr '15

09 Apr '15

Hi guys, Ever since my users switched to Chrome 40, 41 or 42 beta, every now and then they complain that they cannot hear audio from FreeSWITCH or send audio to FreeSWITCH. I cannot reproduce this at will but was able to catch several occurrences. It happens on both Windows and Macs. Here are some details" 1. Usually the problem is solved if we close Chrome and make sure all the Chrome processes are killed. 2. When the problem happens, FS logs this on connection: [ERR] switch_rtp.c:2976 audio DTLS packet not written 3. I tried using this WebRTC test on an affected computer: https://janus.conf.meetecho.com/echotest.html and that worked, so it is not a general WebRTC issue. 4. I tried FS WebRTC demo page https://webrtc.freeswitch.org/verto/index.html#page-main and on the affected computer the problem was still there - no audio. 5. This happened both on an old FS version from last May and on a very recent master FreeSWITCH Version 1.5.15b+git~20150316T164411Z~b32abaadd9~64bit (git b32abaa 2015-03-16 16:44:11Z 64bit) Any ideas?

4 7

Re- Freetdm version inside FreeSWITCH
by Andrew Keil 08 Apr '15

08 Apr '15

To FreeSWITCH users, I am not sure if this should be posted inside the developers mailing list, however I thought I would start here first. Currently I am setting up a FreeSWITCH system (using the current production release version) in the UK which will be running off BT (British Telecom) and use both SIP and ISDN30e. The ISDN side will use Sangoma A104DE cards (which I have just purchased and I am testing currently). Now the fun begins... I came across an Open Circuit alarm issue, which Sangoma are investigating, which then caused an unmapped debug message to happen (correctly from inside FreeSWITCH ftmod_wanpipe.c). Sangoma then suggested I run: ftp://ftp.sangoma.com/linux/custom/freetdm/freetdm-1.9.0.tar.gz {since this has a temporary fix to skip open circuit alarms and let FreeSWITCH continue to run with no DEBUG messages, also it adds a header (ftmod_wanpipe_defines.h) to enable mapping of alarm status values to plain English - instead of the current unmapped message} Which brings me to my primary question: Sangoma have advised me that they manage freetdm and I should use their version. However when I look at the source to ftmod_wanpipe.c (for example) their version (freetdm-1.9.0) has the following header: Copyright (c) 2007-2012, Anthony Minessale II vs. FreeSWITCH current version: Copyright (c) 2007-2014, Anthony Minessale II. Obviously there are other additions/changes done by Sangoma (including the Open Circuit alarm skipping). So what is FreeSWITCH's point of view on this? Do you merge the changes every <x> months from Sangoma's freetdm or simply expect the freetdm component to be overwritten by Sangoma's current version prior to FreeSWITCH being built? My main aim is to have a reasonably well supported system and not come across issues that have already been found and fixed by either Sangoma or FreeSWITCH. Thanks in advance for any feedback on this, as I just want to make sure I am using an agreed best version of freetdm. Andrew Keil Visytel Pty Ltd

3 4

Re: [Freeswitch-users] WebRTC + Chrome = RTP Socket Error!
by Luiz Fernando Softov 08 Apr '15

08 Apr '15

**** Logs? AS i send before, there is the trace: https://pastebin.freeswitch.org/24100 **** Does it look the same as before? YES **** With IPv6 candidates and what have your set you candidates ACL to? I'm not sure where i can define candidate-acl... The only thing i enable to work with websocket is ws-binding in profile/internal.xml <param name="ws-binding" value=":5066"/> There is anything i'm missing in conf? **** Still wan.auto? [WARNING] switch_core_media.c:2791 NO candidate ACL defined, Defaulting to wan.auto -- Luiz Fernando Softov http://www.softov.com.br fernando(a)softov.com.br

1 0

Cannot call OUT
by T Fred Farmington 08 Apr '15

08 Apr '15

My FS Server is working to: * Allow 'internal' extensions to call each other * Allow 'external' calls to be received and routed as intended But I cannot place a call from 'inside' to an 'outside' number In the file conf/sip_profiles/external/velocity_gateway.xml I have my Gateway defined Within that file I have defined: -<gateway name="velocity_outbound"> username password Proxy IP address register=false (not needed) And in conf/sip_profiles/external.xml I have the command <X-PRE-PROCESS cmd="include" data="external/velocity_gateway.xml"/> In my conf/dialplan/default/outbound_via_velocity.xml I have defined the 'extensions' local_call.velocity long_distance_call.velocity Among the other 'extension' parameters is includes: <action application="bridge" data="sofia/gateway/velocity_outbound/$1"/> In my conf/dialplan/default.xml I have the command: <X-PRE-PROCESS cmd = "include" data = "default/outbound_via_velocity.xml"/> As a result, within FS_CLI, when I do a sofia status I see the following: Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) external-ipv6::velocity_outbound gateway sip:FreeSWITCH@velocity_outbound NOREG external profile sip:mod_sofia@192.168.nnn.nn:5080 RUNNING (0) external::velocity_outbound gateway sip:FreeSWITCH@velocity_outbound NOREG external::velocity_outbound gateway sip:ClancySystems@208.nn.nnn.nn NOREG 192.168.nnn.nn alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia@192.168.nnn.nn:5060 RUNNING (0) ================================================================================================= BUT when I attempt to make a call to an 'outside' number I find the following..... In my log I see where the outbound 'extension' is found (PASS) [local_call.velocity] destination_number(9nnnnnnnnnn) =~ /^9(\d{10})$/ break=on-false and it utilized the defined Gateway (velocity_outbound) Action bridge(sofia/gateway/velocity_outbound/nnnnnnnnnn) >From that point in the log, things continue, but do not lead to success. In the log I NEVER SEE that Gateway's Proxy IP utilized to make the call I do see where it looks like things are trying to move forward [NOTICE] switch_channel.c:1055 New Channel sofia/external-ipv6/nnnnnnnnnn And subsequent log lines continue to reference sofia/external-ipv6/...... Such as: Send signal sofia/external-ipv6/nnnnnnnnnn [BREAK] BUT at the network level my server is only using IPV4 and not utilizing IPV6 at all. And NOT defined to be included in external-ipv6.xml Why is this not utilizing the Gateway defined in external.xml ?

1 0