- FreeSWITCH-users - FreeSWITCH Mailing Lists

Verto on the iPhone 6 with Safari 11?
by Mundkowsky, Robert 19 Oct '17

19 Oct '17

Just saw the email about "Anthony Minessale Dangerously Demoed Verto on the iPhone 6 with Safari 11 at AstriCon". Glad to use iOS 11 works now. Did you have to do anything unique to get getUserMedia to work? Last time I tried Verto on a iOS 11 beta it did not work. Robert ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________

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Help with Freeswitch Installation
by Alexander Perkins 19 Oct '17

19 Oct '17

Hi All. I am attempting to install FS and I am running into issues. I am attempting to install FS following the instructions on this page- https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7. When I attempt to run the second line, yum install -y freeswitch-config-vanilla freeswitch-lang-* freeswitch-sounds-*, I am receiving the following error- http://drops.tiltx.org/BUUzG0. When I look at the freeswitch.repo file in /etc/yum.repos.d, I see the baseURL pointing to http://files.freeswitch.org/yum-1.6/$releasever/$basearch. However, when I visit http://files.freeswitch.org/yum-1.6/latest, I get a 'Not Found' error- http://drops.tiltx.org/aqwKWN. I have attempted to update the baseURL to http://files.freeswitch.org/yum-1.6/7/x86_64/ as well as http://files.freeswitch.org/yum-1.6/7server/x86_64/, but then receive a 'libpcre.so.1 not found' error. I know something is not right, but I do not know what that is. Can someone point me in the right direction? Thank you, Alex

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Re: [Freeswitch-users] outbound route question
by Ahmed habiba 19 Oct '17

19 Oct '17

Hi, looks like that the call is handled by another extension, please be sure to put your new extension before the part that is handling the default extension i.e. 1000 to 1009 i.e. before the below line: <extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9])$"> Thanks, Ahmed Habiba. > > > > From: alex <hiastar_alex(a)163.com> > Subject: [Freeswitch-users] outbound route question > Date: October 18, 2017 at 6:45:02 AM GMT+3 > To: freeswitch-users(a)lists.freeswitch.org > > > I create two sip extension 1000 and 1001,i have to calls 1000 to 1001. > This is my outbount route: > <include> > <context "name=default"> > <extension name="outgoing"> > <condition field="destination_number" expression="^(100)$"> > <action application="bridge" data="sofia/freetdm/wp1/a/$1"/> > </condition> > </extension> > </context> > </include> > But it didn't take the route. > Why is that? what should i do? > This are CLI log: > 2017-10-13 05:02:36.296708 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1000(a)192.168.1.70 [b12d303a-1f58-4731-b568-78e2b256c28b] > 2017-10-13 05:02:36.316705 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1234 in context public > 2017-10-13 05:02:36.316705 [NOTICE] switch_ivr.c:2201 Transfer sofia/internal/1000(a)192.168.1.70 to XML[1001@default] > 2017-10-13 05:02:36.316705 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1001 in context default > 2017-10-13 05:02:36.316705 [CRIT] mod_dptools.c:1782 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING > 2017-10-13 05:02:36.316705 [CRIT] mod_dptools.c:1782 Open /usr/local/freeswitch/conf/vars.xml and change the default_password. > 2017-10-13 05:02:36.316705 [CRIT] mod_dptools.c:1782 Once changed type 'reloadxml' at the console. > 2017-10-13 05:02:36.316705 [CRIT] mod_dptools.c:1782 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING > 2017-10-13 05:02:46.336708 [INFO] switch_ivr_async.c:4273 Bound B-Leg: *1 execute_extension::dx XML features > 2017-10-13 05:02:46.336708 [INFO] switch_ivr_async.c:4273 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1000.2017-10-13-05-02-46.wav > 2017-10-13 05:02:46.336708 [INFO] switch_ivr_async.c:4273 Bound B-Leg: *3 execute_extension::cf XML features > 2017-10-13 05:02:46.336708 [INFO] switch_ivr_async.c:4273 Bound B-Leg: *4 execute_extension::att_xfer XML features > 2017-10-13 05:02:46.336708 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1001@192.168.1.69:5060 [4e2e818f-98f5-461b-ad4c-ad34eb933191] > 2017-10-13 05:02:46.336708 [NOTICE] switch_ivr_originate.c:2868 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] > 2017-10-13 05:02:46.396708 [NOTICE] sofia.c:7391 Ring-Ready sofia/internal/1001@192.168.1.69:5060! > 2017-10-13 05:02:46.396708 [INFO] switch_ivr_originate.c:1220 Sending early media > 2017-10-13 05:02:46.416708 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000(a)192.168.1.70! > 2017-10-13 05:02:50.016708 [NOTICE] sofia.c:8474 Hangup sofia/internal/1000(a)192.168.1.70 [CS_EXECUTE] [ORIGINATOR_CANCEL] > 2017-10-13 05:02:50.016708 [NOTICE] switch_ivr_originate.c:3629 Hangup sofia/internal/1001@192.168.1.69:5060 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] > 2017-10-13 05:02:50.036718 [NOTICE] switch_ivr_originate.c:2868 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] > 2017-10-13 05:02:50.036718 [INFO] mod_dptools.c:3508 Originate Failed. Cause: ORIGINATOR_CANCEL > 2017-10-13 05:02:50.036718 [NOTICE] switch_core_session.c:1731 Session 2 (sofia/internal/1001@192.168.1.69:5060) Ended > 2017-10-13 05:02:50.036718 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1001@192.168.1.69:5060 [CS_DESTROY] > 2017-10-13 05:02:50.056713 [NOTICE] switch_core_session.c:1731 Session 1 (sofia/internal/1000(a)192.168.1.70) Ended > 2017-10-13 05:02:50.056713 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1000(a)192.168.1.70 [CS_DESTROY] > > Thanks for answer. > > > 【网易自营|30天无忧退货】仅售同款价1/4！MUJI制造商“2017秋冬舒适家居拖鞋系列”限时仅34.9元>> <http://you.163.com/item/detail?id=1165011&from=web_gg_mail_jiaobiao_9> >

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[mod_httpcache][Files Life]
by Bilal Abbasi 18 Oct '17

18 Oct '17

Hi Users, I am using mod_http cache to cache my sound files, i can see that once the files are downloaded by the module that remains on the cache folder, is there any way to actually to remove/delete the files after default-max-age. If there is no way in Freeswitch, then i guess i need to write a cronjob to delete the older files, does not seems to have good way to do. Regards Abbasi

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Problems accessing files.freeswitch.org
by Shaun Stokes 18 Oct '17

18 Oct '17

files.freeswitch.org (209.105.235.7) appears to be down and freeswitch.org (209.105.235.6) is running like a slug. Presume there is some ongoing server maintenance? ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________

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Late Negotiation and ReINVITE
by Andrew Cassidy 17 Oct '17

17 Oct '17

Hi Guys, I wonder if this is something anyone else has come across: We make a call out to this particular customer via our wholesale SIP provider. Other end picks up and tries to transfer the call. The call is dropped out end by FreeSWITCH following a reinvite from the remote end with no SDP. FreeSWITCH sends 200 with SDP, other end ACKs then FreeSWITCH drops the call "No answer to offer" We use BT Wholesale's IPEX product and I think we can assume that this call is SIP end-to-end (this is something that I'm finding increasingly common) and what is happening is FreeSWITCH is seeing the reinvite as a late negotiation invite and when the ACK comes back with no SDP, thinks the other end has refused and hanging up the call. I've not tested with inbound-late-negotiation disabled and am hoping to avoid doing so. I have tried setting renegotiate-codec-on-reinvite to false on the outgoing profile which has made no difference. I'm using FreeSWITCH 1.6.19 Kind regards,

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Calls only between members of the same group
by Jose David Jurado Alonso 17 Oct '17

17 Oct '17

Hi, I would like to restrict all calls so that they can only be made between users in the same group (there may be many groups) but I have not found anything in particular. I have been playing with the parameters "toll_allow" and "callgroup" but still calling between members of different groups. Does anyone know how to do this?

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outbound route question
by alex 17 Oct '17

17 Oct '17

Hello: I want to make a loop test from port 1 to port 2. Now, the pri show OK and UP. But the dialplan does not work. the call flow is : SIP A->Port 1->Port 2-> SIP B. it looks SIP A directly call to SIP B without going E1 ports. The attached file are inbound route file 、outbound route file and log file. Is there a problem with outbound route? what should i do? I hope you can help me,thank you.

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newbie Question in two parts
by Shawn Wheeler 17 Oct '17

17 Oct '17

I have three separate FS systems I running. I would like to be able to make a Mad Boss call from server A to Servers B and C. I found this page that tells how to connect two boxes. https://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes Question 1 - on this page the two server have difference extension. Is there a way to do this if they similar e.g. 1001, 1002 etc.. Question 2 - Can I make a Mad Boss call from one to multiple servers? If yes, can you point me in a direction to learn how to accomplish this task? Thank you in advance. Shawn

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Connection problem between FreeSwitch Server and Asterisk Server
by 王聡 16 Oct '17

16 Oct '17

I had built a set of server which includes a FS server and an Asterisk Server, and I hope to use FS on application call, Asterisk on public call. Both 2 servers had installed 2 Ethernet controllers(one for public network, and one for LAN). During test I found the App-App call works, but App-public not. Asterisk and FreeSwitch can’t communicate well via LAN. I thought it might a firewall problem, but it still not work since I disabled the iptables on both server. FS is listening a public ip with 5080 and a local ip with 5060, while Asterisk sends the message by local ip with 5080. In this situation, FS can’t receive messages from Asterisk. After I set port to 5060 on Asterisk, it can’t receive messages from FS. Anyone has idea about settings? Do I need to make another sip profile to listen local:5080?

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