- FreeSWITCH-users - FreeSWITCH Mailing Lists

First Inbound DID
by Don Hawkins 07 Aug '16

07 Aug '16

I feel like the directions on setting up a DID for inbound couldn't be clearer but yet I've been unable to get it working. * 1. SIP Invite: ------------------------------------------------------------------------ ACK sip:19722364002@mymyserver.com SIP/2.0 Via: SIP/2.0/UDP 162.217.100.10;branch=z9hG4bK1bf3.34c008d0cd01ff7b9bec681d865ae587.0 Max-Forwards: 69 To: <sip:19722364002@myserver.com>;tag=cZryjBv7B58aB From: <sip:+12144630000@162.217.100.10>;tag=as74def9f2 Call-ID: 0a7173f94045e4985f82b31525b0d93f(a)192.170.152.170 CSeq: 102 ACK Content-Length: 0 ------------------------------------------------------------------------ * 2. In public.xml AND in /public/inboundtest.xml <include> <extension name="public_did"> <condition field="destination_number" expression="^(19722364002)$"> <action application="set" data="domain_name=$${domain}"/> <action application="transfer" data="1000 XML default"/> </condition> </extension> </include> also tried... <condition field="${sip_to_user}" expression="^(19722364002)$"> * 3. Either way when I call the DID: 2016-08-06 23:05:04.089527 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/+14693380000(a)162.217.100.10 [ca1b456f-e9fc-4c99-a72d-d594835feb10] 2016-08-06 23:05:04.089527 [DEBUG] switch_core_state_machine.c:543 (sofia/internal/+14693380000(a)162.217.100.10) Running State Change CS_NEW 2016-08-06 23:05:04.089527 [DEBUG] sofia.c:9456 sofia/internal/+ 14693380000(a)162.217.100.10 receiving invite from 162.217.100.10:5060 version: 1.6.9 -16-d574870 64bit 2016-08-06 23:05:04.089527 [DEBUG] sofia.c:9623 IP 162.217.100.10 Rejected by acl "domains". Falling back to Digest auth. NOTE: 4693380000 is the number I am calling from, not the DID number. The DID number is 9722364002. Any help is appreciated, thanks!

3 4

"Missing local host" error message while FS establishing early media
by Anthony Brodi 07 Aug '16

07 Aug '16

Hi, Currently my FS is set up that way that it does not sent 183 Session Progress but 180 Ringing at once. It makes the calling side sometimes confused whether the call is being processed or not. To change this, I use pre_answer() before bridge, but then run onto this issue - FS jump up with a AUDIO RTP REPORTS ERROR: [Missing local host] as you can see in the log below. Can anyone help me on this. Thanks. EXECUTE sofia/internal/1000(a)Registar_IP.X.X.X pre_answer() [INFO] mod_dptools.c:1327 Sending early media [DEBUG] switch_core_media.c:2507 Set Codec sofia/internal/1000(a)Registar_IP.X.X.X PCMU/0 0 ms 160 samples 64000 bits 1 channels [DEBUG] switch_core_codec.c:111 sofia/internal/1000(a)Registar_IP.X.X.X Original read codec set to PCMU:0 [DEBUG] switch_core_media.c:5244 PROXY AUDIO RTP [sofia/internal/1000(a)Registar_IP.X.X.X] Softphone_local_IP.X.X.X:5930->Softphone_local_IP.X.X.X:5930 codec: 0 ms: 20 [ERR] switch_core_media.c:5844 AUDIO RTP REPORTS ERROR: [Missing local host] freeswitch@internal> sofia status profile internal ================================================================================================= Name internal Domain Name N/A Auto-NAT false DBName sofia_reg_internal Pres Hosts FS_external_IP.x.x.x,FS_external_IP.x.x.x Dialplan XML Context public Challenge Realm auto_from RTP-IP FS_external_IP.x.x.x SIP-IP FS_external_IP.x.x.x URL sip:mod_sofia@FS_external_IP.x.x.x:5060 BIND-URL sip:mod_sofia@FS_external_IP.x.x.x :5060;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN PCMU, PCMA,G729,G723,GSM,G722,OPUS CODECS OUT PCMU, PCMA,G729,G723,GSM,G722,OPUS TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA true ZRTP-PASSTHRU true AGGRESSIVENAT true

2 8

Group Call
by Shawn Wheeler 06 Aug '16

06 Aug '16

I have an odd situation. I am using the mad boss group call. The phones are set to auto answer. When I make the call, some phone will connect but others will not. However, I can call each phone individually and the phone will answer. Thoughts? Thank you in advance Shawn

2 5

Conference Call Limits
by Deepika Yadav 05 Aug '16

05 Aug '16

Hi, Can anyone tell me, what is the maximum number of mobile phone numbers that can be connected in a conference call and how many connected members can give same DTMF input simultaneously. I have a requirement where I need to call 22 phone numbers in a conference call and receive DTMF input(Digit 1) from these callers at once. I tested with 17 callers, when all of these 17 members pressed 1, my esl program received only DTMF event from only 8 callers. Just wanted to confirm if there are any sort of constraints regarding conference calls. Regards, Deepika

2 1

uuid_dump returns blank
by Don Hawkins 05 Aug '16

05 Aug '16

Using the web API uuid_dump returns a blank page (unless the uuid does not exist then it returns a error) http://192.00.22.22:8080/api/uuid_dump?0c35eaeb-4082-4e2c-9421-ddf614e188ab… uuid_dump woth same uuid is working fine from fs_cli Is this a bug?

1 0

Interesting stats ..
by Muhammad Naseer Bhatti 05 Aug '16

05 Aug '16

I did not knew FreeSWITCH is that old but looks interesting .. on an old version where enable-use-system-time is enabled for some testing. freeswitch@internal> status *UP 93 years, 92 days, 0 hours, 46 minutes, 41 seconds, 34 milliseconds, 220 microseconds* FreeSWITCH (Version 1.4.26 64bit) is ready 0 session(s) since startup 0 session(s) - peak 0, last 5min 0 0 session(s) per Sec out of max 30, peak 0, last 5min 0 1000 session(s) max min idle cpu 0.00/97.93 Current Stack Size/Max 240K/8192K -- Sent with Airmail

5 6

Which module to use?
by Don Hawkins 05 Aug '16

05 Aug '16

Very simple situation... While on a call I want to press * to post to an external URL, I want to post global variables, ie. UUID. I have this working with httapi: <execute application="bind_digit_action" data="httapi,*,exec:httapi, http://myurl.com,aleg,aleg\"></execute> Only problem is, httapi expects XML to be returned and I don't actually want to return anything, I just want to send the information to the URL and for FreeSwitch to go on about it's business. What module is best for this job?

3 13

Hostname vs IP in SDP
by Colin Morelli 05 Aug '16

05 Aug '16

Hey all, So it looks like WebRTC has issues dealing with hostnames in the SDP instead of IP addresses. Particularly, when including a hostname in the SDP there's no audio in either direction. This is the case in both the latest Firefox + Chrome browsers. This is particularly problematic for applications that are WebRTC + mobile, now that Apple is enforcing applications to support IPv6-only networks. Using hostnames in the record-route and SDP addresses meets the requirement since the OS will synthesize an IPv6 address when a DNS query is performed, but according to the documentation they strongly urge applications to avoid using address literals in their applications. Has anyone had success with methods to switch between advertising a hostname vs IP address in the SDP? Or other methods for solving this problem? Best, Colin

5 11

Press 1 to callback
by Kamil Nigmatullin 05 Aug '16

05 Aug '16

Hi. Does anyone know how can I do th following: When client arrives at FS I put him into the callcenter queue. And at the time he wait I want client to be able to press 1. After pressing FS should played mp3 and hanhup. This event somehow shuld be fired in lua or something else so that our operator could call client back. Thanks -- Kamil Nigmatullin Skype: kamil.nigmatullin

1 0

CURL from Lua Script
by Don Hawkins 05 Aug '16

05 Aug '16

root@debian:/usr/share/freeswitch/scripts# vi test.lua api = freeswitch.API(); api:execute("curl", "http://myurl.com/test3.php post name1=value1") Snatched the code straight from the the example shown at the URL below. FS does execute the script but never reaches my URL with the POST request. Did I miss something? https://freeswitch.org/confluence/display/FREESWITCH/mod_curl

4 10