- FreeSWITCH-users - FreeSWITCH Mailing Lists

originating an automated call, default value for origination_caller_id_number
by Rafal Gwizdala 14 Jul '13

14 Jul '13

Hi, Thanks for great support so far. I have another two questions to you: 1. I'm originating a call to a mobile phone where the second leg is a httapi-based IVR service. My originate command looks like this: originate {origination_caller_id_number=2234567890}sofia/gateway/ipfon/606999888 6668 The '6668' extension is an httapi script that plays an audio file and collects some digits. The problem is that my httapi ivr starts the playback even before the mobile phone is answerred. As a result, the called person doesn't hear the beginning of the recording. Can I somehow make my httapi script wait for the mobile to be answerred? 2. If origination_caller_id_number is not specified in the originate call FreeSWITCH will send Remote-Party-ID: <sip:0000000000@sip.ipfon.pl> in INVITE request that goes to the voip gateway. Is there a way to specify a default caller id for all outgoing connections that go through a particular gateway, so I wouldn't have to specify it in originate options? Thanks RG

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Building fscomm : can't find "stfu.h"
by arif hossain 14 Jul '13

14 Jul '13

Hi I'm trying to build fscomm. I'm on this commit : commit 7c88f35451b52cad104bb1794e2a2a2abef306c1 Author: Ken Rice <krice(a)freeswitch.org> Date: Mon May 6 18:30:47 2013 -0500 1.2.9 rev bump The error showed up : In file included from mainwindow.h:38:0, from main.cpp:32: ../src/include/switch.h:110:18: fatal error: stfu.h: No such file or directory compilation terminated. make: *** [main.o] Error 1 But : arif@khost:~/src/freeswitch/fscomm$ tree ../libs/stfu/ ../libs/stfu/ ├── CMakeLists.txt ├── stfu.c └── stfu.h Thanks in advance. ---------- Cheers Arif

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mod_sofia: Gateway variables not set for inbound calls
by Alexander Haugg 13 Jul '13

13 Jul '13

Hi, for my gateway in the sofia.conf.xml the variables will not set for an inbound call. for outbound it works successfully. here my profiles and gateway settings. is there a config fault? <configuration name="sofia.conf" description="sofia Endpoint">  <global_settings> <param name="log-level" value="0"/>  <param name="debug-presence" value="0"/> </global_settings> <profiles> <profile name="profile_H3K_42"> <param name="debug" value="0"/> <aliases> </aliases> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="user-agent-string" value="FreeSWITCH"/> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="$${external_sip_port}"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="rtp-timer-name" value="soft"/> <param name="local-network-acl" value="localnet.auto"/> <param name="manage-presence" value="false"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/>  <param name="enable-3pcc" value="true"/>  <param name="inbound-use-callid-as-uuid" value="true"/>  <param name="outbound-use-uuid-as-callid" value="true"/> </settings> <gateways> <gateway name="H3K_42"> <param name="realm" value="${H3K_ip}:5060" /> <param name="username" value="" /> <param name="password" value="" /> <param name="from-user" value="" /> <param name="from-domain" value="${H3K_ip}" /> <param name="expire-seconds" value="600" /> <param name="register" value="false" /> <param name="retry-seconds" value="30" /> <param name="extension" value=""/> <param name="register-transport" value="udp"/> <param name="extension-in-contact" value="true"/> <param name="ping" value="30" /> <variables> <variable name="prefix_pattern" value="^42(.+)$" /> </variables> </gateway> </gateways> </profile> </profiles> </configuration> Thanks for your Help!

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Re: [Freeswitch-users] RFC4575 and RFC4579 Freeswitch Version 1.5
by Jonathan Hunter 13 Jul '13

13 Jul '13

Hi Anthony, Thank you for replying. I have tried testing this with blink,Jitsi and Sipp script and seem to get same results each time. My configuration is basically terminate on 07917 which has; <action application="conference" data="$1@default++flags{rfc-4579}"/> However please see attached output of sofia global sip trace on.(RFC4575sofiasip). I can see Freeswitch responds with a 202 accepted to the SUBSCRIBE for event conference, then I see no NOTIFY responses. Any advice much appreciated as I want to ensure I haven't missed something in configuration. Many thanks Jon From: freeswitch-users-request(a)lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 85, Issue 123 To: freeswitch-users(a)lists.freeswitch.org Date: Sat, 13 Jul 2013 02:36:56 +0400 --Forwarded Message Attachment-- From: anthony.minessale(a)gmail.com To: freeswitch-users(a)lists.freeswitch.org Date: Fri, 12 Jul 2013 17:36:23 -0500 Subject: Re: [Freeswitch-users] RFC4575 and RFC4579 Freeswitch Version 1.5 I tested this and it does seem to work. Are you enabling sip trace on the console to see this sofia global siptrace on Also I had to update my Blink Lite for MAC up to the latest version in the app store. On Fri, Jul 12, 2013 at 7:24 AM, Jonathan Hunter <hunterj91(a)hotmail.com> wrote: Hi Guys, Hope you are all well? Im currently testing Freeswitch with a blink client to look to send events following RFC4575 /4579. I am making basic calls to a conference with flag rfc-4579 defined and the client connects to the conference, and FS as expected sends back the ;isfocus parameter on its contact address in the 200 ok. As a result the client I'm using (blink) sends a SUBSCRIBE, with Event: conference, which FS in turn sends a 202 Accepted response to. However after this point I get no NOTIFY messages at all to the initial SUBSCRIBE and for subsequent changes in the conference. I notice if a SUBSCRIBE for messages, or presence are sent, they are responded to without issue with a NOTIFY from Freeswitch. Please can someone advise if this is a configuration issue or something else that needs exploring? Any help would be great. Many thanks Jon _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

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RFC4575 and RFC4579 Freeswitch Version 1.5
by Jonathan Hunter 13 Jul '13

13 Jul '13

Hi Guys, Hope you are all well? Im currently testing Freeswitch with a blink client to look to send events following RFC4575 /4579. I am making basic calls to a conference with flag rfc-4579 defined and the client connects to the conference, and FS as expected sends back the ;isfocus parameter on its contact address in the 200 ok. As a result the client I'm using (blink) sends a SUBSCRIBE, with Event: conference, which FS in turn sends a 202 Accepted response to. However after this point I get no NOTIFY messages at all to the initial SUBSCRIBE and for subsequent changes in the conference. I notice if a SUBSCRIBE for messages, or presence are sent, they are responded to without issue with a NOTIFY from Freeswitch. Please can someone advise if this is a configuration issue or something else that needs exploring? Any help would be great. Many thanks Jon

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Call redirect
by Shishko 12 Jul '13

12 Jul '13

Hi, when a user redirects its phone to mobile number, ITSP rejects call because caller id is not in our range of numbers. If I change CID to one of our numbers, call is accepted, but users would like to see original CID . Is there a way I could deflect a call to ITSP so user could see original caller id? Thanks, Denis

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RTP NAT problem in Freeswitch 1.2.3
by Nuno Reis 12 Jul '13

12 Jul '13

Good day all. I'm experiencing the following sinptom when using some softphones behind nat on a private LAN, sometimes the same happen with hardphones. Here's the scenario: FS : <public IP> _______ <public IP>LAN ROUTER <private LAN> --- softphone Basically when softphone makes an INVITE to FS it always sends the private IP on the SDP and when the media flow starts it's being sent out by FS to the public lan address resulting on a audioless call. However if the phone sends the public IP on the SDP there's no issue at all. I know there's a variable available disable_rtp_auto_adjust that shoud make freeswitch ignore the SDP IP and use the INVITE IP instead, but it isn't working for me. Here's what i currently have on my internal SIP profile: <profile name="internal"> <aliases> </aliases> <gateways> </gateways> <domains> <domain name="all" alias="true" parse="false"/> </domains> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="sip-capture" value="no"/> <param name="watchdog-enabled" value="no"/> <param name="watchdog-step-timeout" value="30000"/> <param name="watchdog-event-timeout" value="30000"/> <param name="log-auth-failures" value="true"/> <param name="forward-unsolicited-mwi-notify" value="false"/> <param name="context" value="public"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="H264,G722,PCMA,GSM"/> <param name="outbound-codec-prefs" value="H264,G722,PCMA,GSM"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="<PUBLIC_IP>"/> <param name="sip-ip" value="<PUBLIC_IP>"/> <param name="hold-music" value="local_stream://moh"/> <param name="apply-inbound-acl" value="domains"/> <param name="apply-nat-acl" value="rfc1918"/> <param name="local-network-acl" value="localnet.auto"/> <param name="record-path" value="/opt/freeswitch/recordings"/> <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <param name="manage-presence" value="true"/> <param name="presence-privacy" value=""/> <param name="inbound-codec-negotiation" value="generous"/> <param name="tls" value="true"/> <param name="tls-only" value="false"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-sip-port" value="5061"/> <param name="tls-cert-dir" value="/opt/freeswitch/conf/ssl"/> <param name="tls-passphrase" value=""/> <param name="tls-verify-date" value="true"/> <param name="tls-verify-policy" value="none"/> <param name="tls-verify-depth" value="2"/> <param name="tls-verify-in-subjects" value=""/> <param name="tls-version" value="sslv23"/> <param name="odbc-dsn" value="freeswitch:user:password"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="inbound-reg-force-matching-username" value="true"/> <param name="auth-all-packets" value="false"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="challenge-realm" value="auto_from"/> <param name="ext-rtp-ip" value="<PUBLIC_IP>"/> <param name="ext-sip-ip" value="<PUBLIC_IP>"/> <param name="presence-hosts" value="_DISABLED_"/> <param name="NDLB-received-in-nat-reg-contact" value="true"/> <param name="NDLB-broken-auth-hash" value="true"/> <param name="dbname" value="share_presence"/> <param name="send-presence-on-register" value="true"/> <param name="manage-shared-appearance" value="true"/> <param name="registration-thread-frequency" value="30"/> <param name="enable-timer" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="send-message-query-on-register" value="true"/> <param name="all-reg-options-ping" value="true"/> <param name="sip-force-expires" value="3600"/> <param name="sip-expires-max-deviation" value="300"/> <param name="multiple-registrations" value="contact"/> </settings> </profile> Any suggestions on how to make FS use the INVITE IP for RTP instead of using the IP on the SDP? Looking forward to hear from you. Best Regards, * Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS<http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…>| www.wavecom.pt** <http://www.wavecom.pt/>* [image: Description: Description: WavecomSignature]<http://www.wavecom.pt/pt/wavecom/premios.php> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>

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ClueCon 2013 Reminder: Register Today to Maximize Your Chances of Winning
by Michael Collins 12 Jul '13

12 Jul '13

This is just a friendly reminder that if you register for ClueCon 2013<http://www.cluecon.com>by the end of today you will have the best chance of winning some of our great prizes. We have Android tablets, SIP phones, paperback and e-books, and more! We can't wait - only 24 days until ClueCon! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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Bypass media when connecting two incoming calls
by Oleg Stolyar 12 Jul '13

12 Jul '13

Hi guys, Is there a way to bypass the media when connecting two incoming calls? For example Caller A calls in and is put into park (or valet park) Then Caller B calls in and is connected to Caller A. Is there a way to enable bypass media in this case or does it only work for a bridge? Thank you *Oleg*

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mod_skypopen skype client stability
by Arsen 12 Jul '13

12 Jul '13

Hi Freeswitchers, Does somebody have experience with running multiple skype interfaces with the same user id and skype version 4.x? Is it stable? We faced a skype crash issue under CentOS 6.4 and skype 2.0.0.72 static oss (provided by interactive installer), so we switched to Ubuntu 12.10 with improved memory and CPU configuration, but unfortunatelly skype interfaces keep crashing unexpectedly. We are using shell script which checks failed skype interfaces and remove them (sk remove) from the skypopen, then start the skype and sk reload, but sometimes it causes freeswitch to segfault :/ Any ideas are very welcome. FreeSWITCH Version 1.5.3b+git~20130705T151245Z~9202f0398f (git 9202f03 2013-07-05 15:12:45Z) skype 2.0.0.72 oss Thanks, Arsen.

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