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Sipwise 3.1 installation problems.
by nh.developer 25 Nov '13

25 Nov '13

I have tried to install sipwise 3.1 on a fresh debian installation and always get the following message: service apache2 restart [ ok ] Restarting web server: apache2 ... waiting . Setting up ngcp-ossbss-system (3.8.1) ... Setting up ngcp-ossbss-xmldispatcher (3.8.1) ... Errors were encountered while processing: ngcp-asterisk ngcp-reminder ngcp-services-ce ngcp-ngcp-ce E: Sub-process /usr/bin/dpkg returned an error code (1) No matter what debian version I install, 7.0, 7.1 or 7.2 I get same error Any help? thx

1 0

Trouble bridging to mod_lcr endpoint
by Victor Chukalovskiy 25 Nov '13

25 Nov '13

Hi, I'm on 1.2.15 git 4da26f5 and the following does not work for me: http://wiki.freeswitch.org/wiki/Mod_lcr#Dialplan.2C_Bridge In the dialplan I do: <action application="bridge" value="lcr/test_profile/15555551234"/> Based on the debug, nothing is passed to the bridge app. I don't see mod_lcr called either: http://pastebin.freeswitch.org/21673 Am I doing something wrong here?? Thank you, -Victor

3 5

Notify message: Event: talk not working?
by Peter Steinbach 25 Nov '13

25 Nov '13

Hello, I tried to remotely anwer physical phones, which are in RINGING state, via event-socket. But the phones are not answering (tried Yealink an Aastra, see below). I send via event-socket sendevent NOTIFY. profile: internal. event-string: talk. user: 200. host: my.domain.com. content-type: application/simple-message-summary. State: ACTIVE I also set the "State" as described in http://wiki.freeswitch.org/wiki/Mod_event_socket#sendevent The following SIP is exchanged then (here Yealink): NOTIFY sip:200@192.168.1.129:5062 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.220;rport;branch=z9hG4bKy8arj9a0SXNZN. Max-Forwards: 70. From: <sip:200@my.domain.com>;tag=515t5XNrSKFQF. To: <sip:200@my.domain.com>. Call-ID: c74eacdc-cca4-1231-c19a-b3a15c58d7e1. CSeq: 52146637 NOTIFY. Contact: <sip:mod_sofia@192.168.1.220:5060>. User-Agent: Freeswitch. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Event: talk. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: terminated;reason=noresource. Content-Type: application/simple-message-summary. Content-Length: 0. SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.220;rport;branch=z9hG4bKy8arj9a0SXNZN. From: <sip:200@my.domain.com>;tag=515t5XNrSKFQF. To: <sip:200@my.domain.com>;tag=669161475. Call-ID: c74eacdc-cca4-1231-c19a-b3a15c58d7e1. CSeq: 52146637 NOTIFY. User-Agent: Yealink SIP-T38G 38.70.0.125. Content-Length: 0. The call was not answered however and the Rsyslog does not show an entry to this message. I then tried an Aastra 6739i Phone and got the following: NOTIFY sip:208@192.168.1.118:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP 192.168.1.220;rport;branch=z9hG4bKpeya6gXmHa3rr. Max-Forwards: 70. From: <sip:208@my.domain.com>;tag=SXXjXe2S7De0K. To: <sip:208@my.domain.com>. Call-ID: 835658b2-cca1-1231-c19a-b3a15c58d7e1. CSeq: 52145935 NOTIFY. Contact: <sip:mod_sofia@192.168.1.220:5060>. User-Agent: Freeswitch. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Event: talk. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: terminated;reason=noresource. Content-Type: application/simple-message-summary. Content-Length: 0. . SIP/2.0 489 Bad Event. Via: SIP/2.0/UDP 192.168.1.220;rport=5060;branch=z9hG4bKpeya6gXmHa3rr;received=192.168.1.220. From: <sip:208@my.domain.com>;tag=SXXjXe2S7De0K. To: <sip:208@my.domain.com>;tag=891443321. Call-ID: 835658b2-cca1-1231-c19a-b3a15c58d7e1. CSeq: 52145935 NOTIFY. Server: Aastra 6739i/3.2.2.2044. Content-Length: 0. Aastra shows me in rsyslog: 00:53:57.550000 : (SIP) INFO: EvOnPacketReceived: CSipEngine::, Packet do not match an existing Dialog, method: 6 00:53:57.550000 : (SIP) FUNC: OnPacketReceived: CSipGenericRequest(0x821DBC60) 00:53:57.550000 : (SIP) FUNC: EvInvalidNotify: CSipSubscribeEventDispatcher(0x821DC770) Reason 8002c405: Enter So I think, the Subscription-State may not be right, so Subscription-State: active might be better, right? mod_sofia.c, around line 1429 (as of 23-Oct-2013) should set this to "active" as far as I can see, but this is not the case here. But how can I solve this? Maybe there is another way? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

5 8

Cannot compile last head mod_avmd - error
by Antonio Silva 25 Nov '13

25 Nov '13

Hi, It's not possible to compile last head with the following error: /home/marces/svn/packages/trunk/src/freeswitch/freeswitch-git/src/mod/applications/mod_avmd/mod_avmd.c: In function ‘avmd_process’: /home/marces/svn/packages/trunk/src/freeswitch/freeswitch-git/src/mod/applications/mod_avmd/mod_avmd.c:542: error: passing argument 1 of ‘switch_core_session_get_channel’ from incompatible pointer type /home/marces/svn/packages/trunk/src/freeswitch/freeswitch-git/src/include/switch_core.h:822: note: expected ‘struct switch_core_session_t *’ but argument is of type ‘struct avmd_session_t *’ make[6]: *** [mod_avmd.lo] Error 1 make[6]: Leaving directory `/home/marces/svn/packages/trunk/src/freeswitch/freeswitch-git/src/mod/applications/mod_avmd' make[5]: *** [all] Error 1 make[5]: Leaving directory `/home/marces/svn/packages/trunk/src/freeswitch/freeswitch-git/src/mod/applications/mod_avmd' I create a jira and a possible patch that fixes the compile... but I'm not sure if is ok... can you revise? JIRA: http://jira.freeswitch.org/browse/FS-5989 Best regards, António

2 2

Conference delay increasing over time
by Erik M. Devane - Comms Guy 25 Nov '13

25 Nov '13

I have a very well specified Windows Server connected to sound card input via mod_portaudio. I want multiple callers to be able to listen (only) to the incoming feed, so I set up dynamic conferences, with the dialplan flags setting mute. An autocall bridges the portaudio. My assumption is that this is the best way. I am happy to be wrong if there's a better plan. The problem is that delay builds up, seconds every minute on the conference. I have RTP-Timer-Name=none and enabling both RTP AutoFlushes. Any other suggestions I could try? Would CentOS have better timing? I don't need callers to hear each other, but I'm sure that I'm missing something. Any thoughts? Erik

4 10

Is there a way to setup Call admission and control on a per endpoint basis ?
by Ashwin Rath 25 Nov '13

25 Nov '13

Hi All I am aware of the max calls and call rate setting in the max sessions and the sessions per second in the switch.conf file. However i would like to have the same on a per endpoint basis , something like a setting at a sip profile level. Is it possible ? --Ashwin

2 1

Help with Google Voice from UK
by Claves do Amaral 24 Nov '13

24 Nov '13

Hello to the list, this is my first post here so please, in case I am off topic or this is not the right place to ask questions redirect me to a more suitable list. I am trying to make calls using Freeswitch via Google Voice. I am based in London, UK. I have a standard Google voice account, I have charged it with £10 :) and successfully made a call via the Google web interface. I have compiled the lastest stable (v1.2.stable) and configured freeswitch on a raspberry pi using the following tutorials http://raspberrypipbx.blogspot.co.uk/ http://wiki.freeswitch.org/wiki/Google_Voice I did not define any new account yet, using the user 1000.I am calling with a softphone (Ekiga). I successfully called the test numbers (5000, 9664) from my pc. Here I paste my configuration: root@raspberrypi:/usr/local/freeswitch# cat conf/jingle_profiles/client.xml <include> <profile type="client"> <param name="name" value="gtalk"/> <param name="login" value="claves.doamaral(a)gmail.com/talk"/> <param name="password" value="********"/> <param name="server" value="talk.google.com" /> <param name="message" value="Hello from Claves do Amaral!" /> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="rtp-ip" value="$${bind_server_ip}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="auto-login" value="true"/>  <param name="sasl" value="plain"/>  <param name="tls" value="true"/>  <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="none"/>  <param name="exten" value="2001"/> <param name="vad" value="both"/> <param name="candidate-acl" value="wan.auto"/> <param name="local-network-acl" value="localnet.auto"/> </profile> </include> root@raspberrypi:/usr/local/freeswitch# cat conf/dialplan/default.xml <?xml version="1.0" encoding="utf-8"?> <include> <context name="default"> <extension name="unloop"> <condition field="${unroll_loops}" expression="^true$"/> <condition field="${sip_looped_call}" expression="^true$"> <action application="deflect" data="${destination_number}"/> </condition> </extension> <extension name="gvoice_out"> <condition regex="any"> <regex field="destination_number" expression="^(\d*)$" /> <regex field="dialed_extension" expression="^\+1(\d{10})(a)voice.google.com$" /> <regex field="destination_number" expression="\+1(\d{10})$" /> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="call_timeout=45" /> <action application="ring_ready"/> <action application="bridge" data="dingaling/gtalk/+$1(a)voice.google.com"/> </condition> </extension> ..... This is the result of trying to call my mobile. I have tried to call using several prefixes (+44, 0044, 44,..) and changing the latest <action> above to different patterns (+1$1(a)voice.google.com, $1(a)voice.google.com, ...) but the result did not change. 2013-11-19 10:53:44.350111 [INFO] mod_dialplan_xml.c:558 Processing Claves do Amaral <1000>->447462295322 in context default 2013-11-19 10:53:44.370101 [NOTICE] mod_sofia.c:2821 Ring-Ready sofia/internal/1000@raspberrypi! 2013-11-19 10:53:44.390105 [NOTICE] mod_dptools.c:948 Ring Ready sofia/internal/1000@raspberrypi! 2013-11-19 10:53:44.390105 [NOTICE] switch_channel.c:1052 New Channel dingaling/gtalk/+447462295322(a)voice.google.com [dc04188c-5108-11e3-a264-1dcc41a01c4a] 2013-11-19 10:53:44.410083 [NOTICE] mod_dingaling.c:1807 Ring-Ready dingaling/gtalk/+447462295322(a)voice.google.com! 2013-11-19 10:53:45.370020 [INFO] mod_dingaling.c:1561 Stun Success 46.64.61.76:53970 2013-11-19 10:53:45.390046 [NOTICE] mod_dingaling.c:864 Hangup dingaling/gtalk/+447462295322(a)voice.google.com [CS_INIT] [NORMAL_CLEARING] 2013-11-19 10:53:45.410085 [INFO] mod_dptools.c:3201 Originate Failed. Cause: NORMAL_CLEARING 2013-11-19 10:53:45.410085 [NOTICE] switch_channel.c:4643 Hangup sofia/internal/1000@raspberrypi [CS_EXECUTE] [NORMAL_CLEARING] 2013-11-19 10:53:45.430173 [NOTICE] switch_core_session.c:1567 Session 11 (sofia/internal/1000@raspberrypi) Ended 2013-11-19 10:53:45.430173 [NOTICE] switch_core_session.c:1571 Close Channel sofia/internal/1000@raspberrypi [CS_DESTROY] 2013-11-19 10:53:45.650107 [INFO] mod_dingaling.c:1561 Stun Success 46.64.61.76:53971 2013-11-19 10:53:45.970013 [INFO] mod_dingaling.c:1561 Stun Success 46.64.61.76:53972 2013-11-19 10:53:46.110013 [INFO] mod_dingaling.c:1561 Stun Success 46.64.61.76:53973 2013-11-19 10:53:46.130083 [NOTICE] switch_core_session.c:1567 Session 12 (dingaling/gtalk/+447462295322(a)voice.google.com) Ended 2013-11-19 10:53:46.130083 [NOTICE] switch_core_session.c:1571 Close Channel dingaling/gtalk/+447462295322(a)voice.google.com [CS_DESTROY] Please, can you help? Claves -- Claves do Amaral Email: claves.doamaral(a)gmail.com LinkedIn: http://www.linkedin.com/pub/claves-do-amaral/a/b3b/1a9

2 3

I got FS "working" but......
by Don Ireland 24 Nov '13

24 Nov '13

I got freeswitch to run on my Raspberry Pi. And I'm getting a dialtone. But when I dial out, I just hear dead silence for about 2 minutes -- then I get just a fast busy signal. I went back to my RasPi and here's what FS put up when I tried to place the call including up to putting the phone back on the hook. 2013-11-16 12:20:47.855963 [INFO] mod_dialplan_xml.c:558 Processing R D IRELAND<2001>->12124182222 in context default 2013-11-16 12:20:47.935944 [NOTICE] switch_channel.c:1052 New Channel sofia/external/12124182222 [d0d91f94-4eeb-11e3-873e-c7fbb7f46cb4] 2013-11-16 12:21:19.975943 [NOTICE] sofia.c:6596 Hangup sofia/external/12124182222 [CS_CONSUME_MEDIA] [RECOVERY_ON_TIMER_EXPIRE] 2013-11-16 12:21:19.995953 [INFO] mod_dptools.c:3201 Originate Failed. Cause: RECOVERY_ON_TIMER_EXPIRE 2013-11-16 12:21:19.995953 [NOTICE] switch_channel.c:4643 Hangup sofia/internal/2001(a)192.168.0.14 [CS_EXECUTE] [RECOVERY_ON_TIMER_EXPIRE] 2013-11-16 12:21:20.015962 [NOTICE] switch_core_session.c:1567 Session 3 (sofia/internal/2001(a)192.168.0.14) Ended 2013-11-16 12:21:20.015962 [NOTICE] switch_core_session.c:1571 Close Channel sofia/internal/2001(a)192.168.0.14 [CS_DESTROY] 2013-11-16 12:21:20.035950 [NOTICE] switch_core_session.c:1567 Session 4 (sofia/external/12124182222) Ended 2013-11-16 12:21:20.035950 [NOTICE] switch_core_session.c:1571 Close Channel sofia/external/12124182222 [CS_DESTROY] -- Don Ireland

3 7

FAILED-CALLS-IN parameter
by Ashwin Jain 24 Nov '13

24 Nov '13

Hi, I am running freeswitch on EC2 medium instance with following configuration: Freeswitch Version: 1.2.53 Uname string: Linux freeswitch 3.2.0-27-virtual #43-Ubuntu SMP Fri Jul 6 14:45:58 UTC 2012 x86_64 x86_64 x86_64 GNU/Linux When I am checking the status of my configured profile, I am getting following stats: freeswitch@localhost@internal> sofia status profile mgprofile ================================================================================================= Name mgprofile Domain Name N/A Auto-NAT false DBName sofia_reg_mgprofile Pres Hosts Dialplan XML Context incoming Challenge Realm auto_to RTP-IP 10.113.51.12 Ext-RTP-IP 54.225.73.72 SIP-IP 10.113.51.12 Ext-SIP-IP 54.225.73.72 URL sip:mod_sofia@54.225.73.72:5060 BIND-URL sip:mod_sofia@54.225.73.72:5060;maddr=10.113.51.12 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN PCMU CODECS OUT PCMU 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 STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 649 FAILED-CALLS-IN 255 CALLS-OUT 531 FAILED-CALLS-OUT 37 REGISTRATIONS 0 Why are so many calls failing? What could be the issue? -- Thanks and Regards, Ashwin Jain

2 2

.Net 4.5.1 mod_managed
by Andre 23 Nov '13

23 Nov '13

Can the next release of Freeswitch up the framework to Mod_Managed to 4.5.1? Thanks Andre

2 1