November 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Answer A-Leg only when B-Leg answers
by Grant Bagdasarian 26 Nov '13

26 Nov '13

Hello, I'm trying to bridge the incoming leg to an outgoing leg. The incoming leg has to be answered only when the outgoing leg is answered. Is this possible? If yes, how would I do this using Dialplan XML?

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recommendations for Wifi SIP phones?
by Nathan Neulinger 26 Nov '13

26 Nov '13

Anyone got any recommendations for decent quality wifi sip phones? Obviously the 'android+soft client' model will work, but what about actual handsets built for the task. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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Global variable in dialplan
by Ali Pey 25 Nov '13

25 Nov '13

Hello, Is there a way to access a global variable in the xml dialplan and have a condition on it? Regards, Ali Pey

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Route header not copied from originating to terminating leg.
by Thomas Titty 25 Nov '13

25 Nov '13

Hi, I am trying to configure FS as an application server in an IMS network. When the S-CSCF forwards a SIP INVITE to FS it adds a Route header of FS followed by a Route header of S-CSCF so that the FS sends the INVITE back to S-CSCF. I have added dialplan in FS to use the fs_path with S-CSCF as the proxy to send back the INVITE. Now, FS adds a new Route header with S-CSCF but does not include the Route heard in the original INVITE. How can I make FS to copy the Route header from the originating leg to the terminating leg. Thanks, Thomas

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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

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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

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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

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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

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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

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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

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