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Question -- DTMF from RFC2833 to SIP INFO
by Dan Lan 29 Aug '11

29 Aug '11

Hi, I am new to the Freeswitch community, just compiled my first FS window version one month ago. I have registered two SNOM 300 IP-Phone with extension 1001, 1002 into FS. I also setup an external gateway as SIP trunk to service provider. (For certain reason, the service provider only accept INFO as the DTMF signal.) I add the following line into the external gateway profile <param name="dtmf-type" value="info"/> I also add the following line into the dail-plan <action application = "set" data="dtmf_type=info"/> When I dail the phone number from extension 1001, the channel bridge through the SIP trunk with both-way audio no problem. However, when I enter the DTMF, the signal did not send to the provider SIP trunk with SIP INFO. from the log file, I can see the DTMF was recorgnized by FreeSwitch from extension 1001 (it was sending RFC2933), and I can see FS try to send signal to the external trunk. (but no SIP INFO or RFC2833 in Wireshark capture) [DEBUG] switch_rtp.c:3303 RTP RECV DTMF 1:1280 2011-08-29 15:44:10.081713 [DEBUG] switch_ivr_bridge.c:391 Send signal sofia/external/2131234567@xxx.xxx.xxx.xxx:5060 [BREAK] 2011-08-29 15:44:11.033081 [DEBUG] switch_rtp.c:3303 RTP RECV DTMF 2:1120 2011-08-29 15:44:11.033081 [DEBUG] switch_ivr_bridge.c:391 Send signal sofia/external/2131234567@xxx.xxx.xxx.xxx:5060 [BREAK] 2011-08-29 Did I do something wrong here? Is FS capable of convert RFC2833 signal from one leg and send SIP INFO to another leg?

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Dialed number identification service
by Aakash 29 Aug '11

29 Aug '11

Hi, Is this possible - Dialed number identification service in freeswitch. For eg: I have configured my two inbound number to the same extension(eg 1000).When calling party dials any of the two inbound number should able to show the DNIS to the routed extension(eg 1000). Thanks Aakash -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dialed-number-identification-… Sent from the freeswitch-users mailing list archive at Nabble.com.

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freeswitch php
by Sam 29 Aug '11

29 Aug '11

Does anyone know how to retrieve channel variables (ie. uuid, etc.) using the php example that was shown in the wiki below? #!/usr/bin/php -q <?php // set a couple of things so we dont kill the system ob_implicit_flush(true); set_time_limit(30); // Open stdin so we can read the data in $in = fopen("php://stdin", "r"); // Connect echo "connect\n\n"; // Answer echo "sendmsg\n"; echo "call-command: execute\n"; echo "execute-app-name: answer\n\n"; // Play a prompt echo "sendmsg\n"; echo "call-command: execute\n"; echo "execute-app-name: playback\n"; echo "execute-app-arg: /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav\n\n"; // Wait sleep(5); // Hangup echo "sendmsg\n"; echo "call-command: hangup\n\n"; fclose($in); ?>

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Strange problem with Freeswitch and external extension
by Charles Bujold 29 Aug '11

29 Aug '11

I have a remote extension which registers with the Freeswitch server. The user can record his greeting in the Freeswitch IVR without a problem. Yet if the user calls me I cannot hear him, but he can hear me. I thought at first that it was a NAT issue so we placed the remote telephone directly on the internet and the same problem remains. The Freeswitch server is behind a NAT/router and I can see the packets from the phone coming into the router and being forwarded to the Freeswitch server. The router is not blocking the packets and is forwarding them to the correct server IP. I them placed wireshark on the Freeswitch server and I can see the packets leave the Freeswitch server and going to the phone But I cannot see the Freeswitch server receive the packets from the PFsense router, even though I can see in the router that the packets are being forwarded properly to the Freeswitch server IP. The Freeswitch server is running Ubuntu with no firewall, so in theory I should see the server accept or reject the packets. Yet I do not see any in wireshark coming from the phone, only those leaving from the Freeswitch server. What I do not understand is why, since he is registered, I cannot hear him, but yet he can interact with Freeswitch and record a greeting? Obviously packets are getting routed properly and Freeswitch recognizes them, some of the time. Can somebody suggest a possible explanation and solution to this problem? Thanks cjb

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CallWithUs
by Robert Huddleston 29 Aug '11

29 Aug '11

I already reviewed the wiki - does anyone have experience with CallWithUs and using IP Auth / not registration based?

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Looking up variable from config
by Rob Hutton 29 Aug '11

29 Aug '11

Is there any way of looking up a variable from the XML config from the dialplan? For instance, I know what number is dialed from ${destination_number} and I know that the directory user entry attribute of ID matches the number because I have it set up that way. Can I lookup the value email-addr parameter from the user and, say, set a variable to the value? Something like an XPATH lookup?

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mod_conference - 2-digit DTMF and events
by Michael Kemp 29 Aug '11

29 Aug '11

I have set up a custom caller-controls group in the my conference.conf.xml, as follows <caller-controls> <group name="mick"> <control action="event" digits="#1"/> <control action="event" digits="#2"/> <control action="event" digits="#3"/> <control action="mute" digits="#6"/> </group> </caller-controls> I have a lua script running to detect these events and perform actions according to which digits it receives. However, unless I press both the 2nd digit very quickly after the first one the event is not raised Is there any way of tuning the inter digit delay for caller-control groups so that multi-digit DTMF sequences are handled correctly?

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freetdm api - ftdm_channel_read()
by Juraj Fabo 29 Aug '11

29 Aug '11

Hi I'm prototyping an application which uses libfreetdm above the sangoma isdn stack and sangoma a104d card. I would like to ask for help about proper reading of voice from the particular channel. According to return value of sangoma_get_rx_queue_sz() the default rx queue size is 10. I did tests when application skipped several reads on given channel, then called ftdm_channel_read() w/ zero timeout in a loop while the return value of read was successfull - it was really 10 frames which were then possible to read. How can an application do this reading of a channel by the means of the freetdm api in more nice way? I did not find a better way yet, than to read it in the loop. Any way to explicitly read the number of available frames? Actually, why the ftdm_channel_read() does not read them ALL at once and sets the number of read bytes via *datalen output parameter? (assuming provided *data buffer is large enough) In my tests, always 160 bytes were returned in one particular read. thank you juraj

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Re: [Freeswitch-users] Error compling jsxml.c
by Jerre Cope 29 Aug '11

29 Aug '11

Be sure you have these packages installed: * git-core subversion build-essential autoconf automake libtool libncurses5 libncurses5-dev libjpeg62-dev libcurl4-gnutls-dev libglobus-openssl-dev libglobus-openssl-module-dev * for perl scripts will need: libcurses-perl On 08/29/2011 08:38 AM, freeswitch-users-request(a)lists.freeswitch.org wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > > Today's Topics: > > 1. Error compling jsxml.c (D Foulkes) > 2. Invitation to connect on LinkedIn (Martin Armstrong via LinkedIn) > 3. Invitation to connect on LinkedIn (Martin Armstrong via LinkedIn) > 4. Inband DTMF and bind_meta_app problem again (Dmitry Sytchev) > 5. Build problem with mod_python CentOS 5.5 Python 2.7 (John O'Brien) > 6. Re: Problem with receiving a NOTIFY after sending a SUBSCRIBE > request (Tom Fayette) > > > _______________________________________________ > 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

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Problem with receiving a NOTIFY after sending a SUBSCRIBE request
by Lorem Ipsum 29 Aug '11

29 Aug '11

Hello, I'm testing a SIP stack for an embedded device. The device, among other things, is capable of informing a user about pending messages on the voicemail. It does that by subscribing to the message-summary. Below some wireshark traces (172.16.30.68 is my device, 172.16.31.10 is FreeSWITCH): REGISTER sip:172.16.31.10 SIP/2.0 Via: SIP/2.0/UDP 172.16.30.68:5080 ;rport;branch=z9hG4bKPj051b000000035ea40edf Route: <sip:172.16.31.10:5060;lr> Max-Forwards: 70 From: <sip:399510002@172.16.31.10>;tag=051b000000025ea40edf To: <sip:399510002@172.16.31.10> Call-ID: 051b000000015ea40edf CSeq: 1 REGISTER Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY User-Agent: My_Sip_Device Contact: <sip:399510002@172.16.30.68:5080>;transport=udp Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.16.30.68:5080 ;rport=5080;branch=z9hG4bKPj051b000000035ea40edf From: <sip:399510002@172.16.31.10>;tag=051b000000025ea40edf To: <sip:399510002@172.16.31.10>;tag=mQrcXUcvrtUmS Call-ID: 051b000000015ea40edf CSeq: 1 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-38e3f5f 2011-08-09 03-09-19 -0400 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="172.16.31.10", nonce="3139f996-c814-11e0-93d2-05aa0ee343d6", algorithm=MD5, qop="auth" Content-Length: 0 REGISTER sip:172.16.31.10 SIP/2.0 Via: SIP/2.0/UDP 172.16.30.68:5080 ;rport;branch=z9hG4bKPj051b000000065ea40edf Route: <sip:172.16.31.10:5060;lr> Max-Forwards: 70 From: <sip:399510002@172.16.31.10>;tag=051b000000045ea40edf To: <sip:399510002@172.16.31.10> Call-ID: 051b000000015ea40edf CSeq: 2 REGISTER Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY User-Agent: My_Sip_Device Contact: <sip:399510002@172.16.30.68:5080>;transport=udp Authorization: Digest username="399510002", realm="172.16.31.10", nonce="3139f996-c814-11e0-93d2-05aa0ee343d6", uri="sip:172.16.31.10", response="1e95409a562c074cbe6df148a85107ef", algorithm=MD5, cnonce="051b000000055ea40edf", qop=auth, nc=00000001 Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.30.68:5080 ;rport=5080;branch=z9hG4bKPj051b000000065ea40edf From: <sip:399510002@172.16.31.10>;tag=051b000000045ea40edf To: <sip:399510002@172.16.31.10>;tag=N0H5ypXZN3H7m Call-ID: 051b000000015ea40edf CSeq: 2 REGISTER Contact: <sip:399510002@172.16.30.68:5080>;transport=udp;expires=180 Date: Tue, 16 Aug 2011 14:29:47 GMT User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-38e3f5f 2011-08-09 03-09-19 -0400 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 SUBSCRIBE sip:172.16.31.10 SIP/2.0 Via: SIP/2.0/UDP 172.16.30.68:5080 ;rport;branch=z9hG4bKPj051b0000000a5ea40edf Max-Forwards: 69 From: "399510002" <sip:399510002@172.16.31.10>;tag=051b000000085ea40edf To: <sip:172.16.31.10> Contact: <sip:399510002@172.16.30.68:5080> Call-ID: 051b000000095ea40edf CSeq: 1 SUBSCRIBE Event: message-summary Accept: application/simple-message-summary Allow-Events: message-summary User-Agent: My_Sip_Device X-Serialnumber: LMZ091218000026 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY Route: <sip:172.16.31.10:5060;lr> Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.30.68:5080 ;rport=5080;branch=z9hG4bKPj051b0000000a5ea40edf From: "399510002" <sip:399510002@172.16.31.10>;tag=051b000000085ea40edf To: <sip:172.16.31.10>;tag=p9ay0He3jc8Sg Call-ID: 051b000000095ea40edf CSeq: 1 SUBSCRIBE Contact: <sip:@172.16.31.10:5060> Expires: 60 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-38e3f5f 2011-08-09 03-09-19 -0400 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=60 Content-Length: 0 After sending the 200 OK, FreeSWITCH does not send the NOTIFY. If you look at the Contact header of the answer to the SUBSCRIBE you will notice that the part before the "@" is missing. I guess this is because SUBSCRIBE request does not contain the whole URI, just the host part. That is because our customer wants it done this way; the request line should look like this: SUBSCRIBE sip:voicemail_server SIP/2.0 and the To: header should look like this: To: <sip:voicemail_server> My question is: how can I make FreeSWITCH (the NOTIFY part anyway, other things are working OK) work with such a device? Thanks. Regards, Tom

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