Re: [asterisk-users] WSS over Asterisk
Hi I tested yesterday the SIPML5 fix and I can confirm it works as expected with Asterisk 12 SVN-trunk-r415192 using chan_sip and no DTLS enabled. Tested with Chrome 35.0.1916.153m. The patch is targeted to Chrome. Firefox still be unable to handle calls in my setup. In my tests I've found some asterisk exceptions when SIMPL5 is used from Chrome with the provided patch AND DTLS is configured for the peer in sip.conf AND certificates are installed in Chrome. I suppose this is something work in progress so I'm not worried about it. I can also confirm the problem with wss where the SIPML5 seems not able to connect to the asterisk box. Thank you and best regards, Marco Signorini. On 06/12/2014 03:21 AM, Steve Ng wrote: I am using Asterisk v12.3. As far as DTLS, I understand that applying the following Javascript will temporarily fix for SIPML5 to Asterisk: https://gist.github.com/steve-ng/14b9b88af43f92db1e46 WS works for me, its just wss which I'm stuck currently. On Thu, Jun 12, 2014 at 4:37 AM, Miguel Molina <mailto:mfmolina-lis...@millenium.com.co>> wrote: El 11/06/2014 1:52 p. m., Matthew Jordan escribió: On Wed, Jun 11, 2014 at 1:32 PM, William Hetherington mailto:w...@willwh.com>> wrote: Chrome 35 broke all of this you need to be using DTLS now I believe. I had working secure web sockets with asterisk 12.2.x and chrome 34 and then google broke eveything :) I have not yet got around to test out DTLS etc. with chrome 35 Just so I don't waste too much time when I go to test, does anyone know if all that's required for DTLS on the asterisk side is the following in sip.conf? dtlsenable=yes dtlsverify=yes dtlsrekey=60 dtlscafile=/usr/local/share/ca-certificates/myCA.crt dtlscertfile=/etc/ssl/mycert.com.pem dtlssetup=actpass I assume I also need TLS configs in http.conf Signalling is independent of the media; DTLS only affects the media. However, there are known issues with Chrome's negotiation of DTLS and Asterisk - see https://issues.asterisk.org/jira/browse/ASTERISK-22961 -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org It is broken in Chrome (firefox never had SDES) because the WebRTC standard favoured the DTLS SRTP implementation instead of the SDES one. The thing is that although Asterisk supports DTLS implementation, it only supports SHA-1 hashing but both Firefox and Chrome work with SHA-256. The patch proposed in ASTERISK-22961 is an effort to solve this issue. Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sipML5, Ast12 and WebRTC: not acceptable here
Hi All. I'm running some tests with the latest Asterisk SVN-branch-12-r410493M compiled with fresh github pjsip and srtp 1.4.2 on an i386 centOS machine (2.6.32-358.18.1.el6.i686). As a client I'm using the sipMLP WebRTC javascript softphone running on Chrome 33.0.1750.146 m. I have the softphone correctly registered on the Asterisk machine but as soon as I try to start a new call from the softphone, Asterisk answers with a 488 not acceptable here. I'm probably missing something but I'm not able to find what and where. Is there someone able to point me to the right direction? Below is my configuration. The sofpthone is registered as 1060. Thanks in advance. Marco Signorini. pjsip.conf: [transport-tls] type=transport protocol=tls bind=0.0.0.0 cert_file=/etc/asterisk/sslcert.pem method=tlsv1 [1060] type=endpoint transport=transport-tls context=from-internal use_avpf=yes media_encryption=sdes disallow=all allow=alaw allow=ulaw aors=1060 auth=1060 [1060] type=auth auth_type=userpass password=1060 username=1060 [1060] type=aor max_contacts=10 [204] http.conf: enabled=yes bindaddr=10.10.5.49 bindport=8088 CLI> pjsip show endpoints Endpoint: 1060 Not in use0 of inf InAuth: 1060/1060 Aor: 1060 10 Contact: 1060/sip:1060@10.10.5.106:54083;transport=ws;rt Unknown nan Transport: transport-tls tls 0 0 0.0.0.0:5061 Endpoint: 204 Not in use0 of inf InAuth: 204/204 Aor: 2041 Contact: 204/sip:204@10.10.5.120:5066;transport=udp Unknown nan Transport: transport-udp udp 0 0 0.0.0.0:5060 *CLI> pjsip show transport transport-tls Transport: = Transport: transport-tls tls 0 0 0.0.0.0:5061 ParameterName : ParameterValue == async_operations : 1 bind : 0.0.0.0:5061 ca_list_file : cert_file : /etc/asterisk/sslcert.pem cipher : cos: 0 domain : external_media_address : external_signaling_address : external_signaling_port: 0 local_net : method : tlsv1 password : priv_key_file : protocol : tls require_client_cert: No tos: CS0 verify_client : No verify_server : No And this is the relevant SIP data exchange (with public IP hidden): *CLI> <--- Received SIP request (2420 bytes) from WS:10.10.5.106:54411 ---> INVITE sip:204@10.10.5.49 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKiBw81ooU7ybSRbRqr8TOqWkMPQRdkMXo;rport From: "John Doe (101)";tag=heMv1HvlT7DeQxPxuqcq To: Contact: "John Doe (101)";impi=1060;ha1=0b2413e6f3c96a0517b4413a6f6ce7ae;+g.oma.sip-im;+sip.ice;language="en,fr,it" Call-ID: 636a5d79-5fda-f79a-cc4b-9ba18d060edc CSeq: 38718 INVITE Content-Type: application/sdp Content-Length: 1827 Max-Forwards: 70 User-Agent: IM-client/OMA1.0 sipML5 v=0 o=- 365893986064703740 2 IN IP4 127.0.0.1 s=Doubango Telecom - chrome t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS dXVhxyOSxULu3iClZayhTeEBzH2voboiJJ28 m=audio 37874 RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 85.0.XXX.XXX a=rtcp:37874 IN IP4 85.0.XXX.XXX a=candidate:296123718 1 udp 2113937151 10.10.5.106 63858 typ host generation 0 a=candidate:296123718 2 udp 2113937151 10.10.5.106 63858 typ host generation 0 a=candidate:3103388307 1 udp 1845501695 85.0.XXX.XXX 37874 typ srflx raddr 10.10.5.106 rport 63858 generation 0 a=candidate:3103388307 2 udp 1845501695 85.0.XXX.XXX 37874 typ srflx raddr 10.10.5.106 rport 63858 generation 0 a=candidate:1596293558 1 tcp 1509957375 10.10.5.106 0 typ host generation 0 a=candidate:1596293558 2 tcp 1509957375 10.10.5.106 0 typ host generation 0 a=ice-ufrag:l8AWdK4ft+AnAYGl a=ice-pwd:3tLKvT97tf0GQr+e8v8bKncd a=ice-options:google-ice a=fingerprint:sha-256 89:D4:EB:6E:9C:41:36:03:A1:44:CD:A2:08:78:CD:86:FE:EC:30:09:53:0F:77:CE:BA:8E:DE:8C:1B:A1:41:10 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=rtcp-mux a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:dYMEPP1zoNS/W70Ln6cnBCtHXDCq6ciLZmHDHdFj a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Gr23SpFGDiukOKyrrfAauWssQ+3pYjD0jwkK9hOo a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000
Re: [asterisk-users] DAHDI and Oslec
Hi. I had the same problem in the past and I've found that there was already an echo.ko module built in my kernel module folder. I've renamed it and replaced with the one compiled with dahdi+oslec and it started working as expected. I was on OpenSuse with kernel 2.6.27.56-0.1... it's very old so I can't tell you if this is something true for Debian 6.06 too. Thanks. Marco Signorini. On 02/26/2013 05:38 PM, Doug Lytle wrote: I'm hoping someone can help me here. I've purchased replacement systems for 3 aging 1.4.x installs. I'm hoping to setup Asterisk 11, dahdi 2.6.1 and Oslec. I'm also moving those installs from Mandriva 10.0 to Debian 6.06 (Squeeze). In my testing, the TE220P PCIe cards that I have, the timing was awful on both slots, so I compiled Kernel 3.6.9 from kernel.org. Timing jumped to what I was expecting, so I moved on to recompiling dahdi complete for Oslec. Browsed the Linux source directory for drivers/staging/echo and copied it to the proper tree in the dahdi complete directory. Did a make distclean;make clean;make all And everything compiled cleanly, including oslec. But, when trying to set my E.C. to oslec, I get: Feb 26 11:21:37 indyvoip modprobe: FATAL: Error inserting dahdi_echocan_oslec (/lib/modules/3.6.9-custom-3.6.9/dahdi/dahdi_echocan_oslec.ko): Unknown symbol in module, or unknown parameter (see dmesg) Feb 26 11:21:37 indyvoip kernel: [ 1395.262334] dahdi_echocan_oslec: Unknown symbol oslec_create (err 0) Feb 26 11:21:37 indyvoip kernel: [ 1395.262348] dahdi_echocan_oslec: Unknown symbol oslec_update (err 0) Feb 26 11:21:37 indyvoip kernel: [ 1395.262365] dahdi_echocan_oslec: Unknown symbol oslec_free (err 0) Feb 26 11:21:41 indyvoip modprobe: FATAL: Error inserting dahdi_echocan_oslec (/lib/modules/3.6.9-custom-3.6.9/dahdi/dahdi_echocan_oslec.ko): Unknown symbol in module, or unknown parameter (see dmesg) Feb 26 11:21:41 indyvoip kernel: [ 1398.799227] dahdi_echocan_oslec: Unknown symbol oslec_create (err 0) Feb 26 11:21:41 indyvoip kernel: [ 1398.799241] dahdi_echocan_oslec: Unknown symbol oslec_update (err 0) Feb 26 11:21:41 indyvoip kernel: [ 1398.799258] dahdi_echocan_oslec: Unknown symbol oslec_free (err 0) And dmesg shows: [ 1395.262334] dahdi_echocan_oslec: Unknown symbol oslec_create (err 0) [ 1395.262348] dahdi_echocan_oslec: Unknown symbol oslec_update (err 0) [ 1395.262365] dahdi_echocan_oslec: Unknown symbol oslec_free (err 0) [ 1398.799227] dahdi_echocan_oslec: Unknown symbol oslec_create (err 0) [ 1398.799241] dahdi_echocan_oslec: Unknown symbol oslec_update (err 0) [ 1398.799258] dahdi_echocan_oslec: Unknown symbol oslec_free (err 0) My Googlng-Fu failed me, as everything was dated from 201 0 and earlier on this error. I'm guessing that I compiled the kernel wrong, I followed these instructions to create .debs http://www.howtoforge.com/kernel_compilation_ubuntu Everything seemed to work well. Coming from a Mandrake/Mandriva background, I'm used to just: make oldconfig make menuconfig (Make my changes) make all make modules_install make install Any hints would be appreciated, Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USB or Ethernet based FXO device ?
Hi. I was following this thread. We normally use Patton SmartNode SN4112 series to interface to FXO ports. But I'm looking for something different for a future setup. Xorcom USB channel banks seems something quite interesting. Is there anyone that could/would share experiences using that? We need to replace an old PBX interfaced to 50 FXS and 8 BRI ISDN in Italy. My concern is about reliability of USB Any success stories with it? Tips and tricks? Thank you and regards, Marco Signorini. -- INGEGNI Tech S.r.l. site http://www.ingegnitech.com mail i...@ingegnitech.com Gilles wrote: On Tue, 30 Aug 2011 10:41:30 -0500, Carlos Chavez wrote: Actually Xorcom makes USB channelbanks of up to 32 FXO/FXS ports. Thanks for the tip. It looks like the smallest option is 8 FXO ports: www.xorcom.com/telephony-interfaces/astribank-models.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: Re: Asterisk as a Condo door opener/intercom
Hi.
To open the door I'm suggesting you to use Arduino if you can't have a
PC near the door opener contacts.
Arduino is something useful for implementing this type of networked
embedded stuffs.
It's not so expensive and easy to use for people familiar with C, C++
and a little bit of electronics.
You can find it worldwide by e-commerce and is very well supported by a
lot of open sourced libraries.
If you stack an Arduino UNO and an Ethernet Shield SD you'll have a
small embedded solution providing you at maximum of four hw based TCP
sockets that you can use for implementing, for example, a little web
server. The digital input/output pins could be used (if properly
buffered by a transistor) to drive a relay to be placed in parallel with
the door opener pushbutton.
To have the best reliability you can use the tricks suggested on the
pachube website, where someone suggest to drive the Ethernet shield
reset pin at regular intervals.
At the asterisk level I've implemented something similar to what's
explained by David. The only difference is that, in order to open the
door, I've used the CURL application to generate a suitable HTTP get to
the IP address associated to the Ethernet shield on top of Arduino.
Thanks,
Marco Signorini
--
http://www.ethermania.com
http://www.ingegnitech.com
David - asterisk list wrote:
> Asterisk as a phone system makes perfect sense in a condo. You can get
>
> all the DID's you want and eliminate costs for the owners. You can offer
> standard FXO for people who don't care and IP sets for people who want
> to "upgrade" to feature sets.
>
> Your door openner is a piece of cake.
> 1. Create an option in your dialplan only in the "from-access-door"
> context that reads DTMF from the called station only.
> 2. Use this to access an external program to turn on a serial port line
> for 10 seconds.
> 3. This line drives a solid state relay (~$30) so you won't blow the
> sink current on the PC port that drives a standard door lock.
>
> A commercial door strike is about $100. The program to run the port is
> childs play. Here is a test prog I used for turning on a power hungry
> last printer. Change the comments and the sleep time and you're done.
>
> /*
>* lpon Lineprinter ON
>* *** test program only **
>*
>* (c) David Cook, 1994
>*
>* Set signlal lines on serial port to turn on 5vdc
>* signal. Used for solid-state relay (low current
>* draw on RS232C port) to switch high voltage/high
>* current load for printer.
>*
>* Part of an intelligent print spooler to only power
>* on/off high draw printer when required.
>*
>* Usage: lpon
>* For example, lpon /dev/cua4 4 to set bit 3 on
>* port /dev/cua4.
>* "4" = 0100 or bit 3 which is DTR
>* "2" = 0010 or bit 2 which is RTS
>* "6" = 0110 or both DRT& RTS
>*/
> #include
> #include
> #include
> #include
> #include
> #include
> #include
> #include
> #include
>
> #include "lpswitch.h"
>
> /* Main program. */
> int main(int argc, char **argv)
> {
> struct termios port_config;
> int fd;
> int set_bits = 2;
>
> /* Open monitor device. */
> if ((fd = open(SWDEV, O_RDWR | O_NDELAY))< 0) {
> fprintf(stderr, "lpswtich: %s: %d\n", SWDEV, strerror(errno));
> exit(1);}
>
> cfmakeraw(&port_config );
> port_config.c_iflag=port_config.c_iflag|IXON;
> port_config.c_oflag=port_config.c_oflag|CLOCAL|~CRTSCTS;
> tcsetattr( fd, TCSANOW,&port_config );
> ioctl(fd, TIOCMSET,&set_bits );
>
> /* wait for printer to warm up */
> sleep(45);
>
> /* not say "ready" and release the printer */
> set_bits = 6;
>
> cfmakeraw(&port_config );
> port_config.c_iflag=port_config.c_iflag|IXON;
> port_config.c_oflag=port_config.c_oflag|CLOCAL|~CRTSCTS;
> tcsetattr( fd, TCSANOW,&port_config );
> ioctl(fd, TIOCMSET,&set_bits );
>
> close(fd);
> }
>
>
>
> On 12/04/2011 8:16 AM, asterisk-users-requ...@lists.digium.com wrote:
>> Message: 3
>> Date: Mon, 11 Apr 2011 18:21:39 -0500
>> From: "Don Kelly"
>> Subject: Re: [asterisk-users] Asterisk as a Condo door opener/intercom
>> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
>>
>> Message-ID:<8E20A6A94C9548C8A0E27B502B18F200@Don
Re: [asterisk-users] Asterisk to switch on electric heaters remotely?
Hi Did you looked at Arduino + Ethernet Shield? Is something you can program in C or C++ to receive a simple TCP and/or HTTP packet and turn on an external relay. >From the dialplan you can run an http query through curl and/or an external AGI command. Best regards, Marco Signorini. -- Marco Signorini http://www.ethermania.com http://www.ingegnitech.com Roberto Piola wrote: > we're using a Damocles Mini > (http://www.hw-group.com/products/damocles/damocles_mini_en.html). of > course, the damocles will have to drive a high-power relay. > > the damocles can be driven via snmp, so you'll have to simply call the > snmpset unix standard utility > > On Mon, Oct 18, 2010 at 1:24 PM, Gareth Blades > wrote: > >> Something like http://www.audon.co.uk/udin.html UDIN-8R. It can only >> control 750W so you will probably need to get it to control a more >> powerfull relay as a heater is going to take a lot of current. >> It can be controlled by a virtual serial port so you just program the >> extension to make a system() call to a simple script which sends a >> string of characters to the serial port. >> >> That device is quite expensive. You could probably find something much >> cheaper on ebay. >> >> >> Gilles wrote: >> >>> Hello >>> >>> I'm sure someone has already tried this: I use a couple of electric >>> heaters to heat my office. >>> >>> I'd like to somehow connect them to Asterisk so that I could switch >>> them on remotely by either calling the IVR or sending an e-mail to the >>> Asterisk host, so that the room is warm when I get to the office :-) >>> >>> Any information appreciated. >>> >>> Thank you. >>> >>> >>> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > > > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with Dahdi 2.3.0.1 trying to load OSLEC
Hello Jose. I've found the same problem on some servers and I solved it renaming (or deleting) the echo.ko driver already present in the binary kernel distribution: In my system is something like: /lib/modules/2.6.27.45-0.1-default/kernel/drivers/staging/echo/echo.ko Hope this helps you. Best regards, Marco Signorini. -- = - http://www.ethermania.com - - http://www.ingegnitech.com - Jose P. Espinal wrote: > Hello list, > > > I'm facing a little issue with dahdi attempting to load the OSLEC echo > canceller into my current kernel. > > After compiling dahdi 2.3.0.1 with OSLEC support, I get the following > error when set 'oslec' as the echocanceller: > > DAHDI_ATTACH_ECHOCAN failed on channel 1: Invalid argument (22) > > - Similar errors are *NOT* present using other echo canncelers. > - I tried adding the 'dahdi_echocan_oslec' line to /etc/dahdi/modules > and the error continues. > > I'm running Slackware Linux 13.0, Kernel 2.6.29.6-smp > > # dmesg > ... > dahdi_echocan_oslec: Unknown symbol oslec_create > dahdi_echocan_oslec: Unknown symbol oslec_update > dahdi_echocan_oslec: Unknown symbol oslec_free > ... > > > Could someone point me to some documentation about this incident? > > > Regards, > > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SuSE Firewall2 - Port Forward Command
Brent A. Torrenga wrote: > > Does anyone know what commands in the config file for a SuSE Firewall > will forward 5060 and RTP ranges to an Asterisk box in the internal LAN? > > > > > I think you need to play with the parameter FW_FORWARD_MASQ in the /etc/sysconfig/SuSEfirewall2 For example: FW_FORWARD_MASQ="0/0,192.168.10.1,udp,5060,80,192.168.2.3" lets you able to forward the udp 5060 from the IP 192.168.10.1 to 192.168.2.3 You need to add all the other RTP relevant rules. Best regards. Marco Signorini -- = EtherMania di Signorini Marco For network enthusiast people - http://www.ethermania.com - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] verifying correct loading of VPMADT032
In some motherboards I've found was not possible to assign different IRQs through BIOS and other software ways. This was related to some technical choices in that particular hardware. In these situations, the only profitable solution was to swap the cards between PCI connectors until a better configuration was found. Regards, Marco. -- http://www.ingegnitech.com http://www.ethermania.com Greg Woods wrote: > On Sat, 2010-01-02 at 20:25 +0100, F6HQZ wrote: > > >> cat /proc/interrupts >> Search the Digium cards drivers and look if several interfaces are using the >> same IRQ number. >> If yes, you risk issues and data losses >> > > What can I do if there is a sharing going on? Looks like my TDM card is > sharing it's IRQ with the video card, and I've been having some > occasional problems with it. > > --Greg > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to use patgen and pattest for PRI card?
Hi, I used the patgen and pattest some months ago to test two PRI cards in two different servers connected together. I don't remember now the details, but for sure I was using a PRI cross cable between the two cards and a loopback connector when testing the single board. I remember I had a lot of troubles and not so good results, even if the two cards are still working perfectly under the normal condition (they are in production since the end of march handling several 10 thousands minutes a month). Please, refer to http://lists.digium.com/pipermail/asterisk-users/2009-March/227920.html and to http://lists.digium.com/pipermail/asterisk-dev/2009-March/037003.html Best regards, Marco Signorini === INGEGNI Tech S.r.l. http://www.ingegnitech.com Chris YM wrote: > hello: > I wan to use the test tools-patgen and pattest for pri cards. > according to Tzafrir Cohen from > http://docs.tzafrir.org.il/man/pattest.8.html, i still does not know > how to use that. > do i need to connect two pri cards with two servers, or use a cable to > connect two cards in one server? > please give me a more details in term of cables and configurations. > thanks! > Chris > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo and static on PRI with errors
Dave Platt wrote: >> Could someone tell me how to set which IRQ the ISDN card picks up? >> > > > It's a multi-stage process. > > Each PCI slot has four interrupt pins: INTA through INTD. A > PCI card can choose to use any of these four (or even more than > .. > bridge architecture might be forcing interrupts from some cards > to use a single line/IRQ. > > > Thank you for your complete description on how PCI IRQ subsystem works. It's probably the best explanation I've found since years. My warm compliments, you've my best appreciation. Regards, Marco Signorini. Ingegni TECH S.r.l. http://www.ingegnitech.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo and static on PRI with errors
Tom O'Connor wrote: > > > On Wed, Jul 1, 2009 at 7:37 AM, Francesco Peeters > mailto:france...@fampeeters.com>> wrote: > > John F. Ervin wrote: > > What do you do if you find things sharing interrupts (IRQ 11) in my > > case with my X100P card. I believe there is some sort of internal > > audio card in my cheap slow PC. > > > Check the BIOS whether you can: > Change the IRQ assignments > Disable the extra hardware using the same IRQ > > Or otherwise try changing the slot it is in... I had very good results > in the past swapping card around > > Good luck! > > > I did a bit of investigation WRT the IRQ settings on this box. > > 00:02.1 USB Controller: nVidia Corporation CK804 USB Controller (rev > a3) (prog-if 20) > Subsystem: Hewlett-Packard Company Device 3207 > Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 11 > -- > 01:05.0 VGA compatible controller: nVidia Corporation NV11 [GeForce2 > MX/MX 400] (rev b2) > Subsystem: Hewlett-Packard Company Device 3207 > Flags: bus master, 66MHz, medium devsel, latency 64, IRQ 11 > -- > 02:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5721 > Gigabit Ethernet PCI Express (rev 11) > Subsystem: Hewlett-Packard Company Device 3209 > Flags: bus master, fast devsel, latency 0, IRQ 11 > -- > 81:01.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN > interface > Subsystem: Device 79fe:0001 > Flags: bus master, medium devsel, latency 64, IRQ 11 > > So basically there's 2 network cards and a USB controller sharing IRQ > 11 with the Openvox card. > > I wasn't able to find any settings in the bios to manually configure > IRQ assignments :( > > Could someone tell me how to set which IRQ the ISDN card picks up? > > -- > Tom O'Connor > > http://www.twinhelix.org > t...@twinhelix.org <mailto:t...@twinhelix.org> Hi, Unfortunately is not always possible and it depends on how the mainboard was realized. For what I can understand a lot of producers decide to route only a subset of physical IRQ lines to the PCI slots (I think is something related to cost reduction) and to share it with other onboard peripherals. This lets impossible to change the IRQ assignment for expansion cards. This is not always true and sometimes swapping add-on cards solves the problem. We had better results with cards based on new Digium technology or with Sangoma cards. Best regards, Marco Signorini. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peer 'iaxfax' is now UNREACHABLE! Time: 3
Lee Howard wrote: > Marco wrote: > >> I've IAXModem and asterisk Asterisk 1.4.24 running on the same machine. >> They are linked together through localhost. I've turned qualify on for the >> iax peer. Periodically I've this message: >> >> [Apr 20 23:47:46] NOTICE[4641]: chan_iax2.c:9049 __iax2_poke_noanswer: >> Peer 'iaxfax' is now UNREACHABLE! Time: 3 >> [Apr 20 23:47:56] NOTICE[4632]: chan_iax2.c:8128 socket_process: Peer >> 'iaxfax' is now REACHABLE! Time: 3 >> >> It happens a lot of times during the day, even when the box is not loaded >> at all. >> > > What does iaxmodem say? (Look at the iaxmodem logs.) > I've some "Registration timed out" events but I think they are not related to this problem (even because they are less than the unreachable/reachable events). The strange think is that, when "UNREACHABLE", the reported Time is 3 (I think milliseconds) and it's the same that's reported when the peer became reachable. Is this a little bit strange? Thank you and best regards, Marco Signorini. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing success rate on PRI
Thank you, Doug, for precious information. Best regards, Marco Signorini. === INGEGNI Tech S.r.l. http://www.ingegnitech.com Doug Lytle wrote: > > Main fax server: > > > Mandriva 2008.1 > Kernel 2.6.24.5 (Compiled for source) > (1) Intel(R) Xeon(TM) CPU 2.80GHz > Digium TE110P (23 b channels 1 data) > > Asterisk 1.4.20.1 > HylaFAX+ 5.2.7 > iaxmodem 1.2.0 > SpanDSP 0.0.4 (The one that came with iaxmodem) > > Doug > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing success rate on PRI
Hi Steve, I was waiting for your answer :-P I started to use your SpanDSP library since some years ago but, unfortunately, my experience was only related to lab or personal use and/or systems with PSTN or BRI cards and low fax volume where it's impossible to have valid statistics. I read the link you provided me and now I'm confident that IAXModem running on the same asterisk box with a PRI board is something I can propose to my customer. There are some other variable I would like to evaluate, like, for example, what type of PRI connection people of this list are using in their fax servers. Thank you for writing SpanDSP and best regards, Marco Signorini === INGEGNI Tech S.r.l. http://www.ingegnitech.com Steve Underwood wrote: > Marco Signorini wrote: > >> Thank you to All People answered me on this subject. >> Analyzing your answers, seems that fax handling is still today >> problematic with IAXModem and Hylafax... or I'm wrong? >> What about other solutions? >> >> > I'm not sure where you got that idea. Most comments about iaxmodem + > HylaFAX are very positive. It does, of course, require a reliable > connection between iaxmodem and the PSTN, but most people set these > things up in a reasonably well controlled environment. There are some > notes at http://www.soft-switch.org/spandsp-soft-fax-performance.html > about the results of serious real world volume testing of iaxmodem. > > Regards, > Steve > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing success rate on PRI
Thanks Doug and Lee, your testimonials are changing my opinion :-) Can you provide some details about your setup? What PRI solution are you using? And what version of Asterisk, IAXModem, SpanDSP? Thank you and best regards, Marco Signorini === INGEGNI Tech S.r.l. http://www.ingegnitech.com Lee Howard wrote: > Marco Signorini wrote: > >> Analyzing your answers, seems that fax handling is still today >> problematic with IAXModem and Hylafax... or I'm wrong? >> > > A single server that I administer, receiving 12,000 pages and sending > 1,000 pages daily would seem to contradict your conclusions. > > Thanks, > > Lee. > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing success rate on PRI
Hi Gordon, thank you for your answer. It's not mandatory to use an external box to handle the PRI. I was thinking to use a Patton device instead of a TE120P just because I would like to be able to switch to T38 in the near future or if working with inband faxes will reveal problems. I'm open to any suggestion, being in the initial requirements analysis stage. External devices, also, let me able to select a server independently by the PCI, PCIXpress or any other connectors will be ready in the future, maximizing the customer investment... but this could be the less interesting part. Thank you to All People answered me on this subject. Analyzing your answers, seems that fax handling is still today problematic with IAXModem and Hylafax... or I'm wrong? What about other solutions? Thank you and best regards, Marco Signorini === INGEGNI Tech S.r.l. http://www.ingegnitech.com Gordon Henderson wrote: > On Sun, 8 Mar 2009, Marco wrote: > > >> Hi List, >> I've been using PSTN-ATA + Asterisk + IAXModem + Hylafax since three years >> on my lab test setup and I appreciate it. Moreover the global quantity of >> fax handled by this setup is not very high. >> >> I'll be involved in a more complex system for a customer and I would like >> to ask to All of you if you have experiences and/or statistical results on >> faxing success and failure rate. >> >> The system I have to deploy will operate in the following context: >> >> - It will be interfaced to an E1 PRI >> - It will be able to send and receive faxes (by e-mail and/or virtual >> printers) >> - It will be able to send faxes from a normal fax machine. >> >> The system will be placed on the same building, i.e. only private ethernet >> trunks. >> >> I'm thinking to this type of solution: >> - Patton external unit for E1 >> > > Out of curiosity, why an external box rather than something like a TE120P > PCI card? > > > >> - Asterisk 1.4 + IAXModem + Hylafax >> - An external ATA for the fax machine >> but I'm open to any other possible solution (I'm thinking to have a >> demodulation on Patton and talk T38 with Asterisk 1.6). >> > > Personally, I think you're adding complexity and can't see why that would > be better than an on-board PRI card... > > Gordon > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] faxing via linksys SPA3102 half page goes through
Joseph wrote: > On 03/04/09 15:44, Marco Signorini wrote: > >> Hi Joseph. >> I've spent some time tuning the SPA3102 FXS line input and output gain >> and I think that this is an important variable. >> Let's try to record incoming and outgoing fax tones with asterisk on SIP >> channel (disabling the fax detection on SPA and sending fax inband) and >> look at the recorded file with a wave editor (Audacity). >> I had better results if the maximum level is near half to the full >> dynamic. Then switch to T38, if you need it. >> > > As I remember I have experimented with gain on PSTN line as well but have > reset back to default. > I have: > SPA To PSTN Gain:0 > PSTN To SPA Gain:0 > > I think "0" is the default. > > Yes, "0" is the default. Is the fax machine connected to the FXS port or do you use the SPA3102 only as a SIP 2 PSTN gateway? If you use the FXS port, please take a look at the gain parameters you can find in the "Miscellaneous" section in the "Regional" page (log in as Administrator then switch to the "advanced" report). Now I've -5 as input gain and -2 as output. I don't know if this could helps you. Best regards, Marco Signorini === INGEGNI Tech S.r.l. http://www.ingegnitech.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] faxing via linksys SPA3102 half page goes through
Hi Joseph. I've spent some time tuning the SPA3102 FXS line input and output gain and I think that this is an important variable. Let's try to record incoming and outgoing fax tones with asterisk on SIP channel (disabling the fax detection on SPA and sending fax inband) and look at the recorded file with a wave editor (Audacity). I had better results if the maximum level is near half to the full dynamic. Then switch to T38, if you need it. Hope this helps you. Best regards, Marco Signorini === INGEGNI Tech S.r.l. http://www.ingegnitech.com Joseph wrote: > On 03/04/09 19:31, Michael wrote: > >> On Wed, 04 Mar 2009 19:25:38 Joseph wrote: >> >>> I'm faxing from stand alone fax machine via linksys SPA3102 but most of >>> the time only half or quarter page goes through. >>> >>> Did anybody have any experience like this? >>> >> Should be obvious but does your up line SIP provide support T.38? >> > > What do you mean it should be obvious? > > I think Linksys SPA3102 does support T.38 > On Line 1 I have: > FAX Enable T38: Yes > FAX T38 Redundancy: 1 > FAX Passthru Codec: G711u > FAX Process NSE: Yes > FAX Passthru Method: NSE > FAX CNG Detect Enable: Yes > FAX CED Detect Enable: Yes > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] patlooptest and TE121P
Hi List. I'm running the patlooptest program I've found in dahdi_tools 2.1.0.2. The target is a TE121P board with a loopback cable inserted on the socket. I suppose that the loopback is working fine because I'm able to see the green led on and dahdi_tool reports no errors. When I run the patlooptest I've a lot of errors (the received values are completely different than the transmitted) and I would like to know if someone in the list had run this test with the TE121P board. This lets me understand if the problem is on my board. The setup I've is what I've found on http://kb.digium.com/entry/138/ for the E1. Thank you and best regards, Marco Signorini === INGEGNI Tech S.r.l. http://www.ingegnitech.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] building a phone
Jon Pounder wrote: > Marco Signorini wrote: > >> It's a dream! >> It's since years that I'm thinking to have an open hardware project >> targeted to a SIP application. >> >> > > there is already a project called openmoko - join it and buy some hardware. > > The phone is large and clunky - the idea is good, but not something > you're ever going to carry in your pocket, and somewhat silly when there > is already smaller hardware out there that runs linux at less cost than > their device. > > Thank you Jon, Really interesting project! I'll follow it. Best regards, Marco Signorini ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] building a phone
It's a dream! It's since years that I'm thinking to have an open hardware project targeted to a SIP application. I'm thinking, for example, to have a modular system that can be targeted to different custom appliances like, for example, (video) door bell opener/intercom, or building/desktop music streamer, or SIP compliant actuators. I have a (very) little experience on electronic projects. Is there something I can do to help starting a similar project? Thank you and best regards. Marco Signorini Tzafrir Cohen wrote: > Hi folks > > A common wisdom here is that one should use a proper hardware phone > rather that an extra software on the user's PC. Why is that such a big > issue? > > One thing that bothers me with the current crop of hardware SIP phones > is that they are hopelessly properitary. > > So what would it take to build a fully-adaptable phone? > > Here are some of my thoughts. This is not anything I plan to do soon (if > at all), but I really find it strange that there aren't such phones > already. > > > == Small Quantities: > When you look at such systems it becomes aparant that you can get much > nicer prices if you buy large quanities. But this is something that will > be a problem. Not only for prototying. The fact that you're limited to a > strict hardware setting is very limiting. No mixing and matching like in > a standard PC. I'm not exactly sure how to overcome that. > > == Platforms: > There are many embedded platforms nowadays. I assume that the relevant > application requires some non-trivial CPU power. I would exclude e.g. a > 486-based systems. My target phone should be able to handle at least two > concurrent Speex calls. Preferrebly 6 speex calls and above. > > OTOH, I can't afford a monster CoreDuo. I need a quiet system with no > fan. Thus the target CPU may be higher end VIA or Atom. Not sure about > Geode. > > There are also some interesting ARM-based boards around. I'm completely > unfamiliar with them but I suspect that they may prove to be cheaper. > > == SIP Software: > Not really sure here. There must be something close to usable already, I > guess. > > == Micro Browser: > Hell no! > > The device should have an LCD display, and the content of that display > should be programmable. Programming it using a HTML renderred is a bad > design decision. > > The device should be a good phone. It should not attempt to be a web > browser, as it will be a lousy one. > > == Handset: > I suppose that an obvious starting point for a handset is "skype phones" > such as USB handsets from yealink. Far from an optimal design, but a > driver already exists. > > > == Ease of Use: > A phone must be usable. The target device must be something my mom can > use. However that does not mean it must be easy to program. It must be > programmable and hackable. But I can live with a complicated user > interface for that. If such phones become successful and useful, better > interfaces will eventually be written. > > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR - Asterisk-Stat and PHP5
Hi Paulo, It's right! I've changed the zend.zel_compatibility_mode to Off, following your suggestion, and asterisk-stat is still working on PHP5. Thank you! Just for clarity: the default values for the two keys on OpenSuse 10.2 (updated to latest revision), and following, is "Off". Best regards, Marco Signorini === INGEGNI Tech S.r.l. http://www.ingegnitech.com Tiago Durante wrote: > On Thu, Feb 26, 2009 at 12:52 PM, Paulo Santos wrote: > >> Marco Signorini wrote: >> >>> Hi Tiago. >>> >>> I've it working on PHP 5.2.6 but only after having modified the php.ini >>> default configuration keys: >>> >>> zend.ze1_compatibility_mode = Off >>> short_open_tag = Off >>> >> Though my zend.ze1_compatibility_mode is set to Off, short_open_tag is >> set to On and it is working. >> >> Those are my defaults, at least I never changed them. Installed with >> apt-get on Debian 4.0, PHP version 5.2.0-8+etch13. >> > > Cool, I'm gonna test it and I let you guys know if worked or not. > > Thanks a lot! > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR - Asterisk-Stat and PHP5
Hi Tiago. I've it working on PHP 5.2.6 but only after having modified the php.ini default configuration keys: zend.ze1_compatibility_mode = Off short_open_tag = Off setting together to On and restarting apache forces PHP5 to behave like PHP 4.x version. regards, Marco Signorini === INGEGNI Tech S.r.l. http://www.ingegnitech.com Paulo Santos wrote: > Tiago Durante wrote: > >> Hi all, >> >> I don't know if its the right place to ask, but... Does any one have >> the asterisk-stat-v2 running with PHP5? >> >> >> Tks! >> >> >> > > # php --version > > PHP 5.2.0-8+etch13 (cli) (built: Oct 2 2008 08:26:18) > Copyright (c) 1997-2006 The PHP Group > Zend Engine v2.2.0, Copyright (c) 1998-2006 Zend Technologies > > Working for me. Don't forget you need php5-gd for the graphics to show. > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Suggestion for a new server for E1 line
Hi All, I'm trying to identify a new server as a replacement for what our customer actually has (DELL PowerEdge 860). The server will mount the Digium board TE121, we already have, with echo cancel onboard. I need to know if someone could suggest a new server that's compatible with this board. With "compatible" I mean that's not having any problem like IRQ sharing, IRQ miss or kernel panic with DAHDI/Zaptel drivers or unwanted hangup or noise during the conversation. My other needs are: 1. At least two RAID disk (preferably hot-swappable but I'm looking also for solution without this feature) 2. Possibly with redundant power supply 3. 1U or 2U size 4. As cheap as possible. Our customer is pushing to have the HP Proliant DL120 but I think it's not fitting the 24/7 needs it has. The server will be used to dispatch calls coming from an 800 free number for one humanitarian organization in Italy. Any suggestion is really welcomed. Thank you very much. Best regards, Marco Signorini http://www.ingegnitech.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Portech MV-378 with Asterisk
Yes. That's the correct way to do it. Placing # as a rule in callnum forces the Portech to use the number defined in the SIP INVITE packet. Bye. Marco. Marco Signorini INGEGNI Tech S.r.l. http://www.ingegnitech.com <http://www.ingegnitechcom/> Pascal Bruno wrote: > Sorry for bothering you, but I got it, I just had to put # in callnum! > > > > On Sat, Jan 17, 2009 at 1:44 AM, Pascal Bruno <mailto:tipas...@gmail.com>> wrote: > > I want to dial out using the sim card. What I did, I have used > the SIP channel ex: > > Channel: SIP/thenum...@mv378 > > It shows the called is being made in the dialplan, but the number > I have entered does not dial, it just goes straight to the > specified dialplan extensions. > > Then what I did, in the Lan to Mobile Table, I put * in url and > the number I wanted to dial in call num, then the call was made to > that number using the sim card properly. > > I was wondering if I cannot supply the number to be dialed using > an asterisk call file, or do I have to put that number in the Lan > to Mobile table. > > Any help would be appreciated. > > Thanks > > > > > > On Sat, Jan 17, 2009 at 12:39 AM, Pascal Bruno <mailto:tipas...@gmail.com>> wrote: > > Marco, > > The configs work fine for me. I can receive calls with no > problem. Now, were you able to dial using the sim card? I > cant figure out how I can do it since asterisk doesnt have a > channel to place call through the portech gateway. > > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Portech MV-378 with Asterisk
Pascal Bruno wrote: > Thanks for your reply! > > Can you tell me what you have in your Portech configuration settings > (Mobile to Lan Settings; Sip Proxy settings etc...) My sip.conf file > is pretty similar to yours but still cant register. > > > > On Fri, Jan 16, 2009 at 3:47 AM, Marco Signorini <mailto:marcota...@libero.it>> wrote: > > Emmanuel Pascal Bruno wrote: >> Has anyone been able to configure portech's mv-378 gateway with >> asterisk? >> >> I did the configuration as per the manual but it does not work. >> >> My server sees the portech gateway, but when the gateway is >> trying to register to my server it fails. It says peer is not >> suppose to register. >> >> The gateway and the asterisk box are on two different location >> (two network, 2 differrent IP address). >> >> I would appreciate any kind of tutorial or advice on how to make >> it work. >> >> Thanks >> >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com >> <http://www.api-digital.com/> -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > Hi, > I've an installation working with Portech MV-370. I'm supposing > it's quite similar to what you have. If it could be useful to you, > this is my sip.conf configuration file. > > [GSMGtw1] > type=friend > context=from-gsm > host=dynamic; we have a DHCP assigned address > secret=reallyverysecret > nat=no ; there is not NAT between phone > and Asterisk > canreinvite=no > dtmfmode=INFO > insecure=invite ; required to overcome > authentication problems in incoming calls > call-limit=1 ; permit only 1 outgoing call > at a time > disallow=all > allow=ulaw > allow=alaw > allow=gsm > qualify=500 > > I remember that I've found a bug on the firmware that prevents to > the unit to register correctly on my asterisk box unless I'm using > the raw IP address instead of the name of the asterisk box. I > remember something wrong in cryptography chiper/dechiper based on > realm... So, if you have problems, let's try to specify the > asterisk raw IP address in the Portech. > > Best regards, > Marco Signorini. > > Hi, I don't know if the problem could be in the Mobile to Lan or Lan to Mobile settings because these settings are related on how calls coming from/to mobile are routed. I didn't use the Portech routing features at all because I need a simple GSM gateway to/from the asterisk box. For this reason: 1. The only rule I've on Mobile to Lan is CID=*; url=...@192.168.0.5 where "mob" is the extension I've generated in the asterisk box under the context where the Portech operates; 2. The only rule I've on Lan to Mobile is URL=*; Call Num=# I think the most relevant parameters for your problem are under the "Service Domain" menu option (assuming that the firmware you have is similar to what I've). On this menu I've compiled the 1st Realm (as I've only one account) like that: UserName: GSMGtw1 RegisterName: GSMGtw1 RegisterPassword: reallyverysecret Domain Server: 192.168.0.5 Proxy Server: 192.168.0.5 Pay attention that, having specified the Domain Server with the raw IP address, asterisk needs to be able to authenticate peers associated to that. For this reason I've set: domain=192.168.0.5 on sip.conf [general] section (remember to issue a sip reload from asterisk cli). Hope this helps! Best regards. Marco Signorini Marco Signorini INGEGNI Tech S.r.l. http://www.ingegnitech.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Portech MV-378 with Asterisk
Emmanuel Pascal Bruno wrote: > Has anyone been able to configure portech's mv-378 gateway with asterisk? > > I did the configuration as per the manual but it does not work. > > My server sees the portech gateway, but when the gateway is trying to > register to my server it fails. It says peer is not suppose to register. > > The gateway and the asterisk box are on two different location (two > network, 2 differrent IP address). > > I would appreciate any kind of tutorial or advice on how to make it work. > > Thanks > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users Hi, I've an installation working with Portech MV-370. I'm supposing it's quite similar to what you have. If it could be useful to you, this is my sip.conf configuration file. [GSMGtw1] type=friend context=from-gsm host=dynamic; we have a DHCP assigned address secret=reallyverysecret nat=no ; there is not NAT between phone and Asterisk canreinvite=no dtmfmode=INFO insecure=invite ; required to overcome authentication problems in incoming calls call-limit=1 ; permit only 1 outgoing call at a time disallow=all allow=ulaw allow=alaw allow=gsm qualify=500 I remember that I've found a bug on the firmware that prevents to the unit to register correctly on my asterisk box unless I'm using the raw IP address instead of the name of the asterisk box. I remember something wrong in cryptography chiper/dechiper based on realm... So, if you have problems, let's try to specify the asterisk raw IP address in the Portech. Best regards, Marco Signorini. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Oslec issue
Ok Joseph. Don't worry, take your time :-) For what's concerning the quality: I can assume my phone line is an exception because it has a lot of echo. I've spent a LOT of time trying to have an SPA3102 and an HT488 working without any reasonable result. I'm playing for fun with zaptel/dahdi ec's since years and I was never able to have a satisfying result with any ec provided with it. Neither the fxotune process, neither any Tx/Rx gain or echo training parameter tuning, neither Digium people that connected to my server 3 or 4 years ago were able to completely solve any echo issue. Some years ago I had the opportunity to test on my line HPEC with a customer's box equipped with a TDM400P and I was impressed by the quality. I think OSLEC is a very good piece of code. It's working fine with my line and my Grandstream phones and, the must important thing, it's open and free to use. Sometimes I can ear some echo or strange effects at the very beginning of a call, but this is something that I can accept. In the past I tried to modify the zaptel sources in order to prevent them to free the oslec instance at each call. I think that my mods were not working on systems where more than one zap channel was present and I was not able to test it on these type of situations. Thank you and bye Marco Signorini Joseph L. Casale wrote: >> I spent some time to understand what's missing in the OSLEC patch for >> dahdi... I can confirm the same problem you reported some days ago and I >> need OSLEC for home personal use. >> > > Wow, > Appreciate the info! I will need a few days to get this done. Out of > curiosity, > how do you find this ec's quality compared to the shipped modules and hpec? > > Thanks! > jlc > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Oslec issue
Hi Joseph.
I spent some time to understand what's missing in the OSLEC patch for
dahdi... I can confirm the same problem you reported some days ago and I
need OSLEC for home personal use.
For what I've understood looking at the code, there is some missing in
the dahdi_echocan_oslec.c file you can find in the
dahdi-linux/drivers/dahdi. I can list below what I did to have it
working. Actually I'm using the trunk revision 5443.
1. In the function "echo_can_create" i've modified the line
*ec = (struct echo_can_state *)oslec_create(ecp->tap_length,0);
with
*ec = (struct echo_can_state *)oslec_create(ecp->tap_length,
ECHO_CAN_USE_ADAPTION | ECHO_CAN_USE_NLP | ECHO_CAN_USE_CLIP |
ECHO_CAN_USE_TX_HPF | ECHO_CAN_USE_RX_HPF);
This instructs OSLEC to have a working modality properly set.
2. I've replaced the function "echo_can_update" with the code below:
static void echo_can_update(struct echo_can_state *ec, short *iref,
short *isig)
{
unsigned int SampleNum;
for (SampleNum = 0; SampleNum < DAHDI_CHUNKSIZE; SampleNum++,
iref++)
{
short iCleanSample;
iCleanSample = (short) oslec_update((struct oslec_state
*)ec, *iref, *isig);
*isig++ = iCleanSample;
}
}
This lets the OSLEC to work on complete DAHDI_CHUNKSIZE buffer.
Please, if you have time, let me know if this solves your problem and,
if yes, I'll appreciate to have it public on trunk. I never did a commit
on asterisk svn so I need some hints on how to do it.
Thank you and best regards.
Marco Signorini.
Joseph L. Casale wrote:
> Yesterday I pulled in the latest svn of Dahdi and added the files
> from a recent kernel in the drivers/staging/echo structure and modified
> the Kbuild file so it would compile without error. I insmod'ed the module
> in, and modified my system.conf has echocanceller=oslec.
>
> cat /proc/dahdi/1 shows:
> Span 1: WCTDM/0 "Wildcard TDM410P Board 1" (MASTER)
> IRQ misses: 1
>
>1 WCTDM/0/0 FXSKS (In use) (EC: OSLEC)
>2 WCTDM/0/1
>3 WCTDM/0/2
>4 WCTDM/0/3
>
> With the reco's from http://www.rowetel.com/ucasterisk/oslec.html#install on
> configuring the chan_dahdi.conf file, the system behaves exactly as if there
> is no ec enabled at all?
>
> Are there any additional steps needed to enable oslec under dahdi, I am
> guessing
> I have missed something?
>
> Thanks,
> jlc
>
>
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Re: [asterisk-users] Problem with DAHDI and OSLEC integration.
Hi Joseph and Tzafrir. Thank you for your suggestions, feedbacks. For Joseph: yes. I had the same warning messages and I solved with the trick I suggested. Now oslec seems working (or at least and I can set it through the dahdi_cfg command ;-) ). For Tzafrir: here are the steps I did: 1. Taken the svn revision 5366 into my temporary folder /home/marco/Install/dahdi-linux 2. Taken the linux-2.6.27 kernel sources baseline and placed in my temporary folder /home/marco/install/linux-2.6.27 3. Taken the Linux kernel patch-2.6.28-rc6.gz, unzipped and applied to the baseline kernel 2.6.27. This generates the folder ...linux-2.6.27/drivers/staging/echo 4. Copied the folder /staging/echo into /home/marco/Install/dahdi-linux/drivers 5. Uncommented the oslec related two lines in the file Kbuild 6. From the folder /home/marco/install/dahdi-linx I've issued the command make The compiler starts and seems not able to compile what's present in the folder /home/marco/Install/dahdi-linux/drivers/staging/echo. This produces the warning already reported by Joseph and the inability to run the oslec module. I've had better results modifying the line: obj-m += ../staging/echo/ with obj-m += ../staging/echo/echo.o in the Kbuild file. I don't know if could be helpful, but I'm running these stuffs on OpenSuse 11. Thank you and best regards, Marco Signorini. Joseph L. Casale wrote: >> Have you copied there the files from the directory drivers/staging/echo >> in a recent (that is: >= 2.6.28-rc1) kernel tree? >> > > Tzafrir, > Thank you for following up on this. I don't have a quick command for only > the three files, I just grabbed the tar ball. But like the OP, the only > difference was that he used 2.6.28-rc6 and I used 2.6.28-rc5. I am pretty > sure we had the same errors which I posted: > http://lists.digium.com/pipermail/asterisk-users/2008-November/222063.html > > Thanks for any pointers! > jlc > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > Return-Path: <[EMAIL PROTECTED]> > Received: from mailrelay07.libero.it (192.168.32.94) by ims67c.libero.it > (8.0.019) > id 489AF14B0071194C for [EMAIL PROTECTED]; Mon, 24 Nov 2008 00:45:09 > +0100 > X-IronPort-Anti-Spam-Filtered: true > X-IronPort-Anti-Spam-Result: > AkgAABB6KUnYz/URlGdsb2JhbACBbZFvAQEBAQkLCAkRBLlNgnyBVA > X-IronPort-AV: E=Sophos;i="4.33,655,1220227200"; >d="scan'208";a="576251938" > Received: from lists.digium.com ([216.207.245.17]) > by mailrelay07.libero.it with ESMTP; 23 Nov 2008 23:45:08 + > Received: from localhost ([127.0.0.1] helo=INXS.digium.internal) > by lists.digium.com with esmtp (Exim 4.63) > (envelope-from <[EMAIL PROTECTED]>) > id 1L4OY6-0007qt-UX; Sun, 23 Nov 2008 17:39:31 -0600 > Received: from idcmail-mo2no.shaw.ca ([64.59.134.9]) > by lists.digium.com with esmtp (Exim 4.63) > (envelope-from <[EMAIL PROTECTED]>) id 1L4OXy-0007qj-L3 > for asterisk-users@lists.digium.com; Sun, 23 Nov 2008 17:39:22 -0600 > Received: from pd6ml1no-ssvc.prod.shaw.ca ([10.0.153.160]) > by pd7mo1no-svcs.prod.shaw.ca with ESMTP; 23 Nov 2008 16:39:17 -0700 > X-Cloudmark-SP-Filtered: true > X-Cloudmark-SP-Result: v=1.0 c=0 a=Kd8dHRva:8 a=OvmmjL8WYY_iC51gqwYA:9 > a=pC6ppZV0J0nBJUWzJgg687EU0CoA:4 a=YPYbZooERpMA:10 > a=AKRigw6aElYA:10 > Received: from s0106001e8c610de2.cg.shawcable.net (HELO > mail.activenetwerx.com) ([68.144.97.215]) > by pd6ml1no-dmz.prod.shaw.ca with ESMTP; 23 Nov 2008 16:39:16 -0700 > Received: from exchange.activenetwerx.com (mail.activenetwerx.com [127.0.0.1]) > by mail.activenetwerx.com (Postfix) with ESMTP id 6EC8768136 > for ; > Sun, 23 Nov 2008 16:39:24 -0700 (MST) > Received: from exchange.activenetwerx.com ([192.168.0.3] > helo=exchange.activenetwerx.com) > by mail.activenetwerx.com; 23 Nov 2008 16:39:24 -0700 > Received: from Mail.activenetwerx.int ([::1]) by Mail.activenetwerx.int > ([::1]) with mapi; Sun, 23 Nov 2008 16:39:15 -0700 > From: "Joseph L. Casale" <[EMAIL PROTECTED]> > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > > Date: Sun, 23 Nov 2008 16:39:14 -0700 > Thread-Topic: [asterisk-users] Problem with DAHDI and OSLEC integration. > Thread-Index: AclNwtn/mA/EFuLiQUqEGjb+9HhEWQAABJlg > Message-ID: <[EMAIL PROTECTED]> > References: <[EMAIL PROTECTED]> <[EMAIL PROTECTED]> > In-Reply-To: <[EMAIL PROTECTED]> > Accept-Langu
[asterisk-users] Problem with DAHDI and OSLEC integration.
Hi List. I've bought a new server for my home asterisk installation and I'm trying to install asterisk with dahdi drivers and OSLEC. To do that I've got the svn dahdi-linux trunk revision 5366 and the echo subproject from the 2.6.28-rc6 Linux kernel sources. As reported in the dadhi README file I've uncommented out the two OSLEC related lines at Kbuild file in the dahdi-linux/drivers/dahdi folder. I don't know if there is a bug in the Kbuild trunk revision or if I did something wrong, but I was not able to successfully build the dahdi_echocan_oslec.ko until I've changed the line: obj-m += ../staging/echo/ with obj-m += ../staging/echo/echo.o With the original one I got some warning messages about oslec symbols not defined. I think that the builder was not able to find the oslec object file. Am I doing something wrong? Thank you and best regards. Marco Signorini ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ExtenSpy? am I doing it correctly?
Hi Steve.
I'm still trying the same because I'm interested in the subject.
For what I can understand the ExtenSpy application is working properly
if the selected extension receives a call. Seems not working, instead,
if the selected extension originates the call.
My actual setup is like that:
Ext12(Soggiorno) <==> Ext13(Camera)
^
|
Ext911-> ExtSpy(12)
Here is the log when the 13 calls the 12 and 911 is called by an other
phone (StudioAV):
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/Camera-08231e60",
"SIP/Soggiorno") in new stack
-- Called Soggiorno
-- SIP/Soggiorno-082560f8 is ringing
-- SIP/Soggiorno-082560f8 answered SIP/Camera-08231e60
-- Packet2Packet bridging SIP/Camera-08231e60 and SIP/Soggiorno-082560f8
-- Executing [EMAIL PROTECTED]:1] ExtenSpy("SIP/StudioAV-0822f350",
"12") in new stack
-- Playing 'beep' (language 'it')
-- Playing 'spy-sip' (language 'it')
== Spying on channel SIP/Camera-08231e60
Unfortunately, in the opposite direction:
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/Soggiorno-0822f350",
"SIP/Camera") in new stack
-- Called Camera
-- SIP/Camera-08231e60 is ringing
-- SIP/Camera-08231e60 answered SIP/Soggiorno-0822f350
-- Packet2Packet bridging SIP/Soggiorno-0822f350 and SIP/Camera-08231e60
-- Executing [EMAIL PROTECTED]:1] ExtenSpy("SIP/StudioAV-082560f8",
"12") in new stack
-- Playing 'beep' (language 'it')
== Spawn extension (from-sip, 911, 1) exited non-zero on
'SIP/StudioAV-082560f8'
== Spawn extension (from-sip, 13, 1) exited non-zero on
'SIP/Soggiorno-0822f350'
The application ExtSpy seems to hang just before playing the 'spy-sip'
and I can't hear anything coming from the selected extension.
I'm using Asterisk version "Asterisk 1.4.20.1 built by root @ Gateway on
a i686".
Is this the correct behavior or a bug?
Thank you and best regards.
Marco Signorini.
Steve Gladden wrote:
> Scratching my head and trying this.
> Asterisk Version: Asterisk 1.4.21.2
>
> Tried:
> exten => 4771,1,ExtenSpy([EMAIL PROTECTED])
> exten => 4771,2,Hangup
>
> Also tried:
> exten => 4771,1,Answer
> exten => 4771,2,ExtenSpy([EMAIL PROTECTED])
> exten => 4771,3,Hangup
>
> Also tried many variations including option ,b
> I think most calls I make are 'bridged'
> extensions 4771 and 4724 are both in mbb context.
> Tried 'cycling' though the channels and volule "*" "#" no change.
>
> Test:
> 4724 places outbound or extension call.
> I dial "4771" from 4772
> I expect to hear audio from 4724's in progress call but hear nothing.
> I hear a recording "beep" when I dial 4771.
> I expect to hear audio from call being made from ext. 4724
> I've obviously got this wrong or the feature is not working :-)
>
> Ao far I've been unable to find much information on the net of anyone
> documenting
> a problem or a working configuration.
> Is there something I'm completely missing here?
>
> Thanks!
>
> Steve
>
>
>
>
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[asterisk-users] t38modem on OpenSuse
Hi All, is there anyone that tried to work with the t38modem project integrated with SIP through OPAL libraries in OpenSuse 10.2? I followed the cookbook at http://www.voip-info.org/wiki/index.php?page_id=5096 and I've a strange behavior. Firs of all when the t38modem starts, I've an error message that I think is related to some library not present in my current OpenSuse installation (but I'm not able to understand which library is still requiring, if anyone is able to help me to understand what's happening I'll be very happy to hear him). The message is: "error loading avcodec - avcodec: cannot open shared object file: No such file or directory" Running a ldd ./t38modem all seems ok. The next problem arises when I send faxes through an HT386 ATA and asterisk 1.4.20.1. Looking at the network traffic through ethereal I can see that the t38modem answer to the first INVITE message with a 100 TRYING message.. but it never send an ACK. At the same time, the t38modem is producing the log I've attached below (sorry for the long post). Any help is appreciated. Thank you. Marco Signorini 2008/09/22 23:53:39.395 Opal Liste...er:80b95c8 SIP PDU Received on udp$192.168.0.5:5060 INVITE sip:[EMAIL PROTECTED]:6060 SIP/2.0 Date: Mon, 22 Sep 2008 21:53:39 GMT CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK2409d6cb User-Agent: Cadore 9 PBX From: "Soggiorno2" ;tag=as6bf57b61 Call-ID: [EMAIL PROTECTED] Supported: replaces To: Contact: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 508 Max-Forwards: 70 v=0 o=root 3222 3222 IN IP4 192.168.0.5 s=session c=IN IP4 192.168.0.5 t=0 0 m=audio 5018 RTP/AVP 0 97 3 8 112 5 10 7 18 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2008/09/22 23:53:39.400 Opal Liste...er:80b95c8 SDP Media session port=5018 2008/09/22 23:53:39.401 Opal Liste...er:80b95c8 SDP Adding media session with 11 formats 2008/09/22 23:53:39.402 Opal Liste...er:80b95c8 SDP Unknown media attribute silenceSupp:off - - - - 2008/09/22 23:53:39.405 Opal Liste...er:80b95c8 SIP Sending PDU on udp$192.168.0.5:5060 SIP/2.0 100 Trying CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK2409d6cb From: "Soggiorno2" ;tag=as6bf57b61 Call-ID: [EMAIL PROTECTED] To: Contact: Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE,INFO,PING,PUBLISH Content-Length: 0 2008/09/22 23:53:39.408 Opal Liste...er:80b95c8 CallCreated Call[4] 2008/09/22 23:53:39.409 Opal Liste...er:80b95c8 MySIPEndPoint::CreateConnection for Call[4] 2008/09/22 23:53:39.409 Opal Liste...er:80b95c8 OpalCon Created connection Call[4]-EP[EMAIL PROTECTED] 2008/09/22 23:53:39.410 Opal Liste...er:80b95c8 RFC2833 Handler created 2008/09/22 23:53:39.411 Opal Liste...er:80b95c8 RFC2833 Handler created 2008/09/22 23:53:39.415 Opal Liste...er:80b95c8 OpalUDP Binding to interface: 192.168.0.5:5651 2008/09/22 23:53:39.416 Opal Liste...er:80b95c8 SIP Created transport udp$0.0.0.0 2008/09/22 23:53:39.417 Opal Liste...er:80b95c8 OpalUDP Started connect to 192.168.0.5:6060 2008/09/22 23:53:39.418 Opal Liste...er:80b95c8 OpalUDP Connect on pre-bound interface: 192.168.0.5 2008/09/22 23:53:39.419 Opal Liste...er:80b95c8 PWLib Created thread 0x80e1690 SIP Transport:%x 2008/09/22 23:53:39.420 Opal Liste...er:80b95c8 SIP Created connection. 2008/09/22 23:53:39.421 Opal Liste...er:80b95c8 SIP Queueing PDU: 102 INVITE sip:[EMAIL PROTECTED]:6060 2008/09/22 23:53:39.422 Opal Liste...er:80b95c8 PWLib Created thread 0x80e3190 SIP Handler:%x 2008/09/22 23:53:39.422 Opal Liste...er:80b95c8 OpalTransport clean up on termination 2008/09/22 23:53:39.423 Opal Liste...er:80b95c8 OpalUDP Close 2008/09/22 23:53:39.423 Opal Liste...er:80b95c8 OpalDeleted transport udp$192.168.0.5:5060 2008/09/22 23:53:39.556 Opal Liste...er:80b95c8 Listen Waiting on UDP packet on udp$192.168.0.5:6060 2008/09/22 23:53:39.557 SIP Transp...rt:80e1690 PWLib Started thread 0x80e1690 SIP Transport:80e1690 2008/09/22 23:53:39.557 SIP Transp...rt:80e1690 SIP Read thread started. 2008/09/22 23:53:39.558 SIP Transp...rt:80e1690 SIP Waiting for PDU on udp$192.168.0.5:6060 2008/09/22 23:53:39.559 SIP Handle...er:80e3190 PWLib Started thread 0x80e3190 SIP Handler:80e3190 2008/09/22 23:53:39.559 SIP Handle...er:80e3190 SIP PDU handler thread started. 2008/09/22 23:53:39.560 SIP Handle...er:80e3190 SIP Awaiting next PDU. 2008/09/22 23:53:39.560 SIP Handle...er:80e3190 SIP Handling PDU 102 INVITE sip:[EMAIL PROTECTED]:6060 2008/09/22 23:53:39.563 SIP Handle...er:80e3190 SIP
Re: [asterisk-users] Magnetic door locks
Hi James I did it some years ago to open an electric door entry. I used an home-made board with embedded tcp-ip stack and a perl AGI scripts that sends some UDP packets containing a "secret" passphrase. The AGI it's still triggered by an internal extension call. I think that today you can find some reasonable price ethernet DIN rail industrial controls that provides HTTP capabilities and you can write a simple AGI script that generates some HTTP transactions to set the board rele' status to whatever you want. Best regards, Marco Signorini. c james wrote: > I have an opportunity to interface asterisk with a security system to > open their magnetic door locks. The security system needs a dry contact > close upon activation to signal the door. Has anyone done this before? > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel problem with pots lines
Hi Tzafrir, you're right. I think I've completely misunderstood the problem. If the problem is that asterisk is not able to write in the CDR the proper line answer status, I can confirm that even my installations behave the same. Sorry Enrico for my fault and thank you to Tzafrir for the correction. Best regards, Marco Signorini. Tzafrir Cohen wrote: > On Mon, Jul 14, 2008 at 08:56:40PM +0200, Enrico Maistro wrote: > >> Hi, >> >> I'm trying to get up and running a TDM400 with a standard italian pots >> line but i'm having >> problems at getting asterisk to detect when the line get answered on >> outgoing calls. >> > > AFAIK chan_zap can only detect answer if it is provided through a > polarity reversal. > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel problem with pots lines
Hi Enrico. I'm quite sure that the differences you have in the zapata.conf doesn't have any effect on the problem. If I'm not wrong: "language=it" tells asterisk to use the italian sounds (if available) for any calls related to this zap channel; "rxgain = 0.0" is related only to perceived audio gain "jbenable = no" is forcing the jitter buffer off for this zap channel. Let us know if you have problems with zaptel-1.4.6. I can assure that with this version and this configuration more than one installation with TDM400P I'm responsible for is working fine (since 1.4.6 came out). Could be that the problem is related to Asterisk 1.6? Unfortunately I never had the possibility to try this new version. Best regards, Marco Signorini. Enrico Maistro wrote: > My zapata.conf differs in: > language = it instead of en > rxgain = 0.0 instead of 3.0 > jbenable = no instead of yes > > Unfortunatly even with your exact same configuration nothing change. > > >> I'm using zaptel-1.4.6 and asterisk-1.4.20.1. >> > > I'll give a try with zaptel-1.4.6... > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel problem with pots lines
Hi Enrico. In Italy the polarity reversal is never used. I'm using the TDM400 with an FXO port in Italy with the config reported below and is working properly in any situations: --- zaptel.conf --- fxsks=1 loadzone=it defaultzone=it --- zapata.conf --- [channels] language=en context=from-tdm-fxo signalling=fxs_ks threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes busydetect=yes busycount=6 usecallerid=yes callerid=asreceived echocancel=yes echocancelwhenbridged=no relaxdtmf=yes rxgain=3.0 txgain=0.0 amaflags=billing group=1 callgroup=1 pickupgroup=1 jbenable=yes faxdetect=no channel => 1 I'm using zaptel-1.4.6 and asterisk-1.4.20.1. I hope this could help you. Best regards, Marco Signorini. Enrico Maistro wrote: > Hi, > > I'm trying to get up and running a TDM400 with a standard italian pots > line but i'm having > problems at getting asterisk to detect when the line get answered on > outgoing calls. > > I'm using asterisk 1.6 beta 9 with zaptel 1.4.11. > > I tried with and without answeronpolarityswitch=yes but it didn't change > anything at all. > > With callprogress=yes answer get never detected. > With callprogress=no line get answered as soon as it start ringing, > regardless if someone > really answer the call. > > > Zaptel channels use fxs_ks signalling . > > Loading wctdm module with debug=1 result in: > > kernel: Freshmaker version: 73 > kernel: Freshmaker passed register test > kernel: ProSLIC on module 0, product 3, version 15 > kernel: VoiceDAA System: 04 > kernel: ISO-Cap is now up, line side: 03 rev 06 > kernel: setting FXO tx gain for card=0 to 0 > kernel: setting FXO rx gain for card=0 to 0 > kernel: DEBUG fxotxgain:0.0 fxorxgain:0.0 > kernel: Module 0: Installed -- AUTO FXO (FCC mode) > kernel: ProSLIC on module 1, product 0, version 0 > kernel: VoiceDAA System: 04 > kernel: ISO-Cap is now up, line side: 03 rev 06 > kernel: setting FXO tx gain for card=1 to 0 > kernel: setting FXO rx gain for card=1 to 0 > kernel: DEBUG fxotxgain:0.0 fxorxgain:0.0 > kernel: Module 1: Installed -- AUTO FXO (FCC mode) > kernel: ProSLIC on module 2, product 0, version 0 > kernel: Module 2: Not installed > kernel: ProSLIC on module 3, product 0, version 0 > kernel: Module 3: Not installed > kernel: Found a Wildcard TDM: Wildcard TDM400P REV I (2 modules) > kernel: 4294908325 Polarity reversed (0 -> -1) > kernel: 4294908326 Polarity reversed (0 -> 1) > kernel: BATTERY on 1/1 (-)! > kernel: NO BATTERY on 1/2! > > Any suggestion? > Am i trying to do something that simply can't be done? > > Thanks, > Enrico Maistro > > > ___ > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
