Re: [asterisk-users] Hardware issue, was Using HASH() and REALTIME_HASH()
On 22/01/2010 19:10, Benoit wrote:
Le 13/01/2010 09:57, Benoit a écrit :
Le 12/01/2010 16:35, Tilghman Lesher a écrit :
On Tuesday 12 January 2010 04:44:36 Benoit wrote:
I just experienced another problem however i have two rnis cards, one
b410p and one te220,
while the later works prefectly i can't really make the first one work,
using DAHDI or mISDN
under asterisk 1.6.
If you're having trouble with any Digium hardware, the best thing to do is
to
call Digium support and get your free installation support provided with our
hardware.
Hi,
I didn't think of this, since it looked like more of an asterisk problem
(asterisk 1.4/misdn = ok asterisk 1.6/misdn = fail, asterisk 1.6/dahdi
= fail).
Audio (both way) is working (voicemail/playback), but it fail when
Dial'ing a device.
Looks like a problem with signalling ...
But anyway i just opened a support case, thanks
Well, in fact it wasn't an hardware issue: when calling thru the B410p
the callerid string is prepended with
an Id, looks like the length of the resulting string is a problem to
initial a SIP call.
Hell, it's even more simple, it was the double quote in the Set() (
Set(CALLERID(name)=- ID - ${CALLERID(name)}) ) that rendered
the sip message invalid ...
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Re: [asterisk-users] IAX ans SS7
ok thanks a lot Alex, I will check SIP-T RFCs Mickael 2010/1/23 Alex Balashov abalas...@evaristesys.com On 01/22/2010 07:12 PM, mickael ropars wrote: Hi all, what is the signalling of IAX? Currently I want to connect two switch through IP using asterik signaling, and I want to transfer SS7 over IP (between the 2 asterisk), will IAX can transfer SS7 signalling through IP (like TDMoIP does) If no which solution can I use? see below the architecture switch1-Asterisk--Asterisk --Switch E1 IP E1 To some extent, yes, but only very basically. IAX2 doesn't have the ability to pass through ISUP attributes. For that, you would need SIP-T. -- Alex Balashov - Principal Evariste Systems LLC Tel: +1 678-954-0670 Direct : +1 678-954-0671 Web: http://www.evaristesys.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 -- _ -- 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] Siemens Gigaset + Asterisk Query?
On Sat, Jan 23, 2010 at 08:08:28AM +0100, Philipp von Klitzing wrote: Hi! I was wondering if you can use the base station as a outbound pots connection for asterisk. I currently have a tdm410 to do fxs/fxo ports and would like to get rid of it, I used to use a spa3102, but it only had 1 fxo (telephone connector). I like the idea of the siemans but I would like to control the pots fallover from asterisk. if not the siemans are there any other bases that would fit the bill ? The AVM Fritz!Box 7270 could do the job, but I am not sure if you can get that with an English language web interface. DECT and SIP registrar (for LAN only) are available with a recent firmware. This box might be a bit oversized for what you are trying to do, though. German wife :), nice (the box) but not exactly what I wanted, I like seperate adsl modem, linux firewall, dlink ap, lets me control things Philipp -- Keybuk Perl 6 scares me doogie you can name your operators anything. the name here is the string '~|_|~' * Lo-lan-2 runs away screaming Keybuk it looks like a diagram of a canal lock :) jaybonci japanese smiley operators? nickr ^_^ -- in #debian-devel signature.asc Description: Digital signature -- _ -- 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] Xorcom problem after update from zaptel to dahdi-2.2.1
Hello, I am trying to update to dahdi 2.2.1. But somehow the xorcom (2BRI) device does not want to work. I pasted below the information I have. p...@loicvoip1:~# /usr/share/dahdi/xpp_fxloader reset p...@loicvoip1:~# /usr/share/dahdi/xpp_fxloader usb 'xpp_fxloader'[4937]: - FIRMWARE LOADING: (usb) [1 devices] ..Got all 1 devices 'xpp_fxloader'[5003]: - FIRMWARE IS LOADED p...@loicvoip1:~# /usr/share/dahdi/xpp_fxloader load 'xpp_fxloader'[5015]: - FIRMWARE LOADING: (load) [1 devices] Got all 1 devices INFO: usb:005/004: ID=E4E4:1141 [Xorcom LTD / Astribank / ] INFO: Loading hexfile to FPGA: /usr/share/dahdi/FPGA_1141.hex (version 6799) INFO: usb:005/004: ID=E4E4:1141 [Xorcom LTD / Astribank / ] INFO: Load PIC: /usr/share/dahdi/PIC_TYPE_1.hex (version 7498) INFO: Load PIC: /usr/share/dahdi/PIC_TYPE_2.hex (version 7107) INFO: Load PIC: /usr/share/dahdi/PIC_TYPE_3.hex (version 7107) INFO: Load PIC: /usr/share/dahdi/PIC_TYPE_4.hex (version 7308) INFO: usb:005/004: ID=E4E4:1141 [Xorcom LTD / Astribank / ] astribank_usb.c:366: ERROR(astribank_close): Releasing interface: usb: could not release intf 1: No such device ..Got all 1 devices 'xpp_fxloader'[5132]: - FIRMWARE IS LOADED p...@loicvoip1:~# lsusb Bus 005 Device 005: ID e4e4:1142 Bus 005 Device 001: ID : Bus 004 Device 001: ID : Bus 003 Device 001: ID : Bus 002 Device 001: ID : Bus 001 Device 001: ID : p...@loicvoip1:~# dahdi_hardware -v usb:005/005 xpp_usb- e4e4:1142 Astribank-BRI FPGA-firmware p...@loicvoip1:~# ls /proc/bus/usb/ -R /proc/bus/usb/: 001 002 003 004 005 devices /proc/bus/usb/001: 001 /proc/bus/usb/002: 001 /proc/bus/usb/003: 001 /proc/bus/usb/004: 001 /proc/bus/usb/005: 001 005 Thanks in advance, Loic Didelot. -- _ -- 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] fax over IP - http/ftp-provisioning - intercom
Dear members of the list, a customer of mine has some questions and I would like to pose some of them further to you guys. What is the situation with Asterisk and fax over IP ? Can Asterisk receive a fax over a POTS or ISDN line ?? Do I then need a Digium TDM-card and an FXO-module or a T38-gateway ? What phones should I use to automatically configure them from a central place via HTTP or FTP ? I know Polycom-phones offer this option. How can I implement intercom functionality ? Which phones have auto-answer ? And how do I implement this in the Asterisk dialplan ?? I'm used to working with Asterisk 1.4. I hope the above is possible in an 1.4 environment. Thank you for your feedback. Jonas. -- _ -- 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] Jabber Server
Hello, I need to get the status of SIP clients (Available,busy,Ideal) ,i have integrated open fire (XMPP server) with Asterisk, i tested it with XMPP client spark web client, the point now i need to work with SIP client (Xlite,SJphone) so how can i handle this issue? Thanks -- Ahmed Magdy Mahmoud -- _ -- 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] Attempted break in ?
H323 seemed to be enabled by default, so I just disabled the H.323 module as we do not use it. Rob How did you disable it? I dont see any module containing h323 in its name. (ast. 1.9) Martin -- _ -- 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] odd issue with the with SIP over VPN
I've run into a odd issue where inbound calls to the SIP client work fine, but outbound from the SIP client do not. The path between the client and the server is as below. N900 SIP client -- OpenVPN -- Asterisk The version of Asterisk in question is 1.6.0.18. Any suggestions? signature.asc Description: PGP signature -- _ -- 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] odd issue with the with SIP over VPN
You're going to be a lot more specific about the precise - if symptomatic - meaning of do not. On 01/23/2010 09:08 PM, Zane C.B. wrote: I've run into a odd issue where inbound calls to the SIP client work fine, but outbound from the SIP client do not. The path between the client and the server is as below. N900 SIP client-- OpenVPN -- Asterisk The version of Asterisk in question is 1.6.0.18. Any suggestions? -- Alex Balashov - Principal Evariste Systems LLC Tel: +1 678-954-0670 Direct : +1 678-954-0671 Web: http://www.evaristesys.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] fax over IP - http/ftp-provisioning - intercom
On 01/23/2010 09:11 AM, jonas kellens wrote: Dear members of the list, a customer of mine has some questions and I would like to pose some of them further to you guys. What is the situation with Asterisk and fax over IP ? Can Asterisk receive a fax over a POTS or ISDN line ?? Do I then need a Digium TDM-card and an FXO-module or a T38-gateway ? Despite what anyone may say about Fax over IP allegedly works for them, save yourself the trouble and make sure you take the POTS and ISDN approach. What phones should I use to automatically configure them from a central place via HTTP or FTP ? I know Polycom-phones offer this option. Most major phone vendors offer provisioning of this nature. How can I implement intercom functionality ? Which phones have auto-answer ? And how do I implement this in the Asterisk dialplan ?? Polycom supports this, and I am quite sure Snom does too. Perhaps some others. As far as how to implement it, that is manufacturer-specific. Look on voip-info.org for Polycom and paging if you want the Polycom-centric answer. For other phones, it will be different. -- Alex Balashov - Principal Evariste Systems LLC Tel: +1 678-954-0670 Direct : +1 678-954-0671 Web: http://www.evaristesys.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] odd issue with the with SIP over VPN
You probably are not advertising the routes across the vpn properly. Does your setup look like this asterisk[network a]openVPN server[network b - vpn]-openVPN client[network c]-sip client where network a, b, and c are all separate subnets? Is your vpn setup for routing or bridging? you need to make sure that the vpn server allows network a to to talk to network c, and that network c can talk to network a. By default, only the client can talk to the server. Attached subnets will not be routed automatically. See this section of the openVPN howto: http://openvpn.net/index.php/open-source/documentation/howto.html#scope I have 3 asterisk servers with sip trunking between them all running over openVPN links, and everything works fine when you make sure you setup the routing right in the vpn. I also have 2 phones that connect to one of the servers over a openVPN link as well - they're not sip (Nortel unistim) but it also works just fine. Andrew On Sat, Jan 23, 2010 at 7:17 PM, Alex Balashov abalas...@evaristesys.com wrote: You're going to be a lot more specific about the precise - if symptomatic - meaning of do not. On 01/23/2010 09:08 PM, Zane C.B. wrote: I've run into a odd issue where inbound calls to the SIP client work fine, but outbound from the SIP client do not. The path between the client and the server is as below. N900 SIP client-- OpenVPN -- Asterisk The version of Asterisk in question is 1.6.0.18. Any suggestions? -- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.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 -- _ -- 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] AOC advise of charge
My Asterisk PBX is connected to 2 ISDN lines by a PATTON gateway, in the Patton specification is written that can manage the AOC message. I would like record the AOC value on the mysql's CDR table, so to record the call costs. Is there some one who has manage this issues and can give me help to configure the patton and Asterisk? Many thanks in advance -- _ -- 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] ReceiveFAX and SendFAX questions
Morning, Have some questions regarding receiving and sending faxes... 1:st example: exten = 101,1,Answer() exten = 101,2,Wait(3) exten = 101,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff) exten = 101,4,System(tiff2pdf -p A4 /var/spool/asterisk/tmp/fax.tiff /var/spool/asterisk/tmp/fax.pdf) exten = 101,5,System(mutt -s 'New FAX for you sir' -a /var/spool/asterisk/tmp/fax.pdf magnu...@inputinterior.se /dev/null) I do receive the fax, the fax got converted to a pdf but 101,5 never get executed, when i look in cli, last line is 101,4... can any1 se why? 2:nd example: exten = 101,1,Answer() exten = 101,2,Wait(3) exten = 101,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff) exten = 101,4,System(fax.sh) cat /usr/bin/fax.sh tiff2pdf -p A4 /var/spool/asterisk/tmp/fax.tiff /var/spool/asterisk/tmp/fax.pdf mutt -s 'New FAX for you sir' -a /var/spool/asterisk/tmp/fax.pdf bo...@inputinterior.se /dev/null That works, i receive the fax as an attachment, but as I asked before why is not example 1 working? SendFAX question: exten = 101,1,Answer() exten = 101,2,Wait(3) exten = 101,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff) exten = 101,4,Some magical way to setup the channel to: SIP/033211101 exten = 101,5,SendFAX(/var/spool/asterisk/tmp/fax.tiff) 033211101 is an ATA (SPA2102) registered to *. I wonder if it is possible to do something like my example or not? Any suggestions? I was looking at: http://www.evilspurv.net/blog/2010/01/sending-pdfs-as-fax-with-asterisk/ I could do something like that but i would prefer to have all in the dialplan without need for an external program. /Magnus -- _ -- 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
