[asterisk-users] OK to have Asterisk and clients behind firewalls?
Hi To investigate the UNREACHABLE issue I'm having, I need to have confirmation that it's OK for the Asterisk server to be behind a NAT router, and also have clients elsewhere on the Net behind their own NAT router? I know that clients must use STUN to resolve their public IP and punch UDP holes in their firewall, but is there something special that must be done in the configuration of Asterisk so it knows it's living in a private network, behind a NAT router? And if someone knows of tools to investigate SIP issues, especially a text-based sniffer (no X available in the Asterisk live CD I'm using), I'm interested :-) Thank you. PS: FWIW, extension 203 (softphone) and 204 (IP phone) are both located on the same network and behind a NAT router, and both connect out to an Asterisk server somewhere on the Net behing its own NAT router: slast*CLI sip show peers Name/username HostDyn Nat ACL Port Status 204/20482.237.x.y D 5060 UNREACHABLE 203/20382.237.x.y D N 46838OK (925 ms) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [RTP] PSTN - Gateway - Phone
Hello I'm using a Linksys 3102 as VoIP gateway to connect a POTS line to a PBX. I also have an IP phone in a remote network across the Net. The PBX + gateway, and the phone are both behind a NAT router. I was wondering: 1. When a customer calls us through the POTS line and I pick up the call with the remote IP phone, do RTP packets go directly from the VoIP gateway to the IP phone, or do they go through the PBX, ie. is it... POST - VoIP gateway - NAT - Net - NAT - IP phone or POST - VoIP gateway - PBX - NAT - Net - NAT - IP phone ? 2. Regardless of the route RTP packets take, do I have to map ports on both NAT routers for RTP packets to be let inside the LAN, or is STUN able to handle this itself? How do I know if the routers are STUN-friendly, or I have to map ports? Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to write data to astdb?
Hello,
I'm trying to fill CID data into the astdb using AsteriskWin32's
asterisk.exe, to no avail: The batch file stops after the first line, and
just waits:
rem c:\cygroot\mystuffimport.bat
rem
rem c:\cygroot\mystuffC:\cygroot\bin\asterisk.exe -rx 'database put
cidname 123 My cellphone'
rem
rem Asterisk module loaded successfully
rem Asterisk entry point foundW2003*CLI Updated database successfully
rem Verbosity is at least 1
rem STUCK HERE!
C:\cygroot\bin\asterisk.exe -rx 'database put cidname 123 My cellphone'
C:\cygroot\bin\asterisk.exe -rx 'database put cidname 456 This is a test'
I don't know why the batch script stops after the first line.
So, I installed ActivePerl and the asterisk-perl package from CPAN, and
tried this, but it doesn't work either:
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
$AGI-database_put('cidname', '', 'my number');
Is there a way to access astdb directly, instead of through an AGI script?
Thank you.
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[asterisk-users] [*Win32 0.60] Sending call notification by e-mail/web?
Hello,
In case there are other users of the AsteriskWin32 port...
I haven't really used the AGI feature of Asterisk to run an application
from extensions.conf. *Win32 supports Perl, which I don't know. Apparently,
it's also possible to write AGI applications as EXE's (there's a
eagi-test.exe file installed by default).
= When a call comes in, I'd like an AGI application to send an e-mail and
send CID name/number to a script on a web server.
Is this the correct way to do it in Perl, with the modules available in
AsteriskWin32? Could I rewrite this in Delphi instead?
---
#!/usr/bin/perl
;---
;Note: Not sure if *Win32 supports LWP::Simple and Net::SMTP!
;Called from extensions.conf
;exten = group,n,AGI(notify.agi|${CALLERID(num)}|${CALLERID(name)})
;---
use strict;
open STDOUT, '/dev/null';
fork and exit;
;---
use LWP::Simple;
my $cidnum = $ARGV[0];
my $cidname = $ARGV[1];
my $url = 'http://www.acme.com/input.php?name=$cidnamenumber=$cidnum';
my $content = get $url;
die Couldn't get $url unless defined $content;
print STDERR Notified web server
;---
use Net::SMTP;
$smtp = Net::SMTP-new('smtp.acme.com'); # connect to an SMTP server
$smtp-mail( '[EMAIL PROTECTED]' ); # use the sender's address here
$smtp-to('[EMAIL PROTECTED]');# recipient's address
$smtp-data(); # Start the mail
# Send the header.
$smtp-datasend(To: [EMAIL PROTECTED]);
$smtp-datasend(From: [EMAIL PROTECTED]);
$smtp-datasend(\n);
# Send the body.
$smtp-datasend(Call received from $cidname/$cidnum\n);
$smtp-dataend(); # Finish sending the mail
$smtp-quit;
print STDERR Send e-mail
---
Thanks for any tip
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Re: [asterisk-users] Mini-ITX board + FXO PCI card?
At 10:09 11/02/2007 -0500, Michelle Dupuis Henderson wrote: We use a lot of mini-itx pc's, including the pCI slot. I don't think any of the systems have shared an irq with the PCI slot Thanks for the tip. In that case, I have a couple of questions for you :-) 1. The smallest mini-ITX case I found that accepts a PCI card is the Travla C138: If you used a mini-ITX with a Digium TDM400P, do you know if it fits? I didn't find its width, and apparently, the C138 will not accept a PCI card bigger than 17,52cm. 2. Can the Via motherboards boot from a USB drive, so I can install Linux from this and fetch the install files from an FTP server? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Fxo] Digium TDM01B vs. OpenVox A400P01?
Hello If someone had the opportunity of trying those two analog cards, how do they compare? Digium's sells for $150 while OpenVox's sells for $95. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mini-ITX board + FXO PCI card?
At 10:09 11/02/2007 -0500, Gordon Henderson wrote: On the CN1000 boards I'm using, the PCI slot seems tobe locked to IRQ10. The on-board USB hardware also seems to be wired to IRQ 10 :-) Using the BIOS to reserve IRQ 10 caused the on-board USB hardware to move to IRQ5 on the old VIA 533MHz boards I use for RD, but not on the new CN1000 boards. You'll need to experiment with this on the EX board... Thanks a lot for the feedback. I'll probably get an ML8000, so hopefully the PCI board can have its own IRQ. I've not tried it (I boot them off a flash IDE device I create on a host system), but can't you just temporarily plug in a CD drive to do the install (onto a local IDE/SATA drive) then unplug it put the lid back on? Right, I hadn't thought of this. Guess that solves the issue then. Looks like the only site that sells the Travla 138 is www.caseoutlet.com . Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mini-ITX board + FXO PCI card?
At 10:09 11/02/2007 -0500, Michelle Dupuis Henderson wrote: Yes on booting from a USB drive/memory stick - we setup the same way. Thanks for the tip. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mini-ITX board + FXO PCI card?
At 10:09 11/02/2007 -0500, Gordon Henderson wrote: Check the processor spec. carefully. [...] Also make sure you compile asterisk for an i586 OK, I'll make sure it has enough cache and I'll recompile the code myself. I'm thinking of getting an ML 8000 http://via.com.tw/en/products/mainboards/motherboards.jsp?motherboard_id=301 . At 10:09 11/02/2007 -0500, Manny A. Wise wrote: I did, and I was NOT happy with the results... Mini-itx have a serious problems with IRQ sharing... I am happily using a embeded system now, but the FXO and FXS have to be external. Those boards only come with one PCI slot. Do you mean it could share an IRQ with some embedded component like the video card? BTW, in this age of big USB drive, I don't really nee a DVD/CDRW combo. Does someone know if the Via motherboards (at least the ML series) supports booting off a USB drive, so I can use this to start Linux and fetch install files from an FTP server? Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mini-ITX board + FXO PCI card?
Hello Before I order a Travla C156 case (http://206.14.132.88/products/Travla/c156/C156.html), a Via mini-ITX motherboard (either the fanless ME6000 http://idotpc.com/TheStore/pc/viewPrd.asp?idcategory=50idproduct=4 or the fan-equipped M1 http://idotpc.com/TheStore/pc/viewPrd.asp?idcategory=50idproduct=163 ) , and a PCI FXO card from Digium or OpenVox... has someone already built that kind of box, and could tell me if it's powerful enough to power a small PBX? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: No sound: X-Lite - Asterisk - VoIP Provider - Cellphone
At 17:01 01/12/2006 +0100, Noah Miller [EMAIL PROTECTED] wrote: Just to double check - have you limited the RTP ports on the asterisk server to 8000-8019 (in rtp.conf)? Thanks. That what was missing. In rtp.conf, I fixed ports 1-10019 and mapped those ports on the router, and it worked. Also, Xlite uses (or used to use) a silence suppresion mechanism that doesn't work too well with asterisk. According to the WIKI: Turn off Silence Supression (to avoid RFC3389 warnings on Asterisk console): Menu | Advanced System Settings | Audio Settings | Silence Settings | Transmit Silence: Yes OK. However, the person on the other end tells me that my voice was very low, barely audible. Do you know what could be done about it? Are there voice-related settings in Asterisk that I should look at? Would playing with canreinvite to remove Asterisk from the loop and have RTP packets go directly from the VoIP provider to my X-Lite client at home make a difference? What should I do if canreinvite=yes means that the VoIP provider doesn't use the RTP ports that I expect to use on my side? Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting RTP ports for Asterisk?
Hello When I make calls from home to the PSTN by going through the Net - Asterisk - the Net - VoIP provider - PSTN, I get no sound either way. I assume it's because I must tell Asterisk to use fixed ranges of UDP ports and map ports accordingly on the NAT firewall under which it is located on the LAN at work. Here's the schema: home NAT Internet NAT Asterisk NAT Internet VoIP provide PSTN callee I took care of the NAT at home by using fixed ports in X-Lite + used STUN, so I guess the problem is located on the Asterisk side. 1. What are the settings (in sip.conf?) to tell Asterisk to use specific ports for RTP? 2. With this kind of setup, does Asterisk stay in the loop to forward RTP packets, or do X-Lite at home and the VoIP provider send RTP to each other directly? Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No sound: X-Lite - Asterisk - VoIP Provider - Cellphone
Hi I have the following setup to make outgoing calls: X-Lite (build 34025) at home behind NAT - Internet - Asterisk at work behind NAT - Internet - VoIP provider - GSM gateway - cellphone. I just tried calling my own cellphone, but there is no sound either way. Here's what I did on the X-Lite at home in the Topology section: IP address : Discover global address STUN server : Discover server Port used on local computer : Manually specify range 8000-8019 Here are the ports that I forwarded from my NAT router at home: UDP 5060 UDP 3478 (STUN; needed?) UDP 8000 to 8019 Is there something else I should do, either on my home setup or at work on the NAT router or Asterisk? Thank you! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Re: Re: Re: Rewriting caller ID from database?
At 10:34 28/11/2006 -0700, Pavel Jezek [EMAIL PROTECTED] wrote: I have done simple ael2 script, tak doing lookup in asterisk database like: find full numer, if cidname isn't found, substract one digit from right and try again, and so on Thanks. If LookupCIDname doesn't come with its own feature, I'll add it to the script. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [VoIP Trunk] No such host
Hello I'm trying to add a VoIP trunk to Asterisk, but I'm getting the following warning in the log file if I leave srvlookup=yes in sip.conf (OK if I comment it out): -- Nov 27 16:40:22 NOTICE[29660] chan_sip.c:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #47) Nov 27 16:40:22 WARNING[29660] chan_sip.c: No such host: freephonie.net Nov 27 16:40:22 WARNING[29660] chan_sip.c: Probably a DNS error for registration to [EMAIL PROTECTED], trying REGISTER again (after 20 seconds) -- Any idea what is causing this? Configuration files and more here: http://paste.lisp.org/display/30790 Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Re: Rewriting caller ID from database?
At 12:00 25/11/2006 -0700, Tom Lynn [EMAIL PROTECTED] wrote: By inverting the relationship, I found it easier to focus on the source of the call and the treatments I want to apply. I can also wipe out entries by family name and remove all attributes in one operation using database deltree. Interesting. I'll give it a shot tomorrow at the office. Anyhow, at this point, I could successfully import all the name + number records, and must find solutions for the following problems: - web interface to add/modify/remove records - find out if LookupCID is able to match prefixes with a record (some of customers have DID, so I'd like to just use 123-45?? to match those incoming calls to Such and such customer instead of adding individual records 123-4501, 123-4502, etc.) Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Rewriting caller ID from database?
At 12:00 25/11/2006 -0700, Tom Lynn [EMAIL PROTECTED] wrote: I like a challenge. I'll let you know if I come up with anything. My eternal gratitude if you find something :-) And don't forget to update the VoIP wiki so others can benefit too. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Rewriting caller ID from database?
At 12:00 25/11/2006 -0700, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Do try asterisk -rx database put cidname 12345676 \Me - cellular\ or asterisk -rx 'database put cidname 3871263 Me - home' These quotations seem to work. Yup, I should have tried before posting. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Rewriting caller ID from database?
At 12:00 25/11/2006 -0700, Michelle Dupuis [EMAIL PROTECTED] wrote: Try using smartCID (www.generationd.com). You'll get the benefit of ranges of numbers mapping to single ID's (good for corporate blocks), action field for blocking/accepting calls, etc). Neat, although 411.com won't do as we're not located in the US: smartCID - A php script to replace callerid information with lookup from a local mysql table, and if that fails then reverse phone number lookup from 411.com. This script links to Asterisk PBX. http://www.ocg.ca/clientfiles/gss/downloads.htm Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Rewriting caller ID from database?
At 22:07 22/11/2006 -0700, Marco Mouta wrote: You can do it using AstDB, just load the database with callerid names and numbers and then include a lookup on database in all incoming calls, so you can override whatever you wanted:) Thanks everyone. Indeed, it seems like using the embedded BerkeleyDB-based AstDB is good enough for what I'm trying to do. However, asterisk barfs on the following script that I used to import data: #/bin/bash asterisk -rx database put cidname 1234567 'Me - cellular' asterisk -rx database put cidname 1234567 'Me - home' etc. Any idea why? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 28, Issue 122
At 22:07 22/11/2006 -0700, Marco Mouta wrote: You can do it using AstDB, just load the database with callerid names and numbers and then include a lookup on database in all incoming calls, so you can override whatever you wanted:) Thanks everyone. Indeed, it seems like using the embedded BerkeleyDB-based AstDB is good enough for what I'm trying to do. However, asterisk barfs on the following script that I used to import data: #/bin/bash asterisk -rx database put cidname 1234567 'Me - cellular' asterisk -rx database put cidname 1234567 'Me - home' etc. Any idea why? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Rewriting caller ID from database?
Hi Most of our customers have generic names like Hospital, so I need to rewrite their caller ID name by looking up the number in a database on the Asterisk server, and rewriting the name such as Reading Hospital so that we know who's calling. Any idea if this can be done with Asterisk, and how to do it? Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [FC5] How to update kernel/kernel-develop for Athlon?
Hello I'm following instructions on how to install Asterisk on Fedora 5, but I'm having a problem: - the host is an older i686 athlon i386 GNU/Linux - /etc/rpm/platform says athlon-redhat-linux - running yum update kernel downloaded kernel i686 2.6.18-1.2200.fc5 - running yum update kernel-devel wants to download kernel-devel i586 2.6.18-1.2200.fc5 I know that I shouldn't mix versions (i686 and i586), but I don't know how else to update the system to make it ready for Asterisk. = Should I use a specific repository for Yum to use, or should I download a couple of RPMs to update those two items before proceeding with Asterisk? Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How does SIP work?
(I'm sorry to ask this question here, but I didn't get a reply in VoIP-related forums and I figured there's a lot of people here who are knowledgeable about VoIP and SIP, and could help me see the light. Please replace Axon PBX server with Asterisk in SIP mode if you will :-) ) I finally got to have a working set up using an Axon Windows PBX software, Linksys 3102 gateway, a GrandStream IP phone and an X-Ten softphone over the Net... but I don't know _why_ it works :-) Here's how I think the whole thing works: 1. I set up the router to map UDP 5060 to the host where the PBX is installed, and I launch the Axon server 2. Remote phones connect through the Net into the Axon server to register their IP address and extension 3. When a call comes in from the PSTN network into the Linksys, the 3102 sends an SIP notification to the PBX. The PBX checks what extensions it must ring, and sends out SIP notifactions to all extensions involved. For this to work, all remote routers must also forward SIP messages to the IP phones that registered (UDP 5060 by defaullt, but each phone needs its own port to be reachable, eg. UDP 5060 for the first phone in the LAN, UDP 5061 for the second phone, etc.) 4. Once a phone goes off-hook, a connection is set up between the phone and the Linksys gateway. During the connection, each device tells the other what UDP ports it will use for RTP, ie. data packets. Provided this is correct so far, here's where things begin to blur: - If I don't set up remote phones to use STUN, connections are made, but I don't get sound in one direction: Is it because without STUN, the misconfigured phone sends its private IP in the data part of an SIP message, eg. 192.168.0.1, and since this is an unroutable address the other device won't be able to route data packets? - I didn't forward any ports for RTP, but calls still work: Is it because I happen to have UPnP-capable routers, hence RTP ports are automagically opened to make things happen? Thanks much for any hint :-) -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.407 / Virus Database: 268.12.9/458 - Release Date: 27/09/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Working Sipura 3000 or Linksys 3102 configuration?
At 15:12 23/08/2006 -0700, Ron Wellsted wrote: Here are mine (with UK regional settings/A-law). Thanks a bunch :-) After more search, it turns out that if you don't need the router feature of the 3102 (I already have a router), the unit must be connected to the LAN through its... WAN plug. It would have been nice that the pathetic 6-page brochur - which is the only known documentation to date for the 3102 - mention that :-/ Next steps: - setting up an IP phone accross the Internet, and handle the firewall issue - see if the Linksys can connect out to a remote IP phone directly, with no PBX at all. Cheers VD. -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.11.5/426 - Release Date: 23/08/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Working Sipura 3000 or Linksys 3102 configuration?
Hello I'm having a problem with the Linksys 3102: With incoming PSTN calls, I can hear the caller through the X-Ten softphone, but he can't hear me. The problem is worse with Sjphone and the GrandStream 100 hardphone, as I get no sound in either direction. FWIW... - the SIP client, the PBX and the Linksys are all connected to a switch, with no firewall anywhere - the only way I can get the Linksys to notify the PBX of an incoming PSTN call is using the following settings: * PSTN Line PSTN-To-VoIP Gateway Setup PSTN Ring Thru Line 1 = yes * User 1 Call Forward Settings Cfwd All Dest = fxo (where fxo is the account also used in PSTN Line Subscriber Information to register with the PBX) Dial plans in either Line 1 or PSTN Line don't make it. Could someone upload his configuration of the Linksys (File Save as file) so I can compare with what I have? Since both ends use G711u as their default codec and there's no firewall between them, I suspect I'm totally wrong when it comes to configuring the Linksys as a simple SIP gateway (no use for the FXS port at this point). Possibly some routing issue. Thank you. -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.11.4/424 - Release Date: 21/08/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Linksys 3102] Couple of issues
Hello I'm several days into configuring this thing, and I still have a couple of issues. FWIW, I upgraded it to 3.2.10, and will only use this device to handle incoming calls from the PSTN, ie. no phone plugged into the Line 1/FXS port, and no need for outgoing calls through the PSTN Line/FXO port. Also, following the couple of documents in the PS section never worked: Using dial plans in either PSTN Line or PSTN User don't do anything. The only way I get the Linksys to notify the PBX of incoming calls is by setting PSTN Line PSTN-To-VoIP Gateway Setup PSTN Ring Thru Line 1 = yes and User 1 Call Forward Settings Cfwd All Dest = extension on PBX. Hopefully, some people here have used this newer brother of the Sipura 3000 and know how to solve them: - when a call comes in through the FXO port, I guess the Linksys first notifies the FXS port. Then, thanks to the Cfwd All Dest setting, it forwards the call to the PBX at the given extension. Problem is, the notifcation will also make the Linksys go off-hook right away while it rings the PBX: If no SIP device actually answers the call, the caller will be needlessly charged for the call (and will only hear silence while the PBX rings the extension). FWIW, I left PSTN Line PSTN-To-VoIP Gateway Setup Off Hook While Calling VoIP = no as is. Why does it do this? Can it be changed? - when an SIP device does answer the call (using the X-Free softphone, in case that matters), I can hear the remote PSTN caller, but he can't hear me. The MIC volume is OK. Does it have something to do with RTP and UDP ports? How do SIP devices and the Linksys know which ports to use on either end? As a bonus, if someone can confirm the philosophy of this device (I think it's really meant for home use, with no PBX : a call comes in from the PSTN or VoIP and rings the FXS phone; The FXS phone is used to make outgoing calls through either the PSTN or VoIP line), and tell me what the User 1 and PSTN Line tabs are really for... Thank you! PS: Here are the documents... that didn't work for me: 1. http://mundy.org/blog/index.php?p=65 : changing Dial Plan 1 = (S0:fxo) doesn't do anything. Calls aren't forwarded. Although the very last comment says This trick is not needed anymore in the lastest firmware release ver. 3.1.3, have a look at the release notes. just foward to asterisk and the sipura wont pickup until asterisk answerd, the Linksys 3102 _does_ pick up the call before the PBX actually answers the call. 2. http://voxilla.com/forum-viewtopic-t-1335.html : No need to doctor the caller ID number for calls to be forwarded; All it takes is the PSTN Ring Thru Line 1 = yes and unconditionnal forwarding in User 1 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.11.3/423 - Release Date: 18/08/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What's the story with X10*P FXO cards?
Hi It looks like the X101P clones I bought from eBay are dogs, so I'll look into buying some FXO-SIP box instead. Hopefully, I won't have the same problems with static, or caller ID and call termination not being detected. Still, considering the number of people having similar problems with those cards, I was wondering what the problem is. Is it because the hardware, no matter what is advertised, is actually not identical from card to card so the zaptel driver doesn't work reliably unless they are among the few remaining authentic cards made by Digium before it stopped manufacturing them? Because they're actually voice softmodems, and hence, very sensitive to the computers in which they're installed (voltage on the PCI slot, sharing IRQ's, etc.)? Other reasons? Thank you VD. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.1.394 / Virus Database: 268.9.10/383 - Release Date: 07/07/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: What's the story with X10*P FXO cards?
At 22:36 09/07/2006 -0700, Michael Graves [EMAIL PROTECTED] wrote: Skip local FXOs altogether. Setup an account with somone who provides DIDs via IP. Call forward your analog line to the IP based number. It will be absolutely painless compared to the troubles of small FXO interfaces. I'll look into this, although 1. I'm not sure VoIP providers do this here yet 2. While the POTS is very reliable, I can't say the same for ADSL. I'm a bit scared to depend on the Net for incoming calls. At 22:36 09/07/2006 -0700, Rich Adamson [EMAIL PROTECTED] wrote: Some folks have had good luck with the spa3000 (and some not). The TDM400 card from digium works pretty good, but can be less then acceptable in some cases. The Sangoma A200D (with hardware EC) works very well on all pstn lines that I've tested, but is rather expensive. Thanks for the info. This little experiment is getting expensive ;-) VD. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.1.394 / Virus Database: 268.9.10/383 - Release Date: 07/07/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Two FXO: How to dial a number when a RING comes in?
Hi
From: Noah Miller [EMAIL PROTECTED]
Sorry for the long delay in responding. I didn't see you message until
now due to the postfix problems on the mailing list.
No problem. I've decided to dump the rPath PoundKey linux distro because it
was still using Asterisk 1.2.5 and it was pointless to try to solve this
issue using older versions. Incidently, at least two people tried to do
what I'd like to do... but failed.
I'm beginning to think no one at Asterisk ever had the idea that maybe
someone would want to use Asterisk just as a simple bridget between two
POTS lines, with no IVR...
I think Eric Wieling is right. You have another problem not related to
what you are trying to do in the dialplan. It sounds like one of your fxo
cards or one of your phone lines is not working properly (or maybe
both). Test both phone lines and both interfaces by dialing into both of
them (make sure they are pointed to a context in the extensions.conf, and
make sure they have something to do there when you try to dial). Can you
get in to the asterisk box at all? Then
try swapping the phone lines with the fxo interfaces. Can you dial in then?
I'll finish installing Asterisk tomorrow (got an error when compiling
Zaptel on Fedora 5, but found the probable reason why on the web forum).
Once it's up and running, I'll go through the tests, including setting up
an SIP softphone on a Windows host and trying to call out or be called in
through both FXO cards.
In the mean time, the config files are really basic:
- FILES -
ZAPTEL.CONF
fxsks=1,2
loadzone=fr
defaultzone=fr
ZAPATA.CONF
[channels]
context=cherbourg
signalling=fxs_ks
usecallerid=yes
echocancel=yes
channel=1,2
EXTENSIONS.CONF
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]
TRUNK=Zap/2 ; Trunk interface
[cherbourg]
;If RING on Zap/1, just dial remote through through Zap/2
exten = s,1,NoOp(Before Dialing out through ${TRUNK})
exten = s,n,Dial(${TRUNK}/01XX)
exten = s,n,NoOp(After Dialing out through ${TRUNK})
- FILES -
Your help is much appreciated :-)
VD.
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[Asterisk-Users] Re: Two FXO: How to dial a number when a RING comes in?
Thanks Noah for the help, but... no go :-/ From: Noah Miller ONE: You should answer an incoming zap line before doing anything with it, so do this: exten = s,1,Answer exten = s,2,Dial(Zap/2/014XX) When I try this, instead of using the Zap/2 interface to ring the other number, Asterisk goes off hook and I hear some kind of static: Jun 19 18:17:46 NOTICE[2186] chan_zap.c: Got event 18 (Ring Begin)... Jun 19 18:17:47 NOTICE[2186] chan_zap.c: Got event 2 (Ring/Answered)... Jun 19 18:17:51 NOTICE[2186] chan_zap.c: Got event 18 (Ring Begin)... TWO: Are there any console messages? Can you dial into the system and get internal extensions? Maybe you could try a testing dialplan like this: exten = s,1,Answer exten = s,2,Waitexten(10) exten = 100,Dial(Zap/2/014XX) Then call in and after you're connected, dial 100 to see if it will dial out on ZAP/2 When I try this, /var/log/asterisk/messages says: Jun 19 18:12:38 NOTICE[1660] pbx.c: Cannot find extension '100' in context '(null)' Jun 19 18:12:38 WARNING[1660] pbx_config.c: Invalid priority/label 'Dial' at line 172 I just realized that I blindly typed the above, without realizing that the second parameter is missing. Regardless, since even the first test doesn't work... Just in case, I'd like to repeat that I don't want Asterisk to answer the call: I just want it to use the second FXO to ring another phone, at a remote location. For reference, I went back to the original configuration that I used, but it picks up the line and remains silent (static noises): --- extensions.conf -- [cherbourg] exten = s,1,Dial(Zap/2/0145815059) --- zaptel.conf --- fxsks=1,2 loadzone=fr defaultzone=fr zapata.conf --- [channels] ;context=default context=cherbourg signalling=fxs_ks usecallerid=yes echocancel=yes callgroup=1 pickupgroup=1 immediate=no callerid=my caller id(123) 123-1234 channel=1 ;context=default context=cherbourg signalling=fxs_ks usecallerid=yes echocancel=yes callgroup=1 pickupgroup=1 immediate=no callerid=my caller id(123) 123-1234 channel=2 and just in case you're wondering if the FXO cards are correctly loaded... - dmesg - Jun 19 18:12:31 localhost syslogd 1.4.1: restart. Jun 19 18:12:31 localhost kernel: klogd 1.4.1, log source = /proc/kmsg started. Jun 19 18:12:31 localhost kernel: Linux version 2.6.13.4-1.x86.i686.cmov ([EMAIL PROTECTED]:1) (gcc version 3.4.4) #1 Wed Nov 23 11:31:48 EST 2005 [...] Jun 19 18:12:31 localhost kernel: Zapata Telephony Interface Registered on major 196 Jun 19 18:12:31 localhost kernel: Zaptel Version: Echo Canceller: KB1 Jun 19 18:12:31 localhost kernel: Registered Tormenta2 PCI Jun 19 18:12:31 localhost kernel: ACPI: PCI Interrupt Link [LNKA] enabled at IRQ 5 Jun 19 18:12:31 localhost kernel: PCI: setting IRQ 5 as level-triggered Jun 19 18:12:31 localhost kernel: ACPI: PCI Interrupt :00:08.0[A] - Link [LNKA] - GSI 5 (level, low) - IRQ 5 Jun 19 18:12:32 localhost kernel: wcfxo: DAA mode is 'FCC' Jun 19 18:12:32 localhost kernel: Found a Wildcard FXO: Generic Clone Jun 19 18:12:32 localhost kernel: ACPI: PCI Interrupt Link [LNKD] enabled at IRQ 10 Jun 19 18:12:32 localhost kernel: PCI: setting IRQ 10 as level-triggered Jun 19 18:12:32 localhost kernel: ACPI: PCI Interrupt :00:09.0[A] - Link [LNKD] - GSI 10 (level, low) - IRQ 10 Jun 19 18:12:32 localhost kernel: wcfxo: DAA mode is 'FCC' Jun 19 18:12:32 localhost kernel: Found a Wildcard FXO: Generic Clone Jun 19 18:12:32 localhost kernel: usbcore: registered new driver wcusb Jun 19 18:12:32 localhost kernel: Wildcard USB FXS Interface driver registered Jun 19 18:12:35 localhost kernel: Registered tone zone 2 (France) = Surely, I can't be the only one in this list who needs to set up Asterisk simply to ring a remote phone when a call comes in at the office. Anybody has a working configuration that I could use as a reference? Thank you :-) VD. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.1.394 / Virus Database: 268.9.1/369 - Release Date: 19/06/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Two FXO: How to dial a number when a RING comes in?
Hello From: Eric \ManxPower\ Wieling [EMAIL PROTECTED] You have a problem unrelated to what you are trying to do. Fix the problem with dialing out of Zap/2 first. As an Asterisk newbie, I have no idea what you mean :-) Besides checking that the two FXO cards seem to be loaded by Asterisk (the [demo] works), and go from RED to OK when I plug in phone lines... how can I check that there is no issue with the HW before I go further? Thx. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.1.394 / Virus Database: 268.9.2/370 - Release Date: 20/06/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Two FXO: How to dial a number when a RING comes in?
Hello From: From: John D. Coleman [EMAIL PROTECTED] Correct me if I'm wrong but I think you would want to use the transfer command instead of dial to get it to call out to a remote office. The reason I used Dial() is because it's what I saw in Asterisk The Future of Telephony.pdf : (page 88) If you are making outbound calls on an FXO Zap channel, you can use the following syntax to dial a number on that channel: exten = 123,1,Dial(Zap/4/5551212) This example would dial the number 555-1212 on the Zap/4 channel. Does someone have an example of how to dial out a remote number when a call comes in through another FXO card? Thx. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.1.394 / Virus Database: 268.9.2/370 - Release Date: 20/06/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Two FXO: How to dial a number when a RING comes in?
Thanks Noah for the help, but... no go :-/ From: Noah Miller ONE: You should answer an incoming zap line before doing anything with it, so do this: exten = s,1,Answer exten = s,2,Dial(Zap/2/014XX) When I try this, instead of using the Zap/2 interface to ring the other number, Asterisk goes off hook and I hear some kind of static: Jun 19 18:17:46 NOTICE[2186] chan_zap.c: Got event 18 (Ring Begin)... Jun 19 18:17:47 NOTICE[2186] chan_zap.c: Got event 2 (Ring/Answered)... Jun 19 18:17:51 NOTICE[2186] chan_zap.c: Got event 18 (Ring Begin)... TWO: Are there any console messages? Can you dial into the system and get internal extensions? Maybe you could try a testing dialplan like this: exten = s,1,Answer exten = s,2,Waitexten(10) exten = 100,Dial(Zap/2/014XX) Then call in and after you're connected, dial 100 to see if it will dial out on ZAP/2 When I try this, /var/log/asterisk/messages says: Jun 19 18:12:38 NOTICE[1660] pbx.c: Cannot find extension '100' in context '(null)' Jun 19 18:12:38 WARNING[1660] pbx_config.c: Invalid priority/label 'Dial' at line 172 I just realized that I blindly typed the above, without realizing that the second parameter is missing. Regardless, since even the first test doesn't work... Just in case, I'd like to repeat that I don't want Asterisk to answer the call: I just want it to use the second FXO to ring another phone, at a remote location. For reference, I went back to the original configuration that I used, but it picks up the line and remains silent (static noises): --- extensions.conf -- [cherbourg] exten = s,1,Dial(Zap/2/0145815059) --- zaptel.conf --- fxsks=1,2 loadzone=fr defaultzone=fr zapata.conf --- [channels] ;context=default context=cherbourg signalling=fxs_ks usecallerid=yes echocancel=yes callgroup=1 pickupgroup=1 immediate=no callerid=my caller id(123) 123-1234 channel=1 ;context=default context=cherbourg signalling=fxs_ks usecallerid=yes echocancel=yes callgroup=1 pickupgroup=1 immediate=no callerid=my caller id(123) 123-1234 channel=2 and just in case you're wondering if the FXO cards are correctly loaded... - dmesg - Jun 19 18:12:31 localhost syslogd 1.4.1: restart. Jun 19 18:12:31 localhost kernel: klogd 1.4.1, log source = /proc/kmsg started. Jun 19 18:12:31 localhost kernel: Linux version 2.6.13.4-1.x86.i686.cmov ([EMAIL PROTECTED]:1) (gcc version 3.4.4) #1 Wed Nov 23 11:31:48 EST 2005 [...] Jun 19 18:12:31 localhost kernel: Zapata Telephony Interface Registered on major 196 Jun 19 18:12:31 localhost kernel: Zaptel Version: Echo Canceller: KB1 Jun 19 18:12:31 localhost kernel: Registered Tormenta2 PCI Jun 19 18:12:31 localhost kernel: ACPI: PCI Interrupt Link [LNKA] enabled at IRQ 5 Jun 19 18:12:31 localhost kernel: PCI: setting IRQ 5 as level-triggered Jun 19 18:12:31 localhost kernel: ACPI: PCI Interrupt :00:08.0[A] - Link [LNKA] - GSI 5 (level, low) - IRQ 5 Jun 19 18:12:32 localhost kernel: wcfxo: DAA mode is 'FCC' Jun 19 18:12:32 localhost kernel: Found a Wildcard FXO: Generic Clone Jun 19 18:12:32 localhost kernel: ACPI: PCI Interrupt Link [LNKD] enabled at IRQ 10 Jun 19 18:12:32 localhost kernel: PCI: setting IRQ 10 as level-triggered Jun 19 18:12:32 localhost kernel: ACPI: PCI Interrupt :00:09.0[A] - Link [LNKD] - GSI 10 (level, low) - IRQ 10 Jun 19 18:12:32 localhost kernel: wcfxo: DAA mode is 'FCC' Jun 19 18:12:32 localhost kernel: Found a Wildcard FXO: Generic Clone Jun 19 18:12:32 localhost kernel: usbcore: registered new driver wcusb Jun 19 18:12:32 localhost kernel: Wildcard USB FXS Interface driver registered Jun 19 18:12:35 localhost kernel: Registered tone zone 2 (France) = Surely, I can't be the only one in this list who needs to set up Asterisk simply to ring a remote phone when a call comes in at the office. Anybody has a working configuration that I could use as a reference? Thank you :-) VD. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.1.394 / Virus Database: 268.9.0/368 - Release Date: 16/06/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Two FXO: How to dial a number when a RING comes in?
Hi I'm a little lost on how to set things up with the two FXO cards I have: I want card #2 to dial a number when a call comes in on card #1. Using the following configuration, card #1 picks up the line and remains silent, instead of dialing out through card #2. Anybody knows what's wrong? - /etc/zaptel.conf - # Zaptel Configuration File # fxsks=1,2 loadzone=fr defaultzone=fr - /etc/asterisk/zapata.conf - [channels] context=cherbourg signalling=fxs_ks usecallerid=yes echocancel=yes callgroup=1 pickupgroup=1 immediate=no callerid=my caller id(123) 123-1234 channel=1 context=cherbourg signalling=fxs_ks usecallerid=yes echocancel=yes callgroup=1 pickupgroup=1 immediate=no callerid=my caller id(123) 123-1234 channel=2 - /etc/asterisk/extensions.conf - [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] CONSOLE=Console/dsp; Console interface for demo IAXINFO=guest ; IAXtel username/password ;Changed from TRUNK=Zap/g2 ; Trunk interface TRUNK=Zap/1 ; Trunk interface TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) [cherbourg] ;Hid the number to protect the innocents exten = s,1,Dial(Zap/2/014XX) - Thank you! -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.1.394 / Virus Database: 268.9.0/366 - Release Date: 15/06/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
