RE: [Asterisk-Users] Unable to create channel of type IAX2
And when it does work, the console says: Mar 5 02:07:08 NOTICE[9962]: chan_iax2.c:7065 iax2_poke_noanswer: Peer 'akralliax' is now UNREACHABLE! Time: 5 Mar 5 02:07:18 NOTICE[9962]: chan_iax2.c:6420 socket_read: Peer 'akralliax' is now REACHABLE! Time: 3 The iaxcomm phone is on the same LAN, so why can it be coming and going? Any ideas? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Sábado, 05 de Marzo de 2005 01:55 a.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Unable to create channel of type IAX2 Guys.. Im trying to setup a fotphone using iaxcomm and when I dial that softphones extension, * complains of this: Mar 5 01:54:54 NOTICE[9962]: app_dial.c:936 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3) Any hints? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Are codec capabilities bitmasks different in IAX and SIP?
I didn't know how else to caption this. I'm trying to play around with codec pass-through. I have two SIP phones, both with g729, behind two Asterisk servers. I set all the configs, SIP and IAX, to disallow=all; allow=g729 on both servers. But the originating server won't even try to call the destination server: -- Executing Dial(SIP/btel-c7d7, IAX2/bris/10101) in new stack Mar 5 02:55:32 WARNING[2786]: channel.c:1942 ast_request: No translator path exists for channel type IAX2 (native 63508) to 256 Mar 5 02:55:32 NOTICE[2786]: app_dial.c:936 dial_exec_full: Unable to create channel of type 'IAX2' (cause 0) == Everyone is busy/congested at this time (1:0/0/1) -- Executing Hangup(SIP/btel-c7d7, ) in new stack == Spawn extension (home, 55, 2) exited non-zero on 'SIP/btel-c7d7' When I show the peer entries on both servers, I see these same values for the codec strings on either end, but they are *different* for the IAX peer than the SIP, e.g. here's a snippet from show peer: iax2 show peer bris * Name : bris Secret : Set other stuff omitted Codecs : 0xf900 (g729) Codec Order : (g729) sip show peer btel * Name : btel Secret : Set ditto Codecs : 0x100 (g729) Codec Order : (g729) ** I'm running CVS-HEAD from yesterday. I get the same result in reverse if I start the call on the other side. I have run the Wiki and list archives route; followed the advice there to a tee (add some lines to the general context in sip.con) but nothing seems to yield anything different than the result shown above. Thanks. B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC questions: Userconfig, sip friends, iax friends and multiple trunks in routes
Can anybody explain how to use in ASTCC Userconfig, Sip Friends, IAX friends and what it does, when you setup multiple trunks in routes? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cant compile app_meetme2
Dear all I am get the following problem when trying to compile app_meetme2 using mysql...it seems to want to use pgsql.? anyone my Makefile looks like app_meetme2.o: app_meetme2.c #$(CC) -pipe $(CFLAGS) -c -o app_meetme2.o app_meetme2.c $(CC) -pipe -I/usr/local/include/mysql -L/usr/local/lib/mysql $(CFLAGS) -c -o app_meetme2.o app_meetme2.c app_meetme2.so: app_meetme2.o $(CC) $(SOLINK) -o $@ $ -lpq -I/usr/local/include/mysql -L/usr/local/li b/mysql -lmysqlclient # $(CC) $(SOLINK) -o $@ $ -lgdbm app_meetme2.o app_meetme2.c app_meetme2.c: In function `launch_query': app_meetme2.c:138: error: `PGconn' undeclared (first use in this function) app_meetme2.c:138: error: (Each undeclared identifier is reported only once app_meetme2.c:138: error: for each function it appears in.) app_meetme2.c:138: error: `conn' undeclared (first use in this function) app_meetme2.c:139: error: `PGresult' undeclared (first use in this function) app_meetme2.c:139: error: `res' undeclared (first use in this function) app_meetme2.c:151: warning: implicit declaration of function `PQsetdbLogin' app_meetme2.c:152: warning: implicit declaration of function `PQstatus' app_meetme2.c:152: error: `CONNECTION_BAD' undeclared (first use in this functio n) app_meetme2.c:154: warning: implicit declaration of function `PQerrorMessage' app_meetme2.c:154: warning: format argument is not a pointer (arg 6) app_meetme2.c:155: warning: implicit declaration of function `PQfinish' app_meetme2.c:162: warning: implicit declaration of function `PQexec' app_meetme2.c:163: warning: implicit declaration of function `PQresultStatus' app_meetme2.c:163: error: `PGRES_TUPLES_OK' undeclared (first use in this functi on) app_meetme2.c:166: warning: implicit declaration of function `PQclear' app_meetme2.c:171: warning: implicit declaration of function `PQntuples' app_meetme2.c:179: warning: implicit declaration of function `PQgetvalue' app_meetme2.c:179: warning: passing arg 1 of `atoi' makes pointer from integer w ithout a cast app_meetme2.c:180: warning: passing arg 1 of `atoi' makes pointer from integer w ithout a cast app_meetme2.c:181: warning: passing arg 1 of `atoi' makes pointer from integer w ithout a cast app_meetme2.c:182: warning: passing arg 1 of `atoi' makes pointer from integer w ithout a cast app_meetme2.c:183: warning: passing arg 1 of `atoi' makes pointer from integer w ithout a cast app_meetme2.c:184: warning: passing arg 1 of `atoi' makes pointer from integer w ithout a cast app_meetme2.c:185: warning: passing arg 2 of `strcpy' makes pointer from integer without a cast app_meetme2.c: In function `launch_query_onefield': app_meetme2.c:254: error: `PGconn' undeclared (first use in this function) app_meetme2.c:254: error: `conn' undeclared (first use in this function) app_meetme2.c:255: error: `PGresult' undeclared (first use in this function) app_meetme2.c:255: error: `res' undeclared (first use in this function) app_meetme2.c:268: error: `CONNECTION_BAD' undeclared (first use in this functio n) app_meetme2.c:270: warning: format argument is not a pointer (arg 6) app_meetme2.c:277: error: `PGRES_COMMAND_OK' undeclared (first use in this funct ion) app_meetme2.c:295: error: `PGRES_TUPLES_OK' undeclared (first use in this functi on) app_meetme2.c:296: warning: format argument is not a pointer (arg 6) app_meetme2.c:302: warning: passing arg 1 of `strlen' makes pointer from integer without a cast app_meetme2.c:307: warning: format argument is not a pointer (arg 3) app_meetme2.c:331: warning: passing arg 1 of `strlen' makes pointer from integer without a cast app_meetme2.c:336: warning: format argument is not a pointer (arg 3) app_meetme2.c: At top level: app_meetme2.c:645: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_I NITIALIZER__' undeclared here (not in a function) app_meetme2.c: In function `count_exec': app_meetme2.c:1547: error: too few arguments to function `ast_say_number' gmake[1]: *** [app_meetme2.o] Error 1 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cant compile app_meetme2
The error messages are Postgres related. You need to have a special postgres include file (postgres-dev files) to make it compile or disable postpres support somehow. I'm using debian and the the concering include file resided in a subdirectory of what asterisk was told. Jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bluetooth phone as SIP handset?
Quoting Jay Milk [EMAIL PROTECTED]: In a word - No. Generally, BT-capable phones can only control a headset or handsfree-set, but not be turned into a headset themselves. It's akin to expecting to watch TV on your remote, as it controls the TV so nicely :) Thanks, that's exactly what I meant. Never mind! It would have been cool to use my mobile as a handset to my home phone line, maybe one day! There is, however, an effort to have asterisk become the headset to a BT capable phone, which would allow the phone to be used as FXO through a $5 USB/BT dongle without further hardware. I was aware of this through the Wiki, however this is of less interest to me though I can see it would be cool to come home, plug your phone in to charge, then receive all calls through asterisk... Thanks for the info, Chris This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Difference between Snom 190 Elmeg 290?
Am Samstag 05 März 2005 07:58 schrieb Remco Barende: Hi list! While looking for the Snom 190 I found another phone, the Elmeg IP 290 (www.elmeg.de). Looking at the pictures the specs they seem to be very similar beasts but the firmware is supposedly not interchangeable. Does anyone know the difference between the 2, do they work with Asterisk? The weird thing is that Elmeg has similar phones with the Snom look but they are ISDN only (no voip) while Snom has several other models that are IP. Who's cloning who? I don't want to end up with phones for which firmware support or update will disappear soon while the 'orginal' will continue to be supported? Can't say anything about support, but my personal research told me that they are the same - no even more they are identical. Jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bluetooth phone as SIP handset?
Quoting dean collins [EMAIL PROTECTED]: I believe the advice that was given to you 5 minutes after you posted your original question was good and valuable and you should utilize bluetooth in this format, otherwise - go read the wiki. The Wiki only covers using the phone as an FXO via bluetooth dongle on the asterisk server, I wanted to use it as an FXS. I also don't want to simply pair a headset with an application on my PC, as I want to be able to use it without involving my laptop. Thanks for the info, it seems that what I wanted isn't possible due to lack of this facility in the BT phones. Ho hum! Chris This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cant compile app_meetme2
At 04:34 AM 3/5/2005, you wrote: The error messages are Postgres related. You need to have a special postgres include file (postgres-dev files) to make it compile or disable postpres support somehow. I'm using debian and the the concering include file resided in a subdirectory of what asterisk was told. if this is the case why dont i see a include file missing error someplace? or am I missing something.. Jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk behind NAT -- SIP config file
I have used the Draytek 2600V router in a few locations where only 1 or 2 phones are required. The router has 2 FXS ports and can be used locally to an * box or via the VPN to a remote * box. The VPN built into the routers just works, and I have 1 user who has had 3 VPN circuits up and running now for 6 months solid. Not bad in this day and age for an ADSL to stay functional for so long without interruptions. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Anton Krall Sent: 05 March 2005 04:56 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk behind NAT -- SIP config file The VPN approach might resolv a lot of nat issues I guess... Depending on the scenario I guess.. You could put another * box inside the second nat and interconnect using IAX, or if using a single phone, just use your setup, and finally, if using 2 or more phones and cant put a second * box, well, the vpn solution, I wonder how to do it if you have ATAs and nost softphone on the second NATted LAN.. Well... In time I guess :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rudolf Ladyzhenskii Sent: Viernes, 04 de Marzo de 2005 10:20 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk behind NAT -- SIP config file Yes, only port 5060. If you do not forward 5060, you can not call this phone from outside. Seem to work OK without other ports being forwarded. You mean on the remote sip phone firewall? What if there arem ore than 1 sip phone on that network behidn that firewall? Then you are in trouble. Asterisk only sees single public IP address. As far as it concerns there is only single phone out there. If you get multiple phones working, let me know. Another option, I think, may be using VPN, but I have not tried that. Then you can potentially have remote SIP phones to be on the virtual network. Don't you need to forward ports 1-2 for voice? Or does the sip phones just open up the ports from inside (by doing the in to out calls and keep alives)? I have mot tried to sniff on the traffic in details. I think, other ports are opened in responce to connection on port 5060. The only port listens at is port 5060. Rudolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SAY DIGITS problem
Hi,
I have a problem using AGI cmd SAY DIGITS. For some reason I cannot
here any thing when the script got executed. However if I use the cmd
SAY NUMBER I can here * reading the number fine.
I am running asterisk-1.0.6 and below is my PHP script.
Help please.
- Natt
#!/usr/bin/php -q
?php
ob_implicit_flush(true);
set_time_limit(10);
/* Standard Input file descriptor */
$stdin = fopen('php://stdin', 'r');
/* Standard Output file descriptor */
$stdout = fopen('php://stdout', 'w');
/* Standard Log output */
$stdlog = fopen('/var/log/asterisk/my_agi.log', 'w');
while(!feof($stdin)) {
$data = fgets($stdin);
fputs($stdlog, $data);
if ($data == || $data == \n) {
break;
}
}
sleep(1);
fputs($stdout, SAY DIGITS 1234);
fflush($stdout);
fclose($stdin);
fclose($stdout);
fclose($stdlog);
return 0;
?
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[Asterisk-Users] Re: Is anyone using asterisk in a small call
Date: Fri, 4 Mar 2005 17:37:16 -0500 From: John Scully [EMAIL PROTECTED] Subject: [Asterisk-Users] Is anyone using asterisk in a small call center Hello - I have just joined the lists and am considering installing quite a few * systems. I am looking for an IP-PBX with both solid standard features and call-center/ACD features. I have read the documentation and the list archives and did not see any references to real call-center type reporting and queuing. It is there. Look for the Queue's plugin it is default loaded in * Is anyone out there using * in this kind of environment? The features I would be looking for would include: Yes I'm running it for a bussiness and it is wokring fine. My agent's are loggin in and out by them self or the manager is putting them to work :-) Skill set routing Think you mean prioritizing of your agents that is called pennalty under asterisk multiple inbound queues. MM not sure you mean with that but you can connect queu's real time displays The data is there, you need an application who intreprt this data. Look for the different gui's and software that is out there opensource or closed. We are build our own tailor made applicaton for this. tracking of lost calls, wait times etc. Yes that is default available just type Queue show And * will show you the numbers :-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.0.3 Periodically Fails Registrations
Asterisk 1.0.3 Sayson 480i running .78 release (problem may not be Sayson specific, it's just that's what's deployed) Problem: Asterisk rejects registrations every so often even though nothing has changed either with Sayson or Asterisk configuration (and previous registrations have succeeded) SIP trace of successful registration: = =OUT=192.168.0.52: Sending SIP packet to: 209.139.212.169:5060 REGISTER sip:209.139.212.169:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.52;branch=z9hG4bKe9dfdb692 Max-Forwards: 9 Content-Length: 0 To: Chak De Display sip:[EMAIL PROTECTED]:5060 From: Chak De Display sip:[EMAIL PROTECTED]:5060;tag=3747511ff645a2c Call-ID: [EMAIL PROTECTED] CSeq: 272720092 REGISTER Contact: Chak De Display sip:[EMAIL PROTECTED];expires=60 Allow: SUBSCRIBE Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: MESSAGE Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Allow: INFO Authorization:Digest response=2d5dd24c01e8db3a1ac1b918d471b1a0,username=cdot-109,realm=asterisk,nonce=7ec20f6d,uri=sip:209.139.212.169:5060 User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 =OUT=END SIP packet =IN=192.168.0.52: Received SIP packet from: 209.139.212.169:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.52;branch=z9hG4bKe9dfdb692;received=69.90.106.130;rport=59749 From: Chak De Display sip:[EMAIL PROTECTED]:5060;tag=3747511ff645a2c To: Chak De Display sip:[EMAIL PROTECTED]:5060;tag=as3177eccc Call-ID: [EMAIL PROTECTED] CSeq: 272720092 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 =IN=END SIP packet =IN=192.168.0.52: Received SIP packet from: 209.139.212.169:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.52;branch=z9hG4bKe9dfdb692;received=69.90.106.130;rport=59749 From: Chak De Display sip:[EMAIL PROTECTED]:5060;tag=3747511ff645a2c To: Chak De Display sip:[EMAIL PROTECTED]:5060;tag=as3177eccc Call-ID: [EMAIL PROTECTED] CSeq: 272720092 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 60 Contact: sip:[EMAIL PROTECTED];expires=60 Date: Fri, 04 Mar 2005 18:04:12 GMT Content-Length: 0 =IN=END SIP packet = SIP trace of unsuccessful registration: = =OUT=192.168.0.52: Sending SIP packet to: 209.139.212.169:5060 REGISTER sip:209.139.212.169:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.52;branch=z9hG4bKc3e1d72f2 Max-Forwards: 9 Content-Length: 0 To: Chak De Display sip:[EMAIL PROTECTED]:5060 From: Chak De Display sip:[EMAIL PROTECTED]:5060;tag=3747511ff645a2c Call-ID: [EMAIL PROTECTED] CSeq: 272720093 REGISTER Contact: Chak De Display sip:[EMAIL PROTECTED];expires=60 Allow: SUBSCRIBE Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: MESSAGE Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Allow: INFO Authorization:Digest response=2d5dd24c01e8db3a1ac1b918d471b1a0,username=cdot-109,realm=asterisk,nonce=7ec20f6d,uri=sip:209.139.212.169:5060 User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 =OUT=END SIP packet =IN=192.168.0.52: Received SIP packet from: 209.139.212.169:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.52;branch=z9hG4bKc3e1d72f2;received=69.90.106.130;rport=59749 From: Chak De Display sip:[EMAIL PROTECTED]:5060;tag=3747511ff645a2c To: Chak De Display sip:[EMAIL PROTECTED]:5060;tag=as2333d070 Call-ID: [EMAIL PROTECTED] CSeq: 272720093 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 =IN=END SIP packet =IN=192.168.0.52: Received SIP packet from: 209.139.212.169:5060 SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.0.52;branch=z9hG4bKc3e1d72f2;received=69.90.106.130;rport=59749 From: Chak De Display sip:[EMAIL PROTECTED]:5060;tag=3747511ff645a2c To: Chak De Display sip:[EMAIL PROTECTED]:5060;tag=as2333d070 Call-ID: [EMAIL PROTECTED] CSeq: 272720093 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 = ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Difference between Snom 190 Elmeg 290?
On 5 Mar 2005, at 06:58, Remco Barende wrote: Hi list! While looking for the Snom 190 I found another phone, the Elmeg IP 290 (www.elmeg.de). Looking at the pictures the specs they seem to be very similar beasts but the firmware is supposedly not interchangeable. Does anyone know the difference between the 2, do they work with Asterisk? The weird thing is that Elmeg has similar phones with the Snom look but they are ISDN only (no voip) while Snom has several other models that are IP. Who's cloning who? I don't want to end up with phones for which firmware support or update will disappear soon while the 'orginal' will continue to be supported? The way I heard it was that Snom had some trouble with the mechanical design of their earlier phones so bought the case design in from an existing ISDN phone maker. I guess that must be Elmeg. I guess the outer look tells you nothing about the hardware let alone the firmware. Tim. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] change proxy after timeout
hi! i got the following problem: the * is registered to a provider - sip.conf ... register = user:[EMAIL PROTECTED] sometimes it happens that the server goes down. when i try to make a call it is not working ... he tries to reregister at the SAME server, times out and tries to reregister again on the SAME server. * is not rereding the srv entry! i want the * to reconnect complete by changing to the next SRV entry. ( if i restart gracefully it works in the same second. if i wait it takes minutes) is this possible??? thanx for helping Markus Dörfler ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: Getting PHP Config to work?
Hi Again, I used my phpconfig setup for a week, and found it a great timesaver for me :-) However I have just gone and broken it, and can't seem to 'fix' it. I was running a xorcom rapid installation, but converted to a semi-standard debian by changing the apt sources; so I could install a couple of extra things. I did apt-setup and choose British FTP sites I did apt-get dist-upgrade which installed a lot of stuff However this changed my box rather more than I was expecting for example it installed caudium as a web server, when I already have apache. I removed that via apt-get --purge remove caudium And after a couple of tweaks, I now have apache running again fine (as far as I can tell), and all my other web things work fine. However now I cannot even browse a .conf file via phpconfig. When clicking on the file I get the following error: Warning: fopen(/etc/asterisk/iax.conf): failed to open stream: Permission denied in /var/www/phpconfig/cls_phpconfig.php on line 127 I have gone over the wiki page, done chmod again etc, but nothing makes a difference. Does anybody have any ideas? Thanks C ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [OT] - [Asterisk-Users] Why should I answer a Newbie question, therethick!
On March 5, 2005 01:39 am, Jonathan Hobbs wrote: Ignore them and they will go away. Only after polluting the list with incessant How do I do X? messages, and then only after subsequently polluting the list with asterisk sucks messages, and then all the bad karma of some clueless twitt who couldn't be bothered to embrace OSS in the first place spewing incorrect information around, all because they should have hired a consultant to do their work for them instead. Ignoring them doesn't work, sorry. Education has a (marginally) better shot. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FW: Getting PHP Config to work?
However now I cannot even browse a .conf file via phpconfig. When clicking on the file I get the following error: Warning: fopen(/etc/asterisk/iax.conf): failed to open stream: Permission denied in /var/www/phpconfig/cls_phpconfig.php on line 127 I have gone over the wiki page, done chmod again etc, but nothing makes a difference. Your apache doesn't have read access on the file. It can't read the file or even worse, it can't go in that dir. Check that /etc/asterisk is readable (and writable) by apache. Also check that the conf files are readable by apache. hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: Getting PHP Config to work?
Thanks, I had thought this, and done the command: chmod -R a+w /etc/asterisk And it still didn't work. However I just set chmod 777 via WinSCP recursively, and it worked :) This is only a testing box I am not worried about the security risks. Strange the chmod didn't work I feel? C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit Sent: 05 March 2005 12:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FW: Getting PHP Config to work? However now I cannot even browse a .conf file via phpconfig. When clicking on the file I get the following error: Warning: fopen(/etc/asterisk/iax.conf): failed to open stream: Permission denied in /var/www/phpconfig/cls_phpconfig.php on line 127 I have gone over the wiki page, done chmod again etc, but nothing makes a difference. Your apache doesn't have read access on the file. It can't read the file or even worse, it can't go in that dir. Check that /etc/asterisk is readable (and writable) by apache. Also check that the conf files are readable by apache. hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] queue log analyser?
I've not released the source yet, I asked last week on the mailing list for people to send me over some example queue_logs, because so far I've only been able to test the software against my own. I have however made a lot of changes to it since last I posted about it. How is the progress on this? Could I have a look, please? roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Automatically send monitored call files by e-mail
Hi, I want to automatically send the sound files generated by asterisks monitor functions to a certain email address. My knowledge of shell scripting leaves a lot to desire, so I was hoping maybe on of you guys already did this and might provide me with an example of what to do :) Best regards, Anders ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P Clone, Which one?
Hi All, Googling X100P Clone I found several information about these cards and seems that some winmodem has the same chip used from the original X100P. Here below a list winmodem which should work as X100P clone: 1057 Motorola5608 SM56 PCI Fax Voice ModemE159 Tiger Jet Network Inc0001 Tiger 300/320 PCI interface I boughtone "Trust 56k V92 PCI Internal Modem MD-1100" which has the 1057 Motorola Chip, and I installed it on my linux box. in my /proc/pci list it is recognized like 1057:3052 (Motorola) (rev 4)., IRQ 11, I/O at 0x6c00 [0x6cff] When I try to load the module wcfxo, I cannot load it (zaptel is already loaded): [EMAIL PROTECTED] misc]# modprobe wcfxo/lib/modules/2.4.22/misc/wcfxo.o: init_module: No such deviceHint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesgmodprobe: insmod /lib/modules/2.4.22/misc/wcfxo.o failedmodprobe: insmod wcfxo failed I cannot undertand if: 1) the message occours because the FXO card use the same IRQ of the VGA card 2) it occurs because the winmodem installed is not a valid FXO card. Some one as experienced this kind of card? Regards MY PCI DEVICES: [EMAIL PROTECTED] root]# cat /proc/pciPCI devices found: Bus 0, device 0, function 0: Host bridge: Intel Corp. 440BX/ZX/DX - 82443BX/ZX/DX Host bridge (rev 3). Master Capable. Latency=64. Prefetchable 32 bit memory at 0xe000 [0xe7ff]. Bus 0, device 1, function 0: PCI bridge: Intel Corp. 440BX/ZX/DX - 82443BX/ZX/DX AGP bridge (rev 3). Master Capable. Latency=64. Min Gnt=129. Bus 0, device 7, function 0: ISA bridge: Intel Corp. 82371AB/EB/MB PIIX4 ISA (rev 2). Bus 0, device 7, function 1: IDE interface: Intel Corp. 82371AB/EB/MB PIIX4 IDE (rev 1). Master Capable. Latency=64. I/O at 0xf000 [0xf00f]. Bus 0, device 7, function 2: USB Controller: Intel Corp. 82371AB/EB/MB PIIX4 USB (rev 1). IRQ 11. Master Capable. Latency=64. I/O at 0x6400 [0x641f]. Bus 0, device 7, function 3: Bridge: Intel Corp. 82371AB/EB/MB PIIX4 ACPI (rev 2). IRQ 9. Bus 0, device 10, function 0: Ethernet controller: 3Com Corporation 3c590 10BaseT [Vortex] (rev 0). IRQ 10. Master Capable. Latency=248. Min Gnt=3.Max Lat=8. I/O at 0x6800 [0x681f]. Bus 0, device 13, function 0: VGA compatible controller: Matrox Graphics, Inc. MGA 1064SG [Mystique] (rev 2). IRQ 11. Master Capable. Latency=64. Non-prefetchable 32 bit memory at 0xe800 [0xe8003fff]. Prefetchable 32 bit memory at 0xe900 [0xe97f]. Non-prefetchable 32 bit memory at 0xea00 [0xea7f]. Bus 0, device 14, function 0: Modem: PCI device 1057:3052 (Motorola) (rev 4). IRQ 11. Master Capable. Latency=64. Min Gnt=1.Max Lat=62. Non-prefetchable 32 bit memory at 0xeb00 [0xeb000fff]. I/O at 0x6c00 [0x6cff]. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Difference between Snom 190 Elmeg 290?
Hello Remco, On Sat, 5 Mar 2005, Remco Barende wrote: Hi list! While looking for the Snom 190 I found another phone, the Elmeg IP 290 (www.elmeg.de). Looking at the pictures the specs they seem to be very similar beasts but the firmware is supposedly not interchangeable. Does anyone know the difference between the 2, do they work with Asterisk? The weird thing is that Elmeg has similar phones with the Snom look but they are ISDN only (no voip) while Snom has several other models that are IP. Who's cloning who? I don't want to end up with phones for which firmware support or update will disappear soon while the 'orginal' will continue to be supported? Elmeg has been for a long time a manufacturer of ISDN Phones and small to medium PBXes in germany. Snom uses the chassis of the elmeg phones and puts their own electronics in them. So it seems very likely that the elmeg IP-Phones are in fact Snom phones. I do not wether the firmware can be changed across elmeg and snom, but if there are no artificial barriers in place that prevent this this could be possible. Torsten Thx! Remco ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Media Online Internet Services Marketing GmbH Torsten Krueger [EMAIL PROTECTED] fon: 49-231-5575100fax: 49-231-55751098 Kurze Str. 10 D-44137 Dortmund ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Getting asterisk-addons installed on Debian?
Title: Getting asterisk-addons installed on Debian? Hi, I am having some trouble installing asterisk addons on Debian. I wish to do this to use mysql billing. I have mysql and mysql-devel packages installed I think!? pbx01:/usr/src/asterisk-addons# dpkg -l mysql-server libmysqlclient*dev Desired=Unknown/Install/Remove/Purge/Hold | Status=Not/Installed/Config-files/Unpacked/Failed-config/Half-installed |/ Err?=(none)/Hold/Reinst-required/X=both-problems (Status,Err: uppercase=bad) ||/ Name Version Description +++-===-===-== ii mysql-server 4.0.23-7 mysql database server binaries un libmysqlclient-dev none (no description available) pn libmysqlclient10-de none (no description available) ii libmysqlclient12-de 4.0.23-7 mysql database development files un libmysqlclient14-de none (no description available) un libmysqlclient6-dev none (no description available) un libmysqlclient9-dev none (no description available) pbx01:/usr/src/asterisk-addons# Which I know you need. I have mysql running etc. The problem seems to be making asterisk-addons I have exported from the cvs and tried both CVS and STABLE versions. I am running asterisk stable, installed via xorcom rapid (which may be why it freaks out?) When googling I didnt find much, bar one similar problem with no replies. Output: pbx01:/usr/src/asterisk-addons# make clean rm -f *.so *.o .depend make -C format_mp3 clean make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3' rm -f *.o *.so *~ make[1]: Leaving directory `/usr/src/asterisk-addons/format_mp3' pbx01:/usr/src/asterisk-addons# make install ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` ./mkdep: line 85: cc: command not found make -C format_mp3 all make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -c -o common.o common.c make[1]: gcc: Command not found make[1]: *** [common.o] Error 127 make[1]: Leaving directory `/usr/src/asterisk-addons/format_mp3' make: *** [format_mp3/format_mp3.so] Error 2 pbx01:/usr/src/asterisk-addons# A few errors.. If anyone could help with any easy way to install asterisk-addons, or just the mysql section, that would be great. I havent been able to find ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting asterisk-addons installed on Debian?
On Sat, Mar 05, 2005 at 01:19:24PM -, C. Tomlinson wrote: Hi, I am having some trouble installing asterisk addons on Debian. I wish to do this to use mysql billing. snip make[1]: gcc: Command not found You need a C compiler, try apt-get install build-essential Hope this helps, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] e164.org and FWD now have peering arrangement
Wolfgang S. Rupprecht wrote: FYI: FWD shows a different inbound prefix: **164 e164.org8781039311 Yes that works as well, and was issued by another company, the contact at FWD asked if we could route that to them as well, I prefer the other range because it's shorter and we've always routed it like that so it's easier for me to remember... Is there enough spare numbering space there for you to assign e164.org dialable numbers to people in the asterisk community too? Technically no, but we do it anyways despite how much the ITU loves us for doing it :) In the telephone world there is no equivalent to private LAN IP ranges, we're hoping if we get enough support for what we're doing to be allocated the +88299 range, (highly unlikely to funnier things have been known to happen) While it might be nice for asterisk home users to have their single DID listed, it strikes me that the real utility would be to have a blocks of 100 or 1000 numbers assigned to folks, so they could have each of their voip phones directly dialable from anyone that queries your db. We have always offered blocks of 100 numbers in the +88299 range for anyone that wants them, catch being that they're not really allocated by anyone/body except our DNS zone... Any number ranges in our zone are also accessible from FWD etc etc etc... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the long run the pessimist may be proved right, but the optimist has a better time on the trip. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] signaling problems
Hi ml, this is my problem: I have an Asterisk on remote site (my office) and two x-lite at home behind a ful cone nat. Both my ua can register, I can place and receive calls from both the phones and I can hear voice, so I don't think I have nat problem but when when i place a call if the called party hangup, calling party doesn't receive the signal and it stays connected. I also experienced the same problem placing a call on hold. Calling party can place the call on hold (called party listen moh) but called party cannot do it. Watching asterisk CLI no called party signals were detected? Why these? Can someone help me? Regards. Marco Ziglioli Alascom Services S.R.L. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with loging on guest account
Hi, I've just compiled and installed Asterisk (1.0.5). After some problems with codecs I could successfully connect to server by: [EMAIL PROTECTED] Next I created account at iaxtel.com and configured iaxcomm to work with this account. Unfortunately after that I had problem with logging as guest. Calling as [EMAIL PROTECTED] tried to connect with my local server through. So I changed it to: 192.168.0.1/guest and 192.168.0.1/[EMAIL PROTECTED] but I had errors: Mar 5 14:58:05 NOTICE[13269]: chan_iax2.c:5461 socket_read: Rejected connect attempt from 192.168.0.2, request '[EMAIL PROTECTED]' does not exist and Mar 5 14:50:49 NOTICE[13243]: chan_iax2.c:5441 socket_read: Rejected connect attempt from 192.168.0.2 Is there something wrong with those adresses? How can I force to connect locally or why asterisk rejects my connections? Thanks for help Marcin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to transfer timed out calls from call parking
I am able to transfer a call to call parking using '#' without any problems since I have the 't' option in my Dial() line. However, if no one picks up the call after it has been parked and a timeout occurs (such that the call is returned back to the original extension), the call is no longer transferrable afterwards. Can anyone suggest how to fix this problem so that I can re-transfer a parked call that has timed out and returned back to the original extension from which it came? Right now, I just include parkedcalls context in extensions.conf. It seems that the problem is that when the call times out and parked call calls back the original extension, there is no equivalent of having the 'T' option specified in Dial(). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Getting asterisk-addons installed on Debian?
Hi, Thanks. I idd that and now get different errors: pbx01:/usr/src/asterisk-addons# make install ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` app_addon_sql_mysql.c:15:27: asterisk/file.h: No such file or directory app_addon_sql_mysql.c:16:29: asterisk/logger.h: No such file or directory app_addon_sql_mysql.c:17:30: asterisk/channel.h: No such file or directory app_addon_sql_mysql.c:18:26: asterisk/pbx.h: No such file or directory app_addon_sql_mysql.c:19:29: asterisk/module.h: No such file or directory app_addon_sql_mysql.c:20:34: asterisk/linkedlists.h: No such file or directory app_addon_sql_mysql.c:21:31: asterisk/chanvars.h: No such file or directory app_addon_sql_mysql.c:22:27: asterisk/lock.h: No such file or directory cdr_addon_mysql.c:17:29: asterisk/config.h: No such file or directory cdr_addon_mysql.c:18:30: asterisk/options.h: No such file or directory cdr_addon_mysql.c:19:30: asterisk/channel.h: No such file or directory cdr_addon_mysql.c:20:26: asterisk/cdr.h: No such file or directory cdr_addon_mysql.c:21:29: asterisk/module.h: No such file or directory cdr_addon_mysql.c:22:29: asterisk/logger.h: No such file or directory cdr_addon_mysql.c:23:26: asterisk/cli.h: No such file or directory cdr_addon_mysql.c:24:22: asterisk.h: No such file or directory make -C format_mp3 all make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o common.o common.c common.c:1:29: asterisk/logger.h: No such file or directory common.c: In function `decode_header': common.c:93: warning: implicit declaration of function `ast_log' common.c:93: error: `LOG_WARNING' undeclared (first use in this function) common.c:93: error: (Each undeclared identifier is reported only once common.c:93: error: for each function it appears in.) make[1]: *** [common.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-addons/format_mp3' make: *** [format_mp3/format_mp3.so] Error 2 pbx01:/usr/src/asterisk-addons# Now this is probably due to me not having compiled * to start with, so I have no /usr/src/asterisk folder. I feel I may be better starting from scratch with a default Debian installation, and then I will know what I have where? What are your opinions? The best thing would be an apt-get install asterisk-addons, but I haven't found that :/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martijn van Oosterhout Sent: 05 March 2005 13:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Getting asterisk-addons installed on Debian? On Sat, Mar 05, 2005 at 01:19:24PM -, C. Tomlinson wrote: Hi, I am having some trouble installing asterisk addons on Debian. I wish to do this to use mysql billing. snip make[1]: gcc: Command not found You need a C compiler, try apt-get install build-essential Hope this helps, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie guidance requested --- Grandstream Budgetone
Hi- I am attempting to setup my Budgettone phone for use with my * server and am having problems obtaining an IP address. I have checked the phones settings to make sure it has dhcp enabled and it is. The display says no IP. I bought the phone but do not have any documentation other than the Wiki, but I am still at a loss. What could be preventing the phone from picking up an IP address? Any help would be appreciated. Thanks, Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie guidance requested --- Grandstream Budgetone
What could be preventing the phone from picking up an IP address? Do you have a DHCP server on your network? Stuart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Block anonymous calls
Hi. I am trying to set up my Asterisk box to block anonymous calls. I am having some grief from telemarketer calls to my number and I would like to block it. I see from my CDR's that some of my callers also have unknown in their FROM field. I would like to let them through. Only block the FROM anonymous that the telemarketers use. Have anyone here done it and would like to drop a line explaining how ? Fredrik _ Last ned MSN Messenger gratis http://www.msn.no/computing/messenger - Den korteste veien mellom deg og dine venner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI HDLC Abort (6) Errors
ztcfg - works, too.. after a timing source change... power cycle works. - Yeah, we rocked the vote all right. Those little bastards betrayed us again. - Hunter S. Thompson on the 2004 election. On Fri, 4 Mar 2005, Steven Critchfield wrote: On Fri, 2005-03-04 at 15:27 -0700, Tom wrote: Hello, I have searched and searched, and come up with nothing. I am running Asterisk with a wcte110p configured for t1. Our PRI is staying up, and we can make calls however our service provider's logs are flooding with errors and we are getting lots of HDLC Abort (6) on Primary D-Channel Errors. Our provider says it looks like our box is trying to be the master timer on the circuit (which is not correct they are providing the timing) we have tried both span=1,1,0,esf,b8zs and span=1,0,0,esf,b8zs in zaptel.conf both produce the same problems. The problem is not in Asterisk per se as the errors start happening as soon as I modprobe the driver and run ztcfg. As soon as the circuit comes up the errors start on the provider's end. Did you make sure to power cycle afterwords? Sometimes the zap cards don't change critical settings like timing once configured. We are running CVS Asterisk/zaptel/libpri from March 2nd 2005 on Fedora Core 3 fully patched as of last night, I was thinking the problem was with the 2.6 kernel getting preempted and therefore the driver not being able to do its timings right, however fc3's kernels have preemption disabled by default. Does Digium hardware really need/expect a real time OS to run properly? Like I said previously I think the problem is in the driver itself not in asterisk. Any help would be appreciated, and I can code a bit in c so if someone can point me in the right direction I might be able to fix it myself... You probably want to dump the FC kernel like a bad habit. Get a plain vanilla kernel and see if that fixes your problems. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting asterisk-addons installed on Debian?
Grab a copy of the Asterisk source, and untar it into /usr/src. Once you've done this, make sure that files such as /usr/src/asterisk/include/asterisk/file.h exist. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ C. Tomlinson wrote: Hi, Thanks. I idd that and now get different errors: pbx01:/usr/src/asterisk-addons# make install ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` app_addon_sql_mysql.c:15:27: asterisk/file.h: No such file or directory app_addon_sql_mysql.c:16:29: asterisk/logger.h: No such file or directory app_addon_sql_mysql.c:17:30: asterisk/channel.h: No such file or directory app_addon_sql_mysql.c:18:26: asterisk/pbx.h: No such file or directory app_addon_sql_mysql.c:19:29: asterisk/module.h: No such file or directory app_addon_sql_mysql.c:20:34: asterisk/linkedlists.h: No such file or directory app_addon_sql_mysql.c:21:31: asterisk/chanvars.h: No such file or directory app_addon_sql_mysql.c:22:27: asterisk/lock.h: No such file or directory cdr_addon_mysql.c:17:29: asterisk/config.h: No such file or directory cdr_addon_mysql.c:18:30: asterisk/options.h: No such file or directory cdr_addon_mysql.c:19:30: asterisk/channel.h: No such file or directory cdr_addon_mysql.c:20:26: asterisk/cdr.h: No such file or directory cdr_addon_mysql.c:21:29: asterisk/module.h: No such file or directory cdr_addon_mysql.c:22:29: asterisk/logger.h: No such file or directory cdr_addon_mysql.c:23:26: asterisk/cli.h: No such file or directory cdr_addon_mysql.c:24:22: asterisk.h: No such file or directory make -C format_mp3 all make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o common.o common.c common.c:1:29: asterisk/logger.h: No such file or directory common.c: In function `decode_header': common.c:93: warning: implicit declaration of function `ast_log' common.c:93: error: `LOG_WARNING' undeclared (first use in this function) common.c:93: error: (Each undeclared identifier is reported only once common.c:93: error: for each function it appears in.) make[1]: *** [common.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-addons/format_mp3' make: *** [format_mp3/format_mp3.so] Error 2 pbx01:/usr/src/asterisk-addons# Now this is probably due to me not having compiled * to start with, so I have no /usr/src/asterisk folder. I feel I may be better starting from scratch with a default Debian installation, and then I will know what I have where? What are your opinions? The best thing would be an apt-get install asterisk-addons, but I haven't found that :/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martijn van Oosterhout Sent: 05 March 2005 13:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Getting asterisk-addons installed on Debian? On Sat, Mar 05, 2005 at 01:19:24PM -, C. Tomlinson wrote: Hi, I am having some trouble installing asterisk addons on Debian. I wish to do this to use mysql billing. snip make[1]: gcc: Command not found You need a C compiler, try apt-get install build-essential Hope this helps, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie guidance requested --- Grandstream Budgetone
Stuart Ford wrote: What could be preventing the phone from picking up an IP address? Do you have a DHCP server on your network? Stuart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users at the risk of staing the obvious... If you manually set the IP address of the phone, can you ping it from the server ? This will prove the TCP/IP connectivity begin:vcard fn:Nigel Taylor n:Taylor;Nigel org:ITAzure Limited adr:15 Warren Park Way;;Dunn House;Enderby;Leicestershire;LE19 4SA;United Kingdom email;internet:[EMAIL PROTECTED] title:Technology Director tel;work:0116 286 3016 url:http://www.itazure.com version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Difference between Snom 190 Elmeg 290?
There is a partnership between Elmeg and snom. We are using their plastic (in the snom 190/200/220), they are using our hard- and software (in the Elmeg 290). Elmeg have a long experience in making phones and we have experience in making hard- and software for VoIP (as long as it can be in the SIP-based industry). A good partnership! We call it snom inside... CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Torsten Krueger Sent: Saturday, March 05, 2005 2:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Difference between Snom 190 Elmeg 290? Hello Remco, On Sat, 5 Mar 2005, Remco Barende wrote: Hi list! While looking for the Snom 190 I found another phone, the Elmeg IP 290 (www.elmeg.de). Looking at the pictures the specs they seem to be very similar beasts but the firmware is supposedly not interchangeable. Does anyone know the difference between the 2, do they work with Asterisk? The weird thing is that Elmeg has similar phones with the Snom look but they are ISDN only (no voip) while Snom has several other models that are IP. Who's cloning who? I don't want to end up with phones for which firmware support or update will disappear soon while the 'orginal' will continue to be supported? Elmeg has been for a long time a manufacturer of ISDN Phones and small to medium PBXes in germany. Snom uses the chassis of the elmeg phones and puts their own electronics in them. So it seems very likely that the elmeg IP-Phones are in fact Snom phones. I do not wether the firmware can be changed across elmeg and snom, but if there are no artificial barriers in place that prevent this this could be possible. Torsten Thx! Remco ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Media Online Internet Services Marketing GmbH Torsten Krueger [EMAIL PROTECTED] fon: 49-231-5575100fax: 49-231-55751098 Kurze Str. 10 D-44137 Dortmund ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting asterisk-addons installed on Debian?
I should have added that you'll be OK with your current build; you'd have to install the source for Asterisk if you went for a default Debian anyway. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Alistair Cunningham wrote: Grab a copy of the Asterisk source, and untar it into /usr/src. Once you've done this, make sure that files such as /usr/src/asterisk/include/asterisk/file.h exist. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ C. Tomlinson wrote: Hi, Thanks. I idd that and now get different errors: pbx01:/usr/src/asterisk-addons# make install ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` app_addon_sql_mysql.c:15:27: asterisk/file.h: No such file or directory app_addon_sql_mysql.c:16:29: asterisk/logger.h: No such file or directory app_addon_sql_mysql.c:17:30: asterisk/channel.h: No such file or directory app_addon_sql_mysql.c:18:26: asterisk/pbx.h: No such file or directory app_addon_sql_mysql.c:19:29: asterisk/module.h: No such file or directory app_addon_sql_mysql.c:20:34: asterisk/linkedlists.h: No such file or directory app_addon_sql_mysql.c:21:31: asterisk/chanvars.h: No such file or directory app_addon_sql_mysql.c:22:27: asterisk/lock.h: No such file or directory cdr_addon_mysql.c:17:29: asterisk/config.h: No such file or directory cdr_addon_mysql.c:18:30: asterisk/options.h: No such file or directory cdr_addon_mysql.c:19:30: asterisk/channel.h: No such file or directory cdr_addon_mysql.c:20:26: asterisk/cdr.h: No such file or directory cdr_addon_mysql.c:21:29: asterisk/module.h: No such file or directory cdr_addon_mysql.c:22:29: asterisk/logger.h: No such file or directory cdr_addon_mysql.c:23:26: asterisk/cli.h: No such file or directory cdr_addon_mysql.c:24:22: asterisk.h: No such file or directory make -C format_mp3 all make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o common.o common.c common.c:1:29: asterisk/logger.h: No such file or directory common.c: In function `decode_header': common.c:93: warning: implicit declaration of function `ast_log' common.c:93: error: `LOG_WARNING' undeclared (first use in this function) common.c:93: error: (Each undeclared identifier is reported only once common.c:93: error: for each function it appears in.) make[1]: *** [common.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-addons/format_mp3' make: *** [format_mp3/format_mp3.so] Error 2 pbx01:/usr/src/asterisk-addons# Now this is probably due to me not having compiled * to start with, so I have no /usr/src/asterisk folder. I feel I may be better starting from scratch with a default Debian installation, and then I will know what I have where? What are your opinions? The best thing would be an apt-get install asterisk-addons, but I haven't found that :/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martijn van Oosterhout Sent: 05 March 2005 13:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Getting asterisk-addons installed on Debian? On Sat, Mar 05, 2005 at 01:19:24PM -, C. Tomlinson wrote: Hi, I am having some trouble installing asterisk addons on Debian. I wish to do this to use mysql billing. snip make[1]: gcc: Command not found You need a C compiler, try apt-get install build-essential Hope this helps, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is anyone using asterisk in a small call center
Asterisk has the ability to do agent queueing and some general ACD functionality. The functionality doesn't come close to the functionality/flexibility of Avaya's Expert Agent functionality, but * won't cost you several hundred thousand dollars for deployment either. As far as reporting, there are also tools to have the agent activity to go to a SQL DB of your choosing, but reporting writing (ala Avaya CMS,etc) isn't there. If you're comfortable doing the SQL thing, or having someone on your staff do that to obtain the data you're looking for, again, you'll likely end up at least another $100,000 ahead. Bottom line - You may be able to get * to do what you're looking to do if you're willing to contribute alot of back-end techie/dev work to get to a finished product. The upside is, of course, a significant cost savings both on the opex and capex side if you choose * over the out of the box PBX from a brand name PBX vendor. You may also want to take a look at www.fonality.com. I don't really know anything about this company and the quality of their product as I personally don't own or have any experience, but they seem to have done alot of work adding bells and whistles around a core asterisk install and ACD functionality was on the roadmap the last time I was to their website about a month ago. On Fri, 4 Mar 2005 17:37:16 -0500, John Scully [EMAIL PROTECTED] wrote: Hello - I have just joined the lists and am considering installing quite a few * systems. I am looking for an IP-PBX with both solid standard features and call-center/ACD features. I have read the documentation and the list archives and did not see any references to real call-center type reporting and queuing. Is anyone out there using * in this kind of environment? The features I would be looking for would include: Skill set routing multiple inbound queues. real time displays tracking of lost calls, wait times etc. Thanks - John Scully CTO isipi.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FWD and SIPPHONE problems after upgrading to CVSHEAD - VERIFIED
Replying to my own post :-(
Yes, I'm top-posting, because no one ever seems to reply to my posts
anyway, I don't want to make you re-read my old post just to find out
what I'm adding.
I have _not_ solved the problem, but I reverted briefly to 1.0.3, and
I can indeed call to FWD without any problems. This is with _no
changes_ to the iax.conf between the two, so something in the recent
CVS HEAD has caused me to be able to receive calls from FWD (via
IAX2), but no longer call FWD.
I can't believe this is only happening to me, but apparently, it must
be...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hadar
Pedhazur
Sent: Thursday, March 03, 2005 5:18 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] FWD and SIPPHONE problems after upgrading to
CVSHEAD
I have been successfully connected (incoming and outgoing)
to FWD for a very long time. A few months ago, I changed
from SIP-based FWD service to IAX2-based, and that went fine
as well, both incoming and outgoing.
At the time, I was running Asterisk 1.0.3 Stable.
I rarely use the service, so other than noticing that I was
always successfully registered to FWD, I didn't make or
receive calls recently.
2 weeks ago, I upgraded to CVS HEAD:
Asterisk CVS-HEAD-02/21/05-09:07:50
Still didn't make or receive calls to FWD since the upgrade,
but everything else has worked flawlessly (including sixTel,
NuFone, etc.). All my softphones (SIP and IAX2) and
Sipura-2000's work perfectly too.
On to the problem... A few days ago, I signed up for an
account with SIPPhone. When I did a sip reload, which had
the register statement, I immediately got a call welcoming
me, so I thought everything was fine. It wasn't.
I have been unable to make any calls to sipphone, and even
though the registration appears to work (and my.sipphone.com
shows me as online), all calls to my number actually claim
that I am unavailable, and go directly to voicemail.
Before I show my configs and CLI output, a few more
background data points:
I can successfully connect to sipphone with their own
download of X-Lite (pre-configured), and I can set a profile
in SJPhone by hand and it works too, both incoming and
outgoing, so I have the correct password, etc.
Today, I tested outgoing calls on FWD (actually to use the
peering to test incoming on sipphone), and my calls to FWD
are failing now as well. Incoming from FWD (via IAX2) still
works correctly. Worse, I also tried to go back to SIP-based
outgoing to FWD, and I get the same error as I do for
sipphone, so now I am starting to suspect that it's Asterisk
CVS HEAD that's possibly the problem...
Finally, the machine that is connected to both FWD and
SIPPHONE is on a public static IP address, so there are no
NAT issues involved here, and no STUN services needed
either.
OK, here is the sip.conf entry:
register=1747XXX:[EMAIL PROTECTED]/4321
[proxy01.sipphone.com]
type=peer
;auth=md5
secret=YYY
username=1747XXX
fromuser=1747XXX
fromdomain=proxy01.sipphone.com
host=proxy01.sipphone.com
nat=no
qualify=no
canreinvite=no
disallow=all
allow=ulaw
;context=default
;callerid=Hadar Pedhazur 1747XXX
(The above has been variously named sipphone, sipphone-out
and now proxy01.sipphone.com, all with the same exact
result! Also, the above has been tried with auth=md5
uncommented as well, and also no password, and
insecure=vary, etc.)
Now extensions.conf:
; Dial SIPPhone with a prefix of 76
exten = _76.,1,SetCallerID(${SIPPHONENUM})
exten = _76.,2,SetCIDName(Hadar Pedhazur)
exten = _76.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
OK, here's the output from a call:
-- Called [EMAIL PROTECTED]
-- Got SIP response 500 I'm terribly sorry, server error occured
(1/SL) back from 198.65.166.131
-- SIP/proxy01.sipphone.com-78d5 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
Notice that at the end of the Got SIP response line, is
the correct IP address of their server, so it's finding the
correct server. As mentioned above, if I switch FWD to call
via SIP, I get the same _exact_ error message, but from
FWD's correct IP address rather than SIPPhone. This seems
very suspicious to me...
Finally, just for completeness, here is the CLI output for
attempting to call FWD via IAX2. This used to work, though I
can't say when it started failing:
-- Called fwd-gw/612
-- Call accepted by 65.39.205.121 (format ulaw)
-- Format for call is ulaw
-- IAX2/fwd-gw-4 is busy
I have called _many_ times, and every time I get an instant
is busy in the CLI, and I can receive calls without a
problem, so I don't think it's that they really are busy.
For now, I'm more interested in fixing the SIPPhone problem,
and if that ends up working, and doesn't shed light on the
FWD problem, I'll move on to that. Of course, PITA that it
would be, my next move if no one here can help will be to
restore my settings from a few weeks back (yes, I back up
RE: [Asterisk-Users] Newbie guidance requested --- Grandstream Budgetone
Mike are you able to log into the phones web server configurator page at all? (Im assuming not if it isnt picking up an ip address). Are you able to assign an ip address via the keypad? Are you able to reset the handset via the mac code/reset command. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Chapman Sent: Saturday, March 05, 2005 9:32 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Newbie guidance requested --- Grandstream Budgetone Hi- I am attempting to setup my Budgettone phone for use with my * server and am having problems obtaining an IP address. I have checked the phones settings to make sure it has dhcp enabled and it is. The display says no IP. I bought the phone but do not have any documentation other than the Wiki, but I am still at a loss. What could be preventing the phone from picking up an IP address? Any help would be appreciated. Thanks, Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SAY DIGITS problem
On Sat, 2005-03-05 at 17:25 +0700, Nattapong Mongkolnavin wrote: I have a problem using AGI cmd SAY DIGITS. For some reason I cannot here any thing when the script got executed. However if I use the cmd SAY NUMBER I can here * reading the number fine. fputs($stdout, SAY DIGITS 1234); SAY DIGITS takes two mandatory parameters: the first contains the digits to say (1234 in your case) the second contains the digits that end the command if pressed by the user Example SAY DIGITS 1234 1 says 1234 and stops as soon as the user presses 1 SAY DIGITS 1234 1# says 1234 and stops as soon as the user presses 1 or # If you don't want the user to interrupt you can pass an empty string as second parameter: SAY DIGITS 1234 (of corse the quotes must be escaped in php) Cheers Stefan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems dialing out - possible settings changes
Thought I had this fixed, but it turns out it is not. I've been wracking my brain. Here is what I have done: - Tried 3 different Qwest PSTN lines (just in case it was a line issue) - Tried calling same number from an analog phone plugged directly into the Qwest line - NO PROBLEMS. - Because of this, I think it is an issue with my * implementation. - I have looked at the Dial plan and have made some modifications, cleaning up some extra stuff that was not needed. What is still happening: - I am calling a single number for testing purposes. The same number each time (yes, I have tried other numbers with the exact same results so I decided to stick with one number) - About 1 out of every 3 or 4 times my call is actually successful. The rest of the time I receive the phone company's the number you called is not in service, please check the number and call again - I look at the CLI readout that appears using asterisk -cg to start and it shows ZAP/g1/the correct number Evidently something is happening in translating the dtmf codes, but I am at a loss. More explanation is in my original post below. Any help would be appreciated as we are going to use this as a production box on a small construction job site. Just a guess here... try inserting a w in the beginning of your dial string to delay the sending of dtmf, etc. In some Qwest cases, their equipment is not ready to receive dtmf immediately after going off hook on a call, and effectively drops the first one or two digits. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [OT] - [Asterisk-Users] Why should I answer a Newbie question, therethick!
It's nice to see that some people think so highly of themselves and are above all others. It's quite amusing to watch people like you give thinking so highly of yourself and so little of others. In the spirit of Asterisk and Mark's organization-Digium, I certainly could understand why you aren't employed by them. Maybe you should take a class on being positive and helpful, but again you maybe above all that, in your own mind, others don't think you are that hot, sorry to burst your bubble. --- Andrew Kohlsmith [EMAIL PROTECTED] wrote: On March 5, 2005 01:39 am, Jonathan Hobbs wrote: Ignore them and they will go away. Only after polluting the list with incessant How do I do X? messages, and then only after subsequently polluting the list with asterisk sucks messages, and then all the bad karma of some clueless twitt who couldn't be bothered to embrace OSS in the first place spewing incorrect information around, all because they should have hired a consultant to do their work for them instead. Ignoring them doesn't work, sorry. Education has a (marginally) better shot. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk behind NAT -- SIP config file
Good success story.. I'll keep in mind that router just in case. Thx David. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David J Carter Sent: Sábado, 05 de Marzo de 2005 04:18 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk behind NAT -- SIP config file I have used the Draytek 2600V router in a few locations where only 1 or 2 phones are required. The router has 2 FXS ports and can be used locally to an * box or via the VPN to a remote * box. The VPN built into the routers just works, and I have 1 user who has had 3 VPN circuits up and running now for 6 months solid. Not bad in this day and age for an ADSL to stay functional for so long without interruptions. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Anton Krall Sent: 05 March 2005 04:56 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk behind NAT -- SIP config file The VPN approach might resolv a lot of nat issues I guess... Depending on the scenario I guess.. You could put another * box inside the second nat and interconnect using IAX, or if using a single phone, just use your setup, and finally, if using 2 or more phones and cant put a second * box, well, the vpn solution, I wonder how to do it if you have ATAs and nost softphone on the second NATted LAN.. Well... In time I guess :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rudolf Ladyzhenskii Sent: Viernes, 04 de Marzo de 2005 10:20 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk behind NAT -- SIP config file Yes, only port 5060. If you do not forward 5060, you can not call this phone from outside. Seem to work OK without other ports being forwarded. You mean on the remote sip phone firewall? What if there arem ore than 1 sip phone on that network behidn that firewall? Then you are in trouble. Asterisk only sees single public IP address. As far as it concerns there is only single phone out there. If you get multiple phones working, let me know. Another option, I think, may be using VPN, but I have not tried that. Then you can potentially have remote SIP phones to be on the virtual network. Don't you need to forward ports 1-2 for voice? Or does the sip phones just open up the ports from inside (by doing the in to out calls and keep alives)? I have mot tried to sniff on the traffic in details. I think, other ports are opened in responce to connection on port 5060. The only port listens at is port 5060. Rudolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Clone, Which one?
On March 5, 2005 08:14 am, Androtech wrote: I bought one Trust 56k V92 PCI Internal Modem MD-1100 which has the 1057 Motorola Chip, and I installed it on my linux box. When I try to load the module wcfxo, I cannot load it (zaptel is already loaded): Not to rub salt in the wound, but do you honestly expect the people on a Digium-run mailing list to rush out and help you after you consciously went and bought a clone card? You specifically denied Digium any income on the purchase of this hardware, and now you're asking them for help! You've got a lot of nerve. Caveat Emptor. As far as I'm concerned, you're on your own. If you're not experienced enough to figure this out on your own, you should have purchased the Dev Kit Lite, which comes with support from Digium for specifically these types of problems. Maybe someone else on this list is more forgiving than I am but I really hope not. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to optimize sip??
I remember reading somewhare that you should disable as many unused codecs on sip phones as possible to reduce bandwidth. I'm not sure what I have to do for each type of device I'm using. I have 2 sipura-2000's and 2 Grandstream BudgeTone 100's and a sipura-3000, and a few Xlite PC clients. What do I need to do for each device to reduce the overhead, and disable codecs that I'm not using. What codecs should I be using? Thanks Pat ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [OT] - [Asterisk-Users] Why should I answer a Newbie question, therethick!
On March 5, 2005 10:48 am, asterisk phones wrote: It's nice to see that some people think so highly of themselves and are above all others. It's quite amusing to watch people like you give thinking so highly of yourself and so little of others. In the spirit of Asterisk and Mark's organization-Digium, I certainly could understand why you aren't employed by them. Maybe you should take a class on being positive and helpful, but again you maybe above all that, in your own mind, others don't think you are that hot, sorry to burst your bubble. Blow it out yer arse; If you had a half a clue you'd see just how much I do try to help Asterisk, both on this list and on #asterisk. I am generally a very helpful and positive person, but when it comes to OSS projects you better be making at least a half-assed attempt to help yourself or I'll attach a rate sheet to my reply, plain and simple. I make (good) money off of people who choose to be ignorant, and I give the same service away for free to those who want to actually learn, since it benefits the community. I specifically said that ignoring newbies is not an answer and you come back with this kind of retort? I think it shows your inexperience not only with this particular OSS product, but with mailing lists and open source in general. It's a plain and obvious fact. You ignore someone who needs clue and they get up in arms because OSS/project X/whatever stinks and nobody will help me and I'm gonna go use $foo, because it's so much cooler. I reread my reply (the one you wrote this tripe about) -- I can't see a single hint of my being high on myself -- I merely disagreed with you and gave reasons. I'd even go as far as to say my reasoning is an excellent example of WHY you don't ignore newbies. You're right, they *do* eventually go away, but the cloud of negativity they leave behind is like a lingering fart; it affects the entire community for quite some time, long after the offender has left. As far as others thinking I'm hot -- no need to burst any bubbles, I let people speak for themselves and I personally live by the prison credo -- no matter how big and tough you think you are, there is always someone bigger and tougher. I've got plenty of friends and acquaintences who think I'm quite helpful. I've also got a couple acquaintences and enemies who think otherwise. Obviously you fit into the latter. Oh well, that's your choice. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with g729 codec
when you use Dial application without tTm options, and two User agent use same g.720 codec, The two User agent will transfer media with passthrough. You will no need to install g.729 codec module If you want some commerical G.729 codec, pls visit, http://www.voip-info.org/wiki-ITU+G.729 On Fri, 04 Mar 2005 11:13:26 -0600, Steven Critchfield [EMAIL PROTECTED] wrote: On Fri, 2005-03-04 at 12:02 -0500, Erick Perez wrote: sorry to ask, but what does it mean in passthrough mode ? data, in this case audio, passes from one side through to the other with no need for modification. A standard serial cable is a passthrough cable. Same for standard network patch cables. The software here behaves much the same way, it picks the audio data out of the packet and passes it through to the other side of the communication. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jacky ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Clone, Which one?
On Sat, 2005-03-05 at 11:01 -0500, Andrew Kohlsmith wrote: On March 5, 2005 08:14 am, Androtech wrote: I bought one Trust 56k V92 PCI Internal Modem MD-1100 which has the 1057 Motorola Chip, and I installed it on my linux box. When I try to load the module wcfxo, I cannot load it (zaptel is already loaded): Not to rub salt in the wound, but do you honestly expect the people on a Digium-run mailing list to rush out and help you after you consciously went and bought a clone card? You specifically denied Digium any income on the purchase of this hardware, and now you're asking them for help! You've got a lot of nerve. Caveat Emptor. As far as I'm concerned, you're on your own. If you're not experienced enough to figure this out on your own, you should have purchased the Dev Kit Lite, which comes with support from Digium for specifically these types of problems. Maybe someone else on this list is more forgiving than I am but I really hope not. Just a question, where's the Dev Lite Kit on Digium's site? The PCI Dev kit would give him an FXO and an FXS which may be more than some people want, perhaps the single FXO option could be pushed as this would then be around the same price as the old X100 card and give the same initial connectivity with the possibility of future expansion. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Clone, Which one?
Andrew Kohlsmith wrote: On March 5, 2005 08:14 am, Androtech wrote: I bought one "Trust 56k V92 PCI Internal Modem MD-1100" which has the 1057 Motorola Chip, and I installed it on my linux box. When I try to load the module wcfxo, I cannot load it (zaptel is already loaded): Not to rub salt in the wound, but do you honestly expect the people on a Digium-run mailing list to rush out and help you after you consciously went and bought a clone card? You specifically denied Digium any income on the purchase of this hardware, Since Digium no longer suppliers this card, they were denied NOTHING! and now you're asking them for help! You've got a lot of nerve. Caveat Emptor. There is a LOT of traffic on this list about products that are not supplied by Digium. Do you want to exclude those also? Not to mention the 40-50 percent of traffic that does NOTHING to further the project, but is simple carping about this person not being able to find X, or that person is a dumbass because he asked Y. As far as I'm concerned, you're on your own. Ultimately we are ALL on our own. The hardware that Digium DOES supply is poorly supported. John Novack ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Codec
Currently I have one server running Fedora Core 3 AMD 64bits (on a 3mbits DSL with 640kbits upload) and the second server is running on Mac OS X (on a 512kbits SDSL) I'll change it soon for a PC with Fedora Core 3 but I know the G.729 isn't available for Mac OS X. Is there another codec that do the job well for now? I'll change the server in less then a month so I'll try all the supported codec if I have to until then... Thanks Martin |From: Dana Olson [EMAIL PROTECTED] |Subject: Re: [Asterisk-Users] IAX Codec | |I've called using G729 SIP phones over my LAN, and I think it sounds |quite good. YMMV. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BroadVoice configuration changes for Outbound
Today, We have added INVITE Authentication. This seems to bring a large amount of problems to people in the way since they can't make outbound calls. Here's what needs to be done. You need to add three variables to your peers or friends, username, authuser, and secret. username=phonenumber authuser=phonenumber secret=registration password Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice over Frame Relay Asterisk
Has anyone done Voice Over Frame Relay with Asterisk. With Frame Relay work reliably with Asterisk? Any experiences? If you're talking about transporting voip calls across a path that includes frame relay links, yes it works just fine if you frame network is not congested. Frame relay networks can and _may_ drop packets if the traffic exceeds the committed information rate (cir), depending upon exactly how your provider has their frame switches configured. Dropped packets is less of an issue now in frame networks then what they use to be, and the primary reason for that is the abundance of inexpensive bandwidth currently deployed between frame switches. There is a pecking order in terms of which packets are candidates for being dropped, with broadcast traffic being high on that list. It is very difficult to determine exactly where packets are dropped as you (the user) are never notified by the frame provider when/if they dropped any in their switches. And, if they do drop packets you won't be able to detemine whether those that were dropped were in fact broadcast packets or tcp/udp packets, etc. The BECN and FECN counts can be used to determine if the frame provider is recognizing whether you exceeded your cir rate, however in most real world implementations a BECN or FECN does _not_ translate into a dropped packet (at least in the US). You might want to download Qcheck (it was originally written by NetIQ but spun off to another company now) to evaluate the end-to-end bandwidth. Its a free utility that will help determine what is actually available in terms of bandwidth. If your frame network is congested, you might be able to implement QoS at the border routers to give some preference to voip packets. Rich ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Clone, Which one?
Ummm... This isn't a Digium Run list. This is a Digium sponsored list. My understanding (someone please correct me if I'm wrong) is that this list *is not* a Digium support list. This list is a forum for Asterisk discussion by users. As such, I would suggest that all topics of discussion for all hardware manufacturers are fair game. If one isn't supposed to talk about clone cards in this forum, then this list should not be called Asterisk Users, it should be called Digium Discussion. Asterisk is an OSS product, and as such should be heavily promoted for use with all products, not just Digium's. If Digium is going to take offence to that, then they shouldn't be OSSing the software. There's plenty of discussion around non Digium hardware on this list. If you're going to get pissed off about people discussing non-Digium hardware then get pissed off about *all* of it, not just the X100P clone cards. Heck, Digium doesn't even make the X100P any more. And it's generally accepted on this list that the TDM cards have problems. So I can't say I blame people for looking for cheap alternatives. In any event, if this is a Digium Only list, then it needs to be identified as such, rather than promoted as an Asterisk list. And if that's the case, then I beg someone to start a non-Digium affilitated list so that we can have free and open discussion without worrying about getting slammed on regards, Paul - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, March 05, 2005 9:01 AM Subject: Re: [Asterisk-Users] X100P Clone, Which one? On March 5, 2005 08:14 am, Androtech wrote: I bought one Trust 56k V92 PCI Internal Modem MD-1100 which has the 1057 Motorola Chip, and I installed it on my linux box. When I try to load the module wcfxo, I cannot load it (zaptel is already loaded): Not to rub salt in the wound, but do you honestly expect the people on a Digium-run mailing list to rush out and help you after you consciously went and bought a clone card? You specifically denied Digium any income on the purchase of this hardware, and now you're asking them for help! You've got a lot of nerve. Caveat Emptor. As far as I'm concerned, you're on your own. If you're not experienced enough to figure this out on your own, you should have purchased the Dev Kit Lite, which comes with support from Digium for specifically these types of problems. Maybe someone else on this list is more forgiving than I am but I really hope not. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk for Live-Stream?
I'm looking into solutions for providing a live stream of an event in Belgium [1] - for example, as follows: * Event -- mobile phone -- software answering machine -- Internet server * Event -- mobile phone -- VOIP -- Internet server The live stream should be available in a format so that people can listen to it using XMMS or similar software. Comments? Would Asterisk fit the bill? Alternatives? [1] It's Monday's EU Council of Ministers with Software Patents on the agenda: http://wiki.ffii.org/Dkparl050304En -- Felix E. Klee ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] music on hold issue
Thank you Wiley. I guess I had the "q" version installed. I removed everything and tried with "r" and followed the wiki instructions. Simple Answer-MusicOnHold works fine, so I guess my problem is resolved. - Original Message - From: Wiley Siler To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, March 04, 2005 6:28 PM Subject: RE: [Asterisk-Users] music on hold issue Did you install the version of mpg123 that is 59r and not 59q or 59g? This is a problem with version of mpg1223 almost assuredly. Did you install just as the installation says to do? http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf More stuff.. http://www.voip-info.org/tiki-index.php?page=Asterisk%20mpg123%20redhat http://www.voip-info.org/wiki-mpg123 Cheers, W From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of CJ TomaSent: Friday, March 04, 2005 4:08 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] music on hold issue Hello, I am quite new to asterisk (I've been playing with it for just about 2 weeks). I am trying to do music on hold, but I get this error: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! I have read onsome forumsthat usually this message comes when all resources are tied up, but top gives 0%. The server is Intel 2.4 Ghz with 512 MB RAM. Any suggestions? Thank you. CJ ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Codec
Currently I have one server running Fedora Core 3 AMD 64bits (on a 3mbits DSL with 640kbits upload) and the second server is running on Mac OS X (on a 512kbits SDSL) I'll change it soon for a PC with Fedora Core 3 but I know the G.729 isn't available for Mac OS X. Is there another codec that do the job well for now? Any of the codecs should work just fine. The g711 codec would consume about 80k bits/second, so your sdsl circuit would support about five or six simultanous calls if the sdsl circuit is not used for anything else. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk for Live-Stream?
Something like this sis similar to what you are looking for I think. C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Felix E. Klee Sent: 05 March 2005 17:17 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk for Live-Stream? I'm looking into solutions for providing a live stream of an event in Belgium [1] - for example, as follows: * Event -- mobile phone -- software answering machine -- Internet server * Event -- mobile phone -- VOIP -- Internet server The live stream should be available in a format so that people can listen to it using XMMS or similar software. Comments? Would Asterisk fit the bill? Alternatives? [1] It's Monday's EU Council of Ministers with Software Patents on the agenda: http://wiki.ffii.org/Dkparl050304En -- Felix E. Klee ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sorry to be a bother ISO root password
As far as I can make out the root password for the ISO download is supposed to be epping or EPPING depending upon which version you are using. I've downloaded an ISO image from the following link but neither passwords seem to work :( http://ovh.dl.sourceforge.net:80/sourceforge/asteriskathome/asteriskathome-0.6.iso any one know the password for this one? -- Regards Phil -- This message was scanned for spam and viruses by BitDefender. For more information please visit http://linux.bitdefender.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Clone, Which one?
Ummm... This isn't a Digium Run list. This is a Digium sponsored list. My understanding (someone please correct me if I'm wrong) is that this list *is not* a Digium support list. This list is a forum for Asterisk discussion by users. As such, I would suggest that all topics of discussion for all hardware manufacturers are fair game. Actually the list is maintained/managed by digium (Martin does most of the work, more or less on a part time basis). Really don't care whose servers or networks it might run on, but they are the defacto owners of the list. If one isn't supposed to talk about clone cards in this forum, then this list should not be called Asterisk Users, it should be called Digium Discussion. There seems to be about a half dozen self-appointed list cops, and none of them speak for Mark, digium or asterisk. Several of those are lurking on this list only to find fresh meat to sell their services to. It's obvious who they are. Much easier to config the mail reader to send those to your favorite trash bucket then it is to keep reading their BS day after day. If there really were any official-sponsored restrictions on the list, the words would come from Mark, Digium, etc. Come to think of it, maybe all of us that are fed up with their postings should just forward those back to their email address. Maybe they would get the hint and some manners. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice over Frame Relay Asterisk
Great, thanks, that was the information I was looking for. --- Rich Adamson [EMAIL PROTECTED] wrote: Has anyone done Voice Over Frame Relay with Asterisk. With Frame Relay work reliably with Asterisk? Any experiences? If you're talking about transporting voip calls across a path that includes frame relay links, yes it works just fine if you frame network is not congested. Frame relay networks can and _may_ drop packets if the traffic exceeds the committed information rate (cir), depending upon exactly how your provider has their frame switches configured. Dropped packets is less of an issue now in frame networks then what they use to be, and the primary reason for that is the abundance of inexpensive bandwidth currently deployed between frame switches. There is a pecking order in terms of which packets are candidates for being dropped, with broadcast traffic being high on that list. It is very difficult to determine exactly where packets are dropped as you (the user) are never notified by the frame provider when/if they dropped any in their switches. And, if they do drop packets you won't be able to detemine whether those that were dropped were in fact broadcast packets or tcp/udp packets, etc. The BECN and FECN counts can be used to determine if the frame provider is recognizing whether you exceeded your cir rate, however in most real world implementations a BECN or FECN does _not_ translate into a dropped packet (at least in the US). You might want to download Qcheck (it was originally written by NetIQ but spun off to another company now) to evaluate the end-to-end bandwidth. Its a free utility that will help determine what is actually available in terms of bandwidth. If your frame network is congested, you might be able to implement QoS at the border routers to give some preference to voip packets. Rich ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web http://birthday.yahoo.com/netrospective/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Clone, Which one?
Rich Adamson wrote: snip There seems to be about a half dozen self-appointed list cops, and none of them speak for Mark, digium or asterisk. Several of those are lurking on this list only to find fresh meat to sell their services to. It's obvious who they are. Indeed. Much easier to config the mail reader to send those to your favorite trash bucket then it is to keep reading their BS day after day. Agreed. They contribute NOTHING! If there really were any official-sponsored restrictions on the list, the words would come from Mark, Digium, etc. Come to think of it, maybe all of us that are fed up with their postings should just forward those back to their email address. Maybe they would get the hint and some manners. Nah! They were never taught any social skills, and it's too late now. John Novack ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sorry to be a bother ISO root password
snip I've downloaded an ISO image from the following link but neither passwords seem to work :( http://ovh.dl.sourceforge.net:80/sourceforge/asteriskathome/as teriskathome-0.6.iso any one know the password for this one? -- Regards Phil The root password for 0.6 is (I think) password HTH Charlie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sorry to be a bother ISO root password
That worked a treat many thanks. -- Regards Phil -- This message was scanned for spam and viruses by BitDefender. For more information please visit http://linux.bitdefender.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Clone, Which one?
Andrew Kohlsmith wrote: On March 5, 2005 08:14 am, Androtech wrote: I bought one Trust 56k V92 PCI Internal Modem MD-1100 which has the 1057 Motorola Chip, and I installed it on my linux box. When I try to load the module wcfxo, I cannot load it (zaptel is already loaded): Not to rub salt in the wound, but do you honestly expect the people on a Digium-run mailing list to rush out and help you after you consciously went and bought a clone card? You specifically denied Digium any income on the purchase of this hardware, and now you're asking them for help! You've got a lot of nerve. Caveat Emptor. As far as I'm concerned, you're on your own. If you're not experienced enough to figure this out on your own, you should have purchased the Dev Kit Lite, which comes with support from Digium for specifically these types of problems. Maybe someone else on this list is more forgiving than I am but I really hope not. Funny thing is that it was the digium website that first led me to the asterisk website and then on this page: http://www.asterisk.org/index.php?menu=hardware I found the link to the X100P clone card. Some searching led me to a vendor where I got 6 of these with shipping fro about $56 total. I got them for experimentation. I am actually hoping that Digium will develop a standalone FXO-IAX2 adaptor someday. I like external standalone stuff because I don't like shutting down servers and removing the covers too much. Anyway, I don't feel guilty at all. I will buy any G.729 licenses I need from them. If they don't ever do a standalone fxo device, I will probably create one using a mini-ITX mainboard and their 4-port card. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Clone, Which one?
On March 5, 2005 12:22 pm, Rich Adamson wrote: There seems to be about a half dozen self-appointed list cops, and none of them speak for Mark, digium or asterisk. Several of those are lurking on this list only to find fresh meat to sell their services to. It's obvious who they are. I have *never* claimed to speak for Digium. Actually I don't think any of the self appointed list cops do. If you can provide me with archive URLs that prove otherwise I'd be surprised. And I also don't think that any of us self appointed list cops actively solicit our services. Again, if you'll care to peruse the list archives you'll find I am very helpful on this list, just as you are. If I was out to make a killing with Asterisk consulting I wouldn't do that, so your argument fails on this point. As I'm not subscribed to asterisk-biz I think that's an even further failure of that point. I think it comes down to tolerances. Some days I am far less tolerant than others when it comes to people who just don't want to put forward any effort. Today was particularly bad and if I was a smarter person, would probably refrain from reading the list at all. I don't always self-regulate so well, and this particular thread's a good example. Perhaps I was a little harsh when I said I hoped that nobody would help this guy out. In fact, I know I was, and I do apologize for that. I do not, however, apologize for saying if he's not able to solve these kinds of problems that he should be buying Digium hardware. When you buy outside you are on your own, and buying a clone X100P is particularly bad for this since there are so many different PCI IDs. Whether the X100P is available from Digium or not is irrelavent, there are solutions Digium has that would solve this guy's particular problem, support Digium *and* enable him to obtain support which was included in the cost of the hardware. Is it as cheap as a $20 clone X100P without its PCI ID in the driver? Hell no, but he would have functioning hardware. How much is his time worth? Digium (well Mark, but he *is* Digium) has provided the PCI IDs of several of the popular clones in the driver already. It seems our original poster either didn't buy one that was compatible, or it's even more out there than the clones Mark's already added support for. Again -- I wonder if it still feels like such a shit-hot deal. Either way -- he's gone and done his research and is stuck. I personally won't help him out of this jam, but others, perhaps even you, will. Aside: I find it an interesting datapoint that you (and everyone else, but you in particular) would reply to this thread to bitch about me rather than giving this guy his solution, especially since I'm so wrong in berating him. Much easier to config the mail reader to send those to your favorite trash bucket then it is to keep reading their BS day after day. Feel free to do so, it won't affect me in any way shape or form. If there really were any official-sponsored restrictions on the list, the words would come from Mark, Digium, etc. Totally agreed. Come to think of it, maybe all of us that are fed up with their postings should just forward those back to their email address. Maybe they would get the hint and some manners. I have plenty of manners, but sometimes I forget them and say stupid things like I did to the original poster. Further, I'm sure you'd know that any kind of stupid auto-reply like that would just escalate into an auto-reply-reply (I'm not beneath tit-for-tat if you care to play that game) and just continue until both of us had our mail filters tuned up so well that we'd just ignore each other's immature flooding, and our auto-repliers would end up wasting the available bandwidth between us, essentially proving nothing. :-) Just in case you missed it, I am sorry that I wrote that I hope that nobody helps this guy. I stand behind my other points in the original email, though. Clone/alternative hardware can work great, but if it doesn't you find yourself in this guy's shoes. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Clone, Which one?
On March 5, 2005 11:01 am, Andrew Kohlsmith wrote: specifically these types of problems. Maybe someone else on this list is more forgiving than I am but I really hope not. I apologize for this remark. I still do feel, though, that if you're this new to asterisk that you should have purchased hardware which offered support from Digium. Once your system is up and running and you have some experience under your belt, feel free to be as cheap as you like. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How do I reload extensions included in a switch statement in extensions.conf?
Hi, I have two Asterisk servers and I forward calls from one to the other. How do I reload extensions included in a switch statement in extensions.con? I have tried extensions reload, reload and restart now, and it's only restart now that works. Is this how it is supposed to work or can it be a misconfiguration? Regards, Mikael Magnusson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cant compile app_meetme2
The code is configured to allow use of either mysql or postgres, so you will need to install the postgres-dev package, or comment out all postgres related code. Once you have the postgres libraries installed you have two more changes to make. line 645 needs to become: AST_MUTEX_DEFINE_STATIC(conflock); And line 1547 should be this: res = ast_say_number(chan, cnt, , chan-language, (char *) NULL); Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jer Sent: Saturday, March 05, 2005 2:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] cant compile app_meetme2 At 04:34 AM 3/5/2005, you wrote: The error messages are Postgres related. You need to have a special postgres include file (postgres-dev files) to make it compile or disable postpres support somehow. I'm using debian and the the concering include file resided in a subdirectory of what asterisk was told. if this is the case why dont i see a include file missing error someplace? or am I missing something.. Jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Clone, Which one?
On March 5, 2005 11:57 am, Dave Cotton wrote: Just a question, where's the Dev Lite Kit on Digium's site? I meant Dev Kit and you're right, it's an FXO and FXS on the carrier card. US$195. The PCI Dev kit would give him an FXO and an FXS which may be more than some people want, perhaps the single FXO option could be pushed as this would then be around the same price as the old X100 card and give the same initial connectivity with the possibility of future expansion. Agreed. A TDM400P with 1 FXO would run him US$133 at the Yahoo! store, and perhaps cheaper if he can find one on Ebay. Still not as cheap as the (I'm guessing $20) winmodem that doesn't work, but is the $113 he saved worth it? -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE110P module woes
Hi, I've no experience with the TE110, but this is a known problem with the TE405 and TE410. They apparently can get locked up, and only a power cycle will clear it. Good hint, I'll take that into account when testing. Hope the TE100 is better built than that, though. At least, for the cost, one would expect that it behaved like a well educated card. Hi, I have been using asterisk for a couple of months now and for thee most part, I love it. However, I'm having a problem with the drivers of the Digium TE110P. I have tried both the Debian package and the CVS. I have tried several kernels, and am now at 2.6.11. This has been working before (with 2.6.8.1), but after a reboot it stopped working and I am not able to consistently make it work or fail. I have make clean, make and make install, no complains from make. The zaptel module loads fine and says so: Zapata Telephony Interface Registered on major 196 But the module for the TE110P fails. If I only modprobe it, it loads silently; but the moment I execute ztcfg, I get: ZT_SPANCONFIG failed on span 1: No such device or address (6) If I ask for more verbose errors, I get: Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: Individual Clear channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: Individual Clear channel (Default) (Slaves: 06) Channel 07: Individual Clear channel (Default) (Slaves: 07) Channel 08: Individual Clear channel (Default) (Slaves: 08) Channel 09: Individual Clear channel (Default) (Slaves: 09) Channel 10: Individual Clear channel (Default) (Slaves: 10) Channel 11: Individual Clear channel (Default) (Slaves: 11) Channel 12: Individual Clear channel (Default) (Slaves: 12) Channel 13: Individual Clear channel (Default) (Slaves: 13) Channel 14: Individual Clear channel (Default) (Slaves: 14) Channel 15: Individual Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Individual Clear channel (Default) (Slaves: 17) Channel 18: Individual Clear channel (Default) (Slaves: 18) Channel 19: Individual Clear channel (Default) (Slaves: 19) Channel 20: Individual Clear channel (Default) (Slaves: 20) Channel 21: Individual Clear channel (Default) (Slaves: 21) Channel 22: Individual Clear channel (Default) (Slaves: 22) Channel 23: Individual Clear channel (Default) (Slaves: 23) Channel 24: Individual Clear channel (Default) (Slaves: 24) Channel 25: Individual Clear channel (Default) (Slaves: 25) Channel 26: Individual Clear channel (Default) (Slaves: 26) Channel 27: Individual Clear channel (Default) (Slaves: 27) Channel 28: Individual Clear channel (Default) (Slaves: 28) Channel 29: Individual Clear channel (Default) (Slaves: 29) Channel 30: Individual Clear channel (Default) (Slaves: 30) Channel 31: Individual Clear channel (Default) (Slaves: 31) 31 channels configured. ZT_SPANCONFIG failed on span 1: No such device or address (6) Any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Clone, Which one?
On March 5, 2005 12:02 pm, John Novack wrote: Since Digium no longer suppliers this card, they were denied NOTHING! They offer comparable hardware. TDM410P is $113. There is a LOT of traffic on this list about products that are not supplied by Digium. Do you want to exclude those also? The Sangoma guys typically handle support for their own product, even on this list. Atacomm's card hasn't hit the market yet. The Sipura people sell hardware that Digium doesn't have similar hardware for. I guess my specific beef was that he spent the time to research clone cards, bought one and is now asking for help solving his problem where if he just bought the Digium hardware he'd be up and running. He's not stupid, and I'm willing to bet he's quite the opposite. Not to mention the 40-50 percent of traffic that does NOTHING to further the project, but is simple carping about this person not being able to find X, or that person is a dumbass because he asked Y. Feel free to modify your mail filters to adjust the reality to your liking. I've already apologized for my remark about hoping nobody'd help him. I still feel my point stands. And if nobody's going to educate the newbies, then how will they ever learn? Do you believe in letting your children do whatever they want, too? There are 'defacto' rules for any system. No, I don't have my shiny ListCoptm badge and I'm not, nor have I ever claimed to, speak on behalf of Digium or anyone else. I'm taking time to try and educate people on what I perceive as normal and proper list ettiquette. Not everyone agrees with me, this I will admit. However most people do, and that's why lists are how they are. You can easily see that I spend a great deal of time not only here but also on IRC helping people out. Sometimes, like my original reply to this thread, I go overboard, but I generally have the decency to apologize. Maybe you don't agree but I feel that my contributions to this list far outweigh what I detract from it. You take the good with the bad. Ultimately we are ALL on our own. The hardware that Digium DOES supply is poorly supported. I've generally had nothing but good support from Digium. Now mind you I haven't had all that many problems with their hardware, so YMMV. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk for Live-Stream?
Hi! I'm looking into solutions for providing a live stream of an event in Belgium [1] - for example, as follows: How about icecast: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Ices Another approach: Dial into a MeetMe conference, and connect some client to that conference that takes care of the streaming part. Cheers, Philipp ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Clone, Which one?
- Original Message - There is a LOT of traffic on this list about products that are not supplied by Digium. Do you want to exclude those also? The Sangoma guys typically handle support for their own product, even on this list. Atacomm's card hasn't hit the market yet. The Sipura people sell hardware that Digium doesn't have similar hardware for. Very true, but I guess the point I'm trying to make is that whether or not Digium supports or runs this list, my understanding is that this list isn't intended to be a Digium hardware support forum, it's intended to be a general Asterisk Users Discussion forum. I would never expect someone to call up Digium's support channels and expect to get support for setting up a clone card. But this isn't Digium Support. The guy wasn't asking Digium to help solve his problem, he was asking 'people who use Asterisk' to help solve his problem. I'd be willing to bet there are a lot of people reading this list who have at least one clone card sitting in a server at home. It's a Users Discussion. ie. People who use Asterisk converge here to talk about Asterisk in all it's forms. Whether or not Digium runs/supports the list is beside the point. I don't really disagree with you about the cost factor - often times the pain saved by buying a $125 device is worth the extra money. However, not everyone has that luxury. While some of us (myself included) prefer to simply fix the problem by throwing money at it, a lot of people use Linux and OSS products not only because it interests them but because they're scraping out a living that doesn't let them work on anything more than the hand-me-down hardware that they've coersed off of the various people they've helped over the years, and they spend the time making it all work because time is what they've got - not money. regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Clone, Which one?
And if nobody's going to educate the newbies, then how will they ever learn? Do you believe in letting your children do whatever they want, too? There are 'defacto' rules for any system. No, I don't have my shiny ListCoptm There is a difference when you are that childs father or mother but you are neither? Do you stop people in the street when you see them doing things wrong and try and tell them you shouldn't be doing it like that, this is how you should do it. I hope not :) Essentially this list is the same. There are lots of very helpful people, keen to provide help and enthusiasm to new asterisk people, then there are those who seem to have so much time on there hands they have to reply to every newbies question telling them to go and look at google before asking that here! badge and I'm not, nor have I ever claimed to, speak on behalf of Digium or anyone else. I'm taking time to try and educate people on what I perceive as normal and proper list ettiquette. Not everyone agrees with me, this I will admit. However most people do, and that's why lists are how they are. And how exactly is that? Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX on netweb EEZEE phone
Hi Nathan, Nathan C. Smith wrote: I'm running asterisk stable 1.0.5 and I'm trying to get the netweb eezee phone version v1.37.008 to talk IAX to asterisk. The pages I saw in the Try the wiki, myself and someone else wrote up a pretty big howto and tips and tricks on these phones. http://voip-info.org/wiki-Atcom HTH, matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with loging on guest account
Dnia 2005-03-05 15:04, Uytkownik Marcin Zajczkowski napisa: I've just compiled and installed Asterisk (1.0.5). After some problems with codecs I could successfully connect to server by: [EMAIL PROTECTED] Next I created account at iaxtel.com and configured iaxcomm to work with this account. Unfortunately after that I had problem with logging as guest. Calling as [EMAIL PROTECTED] tried to connect with my local server through. So I changed it to: 192.168.0.1/guest and 192.168.0.1/[EMAIL PROTECTED] but I had errors: (...) Ehhh... After hours of fight with register IAX users I've found that [EMAIL PROTECTED]/ (with / at the end) forces direct connection... Marcin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Clone, Which one?
On Sat, 2005-03-05 at 19:30 +, Mike Dent wrote: And if nobody's going to educate the newbies, then how will they ever learn? Do you believe in letting your children do whatever they want, too? There are 'defacto' rules for any system. No, I don't have my shiny ListCoptm There is a difference when you are that childs father or mother but you are neither? Seems shop owners don't mind telling a person how to behave or leave. Any time you are in an enclosed space, people attempt to enforce rules. Do you stop people in the street when you see them doing things wrong and try and tell them you shouldn't be doing it like that, this is how you should do it. I hope not :) Essentially this list is the same. While you may not stop people who haven't asked for help, but once you enter a public forum and request the help you need to be following the rules or at the least be respectful of those willing to give the help. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Clone, Which one?
Mike Dent wrote: And if nobody's going to educate the newbies, then how will they ever learn? Do you believe in letting your children do whatever they want, too? There are 'defacto' rules for any system. No, I don't have my shiny ListCoptm There is a difference when you are that childs father or mother but you are neither? Do you stop people in the street when you see them doing things wrong and try and tell them you shouldn't be doing it like that, this is how you should do it. I hope not :) Essentially this list is the same. There are lots of very helpful people, keen to provide help and enthusiasm to new asterisk people, then there are those who seem to have so much time on there hands they have to reply to every newbies question telling them to go and look at google before asking that here! Maybe one of the free web-based forum packages will eventually offer an elitist or impatient mode. Before you can post, you do the required reading and pass online exams. The idea is to weed out people who think README is just another geek buzzword. You have to know what a FAQ is and you have to know what RTFM means. Most of those people who tend to scold every newbie are probably elitist pretenders. If they were truly elite, they would be too busy to read and reply to such posts. Of course some of us are just impatient, regardless of our skill level. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Softphones
What iax2 softphones are you guys using? Ive trying some but I find some lack certain features and others have them but lacks others. For example, I tried firefly, simple interface but seems it can only handle 1 line, no MWI. IAX Phone has multiple lines and MWI but seems it can only handle gsm and not iLBC.. Am I right? So.. My question is... Which client can handle multiple lines, IAX2, MWI, multiple codecs including iLBC and has a more telephone like interface (like the one in IAX Phone? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Capi installation with Fedora Core 3 (AVM Fritz!)
Hello! I am a newbie with asterisk, I´d like to install capi on FC3, I´ve tried to follow a little howto (http://voip-info.org/wiki-Asterisk+Linux+Fedora), but it is for FC1, and when I do a modprobe fcpci it fails (module not found). Please some help!! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dead SCCP client since upgrade to Asterisk 1.0.6-BRIstuffed-0.2.0-RC7j?
Hi list! I'm using phones that emulate a Cisco 7940 with chan_sccp. When I was using Asterisk 1.0.5 (bristuffed) I never had any such message on the console. The phones do work. Is this a bug in chan_sccp or a feature of asterisk 1.0.6? Thx! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Clone, Which one?
On March 5, 2005 02:30 pm, Mike Dent wrote: There is a difference when you are that childs father or mother but you are neither? You have a point, but... (read on) Do you stop people in the street when you see them doing things wrong and try and tell them you shouldn't be doing it like that, this is how you should do it. I hope not :) Essentially this list is the same. It's not the same at all -- Random people in the street aren't in a forum asking for help. If you're making a ruckus in a public library I can and have asked if you could tone it down, as it's a library and there are certain de-facto rules which the majority of people adhere to. c.f. for movie theatres, bars, strip clubs, sports avenues, etc. There are lots of very helpful people, keen to provide help and enthusiasm to new asterisk people, then there are those who seem to have so much time on there hands they have to reply to every newbies question telling them to go and look at google before asking that here! I implore you to go do some of this research yourself before trying to paint me with a certain brush. This thread aside, what percentage of my posts to -users are helpful, and what percentage are for educating newbies in ettiquette? Finally, what percentage are me just out-and-out flaming a newbie? Perhaps it's just my rants that seem to catch your eye? And how exactly is that? A generally helpful forum for people WHO HAVE SHOWED A MODICUM OF RESEARCH to obtain assistance in setting up, running and diagnosing problems with their Asterisk installations. Without some form of rules or policing as you want to call it, this forum would not be as effective as it is. I think where the problem comes in is that people take this forum to be asterisk-biz half the time. I need X done **RIGHT NOW**!! I DEMAND HELP!! -- take it to -biz, there are dozens if not hundreds of consultants who will happily exchange money for experience. This forum is for people who want to learn. Really really want to learn, not just gesticulate the intention. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Clone, Which one?
On March 5, 2005 02:46 pm, Paul wrote: Maybe one of the free web-based forum packages will eventually offer an elitist or impatient mode. Before you can post, you do the required reading and pass online exams. The idea is to weed out people who think README is just another geek buzzword. You have to know what a FAQ is and you have to know what RTFM means. That wouldn't help. If you're unwilling to learn, then this list isn't for you. Plain and simple. OSS grows when people learn. Most of those people who tend to scold every newbie are probably elitist pretenders. If they were truly elite, they would be too busy to read and reply to such posts. Of course some of us are just impatient, regardless of our skill level. Believe what you want, and I'l continue using my Asterisk servers and providing service to my customers. As I mentioned to another poster, feel free to manipulate your MUA's filters to adjust reality to your liking. FWIW, I think only the truly elite possess the time to help out others, since the elitist pretenders are too busy fiddling with their systems and trying to get them to work properly to be able to help. :-) -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P Clone, Which one?
Well, considering I'm on topic, I shouldn't get flamed to badly for this. I have a bunch of these working well in my home experiments: http://www.laptops4me.com/product_info.php/products_id/1444 And yes that price is correct and they do arrive. :) Not everyone can justify buying the supported hardware to kick the tires and try it out * at home. On the commercial side support is worth every penny I'm sure. However I think it helps the community to have low cost entry options for people to learn. I know it helped me. -James ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE110P module woes
On 5 Mar 2005, at 18:44, Alfredo Sola wrote: Hi, I've no experience with the TE110, but this is a known problem with the TE405 and TE410. They apparently can get locked up, and only a power cycle will clear it. Good hint, I'll take that into account when testing. Hope the TE100 is better built than that, though. At least, for the cost, one would expect that it behaved like a well educated card. I'm not sure it is a card quality issue. I had exactly the same sort of problems with a Dialogic E1 card that cost ten times as much. My feeling (unsupported) is that the powercycle does a better job of forcing the far end of an E1 (e.g. the PTT's equipment) to start afresh than just reinitializing the cards. If you turn the power off you can be sure that you are going to drop carrier, clock and any control lines totally. Tim. http://www.westhawk.co.uk/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zultys Zip 2
Anyone using this Sip phone with Asterisk? If you have had success getting the message waiting indication to work, please contact me off list. TIA John Novack ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE110P module woes
On March 5, 2005 04:15 pm, tim panton wrote: My feeling (unsupported) is that the powercycle does a better job of forcing the far end of an E1 (e.g. the PTT's equipment) to start afresh than just reinitializing the cards. If you turn the power off you can be sure that you are going to drop carrier, clock and any control lines totally. While I agree, I also feel that any proper design should be able to do that from within the driver. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sorry to be a bother ISO root password
As far as I can make out the root password for the ISO download is supposed to be epping or EPPING depending upon which version you are using. I've downloaded an ISO image from the following link but neither passwords seem to work :( http://ovh.dl.sourceforge.net:80/sourceforge/asteriskathome/asteriskathome-0.6.iso any one know the password for this one? Hi Phil, http://asteriskathome.sourceforge.net/install_doc.html says that the password is password. Don't know for sure, because I haven't installed it yet. Good luck, Stewart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DVG-1120M - S
Does any have ... or know where I can find firmware to convert a DVG-1120M (MGCP) to a DVG-1120S (SIP)?? Thanks, Rob ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digium Reseller in the UK ?
Can anyone recommend a Digium Reseller in the UK ? Thanks in Advance Nigel begin:vcard fn:Nigel Taylor n:Taylor;Nigel org:ITAzure Limited adr:15 Warren Park Way;;Dunn House;Enderby;Leicestershire;LE19 4SA;United Kingdom email;internet:[EMAIL PROTECTED] title:Technology Director tel;work:0116 286 3016 url:http://www.itazure.com version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Clone, Which one?
- Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] I think where the problem comes in is that people take this forum to be asterisk-biz half the time. I need X done **RIGHT NOW**!! I DEMAND HELP!! -- take it to -biz, there are dozens if not hundreds of consultants who will happily exchange money for experience. This forum is for people who want to Geepers man. Looking at the last couple of times you tried to 'educate' someone, I don't see anything in their messages that sound like I need X done **RIGHT NOW**!! I DEMAND HELP!!. Uneducated and demanding are not necessarily the same thing. Whatever, we're just going around in endless circles here. I'll get out of this discussion now and leave it up to those who wish to continue it regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BroadVoice configuration changes for Outbound
Thanks for this info, Dan! I noticed immediately that outbound was broken, and inbound was OK. I saw your posting just prior to going berserk... a warning email from Broadvoice would have been nice, they knew to email me when that SIP-patch from edvina came out some months ago. Anyway, thanks for the posting! Jerry On Sat, 5 Mar 2005, Dan Weber wrote: Today, We have added INVITE Authentication. This seems to bring a large amount of problems to people in the way since they can't make outbound calls. Here's what needs to be done. You need to add three variables to your peers or friends, username, authuser, and secret. username=phonenumber authuser=phonenumber secret=registration password Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type IAX2
I don't know if this is still true, but Iax clients had problems when you check them with qualify (set latter to no)... HTH, Rob. - Original Message - From: Anton Krall [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Saturday, March 05, 2005 9:05 AM Subject: RE: [Asterisk-Users] Unable to create channel of type IAX2 And when it does work, the console says: Mar 5 02:07:08 NOTICE[9962]: chan_iax2.c:7065 iax2_poke_noanswer: Peer 'akralliax' is now UNREACHABLE! Time: 5 Mar 5 02:07:18 NOTICE[9962]: chan_iax2.c:6420 socket_read: Peer 'akralliax' is now REACHABLE! Time: 3 The iaxcomm phone is on the same LAN, so why can it be coming and going? Any ideas? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Sábado, 05 de Marzo de 2005 01:55 a.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Unable to create channel of type IAX2 Guys.. Im trying to setup a fotphone using iaxcomm and when I dial that softphones extension, * complains of this: Mar 5 01:54:54 NOTICE[9962]: app_dial.c:936 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3) Any hints? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Block anonymous calls
Fredrik wrote:
I see from my CDR's that some of my callers also have unknown in
their FROM field. I would like to let them through. Only block the
FROM anonymous that the telemarketers use.
Fredrik, I found something on the Wiki a while back... Try this...
exten = s,1,Answer
exten = s,2,NoOp(${CALLERID})
exten = s,3,ResponseTimeout(10)
exten = s,4,GotoIf($[${CALLERIDNUM} = ]?|1000)
exten = s,5,GotoIf($[${CALLERIDNUM:0:3} = 877]?s|1000)
exten = s,6,GotoIf($[${CALLERIDNUM:0:3} = 866]?s|1000)
exten = s,7,GotoIf($[${CALLERIDNUM:0:3} = 800]?s|1000)
exten = s,8,Macro(stdexten,${SIP0})
exten = s,9,Hangup
exten = s,1000,Background(SPAMSTOPPER)
exten = s,1001,Hangup
I have used this for a few months at home, and it works great...
Blake
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Re: [Asterisk-Users] X100P Clone, Which one?
Andrew Kohlsmith wrote: On March 5, 2005 02:46 pm, Paul wrote: Maybe one of the free web-based forum packages will eventually offer an elitist or impatient mode. Before you can post, you do the required reading and pass online exams. The idea is to weed out people who think README is just another geek buzzword. You have to know what a FAQ is and you have to know what RTFM means. That wouldn't help. If you're unwilling to learn, then this list isn't for you. Plain and simple. OSS grows when people learn. I was merely attempting to be sarcastically humorous. Most of those people who tend to scold every newbie are probably elitist pretenders. If they were truly elite, they would be too busy to read and reply to such posts. Of course some of us are just impatient, regardless of our skill level. Believe what you want, and I'l continue using my Asterisk servers and providing service to my customers. As I mentioned to another poster, feel free to manipulate your MUA's filters to adjust reality to your liking. FWIW, I think only the truly elite possess the time to help out others, since the elitist pretenders are too busy fiddling with their systems and trying to get them to work properly to be able to help. :-) FWIW - some of us have paying customers and helping them takes absolute over helping newbies who won't try to help themselves first. Some of them actually are capable of reading and studying the information already out there, but they are also selfish and lazy. A good illustration of this attitude happened last week: I had 3 paid technicicians who didn't have enough common sense to pick up a phone and listen for modem tone when a customer had problems with a dedicated dialup. The customer had neglected to pay the long distance carrier. Do you think the majority of us want to babysit people like this for free? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digium hardware in the UK ?
Can anyone recommend a source of Digium hardware in the UK ? Thanks in advance Nigel begin:vcard fn:Nigel Taylor n:Taylor;Nigel org:ITAzure Limited adr:15 Warren Park Way;;Dunn House;Enderby;Leicestershire;LE19 4SA;United Kingdom email;internet:[EMAIL PROTECTED] title:Technology Director tel;work:0116 286 3016 url:http://www.itazure.com version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BroadVoice configuration changes for Outbound
An Open letter to Broadvoice from an Asterisk user... (This is not a solicitation for support from the Asterisk list. The specifics of my problems have already been emailed to their support team.) May I suggest: 1) Updating your website that tells how to configure Asterisk for Broadvoice. 2) Answering emails to [EMAIL PROTECTED] 3) Emailing your users that signed up as BYOB when you think a change might break stuff. I *really* want to be a happy Broadvoice user. Your willingness to openly support Asterisk users is a huge selling point and the thing that gives you an edge in my mind. As an early adopter kinda guy, I'm happy to tweak stuff to make things work. I can't however explain to my wife why the phone doesn't work *again*. I'm going to hang in there a bit longer in hopes that things will get better, if they don't, it's off to another VSP. Best Wishes, Gabe On Sat, 5 Mar 2005 12:13:08 -0500 (EST), Dan Weber [EMAIL PROTECTED] wrote: Today, We have added INVITE Authentication. This seems to bring a large amount of problems to people in the way since they can't make outbound calls. Here's what needs to be done. You need to add three variables to your peers or friends, username, authuser, and secret. username=phonenumber authuser=phonenumber secret=registration password Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 (Variables)
Anyone knows what are the variables in an inbound IAX2 call who reflect the actual codec and DNID, DNIS, original peer description, I'm only able to see it during an iax debug Timestamp: 3ms SCall: 1 DCall: 0 [XX.XX.XX.XX:5036] VERSION : 2 CALLED NUMBER : XX CALLING NUMBER : asterisk CALLING NAME: asterisk LANGUAGE: en USERNAME: xxx FORMAT : 2 CAPABILITY : 18 ADSICPE : 2 DATE TIME : 173807980 Tkx. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
