Re: [Asterisk-Users] Soekris net4801 and analog interface?
X100P is 3.3v not 5v, at least the one I have. Works fine in a 4801. John Simon wrote: Is anyone using a net4801 and an analog only setup? I am looking for a modem that is PCI 3.3V, apparently the X100P is 5.0V PCI only so it won't work with the net4801. ___ 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] ASTCC: perl / mysql or me???
I try to change something in ASTCC, but I am now totally blind,
I hang on one line now. I changed:
vpbx:/var/lib/asterisk/agi-bin # diff astcc-original.agi astcc.agi
22c22
# exten = _00X,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN})
---
# exten =
_00X,1,DeadAGI(astcc.agi,${CALLERIDNUM},${TARIFF},${EXTEN})
35c35
# exten = _00X,1,DeadAGI(astcc.agi,${CALLERIDNUM},BALANCE,1)
---
# exten = _00X,1,DeadAGI(astcc.agi,${CALLERIDNUM},BALANCE,'',1)
273,274c273,276
I added one parameter ${TARIFF}
my ($number) = @_;
my $sth = $dbh-prepare(SELECT * FROM routes WHERE .
$dbh-quote($number) . RLIKE pattern ORDER BY LENGTH(pattern) DESC);
---
my ($number, $tariff1) = @_;
my $sth = $dbh-prepare(SELECT * FROM . $tariff1 . WHERE
. $dbh-quote($number) . RLIKE pattern ORDER BY LENGTH(pattern) DESC);
print STDERR sth = $sth\n;
277a280
print STDERR res = $res\n;
413c416
($calleridnum, $phoneno, $quiet) = @ARGV;
---
($calleridnum, $phoneno, $tariff, $quiet) = @ARGV;
521c524
print STDERR Phone number is $phoneno\n;
---
print STDERR 1. Phone number is $phoneno\nTariff is
$tariff\n;
526c529
$numdata = getphone($phoneno);
---
$numdata = getphone($phoneno, $tariff);
554c557,560
$numdata = getphone($phoneno);
---
print STDERR 2. Phone number is $phoneno\nTariff is $tariff\n;
$numdata = getphone($phoneno, $tariff);
print STDERR 2.a numdata = $numdata\n;
print STDERR 2.b Matching pattern is $numdata-{pattern}\n;
555a562
print STDERR 2.c numdata = $numdata\n;
556a564
print STDERR 2.d quiet = $quiet\n;
vpbx:/var/lib/asterisk/agi-bin #
What happens is, when I use the $TARIFF=routes (what was the original
name) it works! If I use the new table name I had added to the database,
than it does not work!
The database has both tables routes and newrates.
With routes I get: You have so much money, your call cost
With newrates I get: You have so much money left, I am sorry that is
not a recognized number
I created the newrates table via mysqldump, changed table name
everywhere and changed the rate, inserted the new table with mysql, ...
I tried to reload mysql, ...
Please, enlighten me!!!
bye
Ronald
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Re: [Asterisk-Users] zaptel PRI drivers
On Sun, 2005-03-20 at 23:12 -0700, Tom wrote: Anyway, I know this isn't a supported setup, so if thats your answer don't bother replying, I'm know this will be a kludge/hack to get working (if I can get it working at all). I'm just trying to do something that will be convienient for me and my users, there are other systems running on the server that I don't want to manage through the CLI, and the users don't know how to manage through the CLI, and there is no web management for them. Just run X on your workstation, then login to your asterisk box (ssh -X asterisk.box.com) and then run your X applications (xedit/etc) This way, X windows runs on your workstation, along with any graphics card stuff, but the X application (xedit) runs on your asterisk box. So just be sure you don't go and run firefox, or mplayer or something silly, and it should be OK Note, should == might... I haven't tested this, and really wouldn't suggest it, but, I hope it is a more helpful solution than simply Don't run X and learn to use the CLI for everything. In case you need it, there are X servers available for MS Windows platforms as well. Used to be one called exceed, but that was about 10 years ago, I just use linux on my desktop now instead :) Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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] Choosing an ISP for Asterisk
However, I don't know the specific requirements for the T1 line or how to split the data and voice. The point of VoIP is to consolidate data and voice onto one network. Combining both allows for economies of scale: * you don't have to use sangoma or digium card, this is the VoIP provider's task! * bandwith for voice + bandwith for data bandwith for (voice + data). * you have only one network to focus your efforts on. So assuming you can have high quality bandwith (guaranteed throughput, guaranteed uptime, good pings) - then just have an IP data pipe. Say, 2 Megs symetrical, contention ratio 1:1. This is to be used for voice and for internet. Then you just need 1 PSTN line for 911 emergencies. Then you'll need to deploy a solid bandwith shaping solution to guarantee enough bandwith for VoIP traffic at all times. An additional advantage of this setup is that should you have offices that are equiped the same way, then communications cost between those offices would drop to zero because they become purely VoIP - VoIP. Another benefit is that you're not limited to 24 channels but by your bandwith. Cheers, Jean-Michel. ___ 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] Log Error
I'm getting the same problem running Asterisk CVS-HEAD-03/21/05-03:24:01 built by [EMAIL PROTECTED] on a i686 running Linux which is the code from yesterday. Robert Goodyear wrote: FWIW I get the same exact error at the end of every VM session as well, thus: -- Playing 'vm-intro' (language 'en') -- Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/501/INBOX/msg format: wav49, 0x8186370 -- x=1, open writing: /var/spool/asterisk/voicemail/default/501/INBOX/msg format: gsm, 0x81634c8 -- x=2, open writing: /var/spool/asterisk/voicemail/default/501/INBOX/msg format: wav, 0x8186570 -- User hung up Mar 11 12:21:32 WARNING[22142]: app_queue.c:374 changethread: Can't change device '**Unknown**' with no technology! Mar 11 12:21:32 WARNING[22142]: app_queue.c:374 changethread: Can't change device '**Unknown**' with no technology! Mar 11 12:21:32 WARNING[22142]: app_queue.c:374 changethread: Can't change device '**Unknown**' with no technology! On Mar 4, 2005, at 7:00 PM, Anton Krall wrote: Guys, this error has been driving me nuts and I see no indication anywhere as to what it may mean. Anybody has any clues on this? -- User ended message by pressing # -- Playing 'auth-thankyou' (language 'en') -- Playing 'vm-review' (language 'en') -- Saving message as is -- Playing 'vm-msgsaved' (language 'en') Mar 4 21:02:06 WARNING[8816]: app_queue.c:374 changethread: Can't change device '**Unknown**' with no technology! ___ 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] DTMF doesn't seem to get through incoming ZAP channels
Hi, I'm running CVS-HEAD-03/19/05-11:15:15 on Fedora Core 3 with Digium TE410P card. Calling into meeting rooms that have been configured with the p option works fine. From ZAP extensions the # key does not work to exit, however from SIP extensions the # key works fine. This makes me believe that somehow the DTMF doesn't get through the ZAP interface. After furter experimenting voicemail also doesn't work through ZAP (the selection of menu-options that is...) So now I definately know that DTMF through ZAP doesn't work (anymore, it used to in the past). Is there any way I can troubleshoot this ? I have already set the relaxdtmf=yes option in /etc/asterisk/zapata.conf, which looks like this: [channels] context=default switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echotraining=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=2.0 immediate=no ; Channels inherit configuration above them ; Span 1 group=1 context=default signalling=pri_net; this is connected to voice switch channel = 1-15 channel = 17-31 Any suggestions and assistance would be very welcome. Thanks in advance Walter Klomp Singapore. ___ 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] Choosing an ISP for Asterisk
On Mon, 2005-03-21 at 12:36 +0400, Jean-Michel Hiver wrote: However, I don't know the specific requirements for the T1 line or how to split the data and voice. The point of VoIP is to consolidate data and voice onto one network. Combining both allows for economies of scale: [SNIP] Say, 2 Megs symetrical, contention ratio 1:1. This is to be used for voice and for internet. Then you just need 1 PSTN line for 911 emergencies. Ummm, in an office of x people, where x is some arbitrary integer 1, is 1 PSTN line for emergency services sufficient ?? Personally, I would think it isn't, but haven't quite determined what number is sufficient. Consider, an office of 50 people, and a small fire breaks out, how many people will call the emergency number? Substitute a small fire for any other emergency, but I suspect that most emergencies would generate multiple reports. Perhaps someone has some data on this? Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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] Choosing an ISP for Asterisk
Ummm, in an office of x people, where x is some arbitrary integer 1, is 1 PSTN line for emergency services sufficient ?? Personally, I would think it isn't, but haven't quite determined what number is sufficient. Consider, an office of 50 people, and a small fire breaks out, how many people will call the emergency number? And how many calls are needed? Is calling the fire station 50 times going to be more efficient? I don't think so... ___ 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] app_nv_backgrounddetect - how to make module
On Sun, 2005-03-20 at 23:40, Joseph wrote: How to compile additional module to asterisk? I have app_nv_backgrounddetect.c file and followed instructions below, but make did not generate app_nv_backgrounddetect.so or app_nv_backgrounddetect.o (1) Drop the code in your /usr/src/asterisk/apps directory (2) Edit the Makefile in the apps directory. Add the following line: APPS+=app_nv_backgrounddetect.so (3) Go to /usr/src/asterisk and run make, then run make install I've noticed that in .../apps directory every module has three files file_name.c file_name.o file_name.so How do I get the last two if I have the first one? When U have done first two steps U need just get back to /usr/src/asterisk/ and execute: make that U will have those .so files Everything is pretty clear described on wiki http://voip-info.org/tiki-index.php?page=NVBackgroundDetect ___ 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] Choosing an ISP for Asterisk
On Mon, 2005-03-21 at 13:51 +0400, Jean-Michel Hiver wrote: Ummm, in an office of x people, where x is some arbitrary integer 1, is 1 PSTN line for emergency services sufficient ?? Personally, I would think it isn't, but haven't quite determined what number is sufficient. Consider, an office of 50 people, and a small fire breaks out, how many people will call the emergency number? And how many calls are needed? Is calling the fire station 50 times going to be more efficient? I don't think so... Really?? What if you happen to be the 4th person calling, and need to inform them that you are trapped in the stationary cupboard, and luckily you were carrying the cordless handset (but not your mobile phone)?? At the end of the day, I wouldn't expect any office to have a 1:1 ratio between users:phone lines, but there must be some ratio which would make some sort of sense Also, it was a bit of a 'heads-up' for those people that thought they had 'solved' the problem by dropping a single phone line in, or plugging a 'red' phone into the fax line, or some-such So, any useful comments... please post them. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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] DTMF doesn't seem to get through incoming ZAP channels
On Mon, 21 Mar 2005, Walter Klomp wrote: I'm running CVS-HEAD-03/19/05-11:15:15 on Fedora Core 3 with Digium TE410P card. Calling into meeting rooms that have been configured with the p option works fine. From ZAP extensions the # key does not work to exit, however from SIP extensions the # key works fine. This makes me believe that somehow the DTMF doesn't get through the ZAP interface. After furter experimenting voicemail also doesn't work through ZAP (the selection of menu-options that is...) I tried CVS on 2005-03-18 and we found a similar problem with Dial with the transfer options enabled. The calling phone could transfer but the called phone could not. Identical phones etc, and the results were the same when the two endpoints were interchanged. We placed debug logging code at various places, including all the way down in zt_read in chan_zap. It seems that the dsp code got called, but did not detect digits on the outbound leg. Perhaps some state in the dsp code is not initialized properly for DTMF detection? If the called phone was a sip phone set to rfc2833 then transfers work. I ran out of time to test further and reverted to an older cvs release. (Much older). Peter ___ 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 and X [was: Re: [Asterisk-Users] zaptel PRI drivers]
On Sun, Mar 20, 2005 at 11:12:22PM -0700, Tom wrote: I have a quick question. I know that running X on an asterisk server is not officially supported, Generally it shouldn't cause errors, but will probably degregate performance, as an X server is probably as close as Asterisk is to the hardware and optimized just as well for minimal latency. however, I've never had any trouble with it until now (8 months, using wctdm cards with fxo and fxs ports, IAX trunks, SIP phones, everything except a PRI card). Now I just installed my first asterisk box that terminates a PRI, and bam, HDLC errors up the wazoo if X is running, if its not, everything is fine, I assume this is because the timing parameters for the PRI are so much more strick. Why do you need the X server running at all? Is Asterisk running as root? With real-time priority? (-p) What distro do you use, BTW? I don't mind if X is a little less responsive (even alot less responsive), but I would really like to be able to run X on a server with a PRI. Is there any way to reduce X11's priority so that it doesn't interfere with the zaptel driver for the PRI... I've tried renicing X as far down as I can and renicing Asterisk up as far as I can, however I fear this won't ever fix the problem since I think the actual kernel module that is running the pri card needs to get higher priority (ie, the kernel itself needs higher priority). What exactly do you run on X? Is the CPU very busy? try a light interface such as icewm, windowmaker or fluxbox with a theme that uses no gradients and no special effects. If your display has a little resolution, try something like matchbox. Is there any way to do this? Am I correct in my analysis? I really don't understand why on a system that averages less than 3% CPU usage with X running, why it can't handle the PRI. I know for whatever reason X always gets a really high priority (although top doesn't show X getting any special treatment its PR 15 NICE 0 by default, lower than most other processes on the system). Another idea is that right now the system is only a single proc, but it is dual proc capable. Would this somehow help if we added the second proc? My thinking is it won't because it's a kernel module we are dealing with, and because of that I can't control the affinity of the driver (I was thinking at one point put X11 on 1 proc and Asterisk on the second, but it's not Asterisk that has the problem I don't think.) My final idea is that currently the system has an onboard 8mb ati graphic card that leaves almost all actual graphics processing to the CPU, could adding a better graphics card possibly help X use less cpu and not get in the way so much? Anyway, I know this isn't a supported setup, so if thats your answer don't bother replying, I'm know this will be a kludge/hack to get working (if I can get it working at all). I'm just trying to do something that will be convienient for me and my users, there are other systems running on the server that I don't want to manage through the CLI, and the users don't know how to manage through the CLI, and there is no web management for them. You want to run a full desktop just be able to manage the Asterisk box? That's what ssh is for. Xorcom Rapid added a menu application for managing the box for those who don't know the command to type. If you have an X server on your workstation you can run X programs on your local X server. There should be no need for a local X server on the Asterisk box. Does anyone have success running X on an asterisk box that terminates a PRI? If so what hardware (video card, cpu, ram, mobo, etc)? Thanks as I know this setup isn't supported, and I'm probably asking alot, don't think I'm just relying on the list for bizarre things, I've been trying various ways of doing this for the last 3 weeks, I can successfully run a vnc server on the box (without X running) and everything works, so for whatever reason it is getting a lower priority or something. I really need to run GDM though as managing VNC passwords/usernames/desktop settings is quite cumbersome and if we can just get GDM running, we can use our ldap authentication server for logins to this box (which is what we were doing previously when we didn't have a PRI terminated on this box). VNC is a protocol for remotely controling a desktop. There are several ways of working with GDM. One useful way is to run a local XVnc server. This requires no GDM at all, unless you want a separate user and separate desktop for each real user (and waste tons of memory on that). Still, why waste all of those resources of your * box? -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend
[Asterisk-Users] Cdr_odbc asterisk 1.0.6
Asterisk Ready. *CLI -- Executing route(SIP/7408-02e3, 370263) in new stack -- odbcquery: query=370263 Query = 370263 : SQLcmd = select routing, ring_timer from ddi_pool where ddi_inbound = '370263' Urgent handler app_route: Query Successful! -- Varname= 55 -- odbcquery: set route 721017101 -- odbcquery: set timer 15 Urgent handler -- Executing Dial(SIP/7408-02e3, OH323/[EMAIL PROTECTED]|15) in new stack -- H.323 call to [EMAIL PROTECTED] with codec(s) g729 -- Called [EMAIL PROTECTED] Urgent handler -- OH323/L24286 is ringing Urgent handler Mar 21 10:39:59 WARNING[17072]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 101 (Non-critical Response) -- Nobody picked up in 15000 ms Urgent handler -- Hungup 'OH323/L24286' Urgent handler -- Executing Dial(SIP/7408-02e3, SIP/[EMAIL PROTECTED]|15) in new stack Urgent handler -- Called [EMAIL PROTECTED] Urgent handler Urgent handler -- SIP/192.168.1.252-da4c answered SIP/7408-02e3 -- Attempting native bridge of SIP/7408-02e3 and SIP/192.168.1.252-da4c Urgent handler -- H.323 call 'ip$localhost/24286' cleared, reason 1 (Cleared by local user) Urgent handler -- Executing Hangup(SIP/7408-02e3, ) in new stack -- Starting Query-- Sucessfully Setup Unique or Nonunique loggin -- Checked DB Connection-- Setup QueryUrgent handler Segmentation fault (core dumped) I'm having some trouble with cdr_odbc I have installed from 1.0.6 stable and it keeps seg faulting when trying to execute the cdr record. Does anyone have any idea or how do I go about debugging what exactly is going on and then fix it. Regards Sean Lowry ___ 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] noice sip to sip only???
i have been using the asterisk for some three weeks. Previously i was using the softphone iax-phone and now i have to shift to the sip phone xlite. The problem is that there's always unbearable noice in sip to sip calls. Is there any way to get rid of this Kindest MM Luqman ___ 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 call rejected.....who was trying to reach 's@'
dear All i signed up with an Aussie provider who gives me a DID in Aust... when I call my number I get the following on the console Mar 21 05:54:15 NOTICE[68071]: chan_iax2.c:6123 socket_read: Rejected connect at tempt from 203.13.163.245, who was trying to reach 's@' the s part i can understand by the @nothing ..?!? my iax.conf looks like this register = aa:[EMAIL PROTECTED] [alphanet] type=friend username= auth=plaintext ; ugh plaintext secret= host=proxy.freecall.net.au context=main disallow=all allow=ilbc my extensions.conf looks like this [default] exten = s,1,Answer exten = s,2,Dial(SIP/me,40,tr) also I have the same under [main] aswell anyone got any thoughts on this one its making me nuts Thanks ___ 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] Log Error
So far, nobody has been able to tell us what this error means. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MF Hulber Sent: Lunes, 21 de Marzo de 2005 02:54 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Log Error I'm getting the same problem running Asterisk CVS-HEAD-03/21/05-03:24:01 built by [EMAIL PROTECTED] on a i686 running Linux which is the code from yesterday. Robert Goodyear wrote: FWIW I get the same exact error at the end of every VM session as well, thus: -- Playing 'vm-intro' (language 'en') -- Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/501/INBOX/msg format: wav49, 0x8186370 -- x=1, open writing: /var/spool/asterisk/voicemail/default/501/INBOX/msg format: gsm, 0x81634c8 -- x=2, open writing: /var/spool/asterisk/voicemail/default/501/INBOX/msg format: wav, 0x8186570 -- User hung up Mar 11 12:21:32 WARNING[22142]: app_queue.c:374 changethread: Can't change device '**Unknown**' with no technology! Mar 11 12:21:32 WARNING[22142]: app_queue.c:374 changethread: Can't change device '**Unknown**' with no technology! Mar 11 12:21:32 WARNING[22142]: app_queue.c:374 changethread: Can't change device '**Unknown**' with no technology! On Mar 4, 2005, at 7:00 PM, Anton Krall wrote: Guys, this error has been driving me nuts and I see no indication anywhere as to what it may mean. Anybody has any clues on this? -- User ended message by pressing # -- Playing 'auth-thankyou' (language 'en') -- Playing 'vm-review' (language 'en') -- Saving message as is -- Playing 'vm-msgsaved' (language 'en') Mar 4 21:02:06 WARNING[8816]: app_queue.c:374 changethread: Can't change device '**Unknown**' with no technology! ___ 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] codec
My objective is to estimate the performances of * How much the trancoded can influence the performances? Thanks, Ale ___ 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] codec
* Alessandra Grasso ([EMAIL PROTECTED]) ha scritto: My objective is to estimate the performances of * How much the trancoded can influence the performances? take a look at translate.c file to see how transcoding costs are calculated. use the command 'show translation' at the CLI to see intercodecs costs. ___ 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] Choosing an ISP for Asterisk
Really?? What if you happen to be the 4th person calling, and need to inform them that you are trapped in the stationary cupboard, and luckily you were carrying the cordless handset (but not your mobile phone)?? At the end of the day, I wouldn't expect any office to have a 1:1 ratio between users:phone lines, but there must be some ratio which would make some sort of sense I don't know really. Say you put a TDM card with 4 FXO modules in, and you repeat same scenario as above? Hey, maybe a T1 with 24 channels isn't enough either. Maybe the fire / earthquake / godzilla will have destroyed your PBX system by the time you try to use the phone. Maybe everybody needs to have some kind of GSM + GPS device around their neck with a red button marked 'I'm in trouble' that would transmit their location to emergency services. Or maybe not. Maybe proper fire evacuation procedures with one emergency line is deemed enough. But then when it comes to safety, what is enough? Where do you draw the line? ___ 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: Re: [Asterisk-Users] Meetme2 compilation problem
Could you email login information anonymous login isnt allowed. Sean From: Asterisk [mailto:[EMAIL PROTECTED] Sent: 18 March 2005 16:20 To: Anil Kumar K; Giovanni Powell Cc: asterisk-users@lists.digium.com Subject: Re: Re: [Asterisk-Users] Meetme2 compilation problem Giovanni, on ftp://ftp.vinkconsult.com/downloads is a patched version of app_meetme2.c. I patched and compiled it against the CVS unstable from today Andre - Oorspronkelijk Bericht - Onderwerp:Re: [Asterisk-Users] Meetme2 compilation problem Afzender: Anil Kumar K [EMAIL PROTECTED] Aan:Giovanni Powell [EMAIL PROTECTED] CC:asterisk-users@lists.digium.com Datum:18-03-2005 16:56 I did the patch also . That didnt help me. I am using CVS head of 17th March . Googling didnt give me much info other than this patch. Thanks On Fri, 18 Mar 2005 10:18:26 -0500, Giovanni Powell [EMAIL PROTECTED] wrote: I'm sure there was a patch for meetme2 regarding compilation... google for meetme2 + patch. It worked for me. ___ 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] IAX call rejected.....who was trying to reach 's@'
* Jer ([EMAIL PROTECTED]) ha scritto: my iax.conf looks like this dunno if it may help, try adding context=default in your iax.conf 'general' section ___ 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] mpg123 home music from stream
Does anyone have an example for using a live mp3 shoutcast stream with mpg123 for hold music? ___ 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] Dial from a URL - Possible?
Is it possible to initiate/receive calls from a url (that is without having to install and configure a PC soft phone) using asterisk? If yes, may I please get some sites, pointers, HOWTOs on how its done? You can also try the Flash Operator Panel, http://www.asternic.org. It supports click-to-dial, screen pops, etc. -- Nicolás Gudiño Buenos Aires - Argentina ___ 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] features.conf
Hello list, i configured correctly the codes in features.conf, loaded successfully res_features, but while in a call (any type of call including zaptel to zaptel, zaptel to sip, sip to sip) both sides hear DTMFs and nothing happens... i'm i missing something? Thanks, Calin. ___ 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] mpg123 home music from stream
1. create a directory inside /var/lib/asterisk or whatever you have configured for that, i.e. /var/lib/asterisk/mohmp3-radio, then 2. create /var/lib/asterisk/mohmp3-radio/dummy.mp3 3. then add live =mp3:/var/lib/asterisk/mohmp3-radio,http://www.yourfavradio.com:port/ into your /etc/asterisk/musiconhold.conf or wherever it is. 4. change your MoH class to 'live' for this example and you're done. - Original Message - From: Matt [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 21, 2005 6:53 AM Subject: [Asterisk-Users] mpg123 home music from stream Does anyone have an example for using a live mp3 shoutcast stream with mpg123 for hold music? ___ 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] Why is asterisk's voice mail called comedian.
Hi list, Is this supposed to be a joke? It doesn't sound very professional. comedian n 1: a professional performer who tells jokes and performs comical acts Moby Thesaurus words for comedian: banana, buffoon, burlesquer, card, caricaturist, choreographer, clown, comedienne, comic, cutup, dramatist, dramatizer, dramaturge, droll, epigrammatist, farcer, farceur, farceuse, farcist, fool, ... ___ 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] Why is asterisk's voice mail called comedian.
There was a short discussion thread on this on / about 8/31/2004 but no real answer was ever given. Some people supposed that since a competing voicemail sounded the same (Meridian) that there might have been some correlation between the two but who knows. I'm sure you're more than welcome to hire the same voice talent and make the voice prompt say whatever you'd like it to say :P ___ 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 outbound to SIP provider problems (still)
Hi I am using cvs and updating it every couple of days Unfortunately I am still getting a 20 second timeout on sip calls placed to various providers, can anyone see anything wrong from sip debugs? Or have any ideas what the problem might be? Cheers Walt sip debug peer of a provider: http://www.walt.9k.com/sip/1_SIP_Provider.html sip debug peer of phone placing the call http://www.walt.9k.com/sip/1_cisco_phone.html The call goes like this: caller: dial caller: SIP code 100 destination: ring caller: 1-2 second delay caller: SIP code 183 (this is what it says on the cisco phone) caller: ring destination: pickup caller: 2 way audio ok destination: 2 way audio ok caller: Sip code 183 (Never 200 connected etc) caller: audio stops destination: chooses to hang up caller: chooses to hang up _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ 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] outbound delay
joerg hanke wrote: hi i wonder why my outbound calls via asterisk-sipgate-german telecom have such high delay rates (about 500 or mor ms) while inbound signals are quite ok (max ca 200ms). any idea? joerg ___ 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 Just a guess but when I saw this, it was down to network latency. ping sipgate from the asterisk box and see what you get Cheers 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: RE: Re: [Asterisk-Users] Meetme2 compilation problem
Try username=guest pass=emailaddr.Andre- Oorspronkelijk Bericht -Onderwerp:RE: Re: [Asterisk-Users] Meetme2 compilation problemAfzender: Sean Lowry [EMAIL PROTECTED]Aan:Asterisk [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comDatum:21-03-2005 13:12 Could you email login information anonymous login isnt allowed. Sean From: Asterisk [mailto:[EMAIL PROTECTED] Sent: 18 March 2005 16:20 To: Anil Kumar K; Giovanni Powell Cc: asterisk-users@lists.digium.com Subject: Re: Re: [Asterisk-Users] Meetme2 compilation problem Giovanni, on ftp://ftp.vinkconsult.com/downloads is a patched version of app_meetme2.c. I patched and compiled it against the CVS unstable from today Andre - Oorspronkelijk Bericht - Onderwerp:Re: [Asterisk-Users] Meetme2 compilation problem Afzender: Anil Kumar K [EMAIL PROTECTED] Aan:Giovanni Powell [EMAIL PROTECTED] CC:asterisk-users@lists.digium.com Datum:18-03-2005 16:56 I did the patch also . That didnt help me. I am using CVS head of 17th March . Googling didnt give me much info other than this patch. Thanks On Fri, 18 Mar 2005 10:18:26 -0500, Giovanni Powell [EMAIL PROTECTED] wrote: I'm sure there was a patch for meetme2 regarding compilation... google for meetme2 + patch. It worked for me. ___ 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] mpg123 home music from stream
Where is my MoH class? I understand what's being done here... but I don't see where that is.. like for meetme conferences, and being placed on hold and such... which file? On Mon, 21 Mar 2005 07:13:16 -0600, Henry Devito [EMAIL PROTECTED] wrote: 1. create a directory inside /var/lib/asterisk or whatever you have configured for that, i.e. /var/lib/asterisk/mohmp3-radio, then 2. create /var/lib/asterisk/mohmp3-radio/dummy.mp3 3. then add live =mp3:/var/lib/asterisk/mohmp3-radio,http://www.yourfavradio.com:port/ into your /etc/asterisk/musiconhold.conf or wherever it is. 4. change your MoH class to 'live' for this example and you're done. - Original Message - From: Matt [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 21, 2005 6:53 AM Subject: [Asterisk-Users] mpg123 home music from stream Does anyone have an example for using a live mp3 shoutcast stream with mpg123 for hold music? ___ 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] OT: No authority found connecting to Freshtel
Hi, Has anyone else experienced problems as of the last couple of months when outbound calling through Freshtel? I've started getting a No authority found error. I've tried contacting them, and they seem to have some serious communication issues with their IT team, infact I think they have serious issues in their IT team full stop. First they can't find my account in their DB, and I keep being promised they will look into my problem and get back to me, and of course they never do so I had no choice but to ask the list just incase it's something on my end, which I seriously doubt as I've triple checked everything. Thanks in advance for any help. Gonzalo ___ 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] Version 0.67 of IPSwitchBoard Released
IPSwitchBoard Version 0.67 Release notes: CRM integration, can call a web page with callerid when there's an incoming call. You can specify the min. and max. length of the callerid. Drop any active call. Help file integrated in IPSwitchBoard. Play button for sound files. Bug fixes - thank you for all your feedback. Download IPSwitchBoard for FREE here: http://www.voip-info.org/tiki-index.php?page=IPSwitchBoard+BETA ___ 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] Last guy to get BV working outbound?
Rich thanks, this makes it a little clearer. My servers are using NAT
behind a Cisco PIX. I only needed the simple patch (see below).
I configured sip.conf from these instructions:
http://www.voip-info.org/wiki-Asterisk+settings+Broadvoice
Hope this helps somebody. Sorry I wasn't clear about using NAT.
Brian
Patch I used:
--- chan_sip.c.fcs 2003-12-13 14:54:37.0 -0800
+++ chan_sip.c 2005-03-10 11:48:40.0 -0800
@@ -,10 +4446,10 @@
}
static int do_proxy_auth(struct sip_pvt *p, struct sip_request *req,
char *header, char *respheader, char *msg, int init) {
- char digest[256];
+ char digest[1024];
p-authtries++;
memset(digest,0,sizeof(digest));
- if (reply_digest(p,req, Proxy-Authenticate, msg, digest,
sizeof(digest) )) {
+ if (reply_digest(p,req, header, msg, digest, sizeof(digest) )) {
/* No way to authenticate */
return -1;
}
On Sat, 2005-03-19 at 09:14, Rich Adamson wrote:
A lot of the BV config confusion is the result of users with registered
IP's vs nat'ed IPs. The patch _was_ only required for those that used
nat'ed systems (proven shortly after that patch was released, and backed
by those that wrote the patch).
So, for those that are still mucking around with BV configs, it would
be helpful to others on this list to understand whether your systems are
nat'ed or not in initial posts.
You can also help yourself by validating some of these recommended
parameters against those listed in /usr/src/asterisk/configs/sip.conf.samples.
(User=phone is one such example of a do-nothing statement that has
no meaning whatsoever.)
Since I no longer subscribe to BV's service, I don't have a clue
which * releases need the patch and which don't.
Thanks John, but I tried adding those and many others. Turned out that
I needed to install a patch even though I tried CVS-3/11/05 and
CVS-3/17/05 code. I'm not sure what release needs what patch to work
but I definitely needed a patch. Thanks to the person on this list who
sent it along. There are many people with many configs posting on many
lists but I can't say I have a handle it.
Brian
On Fri, 2005-03-18 at 12:30, John Sawa wrote:
Brian,
You will need to add the following to your broadvoice peer:
user=phone
insecure=very
dtmf=inband
For more info check out:
http://geekgazette.com/index.php?option=com_contenttask=viewid=20Itemid=26
Hope this helps. -john
Brian G wrote:
I have tried everything to get BV working outbound. All worked fine
until the BV change last week. I called BV and they changed me to sip
gen with a new password. I stripped my Asterisk server to one phone on
Zap/1 until I get this working. The same BV account works fine with a
SPA-3000 so I don't suspect a firewall problem.
Symptoms: Asterisk registers with BV Ok
Incoming calls work
Outbound calls send Invite, receive 100, then 401
Asterisk sends an ACK instead of another Invite with credentials
If anyone knows what specifically makes Asterisk respond to the 401 with
credentials for an authenticated Invite, I'd appreciate it. I can't
seem to find this out.
Thanks in advance,
Brian
Here is my sip.conf:
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind SIP channel to
context = default ; Default context for incoming calls
srvlookup = yes ; Enable DNS SRV lookups on outbound
calls
disallow=all; Disallow all codecs
allow=ulaw ; Allow codecs in order of preference
;
; Configuration for BroadVoice
;
register =
[EMAIL PROTECTED]:pword:[EMAIL PROTECTED]
;
[broadvoice]
type=peer
host=sip.broadvoice.com
secret=pword
fromuser=508XXX
username=508XXX
authuser=508XXX
fromdomain=sip.broadvoice.com
context=incoming
canreinvite=no
dtmfmode=inband
qualify=yes
in extensions.conf:
[default]
exten = _81XX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _81XX,2,Congestion()
exten = _81XX,102,busy()
Other Asterisk info:
*CLI sip show registry
Host Username Refresh State
147.135.0.128:5060508XXX 120 Registered
*CLI
*CLI show version
Asterisk CVS-03/11/05-16:07:49 built by [EMAIL PROTECTED] on a i686
running Linux
*CLI
*CLI Mar 17 10:35:08 NOTICE[-245486672]: chan_sip.c:5047
handle_response: Failed to authenticate on INVITE to 'Analog1
sip:[EMAIL PROTECTED];tag=as212bf17
[Asterisk-Users] CallerID Name with IAX Providers
I am pretty sure that there are no IAX providers that offer CallerID name but wanted to double check with the list in case something has changed recently. Is anyone aware of an IAX provider that offers incoming CallerID name? Is there a technical limitation within IAX which is preventing IAX providers from offering CallerID Name? Why is no one offering this? ___ 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] CallerID Name with IAX Providers
I am pretty sure that there are no IAX providers that offer CallerID name but wanted to double check with the list in case something has changed recently. Is anyone aware of an IAX provider that offers incoming CallerID name? Xetricom Networks, who only have Toronto DIDs, do provide incoming CallerID name using IAX. Nabeel ___ 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] VoIP service through Asterisk?
Peter Loron wrote: Greetings. I did some digging with Google, the wiki, and on the archives, but didn't find a recent conclusive answer. If this is answered in the wiki or archives somewhere, please point me to it. I'm in the process of setting up an Asterisk box for home use. I've got a X100P card on the way. I've not decided what analog adapter(s) to get yet. The only phone service to hook up is currently POTS. I'm interested in integrating a VoIP provider into the system (using it as a service for inbound and outbound calls). I understand that I can use Broadvoice (BYOD plan), however I'm also considering other providers. Other than Broadvoice, are there any VoIP providers (Vonage, Packet8, etc) that can be hooked into Asterisk directly? I read about a scheme for Packet8 that involved routing it in through an analog connection on a FXO port...I'd rather have something I can connect in directly. Thanks! -Pete Hi Pete, I use the Voicepulse Connect! service, and I don't have any issues with it. It *is* a bit pricy ( ~3 cents a minute, 7.99 a month for an incoming number ), but I get great voice quality, and I have yet to have an instance where I *can't* dial a number. However, for reference, excluding the incoming number charge, I think I've paid 6 bucks over the past three months in call charges. Your milage will vary of course, but one thing to keep in mind: You don't pay for 1-8xx numbers. So for a business, this would be an awsome plan. And yes, they have direct iax connections. Given my relative noob status, I wouldn't bother with anything else. :) Sean ps- I don't know if the other services let you do this, but voicepulse lets you set your own callerid. Which is, for me, a deal breaker. ___ 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] CallerID Name with IAX Providers
Nabeel Jafferali wrote: Xetricom Networks, who only have Toronto DIDs, do provide incoming CallerID name using IAX. We also provide Calling Name delivery for our DIDs. It's not an issue of the VOIP protocol in use (it can be done over SIP or IAX), it's an issue of what sort of PSTN connectivity the provider is using and whether they want to incur the cost of Calling Name lookups or not. ___ 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] Net2Phone / Vonage
I guess I should supply my current sip.conf file for net2phone. [general] ;useragent = X-Lite release 1103m useragent = Cisco ATA186 register = accountum:pin@sip.net2phone.com [net2phone] type = peer host = sip.net2phone.com username = accountnum secret = pin fromuser = accountnum fromdomain = net2phone.com insecure = very canreinvite = yes context = Home Anyone who is using n2p at all? Thanks Russell Handorf wrote: Greetings all, I've got Net2Phone and Vonage pitting against each other right now. At the moment, with the Vonage's Softphone account, I can only make incoming or outgoing phone calls (one config for incoming, one for out); in otherwords I cant seem to have one sip.conf file that will allow asterisk to recieve incoming and outgoing calls without rewriting sip.conf and restarting asterisk to allow for an incoming or outgoing call instance. With Net2Phone, I cannot get inbound calls to work (goes to the net2phone voicemail), and outgoing calls report This is an invalid account. I've followed the two configs based on what is reported in voip-info.org, googled the heck out of either configuration options, and have been trying to get either one to allow both incoming and outgoing calls off of one sip.conf and extensions.conf file. Does anyone have and up to date sip.conf file, or and tips or tricks to get this to work? Thanks all ___ 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] Some IAX questions
On Sat, Mar 19, 2005 at 09:47:49PM -0700, Tim Pushor wrote: Hi, Is this a silly question? I am trying to come up with an elegant way of joining a few small * servers in a peer to peer arrangement, and I am just curious as to what algorithm * uses to determine which channel (and therefore context) an inbound call belongs to (IAX and SIP).. Also, knowing when name resolution happens would be beneficial if the peer * boxes had dynamic IP's and dynamic dns ... I googled for Asterisk iax authentication which returned, amongst others: http://www.voip-info.org/wiki-Asterisk+IAX+authentication It should tell you all you need to know... -- 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] H323 gateway thru NAT
Hi all, I am wondering if chan_oh323 or chan_h323 supports NAT traversal the following setup: H323 phone - Asterisk --- NAT router - H323 gateway - PSTN I am trying to register a H323 gateway through a NAT to Asterisk for outgoing calls to PSTN. How can I achieve the above? Please help and advise. Many Thanks. Newbie ___ 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] CallerID Name with IAX Providers
More of a case that in many cases the voip carrier would have to do lookups for CNAM from either their telco or an external CNAM service. These tend to carry an extra cost so that's why it's not wide spread on dids via VOIP. -- William ___ 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] H323 gateway thru NAT
This is possible. But success depends also on whether the router can do port forwarding and whether the H323 Gateway supports NAT. This is possible with Quintum for instance with some port forwarding rules on router level. Selon VoIP Newbie [EMAIL PROTECTED]: Hi all, I am wondering if chan_oh323 or chan_h323 supports NAT traversal the following setup: H323 phone - Asterisk --- NAT router - H323 gateway - PSTN I am trying to register a H323 gateway through a NAT to Asterisk for outgoing calls to PSTN. How can I achieve the above? Please help and advise. Many Thanks. Newbie ___ 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] Soekris net4801 and analog interface?
I have 2 X100p clones that do not work in the net4801. The 4801 will not even power up with them installed. Both cards work fine a a standard desktop PC. On Sun, 2005-03-20 at 21:58 -1000, John Breeden wrote: X100P is 3.3v not 5v, at least the one I have. Works fine in a 4801. John Simon wrote: Is anyone using a net4801 and an analog only setup? I am looking for a modem that is PCI 3.3V, apparently the X100P is 5.0V PCI only so it won't work with the net4801. ___ 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] Follow-Me Script
It never dials the other number and instead goes straight into voicemail.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Sunday, March 20, 2005 8:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Follow-Me Script
Which part is not working?
On Sun, 20 Mar 2005 16:36:08 -0800, Kerry Garrison [EMAIL PROTECTED]
wrote:
I am trying to implement a follow-me script
(http://www.voip-info.org/wiki-Asterisk+Tips+follow+me) but I am
having a brain fart as I haven't a clue where to get started with what
to do with this. From my main menu, I want the extension 300 to
execute the script as
follows:
exten = 300,1,dial(sip/200,20)
exten = 300,2,playback(pls-wait-connect-call)
exten = 300,3,Setvar(NewCaller=${CALLERIDNUM})
exten = 300,4,SetCIDNum(0${CALLERIDNUM}) exten =
300,5,dial(${TRUNK}c/2831385,20,r)
exten = 300,6,SetCIDNum(${NewCaller}) exten =
300,7,voicemail2([EMAIL PROTECTED]) exten =
300,101,voicemail2([EMAIL PROTECTED]) exten = 300,102,hangup
Regardless of what (and where) I have tried to implement this, I just
cant get it to work properly. Does anyone have some tips on this or a
nicer follow-me type of script?
Kerry Garrison
http://www.geekgazette.com
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[Asterisk-Users] astcc sip
Hello, Can anyone point me to any documentation with regards to using sip_friends on astcc. astcc already working on our test * server but im trying to figure out how to sql-ize sip user config. I have thought of using Asterisk Realtime but is not yet available on stable release. Appreciate any pointers on this subject. Thanks! -- Cheers, Paul P. Pongco Mosaic Communications Inc. ___ 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] US pstn = voip
Hi I believe this is due to the way US phone systems work, however I'm going to ask anyway. In the UK there are several providers who provide national rate PSTN = Voip gateways which are free to receive calls on, (for the recipient), the caller pays the cost of calling. E.g 0844 0870 etc. I am looking for a US provier who offers the same sort of system. I don't call the US but I have people in the US who call me. I wanted to set up a number where they could call, which would route to my * box, preferable via IAX or failing that SIP, (I haven't manged to get sip to work with DTMF yet, hence my preference). However I can only find providers of local numbers, or toll free numbers, both of which incur an inbound call rate of between 2c and 10c / minute, plus a monthly charge. Does anyone know of a provider who provides such a service either in US or Canada? I have googled for days on this one, and have come up with next to nothing. Many Thanks Mark ___ 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] Choosing an ISP for Asterisk
Personal opinion here, but in an office of 50 users we just built, here is what we did. First, the PBX was equipped with 4 analog lines that were setup as a failover in case the T1 for voice data failed. Secondly, another 4 analog lines were dropped in a central location of the office with analog handsets stored in a cabinet close by in case the entire network was down, as well as the two fax machine lines both equipped with handsets. Now if I was still really concerned about 911 calls after all this, I would issue a memo telling everyone that in case of emergency, use your cell phone to dial 911 as out of 50 people, all but I think 4 people carry cell phones there. As a sidenote to all of this, with 50 people, it was not practical to combine both voice and data over the same line. Doing the math simply didn't add up. We calculated a maximum usage at any one time being around 20 voice calls. This justified bringing in a voice T1 into a digital line card. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Goryachev Sent: Monday, March 21, 2005 2:27 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Choosing an ISP for Asterisk On Mon, 2005-03-21 at 13:51 +0400, Jean-Michel Hiver wrote: Ummm, in an office of x people, where x is some arbitrary integer 1, is 1 PSTN line for emergency services sufficient ?? Personally, I would think it isn't, but haven't quite determined what number is sufficient. Consider, an office of 50 people, and a small fire breaks out, how many people will call the emergency number? And how many calls are needed? Is calling the fire station 50 times going to be more efficient? I don't think so... Really?? What if you happen to be the 4th person calling, and need to inform them that you are trapped in the stationary cupboard, and luckily you were carrying the cordless handset (but not your mobile phone)?? At the end of the day, I wouldn't expect any office to have a 1:1 ratio between users:phone lines, but there must be some ratio which would make some sort of sense Also, it was a bit of a 'heads-up' for those people that thought they had 'solved' the problem by dropping a single phone line in, or plugging a 'red' phone into the fax line, or some-such So, any useful comments... please post them. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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] Version 0.67 of IPSwitchBoard Released
Thorben, Please check the behaviour of a Park button. If you do a vertical resize of a window (application) Park button gets dislocated. Ivan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Thorben Jensen Sent: Monday, March 21, 2005 3:05 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Version 0.67 of IPSwitchBoard Released IPSwitchBoard Version 0.67 Release notes: CRM integration, can call a web page with callerid when there's an incoming call. You can specify the min. and max. length of the callerid. Drop any active call. Help file integrated in IPSwitchBoard. Play button for sound files. Bug fixes - thank you for all your feedback. Download IPSwitchBoard for FREE here: http://www.voip-info.org/tiki-index.php?page=IPSwitchBoard+BETA ___ 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] mpg123 home music from stream
Matt wrote: Where is my MoH class? I understand what's being done here... but I don't see where that is.. like for meetme conferences, and being placed on hold and such... which file? musiconhold.conf But once you get it going, it doesn't work anyway. Would love to have someone prove me wrong. Asterisk stops MOH (closes the stream) when channel hangs up. This is great for all other MOH uses, but drops the mp3 stream and doesn't reconnect to streaming sever. (as noted in original patch/bug #413) I was streaming XM radio thru MOH via shoutcast. Unless someones fix this problem. ___ 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] US pstn = voip
voicepulse? We get free inbound on them. However, every once in a while the service degrades for quite some time and they blame it on their upstream provider; the issue just goes away without any real resolution. Mark Charlton wrote: Hi I believe this is due to the way US phone systems work, however I'm going to ask anyway. In the UK there are several providers who provide national rate PSTN = Voip gateways which are free to receive calls on, (for the recipient), the caller pays the cost of calling. E.g 0844 0870 etc. I am looking for a US provier who offers the same sort of system. I don't call the US but I have people in the US who call me. I wanted to set up a number where they could call, which would route to my * box, preferable via IAX or failing that SIP, (I haven't manged to get sip to work with DTMF yet, hence my preference). However I can only find providers of local numbers, or toll free numbers, both of which incur an inbound call rate of between 2c and 10c / minute, plus a monthly charge. Does anyone know of a provider who provides such a service either in US or Canada? I have googled for days on this one, and have come up with next to nothing. Many Thanks Mark ___ 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] CallerID Name with IAX Providers
You can roll your own -- for US numbers, my cid_rewrite agi-script does this nicely: http://muware.com/asterisk -Original Message- From: Keith O'Brien [mailto:[EMAIL PROTECTED] Sent: Monday, March 21, 2005 8:29 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] CallerID Name with IAX Providers I am pretty sure that there are no IAX providers that offer CallerID name but wanted to double check with the list in case something has changed recently. Is anyone aware of an IAX provider that offers incoming CallerID name? Is there a technical limitation within IAX which is preventing IAX providers from offering CallerID Name? Why is no one offering this? ___ 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] Why is asterisk's voice mail called comedian.
Is this supposed to be a joke? Probably. It doesn't sound very professional. Then change it -- all you need to do is re-record the greeting, or have Allison do it. ___ 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] US pstn = voip
Someone may correct me, but in the US, the concept of Calling Party Pays went away with early cell phones. Many companies opted for the Calling Parry Pays plans so their clients could call them at will. However, callers were unaware of the charge which created quite a fuss until the cell providers stopped offering that service. I don't see why you couldn't get an account through a VOIP provider in the US for a decent price. Not meaning to start another flame war here, but I am the most familiar with BroadVoice so I will use that as an example. If you are not calling the US, then you don't need to worry about where the phone number is terminated and you could go with the $9.95 unlimited in-state plan since that has an unlimited number of inbound minutes. Setup the inbound trunk to BV and in a few minutes you have a working US number that doesn't cost your callers anything and only sets you bac $10 a month. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Charlton Sent: Monday, March 21, 2005 7:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] US pstn = voip Hi I believe this is due to the way US phone systems work, however I'm going to ask anyway. In the UK there are several providers who provide national rate PSTN = Voip gateways which are free to receive calls on, (for the recipient), the caller pays the cost of calling. E.g 0844 0870 etc. I am looking for a US provier who offers the same sort of system. I don't call the US but I have people in the US who call me. I wanted to set up a number where they could call, which would route to my * box, preferable via IAX or failing that SIP, (I haven't manged to get sip to work with DTMF yet, hence my preference). However I can only find providers of local numbers, or toll free numbers, both of which incur an inbound call rate of between 2c and 10c / minute, plus a monthly charge. Does anyone know of a provider who provides such a service either in US or Canada? I have googled for days on this one, and have come up with next to nothing. Many Thanks Mark ___ 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] US pstn = voip
Get a local number for $3-$10/month with unlimited incoming minutes, and the caller will surely pay the cost of the call (unless they're local). If you have fewer than 100 minutes, go with a metered DID and pay $1-$2/month plus around 1c/minute. Or go with Stanaphone or IPKall and get a free number (with the provision that neither of these may be forever). -Original Message- From: Mark Charlton [mailto:[EMAIL PROTECTED] Sent: Monday, March 21, 2005 9:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] US pstn = voip Hi I believe this is due to the way US phone systems work, however I'm going to ask anyway. In the UK there are several providers who provide national rate PSTN = Voip gateways which are free to receive calls on, (for the recipient), the caller pays the cost of calling. E.g 0844 0870 etc. I am looking for a US provier who offers the same sort of system. I don't call the US but I have people in the US who call me. I wanted to set up a number where they could call, which would route to my * box, preferable via IAX or failing that SIP, (I haven't manged to get sip to work with DTMF yet, hence my preference). However I can only find providers of local numbers, or toll free numbers, both of which incur an inbound call rate of between 2c and 10c / minute, plus a monthly charge. Does anyone know of a provider who provides such a service either in US or Canada? I have googled for days on this one, and have come up with next to nothing. Many Thanks Mark ___ 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] Soekris net4801 and analog interface?
This is the same experiance I had with my net4801 and X100P. Do you know of any 3.3V PCI modems that will work with Asterisk? --- Matt Ryanczak [EMAIL PROTECTED] wrote: I have 2 X100p clones that do not work in the net4801. The 4801 will not even power up with them installed. Both cards work fine a a standard desktop PC. On Sun, 2005-03-20 at 21:58 -1000, John Breeden wrote: X100P is 3.3v not 5v, at least the one I have. Works fine in a 4801. John Simon wrote: Is anyone using a net4801 and an analog only setup? I am looking for a modem that is PCI 3.3V, apparently the X100P is 5.0V PCI only so it won't work with the net4801. ___ 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] Soekris net4801 and analog interface?
The intel v90 56k pci modem, with the MD3200 chipset. I'm using it. John Simon wrote: This is the same experiance I had with my net4801 and X100P. Do you know of any 3.3V PCI modems that will work with Asterisk? --- Matt Ryanczak [EMAIL PROTECTED] wrote: I have 2 X100p clones that do not work in the net4801. The 4801 will not even power up with them installed. Both cards work fine a a standard desktop PC. On Sun, 2005-03-20 at 21:58 -1000, John Breeden wrote: X100P is 3.3v not 5v, at least the one I have. Works fine in a 4801. John Simon wrote: Is anyone using a net4801 and an analog only setup? I am looking for a modem that is PCI 3.3V, apparently the X100P is 5.0V PCI only so it won't work with the net4801. ___ 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 ___ 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] Why is asterisk's voice mail called comedian.
Its a take-off from Nortel's 'Meridian Mail'. Personally, I think its very funny, and its only your users who hear it, outside callers don't hear anything except the greetings you record. Jay Milk ([EMAIL PROTECTED]) wrote: Is this supposed to be a joke? Probably. It doesn't sound very professional. Then change it -- all you need to do is re-record the greeting, or have Allison do it. ___ 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 -- Joe Dennick [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] zaptel PRI drivers
Quoting Roger Gulbranson [EMAIL PROTECTED]: On Mon, 2005-03-21 at 19:03 +1100, Adam Goryachev wrote: In case you need it, there are X servers available for MS Windows platforms as well. Used to be one called exceed, but that was about 10 years ago, I just use linux on my desktop now instead :) CygWin (http://www.cygwin.com/) has X server support. Is it greatly improved in the last 6 months? We've tried the cygwin x server previously to no avail, the apps we need to run wouldn't run on cygwin's x. ___ 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] G726-16 passthrough...
Brian McCrary wrote: Hello, I'm wondering if anyone has benn able to successfully get g726-16 passthrouhg to work? I am wanting to use this codec instead of g729 as I'm running out of DSPs using a high complexity codec on the Ciscos. I would think it would work just as g729 does, which has been working fine for me, but it does not. G726-32 does work great however, but it's like Asterisk doesn't recognize the payload tpyes for G726-16. Asterisk does not support G726-16. It only supports G726-32. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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-h323 and h323_id
Hi all Has anyone managed to send an outgoing call using asterisk-h323 and successfully sent the H323_id ? Sam ___ 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] mpg123 home music from stream
Hi Ken, This has worked fine for me for about 6 months, maybe I just didn't notice a problem. As far as I know there has been music playing when people are being put on hold every time. - Original Message - From: Ken Godee [EMAIL PROTECTED] To: Matt [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 21, 2005 9:17 AM Subject: Re: [Asterisk-Users] mpg123 home music from stream Matt wrote: Where is my MoH class? I understand what's being done here... but I don't see where that is.. like for meetme conferences, and being placed on hold and such... which file? musiconhold.conf But once you get it going, it doesn't work anyway. Would love to have someone prove me wrong. Asterisk stops MOH (closes the stream) when channel hangs up. This is great for all other MOH uses, but drops the mp3 stream and doesn't reconnect to streaming sever. (as noted in original patch/bug #413) I was streaming XM radio thru MOH via shoutcast. Unless someones fix this problem. ___ 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] Follow-Me Script
On Sun, 20 Mar 2005 16:36:08 -0800, Kerry Garrison [EMAIL PROTECTED] wrote: I am trying to implement a follow-me script (http://www.voip-info.org/wiki-Asterisk+Tips+follow+me) but I am having a brain fart as I haven't a clue where to get started with what to do with this. Kerry, I'm more of a fan of anthm's patch that does this. You need to be running CVS-Head to get it though. http://bugs.digium.com/bug_view_page.php?bug_id=0002905 -Chuji ___ 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] Why isasterisk's voice mail calledcomedian.
-Original Message-
From: Mark Charlton [mailto:[EMAIL PROTECTED]
Plus if you send your users to VoicemailMain(${CALLERIDNUM})
they don't hear
it at all.
They just get enter password.
Yup. If you do that, the only time they hear it is during the initial setup
call (if you have forcename=yes or forcegreetings=yes set.)
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Re: Asterisk and X [was: Re: [Asterisk-Users] zaptel PRI drivers]
On Mon, 2005-03-21 at 08:57 -0700, Tom wrote: We don't want to have to spend an extra 3 grand for another server just to take up more space when we have this box that is sitting here idle 99% of the time, and as it has worked spectacularly well with the wctdm cards, I don't see why it can't with the wcte110p/PRI. The wctdm only has to transfer data for 4 channels. The wcte110p has to do 24 (23+1). Your probability of having problems just went up by a factor of 6. ___ 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] Why isasterisk's voice mail calledcomedian.
Yep :)
Use a grandstream and [EMAIL PROTECTED] and you only need to push a single
button and go straight through to messages.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Charlton
Sent: Monday, March 21, 2005 11:18 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Why isasterisk's voice mail
calledcomedian.
Plus if you send your users to VoicemailMain(${CALLERIDNUM}) they don't
hear
it at all.
They just get enter password.
My 2c
Mark
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe
Dennick
Sent: 21 March 2005 16:05
To: Asterisk Users Mailing List -Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Why isasterisk's voice mail
calledcomedian.
Its a take-off from Nortel's 'Meridian Mail'. Personally, I think its
very
funny, and its only your users who hear it, outside callers don't hear
anything except the greetings you record.
Jay Milk ([EMAIL PROTECTED]) wrote:
Is this supposed to be a joke?
Probably.
It doesn't sound very professional.
Then change it -- all you need to do is re-record the greeting, or
have Allison do it.
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--
Joe Dennick
[EMAIL PROTECTED]
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Re: [Asterisk-Users] zaptel PRI drivers
On Monday 21 March 2005 16:09, Tom wrote: Quoting Roger Gulbranson [EMAIL PROTECTED]: On Mon, 2005-03-21 at 19:03 +1100, Adam Goryachev wrote: In case you need it, there are X servers available for MS Windows platforms as well. Used to be one called exceed, but that was about 10 years ago, I just use linux on my desktop now instead :) CygWin (http://www.cygwin.com/) has X server support. Is it greatly improved in the last 6 months? We've tried the cygwin x server previously to no avail, the apps we need to run wouldn't run on cygwin's x. [... Please delete old signatures ...] The only apps I've seen fail to run are those which require DisplayPostscript. B ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why isasterisk's voice mail calledcomedian.
David Brodbeck wrote:
-Original Message-
From: Mark Charlton [mailto:[EMAIL PROTECTED]
Plus if you send your users to VoicemailMain(${CALLERIDNUM})
they don't hear
it at all.
They just get enter password.
Yup. If you do that, the only time they hear it is during the initial setup
call (if you have forcename=yes or forcegreetings=yes set.)
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Thanks for everyone's responses.
I just thought it should maybe say asterisk mail or digium mail.
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Re: [Asterisk-Users] Some IAX questions
Argh. I should have known better. Sorry, Tim Martijn van Oosterhout wrote: On Sat, Mar 19, 2005 at 09:47:49PM -0700, Tim Pushor wrote: Hi, Is this a silly question? I am trying to come up with an elegant way of joining a few small * servers in a peer to peer arrangement, and I am just curious as to what algorithm * uses to determine which channel (and therefore context) an inbound call belongs to (IAX and SIP).. Also, knowing when name resolution happens would be beneficial if the peer * boxes had dynamic IP's and dynamic dns ... I googled for Asterisk iax authentication which returned, amongst others: http://www.voip-info.org/wiki-Asterisk+IAX+authentication It should tell you all you need to know... ___ 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] iLBC codec and mute issues
I tried using the iLBC codec, and whlie I like it, I ran into a strange issue. I did a few searches on Google and haven't found anyone with the same issue as this. Anyhow, I was using a Plantronics analog headset and box plugged into a Digium TDM card, dialed out over my VoIP provider's IAX channel to the PSTN. I was in a conference call which is running on an Avaya PBX (which shouldn't matter), and so I muted myself with the mute button on the headset box. After a minute or two, I was asked to speak again, and so I unmuted, but no one could hear me. I tried hitting mute a bunch more times, but still nothing. It was making a difference in the headset though; I could hear myself a little bit when unmuted, but not when I was muted, leading me to believe it was something with the Asterisk box. I switched codecs and the issue disappeared. Does anyone know what the problem is there, seemingly it is iLBC, but I was wondering if that's a common thing or not. Is this codec unstable like this? Thanks, Dana ___ 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 get message on hold class to work
I can't seem to get the message on hold class to work for anything but default.. it works if I specify default but if I specify anything else it hangs up on me: == Spawn extension (from-internal, 9472, 3) exited non-zero on 'SIP/200-9f2c' -- Executing Macro(SIP/200-9f2c, hangupcall) in new stack -- Executing ResetCDR(SIP/200-9f2c, w) in new stack -- Executing NoCDR(SIP/200-9f2c, ) in new stack -- Executing Wait(SIP/200-9f2c, 5) in new stack -- Executing Hangup(SIP/200-9f2c, ) in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/200-9f2c' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/200-9f2c' [classes] ;default = quietmp3:/var/lib/asterisk/mohmp3 default = mp3:/var/lib/asterisk/mohmp3-radio,http://wgrc.swift-networks.com:8000/ wgrc = mp3:/var/lib/asterisk/mohmp3-radio,http://wgrc.swift-networks.com:8000/ ;loud = mp3:/var/lib/asterisk/mohmp3 ;random = quietmp3:/var/lib/asterisk/mohmp3,-z exten = 9472,1,Answer exten = 9472,2,SetMusicOnHold(wgrc) exten = 9472,3,MusicOnHold() That hangs up.. if I change wgrc to 'default' then it works... but I don't want default specified there.. any thoughts? (yes I know default and wgrc are the same at the moment.. I wanted to make sure my streaming syntax was correct). ___ 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] H323 gateway thru NAT
Thanks. Is there any native solution that is also cheap? I need it for my small office with only a few staff. My H323 gateway is not even a cisco one but costs only $200. Thanks. On Mon, 21 Mar 2005 15:57:13 +0100, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: This is possible. But success depends also on whether the router can do port forwarding and whether the H323 Gateway supports NAT. This is possible with Quintum for instance with some port forwarding rules on router level. Selon VoIP Newbie [EMAIL PROTECTED]: Hi all, I am wondering if chan_oh323 or chan_h323 supports NAT traversal the following setup: H323 phone - Asterisk --- NAT router - H323 gateway - PSTN I am trying to register a H323 gateway through a NAT to Asterisk for outgoing calls to PSTN. How can I achieve the above? Please help and advise. Many Thanks. Newbie ___ 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] Follow-Me Script
THANKS! I had heard of that but couldn't find it. I love that whisper feature. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Roy Sent: Monday, March 21, 2005 8:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Follow-Me Script On Sun, 20 Mar 2005 16:36:08 -0800, Kerry Garrison [EMAIL PROTECTED] wrote: I am trying to implement a follow-me script (http://www.voip-info.org/wiki-Asterisk+Tips+follow+me) but I am having a brain fart as I haven't a clue where to get started with what to do with this. Kerry, I'm more of a fan of anthm's patch that does this. You need to be running CVS-Head to get it though. http://bugs.digium.com/bug_view_page.php?bug_id=0002905 -Chuji ___ 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] mpg123 home music from stream
Henry Devito wrote: Hi Ken, This has worked fine for me for about 6 months, maybe I just didn't notice a problem. As far as I know there has been music playing when people are being put on hold every time. Ah but you might want to take a closer look. If you can, watch the active connections on your streaming server. When you first start asterisk, you'll see connections formed from ast to your streaming server. Test music on hold, all is working, cool. Listen to stream as long as you want, works great. Now hang up, wait about 30 secs. and watch the connections drop off your streaming server. Test music on hold.When you test you will still hear music, but you won't see any new connections back to the streaming server, you'll just be listening to a buffered loop that was streamed in previously. Last * ver. I tried was 1.0.3 and I have not seen anything in the change logs thru 1.0.7 Did you take look at patch/bug #413? this describes the above problem. Same as I'm having. http://bugs.digium.com/bug_view_page.php?bug_id=413 ___ 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] Log Error
It means the caller hung up in the middle of the voicemail app. Anton Krall wrote: So far, nobody has been able to tell us what this error means. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MF Hulber Sent: Lunes, 21 de Marzo de 2005 02:54 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Log Error I'm getting the same problem running Asterisk CVS-HEAD-03/21/05-03:24:01 built by [EMAIL PROTECTED] on a i686 running Linux which is the code from yesterday. Robert Goodyear wrote: FWIW I get the same exact error at the end of every VM session as well, thus: -- Playing 'vm-intro' (language 'en') -- Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/501/INBOX/msg format: wav49, 0x8186370 -- x=1, open writing: /var/spool/asterisk/voicemail/default/501/INBOX/msg format: gsm, 0x81634c8 -- x=2, open writing: /var/spool/asterisk/voicemail/default/501/INBOX/msg format: wav, 0x8186570 -- User hung up Mar 11 12:21:32 WARNING[22142]: app_queue.c:374 changethread: Can't change device '**Unknown**' with no technology! Mar 11 12:21:32 WARNING[22142]: app_queue.c:374 changethread: Can't change device '**Unknown**' with no technology! Mar 11 12:21:32 WARNING[22142]: app_queue.c:374 changethread: Can't change device '**Unknown**' with no technology! On Mar 4, 2005, at 7:00 PM, Anton Krall wrote: Guys, this error has been driving me nuts and I see no indication anywhere as to what it may mean. Anybody has any clues on this? -- User ended message by pressing # -- Playing 'auth-thankyou' (language 'en') -- Playing 'vm-review' (language 'en') -- Saving message as is -- Playing 'vm-msgsaved' (language 'en') Mar 4 21:02:06 WARNING[8816]: app_queue.c:374 changethread: Can't change device '**Unknown**' with no technology! -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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 and X [was: Re: [Asterisk-Users] zaptel PRI drivers]
Quoting Roger Gulbranson [EMAIL PROTECTED]: On Mon, 2005-03-21 at 08:57 -0700, Tom wrote: We don't want to have to spend an extra 3 grand for another server just to take up more space when we have this box that is sitting here idle 99% of the time, and as it has worked spectacularly well with the wctdm cards, I don't see why it can't with the wcte110p/PRI. The wctdm only has to transfer data for 4 channels. The wcte110p has to do 24 (23+1). Your probability of having problems just went up by a factor of 6. We had 3 wctdm's in this box, so that's 12 channels, and the problems occur with 0 calls up (IE no data is being passed across any of the channels). Anyway, at most the chances of problems doubled, but the box is never under any sort of heavy load, we had 8 calls up at one point with the wctdms, it never flinched. ___ 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] mpg123 home music from stream
Yeah I got it to work... but I can't get the set command to work.. like when I try to set the hold music class it just does nothing and then when I do musiconhold() it hangsup! On Mon, 21 Mar 2005 10:19:13 -0600, Henry Devito [EMAIL PROTECTED] wrote: Hi Ken, This has worked fine for me for about 6 months, maybe I just didn't notice a problem. As far as I know there has been music playing when people are being put on hold every time. - Original Message - From: Ken Godee [EMAIL PROTECTED] To: Matt [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 21, 2005 9:17 AM Subject: Re: [Asterisk-Users] mpg123 home music from stream Matt wrote: Where is my MoH class? I understand what's being done here... but I don't see where that is.. like for meetme conferences, and being placed on hold and such... which file? musiconhold.conf But once you get it going, it doesn't work anyway. Would love to have someone prove me wrong. Asterisk stops MOH (closes the stream) when channel hangs up. This is great for all other MOH uses, but drops the mp3 stream and doesn't reconnect to streaming sever. (as noted in original patch/bug #413) I was streaming XM radio thru MOH via shoutcast. Unless someones fix this problem. ___ 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] Modify CallerID (on SIP phone) during call
Is it possible to modify the caller id on the phone during a call (session) ? If not does anybody know with which SIP request this could be handled ? I'm know investigating RFC3311 which seems to offer an answer but if somebody already has an answer ... Michael ___ 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] Modify CallerID (on SIP phone) during call
Michael Devenijn wrote: Is it possible to modify the caller id on the phone during a call (session) ? If not does anybody know with which SIP request this could be handled ? Do you mean what is displayed on the phone's display? If so, yes, with some phones this is possible, by performing a re-INVITE or UPDATE with the new caller information included. However, Asterisk currently has no way to request that to happen, it only knows how to send re-INVITE/UPDATE to change the media path. ___ 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] Soekris net4801 and analog interface?
Strange; It works for me. The x100p (Digium 100 buck model) I have is slotted for 3.3v and works fine. I'm running gentoo with udev and the 2.6.11 kernel with soekris patches (udev is cool, coldplug automagically loads the drivers). The 4801 is flashed with whatever the latest bios is from Sorin. Using a 40G 2.5 inch laptop drive, no CF Card, boot directly from HD. Asterisk is CVS HEAD. lspci shows: :00:00.0 Host bridge: Cyrix Corporation PCI Master :00:06.0 Ethernet controller: National Semiconductor Corporation DP83815 (MacPhyter) Ethernet Controller :00:07.0 Ethernet controller: National Semiconductor Corporation DP83815 (MacPhyter) Ethernet Controller :00:08.0 Ethernet controller: National Semiconductor Corporation DP83815 (MacPhyter) Ethernet Controller :00:0a.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface -- Digium X100P :00:12.0 ISA bridge: National Semiconductor Corporation SC1100 Bridge :00:12.1 Bridge: National Semiconductor Corporation SC1100 SMI :00:12.2 IDE interface: National Semiconductor Corporation SCx200 IDE (rev 01) :00:12.5 Bridge: National Semiconductor Corporation SC1100 XBus :00:13.0 USB Controller: Compaq Computer Corporation ZFMicro Chipset USB (rev 08) kernel messages: Mar 20 22:36:55 ast-soekris Zapata Telephony Interface Registered on major 196 Mar 20 22:36:56 ast-soekris wcfxo: DAA mode is 'FCC' Mar 20 22:36:56 ast-soekris Found a Wildcard FXO: Wildcard X101P lsmod: Module Size Used by ohci_hcd 16648 0 wcfxo 11040 0 --- zaptel220292 5 wcfxo - crc_ccitt 1920 1 zaptel natsemi23904 0 Did you make your modem donation to Digium? :-) John Breeden Hawaii Matt Ryanczak wrote: I ended up using a sipura spa-3000 for FXO/FXS. It works great. http://www.sipura.com/products/spa3000.htm -Matt On Mon, 2005-03-21 at 09:40 -0600, John Simon wrote: This is the same experiance I had with my net4801 and X100P. Do you know of any 3.3V PCI modems that will work with Asterisk? --- Matt Ryanczak [EMAIL PROTECTED] wrote: I have 2 X100p clones that do not work in the net4801. The 4801 will not even power up with them installed. Both cards work fine a a standard desktop PC. On Sun, 2005-03-20 at 21:58 -1000, John Breeden wrote: X100P is 3.3v not 5v, at least the one I have. Works fine in a 4801. John Simon wrote: Is anyone using a net4801 and an analog only setup? I am looking for a modem that is PCI 3.3V, apparently the X100P is 5.0V PCI only so it won't work with the net4801. ___ 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 ___ 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] Soekris net4801 and analog interface?
[EMAIL PROTECTED] wrote: Strange; It works for me. The x100p (Digium 100 buck model) I have is slotted for 3.3v and works fine. I'm running gentoo with udev and the 2.6.11 kernel with soekris patches (udev is cool, coldplug automagically loads the drivers). The 4801 is flashed with whatever the latest bios is from Sorin. Using a 40G 2.5 inch laptop drive, no CF Card, boot directly from HD. Asterisk is CVS HEAD. Hi, Where did u buy that X100P from? Ta Senad ___ 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] Replacement 7960 Handset
After 4 hours of debugging codecs, changing config files, etc. as a result of not being able to capture voice from a Cisco 7960, I eventually found that the mic in the handset appears to be dead. Does anyone know where I can get a new handset (just the part you hold to your head, everything else on the phone works fine)? Or, does anyone know how to open one up? I tried doing a little prying with a screwdriver but gave up after marring the plastic a bit. I've googled but can't seem to find anyone selling just the handset. If you have one from a broken phone, I'd be more than happy to pay shipping, etc. if you want to sell it. 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
[Asterisk-Users] Doubts Configuration SIP
Dear Sirs We are doing some tests in our lab with Digium/Asterisk boards and we have some doubts regarding Asterisk´s SIP server configuration, could you help us please? See attached our topology. Thanks in advance. Best regards Marcia Configuration_serv_sip.doc Configuration_serv_sip.doc Description: Configuration_serv_sip.doc ___ 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 and X [was: Re: [Asterisk-Users] zaptel PRI drivers]
On Mon, 21 Mar 2005, Roger Gulbranson wrote: On Mon, 2005-03-21 at 08:57 -0700, Tom wrote: We don't want to have to spend an extra 3 grand for another server just to take up more space when we have this box that is sitting here idle 99% of the time, and as it has worked spectacularly well with the wctdm cards, I don't see why it can't with the wcte110p/PRI. The wctdm only has to transfer data for 4 channels. The wcte110p has to do 24 (23+1). Your probability of having problems just went up by a factor of 6. Also, you may not notice if you miss a ms worth of audio data, but the digital signalling on a pri will. Ideally this should not be a problem but with standard kernels it will be. Peter ___ 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] mpg123 home music from stream
Really? I just tried it and WHEN it's working.. it is streaming.. and even when I hang up it keeps mpg123 up and running in the background. ___ 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] Why is asterisk's voice mail called comedian.
On Mon, 21 Mar 2005 08:27:17 -0500, Steve Clark [EMAIL PROTECTED] wrote: snip It doesn't sound very professional. comedian n 1: a professional performer snip What's not professional about that? :) ___ 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] codec
Alessandra Grasso wrote: My objective is to estimate the performances of * How much the trancoded can influence the performances? Thanks, show translation recalc 30 -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] why even use SIP
I am setting up a new asterisk based call center. I just read: http://www.voip-info.org/wiki-IAX+versus+SIP After reading this and other google results for IAX vs SIP is there any reason why i should use SIP anywhere !! t ___ 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] why even use SIP
Sys Admin wrote: After reading this and other google results for IAX vs SIP is there any reason why i should use SIP anywhere !! Well, let's see.. 99.99% of the available VOIP hardware only support SIP, MGCP and H.323, but not IAX2. Is that a good reason? IAX2 calls between servers carry the signaling and media in the same connection, which is good for NAT issues, but bad for CDR and traffic control issues. SIP handles them separately, so you can keep complete CDR without forcing the media to follow the same path. Is that a good reason? ___ 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] why even use SIP
On Mon, 21 Mar 2005 10:01:04 -0800, Sys Admin [EMAIL PROTECTED] wrote: I am setting up a new asterisk based call center. I just read: http://www.voip-info.org/wiki-IAX+versus+SIP After reading this and other google results for IAX vs SIP is there any reason why i should use SIP anywhere !! t Do you have your voip hardphones picked out yet? ___ 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 and X [was: Re: [Asterisk-Users] zaptel PRI drivers]
On Mon, 2005-03-21 at 08:57 -0700, Tom wrote: This box never was primarily an * box, it is a server that people have used VNC from windows desktops to run a couple of apps that are X11 only that we need in house. We just have been trying to get off of our old PBX, and onto * as our primary system, and it's been working fine with the wctdm haven't seen any degredation of voice quality, call quality, anything previous to this. We run the GDM system so that users can sign on with their same username/password, and they get their same groups/restrictions etc all through LDAP, this has been working for 2 years now. We don't want to set up 45 user accounts locally on the box, set up separate passwords, have the users manually keep those passwords in sync, and then have separate passwords (again!) for vnc, which is what we have to do if we can't get GDM/xdm/kdm and XDMCP to work. There are never more than 3-5 people logged in at once, and as I said previously this was all working just fine with wctdm cards, its just the wcte110p that has issues, and those are that it can't keep the timing right (according to our provider) when X is enabled. Our provider and our asterisk box get flooded with HDLC Abort(6) errors. We don't want to have to spend an extra 3 grand for another server just to take up more space when we have this box that is sitting here idle 99% of the time, and as it has worked spectacularly well with the wctdm cards, I don't see why it can't with the wcte110p/PRI. Tom Christensen Maybe you need to look at an inexpensive dell 1u machine. You shouldn't have to spend more than $1k for a machine to dedicate to asterisk. As you have seen, asterisk needs realtime speeds and when other apps get in it's way something gets dropped. If you don't need a rack mount server, you can find even cheaper machines around to dedicate to it. -- 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] why even use SIP
Sys Admin wrote: I am setting up a new asterisk based call center. I just read: http://www.voip-info.org/wiki-IAX+versus+SIP After reading this and other google results for IAX vs SIP is there any reason why i should use SIP anywhere !! Because most equipment doesn't support IAX -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Re: asterisk-1.0.7 make install on fedora corre 3 give errors
an update: since it might help others
I did the same make on another machine and it worked fine. So it seems
to be a problem with my tool-chain
t
On Sun, 20 Mar 2005 22:18:50 -0800, Sys Admin [EMAIL PROTECTED] wrote:
I am trying to install asterisk on fedora core 3 these are the steps i took:
1. download asterisk-1.0.7.tar.gz
2. make clean and make install and then it gives me these errors:
{standard input}:9975: Error: symbol `i' is already defined
{standard input}:9978: Error: symbol `__result' is already defined
{standard input}:9979: Error: symbol `__result' is already defined
{standard input}:9981: Error: symbol `__result' is already defined
{standard input}:9982: Error: symbol `__result' is already defined
{standard input}:9984: Error: symbol `__result' is already defined
{standard input}:9985: Error: symbol `__result' is already defined
{standard input}:9987: Error: symbol `__result' is already defined
{standard input}:9988: Error: symbol `__result' is already defined
{standard input}:9990: Error: symbol `__result' is already defined
{standard input}:9991: Error: symbol `__result' is already defined
{standard input}:9993: Error: symbol `__result' is already defined
{standard input}:9994: Error: symbol `__result' is already defined
{standard input}:9996: Error: symbol `__result' is already defined
{standard input}:9997: Error: symbol `__result' is already defined
{standard input}:: Error: symbol `__result' is already defined
{standard input}:1: Error: symbol `__result' is already defined
{standard input}:10002: Error: symbol `__result' is already defined
{standard input}:10003: Error: symbol `__result' is already defined
{standard input}:10005: Error: symbol `__result' is already defined
{standard input}:10006: Error: symbol `__result' is already defined
{standard input}:10008: Error: symbol `__result' is already defined
{standard input}:10009: Error: symbol `__result' is already defined
{standard input}:10011: Error: symbol `__result' is already defined
{standard input}:10012: Error: symbol `__result' is already defined
{standard input}:10014: Error: symbol `__result' is already defined
{standard input}:10015: Error: symbol `__result' is already defined
{standard input}:10017: Error: symbol `__result' is already defined
{standard input}:10018: Error: symbol `__result' is already defined
{standard input}:10020: Error: symbol `__result' is already defined
{standard input}:10021: Error: symbol `__result' is already defined
{standard input}:10023: Error: symbol `__result' is already defined
{standard input}:10024: Error: symbol `__result' is already defined
{standard input}:10030: Error: symbol `i' is already defined
{standard input}:10042: Error: symbol `i' is already defined
{standard input}:10048: Error: symbol `arg_cols' is already defined
{standard input}:10054: Error: symbol `i' is already defined
{standard input}:10060: Error: symbol `arg_rows' is already defined
make[1]: *** [editline.o_a] Error 1
make[1]: Leaving directory `/root/download/asterisk/asterisk-1.0.7/editline'
make: *** [editline/libedit.a] Error 2
any help would be appreciated,
sysadmin
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Re: [Asterisk-Users] why even use SIP
On March 21, 2005 01:07 pm, Kevin P. Fleming wrote: Well, let's see.. 99.99% of the available VOIP hardware only support SIP, MGCP and H.323, but not IAX2. Is that a good reason? No. 95% of the marketplaces uses Windows. Drive the marketplace to use better protocols. IAX2 calls between servers carry the signaling and media in the same connection, which is good for NAT issues, but bad for CDR and traffic control issues. SIP handles them separately, so you can keep complete CDR without forcing the media to follow the same path. Is that a good reason? Yes, but the dynamic port allocation and ABSOLUTELY INSANE WORDINESS and WASTAGE of the SIP control protocol is reason enough for me to never want to support it. While perhaps not worth much on my own, I am voting with my wallet and my feet. I will not support SIP, nor will I purchase products or services which require 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
[Asterisk-Users] audio frequency with wcfxs and K8t
Friday and Saturday I was wrestling with a VoIP system that was having very strange problems. It was playing the outgoing IVR audio at 2-5x faster than it should have been. I found that if I stopped asterisk, removed the wcfxs driver and installed the ztdummy driver, the audio would play fine. I tested this in and out several times and it always worked fine with ztdummy and never worked right with wcfxs. I cannot find any references on the 'net about such a problem. Anyone else run into this? Details: Asterisk 1.0.6 MSI K8T Master2 motherboard Single AMD Opteron installed on the motherboard (other socket empty) TDM card with a single FXO installed on it The system is working fine now (SIP in and out), but I want to put a PSTN line into the FXO port as a land-line fallback. I also figured the TDM card could be the timing device for meetme and such, but I think ztdummy will do just as well there. Anyone else run into this? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] Soekris net4801 and analog interface?
Make sure you are feeding your 4801 enough juice to support the pci card, I'm currently feeding 15v @ 1A. I got the x100p directly from Digium a few years ago, at the time it was a component of Digium's developer package. lspci identifies the card as a Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface . From what I understand, digium sold (and what I have) is a stock Tiger modem. I know a couple of years ago you could pickup the Tiger modem for $10-12 (thus the quip about the digium modem donation). I think digium stopped selling the x100p after the fxo module came out for the tdm400p. But then the tdm400p is also a 3.3v card and can support a mix of 4 fxo/fxs modules. Just for yucks I plugged a tdm400p with an fxs module into my 4801 and the tdm400p driver loaded just fine: lsmod from 4801 w/ tdm400p (CVS HEAD): Module Size Used by ohci_hcd 16648 0 wctdm 34496 0 ---tdm400p zaptel220292 1 wctdm crc_ccitt 1920 1 zaptel natsemi23904 0 With the tdm400p, you'll need to wire up a 12v feed to the connector on the back of the card and you'll also need a wider box :-) Pretty cool though, a 4801 with a mix of 4 fxo/fxs ports. Someone somewhere also makes a pci header card that has the silicon on board to support two pci cards on a 4801, might be interesting. Let's see now, mix 'o 4 fxs/fxo ports and a Sogoma w/ 2 T1/FR ports. Now, if someone could fix the dang'd linux driver for soekris's vpn1411 hardware encryption card, you could do codec compression without taxing the geode processor. (hint, hint). Russ Nelson isn't busy right now.. :-) Ah yes, I'ts currently only a dream: Soekris with 2 T1s, 4 fxo/fxs ports and gsm running to 20 cisco eye-candy phones All from this itty-bitty boxen -) John Breeden Hawaii Senad Jordanovic wrote: [EMAIL PROTECTED] wrote: Strange; It works for me. The x100p (Digium 100 buck model) I have is slotted for 3.3v and works fine. I'm running gentoo with udev and the 2.6.11 kernel with soekris patches (udev is cool, coldplug automagically loads the drivers). The 4801 is flashed with whatever the latest bios is from Sorin. Using a 40G 2.5 inch laptop drive, no CF Card, boot directly from HD. Asterisk is CVS HEAD. Hi, Where did u buy that X100P from? Ta Senad ___ 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] CallingCaed Application
can any body refere me please to a callingcard application that has user's manual or some clear documentation. i have installed areskicc and i have been struggling to make it work. i want to try something else. __ Do you Yahoo!? Yahoo! Mail - 250MB free storage. Do more. Manage less. http://info.mail.yahoo.com/mail_250 ___ 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] Soekris net4801 and analog interface?
I ended up using a sipura spa-3000 for FXO/FXS. It works great. http://www.sipura.com/products/spa3000.htm -Matt On Mon, 2005-03-21 at 09:40 -0600, John Simon wrote: This is the same experiance I had with my net4801 and X100P. Do you know of any 3.3V PCI modems that will work with Asterisk? --- Matt Ryanczak [EMAIL PROTECTED] wrote: I have 2 X100p clones that do not work in the net4801. The 4801 will not even power up with them installed. Both cards work fine a a standard desktop PC. On Sun, 2005-03-20 at 21:58 -1000, John Breeden wrote: X100P is 3.3v not 5v, at least the one I have. Works fine in a 4801. John Simon wrote: Is anyone using a net4801 and an analog only setup? I am looking for a modem that is PCI 3.3V, apparently the X100P is 5.0V PCI only so it won't work with the net4801. ___ 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 ___ 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] CallingCard Application
can anybody please refere me to a callingcard application that has user's manual or clear documentation. i have installed areskicc, but it didn't work for me and i need to try something else. __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ 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 and X [was: Re: [Asterisk-Users] zaptel PRI drivers]
Quoting Tzafrir Cohen [EMAIL PROTECTED]: On Sun, Mar 20, 2005 at 11:12:22PM -0700, Tom wrote: I have a quick question. I know that running X on an asterisk server is not officially supported, Generally it shouldn't cause errors, but will probably degregate performance, as an X server is probably as close as Asterisk is to the hardware and optimized just as well for minimal latency. We experience the exact opposite, even when we are getting these HDLC errors we never have experienced call quality issues. I would be willing to ignore the errors, but our provider doesn't see it that way as their techs get pages constantly about them. however, I've never had any trouble with it until now (8 months, using wctdm cards with fxo and fxs ports, IAX trunks, SIP phones, everything except a PRI card). Now I just installed my first asterisk box that terminates a PRI, and bam, HDLC errors up the wazoo if X is running, if its not, everything is fine, I assume this is because the timing parameters for the PRI are so much more strick. Why do you need the X server running at all? Is Asterisk running as root? With real-time priority? (-p) What distro do you use, BTW? This is running on Fedora Core 3, and yes asterisk is running as root with -p I get PRI HDLC errors whenever X is enabled. I don't mind if X is a little less responsive (even alot less responsive), but I would really like to be able to run X on a server with a PRI. Is there any way to reduce X11's priority so that it doesn't interfere with the zaptel driver for the PRI... I've tried renicing X as far down as I can and renicing Asterisk up as far as I can, however I fear this won't ever fix the problem since I think the actual kernel module that is running the pri card needs to get higher priority (ie, the kernel itself needs higher priority). What exactly do you run on X? Is the CPU very busy? try a light interface such as icewm, windowmaker or fluxbox with a theme that uses no gradients and no special effects. If your display has a little resolution, try something like matchbox. The CPU is never very busy as I stated in my original post ~3% average usage Is there any way to do this? Am I correct in my analysis? I really don't understand why on a system that averages less than 3% CPU usage with X running, why it can't handle the PRI. I know for whatever reason X always gets a really high priority (although top doesn't show X getting any special treatment its PR 15 NICE 0 by default, lower than most other processes on the system). Another idea is that right now the system is only a single proc, but it is dual proc capable. Would this somehow help if we added the second proc? My thinking is it won't because it's a kernel module we are dealing with, and because of that I can't control the affinity of the driver (I was thinking at one point put X11 on 1 proc and Asterisk on the second, but it's not Asterisk that has the problem I don't think.) My final idea is that currently the system has an onboard 8mb ati graphic card that leaves almost all actual graphics processing to the CPU, could adding a better graphics card possibly help X use less cpu and not get in the way so much? Anyway, I know this isn't a supported setup, so if thats your answer don't bother replying, I'm know this will be a kludge/hack to get working (if I can get it working at all). I'm just trying to do something that will be convienient for me and my users, there are other systems running on the server that I don't want to manage through the CLI, and the users don't know how to manage through the CLI, and there is no web management for them. You want to run a full desktop just be able to manage the Asterisk box? That's what ssh is for. Xorcom Rapid added a menu application for managing the box for those who don't know the command to type. If you have an X server on your workstation you can run X programs on your local X server. There should be no need for a local X server on the Asterisk box. This is not to manage asterisk. Asterisk has plenty of web based admin suites, none of which are installed, as I generally like working on the CLI, and manage asterisk that way just fine. However, we have a couple of very large in-house apps that run on X to manage some other things (in-house proprietary stuff). That is the primary function of this box, and we added * to this box after the fact with a couple wctdm cards, it worked very well but we just upgraded our pstn interface from old analog lines to a PRI, so we needed to upgrade the asterisk box as well... Does anyone have success running X on an asterisk box that terminates a PRI? If so what hardware (video card, cpu, ram, mobo, etc)? Thanks as I know this setup isn't supported, and I'm probably asking alot, don't think I'm just relying on the list for bizarre things, I've been trying
