Re: [asterisk-users] PRI dialout problem with some numbers...
Carlos Chavez wrote: I have an Asterisk server (1.4.13) using PRI in Monterrey, Mexico. This is really the first server I have used with PRI in Mexico as we normally use MFC/R2. Everything seems to be working except that some numbers always seem to be busy when you dial them. All these numbers belong to different phone companies. I know that with R2 this problem is present if you have a #define DEFAULT_T1 value under 15000 in mfcr2.c (the default used to be 5000). Is there an equivalent value for PRI? The company we are using is Alestra. Here is what I get when we dial a number that belongs to a company called Protel: -- Executing [EMAIL PROTECTED]:1] Set(SIP/199-08be6c00, TIMEOUT(absolute)=3600) in new stack -- Channel will hangup at 2007-11-05 22:03:34 UTC. -- Executing [EMAIL PROTECTED]:2] Dial(SIP/199-08be6c00, Zap/g1/11070665||Ww) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/11070665 -- Zap/1-1 is proceeding passing it to SIP/199-08be6c00 -- Channel 0/1, span 1 got hangup request, cause 31 -- Hungup 'Zap/1-1' [Nov 5 15:03:34] NOTICE[22300]: cdr.c:434 ast_cdr_free: CDR on channel 'Zap/1-1' not posted == Everyone is busy/congested at this time (1:0/0/1) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.503 / Virus Database: 269.15.22/1112 - Release Date: 11/5/2007 7:11 PM We have several sites setup with Alestra. Contact me off list if you like and I'll see what I can do to help you out. Alex ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk OpenVZ
On Mon, Nov 05, 2007 at 08:10:33PM -0600, JR Richardson wrote: Hi All, I've got debian (etch), openvz and asterisk up and running using the openvz wiki guides. The examples use `apt-get install asterisk` and this will install 1.2.13. Has anyone gotten an VPS to compile the latest versions from source? Also, I'm unsure how the zaptel modules come into play, could use some guidance there as well. I don't know about openvz, but from my experince with vserver (which is supported rather well and painlessly in Debian, with vanilla kernels) zaptel should basically be a matter of: m-a a-i zaptel . One gotcha is the generation of the /dev/zap files. If possible, generate static ones by the host and avoid udev at all. Alternatively, your vserver may need device files generation capabilities (which is naturally not that secure, as it allows access to disks). -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk OpenVZ
I've got debian (etch), openvz and asterisk up and running using the openvz wiki guides. The examples use `apt-get install asterisk` and this will install 1.2.13. Has anyone gotten an VPS to compile the latest versions from source? No problem -- we're running the latest 1.4.x in multiple VEs on Dual and Quad-Core Xeons with Debian and OpenVZ as part of our Virtual Private Asterisk Server service offering (see www.vpas.ca). Indeed our customers can install gcc and build their own Asterisk if they want (and some do). Also, I'm unsure how the zaptel modules come into play, could use some guidance there as well. We chose to have a separate servers with Quad PRI cards to act as gateways which IAX to customer VPASs so the only zaptel module we needed was the ztdummy to provide timing for MeetMe and trunking. You can easily install this module in VE0 and grant access to the appropriate /dev/zap entries in the various VEs. You could put your Digium zaptel hardware on your OpenVZ system but I'm note sure how you would control/limit access to different channels to different VEs so I would think (and I now depart from empirical knowledge into speculation) one would probably want to run a gateway Asterisk in VE0 that would interface to the Digium hardware and pass the connections to other VEs using SIP or IAX. It would be interesting to hear from other Asterisk/OpenVZ users who have put Digium zaptel hardware on their box and managed to securely limit access to specific channels to different VEs. g. -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.ip-pbx.ca www.vpas.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linksys SPA-941 Unavailable
Hello! We are using several Linksys SPA-941 in our office. After IP change occur devices seems not to be reachable, actually unavailable! Devices is connected, e.g. we can place a call using SPA-941 but can not receive any calls... Kim __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and Grandstream both behind different NAT
Hi, i am facing some problem configuring 2 Grandstrem phones 101 behind NAT. I have put server ip as 192.x.x.x and OutGoing proxy as 72.x.x.x, i have forwarded ports on both Grandstream and Asterisk sides, and using those ports on Grandstream for SIP and RTP with random ports =no. This setup is working however at a time only one phone gets registered. Has someone experienced the same problemany suggestions? Thanks in advance, Viv ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recording just first part of call?
I know that I can record the contents of a call by calling Monitor() or MixMonitor() from the dialplan just before invoking Dial(). I have a potential customer who wants only the first minute of each call recorded (for identification purposes, without the storage overhead of keeping the complete call). Can anyone here think of the easiest way to do this? The only possibilities I can think of are: a) Add a new option to Monitor() or MixMonitor() to stop recording after a specified length of time. b) Record the whole call and post-process the recording file to discard all except the required first part. Any better ideas? Thanks in advance! Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Kirk IP600/3 Wireless Server SIP config
RB == Remco Barendse [EMAIL PROTECTED] writes: RB Did you manage to make a dump of a working configuration from the RB IP600/3? RB Would be really useful, can't seem to get it to work properly. Ok, there is one at http://amorsen.dk/complete-IP1200-0f-07-eb.txt. I'm not sure it's very useful, I had to replace lots of things with DELETED, since it's a production configuration. I also removed most of the connected phones. Notice that most of the network configuration is ignored, because the device is set for DHCP. You can at least use it to compare with yours. /Benny ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording just first part of call?
Am Dienstag, den 06.11.2007, 11:49 + schrieb Tony Mountifield:
I know that I can record the contents of a call by calling Monitor()
or MixMonitor() from the dialplan just before invoking Dial().
I have a potential customer who wants only the first minute of each
call recorded (for identification purposes, without the storage overhead
of keeping the complete call).
Can anyone here think of the easiest way to do this? The only possibilities
I can think of are:
a) Add a new option to Monitor() or MixMonitor() to stop recording after
a specified length of time.
b) Record the whole call and post-process the recording file to discard
all except the required first part.
The asterisk manager API seems to offer a StopMonitor command, which
is basically the same as the StopMonitor() extensions.conf command,
afaik.
A quick ugly hack (and well, I did not have my coffee yet, so caveat
emptor):
Before calling the Monitor() in extensions.conf, call an AGI that kind
of starts a timer. This AGI would have to know about the Channel used
(you surely figure how to do that, I am to lazy to look it up right
now).
Something like
8
#!/usr/bin/php -q
GLOBAL $stdin, $stdout;
ob_implicit_flush(false);
set_time_limit(30);
error_reporting(0);
$stdin = fopen( 'php://stdin', 'r' );
$stdout = fopen( 'php://stdout', 'w' );
while ( !feof($stdin) )
{
$temp = fgets( $stdin );
$temp = str_replace( \n, , $temp );
$s = explode( :, $temp );
$agivar[$s[0]] = trim( $s[1] );
if ( ( $temp == ) || ($temp == \n) )
{
break;
}
}
$channel = $agivar[agi_channel];
system (screen -d -m /usr/local/bin/stop-recording .$channel);
exit(0);
8
The script at /usr/local/bin/stop-recording could be a bash script:
8
#!/bin/bash
sleep 60
# Before Stopping the monitor, you want to make sure that
# about 60 seconds went past
# Perhaps add some leeway if the other party answered
# after ring no. 5 or so
# The following should all be on one line, but emails tend to break...
( echo -e Action: login\nUsername: foo\nSecret: bar\nEvents: off\n\n ;
sleep 1 ; echo -e Action: StopMonitor\nChannel: $1\n\n ; sleep 1 ) |
netcat localhost 5038 /dev/null 2/dev/null
8
You would want to add a check that the original call is the one to be
StopMonitored() - e.g. if the caller hangs up and redials within a few
seconds, the second call would possibly be terminated. You could manage
this by writing the channel to a temporary file in the AGI, removing
the file after call termination. The Bash script would then read the
channel from the file, or just silently terminate if the file is not
there.
This is just an idea. It needs some tweaking here and there, and there
probably are way more elegant methods for solving the task... :-)
BR
Anselm
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Re: [asterisk-users] Asterisk and Grandstream both behind different NAT
On 11/6/07, Vivek Shrivastava [EMAIL PROTECTED] wrote: Hi, i am facing some problem configuring 2 Grandstrem phones 101 behind NAT. I have put server ip as 192.x.x.x and OutGoing proxy as 72.x.x.x, i have forwarded ports on both Grandstream and Asterisk sides, and using those ports on Grandstream for SIP and RTP with random ports =no. This setup is working however at a time only one phone gets registered. Has someone experienced the same problemany suggestions? use ngrep to do network trace ram ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA-941 Unavailable
On 11/6/07, Kim Joung-il [EMAIL PROTECTED] wrote: Hello! We are using several Linksys SPA-941 in our office. After IP change occur devices seems not to be reachable, actually unavailable! Devices is connected, e.g. we can place a call using SPA-941 but can not receive any calls... is the phone behind NAT ram ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording just first part of call?
If you're up to using the Manager interface and your programming language of choice, you could poll the list of active calls and stop recording when their duration exceeds a minute. According to my docs, res/res_monitor.c implements manager commands that could be used to halt current recordings. The Asterisk-Java library has a StatusAction and StopMonitorAction, if Java is a language candidate for an application you might write. Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Tuesday, November 06, 2007 6:50 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Recording just first part of call? I know that I can record the contents of a call by calling Monitor() or MixMonitor() from the dialplan just before invoking Dial(). I have a potential customer who wants only the first minute of each call recorded (for identification purposes, without the storage overhead of keeping the complete call). Can anyone here think of the easiest way to do this? The only possibilities I can think of are: a) Add a new option to Monitor() or MixMonitor() to stop recording after a specified length of time. b) Record the whole call and post-process the recording file to discard all except the required first part. Any better ideas? Thanks in advance! Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 7960 Queue Issue
On Mon, Nov 05, 2007 at 12:09:48PM +1100, Nick Brown wrote: Thanks Eric, this is the case. A bit of a shame that it removes the functionality for the member to see calls that have not come from a queue however there is not much choice in the matter. It works for me... somehow... I have Cisco 7960 phones also. I think I add the Local/xx instances into the queue instead of the SIP/ device names, and then have a context that checks the state of the SIP channel before trying to place a call to it. (So, member = Local/[EMAIL PROTECTED]) where agent_call is the context to go to in the dialplan that handles the agent calls (and passes it to another queue/voicemail if the queue drops out with full/unavailable etc.) [agent_call] does some stuff with ChanIsAvail checking if the channel is free before placing a call, and if it is found to be busy, it returns goes to a step which returns Busy() which causes the queue processor to move on to the next person in the queue. (It will go to agent_call again for the next destination, and so on.) That way, users can have DDI numbers with call waiting functionality enabled on the handset if they wish, but for queue calls, it goes to the next available queue member rather than stacking up all the calls on one phone. What I have is a simplified (and 1.4/1.2 compatible) version of Example 2 at: http://www.voip-info.org/wiki/view/Agents+without+agent+channel (just look in the [agent_call] bit of this, and you'll see it is using ChanIsAvail to check the status.) I did not need all the functionality of this example, so removed a bit of it, but used it because encountered a few limitations with chan_agent which meant I couldn't use Agents, so replaced the functionality in dialplan logic. (which was bit difficult to do, but it works!) I can send you what I have if you like, but my dialplan is quite complicated as the setup here allows 'agents' to log in and out from any phone, so the users extn numbers are essentially portable. (i.e, the handsets have some meaningless (to the user) extension like 42105 and the user logs in as 710 from that handset. Some database work is done when they log in to map 710 - SIP/42105, fix the outgoing caller ID, and add them to their queues. Alternatively, you might be able to use Agents, but I really cannot recommend it, as for me, it caused more problems than it solved (problems with call waiting, transfers, and the fact that the feature it relies on, AgentCallbackLogin() is deprecated in 1.4 anyway. Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to delete voice mail messages?
On Mon, Nov 05, 2007 at 12:47:52PM +0100, Michiel van Baak wrote: On 12:15, Mon 05 Nov 07, voip crazy wrote: Hello all, Could I create a script to delete the first messages on my voice mail? In this script should I update any messages index file or there isn't any file to index them? Could you share any script to do that? Hi, Voicemails are stored in /var/spool/asterisk/voicemail/context/vmbox by default. There's some .wav files and a .txt file for every message. You can easily delete them using some shellscript. Yes, but you must not just barge in and start deleting them, they have to be renumbered in sequence after you delete the ones you want, otherwise the vm app breaks when the user is listening to their messages. I think there is also a way to lock the files (I think with .LCK files) so that the vm app does not try to write them while you are manupulating. (and so your script can detect that there is a message being created.) I expect you will be able to find some code out there that does it without breaking it. (vmspool_manager) ? Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA-941 Unavailable
Post the relevant configuration files we'd be glad to help. On 11/6/07, Kim Joung-il [EMAIL PROTECTED] wrote: Hello! We are using several Linksys SPA-941 in our office. After IP change occur devices seems not to be reachable, actually unavailable! Devices is connected, e.g. we can place a call using SPA-941 but can not receive any calls... Kim __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Testcall
Ok, Moy, Thank you for your time!!! Speaking spanish?? Cristian. - Original Message - From: Moises Silva [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, November 05, 2007 5:42 PM Subject: Re: [asterisk-users] Testcall You have other process using at least one of those 1-10 channels. If some other process have it, testcall cannot grab it. Other process could be other testcall instance or Asterisk itslef. On 11/5/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: # ./testcall testcall.conf Chan 1, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025860' to '013331339767' Chan 2, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025861' to '013331339768' Chan 3, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025862' to '013331339769' Chan 4, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025863' to '013331339770' Chan 5, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025864' to '013331339771' Chan 6, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025865' to '013331339772' Chan 7, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025866' to '013331339773' Chan 8, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025867' to '013331339774' Chan 9, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025868' to '013331339775' Chan 10, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025869' to '013331339776' Loading protocol mfcr2 Failed to open channel: Device or resource busy Why??? My testcall.conf is: destination-no 013331339767 protocol-class mfcr2 protocol-variant ar,10,4 protocol-end cpe caller no originating-no 30025860 on-offered accept circuits 1-10 Thanks!!! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Within C++, there is a much smaller and cleaner language struggling to get out. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Statistics reporting
We offer a very comprehensive reporting and real-time monitoring commercial solution called QueueMetrics that has also a free mode for smaller CCs and hobbysts and scales well to multi-server setups with hundreds of live agents. See http://queuemetrics.com As a completely free alternative, try Asterisk Guru's Stats - http://www.asteriskguru.com/tutorials/installation_guide.html I hope this helps l. On Mon, 05 Nov 2007 22:54:24 +0100, Bob Pierce [EMAIL PROTECTED] wrote: Anyone know of a good package for reporting on Queue statistics from Asterisk? Bob -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MeetMe CPU resources
Hello, We would like to have a conference with 15 users aprox. We think that Pentium 4 3GHz and 1GB of RAM should be enough. Only Asterisk running. We wonder if somebody has some other experience, good or bad. We will use Asterisk 1.2 (it is a small and short project for only this). Thanks! -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe CPU resources
Carles Pina i Estany wrote: Hello, We would like to have a conference with 15 users aprox. We think that Pentium 4 3GHz and 1GB of RAM should be enough. Only Asterisk running. I'd consider that over-kill (Means it'll work fine). We've used a Pentium 3 866mhz with 512mb memory and have had up to 10 in a conference without any issues. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Testcall
Asi es, hablo español, soy de México. Anyone interested in R2 support for Asterisk can find more information at: http://www.moythreads.com/astunicall/ - Moy On 11/6/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Ok, Moy, Thank you for your time!!! Speaking spanish?? Cristian. - Original Message - From: Moises Silva [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, November 05, 2007 5:42 PM Subject: Re: [asterisk-users] Testcall You have other process using at least one of those 1-10 channels. If some other process have it, testcall cannot grab it. Other process could be other testcall instance or Asterisk itslef. On 11/5/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: # ./testcall testcall.conf Chan 1, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025860' to '013331339767' Chan 2, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025861' to '013331339768' Chan 3, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025862' to '013331339769' Chan 4, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025863' to '013331339770' Chan 5, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025864' to '013331339771' Chan 6, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025865' to '013331339772' Chan 7, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025866' to '013331339773' Chan 8, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025867' to '013331339774' Chan 9, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025868' to '013331339775' Chan 10, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025869' to '013331339776' Loading protocol mfcr2 Failed to open channel: Device or resource busy Why??? My testcall.conf is: destination-no 013331339767 protocol-class mfcr2 protocol-variant ar,10,4 protocol-end cpe caller no originating-no 30025860 on-offered accept circuits 1-10 Thanks!!! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Within C++, there is a much smaller and cleaner language struggling to get out. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Within C++, there is a much smaller and cleaner language struggling to get out. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe CPU resources
In article [EMAIL PROTECTED], Carles Pina i Estany [EMAIL PROTECTED] wrote: Hello, We would like to have a conference with 15 users aprox. We think that Pentium 4 3GHz and 1GB of RAM should be enough. Only Asterisk running. We wonder if somebody has some other experience, good or bad. We will use Asterisk 1.2 (it is a small and short project for only this). It will depend on whether you are using VoIP or a PRI card. I have a number of systems that have a single Pentium 4 @ 2.8GHz (with HT), 1GB RAM and a 4xE1 PRI card (TE410P), and they regularly have conferences with up to 90 participants. I would expect them easily to handle the full 120 if needed. We are not using echo cancellation. Using VoIP will consume more CPU due to the networking overhead, and even more if you are transcoding a compressed codec. However, with the spec of your machine, you should be ok with 15 users even using VoIP and compressed codecs. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Statistics Reporting
On 6th November Bob Pierce wrote: Anyone know of a good package for reporting on Queue statistics from Asterisk? Bob Hi Bob You can get free real-time queue statistics from www.orderlyq.com. Just click 'sign-up' button to connect. Rgds Russell ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Everyone is busy/congested: IP Trunk
after a copious loss of follicles :-), I finally got outbound working. Basically the channel statement in the call file needs to have the number to be called. For eg., in test.call format the statement as follows : Channel: SIP/3012345678@your-sip-provider And there is no need for a DIAL statement in extensions.conf unless you need to dial an additional number / extension. Then in sip.conf you need a para that matches your-sip-provider with the relevant auth info. These two wiki pages, they were very helpful in figuring out a solution to the problem : http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out+deliver+message hth, -baji. -- On Oct 30, 2007 8:43 AM, Gabriel Natale wrote: I have the same problem. I trying with more 4 SIP providers, the account is registering, receive inboud calls, but can`t make outbound calls for congestion. Can be the out call id the problem? Thanks Gabriel ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking
Hans Feringa wrote: I understood that a timing device (ztdummy if no zaptel hardware is present) was not necessary anymore with linux kernel 2.6. When I enable iax2 trunking I get this warning chan_iax2.c:8908 build_user: Unable to support trunking on user 'xx' without zaptel timing The linux kernel is 2.6.22-14-386 Can I ignore this message, and is trunking working despite this warning? The ztdummy module is not part of the zaptel ubuntu package, so it cannot be loaded. I wanted to install from ubuntu packages for a change and not compile it from source. You still need ztdummy. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking
I understood that a timing device (ztdummy if no zaptel hardware is present) was not necessary anymore with linux kernel 2.6. When I enable iax2 trunking I get this warning chan_iax2.c:8908 build_user: Unable to support trunking on user 'xx' without zaptel timing The linux kernel is 2.6.22-14-386 Can I ignore this message, and is trunking working despite this warning? The ztdummy module is not part of the zaptel ubuntu package, so it cannot be loaded. I wanted to install from ubuntu packages for a change and not compile it from source. rgds, Hans Feringa ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Testcall
Ok, muchas gracias, yo soy de Argentina. Estamos en contacto!! - Original Message - From: Moises Silva [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, November 06, 2007 1:31 PM Subject: Re: [asterisk-users] Testcall Asi es, hablo español, soy de México. Anyone interested in R2 support for Asterisk can find more information at: http://www.moythreads.com/astunicall/ - Moy On 11/6/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Ok, Moy, Thank you for your time!!! Speaking spanish?? Cristian. - Original Message - From: Moises Silva [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, November 05, 2007 5:42 PM Subject: Re: [asterisk-users] Testcall You have other process using at least one of those 1-10 channels. If some other process have it, testcall cannot grab it. Other process could be other testcall instance or Asterisk itslef. On 11/5/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: # ./testcall testcall.conf Chan 1, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025860' to '013331339767' Chan 2, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025861' to '013331339768' Chan 3, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025862' to '013331339769' Chan 4, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025863' to '013331339770' Chan 5, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025864' to '013331339771' Chan 6, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025865' to '013331339772' Chan 7, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025866' to '013331339773' Chan 8, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025867' to '013331339774' Chan 9, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025868' to '013331339775' Chan 10, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025869' to '013331339776' Loading protocol mfcr2 Failed to open channel: Device or resource busy Why??? My testcall.conf is: destination-no 013331339767 protocol-class mfcr2 protocol-variant ar,10,4 protocol-end cpe caller no originating-no 30025860 on-offered accept circuits 1-10 Thanks!!! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Within C++, there is a much smaller and cleaner language struggling to get out. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Within C++, there is a much smaller and cleaner language struggling to get out. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking
On Tue, 2007-11-06 at 18:30 +0100, Hans Feringa wrote: I understood that a timing device (ztdummy if no zaptel hardware is present) was not necessary anymore with linux kernel 2.6. Not quite... this is commonly misunderstood, so let me clarify. Under the 2.6 kernel, ztdummy gets it timing directly from the kernel, and not from certain USB controllers like ztdummy does under the 2.4 kernel. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help: Asterisk info
I am getting this error under system info: File Line Command Message common_functions.php 314 file_exists(/proc/scsi/scsi) the file does not exist on your machine Does anybody know how to fix this? Thank you in advance Jarga image001.jpg___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe CPU resources
On Tue, 6 Nov 2007, Carles Pina i Estany wrote: We would like to have a conference with 15 users aprox. We think that Pentium 4 3GHz and 1GB of RAM should be enough. Only Asterisk running. Your configuration should be more than sufficient. I think (gut feel, no hard stats) I get better audio in conferences with more than 10 users on hosts that have a hardware (Digium) timer. Just a couple of data points... On a dual Intel(R) Xeon(TM) CPU 3.40GHz, all calls are SIP, te410p for timing only, CentOS 4.5, Kernel 2.6.9-55.ELsmp, Asterisk 1.2.18, MySQL, various AGI's. According to top Asterisk is taking about 80mb and between 5% and 10% of a cpu. -u1::sedwards:~$ uptime 09:57:46 up 168 days, 17:36, 1 user, load average: 0.42, 0.22, 0.19 -u1::sedwards:~$ /usr/bin/free total used free sharedbuffers cached Mem: 20749202056944 17976 0 954401779420 -/+ buffers/cache: 1820841892836 Swap: 779144292 778852 -u1::sedwards:~$ sudo /usr/sbin/asterisk -r -x meetme Conf Num PartiesMarked Activity Creation xx 0003 N/A00:00:40 Static xx 0001 N/A01:50:37 Static xx 0001 N/A276:00:56 Dynamic xx 0001 N/A2817:21:50 Static xx 0005 N/A3399:34:11 Static xx 0014 N/A3429:47:26 Static xx 0002 N/A3429:56:07 Static xx 0003 N/A3430:08:10 Static xx 0001 N/A3430:20:55 Static xx 0004 N/A3432:08:31 Static xx 0002 N/A3491:17:05 Static * Total number of MeetMe users: 37 On a dual Intel(R) Xeon(TM) CPU 3.40GHz, all calls are IAX, t100p for timing only, CentOS 4.5, Kernel 2.6.9-55.0.2.ELsmp, Asterisk 1.2.7.1, various AGI's, 6 supervisors monitoring constantly refreshing heavily database driven web pages. According to top Asterisk is taking about 140mb (I know there is a leak somewhere) and between 10% and 55% of a cpu. -ap10::sedwards:~$ uptime 10:06:09 up 88 days, 18:11, 3 users, load average: 1.54, 1.20, 1.00 -ap10::sedwards:~$ /usr/bin/free total used free sharedbuffers cached Mem: 40864523588164 498288 0 407842972616 -/+ buffers/cache: 5747643511688 Swap: 20316082242031384 -ap10::sedwards:~$ sudo /usr/sbin/asterisk -r -x meetme Conf Num PartiesMarked Activity Creation xx 0001 N/A00:01:11 Dynamic xx 0001 N/A00:01:27 Dynamic xx 0001 N/A00:03:12 Dynamic xx 0001 N/A00:06:06 Dynamic xx 0001 N/A00:08:27 Dynamic xx 0002 N/A00:11:04 Dynamic xx 0001 N/A00:14:15 Dynamic xx 0002 N/A01:02:27 Dynamic xx 0001 N/A01:04:08 Dynamic xx 0001 N/A01:04:20 Dynamic xx 0001 N/A01:06:57 Dynamic xx 0001 N/A01:15:05 Dynamic xx 0001 N/A01:23:40 Dynamic xx 0001 N/A01:35:54 Dynamic xx 0001 N/A01:40:09 Dynamic xx 0002 N/A01:47:36 Dynamic xx 0001 N/A01:50:59 Dynamic xx 0001 N/A01:57:01 Dynamic xx 0001 N/A02:01:07 Dynamic xx 0001 N/A02:01:16 Dynamic xx 0001 N/A02:03:44 Dynamic xx 0002 N/A02:05:02 Dynamic xx 0001 N/A02:05:48 Dynamic xx 0001 N/A02:05:54 Dynamic xx 0003 N/A02:06:19 Dynamic xx 0001 N/A30:00:27 Dynamic xx 0001 N/A36:46:29 Dynamic xx 0001 N/A83:03:38 Dynamic xx 0001 N/A86:45:35 Dynamic xx 0001 N/A87:06:51 Dynamic xx 0002 N/A123:09:00 Dynamic xx 0002 N/A139:05:16 Dynamic xx 0001 N/A157:04:56 Dynamic xx 0001 N/A159:57:13 Dynamic xx 0003 N/A180:43:04 Static xx 0001 N/A219:05:01 Dynamic xx 0001
[asterisk-users] Asterisk 1.4 + Presence
Dear all, I'm using Asterisk 1.4 as my SIP server of my LAN users. The SIP clients are using different operating systems such Debian, Gentoo and Windows XP so they use different SIP softphones like SJPhone, Twinkle and X-Lite. In order to let SIP clients to see the presence status to each other, do I have to establish any special setting in Asterisk 1.4 ??? Or the presence status (online, offline, away, etc.) is only up to the SIP clients and not up to the Asterisk ??? Really thanks Alejandro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma S200 and Digium TDM400P together
Hi, I have these two cards, the Sangoma has 4 fxo interfaces and the digium has 1 fxo and 1 fxs. After install the sangoma card, my zaptel.conf was configured for that card. I'm trying to configure the Digium one together thinking that the Digium ports should be 5 and 8 but it doesn't works. Someone has some example about this? Thanks in advance Pauçp ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Help
Under asterisk info: Sip registry 12/12 76.xxx.xxx.xxx D N 5066 UNREACHABLE 11/11 76.xxx.xxx.xxx D N 5064 UNREACHABLE 10/10 76.xxx.xxx.xxx D N 5062 UNREACHABLE All these IP phones are behind NAT. What could be the problem? Thanks in advance. Jarga image002.jpg___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe CPU resources
On Tue, 6 Nov 2007, Steve Edwards wrote: On a dual Intel(R) Xeon(TM) CPU 3.40GHz, all calls are SIP, te410p for timing only, CentOS 4.5, Kernel 2.6.9-55.ELsmp, Asterisk 1.2.18, MySQL, various AGI's. According to top Asterisk is taking about 80mb and between 5% and 10% of a cpu. This host is also running OpenSER, distributing calls between this host and it's twin. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4 SIP Jitter Buffer
Are you running the SIP Jitter Buffer? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Luc Moreira Sent: Monday, November 05, 2007 10:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.4 SIP Jitter Buffer Gregory We have many Asterisk 1.4.13 in production solid like a rock. Couples examples: a) Asterisk 1.4.13 + Unicall + 2 E1 MFCR2 Digium + Legacy PBX 60+ Extentions / IVR / 10~30 concorrent calls b) Asterisk 1.4.11 + 1 E1 ISDN PRI Digium 50+ Extentions / IVR / 5 Queues / ~2000 call/day c) Asterisk 1.4.13 + 4 E1 ISDN Digium (working in progress) CallCenter / 150 PAs / 15 Queues / expected 8000 calls/day -- Luc Gregory Boehnlein escreveu: Hello, I'm running into a few situations on lossy network links where a SIP jitter buffer w/ some PLC would be helpful. My main TDM gateways are running 1.2 (which is solid, stable, reliable and very very very well behaved when you know it's limitations), but I'm considering upgrading them before the end of the year to 1.4. Two of the main reasons that I would do this are Variable Length DTMF and SIP Jitter Buffering. I would be very interested in hearing from anyone that is actually running 1.4 in a PRODUCTION environment, gatewaying SIP to TDM using Digium cards. To me, production means being able to have 3-4 PRI circuits maxed out for 12+ hours a day and 7+ call setups / second. Anything less than that is not really going to be an accurate comparison to what I have running. Anyone have any feedback about this combination? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by N2Net Mailshield, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Selecting OSLEC for zaptel-1.4.6
Hello list, Can someone outline the steps for selecting OSLEC canceller in 1.4.6? I know there was a bug fix for this but I can't figure out how to select it. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Telus (Alberta) PRI Caller ID NAME, Display IE, Facility ID
We are trying to send caller ID NAME information over a Telus PRI in Alberta. The PRI tech says that he sees the NAME information, and for calls over the same network, that NAME info should be reaching the receiving station, but it is not. The technician was stumped. I suspect there's something specific that I need to do to make it work, since many PBXs can do this. The switch is a Nortel DMS 100 in National ISDN 2 mode. I've put some 'pri intense debug' output below. Names and numbers have been changed to protect the innocent :) Is there anybody out there using a Sangoma A10X series card on a Telus PRI in Alberta, and do you have CID NAME working? Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 043 0: 0 N(R): 039 P: 0 90 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: Q.931 (8) len=90 Call Ref: len= 2 (reference 4/0x4) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [1c 1a 9f 8b 01 00 a1 14 02 01 04 02 01 00 80 0c 41 63 75 72 65 20 48 65 61 6c 74 68] Facility (len=28, codeset=0) [ 0x9f, 0x8b, 0x01, 0x00, 0xa1, 0x14, 0x02, 0x01, 0x04, 0x02, 0x01, 0x00, 0x80, 0x0c, 'Customer', 0x20, 'Health' ] [1e 02 80 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [28 0d b1 41 63 75 72 65 20 48 65 61 6c 74 68] Display (len=13) Charset: 31 [ Customer Name ] [6c 0c 21 80 34 30 33 35 33 39 35 37 39 37] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '403814' ] [70 0c a1 31 36 30 34 32 39 38 32 37 39 34] Called Number (len=14) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '16045552794' ] -- Called g0/16045552794 pbx*CLI [ 00 01 01 58 ] You can see that it's sending both Facility IE and Display IE name information. The technician was suggesting that sending both might be the problem. If so, I have no idea how to turn off the Display IE, and I solicit suggestions :) The rest of the PRI stuff is just call setup. Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 044 P/F: 0 0 bytes of data -- ACKing all packets from 42 to (but not including) 44 -- ACKing packet 43, new txqueue is -1 (-1 means empty) -- Since there was nothing left, stopping T200 counter -- Nothing left, starting T203 counter -- Restarting T203 counter pbx*CLI [ 02 01 4e 58 08 02 80 04 02 18 03 a9 83 81 ] pbx*CLI Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 N(S): 039 0: 0 N(R): 044 P: 0 10 bytes of data -- ACKing all packets from 43 to (but not including) 44 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 4/0x4) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] Sending Receiver Ready (40) [ 02 01 01 50 ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 040 P/F: 0 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter -- Zap/1-1 is proceeding passing it to SIP/121-082399e8 pbx*CLI [ 02 01 50 58 08 02 80 04 01 1e 02 80 88 ] pbx*CLI Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 N(S): 040 0: 0 N(R): 044 P: 0 9 bytes of data -- ACKing all packets from 43 to (but not including) 44 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 4/0x4) (Terminator) Message type: ALERTING (1) [1e 02 80 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0)
Re: [asterisk-users] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking
On 11/6/07, Hans Feringa [EMAIL PROTECTED] wrote: I understood that a timing device (ztdummy if no zaptel hardware is present) was not necessary anymore with linux kernel 2.6. When I enable iax2 trunking I get this warning chan_iax2.c:8908 build_user: Unable to support trunking on user 'xx' without zaptel timing The linux kernel is 2.6.22-14-386 Can I ignore this message, and is trunking working despite this warning? The ztdummy module is not part of the zaptel ubuntu package, so it cannot be loaded. I wanted to install from ubuntu packages for a change and not compile it from source. rgds, I believe that's OpenPBX that tries to derive its timing without Zaptel devices, however then you need to recompile your Kernel with 1000Hz timing as most use ~250Hz by default. Linux 2.6 + Ztdummy works fine and I'll take that over having to recompile the Kernel any day. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe CPU resources
Hello, First of all: also thanks to Doug Lytle and Steve Edwards. Just answering one time to all of you. I had the feeling that this computer, for 15 Meetme users, was more than enough... but we wanted to avoid any last-minute surprises! Now we are more sure that everything will work fine. Ah yes, we will use VoIP, without transcoding (I hope!), without Digium Timer card (but I will check, just in case we need it) On Nov/06/2007, Tony Mountifield wrote: In article [EMAIL PROTECTED], Carles Pina i Estany [EMAIL PROTECTED] wrote: It will depend on whether you are using VoIP or a PRI card. I have a number of systems that have a single Pentium 4 @ 2.8GHz (with HT), 1GB RAM and a 4xE1 PRI card (TE410P), and they regularly have conferences with up to 90 participants. I would expect them easily to handle the full Wow, 90 participants. Do you use just MeetMe in Asterisk? Just for curiosity: All of them can talk to conference? or only some of them? I thought about it, and for me, 90 open microphone participants looks like some white noise :-) Not tried here... just wondering how do you do. Thanks! -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dtmf / misdn
Hi all, Perhaps someone can give me a hint i the right direction... Sometimes dtmf is recognized, sometimes not. I'm using 1.2.19 asterisk with misdn for my hfc card. When i got in incoming sip-call, dtmf is recognized, When i phone my self (isdn-phone or gsm-phone) no problem with dtmf When SOME (not all) people phone me (isdn-incoming) DTMF is not recognized. How come? Either it works for a particular configuration, or it doesn't. It doesn't make sense to me that it works sometimes... Hans ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mystery phone!
We are in need of an IAX based hard phone. We have used softphones and USB headsets already and they are greatly affected by the other software running on the Windooz laptops and PCs of our users. Does anyone know where we can go to find IAX based hard phones in the US? The one on this link looks very nice. Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Administrator TOOTAI Sent: Tuesday, November 06, 2007 4:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Mystery phone! Kyle Sexton a écrit : Does anyone know who really makes this phone: http://www.hybsys.bg/Products/VoIP/IP/Phones/5000/ Not so mysterious: we import those phones in Europe ;-) POE, 5 accounts, SIP and IAX able, nice audio Good product. -- Daniel ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help: Asterisk info
And why are you asking in the Asterisk list? The absence of that file means you don't have any scsi adapters in your system. Lyle Jarga Jallow wrote: I am getting this error under system info: File Line Command Message common_functions.php 314 file_exists(/proc/scsi/scsi) the file does not exist on your machine Does anybody know how to fix this? Thank you in advance Jarga ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Extracting custom headers from SIP REFER
Asterisk 1.4.12
I wish to extract some custom headers from a SIP REFER message but am unable
to do so. However I can extract them from an INVITE. The code is:
exten = _.,n,Set(custom-id=${SIP_HEADER(custom-id)}) ;
exten = _.,n,Set(custom-valid=${SIP_HEADER(custom-valid)}) ;
Examples of the INVITE (works) and REFER (doesn't) messages are below.
U 147.202.001.001:5060 - 127.0.0.1:5065
INVITE sip:[EMAIL PROTECTED]:5065 SIP/2.0
Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK8b04.6e642c74.0
To: sip:[EMAIL PROTECTED]:5065
From: sip:[EMAIL PROTECTED];tag=119438778730084
CSeq: 1 INVITE
Call-ID: 119438778730084
Content-Length: 142
User-Agent: OpenSer (1.1.1-notls (i386/linux))
Contact: sip:[EMAIL PROTECTED]:5060
Custom-id: 1100012
Custom-valid: 24702670246
Content-Type: application/sdp
v=0
o=click-to-dial 0 0 IN IP4 0.0.0.0
s=session
c=IN IP4 0.0.0.0
t=0 0
m=audio 9 RTP/AVP 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
U 147.202.001.001:5060 - 147.202.001.001:5065
REFER sip:[EMAIL PROTECTED]:5065 SIP/2.0
Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK5b04.66fc0aa2.0
To: sip:[EMAIL PROTECTED]:5065;tag=as383b22fe
From: sip:[EMAIL PROTECTED];tag=119438778730084
CSeq: 2 REFER
Call-ID: 119438778730084
Content-Length: 0
User-Agent: OpenSer (1.1.1-notls (i386/linux))
Contact: sip:[EMAIL PROTECTED]:5060
Custom-id: 1100012
Custom-valid: 24702670246
Referred-By: sip:[EMAIL PROTECTED]
Refer-To: sip:[EMAIL PROTECTED]:5065
Is this a limitation of Asterisk or am I missing something?
Regards
Cameron
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Re: [asterisk-users] Mystery phone!
Kyle Sexton a écrit : Does anyone know who really makes this phone: http://www.hybsys.bg/Products/VoIP/IP/Phones/5000/ Not so mysterious: we import those phones in Europe ;-) POE, 5 accounts, SIP and IAX able, nice audio Good product. -- Daniel ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] use dial plan passed arg value in C agi code
Hi,
thanks for the replies although i am still confused.
my agi script is
exten = _.,1,AGI(simple_c_prgm|999);
exten = _.,2,NoOp(${MYAGIVAR});
Now i want to set the value of MYAGIVAR to 999 in my c program called
simple_c_prgm. This is what I am doing:
#include stdio.h
int main(int argc, char *argv[])
{
charline[80];
int i;
/* use line buffering */
setlinebuf(stdout);
setlinebuf(stderr);
while (1) {
/*gives me all the agi env var*/
fgets(line,80,stdin);
//prints the variables
printf(-- %s,line);
if (strlen(line) = 1) break;
}
*//according to me the passed variable should be available here but i am not
getting anything here
//nothing gets printed on the agi debug command
* for(i=0 ;i=argc ;i++)
{
printf(%s, argv[i]);
}
/* Send asterisk a command */
printf(SET VARIABLE AGIVAR %s,argv[1] );
/* Read response from Asterisk and show on console */
fgets(line,80,stdin);
fputs(line,stderr);
}
It would be great if someone could tell me what the problem is
Thanks
Arpit
On Nov 2, 2007 5:24 PM, Steve Edwards [EMAIL PROTECTED] wrote:
On Fri, 2 Nov 2007, James FitzGibbon wrote:
I handle this (in Perl) using Getopt::Long, which knows nothing of AGI's
stdin/stdout mechanics:
The AGI variables are passed via stdin (similar to an HTTP GET request)
and can be observed using agi debug at the Asterisk prompt. It's a fixed
list you get whether you want them or not. It's a common mistake for the
beginning AGI programmer to not read them (thus emptying stdin) and then
wonder why their AGI's don't work correctly.
Arguments are passed to the AGI as elements of argv[] (for C, my native
tongue). Thus, Getopt:Long (for Perl) or getopt_long() (for C) work
as expected. Using a comma to separate the arguments is an asteriskism.
Passing multiple arguments this way has worked for as long as I've been
using Asterisk -- several years.
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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--
Arpit Mehta
Graduate Student
Department of Computer Science
Columbia University
Tel: 1-646-387-5998
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Re: [asterisk-users] Telus (Alberta) PRI Caller ID NAME, Display IE, Facility ID
On 11/6/07, Stephen Bosch [EMAIL PROTECTED] wrote: It survives if it goes to a Telus customer, but not if it crosses over to Bell, Rogers, etc. Well -- here's where you can help me, because our name info is not even surviving on Telus' own network. I don't really care too much about Bell and Rogers, since Bell barely has a footprint out here and Rogers doesn't provide CNAM on its mobile network anyway (and nobody is using Rogers home phone ;) ). So -- if you had it working on Telus, what did you do? As soon as I turned on facilityenable=yes, outbound name display started to work for me. One tech claimed it was because I was sending calling name in addition to the IE, He probably meant the Display IE *and* the Facility IE. If you see my post it's what the technician I was working with suggested. Would be great if I knew a way of turning off the Display IE, if that's even possible/allowed. If it's not, then the don't send both idea is wrong. Probably correct, as that's what I'm sending now: [1c 15 9f 00 00 00 00 00 00 00 00 00 00 00 00 00 43 6f 6d 77 61 76 65] Facility (len=23, codeset=0) [ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 'Namehere'] [00 00 00 00 00 00 00 00 00 00 00] Display (len= 8) Charset: 31 [ Namehere ] I just checked dialing out of and back into my system, and the Facility IE comes back in the SETUP message, but the Display IE does not. Did you end up adding name records to the LIDB? For our purposes name display was a nice to have, so we didn't go down this path. Anyway -- again -- what did you do to get it working on Telus' network? Do you know what kind of switch you were connected to? To the best of my knowledge, we're connected to a 5ESS running NI-2. Here's the relevant zapata bits I use: facilityenable = yes pridialplan=unknown priindication=outofband overlapdial=no resetinterval=86400 echocancel=yes switchtype=national signalling=pri_cpe callerid=asreceived Hope that helps. -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Telus (Alberta) PRI Caller ID NAME, Display IE, Facility ID
Hi, James -- thanks for your comments. James FitzGibbon wrote: On 11/6/07, *Stephen Bosch* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: We are trying to send caller ID NAME information over a Telus PRI in Alberta. The PRI tech says that he sees the NAME information, and for calls over the same network, that NAME info should be reaching the receiving station, but it is not. I've had no end of trouble getting CNAM out of NI-2 PRIs with Telus. We're in Ontario, but the switch configs are the same across the country I believe. Well, that's not strictly true. Configurations in BC and Alberta are different, a legacy of the pre BCTel/Telus merger days. BCTel had a lot of GTE Automatic Electric equipment in its network. It survives if it goes to a Telus customer, but not if it crosses over to Bell, Rogers, etc. Well -- here's where you can help me, because our name info is not even surviving on Telus' own network. I don't really care too much about Bell and Rogers, since Bell barely has a footprint out here and Rogers doesn't provide CNAM on its mobile network anyway (and nobody is using Rogers home phone ;) ). So -- if you had it working on Telus, what did you do? One tech claimed it was because I was sending calling name in addition to the IE, He probably meant the Display IE *and* the Facility IE. If you see my post it's what the technician I was working with suggested. Would be great if I knew a way of turning off the Display IE, if that's even possible/allowed. If it's not, then the don't send both idea is wrong. while another claimed it was just a problem when the call passes from a NI-2 circuit to NI-1 (which some of the other carriers still use). Yeah, I've confirmed that this is an issue through a number of different people. So, no real solution for you, but at least you know it's not something obvious you're doing. I've tweaked my zaptel settings back and forth and tested with Telus on the phone to no avail. In the end, we deemed the effort to not be worth it. Did you end up adding name records to the LIDB? Anyway -- again -- what did you do to get it working on Telus' network? Do you know what kind of switch you were connected to? -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please explain the correct LED color for B410P
[EMAIL PROTECTED] wrote: Hi. I have installed B410P in Europe and the cards works more or less ok. My question is what color should the LED's on the back of the card be when connected to the PSTN NT box? Is there anywhere some information on the expected LED color in any given state (idle, call active, cord unplugged etc.)? On my card the lights are shining Red(orange-ish) but flashing to green every now and then and then shining green when there is a call on one of the lines for that port. That is correct. On zaptel-1.4/misdn-1.1.x you should see a blinking red when layer 2 is not up, constant red when layer 2 is up, and it will flash green when D-channel messages are sent on the port. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Help
At 13:25 11/6/2007, Jarga Jallow wrote: Content-class: urn:content-classes:message Content-Type: multipart/related; type=multipart/alternative; boundary=_=_NextPart_001_01C820AA.C8700D98 Under asterisk info: Sip registry 12/12 76.xxx.xxx.xxx D N 5066 UNREACHABLE 11/11 76.xxx.xxx.xxx D N 5064 UNREACHABLE 10/10 76.xxx.xxx.xxx D N 5062 UNREACHABLE All these IP phones are behind NAT. What could be the problem? sip show peers ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Telus (Alberta) PRI Caller ID NAME, Display IE, Facility ID
On 11/6/07, Stephen Bosch [EMAIL PROTECTED] wrote: We are trying to send caller ID NAME information over a Telus PRI in Alberta. The PRI tech says that he sees the NAME information, and for calls over the same network, that NAME info should be reaching the receiving station, but it is not. I've had no end of trouble getting CNAM out of NI-2 PRIs with Telus. We're in Ontario, but the switch configs are the same across the country I believe. It survives if it goes to a Telus customer, but not if it crosses over to Bell, Rogers, etc. One tech claimed it was because I was sending calling name in addition to the IE, while another claimed it was just a problem when the call passes from a NI-2 circuit to NI-1 (which some of the other carriers still use). So, no real solution for you, but at least you know it's not something obvious you're doing. I've tweaked my zaptel settings back and forth and tested with Telus on the phone to no avail. In the end, we deemed the effort to not be worth it. -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Selecting OSLEC for zaptel-1.4.6
Anciso, Roy wrote: Hello list, Can someone outline the steps for selecting OSLEC canceller in 1.4.6? I know there was a bug fix for this but I can't figure out how to select it. Thanks Roy Anciso You shouldn't need to. As long as you have applied the oslec-zaptel patch it should be selected automatically. You can double check it by looking in zconfig.h -Dave ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Selecting OSLEC for zaptel-1.4.6
Dave Fullerton wrote: Anciso, Roy wrote: Hello list, Can someone outline the steps for selecting OSLEC canceller in 1.4.6? I know there was a bug fix for this but I can't figure out how to select it. snip / Roy Anciso You shouldn't need to. As long as you have applied the oslec-zaptel patch it should be selected automatically. You can double check it by looking in zconfig.h I can confirm this - I just upgraded from 1.4.5.1 to 1.4.6 today and the zaptel-1.4.4 oslec patch applied cleanly to the zaptel source tree (with a bit of fuzz) and the build was good too. When the zaptel module loads (the from the init script) OSLEC is installed automatically. Alan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pickup Command not working
When I execute a pickup on a ringing phone I get CALL FAILED REASON CODE
603. I am dialing **212 with the following config. Anyone have a
suggestion?
EXTENSIONS.CONF
-snip-
[BLF_Group_Pickup]
; Defines how the extension to pick up a ringing phone in your BLF group
exten = _**XXX,1,Pickup(${EXTEN:2})
exten = _**XXX,n,Hangup()
[BLF]
; Defines a BLF Hint for phones
exten = 212,hint,SIP/sam
-snip-
SIP.CONF
-snip-
[sam]
type=friend
username=sam
fromuser=sam
callerid=sam
host=dynamic
dtmfmode=RFC2833
disallow=all
allow=ulaw
call-limit=20
subscribecontext=BLF
Thanks in advance for any help.
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Re: [asterisk-users] Selecting OSLEC for zaptel-1.4.6
Thanks I was trying to patch 1.4.6 using the 1.4.1.patch. The 1.4.4 patch did the trick:) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alan Lord Sent: Tuesday, November 06, 2007 4:53 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Selecting OSLEC for zaptel-1.4.6 Dave Fullerton wrote: Anciso, Roy wrote: Hello list, Can someone outline the steps for selecting OSLEC canceller in 1.4.6? I know there was a bug fix for this but I can't figure out how to select it. snip / Roy Anciso You shouldn't need to. As long as you have applied the oslec-zaptel patch it should be selected automatically. You can double check it by looking in zconfig.h I can confirm this - I just upgraded from 1.4.5.1 to 1.4.6 today and the zaptel-1.4.4 oslec patch applied cleanly to the zaptel source tree (with a bit of fuzz) and the build was good too. When the zaptel module loads (the from the init script) OSLEC is installed automatically. Alan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Everyone is busy/congested: IP Trunk
yeah i found openvpn helpful in NAT cases. -Vivek On 11/6/07, Baji Panchumarti [EMAIL PROTECTED] wrote: after a copious loss of follicles :-), I finally got outbound working. Basically the channel statement in the call file needs to have the number to be called. For eg., in test.call format the statement as follows : Channel: SIP/3012345678@your-sip-provider And there is no need for a DIAL statement in extensions.conf unless you need to dial an additional number / extension. Then in sip.conf you need a para that matches your-sip-provider with the relevant auth info. These two wiki pages, they were very helpful in figuring out a solution to the problem : http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out+deliver+message hth, -baji. -- On Oct 30, 2007 8:43 AM, Gabriel Natale wrote: I have the same problem. I trying with more 4 SIP providers, the account is registering, receive inboud calls, but can`t make outbound calls for congestion. Can be the out call id the problem? Thanks Gabriel ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma S200 and Digium TDM400P together
What's the result if you do cat /dev/zap ? On 11/6/07, Paulo Garcia [EMAIL PROTECTED] wrote: Hi, I have these two cards, the Sangoma has 4 fxo interfaces and the digium has 1 fxo and 1 fxs. After install the sangoma card, my zaptel.conf was configured for that card. I'm trying to configure the Digium one together thinking that the Digium ports should be 5 and 8 but it doesn't works. Someone has some example about this? Thanks in advance Pauçp ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma S200 and Digium TDM400P together
Sorry I mean ls /dev/zap On 11/6/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: What's the result if you do cat /dev/zap ? On 11/6/07, Paulo Garcia [EMAIL PROTECTED] wrote: Hi, I have these two cards, the Sangoma has 4 fxo interfaces and the digium has 1 fxo and 1 fxs. After install the sangoma card, my zaptel.conf was configured for that card. I'm trying to configure the Digium one together thinking that the Digium ports should be 5 and 8 but it doesn't works. Someone has some example about this? Thanks in advance Pauçp ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe CPU resources
Just remember if you don't have any Zaptel cards you are going to have to use ztdummy to run app_meetme. Ztdummy essentially requires Linux 2.6, which you should be using anyways. On 11/6/07, Carles Pina i Estany [EMAIL PROTECTED] wrote: Hello, First of all: also thanks to Doug Lytle and Steve Edwards. Just answering one time to all of you. I had the feeling that this computer, for 15 Meetme users, was more than enough... but we wanted to avoid any last-minute surprises! Now we are more sure that everything will work fine. Ah yes, we will use VoIP, without transcoding (I hope!), without Digium Timer card (but I will check, just in case we need it) On Nov/06/2007, Tony Mountifield wrote: In article [EMAIL PROTECTED], Carles Pina i Estany [EMAIL PROTECTED] wrote: It will depend on whether you are using VoIP or a PRI card. I have a number of systems that have a single Pentium 4 @ 2.8GHz (with HT), 1GB RAM and a 4xE1 PRI card (TE410P), and they regularly have conferences with up to 90 participants. I would expect them easily to handle the full Wow, 90 participants. Do you use just MeetMe in Asterisk? Just for curiosity: All of them can talk to conference? or only some of them? I thought about it, and for me, 90 open microphone participants looks like some white noise :-) Not tried here... just wondering how do you do. Thanks! -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] use dial plan passed arg value in C agi code
On Tue, 6 Nov 2007, Arpit Mehta wrote:
Hi,
thanks for the replies although i am still confused.
The first one is always the hardest :)
On Nov 2, 2007 5:24 PM, Steve Edwards [EMAIL PROTECTED] wrote:
The AGI variables are passed via stdin (similar to an HTTP GET request)
and can be observed using agi debug at the Asterisk prompt. It's a
fixed
--^
agi debug is your new best friend.
AGI Rx -- agi_accountcode:
AGI Tx 510 Invalid or unknown command
AGI Rx --
AGI Tx 510 Invalid or unknown command
AGI Rx am1000(null)SET VARIABLE AGIVAR 1000
AGI Tx 510 Invalid or unknown command
You are confusing Asterisk by printing to stdout. Don't do that. I use the
AGI command VERBOSE or syslog() when I want to see what's going on in an
AGI. When your AGI is totally hosed, syslog() is a better choice since it
doesn't interfere with the Asterisk/AGI control path (stdin and stdout).
Also, you are setting AGIVAR in your AGI, but displaying MYAGIVAR in your
dialplan.
I'm guessing you're fairly new to C. In the interest of speeding up your
learning process, let's go over your AGI in detail. My comments to your
code are preceded with .
#include stdio.h
#includestdlib.h
defines a bunch of cool stuff. We need it for EXIT_SUCCESS
further down.
#includesyslog.h
this will come in handy further down.
int main(int argc, char *argv[])
{
charline[80];
while this is sufficient for this agi, the agi set variable
command can be much longer. I didn't take the time to read the
code, but setting a variable with a value of 1,000 bytes worked
fine.
int i;
/* use line buffering */
setlinebuf(stdout);
setlinebuf(stderr);
// while (1) {
///*gives me all the agi env var*/
//fgets(line,80,stdin);
// // //prints the variables
//printf(-- %s,line);
//if (strlen(line) = 1) break;
// }
Aside from the printing that is confusing Asterisk, this would be
better as:
while (0 != (int)fgets(line, sizeof(line), stdin))
fgets returns a value -- use it. Also, note the sizeof()
instead of the constant 80. You already told the compiler how
big line was. Using 80 again only invites bugs when something
changes. sizeof() is an operator, not a function so there is no
additional overhead. Also, your intent is more obvious. Using the
sizeof() makes it obvious that the entire line is available to
fgets(). It would be valid (but obtuse) to limit fgets() to
some substring of line.
{
syslog(LOG_ERR, %s, line);
since you don't have the AGI stuff working yet, let's use syslog.
if ('\n' == *line)
if the first byte of line is a newline, we are done
{
break;
}
}
//according to me the passed variable should be available here but i
//am not getting anything here
//nothing gets printed on the agi debug command
// for(i=0 ;i=argc ;i++)
for (i = 0; i = argc; i++)
whitespace is free, use it in the right places.
{
// printf(%s, argv[i]);
syslog(LOG_ERR, %s, argv[i]);
since you don't have the AGI stuff working yet, let's use syslog.
}
/* Send asterisk a command */
don't mix commenting styles -- /**/ is so last century.
// printf(SET VARIABLE AGIVAR %s,argv[1] );
printf(SET VARIABLE AGIVAR %s\n, argv[1]);
printf does not append a newline for you so you have to
explicitly add it.
/* Read response from Asterisk and show on console */
// fgets(line,80,stdin);
fgets(line, sizeof(line), stdin);
same as above.
// fputs(line,stderr);
syslog(LOG_ERR, %s, line);
same as above.
return(EXIT_SUCCESS);
every non-void function (main() included) should end with a
return statement. While Asterisk currently ignores the return
value, return something meaningful. Maybe someday Asterisk will
use it and your AGI's will still work.
}
Strip out most of my comments and your code that I commented out and you
should have a better view of what's going on.
While this printf/fgets cycle works for getting a good understanding of
how the Asterisk Gateway Interface works, most people write a library (or
use someone else's) to hide the ugly details.
Also, it's a good idea for AGI's to trap SIGHUP and do something
appropriate. This is how Asterisk will tell your AGI that the caller has
hung up before your AGI is finished.
And, adding the compiler flags -Wall -Wstrict-prototypes
-Wno-unknown-pragmas will help keep you honest.
Good luck in your journey :)
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
Re: [asterisk-users] Mystery phone!
We also sell these phones and ship world wide www.voipperiod.com (See IP0027) Administrator TOOTAI wrote: Kyle Sexton a écrit : Does anyone know who really makes this phone: http://www.hybsys.bg/Products/VoIP/IP/Phones/5000/ Not so mysterious: we import those phones in Europe ;-) POE, 5 accounts, SIP and IAX able, nice audio Good product. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] wifi
I'd like to survey those on-list who actually use wifi SIP handsets. What type of wifi access point do you use? Are you happy with it? I presently use some older Linksys WAP54G APs. I'd like to replace these but in doing so I'd like to be moving in a VOIP friendly direction. I've yet to find a handset that I'd buy in quantity, but my last round of access points lasted 4 years so changing these now will merits the voip consideration. Thanks, Michael -- Michael Graves mgravesatmstvp.com o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dtmf / misdn
This may be what you need: http://www.misdn.org/index.php/FAQ#Why_are_my_dtmf_tones_not_detected_everytime.3F Also, something here may be helpful: http://www.voip-info.org/wiki/view/Asterisk+DTMF#Troubleshooting -jr On Nov 6, 2007 2:12 PM, Hans Witvliet [EMAIL PROTECTED] wrote: Hi all, Perhaps someone can give me a hint i the right direction... Sometimes dtmf is recognized, sometimes not. I'm using 1.2.19 asterisk with misdn for my hfc card. When i got in incoming sip-call, dtmf is recognized, When i phone my self (isdn-phone or gsm-phone) no problem with dtmf When SOME (not all) people phone me (isdn-incoming) DTMF is not recognized. How come? Either it works for a particular configuration, or it doesn't. It doesn't make sense to me that it works sometimes... -- Josh Richards - Grover Beach, California US [EMAIL PROTECTED] (don't forget the middle 't' initial when writing) http://blog.joshrichards.org/ 805/471-6923 (cell) Geek Research (Technology Management Consulting) - http://www.geekresearch.com/ Support These Nifty Causes: http://Kiva.org http://RoomToRead.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking
That was my (mis)understanding as well. It seems that it is currently not possible to compile the zaptel modules for a 2.6.22 linux kernel. For now I will not use the trunking option. Thanks, Hans Feringa On Tue, 2007-11-06 at 18:30 +0100, Hans Feringa wrote: I understood that a timing device (ztdummy if no zaptel hardware is present) was not necessary anymore with linux kernel 2.6. Not quite... this is commonly misunderstood, so let me clarify. Under the 2.6 kernel, ztdummy gets it timing directly from the kernel, and not from certain USB controllers like ztdummy does under the 2.4 kernel. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma S200 and Digium TDM400P together
On Tue, Nov 06, 2007 at 07:53:34PM -0500, [EMAIL PROTECTED] wrote: What's the result if you do cat /dev/zap ? You mean: cat /proc/zaptel/* But that still won't be good enough. Use genzaptelconf / zapconf included with latest versions of zaptel. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Selecting OSLEC for zaptel-1.4.6
On Tue, Nov 06, 2007 at 03:29:21PM -0500, Anciso, Roy wrote: Hello list, Can someone outline the steps for selecting OSLEC canceller in 1.4.6? I know there was a bug fix for this but I can't figure out how to select it. make ECHO_CAN_NAME=OSLEC (after you've applied the patch, that is) This is mostly useful for package builder at this point. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking
On Wed, Nov 07, 2007 at 07:59:55AM +0100, Hans Feringa wrote: That was my (mis)understanding as well. It seems that it is currently not possible to compile the zaptel modules for a 2.6.22 linux kernel. For now I will not use the trunking option. Zaptel sure can, if you use zaptel 1.4.6 / zaptel 1.2.21 . -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking
On Tue, Nov 06, 2007 at 05:16:41PM -0500, [EMAIL PROTECTED] wrote: I believe that's OpenPBX OpenPBX is a PBX software written in perl by VoiceTronix . I believe you refer to Callweaver. that tries to derive its timing without Zaptel devices, however then you need to recompile your Kernel with 1000Hz timing as most use ~250Hz by default. Linux 2.6 + Ztdummy works fine and I'll take that over having to recompile the Kernel any day. I'm not really sure if Callweaver has this limitation or not. But they did aim at using high-resolution timers from the Linux kernel. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] detecting voltage on fxo
hi is there any way to find out that an fxo module is connected to telco line or not? paradise dove ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
