Re: [asterisk-users] List delays
There is definitely something wrong with this list. I have my emails sorted by date, and every day, the emails do not just come on top, but get slotted in. Today (10 July 2007), I received about 6 emails from 29th of June, couple from 30th, up until the 5th of July, nothing of today's, or, well, for the last 5 days. Admin, get your act together ! ;) -- This message has been scanned for viruses and dangerous content by Mail Call antivirus software, 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
Re: [asterisk-users] How many number of parallel calls can make through asterisk
Hi, You can see some sample configurations from the link below http://www.voip-info.org/wiki/index.php?page=Asterisk+hardware+recommendations It really depends on the hardware, the codecs, what you are going to use it for, etc,etc. I, for example, have a Pentium IV 2.4Ghz, 2Gb RAM, using G729 codec and putting through up to 15 simultaneous calls, with CPU load under 10%. Are you going to be connecting your Asterisk through phone lines or through a VoIP provider? If it is VoIP provider then you would need to consider your network path to the provider - bandwidth, latency, loads. Dimitri Santosh S Kumar wrote: Hi, We are planning to develop a product making asterisk as base, I love that asterisk is open source and eager to start working on it. But before even we get into start working on asterisk we want to know how many number of parallel calls can be made from a single asterisk box, considering we install the latest stable version of asterisk (we are ready to buy the enterprise version if there is any) on a highly configured box. So, how many number of parallel calls can we make through asterisk?? Regards, ___ --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 Mail Call antivirus software, 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
Re: [asterisk-users] Query
Hi, It looks like your configuration file zapata.conf syntax is wrong. Have a look in the sample files how to set it up correctly, and if you are still having troubles, paste your zapata.conf here. Cheers, Dimitri [EMAIL PROTECTED] wrote: Hi, I have put Digium TE120P card in PCI slot. So, lspci command gives the information in followimg format. 02:0a.0 Ethernet controller: Unknown device d161:0120 (rev 11) Following modules are running when seen through lspci wcte11xp 22304 - ztdynamic 9804 - ztdummy 3468 - ip_conntrack_irc6640 - ip_conntrack_ftp7312 - ipt_state 1864 - iptable_mangle 2696 - ipt_REJECT 5160 - ipt_LOG 6280 - ipt_multiport 2376 - ip_conntrack 47524 - iptable_filter 2856 - ipt_limit 2280 - ip_tables 18168 - wcte12xp 44352 - zaptel180036 - but on running asterisk -vvvgc it stops by printing the following errrors '###' at line 41 of /etc/asterisk/zapata.conf Jun 26 15:19:38 WARNING[1691]: config.c:497 process_text_line: Unknown directive '###' at line 43 of /etc/asterisk/zapata.conf Jun 26 15:19:38 NOTICE[1691]: chan_zap.c:10388 setup_zap: Ignoring unknown keyword 'group' in trunkgroups Jun 26 15:19:38 WARNING[1691]: chan_zap.c:1072 zt_open: Unable to specify channel 1: No such device or address Jun 26 15:19:38 ERROR[1691]: chan_zap.c:7050 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Jun 26 15:19:38 ERROR[1691]: chan_zap.c:10484 setup_zap: Unable to register channel '1-31' Jun 26 15:19:38 WARNING[1691]: loader.c:415 __load_resource: chan_zap.so: load_module failed, returning -1 Jun 26 15:19:38 WARNING[1691]: loader.c:555 load_modules: Loading module chan_zap.so failed! What is the problem actually can anybody tell me. Thanx and regards sanchal ___ --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 Mail Call antivirus software, 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
Re: [asterisk-users] Ring/Off-hook in strange state 6
Hi, I had a similar problem when I had signalling set at FXS_KS for my 4 FXS port TDM400P card. I've read long time ago that the signalling in Australia (where I am) is FXS_LS, so that solved that for me. Try different signalling methods, hopefully that will solve your problem. Apparently, the Digium tech support has a 2 week queue of requests. Cheers, Dimitri Alex Mcdowell wrote: Can anybody at least point me in a direction?? On 6/25/07, Alex Mcdowell [EMAIL PROTECTED] wrote: I don't think my cards are bad, but maybe there is a problem with the one. It has been two weeks since I put my ticket in with Digium...and still no word. I am starting to get frustrated. On 6/22/07, Daniel Hazelbaker [EMAIL PROTECTED] wrote: Alex, I had this problem with a new TDM2400 card that we purchased. Specifically I would get that message and then it would pick up the ringing line AND the line next to it. Basically, lines 1 2 had been cross-linked somehow. After a few weeks of trouble-shooting with Digium tech support they cross-shipped me a new card and the problem (and that message) went away. Daniel Hazelbaker High Desert Church On Jun 22, 2007, at 1:22 PM, Alex Mcdowell wrote: HI I have two servers both of which get this message on one of the lines. Ring/Off-hook in strange state 6. The one server seems to be ok with it, but the other one when an extension picks up there is no one there and the incoming call keeps ringing. I tried to adjust the levels in wcfxo.c like someone had suggested, but it didn't do anything. I also upgraded zaptel to the latest. 1.2.18 and asterisk is running at 1.2.14. callprogess is set to no, as well as busydetect=no. This is a major problem since this box only has 1 other line, but at least it works. I can't seem to find much info on this issue. I can't believe others haven't run into it. I started a ticket with digium, but I guess they are pretty backed up. Here is what I am getting in the CLI: Thanks for any help -Alex -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 Jun 22 13:44:45 WARNING[24296]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 is ringing Jun 22 13:44:46 WARNING[24296]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 answered Zap/4-1 == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 Jun 22 13:44:57 WARNING[24303]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 is ringing Jun 22 13:44:58 WARNING[24303]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 answered Zap/4-1 == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 -- SIP/4125-09559118 is ringing -- This message has been scanned for viruses and dangerous content by Mail Call antivirus software, 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
Re: [asterisk-users] Customized Ring Tone
You can use Queues. Put them in a queue and let them listen to music on
hold.
Cheers,
Dimitri
GNUbie wrote:
Hello all,
I'm running Asterisk 1.4.5 and Zaptel 1.4.3 on Debian Etch i386 with the
Digium's Dev Kit that comes with 1 FXO and 1 FXS. How do I configure my
home PBX in such a way that whenever someone calls on my trunkline (PSTN)
number, he/she will hear a customized ring tone, probably playing an MP3
file, instead of a boring standard ring tone while the extension
number that
is forwarded the call is still ringing? My current
/etc/asterisk/extensions.conf file looks like this:
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
[pstn]
exten = s,1,NoOp(Caller ID is ${CALLERID(num)})
exten = s,2,Dial(Zap/1,15,g2)
exten = s,n,Congestion
[local]
ignorepat = 9
exten = _9.,1,Dial(Zap/g1/${EXTEN:1})
exten = _9.,n,Congestion
exten = 11,1,Dial(Zap/1,20,rt)
Thank you in advance.
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