search for: u100

Hi Everyone, We are using two Sangoma U100 (USB FXO) units connected to an Acer Aspire Revo (little PC running on Atom). The units work beautifully except for Monday :-) It maybe a conincedence or maybe the fact that Saturday/Sunday is off and something happens where one of these U100 modules goes into sleep and that's when all the 4 D...

Hi, I recently purchased an MSI Wind Netbook, just slightly bigger than an EeePC, and IMHO just one step above the mere toy category. It's got a 10" monitor, 80 GB SATA HD, and the keyboard is just big enough so I can type with ten fingers (with a little exercise). The thing came preinstalled with Windows XP, and I've spent the last two days figuring out how I could possibly

...as 3 incoming analog lines and less than 10 extensions. For the Asterisk server I am going to use a small form factor PC with no-PCI slots so the FXO interface needs to be either FXO->SIP or USB. Can anyone make suggestions? I am looking at an AudioCodes MP114 FXO or possibly two Sangoma U100's but don't have experience with either. ===== Eric Merkel ejmerkel.lists at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100813/2d390658/attachment.htm

...nclude => 'extensions' [pbx_config] [ Context 'extensions' created by 'pbx_config' ] '.' => 3. Hangup() [pbx_config] '0' => 1. Dial(SIP/jsantacapita|20|Tt) [pbx_config] 2. Voicemail(u100) [pbx_config] 102. Voicemail(b100) [pbx_config] '100' => 1. Dial(SIP/jsantacapita|20|Tt) [pbx_config] 2. Voicemail(u100) [pbx_config] 102. Voicemail(b100) [pbx_config] '101' => 1. Dial(SIP/mthoma...

...700688nnn,2,playback(pls-wait-connect-call) exten => 08700688nnn,3,Setvar(NewCaller=${CALLERIDNUM}) exten => 08700688nnn,4,SetCIDNum(0${CALLERIDNUM}) exten => 08700688nnn,5,Dial(${TRUNK}/07961106nnn,20,r) exten => 08700688nnn,6,SetCIDNum(${NewCaller}) exten => 08700688nnn,7,Voicemail(u100) exten => 08700688nnn,8,Hangup() exten => 08700688nnn,101,Voicemail(b100) exten => 08700688nnn,102,Hangup() (where nnn is a real number) The sip channel is set to time out quickly for testing. And I don't appear to have the pls-wait-connect-call audio file - but that isn't an iss...

I have one analog line hooked in my Asterisk box using an x100p (I think that's the model number). When I do this in my extensions.conf: exten => 1200,1,playback(pls-wait-connect-call) exten => 1200,2,Dial(Zap/1/5555551212,20,rTt) exten => 1200,3,VoiceMail(u100@lightwavetech.com) exten => 1200,4,Goto,t|1 The phone rings beyond the 20 second timeout and never really goes to the * voicemail. I can't seem to get it to timeout regardless of how many seconds I set it to. I assume this has something to do with the fact that * considers the call answere...

hello, I''m running XEN 2.0 on IBM ThinkPad T23. Now the weird thing is that I get two different outputs from /sbin/lspci depending on whether I run 2.6.8.1-xen0 or 2.6.8.1-bproc. In particular the output from 2.6.8.1-xen0 seems to be missing those 4 lines 0000:00:01.0 PCI bridge: Intel Corp. 82830 830 Chipset AGP Bridge (rev 02) 0000:00:1e.0 PCI bridge: Intel Corp. 82801BAM/CAM PCI

...Here are the respective sip.conf entries: [ata3] type=friend username=ata3 secret=asecret host=129.91.0.164 context=home [sjcata] type=friend username=sjcata secret=anothersecret host=129.91.0.161 context=home In my extensions.conf: exten => 83,1,Dial,SIP/sjcata|25 exten => 83,2,Voicemail,u100 exten => 83,3,Hangup exten => 84,1,Dial,SIP/ata3|25 exten => 84,2,Voicemail,u100 exten => 84,3,Hangup

hi, i am trying to do autoattendant but failing. as in the manual i inserted the background(welcome-mainmenu) file so that after the sound the caller can dial the extension he wants to call. i figured that the background sound wasn't coming in the asterisk. how do we do this without first loading the welcome message? for example after certain rings the caller can dial the extension no to

hello i'm on debian wine 1.3.37 and i try to use iriverplus4 for an iriver U100 i'm french and my english is not very good sorry. papa at debian:~/.wine/drive_c/windows/system32$ wine iriverplus4.exe err:ole:CoGetClassObject class {25baad81-3560-11d3-8471-00c04f79dbc0} not registered err:ole:CoGetClassObject no class object {25baad81-3560-11d3-8471-00c04f79dbc0} could be...

...1-1", "default") in new stack -- Executing Dial("Zap/1-1", "Zap/2|30") in new stack -- Called 2 -- Zap/2-1 is ringing -- Hungup 'Zap/2-1' == No one is available to answer at this time -- Executing VoiceMail2("Zap/1-1", "u100") in new stack -- Playing 'voicemail/default/100/unavail' WARNING[1192484144]: File chan_zap.c, Line 2853 (zt_handle_event): Ring/Off-hook in strange state 6 on channel 1 -- Playing 'vm-intro' -- Playing 'beep' -- x=0, open writing: /var/spool/asteris...

I have a small * system in my home (1 U100S, 1 X100P, 1 BT101, and 1 SPA2000) to handle my requirements. I would like to add Music on Hold and have been watching the forum to see if something would come across on this topic. The difference I am interested in is getting the music from a radio or someother external source. All references t...

...TMF sequence from a user and doing anything with it. Right now the pertinent DP features look like this: exten => s,1,Answer exten => s,2,SetMusicOnHold(default) exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,2 exten => s,5,Background(greeting) exten => 1,1,Voicemail(u100) ; Press 1 to leave a message. exten => 2,1,Voicemail(u6003) ; Press 2 to send an emergency page exten => t,1,Dial(SIP/person,30,t) ; Ring my extension on timeout Obviously extension 2 needs to be changed, right now it just leaves a message in my mailbox. I'm figuring I'll add a...

...s, or in any shared library) should show up in strace, but I am not that proficient with strace. Here you have some details. First, the asterisk traces: -- Registered SIP 'gonvaled' at 192.168.1.200 port 5060 expires 900 -- Executing VoiceMail("SIP/gonvaled-e0c6", "u100") in new stack Jul 2 11:53:32 WARNING[81926]: file.c:462 ast_openstream: File vm-theperson does not exist in any format Jul 2 11:53:32 WARNING[81926]: file.c:750 ast_streamfile: Unable to open vm-theperson (format ULAW): File exists == Spawn extension (from-sip, 100, 1) exited non-zero on...

...s context=pstn channel=>4 My /etc/asterisk/extensions.conf is [pstn] exten => s,1,NoOp(Comment Only: Call from ${CALLERIDNUM}) ; Just put a comment in the CLI for info. exten => s,2,Dial(Zap/g1,45,t) ;Dial the group=1 zap card mod above exten => s,3,Hangup ;exten => s,3,VoiceMail(u100) ;Whatever box you want. [internal] exten => i,1,Playback(invalid) exten => i,2,Hangup exten => t,1,Hangup exten => 099,1,Echo ;simple echo test when you dial 099 on your phone include => outgoing include => voip [outgoing] exten => _9X.,1,Dial(Zap/g2/${EXTEN:1}) e...

...l, I don't know if this is the right list, but here goes what happened recently to me. I have an IDE disk (hda: QUANTUM FIREBALLP AS20, ATA DISK drive / hda: 39851760 sectors (20404 MB) w/1902KiB Cache, CHS=2480/255/63, UDMA(100)) connected to 00:1f.1 IDE interface: Intel Corp. 82801BA IDE U100 (rev 12) I've decided to make a partition, so I've fired up fdisk and make a new one (hda7) of 4GB, as you can see in this line: /dev/hda7 1170 1679 4096543+ 83 Linux Then I've run mke2fs -j /dev/hda7, mounted it, copied data to it, umount it. I've not reboote...

I have a simple dial plan to cascade calls when the first phone does not answer: exten => 100,1,Dial(SIP/1000,10,tr) exten => 100,2,Dial(SIP/1000&SIP/1001,10,tr) exten => 100,3,Dial(SIP/1000&SIP/1001&SIP/1002,10,tr) exten => 100,4,Voicemail(u100) Problem is that the once the call goes onto the second and subsequent steps exten 1000 cannot answer the call. When the user picks up the phone it is just dead, no dial tone, nothing. Occasionally the handset will hang and need to be power-cycled. I've swapped out the phone, the power supply,...

Can anyone recommend some phone sets that are adsi compliant and work well with asterisk?

...oto(${ARG1},1) ; If they press #, return to start Here is what the log shows: -- Zap/1-1 is ringing -- Nobody picked up in 20000 ms -- Hungup 'Zap/1-1' -- Executing Ringing("Zap/49-1", "") in new stack -- Executing VoiceMail("Zap/49-1", "u100") in new stack -- Playing 'vm-theperson' (language 'en') May 2 18:36:45 WARNING[163851]: chan_zap.c:1193 zt_set_hook: zt hook failed: Device or resource busy -- Starting simple switch on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Playing 'digits/1'...

...ignorepat => 9 [globals] MYPHONENUMBER=12407440600 MYNAME=Chris HARIGA [incoming] exten => s,1,Answer() exten => s,1,Wait(0) exten => s,2,Dial(SIP/jim&SIP/jimoffice&SIP/sean&SIP/seanhome&SIP/chariga&SIP/nada&SIP/laurie&SIP/xten|40) exten => s,3,Voicemail,u100 [internal] ignorepat => 9 exten => toti,1,Dial(SIP/jim&SIP/jimoffice&SIP/sean&SIP/seanhome&SIP/chariga&SIP/nada&SIP/laurie&SIP/xten|40) exten => 0,1,Meetme,123 exten => _2.,1,SetCallerID(${MYPHONENUMBER}) exten => _2.,2,AbsoluteTimeout(6000) exten => _...