similar to: Random DTMF Tones Only on heard on ATA

I have an Asterisk server it runs great with SIP phones, soft SIP phones (twinkle) and a soft SIP phone app on my Android phone but I am having problems getting two ATA boxes working. I have a Linksys PAP2T, it is unlocked and I have used them before with no problems. I was able to receive calls with from any local SIP phone or from my Link2VoIP connection via the Internet but it could not call

I have a weird problem with a client. I recently upgraded to Asterisk 1.4.22 and Zaptel 1.4.12.1 on their server and now there is a problem when a fax call is received. Basically when faxdetect=incoming is set in zapata.conf the call comes in and the fax extension dials a Linksys PAP2T where a fax machine is connected. The fax answers and almos immediately it hangs up and I get this: --

I was having problems getting a Linksys PAP2T-NA to work with Pitney Bowes mailing station so it could use its modem to dial home and download postage/software updates. After scowering the web, I couldn't seem to find a definite how to article on what settings were needed. I finally came up some settings by combining the information from various places around the 'net. I have typed out

I am trying to integrate a Nortel Norstar system with an Asterisk service using a TDM04B card (4 fxo). So far everyting works from Asterisk to Nortel. The problem is when someone dials from the Nortel PBX to the Asterisk server. Asterisk answers the call and provides a dialtone (with DISA) but appartently the DTMF tones are not passed to asterisk and the call cannot proceed. This only

I'm considering a situation where I buy about twenty ATA devices. I've played with the Linksys / Cisco PAP2T, and got that working fine with some inbound and outbound faxing. The web GUI was okay. I'm seeing prices around $45 to $50 for this thing. It comes with two FXS ports, but I only need one FXS. I've seen the Grandstream Handytone 286 online. It looks promising as an

Hi, I'm trying to manage LXC instances through OpenStack, which use libvirt as a virtualization driver layer. After launching LXC instance, I simply could not attach to the console. virsh # list Id Name State ---------------------------------- 14366 instance-00000078 running virsh # console 14366 Connected to domain instance-00000078 Escape character is ^] And it keeps

Hi, I am looking for a good ATA recommendation, ideally something: 1) with one FXS and one LAN port (so it's as inexpensive as possible) 2) That can be provisioning using FTP (configuration and firmware upon reboot, ideally remote reboot from a sip notify) 3) Supports T.38 Nice to have would be: a) PoE powered and AC powered (my choice) b) Small size-wise I have been

Dear colleagues, This is not strictly a R question, but I hope it is ok to ask on the list. I fed a vector of p-values from about 20 million anova tests to the package q-value and obtained this output: > qsummary(asso_p.qvalue) Call: qvalue(p = asso_p.vec) pi0: 1 Cumulative number of significant calls: <1e-04 <0.001 <0.01 <0.025 <0.05 <0.1 <1

https://bugzilla.samba.org/show_bug.cgi?id=9744 Summary: Support Git, Mercurial, Subversion ignore lists Product: rsync Version: 3.0.9 Platform: All OS/Version: All Status: NEW Severity: normal Priority: P5 Component: core AssignedTo: wayned at samba.org ReportedBy: jglick at apache.org

Adam Jackson (1): Fix XTS regression in XCopyColormapAndFree Alan Coopersmith (1): Fix spelling/wording issues Alex Henrie (1): Handle small final sigma in XConvertCase Marko Myllynen (1): Update Finnish compose sequences for SFS 5966:2019 standard Matthieu Herrb (2): Change the data_len parameter of _XimAttributeToValue() to CARD16 libX11 1.6.10 Peter

Hello everybody, I have this problem where I can't get a ring tone when SIP devices dial an IAX2 route. I get the ring tone using IAX2 devices to dial the same route. The call completes normally in both cases... Facts: - Asterisk 1.0.9 - The Dial command is within an AGI. - ATA (grandstream) and firefly (SIP mode) would not give me the ring tone at all - Switching to a SIP route works ok -

CentOS Errata and Bugfix Advisory 2019:1888 Upstream details at : https://access.redhat.com/errata/RHBA-2019:1888 The following updated files have been uploaded and are currently syncing to the mirrors: ( sha256sum Filename ) x86_64: 0f909c7851cae3d7691e090e864ceede1ef0c3c9d9e01392075f46e28778ed83 ModemManager-1.6.10-3.el7_6.i686.rpm

When a Windows application crashes (a rare occurrence) I see a set of messages like these: wine: Unhandled page fault on read access to 0x429a3276 at address 0x4ba093 (thread 0009), starting debugger... err:seh:start_debugger Couldn't start debugger ("debugger/winedbg 8 9744") (2) Read the Wine Developers Guide on how to set up winedbg or another debugger I have not been able to

You need to add dom0_mem=XXXm to xen or scale back the memory of domain0 with using "xm mem-set <DomId> <Mem>" command. Your domain0 used up almost all physical mem. >total_memory : 1023 >free_memory : 62 Thanks Yunfeng >-----Original Message----- >From: xen-devel-bounces@lists.xensource.com

Hi everyone, I have a dataset of friendship with this format: ego alter 4746 1 2 9742 1 3 14738 1 NA 4747 2 NA 9743 2 3 14739 2 1 4748 3 13 9744 3 5 14740 3 14 4749 4 NA 9745 4 NA 14741 4 NA 4750 5 NA 9746 5 13 14742 5 10 4751 6 12 9747 6 7 ... NA means that individuals don't select any

Hello, Over the last month or so, I've noticed the following error showing up repeatedly in my system logs under kernel 2.4.18-ac3 and more recently under 2.4.19-rc1: EXT3-fs error (device ide0(3,3)) in ext3_new_inode: error 28 I've now been able to capture the following Oops before the system went down entirely: Assertion failure in do_get_write_access() at transaction.c:611:

Hello. I'm looking to buy a FXO card to do some testing with two phone lines I have at home and was looking in ebay some and found some cheap ones but, the I've never heard of the brand or manufacturer: chinaroby. They run for about $99 plus shipping. Have any one used these? or please recommend one... Money IS an issue. Thanks.

Hi, I compiled * and chan_alsa.so is loaded. But I can't hear any busy signal messages when calls cannot connect. Do I need to record my own message, or does * use some default ones? May I ask where can I find them? Regards, Norman Zhang

When I use AGI's Record File with DTMF termination, you can hear a snippet of DTMF at the end of the message. Actually, you hear DTMF at the beginning too, but I work around that by streaming from a few hundred milliseconds into the file. However, I haven't been able to come up with a way around hearing it at the end, and the customer is complaining. If there's no workaround,

Hi all I complile asterisk 1.2.4 successfully.I install festival successfully and i configure asterisk to work with festival.But When i call the festival extension configured in extensions, the festival application is executed well (i see it in the log) and must read the text (hello world).But i'm hear no voice. What's the problem? Thanks