similar to: 1.6.0.18-rc3: SendFAX causes restart

I'm trying to connect a sip call from sipgate to Asterisk A to Asterisk B. Both are behind NAT, but port forwarded. I get the connection, but no voice - either in or out. I can call on SIP from A to B (and from B to A). Do it all the time. Asterisk A receives SIP calls from Junction and Teliax. CLI on A looks right: == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 ==

I have two internal analogue extensions off a TDM400P. If the first is busy, I'd like to ring the second. So: [incoming] exten =>s,1,Answer() exten =>s,n,Dial(${mainline},60) exten =>s,n,ExecIf($["${DIALSTATUS}" = "BUSY"]?Dial(${secondline},30)) But it doesn't work because * first tries Call Waiting on the main line. Here I dial out: -- Starting

Upgraded from 16.2.14 to 1.6.2.15 on Fedora 13, with aastra 9133i and 57i. On 9133i and 57i: #<extension># works for a blind transfer. Xfer<extension>Xfer doesn't! All this worked on 1.6.2.14. Nothing useful on cli, verbose 3, DEBUG. Here extension 169 answers an outside call, and tries to transfer it to 145 using the Xfer button: -- SIP/169-0000009c answered

DEAR: I SEEM A BUG FOR speex-1.2rc1.tar.gz ON BUILD IN USING IT. PLEASE LOOK: ===================================================================== make all-recursive make[1]: Entering directory `/tmp/speex-1.2rc1' Making all in libspeex make[2]: Entering directory `/tmp/speex-1.2rc1/libspeex' /bin/sh ../libtool --mode=link gcc -g -O2 -fvisibility=hidden -O3 -msse -o testenc

Hi,I am currently doing a project in which we are to investigate the size and power of three different one sample tests over three different distributions using a number of different sample sizes and values for mu1. I have written a function and was trying to get my answer for each test into the right position in an array so the output is the power of each combination of test, distribution, sample

I'm running Samba 4.0.9 as a domain controller in a Windows 2008 R2 functional level single domain forest with a Windows 2012 domain controller. I am using Bind 9.9.3-P2 as the DNS backend with dlz_bind9_9.so driver. When I add my Samba DC to the DNS Manager snap-in, my main domain.com zone has a red X and I cannot expand it. The _msdcs.domain.com zone seems to be ok. Also, one of my reverse

The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.27, 1.6.0.18, and 1.6.1.10. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases resolve a large assortment of issues reported by the community. For a summary of the changes in these releases, please see the release summaries:

The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.27, 1.6.0.18, and 1.6.1.10. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases resolve a large assortment of issues reported by the community. For a summary of the changes in these releases, please see the release summaries:

Hi, I'm having an issue with my Thinkpad P70 laptop/workstation. This system is a dual boot, windows 10 pro and centos 7. I have not needed to use the cdrom until now, however the system does not see it in /dev. I know the cdrom works as I can use it in windows 10 and when initially installing centos 7, I installed from dvd media. I'm not sure what to check. I do not see anything related

Hello all, I recently bought a brand new Dell Precision 7540 laptop.? Before buying I checked the specifications from the RedHat hardware list and they said that it was "fully RedHat 8 Certified" and listed all the hardware that was "certified" to work with that laptop. https://access.redhat.com/ecosystem/hardware/3979771 I have had good results the last few years with

On Nov 30, 2007 6:35 PM, Kingsly John <member+centos at kingsly.net> wrote: > > What you have isn't a "real" modem .. it's just a winmodem that is usable > under linux. "real" modems don't need drivers to work. > > 11/30 00:31:52 dem mdm_command: string 'AT+FCLASS=2.0' > 11/30 00:31:52 dem mdm_command: string 'ERROR' ->

On Dec 1, 2007 5:23 PM, Robert <kerplop at sbcglobal.net> wrote: > > All the more reason to quit wasting your time with that junk and go shopping for a USRobotics Courier v.everything -- external if you have a serial port available. I notice that there are several on ebay right now. If it has no manual included, download one from the mfr. > Lesson learned - I just hate to give

Hello, after I solved my problem with the fax processing after receiving, I got another problem while sending a fax: the header is not set properly. I use a PHP_Script to upload a PDF file and to generate a call file. A bash script is looking for existent call files in the web directory and moves them the asterisk's outgoing directory. Ok, my call file looks like this: ====== $cf_commands

Hi, On a new Raspberry2 with Asterisk 11.13.1 binary-installed, ReceiveFAX exits with FAILED status. Each attempt is made with something like: channel originate Local/12345 at faxgateway application SendFAX myfile.tif I'm not sure if failure "mostly" comes from SendFAX or from ReceiveFAX but it fails with 100% rate. uname -a output is: Linux myhost 3.18.7-v7+ #755 SMP PREEMPT

I followed the Faxing.txt that comes with samba and my MS Word docs print just fine but are not spooled automatically by faxspool. Does anyone have this working with samba? Thanks! Alex. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 1272 bytes Desc: not available Url :

Hi everybody, I've installed Free Fax For Asterisk in my Asterisk box but I don't understand how it works as when using SendFax application from dialplan, I can't find how to introduce destination fax number. How this application works? Do I have to call destination fax using Dial application, detect somehow that it's a fax and then use SendFAX application specifying FAXOPTs and

Using Digium fax I've tried a simple dialplan: '8447' => 1. Answer() [pbx_config] 2. Set(CALLERID(num)=xxxyyyzzzz) [pbx_config] 3. Dial(DAHDI/g0/1bbbcccdddd,,G(send)) [pbx_config] [send] 4. SendFax(/var/spool/asterisk/fax/20090922_1301.tif) [pbx_config] 5. HangUp() But I doesn't work. It executes

Hello, I'm trying to send a tif file, using Fax for Asterisk and the call is executed, but when I get the reINVITE with T.38 data, the local server doesn't recognize that we have this capability and sends a 488 message. These are the logs: <--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 ---> INVITE sip:1234567 at 10.0.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP

Hi, For the project I've used the Eicon DIVA card. It has 8 BRI ports, and for about 25% of the time there are 7 or 8 in use. So we want to replace it with an E1 card. Only issue is, replace it with what? The idea we have been playing with was to get a Digium E1 card (we already have bought lot of Quad E1 cards :-) and then just put it back to back against Asterisk server. And instead of

[big snip] > There is a sysfs attribute called "d3cold_allowed" which can be used > for "blocking" D3cold, so can you please retest using that? > Hey-this is almost certainly not the right place in this thread to respond, but this thread has gotten so deep evolution can't push the subject further to the right, heh. So I'll just respond here. I've been