Re: [asterisk-users] Outbound T.38 via RTP with pjsip does not work as expected

On 04/06/2017 at 08:33 PM, Joshua Colp wrote: > On Thu, Apr 6, 2017, at 03:15 PM, Michael Maier wrote: >> Hello! >> >> I'm trying to send a fax via T.38 to a destination, which should be T.38 >> capable. My provider supports T.38, too. Unfortunately, it doesn't work. >> This means: >> >> Call is

Re: [asterisk-users] restart system from extension

Please don't top post. On Thu, Apr 06, 2017 at 08:16:34PM +0300, Atux Atux wrote: hi. i would like to be able to reboot the system from my extension. is that possible? if yes, how? Στις 6 Απρ 2017 8:25 μ.μ., ο χρήστης "Tzafrir Cohen" έγραψε: System('sudo

Re: [asterisk-users] Outbound T.38 via RTP with pjsip does not work as expected

On Thu, Apr 6, 2017, at 03:15 PM, Michael Maier wrote: > Hello! > > I'm trying to send a fax via T.38 to a destination, which should be T.38 > capable. My provider supports T.38, too. Unfortunately, it doesn't work. > This means: > > Call is started and SDP is negotiated w/ alaw. Callee sends

[asterisk-users] Outbound T.38 via RTP with pjsip does not work as expected

Hello! I'm trying to send a fax via T.38 to a destination, which should be T.38 capable. My provider supports T.38, too. Unfortunately, it doesn't work. This means: Call is started and SDP is negotiated w/ alaw. Callee sends reinvite - for alaw again (and not for T.38)!! After about 30s, callee

Re: [asterisk-users] restart system from extension

Could you give some more details please? Στις 6 Απρ 2017 8:25 μ.μ., ο χρήστης "Tzafrir Cohen" < tzafrir.co...@xorcom.com> έγραψε: > On Thu, Apr 06, 2017 at 08:16:34PM +0300, Atux Atux wrote: > > hi. i would like to be able to reboot the system from my extension. is > that > > possible? if yes,

Re: [asterisk-users] Define SIP fromuser field in Dial()-command

Hello in what way does this set the 'fromuser' field in the SIP INVITE ? Kind regards. J. On 05-04-17 22:05, Pete Mundy wrote: Hi Jonas Does the information at this link help? http://the-asterisk-book.com/1.6/funktionen-callerid.html Pete On 5/04/2017, at 8:11 pm, Jonas Kellens

Re: [asterisk-users] Commit dialplan & other config. in memory to disk?

On Thu, 6 Apr 2017, Nathan Anderson wrote: In the future, if I were to ever run into a similar situation, is there any way to request or instruct Asterisk to write the current dialplan that is in memory and other important config files (e.g., users.conf) to disk in a *different* location than

Re: [asterisk-users] restart system from extension

On Thu, Apr 06, 2017 at 08:16:34PM +0300, Atux Atux wrote: > hi. i would like to be able to reboot the system from my extension. is that > possible? if yes, how? System('sudo /sbin/reboot') You need to allow that in a sudoers file, of course. This may or may not be a good idea. There are a host

[asterisk-users] restart system from extension

hi. i would like to be able to reboot the system from my extension. is that possible? if yes, how? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at:

Re: [asterisk-users] feature codes

hi. thanks for the reply. when you say configure them in features.conf? On Thu, Apr 6, 2017 at 11:57 AM, Marcelo Terres wrote: > You can configure the features in the features.conf file, but some > features like DND and call forward are not available, so, or you use > the

Re: [asterisk-users] Commit dialplan & other config. in memory to disk?

On Thu, Apr 06, 2017 at 09:54:25AM +, Nathan Anderson wrote: > 'lo, > > So yesterday, one of our clients had the misfortune of having the disk that > their Asterisk config (*.conf) was stored on take a dirt nap. Of course, > Asterisk was still running at the time, and everything continued

Re: [asterisk-users] Issues with Siren14 codec in Asterisk 14.3.0

On Thu, Apr 6, 2017, at 12:17 PM, Richard Kenner wrote: > > I would say this is a bug in func_speex and not in codec_siren14. This > > is because the datalen is zero. > > Ah! So, like? > > *** func_speex.c.orig 2017-02-13 15:00:19.0 -0500 > --- func_speex.c2017-04-06

Re: [asterisk-users] Issues with Siren14 codec in Asterisk 14.3.0

> I would say this is a bug in func_speex and not in codec_siren14. This > is because the datalen is zero. Ah! So, like? *** func_speex.c.orig 2017-02-13 15:00:19.0 -0500 --- func_speex.c2017-04-06 11:16:03.0 -0400 *** *** 185,189 } !

Re: [asterisk-users] Issues with Siren14 codec in Asterisk 14.3.0

On Thu, Apr 6, 2017, at 10:57 AM, Richard Kenner wrote: > I'm seeing Asterisk crashes with the following frame at func_speex.c:188: > > (gdb) p *frame > $6 = {frametype = AST_FRAME_VOICE, subclass = {integer = 0, > format = 0xe2f9e20, frame_ending = 0}, datalen = 0, samples = 640, >

[asterisk-users] Issues with Siren14 codec in Asterisk 14.3.0

I'm seeing Asterisk crashes with the following frame at func_speex.c:188: (gdb) p *frame $6 = {frametype = AST_FRAME_VOICE, subclass = {integer = 0, format = 0xe2f9e20, frame_ending = 0}, datalen = 0, samples = 640, mallocd = 1, mallocd_hdr_len = 232, offset = 64, src = 0x2ac07413e7f8

Re: [asterisk-users] Commit dialplan & other config. in memory to disk?

Nathan Anderson wrote: 'lo, So yesterday, one of our clients had the misfortune of having the disk that their Asterisk config (*.conf) was stored on take a dirt nap. Of course, Asterisk was still running at the time, and everything continued to work (except for voicemail, which was stored

Re: [asterisk-users] Commit dialplan & other config. in memory to disk?

You can execute something like asterisk -rx "dialplan show" > some_file.conf, but unfortunately the result cannot be directly parsed by Asterisk. Still it will give you a readable snapshot of your current dialplan. Le 06/04/2017 à 11:54, Nathan Anderson a écrit : 'lo, So yesterday, one of

[asterisk-users] Commit dialplan & other config. in memory to disk?

'lo, So yesterday, one of our clients had the misfortune of having the disk that their Asterisk config (*.conf) was stored on take a dirt nap. Of course, Asterisk was still running at the time, and everything continued to work (except for voicemail, which was stored on the same disk) right up

Re: [asterisk-users] feature codes

You can configure the features in the features.conf file, but some features like DND and call forward are not available, so, or you use the SIP client own functionalities for that (if available), or you will have to develop your own features. Regards, Marcelo H. Terres IM:

[asterisk-users] feature codes

hi. i am running asterisk 11 and i am stuck with the feature codes. how do i setup them. Now the system has. PBX*CLI> features show Builtin Feature Default Current --- --- --- Pickup *8 *8 Blind Transfer # # Attended Transfer One Touch Monitor Disconnect Call * * Park Call