Re: [asterisk-users] dialplan reloading

Hi Jerry, From the Asterisk CLI, enter the command core restart when convenient, this command will restart asterisk only when there is no incoming call, and when it will close all outgoing calls. With a restart of asterisk should reload all the information: extensions, sip, agi, iax,

[asterisk-users] Unable to create channel of type 'DAHDI' (cause 17 - User busy)

Hi, I have 6 Red FXO with TDM2400p in my PC. I have install asterisk and dahdi driver. Scenario is jitsi- asterisk server- analog PBX landline phone I configured this scenario as follow in chan_dahdi.conf file ; General options [channels] usecallerid=yes

Re: [asterisk-users] dialplan reloading

Hi Jerry, From the Asterisk CLI, enter the command core restart when convenient, this command will restart asterisk only when there is no incoming call, and when it will close all outgoing calls. With a restart of asterisk should reload all the information: extensions, sip, agi, iax, voicemail,

Re: [asterisk-users] dialplan reloading

On Fri, 2012-11-02 at 06:25 -0400, Jerry Geis wrote: Hi Jerry, From the Asterisk CLI, enter the command core restart when convenient, this command will restart asterisk only when there is no incoming call, and when it will close all outgoing calls. With a restart of asterisk should

[asterisk-users] Asterisk with R2D configuration

Hi, Has anybody worked on R2D Brazillian setup. I have configured R2 using OpenR2 with Asterisk. While doing some analysis I found R2D is already included in libopenr2. Have anyone tested the same. Regards, Gopal. -- _ --

Re: [asterisk-users] dialplan reloading

Sorry to step in here but I think the 2 of you are talking at cropp purposes I initial query was about a dialplan reload, not an asterisk restart. Jerry, how long does your system take to perform a dialplan reload? surely it is under a second. If you look in the logs, at the end of any

Re: [asterisk-users] dialplan reloading

Jerry Geis wrote: Actually my mistake - looks like based on my code certain things happen and I issue two dialplan reload commands. So the second is killing the first. Then asterisk looses information. So certainly I should not be doing that - but I'm surprised asterisk lets another reload

Re: [asterisk-users] dialplan reloading

What version of Asterisk are you running? There was an issue found in February where this exact behavior could occur, two dialplan reload commands would clobber each other. It was also resolved back then in all supported branches (1.8, 10, and trunk).

Re: [asterisk-users] Different codec for different type of calls

Qasim, Thank you for your response. I tried it but still doesn't work. This is what I have: exten = _XXX.,1,NoOP(Set G711 codec) exten = _XXX.,n,Set(SIP_CODEC=ulaw) exten = _XXX.,n,Set(SIP_CODEC_OUTBOUND=ulaw) exten = _XXX.,n,Dial(DAHDI/g1/$EXTEN) Then I get this error: WARNING[12156]:

Re: [asterisk-users] Different codec for different type of calls

SIP_CODEC is only useable on a SIP channel. You can specify DAHDI codecs in users.conf. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ali Pey Sent: Friday, November 02, 2012 12:28 PM To: Asterisk Users Mailing List - Non-Commercial

[asterisk-users] Outgoing Google Motif Calls connect but continue ringing on outgoing side

I upgraded from Asterisk 10 to 11 and switched from gtalk.conf and jabber.conf to use motif.conf and xmpp.conf. I disabled gtalk and jabber from loading in modules.conf noload = res_jabber.so noload = chan_gtalk.so After copying my settings to the new conf files and restarting Asterisk I had

Re: [asterisk-users] Outgoing Google Motif Calls connect but continue ringing on outgoing side

Roy Abshire wrote: I upgraded from Asterisk 10 to 11 and switched from gtalk.conf and jabber.conf to use motif.conf and xmpp.conf. I disabled gtalk and jabber from loading in modules.conf noload = res_jabber.so noload = chan_gtalk.so After copying my settings to the new conf files and

Re: [asterisk-users] Outgoing Google Motif Calls connect but continue ringing on outgoing side

Roy Abshire wrote: I copied my settings over, and looked at the guide over and over to change the settings. But what the guide doesn't tell you is what you don't need anymore. So I didn't know if /Talk was ok or needed to be omitted and externip or bindaddr was ok still because I had to have it

Re: [asterisk-users] Outgoing Google Motif Calls connect but continue ringing on outgoing side

Roy Abshire wrote: I do have one thing I'm really unsure about. I'm using my Google Account for Asterisk and I'm also logged into it from my Desktop Computer. Am I not supposed to be logged into this account and strictly use it for the Asterisk Server only? Does Asterisk have a problem knowing

[asterisk-users] PRI got event HDLC Abort

hi folks. recently some of our customers complained about bad voice quality on the phone system. i looked at the logs and found a lot of these: [2012-11-03 08:26:38] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 [2012-11-03 08:26:45] NOTICE[11305]

Re: [asterisk-users] PRI got event HDLC Abort

is there static on the line?? is there timing slips and crc4 errors? are they increasing throughout the day? are you getting timing slips during the day when users are using the phones and not off-peak hours? are you getting hdlc abort erros when you hear a static noises?? is the card sharing