Re: [asterisk-users] Asterisk not starting (illegal instruction core dumped)
On Tuesday 27 November 2012, Adolphus Enaboifo wrote: Hi List members, Thanks for the support so far as I try to install and test my first asterisk system. I was able to finally install asterisk-1.8.18.0 with libpri-1.4.13 and dahdi-linux-complete-2.6.1+2.6.1 according to the instructions given in the online documentation (asterisk the definitive guide). But while trying to start asterisk with the following command /usr/sbin/asterisk -cvvv or /usr/sbin/asterisk -c I get the message Illegal instruction (core dumped) Kindly advice on what to do. I've had this trying to install Asterisk on a machine with a VIA processor. Turned out to be VIA's fault for pretending to be an i686 when in fact it only supported a subset of i686 instructions (and Asterisk wanted to use one of the ones that it didn't). The cure was to make distclean and re-compile for an i586 target. Please post the contents of /proc/cpuinfo on the machine which is failing to run Asterisk. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] callerid not received from dahdi
Hi,
my scenario is below
analog phone (10 to 99)-- pbx--(77)asterisk jitsi(2000)
i have analog telephone interface numbered 77 attached with asterisk and
other sip user is 2000 on jitsi.
I can call from any number from 10 to 99(in intercom) on 77 and ivr
response will come then i can typed 2000# and call go to 2000 named user
in asterisk.
Now my problem is when i am calling from 10 to 99 (any number) this number
should display to sip 2000's user. But its not showing to user. Its shows
asterisk@my_asterisk_server_ip.
my config. as follow
extension.conf
exten = s,1,Goto(phrase-menu,s,1)
[phrase-menu]
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Read(PHRASEID,/var/lib/asterisk/sounds/custom/soip)
exten = s,4,Wait(2)
exten = s,5,Set(CALLERID(num,CID)=${CALLERID})
exten = s,6,Dial(SIP/${PHRASEID},40,tT)
exten = h,1,Hangup()
and in chan_dahdi.conf
; General options
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
cidsignalling=dtmf
cidstart=polarity
callerid=asreceived
rxgain=0.0
txgain=0.0
;FXO Modules
group=1
echocancel=yes
signalling=fxs_ks
context=default
channel=1-20
#include dahdi-channels.conf
any help
thanks..
Do not bother about below message. That is auto-generated by my mail
server.
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Harish Mandowara
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[asterisk-users] Asterisk connected to TalkMaster from Digital Accoustics
Does anyone have information or successfully connected Asterisk to TalkMaster from Digital Accoustics? Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need qualifications of SIP trunk providers
Thank you Carlos, What does mean 'por-out'? I'm expecting 1 min/month in out. Elder On Thu, Nov 29, 2012 at 5:50 PM, Carlos Alvarez car...@televolve.comwrote: On Thu, Nov 29, 2012 at 3:22 PM, Daniel - Asterisk earohua...@gmail.comwrote: Hello List, Since I'm looking for a new VoIP provider for US origination/termination, I will very appreciate if you can chare your experience with Flowroute, Vitelity and Voip.ms Vitelity is reliable and decent, but no phone support. Have not used the others. Oh also if you lose a number on Vitelity to a port-out, they won't know and won't stop billing you for it. What's your expected volume in/out? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need qualifications of SIP trunk providers
On Fri, Nov 30, 2012 at 7:10 AM, Daniel - Asterisk earohua...@gmail.comwrote: Thank you Carlos, What does mean 'por-out'? I'm expecting 1 min/month in out. Port out means a number was ported to another carrier. 10k minutes is not huge but a decent number that should get you a reasonable rate with the carrier of your choice. Vitelity is a good fit for that size. I can't say they are better than the others because I haven't used them, but we had a few hundred DIDs and probably did 50k minutes with them at one point. As we've grown we have moved to others (we're around half a million min/mo now). -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need qualifications of SIP trunk providers
Daniel - Asterisk wrote: Thank you Carlos, What does mean 'por-out'? I'm expecting 1 min/month in out. Elder PORT out = Port or move the number away from a provider Seems this provider is unaware that one may have moved the number to another provider, and continues to charge when they no longer have the number. One has to wait until the port is complete and successful to cancel the previous providers account. It seems the user needs to cancel the account with the mentioned provider John Novack On Thu, Nov 29, 2012 at 5:50 PM, Carlos Alvarez car...@televolve.com mailto:car...@televolve.com wrote: On Thu, Nov 29, 2012 at 3:22 PM, Daniel - Asterisk earohua...@gmail.com mailto:earohua...@gmail.com wrote: Hello List, Since I'm looking for a new VoIP provider for US origination/termination, I will very appreciate if you can chare your experience with Flowroute, Vitelity and Voip.ms Vitelity is reliable and decent, but no phone support. Have not used the others. Oh also if you lose a number on Vitelity to a port-out, they won't know and won't stop billing you for it. What's your expected volume in/out? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] callerid not received from dahdi
On Fri, Nov 30, 2012 at 04:54:28PM +0530, Harish Mandowara wrote: Do not bother about below message. That is auto-generated by my mail server. [snip] --- This e-mail is for the sole use of the intended recipient(s) and may contain confidential and privileged information. If you are not the intended recipient, please contact the sender by reply e-mail and destroy all copies and the original message. Any unauthorized review, use, disclosure, dissemination, forwarding, printing or copying of this email is strictly prohibited and appropriate legal action will be taken. --- I realize this probably seems silly, but I do not think it's in my best interest to ignore threats of appropriate legal action for forwarding this email to someone who might be able to help or archiving on a message board, etc.. Do you think you could talk to the people who manage your mail server and have the disclaimer removed? They may be interested in: http://www.goldmark.org/jeff/stupid-disclaimers -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] why number type always changed from subscriber user to national in libpri
I used libpri 1.4.12 version with asterisk 1.8.7, after the pcap files, i found that in wireshark setup message, the number type always changed from subscriber to national number. i have set pridialplan= local and prilocaldialplan=local in chan_dahdi.conf already. because that, the system sometimes can not make outgoing calls. anyone can clarify that? Use of Asterisk v1.8.7 with libpri is *not* recommended. That version of Asterisk has a regression in its ./configure script that does not setup Asterisk to use libpri correctly. Also, Asterisk v1.8.7 is quite old now. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] callerid not received from dahdi
my scenario is below
analog phone (10 to 99)-- pbx--(77)asterisk
jitsi(2000)
i have analog telephone interface numbered 77 attached with asterisk
and
other sip user is 2000 on jitsi.
I can call from any number from 10 to 99(in intercom) on 77 and ivr
response will come then i can typed 2000# and call go to 2000 named
user
in asterisk.
Now my problem is when i am calling from 10 to 99 (any number) this
number
should display to sip 2000's user. But its not showing to user. Its
shows
asterisk@my_asterisk_server_ip.
my config. as follow
extension.conf
exten = s,1,Goto(phrase-menu,s,1)
[phrase-menu]
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Read(PHRASEID,/var/lib/asterisk/sounds/custom/soip)
exten = s,4,Wait(2)
exten = s,5,Set(CALLERID(num,CID)=${CALLERID})
Remove the CID option. It does nothing in this case because
it does not apply. The CID option here only applies to reading
not writing. Please re-read the documentation for CALLERID().
exten = s,6,Dial(SIP/${PHRASEID},40,tT)
exten = h,1,Hangup()
and in chan_dahdi.conf
; General options
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
cidsignalling=dtmf
cidstart=polarity
callerid=asreceived
rxgain=0.0
txgain=0.0
;FXO Modules
group=1
echocancel=yes
signalling=fxs_ks
context=default
channel=1-20
#include dahdi-channels.conf
From your description, the link between the pbx and (77)asterisk
is analog. Analog can only pass caller id information in one
direction. It looks like you have it setup to pass caller id
from the pbx to (77)asterisk. Is the pbx even sending caller id?
Is it sending it in the form you have configured in Asterisk?
(dtmf, polarity start, dtmfcidlevel=???)
Richard
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[asterisk-users] default files for voicemail box creation like /etc/skel
Is there a way to specify default files to use for new mailbox creations? For example, when a mailbox's directory structure is created, there is no greeting, unavailable, or busy messages, so the incoming calls get the message: The person at extension XX is not available. I'd like to be able to specify default files to be used and copied into the voicemail structure, something similar to /etc/skel for user accounts. Does anybody know if such a feature exists and how to use it? Thanks -Justin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] default files for voicemail box creation like /etc/skel
When you set up a new mailbox, the program copies default files. Just overlay those files with what you want. Look in apps/app_voicemail.c for guidance. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen Sent: Friday, November 30, 2012 3:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] default files for voicemail box creation like /etc/skel Is there a way to specify default files to use for new mailbox creations? For example, when a mailbox's directory structure is created, there is no greeting, unavailable, or busy messages, so the incoming calls get the message: The person at extension XX is not available. I'd like to be able to specify default files to be used and copied into the voicemail structure, something similar to /etc/skel for user accounts. Does anybody know if such a feature exists and how to use it? Thanks -Justin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
