[asterisk-users] param sayduration of mailbox
Hello, what exactly is the function of the parameter 'sayduration' in the voicemail box configuration ? Whether I put this to 'yes' or to 'no', nothing changes. I do not get the announcement of duration at the beginning of the voicemail message. Kind regards, Jonas. -- _ -- 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] AGI command
Hi, in CLI, I type agi show or other agi commad, but response me command not found. How can see agi is work normally in my server? -- _ -- 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] AGI command
you need to run full command, like agi show commands topic answer agi show commands topic gosub agi set debug on Regards, Zohair Raza On Tue, Jan 15, 2013 at 1:05 PM, Muhammad mohammad.ghaz...@gmail.comwrote: Hi, in CLI, I type agi show or other agi commad, but response me command not found. How can see agi is work normally in my server? -- _ -- 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
[asterisk-users] POSTing recorded audio stream
Hello,
I quite don't understand how to send a recorded message during a call off to an
HTTP handler using HTTP POST.
How do I access this file/audiostream in the dialplan?
I tried this:
exten = rpm,1,Set(RecordedPersonalMessage=${EPOCH})
exten =
rpm,n,Record(/var/lib/asterisk/sounds/recordings/${RecordedPersonalMessage}:wav)
exten =
rpm,n,Set(Result=${CURL(soundfragmenthandler.company.local/soundfragmenthandler.ashx,postdata_var1=/var/lib/asterisk/sounds/recordings/${RecordedPersonalMessage}.wav)})
I even tried it with the variable ${RECORDED_FILE} but that only posted the
filename.
Am I accessing the file in the wrong way or is CURL not able to handle these
types of things?
Regards,
Grant
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Re: [asterisk-users] POSTing recorded audio stream
I've come up with a solution for this:
exten = rpm,1,System(curl --request POST --form
file=@/var/lib/asterisk/sounds/recordings/${RecordedPersonalMessage}.wavmailto:file=@/var/lib/asterisk/sounds/recordings/$%7bRecordedPersonalMessage%7d.wav
http://soundfragmenthandler.company.local/soundfragmenthandler.ashx)
Not sure if it's the best way to do, but it works.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Grant Bagdasarian
Sent: dinsdag 15 januari 2013 12:01
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] POSTing recorded audio stream
Hello,
I quite don't understand how to send a recorded message during a call off to an
HTTP handler using HTTP POST.
How do I access this file/audiostream in the dialplan?
I tried this:
exten = rpm,1,Set(RecordedPersonalMessage=${EPOCH})
exten =
rpm,n,Record(/var/lib/asterisk/sounds/recordings/${RecordedPersonalMessage}:wav)
exten =
rpm,n,Set(Result=${CURL(soundfragmenthandler.company.local/soundfragmenthandler.ashx,postdata_var1=/var/lib/asterisk/sounds/recordings/${RecordedPersonalMessage}.wav)})
I even tried it with the variable ${RECORDED_FILE} but that only posted the
filename.
Am I accessing the file in the wrong way or is CURL not able to handle these
types of things?
Regards,
Grant
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[asterisk-users] Telephony card in Thecus N4800
Hello, I've seen this Atom-based NAS appliance with one PCIe 1x slot : http://www.thecus.com/product.php?PROD_ID=65 Has someone successfully added a Digium, Sangoma or other telephony card in it along asterisk of course ? Regards -- _ -- 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] Reporting Utility
When CDR reporting was raised a few days ago, it prompted me to add a section on how ADTransform could be used to address the problem raised about consolidating CDR information from various divisional PBXs and producing consolidated reports. I wrote a short Use Case article. http://www.artifact-software.com/?page_id=1666 I also thought about the problem of getting the configuration files into a readable format such as phone lists that could be distributed and added a second Use Case. I am only maintaining our internal PBX so reporting configurations is not a big issue but I would imagine that for someone maintaining many clients or many corporate PBXs, having a batch tool that can collect the data files, produce nice looking reports and automatically upload them to a portal might be helpful. I would be grateful for any comments about content, format or my skill as an artist.:-) Ron -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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] Followme Killing Asterisk
On Mon, Jan 14, 2013 at 9:36 PM, A E G all.efor...@gmail.com wrote:
Hi Guys,
this has been a weekend destroyer for me. I've struggled this all day and
most of today.
From your discussion below, it sounds like the real problem is the Asterisk
crashing.
So, as a first step to solving **that** problem, make sure asterisk is
compiled with debug
flags, dumps another core file, and then you do the gdb asterisk
corefilename, and
get a stack trace. That should give us some idea of what happened.
I have a fairly simple Followme sequence in place to see how it works
before I get into the complex scenarios.
extensions.conf
---
[Incoming]
exten = MyDID,1, Answer()
same = n, Set(CHANNEL(language)=en_AU)
same = n, Followme(TestFollow)
same = n, NoOp(++Back after Followme: DIALSTATUS =
${DIALSTATUS}, Hangupcause = ${HANGUPCAUSE})
same = n, Hangup()
[Followme-Dialout]
exten = _1NXXNXX,1,Set(CHANNEL(language)=en_AU)
same = n, Dial(SIP/GW-1/${EXTEN})
followme.conf
[TestFollow]
context = Followme-Dialout
number = my landline,30
number = my cell phone,20
The call goes out, and rings my first phone. If I answer it, the Asterisk
core dumps, the calls stay up!
snip
[Jan 15 04:19:48] -- Called SIP/GW-1/1203555
[Jan 15 04:19:51] -- SIP/GW-1-0007 is making progress passing it
to Local/1203555@Followme-Dialout-0004;2
[Jan 15 04:19:51] -- Local/1203555@Followme-Dialout-0004;1 is
making progress
[Jan 15 04:20:05] -- SIP/GW-1-0007 answered Local/1203555
@Followme-Dialout-0004;2
[Jan 15 04:20:05] -- Local/1203555@Followme-Dialout-0004;1
answered SIP/DIDProvider-1-0006
[Jan 15 04:20:05] -- Starting playback of followme/call-from
[Jan 15 04:20:05] -- Local/1203555@Followme-Dialout-0004;1
Playing 'followme/no-recording.ulaw' (language 'en_AU')
[Jan 15 04:20:05] -- Local/1203555@Followme-Dialout-0004;1
requested a source update
ast00*CLI
Disconnected from Asterisk server
Bus error (core dumped)
...snip
I have been playing with Local channels over the weekend, and as cool as
they sound, they have caused me nothing but pain. Once again, following the
console log, I notice that Followme indeed uses Local channel to make these
calls and returns control when the call times out etc.
The ONLY time it gets anywhere is if I use the 'l' option with Followme
application.
In that case, the call connect and I can have a conversation but the
minute the remote party hangs up, asterisk dumps core again.
it may be something to do with the after return to handle next steps but
what are they supposed to be? I don't want anything to happen like go to VM
or anything.
Have tried this with 10.3.0 and 10.11.1. I noticed new changes have been
made in v11...but this should work
How does this work?? Do I need fancy options with the Dial command doing
GoSub and what not? and Why does it insist on playing all these prompts I
have commented them all out from followme.conf, but it's still looking to
play them
Thanks in advance
\A
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--
Steve Murphy
ParseTree Corporation
57 Lane 17
Cody, WY 82414
✉ m...@parsetree.com
☎ 307-899-5535
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Re: [asterisk-users] Followme Killing Asterisk
On Tue, Jan 15, 2013 at 11:05 AM, Steve Murphy m...@parsetree.com wrote:
On Mon, Jan 14, 2013 at 9:36 PM, A E G all.efor...@gmail.com wrote:
Hi Guys,
this has been a weekend destroyer for me. I've struggled this all day and
most of today.
From your discussion below, it sounds like the real problem is the
Asterisk crashing.
So, as a first step to solving **that** problem, make sure asterisk is
compiled with debug
flags, dumps another core file, and then you do the gdb asterisk
corefilename, and
get a stack trace. That should give us some idea of what happened.
Thanks for the note Steve. It doesn't sound like there's tremendously wrong
that I'm doing as far s the configuration is concerned then? and it won't
be too surprising since the configuration of Followme is quite simple
assuming the complexities are all handled by the Followme app.
I tried a whole lot of options that made sense as Dial options that the
Local channel dial from Followme is being hooked into but it appears
that, the cause of the crash is most likely that Followme:
1. Is looking for something to do; bill, log or something after it
returns from Dial/call termination but not finding it. I tried using
Answer(nocdr) at the time the call on the DID is being answered but that
didn't help. I have also tried the 'g', 'c', 'C', 'I' and 'i' etc options
with the Dial but they don't help either. I had real hopes in the 'g'
option to tell it to proceed with the dial plan where I was simply making
it return a couple of call status related variables and then just Hangup,
but regardless of the 'calling' or the called party hanging up, these
number get printed, which means that despite the 'g' option, the call does
NOT proceed with the normal/rest of the dialplan
2. Maybe Followme is not built for this purpose where the caller is
unknown (which it would be in most cases) but at least the called party
is usually known AND is a subscriber/registered user of the system who is
then using the Followme feature to find them when they don't answer their
PBX registered phone. What I'm doing calling from outside, having the
system answer the call, allow the caller to put in a number and then
calling those numbers associated with that extension if it's a Followme
extension but the extension itself isn't a registered user in sip.conf or
users.conf, and maybe followme app has some procedures it needs to run
through as a matter of housekeeping (i.e. accounting, billing, logging etc)
that it's not finding info for
Will do a gdb and see what I can find...I'm not a developer so I may not be
able to pick up a lot from the stack-trace but will pastebin it and see if
one of the community/developer members can figure out why it's taking a dump
Cheers
\a
I have a fairly simple Followme sequence in place to see how it works
before I get into the complex scenarios.
extensions.conf
---
[Incoming]
exten = MyDID,1, Answer()
same = n, Set(CHANNEL(language)=en_AU)
same = n, Followme(TestFollow)
same = n, NoOp(++Back after Followme: DIALSTATUS =
${DIALSTATUS}, Hangupcause = ${HANGUPCAUSE})
same = n, Hangup()
[Followme-Dialout]
exten = _1NXXNXX,1,Set(CHANNEL(language)=en_AU)
same = n, Dial(SIP/GW-1/${EXTEN})
followme.conf
[TestFollow]
context = Followme-Dialout
number = my landline,30
number = my cell phone,20
The call goes out, and rings my first phone. If I answer it, the Asterisk
core dumps, the calls stay up!
snip
[Jan 15 04:19:48] -- Called SIP/GW-1/1203555
[Jan 15 04:19:51] -- SIP/GW-1-0007 is making progress passing it
to Local/1203555@Followme-Dialout-0004;2
[Jan 15 04:19:51] -- Local/1203555@Followme-Dialout-0004;1
is making progress
[Jan 15 04:20:05] -- SIP/GW-1-0007 answered Local/1203555
@Followme-Dialout-0004;2
[Jan 15 04:20:05] -- Local/1203555@Followme-Dialout-0004;1
answered SIP/DIDProvider-1-0006
[Jan 15 04:20:05] -- Starting playback of followme/call-from
[Jan 15 04:20:05] -- Local/1203555@Followme-Dialout-0004;1
Playing 'followme/no-recording.ulaw' (language 'en_AU')
[Jan 15 04:20:05] -- Local/1203555@Followme-Dialout-0004;1
requested a source update
ast00*CLI
Disconnected from Asterisk server
Bus error (core dumped)
...snip
I have been playing with Local channels over the weekend, and as cool
as they sound, they have caused me nothing but pain. Once again, following
the console log, I notice that Followme indeed uses Local channel to make
these calls and returns control when the call times out etc.
The ONLY time it gets anywhere is if I use the 'l' option with Followme
application.
In that case, the call connect and I can have a conversation but the
minute the remote party hangs up, asterisk dumps core again.
it may be something to do with the after
[asterisk-users] Asterisk, DNS SRV, 1.8
From voip-info.org: If srvlookup is turned on, Asterisk supports DNS SRV lookups partially. Currently, Asterisk only reads the first SRV entry without bothering with priorities and weights. Is this still the case with Asterisk 1.8? -- _ -- 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] Call parking in a multi-tenant system
We use Asterisk as a hosted PBX. We've had a couple of requests for parking, but none of the documentation shows any way to make it aware of contexts or otherwise make it multi-tenant. Have I missed something and does anyone know how to make this work? Would be on Asterisk 1.6 for now, 1.8 some time soon. -- 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
[asterisk-users] Getting UDPTL (SIP): Transmission error: Resource temporarily unavailable
Hi, I configured Asterisk 10 for inbound fax, for couple of weeks I didn't see any issues until today. The setup I configured for inbound fax is quite simple i.e. Cisco Voice GW sends the fax calls to Asterisk using T.38 protocol and later Asterisk stores/forwards the fax to specific end user. The configuration I made in sip.conf for enabling T38 is listed below; t38pt_udptl = yes,fec,maxdatagram=400 faxdetect = t38 And in udptl.conf, I just uncommented 'use_even_ports = yes ;' and rest of it set as default. Here is the error I'm usually seeing in Asterisk side; [Jan 15 14:13:28] NOTICE[20514] udptl.c: UDPTL (SIP/10.3.22.6-0ad6): Transmission error to 10.3.22.6:18428: Resource temporarily unavailable If this notice comes, it occurs repeatedly unless I need to restart the asterisk service. For some reason it also effect the V-GW. Please advise what is the reason that I'm getting this message and how can I avoid it? -- Regards, Ahmed Munir Chohan -- _ -- 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] Call parking in a multi-tenant system
Hello, from 1.6.2 version, Asterisk suport multi-tenant parking Look at features.conf for a example. Regards El 15/01/2013 15:58, Carlos Alvarez escribió: We use Asterisk as a hosted PBX. We've had a couple of requests for parking, but none of the documentation shows any way to make it aware of contexts or otherwise make it multi-tenant. Have I missed something and does anyone know how to make this work? Would be on Asterisk 1.6 for now, 1.8 some time soon. -- 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] TE110P Wildcard does not work with Ubuntu 12.04 server
Michel Verbraak michel at verbraak.org writes: Op 22-08-12 12:09, Shitian Long schreef: I am trying to setup TE110P wildcard on a PBX running ubuntu 12.04 server edition. I followed the procedure from http://docs.digium.com/misc/ADL_quickstart.pdf step by step. During the process of installing dahdi-linux-complete I got following warnings: root at ubuntu:/usr/local/src/dahdi-linux-complete-2.6.1+2.6.1# make perl: warning: Setting locale failed. perl: warning: Please check that your locale settings: LANGUAGE = en_US:en, LC_ALL = (unset), LC_CTYPE = UTF-8, LANG = en_US.UTF-8 are supported and installed on your system. perl: warning: Falling back to the standard locale (C). Frist of, I am wondering if this error matters? Second question, after installation process complete, and reboot the machine I got the following error, when machine boot up: Loading DAHDI hardware modules: wcte11xp: error I think the TE110P card is no properly loaded. I try to confirm my thought by using root at ubuntu:~# dahdi_tool There is no interface listed on the table. I am wondering if anyone got idea about this issue. Thanks. longst -- _ Having the same problems: /# dmesg -c /dev/null /# /etc/init.d/dahdi stop Unloading DAHDI hardware modules: done /# modprobe wte11xp FATAL: Module wte11xp not found. /# dmesg [ 2775.316252] dahdi: Telephony Interface Unloaded /# lspci | grep Dig /# however: # make config install -D dahdi.init /etc/init.d/dahdi /usr/sbin/update-rc.d dahdi defaults 15 30 System start/stop links for /etc/init.d/dahdi already exist. DAHDI has been configured. List of detected DAHDI devices: pci::03:02.0 wcte11xp-e159:0001 Digium Wildcard TE110P T1/E1 Board run 'dahdi_genconf modules' to load support for only the DAHDI hardware installed in this system. By default support for all DAHDI hardware is loaded at DAHDI start. any ideas? doesnt look like my system recognized the card at all. no lights on the card. card was working perfectly. i simply moved it from one box (gentoo) to another (ubuntu 12.04) -- _ -- 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] TE110P Wildcard does not work with Ubuntu 12.04 server
Hi, On Tue, Jan 15, 2013 at 09:18:46PM +, ted wrote: Michel Verbraak michel at verbraak.org writes: Op 22-08-12 12:09, Shitian Long I am trying to setup TE110P wildcard on a PBX running ubuntu 12.04 server edition. I followed the procedure from http://docs.digium.com/misc/ADL_quickstart.pdf step by step. During the process of installing dahdi-linux-complete I got following warnings: root at ubuntu:/usr/local/src/dahdi-linux-complete-2.6.1+2.6.1# make perl: warning: Setting locale failed. perl: warning: Please check that your locale settings: LANGUAGE = en_US:en, LC_ALL = (unset), LC_CTYPE = UTF-8, LANG = en_US.UTF-8 are supported and installed on your system. perl: warning: Falling back to the standard locale (C). Frist of, I am wondering if this error matters? I do not think this error matters. You probably should still make sure your locale is set properly. Second question, after installation process complete, and reboot the machine I got the following error, when machine boot up: Loading DAHDI hardware modules: wcte11xp: error There should be some output in dmesg that indicates what the problem is when you get an error here. I think the TE110P card is no properly loaded. I try to confirm my thought by using root at ubuntu:~# dahdi_tool There is no interface listed on the table. I am wondering if anyone got idea about this issue. Thanks. Having the same problems: /# dmesg -c /dev/null /# /etc/init.d/dahdi stop Unloading DAHDI hardware modules: done /# modprobe wte11xp FATAL: Module wte11xp not found. The above is a differnt error than what Shitian reported. It doesn't look like the driver was installed properly for the current kernel. /# dmesg [ 2775.316252] dahdi: Telephony Interface Unloaded /# lspci | grep Dig /# however: # make config install -D dahdi.init /etc/init.d/dahdi /usr/sbin/update-rc.d dahdi defaults 15 30 System start/stop links for /etc/init.d/dahdi already exist. DAHDI has been configured. List of detected DAHDI devices: pci::03:02.0 wcte11xp-e159:0001 Digium Wildcard TE110P T1/E1 Board run 'dahdi_genconf modules' to load support for only the DAHDI hardware installed in this system. By default support for all DAHDI hardware is loaded at DAHDI start. any ideas? doesnt look like my system recognized the card at all. no lights on the card. card was working perfectly. i simply moved it from one box (gentoo) to another (ubuntu 12.04) Best guess based on what you said that the driver really isn't installed for the current kernel. Cheers, Shaun -- 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] Getting UDPTL (SIP): Transmission error: Resource temporarily unavailable
Can you be more specific about your Asterisk version? 10.xx.yy ? Sounds like some sort of resource leak. On Tue, Jan 15, 2013 at 3:02 PM, Ahmed Munir ahmedmunir...@gmail.comwrote: Hi, I configured Asterisk 10 for inbound fax, for couple of weeks I didn't see any issues until today. The setup I configured for inbound fax is quite simple i.e. Cisco Voice GW sends the fax calls to Asterisk using T.38 protocol and later Asterisk stores/forwards the fax to specific end user. The configuration I made in sip.conf for enabling T38 is listed below; t38pt_udptl = yes,fec,maxdatagram=400 faxdetect = t38 And in udptl.conf, I just uncommented 'use_even_ports = yes ;' and rest of it set as default. Here is the error I'm usually seeing in Asterisk side; [Jan 15 14:13:28] NOTICE[20514] udptl.c: UDPTL (SIP/10.3.22.6-0ad6): Transmission error to 10.3.22.6:18428: Resource temporarily unavailable If this notice comes, it occurs repeatedly unless I need to restart the asterisk service. For some reason it also effect the V-GW. Please advise what is the reason that I'm getting this message and how can I avoid it? -- Regards, Ahmed Munir Chohan -- _ -- 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 -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] TE110P Wildcard does not work with Ubuntu 12.04 server
Shaun Ruffell sruffell at digium.com writes: Best guess based on what you said that the driver really isn't installed for the current kernel. Cheers, Shaun thanks much! this lead me to find this: http://totalticketsystem.com/blog/technical-articles/how-to-install-asterisk-on-ubuntu-from-scratch/ which i followed and things are working now. -- _ -- 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] special conference room
Hi list, I am in need of a special asterisk conference room with the following constraints: - there is one admin / moderator and several normal callers. - the callers must not hear any other caller, only the moderator - the moderator must be able to mute and unmute any caller at any time - the moderator must be able to talk to all callers or to a specific caller. - the modetator must be able to kick off any caller at any time... Any hints on how to realize that are highly appreciated.. Thanx in advance, yves -- _ -- 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] AGI command
*Thanks Zohair! I wrote some php code to working with AGI, but it dosen't work. I don't know how can run it. please explain me when I put my php code inside /var/lib/asterisk/agi-bin so, what should I do after that. and the second one, how can limit users to call just my number in list at database and permit to call another numbers.** * On Tue, Jan 15, 2013 at 12:39 PM, Zohair Raza engineerzuhairr...@gmail.comwrote: you need to run full command, like agi show commands topic answer agi show commands topic gosub agi set debug on Regards, Zohair Raza On Tue, Jan 15, 2013 at 1:05 PM, Muhammad mohammad.ghaz...@gmail.comwrote: Hi, in CLI, I type agi show or other agi commad, but response me command not found. How can see agi is work normally in my server? -- _ -- 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 -- _ -- 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] AGI command
On Wed, Jan 16, 2013 at 11:01 AM, Muhammad mohammad.ghaz...@gmail.comwrote: *Thanks Zohair! I wrote some php code to working with AGI, but it dosen't work. * *I don't know how can run it. please explain me when I put my php code inside /var/lib/asterisk/agi-bin so, what should I do after that. * Make sure Asterisk has access to your AGI script, and make it executable (chmod u+x agi.php). Also make sure it has shebang (!#/usr/bin/php) *and the second one, how can limit users to call just my number in list at database and permit to call another numbers.* * * That depends on logic in your script, you can also separate users by contexts * * On Tue, Jan 15, 2013 at 12:39 PM, Zohair Raza engineerzuhairr...@gmail.com wrote: you need to run full command, like agi show commands topic answer agi show commands topic gosub agi set debug on Regards, Zohair Raza On Tue, Jan 15, 2013 at 1:05 PM, Muhammad mohammad.ghaz...@gmail.comwrote: Hi, in CLI, I type agi show or other agi commad, but response me command not found. How can see agi is work normally in my server? -- _ -- 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 -- _ -- 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] AGI command
Hi, Please see my comments in line. Regards, Sammy On Wed, Jan 16, 2013 at 12:13 PM, Zohair Raza engineerzuhairr...@gmail.comwrote: On Wed, Jan 16, 2013 at 11:01 AM, Muhammad mohammad.ghaz...@gmail.comwrote: *Thanks Zohair! I wrote some php code to working with AGI, but it dosen't work. * *I don't know how can run it. please explain me when I put my php code inside /var/lib/asterisk/agi-bin so, what should I do after that. * Make sure Asterisk has access to your AGI script, and make it executable (chmod u+x agi.php). Also make sure it has shebang (!#/usr/bin/php) Besides that you'll need to create SIP users and define their * context=my-agi* and in your context call this AGI. [my-agi] exten = _X.,1,NOOP(Invoking AGI Script now) same = n,AGI(my-agi-filename.php) same = n,NOOP(Any other post AGI things here) same = n,Hangup() *and the second one, how can limit users to call just my number in list at database and permit to call another numbers.* * * That depends on logic in your script, you can also separate users by contexts Alternative to the above approach there are other possibilities, like applying a GotoIF() condition in the *[my-agi]* dialplan code above restricting a particluar number, or Instead of *_X. *in the above code implement some regex to enable only local area dialing i.e exten = _78XX,1,NOOP(Invoking AGI Script now for dialled string starting with 78 and 8 digit in length) If you don't like any of the above because you've a list of numbers which should be dialled only in a DB.table then in your php-AGI script do some restrictions based on that table. * * On Tue, Jan 15, 2013 at 12:39 PM, Zohair Raza engineerzuhairr...@gmail.com wrote: you need to run full command, like agi show commands topic answer agi show commands topic gosub agi set debug on Regards, Zohair Raza On Tue, Jan 15, 2013 at 1:05 PM, Muhammad mohammad.ghaz...@gmail.comwrote: Hi, in CLI, I type agi show or other agi commad, but response me command not found. How can see agi is work normally in my server? -- _ -- 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 -- _ -- 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 -- _ -- 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] AGI command
On Wed, 16 Jan 2013, Muhammad wrote: I wrote some php code to working with AGI, but it dosen't work. When you say 'doesn't work' do you mean 'doesn't do what I want' or 'does not execute?' If you enable AGI debugging, what does the Asterisk console log look like? Did you use an established PHP library or 'roll your own?' A good way to test an AGI is to create a text file containing all the cruft (the AGI 'environment') Asterisk sends to the AGI along with the expected responses. Then you can execute your AGI completely external from Asterisk with a shell command line like: /var/lib/asterisk/agi-bin/my-firs-agi example environment -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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
