[asterisk-users] IPKall using iax
Is it possible to receive a call via IPKall through IAX connectivity without registration? If so how to set it up. I've run-into and old link; http://forum.voxilla.com/ipkall-support-forum/ipkall-beta-testing-iax-connectivity-without-registration-26728.html -- Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF end '1' has duration 57 but want minimum 80, emulating on IAX2/a16-q1-9657
Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digiuim card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over SIP trunk from which calls get routed to third server (C) (1.6.0.9) via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is where SIP clients connect. SIP clients can also dial outside and call goes like C - B - A - PSTN. I have an occasional problem where DTMF is not recognized, ie if clients type a digit while in menu the system does not register it. In my C server I saw a log line like this today: DTMF end '1' has duration 57 but want minimum 80, emulating on IAX2/a16-q1-9657 Is the above message an indication of this problem? How can I fix it? with regards, raj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Echo cancellation on DAHDI
hey , all i have one issue on incoming DAHDI PRI it works fine many times but sometimes it creates bad audio and also having echo in line also recording going to be disturbed by this i cannot understand this properly can any one have solutions and how to improve this also how to monitor lines All dahdi lines and any causes for telco i am fro INDIA . regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E65 fails registration, soft phone works
also sprach Luki lugos...@gmail.com [2009.09.19.0745 +0200]: sounds like the hiccup my E71 had once. I think the symptoms were identical. Changing the transport type from Auto to UDP solved the problem for me. The Auto setting worked, but only sometimes. Maybe the E65 is similar... I've tried this before, but no change to the behaviour. :( Thank you for taking the time to reply. -- martin | http://madduck.net/ | http://two.sentenc.es/ give a man a fish, and you'll feed him for a day. teach a man to fish, and he'll buy a funny hat. talk to a hungry man about fish, and you're a consultant. -- scott adams spamtraps: madduck.bo...@madduck.net digital_signature_gpg.asc Description: Digital signature (see http://martin-krafft.net/gpg/) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma A200 and battery removal detection ??!!!
Dear Folks, Anyone knows if Sangoma supports or going to provide support for battery removal detection on FXO lines?? As Tzafrir said earlier DAHDI supports it, which is a very nice feature but what about Sangoma? Regards. -- M. Shokuie Nia. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help sending call to local server
hey paulh,
i think this would not help
because he wants such a dial command which forwards a call to local server
if server_ip is of same server
i have same kind of problem but still dont found proper solution
in,fact i need dialing on IP base in which dialing by using IP address will
send call to remote machine or same machine
regards
Dhaval
On Fri, Sep 18, 2009 at 5:59 PM, Paul Hales pdha...@optusnet.com.au wrote:
I have used the SIPPEER function to find if a phone is local and
available before.
PaulH
Asterisk User wrote:
Hi,
I have a generalized syntax for dial application in my dialplan where
I send calls to particular server.
Here is my dial sysntax...
exten =
_x.,1,Dial(${Dial_technology}/${extension_to_ca...@${server_ip},30,r)
I can send a call to remote server using register statement in
sip.conf or iax.conf and it works as calls get landed in particular
context of remote server.
Would you please suggest me changes to be made in .conf file(s) if I
want the calls to be landed in context of local server if Server_ip is
the IP of a server running asterisk?
Thanking you
--ASTERISK USER
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[asterisk-users] Sunday 20th Global Asterisk Mtg via VOIP - BerkeleyTIP - for forwarding
Get a VOIP headset, Install VOIP client SW, join the global Asterisk meeting this Sunday Sept 20, 12N-3P Pacific Daylight Savings Time (UTC-8), 3P-6P Eastern, (7P-10P UTC?) http://sites.google.com/site/berkeleytip/remote-attendance Lots of great, exciting new things for Asterisk users, as we start Year 2 of the Global FSW GNU(Linux)/BSD, Free HW, Free Culture, TIP meetings: TIP = Talks, Installfest, Project/Programing Party. Educational, Productive, Social. Join with the meeting from your home via VOIP, or create a local meeting at your local college wifi cafe. = Quick announcement. We're starting up BTIP year 2, for the 2009-10 school year. http://sites.google.com/site/berkeleytip/home September Videos: Puppet language, Python mystery talks, CampKDE http://sites.google.com/site/berkeleytip/talk-videos This year 2 we'll be focusing on 1) Inviting UC Berkeley students via poster/flyers 2) Getting local meetings going at California colleges 3) Getting invitations out to more American countries 4) Getting topic groups (OLPC, Python, KDE GNOME, BSD, Ubuntu, etc) having simultaneous meetings. 5) Improving our VOIP server, perhaps upgrading to FreeSwitch. == Come join the Sept 20 Sunday meeting, get on voip, chat, discuss the videos, work on your own projects share them with others, help educate students, help work on the group projects. Join #berkeleytip on irc.freenode.net, we'll help you get your VOIP HW SW working. :) Join the mailing lists say hi, tell us what you are interested in. http://groups.google.com/group/BerkTIPGlobal You are invited to forward this message anywhere it would be welcomed. :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E65 fails registration, soft phone works
Martin, Try to put qualify=yes. Torintino Date: Fri, 18 Sep 2009 22:45:05 -0700 From: lugos...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] E65 fails registration, soft phone works Martin, sounds like the hiccup my E71 had once. I think the symptoms were identical. Changing the transport type from Auto to UDP solved the problem for me. The Auto setting worked, but only sometimes. Maybe the E65 is similar... Luki 2009/9/12 martin f krafft madd...@madduck.net: Hey folks, I am trying to get an E65 to connect to asterisk, and I would really appreciate a second set of eyes. The SIP dialog completes fine, but the phone subsequently says Registration failed. I am in a network that has what seems to be a SIP-capable NAT gateway, but the asterisk is configured nat=yes anyway. Using a softphone (twinkle), I can connect just fine, SIP and RTP work. But when the E65 tries to connect, it seems to complete the SIP REGISTER dialog, but then it'll say Registration failed: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Drag n’ drop—Get easy photo sharing with Windows Live™ Photos. http://www.microsoft.com/windows/windowslive/products/photos.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E65 fails registration, soft phone works
also sprach Torintino T torinti...@hotmail.com [2009.09.19.1356 +0200]: Try to put qualify=yes. I had qualify=2000, but even with the default, the problem prevails. Thanks for taking the time to reply, -- martin | http://madduck.net/ | http://two.sentenc.es/ den stil verbessern, das heißt den gedanken verbessern. - friedrich nietzsche spamtraps: madduck.bo...@madduck.net digital_signature_gpg.asc Description: Digital signature (see http://martin-krafft.net/gpg/) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A200 and battery removal detection ??!!!
M Shokuie wrote: DAHDI supports it, which is a very nice feature but what about Sangoma? I would suggest you ask Sangoma, they are very responsive. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Switchboard/advanced answering machine setup with Asterisk?
Hi, I'm a noob to Asterisk and this list so I apologize if I'm out of place with my questions. I'm planning to utilise Asterisk to build a switchboard (of sorts) with it and was wondering if this would be feasible (from what I've read about Asterisk it seems possible). What I would like to achieve: All analog calls routed through Asterisk. Callers will need a login pin code to progress (preferably a unique pin code for each caller that I can give out). If I'm unable to answer, the caller should be able to make various choices (i.e. leave a message etc.). Is this possible? I was thinking to utilise a low power VIA mini-itx or something like that. Would a card like this be appropriate for my purpose?: http://www.digium.com/en/products/analog/tdm410.php Also, I'm living in Sweden and I was wondering where I could buy digium cards? Any tips, recommendations are welcome! Best regards Peter K ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF end '1' has duration 57 but want minimum 80, emulating on IAX2/a16-q1- 9657
On Saturday 19 September 2009 01:07:54 Rajkumar S wrote: I have an occasional problem where DTMF is not recognized, ie if clients type a digit while in menu the system does not register it. In my C server I saw a log line like this today: DTMF end '1' has duration 57 but want minimum 80, emulating on IAX2/a16-q1-9657 Is the above message an indication of this problem? How can I fix it? It isn't evidence of this problem, but it might be indicative of it. What this message says is that the DTMF lasted for 57ms, but Asterisk normally doesn't detect DTMF that lasts for under 80ms, so it is increasing the duration of the DTMF to compensate (because as a digital signal, DTMF is reliable, but when sent as audio, it might not be). What it probably indicates is that the DTMF sent to your system is _incredibly_ short, and if a DTMF detector is employed, it's possible that the DTMF audio is simply too short to be reliably detected. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF end '1' has duration 57 but want minimum 80, emulating on IAX2/a16-q1-9657
FWIW: From old, old memory, DTMF was 60 ms on, 40 ms off, way back when. With modern technology, shorter durations could work. Most phones of all types don't make a standardized tone burst but produce tones only while the button is pressed. Fast punching will produce short tones. On the other hand, a redialed number will be very well formatted. Reliability of TT data transfer for audio applications (over the phone voice mail, credit card, IVR, etc) would be better if the phones would run button pushes through the redial buffer/formatter. But they don't. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Saturday, September 19, 2009 9:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]DTMF end '1' has duration 57 but want minimum 80, emulating on IAX2/a16-q1-9657 On Saturday 19 September 2009 01:07:54 Rajkumar S wrote: I have an occasional problem where DTMF is not recognized, ie if clients type a digit while in menu the system does not register it. In my C server I saw a log line like this today: DTMF end '1' has duration 57 but want minimum 80, emulating on IAX2/a16-q1-9657 Is the above message an indication of this problem? How can I fix it? It isn't evidence of this problem, but it might be indicative of it. What this message says is that the DTMF lasted for 57ms, but Asterisk normally doesn't detect DTMF that lasts for under 80ms, so it is increasing the duration of the DTMF to compensate (because as a digital signal, DTMF is reliable, but when sent as audio, it might not be). What it probably indicates is that the DTMF sent to your system is _incredibly_ short, and if a DTMF detector is employed, it's possible that the DTMF audio is simply too short to be reliably detected. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi + SIP Realtime
Good afternoon gentlemen (and ladies).
A costumer of mine has many servers and each one maps their SIP extensions
to the others via DUNDi. It works like a charm. SIP extensions can only
register at one server, the one they belong to. In case one extension
wants to call other that is registered in another server, DUNDi takes care
of that by calling the other server using IAX2 and G.729 codec. It's
important to clarify that each server today works as a completely
independent PBX, talking to each other using IAX2 that is routed via a
MPLS network. Also, the servers are very distant from each other
physically.
Today the extensions are being mapped in DUNDi like this:
[dundi-internal]
exten = _70XX,1,Noop()
And all extensions configurations are done in sip.conf. No realtime is
being used, yet.
Now the customer wants to take a step further and make it possible that
*every* SIP extension could be able to register in *every* server. That
would make possible for them to use DNS to automatically find the
closest PBX and make the extension register on that one.
So far I considered the following for this project:
- Moving all SIP extensions from individual sip.confs to one MySQL
database, and point all servers to that one
- Configure sip.conf on each machine like this:
regcontext=dundi-internal
rtcachefriends=yes
rtsavesysname=yes
rtupdate=no
rtautoclear=yes
ignoreregexpire=no
That way each time an extension registers, Asterisk would add an extension
to the dundi-internal context, which as you guessed, is the one being
mapped to the other servers. So instead of mapping extensions using
wildcards, the extensions will be mapped individually.
extensions.conf would be something like this:
[internal]
;Tries to make the call using SIP, in the case
;the extension is registered in this server
;If it's not, switches to DUNDi
exten = _,1,Dial(SIP/${EXTEN},60)
exten = _,n,NoOp(DIALSTATUS = ${DIALSTATUS})
exten = _,n,NoOp(FROM_DUNDI = ${FROM_DUNDI})
exten = _,n,GotoIf($[${FROM_DUNDI} = 1]?end:start)
exten = _,n(start),Answer()
exten = _,n,Playback(vm-dialout)
exten = _,n,Goto(dundi-internal-helper,${EXTEN},1)
exten = _,n(end),Noop(Loop detected. Hanging up.)
exten = _,n,Hangup()
[dundi-internal-helper]
switch = DUNDi/dundi_internal
[from-dundi]
exten = _,1,Set(FROM_DUNDI=1)
exten = _,n,Dial(SIP/${EXTEN},60)
So far it's working fine in a test lab with 2 servers running Asterisk
1.6.0.15.
For the gurus out there: is there something that I'm doing terribly wrong,
that would break everything and make the universe collapse into itself
when I apply the same principle on production?
I'll be happy to provide more details in case there are any doubts. I
really appreciate your feedback, no matter what is it. :)
Vin?cius Fontes
www.asteriskforum.com.br - Informa??es e discuss?o sobre Asterisk e
telefonia IP
[JR Richardson]
I used to do this exact thing a few years ago, wrote a couple of papers
about it. Realtime + DINDi works great for this, I would add in MySQL
replication to the mix so each server writes the SIP cache info to a Master
database that is replicated out to all the servers. Each server will have a
copy of the same database and be able to contact the phones if DUNDi queries
become unavailable.
The tricky problem you may run into, if you haven't figured it out yet, is
what to do about voicemail and where the storage will be, distributed
voicemail will be problematic in a dynamic sip ua registration environment
across multiple servers. Centralize voicemail using DUNDi can help this out
as well.
I'll send you some papers off line
Hope this helps.
JR
Engineering for the Masses
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Re: [asterisk-users] DUNDi + SIP Realtime
- JR Richardson jmr.richard...@gmail.com escreveu:
Good afternoon gentlemen (and ladies).
A costumer of mine has many servers and each one maps their SIP
extensions
to the others via DUNDi. It works like a charm. SIP extensions can
only
register at one server, the one they belong to. In case one
extension
wants to call other that is registered in another server, DUNDi
takes care
of that by calling the other server using IAX2 and G.729 codec.
It's
important to clarify that each server today works as a completely
independent PBX, talking to each other using IAX2 that is routed via
a
MPLS network. Also, the servers are very distant from each other
physically.
Today the extensions are being mapped in DUNDi like this:
[dundi-internal]
exten = _70XX,1,Noop()
And all extensions configurations are done in sip.conf. No realtime
is
being used, yet.
Now the customer wants to take a step further and make it possible
that
*every* SIP extension could be able to register in *every* server.
That
would make possible for them to use DNS to automatically find the
closest PBX and make the extension register on that one.
So far I considered the following for this project:
- Moving all SIP extensions from individual sip.confs to one MySQL
database, and point all servers to that one
- Configure sip.conf on each machine like this:
regcontext=dundi-internal
rtcachefriends=yes
rtsavesysname=yes
rtupdate=no
rtautoclear=yes
ignoreregexpire=no
That way each time an extension registers, Asterisk would add an
extension
to the dundi-internal context, which as you guessed, is the one
being
mapped to the other servers. So instead of mapping extensions using
wildcards, the extensions will be mapped individually.
extensions.conf would be something like this:
[internal]
;Tries to make the call using SIP, in the case
;the extension is registered in this server
;If it's not, switches to DUNDi
exten = _,1,Dial(SIP/${EXTEN},60)
exten = _,n,NoOp(DIALSTATUS = ${DIALSTATUS})
exten = _,n,NoOp(FROM_DUNDI = ${FROM_DUNDI})
exten = _,n,GotoIf($[${FROM_DUNDI} = 1]?end:start)
exten = _,n(start),Answer()
exten = _,n,Playback(vm-dialout)
exten = _,n,Goto(dundi-internal-helper,${EXTEN},1)
exten = _,n(end),Noop(Loop detected. Hanging up.)
exten = _,n,Hangup()
[dundi-internal-helper]
switch = DUNDi/dundi_internal
[from-dundi]
exten = _,1,Set(FROM_DUNDI=1)
exten = _,n,Dial(SIP/${EXTEN},60)
So far it's working fine in a test lab with 2 servers running
Asterisk
1.6.0.15.
For the gurus out there: is there something that I'm doing terribly
wrong,
that would break everything and make the universe collapse into
itself
when I apply the same principle on production?
I'll be happy to provide more details in case there are any doubts.
I
really appreciate your feedback, no matter what is it. :)
Vin?cius Fontes
www.asteriskforum.com.br - Informa??es e discuss?o sobre Asterisk e
telefonia IP
[JR Richardson]
I used to do this exact thing a few years ago, wrote a couple of
papers
about it. Realtime + DINDi works great for this, I would add in
MySQL
replication to the mix so each server writes the SIP cache info to a
Master
database that is replicated out to all the servers. Each server will
have a
copy of the same database and be able to contact the phones if DUNDi
queries
become unavailable.
The tricky problem you may run into, if you haven't figured it out
yet, is
what to do about voicemail and where the storage will be, distributed
voicemail will be problematic in a dynamic sip ua registration
environment
across multiple servers. Centralize voicemail using DUNDi can help
this out
as well.
I'll send you some papers off line
Hope this helps.
JR
Engineering for the Masses
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Thank you so much for your answers. I really wasn't aware of the problem with
voicemail, but storing it on IMAP or even MySQL will certainly help.
Your case is really close to my own, with a few differences. The main objective
in my case is not redundancy, but saving bandwidth. That because if a user of
pbxA is physically in the same network of pbxB, it will register directly to
pbxA. And as the dialplan uses the options tT on Dial(), if this user calls an
extension registered at pbxB, the audio needs to go to pbxA and come back to
pbxB. To make things worst, in this case the codec used is alaw.
So my idea is to force all inter-server communication to be done
Re: [asterisk-users] Sangoma A200 and battery removal detection ??!!!
On Sat, Sep 19, 2009 at 3:33 AM, M Shokuie sena...@gmail.com wrote: Dear Folks, Anyone knows if Sangoma supports or going to provide support for battery removal detection on FXO lines?? As Tzafrir said earlier DAHDI supports it, which is a very nice feature but what about Sangoma? Hello M. Shokuie, In Asterisk installations, Sangoma boards use DAHDI drivers to hook up to Asterisk ( because chan_dahdi works only with DAHDI devices). If you were using FreeSWITCH you could use native wanpipe devices /dev/wanpipex_xx instead of /dev/dahdi/x, which uses native Sangoma drivers. In both modes the drivers notify on-hook, off-hook events depending on the battery status. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Stop / Resume in Dialplan / AMI
Hello. I'd like to know if the two following functionalities are available in Asterisk. -1- A stop/wait/halt functionality in the Dialplan. Like: exten = myexten, n, Halt where execution of the dialplan would wait indefinitely. I guess a Wait would be OK, but I'd like this wait to wait indefinitely. -2- A Goto functionality from the AMI: You give the channel, and you can ask it to change its priority. -3- Or a WaitForEvent: The AMI sends an event, and the Dialplan resumes dialplan execution after this event. These are a few questions I have about AMI / Dialplan asynchronous integration, and guidance would be appreciated. All the best, -- Guillaume Yziquel http://yziquel.homelinux.org/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk + Nonoh
Hello; Asterisk voip configuration settings for nonoh did, but somehow could not manage. I would like your example of working configuration. Best regards M.B. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Experience with Sangoma's USBfxo
Hi, I've seen this USB product from Sangoma : http://www.sangoma.com/products_and_solutions/hardware/analog_telephony/usb_fxo.html Is it working ok ? Is it easy to integrate it with Asterisk ? How would you rate your experience with it ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
