Re: [asterisk-users] Sending SMS from asterisk
Hi Bilal, It's not necessary to use a FXS port, you can compile install chan_dongle and buy a Huawei 3G dongle. We have running here a SMS solution with four 3G dongles, which sends over 20.000 SMS a month. In addition, I wrote an script able to send up to 12000 characters in concatenated SMS (the recipient receives a single SMS only) chan_dongle works very well. -- == Miguel Oyarzo Senior [ Network | Systems Design ] Engineer http://www.linkedin.com/in/mikeaustralia Linux User: # 483188 - counter.li.org Melbourne, Australia On 3/9/2013 1:09 PM, Gerardo Barajas wrote: Yes, you can check solutions from sangoma and khomp. Saludos/Regards -- Ing. Gerardo Barajas Puente Proyectos Especiales/Preventa | www.neocenter.com http://www.neocenter.com T:+52 (55) 8590-9000 x 7003 On Fri, Mar 8, 2013 at 6:32 PM, bilal ghayyad bilmar...@yahoo.com mailto:bilmar...@yahoo.com wrote: Hi; If my landline service provider does not provide the ability to send the SMS, and I need to send SMS from asterisk, then what is the required? How? Is it possible to send SMS from asterisk using SIM card to be connected via GSM adaptor connected to FXS ports? Or HOW? From the other side, this is existed only in asterisk 1.8 or it is existed in asterisk 1.4? Regards Bilal -- _ -- 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] Sending SMS from asterisk
Hi there, for sending SMSes I am using a 3G Modem and SMSlib... it is not bound to asterisk in any way, but I always wanted to integrate this possibility some day. I could not do so, because our landline provider (Vodafone) does not support it via E1 PRI lines... by I thought (never tried...) if it would be possible to use the SMS ServiceNumber from my mobile Provider...? I have a valid mobile contract, the number of the SMScc , my Cardnumber (t-mobile), my phonenumber and so on... so it should be possible, I think... but how? Has anybody a clue? regards, yves Am 09.03.2013 11:03, schrieb Miguel Oyarzo: Hi Bilal, It's not necessary to use a FXS port, you can compile install chan_dongle and buy a Huawei 3G dongle. We have running here a SMS solution with four 3G dongles, which sends over 20.000 SMS a month. In addition, I wrote an script able to send up to 12000 characters in concatenated SMS (the recipient receives a single SMS only) chan_dongle works very well. -- == Miguel Oyarzo Senior [ Network | Systems Design ] Engineer http://www.linkedin.com/in/mikeaustralia Linux User: # 483188 - counter.li.org Melbourne, Australia On 3/9/2013 1:09 PM, Gerardo Barajas wrote: Yes, you can check solutions from sangoma and khomp. Saludos/Regards -- Ing. Gerardo Barajas Puente Proyectos Especiales/Preventa | www.neocenter.com http://www.neocenter.com T:+52 (55) 8590-9000 x 7003 On Fri, Mar 8, 2013 at 6:32 PM, bilal ghayyad bilmar...@yahoo.com mailto:bilmar...@yahoo.com wrote: Hi; If my landline service provider does not provide the ability to send the SMS, and I need to send SMS from asterisk, then what is the required? How? Is it possible to send SMS from asterisk using SIM card to be connected via GSM adaptor connected to FXS ports? Or HOW? From the other side, this is existed only in asterisk 1.8 or it is existed in asterisk 1.4? Regards Bilal -- _ -- 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 byhttp://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
[asterisk-users] Digium Wildcard TDM800P not working with DAHDI
Hello everyone, How can I let Digium Wildcard TDM800P work successfully with DAHDI? Because the Centos recognizes the card but I can't get the analog card working with DAHDI. Thanks in advance, Gilberto -- _ -- 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] About Zitter Control
Hello there, I have built a kernel module that is responsible for sip and rtp encryption/decryption, padding/depadding, ptime decrease (splitting a large rtp packet to splitting into smaller packets of ptime 20) for incoming packets and ptime increase (merging small rtp packets into a big rtp packet of desired ptime) for outgoing packets. This is meant to be a faster process and indeed it is. The whole purpose is to taking the load from the sip server and to manage this things in kernel space. Now my problem is that, when packets are splitted in kernel space, no harm is done. But when packets are merged, i see somehow large zitter and delta that makes the voice robotic. I am quite new to this line. So from my experience I can say nothing on which side it is the problem. I carefully observed RTP sequencing and timestamping, and yes, it is handled perfectly. To have a work around, I suspect the following things: 1. The zitter control might be the problem. But on which side I should concentrate? Should I look the client's zitter buffer handling or server side zitter buffer handling? 2. If asterisk is run from a RT (real time) kernel, should it make any problem? Or, is it necessary to run asterisk from an RT kernel to mitigate my issue? 3. If both RT kernel and zitter control mechanism is taken to the asterisk server, can I hope to solve this problem? I am in doubt whether the process gets somehow expensive while ptime increase. Because I have to drop a few packets, making lists of packets referring to a communication, and etc. etc. I am running out of ideas. Because debugging in kernel space is not easy. -- Rifat Rahman -- _ -- 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] digium card and virualbox
On 08/03/13 21:21, bilal ghayyad wrote: Hi All; How to let the virualbox (ubuntu OS) to be able to see the digium card? Because when I install elastix or asterisk with dahdi, it is not able to see the digium card if the installation though the virualbox .. What is the solution? The solution is to run Ubuntu and Asterisk on your hardware natively, not through VirtualBox. Virtualisation and high-performance hardware such as telephony cards (it will be creating 8000 interrupts per phone line per second) do not mix, I am afraid. -- AJS. -- _ -- 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] Recording with MixMonitor and AGI
Hi,
Thanks for your answer!
1.
so you want to establish a call (triggered by ami) between two partys, record
the conversation
and save the file to a(nother) server (afterwards), right?
Yes this is correct, and I prefer to do the transferring of the file to another
server with my existing AGI.
My AGIs are written in java. Today I the upload is done over http.
Today I schedule the upload in the AGI script a couple of seconds after the
channel is hang up. But the two
lines might not be hung up at the same time.
Your suggestion of always fixing the file is wise, it now seems to work fine
after having been processed with sox.
So now I think that this case 1 is ok for me :-)
2.
and another task is to establish (also ami triggered) a call to a mobile and
play, lets say a voicefile.
this conversation should also be recorded and saved on a(nother) server
(afterwards), right?
The idea is to perform a probe call with the only task of recording what the
other party says.
It will be merged by hand on a mobile phone to an ongoing call with another
party.
This could be done by calling out and letting AGI execute a RECORD FILE but if
there is a way to just
dial out and then let the server side of the call Keep the channel up but do
nothing forever until the call is hang up
Then I could easily use the MixMonitor and write the whole conversation in the
dialplan with uploading similar to the first case.
Any suggestions?
Regards,
Henrik
Från: Yves A. yves...@gmx.demailto:yves...@gmx.de
Svara till: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com
Datum: torsdag 7 mars 2013 20:10
Till: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com
Ämne: Re: [asterisk-users] Recording with MixMonitor and AGI
hi,
hard to understand, what your objective is... at least for me ;-)
so you want to establish a call (triggered by ami) between two partys, record
the conversation
and save the file to a(nother) server (afterwards), right?
and another task is to establish (also ami triggered) a call to a mobile and
play, lets say a voicefile.
this conversation should also be recorded and saved on a(nother) server
(afterwards), right?
let me know, if i understood you right, the solution is not so hard to
implement.
In what language do you preferrably write your AGIs? (although there is no
absolute need for using an
agi... you can all write down in your dialplan...)
is there a special protocol requirement for saving/transferring the recorded
voicefile (e.g. ftps)?
One obstacle is, that the recorded file is not fully written _immediately_
after stopmixmonitor or hangup...
this has to be taken care of and depending on your agi... it might be
interrupted, if the call is hungup...
but as you did not show your agi... these are just hints..
regards,
yves
Am 07.03.2013 16:21, schrieb Henrik Westerberg:
Hi,
I am developing a call recording application on Asterisk 11.2 and have this
configuration in my dialplan:
[macro-ccdev2-rec]
exten = s,1,MixMonitor(${ARG1},b)
[outgoing-originate]
exten = _X.,1,NoOp(Will send call to ${EXTEN})
exten = _X.,n,Dial(SIP/${EXTEN}@x.y.zmailto:EXTEN}@x.y.z)
[outgoing-originate-rec]
exten = h,1,Agi(agi://localhost/ajpbx.agi?path=uploadreccallid=${CC_CALLID})
exten = _X,1,NoOp(Will send call to ${EXTEN}, CC_CALLID is ${CC_CALLID},
CC_FILENAME is ${CC_FILENAME})
exten =
_X,n,Dial(SIP/${EXTEN}@x.y.z,60,Mmailto:EXTEN}@x.y.z,60,M(ccdev2-rec^${CC_FILENAME})e)
If I want to make a recorded server callout from 0 to 08 I then
originate a call via AMI to Local/0@outgoing-originate with context set
to outgoing-originate-rec and extension to 08.
The result will be something like this:
-- Executing [s@macro-ccdev2-rec:1]
MixMonitor(SIP/upps-ccm-tq01-003f, cbrec-15605.wav,b) in new stack
== Begin MixMonitor Recording SIP/upps-ccm-tq01-003f
-- Executing [h@outgoing-originate-rec:1] AGI(SIP/upps-ccm-tq01-003e,
agi://l4574/ajpbxtest.agi?path=uploadreccallid=15605) in new stack
-- SIP/upps-ccm-tq01-003eAGI Script
agi://localhost/ajpbxtest.agi?path=uploadreccallid=15605 completed, returning 0
-- Executing [h@outgoing-originate-rec-dev2:1]
AGI(SIP/upps-ccm-tq01-003f,
agi://4574/ajpbxtest.agi?path=uploadreccallid=) in new stack
-- SIP/upps-ccm-tq01-003fAGI Script
agi://localhost/ajpbxtest.agi?path=uploadreccallid= completed, returning 0
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/upps-ccm-tq01-003f
Unfortunately I get two different calls to the h extension, but this I can cope
with. The one without called is not interesting.
The uploading will fail since the MixMonitor is still on when I try to upload
the file. The file will not have a duration. It works when I schedule the
uploading a while after from my agi application but I
Re: [asterisk-users] Sending SMS from asterisk
Hi, I have some singe port GoIP GSM gateways, good price and after numerous updates their run several weeks before radio hangs and a cold start is needed. It has a WEB based feature to send and receive SMS, it had been nice to be able to send SMS from Asterisk with it. Tips on this wanted:) Best regards HB asterisk-users-requ...@lists.digium.com wrote: Re: Sending SMS from asterisk -- _ -- 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] Recording with MixMonitor and AGI
Hi,
Ok but when I use the macro the recording doesn´t start until the call is
answered which is a plus. It´s easy to trim away silence of course though.
But according to the documentation it seems like DeadAgi is obsolete in
Asterisk 1.6 and later, that AGI should be used instead.
Regards,
Henrik
Den 2013-03-08 05:30 skrev Bharat Lalcheta bharatlalch...@gmail.com:
As far as i understand your requirements, there is no need to use
macro for recording, You can directly call mixmonitor before Dial
application in your dialplan with required options. For transfer of
file, you are using AGI in h priority. However, you have to use
DeadAgi in h extension. As your channel already hangup, it can not
run on AGI.
Hope it will help you.
Regards,
Bharat Lalcheta
On Thu, Mar 7, 2013 at 8:51 PM, Henrik Westerberg
henrik.westerb...@ain.se wrote:
Hi,
I am developing a call recording application on Asterisk 11.2 and have
this
configuration in my dialplan:
[macro-ccdev2-rec]
exten = s,1,MixMonitor(${ARG1},b)
[outgoing-originate]
exten = _X.,1,NoOp(Will send call to ${EXTEN})
exten = _X.,n,Dial(SIP/${EXTEN}@x.y.z)
[outgoing-originate-rec]
exten =
h,1,Agi(agi://localhost/ajpbx.agi?path=uploadreccallid=${CC_CALLID})
exten = _X,1,NoOp(Will send call to ${EXTEN}, CC_CALLID is
${CC_CALLID},
CC_FILENAME is ${CC_FILENAME})
exten = _X,n,Dial(SIP/${EXTEN}@x.y.z,60,M(ccdev2-rec^${CC_FILENAME})e)
If I want to make a recorded server callout from 0 to
08 I
then originate a call via AMI to Local/0@outgoing-originate with
context set to outgoing-originate-rec and extension to 08.
The result will be something like this:
-- Executing [s@macro-ccdev2-rec:1]
MixMonitor(SIP/upps-ccm-tq01-003f, cbrec-15605.wav,b) in new
stack
== Begin MixMonitor Recording SIP/upps-ccm-tq01-003f
-- Executing [h@outgoing-originate-rec:1]
AGI(SIP/upps-ccm-tq01-003e,
agi://l4574/ajpbxtest.agi?path=uploadreccallid=15605) in new stack
-- SIP/upps-ccm-tq01-003eAGI Script
agi://localhost/ajpbxtest.agi?path=uploadreccallid=15605 completed,
returning 0
-- Executing [h@outgoing-originate-rec-dev2:1]
AGI(SIP/upps-ccm-tq01-003f,
agi://4574/ajpbxtest.agi?path=uploadreccallid=) in new stack
-- SIP/upps-ccm-tq01-003fAGI Script
agi://localhost/ajpbxtest.agi?path=uploadreccallid= completed,
returning 0
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/upps-ccm-tq01-003f
Unfortunately I get two different calls to the h extension, but this I
can
cope with. The one without called is not interesting.
The uploading will fail since the MixMonitor is still on when I try to
upload the file. The file will not have a duration. It works when I
schedule
the uploading a while after from my agi application but I would rather
not
rely on a timeout.
When I tried to run StopMixMonitor before the Agi call in the h
extension,
the first call fail and I never get any uploading with callid.
-- Executing [s@macro-ccdev2-rec:1]
MixMonitor(SIP/upps-ccm-tq01-0043, cbrec-15607.wav,b) in new
stack
== Begin MixMonitor Recording SIP/upps-ccm-tq01-0043
-- Executing [h@outgoing-originate-rec-dev2:1]
StopMixMonitor(SIP/upps-ccm-tq01-0042, ) in new stack
== Spawn extension (outgoing-originate-rec-dev2, h, 1) exited
non-zero on
'SIP/upps-ccm-tq01-0042'
-- Executing [h@outgoing-originate-rec-dev2:1]
StopMixMonitor(SIP/upps-ccm-tq01-0043, ) in new stack
== MixMonitor close filestream (mixed)
-- Executing [h@outgoing-originate-rec-dev2:2]
AGI(SIP/upps-ccm-tq01-0043,
agi://localhost/ajpbxtest.agi?path=uploadreccallid=) in new stack
Am I missing something here? I also looked at the possibility to
specify a
command to execute when MixMonitor stops but I would rather handle the
file
uploading in my agi application.
I also have another case: I want to dial out a call and record it. It
will
be a oneway-call from the server to a mobile. Do I need to get
AGI-control
of it and record with an AGI command or how can I hack it directly in
the
dial plan using MixMonitor?
Best Regards,
Henrik
--
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Re: [asterisk-users] Digium Wildcard TDM800P not working with DAHDI
What have you done so far to try and make it work? What version of CentOS are you using, what version of DAHDI, etc? On Sat, Mar 9, 2013 at 5:54 AM, Gilberto Sanches gisanch...@gmail.comwrote: Hello everyone, How can I let Digium Wildcard TDM800P work successfully with DAHDI? Because the Centos recognizes the card but I can't get the analog card working with DAHDI. Thanks in advance, Gilberto -- _ -- 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 -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- 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] Digium Wildcard TDM800P not working with DAHDI
I'm working for over 4 months now to fix this problem. The machine specs: On a P4 2.8Ghz, 1.5 GB RAM, 40 Gb HDD I installed: - I've tried AsteriskNOW 2.02 - Installed CentOS 6.3 and then compiled Asterisk 11, Libpri 1.4 and DAHDI 2.6 - Installed PBX in a Flash - Purple 20631 After the installations I used these commands: - dahdi_genconf - dahdi_cfg -v - /etc/init.d/dahdi restart - /etc/init.d/asterisk restart - shutdown -r now Created and configured: Extensions, Trunk, inbound route, outbound route. rebooted the system and then went into Asterisk console and check if channels are configured with these commands: - dahdi show status - dahdi show channels But didn't showed me any configured channels On Sat, Mar 9, 2013 at 9:06 PM, Warren Selby wcse...@selbytech.com wrote: What have you done so far to try and make it work? What version of CentOS are you using, what version of DAHDI, etc? On Sat, Mar 9, 2013 at 5:54 AM, Gilberto Sanches gisanch...@gmail.comwrote: Hello everyone, How can I let Digium Wildcard TDM800P work successfully with DAHDI? Because the Centos recognizes the card but I can't get the analog card working with DAHDI. Thanks in advance, Gilberto -- _ -- 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 -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- 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] Register Free Opensips/Asterisk Integration
Hello Everyone, I have gone through a few really good tutorials from the OpenSIPS site, Asterisk resources etc.. The unanswered question (and final piece of our puzzle) is if it's possible to have a register free environment in an OpenSIPS/Asterisk integration. Most approaches have OpenSIPS relay the UA's REGISTER request to Asterisk which has host=dynamic set for the Friend/Peer and everything works as expected. Where I run into problems is in Inbound calls. When I try to call the extension from a DID I am receiving Unable to create channel of type 'SIP' (cause 20 - Unknown). And rightfully so! Reason being: SIP Show Peers Yields: Name/username HostDynForcerport ACL Port Status Realtime 1001/1001 192.168.2.5N 5060 UNREACHABLE Cached RT TTrunk/sip.exp.com 192.168.2.5N 5060 UNKNOWN Cached RT As for who will keep track of the UA location, the OpenSIPS `location` table has the correct info: select username,domain,contact,socket from location; +--+++--+ | username | domain | contact| socket | +--+++--+ | 1001 | sip.exp.com | sip:1001@192.168.2.11:5060 | udp:192.168.2.5:5060 | +--+++--+ OpenSIPS: sip.exp.com OpenSIPS: 192.168.2.5 Asterisk: 192.168.2.10 UA: 192.168.2.11 I have set `host=sip.exp.com' for the UA but the UA is still `UNREACHABLE` by asterisk As for the rest of the media related stuff, everything works perfectly. Outbound works fine. As you know, this only poses a problem with inbound calls to the UAs. Your Help is Greatly Appreciated, Nick. -- _ -- 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] digium card and virualbox
Yes I installed Elastix on virualbox but was not able to capture the digium card. But really I did not check well how is the overall situation (accessing from web based, if any IP Phone can register, .. etc). If ubuntu can see the digium or not, also I did not try this. But meanwhile, I am using ubuntu 12.04 LTS. I do not know if someone who used Ubuntu can advise us in this issue? Regards Bilal --- hello regardless the virtual box, just in terms of Ubuntu, I have experience that Digium TP110p does now work with Ubuntu. it was long time a go I had this experience, I hardly could remember that what Ubuntu version I was using. my experience was on the Ubuntu system would not able to load DAHDI driver. for example: if you issue command dmesg |grep TE110 then it would say wct1xxp :04:00.0: Not Found something.. hope my experience would help you something by the way did you install Elastix in the virtual box ? Sent from Shitian Long On Mar 8, 2013, at 10:21 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; How to let the virualbox (ubuntu OS) to be able to see the digium card? Because when I install elastix or asterisk with dahdi, it is not able to see the digium card if the installation though the virualbox .. What is the solution? Regards Bilal -- _ -- 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] Sending SMS from asterisk
Dears; We have running here a SMS solution with four 3G dongles, which sends over 20.000 SMS a month. What are the elements of this solution? Is it only: 3G dongles and chan_dongle only? Or there are something else? About the script that you wrot it: This script is using asterisk (through AMI) to send the SMS? Or it is working without need for asterisk? In this case, where is the benifit of using Asterisk to send the SMS? Regarding to Sangoma and khomp: Do u mean that they have something like Huwewi 3G dongles? Regards Bilal -- Hi Bilal, It's not necessary to use a FXS port, you can compile install chan_dongle and buy a Huawei 3G dongle. We have running here a SMS solution with four 3G dongles, which sends over 20.000 SMS a month. In addition, I wrote an script able to send up to 12000 characters in concatenated SMS (the recipient receives a single SMS only) chan_dongle works very well. -- == Miguel Oyarzo Senior [ Network | Systems Design ] Engineer http://www.linkedin.com/in/mikeaustralia Linux User: # 483188 - counter.li.org Melbourne, Australia On 3/9/2013 1:09 PM, Gerardo Barajas wrote: Yes, you can check solutions from sangoma and khomp. Saludos/Regards -- Ing. Gerardo Barajas Puente Proyectos Especiales/Preventa | www.neocenter.com http://www.neocenter.com T:+52 (55) 8590-9000 x 7003 On Fri, Mar 8, 2013 at 6:32 PM, bilal ghayyad bilmar...@yahoo.com mailto:bilmar...@yahoo.com wrote: Hi; If my landline service provider does not provide the ability to send the SMS, and I need to send SMS from asterisk, then what is the required? How? Is it possible to send SMS from asterisk using SIM card to be connected via GSM adaptor connected to FXS ports? Or HOW? From the other side, this is existed only in asterisk 1.8 or it is existed in asterisk 1.4? Regards Bilal -- _ -- 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] digium card and virualbox
Hello Gertjan; I've heard a lot about it but I'm running Asterisk on ESXi5 Dell boxes without problems * How your ESXi saw the digium? Is it using PCI Passthru? Regards Bilal - It's called PCI Passthru and from what I've tried, the timing is horrible in a virtualized environment. VirtualBox and ESXi 5 Doug What are your experiences, Doug. I've heard a lot about it but I'm running Asterisk on ESXi5 Dell boxes without problems. Did you encouter the timing issues with a lot of concurrent calls? Where the boxs slammed bij other vm's at the time? --Gertjan On Fri, Mar 8, 2013 at 10:29 PM, Doug Lytle supp...@drdos.info wrote: How to let the virualbox (ubuntu OS) to be able to see the digium card? It's called PCI Passthru and from what I've tried, the timing is horrible in a virtualized environment. VirtualBox and ESXi 5 Doug -- _ -- 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] 回覆︰ 回覆︰ Directmedia question
I mean set DTMF =sip info ... not inband .. it sis work .. it do not relay on what codec you use .. it work I test before ... 寄件人︰ Mark Henry markhenry...@gmail.com 收件人︰ kingman chui chuiking...@yahoo.com.hk; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com 傳送日期︰ 2013年03月9日 (週六) 5:21 PM 主題︰ Re: [asterisk-users] 回覆︰ Directmedia question But that is not supported in g729 Inband DTMF is not supported on codec g729. Use RFC2833 Still media is through Asterisk On Sat, Mar 9, 2013 at 3:34 AM, kingman chui chuiking...@yahoo.com.hk wrote: If you want to use direcmedia = yes , in order take to effect.You must not set dtmf = rfc2833 .You should set it dtmf = info. It should work then. Regard/chui king man 寄件人︰ Mark Henry markhenry...@gmail.com 收件人︰ asterisk-users@lists.digium.com 傳送日期︰ 2013年03月9日 (週六) 7:23 AM 主題︰ [asterisk-users] Directmedia question Hello List, I have some doubt about direct media settings. I have an asterisk 1.8.14 instance running on 172.20.255.50, a soft phone on IP 10.100.210.51 and a gateway at 10.100.210.254 I have set both gateway and peer to directmedia=yes but still on gateway I see RTP from asterisk's IP, have tried setting nat=yes/no and also specifying localnet values but not sure where I am doing wrong. Also directrtpsetup is set to yes A sip debug and sip show peer output is here http://pastebin.com/5PwqJ1KW Please assist Thanks -- _ -- 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
