similar to: New Asterisk bounty: SIP simultaneous

Yes I used. The problem can be the version of Asterisk? I use Asterisk 13 instead of 11. 2016-02-12 14:31 GMT-02:00, Shabbir abbasi <shabbirabbasi92 at gmail.com>: > have changed this > [dongle0] > audio=/dev/ttyUSB1 > data=/dev/ttyUSB2 > > To > > [dongle0] > imei=123456789012345 > > and imei exact same as on your device ? > > On Fri, Feb 12, 2016

I tried this [dongle0] ;audio=/dev/ttyUSB1 ; tty port for audio connection; no default value ;data=/dev/ttyUSB2 ; tty port for AT commands; no default value ; or you can omit both audio and data together and use imei=123456789012345 and/or imsi=123456789012345 ; imei and imsi must contain exactly 15 digits ! ; imei/imsi discovery is available on Linux only

Yes, I used IMEI. But in CLI appearing nothing and it not register. 2016-02-12 14:27 GMT-02:00, Shabbir abbasi <shabbirabbasi92 at gmail.com>: > have you tried imei discovery > imei=123456789012345 > > > write imei number instaed of 12345... > > On Fri, Feb 12, 2016 at 8:51 PM, Vitor Mazuco <vitor.mazuco at gmail.com> > wrote: > >> Hi! >>

Hi! I'm trying to use dongle in my Asterisk But appear for me all time this error [Feb 12 13:49:10] ERROR[13347]: chan_dongle.c:442 do_monitor_phone: [dongle0] timedout while waiting 'OK' in response to 'AT' -- [dongle0] Error initializing Dongle -- [dongle0] Dongle has disconnected -- [dongle0] Trying to connect on /dev/ttyUSB1... -- [dongle0] Dongle has

Any body used patch, http://bugs.digium.com/bug_view_page.php?bug_id=0001719 to get the callerid for BT Line. I applied the patch successfully but could not get it to work. Any help. Here are the logs: -- Starting simple switch on 'Zap/1-1' Jun 17 18:22:31 NOTICE[426000]: chan_zap.c:4811 ss_thread: Got event 2 (Ring/Answered)... Jun 17 18:22:34 NOTICE[426000]: chan_zap.c:4811

Hi. I've searched the archives and found nothing regarding collaborating Asterisk with a Panasonic PBX (TD1232 to be exact) Here's my question: Can I use a Wildcard X100P to connect an outgoing line jack (on the Panasonic) to Asterisk, so I can route calls from the PBX to Asterisk, and calls from Asterisk to the PBX? On the hardware page for the X100P card is says it's great for

Andrew Kohlsmith wrote: >I wasn't talking about bandwidth but rather lengthy >Dial() commands... > >exten => s,1,Dial(SIP/someuser&SIP/someuser&SIP ...... > >kind of thing... seems awfully unwieldy. That's why you would stick the members into a global variable [globals] DIYCALLGROUP => SIP/111&SIP/112&SIP113 etc. then dial using

I need a cheap simple adaptor for analog phones to use with Asterisk. It should be some kind of "configure and forget" type of device, to use at the office, or just throw it in a road warrior's bag and use it while travelling, to call back to the "mothership". I can't decide between iaxy and sipura. Can you guys help? Which one would you use? (and why?) I feel that iaxy

Dear R users, I'd like to make this data rem.y = c(-1,0,2,4,5) from y = c(-1,-1,0,2,2,2,2,4,4,5,5,5,5,5). That is, I need to remove repeated values. Here is my code, but I don't think it is efficient. How could I improve this? #------------------------------------------------------------------------ y = c(-1,-1,0,2,2,2,2,4,4,5,5,5,5,5) n=length(y) for (i in 1:n) #

Dear List, I frequently use the data() function to load csv files (with separator ";") into R session, typically data(myfile) loads myfile.csv from my working/data directory into R. Now, in 1.4.0 version, everything works as expected, but with one difference: The values readed in older versions in "num" mode are now readed as "int" mode, converting the values

Hi there, I tried to get a few "Optipoint 400 SIP" working with *, but it refused to work properly. In my testing-network i have two Sjphones (they are working really fine) and three optipoints. I?m able to dial the number of a Sjphone on all of the optipoints. The call is signalled at the Sjphone with the right number of the optipoint and an connection can be established. But when I

I know someone who's at VON this week. Apparently Mark Spencer was up there talking about how Asterisk supports SRV. Sounds like vaporware to me. > -----Original Message----- > From: David Thomas [mailto:punknow@gmail.com] > Sent: Thursday, March 16, 2006 11:54 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Feedback from VON

I have coded chan_sip.c so that you can have // sip.conf register => username:password@somedomain.com/redirectconfig [redirectconfig] redirect=yes redirecturi=sip:12345@domain1.com redirecturi=sip:34556@domain2.com redirecturi=sip:87877@domain3.com .... so when you receive a call it will redirect to the alternating uri's with a SIP 300 Message. It works with the following sequence,

Hi everybody, I am testing Optipoint 400 Standard SIP (Firmware 2.3.14) with Asterisk. It is posible to dial from another Phone (x-lite) to the Optipoint, but when I try to dial from the Optipoint there is no dialtone and there is only a short beep when I dial Numbers. The Optipoint shows "no Server..." (Registrar?) in Display. Sip debug shows no unusual (to me) Messages. Sip show

I want to create a general model ''Product'' that I will be able to store several attributes in it. For example the attributes :id :name :description are pretty stantard. So each product will have this attributes. However, I wanted to know if it is possible to create dynamic attributes. In the case of a cell phone device I might want to store the IMEI of the phone, this

Have anyone tried to interface BT's Broadband Voice with asterisk? Kannaiyan

I'm trying to use OpenBTS with Asterisk. I have two phones that are connecting to OpenBTS correctly, but on the Asterisk side the phones can't call each other. I followed this guide: http://gnuradio.org/redmine/projects/gnuradio/wiki/OpenBTSSettingUpAsterisk I set up two phones in sip.conf and extensions.conf. In my SIP output I see this: WARNING[1689]: app_dial.c:2041 dial_exec_full:

I am thinking about using the G729 codecs on my endpoint devices and purchasing some G729 licenses for Asterisk but I have several questions: 1. Which G729 codec is sold by Digium for Asterisk, G729, G729A, B...I? 2. If I have G729A on one end and G729B on the other, are they compatible? I have looked all over the place for question 2, but without buying the ITU docs I cannot seem to find this

Hello, I?m looking for Optipoint 420 Standard SIP Firmware to make my first tests with Asterisk and SIP, but I?m unable to find it. Could someone help me? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070415/c2da9cc0/attachment.htm

hi! anyone try siemens optipoint 400 economy SIP phone with * ? -- http://www.siemens.com/Daten/siecom/HQ/ICN/Internet/Enterprise_Networks/WORKAREA/skuch_c/templatedata/English/document/binary/a31002-h1000-a250-2-7629.pdf Thomas