Re: [asterisk-users] Mailing List Future

On Mon, 2023-12-04 at 08:00 -0400, Joshua C. Colp wrote: > To that end, we’ve decided to discontinue the mailing lists effective > February 1st, 2024. That's a very sad news! :-( -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] audio from soft phone actual phone from cloud

On Wed, 2023-07-19 at 12:42 -0400, Jerry Geis wrote: > Why might I not be getting audio ? Make sure the RTP port range is correctly configured and open on your server's firewall. The port range is defined in /etc/asterisk/rtp.conf The same range of UDP ports must be correctly forwarded on your

Re: [asterisk-users] Intro and question

Even I was confused, and the directions in that book seem like a complication of a simple affair, at least for my modest needs. Finally, I installed Asterisk with apt and created extensions.conf and pjsip.conf files. -- _ --

Re: [asterisk-users] Dial() after the h extension has been invoked?

On Fri, 2021-11-12 at 16:56 +, Antony Stone wrote: > I use Dial() commands with custom SIP headers to pass information > (eg: about > the current state of a call) between the front-end and back-end > machines, and > this works very well. > > I need to perform a Dial() > command after an

Re: [asterisk-users] Notifying missed calls

On Sat, 2021-11-06 at 14:46 +0100, Luca Bertoncello wrote: > Really, I can't understand what you mean... I'm feeling really > dumb... No need to feel dumb. I'm not an expert and when I look to my extensions.conf... well... countless pulling my hairs out, head banging on the keyboard,,, :-) The

Re: [asterisk-users] Notifying missed calls

On Fri, 2021-11-05 at 10:50 +0100, Luca Bertoncello wrote: > 1) The E-Mails will be sent "double" It sends the first mail by executing "noanswer,2" and a second mail because because of "main-incoming,h,2" > 2) The E-Mails will be sent for outgoing unanswered calls, too. Use the "h" extension

Re: [asterisk-users] Notifying missed calls

Here my configuration: [incoming] ; Incoming from Swisscom exten => +4191xxx,1,NoOp(Call from ${CALLERID(num)}) same => n,Dial(SIP/deskphone,120) same => n,Hangup() exten => h,1,GotoIf($["${DIALSTATUS}" = "ANSWER"]?done) exten => h,n,System(echo "Missed Call from ${CALLERID(num)}" |

Re: [asterisk-users] Detect if people is talking

On Wed, 2020-12-30 at 12:09 -0300, Valter Nogueira wrote: > Is there any way to detect if an agent is speaking? https://wiki.asterisk.org/wiki/display/AST/Application_WaitForSilence https://wiki.asterisk.org/wiki/display/AST/Application_WaitForNoise --

Re: [asterisk-users] which linux for asterisk?

On Wed, 2020-12-09 at 11:03 +0400, Dmitry Melekhov wrote: > what is best choice ? Oracle? Ubuntu? I'm running Asterisk since several years on Ubuntu without any issues. Debian should be fine too. -- _ -- Bandwidth and

Re: [asterisk-users] Anyone that know of DECT "client" for asterisk?

On Sat, 2020-10-03 at 22:25 +0200, Sebastian Nielsen wrote: > many providers in sweden have started disabling SIP account details > and now require usage of their own ”router’s”. That's very irritating and make me angry. Few of my client had the same problem. The solution: write a letter asking

Re: [asterisk-users] Mail2Fax

On Wed, 2020-06-17 at 18:10 +0200, basti wrote: > txfax seem to be a port of spandsp. it is also old. > Is there a newer way to send fax via asterisk. I don't know if it's newer, but I use "sendfax" -- _ -- Bandwidth and

Re: [asterisk-users] email notification on missed call

On Sat, 2019-11-02 at 11:42 +0100, Antony Stone wrote: > Doesn't that send an email for every call once it ends, not just > unanswered ones? Whoops! You are right! :-) exten => h,1,GotoIf($["${DIALSTATUS}" = "ANSWER"]?done) exten => h,n,System(echo "Missed Call Open on Asterisk from

Re: [asterisk-users] email notification on missed call

On Wed, 2019-10-30 at 05:10 +0100, Fourhundred Thecat wrote: > what is the best way to implement email notification on missed call ? > Is there perhaps a better way to this than described above ? This is my way: exten => h,1,System(echo "Missed Call Open on Asterisk from ${CALLERID(num)}" |

Re: [asterisk-users] Asterisk 13.28.0 Now Available

Thank you, dear Asterisk Development Team, for this great software! > The Asterisk Development Team would like to announce the release of > Asterisk 13.28.0. -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Sending SMS and SIM card

On Fri, 2019-04-26 at 14:39 +, bilal ghayyad wrote: > Any small example how to send gsm calls through chan_dognle and how > to send sms through chan_dongle? To send SMS, there is a CLI command. You can use the commands in your extensions.conf accordingly your needs.

Re: [asterisk-users] Sending SMS and SIM card

On Fri, 2019-04-26 at 14:39 +, bilal ghayyad wrote: > Any small example how to send gsm calls through chan_dognle and how > to send sms through chan_dongle? In dongle.conf: [gsmgateway] context=gsm imei=123456789012345 imsi=098765432112345 In extensions.conf: [gsm] ; Incoming calls from

Re: [asterisk-users] Sending SMS and SIM card

You can use a cheap 3G-USB-dongle and chan_dongle. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk?

Re: [asterisk-users] Asking

On Tue, 2018-09-18 at 20:28 +0200, modou lo wrote: > Hello, please can i have a code which help me to tax user every voip > services in asterisk means when user starts to call someone Check Asterisk2billing  http://www.asterisk2billing.org/ --

Re: [asterisk-users] Decoding SIP register hack

On Thu, 2018-05-17 at 11:18 -0400, sean darcy wrote: > 3. How do I set up the server to block these ? > > 4. Can I stop the retransmitting of the 401 Unauthorized packets ? I'm happy with Fail2Ban protecting my Asterisk 13. Here is my configuration: in /etc/asterisk/logger.conf: messages =>

Re: [asterisk-users] invite to conference by a call file

Maybe something like a local web page where your secretary can enter the list of phone numbers to call and a script that generates a call file and moves it to the Asterisk spool folder. But that's not an Asterisk issue. It's more a programmer's issue. :-)  On Thu, 2018-03-22 at 16:06 +0200,

Re: [asterisk-users] invite to conference by a call file

outgoing folder of LINUX shared through samba in LAN. i need to > make it as easy as possible, please. > > On Tue, Mar 20, 2018 at 5:41 PM, Frank Vanoni <mailinglist@linuxista. > com> wrote: > > Here I'm using the "Page" application to make a conference call &quo

Re: [asterisk-users] invite to conference by a call file

Here I'm using the "Page" application to make a conference call "on the fly". [office] exten => ,1,Dial(SIP/desk2,150)    same => n,Hangup() exten => ,1,Dial(SIP/desk3,150)    same => n,Hangup() exten => ,1,Dial(SIP/desk4,150)    same => n,Hangup() exten =>

Re: [asterisk-users] Blacklist failed attempts

On Thu, 2018-03-01 at 15:02 +0200, Atux Atux wrote: > I have tried to implement it through fail2ban, but it doe snot seem > to work for my asterisk implementation. I'm happy with Fail2Ban protecting my Asterisk 13. Here is my configuration: in /etc/asterisk/logger.conf: messages =>

Re: [asterisk-users] email when certain numbers are called

On Mon, 2018-01-15 at 14:26 +0200, Atux Atux wrote: > [DefaultPlan] exten => _XX,1,System(echo "Dialed number ${EXTEN} on Asterisk from ${CALLERID(num)}" | mail -s "Dialed number ${EXTEN} on Asterisk from ${CALLERID(num)}" -a "From: Asterisk PBX " yo

Re: [asterisk-users] SIP invite timeouts : how is someone sending invites from our server ??

> fail2ban is most useful for blocking registration attempts.    I > handle  > non-registration call attempts by allowing guests, point them to a > jail  > context, which runs Log(WARNING,fail2ban='${CHANNEL(peerip)}')   I > set a  > fail2ban rule to match that line logged from Asterisk. Thanks

Re: [asterisk-users] SIP trunks going to the wrong context

I don't know if it applies to your problem, but I also had some troubles with multiple account on same SIP provider.  Here what works for me: In sip.conf: register => 11:qwe...@sip.provider.zz/11 ; Trunk1 register => 22:asd...@sip.provider.zz/22 ; Trunk2 register =>

Re: [asterisk-users] Looking for the carrier that owns a particular DID

On Thu, 2017-11-02 at 11:33 -0400, Tech Support wrote: > How do I find out which carrier owns the DID in question? Try here: https://www.twilio.com/lookup -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] PJSIP, NAT and STUN/ICE

On Tue, 2017-10-10 at 11:32 +0200, Frank Vanoni wrote: > On Mon, 2017-10-09 at 23:56 +0200, O. Hartmann wrote: > > > > > local_net=  192.168.254.1/24 > > It should be: > > localnet = 192.168.254.0/255.255.255.0 Whoops.

Re: [asterisk-users] PJSIP, NAT and STUN/ICE

On Mon, 2017-10-09 at 23:56 +0200, O. Hartmann wrote: > local_net=  192.168.254.1/24 It should be: localnet = 192.168.254.0/255.255.255.0 -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Dial an extension to modify dialplan

On Wed, 2017-05-10 at 12:56 +0200, Frank Vanoni wrote: > exten => 2001,1,Dial(SIP/Dial(SIP/deviceA/deviceB/deviceC) > > exten => 2002,1,Dial(SIP/Dial(SIP/deviceA/deviceB) Whoops... sorry for the typo (in the hurry of copy & paste)! exten => 2001,1,Dial(SIP/deviceA/d

Re: [asterisk-users] Dial an extension to modify dialplan

Dear Digium List First of all, I thank all of you for all the replies and the interesting suggestions. I thank you very much. I can only learn from people like you. :-) I will remember all the different solutions for a future use in other scenarios. On Mon, 2017-05-08 at 16:35 +0200, Frank

[asterisk-users] Dial an extension to modify dialplan

=> 4001,3,System(/usr/sbin/asterisk -rx "dialplan reload") exten => 4001,4,Playback(service) -- end snip - twodevices.txt contains exten => ring,1,Dial(SIP/deviceA) alldevices.txt contains exten => ring,1,Dial(SIP/deviceA/deviceC) By dialing 4000 or 4001,

Re: [asterisk-users] log incoming calls without answering

On Thu, 2017-04-20 at 17:26 -0300, Fabio Moretti wrote: > Any idea? I used to play with an analog telephone line and Asterisk by using a Linksys SPA-3102 Voice Gateway. I think it is no longer manufactured, but maybe you con buy a used one on eBay or you can find an equivalent device from

Re: [asterisk-users] Disallow CALLS without registry

On Sat, 2017-02-11 at 12:25 +1300, Pete Mundy wrote: > > sip.conf configuration > > In the [general] section, define: > > [general] > > ... > > allowguest=no > > alwaysauthreject=yes > > ... > With the above configuration on my Asterisk, I obtain the following result: - if the phone is

Re: [asterisk-users] Disallow CALLS without registry

On Thu, 2017-02-09 at 14:58 +0200, Антон Сацкий wrote: > so the main question is -- how to Disallow CALLS without registering > on PBX sip.conf configuration In the [general] section, define: [general] ... allowguest=no alwaysauthreject=yes ... The "allowguest" line disables anonymous SIP

Re: [asterisk-users] MOBILE SIMCARD ON ASTERISK

Hi Chris On Tue, 2016-12-06 at 04:36 +0200, christopher kamutumwa wrote: > Is it possible to have a simcard configured and become incoming line > and outgoing on asterisk and also have the IVR function? Yes, it is possible! :-) A cheap solution is using a 3G-UBS-dongle. I have two SIM cards

Re: [asterisk-users] missed call notification

On Mon, 2016-11-28 at 14:31 +0100, tux john wrote: > Hi. i am running asterisk 11 in debian and i would like have a missed > call notification down to extension level. > so if i get a missed call to extension 6589 then send an email to the > user's email address with a subject and a text message.

Re: [asterisk-users] Blacklist callers from file

On Sat, 2016-08-27 at 17:59 +0200, tux john wrote: > Hi. I would like to blacklist a few callers Example: callers with CallerID 0123456789, 9876543210 and 7410258963 are sent to tt-monkeys. Callers from area code 555 are also blocked. In "extensions.conf" file add #include "blacklist.conf"

Re: [asterisk-users] IAX UNREACHABLE : Ignoring bindport/bindaddr on reload

On Fri, 2016-08-26 at 10:12 -0300, Vitor Mazuco wrote: > bindaddr = all Try: bindaddr=0.0.0.0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon,

Re: [asterisk-users] rasberry pi

I'm currently using Asterisk 11.7.0 on a Raspberry Pi 2 Model B with Ubuntu Server 14.04. Works fine! :-) Frank On Wed, 2016-07-06 at 01:10 -0700, Thufir wrote: > I'm debating between a cloud PBX or, perhaps, rasberry pi. For a > SOHO, maybe three hardphones, rasberry pi would suffi

Re: [asterisk-users] Including doesn't have any effect

esting and detailed explanation. I'll go over the books again and rewrite the little "black box" taking in consideration your suggestions. Thanks again! Best regards Frank -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Including doesn't have any effect

area code or a block of numbers (eg. 321-654-8XXX) the blacklist function is useless. Correct me if I'm wrong. Frank -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us f

Re: [asterisk-users] Including doesn't have any effect

ge :) Well... I'm not an expert and my approach is by "trial and error". It works perfectly. :-) Frank -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live intr

Re: [asterisk-users] Including doesn't have any effect

Another possible approach to blacklist two (or more) specific callers (098765432 and 012345678 as example) In extension.conf #include "blaklist.conf" exten => _+x.,1,Gosub(blacklist,s,1) exten => _+x.,n, exten => black,1,playback(tt-monkeys) In blacklist.conf exten =>

Re: [asterisk-users] How to set outgoing sip callid ?

In sip.conf [devicename] callerid="Jon Doe" <+123456789> or in extensions.conf exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) exten => 1234,n,Dial(SIP/ -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] VoipRaider is true for FREE calls?

On Mon, 2016-05-09 at 19:43 -0300, Vitor Mazuco wrote: > VoipRaider the site, says calls to landlines in Brazil... I hope I'm not infringing any mailing list rule by recommending you to take a look to the following providers. I use them with my Asterisk, the rates are good and they allow

Re: [asterisk-users] VoipRaider is true for FREE calls?

VoipRaider is a service from DELLMONT SARL. This company offers voip services under dozens of different domains (voipcheap, voipdiscount, onevoip,...) Search "Dellmont Sarl" in Google and read the user's reviews. Personally, I would never send a penny to them... Franky --

Re: [asterisk-users] "Follow me" with Asterisk that detects cellphone voicemail and similar announcements

ts IAX (for example, Zoiper). Frank -- _ -- 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

Re: [asterisk-users] How to install Huawei E153 in a Asterisk 11 or 13?

On Wed, 2016-03-02 at 19:12 -0300, Vitor Mazuco wrote: > I tried to install chan_dongle for Asterisk 11 in a Ubuntu 14.04, but > my Huawei E153 is not working in my Asterisk. > But not successes. A little more information from you would be helpful to identify the problem. I have a Huawei USB

Re: [asterisk-users] Handle a call if one phone of a ring group is busy

On Sun, 2016-02-28 at 01:43 +0100, Frank wrote: > Question: How to give a "busy signal" back to the caller if one > extension of a ring group is in use? Or redirect the call to voice mail? Found a solution! :-) exten => 7654321,1,GotoIf($["${DEVICE_STATE(SIP/111)}

[asterisk-users] Handle a call if one phone of a ring group is busy

to the caller if one extension of a ring group is in use? Or redirect the call to voice mail? Any hint? Thanks in advance Frank -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join

Re: [asterisk-users] Voice recognition IVR Is it possible?

I gave up since everything seems a little too complicated for something I just wanted to try as a test. Thanks again! Frank :-) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

Re: [asterisk-users] Voice recognition IVR Is it possible?

On Tue, 2016-02-23 at 17:06 +, Steve Howes wrote: > Google?... Yeah... searched "google voice recognition api asterisk", browsed though various results. Nothing helpful for a beginner, very confusing bla bla... Thanks anyway for your help. F. --

Re: [asterisk-users] Voice recognition IVR Is it possible?

On Tue, 2016-02-23 at 00:43 +0100, Laszlo wrote: > > Requirements > > ... > Speech API key from Google Yes... OK... but... where and how can I obtain this API Key? Where and how do I install it into my Asterisk box? --

Re: [asterisk-users] Voice recognition IVR Is it possible?

g. Aborting. " :-( What am I missing? Any hint? Frank -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.as

Re: [asterisk-users] Windstream SIP Trunk settings

On Mon, 2016-02-22 at 08:20 -0500, James Cass wrote: > register string: :@:5060 Try: register => 5551231234:sec...@sipdomain.com/5551231234 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to

Re: [asterisk-users] [dongle0] timedout while waiting 'OK' in response to 'AT'

On Fri, 2016-02-12 at 14:33 -0200, Vitor Mazuco wrote: > Yes I used. > > The problem can be the version of Asterisk? > > I use Asterisk 13 instead of 11. Try [dongle0] imei=347654458453667 imsi=976895757545778 -- _ --

Re: [asterisk-users] NAT traversal for mobile app softphones - best strategy?

disallow=all allow=alaw allow=ulaw tcpenable=yes Here the configuration for a mobile device with a softphone (Android and Zoiper) ;Mobile phone [mobile1] type=peer callerid="Frank " <+987654321> nat=force_rport,comedia qualify=6000 host=dynamic secret=mysupersecretpassword canrein

[asterisk-users] Peer Reachable / Unreachable on TLS

P is fine, TLS is intermittent. Any idea? Thanks for any hint Frank [android1] type=peer callerid="Abcd Efgh" <+1234567890> nat=force_rport,comedia qualify=6000 host=dynamic secret=qt528frh3bAW3 tcanreinvite=no context=venomp

Re: [asterisk-users] Asterisk 11 and old Thomson 2030S Hardphone => SIP Register/Auth Problem against V11

On Mon, 2016-01-11 at 14:52 +0100, Juergen Sauer wrote: > It seems to be, that this fw can not deal with not-numeric-sip accounts. > I entered the extension number as name, account and it works. Glad to hear that. Very interesting. Good to know! > Solved by my self, using Try-and-error Metodic.

Re: [asterisk-users] ST2030 replacement

On Thu, 2016-01-07 at 17:35 +0100, Sil wrote: > Can you give me a return on the models you use ? Yealink T26P -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] placing calls with linphone.org SIP account

On Wed, 2016-01-06 at 11:27 +0100, Yves wrote: > how can I call other users > registered at other SIP-Providers? > I tried all well-known SIP URI Syntaxes but none worked... does anyone > reliably know, if it is possible at all and if so, what is the > dialstring looking like? It depends if

[asterisk-users] Manipulating of a dialed sequence

Hi Asterisk List Given, as an example, the following sequence 012345*543210 I would like to store into a variable all digits before "*" (012345) and in a different variable all digits after the "*" (543210) for further processing in the dial plan. The length of the dialed sequence may be

Re: [asterisk-users] Manipulating of a dialed sequence

Steve and Rafael, that is exactly what I was looking for! Many thanks for your help! :-) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar

[asterisk-users] How to hang-up a FXO call without answering it?

-pstn, s, 5) exited non-zero on 'DAHDI/4-1' [Sep 19 11:17:49] -- Hanging up on 'DAHDI/4-1' [Sep 19 11:17:49] -- Hungup 'DAHDI/4-1' The caller just hears the line ring and ring and the SIP extensions are dialed over and over until the caller hangs-up. Is there anyway to force a hang-up or di

[asterisk-users] Anyone running Asterisk on KVM virtual machine under SmartOS?

I'm thinking of condensing some of my boxes down to KVM virtual machines running under SmartOS. My Asterisk box is running Centos 6.4 and I'd like to include it. Is anyone running Asterisk on a virtual machine under SmartOS? Does DAHDI work? Thanks in advance. Frank

Re: [asterisk-users] GSM Sip Gateway

So what is the official page to get those GoIP ? All I can find is on ebay.. On 2/24/13 5:23 AM, longst wrote: I am using GoIp 1 channel gateway. it is fine Sent from Shitian Long On Feb 24, 2013, at 1:54 AM, Frank fr...@efirehouse.com wrote: Hi all, Anyone ever used GoIP GSM SIP Gateways

Re: [asterisk-users] GSM Sip Gateway

Ha... Here is the other company I was looking for: http://www.yx.cl Anyone is using their GSM gateways ? On 2/26/13 11:56 AM, Frank wrote: So what is the official page to get those GoIP ? All I can find is on ebay.. On 2/24/13 5:23 AM, longst wrote: I am using GoIp 1 channel gateway

Re: [asterisk-users] GSM Sip Gateway

USA, this will be use with a 4G network. On Feb 24, 2013, at 5:24 AM, longst longst...@gmail.com wrote: where are you from by the way Sent from Shitian Long On Feb 24, 2013, at 1:54 AM, Frank fr...@efirehouse.com wrote: Hi all, Anyone ever used GoIP GSM SIP Gateways ? If yes

Re: [asterisk-users] GSM Sip Gateway

So anyone would know a gateway working on 3G/4G network ? I remember a website called XY something (I cant find it anymore. I don t remember if it was xywireless.com , or xytelecom.com , or something else) where they seemed to have good gateways, but I can't find it anymore. On 2/24/13 9:15

[asterisk-users] GSM Sip Gateway

Hi all, Anyone ever used GoIP GSM SIP Gateways ? If yes, what was your experience with those ? I'm looking at this: http://www.ebay.com/itm/HOT-GSM-VOIP-GoIP-Gateway-SIP-Trunk-to-Asterisk-iP-PBX-/280736774012?pt=US_VoIP_Business_Phones_IP_PBXhash=item415d37377c If anyone has any (good)

Re: [asterisk-users] Asterisk Realtime Extension... strange behaviour

Remove the line _X. , and try 3 digits other than 110 112 , let us know if it works. On 2/12/13 5:55 AM, Yves A. wrote: Hi, I encountered a strange behaviour using realtime extensions... (on Asterisk 11.2) when I use the following static dialplan, everything works as expected..: [from-sip]

[asterisk-users] Google talk not (re)connecting after network down

Hi all, I notice yesterday night while doing tests of uptime that if I unplug my network from the internet, then plug it back, my jabber still shows connection to google, but no outgoing calls are going out, and nothing is coming in (calls are going in google vmail since there is no

[asterisk-users] Asterisk calls between 2 private networks

My apologies if this topic was already discussed in the past. Here is my scenario: Network A - 192.168.1.0 1 Asterisk 1 Digium phone Router does NAT from the public IP to asterisk, and forward ports 5060tcp/udp and 10k-20k udp Network B - 192.168.1.0 1 Digium phone, registering to the public

Re: [asterisk-users] Asterisk calls between 2 private networks

. On 2/7/13 10:46 AM, A J Stiles wrote: On Thursday 07 February 2013, Frank wrote: My apologies if this topic was already discussed in the past. Here is my scenario: Network A - 192.168.1.0 1 Asterisk 1 Digium phone Router does NAT from the public IP to asterisk, and forward ports 5060tcp/udp

Re: [asterisk-users] Asterisk calls between 2 private networks

...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Thursday, February 07, 2013 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk calls between 2 private networks AJS, That is a solution that I am

Re: [asterisk-users] Asterisk calls between 2 private networks

...@lists.digium.com] On Behalf Of Frank Sent: Thursday, February 07, 2013 9:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Eric Wieling Subject: Re: [asterisk-users] Asterisk calls between 2 private networks I thought about that. I will give it a shot

Re: [asterisk-users] Asterisk calls between 2 private networks

-8208 From: Frank fr...@efirehouse.com To: ch...@acsdi.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Date: 02/07/2013 12:06 PM Subject: Re: [asterisk-users] Asterisk calls between 2 private networks Sent by: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] Asterisk calls between 2 private networks

And actually I did not have directmediadeny=0.0.0.0 But I had directmedia=no. So I will add the directmediadeny line, and will check it out again tonight. On 2/7/13 1:22 PM, Frank wrote: i think canreinvite is not part of Asterisk 1.8 anymore. Asterisk 1.8 added directmediapermit

[asterisk-users] [FIXED] Asterisk calls between 2 private networks

Got it to work tonight. So once again this is my network: Network A: 192.168.1.x Network B: 192.168.1.x In between, the internet. Asterisk is in Network A. 1 Digium phone is in network A. Router from network A does NAT and forward (for now): - 5060 TCP/UDP to internal IP of asterisk - 10k-20k

Re: [asterisk-users] CallerID external call after Attended Transfer

What is the PAI option below that you are talking about, for sendrpid ? The manual only says that yes or no can be used.. On 2/4/13 9:39 AM, Kevin Larsen wrote: One thing you can try is to set the following in your sip.conf. sendrpid=pai trustrpid=yes You can put that on individual phone

Re: [asterisk-users] Google voice with no voice

server in the internet with a public IP Use Google Voice Even if you have asterisk on a private network, but have the same kind of solution working for you, I'd love to hear your story.. On 1/22/13 9:55 AM, Christopher Harrington wrote: On Mon, Jan 21, 2013 at 9:59 PM, Frank fr...@efirehouse.com

Re: [asterisk-users] Google voice with no voice

[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Tuesday, January 22, 2013 10:18 AM To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Chris, I covered the whole 74.125.225.* subnet. Even if I open

Re: [asterisk-users] Google voice with no voice

. This will (hopefully) show you where the out of range condition is occurring. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Danny Nicholas Subject: Re: [asterisk-users] Google

Re: [asterisk-users] Google voice with no voice

/13 2:02 PM, Danny Nicholas wrote: If you needed a MITM, nothing would work now. The incoming call is connecting, but no voice or no connection at all? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 11:56 AM To: Danny Nicholas Subject: Re

Re: [asterisk-users] Google voice with no voice

commands? GV isn't a SIP call per se, so the incoming line would be a gtalk peer. Try these commands from CLI Gtalk show peers Core help gtalk -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:04 PM To: Danny Nicholas Cc: Asterisk Users Mailing

Re: [asterisk-users] Google voice with no voice

*CLI jabber show connections Jabber Users and their status: [asterisk] r...@gmail.com - Connected Number of users: 1 On 1/22/13 2:14 PM, Danny Nicholas wrote: What about jabber show channels? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday

Re: [asterisk-users] Google voice with no voice

-MTAuMTIu ulaw ulaw 1 active gtalk channel The only difference is the WRITE column that changes from SLIN to ULAW On 1/22/13 2:22 PM, Danny Nicholas wrote: This is incoming, outgoing or idle (no call)? -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday

Re: [asterisk-users] Google voice with no voice

for everything to work. It slightly sucks, but I'll take it. On 1/22/13 2:29 PM, Danny Nicholas wrote: This sounds like a codec issue. Set your verbose to 10 and retry the incoming call. -Original Message- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:26 PM

Re: [asterisk-users] Google voice with no voice

. If there anything special to compile for this ? Thanks On 1/22/13 2:54 PM, Joshua Colp wrote: Frank wrote: OK, so here is the new.. By mistake, when I picked up the D70 , I pushed the 2 button. I suddenly heard google voice saying Okay, I'll send the caller to voicemail. So I called again.. picked

Re: [asterisk-users] Google voice with no voice

My bad, I found it not loaded in my modules.conf. This is now working. What a pain. Is there a wiki page I can update in order to share the configuration and how to have this work, with everybody ? On 1/22/13 2:58 PM, Joshua Colp wrote: Frank wrote: Hi , So I tried Answer() Wait(1

[asterisk-users] Asterisk, Digium phones, and voicemail.

Hi all, I registered my Digium D70 using a name (D70) instead of a number. Is there a way to program Asterisk (or the phone?) so when I press the MSGS button, it automatically goes to the correct voicemail, with or without asking me for a password ? As of now, it asks me for my mailbox

Re: [asterisk-users] Asterisk, Digium phones, and voicemail.

- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Sent: Tuesday, January 22, 2013 2:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk, Digium phones, and voicemail. Hi all, I

Re: [asterisk-users] Asterisk, Digium phones, and voicemail.

module res_digium_phone.so Command 'module load res_digium_phone.so' failed. Any way to get more information about what's wrong ? Thanks On 1/22/13 4:27 PM, Christopher Harrington wrote: On Tue, Jan 22, 2013 at 2:27 PM, Frank fr...@efirehouse.com mailto:fr...@efirehouse.com wrote: Hi all

[asterisk-users] Google voice with no voice

Greetings all, I was reading the documentation tonight, and decided to try Google voice with my asterisk. I was able to setup iksemel, connect to google using jabber, and connect to google voice using gtalk. Here is my physical configuration: Digium D70 -- private network 192.168.1.x --

Re: [asterisk-users] Google voice with no voice

Actually, the funny thing is that it works randomly. I just tried out of the blue calling from D70 through Google Voice to a cell phone, and it worked. I hung up, redial, and no audio at all. On 1/21/13 10:38 PM, Frank wrote: Greetings all, I was reading the documentation tonight

Re: [asterisk-users] DIDForSale spam

With all my respect guys, I do have my asterisk mailing list setup as send-as-soon-as-their-is-a-message. I'm getting too many email from this thread that I seriously don't care about, and that should be taking out of here. If you guys want to discuss, I suggest you email between each other,

[asterisk-users] Auto ban IP addresses

Greetings all, I have been seeing a lot of [Jan 2 16:36:31] NOTICE[7519]: chan_sip.c:23149 handle_request_invite: Sending fake auth rejection for device 100sip:100@108.161.145.18;tag=2e921697 in my logs lately. Is there a way to automatically ban IP address from attackers within asterisk

[asterisk-users] Should a Linksys Sipura 2102 be configured with nat=yes even if it is on the local network?

Should a Linksys Sipura 2102 be configured with nat=yes even if it is on the local network? I have been having some troubles with a Linksys Sipura 2100 series, which suffers from NO AUDIO after a few calls.. Because it is on the same subnet as Asterisk it is configured with nat=no. When you think

Re: [asterisk-users] Asterisk V/s FreeSwitch

Freeswitch was engineered from scratch by some Asterisk developers who wanted to start afresh on a cleaner programming base. Asterisk is like Topsy, She just growed and had to maintain backward compatibility. The latest versions of Asterisk are reported to be much improved in that respect. On 7

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