[asterisk-users] Missed call

... Is it possible to avoid that and signaling the other phone, that the call was not missed? Thanks a lot Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

[asterisk-users] Accessing an account from more than one phone

than one device? If yes, I can just configure my mobile phone with the same login of my phone at home and all works as expected. If not, I have to create another user and to forward all calls to this user, too... Thanks Luca Bertoncello (lucab...@lucabert.de

Re: [asterisk-users] תשובה: Missed call

explain, maybe with an example? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] תשובה: תשובה: Accessing an account from more than one phone

Zitat von Israel Gottlieb isr...@gmail.com: Shalom to you too So that's the way to go Toda! :) I'll try this weekend... Regards Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] תשובה: Accessing an account from more than one phone

... :( I'll create another user. Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] תשובה: Missed call

Zitat von Israel Gottlieb isr...@gmail.com: At the end of the Command you could use options one of them is the c (not apital) which sends a cancel event to the phone http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Ach! Thank you! I'll try this evening Luca Bertoncello (lucab

[asterisk-users] Logging in local time

? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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

[asterisk-users] Am I cracked?

? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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

Re: [asterisk-users] Am I cracked?

, too... Hopefully I solved my problem... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every

Re: [asterisk-users] Am I cracked?

to authenticated users. I was sure, that Asterisk already do that, but I'm not sure anymore... How can I restrict it? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api

[asterisk-users] Problem with NAT - Part 2

-t nat -A PREROUTING -p udp --dport 5060 -j DNAT --to-destination 192.168.20.120:5060 What can be the problem? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] Curious problem with NAT

Zitat von Steve Totaro stot...@totarotechnologies.com: Not without seeing some SIP debug output. I'm currently not at home. If you say me which debug output you wish, I can send them as soon I'll be back... Thanks Luca Bertoncello (lucab...@lucabert.de

Re: [asterisk-users] Connecting two Asterisk

quality on my mobile phone... On the other phone however, the quality is very good... I'm very very puzzled... Thanks for any help! Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api

[asterisk-users] Connecting two Asterisk

-- SIP/0049177333-0008 answered IAX2/lucabert-94 == Spawn extension (default, 0049177333, 3) exited non-zero on 'IAX2/lucabert-94' -- Hungup 'IAX2/lucabert-94' Well, I'm very puzzled... Can someone help me? Thank you very much! Luca Bertoncello (lucab...@lucabert.de

Re: [asterisk-users] Curious problem with NAT

. Any idea? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

[asterisk-users] No reply to our critical packet

yesterday, as all worked... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

[asterisk-users] Problem with SIP-TLS

connection: error:140760FC:lib(20):func(118):reason(252) [Jun 5 20:16:25] WARNING[20826]: tcptls.c:669 handle_tcptls_connection: FILE * open failed! And of course it does NOT connect... Any idea? Thanks Luca Bertoncello (lucab...@lucabert.de

Re: [asterisk-users] תשובה: Missed call

: exten = _0049351222,n,Dial(SIP/0049351222SIP/004935,,Rc) both phones ring, but if I answer from one phone, the other one say 1 missed call... Any other idea? Thanks Luca Bertoncello (lucab...@lucabert.de

Re: [asterisk-users] Problem with SIP-TLS

ricky gutierrez xserverli...@gmail.com schrieb: compilation problems with the module srtp , check the module module show like srtp Now available on OpenWRT... :( Thanks Luca Bertoncello (lucab...@lucabert.de

Re: [asterisk-users] Problem with SIP-TLS

ricky gutierrez xserverli...@gmail.com schrieb: Hi lucas , dou you try this: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial Tested right now. Same problem... I think it is a problem on Asterisk for OpenWRT... :( Regards Luca Bertoncello (lucab...@lucabert.de

Re: [asterisk-users] תשובה: תשובה: Missed call

Israel Gottlieb isr...@gmail.com schrieb: It looks like you are dialing a external # then that won't work No, both number are internal... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation

[asterisk-users] Almost solved: using my Asterisk from Internet

idea, what can be wrong now? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] Connecting two Asterisk

Steve Edwards asterisk@sedwards.com schrieb: On Sun, 7 Jun 2015, Luca Bertoncello wrote: Now the problem: on my phones at wrt I can hear what the mobile phone at lucabert sends (with a very good audio-quality), but on this mobile phone I cannot hear a single word spoken

[asterisk-users] Peer unreachable after IP change

the problem? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] Almost solved: using my Asterisk from Internet

this evening, too and report to the list... I'm very happy, that now I can login in my Asterisk at home and I don't need another Asterisk on a separate server. Firewall can be very difficult to setup, sometimes, for a SysAdmin as I be, too... :( Regards Luca Bertoncello (lucab...@lucabert.de

[asterisk-users] Connecting peer if the peer is already connected

, that is already registered or if the login was NOT successful, or even if my cellphone successfully registered (for example, to send me an E-Mail)? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Am I cracked?

Zitat von Olivier oza.4...@gmail.com: 2015-06-08 22:35 GMT+02:00 D'Arcy J.M. Cain da...@vex.net: On Mon, 8 Jun 2015 22:24:33 +0200 Luca Bertoncello lucab...@lucabert.de wrote: Kevin Larsen kevin.lar...@pioneerballoon.com schrieb: Basically, they are hoping that you are running

Re: [asterisk-users] Am I cracked?

08:06 AM, A J Stiles wrote: On Wednesday 10 Jun 2015, Luca Bertoncello wrote: I'm very sorry to write that, but these answers are really NOT helpful... I searched two days long how can I check it and didn't found anything useful... Could someone suggest me a way to check if my Asterisk

Re: [asterisk-users] Connecting peer if the peer is already connected

Zitat von A J Stiles asterisk_l...@earthshod.co.uk: On Tuesday 09 Jun 2015, Luca Bertoncello wrote: Now, I tried to register the user of my cellphone using a PC, as my cellphone was already registered. And Asterisk accepted this registration... :( Did you actually reboot the server

Re: [asterisk-users] Am I cracked?

Zitat von Dereck D derec...@gmail.com: For such cases i created a dialplan in the default dialplan which blocks the ip of the hacker with iptables. That's interesting... Could you explain me how do you did it? Thanks Luca Bertoncello (lucab...@lucabert.de

Re: [asterisk-users] Curious problem with NAT

Ashwin Surendran ashwin.surend...@now-health.com schrieb: What settings have you got for directmedia? Could you try nat=force_rport,comedia directmedia=no Tried. Peer always unreachable, call not possible... :( Other idea? Thanks Luca Bertoncello (lucab...@lucabert.de

[asterisk-users] Curious problem with NAT

that the problem is on my network, but I can't find it... What am I doing wrong? Thanks a lot Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] Curious problem with NAT

Ashwin Surendran ashwin.surend...@now-health.com schrieb: Have you tried NAT=force_rport ? No, not yet... I'll try later and report to the list... Have I to define (in Asterisk or Gateway) the ports? Thanks Luca bertoncello (lucab...@lucabert.de

Re: [asterisk-users] Curious problem with NAT

Ashwin Surendran ashwin.surend...@now-health.com schrieb: Have you tried NAT=force_rport ? OK, tried... I can transmit from my phone (aka: I hear my voice on another phone), but I'm not able to receive data (aka: I cannot hear what I say on the other phone). Other suggestion? Thanks Luca

Re: [asterisk-users] Allowing calls - maybe I'm just stupid...

(maybe just if I'm in holiday), I find this limitation meaningful. Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] Connecting peer if the peer is already connected

of time... the PC could lock you out, and the cellphone could lock *itself* out every time it moved from one IP network to another. Well, as I said, this is not a problem for me... How can I do that? And, how can I for example send an E-Mail if the client connect? Thanks Luca Bertoncello (lucab

[asterisk-users] German sounds on Asterisk

why it works so? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] German sounds on Asterisk

structure: /var/lib/asterisk/sounds/de/ /var/lib/asterisk/sounds/de/digits /var/lib/asterisk/sounds/de/letters /var/lib/asterisk/sounds/de/phonetics and it works... Regards Luca Bertoncello (lucab...@lucabert.de

Re: [asterisk-users] Peer unreachable after IP change

don't find anything for the other ports... Thanks Luca Bertoncello (lucab...@lucabert.de) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.10 (GNU/Linux) iEYEARECAAYFAlV9FZMACgkQ8Ggznj+1EDgLiQCfdeeRUuERnrJyAB0BMk1d+nF6 UIEAoIxq2SLdanDobMQ20FioqW3H/Z3G =CEgn -END PGP SIGNATURE

Re: [asterisk-users] Peer unreachable after IP change

ports are supported by your provider. Sipgate for example is telling there customers to use 5060, 5160, 5260 etc. OK, thanks! Another question: do you use Asterisk on the DSL of Deutsche Telekom? Could we compare our configuration? Thanks Luca Bertoncello (lucab...@lucabert.de) -BEGIN PGP

Re: [asterisk-users] Asterisk and Deutsche Telekom

your parachute now, then just take an airplane, climb at FL95 and jump, what's the problem? :) The problem is: that if the parachute does NOT work, you'll be dead... If my Asterisk-configuration don't work, I don't have a phone and my wife cannot work... Regards Luca Bertoncello (lucab

Re: [asterisk-users] Asterisk and Deutsche Telekom

can receive calls without any problem... So, I don't think, I have to expect problem on my NAT (anymore... initially I had some problems...). Regards Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Asterisk and Deutsche Telekom

, this will be (eventually) a future problem... :) First, I want to get my Asterisk working with Deutsche Telekom. Regards Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

[asterisk-users] Asterisk and Deutsche Telekom

, but I cannot be sure, since I can't test it... So my question: can someone using Asterisk with Deutsche Telekom contact me (PN), so that we can compare the sip.conf? Thanks a lot! Luca Bertoncello (lucab...@lucabert.de

[asterisk-users] Call accepted from not registered peers?

,DEFAULT) include = internal_calls include = luca_incoming include = fax_incoming include = anika_incoming include = messagenet_incoming include = myproxy What's wrong, now? Many thanks for your help! Luca Bertoncello (lucab...@lucabert.de

[asterisk-users] no samples for gsmtolin

, that it's already turned off... I tried to change the settings for the users, allowing just ulaw and alaw, but it's the same... Can someone say me what does this message mean and how can I suppress it? Thanks Luca Bertoncello (lucab...@lucabert.de

Re: [asterisk-users] Allowing calls - maybe I'm just stupid... [almost solved]

Zitat von Luca Bertoncello lucab...@lucabert.de: Now my problem is to check in my dialplan if the peer, that originate the call, is reachable, and if not, to give an error... Is there any function to know if the peer is reachable? I answer myself... I did that (in my [myproxy]-context

Re: [asterisk-users] Allowing calls - maybe I'm just stupid...

Zitat von A J Stiles asterisk_l...@earthshod.co.uk: On Thursday 11 Jun 2015, Luca Bertoncello wrote: Now my problem is to check in my dialplan if the peer, that originate the call, is reachable, and if not, to give an error... Is there any function to know if the peer is reachable? The peer

[asterisk-users] Allowing calls - maybe I'm just stupid...

% correct, but it tries the right way... Now my problem is to check in my dialplan if the peer, that originate the call, is reachable, and if not, to give an error... Is there any function to know if the peer is reachable? Thanks Luca Bertoncello (lucab...@lucabert.de

[asterisk-users] Peer is UNREACHABLE

online, 3 offline Unmonitored: 4 online, 0 offline] Asterisk connects to another Test-VM with AsteriskNOW and to the italian provider Messagenet. Can someone suggest me, what can I do? I can send the configuration file, if they are needed. Thanks Luca Bertoncello (lucab...@lucabert.de

Re: [asterisk-users] Peer is UNREACHABLE

already sent... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] Peer is UNREACHABLE

configured something wrong in Asterisk... Do you see anything in the asterisk logs or the logs of the phone itself (providing the phone puts logs somewhere) that indicate a failure to register or to resolve the ip address of the asterisk server? Unfortunately not... Just UNREACHABLE... Thanks Luca

Re: [asterisk-users] Peer is UNREACHABLE

device 0049351222 sip:0049351222@172.16.34.132;tag=as193c26b0 == Everyone is busy/congested at this time (1:0/1/0) == Spawn extension (myproxy, 004935, 15) exited non-zero on 'SIP/0049351222-000a' Maybe can these information help someone helping me? Thanks a lot! Luca

Re: [asterisk-users] Peer is UNREACHABLE

and try to call my phone with the same number I use from Twinkle, which works). Very puzzled... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] Peer is UNREACHABLE

can try tomorrow. Thanks a lot Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] Peer is UNREACHABLE

, you might want to obscure the secret. Which part of the configuration do you need? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

Re: [asterisk-users] Peer is UNREACHABLE

wife, and 172.16.34.133 the IP of the Asterisk server. Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] Peer is UNREACHABLE

check_auth: username mismatch, have 1234, digest has luca [May 28 22:49:51] NOTICE[4135]: chan_sip.c:20083 handle_request_invite: Failed to authenticate device Test1 sip:1234@172.16.34.132;tag=as6dd12e05 Thanks Luca Bertoncello (lucab...@lucabert.de

Re: [asterisk-users] Debugging dialplan

it will handle a call... Once you are more familiar with *, you might want to have a look what you can do with logger.conf. Maybe later... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Debugging dialplan

the same. I know the source and the destination number. I'm not sure how can I know the context. How can I say Asterisk what do you want to do with the call from X to Y? Thanks Luca Bertoncello (lucab...@lucabert.de

[asterisk-users] Debugging dialplan

in Asterisk? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] Peer is UNREACHABLE

Well, the same happens with my wife's phone... I'll try later again... Regards Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] Signaling incoming call

much! I'll try it and report to the list. Regards Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every

[asterisk-users] Getting a list of availabe SIP-Header on phone

Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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

Re: [asterisk-users] Signaling incoming call

: the normal and this one, but it is better than nothing! Now, if it will be possible to add a text on the display, it will be perfect, but I didn't found any option for that... Thanks Luca Bertoncello (lucab...@lucabert.de

Re: [asterisk-users] Signaling incoming call

the right number and the name from address book... If I change it, I'll not get the right data on the display, isn't it? Anyway, I'll try tomorrow... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation

[asterisk-users] Asterisk as Proxy and more device for a number

to both phones? Unfortunately, I didn't found any HowTo for my problems... Thank you very much for your help! Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] Asterisk as Proxy and more device for a number

with IAXModem, maybe I'll got it... Regards Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

[asterisk-users] Calling from extern

/${EXTEN},30,r) exten = _X.,n,Hangup exten = _X.,n(dialanika),Verbose(2,Outgoing using pbxanika) exten = _X.,n(dialanika),Dial(SIP/pbxanika/${EXTEN},30,r) exten = _X.,n,Hangup I really hope, someone can help me... Thanks Luca Bertoncello (lucab...@lucabert.de

[asterisk-users] Signaling incoming call

I get a message on the display or the phone ring with a particular tone, and if I receive a call for +49351222 the phone write something other on the display or ring with another tone. Is it possible? Maybe it depends from phone... I use a Thomson ST2022. Thanks a lot Luca Bertoncello (lucab

Re: [asterisk-users] Signaling incoming call

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Guenther Boelter gboel...@gmail.com schrieb: -BEGIN PGP SIGNED MESSAGE- Hash: SHA256 On 05/31/2015 02:31 PM, Luca Bertoncello wrote: Hi list! Now all works as expected, at least in the simulation I did with AsteriskNOW

Re: [asterisk-users] Asterisk as Proxy and more device for a number

, faxing is no fun, especially considering that the problems can be caused before the fax ever reaches your system. Hopefully your provider supports T.38 properly, in which case faxing will be much nicer. I know, but since I don't had a choice, I must try... Hopefully, it works... Regards Luca

Re: [asterisk-users] Signaling incoming call

. And, it the number is in the address book, I see the name, too. Perfect! Regards Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

[asterisk-users] Voicemail: saycid without prefix

set Asterisk to just read the prefix if it's necessary (so that calls from german numbers will not have 0049)? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] Choosing codecs

stick a semicolon in front of the ones you do *not* want, to comment them out. Then issue core reload in Asterisk CLI, and all your calls should be A-law from now on. OK, thanks a lot! Luca Bertoncello (lucab...@lucabert.de

Re: [asterisk-users] Choosing codecs

if you are trying to use A-law to call a mobile phone, the transcoding to GSM for the final leg to and from the handset will be taken care of by the mobile company's equipment. OK, I'll change the settings! Thanks Luca Bertoncello (lucab...@lucabert.de

Re: [asterisk-users] Choosing codecs

Zitat von A J Stiles asterisk_l...@earthshod.co.uk: On Monday 06 Jul 2015, Luca Bertoncello wrote: Well, but for voice quality, which codec is better? alaw or gsm? A-law is better for voice quality (sorry, thought my original explanation was obvious). But note that if the destination

Re: [asterisk-users] Choosing codecs

is for the network... How can I change this setting? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] Choosing codecs

by arranging the steps closer together near the zero line, and further apart away from it; so the difference between the actual signal and the nearest digital representation is small in proportion to the signal. Well, but for voice quality, which codec is better? alaw or gsm? Thanks Luca

[asterisk-users] Choosing codecs

MCBsoNI2Cj266BB 0x2 (gsm)No Rx: ACK0049351222 Could someone explain me why? Second question: I think, ulaw/alaw are better then gsm, isn't it? If so, how can I change it? Thanks Luca Bertoncello (lucab...@lucabert.de

Re: [asterisk-users] Voicemail: saycid without prefix

had this idea and implemented it. It works... Regards Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar

[asterisk-users] CDR in an MySQL-Database

* Registered Backends --- cdr-custom Asterisk 1.8 runs on an OpenWRT-Switch. Any idea? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] CDR in an MySQL-Database

that first? I think, the team of OpenWRT did NOT prepare the CDR-MySQL-Module, since I could not find cdr_addon_mysql.so... I resolved writing the data in a CSV, and then importing the data in the MySQL-DB with a script... Thanks Luca Bertoncello (lucab...@lucabert.de

Re: [asterisk-users] Recording INCOMING calls

to the previous Dial and all work! Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

[asterisk-users] Recording INCOMING calls

*3 nothing happens... In the console I can't see anything, too. Could you suggest me what is wrong? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

[asterisk-users] Problem no voice

allow=g729 allow=g723 allow=gsm I tried with allow=all, too, but it results in no communication on all numbers... Could someone help me? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Problem no voice

jg webaccounts...@jgoettgens.de schrieb: How is the 4th phone configured? It's not a phone, just a number routed on a phone that receives calls for other number, too (without any problem). You could also enable SIP debugging to get more information about the problem. I already set core set

[asterisk-users] Help with voicemail

, I have a file /var/spool/asterisk/voicemail/default/0039015111/unavail.gsm... Can someone help me to solve my problem? Thanks a lot! Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] Help with voicemail

ce I don't have a g729 codec, I changed the properties of this peer enabling other codecs. Now the voicemail works as expected... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.

[asterisk-users] Sending E-Mail from voicemail

attach=no, but I'm not sure how can I use it... Am I right, if I just write so: 004935 = SECRET,John Doe,fi...@email.de,attach=no|sec...@email.de so that fi...@email.de receive the E-Mail WITH the attachment and sec...@email.de not? As I said, I can't just try... Thanks Luca Bertoncello

[asterisk-users] Sending E-Mail from voicemail with AND without attachment

help is appreciated Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] Sending E-Mail from voicemail with AND without attachment

Zitat von Luca Bertoncello lucab...@lucabert.de: I'm trying to send two E-Mails when a message comes in the voicemail, the first WITH the attachment, and the second WITHOUT. But I don't get it working... OK, I'm __VERY__ stupid... I can write two addresses, and the second is for pager

[asterisk-users] Low quality using mobile phone

big problem to understand me. This if I use my own WLAN, too (same network of the other VoIP-phones). I don't find anything in the logs. Any idea, where could be the problem? Thanks a lot Luca Bertoncello (lucab...@lucabert.de

[asterisk-users] Forwarding call if extension busy

ned my problem... Any suggestion? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webin

Re: [asterisk-users] Forwarding call if extension busy

; n(voice),GotoIf($["${DEVICE_STATE(sip/${EXTEN})}" != "NOT_INUSE"]?voicebusy) I think, this is EXACTLY what I'm looking for... I'll try this evening! Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Band

[asterisk-users] Signaling ringing on other extension

two phones "Thomson ST2022". Thanks a lot Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] Forwarding call if extension busy

Kevin Larsen <kevin.lar...@pioneerballoon.com> schrieb: > I am not sure if I completely understand what you are trying to do, but it > sounds like you want to query the DEVICE_STATE function. IT WORKS Thank you very much! Luca Bertoncello (lucab...

Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

ip > reload". Sorry, I forgot to mention that... I already have this setting: session-refresher=uac session-timers=refuse > (I assume You are using chan_sip. I don't know how to disable session > timer in pj sip). I use chan_sip. Thanks L

[asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

on (default, +3901522, 3) exited non-zero on 'SIP/004935-0125' -- fixed jitterbuffer destroyed on channel SIP/004935-0125 My number is the 004935 and I called the 003901522. Any idea? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _

Re: [asterisk-users] Deutsche Telekom: calls dropped after 15

Marlon Araujo <marlon...@me.com> schrieb: > 15 minutes, sure sounds like reinvite could be the villain. > > Can you paste your sip.conf Very strange... I didn't change anything, but now the calls are NOT dropped anymore... Maybe Telekom changed somewhat... Thanks Luca Ber

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