Re: [asterisk-users] Peer is UNREACHABLE

.4 Date: Thu, 28 May 2015 20:39:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 repeated in loop... Help that? 192.168.200.11 is the IP of the phone of my wife, and 172.16.34.133 the IP of the Asterisk serve

Re: [asterisk-users] Peer is UNREACHABLE

35]: chan_sip.c:12800 check_auth: username mismatch, have <1234>, digest has [May 28 22:49:51] NOTICE[4135]: chan_sip.c:20083 handle_request_invite: Failed to authenticate device "Test1" ;tag=as6dd12e05 Thanks Luca Bertoncello (lucab...@lucabert.de) --

Re: [asterisk-users] Peer is UNREACHABLE

s [May 28 23:05:59] NOTICE[4135]: chan_sip.c:20083 handle_request_invite: Failed to authenticate device "Test1" ;tag=as7855ffe5 (the phone of my wife is now logged in on AsteriskNOW with the user "1234" and try to call my phone with the same number I use from Twinkle, which w

Re: [asterisk-users] Peer is UNREACHABLE

e Telekom activate the new line... So I installed AsteriskNOW on a VM and configured it to serve a couple of number. Then I installed Asterisk on a second VM and configured it to connect to AsteriskNOW (later will be Telekom) and Messagenet. Dialplan and the other configuration were already sent... Th

Re: [asterisk-users] Peer is UNREACHABLE

to the AsteriskNOW... Well, now I must sleep... Hope someone can suggest me something that I can try tomorrow. Thanks a lot Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital

Re: [asterisk-users] Peer is UNREACHABLE

eck_auth: username mismatch, have <0049351222>, digest has <004935> [May 29 00:12:02] NOTICE[1106]: chan_sip.c:20083 handle_request_invite: Failed to authenticate device "0049351222" ;tag=as193c26b0 == Everyone is busy/congested at this ti

[asterisk-users] Debugging dialplan

option 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:

Re: [asterisk-users] Debugging dialplan

e. 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) -- __

Re: [asterisk-users] Debugging dialplan

how 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 P

Re: [asterisk-users] Peer is UNREACHABLE

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 for a live in

[asterisk-users] Calling from "extern"

,r) exten => _X.,n,Hangup exten => _X.,n(dialluca),Verbose(2,Outgoing using pbxluca) exten => _X.,n(dialluca),Dial(SIP/pbxluca/${EXTEN},30,r) exten => _X.,n,Hangup exten => _X.,n(dialfax),Verbose(2,Outgoing using pbxfax) exten => _X.,n(dialfax),Dial(SIP/pbxfax/${EXTEN},30,r) exte

[asterisk-users] Signaling incoming call

a call for +4935 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 B

Re: [asterisk-users] Signaling incoming call

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Guenther Boelter 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

Re: [asterisk-users] Signaling incoming call

alls-from-particular- Thank you very 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

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

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: http://www.asterisk.org/

Re: [asterisk-users] Signaling incoming call

And, unfortunately, I just have two melody: 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

nd 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 Coloca

Re: [asterisk-users] Signaling incoming call

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 introductory webinar

[asterisk-users] Missed call

uot;1 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

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

count from more 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

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

be 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 : 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: Missed call

Zitat von Israel Gottlieb : 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...@lucabe

[asterisk-users] Logging in "local time"

someone help me? 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://w

Re: [asterisk-users] תשובה: Missed call

e: exten => _0049351222,n,Dial(SIP/0049351222&SIP/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) --

[asterisk-users] Problem with SIP-TLS

p ssl 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...@lucabe

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

ricky gutierrez 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...@lu

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

ricky gutierrez schrieb: > compilation problems with the module srtp , check the module > > module show like srtp Now available on OpenWRT... :( Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Band

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

Israel Gottlieb 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 Provided

[asterisk-users] Curious problem with NAT

, so I'm sure 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

Re: [asterisk-users] Curious problem with NAT

Ashwin Surendran 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...@luc

Re: [asterisk-users] Curious problem with NAT

Ashwin Surendran 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 Bertoncello (lucab...@luc

Re: [asterisk-users] Curious problem with NAT

Ashwin Surendran 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 (lu

[asterisk-users] Problem with NAT - Part 2

/sbin/iptables -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

Re: [asterisk-users] Curious problem with NAT

direct in Internet. But maybe my Provider does a NAT, too... Very strange is, that I have a very poorly audio-quality, if I use my cellphone in my WLAN and connect to my Asterisk. With THE SAME USER, but from a PC in the same Network, the audio quality is perfect. Any idea? Thanks Luca

Re: [asterisk-users] Curious problem with NAT

Zitat von Steve Totaro : 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...@l

[asterisk-users] Connecting two Asterisk

inging -- SIP/0049177333-0008 is ringing -- 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 h

Re: [asterisk-users] Connecting two Asterisk

ith a VERY poor 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 Colocat

[asterisk-users] Peer unreachable after IP change

e help me to understand 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 Thu

Re: [asterisk-users] Connecting two Asterisk

Steve Edwards 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

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

ation 192.168.20.120 Any 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

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

tion 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... :

[asterisk-users] Am I cracked?

. On my Firewall I see a SIP packet coming from an IP in Palestine... Am I cracked? I think I disabled all "guest" access. How can I check if my Asterisk allows guest to originate calls? Thanks Luca Bertoncello (lucab...@lucabert.de) -- __

Re: [asterisk-users] Am I cracked?

Now I log the SIP-pakets coming from Internet, too... Hopefully I solved my problem... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] Am I cracked?

, I need to be able to call any phone on this world... On a Mail-Server I'd restrict outgoing calls 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) --

[asterisk-users] No reply to our critical packet

rday, 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] Connecting peer if the peer is already connected

to register as a peer, 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) -- _ -- Bandwi

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

Zitat von A J Stiles : 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, as opposed to simply reloading your

Re: [asterisk-users] Am I cracked?

Zitat von Olivier : 2015-06-08 22:35 GMT+02:00 D'Arcy J.M. Cain : On Mon, 8 Jun 2015 22:24:33 +0200 Luca Bertoncello wrote: > Kevin Larsen schrieb: > > Basically, they are hoping that you are running the equivalent of a > > mail server open relay. They are trying to us

Re: [asterisk-users] Am I cracked?

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 is an "O

Re: [asterisk-users] Am I cracked?

Zitat von Dereck D : 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...@lucabe

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

tered", you could quite easily find yourself unable to register the > cellphone with Asterisk for a prolonged period 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 pro

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

og, but it calls, too... My [default] exten => _X.,1,Verbose(2,DEFAULT) include => internal_calls include => luca_incoming include => fax_incoming include => anika_incoming include => messagenet_incoming include => myproxy

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

t 100% 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...@lucabe

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

Zitat von Luca Bertoncello : 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): exten => _X.,n,

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

Zitat von A J Stiles : 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 that *originated* the call

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

[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...@lucabe

Re: [asterisk-users] Asterisk and Deutsche Telekom

problem? :) If you can't test 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

Re: [asterisk-users] Asterisk and Deutsche Telekom

ve 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 Provi

Re: [asterisk-users] Asterisk and Deutsche Telekom

frastructure > for finer control. OK, understood... Well, this will be (eventually) a future problem... :) First, I want to get my Asterisk working with Deutsche Telekom. 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 -E

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

nds from which 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.d

[asterisk-users] German sounds on Asterisk

"CHANNEL(language)=de") in new stack -- Executing [24@default:3] SayUnixTime("SIP/004935-0003", "") in new stack -- Playing 'digits/day-0.gsm' (language 'de') -- Playing 'digits/h-14.gsm' (language 'de'

Re: [asterisk-users] German sounds on Asterisk

work so... Right now I got it, using a new 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) -- _

[asterisk-users] no samples for gsmtolin

uppression", 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 Bertonce

[asterisk-users] Voicemail: saycid without prefix

49177...". How can I 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 a

[asterisk-users] Choosing codecs

22 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...@

Re: [asterisk-users] Choosing codecs

ot; 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 Bertoncello (lucab...@l

Re: [asterisk-users] Choosing codecs

Zitat von A J Stiles : 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 is a mobile phone, GSM will be

Re: [asterisk-users] Choosing codecs

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...@l

Re: [asterisk-users] Choosing codecs

ow 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

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) -- _ -- Bandwidth and

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

Thanks! I already 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

[asterisk-users] CDR in an MySQL-Database

il Record (CDR) settings -- Logging:Enabled Mode: Simple Log unanswered calls: Yes * Registered Backends --- cdr-custom Asterisk 1.8 runs on an OpenWRT-Switch. Any idea? Thanks Luca Bertoncello (lucab

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

need to > fix 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...

[asterisk-users] Sending E-Mail from voicemail

ption "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

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

t from the second E-Mail... Any 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

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

Zitat von Luca Bertoncello : 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 and it doe

[asterisk-users] Problem "no voice"

allow=ilbc 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 P

Re: [asterisk-users] Problem "no voice"

jg 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 debug 42 and core set verb

[asterisk-users] Recording INCOMING calls

be called, and press *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://

Re: [asterisk-users] Recording INCOMING calls

o 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] 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...@lucabe

[asterisk-users] Help with voicemail

ourse, 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

Re: [asterisk-users] Help with voicemail

29 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.api-digital.com

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

xtension (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 minutes

uot;. 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 Lu

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

"Brian ::" schrieb: > sip trace? Could you please explain? I'm not a VoIP-expert... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

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

sip set verbose 42 The result was in my first E-Mail... 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] Deutsche Telekom: calls dropped after 15 minutes

" (whatever it means). As result I was without Internet and phone for over an hour... Then I tried to call my cousin in Italy and the call was NOT dropped after 15 minutes... I'll try this evening again. Maybe it was a problem by Deutsche Telekom... Thanks Luca Ber

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

list, make sure to clean it up first (remove passwords, user names, phone numbers, digest authentication info etc). OK, I'll try and report to the list Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and

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

Marlon Araujo 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 Bertoncello (lucab.

[asterisk-users] Transfer calls "on demand"

ch the call from the other phone, if for example my wife is not at home, for example pressing "*5#" or other key combination. Thanks a lot for your suggestion! Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth a

Re: [asterisk-users] Transfer calls "on demand"

t exten => _222,n,VoiceMail(0049351222,us) exten => _222,n,Hangup Then I called the 222 with my mobile phone and I tried to get the call from the other phone, calling the *8. Unfortunately I get an error (invalid number) on the display of the phone, and the phone 222 c

Re: [asterisk-users] Transfer calls "on demand"

Doug Lytle schrieb: > Keep it simple for testing. My sip.conf on a working Asterisk system below: IT WORKS!!! Thanks a lot! Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.

[asterisk-users] Signaling ringing on other extension

use 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 introdu

Re: [asterisk-users] Signaling ringing on other extension

51222,us) exten => _222,n,Hangup then I reloaded the core (core reload), SIP (sip reload) and Dialplan (dialplan reload) and I called the 0351222 from my mobile phone. It rings, but on the other phone (035) is nothing to see... Where is my error? Thanks Luca Bertoncello (luca

Re: [asterisk-users] Signaling ringing on other extension

oo. No changes... > Also, have you configured the phones as well? What do you mean? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join

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