[asterisk-users] [Maybe OT]: SIP Provider

service, but if not, I don't want to pay too much... As said: I need a SIP Provider to have an italian number (better if I can choose the prefix) only to receive calls. Any suggestion? Thanks a lot Luca Bertoncello (lucab...@lucabert.de

[asterisk-users] Cannot send faxes

ug 16 18:34:37.51: [26819]: <-- [16:ATDT0177yyy\r] Aug 16 18:34:52.66: [26819]: --> [10:NO CARRIER] Aug 16 18:34:52.66: [26819]: SEND FAILED: JOB 39 DEST 0177yyy ERR [2] No carrier detected Aug 16 18:34:53.66: [26819]: <-- [5:ATH0\r] Aug 16 18:34:53

Re: [asterisk-users] Called number changed on SNOM 821

Am 31.12.2021 um 16:04 schrieb Antony Stone: Hi Antony > Check the Dial() command which places the call to the phone. Does it contain > the "c" option? So, I tested it right now and it works... Just removing the "c"... Thanks a lot for your help and of course happy

Re: [asterisk-users] Called number changed on SNOM 821

Am 31.12.2021 um 16:07 schrieb Luca Bertoncello: > I'll try to remove it, but I can't test it today... > > I'll let you know if it works. At least a call without anser does not contain the Header anymore... I'll ask if the number is shown in the missed calls. Regards Luca Bertoncel

Re: [asterisk-users] Called number changed on SNOM 821

E-Mail der AB nicht den Namen steht exten => _529874,n,VoiceMail(74,us) exten => _529874,n,Hangup I'll try to remove it, but I can't test it today... I'll let you know if it works. Thanks Luca Bertoncello (lucab...@lucabert.de) -- __

Re: [asterisk-users] Called number changed on SNOM 821

${UNIQUEID} / DATE: ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})) exten => s,n,System(echo "Verpasster Anruf vom ${CALLERID(NUM)} um ${STRFTIME(${EPOCH},,%H:%M)}" | mail -s "Verpasster Anruf" i...@xxx.de) exten => h,1,GotoIf($[“${DIALSTATUS}” = “ANSWER”]?done)

Re: [asterisk-users] Called number changed on SNOM 821

r does not change anymore. Last very strange problem is, that the list of missed calls on the phone is always empty... But it can be a problem of the phone hisself... Maybe has someone an idea? The phone is a Snom 821-SIP Thanks and happy new year! Luca Bertoncello (lucab...@lucab

Re: [asterisk-users] Called number changed on SNOM 821

s being the source > of the problem, which I think is good. Well, this means, that the problem is in the Asterisk... Very huge part of the infrastructure... :( Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth a

Re: [asterisk-users] Called number changed on SNOM 821

one_ "Ringing" and sends the phone _two_ "Ringing", the second one with the P-Asserted-Identity... Maybe help it to identify the problem? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth an

Re: [asterisk-users] Called number changed on SNOM 821

to call me from the phone when I sniff the traffic... I hope, I find someone tomorrow. Regards Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the ne

Re: [asterisk-users] Called number changed on SNOM 821

and what you mean... You mean that I should compare what the "180 ringing" in the internal network (phone to asterisk) and the external one (asterisk to Telekom)? If so, then I have to check again, since I only sniffed the internal traffic... If not, I didn't understand what you m

Re: [asterisk-users] Called number changed on SNOM 821

LAN for the phones. All traffic captured. Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.as

Re: [asterisk-users] Called number changed on SNOM 821

L, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Contact: Content-Length: 0 So, I see, there is a "P-Asserted-Identity"... But I can't understand why... Any idea? Thanks Luca Bertoncello (lucab...@lucabert.de) -- ___

Re: [asterisk-users] Called number changed on SNOM 821

I don't see anything strange... Btw, what do you mean with "180 response"? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new As

Re: [asterisk-users] Called number changed on SNOM 821

Am 28.12.2021 14:30, schrieb Luca Bertoncello: Hi again, If I call a number I can see in the display the called number, after a few seconds the number changes to the own numer. After hangup I just see my own number in the call log. The same if I receive a call. Very very strange

[asterisk-users] Called number changed on SNOM 821

seconds the number changes to the own numer. After hangup I just see my own number in the call log. The same if I receive a call. On the old Server (with Asterisk 11.7.0) with the same phones there was no problem. Do someone have any idea what can be the problem? Thanks a lot Luca Bertoncello

Re: [asterisk-users] Notifying missed calls

Am 06.11.2021 um 21:15 schrieb Łukasz Grzywański: Hi Łukasz, Dziękuję > two legs in this same context > ( exten => _03529123456,n,Dial(local/123456@main_incoming,,xX) ) > > PJSIP/pbxmichael_in-0418 > and  > Local/123456@main_incoming-0268 > > [main_incoming] > exten =>

Re: [asterisk-users] Notifying missed calls

e") in new stack -- Executing [s@noanswer:1] NoOp("Local/123456@main_incoming-0268;2", "UID CALL: 1636222382.6032 / DATE: 20211106-191306)") in new stack -- Executing [s@noanswer:2] System("Local/123456@main_incoming-0268;2", "echo "

Re: [asterisk-users] Notifying missed calls

Am 06.11.2021 um 14:43 schrieb Frank Vanoni: Hi Frank > 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

Re: [asterisk-users] Notifying missed calls

) exten => h,n(done),NoOp() exten => h,n,HangUp() ... It works, but I have two problems: 1) The E-Mails will be sent "double" 2) The E-Mails will be sent for outgoing unanswered calls, too. Do someone has an idea what is

Re: [asterisk-users] Notifying missed calls

Am 03.11.2021 um 21:34 schrieb Antony Stone: > On Wednesday 03 November 2021 at 21:29:46, Luca Bertoncello wrote: > >> I tried so: >> >> exten => h,n(hang),Gosub(noanswer,s,1) > > The n there should be 1, surely? Ach, you're right! Now it works! Tha

Re: [asterisk-users] Notifying missed calls

n,Dial(SIP/74,39,RcxX) exten => _xx,n,Verbose(2,Voicemail for Main) exten => _xx,n,Set(CALLERID(name)=) exten => _xx,n,Gosub(noanswer,s,1) exten => _xx,n,VoiceMail(74,us) exten => _xx,n,Ha

[asterisk-users] Notifying missed calls

email, the Subrouting "noanswer" will not called... Any ideas? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community

Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?

Am 18.02.2021 um 18:59 schrieb Michael Maier: > On 17.02.21 at 21:46 Luca Bertoncello wrote: >> Am 16.02.2021 um 22:32 schrieb Michael Maier: >> >> Hi Michael >> >>>> Maybe could you send me an abstract of your configuration? >>> >>> Tak

Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?

.de,,R" and it does NOT work... Is it correct, that I have to leave "sip:..."? Thank you very much for your help!! Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.ap

Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?

is it correct? > The script unregisters and registers the telekom trunks, if a change is > detected. This is done as long as there is no call active. This works > for me - but may not wort for others - feel free to change the code. OK, I'll check it... >

Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?

tel.t-online.de in my Bind with these settings? Looks like dangerous, if they changes something... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out

[asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?

tion? Thanks a lot Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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-users] [OT?] Elmeg IP290: do someone know this telephone?

Hi! I have a little problem with the given phone... Do someone know it? My problem is that I'd like to display the name of the caller (if it is saved in the address book, of course), but it always display just the number... Thanks Luca Bertoncello (lucab...@lucabert.de

Re: [asterisk-users] Voice broken during calls (again...)

Hi list! Am 22.06.2020 um 16:48 schrieb Luca Bertoncello: > Hi list! > > So, now I have a business contract and a technician was here to check > the DSL... > Nothing found, except that for 50Mbps I need now vectoring. Really > nice... A couple of years ago I could get 50Mbps

Re: [asterisk-users] Voice broken during calls (again...)

to understand/learn how to check the involved parts and search for the problem? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk

Re: [asterisk-users] Voice broken during calls (again...)

Am 23.06.2020 um 21:08 schrieb Michael Maier: > On 23.06.20 at 08:05 Luca Bertoncello wrote: >> Am 23.06.2020 07:27, schrieb Luca Bertoncello: >> >> I again >> >>>> Do not change MTU. Probably there will be another problem. I expect >>>> packet

Re: [asterisk-users] Voice broken during calls (again...)

genet/5 register => lucabertoncello:x@rebvoice/lucabertoncello jbenable = no jbmaxsize = 200 jbresyncthreshold = 1000 jbimpl = fixed Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Voice broken during calls (again...)

and a peer connected via LTE and the other in LAN, then maybe it's possible to find the problem... But if you have any other idea, I'm very happy to hear it! ;) Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth

Re: [asterisk-users] Voice broken during calls (again...)

he list is an expert with iptables and can check it? I know this program, but I'm not really an expert... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check

Re: [asterisk-users] Voice broken during calls (again...)

, communication with both peers in the same interface work correct, but maybe my firewall script... If you can reproduce this can you send me a few more packet traces, from each of the VLAN interfaces involved? Of course, I can do that! Maybe I get it this evening. Regards Luca Bertoncello (lucab

Re: [asterisk-users] Voice broken during calls (again...)

akets in the internal networks... Thanks a lot Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community

Re: [asterisk-users] Voice broken during calls (again...)

... :( Everyway: you think, my network works as expected? At least the part using DSL? Any idea, where could be the problem? Thanks a lot Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Voice broken during calls (again...)

transmitted, 0 received, +4 errors, 100% packet loss, time 3965ms pipe 2 With paket size of 1464 it works... You know MTU is a size of l2 frame, so using ipv6 you are able to use higher payload sizes because of ip header size. OK, thanks! Luca Bertoncello (lucab...@lucabert.de

Re: [asterisk-users] Voice broken during calls (again...)

-flags SYN,RST SYN -j TCPMSS --set-mss 128 ? Or I just have to use: iptables -A FORWARD -p tcp --tcp-flags SYN,RST SYN -j TCPMSS --set-mss 128 instead of: iptables -A FORWARD -p tcp --tcp-flags SYN,RST SYN -j TCPMSS --clamp-mss-to-pmtu ? Thanks Luca Bertoncello (lucab...@lucabert.de

Re: [asterisk-users] Voice broken during calls (again...)

Am 23.06.2020 08:43, schrieb Luca Bertoncello: And another thing, I discovered right now... Could you suggest me something to restrict the problem? Currently, I think the problem can be: 1) on Asterisk 2) on my Gateway/Firewall A couple of years ago I added this entry in my firewall: /sbin

Re: [asterisk-users] Voice broken during calls (again...)

Am 22.06.2020 20:09, schrieb Luca Bertoncello: A couple of other ideas... Conclusion (maybe!): it can *not* be a problem in the DSL connection and *maybe* it is not a problem in the communication with the Server of Deutsche Telekom, since I have many problems to communicate between two peers

Re: [asterisk-users] Voice broken during calls (again...)

Am 23.06.2020 07:27, schrieb Luca Bertoncello: I again Do not change MTU. Probably there will be another problem. I expect packet size 1466 would pass and higher will have the same result. It I checked it, and I see, that the maximum I can use is a paket size of 1464 with all hosts via IPv4

Re: [asterisk-users] Voice broken during calls (again...)

gt; ping. I don't understand what you mean, could you explain? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk communi

Re: [asterisk-users] Voice broken during calls (again...)

hould I reduce the MTU?!? Maybe I didn't understood what you mean... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new A

Re: [asterisk-users] Voice broken during calls (again...)

A thing I forgot to report... My Asterisk listen on an high port (*not* 5060), since I had many problems in the past with someone trying to use my Asterisk with brute force attack... I really don't think, this can be the problem, but better to report all... Regards Luca Bertoncello (lucab

Re: [asterisk-users] Voice broken during calls (again...)

too... Regarding the ping time: wich line do you have? I have a DSL 50Mbps. Maybe your times are better due to a faster line? What is your opinion about the tests I did today with the friend and his phone as VoIP-peer? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _

Re: [asterisk-users] Voice broken during calls (again...)

ons? 2) assuming are my conclusions correct, can someone suggest me where can I search the problem? Thanks a lot Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check o

Re: [asterisk-users] Voice broken during calls (again...)

e "internal number") the quality is excellent. If I call my wife using the "external number", the quality is very bad... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by ht

[asterisk-users] Voice broken during calls (again...)

I can do... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New

Re: [asterisk-users] Voice "broken" during calls

s contract). The quality is disturbed from the first second... I had the problem, that the connection will be *dropped* after 15 minutes, and I solved it with "session-timers = refuse" Bye Luca Bertoncello (

Re: [asterisk-users] Voice "broken" during calls

Am 16.06.2020 10:48, schrieb Antony Stone: On Tuesday 16 June 2020 at 08:18:51, Luca Bertoncello wrote: > sudo tcpdump -i eth0 -s 0 -w /tmp/test0.pcap & > sudo tcpdump -i eth1 -s 0 -w /tmp/test1.pcap & eth0 is my DSL interface and eth1 my phone interface? Well, one is i

Re: [asterisk-users] Voice "broken" during calls

xx (IP of my phone) & is it correct? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://commun

Re: [asterisk-users] Voice "broken" during calls

since now I must go to the office... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.o

Re: [asterisk-users] Voice "broken" during calls

mit? No, I tried the test disabling the traffic shaper, too... no changes... > I'm very much agreeing with you here, that DT appears to be the problem, and > I > think Jeff's suggestion / offer to capture the audio data and

Re: [asterisk-users] Voice "broken" during calls

Am 15.06.2020 um 21:50 schrieb Luca Bertoncello: > What do you mean now? If I can use the full available band or if I can > download exactly 50Mbs? > The answer to the first question is: YES! That's why I use a traffic > shaper... ;) > The answer to the second question is: NO. I m

Re: [asterisk-users] Voice "broken" during calls

Am 15.06.2020 um 21:28 schrieb Antony Stone: > On Monday 15 June 2020 at 21:19:51, Luca Bertoncello wrote: > >> But I'm not really sure, that Asterisk could be the problem, since, as I >> said, the problem happens even if I connect the phone direct to the >> server of

Re: [asterisk-users] Voice "broken" during calls

Am 15.06.2020 um 21:24 schrieb Antony Stone: > On Monday 15 June 2020 at 18:55:23, Luca Bertoncello wrote: > >> Absolutly *no changes* on the behaviour compared with my Thomsons... > > Okay, I'm glad we can rule out the specific make / model of phone - that > would > ha

Re: [asterisk-users] Voice "broken" during calls

nage the data transfer, isn't it? Regards Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.

Re: [asterisk-users] Voice "broken" during calls

Am 14.06.2020 um 17:33 schrieb Luca Bertoncello: Hi So, I got a phone (Elmeg IP290) from a collegue and tested it... > What I'll do tomorrow with a test phone is: > > 1) connecting it to my Asterisk and try to make a call > 2) connecting it directly to the servers of Deutsche Telek

Re: [asterisk-users] Voice "broken" during calls

erisk and try to make a call 2) connecting it directly to the servers of Deutsche Telekom (using my network) and try to make a call Thanks a lot for your help Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Voice "broken" during calls

it does not depend > on your DSL account (as it is standard with most other VoIP providers). OK, I really don't think I want to subscribe this option just to check if the problem is in my account... :D Any other suggestion how to find *where* the problem is? Thanks Luca Bertoncello

Re: [asterisk-users] Voice "broken" during calls

, but the problem exists an almost all calls, incoming or outgoing, no matter from/to which network provider... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.c

Re: [asterisk-users] Voice "broken" during calls

son connected on the same network to the same Asterisk server, > or is it somewhere else altogether? Yes, both telefons are in the same VLAN and Asterisk, too. > Why do you have: > >> allow=ilbc > > in sip.conf? I can't really remember why I a

Re: [asterisk-users] Voice "broken" during calls

What's the call quality like then? The quality is terrible. It is not possible to understand any word... BUT: if I call my wife using the Thomson (she uses a Thomsons, same model, too!) the quality is excellent... > In regard to: > > On Saturday 13 June 2020 at 18:25:32, Luca Bertoncello wrote:

Re: [asterisk-users] Voice "broken" during calls

le that there is a problem on Telekom-side, but it does not explain why I have the same problems, altought not often as by Telekom, by MessageNet, too... Thanks a lot Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Col

Re: [asterisk-users] Voice "broken" during calls

lly puzzled... Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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? St

Re: [asterisk-users] Voice "broken" during calls

gt; That looks a little more standard. The questions are: 1) why the mobile phone, with "too many things" has a better quality 2) where can I change these settings? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- B

Re: [asterisk-users] Voice "broken" during calls

ACK Promiscuous Redir: No Route: DTMF Mode: rfc2833 SIP Options:(none) Session-Timer: Inactive Transport: UDP Media: RTP So, I'd say, the codecs are the same... Do you see something strange

Re: [asterisk-users] Voice "broken" during calls

Am 13.06.2020 09:30, schrieb Luca Bertoncello: Hi again (again) I noticed right now another strange detail... I made a call using my mobile phone (connected to the Asterisk). The quality was top... Maybe is the problem in a codec used from our phones at homes? Could someone suggest me how

Re: [asterisk-users] Voice "broken" during calls

Am 13.06.2020 um 08:28 schrieb Luca Bertoncello: > Hi! > > I have a Asterisk installation to manage my phones at home (provider is > Deutsche Telekom). > It works, but very often the voice is "broken"... > Yesterday during a call it was very difficult to under

[asterisk-users] Voice "broken" during calls

if needed. Thanks a lot Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Aster

Re: [asterisk-users] Delay on speak with Asterisk

ccepting this offer! So, back to alaw... :( > Ah, but SIP is not RTP :) OK, I forgot it... I privilege RTP, too... ;) Regards Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.ap

Re: [asterisk-users] Delay on speak with Asterisk

( > 2. What is the bandwidth (upstream is more important than downstream) of your > Internet connection? Down 50Mbps Up 10Mbps On my Router (Debian 9) I configured a traffic shaper that privileges the SIP-Packets. Thanks Luca Bertoncello (lucab...

Re: [asterisk-users] Delay on speak with Asterisk

Am 03.12.2019 um 19:28 schrieb Luca Bertoncello: Hi again > This delay happens on every peer, Deutsche Telekom and Messagenet, so I > think the problem is NOT by the Provider, but in my configuration... Maybe I got the solution... I see, that I had the jitter buffer active. As I deact

Re: [asterisk-users] Delay on speak with Asterisk

SIP Options:timer Session-Timer: Inactive Transport: UDP Media: RTP Maybe it helps to find the problem? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation

[asterisk-users] Delay on speak with Asterisk

but in my configuration... Can someone suggest me where can I search the problem? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk

Re: [asterisk-users] High delay and some echo

useful to know. As I said, I have a BananaPI with a Debian 9, minimal installed from me with some scripts to manage the DSL. Asterisk was installed from Debian Repositories. Thanks Luca Bertoncello (lucab...@lucabert.de) -- __

Re: [asterisk-users] High delay and some echo

if you can identify where the latency comes in? I must say, that I'm not an expert in VoIP, so I really don't know this tool and don't have any idea how to analyze the problem... Thanks Luca Bertoncello (lucab...@lucabert.de) -- ___

Re: [asterisk-users] High delay and some echo

he other party use VoIP, too, since they are in Germany (and Italy) and here there are just VoIP... Sigh! Now I disabled the jitter (jbenable = no), and I called my father in law. He sayd me, the quality is really better, but I hear sometimes little noises... Any other suggestion? Thanks Luca

[asterisk-users] High delay and some echo

what can I check and what can be the problem. The problem exists since a very long time, but in the last months it got worse... Thank you for your help, I can send abstracts of my configuration, if you say me what should I send. Luca Bertoncello (lucab...@lucabert.de

[asterisk-users] Unable to use VoIP-device

twork "phone0" (192.168.200.0/24) and the mobile phone in the network "intlan0" (192.168.10.0/24). The BananaPI hat IPs on bot networks and I configured Asterisk to bind to 0.0.0.0. And, as I said, the mobile phone CAN register in Asterisk...

Re: [asterisk-users] Problem with DAHDI

Luca Bertoncello <lucab...@lucabert.de> schrieb: > But if I try to call another VoIP-phone it rings but no voice will be > transferred... Got it! A "little" firewall problem... :( Regards Luca Bertoncello

Re: [asterisk-users] Problem with DAHDI

0 left 'simple_bridge' basic-bridge <0ef9a447-b1b3-45af-a4af-7c4ac4d10546> == Spawn extension (default, 00493517654321, 1) exited non-zero on 'SIP/00493511234567-' Where is the error?!? Thanks Luca Bertoncello (lucab...@lucabert.de) --

[asterisk-users] Problem with DAHDI

code: No such file or directory Asterisk Ready. it does not seems to be normal, but I can't understand why /dev/dahdi/channel does not exists... I installed the Paket asterisk-dahdi, of course... Other question: what does the error about res_phoneprov.c means? Can someone help me? Thank you very much

Re: [asterisk-users] chan_oss.c: Unable to register channel type 'OSS'

Zitat von Tzafrir Cohen <tzafrir.co...@xorcom.com>: Yes. It is useful if you want to call using a local sound device. On a Banana PI? ;) Consider editing /etc/asterisk/modules.conf and disable ('noload =>') chan_oss.so . So I did... Thanks Luca Bertoncello (lucab...@lu

Re: [asterisk-users] chan_oss.c: Unable to register channel type 'OSS'

d I really don't think, I need this module... As I undestand, I just need it, if I want to call/answer call using the console, and I really don't need this... Or I understood wrong? Regards Luca Bertoncello (luca

[asterisk-users] chan_oss.c: Unable to register channel type 'OSS'

t for your help! Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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?

[asterisk-users] Remote Phonebook with Thomson ST2022

- [25/Oct/2017:19:38:40 +0200] "GET /phonebook.xml HTTP/1.1" 200 36611 Can someone help me? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -

Re: [asterisk-users] Problem using Android SIP-Client

nes as its port range, not your > phone. If you get one way voice (remote hears phone) then you are on the > right direction. You'll then need to open the incoming ports too for the > ports that your phone is expecting to get its RTP from. OK, tomorrow I'll check it..

Re: [asterisk-users] Problem using Android SIP-Client

Luca Bertoncello <lucab...@lucabert.de> schrieb: Hallo again > I configured an user for my mobile phone and I can call, but as soon > as the other party answer, I get this error in Log: > > [Oct 24 15:32:41] NOTICE[3731]: channel.c:4280 __ast_read: Dropping > incompatib

[asterisk-users] Problem using Android SIP-Client

phone with Android 7... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org

Re: [asterisk-users] Voicemail: search for name in a phonebook

ve data sources using the CALLERID(NUM) and change CALLERID(NAME) > to be the name you set. Thanks a lot! I found this page: http://deepliquid.com/blog/archives/59 and I successfully got it working! Regards Luca Bertoncel

[asterisk-users] Voicemail: search for name in a phonebook

to the number in the E-Mail of voicemail. Is it possible? I currently use ${VM_CALLERID} in emailbody and it gives, of course, the phone number... Thanks a lot! Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth

Re: [asterisk-users] Need to restart Asterisk if remote server not working?

e WRT device or just restarted the Asterisk > service to resolve your problem? Maybe it's less an Asterisk issue but one > with DNS caching on this device? I just restarted Asterisk... Thanks Luca Bertoncello (lucab...@lucabert.de) pgpqpqFm8dZW7.pgp Description: Digitale Signatur von

Re: [asterisk-users] Need to restart Asterisk if remote server not working?

our SIP channel driver > supports it ;-) Could you say me how can I disable the SRV lookups? I use Asterisk 1.8.30.0 on an OpenWRT device. > You should also use the dnsmgr of Asterisk, resp. configuring it to > reasonable values. In dnsmgr.conf I set: The version of Asterisk on

Re: [asterisk-users] Need to restart Asterisk if remote server not working?

ine.de > 4. Did the IP address of Telekom's end of the connection change? I really don't know, but I suppose not > 5. Did the IP address of your end of the connection change? No. Thanks Luca Bertoncello (lucab...@lucabert.de) --

[asterisk-users] Need to restart Asterisk if remote server not working?

... I think, this should not be normal... Can someone explain me why it happens and what I have to change in the configuration to avoid this problem? Thanks a lot Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth

[asterisk-users] [OT] Suggestion for VoIP-App

App I found always ask if I want to phone via GSM or VoIP... Thanks for your suggestion and sorry for the OT! Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check

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

Luca Bertoncello <lucab...@lucabert.de> schrieb: Hi again! > The problem: after 15 minutes will the call dropped, but only if the call is > to another nation! If I just call another phone in Germany, I can speak > longer than 15 minutes... After a long work, and with the huge

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