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

responding 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) -- _ -- Ban

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

LLERID(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 Bertoncello (lu

[asterisk-users] Problem using Android SIP-Client

use my new 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.as

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

Luca Bertoncello 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 > incompatible voice frame on SIP/mess

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

es 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,

[asterisk-users] Remote Phonebook with Thomson ST2022

8.200.10 - - [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-dig

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

.. Thanks a lot 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/ Ne

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

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 (lucab...@lucabert.de) -- _ -- Ban

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

Zitat von Tzafrir Cohen : 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..

[asterisk-users] Problem with DAHDI

onfig: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs. [Feb 15 18:43:29] ERROR[3428]: codec_dahdi.c:820 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory Asterisk Ready. it does not seems to be normal, but I can't understand why /dev/dahdi/c

Re: [asterisk-users] Problem with DAHDI

1234567- 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) --

Re: [asterisk-users] Problem with DAHDI

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

[asterisk-users] Unable to use VoIP-device

voip-0013;2' The desk-VoIP-phone is in the network "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.

[asterisk-users] High delay and some echo

know 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...@lucabe

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

bps up). The 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? Th

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

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

> for very small setups such as yours, but it might still be 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) -- _

[asterisk-users] Delay on speak with Asterisk

ider, 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 Ast

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

c2833 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 Coloca

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 deactiva

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

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

[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] 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

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 to

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 str

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

> 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) -- _

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

m really 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 Asteris

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

ourse, it could be possible 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) -- _

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

son phone > from it? 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

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

s her Thomson 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 r

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

ce he had time for me today, 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 Provid

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 Berton

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

one is: 1) connecting it to my Asterisk 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) -- _ -

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

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

ot have any problem to manage 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 communit

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 &g

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 21:50 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

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

ic shaper which is imposing this limit? 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 sug

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

ince 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.aster

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

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 intern

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

siness 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 Bertonce

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

ow anymore what 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.asteri

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

g the "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

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

ntradict my conclusions? 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-dig

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

Gateway has to manage the internal domains, 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 (luc

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 (

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

m. Should 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

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 co

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...)

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 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...)

--tcp-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...@lucabe

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...)

comes... :( 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://ww

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

ets 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...)

l, probabilly not on the PI, since, as I sayd, 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 t

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

st 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 -- C

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

same LAN 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) -- _

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...)

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...)

rt has errors... Now the question: can someone help me to understand/learn how to check the involved parts and search for the problem? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided

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

[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...@lucabe

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

risk configuration? 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.asteris

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

e to create a "fake" Zone 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

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

in", 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 ch

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

line.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:/

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? >>> >>> Take a

[asterisk-users] Notifying missed calls

voicemail, 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 comm

Re: [asterisk-users] Notifying missed calls

al(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

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!

Re: [asterisk-users] Notifying missed calls

r,s,1) 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 wha

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

uot; i...@mydomain.de") 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"

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 => _+49X.,1,got

[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] 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... The

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

t but 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

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

the VLAN 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://communi

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

derstand 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

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

meone 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 t

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

kom as 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) -- _ -- Bandwi

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

er 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...@

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

CALL: ${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”]?

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

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. Re

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

[asterisk-users] Cannot send faxes

6 18:34:37.51: [26819]: --> [2:OK] Aug 16 18:34:37.51: [26819]: DIAL 0177yyy Aug 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 [

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

will be a free 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 (

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

the calls to 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

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

th 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 webin

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

e > inherently worse than over a regular phone line. They can be made to be > almost as good or they can just be horrible, but either way, faxing is no > fun, especially considering that the problems can be caused before the fax > ever reaches your system. Hopefully your provider supports

[asterisk-users] Peer is UNREACHABLE

nitored: 1 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...@l

Re: [asterisk-users] Peer is UNREACHABLE

igured 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.

Re: [asterisk-users] Peer is UNREACHABLE

. If you > do post the relevant parts of your config in here, you might want to > obscure the secret. Which part of the configuration do you need? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Coloc

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