[asterisk-users] [Maybe OT]: SIP Provider
Hi all! Currently I'm using Messagenet, a SIP-Provider in Italy, to have an italian number via VoIP, _to receive calls only_. I use it to allow my friends and parents in Italy to call me in Germany without paying too much. This service was free of charge in the last years. Now will Messagenet beginning from end of november, to cancel this free service and only offer paying services (for receiving and calling). Since I don't need to call from this number (using Deutsche Telekom I already can call Italy for free), I'm currently searching an alternative. The best 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 (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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cannot send faxes
Hi!
Not sure how long I have the problem, since I really don't send many
faxes, but today I need to send one, and it does not work...
I use HylaFAX 6.0.6-8.1 from Debian repositories and IAXModem
1.2.0~dfsg-3 even from Debian repos.
IAXModem is configured so:
device /dev/ttyIAX
owner uucp:uucp
mode660
port4569
refresh 60
server pbxint.lucabert.intra
peername0049351xxx
secret myverypassword
cidname Familie Bertoncello
cidnumber 0049351xxx
codec alaw
I tried (to debug) to send me a "fax" to my mobile phone.
The number will be called, but if I answer, instead of hearing the usual
"sound" of a fax, I only hear a "wait tune".
In Asterisk I see:
-- Accepting AUTHENTICATED call from 192.168.10.3:4569:
--> requested format = alaw,
--> requested prefs = (),
--> actual format = alaw,
--> host prefs = (alaw),
--> priority = mine
-- Executing [0177yyy@fax-out:1]
NoOp("IAX2/0049351xxx-2884", "") in new stack
-- Executing [0177yyy@fax-out:2]
Verbose("IAX2/0049351xxx-2884", "2,Call from FAX") in new stack
== Call from FAX
-- Executing [0177yyy@fax-out:3]
Dial("IAX2/0049351xxx-2884",
"PJSIP/pbxfax/sip:0177...@tel.t-online.de,,R") in new stack
-- Called PJSIP/pbxfax/sip:0177...@tel.t-online.de
> 0x7f8ec402c000 -- Strict RTP learning after remote address set
to: 217.0.5.212:27590
-- PJSIP/pbxfax-0086 is ringing
> 0x7f8ec402c000 -- Strict RTP switching to RTP target address
217.0.5.212:27590 as source
-- PJSIP/pbxfax-0086 is ringing
-- PJSIP/pbxfax-0086 answered IAX2/0049351xxx-2884
-- Channel PJSIP/pbxfax-0086 joined 'simple_bridge' basic-bridge
<9c5b5ca2-d4d9-42e9-b5f8-9532e09cc360>
-- Channel IAX2/0049351xxx-2884 joined 'simple_bridge'
basic-bridge <9c5b5ca2-d4d9-42e9-b5f8-9532e09cc360>
> 0x7f8ec402c000 -- Strict RTP learning complete - Locking on
source address 217.0.5.212:27590
-- Channel PJSIP/pbxfax-0086 left 'simple_bridge' basic-bridge
<9c5b5ca2-d4d9-42e9-b5f8-9532e09cc360>
-- Channel IAX2/0049351xxx-2884 left 'simple_bridge'
basic-bridge <9c5b5ca2-d4d9-42e9-b5f8-9532e09cc360>
== Spawn extension (fax-out, 0177yyy, 3) exited non-zero on
'IAX2/0049351xxx-2884'
-- Hungup 'IAX2/0049351xxx-2884'
HylaFAX said:
Aug 16 18:34:37.51: [26819]: SESSION BEGIN 00371 491773218409
Aug 16 18:34:37.51: [26819]: HylaFAX (tm) Version 6.0.6
Aug 16 18:34:37.51: [26819]: SEND FAX: JOB 39 DEST 0177yyy COMMID
00371 DEVICE '/dev/ttyIAX' FROM 'lucabert '
USER www-data
Aug 16 18:34:37.51: [26819]: <-- [12:AT+FCLASS=1\r]
Aug 16 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
[2] No carrier detected
Aug 16 18:34:53.66: [26819]: <-- [5:ATH0\r]
Aug 16 18:34:53.66: [26819]: --> [2:OK]
Aug 16 18:34:53.66: [26819]: SESSION END
Any idea what can be the problem?
Thanks a lot
Luca Bertoncello
(lucab...@lucabert.de)
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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 new year! 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Called number changed on SNOM 821
Am 31.12.2021 um 16:04 schrieb Antony Stone:
Hi
> Check the Dial() command which places the call to the phone. Does it contain
> the "c" option?
Jup...
exten => _529874,1,Verbose(2,Call for Main - [${CALLERID(num)}])
exten => _529874,n,Set(CALLERID(num)=${IF($[ "${CALLERID(num):0:3}" =
"+49" ]?0${CALLERID(num):3}:${CALLERID(num)})})
exten => _529874,n,Set(CHANNEL(musicclass)=default)
exten => _529874,n,Dial(SIP/74,39,RcxX)
exten => _529874,n,Verbose(2,Voicemail for Main)
exten => _529874,n,Set(CALLERID(name)=)
; Damit in der 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)
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Re: [asterisk-users] Called number changed on SNOM 821
Am 31.12.2021 um 14:39 schrieb Antony Stone:
Hi Antony
>> Last very strange problem is, that the list of missed calls on the phone
>> is always empty...
>
> Check the SIP notifications which are being sent to the telephone for these
> calls, and whether any of them contain a "Reason" code for "Answered
> elsewhere".
Got it...
Via: SIP/2.0/UDP 192.168.60.1:5060;branch=z9hG4bK5d77ab07;rport
Max-Forwards: 70
From: ;tag=as7a4dc11e
To:
Call-ID: 5372968b4a66dac5051ce91e29f8b283@192.168.60.1:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u2
Reason: SIP;cause=200;text="Call completed elsewhere"
Content-Length: 0
> This "answered elsewhere" code is usually used when telephones are in a ring
> group or agents subscribed to a queue, and nobody wants to know about the
> calls which someone else answered, even if their telephone rang, so the phone
> sees this code and eliminates the call from its history.
Now the very question is how to remove this header...
Can the problem be that I added a function to send an E-Mail if a call
wasn't answered?
[noanswer]
exten => s,1,NoOp(UID 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”]?done)
exten => h,n,Goto(noanswer,s,1)
exten => h,n(done),NoOp()
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
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Re: [asterisk-users] Called number changed on SNOM 821
Am 28.12.2021 um 21:21 schrieb Antony Stone: Hi > However, at least you've got as far as ruling out Telekom as being the source > of the problem, which I think is good. So, I setted: sendrpid=rpid instead of: sendrpid=pai and now it seems to work. The called number 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...@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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Called number changed on SNOM 821
Am 28.12.2021 um 21:21 schrieb Antony Stone: > I would look at whatever part of the dial plan is responsible for inserting > "Sekretariat", and also check whether you have "sendrpid=yes" in sip.conf. I find "Sekretariat" in the Ringing sent from Asterisk: Via: SIP/2.0/UDP 192.168.60.53:3072;branch=z9hG4bK-igym6msxn88p;received=192.168.60.53;rport=3072 From: "Sekretariat" ;tag=ts2ye4krhs To: ;tag=as32fe51ba Call-ID: 313634303731393637343630373636-ex7145moy1mt CSeq: 2 INVITE Server: Asterisk PBX 16.2.1~dfsg-1+deb10u2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Contact: Content-Length: 0 In the users.d/74.conf (the configuration of the SIP client for "Sekretariat") I have: sendrpid=pai > However, at least you've got as far as ruling out Telekom 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) -- _ -- 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Called number changed on SNOM 821
Am 28.12.2021 um 20:24 schrieb Antony Stone: > No, you want to look at the "180 Ringing" response in both cases - what goes > in to Asterisk, and what comes out of it. OK > No, data FROM Deutsche Telekom. They are the ones sending the "180 Ringing" > back to you once they think the external telephone is ringing. OK. So I sniffed data from internal network and from DSL, then I started the call using the web management system of the SNOM. I see Asterisk sends to the phone: Via: SIP/2.0/UDP 192.168.60.53:3072;branch=z9hG4bK-igym6msxn88p;received=192.168.60.53;rport=3072 From: "Sekretariat" ;tag=ts2ye4krhs To: ;tag=as32fe51ba Call-ID: 313634303731393637343630373636-ex7145moy1mt CSeq: 2 INVITE Server: Asterisk PBX 16.2.1~dfsg-1+deb10u2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Contact: Content-Length: 0 After about 6 seconds I get from the Telekom: Via: SIP/2.0/UDP 87.191.224.158:5060;received=87.191.224.158;rport=5060;branch=z9hG4bKPj43be873a-cf55-4348-8867-5c2bb97bd76a To: ;tag=h7g4Esbg_p65544t1640719676m169304c9321s1_3514393582-932943693 From: ;tag=4781eb96-b155-421e-8206-593d44c9f7c4 Call-ID: 478ba582-946c-46ac-984d-6f1835e3391b CSeq: 15716 INVITE Contact: Record-Route: P-Early-Media: sendrecv, gated Require: 100rel RSeq: 2 Content-Type: application/sdp Content-Length: 281 Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, PUBLISH, MESSAGE, UPDATE, PRACK, INFO, INVITE, ACK, OPTIONS, CANCEL, BYE Then I see Asterisk sends this to the phone: Via: SIP/2.0/UDP 192.168.60.53:3072;branch=z9hG4bK-igym6msxn88p;received=192.168.60.53;rport=3072 From: "Sekretariat" ;tag=ts2ye4krhs To: ;tag=as32fe51ba Call-ID: 313634303731393637343630373636-ex7145moy1mt CSeq: 2 INVITE Server: Asterisk PBX 16.2.1~dfsg-1+deb10u2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Contact: P-Asserted-Identity: "03529529874" Content-Length: 0 So, it seems Asterisk receives from Deutsche Telekom _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 and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Called number changed on SNOM 821
Am 28.12.2021 um 20:00 schrieb Antony Stone: > Which way round are you making the telephone call? > > From your earlier packet capture, it looked to me like you were dialling an > external number from an internal telephone. This is correct! I called my mobile phone using a VoIP phone connected to an Asterisk. > If that is true, then you should be looking for a packet *from Telekom* > coming > in to Asterisk, and a packet *from Asterisk* to the internal telephone - > remember that these packets are the _reply_ to the INVITE. > > INVITE goes from callING telephone to callED telephone. > > Response "180 Ringing" goes from the callED telephone to the callING > telephone. So do I have to compare the INVITE with the Ringing? > I think it's important to find out what Asterisk is receiving from your > upstream provider, and whether it is then changing this in what it sends on > to > the calling telephone (the one on which you see the unexpected display). OK, so I have to sniff the data to Deutsche Telekom and not the internal network... Since the Asterisk is not my own, but of a company, I have to ask someone 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 new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Called number changed on SNOM 821
Am 28.12.2021 um 19:41 schrieb Antony Stone: Hi Antony, > Okay, so, returning to my question, do yu see any difference between the > packet > inbound to Asterisk from the called telephone, and the packet outbound from > Asterisk to the calling telephone? I'm trying to understand 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 mean... :( 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 to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Called number changed on SNOM 821
Am 28.12.2021 um 17:35 schrieb Antony Stone: >> So, I see, there is a "P-Asserted-Identity"... But I can't understand >> why... >> >> Any idea? > > Where exactly were those packets captured? tcpdump on the Asterisk-Server on the interface of 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://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Called number changed on SNOM 821
Am 28.12.2021 um 17:22 schrieb Antony Stone: Hi Antony > I mean the response from the called telephone in reply to the INVITE, which > contains the SIP code "180 Ringing" and may optionally have an RPID header. OK, I see something strange... Here what I see if I call my mobile phone (then the number "changes"): Via: SIP/2.0/UDP 192.168.60.53:3072;branch=z9hG4bK-tzms3vciubaj;received=192.168.60.53;rport=3072 From: "Sekretariat" ;tag=8o3bow73en To: ;tag=as234ad778 Call-ID: 313634303730333537323534383539-zw0qs5wmmpd2 CSeq: 2 INVITE Server: Asterisk PBX 16.2.1~dfsg-1+deb10u2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Contact: P-Asserted-Identity: "03529529874" Content-Length: 0 and here what I see if I call another mobile phone (then the number does NOT change): Via: SIP/2.0/UDP 192.168.60.53:3072;branch=z9hG4bK-qjbxwwkv3n3p;received=192.168.60.53;rport=3072 From: "Sekretariat" ;tag=fararstgh4 To: ;tag=as7da4425c Call-ID: 313634303730333630353537373731-7x9ey0nf5gm2 CSeq: 2 INVITE Server: Asterisk PBX 16.2.1~dfsg-1+deb10u2 Allow: INVITE, ACK, CANCEL, 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) -- _ -- 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Called number changed on SNOM 821
Am 28.12.2021 um 15:42 schrieb Antony Stone: Hi Antony, > Sounds like something strange is happening with Remote-Party-ID.> > Do a packet capture and see whether the 180 response from the callee's phone > contains an RPID header with silly content. I captured the packet 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 new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 problem happens only on some numbers, but not on some other... Any idea? 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 to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Called number changed on SNOM 821
Dear list, I have a very strange problem and no idea where the problem can be... I have a Debian Server with Asterisk 16.2.1 from Debian repos and some SNOM phones (SNOM 821, last firmware snom821-SIP 8.7.5.35). 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. 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 (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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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,goto(${EXTEN:3},1)
> exten => _0049X.,1,goto(${EXTEN:4},1)
> exten => _03529X.,1,goto(${EXTEN:1},1)
> exten => _3529X.,1,goto(${EXTEN:4},1)
>
> exten => _123456,1,Verbose(2,Call for Main - [${CALLERID(num)}])
> exten => _123456,n,Set(CALLERID(num)=${IF($[ "${CALLERID(num):0:3}" =
> "+49" ]?0${CALLERID(num):3}:${CALLERID(num)})})
> exten => _123456,n,Set(CHANNEL(musicclass)=default)
> exten => _123456,n,Dial(SIP/74,39,RcxX)
> exten => _123456,n,Verbose(2,Voicemail for Main)
> exten => _123456,n,Set(CALLERID(name)=)
> exten => _123456,n,VoiceMail(74,us)
> exten => _123456,n,Hangup
You are my hero!
It works as expected!
Thank you very very much!
Luca
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Re: [asterisk-users] Notifying missed calls
Am 06.11.2021 um 15:06 schrieb Frank Vanoni:
Hi Frank
> The "h" extension is executed whenever a call is hang up in that
> contexts.
>
> In your configuration it executes first the "s" extension (where you
> GoTo h,1) and once that is executed, the "h" extension is executed
> again.
OK, I modified my configuration so:
[main_incoming]
exten => _00493529123456,1,Verbose(2,Call for Main - [${CALLERID(num)}])
exten => _00493529123456,n,Dial(local/123456@main_incoming,,xX)
exten => _03529123456,1,Verbose(2,Call for Main - [${CALLERID(num)}])
exten => _03529123456,n,Dial(local/123456@main_incoming,,xX)
exten => _123456,1,Verbose(2,Call for Main - [${CALLERID(num)}])
exten => _123456,n,Set(CALLERID(num)=${IF($[ "${CALLERID(num):0:3}" =
"+49" ]?0${CALLERID(num):3}:${CALLERID(num)})})
exten => _123456,n,Set(CHANNEL(musicclass)=default)
exten => _123456,n,Dial(SIP/74,39,RcxX)
exten => _123456,n,Verbose(2,Voicemail for Main)
exten => _123456,n,Set(CALLERID(name)=)
exten => _123456,n,VoiceMail(74,us)
exten => _123456,n,Hangup
include => fax_incoming
include => michael_incoming
include => internal_calls
exten => h,1,GotoIf($[“${DIALSTATUS}” = “ANSWER”]?done)
exten => h,n,Goto(noanswer,s,1)
exten => h,n(done),NoOp()
Unfortunately two E-Mails are sent anyway...
This is the Asterisk log:
-- Executing [00493529123456@michael_incoming:1]
Verbose("PJSIP/pbxmichael_in-0418", "2,Call for Main -
[+4935]") in new stack
== Call for Main - [+4935]
-- Executing [00493529123456@michael_incoming:2]
Dial("PJSIP/pbxmichael_in-0418", "local/123456@main_incoming,,xX")
in new stack
-- Called local/123456@main_incoming
-- Executing [123456@main_incoming:1]
Verbose("Local/123456@main_incoming-0268;2", "2,Call for Main -
[+4935]") in new stack
== Call for Main - [+4935]
-- Executing [123456@main_incoming:2]
Set("Local/123456@main_incoming-0268;2",
"CALLERID(num)=035") in new stack
-- Executing [123456@main_incoming:3]
Set("Local/123456@main_incoming-0268;2",
"CHANNEL(musicclass)=default") in new stack
-- Executing [123456@main_incoming:4]
Dial("Local/123456@main_incoming-0268;2", "SIP/74,39,RcxX") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/74
-- Local/123456@main_incoming-0268;1 is ringing
-- SIP/74-0462 is ringing
-- Local/123456@main_incoming-0268;1 is ringing
-- SIP/74-0462 is ringing
-- SIP/74-0462 is ringing
-- SIP/74-0462 is ringing
== Spawn extension (michael_incoming, 00493529123456, 2) exited
non-zero on 'PJSIP/pbxmichael_in-0418'
-- Executing [h@michael_incoming:1]
GotoIf("PJSIP/pbxmichael_in-0418", "0?done") in new stack
-- Executing [h@michael_incoming:2]
Goto("PJSIP/pbxmichael_in-0418", "noanswer,s,1") in new stack
-- Goto (noanswer,s,1)
== Spawn extension (main_incoming, 123456, 4) exited non-zero on
'Local/123456@main_incoming-0268;2'
-- Executing [h@main_incoming:1]
GotoIf("Local/123456@main_incoming-0268;2", "0?done") in new stack
-- Executing [s@noanswer:1] NoOp("PJSIP/pbxmichael_in-0418",
"UID CALL: 1636222382.6030 / DATE: 20211106-191306)") in new stack
-- Executing [h@main_incoming:2]
Goto("Local/123456@main_incoming-0268;2", "noanswer,s,1") in new stack
-- Goto (noanswer,s,1)
-- Executing [s@noanswer:2] System("PJSIP/pbxmichael_in-0418",
"echo "Verpasster Anruf vom +4935 um 19:13" | mail -s
"Verpasster Anruf" 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", "echo "Verpasster Anruf
vom 035 um 19:13" | mail -s "Verpasster Anruf"
i...@mydomain.de") in new stack
Any other idea?
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
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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 of "main-incoming,h,2"
Really, I can't understand what you mean... I'm feeling really dumb...
>> 2) The E-Mails will be sent for outgoing unanswered calls, too.
>
> Use the "h" extension only in the context for incoming calls
I have just one "h" extension:
[main_incoming]
exten => h,1,GotoIf($[“${DIALSTATUS}” = “ANSWER”]?done)
exten => h,n,Goto(noanswer,s,1)
exten => h,n(done),NoOp()
Could you explain what you mean?
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
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Re: [asterisk-users] Notifying missed calls
Am 03.11.2021 21:34, schrieb Antony Stone:
Hi again
The n there should be 1, surely?
exten => h,n,Hangup
I would say "remove that line". The call has already been hung up, so
calling
Hangup is at best going to go into a recursive loop - it certainly
isn't going
to help.
This is my current configuration:
[cch]
exten => _X.,1,Verbose(2,DEFAULT)
include => internal_calls
include => main_incoming
include => fax_incoming
include => michael_incoming
include => myproxy
[noanswer]
exten => s,1,NoOp(UID 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...@.de)
[main_incoming]
exten => h,1,GotoIf($[“${DIALSTATUS}” = “ANSWER”]?done)
exten => h,n,Goto(noanswer,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 what is wrong in my configuration?
Thanks a lot
Luca Bertoncello
(lucab...@lucabert.de)
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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! 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Notifying missed calls
Am 03.11.2021 um 21:24 schrieb Doug Lytle:
>>>> but if the called hangs up prior the timeout for the voicemail, the
>>>> Subrouting "noanswer" will not called...
>
> You can use the h priority for that.
>
> https://wiki.asterisk.org/wiki/display/AST/Special+Dialplan+Extensions
Hi Doug,
could you send me an example?
I tried so:
exten => h,n(hang),Gosub(noanswer,s,1)
exten => h,n,Hangup
exten => _xx,1,Verbose(2,Call for Main - [${CALLERID(num)}])
exten => _xx,n,Set(CHANNEL(musicclass)=default)
exten => _xx,n,Dial(SIP/74,39,RcxX)
exten => _xx,n,Verbose(2,Voicemail for Main)
exten => _xx,n,Set(CALLERID(name)=)
exten => _xx,n,Gosub(noanswer,s,1)
exten => _xx,n,VoiceMail(74,us)
exten => _xx,n,Hangup
But it does not work...
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
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[asterisk-users] Notifying missed calls
Hi list!
I have this problem: I'd like to notify the user about missed calls.
With "missed call" I mean: the caller calls, wait a while and hangup
_before_ the voicemail starts.
I got it call a script just before the voicemail starts, so:
exten => s,1,Verbose(2,Call for Main - [${CALLERID(num)}])
exten => s,n,Set(CHANNEL(musicclass)=default)
exten => s,n,Dial(SIP/74,39,RcxX)
exten => s,n,Verbose(2,Voicemail for Main)
exten => s,n,Set(CALLERID(name)=)
exten => s,n,Gosub(noanswer,s,1)
exten => s,n,VoiceMail(74,us)
exten => s,n,Hangup
[noanswer]
exten => s,1,NoOp(UID 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)}" > /tmp/calllog.txt)
exten => s,n,Return()
but if the called hangs up prior the timeout for the voicemail, the
Subrouting "noanswer" will not called...
Any ideas?
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
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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 look here [1] >> >> So, maybe I got it... >> I tested the configuration with my Fax number and it seems to work (= I >> can call the fax and can call my mobile phone from the fax with >> "originate..."). > > Congrats! So, it seems it does NOT work as expected... I tried to activate the FAX and it works, then I activated my number and it works, too. Finally I activated the number of my wife and it does not work anymore... If I call the number I can only see (verbose 42): [Feb 18 19:57:12] NOTICE[19379] res_pjsip/pjsip_distributor.c: Request 'INVITE' from '' failed for '217.0.21.64:5060' (callid: p65550t1613674632m753568c93349s2) - No matching endpoint found and no phone rings... After that, even if I restore the single number to SIP I only get the error and nothing work, until I restored _ALL_ numbers to SIP. Do someone has an explanation and (better!) a solution to the problem? 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 to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?
Am 16.02.2021 um 22:32 schrieb Michael Maier:
Hi Michael
>> Maybe could you send me an abstract of your configuration?
>
> Take a look here [1]
So, maybe I got it...
I tested the configuration with my Fax number and it seems to work (= I
can call the fax and can call my mobile phone from the fax with
"originate...").
On the registration I have:
[pbxfax]
type = registration
retry_interval = 20
max_retries = 10
contact_user = 00493514977291
expiration = 120
transport = transport-udp
outbound_auth = pbxfax
client_uri = sip:03514977...@tel.t-online.de
server_uri = sip:tel.t-online.de
First: can I use tel.t-online.de or _MUST_ I change it? If I understand
your previous E-Mail, I'd say that I can leave tel.t-online.de...
Then I have a question by the Dialplan... Currently I have:
[fax-out]
exten => _X.,1,NoOp()
exten => _X.,n,Verbose(2,Call from FAX)
exten => _X.,n,Dial(SIP/pbxfax/${EXTEN},,R)
And I'll replace it with:
[fax-out]
exten => _X.,1,NoOp()
exten => _X.,n,Verbose(2,Call from FAX)
exten => _X.,n,Dial(PJSIP/pbxfax/sip:${EXTEN}@tel.t-online.de,,R)
Is it correct? I tried with
"PJSIP/pbxfax/pjsip:${EXTEN}@tel.t-online.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)
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Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?
Am 16.02.2021 um 19:56 schrieb Michael Maier: Hi Michael, >> Do I use pjsip? > > pjsip show registrations gw*CLI> pjsip show registrations No objects found. So I don't use pjsip... :( Maybe could you send me an abstract of your configuration? >> You mean, I have to create a "fake" Zone tel.t-online.de in my Bind with >> these settings? Looks like dangerous, if they changes something... > > If you do that statically -> yes, you're right. You have to do it > dynamically. I attached a script, which can be used to dynamically build > a rpz each 15 minutes e.g. It directly asks the telekom nameserver for > naptr and srv entries. It looks like this: > > server 192.168.62.13 > zone rpz-tonline > update delete tel.t-online.de.rpz-tonline. > update delete _sips._tcp.tel.t-online.de.rpz-tonline. > update delete _sip._tcp.tel.t-online.de.rpz-tonline. > update add tel.t-online.de.rpz-tonline. 60 NAPTR 10 0 "s" > "SIPS+D2T" "" _sips._tcp.tel.t-online.de. > update add tel.t-online.de.rpz-tonline. 60 NAPTR 30 0 "s" > "SIP+D2T" "" _sip._tcp.tel.t-online.de. > update add _sips._tcp.tel.t-online.de.rpz-tonline. 60 SRV 10 0 > 5061 s-eps-110.edns.t-ipnet.de. > update add _sip._tcp.tel.t-online.de.rpz-tonline. 60 SRV 10 0 > 5060 s-epp-110.edns.t-ipnet.de. > send So if I undestand what you mean, you check the NAPTR and SRV für _sips._tcp.tel.t-online.de and save the record in a "virtual domain" rpz-tonline, is it correct? Then I suppose you use this domain instead of tel.t-online.de in the SIP configuratione as "host", "outboundproxy" and "fromdomain", 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... > Independently you have to add your own trunk names to get it working > (telekomPJSIP-a, ...). Could you explain me that? I'm not an expert of Asterisk... :( 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?
Am 15.02.2021 um 21:40 schrieb Michael Maier: Hi Michael, > They're switching to DNS NAPTR / SRV[1]. If you are using Asterisk / > pjsip and hostnames (tel.t-online.de e.g. for the AllIP service), you Mmm... I'm using tel.t-online.de, but I'm not sure I'm using pjsip... module show say me: res_pjsip.so Basic SIP resource 46 Running core Do I use pjsip? > won't have any problem (using asterisk 14 or higher), because it's > default. But you may have problems with the handling of the calls, > because Telekom needs the client always to use the same server for all > activities after the register has been done (the SRV entries contain 3 > servers and asterisk will use them "randomly" if it detects a problem - > regardless which server of the list has been used for registration - > this won't work with Telekom and will lead to not working outbound calls > / interrupted calls e.g.). This won't happen very often (because they > have been extremely stable in the past), but I could see it nevertheless > already. If you want to be really sure to not face this problem, you > have to create a workaround by adding a rpz zone e.g. with an own bind, > which is fed by an own job and presents asterisk just one server when > looking up the SRV entries after the NAPTR call. NAPTR / SRV works like > this (example for tel.t-online.de): You mean, I have 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 http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?
Hi list! I received a letter from Deutsche Telekom where they say me, that I need to change "something" on my router until 28.02.2021, otherwise I cannot phone anymore. Since I use Asterisk and I don't have a router, I'm not sure what I need to do... In the letter there is an URL to "explain" how to change the configuration if I use a VoIP-phone, but they only say, that I don't have to use Port 5060, but Port 0... Surely there are in this list someone other using Deutsche Telekom... Does someone of them understand what I should change in the Asterisk 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.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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) -- _ -- 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 without vectoring. > Of course, Deutsche Telekom said nothing about this change... > > Well, I got it working, and now I have 48Mbps down and 10Mbps up. > I _REALLY CAN'T_ believe, that this is not enough... > > The problem with many little disruptions during calls is always here. > > I tried changing the codecs and changing some settings in the SIP > configuration of the peers. > No changes... > > On the Gateway (Banana PI), where the Asterisk server also runs, the > load is about 0.50 during calls and it has a Gbps LAN. > I can't believe, the problem is here... So, now I know what was the problem and I solved it... The problem was: the Banana PI... :( I checked it with mtr and I see really bad times to communicate with other devices im same networks (~2 - 380 ms!!). Many tries with other Switch ports and so on didn't solved the problem. So I bought a mini PC and I configured it as Firewall with Asterisk. mtr give now really good times (~0.2 - 0.4 ms). And Asterisk works very good... I tried right now with my father in law and the communication was excellent, without any disruptions. So, I really thank you for the idea, that my Banana PI can be the problem. It was! ;) Have a nice weekend! 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice broken during calls (again...)
Am 24.06.2020 05:05, schrieb Michael Maier: Hi Your basic architecture looks good to me - now you have to start the Nice to hear it... analysis of the problem with pcapsipdump and wireshark as I wrote before to get an idea what actually happens at all. You most probably won't come any further without doing any analyzing. You have to learn it. It will take some, or even more, time. You can't do it in just few hours or maybe even days or weeks. It is work or even hard work to learn and to do it. Well, that's the very problem... I don't know *how* to analyze it... Or, better, how to read the data... I know, I can use tcpdump, sngrep and many other tools, but I don't know what I have to expect and how to decide, that a paket is wrong... Can someone help me to learn it? That's my problem: It's impossible for me to assist because I can't see any effort on your side to learn. I won't fix your problem. You have to fix it yourself. All I can do, is, to show you a way to *find* your problem (I can't know your problem) and may be to give some hints how to fix it (once you've found it). Finding / localizing problems and fixing them are two completely different things. But before you fix a problem, you have to know the problem. Therefore: go and find your problem by starting the analysis. That's the first thing to do. Of course, the first thing to do is to locate the problem... I think, the problem can *not* be: 1) by Deutsche Telekom (since I have the problem even on local tests) 2) on the devices (since I tested many devices) so the problem can be: 1) in the Firewall configuration 2) in the Asterisk configuration 3) in the BananaPI I think I can exclude my switch, since I checked right now and no port 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 by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 size 1466 would pass and higher will have the same result. It > > RTP-VoIP-packets never reach this size. Size is about 214 bytes. OK, so it must be something other... But I really don't have any idea what... :( Thanks Luca -- _ -- 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice broken during calls (again...)
Am 23.06.2020 um 17:04 schrieb Marek Greško: > I interchanged LAN and LTE in the sentence. OK... > Do you have some kind of NAT in fron of asterisk? Or is your asterisk No, Asterisk has a public IP. No NAT in front of Asterisk... > having public IP? Could you share sip.conf (without passwords)? One > LAN client, one LTE and general section. Of course: my outgoing configuration: [pbxluca] type=peer defaultuser= secret= dtmfmode=rfc2833 host=tel.t-online.de context=luca_incoming outboundproxy=tel.t-online.de port=5060 fromuser=035 fromdomain=tel.t-online.de usereqphone=yes canreinvite=yes insecure=port,invite nat=no qualify=yes qualifyfreq=600 disallow=all allow=alaw allow=ulaw my phone configuration: ; Lucas Telefon [004935] fullname = 004935 secret = hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = default host = dynamic dtmfmode=rfc2833 canreinvite=no sendrpid=pai type=friend nat=force_rport,comedia qualify=yes qualifyfreq=60 avpf=no force_avp=no icesupport=no encryption=no callgroup=1 pickupgroup=1 deny=0.0.0.0/0.0.0.0 permit=192.168.200.0/255.255.255.0 dial=SIP/004935 my mobile phone: ; Lucas Handy [0049177222] fullname = 0049177222 secret = hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = default host = dynamic dtmfmode=rfc2833 canreinvite=no sendrpid=pai type=friend nat=force_rport,comedia qualify=yes qualifyfreq=60 avpf=no force_avp=no icesupport=no encryption=no callgroup=1 pickupgroup=1 dial=SIP/0049177222 allow = all sip.conf: [general] context=default allowoverlap=no udpbindaddr=0.0.0.0:25572 tcpenable=yes tcpbindaddr=0.0.0.0:25572 tlsenable=no tlsbindaddr=0.0.0.0:25573 transport=udp srvlookup=no minexpiry=480 defaultexpiry=480 disallow=all allow=alaw allow=ulaw allow=ilbc allow=g729 allow=g723 allow=gsm language=de alwaysauthreject = yes tlscertfile=/etc/asterisk/ssl/asterisk.pem tlscafile=/etc/asterisk/ssl/ca.crt tlscipher=ALL tlsclientmethod=tlsv1 callcounter = yes t38pt_udptl = yes faxdetect = no register => 035::-0...@t-online.de@pbxluca/004935 register => 0351112::-0...@t-online.de@pbxfax/0049351112 register => 0351113::-0...@t-online.de@pbxanika/0049351113 register => 5:@messagenet/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 http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice broken during calls (again...)
Am 23.06.2020 16:22, schrieb Marek Greško: It seems your problems lie in something other. Most probably it is not mtu problem. All my suspections are contradicted. If it is true you have inter vlan voice quality problems, it is definitely something different. Formerly I assumed you were trying only LTE vs LAN using internet. I'm not sure what you mean with the last sentence... I tried to connect to my Asterisk via LAN or via DSL (either via LTE or other DSL). Then I noticed that if I call another peer in same network (= both peers via DSL or both peers in the same VLAN), the quality is very good, otherwise is very poor. But why should Asterisk have problem if the peers are in different networks it's for me a really big mistery... This evening I'll try to capture the pakets in a call between two peers connected to Asterisk via LTE, two peers connected in the 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) -- _ -- 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice broken during calls (again...)
Am 23.06.2020 15:43, schrieb Marek Greško: Hi Do you mean "my Linux-Box ignores ICMP packet unreachable" or "Deutsche Telekom ignores them"? I meant DT, but this was a speculation. I did not say they do. I consider it highly improbable. Then I was asking whether you do. As per configuration you sent you are not blocking icmp type 3 so this should not be an issue. OK, so this should not be the problem... What can we check now? If you want, I can send my iptables-script. It is possible, that I have there an error causing this behaviour... Maybe someone in the 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 out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice broken during calls (again...)
Am 23.06.2020 15:15, schrieb Jeff LaCoursiere: Hi Jeff, I have problem calling someone outside my networks and I have problem if the peers are in different networks... I may have missed this originally - are you saying you have trouble when internal phones call each other, if they are on different VLAN's? That's a pretty big deal. There were the results of my yesterday's tests... If both mobile phones using SIP via LTE or both phones are in the same VLAN, the quality is excellent, otherwise it's bad to very bad... But the very problem is, that all other communication between the VLANs don't have any problem?!? I can transfer GB and don't have any issue... I'm really confused... I didn't see my post with the graphs of inter-packet latency make it to the list (moderator?), I think the images were too large. Recall that clearly showed half of the packets coming inbound from DT were *missing*, which confirms your audio experience. I don't think that fact has been addressed properly - it is the only smoking gun you have so far. If that is also happening inter-VLAN, something is seriously wrong on the Pi. Well, 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 this evening. 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.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice broken during calls (again...)
Am 23.06.2020 14:49, schrieb Marek Greško: Hi Marek, this could be ip address of the different interface on the same box. I think it works like expected. The only exception would be if the sip peer ignores the icmp packet unreachable. But I doubt this is the Do you mean "my Linux-Box ignores ICMP packet unreachable" or "Deutsche Telekom ignores them"? case. Anyway you get problems also when calling to LTE phone without using sip provider. I have problem calling someone outside my networks and I have problem if the peers are in different networks... Let first concentrate on these calls LTE to LAN. Are you sure you do not block incoming icmp unreachables? At least verify type 3 subtype 4 is enabled. If it is, I have no clue what is going on. Well, I limit incoming ICMP packets and I block some hosts (known crackers)... If you think, I can send you the script I use (with iptables) to manage my firewall, so you can check it... The only entries I have, having something to do with ICMP, are: -- /bin/echo -n "Disable ICMP Redirect acceptance..." for f in /proc/sys/net/ipv4/conf/*/accept_redirects; do /bin/echo 0 > $f done /bin/echo "done." /sbin/iptables -A INPUT -i dsl0 -p icmp --icmp-type echo-request -m limit --limit 6/m --limit-burst 5 -j ACCEPT /sbin/iptables -A FORWARD -o dsl0 -p icmp -j ACCEPT -- and of course other rules to allow ICMP pakets in the internal networks... Thanks a lot Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice broken during calls (again...)
Am 23.06.2020 10:07, schrieb Marek Greško: Hi this is a correct response: From 62.156.246.57 (62.156.246.57) icmp_seq=1 Frag needed and DF set (mtu = 1492) So PMTU discovery is working. No problem here. You got correct message to lower the packet size from 62.156.246.57. This is probably the last hop before your site. No, the last hop is 62.156.246.65: lucabert@ns:~$ mtr -4nr bpi.d.lucabert.com Start: Tue Jun 23 10:10:16 2020 HOST: ns.lucabert.de Loss% Snt Last Avg Best Wrst StDev 1.|-- 185.242.112.1 0.0%100.4 1.1 0.3 4.4 1.2 2.|-- 84.200.230.82 0.0%100.8 0.7 0.5 0.8 0.0 3.|-- 87.190.233.113 0.0%101.6 1.7 1.4 2.5 0.0 4.|-- 217.5.82.940.0%107.9 7.6 7.4 7.9 0.0 5.|-- 217.5.82.940.0%107.7 7.5 7.2 7.7 0.0 6.|-- 62.156.246.49 0.0%107.4 7.4 7.3 7.4 0.0 7.|-- 62.156.246.65 0.0%107.6 7.6 7.4 7.8 0.0 8.|-- 93.241.91.232 0.0%10 21.4 21.9 21.4 24.3 0.7 Don't know where this 62.156.246.57 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://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice broken during calls (again...)
Am 23.06.2020 09:28, schrieb Marek Greško: Hi if you need clampmss then it is highly probable there is a PMTU discovery problem. The clampmss does not work for UDP. Is there a way to check if I have this problem? I probably counted the size incorrectly. So you are able to ping with size 1464 and not with 1466. How about trying same ping sizes from the internet towards your site? I mean trying to ping from sites with higher MTU than yours without lower MTU links in the path. lucabert@ns:~$ ping -4 -M do -s 1465 bpi.d.lucabert.com PING bpi.d.lucabert.com (93.241.91.232) 1465(1493) bytes of data. From 62.156.246.57 (62.156.246.57) icmp_seq=1 Frag needed and DF set (mtu = 1492) ping: local error: Message too long, mtu=1492 ping: local error: Message too long, mtu=1492 ping: local error: Message too long, mtu=1492 ^C --- bpi.d.lucabert.com ping statistics --- 4 packets 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) -- _ -- 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice broken during calls (again...)
Am 23.06.2020 09:19, schrieb Administrator: Hi Daniel Audio has nothing to do with SIP signaling 5060 port. Look at your rtp.conf You're right... I have to restrict to the ports I configured in rtp.conf... So like: iptables -A FORWARD -p tcp -m multiport --ports -ports 1:15100 --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...@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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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/iptables -A FORWARD -p tcp --tcp-flags SYN,RST SYN -j TCPMSS --clamp-mss-to-pmtu since I had the problem downloading data from an Internet site using my tablet. I found this site explaining that: https://lartc.org/howto/lartc.cookbook.mtu-mss.html I really forgot this entry, but now I checked all entries in my Firewall, and I see it, with my remark... Now, the last line of the HowTo: # iptables -A FORWARD -p tcp --tcp-flags SYN,RST SYN -j TCPMSS --set-mss 128 This sets the MSS of passing SYN packets to 128. Use this if you have VoIP with tiny packets, and huge http packets which are causing chopping in your voice calls. Could it be the problem? Right now I'm not at home, so I cannot test it, but maybe I can add an entry like: iptables -A FORWARD -p tcp -m multiport --ports 5060,SIP> --tcp-flags SYN,RST SYN -j TCPMSS --set-mss 128 and change the previous entry like: iptables -A FORWARD -p tcp -i intlan0 --tcp-flags SYN,RST SYN -j TCPMSS --clamp-mss-to-pmtu to limit the behaviour on the internal LAN... Your opinion? 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 in local Asterisk if one is over LTE and the other in local LAN (but curiously *not* if both peers are in local LAN or both via LTE). I think, the problem with bad quality and broken voice just happens if the peers are in different LANs, since if I call my wife's phone (VLAN "phone") using my mobile phone via SIP (in VLAN "intlan") the quality is bad, but if I call her using my phone in VLAN "phone" or if both peers use SIP via LTE the quality is very good... Could you suggest me something to restrict the problem? Currently, I think the problem can be: 1) on Asterisk 2) on my Gateway/Firewall At home I have many VLANs, that normally *not* communicate together (some exceptions are of course implemented). The phones don't reach the Internet via NAT (VLAN "phone" has no routing in Internet). The mobile phones are in VLAN "intlan", with routing in Internet. Any idea? 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 to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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. Via IPv6 I can use higher MTU, but I really can't explain why... Can someone explain me what does it mean, if this is a problem for VoIP and how I can solve it? 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 to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice broken during calls (again...)
Am 22.06.2020 um 22:42 schrieb Marek Greško: Hi Marek, > there is no need to change canreinvite for provider configuration. OK, so I leave it... > 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 root@bpi:~# ping -M do -s 1466 tel.t-online.de PING tel.t-online.de (217.0.128.133) 1466(1494) bytes of data. ping: local error: Message too long, mtu=1492 ping: local error: Message too long, mtu=1492 ping: local error: Message too long, mtu=1492 ping: local error: Message too long, mtu=1492 ^C --- tel.t-online.de ping statistics --- 4 packets transmitted, 0 received, +4 errors, 100% packet loss, time 3077ms Do I have a problem? > would be interesting to make the same test from the outside towards > your asterisk with size 2 bytes larger the highest you are able to > 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 community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice broken during calls (again...)
Am 22.06.2020 um 22:12 schrieb Marek Greško:
Hi Marek
> Would you mind repeating the test with canreinvite=no set for all you
> phones and mobile phones?
All my peers have already canreinvite=no...
I only have canreinvite=yes on the SIP configuration on the Telekom part:
[pbxluca]
type=peer
defaultuser=11...@t-online.de
secret= xx
dtmfmode=rfc2833
host=tel.t-online.de
context=luca_incoming
outboundproxy=tel.t-online.de
port=5060
fromuser=035
fromdomain=tel.t-online.de
usereqphone=yes
canreinvite=yes
insecure=port,invite
nat=no
qualify=yes
qualifyfreq=600
disallow=all
allow=alaw
allow=ulaw
Should I change canreinvite=no there?
> What is your upload bitrate? Is it guaranteed?
Currently 12Mbps. Guaranteed should be about 10Mbps...
> I would try also to test the PMTU:
>
> Try:
>
> ping -M do -s 2000 ${ip address of the sip server}
>
> You should receive icmp asking for lowering the packet size.
root@bpi:/etc/asterisk# ping -M do -s 2000 tel.t-online.de
PING tel.t-online.de (217.0.128.133) 2000(2028) bytes of data.
ping: local error: Message too long, mtu=1492
ping: local error: Message too long, mtu=1492
ping: local error: Message too long, mtu=1492
ping: local error: Message too long, mtu=1492
ping: local error: Message too long, mtu=1492
ping: local error: Message too long, mtu=1492
^C
--- tel.t-online.de ping statistics ---
6 packets transmitted, 0 received, +6 errors, 100% packet loss, time 5103ms
Mmmm... it seems not good, isn't it?
For information, here the output of ifconfig:
dsl0: flags=4305 mtu 1492
inet 93.241.x.y netmask 255.255.255.255 destination 62.156.z.k
inet6 fe80::9565:3024:4deb:ebc7 prefixlen 10 scopeid 0x20
ppp txqueuelen 3 (Point-to-Point Protocol)
RX packets 852397 bytes 480197087 (457.9 MiB)
RX errors 0 dropped 0 overruns 0 frame 0
TX packets 967912 bytes 170822532 (162.9 MiB)
TX errors 0 dropped 0 overruns 0 carrier 0 collisions 0
> The LTE phones could have lower MTU and thus overcome PMTU problem.
Should I reduce the MTU?!?
Maybe I didn't understood what you mean...
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
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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...@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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice broken during calls (again...)
Am 22.06.2020 um 21:30 schrieb Michael Maier: > Did you check to prevent transcoding? could you explain what do you mean and how to check it? >> On the Gateway (Banana PI), where the Asterisk server also runs, the >> load is about 0.50 during calls and it has a Gbps LAN. > > What's running on this device on parallel? What about other network > traffic - not necessarily to the internet interface? On the BananaPI? Nothing other PPP, Bind, NTP, Firewall (iptables) and Asterisk. >> I can't believe, the problem is here... > > That's irrelevant. You have to ensure, that the driver doesn't have any > problems. Reducing the queue sizes of the interface may help. I don't understand what you mean... > - Are you using NAT or is asterisk running on the device which runs the > ppp-interface? Asterisk runs on PPP interface > - What's the modem you are using? What about the wiring between APL and > modem? Is it done correctly? [2] Zyxel VMG1312B30A. It works correctly and using the Internet (upload and download) is not a problem > - Did you configure prioritization for the up-stream regarding RTP and > SIP? This is done with the tc tool. Yes > - Did you correctly configure tos? For Deutsche Telekom you may use > tos=0xb8 (pjsip). You have to verify it with Wireshark with your traces. > You have to set it to the same value as the packages which are received > from their server. I use SIP, not PJSIP... Do I have to do that, too? Which value? > - You have to use the DNS of Deutsche Telekom which they provide during > the ppp-login because they usually provide optimal sip servers for you > (regarding distance). You're RTT of ping (18 ms) is pretty bad. I'm > having here 5 ms to the primary server (Telekom provides 3). See > > dig +noall +answer _sip._udp.tel.t-online.de SRV > > e.g. (don't know the hostname for the business infrastructure) I have a forwarding to the DNS servers of Telekom configured in my bind, since the 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 (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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice broken during calls (again...)
be a problem in my Asterisk... So the questions: 1) can someone confirm or contradict 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-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice broken during calls (again...)
Am 22.06.2020 um 17:01 schrieb Telium Technical Support: > I don't know if there was a prior email with more details, but > > Latency is as important as speed. Have you checked latency between your > device and pop? What about QoS at your location, and does your ITSP > support/respect QoS? That's a very good idea... Could you suggest me how can I check it? The Gateway is a Linux with Debian 9. > Could problem be inside your network? Have you tested/optimized internal? Really difficult to believe... If I call another VoIP-phone in my network (using 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 by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voice broken during calls (again...)
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 without vectoring. Of course, Deutsche Telekom said nothing about this change... Well, I got it working, and now I have 48Mbps down and 10Mbps up. I _REALLY CAN'T_ believe, that this is not enough... The problem with many little disruptions during calls is always here. I tried changing the codecs and changing some settings in the SIP configuration of the peers. No changes... On the Gateway (Banana PI), where the Asterisk server also runs, the load is about 0.50 during calls and it has a Gbps LAN. I can't believe, the problem is here... @all german users using Telekom: how did you configured your Asterisk? @all: thank you for all your suggestion, I really don't know 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.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice "broken" during calls
Am 17.06.2020 14:37, schrieb Karsten Wemheuer: Hi Karsten! The product is "All-IP" and not the SIP trunk, right? The call starts normally and after about 15 minutes the quality is disturbed? No, current we have Magenta Zuhause. Tomorrow we'll change to DeutschlandLAN IP (business 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 (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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 internal (phone) and the other is external (DT), doesn't matter which way round. This was what I meant... tcpdump -i dsl0 -s 0 -w /tmp/test0.pcap host tel.t-online.de & tcpdump -i phone0 -s 0 -w /tmp/test1.pcap host 192.168.200.xx (IP of my phone) & Looks like you name your Banana interfaces very similarly to mine :) I think, we are not alone... :D However, I would be careful with that first one, containing "host tel.t- online.de". I don't use DT, so I can't be sure, but I guess this is the SIP server to which you register with the account credentials... It *may not* be the same machine as handles the RTP packets - that is negotiated separately between Asterisk (or the Thomson, when it's connected directly to DT) as part of the SIP INVITE / Acknowledge. So, you *could* find that you capture all of the SIP traffic and none of the RTP traffic. On the other hand, you might get everything. You can be pretty sure it's worked if you do the above and then find that the two packet capture files are approximately the same size. If the DT one is significantly smaller (by which I mean a factor of at least ten different), then omit the "host" parameter on that capture and try again... OK, I'll check it... 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 to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice "broken" during calls
Am 15.06.2020 23:15, schrieb Jeff LaCoursiere: Hi again, just a question, to be sure... 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? Try to limit the traffic to just your phone call tests (to reduce the size of the capture files). Make all your tests, then: Well, assuming eth0 is the DSL interface and eth1 the phone interface, I can so that: tcpdump -i dsl0 -s 0 -w /tmp/test0.pcap host tel.t-online.de & tcpdump -i phone0 -s 0 -w /tmp/test1.pcap host 192.168.200.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://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice "broken" during calls
Am 15.06.2020 um 23:15 schrieb Jeff LaCoursiere: Hi > Yes, sure, please use (replace with correct interface names): > > sudo tcpdump -i eth0 -s 0 -w /tmp/test0.pcap & > sudo tcpdump -i eth1 -s 0 -w /tmp/test1.pcap & OK, I'll do it this evening (german time), 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.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice "broken" during calls
Am 15.06.2020 um 22:30 schrieb Antony Stone: >> What do you mean? In which sense this is "significant"? > > Because if/when the Thomson is registered directly to DT, then Asterisk is > not > doing anything on the Banana. Yes, I think it too >> What do you mean now? If I can use the full available band or if I can >> download exactly 50Mbs? > > I don't see the difference - I'm asking whether the Banana can manage to > route > 50Mbps of traffic. I think, it should be possible... the BananaPI has a Gbit ethernet... >> The answer to the first question is: YES! That's why I use a traffic >> shaper... ;) > > So, you're saying that your Banana Pi *can* provide the full 50Mbps > throughput > if you don't enable the traffic shaper? If I try to transfer a big file between two devices in different VLANs (through the BananaPI as Gateway), there is no problem. I don't really measured it, but it's bigger than 50Mbps... >> The answer to the second question is: NO. I made a speedtest right now >> and I get only ~18Mbps download. > > So, is it the traffic 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 suggestion / offer to capture the audio data and do an analysis > on > it would likely show that. OK! 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 to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 made a speedtest right now > and I get only ~18Mbps download. And some other information, too. I checked the xDSL-statistics of my DSL-Modem (which use the BananaPI to establish the PPPoE connection): adsl: ADSL driver and PHY status Status: Showtime Last Retrain Reason:2 Last initialization procedure status: 0 Max:Upstream rate = 1709 Kbps, Downstream rate = 19888 Kbps Bearer: 0, Upstream rate = 1626 Kbps, Downstream rate = 20113 Kbps Bearer: 1, Upstream rate = 0 Kbps, Downstream rate = 0 Kbps So it seems, that my connection is about the half of the theorical one... I think, I must call Deutsche Telekom, but since I'll change my contract at 18.06., I'll wait some days. Then I'll have a "business" contract, and I hope I don't must speak with someone that can just say "you have to reboot your Fritzbox. What? You don't have a Fritzbox? That's not possible. Please check your Fritbox, I can't reach it"... ;) Bye 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 Telekom... > > I think that is significant, even if the routing is still going through the > Banana. What do you mean? In which sense this is "significant"? >> Well, during the calls, the BananaPI has a load of max 1, and it have 2 >> cores... > > Multi-core CPUs are only a benefit if you can run separate applications (or > at > least separate threads of an application) on the separate cores. Yes, of course. > I'm not sure Asterisk can do this for a single call. But of course the BananaPI can handle that Asterisk uses a core and reserve the rest (or part of it) to manage the PPPoE connection... >> The LAN interface is Gbps, and my DSL is only 50Mbps, so it is not >> possible to get it full of band... > > Can you get the full 50Mbps through the Banana when you're doing a download > of > something biggish? 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 made a speedtest right now and I get only ~18Mbps download. >> Last but not least: I tried calls via Skype and WhatsApp (with my phone >> in my WLAN). No problem and very good quality, so the BananaPI does not >> have any problem to manage the data transfer, isn't it? > > The big difference there, though, is that Asterisk (running on the Banana) is > not handling the call, so you have the traffic being routed through the > Banana, > but Asterisk is not being asked to do anything with it in the middle. Yes, and the voice quality is excellent, but *just* if I'm *not* using DT... As soon as I use DT (configuring the phone to connect directly to the server) both partners can hear the "interruptions"... 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 to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 > have been bizarre. Yes, I really didn't believe, it could be the problem, but know is better than believe... ;) >> 2) Asterisk seems not to be the problem, too, since I have the same >> behaviour if I connect to phone directly to the server of Deutsche Telekom. > > Is that also via the Banana, or with the phone directly on a DSL modem? Always via the BananaPI I cannot connect the phone directly to a DSL modem, since the phone does not have any program to establish a PPPoE >> 4) The problem happens *only* on active call, not by voicemail. > > So, only when there are two SIP clients active on each side of the Asterisk > server... Yes, you can say it, too... I think, this is the same with other words... >> 4a) To test it I read a text and my partner just listen it, and then he >> read a text and I listen it. *No* simulaneously speak! > > But, what were the results - each of you could hear the other perfectly well? No! Maybe I didn't explained well... All the tests I done with my father in law, during that we experienced the "interruptions" were made as I described, one of us spoke, the other counts the "interruptions". > This sounds interesting - more ideas below. > >> 5) A *single call* (since I couldn't reproduce it anymore), made using >> my Android phone as SIP-client connected to my Asterisk, had not the >> problem. Any other try to call someone using my mobile phone via SIP had >> the problem. > > You seem to have the problem in general, so a single (or small number of) > instances of no problem doesn't mean there isn't something to be resolved. Yes, this is a general problem, happening using the phone of my wife, too, btw... >> I really think, the problem should be by Deutsche Telekom... > > Especially since you say you do not get the problem when you have calls in > via > Messagenet for your Italian calls. Sometimes I experienced problem with MessageNet, too, but not so frequently as with Deutsche Telekom... > What happens if: > > a) you call someone external, speak for about 30 seconds without them making > any sound, then they start speaking *at the same time as you*, then you stop > talking and they carry on. Are these 30 seconds of "silence" important? If not, it happens very frequently that both partners speak at the same time. In this case, the quality is a little bit less than normal, but very very little!, and I hear "interruptions" in this case, too... > b) exactly the same, except this time they call you, so it's an inbound call. I experienced these "interruptions" if I call even if I receive the call, if this was your question... > Do you get good quality while only one person speaks, and bad while both do? > Does the quality return to good when one person stops speaking? Actually I don't get a good quality at all, expect for some isolated calls... 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 to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice "broken" during calls
Am 15.06.2020 um 20:15 schrieb Jeff LaCoursiere: Hi Jeff, > We are working on a product to analyze pcap files of VoIP calls. So far > it does a reasonable job of analyzing the frequency distribution of > packets in both directions, pointing out which direction packet loss / > bad jitter occurs. If you can trap the traffic on the outside and the > inside of your Banana Pi and send me the pcap files, I would be happy to > run it through our analyzer as further information for you. If it shows > DTK isn't sending packets when it should, that will be obvious, and you > can send to them as solid evidence of their guilt :) Thank you for your offer. Could you say me which options I should pass to tcpdump to get all information you need? 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 Telekom... > Beyond that, are you certain you aren't taxing the Banana Pi? It really > *should* be able to handle a single call while handling your LAN's > routing/firewall tasks, but you are probably skating the edge. The > results of the above might point out that the Pi isn't *sending* packets > it should be, or sending them way late, in which case the issue is > actually your hardware. Well, during the calls, the BananaPI has a load of max 1, and it have 2 cores... The LAN interface is Gbps, and my DSL is only 50Mbps, so it is not possible to get it full of band... And during the test as I connected the phone to the Telekom servers the load of the BananaPI was lower as 1. Last but not least: I tried calls via Skype and WhatsApp (with my phone in my WLAN). No problem and very good quality, so the BananaPI does not 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 community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 Telekom (using my > network) and try to make a call Absolutly *no changes* on the behaviour compared with my Thomsons... I try to summarize: 1) Phones are not the problem, since 3 phones of 2 different companies/model have the same issue. 2) Asterisk seems not to be the problem, too, since I have the same behaviour if I connect to phone directly to the server of Deutsche Telekom. 3) Traffic shaping seems not to be the problem, too, since I tried to deactivate it. 4) The problem happens *only* on active call, not by voicemail. 4a) To test it I read a text and my partner just listen it, and then he read a text and I listen it. *No* simulaneously speak! 5) A *single call* (since I couldn't reproduce it anymore), made using my Android phone as SIP-client connected to my Asterisk, had not the problem. Any other try to call someone using my mobile phone via SIP had the problem. I could *not* test connecting to the server of Deutsche Telekom using the Internet connection of someone other, since Telekom bounds my VoIP-login to my IP. I really think, the problem should be by Deutsche Telekom... What is your opinion? Do you see some other tests I should try? 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice "broken" during calls
Am 14.06.2020 um 17:05 schrieb Antony Stone: Hi Antony, > You mean that the Thomson phone is registering to Deutsche Telekom? > > I thought it was registering to your Asterisk server. Sorry, I didn't read correctly your test 2b... Normally my Thomson phone is registering to my Asterisk server. I tried to register the Thomson phone directly to Telekom's server, to check if the problem could be in my Asterisk... > Maybe it would be a good idea to tell *exactly* what your network setup is, > because I'd certainly assumed something that's clearly not true; maybe others > here have as well. Well, I'll try: - DSL-Modem, connected to a BananaPI with Debian 9 - On the BananaPI, PPPoE to connect to the Internet, iptables and some scripts to manage the Gateway and Firewall - Many VLANs, some of them can use the Internet via NAT - The phones are in an own VLAN without any routing to the Internet (exception for my phone was temporarily made to allow the tests) - In the phone's VLAN there is the Asterisk server, running on the same BananaPI the act as Gateway/Firewall - Mobile phone connected via WLAN in the same VLAN used from the PCs, and with routing to the Internet via NAT > Basically, what SIP phones (hardware or software) are you using, what are > they > registering to, and what role is Asterisk playing in all of this? How do > calls to/from the public phone network get routed from/to your telephones? We have two phones Thomson ST2022, registering to the Asterisk server. The Asterisk server registers to Deutsche Telekom and MessageNet. All calls are normally routed by Asterisk to Deutsche Telekom. Some *incoming* calls to an italian number arrives via MessageNet and will be directed to my Thomson phone. What I tried connecting the phone directly to the Internet and the servers of Deutsche Telekom was just a test, not the normal situation. Do I have explained my current situation? Of course I can send extract of configurations, if needed... 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 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 by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice "broken" during calls
Am 14.06.2020 um 16:48 schrieb Michael Keuter: Hi Michael, > the standard Deutsche Telekom SIP-account (former ISDN Mehrgeräteanschluß > PTMP with 3-10 numbers) is always tied to your DSL account. I supposed it... > There is a special "DeutschlandLAN SIP-Trunk Pure" where 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 (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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice "broken" during calls
Am 13.06.2020 um 22:56 schrieb Antony Stone: Hi again, > 2b. Take your Thomson telephone to some other location with Internet access, > let it register to your home Asterisk server, and them make a call to the > same > number yet again. I'm sure you can get the Thomson to connect to Asterisk > via > some external network, since you say you can do this from your Android phone. > > Again, check the call quality. I tried it on the network of a friend. Not possible to establish a connection at all... I *suppose* Deutsche Telekom just allow a logon on their servers from the IP of the user, who tries to log on (with other words: my VoIP login can just log on from my current IP)... This would explain why I didn't got my mobile phone connecting to the Telekom's server and establish a call... I also tried to stop Asterisk and all other network services on my Linux-Box Firewall/Gateway, including the traffic shaper (in the case, this was the problem), then connect my Thomson phone to the Telekom's server and call my father in law. Always the same problem... So, tomorrow I'll get another VoIP phone from a colleque (Elmeg IP 290). I'll connect it to my network and my Asterisk and will try to call my father in law for a test. I really do *not* expect any change in the situation... I think, the problem should be somewhere by Deutsche Telekom... What is your opinion? Btw: I did all tests with my father in law, since 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 Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice "broken" during calls
Am 14.06.2020 um 10:49 schrieb Antony Stone: >> So, the module voicemail in Asterisk does *not* have the same problems >> as the other phones. >> And the Thomson VoIP-phone has more problems than my Android connected >> to the Asterisk... > > So, you don't get "consistently good quality" in any situation, but it seems > like the Thomson phone being involved makes things worse. Correct >>> what happens if you let your Android phone connect via LTE to your home >>> Asterisk server and you dial your (home, cabled) Thomson 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... > > And is 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 remember why I added it... I try to remove it now and call my father in law. No changes... 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 to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice "broken" during calls
Am 13.06.2020 um 22:56 schrieb Antony Stone: Hi Antony, > I would like to see a much simpler one-for-one comparison: only change one > thing at a time, and see what the difference is. > > So: I suggest you try *two* independent *pairs* of tests: OK > 1a. Using your Android phone, connect using your home wireless network (I > assume you have a wireless network, if not then skip to test 2) to your home > Asterisk server, make a phone call to some external number, check the call > quality. Today was the quality not so excellent as yesterday... Both I and my father in law could hear "interruptions", but so so much as if I call with the Thomson phone... > 1b. Using your Thomson phone, connected using your home cabled network to > your > home Asterisk server, make a phone call to the same external number and check > the call quality. This was the same as always... More little "broken voice" on both parts... > 2a. Using your Android phone, connect from outside your home wireless network > over LTE to your home Asterisk server and make a phone call to the same > number > again (you'll need someone with a bit of patience and understanding on the > other end of this number ...) Check the call quality. It was a little bit better. I didn't hear any "interruptions", but my father in law does. Not many, but somes... > 2b. Take your Thomson telephone to some other location with Internet access, > let it register to your home Asterisk server, and them make a call to the > same > number yet again. I'm sure you can get the Thomson to connect to Asterisk > via > some external network, since you say you can do this from your Android phone. > > Again, check the call quality. Right now I don't have the possibility to do that... :( I did another test, today: I called my leased line number using my mobile phone (over GSM, not VoIP) and wait for the answering maschine. So, as a normal call from outside if I'm not at home. Result: the quality is *excellent*. I didn't hear any "interruptions" in the message of the answering maschine and, as I played the message I spoked there were no "interruptions", too... So, the module voicemail in Asterisk does *not* have the same problems as the other phones. And the Thomson VoIP-phone has more problems than my Android connected to the Asterisk... Maybe helps this behaviour to find the problem? > PS; Just for fun, what happens if you let your Android phone connect via LTE > to your home Asterisk server and you dial your (home, cabled) Thomson 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 18:25:32, Luca Bertoncello wrote: > >> 2) where can I change these settings? > > sip.conf > > Look for lines such as > > disallow=all > allow=ulaw > allow=alaw > allow=h263 > > They may be in the [general] section, or they may be in the client (Android / > Thomson) specific sections. In my [general] section I have: disallow=all allow=alaw allow=ulaw allow=ilbc allow=g729 allow=g723 allow=gsm ; Messagenet need gsm... 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 to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice "broken" during calls
Am 13.06.2020 um 22:09 schrieb Antony Stone: Hi Antony > You are *assuming* that it's the codec causing the difference. Well, I really don't know what I can think, now... > We don't know that. > > Let me get this clear, to make sure I understand (differences emphasised): > > 1. You use *a VoIP softphone app* on your mobile, which is registered by SIP, > to your Asterisk server, over your home *wireless network*, to place a call > to > some external number, you have a conversation and *the quality is excellent*. > > 2. You use your *Thomson ST2022*, which is also registered by SIP, to your > home Asterisk server, over your home *cabled* network, to place a call to > some > (the same???) external number, you have a conversation and the quality is > *not > excellent*. > > > Is that an accurate summary of your situation? Not really... 1) I have an Android phone, using the integrated Android VoIP-subsystem, connected to my Asterisk at home, over LTE or other network *outside my home network*. Today I called my mother using this method (I was in the home network of my parents in law, about 20km von my home network, so definitly *not* in my wireless...). The quality was excellent and it was confirmed from my father in law, too... 2) I have a Thomson ST2022 connected to my Asterisk over Ethernet (cabled network). If I call for example my mother or my parents in law, the conversation is "broken", eg: both partner can hear little "interruption", about 1/10 seconds in the conversation... This is the situation... I tried to connect the Thomson ST2022 directly to the server of Deutsche Telekom via VoIP (excluding the Asterisk, but of couse using NAT, since the phone does not have a public IP but just an IP in my internal network) and then I called my father in law. Same problem... :( I didn't get my Android phone connected to the server of Deutsche Telekom to check how it works *outside my home network*... Not sure why it doesn't work... Some other information: 1) Asterisk runs on a Linux-Box (on a BananaPI) with Debian 10. Asterisk was installed from Debian repositories. 2) The Linux-Box is directly connected to the Internet (no NAT) with a DSL-Modem and PPPoE. Public IPv4 and IPv6 addresses are configured in a network interface of the Linux-Box. 3) I use iptables+tc to manage a traffic shaping, privileging the VoIP connection. If you want, I have no problem to send the traffic-shaping-script to the list. 4) The DSL connection has a speed of 50Mbps down and 10Mbps up, and I really think, it should be enough... 5) The phones are connected with Gbps-Ethernet to the Linux-Box. 6) On my Asterisk I configured a second VoIP-Provider (MessageNet, in Italy), but just to *receive* calls. My contract with MessageNet does not allow me the call someone using this connection. If someone calls my number by MessageNet, I have the same problem I have with Deutsche Telekom, altought not so strong, eg. the "interruptions" are not so frequent as by calls via Deutsche Telekom... Btw: by MessageNet I must use *gsm* as Codec, otherwise a connection will be extablished, but no Voice can be heared... I really appreciate any idea. Of course, 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) -- _ -- 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice "broken" during calls
Am 13.06.2020 um 18:06 schrieb Michael Keuter: > So the call used Alaw as Codec. Yes, so seems it to be... It should has the better quality... But the calls done using my mobile phone in VoIP with the Asterisk have better quality as the calls done using the normal VoIP-telefon... I'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 Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice "broken" during calls
Am 13.06.2020 um 18:20 schrieb Antony Stone: Hi >> bpi*CLI> sip show peer 0049177xxx >> Codecs : >> (alaw|ulaw|ilbc|g729|g723|gsm|amr|amrwb|g726|g726aal2|adpcm|slin|slin|slin| >> slin|slin|slin|slin|slin|slin|lpc10|speex|speex|speex|g722|siren7|siren14|t >> estlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|red|t140|silk|silk >> |silk|silk) > > That strikes me as somewhat unlikely. Too much things, isn't it? >> bpi*CLI> sip show peer 0049351xxx >> Codecs : (alaw|ulaw|ilbc|g729|g723|gsm) > > 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) -- _ -- 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice "broken" during calls
SIP User agent: Username: 00493501xxx Peername: pbxluca Original uri: sip:sg...@217.0.27.xx Need Destroy: No Last Message: Tx: 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 that I should check/change? Thank you very very much 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 check the codec used by my mobile phone and the codec used by the phones at home? 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 to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 understand what my > partner sayd... > > It can NOT be a problem of other downloads/uploads, since in that moment > there were no ones... Hi again! Just a detail: I tried an internal call (from my phone, to my wife's phone) and it works wonderful, no broken, no delay, top quality. So the problem _MUST_ be in the settings of the communication with Deutsche Telekom and MessageNet (the providers I used). The settings for Deutsche Telekom are: [pbxluca] type=peer defaultuser=-0001 secret= dtmfmode=rfc2833 host=tel.t-online.de context=luca_incoming outboundproxy=tel.t-online.de port=5060 fromuser=0351xxx fromdomain=tel.t-online.de usereqphone=yes canreinvite=yes insecure=port,invite nat=force_rport,comedia qualify=yes qualifyfreq=600 disallow=all allow=alaw allow=ulaw and the settings for MessageNet are: [messagenet] type=peer defaultuser= secret= dtmfmode=rfc2833 host=sip.messagenet.it context=messagenet_incoming outboundproxy=sip.messagenet.it port=5060 fromuser= fromdomain=sip.messagenet.it usereqphone=yes canreinvite=yes insecure=invite qualify=yes qualifyfreq=60 disallow=all allow=alaw allow=ulaw allow=gsm Any idea? 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voice "broken" during calls
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 understand what my partner sayd... It can NOT be a problem of other downloads/uploads, since in that moment there were no ones... I already had the problem in the past, solved it enabling the jitter, but the problem with jitter enabled was a long delay (1-2 seconds) in the communication, so I disabled it. Can someone suggest me what can I do? I can send extract of my configuration 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 Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delay on speak with Asterisk
Am 04.12.2019 um 11:14 schrieb Antony Stone: > Hm, I was judging based on what you posted previously: > > Our Codec Capability: (alaw) > Their Codec Capability: (ulaw|gsm|alaw|amr) > Joint Codec Capability: (alaw) > > which suggested to me that if you offered GSM, that could be agreed with the > other side. I tried again right now: disallow=all allow=alaw allow=gsm If I call my phone I can see, alaw is used. If I allow just gsm I get the error: [Dec 4 11:23:17] NOTICE[14060][C-012e]: chan_sip.c:10798 process_sdp: No compatible codecs, not accepting 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.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delay on speak with Asterisk
Am 04.12.2019 um 10:53 schrieb Antony Stone: Hi Antony! > 1. Try using codec GSM (which is pretty good quality but lower bandwidth than > alaw, which is currently the only one you are offering). gsm seems to be unsupported from Deutsche Telekom... Already tried, it does not work... :( > 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...@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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 deactivated it, I have no delay anymore. Unfortunately is the audio quality now a little bad than with the jitter buffer... Any suggestion how can I improve the audio quality without add the delay? 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 to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delay on speak with Asterisk
Am 03.12.2019 um 19:57 schrieb Antony Stone: Hi Antony, thank you for your answer. > I would firstly look at whether your Asterisk box is doing transcoding - > converting from oe codec (supported by your phones) and another codec > (supported by the provider) because no codec can be found in common between > the two. > > Secondly I would put a full packet sniffer (by which I mean collect all the > RTP > data as well as SIP) on each of your interfaces (internal and external) to > see > whether the delay really is happening inside your Asterisk server - if you > see > RTP data on your internal interface, then appearing 1-1.5 seconds later on > the > external interface, and vice versa, then you know the delay is inside your > system. I'm really not an expert on Asterisk... Could you please say me HOW can I check the codecs? I tried to get the information of the channel: bpi*CLI> sip show channel p65551t1575398506m6025c4749452s2 * SIP Call Curr. trans. direction: Incoming Call-ID:p65551t1575398506m6025c4749452s2 Owner channel ID: SIP/pbxanika-021e Our Codec Capability: (alaw) Non-Codec Capability (DTMF): 1 Their Codec Capability: (ulaw|gsm|alaw|amr) Joint Codec Capability: (alaw) Format: (alaw) T.38 supportNo Video support No MaxCallBR: 384 kbps Theoretical Address:217.x.x.x:5060 Received Address: 217.x.x.x:5060 SIP Transfer mode: open Force rport:Auto (No) Audio IP: 217.y.y.y (local) Our Tag:as45e11359 Their Tag: h7g4Esbg_p65551t1575398506m6025c4749452s1_206873930-910452977 SIP User agent: Username: 550293777072-0001 Peername: pbxanika Original uri: sip:sgc_c@217.x.y.z Caller-ID: +49177 Need Destroy: No Last Message: Rx: ACK Promiscuous Redir: No Route: DTMF Mode: rfc2833 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 Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Delay on speak with Asterisk
Hi list! I'm using Asterisk 13.14.1 from Debian 9 repositories. The provider is Deutsche Telekom und Messagenet (just for receive). I can call and receive calls, but I have a little problem: there is a "delay" of about 1-1,5 seconds between the time the voice is sent and the time when the voice is received, so that it happens very often that the peer does not get my voice and try to repeat the question, then it get my voice, and breaks the question, and so on... This delay happens on every peer, Deutsche Telekom and Messagenet, so I think the problem is NOT by the Provider, 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 community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] High delay and some echo
Am 11.06.2019 um 21:28 schrieb Antony Stone: Hi, > Well, my starting point, given the hardware setup you've confirmed above, > would > be to plug an analogue phone into your FritzBox (assuming that's what DT gave > you) and see whether the problem exists without Asterisk in the picture at > all. I don't have any FritzBox. I have a little BananaPI with Debian 9 configured as Router and connected to a DSL-Modem. On the BananaPI I installed Asterisk, to have it directly connected to the Internet. > The second thing I would try is to put the SIP credentials given to you by DT > into the SIP phone itself (most can support at least two lines, so you don't > need to over-write the credentials for your Asterisk server account) and > again, soo whether the problem persists with the Asterisk server removed from > the signal path. I can try it... Now it's too late for the test. I'll try tomorrow. > That will at least tell you whether Asterisk is causing the problem, because > if it isn't: > > a) there isn't much you can do about it except report it to DT, and Bwahahahahahah The technician of DT, at least the people answering the Hotline, don't have any idea _WHAT_ is VoIP and so on... They only can say "you have to power off your FritzBox, wait 30 seconds and power it on again". If I say, that I don't have any FritzBox they give a Brain core dumped... > b) there's very little the good people here on this mailing list will be able > to help you with. Really a pity... :( > Just out of interest, what hardware are you running Asterisk on? It's > unlikely to be the cause of the problem, because I've run it on Raspberry > Pies > 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) -- _ -- 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] High delay and some echo
Am 11.06.2019 um 21:10 schrieb Antony Stone: Hi, > So, you have a SIP phone, connected to an Asterisk server on your local > network, which then connects to D Telekom's SIP server over the DSL line? Correct! >> The other party use VoIP, too, since they are in Germany (and Italy) and >> here there are just VoIP... Sigh! > > Are they also using a SIP phone? My mother yes, my father in law uses an ISDN phone connected to a FritzBox that convert the signal in VoIP. > Do they also have an Asterisk server on their local network? > >> 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? > > Have you considered trying some tool such as http://sipcapture.org/#about to > see 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) -- _ -- 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] High delay and some echo
Am 11.06.2019 um 20:42 schrieb Antony Stone: Hi Antony, > I think the main question here is: how are you connecting Asterisk to the > telephone system? Via VoIP... > You mention that you're on DSL from Deutsche Telekom, but is the call going > over this DSL link to soem SIP provider, who then connects you to the PSTN, > or > are you connecting Asterisk locally to the phone line via some ATA device? Deutsche Telekom uses since years just VoIP. No ISDN, PSTN, and so on... :( I'm connecting to the VoIP-Server of Deutsche Telekom via DSL (50Mbps down, 10Mbps 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? 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 to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] High delay and some echo
Hi list! I use Asterisk 13.14.1 from Debian repository on a DSL from Deutsche Telekom. Asterisk works well, but I have really often an high delay (I understand it since the other party speak some seconds before he hears my question and answer) and sometimes I hear an echo. I really don't 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...@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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unable to use VoIP-device
Hi list!
Today I replaced my old Asterisk 1.8.30.0 on a OpenWRT switch with Asterisk
13.14.1 running on a Banana PI.
Well, I'm trying since hours to connect my mobile phone to the new Asterisk,
but I can't...
I can register it, and I can see it with "sip show peers".
If I try to call my mobile phone from my desk VoIP-phone I see that:
[Feb 17 19:54:10] NOTICE[15630]: chan_sip.c:24457 handle_response_peerpoke:
Peer '00491777654321' is now Reachable. (16ms / 2000ms)
== Using SIP RTP CoS mark 5
> 0xb44ce478 -- Strict RTP learning after remote address set to:
192.168.200.10:41000
-- Executing [4@default:1] Dial("SIP/00493511234567-001a",
"local/4@luca_mobile") in new stack
-- Called local/4@luca_mobile
-- Executing [4@luca_mobile:1] NoOp("Local/4@luca_mobile-000f;2", "")
in new stack
-- Executing [4@luca_mobile:2] Verbose("Local/4@luca_mobile-000f;2",
"2,Call for Mobile - [00493511234567]") in new stack
== Call for Mobile - [00493511234567]
-- Executing [4@luca_mobile:3] Set("Local/4@luca_mobile-000f;2",
"available=NOT_INUSE") in new stack
-- Executing [4@luca_mobile:4] GotoIf("Local/4@luca_mobile-000f;2",
"0?unavailable") in new stack
-- Executing [4@luca_mobile:5] Dial("Local/4@luca_mobile-000f;2",
"SIP/00491777654321,,RxX") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/00491777654321
-- Local/4@luca_mobile-000f;1 is ringing
-- SIP/00491777654321-001b is ringing
-- Local/4@luca_mobile-000f;1 is ringing
== Spawn extension (default, 4, 1) exited non-zero on
'SIP/00493511234567-001a'
== Spawn extension (luca_mobile, 4, 5) exited non-zero on
'Local/4@luca_mobile-000f;2'
but my mobile phone does NOT ring...
If I try to call another VoIP-phone in my Asterisk from my desk-phone it
works:
== Using SIP RTP CoS mark 5
> 0xb44ce478 -- Strict RTP learning after remote address set to:
192.168.200.10:41000
-- Executing [2@default:1] Dial("SIP/00493511234567-0022",
"local/2@anika_voip") in new stack
-- Called local/2@anika_voip
-- Executing [2@anika_voip:1] Verbose("Local/2@anika_voip-0013;2",
"2,Call for Anika - [00493511234567]") in new stack
== Call for Anika - [00493511234567]
-- Executing [2@anika_voip:2] Dial("Local/2@anika_voip-0013;2",
"SIP/00493511234765,20,RxX") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/00493511234765
-- Local/2@anika_voip-0013;1 is ringing
-- SIP/00493511234765-0023 is ringing
-- Local/2@anika_voip-0013;1 is ringing
== Spawn extension (default, 2, 1) exited non-zero on
'SIP/00493511234567-0022'
== Spawn extension (anika_voip, 2, 2) exited non-zero on
'Local/2@anika_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.
And, as I said, the mobile phone CAN register in Asterisk...
Can someone help me to understand WHY using my mobile phone on my new
Asterisk doesn't work?
And, of course, to solve my problem... ;)
Thank you very very much for your help!
Luca Bertoncello
(lucab...@lucabert.de)
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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 (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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with DAHDI
Tzafrir Cohen <tzafrir.co...@xorcom.com> schrieb:
> This means that you have configured a dahdi channel in
> /etc/asterisk/chan_dahdi.conf . The default configuration does not
> include one. Do you have any DAHDI device on the system?
I think not...
> If /dev/dahdi/channel itself does not exist, it means that the
> kernel-level support is not loaded (or not even configured). It is
> generally from the kernel module dahdi, which is an out-of-tree one.
>
> If you really want it, you may need to run:
>
> m-a a-i dahdi
>
> But do you really have a DAHDI device?
No, I haven't...
I remove the configuration
dahdichan = 1
for every user and I don't have any error anymore.
But if I try to call another VoIP-phone it rings but no voice will be
transferred...
Any idea?
I tryed to enable the debugging and I see that:
> 0xaa90a380 -- Strict RTP learning after remote address set to:
192.168.200.10:41000
-- Executing [00493517654321@default:1] Dial("SIP/00493511234567-",
"SIP/00493517654321") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/00493517654321
-- SIP/00493517654321-0001 is ringing
> 0xb4606f38 -- Strict RTP learning after remote address set to:
192.168.200.11:41000
-- SIP/00493517654321-0001 answered SIP/00493511234567-
-- Channel SIP/00493517654321-0001 joined 'simple_bridge' basic-bridge
<0ef9a447-b1b3-45af-a4af-7c4ac4d10546>
-- Channel SIP/00493511234567- joined 'simple_bridge' basic-bridge
<0ef9a447-b1b3-45af-a4af-7c4ac4d10546>
-- Channel SIP/00493517654321-0001 left 'simple_bridge' basic-bridge
<0ef9a447-b1b3-45af-a4af-7c4ac4d10546>
-- Channel SIP/00493511234567- 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)
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[asterisk-users] Problem with DAHDI
Hi again! I tried to attach two VoIP-phones to my new Asterisk 13.14.1 on a Banana PI with Armbian/Debian 9. First test was to call a test service that say the time. Works! Second test was to record my voice and play it again. Works! Third test was to call the other VoIP-phone. It does NOT work... :( Then I noticed that, by starting, Asterisk says the following messages: [Feb 15 18:42:54] NOTICE[3428]: loader.c:1446 load_modules: 319 modules will be loaded. [Feb 15 18:43:01] NOTICE[3428]: res_odbc.c:1089 load_module: res_odbc loaded. [Feb 15 18:43:12] WARNING[3428]: res_phoneprov.c:1231 get_defaults: Unable to find a valid server address or name. [Feb 15 18:43:15] NOTICE[3428]: pbx_lua.c:1640 load_or_reload_lua_stuff: Lua PBX Switch loaded. SIP channel loading... [Feb 15 18:43:24] NOTICE[3428]: chan_skinny.c:8429 config_load: Configuring skinny from skinny.conf [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:4126 dahdi_open: Unable to open '/dev/dahdi/channel': No such file or directory [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:12110 mkintf: Unable to open channel 1: No such file or directory here = 0, tmp->channel = 1, channel = 1 [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:17518 build_channels: Unable to register channel '1' [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:19041 process_dahdi: Dahdichan '1' failure ignored: ignore_failed_channels. [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:4126 dahdi_open: Unable to open '/dev/dahdi/channel': No such file or directory [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:12110 mkintf: Unable to open channel 1: No such file or directory here = 0, tmp->channel = 1, channel = 1 [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:17518 build_channels: Unable to register channel '1' [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:19041 process_dahdi: Dahdichan '1' failure ignored: ignore_failed_channels. [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:4126 dahdi_open: Unable to open '/dev/dahdi/channel': No such file or directory [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:12110 mkintf: Unable to open channel 1: No such file or directory here = 0, tmp->channel = 1, channel = 1 [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:17518 build_channels: Unable to register channel '1' [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:19041 process_dahdi: Dahdichan '1' failure ignored: ignore_failed_channels. [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:4126 dahdi_open: Unable to open '/dev/dahdi/channel': No such file or directory [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:12110 mkintf: Unable to open channel 1: No such file or directory here = 0, tmp->channel = 1, channel = 1 [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:17518 build_channels: Unable to register channel '1' [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:19041 process_dahdi: Dahdichan '1' failure ignored: ignore_failed_channels. [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:4126 dahdi_open: Unable to open '/dev/dahdi/channel': No such file or directory [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:12110 mkintf: Unable to open channel 1: No such file or directory here = 0, tmp->channel = 1, channel = 1 [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:17518 build_channels: Unable to register channel '1' [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:19041 process_dahdi: Dahdichan '1' failure ignored: ignore_failed_channels. [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:4126 dahdi_open: Unable to open '/dev/dahdi/channel': No such file or directory [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:12110 mkintf: Unable to open channel 1: No such file or directory here = 0, tmp->channel = 1, channel = 1 [Feb 15 18:43:24] ERROR[3428]: chan_dahdi.c:17518 build_channels: Unable to register channel '1' [Feb 15 18:43:24] WARNING[3428]: chan_dahdi.c:19041 process_dahdi: Dahdichan '1' failure ignored: ignore_failed_channels. [Feb 15 18:43:28] NOTICE[3428]: confbridge/conf_config_parser.c:2094 verify_default_profiles: Adding default_menu menu to app_confbridge [Feb 15 18:43:28] NOTICE[3428]: cel_tds.c:452 tds_load_module: cel_tds has no global category, nothing to configure. [Feb 15 18:43:28] WARNING[3428]: cel_tds.c:557 load_module: cel_tds module had config problems; declining load [Feb 15 18:43:28] NOTICE[3428]: cel_custom.c:97 load_config: 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/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! Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check ou
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...@lucabert.de)
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Re: [asterisk-users] chan_oss.c: Unable to register channel type 'OSS'
Zitat von Tzafrir Cohen <tzafrir.co...@xorcom.com>: Hi, Off-topic: any reason you don't use chan_alsa? This was the "Armbian installation", I didn't configured it extra... Are you sure you quote the error message right? Copy+Paste... ;) But I searched a little bit and 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 (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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_oss.c: Unable to register channel type 'OSS'
Hi list! Currently I use Asterisk 1.8.30.0 on an OpenWRT-Switch. Now I want to change to Asterisk 13.14.1 on a Banana PI (with Armbian/Debian 9). Well, I copied the configuration and changed what needed, so basically, it works, at least with my tests. But when Asterisk will be started, in the message log I get this error: [Feb 15 08:40:15] ERROR[3971] chan_oss.c: Unable to register channel type 'OSS' Unfortunately I cannot find WHY Asterisk was unable to register a channel type "OSS". And then: do I need this? On the old Asterisk I didn't had that... 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/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Remote Phonebook with Thomson ST2022
Hi list! I'm a little bit OT, I know, but maybe someone here can help me... I have a Thomson ST2022 and I'd like to have the phonebook on a Server, synchronized with my DAV phonebook. I created a phonebook.xml so: Test number 12345678 Then I set http://www.myserver.de/phonebook.xml in the configuration for the phonebook. As soon as I search an entry, the phone crashes... :( In the Apache-Log I see very curiously this: 192.168.200.10 - - [25/Oct/2017:19:38:40 +0200] "GET http://www.myserver.de/phonebook.xml HTTP/1.1" 200 36611 instead of: 192.168.200.10 - - [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-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem using Android SIP-Client
Harel Cohen <ha...@mayorcom.com> schrieb: > Is the Sophos a home router or professional one? In many cases what home Of course the professional firewalls (we have two Sophos in Cluster, to manage our two SDSLs) > router does by default needs to be configured manually on professional one. > E.G. a home router will allow outgoing sessions and create a return path by > default where professional one won't. > Two things I would look for: > 1. Look for, and disable, ALG for SIP. The idea of ALG is nice but I > haven't encountered a device that implements this properly (I'm not a > network expert so it doesn't mean that there isn't such a router out there). > 2. On the Sophos try to statically open the UDP port range of your RTP to > outgoing traffic from your phone to your SIP server. Note that outgoing > port range is what your SIP server defines 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... 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 to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 > incompatible voice frame on SIP/messagenet-028e of format gsm > since our native format has changed to 0x8 (alaw) > > and I can't hear anything... I tried to call the same number I called before using LTE instead of WLAN, and it worked... Then I tried to call the same number again using my WLAN at home, and it worked again. So, I must conclude that the problem is somewhere in the WLAN at office... Very curiously I can initiate the SIP-communication, but as soon as the other party answer the connection will be closed... Since I'm one of the admins at office, I'd like to solve this problem. Can someone give me some advice what can be wrong in our firewall (Sophos)? 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem using Android SIP-Client
Hi list! I use Asterisk 1.8.30.0 on an OpenWRT-Router (I know, not the last version, but I can't upgrade). It always runned very well, and it runs very well with our home phones, too, but now I have problems using the native Android SIP-Client... 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/messagenet-028e of format gsm since our native format has changed to 0x8 (alaw) and I can't hear anything... This is the configuration of the user: [00491771234567] fullname = 00491771234567 secret = MYVERYSECRET dahdichan = 1 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = default host = dynamic dtmfmode=rfc2833 canreinvite=no sendrpid=pai type=friend ;nat=force_rport,comedia nat=yes qualify=yes qualifyfreq=60 ;transport=Auto avpf=no force_avp=no icesupport=no encryption=no callgroup=1 pickupgroup=1 dial=SIP/00491771234567 allow = all Any idea? The user worked very well with my old mobile phone (Android 4), I __THINK__ the problem happens since I 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.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail: search for name in a phonebook
John Kiniston <johnkinis...@gmail.com> schrieb: > Yes, You could do easily this either with the internal asterisk database or > with something like func_odbc as a source for the data. > > In the context you receive your incoming calls you do a lookup against one > of the above 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 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail: search for name in a phonebook
Hi list!
I'm using Asterisk 1.8.30.0 on a OpenWRT device and it works perfectly.
I configured a voicemail and I receive an E-Mail with some information about
the call.
Again, wonderful!
Now my wish: I'd like to have Asterisk to search the caller in a list file
and send me the name corresponding 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)
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Re: [asterisk-users] Need to restart Asterisk if remote server not working?
Max Grobecker <max.grobec...@ml.grobecker.info> schrieb: > > Maybe should I configure a forwarder for the zone t-online.de? It not > > difficult, and if you mean it can help, I'll do that... > > In the meantime, I setup forwarding requests to "t-online.de" and > "t-ipnet.de" to the address 194.25.2.129. That is kind of a global DNS > resolver for all customers and is working since the 90s without address > changes. OK, I'll set it up now... ;) > > Could you say me how can I disable the SRV lookups? > > I use Asterisk 1.8.30.0 on an OpenWRT device. > > In your sip.conf, simply add > > srvlookup = no > > To your DTAG peer configuration. > If set globally, you may break the ability to directly call SIP addresses. I see now, that I already have this globally set... > Have you rebooted the whole 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 OpenPGP -- _ -- 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need to restart Asterisk if remote server not working?
Max Grobecker <max.grobec...@ml.grobecker.info> schrieb: Hello Max, > I'm also a customer of the DTAG. > Yesterday, the messed a bit with their DNS entries... Maybe they tried again to repair a working system... :) > If you are NOT using their DNS resolvers you got a "wrong" IP address back > that was not working. Besides that, you should disable SRV lookups for > their SIP peers. Since Asterisk's chan_sip.c does not honour the weight of > the SRV entries, nor it failovers to the other records, you might just end > up with a not working server. PJSIP might work with that, but it depends on > your version. I'm running an own BIND on my Linux-PC... Maybe should I configure a forwarder for the zone t-online.de? It not difficult, and if you mean it can help, I'll do that... > ims001 should be the preferred one based on the SRV weight. But Asterisk > only looks at the first record that comes as an answer, so if ims002 is at > the first position it will be used for registration, regardless that the > other record is weighted better. And if that one is not answering... So: > Better disable SRV lookups if you are not sure if your 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 my OpenWRT unfortunately does not support dnsmgr... Thanks Luca Bertoncello (lucab...@lucabert.de) pgpnFFa7pU8Hg.pgp Description: Digitale Signatur von OpenPGP -- _ -- 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need to restart Asterisk if remote server not working?
Antony Stone <antony.st...@asterisk.open.source.it> schrieb: > What was Asterisk doing until you restarted it? What happened when it > tried to use the (stale, but now restored) connection? Just saying, that it was not possible to connect to the remote server... [May 6 07:26:50] NOTICE[3376] chan_sip.c:-- Registration for '03514977290@pbxluca' timed out, trying again (Attempt #1769) > > 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? > > 1. How is your Asterisk server connecting to Deutsche Telekom (SIP, IAX2, > other...)? SIP > 2. How do you authenticate on that connection (password, certificate, IP > address...)? Password > 3. Do you connect to an IP address at Telekom, or to a hostname? Hostname: tel.t-online.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) -- _ -- 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need to restart Asterisk if remote server not working?
Hi list! Yesterday Deutsche Telekom had a really big problem and Asterisk couldn't connect to the remote Server (by Telekom) until today about 7:30. Well, it could happen... What I find really annoying was that I needed to restart Asterisk as I checked with sipsak that the Telekom-Server works... 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 and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [OT] Suggestion for VoIP-App
Hi list! I think many people here connect their mobile phone with Asterisk... Can someone suggest me an App that allow me to add a VoIP-number in the contact? With my old Samsung Galaxy S2 it was possible direct in system without additional Apps, but with the Galaxy S5 I can't do that and any 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 out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 help of Michael Maier, I found the problem... I write here the description of the problem and my solution, maybe can this help someone other having the same problem... The problem: after a successfully INVITE with the complete list of all supported Codecs, I receive about 15 minutes after call start, another INVITE (re-INVITE) from Telekom with __JUST__ one Codec: the one used by the call (currently: alaw). My Asterisk sends an "200 OK" with the same Codec and Telekom apparently has a problem with my answer, since the connection will be closed... __MY__ solution: I configured Asterisk to use just __ONE__ Codec (alaw) for the communication with Deutsche Telekom. Now it seems to work, then I can call Italy and can speak longer than 15 minutes. I'm really puzzled and can't understand why Telekom has problem with my answer __JUST__ on calls outside Germany, but that is... So, if someone other has the same problem, can try with my solution. Hope to help! 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.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
