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
On Friday 31 December 2021 at 15:54:01, Luca Bertoncello wrote: > 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... > Now the very question is how to remove this header... Check the Dial() command which places the call to the phone. Does it contain the "c" option? https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Dial Antony. -- In science, one tries to tell people in such a way as to be understood by everyone something that no-one ever knew before. In poetry, it is the exact opposite. - Paul Dirac Please reply to the list; please *don't* CC me. -- _ -- 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 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
On Friday 31 December 2021 at 12:29:19, Luca Bertoncello wrote: > 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. Glad you found a solution to this. > 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". See https://datatracker.ietf.org/doc/html/rfc3326#section-2 - the first example at the top of page 3 shows the sort of thing I mean. 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. Antony. -- Pavlov is in the pub enjoying a pint. The barman rings for last orders, and Pavlov jumps up exclaiming "Damn! I forgot to feed the dog!" Please reply to the list; please *don't* CC me. -- _ -- 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: 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
On Tuesday 28 December 2021 at 20:39:37, Luca Bertoncello wrote: > 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=z9hG4bKPj43be > 873a-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 So, no PAID header there, and no mention of "Sekretariat" either. > 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... Indeed. > Maybe help it to identify the problem? 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. However, at least you've got as far as ruling out Telekom as being the source of the problem, which I think is good. Antony. -- I bought a book about anti-gravity. The reviews say you can't put it down. Please reply to the list; please *don't* CC me. -- _ -- 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
On Tuesday 28 December 2021 at 20:07:22, Luca Bertoncello wrote: > Am 28.12.2021 um 20:00 schrieb Antony Stone: > > > > 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? No, you want to look at the "180 Ringing" response in both cases - what goes in to Asterisk, and what comes out of it. The INVITE does not contain the data which gets displayed on the calling telephone. > OK, so I have to sniff the data to Deutsche Telekom and not the internal > network... No, data FROM Deutsche Telekom. They are the ones sending the "180 Ringing" back to you once they think the external telephone is ringing. > 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. I don't quite follow that, but what I am saying is that the callEE's number (the number of the telephone you are calling) may be contained in an RPID header of the "180 Ringing" packet which comes back from the telephone being called. You want to find out whether this header, and the strange number you are not expecting, exists in the packets coming from your upstream provider IN to Asterisk, and also whether it exists in the packet coming FROM asterisk to your internal telephone which made the call, and is showing the strange data. I hope that is clear. Antony. -- These clients are often infected by viruses or other malware and need to be fixed. If not, the user at that client needs to be fixed... - Henrik Nordstrom, on Squid users' mailing list Please reply to the list; please *don't* CC me. -- _ -- 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
On Tuesday 28 December 2021 at 19:52:46, Luca Bertoncello wrote: > Am 28.12.2021 um 19:41 schrieb Antony Stone: > > Hi Antony, > > > Okay, so, returning to my question, do you 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)? 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. 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. > If so, then I have to check again, since I only sniffed the internal > traffic... 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). Antony. -- "It would appear we have reached the limits of what it is possible to achieve with computer technology, although one should be careful with such statements; they tend to sound pretty silly in five years." - John von Neumann (1949) Please reply to the list; please *don't* CC me. -- _ -- 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
On Tuesday 28 December 2021 at 18:17:00, Luca Bertoncello wrote: > Am 28.12.2021 um 17:35 schrieb Antony Stone: > > > > Where exactly were those packets captured? > > tcpdump on the Asterisk-Server on the interface of the VLAN for the phones. > All traffic captured. 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? Antony. -- I have an excellent memory. I can't think of a single thing I've forgotten. Please reply to the list; please *don't* CC me. -- _ -- 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
On Tuesday 28 December 2021 at 17:28:47, Luca Bertoncello wrote: > 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"): . > and here what I see if I call another mobile phone (then the number does > NOT change): . > So, I see, there is a "P-Asserted-Identity"... But I can't understand > why... > > Any idea? Where exactly were those packets captured? On the connection from the called telephone sending the reponse in to Asterisk? Or on the connection from Asterisk sending the response out to the calling telephone? (It looks like the latter to me.) It may well be a good idea to do both and compare, to see whether Asterisk is actually adding the header. Antony. -- 3 logicians walk into a bar. The bartender asks "Do you all want a drink?" The first logician says "I don't know." The second logician says "I don't know." The third logician says "Yes!" Please reply to the list; please *don't* CC me. -- _ -- 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
On Tuesday 28 December 2021 at 16:58:01, Luca Bertoncello wrote: > 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"? 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. Antony. -- "Measuring average network latency is about as useful as measuring the mean temperature of patients in a hospital." - Stéphane Bortzmeyer Please reply to the list; please *don't* CC me. -- _ -- 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
On Tuesday 28 December 2021 at 14:30:17, Luca Bertoncello wrote: > 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. 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. Antony. -- "Remember: the S in IoT stands for Security." - Jan-Piet Mens Please reply to the list; please *don't* CC me. -- _ -- 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
