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Hi all Apparently there is a patch for those 1.2.4/5 MeetMe Freezes: http://bugs.digium.com/view.php?id=5884 Haven't tried it out yet. Benoit Panizzon -- I m p r o W a r e A G - System Services ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ____________________________...

...as well, so the line cannot be hijacked. So I'm looking for something like On Register: If check_allowed_ip(auth_username) { return; } else { Reply(403 Wrong IP for this user); } Any ideas how to do that? (Yes, I asked Google and found nothing useful yet) Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ____________________...

...efore the call gets connected through. Is there a way to do this? Example: I'm registered to multiple SIP providers. They come in to a context each and then get through to my phone. Now I would like to send myself an announcement about from which SIP provider this call came from. -- Beno?t Panizzon, <bp@imp.ch> ------------------------------------------------------------------------ ImproWare AG, UNIXSP & ISP Phone: +41 61 826 93 00 Zurlindenstrasse 29 Fax: +41 61 826 93 01 CH-4133 Pratteln Net: http://...

...signaling_address=[local IP] Assuming this is the own interface IP that would be told to external endpoints was obviously wrong. In the end this caused the Proxy-SBC to believe it was not getting an OK to it's forwarded registration and discard this session. Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ____________________...

...think having called the wrong number when they hear the announcement. Is there a way (like in many other PBXes) that the VoiceMail user could record his own announcement? (like, hello, this is the Voicebox of John Smith, please leave a message after the tone). Mit freundlichen Gr?ssen Benoit Panizzon -- I m p r o W a r e A G - System Services ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ____________________________...

...s to solve that issue? I see the asterisk sending out a challenge and getting a proper reply from the patton, but then stills complains about the endpoint not matching. According to the manual there is no type=identify match=authentication_username or similar. Mit freundlichen Gr?ssen -Beno?t Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ____________________...

...ecially when an 183 message is sent, the Contact header does contain the wrong IP Address. Is this a known issue 13.18.3? Or is there a way to make absolutely sure the IP addresses within the Contact header is corresponding to the endpoint the packet is sent out? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ____________________...

...they generated we would need to know exactly which calls originated from which IP address to put an exact sum of damage done by each of the abusers. Well for this case it is too late now. But is there a way to get the IP Address of the SIP Client being logged in each CDR? Kind regards Benoit Panizzon

...xact subscribe message and reply as recored by the Snom M9 Logging facility. 2017/12/02 11:58:00 [SIP-Reg:5]: SIP Tx udp:[2001:4060:dead:beef::1]:5060: SUBSCRIBE sip:11 at woody.ch SIP/2.0 Via: SIP/2.0/UDP [2001:4060:dead:d1d0:204:13ff:fe30:228d]:2799;branch=z9hG4bK-88k838;rport From: "Beno?t Panizzon" <sip:11 at woody.ch>;tag=fg0ojl To: "Beno?t Panizzon" <sip:11 at woody.ch> Call-ID: 6lrsku1p at snom CSeq: 933701145 SUBSCRIBE Max-Forwards: 70 Contact: <sip:11@[2001:4060:dead:d1d0:204:13ff:fe30:228d]:2799;transport=udp;line=u5go22>;reg-id=1;+sip.instance="&l...

...ot;<sip:1000 at 1.2.3.4:5060;user=phone>" in the variable $pai. P-Asserted-Identity: "John Doe" <sip:2000 at 1.2.3.4:5060;user=phone> Is getting me $pai containing just "John". Anyone a clue how I could get the whole header? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ____________________...

On 2013-06-16 21:46, Timo Sirainen wrote: > On 14.6.2013, at 9.15, Benoit Panizzon <benoit.panizzon at imp.ch> wrote: > >> Is there a way to get quota-status to also use the proxy feature to >> request >> the quota information from the correct machine? > > Looks like this is a missing feature. I first thought quota-status > would go through...

...List We have some CPE which run an embedded asterisk 13 with chan_sip. Unfortunately, when a registration is rejected, those stop trying. I am familiar with pjsip which allows to configure: auth_rejection_permanent=no How do I achieve the same with chan_sip? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ____________________...

Hello Le 04/12/2023 à 10:56, Benoit Panizzon (by way of Benoit Panizzon <benoit.panizzon at imp.ch>) a écrit : > Hi List > > We have some CPE which run an embedded asterisk 13 with chan_sip. > > Unfortunately, when a registration is rejected, those stop trying. > > I am familiar with pjsip which allows to configure...

Hi Joshua thank you for the quick reply > Have you checked the Asterisk console when PJSIP is loaded to see if > the endpoint did not load for some reason? Does it show up in "pjsip > show endpoints"? Yes, the endpoint shows up. Endpoint: 11/(scrubbed from mail) Not in use 0 of inf InAuth: 11/11 Aor: 11

...39; value would do for an example :-) Could you please make an example how to do that? PS: CONNECTED_LINE_CALLER_SEND_MACRO is depreciated and should be replaced by: CONNECTED_LINE_CALLER_SEND_SUB Do I have to call my routine [sub-routine] and end with ExitSub? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ____________________...

...om and are not set to the context [from-ISP-X] but probably to [default] and challenged. Of course, I could allow guests, but that would bring back the problem of having unwanted calls from sip scanners. So how do I tell the asterisk to also match calls from the ip of the outbound proxy? -Beno?t Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ____________________...

...e this problem by setting up a openldap server and get winbindd to store username => Posix uid's into this directory? Has somebody allready got such an environement running? I'm not so interrested in theoretical possibilities to solve it, I've read them too :-) Regards -- Beno?t Panizzon, <bp@imp.ch> ------------------------------------------------------------------------ ImproWare AG, UNIXSP & ISP Phone: +41 61 826 93 00 Zurlindenstrasse 29 Fax: +41 61 826 93 01 CH-4133 Pratteln Net: http://...

...MUST set this to the customers real phone number) I can correct ${CALLERID(number)} on the calling channel and this is preserved on the outgoing channel. But I have not yet found any 'easy' way to pass the corrected ${PAI} Variable to the callee channel. Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ____________________...

...the trunk to 4 channels so 400kbit/s can be assured by QOS measure. But when testing, I noticed caller don't get busy. pjsip show endpoints happily shows 5 of 4 channels in use. Do I have to query the device state from the dialplan and manually return busy? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ____________________...

...; XX,n,set(REDIRECTING(reason)=cfb) exten => XX,n,Transfer(SIP/YY) I did try with 'reason' 'orig-reason' I added cfb with and only quotes, I did try cfnr. But the 302 message generated this way allways contains reason=unknown in the diversion header. Any hint welcome. -Beno?t Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ____________________...