similar to: Dropped SIP connections never being closed?

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

Hi all I noticed that most caller are quite confused by the standard voicemail announcement text. Especialy as the number read is the 'internal' number. Callers often hang up because they 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

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 doveadm

Hello We had this situation: Some bot-net did try to guess SIP logins and finally succeeded. The Asterisk Server was abused to call a large number of expensive destinations. It is clear that the sip logins have been passed to various persons (probably posted on a forum somewhere inviting to do 'free calls'). Right after the affected password was changed, the message log shows which

Hi all We have the following environement: Microsoft ADS for Windows Users, NIS for Un*x Users. Samba 3.x Fileservers. Win2k/XP Clients which use CIFS to connect to the Fileserver. FreeBSD/Linux Clients which use NFS to connect to the Fileserver. For the moment, Windows User authenticate against the ADS and Un*x users authenticate against a NIS Server. Everything runs fine. But we would like

Hi all There seam to be a very short timeout waiting for digits being dialed. (about 6 seconds). Is there a way to increase that time? I have a phone with integrated address book and my fingers are just not fast enough to open the menue, select an entry and hit 'dial'. -Benoit-

Hi After migrating from 1.2.4 to 1.2.5 I noticed that: show application dial does not show the 'R' option anymore. Has this become an undocumented feature or has it gone completely? 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

Hi out there To play the correct announcement in app_voicemail I whould be able to read the SIP Diversion Reason which ist sent by another PBX: Invite contains: Diversion: <sip:+41315995003 at 157.161.10.190>;reason=no- answer;privacy=off;counter=1 Asterisk Logs: RDNIS for this call is is +41315995003 (reason no-answer;privacy=off;counter=1) From what I see in the source of chan_sip

Hi all I don't manage to get asterisk 1.2.5 or 1.2.6 running with the zaphfc driver.... The scripts from junghanns.net do download a very old libpri and asterisk version which is too buggy for me to use. Isn't there an acutal patch to get zaphfc support in *? -Benoit-

In article <200603181001.08589.benoit.panizzon@imp.ch>, benoit.panizzon@imp.ch says... > Thank you for the hint. Now finaly I can 100% reproduce the problem. Yes, if I > hang up during Playing 'conf-onlyperson' my machine freezes. It's not a GSM > Enconding problem as I suspected first, this happens with every encoding. > > magma*CLI> > -- Executing

Hi I would like to send a text to the called person when he picks up the phone before 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,

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

Hi Gang To increase security against phished passwords and similar attacks, we consider offering customers to define IP ranges (or GeoIP locations) from which their dynamic registrations are being accepted. I can already look at the source IP in the dial plan, so no issue with validate an INVITE against a source IP. But I would also like to prevent registrations from outside of this

Hi Joshua Thank you for your reply. Indeed, Ubuntu only ships with this old version. Upgraded to 16.2. via PPA. Problem persisted. Well, I already mentioned that this is a machine with two physical interfaces with different routes which on the 'external' side handles SIP customer registrations and has an 'internal' IC Trunk to a commercial Voice Switch via private IP Range. I

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: 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

Hi Gang I have stumbled of this problem. I need the P-Asserted-Identity header in an AGI scrip. In the Dial-Plan I do: same => n,Set(PAI=${PJSIP_HEADER(read,P-Asserted-Identity)}) In the AGI I do: my $pai = $AGI->get_variable(PAI); This works fine, unless the PAI contains quotes: P-Asserted-Identity: <sip:1000 at 1.2.3.4:5060;user=phone> I get "<sip:1000 at

Hi there I europe alaw is usual. I have a SIP Phone which perferes ulaw. When my * box has to transcode alaw to ulaw the sound get's one way choppy. (alaw => ulaw is choppy, ulaw => alaw is fine). I managed to fix the issue by forcing my SIP phone to use alaw only, but is this a know issue with asterisk 1.2.13? -Benoit-

Hello We have migrated our email services from a server, which did not support IMAP and folders, therefore threated the plus sign + as a normal character in a part of an email address. Our new server delivers the emails via lmtp to dovecot. the few users which got a + character in the username first could not log-in (fixed by adding + to auth_username_chars). Now the next problem turn out to

Hi all Hmm, often when my Asterisk tryes to register, it get's the answer back: Got SIP response 302 "Moved temporarely" (and an IP). But it looks like it's not respecting this redirection and tryes again and again to register to the server configured in sip.conf instead of the one the SIP provider tryes to redirect to. Any known issues? Mit freundlichen Gr?ssen Benoit

Hi All I'm tracing a very strange problem which I could reproduce with different versions up to 1.2.5 (sorry, didn't update to a newer one yet). Scenario 1: Problem does not occure. ============================= Sip Phone registered directly to the Asterisk. exten => i,1,Zapateller() exten => i,n,Playback(invalid,noanswer) exten => i,n,Hangup() Works like expected. I dial an