similar to: How to set overlap dial timeout in bristuff zaptel?

has anyone seen a bristuff version compatible to the actual *1.2.6/zaptel 1.2.5 ? the actual bristuff-patches version 0.3.0 PRE 1l doesn't apply correctly anymore...

Hello We would like to customize the voicemail menues. So the intro should not be played if some user has recorded an own greeting message and we would also like to remove some options from the menue. Is this all hardcoded or is it somehow possible to redefine the voice menues and the order how messages are played via voicemail.conf? Mit freundlichen Gr?ssen Benoit Panizzon -- I m p r o W

Next strange problem... W2k3 ADS. Sambe as ADS Member. pam_krb5 nss_ldap winbindd all seam to working correctls. Windows Users can access the shares on the Samba Server and can login using pam. smbclient works for all users... except from the Domain Administrator. smbclient //server/user -U user => is fine.... smbclient //server/Administrator -U Administrator [2005/03/23 17:33:30, 0]

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 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-

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

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,

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi All Well, it looks like I run into the same problem as many others... Unfortuately the FAQ referenced in the mailinglist archive with the solution does not seam to be online anymore. What I did try: rpm --import /usr/share/rhn/RPM-GPG-KEY* rpm --import /usr/share/doc/centos-release-3.3/RPM-GPG-KEY And also: rpm --import

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

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 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

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

Dear fellow list readers This is the situation: ISDN Devices => Patton ISDN to SIP GW => Asterisk PJSIP The Patton GW resides on a dynamic IP address, so I cannot really use match=ip in the identify section. The Patton does not send a line parameter. The ISDN Devices behind the patton have different MSN and should be able to send them in the From: Header, so the default endpoint

Hello List I work at an SIP Provider and we have added and SBC in front of our Voice Switch to protect it. This requires all our SIP Trunk customers to register via a 'proxy'. I struggle with Asterisk to work over a proxy. This is what I have done so far. register => username at sip.example.com:password at sbc.example.com This works fine, asterisk is sending registrations via the

Hi Joshua I had a shot at your suggestion, bug still no success. I fear the 181 is sent before the macro is called. I want to change the Diversion Header in the 181 message sent back to the caller to put the number it contains in the correct e164 format (stripping the 0 and adding +41 for Switzerland) but just any 'dialplan set' value would do for an example :-) Could you please make

Hi Gang Server, two interfaces, routing to two different networks. Two transports defined, each bound to the corresponding ip assigned to the interface. But still, especially 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

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 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