similar to: Asterisk -> Office 365 Unified Messaging... anyone done it?

Hi gurus!!! I have a Freepbx with Asterisk 1.8.25.0 with a sip trunk on the pstn Every minute asterisk sends an OPTION Request, i beleived that it's related to qualify functions. The every minute annoyng answer of the pstn is "403 Forbidden". Some people told that asterisk is not sending the username in the OPTION, required by the pstn. Taking a look of the example of rfc3261.txt

Hello I used https://wiki.samba.org/index.php/Migrating_a_Samba_NT4_Domain_to_Samba_AD_(Classic_Upgrade) to migrated my old samba 3, i created a dc and a member file server, but on member i can't see id for usernames and groups. This is my smb.conf on dc [global]         netbios name = DC1         realm = LXCERRUTI.COM         server role = active directory domain controller        

http://bugs.freedesktop.org/show_bug.cgi?id=17999 Summary: HIgh memory and cpu use Product: xorg Version: unspecified Platform: x86 (IA32) OS/Version: Linux (All) Status: NEW Severity: normal Priority: medium Component: Driver/nouveau AssignedTo: nouveau at lists.freedesktop.org ReportedBy:

Hi all, I'm getting one way audio when calling over the SIP trunk i.e. end device B (remote end of SIP trunk) can hear device A (softphone registered with Asterisk) but device A can't hear device B. Even though I configured same NAT configurations on other servers and they are working good. The NAT configuration is listed below; localnet=130.0.0.0/130.0.0.0 externhost=12.131.12.13

We are running Asterisk 1.8.5.0 with an uptime of 40 weeks. Just today our streaming music on hold stopped working. I remember when we had first installed 1.8 we had an issue where the streaming music on hold would not work because Music On Hold was using the DAHDI timing module. We needed the DAHDI timing module loaded so that paging would work. However, at that time we upgraded to 1.8.5.0 and

Hi, I have 8 E1 PRI Lines and i have 200 phone numbers and 200 channels (25 channels per PRI). is there a utility available in Asterisk to dial out 200 numbers and run a campaign for 200 numbers concurrently and play a mp3 file ? Please suggest/guide Regards Kaushal

On a BT line, how do I determine whether the number on an incoming call has been deliberately withheld (by dialling 141) or is merely unavailable (e.g. because it originated from overseas or passed through some ancient switching equipment) ? In the first case, I want the caller to be played a message to the effect that we are not at home to anonymous cowards but if their business is

Hello guys. When I am trying to send fax through T38 to linksys SPA (properly configured etc. - I have tried it with other systems), I'm getting error and fax is not delivered. I'm getting this errors in asterisk.log: WARNING[687] udptl.c: No UDPTL ports remaining ERROR[687] chan_sip.c: UDPTL creation failed WARNING[687] udptl.c: No UDPTL ports remaining then, couple lines down:

On Fri, 26 Oct 2018 16:47:52 +0200 Corrado Ravinetto via samba <samba at lists.samba.org> wrote: > thank you for your comprehension > > Il 26/10/2018 16:40, Rowland Penny via samba ha scritto: > > OK, two further ldbsearches: > > > > ldbsearch -Hldap://$(hostname -s) -k yes -P > > '(&(samaccountname=*)(uidNumber=*))' uidNumber | grep uidNumber

Hi, When starting Asterisk (1.8.5.0) I see in messages: [Jul 18 15:47:50] WARNING[15491] loader.c: Error loading module 'chan_gtalk.so': /usr/lib/asterisk/modules/chan_gtalk.so: undefined symbol: ast_aji_get_client [Jul 18 15:47:50] WARNING[15491] loader.c: Module 'chan_gtalk.so' could not be loaded. Yet I do have iksemel installed: ls -l /usr/local/lib/libik* -rw-r--r-- 1

Dear all, i have a problem with a system running - Ubuntu 10.04 ( all updates done ) - ii asterisk 1:1.8.5.0-1digium1~lucid Open Source Private Branch Exchange (PBX) - ii asterisk-dahdi 1:1.8.5.0-1digium1~lucid DAHDI devices support for the Asterisk PBX I also use freepbx 2.9 for the configuration. Hardware is a Dell R410 and a Digium

Hello Rowland and thanks for fast answer according with your suggestion i modified my smb.conf like this: [global]         lanman auth = Yes         log file = /var/log/samba/%m.log         ntlm auth = ntlmv1-permitted         realm = LXCERRUTI.COM         security = ADS         winbind offline logon = Yes         winbind use default domain = Yes         workgroup = LXCERRUTI        

Use the command bellow to check if is Asterisk opening the port. netstat -nap | grep 4569 You need to see something like this output, otherwise your asterisk is not opening the port. udp 0 0 0.0.0.0:4569 0.0.0.0:* 10244/asterisk Att, H?lvio Junior dCAA - Digium Certified Asterisk Administrator SafeId - Gest?o de identidades e Acessos +55 41 | 9

Trying to setup UM with Office 365 which requires TLS. I've tried under 1.8.5.0 and under 1.6.2.16.1 and I get the same error: [Aug 11 06:50:20] VERBOSE[3023] tcptls.c: SSL certificate ok [Aug 11 06:50:20] VERBOSE[3023] tcptls.c:?? == Problem setting up ssl connection: error:00000000:lib(0):func(0):reason(0) [Aug 11 06:50:20] WARNING[3023] tcptls.c: FILE * open failed! Following the

Hi all, i have the following setup PSTN -> routing server (asterisk 1.6.2.11) -> IAX -> callcenter asterisk 1.6.2.9 -> SIP -> agent Does work quit fine - then agent does have the abibility to transfer a call to a third party - the agent can initiate the transfer over a web interface - it does generate a asterisk manager atxfer request... So agent does initiate transfer - call

I'm using FreePBX/Trixbox with Asterisk 1.4.17-1 trying to register against a Sylantro server in front of a Metaswitch. I'm able to register and receive inbound calls but outbound calls are rejected by the far end. The username and password have been checked repeatedly. Putting the same authentication and server IP into a softphone or polycom phone work fine for inbound and outbound

Hi, We are running a samba fileserver, access controlled using posix acl (right 770, with users/groups on the filesystem level. Therefore samba shares look like this: [share] path = /srv/academic read only = no writable = yes create mask = 0770 directory mask = 0770 Now certain users complain that they cannot access certain folders, but looking at the folders from the linux fileystem, their

Anyone have any testing experience with T38 and HT-502 Grandstream? I just want to confirm that t.38 is working on this device. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110801/fea18d8c/attachment.htm>

The Asterisk Development Team has announced the release of Asterisk 1.6.2.18. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.18 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

The Asterisk Development Team has announced the release of Asterisk 1.6.2.18. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.18 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this