similar to: Asterisk 1.4.26 final release - What is blocking?

Good afternoon, I'm trying to use the CLI command file convert on an Asterisk 1.4.26 server with a TC400B transcoding card. The transcoding card is working well for calls but I have some trouble converting sound files from alaw to g729. The command creates empty file as you can see below... CLI> file convert /var/lib/asterisk/sounds/fr/service_notactivated.alaw

Hello everybody, I have 2 users connected on the same Asterisk server that are connected with 2 different Asterisk servers. For outgoing calls, one in receiving 183 Session Progress and the other not! Do you have any idea why? Thanks. I have tried to understand by myself and in their INVITE they have almost the same Allow and Supported SIP Headers The one that works: Allow: INVITE, ACK,

Hello all, I'm tring to retrieve a formula to calculate a MOS value from Asterisk RTCP stats... Have you got any idea how to do it? Thanks I'm reading all G.107 ITU docs to retrieve something... I'm saving the SIP RTCP stats with: [macro-hangupcall] exten => s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)}) exten => s,n,ResetCDR(vw) exten => s,n,NoCDR() So I retrieve

Hello All, I have a little complicated question about the Dial command. I use OpenSIPs to loadbalance Asterisk Servers, and Users are registered on Asterisk servers. Asterisk use the Reg. Contact entry to reach the UAC via the OpenSIPs server. Everything works except for trunk numbers: For each peer on Asterisk, "Addr->IP" is IP of the Proxy and "Reg. Contact" is the IP

Hello, I'm using samba 3.5.11 to connect a Windows 2003 Active Directory. With cups, samba is an part of a print server used to print to windows desktop shared printers. DNS are Active Directory Integrated. Network is both IPV4 and IPV6, IPV6 for Linux and Windows Vista and above. Some times, some users are not able to print. In logs of cups, I see to thinks "Unable to connect CIFS

Hello all, I have a TC400B Digium card in order to deal with transcoding and I have some trouble using it, I have a timer synchronisation problem! I would be very grateful if you have any idea to help me? It seems that the card is not correctly synchronised to the system because when I speak to one side, the sound takes 5 seconds to go to the other side, and increasing, after 30 seconds of call,

Hi Using SSH_ORIGINAL_COMMAND in AuthorizedKeys is so helpful, I'd like to know if it might be possible to access it in the AuthorizedKeysCommand context (via env ?). Is this possible ? can anybody give me advice on going into this ? If possible, I'll use this SSH_ORIGINAL_COMMAND to send client specifics information to the AuthorizedKeysCommand script. Currently, the only alternative

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good Morning, Any help would be grateful to help me understanding what's wrong... I have bought 2 g729a licenses to digium and I would like to have them works... My processor is an Intel(R) Xeon(R) CPU E5310 @ 1.60GHz (4 processors) so I have downloaded the

Hello, I've been looking into OpenSIPS to see if it's a worthwhile addition to our setup. We're currently running a cluster, using Heartbeat, between two servers. It works well but I'm interested in seeing if we can improve it. My manager heavily uses RPM's for installations rather than source, particularly using yum to update. I'm trying to actually install OpenSips via

Asterisk Project Security Advisory - AST-2011-001 Product Asterisk Summary Stack buffer overflow in SIP channel driver Nature of Advisory Exploitable Stack Buffer Overflow Susceptibility Remote Authenticated Sessions Severity Moderate

Hi, I'm having a problem trying to get a Cisco 7965 phone registered on Asterisk 1.4.26. As we know, Cisco now, for security reasons, has made the phone ports non-symmetric, in that it sends out UDP requests on a high port and receives them on a different port. It seems that, even with 'nat' set to 'no', that Asterisk is not honoring the Contact header and keeps attempting to

The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.26. Asterisk 1.4.26 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ This release resolves a large assortment of issues reported by the community. For a summary of the changes in this release, please see the release summary:

The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.26. Asterisk 1.4.26 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ This release resolves a large assortment of issues reported by the community. For a summary of the changes in this release, please see the release summary:

The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 1.4.26. Asterisk 1.4.26-rc1 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ This release is primarily a fix for an issue (#14867, #14717) related to security fix AST-2009-001 where IAX was not sending REGREJ to terminate invalid registrations. Instead it sent

The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 1.4.26. Asterisk 1.4.26-rc1 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ This release is primarily a fix for an issue (#14867, #14717) related to security fix AST-2009-001 where IAX was not sending REGREJ to terminate invalid registrations. Instead it sent

Good evening, Have you got any idea which prepaid application will be the best to do simple prepaid calls with a MySQL storage...? PS: I have a compiled by hand Asterisk 1.4.13 on a Debian Etch Thanks

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good evening, I have something strange, I have unread message in my voicemail box but the SIP NOTIFY that are received by my telephone are like: whereas there is voice messages inside! Any idea how to solve that? Thanks PS: I'm using asterisk 1.4.13 + Freepbx # U 192.168.95.235:5060 -> 192.168.95.73:5060 NOTIFY sip:9755 at

Hello, I make the migration of linux machine where I use smbfs to mount an IFS drive from an iSeries and I have troubles with Samba 3.0.9 The "Old" Computer : RH 8.0, Samba 2.2.5 The "New" Computer : Sarge, kernel 2.6.8, Samba 3.0.9 When I make "mount -t smbfs -o //as400/qdls /mnt/as400", all seems good but when I make an ls /mnt/as400/*, I receive this error

Hello everybody, I try to install -Ubuntu 8.04 server- a B410P and a TDM2400P together with Asterisk 1.4.26-2, dahdi-linux-2.2.0.2 and dahdi-tools-2.2.0. Problem I face is the following one: CLI> module load chan_dahdi.so == Registered application 'DAHDISendKeypadFacility' == Registered application 'ZapSendKeypadFacility' == Parsing

Hello, I'm trying to setup shared lines with Asterisk 1.4.26 and Snom phones. It seems that Asterisk works correctly (I get "State: SLA_TRUNK_STATE_RINGING" from the CLI) but the lamps on the phone are not blinking even if I setup one function key on my phones as shared line with number: <sip:line1 at 192.168.0.123> where line1 is my DAHDI/32 Any hints appreciated. :)