similar to: Can't get G729 working after buying a license.

Hello all, I am going to register asterisk sip users through active directory accounts LDAP (that is a separated server with ip : 192.168.11.17) So I have followed the below link as well: https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html

Hi, I am trying to post this again as I am getting no answers and the support@digium.com bounces... (I have searched the whole list and can't find the answer either) I have installed a 5 user license for G.729 and want to route calls through Asterisk from my G.729 phone to Cisco AS5300 also using G729. Both Cisco and the phone connect using this codec if I do not force the call to go

Hi I seem to have installed the G729 codec properly but can't seem to get it to work .. in the Asterisk startup I see .. [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) == Detected 1 licensed G.729 transcoders WARNING[1074408160]: File translate.c, Line 218 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator

Hi, I have a Cisco 5300 which I want to make a call THROUGH the Asterisk PBX (security) to an IP phone which supports g729, and vice versa. Both Cisco and the phone talk this codec if I do not force the call to go through * However if I say canreinvite=no in the sip.conf for either of these gadgets, the call will fail with No compatible codecs! I have bought a 5 user license just to

Hi, I have licensed the digium G.729A codec. But for some reason incoming and outgoing calls will ALWAYS use G.711a. When I force my phone to only accept G.729 then an incoming call from ZAP goes straight to my voicemailbox as the phone doesn't accept the codec Asterisk wants, even if I force it in sip.conf. Is there anywhere else I need to look to tell ZAP to use G.729 preferrably ? The

Hi, I am trying to replace my Cisco 5300 gateway with my new Zap TE405P card. Incoming calls and outgoing calls between my cisco and my SIP phone works fine on G.729. Recording messages in the asterisk voice-mailbox also works fine from both my SIP phone as well as PSTN -> Cisco -> Asterisk. I have licensed the digium G.729A codec. When I connect my ISDN PRI to my Zap card and I call

Firstly, I'm running Asterisk 1.4.4 on Solaris 10. I have several different internal SIP phones all sharing a single IAX2 VoIP channel. PHONES |------------- <SIP/uLAW> --------------| ASTERISK |-------------- <IAX2/g729> ------------|VoIP/ISP The g729 codec has been registered successfully and appears to be detected by Asterisk (NOTE: I have changed what I thought might have

Hi folks, In an effort to save bandwidth (our 7905s run over a WAN) we've switched from ulaw to g729a. We purchased 4 licenses from Digium (4 SIP clients, low call volume), and they seem to have been accepted: [codec_g729a.so] => (Annex A/B (floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: 07:53:aa:d3:e2:f2:bd:cc:27:60:9d:5f:da:eb:5d:e9:6e:09:a1:4e == Found license

Very sporadicially (1 out of every 40-5000 boots), our blade system will hang indefinately while PXE Linux is attempting to locate it's configuration file. This is causing our automated testing to hang and generate failures. I have put more details below. My concerns are threefold: 1) System hangs 2) PXE Linux bootstrap never seems to retry the transaction 3) PXE Linux bootstrap never seems

Hi Sir, We have integrated Skype with Asterisk (skype user id:- rexesbposolutions). Each call which is coming to skype account is getting transfered to Asterisk Queue. It has following two cases: case 1: When we call from normal skype account to skype account (rexesbposolutions), everything is working fine. case 2: This skype account (rexesbposolutions) has been assigned with a online virtual

Hi, I purchased a G.729 1 channel codec license from digium. And installed as per the documentation. Then configured the sip.conf to use the new codec. For that, I am added the following entries in sip.conf (via web interface, as i am using asterisknow 1.5) disallow=all allow=g729 allow=ulaw allow=alaw allow=gsm After that, when try to call through the PSTN line I can hear the voice of

Hi to all, I got a Cisco 2651XM wired to an E1 PRI. What I want to do is to pass all incoming calls to my asterisk. This is my actual conf: http://pastebin.ca/270677 with this I'm able to call my number from outside, but the call stop on the 2600, infact I can hear the tone, but I'm not able to forward calls to my asterisk. Anyone got an idea of my errors? Thanks to all. -- .:FaberK:.

I apologise for writing again about the problem with using stepAIC + multinom, but I think the reason why I had it in the first place is perhaps there may be a bug in either stepAIC or multinom. Just to repeat the problem, I have 126 variables and 99 cases. I don't know if the large number of variables could be the problem. Of couse the reason for doing a stepwise method is to reduce this

Hi, I have a query from a client that can he use the E100P card to terminate the 2Mbps bandwidth in a linux box, thus reducing the cost of cisco router ?? The other end is a cisco 2620 router with E1 VWIC-1MFC. Can anyone explain if its possible with Asterisk and further any configuration help. Applreciated. Regards Azher Amin ------------------------------- http://www.consulttech.com.pk

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For some reason I cannot kickstart via pxe after installing a new SMC 48Port GigE switch. I swapped out an old hub to test and kickstarting still works with the old hub. Does anyone have any suggestions, --

Sorry about this, but do any users have more detailed iax.cc information? Will they do trunking? What are the hosts that I will be logging into? I want to make sure that they will work well for me, and I would like to do some traceroutes to make sure that they are close! Thanks. -- Kristian Kielhofner

https://www.cnbc.com/2018/10/28/ibm-is-reportedly-nearing-deal-to-acquire-red-hat.html

> -----Original Message----- > From: CentOS [mailto:centos-bounces at centos.org] On Behalf Of Larry > Martell > Sent: Sunday, October 28, 2018 4:11 PM > To: CentOS mailing list <centos at centos.org> > Subject: [CentOS] IBM buying RedHat > > https://www.cnbc.com/2018/10/28/ibm-is-reportedly-nearing-deal-to- > acquire-red-hat.html Damn, this is bad enough to make

On Sun Oct 28 10:20:31 PM, Alain p?an wrote: > Le 28/10/2018 ? 22:10, Albert McCann a ?crit?: > > Damn, this is bad enough to make one weep. > > Red Hat would stay as a distinct entity inside IBM. IBM has also > contributed to Free software, and especially Linux kernel. > I don't know how bad it is and the implications for CentOS... That old war wound started aching