similar to: G729 and snom

Skipped content of type multipart/related-------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 2892 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20030403/46e84bf5/smime.bin

What is this? chan_iax2.c line 4695 (build_peer): Unable to support trunking on peer 'lamas-tigris' without zaptel timing codec_g729b.c Line 413 (load_module): Unable to initialize va stuff: -1 This is why I can't start asterisk in the background -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, there were some postings a few weeks ago telling that the GSM codec problem with snom100 will be fixed. But it still seems to be very quality. Will be any change in this subject? THX -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030516/2fa9d206/attachment.htm

What is the no authority found problem? And how can I register with * on IAX. It keeps rejecting the request telling that XXX not dynamic host. rejected any idea THX -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030520/a8a5907d/attachment.htm

I've had the same thing happen, only on the single port T1 card and a channel bank, and one of the FXO channels also having a phone attached elsewhere... I just wound up putting that channel in a different context and running Exten => s,1,Hangup (I'm just using the line for outbound dialing) -----Original Message----- From: asterisk-users-admin@lists.digium.com

Hello, I have that problem with codec G729. Please can somebody help me! WARNING[16384]: File codec_g729b.c, Line 413 (load_module): Unable to initialize va stuff: -1 == Detected 4 licensed G.729 transcoders WARNING[16384]: File translate.c, Line 219 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator 'g729tolinb' from format G729A to

Trying to configure voicemail with H.323 all I get is the following errors when I call 123, 666, 665, 664 or 031. I'm a newbie at this so, I think it might be a simple fix. [chan_oh323.so] => (OpenH323 Channel Driver) == Parsing '/etc/asterisk/oh323.conf': Found 0:00.004 OpenH323 Wrapper OpenH323 Wrapper Version 0.0alpha0 by inAccess Networks

The problem I mentioned a day ago is now getting worse. message log: kernel: PCI Master abort kernel: Unable to handle kernel paging request at virtual address 00230004 kernel : printing eip c02060a6 kernel : Oops: 0002 kernel : CPU: 1 kernel : EIP: 0010:[<c02060a6>] Not tainted ,... Some register dumps kernel: Process asterisk (pid: 23917, stackpage=c5aff0000) kernel: Stack: numbers:)

Hello, I've successfully installed Asterisk from last CVS and configured it for using with DLINK-DG104S as mgcp CPE and PGW2200 as external sip server. All are work fine at G711 codecs, but then I disable all codecs except g729 some calls failed (Not all calls. Some calls passed at g729 succesfully). All my devices configred to use only g729 and I don't see other codecs at mgcp or sip

Hi There is a Centos 7 up-to-date box with 2 interfaces, let's say 192.168.1.12 - enp2s0, 192.168.1.13 on enp3s0. Default gateway on enp2s0. The gateway is pfsense, IP is 192.168.1.1 with 2 WAN connections On the gateway the outgoing traffic is routed by source ip to different WAN, 192.168.1.12 to WAN1 and 192.168.1.13 to WAN2 On the centos box are set all the route and routing rules:

OK I did some researches and tests with it, and finally: I registered my messenger to the asterisk and called if from the snom. The audio from the snom to the messenger was PERFECT. By the time of the call This message was running on the asterisk console: WARNING[16400]: File dsp.c, Line 1107 (ast_dsp_process): Unable to detect process 2 frames My conclusion is that the snom100 utilizes MSGSM

On 09/08/2016 20:01, Gordon Messmer wrote: > On 08/08/2016 04:05 AM, Levente Birta wrote: >> Can I add this in any config files (ex: route-enp2s0)? > > > Yes. Add a route file for each interface, and set up rules to send > packets out the corresponding physical interface: > > https://blogs.oracle.com/networking/entry/advance_routing_for_multi_homed > Yes, but no

Dear i have buy two license of G729 codec and i have install/registered as documented but after i start "Asterisk -vvvcng" i notice this warning and if i made call the CLI say "No compatible codec!" How can i solve this problem? Thanks in advance Dimitri ------------------------------------------ [app_datetime.so] => (Date and Time) == Registered application

Hi I have a supermicro server, motherboard is with C612 chipset and beside that with LSI3108 raid controller integrated. Two Intel SSD DC S3710 200GB. OS: Centos 7.1 up to date. My problem is that the Intel SSD Data Center Tool (ISDCT) does not recognize the SSD drives when they connected to the standard S-ATA ports on the motherboard, but through the LSI raid controller is working. Does

Hi, Does anyone has a *working* patch to set the dnid from extension when it is now supplied, cos this plugin doesn't working without it. I tried and it passes empty string. THX -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030516/aae0a7f4/attachment.htm

Hi all Can someone explain me this: ifcfg-eth0-range1: ONBOOT="yes" IPADDR_START="192.168.1.48" IPADDR_END="192.168.1.55" CLONENUM_START="1" Why Bcast is 192.168.1.51 and why Mask is 255.255.255.252 ? OS: Centos 6.3/64bit Thanks Levi # ifconfig eth0:1 Link encap:Ethernet HWaddr 00:9C:02:99:FA:00 inet addr:192.168.1.48

On 09/08/2016 06:56, Anthony K wrote: > On 08/08/16 21:05, Levente Birta wrote: >> >> But how can I add achieve this only with ip route command ... without >> route? >> Can I add this in any config files (ex: route-enp2s0)? >> > Hi Levente. > > The iproute2 man page for each command is rather well documented on > CentOS 7. For instance, to view the

I left these information out sorry = Detected 1 licensed G.729 transcoders WARNING[1024]: File translate.c, Line 218 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator 'g729tolinb' from format 8 to 6, cost 99999 == Registered translator 'lintog729b' from format 6 to 8, cost 18 -------------- next part -------------- An HTML

Hello everyone. I'm struggling to get T.38 faxing to work in Asterisk 1.6.1.13 with a SIP DID provider here in Brazil (GVT - Vox IP service). Here's my scenario: When faxes arrive by a specific DID, they are routed thru this simple macro: [macro-recebefax] exten => s,1,Set(DB(fax/count)=$[${DB(fax/count)} + 1]) exten => s,n,Set(FAXCOUNT=${DB(fax/count)}) exten =>

Just wondering if there is a reason PPP support is compiled into zapata by *default*: # Uncomment for Generic PPP support (i.e. ZapRAS) # KFLAGS+=-DCONFIG_ZAPATA_PPP Especially since the comments imply that it should be commented out by default... The main reason I ask is because I usually try to re-compile the kernel to only include the bits that I need, and so I don't include PPP...