similar to: Intermittent Low volume

Hi List, I'm finding a solution to provide failover and load balancing features to my IVR system. Anyone suggest me what is the best solution please?. what the hardware I should use ?. I heard about RedFone, but someone on the mail list said that it is not good because TDMoE module in asterisk is not so stable and TDMoE is stale. And It seems that RedFone doesn't not support load

I wont say we are extremely high volume (40 concurrent calls) but I get occasional complaints about quality. Setup (at same location): Asterisk 1.4.26.2 FrontEnd Asterisk 1.4.26.2 Gateway with Sangoma A108D card with 2 PRI and LDT1 Connected via IAX2 trunking on its own VLAN Is IAX2 the way to go or would SIP trunking be better. I know its a pretty vague question but I am just trying to

Hi Does Sangoma 8-port card A108 support PCIe version 2.0 ? The cards is here And we want to use 3 such cards in this motherboard because it has 3 PCIe slots of version 2.0 http://www.intel.com/products/desktop/motherboards/DX58SO/DX58SO-overview.htm Is this a good idea ? Do you have any experience with multiple A108 with PCIe on the same motherboard that supports PCIe 2.0 ? Any comments

Hello, All. I am currently running a test configuration for the telephony engineer here. I have two Dell PE servers, each with 2 A108 Sangoma T1/E1 cards. Here is a rough drawing: Ameritec Call Generator 8 T1 ISDN/PRI lines ---> Asterisk ---> Asterisk ---> Ameritec Call Generator Sangoma A108 Sangoma A108 Terminating 8 ISDN

Hello, I need to convert the dahdi_monitor output to dBm0, so I can measure Echo Return Loss in dB. I've read a formula that calculate S(k) in ITU-T G.168 recommendation, where S(k) is the signal level in dBm0. Can I use this formula to convert it? If yes, what value should I use to the number of samples if I want to convert a single output from dahdi_monitor? -- Thanks in advance, Gustavo

Hi Does Sangoma 8-port card A108 support PCIe version 2.0 ? The card is here http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a108.html And we want to use 3 such cards in this motherboard because it has 3 PCIe slots of version 2.0 http://www.intel.com/products/desktop/motherboards/DX58SO/DX58SO-overview.htm Is this a good idea ? Do you have any experience

Hi, I'm experiencing "frequent" kernel panics on a system with Asterisk 1.4.26.3. There is no core dump, "just" a kernel panic. This is the only data I could copy from the screen: EIP: 0060: [<f8e248b4>] Tainted: P VLI EFLAGS: 00210297 (2.6.23-gentoo-r8 #1) eax: 00000130 ebx: 00000000 ecx: 00220028 edx: 00000978 esi: 346e5802 edi: 00000000 ebp: c3b45500 esp:

Hello everyone, I'm considering getting me a quad-core Dell PowerEdge 860 to run Asterisk. Anyone else using this model? Any tales of woe and sorrow I should know about? Then, in a couple of weeks, I'm thinking of getting a Sangoma A108 and giving that a try. Same question with that one - any quirks I should be aware of? Girts -------------- next part -------------- An HTML

Hi Eric, > On 12 avr. 2016, at 17:48, Eric Cooper <ecc at cmu.edu> wrote: > > On Tue, Apr 12, 2016 at 04:36:58PM +0200, Bertrand LUPART - Linkeo.com wrote: >> I just made a asterisk / dahdi fresh install on Debian 8.4, and ended up with the following packages : >> [...] >> However, i can't find any dahdi startup script, neither init.d neither systemd

Could anyone help me set up Asterisk in such a way that it makes SIP-SIP transfers using the REFER / NOTIFY method according to RFC-3515 ? SCANARIO: - Asterisk registers with PSTN<->SIP VoIP provider "V" (Vonage) as a friend - Asterisk is located in Europe, Vonage in located US. - Asterisk acts as an autoattendant located in Europe. - Asterisk answers and incoming call from

I am studying about echo cancellation in asterisk and I want to use the numeric information from dahdi_monitor verbose for my research. Unfortunately, I couldn't find anything about the unit of measurement used in this tool. Which unit is used to measure the signal level? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Does anyone currently use Vonage with Asterisk? I've tried to set it up but it looks like Asterisk (at least the version that I have) does not handle well the SIP call dialog, sending a BYE with the wrong tag. As a result, when I hang up, Vonage sends back a 400 Bad Request and the call on the PSTN side does not hang up. I know that Vonage does a lot of nasty stuff which impacts UA's

Hi, on one of our clients asterisk server we have the problem that you hear nothing on external calls. Here are the details abount the system: Asterisk 1.6.0.22 DIGIUM Wildcard B410 quad-BRI card (rev 01) dahdi-linux-complete-2.2.0+2.2.0 I have setup the following test extension: exten => 9216992,1,ANSWER() exten => 9216992,2,WAIT(2) exten =>

Hi, I'm trying to get Asterisk to connect to Vonage (softphone acct) to allow me to place and receive calls. I have successfully configured Asterisk to route inbound calls and send them to the correct extension, but I can't get outbound calls to work. I have Asterisk successfully registering with Vonage, but when an INVITE is sent out, I get a "404 Not Found" back from Vonage

I just got setup with vonage.ca with the motorola ata unit.. I fired up ethreal and checked out what's flying over the network... The sniff below would lead me to believe that it might be possible to have asterisk spoof the User-Agent field and register itself? Any thoughts/feedback? Thanks. > > No. Time Source Destination Protocol Info >

Hello everyone. I've seen postings for connecting asterisk to vonage but I'm still having trouble achieving that. I have a vonage softphone and I'm trying to register to vonage using asterisk. I have not had any luck. I am behind a firewall. I've successfully gotten xlite to connect and work from the same network. When I change the port setting in [general] to 5061, I am able to

hi group , i am working on dahdi_monitor for measuring voice quality , so i want to know that on which data i can tell that this PRI lines are working properly, is there any measurement on basis of that i can make MOS. i am working from last 2-3 days but i only get idea about making .raw file and making .wav file and visulal mode of RX and TX of PRI line. what i want is measurement of voice

After two weeks of reading about asterisk and joining this mailing list, I finally decided jumping on the asterisk bandwagon... Asterisk rocks!!! I have a www.Stanaphone.com SIP (free) for incoming line and a www.VOIPJET.com IAX line for outbound. I also have a www.Vonage.com line (gives me 500 outbound minutes) and a Cingular cell phone (gives me 800 minutes) and I also use Skype fairly

Now that I have a new card and my echo problems are 'mostly' solved, I have another major issue to deal with. After about an hour or so the card will stop detecting DTMF tones on incoming calls. dahdi_monitor shows the following: [root at asterisk wctdm24xxp]# dahdi_monitor 1 -v Visual Audio Levels. -------------------- Use chan_dahdi.conf file to adjust the gains if needed. ( # =

Hi All I am trying to dial out using SIP and Vonage using the instructions : <a href="http&#58;&#47;&#47;www.voip-info.org&#47;wiki&#47;view&#47;Asterisk&#43;and&#43;Vonage" target="_blank"