similar to: RE: [PROBLEM] Still exist --> DTMF Tones, occures in Asterisk - Channelwide

> I don't see anything obviously wrong with your configs. > You don't want relaxdtmf. That can cause the problem, not fix it. POST 2 --> got no response Hi Eric, at the begining -> Thanks for your help. relaxdtmf is not written in my config, so it should be at the default, i guess i remember default is yes ? However, the dtmfmode should be the same, i think so, too, but

Hi, I have asked this question months ago - i have "toggled down" all DTMF Recognizations in my Asterisk (no more features etc) and found more people which recognized the same problem, but i cant find any help for them and me. The Problem (short as possible) : In a randomly call in my business day some unit in my Asterisk System sends an randomly DTMF Tone, like "A"

Hi, I have Asterisk-1.2.18 install with FreePBX & more than 75 extnsion, daily I come accross an issue & try resolving them its either user learning curve or my ignorance. But, I dont know what to say regarding this issue. I have my Dial Plan for internal users to have a 3 Digit Extensions. So instance my Ext is 239 & someone dials the main #, its gets the

Hi, i am from Germany, so excuse my School English. I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update of Asterisk 2 wooks ago, Echos accure in my SIP Calls. I use SNOM 360, sometimes there is no echo (for example if i call myself via SIP->Asterisk->SIPProvider->TELEKOM->ISDN) but if i call other people there occures Echo many times. The Routing is always the

Hey Everyone, i am using Asterisk 1.4.21.1 with a old Patton SN1400 and some SNOM 300,320,360 Devices. In the combination with asterisk and the patton, there are occuring some strange behaviour, due to the calling and answering everything works good, clear voice, great availability. All the devices have to use ulaw, alaw and slinear is available but never the first choice since i use my

Hi List, I am using Asterisk 1.6.2.18. One strange problem come into my knowledge after using this version of asterisk. Without pressing any digits or key from my mobile, I am getting DTMF into my asterisk server. For getting DTMF I have use one opensourse application which gets events from asterisk server and store into database. And after that I made my own script to gets these DTMF keys and

I am using Zaptel for my Wildcard and got DTMF Tones in my conversations since a long time. To control this , i search for a possibility to control the detection of DTMF and to control talkoff, called "dtmfthreshold". As seen in mISDN* i am in need of an Patch to control the "dtmfthreshold" in zaptel, it must be possible, because there is an value, staticly set to 1000,

I got an up2date Asterisk with SNOM360 as SIP and mISDN with 2 ISDN Cards, if i press in a call the "*" Asterisk, Asterisk destroys the call not, Asterisk lets him hang and do nothing, if i hangup, Asterisk tell me in the warnings-log that the bridging was not successfull ?! If have disabled the function to hangup in the features.conf, but the key is still available, can someone

Hi, I have 2 servers: one is running 2 B410P cards with 8 euroisdn lines (mISDN) connected on it, everything runs fine. I prepare a new server - HP 360 G8- with 2 HA8 cards each of them 1 module of 4 lines. Already had with this machine an RMA on both cards as they was faulty and crashed the server. What happends is that when I connect cables on the HA8 modules (those cables are unpluged

Good Morning, I've recently gotten Asterisk installed and configured our IVR using FreePBX. Things seem to be going well except a few of our inbound callers are ending up in the wrong place when trying to connect to a specific extension. The example I had this morning was someone trying to call extension 212 and getting connected to the Sales queue which is option 2 on the IVR. I looked in

I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of the three perform well in all enviroments. Between stablity issues, T38 and DTMF talkoff all three suffer some combination of issues. I am looking at Patton and Innomedia. Has any one tried either brand and what is your experience with them. Which would be the base for stability, audio quality, provisioning, DTMF

Hi, I am trying to configure * 1.0.7 box with a Digium Wildcard TE110P T1/E1 and libunicall latest code. All libs compiled successfully and the E1 have a green light! I am able to receive a call (or at least) testcall shows some information when an incoming call is received so the drivers and basic configuration is working. My * box has 2 cards, one TDM400B (2 fxs and 2 fxo) and one TE110P

Hello! We are using the Digium 405PP card, and getting the following messages: Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1 Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 My config file is below. We are trying to set up D-Channel on channel 24, 1-23 in trunk group 1,

Regards Bilal ---------------------- On Thu, 2008-06-12 at 01:23 -0700, bilal ghayyad wrote: > Where did u find a good IAX IP Phone? I've had good success with my Allnet IP-7960 phones. They have the ability in the firmware to either do SIP or IAX, and they even have a mode where you dial one prefix to send the call out using the SIP protocol, and another prefix to send the call out over

Just to add.... P[ 1] Transmitting 128 samples 2 misdn P[ 1] writing 128 bytes 2 asterisk P[ 1] Sending :160 bytes 2 MISDN P[ 0] misdn_jb_fill: written:160 | Buffer status:256 p:861fee0 P[ 0] misdn_jb_empty: read:128 | Buffer status:128 p:861fee0 P[ 1] Transmitting 128 samples 2 misdn P[ 1] writing 128 bytes 2 asterisk P[ 1] PH_CONTROL: channel:1 oad2:07nnnnnnnnn dad0:820055 P[ 1] --> DTMF

This was submitted to the Dev list last week, but there was no response, and perhaps it wasn't the right group. I am developing an application in which I need asterisk to pass on an incoming call to a separate IVR server. The problem is that asterisk appears to hang up while the IVR is playing back a sequence of recorded voice and systhesized voice prompts. My setup is: Analog line

Hi All, I have just migrated from Asterisk 1.0.0 to Asterisk 1.0.5 and I have an X100P installed. The old asterisk was working, but now the new version isn't picking up any calls! However, I did notice that after installation, I performed modprobe zaptel and modprobe wcfxo and they worked fine, but when I executed ztcfg, I get the following errors: ioctl(ZT_LOADZONE) failed: Invalid

Hi, Probably a stupid syntax problem.. but can't seem to make these files work for more than the first T1 (ok, if I comment out the 2nd, 3rd, and 4th span). Can someone proofread this for me? Thanks! ====================== zaptel.conf span=1,0,0,esf,b8zs bchan=1-23 dchan=24 span=2,0,0,esf,b8zs bchan=25-47 dchan=48 span=3,0,0,esf,b8zs bchan=49-71 dchan=72 span=4,0,0,esf,b8zs bchan=73-95

I'm a bit confused on how to setup Zaptel.conf and Zapata.conf when there is a TDM400P and a TE410P installed after upgrade. The TDM400P has 2 FXS in position 1 & 2 and 1 FXO in the fourth position. I see boot, WCT4xxP loading first and WCFXS loading second. According to my understanding, given above, the TE410P should be configured first, then the TDM400P. However, I'm not sure

Hi, I am using a ALLNET 91500 UPS with blazer_usb driver /usr/lib/ups/driver/blazer_usb -a ups91500 -L Network UPS Tools - Megatec/Q1 protocol USB driver 0.10 (2.7.1) I have this settings: upsc ups91500 at localhost battery.charge: 100 battery.runtime: 12000 battery.voltage: 27.50 battery.voltage.high: 27.3 battery.voltage.low: 18.0 device.type: ups driver.flag.novendor: enabled driver.name: