similar to: Non-codec capabilities (dtmf): us - 0x1 (telephone-event),

Hello, does some people here use https://voicetrading.com which is a Dellmont service from Netherlands. At the high begining they were also known as Finarea (CH and DE mixed Co) Anyway, after moving from Asterisk13/chan_sip to Asterisk16/PJSIP our callerID is no more seen by them. We use Set(CALLERID(num)=+331234356789) and Set(CALLERID(name)=Co name) or equal to CALLERID(num). We tried

On Thursday 18 June 2020 at 19:57:03, Administrator wrote: > does some people here use https://voicetrading.com which is a Dellmont > service from Netherlands. At the high begining they were also known as > Finarea (CH and DE mixed Co) > Set(CALLERID(num)=+331234356789) and Set(CALLERID(name)=Co name) or > equal to CALLERID(num). We tried replacing + with 00, same problem. >

Aloha, I'm not sure why I'm getting this error, but I can't seem to get chan_dahdi to load. SIP & IAX2 are working fine. Debian 4 w/ 2.6.28 kernel. Asterisk 1.6.1.5, dahdi-linux 2.2.0.2, dahdi-tools-2.2.0 CLI> module load chan_dahdi.so Unable to load module chan_dahdi.so Command 'module load chan_dahdi.so' failed. [Sep 1 10:57:51] WARNING[31696]: pbx.c:4550

Hello, I hope this mailing list is correct. I've a cluster pacemaker with a clone resource ocfs2 with ocfs2-tools-1.4.1-25.6.x86_64 ocfs2-tools-o2cb-1.4.1-25.6.x86_64 on Opensuse 11.2 After some network problem on my switch I receive on one of 4 nodes of my cluster the following messages Aug 18 13:12:28 nodo1 openais[8462]: [TOTEM] The token was lost in the OPERATIONAL state. Aug 18 13:12:28

Hi, sorry for this a bit OT. I'm using VoiceTrading for some calls -premium route- and can't get CID to work despite the fact that CALLERID(num) and CALLERID(name) are setted. I ask in VT->myAccount to accept calls from my IP without checking username & secret. On incoming calls the CID is setted to 0100000000 If I accept calls from username & secret and no IP relation,

Here's a little story on all the cheap guys trying to get the best rate on any route out there ( lcr and others). Anyone have 0.000001 to Mexico billed 1/1 ????? " When customers call us to ask if we sell Cuba termination for 50c/min, I sometimes joke and tell them "sure, I'll give you Cuba for 1c/min, but it will have 0% ASR". Or some other times, I

Hi, We are using MixMonitor to record the call. When the call is bridged, the latency is significant. We tried to increase the internet speed and the server RAM and processor speed and still we are having that issue. We use VoiceTrading and Gafachi's Termination minutes to make calls. As we are in US and VoiceTrading in Europe, somebody suggested to move the termination minute provider

Hello, We have a quadBRI in NT mode with bri_cpe_ptmp signalling and when connect a ISDN telephone to this nothings happen. What can I do? My config files are this: Zaptel.conf: loadzone=es defaultzone=es # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,1,3,ccs,ami span=3,1,3,ccs,ami span=4,1,3,ccs,ami bchan=1,2 dchan=3

I connect Asterisk to my analog PBX using X100P. In my analog PBX, I need to dial 0 (zero) to pick up the line. How can I use Dial command to dial (0 + telephone number) directly? I used exten => 10,1,Answer() exten => 10,2,Dial(Zap/1/0) exten => 10,3,Hangup It works, but I need to dial 10 and after the ring tone, the telephone number How can I do?

I'm new to this list. Reading the asterisk handbook pdf (good work) but but still have some questions. Using Trustix 2.1 and installed Asterisk via CVS, zaptel and libpri. We have a dedicated server which is connected to our telephone company. It makes us able to call ordinary phones via VOIP using Ericsson DRG22. Would like to make people able to call me - and get a message "dial 1

Hello, I'm really new on Asterisk. Is it possible to use a telephone machine connected to a modem as an asterisk voice input output device? I do not need PSTN connection. The scheme i'm thinking about is; user -> phone -> modem -> asterisk -> ip -> vice versa. If it is possible can a user dial another asterisk user via the phone? I've searched astersik lists but

Hi, I have a Cisco 7960G phone for which I know the pinout of the headset socket. I have a couple of standard telephone headsets which I do not know the pinout of. I'd like to connect the two. If I have the pinout of a normal/standard headset I can rewire the ones I have to match the cisco. Thanks Mike

Hello All, I am looking for a IAX2 analog telephone adapter, just want to ask your views on which ones are bad, good and the best. Thanks in advance, Dinesh Birlasekaran Network Engineer, ComIT, Institute of Molecular and Cell Biology 61 Biopolis Drive, Singapore 138673 HP : 92962676 DID : 65869804 Fax : 67791117 Email : dinesh@imcb.a-star.edu.sg WWW: www.imcb.a-star.edu.sg

Hello I am using a Snom 190 and the quality seems OK. Trouble is the volume is quite low and full volume on the Snom is still too low. Is there something I can do on the asterisk to increase the volume? Angus

I have Asterisk setup with two PRI's one going to my telco and the other going to a Norstar Meridian system. The Norstar Meridian is sending a BTN number that needs to be passed to the Telco. Is there a way to pass the BTN as a variable in the dial plan? Like CallerID(num)? What is the variable for BTN if so? Many Thanks. -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck@gmail.com

I have an interesting task for my son's lacrosse team... it is the time-old telephone tree... I am pretty sure someone has already done this w/*, why re-invent the wheel?... a) coach calls in leaves a msg, others call in retrieve the msg b) coach calls in leaves a msg, kicks of a call to every parent plays msg c) coach calls in leaves a msg, kicks off a call to every parent, checks for

Hello everybody. I have two AT530 telephones and one X-Lite extension conected to my Asterisk. This is part of my extensions.con. exten => 105,1,Answer exten => 105,2,Background(/home/user/suport) exten => 1,1,Dial(SIP/101,30,Ttm) exten => 2,1,Dial(SIP/102,30,Ttm) When I call to 105 extension from the AT530 telephones and I select option "1" it doesn't redirect to

Hi Lists, I have one box with two FXO and two FXS ports, it is running asterisk inside. I have one sinle POTS line connected to the one FXO and two phone sets connected to the FXS port. Extension 6003 is asigned to one fxs and 6004 is asigned to another fxs, the two extensions can call each other, they can both make/receive PSTN call, but they can't make PSTN call simultaneously. Is it

My wife really likes the fit and feel of my SPA-942. Anyone know of a POTS telephone with similar rugged construction?

Hello, I have some zaptel cards, and I would like to install them in some user's computers. Is there any way to connect those cards with asterisk server (which is in another computer)? All manuals I have read explain how to connect asterisk and zaptel cards in the same computers, but not on different ones. Thank you very much.