similar to: outgoing calls

Hi all, I registered a SIP account at budgetphone.nl/talkin2ya.nl Receiving calls works like a charm, I even redirected my normal PSTN number to the number I got from them so everything ends up in my * server. Before I ask them to take over my normal phone number I wanted to test all of it, so I ordered some calling minutes to test. Now I cannot get outbound calling to work with them. Anyone here

(this time with subject....) Hello, I?m trying to get Asterisk to accept incoming calls from budgetphone.nl. When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy tone. I tried X-lite, which worked perfect, so my modem (with nat) probably is not the problem. I did a sip debug and got the following output. Because I?m new to Asterisk I can?t get the error why this is not

Hi all! Quite a mystery. The following happened when I was on holiday, and no one else has changed any configs of either Asterisk or the Cisco's in the building... The situation: Incoming works fine on all phones. Outgoing only works from non-Cisco phones. When calling from a Cisco phone to an external phone, all the Cisco-user hears is a ticking crackle and after about a minute the phone

Hi, Asterisk is new for me. I had a working configuration, but suddenly I can't call anymore with my voip provider. I am not aware that I changed anything in the configuration, but who knows. Can somebody explain me what is happening here? I changed username, password and number. -- Executing Dial("Zap/2-1",

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

Is plainvoip down? I've tried to contact them via email and their 800-956-3285; nobody is answering or replying to emails -- #Joseph

I seem to be having a problem with transcoding and/or agreeing on a valid codec. I am running a new image pulled from CVS at 1:30 PM CST. The issue occurs when I try to make a call to a toll-free number over sipphone.com. Here's what I see in the console: NOTICE[1259545280]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from G729A to ULAW NOTICE[1259545280]: File

I have come to realize that I don't have to have a g729a license in order to make use of an ATA-186 or 7460 connecting to another 7460. I just need to allow the codec in sip.conf. Now what ramification does that have when I dial out over one of my analog line (connected to * by a channelbank and a T100P) using my 7460 or ATA-186. The only benefit I am looking for is reduced bandwidth

Hi, I read a lot about US providers that can terminate a PSTN number for you and offer IAX or SIP connectivity. Does anyone know such a company in The Netherlands ? I read about Unet. Anyone with experience with them ? Any information is welcome. -- Michiel van Baak http://lunteren.vanbaak.info michiel@vanbaak.info GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

Hi Friends, At present, I am using VoIPJET.COM provider for make calls to USA. I have two doubts. 1) I am unable to make call to UK Mobile phone. Why? 2) I want to make calls to "Turkey" country from "India". With VoIPJET, I am unable to make call to "Turkey" and unable to find VoIP provider for Turkey. Please tell me VoIP Provider for Turkey from India.

Hello all, This afternoon I had a BRI line installed by Telstra (our telco in Australia). I'm using an Eicon Diva 2.02 PCI ISDN card with the isdn4linux driver. Incoming and outgoing calls with Asterisk work fine (and with no echo - my main reason for getting ISDN). However, I can't seem to get outgoing DTMF working (incoming works fine). I made a call from my desk phone (Cisco 7940G)

Hi, guys. May be someone could give me advise? I am trying to find good VOIP provider ONLY for OUTGOING calls with low per channel cost and cheap rates on Eastern Europe, Turky and xUSSR. Should support g729 or g723 codecs, SIP or IAX connectivity. -- ========================================================================= = Best regards, Nikolay Pavlov.

Well, it has been a long time since I used my NetJet S cards with asterisk (moved to the quad PRI card) but I am trying to get this working again for home use. Basically, what I am using is this: Linux 2.6.8.1: config lines: # ISDN subsystem CONFIG_ISDN=y # Old ISDN4Linux CONFIG_ISDN_I4L=y # CONFIG_ISDN_PPP is not set CONFIG_ISDN_AUDIO=y # CONFIG_ISDN_TTY_FAX is not set CONFIG_ISDN_DRV_HISAX=y

Hello, I just got my isdn-card working together with i4l and asterisk. Everything seems to be working fine: I can accept calls coming from the outside and I can dial out. Even setting the msn works like charm but my problem is that I cannot hear a word. There's complete silence in both directions. Any idea what could be the cause? Thanks for your help, Gunther Lspci: 0000:01:07.0 Network

Hi. I'm working with i4l with asterisk CVS-02/21/03-13:59:12, plus i4l (chan modem i4l *dsp patched* and kernel 2.4.19 patched to disable dtmf). All seems ok (apart some echo issues that seems gone with mec2 aggressive suppressor), but outgoing dtmf doesn't work . or at least I hear the very first part of the dtmf, but then it seems suppressed. here's my modem.conf [interfaces]

I posted a message a little while ago but got no response (that I can recall), I've also seen other people mention this issue. Basically, when you have music on hold, it doesn't play the music on hold, the debug info shows it is starting and then stops straight away.. # My extensions.conf ... exten => s,1,Answer exten => s,2,DigitTimeout,5 exten => s,3,ResponseTimeout,10 exten

Hi to everyone! I am new to this list and i want to greet you all! I am trying to install Asterisk on a Fedora Core 3 Distribution with a Digi Datafire Micro V ISDN Card. I know that I have to use the ISDN4Linux so in modem.conf I set the I4l driver ,the msn and device. Modem.conf [interfaces] context=remote ;driver=aopen ; modem by AOpen driver=i4l ; isdn4linux - an

I've the following installation : |asterisk client| --- > |asterisk server| --- > |other asterisk server| all the connections are made in IAX, the client and first server allows 711 and 729 the other server only allows 729 since it has low bandwidth at disposal all the numbers but a few are routed to a digium card in the first server, the others are routed to the other server, this

Hey All, I'm still (since April) having problems getting RxFax to work over an ISDN4Linux channel. Just wondering if anyone has had any luck getting it to work? I have done a CVS update today (about half hour ago) and made sure I have the latest version of spandsp according to Steve's website (spandsp-0.0.1k). When I was compiling asterisk, I got the following warnings:

I receive the following error when I try to call another H323 extension from another H323 when going through *. NOTICE[27669]: File channel.c, Line 1325 (ast_set_read_format): Unable to find a path from 1 to 8 NOTICE[27669]: File channel.c, Line 1296 (ast_set_write_format): Unable to find a path from 8 to 1 WARNING[27669]: File chan_h323.c, Line 528 (oh323_write): Asked to transmit frame type 1,