similar to: Asterisk and Sipgate problem...

Anyone using Simpletelecom with *? I had a nice working system with them, my credit can out so I apply another $5 to continue testing. Since then nothing has worked. I always get: -- Executing SetCallerID("SIP/line1-74ac", ""myname"|<>|a") in new stack -- Executing Dial("SIP/line1-74ac",

I'm having a very annoying problem with access my asterisk system from work. Our phone system here only produces very very short DTMF tones. The phones work fine for other IVR systems (Dell Support, HP Support, etc, etc). However, tones to Asterisk just never make it. The way I'm calling into my Asterisk server is such: OFFICE PHONE => CALLUK.COM 0870 => IAX Inbound The

I'm working on an IVR that gives the users the option (number 5 in the main menu) to dial by extension: exten => 5,1,Set(TIMEOUT(digit)=5) ; Dial Extension exten => 5,2,Set(TIMEOUT(response)=10) exten => 5,3,Background(LCL/prompt-60) exten => 5,4,WaitExten(15) When going option 5 you can dial some extensions such as 2802, it goes to the extension (all extens start with 28 on the

Hello all. I'm having a problem debugging an IVR I'm building. I can't see any reason this shouldn't be working. Firstly the asterisk version is: Asterisk SVN-trunk-r7230 built by root @ localhost.localdomain on a i686 running Linux on 2006-02-17 22:44:48 UTC Basically the problem is this. While the playbacks are happening you can push any one of the options and to happily

I've noticed that my * box isn't logging all that it use to / should. I'm running version Asterisk CVS-v1-0-03/07/05-22:42:03, prior versions would log everything including connections to voicemail and such. This version of (or more likely my configs) seems only to be logging certain things. It logs most calls to both Master.csv and MySQL, but there are still loads of calls

Does anyone here use either Gradewell or inWeb for service? They are both UK based. I'm trying to get a couple of inbound IAX2 based numbers from both of them to work with no luck at all. The one thing that sets these guys apart from the rest of companies offering inbound numbers is they tie the account to the IP of the asterisk server. Neither use register lines in iax.conf, there appears

Hello, I'm having a problem I can't seen to figure out. In a nut shell, I have asterisk running with 4 accounts configured. All accounts work fine for local calling to each other and voicemail. However, only 1 account can make outgoing calls. All the others fail with the following error. If anyone can shed some light on the possible problem or where to look for more info it

I just setup an Asterisk system on a small Shuttle box; I am only using SIP channels and have no FXO/FXS cards. The system works fine in that I can call my inbound number (Broadvoice) and have the system answer and I can make outgoing calls. The problem is that every time I want to change something in the sip.conf file, I have to do a 'restart now' instead of a 'reload' or

I can not register via sip to iptel or sipgate and do not see sip into ethereal. I do not unterstand why thats Wudu .. but i am new to asterisk and sip. I am behind a susefirewall2 but asterisk even do not register if it is down. The asterisk is running onto the machine witch is connected to the internet. No answer seems coming back from iptel (sip debug in asterisk). Ports are open (5060,

Hi Does anyone know a UK Voip Proivder that will give me more than 1 telephone number and point it to my sip account. www.SipGate.co.uk are great but they only allow 1 telephone number per user, you can register another telephone number by registering as another user but Asterisk doesn't allow multiple registrations. Many Thanks Scott

Hi, have some problem with incoming calls from sipgate. This was working in 1.4 but in 1.6 I get a 401 Unauthorized :-(. Sipgate has mentioned that I have to change the type to friend, but it is already friend, so what's wrong? Kind regards, Michael Here is the sip.conf: [sipgate_out] type=friend nat=yes username=1234567 fromuser=1234567 fromdomain=sipgate.de secret=secret host=sipgate.de

Hello i am trying to setup sipgate gateway i can get incoming calls fine, but when i dial in and then try to dial out i get this in asterisk command line -- Called 01179248615 at sipgate [Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885 handle_response_invite: Failed to authenticate on INVITE to '"01179553708" <sip:SIP-ID at sipgate.co.uk>;tag=as30eb9dd1' --

Hello. My setup: _ one Samba 3.5 domain (XXXXXXXX), with a PDC and a BDC, both running FreeBSD; _ one AD domain (YYYYYYYY) running on two Windows 2003 DCs; _ bidirectional trust between the two domains. Everything used to work until I moved the PDC from Samba 3.5 (EOL'ed) to 3.6; now, users from domain YYYYYYYY cannot access the PDC's shares. I used to have in smb.conf: >

Hi, Anyone here using sipgate.de ? It worked for months, but for a couple of days now I'm unable to register with them. My account is ok, because I can login to the website. Asterisk keeps showing me: Feb 25 23:50:18 NOTICE[5144]: chan_sip.c:5269 sip_reg_timeout: -- Registration for 'XXXXX@sipgate.de' timed out, trying again (Attempt #n) I looked at the sip debug stuff, and all I

Hi, Using a couple of sip phones and using asterisk to connect them to a single sipgate.de account. if I call a mobile I have no problem makeing conversions. If the mobile rejects the call (by pressing hangup while it rings), something strange happens: the following is seen in the logfile, everytime a rejected mobile call happens: ----------------- Mar 20 22:52:29 WARNING[4682]: Forbidden

Hello, I've been looking at the DialPlans by some poeple using Asterisk with SipGate, but the new Asterisk@home 1.0 allows you to create Outbound routes etc, does using the web admin give the same effects? When I add a SIP Trunk with my Sipgate settings and use a pattern of "8|." to place all calls with a 8 prefix tot he sipgate account the softphones dial the number, the Asterisk

Hi all, I have two sipgate accounts (numbers), if I have both accounts register only one will work for incoming calls (which is all i'm interested in). However if I disable either account the other account will work perfectly. Am I missing something obvious? Thanks in advance, Ray. Excerpts from sip.conf - [general] 8<---- SNIP! ---->8 Register => 1212121:aaaaaaaa at

All, I've been experimenting with shortcodes for SIPgate etc. Passing calls to SIPbroker seems a good way to go, but the message I've had back from SIPgate is "we don't support SIPBroker"... So whats the easiest way to support SIP <> SIP network calling? At the moment, I've setup some local shortcodes (eg dial **777. to goto sipgate.co.uk) based on what Gradwell

hi anybody running with german SIPGATE? my configuration don't works :-( regards thorsten@gehrig.de

I could call a regular phone through sipgate. Now i can not: Failed to authenticate on INVITE to '"xyz" <sip:4xxxxxx@sipgate.net>;tag=as4ddd4a6f' A call from outside to my sip-phone through sipgate is OK. Can anyone verify ? Is it a sipgate problem ? greetings nicolas