search for: voipprovid

...he external ip address. I have FORWARD to ACCEPT in the router and I still cant establish a connection. My sip.conf file looks like this: [general] externhost=optimumwireless.com localnet=172.16.0.0/16 register => username:secret at my.service_provider.tld language=es ;allow=gsm allow=all [voipprovider] type=friend host=208.78.163.3 username=username fromuser=username secret=password port=5060 dtmfmode=rfc2833 nat=yes insucure=port,invite allow=all careinvite=yes I don't know what else to try. When I try to call I get this at the cli: == Using SIP RTP CoS mark 5 -- Executing [91xxx763xxx...

...ot;test" server, we've been able to put through a call w/o registration, so it seems some of this can be compatible. I'm wondering if I can use "md5secret" with a "register => " statement. The current busted config: [general] ;register => userid:pass:acctid@voipprovider.com:5069 [myipsolution] type=friend authuser=acctid username=userid secret=pass md5secret=XXXMD5HASH of userid:asterisk:pass XXXXX nat=yes host=voipprovider.com port=5069 insecure=very canreinvite=no The error on the console is: Jul 16 11:29:20 NOTICE[3361]: -- Registration for 'userid@v...

...external include => parkedcalls switch => DUNDi/dundified exten => 300,1,Dial(SIP/300) exten => 300,n,Hangup() exten => 5551234567,1,Goto(300,1) exten => 301,1,Dial(SIP/301) exten => 301,n,Hangup() exten => 8885551212,1,Goto(301,1) exten => _NXXNXXXXXX,1,Dial({$EXTEN}@voipprovider) exten => _NXXNXXXXXX,n,Hangup() [external] exten => 5551234567,1,Goto(internal,300,1) ----------------------------------- sip.conf [dundified] type=friend dbsecret=dundi/secret context=internal [voipprovider] type=friend host=voipprovider.web dtmfmode=rfc2833 insecure=port,invite disal...

...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 same : SIP (SNOM) -> Asterisk -> VoIPProvider -> ISDN/POTS Can i control the cancellation with the zapata.conf ? I have tried this with "echocancel=..." and so on, no luck :( I would be glad to get some help, the Docs of Asterisk dont explain how to cancel Echos in ! SIP !

...1122332. In this case works fine. How i can solve this problem ?? Bellow i list my extension.conf [default] ignorepat=>0 ### Internal Calls ## Input Calls exten=> 312120XX,1,Dial(Zap/g1/${EXTEN:-4}) exten=> 312120XX,2,Hangup ### External Calls exten=> _XXXXXXXX,1,Dial(SIP/${EXTEN}@voipprovider,60,Tt) # Local calls exten=> _XXXXXXXX,2,Hangup exten=> _0XXXXXXXXXX,1,Dial(SIP/${EXTEN}@voipprovider,60,Tt) # Long distance Calls exten=> _0XXXXXXXXXX,2Hangup exten=> _00XXXXXXXXXX,1,Dial(SIP/${EXTEN}@voipprovider,60,Tt) # Internacional Calls exten=> _00XXXXXXXXXX,2Hangup Thank...

...s") in new stack But cdr-csv/Master.csv has logged time values for duration and billsec: "","5105550000","+4155550001","pop-inbound","""15105550000"" <5105550000>","SIP/10.10.10.170-b7d94f78","SIP/ voipprovider.com-089ae8a0","Dial","SIP/15105550000:password::authname at voipprovider.com ,,M(post-dial)","2008-10-09 20:59:00","2008-10-09 20:59:03","2008-10-09 20:59:08", 8,5,"ANSWERED","DOCUMENTATION","1223585940.35"...

...going sip proxy but it failing with subject error. Can any one please correct me what i am doing wrong in below config. SIP.conf [Inbound] type=peer context=introuting host=184.107.XXX.XXX disallow=all allow=all [astinside] type=peer context=introutingB host=localhost disallow=all ;allow=speex [voipprovider] type=peer host=XXX.X.XXX.159 disallow=all allow=g723:120 Extentions.config [introuting] exten => _X.,1,Dial(SIP/astinside) [introutingB] exten => s,n,Dial(SIP/voipprovider) I will appreciate for your help. -------------- next part -------------- An HTML attachment was scrubbed... URL:...

...ge----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Gonzalo Servat Sent: Friday, March 21, 2008 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ----www.cdsportal.net---- wholesale voipprovider --starting at 1.1 cent per min I think this type of abuse is well deserved due to the way he intended to advertise his "business", so I'll add a bit of wood to the fire. How about the sign-up form?? Some serious HTML design work going on there. - Gonzalo On Fri, Mar 21, 2008 at...

...led immediately, for example "011862", when I actually ment to dial 0118620xxxxxxxx. Thus left the remaining numbers "0xxxxxxxx" unsent. The handset had its dial plan disabled. It configured to use iax protocol. My extensions.conf has this: exten=_01186.,1, dial(SIP/<voipprovider>,60) and it works fine with other iaxy and Cisco ATA. Anyone encounter this symptom? Can you share your experience? Thanks, Min

Hi, I am attempting to make about ten calls simultaneously and intermittently get 'SIP/voipprovider is circuit-busy' followed by 'everyone is busy/congested at this time" I am not sure if this is related to my bandwidth to my voip provider, a configuration issue or something else. Has anyone seen this before and have any suggestions. Thanks in advance. -------------- next...

...gt; setup IAX2 channel to remote * server (which then rings extension) Pickup call on extension on remote * server --> main server sip extension stops ringing --> ast console on main server I get : ------------------------------------------------- -- Attempting native bridge of IAX2/voipprovider/6 and IAX2/remote*server/7 -- Channel 'IAX2/voipprovider/6' unable to transfer -- Channel 'IAX2/remote*server/7' unable to transfer ------------------------------------------------- In the user/friend declarations (user for incoming voip provider, friend for remote * serv...

Hi people, I would like to read your suggestions as to where the issue might be. ATA286=Grandstream Budgetone ATA 286 in SIP mode. One Lan Port and one FXS port. TDM04B= 4 FXO signal fxls There is a 8FXO-to-SIP unit in this scenario that works perfectly so i will not make mention of it. PSTN----VOIPprovider---Internet---ATA286------tdm04b---Asterisk1.2.-13 Asterisk is being used as a meetme server for 8 more calls. Everything works fine in terms of the asterisk/meetme. The issue arises when the calls comes in via the ATA286 box and in any part of the meeting the CALLER hangs up but the ata286 does...

Hi list, My Asterisk config for outgoing calls is the following: exten => s,1,Dial(SIP/${MACRO_EXTEN}@voipprovider,60,g) exten => s,n,GotoIf($[\"${ANSWEREDTIME}\" = \"\"]?pri:hang) exten => s,n(pri),NoOp(Problems with voip provider trying PRI) exten => s,n,Dial(Zap/g2/${MACRO_EXTEN},60,g) exten => s,n(hang),HangUp in most cases it works well but, if my internet connection is...

Hello. >> ubuntu*CLI> sip show peers >> Name/username Host Dyn Nat ACL Port Status >> voipprovider xxx.xxx.xxx.xxx A 5060 Unmonitored I've ben trying to connect an asterisk server to a voip provider, and I'm currently wondering what the 'A' in the ACL field of the 'sip show peers' command might be. I've been unable to find this inform...

...---- From: "Doug Crompton" <doug@crompton.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Monday, February 12, 2007 10:33 PM Subject: [asterisk-users] Using Asterisk/callerid with "pay as you go" VOIPproviders >I am curious how others handle "call out" VOIP and callerid. I have found > numerous providers that are cheap and seem to have good voice quality but > offer no provisions for callerid. I find it almost useless to use call > out when the receiving party gets some bogus...

...calls. Outgoing audio works fine, incoming audio is not working. My setup is the following : incoming calls : PSTN -- FXOport -- HT503 -- WANport -- Asterisk -- WANport -- HT503 (the same) -- FXSport -- DECTphone outgoing calls : DECTphone -- FXSport -- HT503 -- WAN-port -- Asterisk -- internet (VoIPprovider) I've done a TCPdump on the Asterisk-server : 18:20:21.189504 IP vds.server.net.11574 > my.asterisk.server.be.20056: UDP, length 172 18:20:21.193065 IP my.asterisk.server.be.20056 > vds.server.net.11574: UDP, length 172 18:20:21.210111 IP vds.server.net.11574 > my.asterisk.server.b...