search for: sipdiscount

Sipdiscount has replaced their asterisk servers for another thing. Then, no more iax. Ok, but I can't make calls using sip also... I'm getting a "forbidden" error when using sip1.sipdiscount.com. Anybody got it working? -- Alejandro Vargas

Hi Someone have luck using Sipdiscount service with IAX ? I only can use sipdiscount IAX service using a free account (only 1 minute call) , I have a normal account and with it can login in the IAX server. I using sip1.sipdiscount.com like IAX server but can make free calls (less 1 minute). Thanks in advance. roberto -- Ing. Rober...

Trying to get SIP realtime working here... I'm connected to the database... *CLI> realtime mysql status Connected to vox180internal@db1.ipt.XXX.com, port 3306 with username voxadmin for 6 seconds. I can get information for the extension in question... *CLI> realtime load sipusers name 2944093 Column Name Column Value

Can you also post information such as: Type of phone (model Number would be idela) How is it conencted, SIP, ZAP, IAX, Channel Bank. Corresponding config files would also help. Help us help you. >>-----Original Message----- >>From: asterisk-users-bounces@lists.digium.com >>[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of >>Paul A Brown >>Sent:

I have a strange problem in that I have put a budgetone out on the internet that connects to my * server that's behind a firewall. They can call me I can call them and it works fine. However, I have setup a link to sipdiscount on my * server. If the budgetone user calls via my * box to sipdiscount all the budgetone user hears is silence and the called person hears silence as well when they pick up the phone. If the budgetone user then hits the hold button then the called party hears the music on hold and then when the...

Hi list, I got a question: When I try to ChanSpy a SIP channel I only listen one channel, for example, I call from 302 extension and I have two active channels: SIP/r1-voip-1b7b (None) Up Bridged Call(SIP/302-f1f1) SIP/302-f1f1 09143213452@prueba-sip- Up Dial(SIP/09143213452@r1-voip|4 When I try to spy this call from another extension:

...ebf0991d3" Contact: <sip:987073366@192.168.0.20:5061> User-Agent: Sipura/SPA3000-3.1.7(GWg) Content-Length: 0 --- (11 headers 0 lines)--- Destroying call 'a459834e-816b2cbd@192.168.0.20' 12 headers, 0 lines Reliably Transmitting (no NAT) to 80.239.235.200:5060: OPTIONS sip:sip1.sipdiscount.com SIP/2.0 Via: SIP/2.0/UDP 81.172.52.3:5060;branch=z9hG4bK0596a5f2;rport From: "Unknown" <sip:Unknown@81.172.52.3>;tag=as6c5807a2 To: <sip:sip1.sipdiscount.com> Contact: <sip:Unknown@81.172.52.3> Call-ID: 43c17608446ba56231766bb82c8e350e@81.172.52.3 CSeq: 102 OPTIONS U...

I am trying to move from IAX2 to SIP for voipbuster, moving at the same time to sip1.voipbuster.com. When I try calling out, I see that there is SIP exchange, and in many cases also RTP data being exchanged. Hover in a very large number of attempts the connection is not established. Half of the time there is no RTP, the rest of the time there *is* RTP data flowing in two ways, but no ringtone is

...the Dial help says) Without options when Dial ends the call, dialplan ends too, meanwhile Dial mantains the control. Now, I've noticed that if I run the application in verbose mode, there are events wich show when I pick up and hung up the phone. When I pick up, the event is: -- SIP/sipdiscount-0d27 answered SIP/localuser-da9c (event that appears in app_dial.c) When I hung up, for example: == Spawn extension (default, 4233398, 3) exited non-zero on 'SIP/localuser-da9c' (but this is not happening in app_dial.c, I've seen its happening on pbx.c) So, what I'm t...

Simple as that please email me direct. voipviews@gmail.com Also looking for a U.S. DID provider as well as orig provider.

Hi all, I just set up a small asterisk box at home and it works as expected, I have some relatives in USA, I usually use skype to speak to them, is there anyway to connect skype and asterisk. I'd like to use skpe as an extension or a channel with asterisk. Thanks in advance for any suggestion. adriano. -------------- next part -------------- An HTML attachment was scrubbed... URL:

what happens when you drop your gains? use /etc/asterisk/zaptel.conf and fiddle with tx and rx values. Works, most of the time. -----Original Message----- From: John McEntee [mailto:john@mcentee87.freeserve.co.uk] Sent: Wednesday, October 11, 2006 12:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Echo problems on ISDN. (mainly incoming calls) OK I