Dear Sirs,
Anybody ever tried running SIP up against Voicepulse? On their
http://connect.voicepulse.com they claim they support both SIP and IAX, but
I can't seem to get SIP running. I have as mentioned before on this list -
huge problems getting any timing devices running on some of my machines, so
IAX is not really an option right now. If I try I get a "Service
Unavailable" back from gw5.voicepulse.com. If I try IAX2 with the same
settings, the call goes through - but sound is horrible.
Regards,
Lars...
--
Lars Boegild Thomsen
Technical Director
JustIT Sdn. Bhd.
Cell Phone (MY): +60 (16) 323 1999
ICQ: 6478559
Yahoo Chat: lars_boegild_thomsen@yahoo.com
MSN Chat: lars_boegild_thomsen@hotmail.com
http://www.justit.ws
Phone: +1 (360) 515 3551 (US) +45 8692 1951 (DK) +60 (3) 2057 2646 (MY)
Fax : +60 (3) 2057 2647 (MY)
SIP used to work fine with VoicePulse. But the funny thing is I could never detect any signs that they were doing call accounting. I could make IAX calls and see them show up in the CDR and the $$ deducted from my account balance. But when I made SIP calls they appeared, by all measures, to be free. I wrote to their support department several times about this and never received a response. But that was pretty much par for the course with those guys so I moved on to another provider. -brian Lars Boegild Thomsen wrote:>Dear Sirs, > >Anybody ever tried running SIP up against Voicepulse? On their >http://connect.voicepulse.com they claim they support both SIP and IAX, but >I can't seem to get SIP running. I have as mentioned before on this list - >huge problems getting any timing devices running on some of my machines, so >IAX is not really an option right now. If I try I get a "Service >Unavailable" back from gw5.voicepulse.com. If I try IAX2 with the same >settings, the call goes through - but sound is horrible. > >Regards, > > Lars... > >-- >Lars Boegild Thomsen >Technical Director >JustIT Sdn. Bhd. >Cell Phone (MY): +60 (16) 323 1999 >ICQ: 6478559 >Yahoo Chat: lars_boegild_thomsen@yahoo.com >MSN Chat: lars_boegild_thomsen@hotmail.com >http://www.justit.ws >Phone: +1 (360) 515 3551 (US) +45 8692 1951 (DK) +60 (3) 2057 2646 (MY) >Fax : +60 (3) 2057 2647 (MY) > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >
Lars Boegild Thomsen wrote:>Dear Sirs, > >Anybody ever tried running SIP up against Voicepulse? On their >http://connect.voicepulse.com they claim they support both SIP and IAX, but >I can't seem to get SIP running. I have as mentioned before on this list - >huge problems getting any timing devices running on some of my machines, so >IAX is not really an option right now. If I try I get a "Service >Unavailable" back from gw5.voicepulse.com. If I try IAX2 with the same >settings, the call goes through - but sound is horrible. > >Welcome to Voicepulse and their lack of jitter buffer. This is the cause of your horrible sound. Will be just as bad with SIP.>Regards, > > Lars... > >-- >Lars Boegild Thomsen >Technical Director >JustIT Sdn. Bhd. >Cell Phone (MY): +60 (16) 323 1999 >ICQ: 6478559 >Yahoo Chat: lars_boegild_thomsen@yahoo.com >MSN Chat: lars_boegild_thomsen@hotmail.com >http://www.justit.ws >Phone: +1 (360) 515 3551 (US) +45 8692 1951 (DK) +60 (3) 2057 2646 (MY) >Fax : +60 (3) 2057 2647 (MY) > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >-- Andres Network Admin http://www.telesip.net "Providing Wholesale Florida SIP/IAX2 Termination for US$0.01/minute"
Lars, I could be quite wrong, but I think you only need a 'timing' source if you want to use trunking over IAX. You can still use IAX without trunking if you don't have any sort of timing device. -Chris On 06:39 PM 5/21/2004, Lars Boegild Thomsen wrote: >Dear Sirs, > >Anybody ever tried running SIP up against Voicepulse? On their >http://connect.voicepulse.com they claim they support both SIP and IAX, but >I can't seem to get SIP running. I have as mentioned before on this list - >huge problems getting any timing devices running on some of my machines, so >IAX is not really an option right now. If I try I get a "Service >Unavailable" back from gw5.voicepulse.com. If I try IAX2 with the same >settings, the call goes through - but sound is horrible. > >Regards, > > Lars... > >-- >Lars Boegild Thomsen >Technical Director >JustIT Sdn. Bhd. >Cell Phone (MY): +60 (16) 323 1999 >ICQ: 6478559 >Yahoo Chat: lars_boegild_thomsen@yahoo.com >MSN Chat: lars_boegild_thomsen@hotmail.com >http://www.justit.ws >Phone: +1 (360) 515 3551 (US) +45 8692 1951 (DK) +60 (3) 2057 2646 (MY) >Fax : +60 (3) 2057 2647 (MY) > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
> Welcome to Voicepulse and their lack of jitter buffer. This is the > cause of your horrible sound. Will be just as bad with SIP.Which providers give you a jitter buffer?
jparr@bgcfreedom.com wrote:>>Welcome to Voicepulse and their lack of jitter buffer. This is the >>cause of your horrible sound. Will be just as bad with SIP. >> >> > >Which providers give you a jitter buffer? > > >In Europe: VoipTalk and Magrathea. In the US: Iconnecthere. I am sure there are more.>_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >
Andres wrote:> jparr@bgcfreedom.com wrote: >> Which providers give you a jitter buffer? >> >> >> > In Europe: VoipTalk and Magrathea. In the US: Iconnecthere. I am sure > there are more. >Clearpath gives jitter buffer as well. http://www.clearpath1.com/ John