Hi, I just got a SPA-2000 in and was finally able to complete my asterisk setup. I'm making my outgoing calls through iconnecthere from the asterisk server however I'm running into a problem when placing calls. I can connect fine but when the person (or answering machine) picks up I hear them talk for a about half a second then there is a half a second pause or muted period and then the audio comes back fine for the rest of the call. This is frustrating though cause in that time period is usually when people say hello when they pickup. It happens on every call and there isn't any error messages in the sip debug console. I think I've narrowed down the problem to some kind of issue with my setup between the asterisk server and iconnecthere. The reason I say that is this doesn't occur when I call fwd numbers or 1800 numbers through iaxtel but will occur on those same 1800 numbers through iconnecthere. Also the problem isn't my phone or SPA-2000 cause I get the same issue if I redirect a incoming fwd call (through ipkall) out through iconnecthere. Another interesting test is if I connect my SPA-2000 directly to iconnecthere without going through asterisk I don't have any problems and there is no pause. So the issue seems to be something with the way my asterisk setup is talking with iconnecthere. My sip.conf is setup to disallow=all, allow=gsm, allow=ulaw, allow=alaw, however it doesn't make a difference if I connect with either codec. Here is my iconnecthere section with user/pass removed: [iconnect] type=friend secretusernamehost=natrelay.deltathree.com dtmfmode=inband canreinvite=no All this seems pretty standard. What settings or debug information should I look at to fix this issue. Has anyone else encountered this problem before. I'd be happy to supply more information about my setup but since I'm not sure where the problem would be from I don't know what to post. Any help would be appreciated. Thanks. -- Justin <justin@sandershosting.com>
Are you actually behind a nat? I have * connecting to iconnecthere just fine. I have an external IP for my * server. Zac On Wed, 2004-04-28 at 14:13, Justin Sanders wrote:> Hi, I just got a SPA-2000 in and was finally able to complete my asterisk > setup. I'm making my outgoing calls through iconnecthere from the > asterisk server however I'm running into a problem when placing calls. I > can connect fine but when the person (or answering machine) picks up I > hear them talk for a about half a second then there is a half a second > pause or muted period and then the audio comes back fine for the rest of > the call. This is frustrating though cause in that time period is usually > when people say hello when they pickup. It happens on every call and > there isn't any error messages in the sip debug console. I think I've > narrowed down the problem to some kind of issue with my setup between the > asterisk server and iconnecthere. The reason I say that is this doesn't > occur when I call fwd numbers or 1800 numbers through iaxtel but will > occur on those same 1800 numbers through iconnecthere. Also the problem > isn't my phone or SPA-2000 cause I get the same issue if I redirect a > incoming fwd call (through ipkall) out through iconnecthere. Another > interesting test is if I connect my SPA-2000 directly to iconnecthere > without going through asterisk I don't have any problems and there is no > pause. So the issue seems to be something with the way my asterisk setup > is talking with iconnecthere. My sip.conf is setup to disallow=all, > allow=gsm, allow=ulaw, allow=alaw, however it doesn't make a difference if > I connect with either codec. Here is my iconnecthere section with > user/pass removed: > > [iconnect] > type=friend > secret> username> host=natrelay.deltathree.com > dtmfmode=inband > canreinvite=no > > All this seems pretty standard. What settings or debug information should > I look at to fix this issue. Has anyone else encountered this problem > before. I'd be happy to supply more information about my setup but since > I'm not sure where the problem would be from I don't know what to post. > Any help would be appreciated. Thanks. > > -- > Justin <justin@sandershosting.com> > > > _______________________________________________ > 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-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040428/19713d76/attachment.htm
Hi, Justin: I use IConnectHere with * without problems. I presume you have a registration line in your sip.conf? Didn't see it in what you have below. Something like the following (I've zapped the real numbers.) register=12225551234:1234@sipauth.deltathree.com/12225551234 Also, I have the following in my extensions.conf [macro-dialiconnect] exten => s,1,SetCallerID(${ICONNECT1}) exten => s,2,SetCIDName(${MYNAME}) exten => s,3,Background(dial-iconnect) exten => s,4,Dial(SIP/${ARG1}@iconnect,${ARG2},r) [iconnect-forced] exten => _61XXXXXXXXXX,1,Macro(dialiconnect,${EXTEN:1},20) exten => _61XXXXXXXXXX,2,Playback(vm-goodbye) exten => _61XXXXXXXXXX,3,Hangup Good luck! John Vogel -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Justin Sanders Sent: Wednesday, April 28, 2004 12:14 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk and Iconnecthere pause Hi, I just got a SPA-2000 in and was finally able to complete my asterisk setup. I'm making my outgoing calls through iconnecthere from the asterisk server however I'm running into a problem when placing calls. I can connect fine but when the person (or answering machine) picks up I hear them talk for a about half a second then there is a half a second pause or muted period and then the audio comes back fine for the rest of the call. This is frustrating though cause in that time period is usually when people say hello when they pickup. It happens on every call and there isn't any error messages in the sip debug console. I think I've narrowed down the problem to some kind of issue with my setup between the asterisk server and iconnecthere. The reason I say that is this doesn't occur when I call fwd numbers or 1800 numbers through iaxtel but will occur on those same 1800 numbers through iconnecthere. Also the problem isn't my phone or SPA-2000 cause I get the same issue if I redirect a incoming fwd call (through ipkall) out through iconnecthere. Another interesting test is if I connect my SPA-2000 directly to iconnecthere without going through asterisk I don't have any problems and there is no pause. So the issue seems to be something with the way my asterisk setup is talking with iconnecthere. My sip.conf is setup to disallow=all, allow=gsm, allow=ulaw, allow=alaw, however it doesn't make a difference if I connect with either codec. Here is my iconnecthere section with user/pass removed: [iconnect] type=friend secretusernamehost=natrelay.deltathree.com dtmfmode=inband canreinvite=no All this seems pretty standard. What settings or debug information should I look at to fix this issue. Has anyone else encountered this problem before. I'd be happy to supply more information about my setup but since I'm not sure where the problem would be from I don't know what to post. Any help would be appreciated. Thanks. -- Justin <justin@sandershosting.com> _______________________________________________ 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