Hi, I am trying to forward calls to another * server with IAX Here is What I want to Do 1- Call SERVER1, let say at 51412345678 2- SERVER1 should transfer the call to SERVER2 in a remote location 3- SERVER2 Receive the call and transfer it to the PSTN number. I have one X100P card on each machine. What is happening is that when the remote party picks up the phone, all he can hear is a weird sound. CONFIGS: SERVER1: zaptel.conf --------------------------------------------- ~ [channels] ~ language=fr ~ context=montr?al ~ signalling=fxs_ks ~ usercallerid=yes ~ callwaiting=yes ~ threewaycalling=yes ~ transfer=yes ~ cancellforward=yes ~ echocancel=yes ~ echocancelwhenbridged=yes ~ echotraining=yes ~ relaxdtmf=yes ~ busydetect=yes ~ busycount=4 ~ callprogress=yes ~ group=1 ~ channel=>1 -------------------------------------------------- (same for SERVER2) IAX.conf ------------------------------------------------ ~ [general] ~ bindport=4569 ~ delayreject=yes ~ language=fr ~ allow=all ~ jutterbuffer=no ~ register => username:password@server2.domain.com ~ tos=lowdelay ~ autokill=yes ~ ~ [quebec] ~ type=friends ~ username = username ~ password=password ~ context=montr?al ~ host=Dynamic ~ secret = password ~ disallow = all ~ allow=ulaw ~ allow=gsm extensions.conf ------------------------------------------(Same for SERVER2 but no registration) ~ [general] ~ static=yes ~ writeprotect=yes ~ autofallthrough=yes ~ [montr?al] ~ exten=>s,1,Answer ~ exten=>s,2,Playback(message-transfer) ~ exten=>s,3,Dial(IAX2/username:password@SERVER2.DOMAIN.COM/51412345678@montr?al) ; always the same number ~ exten=>s,4,Hangup My remote server receive the call, answer the line and then Dial(ZAP/1/51412345678). So far so good. But when 51412345678 pickup the phone, all she can hear is a weird sound. What am I doing wrong ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050206/ede56e17/attachment.htm
> I am trying to forward calls to another * server with IAX > Here is What I want to Do > 1- Call SERVER1, let say at 51412345678 > 2- SERVER1 should transfer the call to SERVER2 in a remote location > 3- SERVER2 Receive the call and transfer it to the PSTN number. > > I have one X100P card on each machine. What is happening is that when the remote party picksup the phone, all he can hear> is a weird sound. > > CONFIGS: > > SERVER1: > zaptel.conf > --------------------------------------------- > ~ [channels] > ~ language=fr > ~ context=montr?al > ~ signalling=fxs_ks > ~ usercallerid=yes > ~ callwaiting=yes > ~ threewaycalling=yes > ~ transfer=yes > ~ cancellforward=yes > ~ echocancel=yes > ~ echocancelwhenbridged=yes > ~ echotraining=yes > ~ relaxdtmf=yes > ~ busydetect=yes > ~ busycount=4 > ~ callprogress=yes > ~ group=1 > ~ channel=>1 > -------------------------------------------------- (same for SERVER2) > > IAX.conf > ------------------------------------------------ > ~ [general] > ~ bindport=4569 > ~ delayreject=yes > ~ language=fr > ~ allow=all > ~ jutterbuffer=no > ~ register => username:password@server2.domain.com > ~ tos=lowdelay > ~ autokill=yes > ~ > ~ [quebec] > ~ type=friends > ~ username = username > ~ password=password > ~ context=montr?al > ~ host=Dynamic > ~ secret = password > ~ disallow = all > ~ allow=ulaw > ~ allow=gsm > > extensions.conf > ------------------------------------------(Same for SERVER2 but no registration) > ~ [general] > ~ static=yes > ~ writeprotect=yes > ~ autofallthrough=yes > ~ [montr?al] > ~ exten=>s,1,Answer > ~ exten=>s,2,Playback(message-transfer) > ~ exten=>s,3,Dial(IAX2/username:password@SERVER2.DOMAIN.COM/51412345678@montr?al) ; alwaysthe same number> ~ exten=>s,4,Hangup > > > > My remote server receive the call, answer the line and then Dial(ZAP/1/51412345678). So far sogood. But when 51412345678 pickup the phone,> all she can hear is a weird sound. > What am I doing wrong ?Difficult to tell without some feedback from the CLI. If you actually copy/pasted the above config statements, I'm assuming you manually added all those "~" at the front of each line. If they are actually in your config, get rid of them. The statement "jutterbuffer=no" should be jitterbuffer=no. One thing you might try to at least eliminate possible problems is to change iax.conf to disallow=all and allow=gsm only. Get rid of the allow=ulaw and do another test. Might as well add trunk=no to this link as well. (Must stop and restart * after making these type changes.) You might try 'iax2 debug' from the CLI on both machines and look at the detail to see if you can spot any conflicts or problems.
One thing I do on remote sites is set up a soft phone so I can call myself, this proves out the link and quality before anything else. DIAX id good for this as you can connect to multiple sites, also good to see if you have problems before anyone else calls you to say there is a problem. It also helps in cases like this, if your return quality is good then the possible fault lies with the ZAP interface. Process of elimination, works for me every time. Regards Dave -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Ousmane Doukara Sent: 06 February 2005 08:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] inter asterisk Hi, I am trying to forward calls to another * server with IAX Here is What I want to Do 1- Call SERVER1, let say at 51412345678 2- SERVER1 should transfer the call to SERVER2 in a remote location 3- SERVER2 Receive the call and transfer it to the PSTN number. I have one X100P card on each machine. What is happening is that when the remote party picks up the phone, all he can hear is a weird sound. CONFIGS: SERVER1: zaptel.conf --------------------------------------------- ~ [channels] ~ language=fr ~ context=montr?al ~ signalling=fxs_ks ~ usercallerid=yes ~ callwaiting=yes ~ threewaycalling=yes ~ transfer=yes ~ cancellforward=yes ~ echocancel=yes ~ echocancelwhenbridged=yes ~ echotraining=yes ~ relaxdtmf=yes ~ busydetect=yes ~ busycount=4 ~ callprogress=yes ~ group=1 ~ channel=>1 -------------------------------------------------- (same for SERVER2) IAX.conf ------------------------------------------------ ~ [general] ~ bindport=4569 ~ delayreject=yes ~ language=fr ~ allow=all ~ jutterbuffer=no ~ register => username:password@server2.domain.com ~ tos=lowdelay ~ autokill=yes ~ ~ [quebec] ~ type=friends ~ username = username ~ password=password ~ context=montr?al ~ host=Dynamic ~ secret = password ~ disallow = all ~ allow=ulaw ~ allow=gsm extensions.conf ------------------------------------------(Same for SERVER2 but no registration) ~ [general] ~ static=yes ~ writeprotect=yes ~ autofallthrough=yes ~ [montr?al] ~ exten=>s,1,Answer ~ exten=>s,2,Playback(message-transfer) ~ exten=>s,3,Dial(IAX2/username:password@SERVER2.DOMAIN.COM/51412345678@montr? al) ; always the same number ~ exten=>s,4,Hangup My remote server receive the call, answer the line and then Dial(ZAP/1/51412345678). So far so good. But when 51412345678 pickup the phone, all she can hear is a weird sound. What am I doing wrong ?
I think it has to do with my ZAP interface. Before my DIAL(ZAP/1/51412345678) I have a Playback(message-transfert) which play nicely. As soon as the ZAP start ringing the PSTN phone, i have that helicopter sound. ----- Original Message ----- From: "David J Carter" <david.carter@codepipe.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Sunday, February 06, 2005 7:18 AM Subject: RE: [Asterisk-Users] inter asterisk One thing I do on remote sites is set up a soft phone so I can call myself, this proves out the link and quality before anything else. DIAX id good for this as you can connect to multiple sites, also good to see if you have problems before anyone else calls you to say there is a problem. It also helps in cases like this, if your return quality is good then the possible fault lies with the ZAP interface. Process of elimination, works for me every time. Regards Dave -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Ousmane Doukara Sent: 06 February 2005 08:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] inter asterisk Hi, I am trying to forward calls to another * server with IAX Here is What I want to Do 1- Call SERVER1, let say at 51412345678 2- SERVER1 should transfer the call to SERVER2 in a remote location 3- SERVER2 Receive the call and transfer it to the PSTN number. I have one X100P card on each machine. What is happening is that when the remote party picks up the phone, all he can hear is a weird sound. CONFIGS: SERVER1: zaptel.conf --------------------------------------------- ~ [channels] ~ language=fr ~ context=montr?al ~ signalling=fxs_ks ~ usercallerid=yes ~ callwaiting=yes ~ threewaycalling=yes ~ transfer=yes ~ cancellforward=yes ~ echocancel=yes ~ echocancelwhenbridged=yes ~ echotraining=yes ~ relaxdtmf=yes ~ busydetect=yes ~ busycount=4 ~ callprogress=yes ~ group=1 ~ channel=>1 -------------------------------------------------- (same for SERVER2) IAX.conf ------------------------------------------------ ~ [general] ~ bindport=4569 ~ delayreject=yes ~ language=fr ~ allow=all ~ jutterbuffer=no ~ register => username:password@server2.domain.com ~ tos=lowdelay ~ autokill=yes ~ ~ [quebec] ~ type=friends ~ username = username ~ password=password ~ context=montr?al ~ host=Dynamic ~ secret = password ~ disallow = all ~ allow=ulaw ~ allow=gsm extensions.conf ------------------------------------------(Same for SERVER2 but no registration) ~ [general] ~ static=yes ~ writeprotect=yes ~ autofallthrough=yes ~ [montr?al] ~ exten=>s,1,Answer ~ exten=>s,2,Playback(message-transfer) ~ exten=>s,3,Dial(IAX2/username:password@SERVER2.DOMAIN.COM/51412345678@montr? al) ; always the same number ~ exten=>s,4,Hangup My remote server receive the call, answer the line and then Dial(ZAP/1/51412345678). So far so good. But when 51412345678 pickup the phone, all she can hear is a weird sound. What am I doing wrong ? _______________________________________________ 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