woodpecker Posted May 17, 2006 Share Posted May 17, 2006 Hi folks, I know a few of you have been trying to get asterisk@home to work with uplink, here is a working example although there is a bug somewhere that causes a freeze when dialing skype out. I suspect this is an uplink bug in which case there is nothing we can do except hope for new release. 1. Setup an a@h trunk as follows:- Outgoing settings Trunk Name: skype PEER Details: allow=all caninvite=yes canreinvite=yes context=extensions host=dynamic insecure=very nat=yes secret=5506 type=friend username=5506 Incoming settings USER Context: 5506 USER Details: allow=all callerid="Incoming Skype" caninvite=yes canreinvite=yes context=ext-queues host=dynamic insecure=very nat=yes secret=5506 type=friend username=5506 Registration Register String: 5506:5506@192.168.1.2/5506 The format of the register string is username:password@uplinkhost/username, ie my machine with the uplink and skype client running is 192.168.1.2, adjust to suit your own network. Note that 5506 is an arbitrary number that is based on the asterisk setup in this forum. Also I am using a context=ext-queues on incoming as I am routing my skype calls into call queue 123. If you just want to ring an extension use context=from-internal etc. 2. Define an outbound route. Route name: skype Dial patters: 8|X. This routes all calls prefixed with an 8 via this route. eg to dial 00441234567890 via skype dial 800441234567890 etc. Trunk Sequence: SIP/skype That completes the basic setup for a@h but we then need to add some bits to the config files, note that these will be overwritten each you time you change settings from within a@h so do these last. Go to Tools -> Config Edit Click on extensions_additional.conf Find the skype outroute, eg outrt-xxx-skype Change the dialout line to read:- exten => _8X.,1,Macro(UpLink,${EXTEN:1},,) If you want to map some fixed extensions to call skype users by their skypename then add some arbitrary extensions in the the [ext-local] section, eg exten => 1977,1,Macro(UpLink,skypename) exten => 1977,hint,SIP/1977 This will call skype user 'skypename' when you dial 1977, note you cannot dial your own skypename to test it, I tried and it doesn't work. Click update. Now open extensions.conf Add this to the end of the file:- [macro-UpLink] exten => s,1,Dial(SIP/${ARG1}@5506,60,) exten => s,2,Congestion If you want to follow my example completely setup a call queue, number 123 before you edit the config files. Uplink setup. General When SIP Calls: Use the dialled number When Skype calls SIP dial the following number: 123 Note 123 is my call queue. SIP Full friendly display name: skype SIP account number (or user): 5506 Server: 192.168.1.5 Use the ip address of your a@h server Password: 5506 Network Local SIP Port to listen on: 5060 Local RTP ports....: 8000 Use static IP address Public IP address:- xxx.xxx.xxx.xxx eg put in your public IP address External SIP Port: 5060 External RTP port starting from: 8000 Restart uplink. If everything is ok then uplink should register as sip:5506@your-a@h-ip That's it. This all works for me with one problem, when I dial out through skype only the first call works and then the Skype client hangs and does not disconnect, it requies a ctrl-alt-delete to kill the uplink process before the skype client will disconnect and respond. With calls to skype users this does not happen, it only happens when dialing skype out, not skype to skype. I don't know whether this is an uplink bug??? I am dialing UK numbers of the format 00441234567890 If I've missed anything please let me know, good luck! Link to comment Share on other sites More sharing options...
sipsllc Posted May 22, 2006 Share Posted May 22, 2006 Hi there, I have follow your steps as below and got error in uplink program with error "register attempt for sip fail". I am confusing whether we need to create a SIP user 5506 in sip.conf or as an extension in a@home? To do the testing, what number from IP phone connect from asterisk to call skype user or from skype user to call internal extension in asterisk. Thanks ... Hi folks, I know a few of you have been trying to get asterisk@home to work with uplink, here is a working example although there is a bug somewhere that causes a freeze when dialing skype out. I suspect this is an uplink bug in which case there is nothing we can do except hope for new release. 1. Setup an a@h trunk as follows:- Outgoing settings Trunk Name: skype PEER Details: allow=all caninvite=yes canreinvite=yes context=extensions host=dynamic insecure=very nat=yes secret=5506 type=friend username=5506 Incoming settings USER Context: 5506 USER Details: allow=all callerid="Incoming Skype" caninvite=yes canreinvite=yes context=ext-queues host=dynamic insecure=very nat=yes secret=5506 type=friend username=5506 Registration Register String: 5506:5506@192.168.1.2/5506 The format of the register string is username:password@uplinkhost/username, ie my machine with the uplink and skype client running is 192.168.1.2, adjust to suit your own network. Note that 5506 is an arbitrary number that is based on the asterisk setup in this forum. Also I am using a context=ext-queues on incoming as I am routing my skype calls into call queue 123. If you just want to ring an extension use context=from-internal etc. 2. Define an outbound route. Route name: skype Dial patters: 8|X. This routes all calls prefixed with an 8 via this route. eg to dial 00441234567890 via skype dial 800441234567890 etc. Trunk Sequence: SIP/skype That completes the basic setup for a@h but we then need to add some bits to the config files, note that these will be overwritten each you time you change settings from within a@h so do these last. Go to Tools -> Config Edit Click on extensions_additional.conf Find the skype outroute, eg outrt-xxx-skype Change the dialout line to read:- exten => _8X.,1,Macro(UpLink,${EXTEN:1},,) If you want to map some fixed extensions to call skype users by their skypename then add some arbitrary extensions in the the [ext-local] section, eg exten => 1977,1,Macro(UpLink,skypename) exten => 1977,hint,SIP/1977 This will call skype user 'skypename' when you dial 1977, note you cannot dial your own skypename to test it, I tried and it doesn't work. Click update. Now open extensions.conf Add this to the end of the file:- [macro-UpLink] exten => s,1,Dial(SIP/${ARG1}@5506,60,) exten => s,2,Congestion If you want to follow my example completely setup a call queue, number 123 before you edit the config files. Uplink setup. General When SIP Calls: Use the dialled number When Skype calls SIP dial the following number: 123 Note 123 is my call queue. SIP Full friendly display name: skype SIP account number (or user): 5506 Server: 192.168.1.5 Use the ip address of your a@h server Password: 5506 Network Local SIP Port to listen on: 5060 Local RTP ports....: 8000 Use static IP address Public IP address:- xxx.xxx.xxx.xxx eg put in your public IP address External SIP Port: 5060 External RTP port starting from: 8000 Restart uplink. If everything is ok then uplink should register as sip:5506@your-a@h-ip That's it. This all works for me with one problem, when I dial out through skype only the first call works and then the Skype client hangs and does not disconnect, it requies a ctrl-alt-delete to kill the uplink process before the skype client will disconnect and respond. With calls to skype users this does not happen, it only happens when dialing skype out, not skype to skype. I don't know whether this is an uplink bug??? I am dialing UK numbers of the format 00441234567890 If I've missed anything please let me know, good luck! <{POST_SNAPBACK}> Link to comment Share on other sites More sharing options...
niter3 Posted May 27, 2006 Share Posted May 27, 2006 Yes, and I have done the same. However, uplink logins no problem. Now I'm trying to place an out going 7 digit call by 85643232 and I'm getting all circuits are busy now. Is there something I can check??? Hi there, I have follow your steps as below and got error in uplink program with error "register attempt for sip fail". I am confusing whether we need to create a SIP user 5506 in sip.conf or as an extension in a@home? To do the testing, what number from IP phone connect from asterisk to call skype user or from skype user to call internal extension in asterisk. Thanks ... <{POST_SNAPBACK}> Link to comment Share on other sites More sharing options...
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