Hello, everybody! I need help connecting my Cisco AS5350 to Asterisk. What i want to do is forward all outgoing calls from Asterisk server to Cisco AS5350, and from Cisco 5350 to my Asterisk server, using SIP. How could this be done? Thanks in advance Atif Shahzad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081125/ca45932c/attachment.htm
Hello, everybody! I need help connecting my Cisco AS5350 to Asterisk. What i want to do is forward all outgoing calls from Asterisk server to Cisco AS5350, and from Cisco 5350 to my Asterisk server, using SIP. How could this be done? Thanks in advance Atif Shahzad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081125/371fb8c6/attachment.htm
Hello, everybody! I need help connecting my Cisco AS5350 to Asterisk. What i want to do is forward all outgoing calls from Asterisk server to Cisco AS5350, and from Cisco 5350 to my Asterisk server, using SIP. How could this be done? Thanks in advance Atif Shahzad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081125/9977b036/attachment.htm
Set up a SIP dial peer and an outbound POTS dial peer. Bear in mind that the gateway shunts calls POTS->VOIP and VOIP->POTS by default, so you can use the same destination pattern matching for both in this simple scenario, but if it gets any more complicated than that, some degree of translation is almost certainly required. The process can be fairly complex, but the general idea, if you have your TDM side set up, is: dial-peer voice 500 voip description Asterisk destination-pattern .T progress_ind setup enable 3 voice-class codec 1 session protocol sipv2 session target ipv4:ip.addr.of.asterisk session transport udp dtmf-relay rtp-nte no vad dial-peer voice 510 pots description Fancy PRI - Outgoing huntstop destination-pattern .T direct-inward-dial forward-digits 10 A T I F wrote:> Hello, everybody! > > I need help connecting my Cisco AS5350 to Asterisk. > > What i want to do is forward all outgoing calls from Asterisk server to > Cisco AS5350, and from Cisco 5350 to my Asterisk server, using SIP. > > How could this be done? > > Thanks in advance > > Atif Shahzad > > > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599
The procedure you explain is for outbound or inbound for Asterisk or Can you tell me the procedure for only outbound from my Asterisk server to Cisco 5350? On Tue, Nov 25, 2008 at 3:10 PM, Alex Balashov <abalashov at evaristesys.com>wrote:> Set up a SIP dial peer and an outbound POTS dial peer. Bear in mind > that the gateway shunts calls POTS->VOIP and VOIP->POTS by default, so > you can use the same destination pattern matching for both in this > simple scenario, but if it gets any more complicated than that, some > degree of translation is almost certainly required. > > The process can be fairly complex, but the general idea, if you have > your TDM side set up, is: > > dial-peer voice 500 voip > description Asterisk > destination-pattern .T > progress_ind setup enable 3 > voice-class codec 1 > session protocol sipv2 > session target ipv4:ip.addr.of.asterisk > session transport udp > dtmf-relay rtp-nte > no vad > > dial-peer voice 510 pots > description Fancy PRI - Outgoing > huntstop > destination-pattern .T > direct-inward-dial > forward-digits 10 > > > A T I F wrote: > > > Hello, everybody! > > > > I need help connecting my Cisco AS5350 to Asterisk. > > > > What i want to do is forward all outgoing calls from Asterisk server to > > Cisco AS5350, and from Cisco 5350 to my Asterisk server, using SIP. > > > > How could this be done? > > > > Thanks in advance > > > > Atif Shahzad > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (706) 338-8599 > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > 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/20081125/71fa927f/attachment.htm
Alex, 1 more thing my gateway is configured with H.323 so tell me how can I configure it with SIP? On Tue, Nov 25, 2008 at 3:10 PM, Alex Balashov <abalashov at evaristesys.com>wrote:> Set up a SIP dial peer and an outbound POTS dial peer. Bear in mind > that the gateway shunts calls POTS->VOIP and VOIP->POTS by default, so > you can use the same destination pattern matching for both in this > simple scenario, but if it gets any more complicated than that, some > degree of translation is almost certainly required. > > The process can be fairly complex, but the general idea, if you have > your TDM side set up, is: > > dial-peer voice 500 voip > description Asterisk > destination-pattern .T > progress_ind setup enable 3 > voice-class codec 1 > session protocol sipv2 > session target ipv4:ip.addr.of.asterisk > session transport udp > dtmf-relay rtp-nte > no vad > > dial-peer voice 510 pots > description Fancy PRI - Outgoing > huntstop > destination-pattern .T > direct-inward-dial > forward-digits 10 > > > A T I F wrote: > > > Hello, everybody! > > > > I need help connecting my Cisco AS5350 to Asterisk. > > > > What i want to do is forward all outgoing calls from Asterisk server to > > Cisco AS5350, and from Cisco 5350 to my Asterisk server, using SIP. > > > > How could this be done? > > > > Thanks in advance > > > > Atif Shahzad > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (706) 338-8599 > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > 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/20081125/6f2d1354/attachment.htm
Use only the second dial peer. A T I F wrote:> The procedure you explain is for outbound or inbound for Asterisk or Can > you tell me the procedure for only outbound from my Asterisk server to > Cisco 5350? > > On Tue, Nov 25, 2008 at 3:10 PM, Alex Balashov > <abalashov at evaristesys.com <mailto:abalashov at evaristesys.com>> wrote: > > Set up a SIP dial peer and an outbound POTS dial peer. Bear in mind > that the gateway shunts calls POTS->VOIP and VOIP->POTS by default, so > you can use the same destination pattern matching for both in this > simple scenario, but if it gets any more complicated than that, some > degree of translation is almost certainly required. > > The process can be fairly complex, but the general idea, if you have > your TDM side set up, is: > > dial-peer voice 500 voip > description Asterisk > destination-pattern .T > progress_ind setup enable 3 > voice-class codec 1 > session protocol sipv2 > session target ipv4:ip.addr.of.asterisk > session transport udp > dtmf-relay rtp-nte > no vad > > dial-peer voice 510 pots > description Fancy PRI - Outgoing > huntstop > destination-pattern .T > direct-inward-dial > forward-digits 10 > > > A T I F wrote: > > > Hello, everybody! > > > > I need help connecting my Cisco AS5350 to Asterisk. > > > > What i want to do is forward all outgoing calls from Asterisk > server to > > Cisco AS5350, and from Cisco 5350 to my Asterisk server, using SIP. > > > > How could this be done? > > > > Thanks in advance > > > > Atif Shahzad > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (706) 338-8599 > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599
My attention to my dial peer. It has nothing about H.323 and much about SIP. A T I F wrote:> Alex, > > 1 more thing my gateway is configured with H.323 so tell me how can I > configure it with SIP? > > On Tue, Nov 25, 2008 at 3:10 PM, Alex Balashov > <abalashov at evaristesys.com <mailto:abalashov at evaristesys.com>> wrote: > > Set up a SIP dial peer and an outbound POTS dial peer. Bear in mind > that the gateway shunts calls POTS->VOIP and VOIP->POTS by default, so > you can use the same destination pattern matching for both in this > simple scenario, but if it gets any more complicated than that, some > degree of translation is almost certainly required. > > The process can be fairly complex, but the general idea, if you have > your TDM side set up, is: > > dial-peer voice 500 voip > description Asterisk > destination-pattern .T > progress_ind setup enable 3 > voice-class codec 1 > session protocol sipv2 > session target ipv4:ip.addr.of.asterisk > session transport udp > dtmf-relay rtp-nte > no vad > > dial-peer voice 510 pots > description Fancy PRI - Outgoing > huntstop > destination-pattern .T > direct-inward-dial > forward-digits 10 > > > A T I F wrote: > > > Hello, everybody! > > > > I need help connecting my Cisco AS5350 to Asterisk. > > > > What i want to do is forward all outgoing calls from Asterisk > server to > > Cisco AS5350, and from Cisco 5350 to my Asterisk server, using SIP. > > > > How could this be done? > > > > Thanks in advance > > > > Atif Shahzad > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (706) 338-8599 > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599
1. dial-peer voice 500 voip
I use this configuration for inbound to asterisk.
2. dial-peer voice 510 pots
description Fancy PRI - Outgoing
huntstop
destination-pattern .T
direct-inward-dial
forward-digits 10
And use this configuration for outbound from asterisk to Cisco 5350 right?
On Tue, Nov 25, 2008 at 3:42 PM, Alex Balashov <abalashov at
evaristesys.com>wrote:
> My attention to my dial peer. It has nothing about H.323 and much about
> SIP.
>
> A T I F wrote:
>
> > Alex,
> >
> > 1 more thing my gateway is configured with H.323 so tell me how can I
> > configure it with SIP?
> >
> > On Tue, Nov 25, 2008 at 3:10 PM, Alex Balashov
> > <abalashov at evaristesys.com <mailto:abalashov at
evaristesys.com>> wrote:
> >
> > Set up a SIP dial peer and an outbound POTS dial peer. Bear in
mind
> > that the gateway shunts calls POTS->VOIP and VOIP->POTS by
default,
> so
> > you can use the same destination pattern matching for both in this
> > simple scenario, but if it gets any more complicated than that,
some
> > degree of translation is almost certainly required.
> >
> > The process can be fairly complex, but the general idea, if you
have
> > your TDM side set up, is:
> >
> > dial-peer voice 500 voip
> > description Asterisk
> > destination-pattern .T
> > progress_ind setup enable 3
> > voice-class codec 1
> > session protocol sipv2
> > session target ipv4:ip.addr.of.asterisk
> > session transport udp
> > dtmf-relay rtp-nte
> > no vad
> >
> > dial-peer voice 510 pots
> > description Fancy PRI - Outgoing
> > huntstop
> > destination-pattern .T
> > direct-inward-dial
> > forward-digits 10
> >
> >
> > A T I F wrote:
> >
> > > Hello, everybody!
> > >
> > > I need help connecting my Cisco AS5350 to Asterisk.
> > >
> > > What i want to do is forward all outgoing calls from
Asterisk
> > server to
> > > Cisco AS5350, and from Cisco 5350 to my Asterisk server,
using
> SIP.
> > >
> > > How could this be done?
> > >
> > > Thanks in advance
> > >
> > > Atif Shahzad
> > >
> > >
> > >
> >
> ------------------------------------------------------------------------
> > >
> > > _______________________________________________
> > > -- Bandwidth and Colocation Provided by
> http://www.api-digital.com --
> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > --
> > Alex Balashov
> > Evariste Systems
> > Web : http://www.evaristesys.com/
> > Tel : (+1) (678) 954-0670
> > Direct : (+1) (678) 954-0671
> > Mobile : (+1) (706) 338-8599
> >
> > _______________________________________________
> > -- Bandwidth and Colocation Provided by
http://www.api-digital.com--
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> >
------------------------------------------------------------------------
> >
> > _______________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
> Alex Balashov
> Evariste Systems
> Web : http://www.evaristesys.com/
> Tel : (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (706) 338-8599
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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