Hi, I have the following config: [7960] <--skinny--> [Cisco CCM] <--SIP_trunk--> [asterisk] <--SIP--> [X-lite] Is there a chance to avoid the RTP stream from passing through the Cisco CCM ? I would like to have all RTP handled by the *. This is just a testbed, for a larger project. What I want to achieve, is actually this: [Cisco Phone] <--skinny--> [Cisco CCM] <--SIP_trunk--> [asterisk] <--IAX2_trunk--> [asterisk] <--SIP_trunk--> [Cisco CCM] <--skinny--> [Cisco Phone] In that case, I need to handle basic calls between the Cisco phones, and would like to offload the RTP from the CallManagers. RTP needs to pass through the asterisk servers and the IAX trunk only. Here is my sip.conf: =====================================[general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown [201] username=201 type=friend secret=xxx qualify=no port=5060 pickupgroupnat=never mailbox=201@default host=dynamic dtmfmode=rfc2833 disallowcontext=from-internal canreinvite=no callgroupcallerid="201" <201> allow [ccm] type=friend qualify=yes nat=no host=a.b.c.d disallow=all context=from-internal canreinvite=yes allow=alaw ======================================
Patrick, I would try to get * to talk SCCP(Skinny). The Cisco phones are either going to talk SIP or SCCP but I'm not aware that they will talk both. Take a look at http://chan-sccp.sourceforge.net/ Good Luck Scott Patrick Zwahlen wrote:> Hi, > > I have the following config: > > [7960] <--skinny--> [Cisco CCM] <--SIP_trunk--> [asterisk] <--SIP--> > [X-lite] > > Is there a chance to avoid the RTP stream from passing through the Cisco > CCM ? I would like to have all RTP handled by the *. > > This is just a testbed, for a larger project. What I want to achieve, is > actually this: > > [Cisco Phone] <--skinny--> [Cisco CCM] <--SIP_trunk--> [asterisk] > <--IAX2_trunk--> [asterisk] <--SIP_trunk--> [Cisco CCM] <--skinny--> > [Cisco Phone] > > In that case, I need to handle basic calls between the Cisco phones, and > would like to offload the RTP from the CallManagers. RTP needs to pass > through the asterisk servers and the IAX trunk only. > > Here is my sip.conf: > =====================================> [general] > > port = 5060 ; Port to bind to (SIP is 5060) > bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) > disallow=all > allow=ulaw > allow=alaw > context = from-sip-external ; Send unknown SIP callers to this context > callerid = Unknown > > [201] > username=201 > type=friend > secret=xxx > qualify=no > port=5060 > pickupgroup> nat=never > mailbox=201@default > host=dynamic > dtmfmode=rfc2833 > disallow> context=from-internal > canreinvite=no > callgroup> callerid="201" <201> > allow> > [ccm] > type=friend > qualify=yes > nat=no > host=a.b.c.d > disallow=all > context=from-internal > canreinvite=yes > allow=alaw > =====================================> _______________________________________________ > 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 > >
Thx Scott, However, and from what I can read, chan_sccp is not really meant for server-to-server trunks, yet... Am I wrong ? BR, - Patrick - -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Scott Herrick Sent: jeudi, 26. mai 2005 04:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RTP path with Cisco CCM Patrick, I would try to get * to talk SCCP(Skinny). The Cisco phones are either going to talk SIP or SCCP but I'm not aware that they will talk both. Take a look at http://chan-sccp.sourceforge.net/ Good Luck Scott Patrick Zwahlen wrote:> Hi, > > I have the following config: > > [7960] <--skinny--> [Cisco CCM] <--SIP_trunk--> [asterisk] <--SIP--> > [X-lite] > > Is there a chance to avoid the RTP stream from passing through the > Cisco CCM ? I would like to have all RTP handled by the *. > > This is just a testbed, for a larger project. What I want to achieve, > is actually this: > > [Cisco Phone] <--skinny--> [Cisco CCM] <--SIP_trunk--> [asterisk] > <--IAX2_trunk--> [asterisk] <--SIP_trunk--> [Cisco CCM] <--skinny--> > [Cisco Phone] > > In that case, I need to handle basic calls between the Cisco phones, > and would like to offload the RTP from the CallManagers. RTP needs to > pass through the asterisk servers and the IAX trunk only. > > Here is my sip.conf: > =====================================> [general] > > port = 5060 ; Port to bind to (SIP is 5060) > bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) > disallow=all > allow=ulaw > allow=alaw > context = from-sip-external ; Send unknown SIP callers to this context> callerid = Unknown > > [201] > username=201 > type=friend > secret=xxx > qualify=no > port=5060 > pickupgroup> nat=never > mailbox=201@default > host=dynamic > dtmfmode=rfc2833 > disallow> context=from-internal > canreinvite=no > callgroup> callerid="201" <201> > allow> > [ccm] > type=friend > qualify=yes > nat=no > host=a.b.c.d > disallow=all > context=from-internal > canreinvite=yes > allow=alaw > =====================================> _______________________________________________ > 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 > >_______________________________________________ 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