Re: [Asterisk-Users] chan_sccp2 testers needed
On Fri, Jul 30, 2004 at 03:40:58AM +0200, Jan Czmok wrote: Date: Fri, 30 Jul 2004 03:40:58 +0200 From: Jan Czmok [EMAIL PROTECTED] Dear Skinny/SCCP lovers :-) I've just completed uploaded to the cvs the newest version with fixed redial key AND implementation of speed dials. please test extensively and report any bugs. i know that the display is not yet set correctly but the buttons are working as expected. Enjoy testing... Is it possible to make chan_sccp work with Cisco's SRST feature? Chan_sccp complains: chan_sccp.c:134 handle_message: Client sent RegisterTokenReq without first registering. when phone tries to register form call-manager-fallback (SRST) router back to Asterisk. Thanks in advance. -- Alexei Chetroi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What happened to the CVS asterisk_stable branch?
On Tue, Jul 06, 2004 at 12:32:20AM -0500, Chris Foster wrote: Date: Tue, 6 Jul 2004 00:32:20 -0500 From: Chris Foster [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] What happened to the CVS asterisk_stable branch? On Mon, 5 Jul 2004 22:02:37 -0700 (PDT), every buddy [EMAIL PROTECTED] wrote: A while ago on the download page on www.asterisk.org, there was a stable branch for the asterisk source tree. It seems to have disappeared now, at least the instructions on that web page are gone. What's the story on this? Can we have it back please? thanks stable's gone because it wasn't too stable. The lastest CVS source is alot more full featured and stable then the old stable branch. Disagree. At least I had Dlink DPH-100M (mgcp phone) working fine with stable. Whith cvs head it is working strange. Doesn't provide tone on handset pickup, strips 1st dialed digit, etc. -- Alexei Chetroi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with CHAN_SCCP
On Fri, Jul 02, 2004 at 12:36:23PM -0300, Lopez Marcelo wrote: Date: Fri, 2 Jul 2004 12:36:23 -0300 From: Lopez Marcelo [EMAIL PROTECTED] X-Mailer: Internet Mail Service (5.5.2657.72) To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Problem with CHAN_SCCP Hi, I have an asterisk running great, with 2 cisco 7912 phones converted to SIP, and a cisco 2600 xl w/ E1 and SIP. I'm thinking to expand the test adding more 7912, but I prefer not convert all the 7912 to SIP, so I'm tying to put CHAN_SCCP to work. I've get the sources from [snip] It seems that chan_sccp doesn't compile with stable version of Asterisk. I had to download head cvs in order to compile chan_sccp. BTW why you don't want to use SIP firmware? I have 7910 which doesn't support SIP and I cannot get all functions working. For example two speed dial buttons do not work. (chan_sccp receives unknown stimuls). Don't know how to change ringer type, depending which line is called. Regards, -- Alexei Chetroi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Causing Cisco 7200 Router to Crash?
On Wed, Jun 30, 2004 at 05:05:26PM -0400, Brian Wilkins wrote: Date: Wed, 30 Jun 2004 17:05:26 -0400 From: Brian Wilkins [EMAIL PROTECTED] Organization: HCC User-Agent: KMail/1.6.2 To: Asterisk-users [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk Causing Cisco 7200 Router to Crash? Hi, We are having an issue here. It seems that whenever we initialize Asterisk on our network, the router that the Asterisk server is connected to (Cisco 7200) crashes and loses it configuration. This has happended five times and each time we have tested it, it is always when Asterisk starts up. Has anyone else seen this problem? It is very odd because this is a very good router and we had the Asterisk server on an exact same router but different network before and it did not cause a crash. We have gone through two different Cisco 7200 series routers and both exhibited the same problems. Any clues? Thanks - I think you should open a TAC case on cisco or contact your cisco representative. IMHO it's a serious problem, if you can crash your cisco just by starting asterisk. BTW, have you saved cisco's configuration in nvram after configuring it. How cisco is configured? is it just ip gateway or you are using it as Voice gateway? in second case what hardware: BRI/PRI? what protocol h323/mgcp? -- Alexei Chetroi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sccp to sip call signalling
Hi, How asterisk decides whether to do media relaying or not? For SIP I've found that canreinvite=yes allows me to use * only for signalling, RTP stream will flow between endpoints only. Are such things possible when calling from SCCP channel to SIP for example? SCCP to SCCP? Thanks in advance! -- Alexei Chetroi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DLink mgcp phone and CVS HEAD
Hi, I'm playing around with Asterisk and DPH-100M (Dlink mgcp phone) on my debian box. I've got stable version of Asterisk (packaged for debian) working with dlink phone and 7910 from cisco (minimalistic extensions.conf and chan_skinny for 7910) Everything works fine. Now I'm trying to get CVS HEAD working with MGCP. I want to test chan_sccp (http://sourceforge.net/projects/chan-sccp/) and this compiles for head only. I've successfuly compiled CVS version and installed it. Chan_sccp works as I see, but strange things happens to dlink phone. When I pick-up the phone I hear nothing, but after some period I hear fast busy (or how is it called). If I press hook for very short period (flash), than I can hear tone, but unable to dial extensions. Anybody experienced this? May mgcp debug be posted on list? Thanks in advance. -- Alexei Chetroi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
