search for: millay

I'm running SIP between my Lucent TNT acting as a gateway, and an asterisk server. We have a PRI coming into the Lucent. Basically the problem I'm having is mostly on inbound calls but some outbound calls as well. I hear echo and sometimes some weird artifacting on calls coming in from the lucent. Everything routed over IAX to VoIP Jet or Nufone sounds fine. It seems like every 3

Trying to use: Asterisk 1.4.2 Zaptel 1.4.0 chan_zap won't compile in asterisk 1.4.2 when used with zaptel 1.4.0. The changelog has this entry: * channels/chan_zap.c, configure, configure.ac: If we receive ZT_EVENT_REMOVED, destroy the specified channel. (issue #7256, tzafrir) Also, update the configure script to make sure that we don't try to build chan_zap

...970 but the status log shows errors parsing the config. I copied a config that was posted to the list but it didn't seem to work. Any help would be appreciated. Jeremiah -- ______________________________________________________________ Rock River Internet Jeremiah Millay 202 W. State St, 8th Floor jeremiah@rockriver.net Rockford, IL 61101 815-968-9888 Ext. 2202 USA fax 968-6888

I'm having trouble building res_snmp (under Asterisk 1.4.0) on a box running Debian Sarge. res_snmp says its dependencies are netsnmp but Debian doesn't seem to have a netsnmp package. I've tried installing pretty much every package available related to snmp and no luck. I'm just wondering if anyone has successfully built the res_snmp module under Debian Sarge stable. Any

I've downloaded: asterisk-1.4.1 zaptel-1.4.0 I've compiled and installed zaptel. When I go to install asterisk I do: ./configure make menuselect I then take a look under the codec selection menu and I see that codec_zap can not be compiled. *************************************

Does anyone know what application I should place this function in? For example with the DB function I currently do something like this to add an entry to the asterisk database: exten => s,n,Set(DB(AGENT/${MACRO_EXTEN:1})=${CALLERID(num)}) To delete the entries I do something like this: exten => s,n,DBDel(AGENT/${MACRO_EXTEN:1}) DBDel is marked as deprecated in favor of the DB_DELETE

...quot;dir-intro"; + else if (last==2) + dirintro = "dir-intro-fnln"; else dirintro = "dir-intro-fn"; } -- ______________________________________________________________ Rock River Internet Jeremiah Millay 202 W. State St, 8th Floor jeremiah@rockriver.net Rockford, IL 61101 815-968-9888 Ext. 2202 USA fax 968-6888

I'm trying to change the ring tone on my 7960 from the dialplan. I've tried the example on the wiki but it doesn't seem to work. Something like: exten => 3010,1,SetVar(ALERT_INFO=<Bellcore-dr1>) ; selects Ringer exten => 3010,2,Dial(SIP/3010,15) I'm not sure what the Bellcore-dr1 ringer is supposed to be. I've tried replacing ALERT_INFO with another ring tone

Has anyone ever encountered an echo on the IP phone side of a call? It is an echo of the user's own voice. I believe that no one else in the office is experiencing this problem. The phone itself is a Snom 320. I've asked Snom for assistance since my source no longer carries Snom, but unlike previous times they've been slow to respond. ----- Mike Hammett Intelligent Computing

I currently have two Cisco phones, a 7960 and 7970. The 7960 has a SIP OS on it and the 7970 has a SCCP. When the 7960 powers up it loads OS79XX.TXT, SIPDefault.cnf, SIP000E3875266C.cnf, RINGLIST.DAT, and dialplan.xml. I have a Cisco SmartNet agreement with the phone so I have access to download the firmware. I recently purchased a Cisco 7970 phone and was in the process of configuring

Has anyone else had problems with the Page() application not working under Asterisk 1.2.3? We use Cisco 7960 phones and set one of the lines to auto answer. When someone dials the paging extension it calls the page app and invites all the lines on the phones that are set to auto answer into a meetme conference where all the members are muted except the original caller. When I try to use the

I know this has been added to SVN but I'm looking for the source for the original module. It used to be at http://redice.krisk.org/ but this page no longer seems to display anything. I'd like to add it to my 1.2.13 stable install. Does anyone have a copy of the original? I used to have this somewhere but have been unable to locate it. Jeremiah

I'm using a mix of Cisco 7960, Linksys SPA-942, Cisco 7961, Cisco 7970 phones in a paging group. I have all the phones set up with an extra line that auto answers the dial from my paging extension when the primary line is not in use. All of these are operating correctly however the 7961/7970s all ring once and then auto answer so the person paging all the phones has the first part of his