search for: radamson

...stalled and running, and am able to place calls via *, etc. However, when upgrading to v4.4.0 I can never get to the point of being able to place a call (eg, no dialtone, etc). I can ping the phone, look at the Network Config, etc, but I can't unlock it to do any configs. Any thoughts? Rich radamson@routers.com

...pply the astrisk diff's, and initiate basic testing. Thoughts are to download v1.4.3 (latest per the fesitval website. If anyone has an existing how-to, install notes, tips, or any suggestions I'd greatly appreciate it. Direct email is fine if you'd rather not post them. Thanks, Rich radamson@routers.com

...t's also expected to accept incoming pstn calls directing those to a single asterisk. (I don't care about an IP dialtone, nat, etc, just a plain-jane two-way sip gateway.) If anyone is designing such a box and need professional eval, we can certainly work with you privately (off list to radamson @ routers dot com) to accomidate those needs. Anyone seen such a beast at a reasonable price? Rich

Hi, I know that SIP is using port 5060 for session initiation, but which port does it use for audio ? is it dynamically assigned ? Thanks, Adrien -- Adrien Laurent - CIO www.modulis.ca 514-284-2020 ext 202 adrien@modulis.ca

Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip FXO 4-port gateway? The archives tend to suggest the box is not very straight forward, and possibly lacks some basic pstn interaction features. Thinking about trying one in place of a pair of x100p's (functioning fine now). CallerId, etc, supported on this gateway? Rich

Anyone interfaced * to the Nortel option 61c pbx via T1's, pri, etc? Need to begin planning the implementation, purchase cards, etc. Any recommended approaches, configuration problems, etc? Off-list is fine if you'd like. Rich radamson @ routers . com

Guys. I would like to hear tips and tricks on extention config best practices, for example, naming, etc. and most of all, how to deal with extention that have a full time hardphone configured and assigned and then a softphone connecting to the same extention, for example, one employee has its hardphone on the office but sometimes when he travel, he uses his softphone to work with, what happens

Hi Rich, I had to change all my nat=yes to nat=route in the sip.conf. nat=yes seems to be ignored in today's CVS. Walter > > Message: 5 > Date: Fri, 27 Aug 2004 08:45:19 -0600 > From: Rich Adamson <radamson@routers.com> > Subject: Re: [Asterisk-Users] sip change? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <Chameleon.1093614365.adar0@vegas> > Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1 > > * and th...

Has anyone had problems with using hold on a 7960 SIP firmware? The problem is when the 7960 puts a call on hold and you take it off hold again, the 7960 outbound audio is delayed on the other end. Sometimes up to a few seconds. I've tried a couple different things, making the "other end" a diff type of trunk ie: 7960sip --> asterisk --> IAX2 --> PRI 7960sip -->

...lication) == Registered application 'MP3Player' <snip> I can't seem to find any other reason for MOH to not work. If someone has MOH working, would it be possible for you to send me a "sip debug" listing of what's going on so I can compare it to mine? (offlist to radamson@routers.com) If anyone can think of anything more I can try it would be greatly appreciated. This system does have two X100P cards, everything else is working fine, its just the MOH that won't work. Thanks Rich

Hello, Does asterisk do transcoding when the call goes through the system, codecs are the same but signaling protocol is changed. example: SIP with GSM ---> IAX with GSM What quality destruction happen when I use transcoding? I know this is not a concrete/precise question, but I would like to know how is it in general. What CPU performance is needed for transcoding 30 channels e.g. from

...d during the day, by just watching the phone lines and waiting for all of them to go free. This of course will get harder as we add more sites into the central system. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting Rich Adamson <radamson@routers.com>: > > Another concern I have on this front is that it seems like some > > updates > require > > an asterisk restart rather than just issueing a reload command from > > the * console. This that correct, or I am just not running the > > system corre...

...for the sake of luck, and it was back to working. In the mean time, thanks for the troubleshooting advice. I will keep this in my log book so I have the steps for next time something goes awry. ------------------------------ Message: 6 Date: Fri, 15 Jul 2005 14:52:40 -0600 From: Rich Adamson <radamson@routers.com> Subject: Re: [Asterisk-Users] 2 TDM04B In Asterisk at home To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <Chameleon.1121457580.adar0@vegas> Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1 > I have seen...

...ehow it is turned on by default. Can I turn this option off ???? In my extensions.conf I have written : exten => 8000,3,Queue(supportq|t) plz help me inthis regard ... Thanks ! Usman. ------------------------------ Message: 3 Date: Thu, 23 Dec 2004 16:51:34 -0600 From: Rich Adamson <radamson@routers.com> Subject: Re: [Asterisk-Users] Voicemail email notification To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <Chameleon.1103842356.adar0@vegas> Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1 > Are there any commo...

Isn't it about time to lock down added functionality to v1.1 and fix the remaining bugs? There has been a significant amount of traffic on the cvs list, the irc and other channels with folks spending time adding new functionality to Head. Think its time to lock it down, fix the bugs that have been introduced, and get to "something" that the _majority_ can agree to call v1.1 Stable

i need to know something in the zaptel configuration as it seems i can configure detecting the busy tone and hangup after number of busy tone counts, that was great but the problem is sometimes the pstn line has no dialtone and when i try to make call it continue dialing while not having a dialtone! while it should say "all lines are busy/congested" how can i configure that?? i already

I've not been able to register with iaxtel.com for the last couple of days. Is anyone else seeing this, or did I miss something?

>> Wouldn't that break everybody's dialplans where they would have to >> replace all occurrences of Voicemail2 with Voicemail and all >> occurrences of Voicemailmain2 with Voicemailmain? > > No, we would register with both names. Is it necessary (with reasonably current cvs) to make any changes in the *.conf files to use Voicemail2, or is that happening

I typically start asterisk with the safe_asterisk script: 22865 pts/3 S 0:00 /bin/sh /usr/sbin/safe_asterisk 22867 pts/3 S 0:31 asterisk -vvvg -c 22871 pts/3 S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 For-You.mp3 for.m 22873 pts/3 S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 For-You.mp3 for.m But when I do a "stop now" from the CLI, the mpg123 causes a

Anyone know of a business class sip hard phone that includes a quality display capable of supporting "push" data (maybe Polycom?). Something like... VM: 3 msgs OurStock (1:43pm): 59.5 somewhere on the display that can be updated (pushed) from a server? Rich