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...a PRI (these are all local calls). I have run a "sip debug" trace of the successful and failing calls, and everything looks normal; there is only one difference for the failing calls. The successful SIP trace looks like this (P-Polycom Phone, A-Asterisk): P-INVITE sip:96027414660@test.starnetworks.us;user=phone SIP/2.0 A-SIP/2.0 100 Trying A-SIP/2.0 183 Session Progress A-SIP/2.0 200 OK P-ACK sip:96027414660@67.137.151.151 SIP/2.0 P-BYE sip:96027414660@67.137.151.151 SIP/2.0 A-SIP/2.0 200 OK The failing SIP trace looks like this: P-INVITE sip:96027414660@test.starnetworks.us;user=phone SIP...

http://host-a.starnetworks.us/Members/kpfleming/spring_von/photoalbum_view Enjoy!

>Message: 9 >Date: Thu, 02 Dec 2004 11:33:16 -0700 >From: "Kevin P. Fleming" <kpfleming@starnetworks.us> >Subject: Re: [Asterisk-Users] Polycom 500, asterisk user opinions? >To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >Message-ID: <41AF5FEC.7050406@starnetworks.us> >Content-Type: text/plain; charset=ISO-8859-1; forma...

I am running Asterisk 1.8.10.1. I am trying to set up some routing in my dial plans and having some issues (the issue being that I don't quite understand all of the syntax and patterns that can be used: Examples: DID1 = 6140000000 DID2 = 6140000001 CNAME1 = 6149999999 CNAME2 = 6149999998 CNAME3 = 6149999997 context1 context2 context3 I have looked at several examples (patterns) and I

I am having a problem with Polycom auto-answer. I have the auto-answer working between PhoneA and PhoneB, but when I try to use the intercom between more then one phone I start having problems. PhoneA dials *3 which calls PhoneB, PhoneC, and PhoneD. All the phones ring, but only one will pick up, the rest will hang up and I get this error on Asterisk: Got SIP response 500 "Internal Server

I searched and read all the relevant posts, but I still don't have a solution to my problem.. I've got a small queue for tech support calls using AddQueueMember. The agents are using IP300's from polycom. In my example, only one agent is logged int. When a call comes into the queue, asterisk sends the call to the one agent logged in. The agent answers and is talking to the

I have a problem where if an agent's extension is busy and has voicemail the queue app will follow the dialplan and send the caller to an agents voicemail. This is really bad, because it takes the caller out of the queue when it hits that agent. But we also would like to have voicemail for some extensions like the shift managers etc. Is there s solution/workaround/patch? Thanks, -Ryan

> -----Original Message----- > From: Kevin P. Fleming [mailto:kpfleming@starnetworks.us] > David Brodbeck wrote: > > > Anyway, I figured it out. The extension was working, but > Background() > > ignores the tones A through D by default. I didn't realize > this because I > > wasn't waiting for message playback to finish. > > Please...

Hello, We use and develop the astGUIclient suite. It is Open-source(as in GPL) and offers Inbound and Outbound call center functions with reports, ACD, monitoring, recording and very basic IVR scripts. Complex IVR functions need to be custom programmed within Asterisk but that is not really that hard. It works across multiple Asterisk servers and we are using it currently at 5 locations including

Ok, so I emailed the list earlier about my polycom phone not ringing when anyone called in. Well, polycom support said that is impossible that this could happen because of a change in a configuration file. However the new phone arrived today (a refurb.) and it also would not ring. So I obviously got rather frustrated and blasted away all of my configuration files from the FTP server. I then copied

Hi there, we are looking for an opensource or commercial * based Call Center. Full ACD, call monitoring, multiple queue, IVR, voicemail, management, reporting, CDR, etc is needed. over 100 seat can be the initial target and will grow in a very short time. SIP phones will be used and multiple E1 lines incoming, so to provide full failover a cluster of * machines or some other form of redundancy

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I have been searching all over for the answer on all sources online and have come to the conclusion that it must be rudimentary or I am asking the wrong question. I cannot figure out how to configure the box to set the "from" address to a correct domain, as my outgoing isp will not pass mail from root@asterisk1.local, as I expect it wouldn't. Any help is appreciated, even just

Can anyone give an example of the difference between the following: canreinvite=no canreinvite=yes canreinvite=update Here is the problem: I have an 800 number sent to me via SIP from a national carrier. Asterisk gets the number and rings my desk phone. Asterisk has 2 NICs, one with public IP and private IP. My phone is on private IP, the inbound call is on public. My phone rings and I answer

...--- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050318/e1446f0e/attachment-0001.htm ------------------------------ Message: 20 Date: Fri, 18 Mar 2005 09:19:33 -0700 From: "Kevin P. Fleming" <kpfleming@starnetworks.us> Subject: Re: [Asterisk-Users] asterisk reload To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <423AFF95.5040200@starnetworks.us> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Asterisk wrote: > > CVS...

Hey Everyone, I downloaded and installed the X-Lite softphone the other day (the lite version) and cannot seem to get it to work well. Don't get me wrong, it registers with my asterisk server and everything seems to work well, except the call quality really is horrible. I thought it may be the place I was trying it at (DSL) so I took it to the office and tried it right next to the asterisk

For any of you that are running CVS (recent enough to have the wildcard config file include change), I have posted another enhancement that makes it even more useful :-) Bug 3020 adds the ability for the same context to appear multiple times, in multiple files, and have them "add together". This allows you to have a master context defined in extensions.conf, that is then filled in

If you run CVS HEAD, and are using ODBC storage for voicemail, please apply and test the patch from bug 3024 in Mantis. This patch should _not_ change the behavior of your system at all, and that's what needs to be tested. If you can try it, please report the results in the bugnotes. Thanks!

For anyone using CVS HEAD, if you are using queue member persistence or agent persistence, your next update will cause the persistence to break. The storage format for these elements has been changed so that it can be more easily extended in the future, but this required breaking compatibility. This should be the last time these features will be broken by an upgrade :-)