Is there any way to use the polycom phones to log into and out of queues? So the polycom phone could show their current status in that queue? logged in / logged out for example. Thanks Julian ----------------------------------------> Subject: RE: [asterisk-users] PRI dropouts > From: pdhales@optusnet.com.au > To: asterisk-users@lists.digium.com > Date: Sat, 15 Jul 2006 20:47:17 +1000 > > > Hmm - I have had 2 bad PRI installs out of about 20, and both times it > was faulty wiring from the Telco. > But getting them to fix it can be a real struggle! > > > Paul Hales > Technical Manager > www.asteriskit.com.au > > > On Sat, 2006-07-15 at 12:23 +1000, James Sturges wrote: > > Have had L O T S of trouble like this, the settings zap config files > > seem to have to e exact, please send email to thinking@1am.com.au and > > I will send config files. > > > > > > > > Thanks > > > > > > > > James > > > > > > > > > > ______________________________________________________________________ > > From:asterisk-users-bounces@lists.digium.com > > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kevin > > Withnall > > Sent: Saturday, 15 July 2006 11:05 AM > > To: asterisk-users@lists.digium.com > > Subject: [asterisk-users] PRI dropouts > > > > > > > > > > Recently we cut over to using asterisk (trixbox 1.1.1) for our > > production system. > > > > > > > > We are using a TE110P digium card (Primary rate) with a Telstra onramp > > 10. > > > > > > > > Sometimes when people call, on their end it doesn?t seem to connect. > > On our end, we get caller id, it passes ok to the sip phone but then > > no-one is there. > > > > > > > > Anyone have any similar problems and worked out how to solve it ? > > > > > > > > Thanks. > > > > > > > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Hi Julian, If the 301's support ACD log in and log out, they should display a soft button showing the current status of the phone, I know for sure the 601's do. Personally with our 601's I used two of the contact lines and made my own log in and logout buttons and wrote my own script to log our agents in. It doesn't display the status, but I have a section on our intranet page showing the status of all members of a queue that are logged in. So it may not be the answer you wanted, and again I don't have any experience with the 301's to say what they can and cannot do, but there are some workarounds that will come close to the same goal. kevin Julian Varanini wrote:> Is there any way to use the polycom phones to log into and out of queues? So the polycom phone could show their current status in that queue? logged in / logged out for example. > > Thanks > > Julian > > > >
Do you have a soft button on the IP301? I use the 501 and it works fine, you do have to use the special asterisk code for it to work correctly. It lets me login, logout, make the agent available/unavailable. You can read about it at http://bugs.digium.com/view.php?id=6119 I found you must also use the trunk version of zaptel and libpri, and make sure you use auth on the phones in the config. Hope thats what you looking for, if so, any problems just ask, its just taken me 2 weeks to get it working great. Regards, Dean. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Julian Varanini Sent: 17 July 2006 00:15 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom IP301 and Queues Is there any way to use the polycom phones to log into and out of queues? So the polycom phone could show their current status in that queue? logged in / logged out for example. Thanks Julian ----------------------------------------> Subject: RE: [asterisk-users] PRI dropouts > From: pdhales@optusnet.com.au > To: asterisk-users@lists.digium.com > Date: Sat, 15 Jul 2006 20:47:17 +1000 > > > Hmm - I have had 2 bad PRI installs out of about 20, and both times it > was faulty wiring from the Telco. > But getting them to fix it can be a real struggle! > > > Paul Hales > Technical Manager > www.asteriskit.com.au > > > On Sat, 2006-07-15 at 12:23 +1000, James Sturges wrote: > > Have had L O T S of trouble like this, the settings zap config files > > seem to have to e exact, please send email to thinking@1am.com.au and > > I will send config files. > > > > > > > > Thanks > > > > > > > > James > > > > > > > > > > ______________________________________________________________________ > > From:asterisk-users-bounces@lists.digium.com > > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kevin > > Withnall > > Sent: Saturday, 15 July 2006 11:05 AM > > To: asterisk-users@lists.digium.com > > Subject: [asterisk-users] PRI dropouts > > > > > > > > > > Recently we cut over to using asterisk (trixbox 1.1.1) for our > > production system. > > > > > > > > We are using a TE110P digium card (Primary rate) with a Telstra onramp > > 10. > > > > > > > > Sometimes when people call, on their end it doesn?t seem to connect. > > On our end, we get caller id, it passes ok to the sip phone but then > > no-one is there. > > > > > > > > Anyone have any similar problems and worked out how to solve it ? > > > > > > > > Thanks. > > > > > > > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
I have been unable to get this branch of asterisk to work properly. I can not get any SIP phone, Polycom or X-Lite, to register with the server. If, on the same server, I recompile and install Trunk the phones register properly. In doing this I made no changes to the conf files at all. I simply recompiled and reinstalled. Is there a trick to getting the phones to register? I made sure that the phone SIP config and the agent config did no overlap. The phone will register if I comment out the secret line. I have not tried getting the ACD functionality to work at this point in time...one issue at a time. Although this will be a big leap forward if it works and I would be willing to put up a bounty to move this forward. Thanks, Michael -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dean @ INKnBITs Sent: Monday, July 17, 2006 3:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom IP301 and Queues Do you have a soft button on the IP301? I use the 501 and it works fine, you do have to use the special asterisk code for it to work correctly. It lets me login, logout, make the agent available/unavailable. You can read about it at http://bugs.digium.com/view.php?id=6119 I found you must also use the trunk version of zaptel and libpri, and make sure you use auth on the phones in the config. Hope thats what you looking for, if so, any problems just ask, its just taken me 2 weeks to get it working great. Regards, Dean. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Julian Varanini Sent: 17 July 2006 00:15 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom IP301 and Queues Is there any way to use the polycom phones to log into and out of queues? So the polycom phone could show their current status in that queue? logged in / logged out for example. Thanks Julian ----------------------------------------> Subject: RE: [asterisk-users] PRI dropouts > From: pdhales@optusnet.com.au > To: asterisk-users@lists.digium.com > Date: Sat, 15 Jul 2006 20:47:17 +1000 > > > Hmm - I have had 2 bad PRI installs out of about 20, and both times it > was faulty wiring from the Telco. > But getting them to fix it can be a real struggle! > > > Paul Hales > Technical Manager > www.asteriskit.com.au > > > On Sat, 2006-07-15 at 12:23 +1000, James Sturges wrote: > > Have had L O T S of trouble like this, the settings zap config files > > seem to have to e exact, please send email to thinking@1am.com.auand> > I will send config files. > > > > > > > > Thanks > > > > > > > > James > > > > > > > > > >______________________________________________________________________> > From:asterisk-users-bounces@lists.digium.com > > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kevin > > Withnall > > Sent: Saturday, 15 July 2006 11:05 AM > > To: asterisk-users@lists.digium.com > > Subject: [asterisk-users] PRI dropouts > > > > > > > > > > Recently we cut over to using asterisk (trixbox 1.1.1) for our > > production system. > > > > > > > > We are using a TE110P digium card (Primary rate) with a Telstraonramp> > 10. > > > > > > > > Sometimes when people call, on their end it doesn't seem to connect. > > On our end, we get caller id, it passes ok to the sip phone but then > > no-one is there. > > > > > > > > Anyone have any similar problems and worked out how to solve it ? > > > > > > > > Thanks. > > > > > > > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Thanks for the response and information. The Asterisk version that I am using is Asterisk SVN-bweschke-polycom_acd_functions-r37228. I went one revision back using the following command: svn checkout -r37228 http://svn.digium.com/svn/asterisk/team/bweschke/polycom_acd_functions PolycomACD-07172006 With no results. I am not as familiar with svn as cvs. I am not sure if the -r option just labels or checks out the requested version. I will do some reading tonight on svn. I have install zaptel and libpri from the latest version of trunk. I am using a Polycom 601 SIP version 1.6.6.0036. The Polycom <reg> tag includes the following for line button one: reg.1.displayName="Helpdesk" reg.1.address="1000" reg.1.label="Agent" reg.1.type="private" reg.1.thirdPartyName="" reg.1.auth.userId="" reg.1.auth.password="1000" reg.1.server.1.address="" reg.1.server.1.port="" reg.1.server.1.transport="DNSnaptr" reg.1.server.2.transport="DNSnaptr" reg.1.server.1.expires="" reg.1.server.1.register="" reg.1.server.1.retryTimeOut="" reg.1.server.1.retryMaxCount="" reg.1.server.1.expires.lineSeize="" reg.1.acd-login-logout="1" reg.1.acd-agent-available="1" reg.1.ringType="2" reg.1.lineKeys="1" reg.1.callsPerLineKey="2" I assumed that the property reg.1.auth.userId="" is what you meant by not putting in a username on the Polycom. I tried it both ways with no luck. I set the server addrss in the Polycom sip.cfg file. The sip.conf entry for the Polycom looks like: [1000] type = friend secret = 1000 context = default callerid = "Helpdesk" <1000> accountcode = 1000 host = dynamic nat = no qualify = 1000 canreinvite = no disallow = all allow = ulaw dtmfmode = rfc2833 agentlogin = yes agentcbcontext = default I also have an agent defined in the agnt.conf as: agent => 2000,1234,Test Agent Thanks again for the assistance! Michael -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dean @ INKnBITs Sent: Monday, July 17, 2006 3:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom IP301 and Queues I had the same problems, first of all, what version of asterisk are you using? If you run the CLI whats the polycom_acd_functions verison 3xxxx. If you did a svn checkout http://........polycom_acd_function, then you most likely got the newest version. I had trouble with that. Have you installed and compiled the zaptel/libpri from the trunk? http://svn.digium.com/svn/zaptel/trunk and http://svn.digium.com/svn/libpri/trunk ? You need these for the ACD part. On the polycom setup, make sure the username field is blank and that set a password. In the Sip.conf, make sure the secret is the same as the polycom, and that you do not put a username= or a authname I can get you all the release/version numbers to download from the svn tomorrow when back in work. It would be easier to talk you through it when in front of the server, but I'm in the UK and the time differences might get in the way! Regards, Dean. ----- Original Message ----- From: "Michael Miller" <Michael.Miller@sungardhe.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Monday, July 17, 2006 6:56 PM Subject: RE: [asterisk-users] Polycom IP301 and Queues I have been unable to get this branch of asterisk to work properly. I can not get any SIP phone, Polycom or X-Lite, to register with the server. If, on the same server, I recompile and install Trunk the phones register properly. In doing this I made no changes to the conf files at all. I simply recompiled and reinstalled. Is there a trick to getting the phones to register? I made sure that the phone SIP config and the agent config did no overlap. The phone will register if I comment out the secret line. I have not tried getting the ACD functionality to work at this point in time...one issue at a time. Although this will be a big leap forward if it works and I would be willing to put up a bounty to move this forward. Thanks, Michael -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dean @ INKnBITs Sent: Monday, July 17, 2006 3:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom IP301 and Queues Do you have a soft button on the IP301? I use the 501 and it works fine, you do have to use the special asterisk code for it to work correctly. It lets me login, logout, make the agent available/unavailable. You can read about it at http://bugs.digium.com/view.php?id=6119 I found you must also use the trunk version of zaptel and libpri, and make sure you use auth on the phones in the config. Hope thats what you looking for, if so, any problems just ask, its just taken me 2 weeks to get it working great. Regards, Dean. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Julian Varanini Sent: 17 July 2006 00:15 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom IP301 and Queues Is there any way to use the polycom phones to log into and out of queues? So the polycom phone could show their current status in that queue? logged in / logged out for example. Thanks Julian ----------------------------------------> Subject: RE: [asterisk-users] PRI dropouts > From: pdhales@optusnet.com.au > To: asterisk-users@lists.digium.com > Date: Sat, 15 Jul 2006 20:47:17 +1000 > > > Hmm - I have had 2 bad PRI installs out of about 20, and both times it > was faulty wiring from the Telco. > But getting them to fix it can be a real struggle! > > > Paul Hales > Technical Manager > www.asteriskit.com.au > > > On Sat, 2006-07-15 at 12:23 +1000, James Sturges wrote: > > Have had L O T S of trouble like this, the settings zap config files > > seem to have to e exact, please send email to thinking@1am.com.auand> > I will send config files. > > > > > > > > Thanks > > > > > > > > James > > > > > > > > > >______________________________________________________________________> > From:asterisk-users-bounces@lists.digium.com > > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kevin > > Withnall > > Sent: Saturday, 15 July 2006 11:05 AM > > To: asterisk-users@lists.digium.com > > Subject: [asterisk-users] PRI dropouts > > > > > > > > > > Recently we cut over to using asterisk (trixbox 1.1.1) for our > > production system. > > > > > > > > We are using a TE110P digium card (Primary rate) with a Telstraonramp> > 10. > > > > > > > > Sometimes when people call, on their end it doesn't seem to connect. > > On our end, we get caller id, it passes ok to the sip phone but then > > no-one is there. > > > > > > > > Anyone have any similar problems and worked out how to solve it ? > > > > > > > > Thanks. > > > > > > > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Dean, Thank you for your help. I have it up and running. As soon as I get some free time lets chat about what we need going forward. I have some dollars to move this forward. If I can accommodate additional requirements, all the better. Michael -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dean @ INKnBITs Sent: Tuesday, July 18, 2006 3:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom IP301 and Queues The setup looks fine, I will run through what I did and the version, there might be an easier way. cd /usr/src svn checkout http://svn.digium.com/svn/asterisk/team/bweschke/polycom_acd_functions/ asterisk-poly -r 30432 this will checkout the 30432 release and put in the the asterisk-poly directory. cd /usr/src/asterisk-poly make clean make - I found you had to run make (2 or 3 times), it does come up on the screen and tells you to re-run. First run I think makes menuconfig, second.... can't remember. make mpg123 (if you want mp3 music on hold) make install The only problem I can find in this release is the meetme (conference centre) does not compile, (but ACD does) and in the newer version the meetme works but not ACD. So I'm going to have two servers one for ACD on old software and one for conference on new software. Not great but least it works. Hope that helps. Regards, Dean. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Michael Miller Sent: 17 July 2006 23:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom IP301 and Queues Thanks for the response and information. The Asterisk version that I am using is Asterisk SVN-bweschke-polycom_acd_functions-r37228. I went one revision back using the following command: svn checkout -r37228 http://svn.digium.com/svn/asterisk/team/bweschke/polycom_acd_functions PolycomACD-07172006 With no results. I am not as familiar with svn as cvs. I am not sure if the -r option just labels or checks out the requested version. I will do some reading tonight on svn. I have install zaptel and libpri from the latest version of trunk. I am using a Polycom 601 SIP version 1.6.6.0036. The Polycom <reg> tag includes the following for line button one: reg.1.displayName="Helpdesk" reg.1.address="1000" reg.1.label="Agent" reg.1.type="private" reg.1.thirdPartyName="" reg.1.auth.userId="" reg.1.auth.password="1000" reg.1.server.1.address="" reg.1.server.1.port="" reg.1.server.1.transport="DNSnaptr" reg.1.server.2.transport="DNSnaptr" reg.1.server.1.expires="" reg.1.server.1.register="" reg.1.server.1.retryTimeOut="" reg.1.server.1.retryMaxCount="" reg.1.server.1.expires.lineSeize="" reg.1.acd-login-logout="1" reg.1.acd-agent-available="1" reg.1.ringType="2" reg.1.lineKeys="1" reg.1.callsPerLineKey="2" I assumed that the property reg.1.auth.userId="" is what you meant by not putting in a username on the Polycom. I tried it both ways with no luck. I set the server addrss in the Polycom sip.cfg file. The sip.conf entry for the Polycom looks like: [1000] type = friend secret = 1000 context = default callerid = "Helpdesk" <1000> accountcode = 1000 host = dynamic nat = no qualify = 1000 canreinvite = no disallow = all allow = ulaw dtmfmode = rfc2833 agentlogin = yes agentcbcontext = default I also have an agent defined in the agnt.conf as: agent => 2000,1234,Test Agent Thanks again for the assistance! Michael -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dean @ INKnBITs Sent: Monday, July 17, 2006 3:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom IP301 and Queues I had the same problems, first of all, what version of asterisk are you using? If you run the CLI whats the polycom_acd_functions verison 3xxxx. If you did a svn checkout http://........polycom_acd_function, then you most likely got the newest version. I had trouble with that. Have you installed and compiled the zaptel/libpri from the trunk? http://svn.digium.com/svn/zaptel/trunk and http://svn.digium.com/svn/libpri/trunk ? You need these for the ACD part. On the polycom setup, make sure the username field is blank and that set a password. In the Sip.conf, make sure the secret is the same as the polycom, and that you do not put a username= or a authname I can get you all the release/version numbers to download from the svn tomorrow when back in work. It would be easier to talk you through it when in front of the server, but I'm in the UK and the time differences might get in the way! Regards, Dean. ----- Original Message ----- From: "Michael Miller" <Michael.Miller@sungardhe.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Monday, July 17, 2006 6:56 PM Subject: RE: [asterisk-users] Polycom IP301 and Queues I have been unable to get this branch of asterisk to work properly. I can not get any SIP phone, Polycom or X-Lite, to register with the server. If, on the same server, I recompile and install Trunk the phones register properly. In doing this I made no changes to the conf files at all. I simply recompiled and reinstalled. Is there a trick to getting the phones to register? I made sure that the phone SIP config and the agent config did no overlap. The phone will register if I comment out the secret line. I have not tried getting the ACD functionality to work at this point in time...one issue at a time. Although this will be a big leap forward if it works and I would be willing to put up a bounty to move this forward. Thanks, Michael -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dean @ INKnBITs Sent: Monday, July 17, 2006 3:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom IP301 and Queues Do you have a soft button on the IP301? I use the 501 and it works fine, you do have to use the special asterisk code for it to work correctly. It lets me login, logout, make the agent available/unavailable. You can read about it at http://bugs.digium.com/view.php?id=6119 I found you must also use the trunk version of zaptel and libpri, and make sure you use auth on the phones in the config. Hope thats what you looking for, if so, any problems just ask, its just taken me 2 weeks to get it working great. Regards, Dean. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Julian Varanini Sent: 17 July 2006 00:15 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom IP301 and Queues Is there any way to use the polycom phones to log into and out of queues? So the polycom phone could show their current status in that queue? logged in / logged out for example. Thanks Julian ----------------------------------------> Subject: RE: [asterisk-users] PRI dropouts > From: pdhales@optusnet.com.au > To: asterisk-users@lists.digium.com > Date: Sat, 15 Jul 2006 20:47:17 +1000 > > > Hmm - I have had 2 bad PRI installs out of about 20, and both times it > was faulty wiring from the Telco. > But getting them to fix it can be a real struggle! > > > Paul Hales > Technical Manager > www.asteriskit.com.au > > > On Sat, 2006-07-15 at 12:23 +1000, James Sturges wrote: > > Have had L O T S of trouble like this, the settings zap config files > > seem to have to e exact, please send email to thinking@1am.com.auand> > I will send config files. > > > > > > > > Thanks > > > > > > > > James > > > > > > > > > >______________________________________________________________________> > From:asterisk-users-bounces@lists.digium.com > > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kevin > > Withnall > > Sent: Saturday, 15 July 2006 11:05 AM > > To: asterisk-users@lists.digium.com > > Subject: [asterisk-users] PRI dropouts > > > > > > > > > > Recently we cut over to using asterisk (trixbox 1.1.1) for our > > production system. > > > > > > > > We are using a TE110P digium card (Primary rate) with a Telstraonramp> > 10. > > > > > > > > Sometimes when people call, on their end it doesn't seem to connect. > > On our end, we get caller id, it passes ok to the sip phone but then > > no-one is there. > > > > > > > > Anyone have any similar problems and worked out how to solve it ? > > > > > > > > Thanks. > > > > > > > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users