Re: [Asterisk-Users] Stutter dialtone but no messages
On Saturday 22 November 2003 12:59 am, SloopJohnB wrote: > There must be a code to turn off message notify?? I'm sure there is. But I have been unable to find the documentation for it. I can only assume that when I deleted a previous last message, the flag wasn't changed. Is this a coding issue ??? Shouldn't there be a check to see if there are new messages before the flag is set ??? ie. atomic Regards...Martin -- Don't steal; thou'lt never thus compete successfully in business. Cheat. -- Ambrose Bierce ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Stutter dialtone but no messages
There must be a code to turn off message notify?? Thank You, John Berry, Owner > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of marrandy > Sent: Friday, November 21, 2003 9:43 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Stutter dialtone but no messages > > Hello, yet again. > > I have 3 voicemail boxes. > > They worked good until today. ie. I had the stutter/dialtonme AND had a > message. > > Now, I am getting the stutter/dialtone, that tells me I have a message, > But > when I check both the new and old messages on each of the 3 mailbox > accounts, > there are no messages. > > How do I resolve this problem > > Regards...Martin > -- > If some day we are defeated, well, war has its fortunes, good and bad. > -- Commander Kor, "Errand of Mercy", stardate 3201.7 > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > BEGIN:VCARD VERSION:2.1 N:Berry;John FN:John Berry (Business Fax) ORG:Lantex Voice & Data Systems TITLE:President TEL;WORK;VOICE:(619) 690-4428 TEL;CELL;VOICE:(619) 247-3685 TEL;WORK;FAX:(619) 741-6823 ADR;WORK:;;312 Billow Dr;San Diego;Ca;92114;United States of America LABEL;WORK;ENCODING=QUOTED-PRINTABLE:312 Billow Dr=0D=0ASan Diego, Ca 92114=0D=0AUnited States of America URL;WORK:http://www.lantex.com EMAIL;PREF;FAX:John [EMAIL PROTECTED] (619) 741-6823 REV:20031015T030219Z END:VCARD
[Asterisk-Users] Stutter dialtone but no messages
Hello, yet again. I have 3 voicemail boxes. They worked good until today. ie. I had the stutter/dialtonme AND had a message. Now, I am getting the stutter/dialtone, that tells me I have a message, But when I check both the new and old messages on each of the 3 mailbox accounts, there are no messages. How do I resolve this problem Regards...Martin -- If some day we are defeated, well, war has its fortunes, good and bad. -- Commander Kor, "Errand of Mercy", stardate 3201.7 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Iconnect (DeltaThree) config on Asterisk
I'm registered on deltathree I think but then I try to call the number I receive a strange message like "check the number and try again". Some sugestions??? Chris HARIGA - Original Message - From: "Brian Capouch" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, November 21, 2003 11:31 PM Subject: Re: [Asterisk-Users] Iconnect (DeltaThree) config on Asterisk > Chris HARIGA wrote: > > Hi, > > > > Is anyone using the iconnect on Asterisk to receive and to place calls? > > > > I use it for both incoming and outbound calls. > > My phones use private IP only; my asterisk server is private but is > NATted to public. > > The only problem I have with it is a very slight delay before it passes > audio when the other side answers. It freaks people out because they > don't hear me in the instant after they say hello. > > Don't know why it does it, but it has been that way since the beginning. > > Otherwise, very good quality when in a jittery situation. It is better > than Vonage in that respect IMO. > > I don't use it for "important" calls because of the limitation mentioned > above, unless they're going to be long ones :-) > > B. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Iconnect (DeltaThree) config on Asterisk
Can U send me the samples of you .conf files please? Chris HARIGA - Original Message - From: "Brian Capouch" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, November 21, 2003 11:31 PM Subject: Re: [Asterisk-Users] Iconnect (DeltaThree) config on Asterisk > Chris HARIGA wrote: > > Hi, > > > > Is anyone using the iconnect on Asterisk to receive and to place calls? > > > > I use it for both incoming and outbound calls. > > My phones use private IP only; my asterisk server is private but is > NATted to public. > > The only problem I have with it is a very slight delay before it passes > audio when the other side answers. It freaks people out because they > don't hear me in the instant after they say hello. > > Don't know why it does it, but it has been that way since the beginning. > > Otherwise, very good quality when in a jittery situation. It is better > than Vonage in that respect IMO. > > I don't use it for "important" calls because of the limitation mentioned > above, unless they're going to be long ones :-) > > B. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Iconnect (DeltaThree) config on Asterisk
Chris HARIGA wrote: Hi, Is anyone using the iconnect on Asterisk to receive and to place calls? I use it for both incoming and outbound calls. My phones use private IP only; my asterisk server is private but is NATted to public. The only problem I have with it is a very slight delay before it passes audio when the other side answers. It freaks people out because they don't hear me in the instant after they say hello. Don't know why it does it, but it has been that way since the beginning. Otherwise, very good quality when in a jittery situation. It is better than Vonage in that respect IMO. I don't use it for "important" calls because of the limitation mentioned above, unless they're going to be long ones :-) B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Iconnect (DeltaThree) config on Asterisk
Hi, Is anyone using the iconnect on Asterisk to receive and to place calls? Best regards, Chris HARIGA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of PBX > Sent: Friday, November 21, 2003 6:33 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy > > > Ok... I know I have asked this question before, but have never gotten an > answer... When I press the hold button on my phone, should the caller > hear music just like when I park the caller or transfer them to another > extension? It depends... If the button uses a function of the phone to hold the call(ie keeps the call active but mutes the speaker & mic),then you will not hear music on hold from *. I have some phones that can operate this way. Andy > > Please assist... > > -gcc > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1 (Alpha)
Here's the structure for the monitor.conf file: [1101] Extension Number (from extensions.conf in Asterisk) UserName=Blah Blah Label. Simply sets the caption for the button. Technology=SIP Technology used for stations (SIP, MGCP, Zap, etc.) DeviceID=1101 Device identifier (from sip.conf in this case) All of the Technology values are normal asterisk values except for APP, which is an application (like Voicemail or MOH or MeetMe) and PSTN, which is a number outside of the Asterisk inside dial plan. I hope this helps. Remember that for PSTN and APP values, the bracketed Extension number and the DeviceID need to be the same. Regards, Steve > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of zoa > Sent: Friday, November 21, 2003 7:41 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1 > (Alpha) > > Could you give me some explanation on how to use the configuration file ? > > I always get INVALID channels if i click on the red icon next to my name. > (Maybe i should use a numeric context or use numeric user names?) > > Tool looks great, this will be a very cool asterisk addition. > > zoa. > > At 16:43 21/11/2003 -0600, you wrote: > >I think the script host gets installed with Windows explorer. If you > >don't have it, you can use the DLL in the dlls download: > > > >http://www.sokol-associates.com/Downloads/Dlls.zip > > > >Hope that helps. > > > >Thanks, > > > >Steve > > > > > -Original Message- > > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > > [EMAIL PROTECTED] On Behalf Of Walker Haddock > > > Sent: Friday, November 21, 2003 4:21 PM > > > To: [EMAIL PROTECTED] > > > Subject: Re: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1 > > > (Alpha) > > > > > > > here: > > > > http://www.sokol-associates.com/Downloads/AstMgr.zip > > > > > > > > It's written in VB6 (yes - barf, gag, whatever). The only thing > > > > required beyond the integral VB6 controls is the Windows Scripting > > > > Runtime which most PCs should have. I will work on an installable > > > > version soon. I may also port it to something more cross-platform. > > > > Please bear with me as I am just learning Gnome/GTK/X-windows. > > > Steve, how do you know if the Windows Scripting Runtime is installed > >in > > > Windows XP Pro? > > > Where do you get it from and how should it be installed? > > > > > > Thanks, Walker > > > > > > -- > > > DataCrest, Inc. -- Technically Superior > >** > > > Walker Haddock http://www.datacrest.com > > > DataCrest, Inc.e-mail: [EMAIL PROTECTED] > > > 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 > > > Birmingham, AL 35216 fax: 1-205-823-7838 > > > > >*** > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo Cancellation, TDMoE fails, X100P works
We have been pretty much able to solve our echo problems, except for the primary mode in which we desire to operate our system. See system diagram at bottom. Prior to making adjustments to cancel echos (all echocancel=no): Call TypeResult (Before) ---- CP <- LEC PRI * TDMoE * FXO -> AP no echo CP <- LEC PRI * TDMoE * SIP -> sip phoneno echo AP <- LEC PRI * TDMoE * FXO -> AP moderate echo on both ends AP <- LEC PRI * TDMoE * SIP -> sip phonebad echo on sip phone no echo on other end AP <- LEC analog FXO * SIP -> sip phonebad echo on sip phone no echo on other end AP <- LEC analog FXO * FXO -> LEC -> APslight echo on both ends (AP stands for analog phone and CP stands for cell phone) After changes to the * with the X100P cards; latest CVS (11/21/03), echocancel=yes and echocancelwhenbridged=yes in zapata.conf, KFLAGS+=-DECHO_CAN_MARK2 and KFLAGS+=-DAGGRESSIVE_SUPPRESSOR in /zaptel/Makefile: Call TypeResult (After) --- CP <- LEC PRI * TDMoE * FXO -> AP no echo CP <- LEC PRI * TDMoE * SIP -> sip phoneno echo AP <- LEC PRI * TDMoE * FXO -> AP slight echo on both ends AP <- LEC PRI * TDMoE * SIP -> sip phoneBAD echo on sip phone (STILL) no echo on other end AP <- LEC analog FXO * SIP -> sip phoneshort echo on sip phone, gets much better after a few seconds no echo on other end AP <- LEC analog FXO * FXO -> LEC -> APnegligible echo (AP stands for analog phone and CP stands for cell phone) It appears that echo cancel does work if an FXO (X100P) channel is involved in the call but does not work for TDMoE <--> * <--> SIP. The TDMoE is configured as a pri_cpe in zapata.conf on the * box with the X100P boards, and has the same echocancel configuration as the X100P channels. Any ideas as to why echo cancel works for the X100P channels but does not work for the TDMoE channels? Do the echo cancel algorithms even work at all on TDMoE channels? I could detect NO difference with echo canceling on for a TDMoE call through * to a sip phone while there was a big difference with echo canceling on for an FXO call through * to a sip phone. I have not done echo canceling on the * box connected by PRI to the LEC because, as I understand, it may cause problems for the modem connections that go through that machine. Also, it is clear that the echo heard on the sip phone originates with the the analog phone at the far end of the connection. As was pointed out in an earlier posting (and my tests confirm), digital cell phones don't generate an echo. System Diagram: +- PRI --> modem bank | LEC <-- PRI --> * with <--+ T400P <--+ | (pri) +- TDMoE --> * with <-- SIP --> sip phones LEC <-- analog line ---> X100P Gary Mart PS The modem connections have been working just fine after some initial problems and a CVS update. Thanks Mark! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI problems
It seems that there's a non-printable character at the beginning of the DNIS stream I'm getting from the SUMA 4 switch. Once I chopped that off, everything works right. > Hi James, > > Try to do > exten => _8005095639,1,Agi(ivr-main.pl) > > > Quoting James Sharp <[EMAIL PROTECTED]>: >> *CLI> show dialplan nonauthenticated >> [ Context 'nonauthenticated' created by 'pbx_config' ] >> '8005095639' => 1. AGI(ivr-main.pl) > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1 (Alpha)
Could you give me some explanation on how to use the configuration file ? I always get INVALID channels if i click on the red icon next to my name. (Maybe i should use a numeric context or use numeric user names?) Tool looks great, this will be a very cool asterisk addition. zoa. At 16:43 21/11/2003 -0600, you wrote: I think the script host gets installed with Windows explorer. If you don't have it, you can use the DLL in the dlls download: http://www.sokol-associates.com/Downloads/Dlls.zip Hope that helps. Thanks, Steve > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Walker Haddock > Sent: Friday, November 21, 2003 4:21 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1 > (Alpha) > > > here: > > http://www.sokol-associates.com/Downloads/AstMgr.zip > > > > It's written in VB6 (yes - barf, gag, whatever). The only thing > > required beyond the integral VB6 controls is the Windows Scripting > > Runtime which most PCs should have. I will work on an installable > > version soon. I may also port it to something more cross-platform. > > Please bear with me as I am just learning Gnome/GTK/X-windows. > Steve, how do you know if the Windows Scripting Runtime is installed in > Windows XP Pro? > Where do you get it from and how should it be installed? > > Thanks, Walker > > -- > DataCrest, Inc. -- Technically Superior ** > Walker Haddock http://www.datacrest.com > DataCrest, Inc.e-mail: [EMAIL PROTECTED] > 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 > Birmingham, AL 35216 fax: 1-205-823-7838 > *** > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] status of distinctive ring
In the archives, I have found discussion of detection of incoming distinctive ring patterns dating back almost 2 years, but I have not been able to figure out whether it is partially functional or even whether it is definitely planned. When I call Asterisk with a distinctive ring, it consistently logs the following: NOTICE[1217669936]: File chan_zap.c, Line 4421 (ss_thread): Got event (Ring/Answered)... However, looking at the code, it appears that this may just be a notice that Asterisk detected the callerid in spite of the distinctive ring. So, I'd be appreciative if anyone could comment on whether there is currently any way to detect incoming distinctive ring, or whether it is planned. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI problems
Hi James, Try to do exten => _8005095639,1,Agi(ivr-main.pl) Quoting James Sharp <[EMAIL PROTECTED]>: > *CLI> show dialplan nonauthenticated > [ Context 'nonauthenticated' created by 'pbx_config' ] > '8005095639' => 1. AGI(ivr-main.pl) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MOH - Hold Button - I think I'm going crazy
Ok... I know I have asked this question before, but have never gotten an answer... When I press the hold button on my phone, should the caller hear music just like when I park the caller or transfer them to another extension? Please assist... -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI problems
*CLI> show dialplan nonauthenticated [ Context 'nonauthenticated' created by 'pbx_config' ] '8005095639' => 1. AGI(ivr-main.pl) [pbx_config] '8005095640' => 1. AGI(ivr-main.pl) [pbx_config] '8005095641' => 1. AGI(ivr-main.pl) [pbx_config] > check 'show dialplan nonauthenticated' > > regards > Martin > > On Fri, 21 Nov 2003, James Sharp wrote: > >> I've got a couple of PRIs coming in from a SUMA 4 switch with some 800 >> numbers routed through it. >> >> When the calls come in, I get the following message on the console and >> the >> call never makes it through: >> >> (800 number is fake) >> >> Extension '8005551212' in context 'nonauthenticated' from '232102749585' >> does not exist. Rejecting the call on span 4, channel 1. >> >> I do have the following line in extensions.conf in [nonauthenticated] >> >> exten => 8005551212,1,AGI,ivr-main.pl >> >> ___ >> Asterisk-Users mailing list >> [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI problems
check 'show dialplan nonauthenticated' regards Martin On Fri, 21 Nov 2003, James Sharp wrote: > I've got a couple of PRIs coming in from a SUMA 4 switch with some 800 > numbers routed through it. > > When the calls come in, I get the following message on the console and the > call never makes it through: > > (800 number is fake) > > Extension '8005551212' in context 'nonauthenticated' from '232102749585' > does not exist. Rejecting the call on span 4, channel 1. > > I do have the following line in extensions.conf in [nonauthenticated] > > exten => 8005551212,1,AGI,ivr-main.pl > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI problems
I've got a couple of PRIs coming in from a SUMA 4 switch with some 800 numbers routed through it. When the calls come in, I get the following message on the console and the call never makes it through: (800 number is fake) Extension '8005551212' in context 'nonauthenticated' from '232102749585' does not exist. Rejecting the call on span 4, channel 1. I do have the following line in extensions.conf in [nonauthenticated] exten => 8005551212,1,AGI,ivr-main.pl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Which ISDM BRI Card for Asterisk?
WipeOut <[EMAIL PROTECTED]> said: >I would recommend you dump i4l and use a CAPI card with the chan_capi >driver.. The cheap solution is a AVM FritzPCI card(this is what I use).. >The other solution is the either the Eicon or AVM active cards.. > I have experienced lots of bus hangups with the Fritz!, where the card doesn't see anything happening on the ISDN bus anymore. For my production system, I've now ordered a AVM B1, which hopefully works better. -- Cees de Groot http://www.tric.nl <[EMAIL PROTECTED]> tric, the new way helpdesk/ticketing software, VoIP/CTI, web applications, custom development ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1 (Alpha)
I think the script host gets installed with Windows explorer. If you don't have it, you can use the DLL in the dlls download: http://www.sokol-associates.com/Downloads/Dlls.zip Hope that helps. Thanks, Steve > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Walker Haddock > Sent: Friday, November 21, 2003 4:21 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1 > (Alpha) > > > here: > > http://www.sokol-associates.com/Downloads/AstMgr.zip > > > > It's written in VB6 (yes - barf, gag, whatever). The only thing > > required beyond the integral VB6 controls is the Windows Scripting > > Runtime which most PCs should have. I will work on an installable > > version soon. I may also port it to something more cross-platform. > > Please bear with me as I am just learning Gnome/GTK/X-windows. > Steve, how do you know if the Windows Scripting Runtime is installed in > Windows XP Pro? > Where do you get it from and how should it be installed? > > Thanks, Walker > > -- > DataCrest, Inc. -- Technically Superior ** > Walker Haddock http://www.datacrest.com > DataCrest, Inc.e-mail: [EMAIL PROTECTED] > 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 > Birmingham, AL 35216 fax: 1-205-823-7838 > *** > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Struggling with grandstream sip to asterisk
Correct, never the less I do not see the same issues other are reporting with SIP not staying registered I only saw this after I had installed 20 phones. When I had only one phone installed during the test and setup phase, it never happened. Either it is rather infrequent, or you have the re-registration interval set really low, or it has to do with more than one phone trying to register at once. One never knows! or the SIP Info issue with the DTMF Alas, maybe I was just too damn impatient to struggle with getting anything but in-band to work. How can I verify this registration issue, its werid that it just seems to work in my install ... Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Outline For Asterisk Book - Please Review & Comment
Will, I have added Clustering/TDMoE and ENUM as two additional topics in the Advanced Configuration section. I have also added the ISDN BRI/CAPI stuff to the Add-On/Optional Components/Hardware chapter of Section 1. Thanks for the suggestions. I would love to have whatever you can write up on all of these topics. Please send as plain text or RTF. Thanks! Steve > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] > Sent: Friday, November 21, 2003 2:09 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Outline For Asterisk Book - Please Review & > Comment > > Great idea! > > I would also like to see some discussion of ISDN BRI and CAPI > hardware, as well as some discussion of "distributed asterisk" > what happens when you start deploying a network of many asterisk > boxes, how to do forwarding and switching properly, TDMoE as well as > E164 enum call routing -- on the latter I can perhaps contribute > some text now that I have gotten it to work properly ;) > > Cheers, > Will > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1 (Alpha)
> here: > http://www.sokol-associates.com/Downloads/AstMgr.zip > > It's written in VB6 (yes - barf, gag, whatever). The only thing > required beyond the integral VB6 controls is the Windows Scripting > Runtime which most PCs should have. I will work on an installable > version soon. I may also port it to something more cross-platform. > Please bear with me as I am just learning Gnome/GTK/X-windows. Steve, how do you know if the Windows Scripting Runtime is installed in Windows XP Pro? Where do you get it from and how should it be installed? Thanks, Walker -- DataCrest, Inc. -- Technically Superior ** Walker Haddock http://www.datacrest.com DataCrest, Inc.e-mail: [EMAIL PROTECTED] 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'SIP'
[EMAIL PROTECTED] wrote:
I recently moved my Asterisk configuration to a new server and re-built
Asterisk from CVS. Now, I'm experiencing the following issue with SIP:
Executing Dial("Zap/1-1", "SIP/100|20") in new stack
NOTICE[-1232077904]: File app_dial.c, Line 518 (dial_exec): Unable to
create channel of type 'SIP'
== Everyone is busy at this time
Has anyone seen this issue before?
cvs update again
Jeremy McNamara
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RE: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1 (Alpha)
>Can I have some installation steps on this > >it looks very good.. > > >Alex Unfortunately I haven't had a chance to work up an install. If you have the core VB runtime file (MSVBVM60.dll) you will only need a couple of additional files: Windows Scripting Runtime: scrrun.dll MS Winsock Control: mswinsck.ocx MS Common Controls: mscomctl.ocx Here is a link to a zip with those files: http://www.sokol-associates.com/Downloads/Dlls.zip First: Dump all 3 files into your WINDOWS\System32 (WINNT\System32) folder. Second: Register the files using the regsvr32.exe utility: Start->Run->regsvr32.exe "C:\Winnt\System32\scrrun.dll" regsvr32.exe Start->Run->"C:\Winnt\System32\mswinsck.ocx" Start->Run->regsvr32.exe "C:\Winnt\System32\mscomctl.ocx" (Replace Winnt\System32 with your system folder) Before you start the program, make sure that the monitor module is loaded and running on your Asterisk. There are plenty of posts on setting this up. Here's the manager.conf file on my machine: = MANAGER.CONF FILE ; ; Asterisk Call Management support ; [general] enabled = yes port = 5038 bindaddr = 0.0.0.0 [steve] secret = [SECRET WORD HERE] ;deny=0.0.0.0/0.0.0.0 ;permit=209.16.236.73/255.255.255.0 read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user = END OF MANAGER.CONF FILE = Now you should be able to execute Astman.exe from wherever you have it stored. It will automatically detect that it is a new installation and pop open the configuration dialog. Configure the following: Device Name = SIP/1100 (or) Zap/3 (or) ... (the channel name you want to define as "your phone") Mailbox = 1100 (your mailbox) Asterisk = sip.sokol-associates.com (enter the IP or name for your Asterisk installation here) Port = 5038 (enter the Asterisk manager port here -- defaults to 5038) Debug Level = 3 (set between 0 and 4 for vb debugging) User Name: steven (Asterisk manager login as defined in manager.conf) Password: mysecretword (Secret as defined in manager.conf) Keep Command Results controls whether command results (in the command output window) are concatenated (and "kept") or replaced with each command. Automatically connect on startup is not currently in use (I think) and would allow you to manually establish your connection after the toolbar deploys. Debug Window allows you to display the manager traffic in realtime. The system will attempt to load the monitor.conf file once the settings have been properly set. You can manually edit that file as described in my original post. I will work on the editor dialog over the weekend. Note: After you enter the basic configuration info, it will open but it appears not to connect. You may need to close the app and re-start it. I will track this as a bug and see if I can't fix it. I hope this helps. Comments Please! Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outline For Asterisk Book - Please Review & Comment
A little more about Linux commands to help troubleshoot installations. Also using gnomephone would be nice. -- Original Message -- From: "Steven Sokol" <[EMAIL PROTECTED]> Reply-To: [EMAIL PROTECTED] Date: Fri, 21 Nov 2003 11:34:35 -0600 >Asterisk Users > >In an attempt to help Asterisk move forward, a number of us have decided >to create a book. It would initially be released as an "ebook" that >could be sent to newbies to help them up the rather steep learning >curve. Ultimately I would like to see it published and sold in >bookstores (preferably by O'Reilly & Co.). > >Below is the outline for the book. We REALLY need as much input as we >can get. I would like to completely flesh-out the outline, then I would >like to start accepting submissions from the user community for each of >the sections/chapters/topics covered in the outline. > >I have to stress here that I AM NOT AN ASTERISK GURU. I need help from >the real gurus (especially: Steven Critchfield, John Todd, Tilghman >Lesher, Olle Johansson, and where possible/necessary Mark and Martin). > >If this works, it will help Asterisk achieve the same kind of global >success as Apache, Samba, and other Linux staples. If you want to chat, >I am lurking in the #asterisk-doc channel on Freenode IRC. I'm >'ssokol'. Others on this project (so far) are Jared Smith and Leif >Madson. > >A "living" copy of this outline can be found at: >http://www.sokol-associates.com/outline.htm . I will try to update it >daily with your suggestions. It will also be the basis for the project >outline (completion %, assignments, etc.). > >Thanks, > >Steve Sokol >Sokol & Associates, LLC > >[Outline Guide] >The following outline describes the layout for the book. > >1. <- Section > a. <- Chapter > 1) <- Sub-Chapter > i. <- Topic Heading > *. <- Sidebar Heading > 1} <- Graphic or Chart > 1> <- Table > >[Outline] >1. Introduction to Asterisk > a. Introductory letter from Mark Spencer > 1) Whatever Mark has to say... > 2) Digium Reference Information > i. Web Site > ii. Phone Number > b. The Business Case For Asterisk > [Somebody From The Business Side Writes This] > c. General concept of asterisk > 1) Asterisk: Swiss Army Knife of Telephony > 2) PBX, IVR, ACD > 3) What To Expect > i. Asterisk Is Not A Turnkey System > ii. Don't Like It? Change It Yourself. > iii. Opensource, GPL and LGPL Licensing > d. Asterisk architecture > 1) The Big Picture > 2) Channels > 3) Codec Conversions > 4) Etc. > e. Key components > 1) Asterisk software > i. Asterisk (Main PBX & Channels) > ii. Zaptel (Drivers for Zaptel Hardware) > iii. Libpri (ISDN PRI Drivers for Zaptel) > 2) Zaptel Hardware > i. Overview > ii. X100P - Single Port FXO Line Interface > iii. S100U - Single Port FXS USB Interface > iv. TDM400P - 4 Port FXS Analog Interface > v. T100P - Single Span T1/E1 Interface > vi. TE410P - Quad-Span T1/E1 Interface > 3) Channels > i. Zaptel Devices/Channels > ii. The IAX Protocol > iii. SIP > iv. MGCP > v. Skinny > vi. H323 > 4) Applications > i. Dial and Other Basics > ii. Voicemail > iii. Dial-Plan Scripting > 5) Extensibility > i. AGI > ii. Custom Applications > f. Add-On/Optional Components > 1) Software > i. Gnophone > ii. VoIP Soft Phones > iii. DIAX > iv. Gastman > v. Open H.323 > 2) Hardware > i. VoIP Hard-Phones > ii. VoIP Gateways > ii. Channel Banks > >2. Installing Asterisk > *. Asterisk Quickstart > 1) Install PC Hardware > 2) Download Asterisk Software > 3) Build Asterisk > 4) Install Asterisk > 5) Configure Autostart > > a. Requirements > *) Picking A Solid System > 1) PC Hardware Requirements > i. SOHO/Residential
[Asterisk-Users] Unable to create channel of type 'SIP'
I recently moved my Asterisk configuration to a new server and re-built
Asterisk from CVS. Now, I'm experiencing the following issue with SIP:
Executing Dial("Zap/1-1", "SIP/100|20") in new stack
NOTICE[-1232077904]: File app_dial.c, Line 518 (dial_exec): Unable to
create channel of type 'SIP'
== Everyone is busy at this time
Has anyone seen this issue before?
Thanks.
___
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Re: [Asterisk-Users] TE410P Errors under load
Again, can you please confirm you are neither running serial console *nor* graphical console (e.g. framebuffer). If you can call into the office we can ssh in and take a look at the configuration. Mark On Fri, 21 Nov 2003, Scott Stingel wrote: > (Apologies: starting this as a new thread - I'm in a new location.) > > Mark- > > Ran latest CVS from today, and sorry to report little improvement with the > changes you made. Running my IVR load test from one span to another on same > system. > > I'm initiating calls on the 2nd span, these are channels 32-62 (skipping the > D channel 47), and receiving on the cooresponding channels on the 1st span, > channels 1-31 (D channel is 16). When I run these 30 channels, I get > hundreds of WARNING's (excerpt below). I'm using a short crossover cable > (1,2 <=> 4,5) > > When I run only 10-15 channels, I get few or no WARNING's... > > Note read error on channel 252(?) > Why is asterisk retransmitting so many frames on each error? > > These symptoms are identical to those that I've been getting from my > customer in the field, while connected to a DMS-100, handling real traffic. > > THANKS > Scott > > WARNING[1167272128]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: Read on > 252 failed: Unknown error 500 > WARNING[1167272128]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: Read on > 252 failed: Unknown error 500 > WARNING[1167272128]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: Read on > 252 failed: Unknown error 500 > WARNING[1167272128]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: Read on > 252 failed: Unknown error 500 > WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: Read on > 253 failed: Unknown error 500 > WARNING[1167272128]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: Read on > 252 failed: Unknown error 500 > WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: Read on > 253 failed: Unknown error 500 > WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got > reject for frame 97, retransmitting frame 97 now, updating n_r! > WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got > reject for frame 97, retransmitting frame 98 now, updating n_r! > WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got > reject for frame 97, retransmitting frame 99 now, updating n_r! > WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got > reject for frame 97, retransmitting frame 100 now, updating n_r! > WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got > reject for frame 97, retransmitting frame 101 now, updating n_r! > WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got > reject for frame 97, retransmitting frame 102 now, updating n_r! > WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got > reject for frame 97, retransmitting frame 103 now, updating n_r! > WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got > reject for frame 97, retransmitting frame 104 now, updating n_r! > WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got > reject for frame 97, retransmitting frame 105 now, updating n_r! > WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got > reject for frame 97, retransmitting frame 106 now, updating n_r! > WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got > reject for frame 97, retransmitting frame 107 now, updating n_r! > WARNING[1167272128]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: Read on > 252 failed: Unknown error 500 > --- > > Scott M. Stingel > Emerging Voice Technology Inc. > Palo Alto, California and London, England > > URL:www.evtmedia.com > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] making outside call with sip phone
Do you have some sort of device that will allow you to make a call to an outside line? - Original Message - From: "Steve Bradwell" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, November 21, 2003 4:12 PM Subject: [Asterisk-Users] making outside call with sip phone Hi All, I have a grandstream sip phone which I have figured out and configured to make internal calls. How do I now configure asterisk to allow this phone to make an outside call? Thanks in advance, Steve. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users [EMAIL PROTECTED])fjåËbú?jË^®+$ºÇ«
asterisk-users@lists.digium.com
Steven Sokol wrote: An excellent suggestion. Would Digium be willing to host this, or should we find a different host? Personally I would like to keep everything on the Digium server, but I can understand if Mark doesn't want to cover the additional bandwidth. What do you think is best? I have a few gigabit to spare, if needed. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE410P Errors under load
(Apologies: starting this as a new thread - I'm in a new location.) Mark- Ran latest CVS from today, and sorry to report little improvement with the changes you made. Running my IVR load test from one span to another on same system. I'm initiating calls on the 2nd span, these are channels 32-62 (skipping the D channel 47), and receiving on the cooresponding channels on the 1st span, channels 1-31 (D channel is 16). When I run these 30 channels, I get hundreds of WARNING's (excerpt below). I'm using a short crossover cable (1,2 <=> 4,5) When I run only 10-15 channels, I get few or no WARNING's... Note read error on channel 252(?) Why is asterisk retransmitting so many frames on each error? These symptoms are identical to those that I've been getting from my customer in the field, while connected to a DMS-100, handling real traffic. THANKS Scott WARNING[1167272128]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: Read on 252 failed: Unknown error 500 WARNING[1167272128]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: Read on 252 failed: Unknown error 500 WARNING[1167272128]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: Read on 252 failed: Unknown error 500 WARNING[1167272128]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: Read on 252 failed: Unknown error 500 WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: Read on 253 failed: Unknown error 500 WARNING[1167272128]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: Read on 252 failed: Unknown error 500 WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: Read on 253 failed: Unknown error 500 WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got reject for frame 97, retransmitting frame 97 now, updating n_r! WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got reject for frame 97, retransmitting frame 98 now, updating n_r! WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got reject for frame 97, retransmitting frame 99 now, updating n_r! WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got reject for frame 97, retransmitting frame 100 now, updating n_r! WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got reject for frame 97, retransmitting frame 101 now, updating n_r! WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got reject for frame 97, retransmitting frame 102 now, updating n_r! WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got reject for frame 97, retransmitting frame 103 now, updating n_r! WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got reject for frame 97, retransmitting frame 104 now, updating n_r! WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got reject for frame 97, retransmitting frame 105 now, updating n_r! WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got reject for frame 97, retransmitting frame 106 now, updating n_r! WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got reject for frame 97, retransmitting frame 107 now, updating n_r! WARNING[1167272128]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: Read on 252 failed: Unknown error 500 --- Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England URL:www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DIAX, IAX2 and latency
Peer, Would you be interested in helping me test diax from Germany to my * box here in Texas? I just want to see about latencty etc.. If so email me offlist so I can set up an extension/registration for you. I am working up a proposal for a conferencing server for a customer whose main office is in Hamburg. I hope to be able to use his existing ADSL line and use DIAX or Hard phones to connect to the * server here. I will also get voip 800 service for others who might need to call in to the conference. The conferences right now are only 4 or 5 people. Andy > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Peer Oliver > schmidt > Sent: Friday, November 21, 2003 11:50 AM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] DIAX, IAX2 and latency > > > Hello, > > today I tried a DIAX -> * -> DIAX connection over the internet (768/128 > ADSL connection on both sides). > > The sound quality was great. However, we had some latency problems, and > also, if both sides where not talking the first words had some problems > getting thru. > > Is this expected, is there anything that can be done on our setup, any > magical iax.conf entry? > > Thanks and best regards from Germany > > pos > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] making outside call with sip phone
Hi All, I have a grandstream sip phone which I have figured out and configured to make internal calls. How do I now configure asterisk to allow this phone to make an outside call? Thanks in advance, Steve. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Outline For Asterisk Book - Please Review & Comment
Title: Message > I hope Outlook doesn't mess this up too much--I've to make it as list-friendly as I know how... Try Outlook QuoteFix at http://home.in.tum.de/~jain/software/outlook-quotefix/ It worked great until this last email :)
RE: [Asterisk-Users] Outline For Asterisk Book - Please Review & Comment
Randy, Great suggestions! Jared is working on integrating the changes since this morning. I think I may drive him nuts, but we will get these integrated too. If you get a chance, drop by #asterisk-doc. Thanks, Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Johnson, Randy Sent: Friday, November 21, 2003 2:25 PM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Outline For Asterisk Book - Please Review & Comment This is a great development. What a good way to develop a book for a great piece of software! I had an ouline slowly developing as my Asterisk implementations broadened, but now I'll jump on board behind yours. I am not an Asterisk guru either, but I'll contribute as I can. Some initial suggestions are in-line below. One general suggestion: Include with the printed book a bootable Knoppix-style CD that would boot up, autoload drivers for the DevKit or DevKit Lite, and be ready to hit with SIP and IAX from DIAX and X-Lite from another PC running Windows on the same network (and maybe a GS phone, too). I think this should be feasible. I hope Outlook doesn't mess this up too much--I've to make it as list-friendly as I know how... Randy Johnson > -Original Message- > From: Steven Sokol [mailto:[EMAIL PROTECTED] > Sent: Friday, November 21, 2003 12:35 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Outline For Asterisk Book - Please > Review & Comment > > > Asterisk Users > > In an attempt to help Asterisk move forward, a number of us > have decided > to create a book. It would initially be released as an "ebook" that > could be sent to newbies to help them up the rather steep learning > curve. Ultimately I would like to see it published and sold in > bookstores (preferably by O'Reilly & Co.). > [snip] > > A "living" copy of this outline can be found at: > http://www.sokol-associates.com/outline.htm . I will try to update it > daily with your suggestions. It will also be the basis for > the project > outline (completion %, assignments, etc.). > > Thanks, > > Steve Sokol > Sokol & Associates, LLC > > [Outline Guide] > The following outline describes the layout for the book. > > 1. <- Section > a. <- Chapter > 1) <- Sub-Chapter > i. <- Topic Heading > *. <- Sidebar Heading > 1} <- Graphic or Chart > 1> <- Table > > [Outline] > 1. Introduction to Asterisk > a. Introductory letter from Mark Spencer > 1) Whatever Mark has to say... > 2) Digium Reference Information > i. Web Site > ii. Phone Number > b. The Business Case For Asterisk > [Somebody From The Business Side Writes This] > c. General concept of asterisk > 1) Asterisk: Swiss Army Knife of Telephony > 2) PBX, IVR, ACD Is this also Telephony 101? FXO, FXS, loop start/ground start, PRI, E&M, RBS, T1, etc. Telephony has a lot of new concepts for someone coming from software or networking... Maybe a basic telephony dictionary at least should be an appendix? > 3) What To Expect > i. Asterisk Is Not A Turnkey System > ii. Don't Like It? Change It Yourself. > iii. Opensource, GPL and LGPL Licensing How about either 4) or iv. Like: 4) Other Related Open Source Alternatives i. VOCAL ii. SER iii.GnuGK iv. Bayonne This will put Asterisk in context for someone familiar with any of these, point out Asterisk's advantages, and show how they can be used together if appropriate. > d. Asterisk architecture > 1) The Big Picture > 2) Channels > 3) Codec Conversions > 4) Etc. > e. Key components > 1) Asterisk software > i. Asterisk (Main PBX & Channels) > ii. Zaptel (Drivers for Zaptel Hardware) > iii. Libpri (ISDN PRI Drivers for Zaptel) > 2) Zaptel Hardware > i. Overview > ii. X100P - Single Port FXO Line Interface > iii. S100U - Single Port FXS USB Interface > iv. TDM400P - 4 Port FXS Analog Interface > v. T100P - Single Span T1/E1 Interface > vi. TE410P - Quad-Span T1/E1 Interface > 3) Channels > i. Zaptel Devices/Channels > ii. The IAX Protocol > iii. SIP > iv. MGCP > v. Skinny >
asterisk-users@lists.digium.com
An excellent suggestion. Would Digium be willing to host this, or should we find a different host? Personally I would like to keep everything on the Digium server, but I can understand if Mark doesn't want to cover the additional bandwidth. What do you think is best? Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Friday, November 21, 2003 2:47 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Outline For Asterisk Book - Please Review &Comment How about this whole discussion getting a mailing list of it's own. I do want to contribute, and would join a new mailing list for this, but would like to take it out of the -users list to cut down on volume and get to answering questions easier. On Fri, 2003-11-21 at 11:34, Steven Sokol wrote: > Asterisk Users > > In an attempt to help Asterisk move forward, a number of us have decided > to create a book. It would initially be released as an "ebook" that > could be sent to newbies to help them up the rather steep learning > curve. Ultimately I would like to see it published and sold in > bookstores (preferably by O'Reilly & Co.). > > Below is the outline for the book. We REALLY need as much input as we > can get. I would like to completely flesh-out the outline, then I would > like to start accepting submissions from the user community for each of > the sections/chapters/topics covered in the outline. > > I have to stress here that I AM NOT AN ASTERISK GURU. I need help from > the real gurus (especially: Steven Critchfield, John Todd, Tilghman > Lesher, Olle Johansson, and where possible/necessary Mark and Martin). > > If this works, it will help Asterisk achieve the same kind of global > success as Apache, Samba, and other Linux staples. If you want to chat, > I am lurking in the #asterisk-doc channel on Freenode IRC. I'm > 'ssokol'. Others on this project (so far) are Jared Smith and Leif > Madson. > > A "living" copy of this outline can be found at: > http://www.sokol-associates.com/outline.htm . I will try to update it > daily with your suggestions. It will also be the basis for the project > outline (completion %, assignments, etc.). > > Thanks, > > Steve Sokol > Sokol & Associates, LLC > > [Outline Guide] > The following outline describes the layout for the book. > > 1. <- Section > a. <- Chapter > 1) <- Sub-Chapter > i. <- Topic Heading > *. <- Sidebar Heading > 1} <- Graphic or Chart > 1> <- Table > > [Outline] > 1. Introduction to Asterisk > a. Introductory letter from Mark Spencer > 1) Whatever Mark has to say... > 2) Digium Reference Information > i. Web Site > ii. Phone Number > b. The Business Case For Asterisk > [Somebody From The Business Side Writes This] > c. General concept of asterisk > 1) Asterisk: Swiss Army Knife of Telephony > 2) PBX, IVR, ACD > 3) What To Expect > i. Asterisk Is Not A Turnkey System > ii. Don't Like It? Change It Yourself. > iii. Opensource, GPL and LGPL Licensing > d. Asterisk architecture > 1) The Big Picture > 2) Channels > 3) Codec Conversions > 4) Etc. > e. Key components > 1) Asterisk software > i. Asterisk (Main PBX & Channels) > ii. Zaptel (Drivers for Zaptel Hardware) > iii. Libpri (ISDN PRI Drivers for Zaptel) > 2) Zaptel Hardware > i. Overview > ii. X100P - Single Port FXO Line Interface > iii. S100U - Single Port FXS USB Interface > iv. TDM400P - 4 Port FXS Analog Interface > v. T100P - Single Span T1/E1 Interface > vi. TE410P - Quad-Span T1/E1 Interface > 3) Channels > i. Zaptel Devices/Channels > ii. The IAX Protocol > iii. SIP > iv. MGCP > v. Skinny > vi. H323 > 4) Applications > i. Dial and Other Basics > ii. Voicemail > iii. Dial-Plan Scripting > 5) Extensibility > i. AGI > ii. Custom Applications > f. Add-On/Optional Components > 1) Software > i. Gnophone > ii. VoIP Soft Phones > iii. DIAX > iv. Gastman > v. Open H.323 >
RE: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1 (Alpha)
I think this is a great addition!!! Thanks for the app! -Original Message- From: Steven Sokol [mailto:[EMAIL PROTECTED] Sent: Friday, November 21, 2003 3:32 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1 (Alpha) If anybody is interested, I have an early version of my Call Manager for Windows application integrated with Asterisk. CMW is an application bar (like the task-bar) that docks to the top of your desktop window. It provides the following functions: 1. View Call-Related Information (Caller ID, Call State, Call Direction) 2. Monitor Status of Asterisk Stations (Channels) -- BLF or "Busy Lamp Field" 3. Place outbound calls. 4. Record (monitor), transfer, or drop connected, active calls. 5. Speed-dial "inside" and "outside" numbers. 6. Remote access to Asterisk CLI functions (show channels, reload, etc.) 7. AstManager COM component (can be used to add Ast functionality to other apps). Here are four screenshots with various features in use: Basic Screen: http://www.sokol-associates.com/images/AstMgr.jpg With Command Window: http://www.sokol-associates.com/images/AstMgr2.jpg With Settings Window: http://www.sokol-associates.com/images/AstMgr3.jpg With Monitor Config: http://www.sokol-associates.com/images/AstMgr4.jpg With Debug Window: http://www.sokol-associates.com/images/AstMgr5.jpg I kind of think of it as a "SoftPhone-Lite" application. It works as a soft-phone enhancement or add-on to your VoIP hard-phone. It is currently buggy and rather feature-poor. I hope to add lots of additional features. These will include: 1. Redial 2. Voicemail Box Monitoring 3. Enhanced Conferencing 4. Outlook/Act/Goldmine Integration (PIM stuff) 5. Call History (both inbound and outbound) 6. Redirect Option on Ring (VM, Application, Transfer, etc.) 7. Automatic mixing and delivery of monitored (recorded) files. I also plan on adding a DDE and COM extension so that it can easily act as a "CTI Client" to execute screen-pops. I may even throw in a scripting function that allows scripts to be executed on each incoming call. A copy of the source code (let's call this LGPL for now) is available here: http://www.sokol-associates.com/Downloads/AstMgr.zip It's written in VB6 (yes - barf, gag, whatever). The only thing required beyond the integral VB6 controls is the Windows Scripting Runtime which most PCs should have. I will work on an installable version soon. I may also port it to something more cross-platform. Please bear with me as I am just learning Gnome/GTK/X-windows. Please let me know what you think of the idea. Constructive criticism only please. I am fully aware that Windoze sucks, VB sucks, Bill Gates sucks, etc. I don't need to be called a "script kiddie" by anyone. (And FWIW: Asterisk will go much further by playing nicely with the evil but predominant operating system out there.) Thanks, Steve P.S. Note: the Monitor editing dialog is not working yet. I shows the devices/users but does not actually edit the file. The file (Monitor.conf) is stored in the root directory for the application and can be edited using notepad or whatever. The format is pretty self-explanatory. Obviously the PSTN and APP technologies can't be monitored. Regs, SMS ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outline For Asterisk Book - Please Review & Comment
How about this whole discussion getting a mailing list of it's own. I do want to contribute, and would join a new mailing list for this, but would like to take it out of the -users list to cut down on volume and get to answering questions easier. On Fri, 2003-11-21 at 11:34, Steven Sokol wrote: > Asterisk Users > > In an attempt to help Asterisk move forward, a number of us have decided > to create a book. It would initially be released as an "ebook" that > could be sent to newbies to help them up the rather steep learning > curve. Ultimately I would like to see it published and sold in > bookstores (preferably by O'Reilly & Co.). > > Below is the outline for the book. We REALLY need as much input as we > can get. I would like to completely flesh-out the outline, then I would > like to start accepting submissions from the user community for each of > the sections/chapters/topics covered in the outline. > > I have to stress here that I AM NOT AN ASTERISK GURU. I need help from > the real gurus (especially: Steven Critchfield, John Todd, Tilghman > Lesher, Olle Johansson, and where possible/necessary Mark and Martin). > > If this works, it will help Asterisk achieve the same kind of global > success as Apache, Samba, and other Linux staples. If you want to chat, > I am lurking in the #asterisk-doc channel on Freenode IRC. I'm > 'ssokol'. Others on this project (so far) are Jared Smith and Leif > Madson. > > A "living" copy of this outline can be found at: > http://www.sokol-associates.com/outline.htm . I will try to update it > daily with your suggestions. It will also be the basis for the project > outline (completion %, assignments, etc.). > > Thanks, > > Steve Sokol > Sokol & Associates, LLC > > [Outline Guide] > The following outline describes the layout for the book. > > 1. <- Section > a. <- Chapter > 1) <- Sub-Chapter > i. <- Topic Heading > *. <- Sidebar Heading > 1} <- Graphic or Chart > 1> <- Table > > [Outline] > 1. Introduction to Asterisk > a. Introductory letter from Mark Spencer > 1) Whatever Mark has to say... > 2) Digium Reference Information > i. Web Site > ii. Phone Number > b. The Business Case For Asterisk > [Somebody From The Business Side Writes This] > c. General concept of asterisk > 1) Asterisk: Swiss Army Knife of Telephony > 2) PBX, IVR, ACD > 3) What To Expect > i. Asterisk Is Not A Turnkey System > ii. Don't Like It? Change It Yourself. > iii. Opensource, GPL and LGPL Licensing > d. Asterisk architecture > 1) The Big Picture > 2) Channels > 3) Codec Conversions > 4) Etc. > e. Key components > 1) Asterisk software > i. Asterisk (Main PBX & Channels) > ii. Zaptel (Drivers for Zaptel Hardware) > iii. Libpri (ISDN PRI Drivers for Zaptel) > 2) Zaptel Hardware > i. Overview > ii. X100P - Single Port FXO Line Interface > iii. S100U - Single Port FXS USB Interface > iv. TDM400P - 4 Port FXS Analog Interface > v. T100P - Single Span T1/E1 Interface > vi. TE410P - Quad-Span T1/E1 Interface > 3) Channels > i. Zaptel Devices/Channels > ii. The IAX Protocol > iii. SIP > iv. MGCP > v. Skinny > vi. H323 > 4) Applications > i. Dial and Other Basics > ii. Voicemail > iii. Dial-Plan Scripting > 5) Extensibility > i. AGI > ii. Custom Applications > f. Add-On/Optional Components > 1) Software > i. Gnophone > ii. VoIP Soft Phones > iii. DIAX > iv. Gastman > v. Open H.323 > 2) Hardware > i. VoIP Hard-Phones > ii. VoIP Gateways > ii. Channel Banks > > 2. Installing Asterisk > *. Asterisk Quickstart > 1) Install PC Hardware > 2) Download Asterisk Software > 3) Build Asterisk > 4) Install Asterisk > 5) Configure Autostart > > a. Requirements > *) Picking A Solid System > 1) PC Hardware Re
Re: [Asterisk-Users] Struggling with grandstream sip to asterisk
On Friday 21 November 2003 03:22 pm, Dave Cotton wrote: > On Fri, 2003-11-21 at 20:14, marrandy wrote: > > My next observation, is many of the buttons on the phone don't work e.g. > > > > Called - After make calls, I expected that pressing this button would show > > something. Nothing is shown. > > Pickup and press called the last 10 numbers called are shown That's what I am doing - Nothing is shown > > Callers - After the grandstream receives calls, I expected that pressing > > this button would show something. Nothing is shown > > As above As above, I have tried it both ways. Nothing is shown. > redial - does nothing > > redials last number or saves the delay after dialling Nothing happens. > > > Message button - does nothing, even when message are waiting (stutter > > dialtone. > > Program with your voice mail extension number Done...it doesn't work. Perhap's my phone is just defective on the buttons. Regards...Martin -- The secret of success is sincerity. Once you can fake that, you've got it made. -- Jean Giraudoux ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Call Manager for Windows 0.0.1 (Alpha)
If anybody is interested, I have an early version of my Call Manager for Windows application integrated with Asterisk. CMW is an application bar (like the task-bar) that docks to the top of your desktop window. It provides the following functions: 1. View Call-Related Information (Caller ID, Call State, Call Direction) 2. Monitor Status of Asterisk Stations (Channels) -- BLF or "Busy Lamp Field" 3. Place outbound calls. 4. Record (monitor), transfer, or drop connected, active calls. 5. Speed-dial "inside" and "outside" numbers. 6. Remote access to Asterisk CLI functions (show channels, reload, etc.) 7. AstManager COM component (can be used to add Ast functionality to other apps). Here are four screenshots with various features in use: Basic Screen: http://www.sokol-associates.com/images/AstMgr.jpg With Command Window: http://www.sokol-associates.com/images/AstMgr2.jpg With Settings Window: http://www.sokol-associates.com/images/AstMgr3.jpg With Monitor Config: http://www.sokol-associates.com/images/AstMgr4.jpg With Debug Window: http://www.sokol-associates.com/images/AstMgr5.jpg I kind of think of it as a "SoftPhone-Lite" application. It works as a soft-phone enhancement or add-on to your VoIP hard-phone. It is currently buggy and rather feature-poor. I hope to add lots of additional features. These will include: 1. Redial 2. Voicemail Box Monitoring 3. Enhanced Conferencing 4. Outlook/Act/Goldmine Integration (PIM stuff) 5. Call History (both inbound and outbound) 6. Redirect Option on Ring (VM, Application, Transfer, etc.) 7. Automatic mixing and delivery of monitored (recorded) files. I also plan on adding a DDE and COM extension so that it can easily act as a "CTI Client" to execute screen-pops. I may even throw in a scripting function that allows scripts to be executed on each incoming call. A copy of the source code (let's call this LGPL for now) is available here: http://www.sokol-associates.com/Downloads/AstMgr.zip It's written in VB6 (yes - barf, gag, whatever). The only thing required beyond the integral VB6 controls is the Windows Scripting Runtime which most PCs should have. I will work on an installable version soon. I may also port it to something more cross-platform. Please bear with me as I am just learning Gnome/GTK/X-windows. Please let me know what you think of the idea. Constructive criticism only please. I am fully aware that Windoze sucks, VB sucks, Bill Gates sucks, etc. I don't need to be called a "script kiddie" by anyone. (And FWIW: Asterisk will go much further by playing nicely with the evil but predominant operating system out there.) Thanks, Steve P.S. Note: the Monitor editing dialog is not working yet. I shows the devices/users but does not actually edit the file. The file (Monitor.conf) is stored in the root directory for the application and can be edited using notepad or whatever. The format is pretty self-explanatory. Obviously the PSTN and APP technologies can't be monitored. Regs, SMS ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Outline For Asterisk Book - Please Review & Comment
Title: RE: [Asterisk-Users] Outline For Asterisk Book - Please Review & Comment This is a great development. What a good way to develop a book for a great piece of software! I had an ouline slowly developing as my Asterisk implementations broadened, but now I'll jump on board behind yours. I am not an Asterisk guru either, but I'll contribute as I can. Some initial suggestions are in-line below. One general suggestion: Include with the printed book a bootable Knoppix-style CD that would boot up, autoload drivers for the DevKit or DevKit Lite, and be ready to hit with SIP and IAX from DIAX and X-Lite from another PC running Windows on the same network (and maybe a GS phone, too). I think this should be feasible. I hope Outlook doesn't mess this up too much--I've to make it as list-friendly as I know how... Randy Johnson > -Original Message- > From: Steven Sokol [mailto:[EMAIL PROTECTED]] > Sent: Friday, November 21, 2003 12:35 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Outline For Asterisk Book - Please > Review & Comment > > > Asterisk Users > > In an attempt to help Asterisk move forward, a number of us > have decided > to create a book. It would initially be released as an "ebook" that > could be sent to newbies to help them up the rather steep learning > curve. Ultimately I would like to see it published and sold in > bookstores (preferably by O'Reilly & Co.). > [snip] > > A "living" copy of this outline can be found at: > http://www.sokol-associates.com/outline.htm . I will try to update it > daily with your suggestions. It will also be the basis for > the project > outline (completion %, assignments, etc.). > > Thanks, > > Steve Sokol > Sokol & Associates, LLC > > [Outline Guide] > The following outline describes the layout for the book. > > 1. <- Section > a. <- Chapter > 1) <- Sub-Chapter > i. <- Topic Heading > *. <- Sidebar Heading > 1} <- Graphic or Chart > 1> <- Table > > [Outline] > 1. Introduction to Asterisk > a. Introductory letter from Mark Spencer > 1) Whatever Mark has to say... > 2) Digium Reference Information > i. Web Site > ii. Phone Number > b. The Business Case For Asterisk > [Somebody From The Business Side Writes This] > c. General concept of asterisk > 1) Asterisk: Swiss Army Knife of Telephony > 2) PBX, IVR, ACD Is this also Telephony 101? FXO, FXS, loop start/ground start, PRI, E&M, RBS, T1, etc. Telephony has a lot of new concepts for someone coming from software or networking... Maybe a basic telephony dictionary at least should be an appendix? > 3) What To Expect > i. Asterisk Is Not A Turnkey System > ii. Don't Like It? Change It Yourself. > iii. Opensource, GPL and LGPL Licensing How about either 4) or iv. Like: 4) Other Related Open Source Alternatives i. VOCAL ii. SER iii. GnuGK iv. Bayonne This will put Asterisk in context for someone familiar with any of these, point out Asterisk's advantages, and show how they can be used together if appropriate. > d. Asterisk architecture > 1) The Big Picture > 2) Channels > 3) Codec Conversions > 4) Etc. > e. Key components > 1) Asterisk software > i. Asterisk (Main PBX & Channels) > ii. Zaptel (Drivers for Zaptel Hardware) > iii. Libpri (ISDN PRI Drivers for Zaptel) > 2) Zaptel Hardware > i. Overview > ii. X100P - Single Port FXO Line Interface > iii. S100U - Single Port FXS USB Interface > iv. TDM400P - 4 Port FXS Analog Interface > v. T100P - Single Span T1/E1 Interface > vi. TE410P - Quad-Span T1/E1 Interface > 3) Channels > i. Zaptel Devices/Channels > ii. The IAX Protocol > iii. SIP > iv. MGCP > v. Skinny > vi. H323 > 4) Applications > i. Dial and Other Basics > ii. Voicemail > iii. Dial-Plan Scripting iv. CDR (does this belong here? It should be somewhere...) > 5) Extensibility > i. AGI > ii. Custom Applications > f. Add-On/Optional Components >
Re: [Asterisk-Users] Struggling with grandstream sip to asterisk
On Fri, 2003-11-21 at 20:14, marrandy wrote: > My next observation, is many of the buttons on the phone don't work e.g. > > Called - After make calls, I expected that pressing this button would show > something. Nothing is shown. Pickup and press called the last 10 numbers called are shown > > Callers - After the grandstream receives calls, I expected that pressing > this button would show something. Nothing is shown As above redial - does nothing redials last number or saves the delay after dialling > Message button - does nothing, even when message are waiting (stutter > dialtone. Program with your voice mail extension number ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Dave Cotton <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Outline For Asterisk Book - Please Review & Comment
Absolutely Right! I just added it to the living outline. Thanks, Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ernest W. Lessenger Sent: Friday, November 21, 2003 1:07 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Outline For Asterisk Book - Please Review & Comment Sounds like a great idea! I'll gladly help if requested (I'm a technical writer). Comment: I don't see anything on echo cancellation. That's a big enough and common enough issue that it deserves some discussion. --Ernest At 10:46 AM 11/21/2003, you wrote: >Hi Steven, > >I think this is a great idea and the best way to make >users more familiar with Asterisk and its configuration and usage. >I can and will provide input for all H.323 related sections. > >Michael. > >Steven Sokol wrote: > > Asterisk Users > > > > In an attempt to help Asterisk move forward, a number of us have decided > > to create a book. It would initially be released as an "ebook" that > > could be sent to newbies to help them up the rather steep learning > > curve. Ultimately I would like to see it published and sold in > > bookstores (preferably by O'Reilly & Co.). > > > > Below is the outline for the book. We REALLY need as much input as we > > can get. I would like to completely flesh-out the outline, then I would > > like to start accepting submissions from the user community for each of > > the sections/chapters/topics covered in the outline. > > > > I have to stress here that I AM NOT AN ASTERISK GURU. I need help from > > the real gurus (especially: Steven Critchfield, John Todd, Tilghman > > Lesher, Olle Johansson, and where possible/necessary Mark and Martin). > > > > If this works, it will help Asterisk achieve the same kind of global > > success as Apache, Samba, and other Linux staples. If you want to chat, > > I am lurking in the #asterisk-doc channel on Freenode IRC. I'm > > 'ssokol'. Others on this project (so far) are Jared Smith and Leif > > Madson. > > > > A "living" copy of this outline can be found at: > > http://www.sokol-associates.com/outline.htm . I will try to update it > > daily with your suggestions. It will also be the basis for the project > > outline (completion %, assignments, etc.). > > > > Thanks, > > > > Steve Sokol > > Sokol & Associates, LLC > > > > [Outline Guide] > > The following outline describes the layout for the book. > > > > 1. <- Section > > a. <- Chapter > > 1) <- Sub-Chapter > > i. <- Topic Heading > > *. <- Sidebar Heading > > 1} <- Graphic or Chart > > 1> <- Table > > > > [Outline] > > 1. Introduction to Asterisk > > a. Introductory letter from Mark Spencer > > 1) Whatever Mark has to say... > > 2) Digium Reference Information > > i. Web Site > > ii. Phone Number > > b. The Business Case For Asterisk > > [Somebody From The Business Side Writes This] > > c. General concept of asterisk > > 1) Asterisk: Swiss Army Knife of Telephony > > 2) PBX, IVR, ACD > > 3) What To Expect > > i. Asterisk Is Not A Turnkey System > > ii. Don't Like It? Change It Yourself. > > iii. Opensource, GPL and LGPL Licensing > > d. Asterisk architecture > > 1) The Big Picture > > 2) Channels > > 3) Codec Conversions > > 4) Etc. > > e. Key components > > 1) Asterisk software > > i. Asterisk (Main PBX & Channels) > > ii. Zaptel (Drivers for Zaptel Hardware) > > iii. Libpri (ISDN PRI Drivers for Zaptel) > > 2) Zaptel Hardware > > i. Overview > > ii. X100P - Single Port FXO Line Interface > > iii. S100U - Single Port FXS USB Interface > > iv. TDM400P - 4 Port FXS Analog Interface > > v. T100P - Single Span T1/E1 Interface > > vi. TE410P - Quad-Span T1/E1 Interface > > 3) Channels > > i. Zaptel Devices/Channels > > ii. The IAX Protocol > > iii. SIP > > iv. MGCP > > v. Skinny > > vi. H323 > > 4) Applications > > i. Dial and Other Basics > > ii. Voicemail > > iii. Dial-Plan Scripting > > 5) Extensibility > > i. AGI > > ii. Custom Applications > > f. Add-On/Optional Components > > 1) Software > > i. Gnophone > >
Re: [Asterisk-Users] Outline For Asterisk Book - Please Review & Comment
Great idea! I would also like to see some discussion of ISDN BRI and CAPI hardware, as well as some discussion of "distributed asterisk" what happens when you start deploying a network of many asterisk boxes, how to do forwarding and switching properly, TDMoE as well as E164 enum call routing -- on the latter I can perhaps contribute some text now that I have gotten it to work properly ;) Cheers, Will ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Struggling with grandstream sip to asterisk
> > > Software Version:Program--1.0.4.20Bootloader--1.0.0.12 > > HTML--1.0.0.19 > > Send DTMF: via SIP INFO > > DTMF Payload Type: 101 > > > In my experience, anything higher then 1.0.4.17 will NOT work properly > with asterisk. You can get the phone to sip register but it will not stay > that way. > > The tftp site you are using is BETA software if I'm not mistaken. Correct, never the less I do not see the same issues other are reporting with SIP not staying registered or the SIP Info issue with the DTMF How can I verify this registration issue, its werid that it just seems to work in my install ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Upgrade CISCO 7960 Question
On Fri, Sep 24, 2004 at 07:13:39AM -0400, [EMAIL PROTECTED] wrote: > Yes, it is. But why would you want to do that when yo said what you > want it to be at 6.0. He's got the Skinny version and wants to change to the SIP version. > > Maybe you didn't expling what and why you want to do it in enough detail > to get a good answer. > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz > Jozwiak > Sent: Friday, November 21, 2003 8:11 AM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Upgrade CISCO 7960 Question > > > Hello, > > My Cisco phone has software: > Boot Load: PC030300 > Ver: 3.2(7.0) The SIP version 6.0 has these versions if you login to the 7960 tusing telnet: Cisco Systems, Inc. Copyright 2000-2003 Cisco IP phone MAC: 0030:94c2:ea67 Loadid: SW: P0S3-06-0-00 ARM: PAS3ARM1 Boot: PC03M030 DSP: PS03AT38 > > And I want to upgrade it to SIP 6.0 > Is it possible or I have to upgrade to ealier then 6.0 and then > to 6.0 ? I upgraded directly from your version of the Skinny image to the SIP 6.0 image and it is working fine. > > bart > -- DataCrest, Inc. -- Technically Superior ** Walker Haddock http://www.datacrest.com DataCrest, Inc.e-mail: [EMAIL PROTECTED] 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ADSI Hold
My issue is programing the ADSI softkeys... But what is the syntax for Hold. Example to create softkeys to transfer I can send the FLASH key. But I am lost when it comes to the Hold. -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Biggs Posted At: Friday, November 21, 2003 12:57 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] ADSI Hold Subject: Re: [Asterisk-Users] ADSI Hold New to ADSI but my guess would be... If you want Asterisk to put the call on hold you could just program the soft keys to send the DTMF tones that a regular phone would use to put the call on hold. If you look at the example that can with Asterisk. (/etc/asterisk/asterisk.adsi ..I have been staring at it for days trying different things) you'll notice that is how the vmail, etc keys work already. Just change the label and the DTMF, etc Hope this helps. BTW I have an annoying ADSI issue looking for help with on Hearing the CAS tone in the voicemail prompts. Do you hear a loud short "Chirp" in the vmail prompts on ADSI phones? --- PBX <[EMAIL PROTECTED]> wrote: > Is there any way to program a soft key in ADSI to > put a caller on hold. > Then able to retreive that caller. > > Example - > > Softkey Hold > Softkey Retreive Call > Softkey End Call > > -gcc > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Free Pop-Up Blocker - Get it now http://companion.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Struggling with grandstream sip to asterisk
On Friday 21 November 2003 06:17 am, Michael T Farnworth wrote: > On Thu, 20 Nov 2003, marrandy wrote: > host=dynamic > defaultip=192.168.254.160 O.K. - that worked, plus I removed the permit and went back to inband (or grandstream calls it, in-audio. So I can make and receive calls. My next observation, is many of the buttons on the phone don't work e.g. Called - After make calls, I expected that pressing this button would show something. Nothing is shown. Callers - After the grandstream receives calls, I expected that pressing this button would show something. Nothing is shown Hold - Pressing hold during a call does nothing. Transfer - pressing transfer does nothing. Conference - does nothing flash -does nothing redial - does nothing Message button - does nothing, even when message are waiting (stutter dialtone. Any clues on getting these features working ? Regards...Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Struggling with grandstream sip to asterisk
On Fri, 21 Nov 2003, TC wrote: > Software Version:Program--1.0.4.20Bootloader--1.0.0.12 > HTML--1.0.0.19 > Send DTMF: via SIP INFO > DTMF Payload Type: 101 In my experience, anything higher then 1.0.4.17 will NOT work properly with asterisk. You can get the phone to sip register but it will not stay that way. The tftp site you are using is BETA software if I'm not mistaken. John - NetRom Internet Services973-208-1339 voice [EMAIL PROTECTED] 973-208-0942 fax http://www.netrom.com - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outline For Asterisk Book - Please Review & Comment
Sounds like a great idea! I'll gladly help if requested (I'm a technical writer). Comment: I don't see anything on echo cancellation. That's a big enough and common enough issue that it deserves some discussion. --Ernest At 10:46 AM 11/21/2003, you wrote: Hi Steven, I think this is a great idea and the best way to make users more familiar with Asterisk and its configuration and usage. I can and will provide input for all H.323 related sections. Michael. Steven Sokol wrote: > Asterisk Users > > In an attempt to help Asterisk move forward, a number of us have decided > to create a book. It would initially be released as an "ebook" that > could be sent to newbies to help them up the rather steep learning > curve. Ultimately I would like to see it published and sold in > bookstores (preferably by O'Reilly & Co.). > > Below is the outline for the book. We REALLY need as much input as we > can get. I would like to completely flesh-out the outline, then I would > like to start accepting submissions from the user community for each of > the sections/chapters/topics covered in the outline. > > I have to stress here that I AM NOT AN ASTERISK GURU. I need help from > the real gurus (especially: Steven Critchfield, John Todd, Tilghman > Lesher, Olle Johansson, and where possible/necessary Mark and Martin). > > If this works, it will help Asterisk achieve the same kind of global > success as Apache, Samba, and other Linux staples. If you want to chat, > I am lurking in the #asterisk-doc channel on Freenode IRC. I'm > 'ssokol'. Others on this project (so far) are Jared Smith and Leif > Madson. > > A "living" copy of this outline can be found at: > http://www.sokol-associates.com/outline.htm . I will try to update it > daily with your suggestions. It will also be the basis for the project > outline (completion %, assignments, etc.). > > Thanks, > > Steve Sokol > Sokol & Associates, LLC > > [Outline Guide] > The following outline describes the layout for the book. > > 1. <- Section > a. <- Chapter > 1) <- Sub-Chapter > i. <- Topic Heading > *. <- Sidebar Heading > 1} <- Graphic or Chart > 1> <- Table > > [Outline] > 1. Introduction to Asterisk > a. Introductory letter from Mark Spencer > 1) Whatever Mark has to say... > 2) Digium Reference Information > i. Web Site > ii. Phone Number > b. The Business Case For Asterisk > [Somebody From The Business Side Writes This] > c. General concept of asterisk > 1) Asterisk: Swiss Army Knife of Telephony > 2) PBX, IVR, ACD > 3) What To Expect > i. Asterisk Is Not A Turnkey System > ii. Don't Like It? Change It Yourself. > iii. Opensource, GPL and LGPL Licensing > d. Asterisk architecture > 1) The Big Picture > 2) Channels > 3) Codec Conversions > 4) Etc. > e. Key components > 1) Asterisk software > i. Asterisk (Main PBX & Channels) > ii. Zaptel (Drivers for Zaptel Hardware) > iii. Libpri (ISDN PRI Drivers for Zaptel) > 2) Zaptel Hardware > i. Overview > ii. X100P - Single Port FXO Line Interface > iii. S100U - Single Port FXS USB Interface > iv. TDM400P - 4 Port FXS Analog Interface > v. T100P - Single Span T1/E1 Interface > vi. TE410P - Quad-Span T1/E1 Interface > 3) Channels > i. Zaptel Devices/Channels > ii. The IAX Protocol > iii. SIP > iv. MGCP > v. Skinny > vi. H323 > 4) Applications > i. Dial and Other Basics > ii. Voicemail > iii. Dial-Plan Scripting > 5) Extensibility > i. AGI > ii. Custom Applications > f. Add-On/Optional Components > 1) Software > i. Gnophone > ii. VoIP Soft Phones > iii. DIAX > iv. Gastman > v. Open H.323 > 2) Hardware > i. VoIP Hard-Phones > ii. VoIP Gateways > ii. Channel Banks > > 2. Installing Asterisk > *. Asterisk Quickstart > 1) Install PC Hardware > 2) Download Asterisk Software > 3) Build Asterisk > 4) Install Asterisk >
Re: [Asterisk-Users] DIAX, IAX2 and latency
Hi, - Original Message - From: "Peer Oliver schmidt" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, November 21, 2003 7:49 PM Subject: [Asterisk-Users] DIAX, IAX2 and latency > Hello, > > today I tried a DIAX -> * -> DIAX connection over the internet (768/128 > ADSL connection on both sides). > > The sound quality was great. However, we had some latency problems, and > also, if both sides where not talking the first words had some problems > getting thru. > > Is this expected, is there anything that can be done on our setup, any > magical iax.conf entry? Check in the config file of DIAX if you have SILENCE=-99 Please wait till Sunday evening when I'll post the new version of DIAX (0.9.4) with IAX2 support. It has some improvements in the sound processing part. > > Thanks and best regards from Germany Thank you too for trying DIAX. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DeltaThree setup on Asterisk
Hi, I get a number from DeltaThree and I want to setup on my Asterisk to receive and to make calls thru this phone #. I need some samples of .conf with please. Best regards, Chris HARIGA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outline For Asterisk Book - Please Review & Comment
Hi Steven, I think this is a great idea and the best way to make users more familiar with Asterisk and its configuration and usage. I can and will provide input for all H.323 related sections. Michael. Steven Sokol wrote: Asterisk Users In an attempt to help Asterisk move forward, a number of us have decided to create a book. It would initially be released as an "ebook" that could be sent to newbies to help them up the rather steep learning curve. Ultimately I would like to see it published and sold in bookstores (preferably by O'Reilly & Co.). Below is the outline for the book. We REALLY need as much input as we can get. I would like to completely flesh-out the outline, then I would like to start accepting submissions from the user community for each of the sections/chapters/topics covered in the outline. I have to stress here that I AM NOT AN ASTERISK GURU. I need help from the real gurus (especially: Steven Critchfield, John Todd, Tilghman Lesher, Olle Johansson, and where possible/necessary Mark and Martin). If this works, it will help Asterisk achieve the same kind of global success as Apache, Samba, and other Linux staples. If you want to chat, I am lurking in the #asterisk-doc channel on Freenode IRC. I'm 'ssokol'. Others on this project (so far) are Jared Smith and Leif Madson. A "living" copy of this outline can be found at: http://www.sokol-associates.com/outline.htm . I will try to update it daily with your suggestions. It will also be the basis for the project outline (completion %, assignments, etc.). Thanks, Steve Sokol Sokol & Associates, LLC [Outline Guide] The following outline describes the layout for the book. 1. <- Section a. <- Chapter 1) <- Sub-Chapter i. <- Topic Heading *. <- Sidebar Heading 1} <- Graphic or Chart 1> <- Table [Outline] 1. Introduction to Asterisk a. Introductory letter from Mark Spencer 1) Whatever Mark has to say... 2) Digium Reference Information i. Web Site ii. Phone Number b. The Business Case For Asterisk [Somebody From The Business Side Writes This] c. General concept of asterisk 1) Asterisk: Swiss Army Knife of Telephony 2) PBX, IVR, ACD 3) What To Expect i. Asterisk Is Not A Turnkey System ii. Don't Like It? Change It Yourself. iii. Opensource, GPL and LGPL Licensing d. Asterisk architecture 1) The Big Picture 2) Channels 3) Codec Conversions 4) Etc. e. Key components 1) Asterisk software i. Asterisk (Main PBX & Channels) ii. Zaptel (Drivers for Zaptel Hardware) iii. Libpri (ISDN PRI Drivers for Zaptel) 2) Zaptel Hardware i. Overview ii. X100P - Single Port FXO Line Interface iii. S100U - Single Port FXS USB Interface iv. TDM400P - 4 Port FXS Analog Interface v. T100P - Single Span T1/E1 Interface vi. TE410P - Quad-Span T1/E1 Interface 3) Channels i. Zaptel Devices/Channels ii. The IAX Protocol iii. SIP iv. MGCP v. Skinny vi. H323 4) Applications i. Dial and Other Basics ii. Voicemail iii. Dial-Plan Scripting 5) Extensibility i. AGI ii. Custom Applications f. Add-On/Optional Components 1) Software i. Gnophone ii. VoIP Soft Phones iii. DIAX iv. Gastman v. Open H.323 2) Hardware i. VoIP Hard-Phones ii. VoIP Gateways ii. Channel Banks 2. Installing Asterisk *. Asterisk Quickstart 1) Install PC Hardware 2) Download Asterisk Software 3) Build Asterisk 4) Install Asterisk 5) Configure Autostart a. Requirements *) Picking A Solid System 1) PC Hardware Requirements i. SOHO/Residential System ii. Small Business System iii. Medium Business/Small Call-Center System iv. Enterprise System v. VoIP Carrier System 2) Linux Requirements *. Linux Installation Is Not Covered i. Tested Distributions ii. Minimum Ker
asterisk-users@lists.digium.com
>Especially with respect to a Orieley & co book, think about a removable >card that lists in short description the applications and the arguments. >This could be the item that gets taped to the side of the monitor of the >new user trying to lay out his dialplan. I have added the pull-out card to the living outline. Great idea. >On Fri, 2003-11-21 at 11:34, Steven Sokol wrote: > e. Key components > 1) Asterisk software > i. Asterisk (Main PBX & Channels) > ii. Zaptel (Drivers for Zaptel Hardware) > iii. Libpri (ISDN PRI Drivers for Zaptel) > 2) Zaptel Hardware > i. Overview > ii. X100P - Single Port FXO Line Interface > iii. S100U - Single Port FXS USB Interface > iv. TDM400P - 4 Port FXS Analog Interface > v. T100P - Single Span T1/E1 Interface > vi. TE410P - Quad-Span T1/E1 Interface > 3) Channels > i. Zaptel Devices/Channels >Suggest that e-2 be removed and merged under this heading. Other than >mentioning of capabilities there is no need for a chapter devoted to >them. Add to that that you can then make the case for buying Diguim >hardware with the channel driver comments. I don't know. I think it's a good place to plug for Digium's hardware and to review the capabilities of the boards. I imagine Sec:1,Ch:e,Topic:2 being made up of no more than six paragraphs, one for each card, with the possible addition of introduction/conclusion paragraphs. I just like the idea of letting the reader (who probably doesn't know much about telephony) what a great deal the Digium card are (especially in comparison to any of the major players in this space). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Struggling with grandstream sip to asterisk
Good day, I have a new Grandstream and am having trouble connecting to * my software version is the same as below... I can get it to connect, but am getting "RTP Read error: Resource temporarily unavailable" errors whenever I dial... Tom - Original Message - From: "Dave Cotton" <[EMAIL PROTECTED]> To: "Asterisk List" <[EMAIL PROTECTED]> Sent: Friday, November 21, 2003 10:15 AM Subject: Re: [Asterisk-Users] Struggling with grandstream sip to asterisk > On Fri, 2003-11-21 at 17:57, TC wrote: > > > I am using the following settings > > > > Software Version:Program--1.0.4.20Bootloader--1.0.0.12 > > HTML--1.0.0.19 > > Where is this firmware from? The GS site is still at > Program--1.0.3.81Bootloader--1.0.0.7HTML--1.0.0.18 > > or is Sipphone ahead of GS? > > -- > Dave Cotton <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN intercepted announcement
As I understand you have PRI connected to asterisk directly.
In my case there is a h323 gateway between them and the h323
driver must recognize the "not in service" signal and made asterisk
aware of it so that asterisk could relay the conditions/recorded messages
to SIP phones. From my experience so far, oh323 driver does it, h323 does not.
Please correct me if I'm wrong.
Thanks
Michael
On Friday 21 November 2003 03:33 am, Josh Rollyson wrote:
> Michael Ulitskiy wrote:
>
> >Hi,
> >
> >I have asterisk functioning as SIP to H.323 gateway for local SIP endpoints and I
> >have H.323 to PSTN
> >gateway (Lucent MAX TNT) connecting my LAN VOIP to PSTN via PRI.
> >Everything works fine with one exception. I seem to be unable to figure out why I
> >cannot hear
> >PSTN intercepted announcement ("number is not in service" etc.) when I'm calling
> >a disconnected number through asterisk.
> >
> AFAIK, A PRI is normally expected to signal number not in service
> conditions out of band, so * should be signalling the out of service
> condition in a manner appropriate for the channel type (as a recording,
> or as the most accurate protocol specific out of service response code
> available, casung the out of service condition to be indicated by the
> target equipment)
>
> With my SNOM phone, the PRI signals not in service or no route or
> whatever, then * relays that in the form of a SIP error response, then
> the SNOM phone executes an internal recording and shows the error on the
> display as well.
>
> This results in more reliable call handling all the way through, and
> some bandwidth savings.
>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
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Re: [Asterisk-Users] Current CVS problem
On Friday 21 November 2003 12:03, Dave Kitchen wrote: > Help: Checkout as of 17:00 UCT > Does anyone know if: > chan_zap.c: In function `zt_train_ec': > chan_zap.c:1081: `ZT_ECHOTRAIN' undeclared (first use in this > function) is expected at the moment? You need to update your zaptel CVS first. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can't get caller id?
You mean you get CallerID on a CallerID capable phone, right? Try this: Load a terminal communications app like minicom, bitcom, telix, or something you're familiar with. Plug the phoneline into a modem that you can see with your temimal software. Ring the phone. I believe in the US, you will see: RING RING So, you should be able to see: RING RING Also take a look at this link: http://www.ainslie.org.uk/callerid/cli_faq.htm#Q_27 It may not exactly apply to you, but it should be good for the archives and might give you an idea of what's going on... - Original Message - From: "C M" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, November 21, 2003 12:46 PM Subject: Re: [Asterisk-Users] can't get caller id? > when i connect the phone directly i get the caller id > after the first ring and b4 the second one. what does > that suggest? > > --- Steven Critchfield <[EMAIL PROTECTED]> wrote: > > On Fri, 2003-11-21 at 08:17, C M wrote: > > > --- Steven Critchfield <[EMAIL PROTECTED]> wrote: > > > > On Fri, 2003-11-21 at 07:14, C M wrote: > > > > > not with *. > > > > > > > > > > i guess its the country issue. like i saw som > > > > posts > > > > > with callerid issues in uk. i am in nepal. how > > can > > > > i > > > > > configure * AGI to make it compatible with my > > > > country. > > > > > has anyone written a script? > > > > > > > > AGI is not configurable, and also would not help > > you > > > > here. What you need > > > > to do is track down what form of CallerId you > > have > > > > and when it is sent > > > > on the line. Then see about tweaking callerid.c > > to > > > > support it. > > > > > > > > > > ok > > > > > > how can i the track callerid information? ask the > > > telco? how does yours work? i want to know how urs > > > happens so that i can figure out how diff is mine. > > > > Callerid in the Us is signaled with FSK between the > > first and second > > ring. For more information, you will need to consult > > the source as it > > will be definitive. > > > > So far, I think it has been established that in the > > UK, the callerid is > > sent after a line reversal and before the first > > ring. > > > > I think it has also been established that some > > locations have callerid > > as either dtmf or mf. I don't usually pay attention > > to this as much > > because I can't help them. > > -- > > Steven Critchfield <[EMAIL PROTECTED]> > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > = > Designs > > __ > Do you Yahoo!? > Free Pop-Up Blocker - Get it now > http://companion.yahoo.com/ > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > [EMAIL PROTECTED])fjåËbú?jË^®+$ºÇ«
Re: [Asterisk-Users] Current CVS problem
On Fri, 2003-11-21 at 12:03, Dave Kitchen wrote: > Help: Checkout as of 17:00 UCT > Does anyone know if: > chan_zap.c: In function `zt_train_ec': > chan_zap.c:1081: `ZT_ECHOTRAIN' undeclared (first use in this function) > is expected at the moment? > Dave Kitchen Google lesson for today Take the part after the ':' in a seeming unique line and plug it directly into the google search and see what pops up. In this case you find 4 messages (at least when I did it), and you will see that you should make sure you have installed recent copies of the zaptel drivers. This is even if you are not going to use the zaptel drivers. Your other option that isn't documented is that you could remove the chan_zap module from the makefile, but this will limit you later on as asterisk really wants a zaptel timing device. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Current CVS problem
Dave, > Help: Checkout as of 17:00 UCT > Does anyone know if: > chan_zap.c: In function `zt_train_ec': > chan_zap.c:1081: `ZT_ECHOTRAIN' undeclared (first use in this function) > is expected at the moment? That variable I think is defined in zaptel directory. I had the same problem and had to rename/delete all current source directories and do a fresh cvs checkout. Probably other ways to handle it, but that worked for me. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outline For Asterisk Book - Please Review & Comment
Especially with respect to a Orieley & co book, think about a removable card that lists in short description the applications and the arguments. This could be the item that gets taped to the side of the monitor of the new user trying to lay out his dialplan. On Fri, 2003-11-21 at 11:34, Steven Sokol wrote: > e. Key components > 1) Asterisk software > i. Asterisk (Main PBX & Channels) > ii. Zaptel (Drivers for Zaptel Hardware) > iii. Libpri (ISDN PRI Drivers for Zaptel) > 2) Zaptel Hardware > i. Overview > ii. X100P - Single Port FXO Line Interface > iii. S100U - Single Port FXS USB Interface > iv. TDM400P - 4 Port FXS Analog Interface > v. T100P - Single Span T1/E1 Interface > vi. TE410P - Quad-Span T1/E1 Interface > 3) Channels > i. Zaptel Devices/Channels Suggest that e-2 be removed and merged under this heading. Other than mentioning of capabilities there is no need for a chapter devoted to them. Add to that that you can then make the case for buying Diguim hardware with the channel driver comments. > ii. The IAX Protocol > iii. SIP > iv. MGCP > v. Skinny > vi. H323 > 4) Applications > i. Dial and Other Basics > ii. Voicemail > iii. Dial-Plan Scripting > 5) Extensibility > i. AGI > ii. Custom Applications > f. Add-On/Optional Components > 1) Software > i. Gnophone > ii. VoIP Soft Phones > iii. DIAX > iv. Gastman > v. Open H.323 > 2) Hardware > i. VoIP Hard-Phones > ii. VoIP Gateways > ii. Channel Banks > > 2. Installing Asterisk > *. Asterisk Quickstart > 1) Install PC Hardware > 2) Download Asterisk Software > 3) Build Asterisk > 4) Install Asterisk > 5) Configure Autostart > > a. Requirements > *) Picking A Solid System > 1) PC Hardware Requirements > i. SOHO/Residential System > ii. Small Business System > iii. Medium Business/Small Call-Center System > iv. Enterprise System > v. VoIP Carrier System > 2) Linux Requirements > *. Linux Installation Is Not Covered > i. Tested Distributions > ii. Minimum Kernel Version > iii. Required Packages > *. Other Operating Systems > - Free BSD > - Mac OS-X > - BeOS? > - Win32/Win64? > b. Hardware Installation > 1) IRQ Sharing Issues > 2) Digium Wildcard Cards > 3) LineJack and PhoneJack Cards > 4) Other Cards (ISDN, VoiceTronix, Etc.) > c. Downloading Asterisk from CVS > 1) What is CVS? > 2) The Asterisk "Versioning" Issues > 3) Your Initial Download > 4) Updates > *. Adding Custom Patches (patch/diff) > d. Compiling Asterisk > *) Why Do I Have To Compile The Code? > 1) Using 'make' > 2) Compiling The Software > i. Zaptel > ii. Libpri > iii. Asterisk > 3) Making The Samples/Demo > 4) Making Code Documentation (Doxygen) > i. Why build code documentation? > ii. What Is Doxygen? > iii. Code Doc Layout > 5) Common Build Errors/Warnings > i. Via C3 Is NOT An i686 > ii. Building on Little-Endian Systems > iii. Etc. > e. Loading drivers (zaptel/ztdummy) > *) Read Ahead (Section 3, Chapter C1) For Zaptel Setup > 1) Linux Kernel Loadable Modules > 2) Using modprobe > 3) Adding zaptel modules to your startup file > *) RedHat Is Weird > f. Starting Asterisk > 1) Manual Starting and the CLI > 2) Starting using safe_asterisk > 3) Accessing the CLI when Aste
[Asterisk-Users] Current CVS problem
Help: Checkout as of 17:00 UCT Does anyone know if: chan_zap.c: In function `zt_train_ec': chan_zap.c:1081: `ZT_ECHOTRAIN' undeclared (first use in this function) is expected at the moment? Dave Kitchen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] One way sound
Hi, I'm having trouble with asterisk: I can't hear both way of a call. here is my current architecture: grandstream -> siproxd -> asterisk -> pstn As I'm just testing for the moment, evrything is on my LAN. I know that there is no need to have a proxy here. But Later, the asterisk will be on a public IP outside of my LAn, so I'm practising... And there is no direct communication between asterisk and my phone (tcpdump is my friend) Well everytime I try a call a pstn number from the grandstream phone, I can get the call and speak, my friend who's got the pstn can hear me... But I can't hear him... Does anybody has any idea of the problem ? chris -- Christophe Sauthier <[EMAIL PROTECTED]> Eikonex - Open Source Engineering (http://www.eikonex.net) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ADSI Hold
New to ADSI but my guess would be... If you want Asterisk to put the call on hold you could just program the soft keys to send the DTMF tones that a regular phone would use to put the call on hold. If you look at the example that can with Asterisk. (/etc/asterisk/asterisk.adsi ..I have been staring at it for days trying different things) you'll notice that is how the vmail, etc keys work already. Just change the label and the DTMF, etc Hope this helps. BTW I have an annoying ADSI issue looking for help with on Hearing the CAS tone in the voicemail prompts. Do you hear a loud short "Chirp" in the vmail prompts on ADSI phones? --- PBX <[EMAIL PROTECTED]> wrote: > Is there any way to program a soft key in ADSI to > put a caller on hold. > Then able to retreive that caller. > > Example - > > Softkey Hold > Softkey Retreive Call > Softkey End Call > > -gcc > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Free Pop-Up Blocker - Get it now http://companion.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can't get caller id?
when i connect the phone directly i get the caller id after the first ring and b4 the second one. what does that suggest? --- Steven Critchfield <[EMAIL PROTECTED]> wrote: > On Fri, 2003-11-21 at 08:17, C M wrote: > > --- Steven Critchfield <[EMAIL PROTECTED]> wrote: > > > On Fri, 2003-11-21 at 07:14, C M wrote: > > > > not with *. > > > > > > > > i guess its the country issue. like i saw som > > > posts > > > > with callerid issues in uk. i am in nepal. how > can > > > i > > > > configure * AGI to make it compatible with my > > > country. > > > > has anyone written a script? > > > > > > AGI is not configurable, and also would not help > you > > > here. What you need > > > to do is track down what form of CallerId you > have > > > and when it is sent > > > on the line. Then see about tweaking callerid.c > to > > > support it. > > > > > > > ok > > > > how can i the track callerid information? ask the > > telco? how does yours work? i want to know how urs > > happens so that i can figure out how diff is mine. > > Callerid in the Us is signaled with FSK between the > first and second > ring. For more information, you will need to consult > the source as it > will be definitive. > > So far, I think it has been established that in the > UK, the callerid is > sent after a line reversal and before the first > ring. > > I think it has also been established that some > locations have callerid > as either dtmf or mf. I don't usually pay attention > to this as much > because I can't help them. > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users = Designs __ Do you Yahoo!? Free Pop-Up Blocker - Get it now http://companion.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DIAX, IAX2 and latency
Hello, today I tried a DIAX -> * -> DIAX connection over the internet (768/128 ADSL connection on both sides). The sound quality was great. However, we had some latency problems, and also, if both sides where not talking the first words had some problems getting thru. Is this expected, is there anything that can be done on our setup, any magical iax.conf entry? Thanks and best regards from Germany pos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] hold music =]
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Areski > Sent: woensdag 19 november 2003 19:27 > To: Asterisk-Users Mailing-list > Subject: RE: [Asterisk-Users] hold music =] > > > http://www.voip-info.org/tiki-index.php?page=Asterisk%20mpg123%20redhat > FYI this tip also applies to Debian users. I'm running Debian stable, used dselect to install mpg123 (did a search in dselect for 'mpg123', found a hit, installed it). Dselect installs mpg321 and places a symlink mgg123, just like redhat. The solution is also simular. Arnold ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can you monitor a call via the asterisk speaker system and do a call pickup if you wish
Reason. I have a fax/ans phone with handset, that lets you monitor the caller, so if you wish, you can pickup the call. The asterisk is undergoing testing, it will then be online tested at the house so I can get more familiar in setting components up, e.g. sip phones, voicemail, transfers etc. But, I really need the monitoring of the Voicemall being left, with the ability to pick up the call, otherwise, someone is going to be unhappy. If this functionality isn't available. Can it be added as a request to the developers. Regards...martin -- Keep the phase, baby. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outline For Asterisk Book - Please Review & Comment
Asterisk Users In an attempt to help Asterisk move forward, a number of us have decided to create a book. It would initially be released as an "ebook" that could be sent to newbies to help them up the rather steep learning curve. Ultimately I would like to see it published and sold in bookstores (preferably by O'Reilly & Co.). Below is the outline for the book. We REALLY need as much input as we can get. I would like to completely flesh-out the outline, then I would like to start accepting submissions from the user community for each of the sections/chapters/topics covered in the outline. I have to stress here that I AM NOT AN ASTERISK GURU. I need help from the real gurus (especially: Steven Critchfield, John Todd, Tilghman Lesher, Olle Johansson, and where possible/necessary Mark and Martin). If this works, it will help Asterisk achieve the same kind of global success as Apache, Samba, and other Linux staples. If you want to chat, I am lurking in the #asterisk-doc channel on Freenode IRC. I'm 'ssokol'. Others on this project (so far) are Jared Smith and Leif Madson. A "living" copy of this outline can be found at: http://www.sokol-associates.com/outline.htm . I will try to update it daily with your suggestions. It will also be the basis for the project outline (completion %, assignments, etc.). Thanks, Steve Sokol Sokol & Associates, LLC [Outline Guide] The following outline describes the layout for the book. 1. <- Section a. <- Chapter 1) <- Sub-Chapter i. <- Topic Heading *. <- Sidebar Heading 1} <- Graphic or Chart 1> <- Table [Outline] 1. Introduction to Asterisk a. Introductory letter from Mark Spencer 1) Whatever Mark has to say... 2) Digium Reference Information i. Web Site ii. Phone Number b. The Business Case For Asterisk [Somebody From The Business Side Writes This] c. General concept of asterisk 1) Asterisk: Swiss Army Knife of Telephony 2) PBX, IVR, ACD 3) What To Expect i. Asterisk Is Not A Turnkey System ii. Don't Like It? Change It Yourself. iii. Opensource, GPL and LGPL Licensing d. Asterisk architecture 1) The Big Picture 2) Channels 3) Codec Conversions 4) Etc. e. Key components 1) Asterisk software i. Asterisk (Main PBX & Channels) ii. Zaptel (Drivers for Zaptel Hardware) iii. Libpri (ISDN PRI Drivers for Zaptel) 2) Zaptel Hardware i. Overview ii. X100P - Single Port FXO Line Interface iii. S100U - Single Port FXS USB Interface iv. TDM400P - 4 Port FXS Analog Interface v. T100P - Single Span T1/E1 Interface vi. TE410P - Quad-Span T1/E1 Interface 3) Channels i. Zaptel Devices/Channels ii. The IAX Protocol iii. SIP iv. MGCP v. Skinny vi. H323 4) Applications i. Dial and Other Basics ii. Voicemail iii. Dial-Plan Scripting 5) Extensibility i. AGI ii. Custom Applications f. Add-On/Optional Components 1) Software i. Gnophone ii. VoIP Soft Phones iii. DIAX iv. Gastman v. Open H.323 2) Hardware i. VoIP Hard-Phones ii. VoIP Gateways ii. Channel Banks 2. Installing Asterisk *. Asterisk Quickstart 1) Install PC Hardware 2) Download Asterisk Software 3) Build Asterisk 4) Install Asterisk 5) Configure Autostart a. Requirements *) Picking A Solid System 1) PC Hardware Requirements i. SOHO/Residential System ii. Small Business System iii. Medium Business/Small Call-Center System iv. Enterprise System v. VoIP Carrier System 2) Linux Requirements *. Linux Installation Is Not Cover
Re: [Asterisk-Users] Struggling with grandstream sip to asterisk
TC wrote: You can't use INFO with the GS. * and GS interpret the INFO standard differently. As a result, the GS does multiple digit transmission. Use either inband or the RFC2833 option. INFO will not work no matter what you do. I am using the following settings Software Version:Program--1.0.4.20Bootloader--1.0.0.12 HTML--1.0.0.19 Send DTMF: via SIP INFO DTMF Payload Type: 101 and I can dial dtmf fine & interact with * Background menus & other PBX IVR systems What test are you doing to show INFO is not supported on GS It's not that INFO is not supported, it's just that it did not work without transmitting multiple copies of digits to *. I have been told (by GS) that this was a problem with *, and it may be that it has been fixed in more recent CVS releases (mine is about 2 months old right now). On the otherhand, unless I use inband, the # key does not work on some (not all) IVR systems that I dial into. Rather than wasting time trying to figure out why, I just use inband. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Struggling with grandstream sip to asterisk
On Fri, 2003-11-21 at 17:57, TC wrote: > I am using the following settings > > Software Version:Program--1.0.4.20Bootloader--1.0.0.12 > HTML--1.0.0.19 Where is this firmware from? The GS site is still at Program--1.0.3.81Bootloader--1.0.0.7HTML--1.0.0.18 or is Sipphone ahead of GS? -- Dave Cotton <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Solved! Snom 200 Busy signal
Now that you mention it, I did observe the PUBLISH message. Can someone please tell me exactly the change that was made that fixed this? (file/lines) I can do a diff -r and see a few changes from CVS but I'd like to be sure. We have a lot of custom changes as well so it's non-trivial to update to CVS. I would definitely like to check on this one patch though. Thanks. - Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Lawson Sent: Thursday, November 20, 2003 7:03 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Solved! Snom 200 Busy signal As a follow up to my earlier posting, the problem with the Snom 200 Busy signal was the firmware! I reverted back to 1.16x and everything's OK - Matt I had the same type of problem you did when I first upgraded to 2.02t. But I did a CVS update on 11/19/03, and everything is fine now. What I did notice (and I have a customer that saw the same thing) is that that been able to determine what they're publishing with it. But Asterisk didn't seem to like it until I did a CVS update. It didn't seem to respond to it. Now Asterisk sends back a "Status: 405 Method Not Allowed" message, and everything is fine. I can't definitively say this was the cause because I didn't go back to the old Asterisk code and verify. But I can tell you that my Snom 200 phones with version 2.02t work fine now. Jerry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Struggling with grandstream sip to asterisk
> You can't use INFO with the GS. * and GS interpret the INFO standard > differently. As a result, the GS does multiple digit transmission. Use > either inband or the RFC2833 option. INFO will not work no matter what > you do. I am using the following settings Software Version:Program--1.0.4.20Bootloader--1.0.0.12 HTML--1.0.0.19 Send DTMF: via SIP INFO DTMF Payload Type: 101 and I can dial dtmf fine & interact with * Background menus & other PBX IVR systems What test are you doing to show INFO is not supported on GS ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP with Cisco GW
Hi, I have a Cisco IAD 2431 which has MGCP protocol; I cannot make it to work againts Asterisk; at least there is some MGCP conversation between them but when I offhook a phone attached to IAD I get no tone at all. As anybody managed to get working Asterisk against an MGCP Cisco gateway ? Which MGCP version should I use ? TIA Attached are the configs: Cisco -- ! version 12.2 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname 192.168.65.200 ! logging queue-limit 100 enable secret enable password ! ip subnet-zero ! ! no ip domain lookup ! isdn switch-type primary-net5 ! ! voice call carrier capacity active ! voice service pots ! voice service voip ! voice class codec 10 codec preference 1 gsmfr codec preference 2 g711alaw ! ! ! ! ! ! ! no voice hpi capture buffer no voice hpi capture destination ! ! mta receive maximum-recipients 0 ! ! controller T1 1/0 shutdown framing esf linecode b8zs ! ! ! interface FastEthernet0/0 ip address 192.168.65.200 255.255.255.0 duplex auto speed auto ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! ip http server ip classless ip route 0.0.0.0 0.0.0.0 192.168.65.1 ! ! ! dialer-list 1 protocol ip permit ! ! call rsvp-sync ! voice-port 2/0 no vad timing hookflash-in 750 ! voice-port 2/1 no vad timing hookflash-in 750 ! voice-port 2/2 no vad timing hookflash-in 750 ! voice-port 2/3 no vad timing hookflash-in 750 ! voice-port 2/4 ! voice-port 2/5 ! voice-port 2/6 ! voice-port 2/7 ! voice-port 2/8 ! voice-port 2/9 ! voice-port 2/10 ! voice-port 2/11 ! voice-port 2/12 ! voice-port 2/13 ! voice-port 2/14 ! voice-port 2/15 ! mgcp mgcp call-agent 192.168.65.100 service-type mgcp version 1.0 mgcp package-capability rtp-package mgcp default-package dtmf-package no mgcp timer receive-rtcp no mgcp validate domain-name mgcp bind control source-interface FastEthernet0/0 ! mgcp profile default ! dial-peer cor custom ! ! ! dial-peer voice 2 pots application mgcpapp port 2/1 ! dial-peer voice 3 pots application mgcpapp port 2/2 ! dial-peer voice 4 pots application mgcpapp port 2/3 ! dial-peer voice 1 pots application mgcpapp port 2/0 ! ! line con 0 line aux 0 line vty 0 4 password login ! end mgcp.conf -- ; ; MGCP Configuration for Asterisk ; [general] port = 2727 bindaddr = 0.0.0.0 [192.168.65.200] host = 192.168.65.200 context = local line = aaln/S2/0 line = aaln/S2/1 line = aaln/S2/2 line = aaln/S2/3 -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Struggling with grandstream sip to asterisk
*CLI> NOTICE[81926]: File chan_sip.c, Line 5215 (handle_request): Registration
from '' failed for '192.168.1.70'
I've seen this in only two cases:
1) when the SIP user ID, the Authenticate ID (on the GS) and the
extension name in sip.conf are not all the same, in your case, 206. The
user name in sip.conf doesn't matter, and in fact doesent seem to do
anything at all with the GS.
2) You are using a secret and the authenticate Password does not
match the secret. This does not appear to be relevant to your situation.
You might also check to be sure that the registration option in the GS
is turned off, since you are hard coding the IP address. There is a bug
in the current GS firmware (supposed to be fixed soon) that sometimes
messes up the registration renewal.
-- Username not entered
-- Executing Hangup("SIP/206-7ecb", "") in new stack
You can't use INFO with the GS. * and GS interpret the INFO standard
differently. As a result, the GS does multiple digit transmission. Use
either inband or the RFC2833 option. INFO will not work no matter what
you do.
Also, if you are hard coding your phone's IP addresses, then the permit
option has no meaning.
[206]
username=206
context=extensions
qualify=yes
incominglimit=1
type=friend
insecure=yes
host=192.168.1.70
permit=192.168.1.0/255.255.255.0
dtmfmode=info
canreinvite=no
reinvite=no
callgroup=1
pickupgroup=1
disallow=all
allow=alaw
allow=ulaw
___
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Re: [Asterisk-Users] Cisco DTMF Issue
I am doing debug on the voip rtp session named-events. The router only does RTP encoding for the dtmf on calls that originate from an FXS interface on the router - not from a phone running SCCP that is connected to the router (Cisco CallManger Express / ITS). However, if I call one of the FXS phones from one of the SCCP phones, my dtmf digits are passed to the FXS phone. Also, if I debug the phones, the router does receive the button press events, it just never chooses to follow the peer's rtp dtmf relay setting. I think this is a bug in the IOS, personally. Anyone else have any experience with this issue? Josh - Original Message - From: "Andrey S Pankov" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, November 21, 2003 4:49 AM Subject: Re: [Asterisk-Users] Cisco DTMF Issue > Hi, > > Try this one: > > "Dual Tone Multifrequency Relay for SIP calls Using Named Telephone Events" > http://www.cisco.com/en/US/products/sw/iosswrel/ps1839/products_feature_guid e09186a0080087edb.html > > Andrey. > > > This is a password-protected document (CCO account required.) Can > > you refer to a non-password protected URL for the sake of the > > archives? > > > > JT > > > > >http://www.cisco.com/en/US/customer/products/sw/iosswrel/ps1839/products_f > > >eature_guide09186a0080087edb.html > > > > > >Andrey. > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Outgoing-call and enter user in Conference
Use the manager interface: Action: Originate Exten: 8320 # meetme room extension in extensions.conf Channel: Zap/g1/3125551212 # outside number to dial Context: default Priority: 1 MATT--- -Original Message- From: Areski [mailto:[EMAIL PROTECTED] Sent: Friday, November 21, 2003 10:41 AM To: Asterisk-Users Mailing-list Subject: [Asterisk-Users] Outgoing-call and enter user in Conference Hi folks, Just wondering if someone have already done something like that : SIP Client_A ---1)call---> ASTERISK ---2)outgoingcall-PSTN-->Client_B | | 3) Enter conference | MeetMe <' with user A Make 2 user in conference, it's definitely easy, but call an other user and put the both in conference,I still don't have any idea how to do it! Every ideas are be welcome ;) Thanks in advance, Aresk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outgoing-call and enter user in Conference
Hi folks, Just wondering if someone have already done something like that : SIP Client_A ---1)call---> ASTERISK ---2)outgoingcall-PSTN-->Client_B | | 3) Enter conference | MeetMe <' with user A Make 2 user in conference, it's definitely easy, but call an other user and put the both in conference,I still don't have any idea how to do it! Every ideas are be welcome ;) Thanks in advance, Aresk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FYI: Simple Small Asterisk install..
Thanks Dorian for the response. Guess you are using Gentoo as any other flavor on fully loaded mode. Though its nice to know Asterisk performs well on Gentoo too, it would be great to see if any customized installation has been made for asterisk, to get it in a min. resource/full throttle mode. i am up such a system this weekend. will share my thoughts once i get it to work. Sri Dorian Gray wrote: Sri wrote: Anyone tried to run Asterisk on Gentoo Linux ? If yes, Howz the performance ? been working fine for about 4 weeks now under 2.4.23_pre8-gss. performance is nominal, I don't feel that the "optimization" part of gentoo is a real benefit (but that's not why I use gentoo, anyway). Any special configuration gotcha's that we have to look out for ? the portage tree only has asterisk-0.2 in it, you might want to try out either of these ebuild sets: (0.5 asterisk snapshot version + 0.8 zaptel snapshot version) http://bugs.gentoo.org/show_bug.cgi?id=30873 http://bugs.gentoo.org/show_bug.cgi?id=30531 --OR-- (latest-cvs version) http://bugs.gentoo.org/show_bug.cgi?id=33345 What was the size of the final installation ? of gentoo? maybe 4gb; but I am using this server for a number of other things as well: iptables, apache, mysql, php, etc. (it even has xfree installed, although I'd like to get rid of that...) Is it bulkier than Trustix ? it's possible to make gentoo pretty small, but it will probably take more effort to do so. you could probably try doing a stage-1 install and leave out absolutely anything unnecessary... (e.g. build with USE="-*" at first, and then slowly add just the pieces you need) it's actually possible to install the portage system onto another distro, but the required python libs are pretty big by default (80-90mb, I think). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Upgrade CISCO 7960 Question
Title: Message Yes, it is. But why would you want to do that when yo said what you want it to be at 6.0. Maybe you didn't expling what and why you want to do it in enough detail to get a good answer. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz JozwiakSent: Friday, November 21, 2003 8:11 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Upgrade CISCO 7960 Question Hello, My Cisco phone has software: Boot Load: PC030300 Ver: 3.2(7.0) And I want to upgrade it to SIP 6.0 Is it possible or I have to upgrade to ealier then 6.0 and then to 6.0 ? bart
Re: Uptime counters (was RE: [Asterisk-Users] Bayonne and Asterisk)
James Sizemore wrote: I did not even know about it! But seeing as it is not in the change log no wonder? You have the bug number the notes are under for usage? ID # 345 10/02/03 - logger_reload.diff Summary - 'logger reload' CLI command Description - Closes and reopens the log files. Good for those wanting to rotate log files. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can't get caller id?
Steven Critchfield wrote: Callerid in the Us is signaled with FSK between the first and second ring. For more information, you will need to consult the source as it will be definitive. So far, I think it has been established that in the UK, the callerid is sent after a line reversal and before the first ring. The UK uses a similar FSK message format to the US (not identical, but similar). However UK ringing doesn't have a huge silence period in it. Therefore, they send the caller ID before the first ring. Does anyone know why they use a reversal to indicate the CLI is coming? Most digital exchange lines are incapable of reversing the line, so it seems like this choice forces the use of a new line card for any sub. who joins the CLI service - weird! Various alternatives seem simpler. In something like * it would not be necessary to look for the reversal. Conintuously running the FSK receiver isn't that big a compute load. It could be run continuously between calls, and pick up any FSK messages which arrive. I think it has also been established that some locations have callerid as either dtmf or mf. I don't usually pay attention to this as much because I can't help them. DTMF is used in some places. Japan uses FSK, but a rather different message format. There isn't a whole lot of global standardisation in CLI! Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P ERRORS under load
Mark- The system is Tyan s2723, which includes an integrated graphics controller, and I've connected a monitor to this port. I don't start the X-windows software however, and I don't use this monitor. For starting my scripts, I connect using SSH via an Ethernet port, however everything operates in background mode, there is no output to any console while these tests are running - just log files. I do receive an occasional Double/missed interrupt message - but very few of these. Will report the results of testing your new driver code! Thanks Scott Scott M. Stingel Emerging Voice Technology Inc. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Mark Spencer > Sent: Friday, November 21, 2003 5:39 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] TE410P ERRORS under load > > > I've made some updates that may help alleviate these problems > on TE410P on > E1. I'd like you to test and tell me if that helps. Also, > I'd like you > to confirm that you are using neither serial console nor frame buffer > (graphical) console. > > Mark > > On Thu, 20 Nov 2003, Scott Stingel wrote: > > > Hi all- > > > > HELP! > > > > This is actually a revisit of a problem that I had earlier > with E400P's at a > > customer site. Customer still gets locked up channel > problem, but has > > learned to live with it (channels clear themselves after > several minutes). > > The symptoms, which I believe are directly related: > > > > I'm having problems with tons of framing and "read" errors on my E1 > > connections (and occasional stuck channels) when I run a > very simple IVR > > script under anything other than light load. Previously, I > had thought this > > problem related to my customer's PBX connections (Nortel > DMS100), but in the > > last week I've had the opportunity to work in-house with a couple of > > TE410P's, with one span making calls to another on the same machine. > > > > I have two simple scripts running under the dialplan. On > span 1, I simply > > answer each call, play a short message, and hang up. On > span 2, I run a > > Perl script that formats and drops calls (into > /var/spool/asterisk/outgoing) > > for all channels at staggered times. These calls are > simple outgoing calls > > that dial a number, wait 2 seconds, and hangup. After a > few seconds, each > > call repeats. > > > > I can run this scenario on up to 10 channels at once with > *no errors*. > > Above 10 channels, I start to get many (several per second > when running 30 > > channels) framing and "read" errors, with text similar to > the following: > > > > WARNING[1167272128]: File chan_zap.c, Line 5670 > (zt_pri_error): PRI: !! Got > > reject for frame 26, retransmitting frame 26 now, up_dating n_r! > > (repeating for each error several times, with ascending > retransmitted frame > > numbers) > > > > and also, less often: > > WARNING[1167272128]: File chan_zap.c, Line 5670 > (zt_pri_error): PRI: Read on > > NN failed: Unknown error 500 (NN is the channel) > > > > MY SETUP: > > Tyan S2723, with dual Xeon's running at 2.4 GHz, 1MB memory > > Redhat 9 > > Two TE410P's, spans set for E1. Problem happens between > spans on different > > boards, or spans on the same board. > > Sending board setup in zaptel.conf, for example: > span=2,1,0,ccs,hdb3 > > (adding crc4 makes no difference) > > Recving board setup in zaptel.conf, for example: > span=1,0,0,ccs,hdb3 " " > > > > In zapata.conf, sender is "pri_net" and receiver is "pri_cpe" > > > > QUESTIONS: > > ANYONE: Has anyone else experienced these framing problems > in any scenario, > > and if so, what did you do about it please? > > > > FOR THE ISDN GURU's: What exactly does the framing error indicate? > > > > THANKS IN ADVANCE FOR HELPING ME SOLVE THIS LOAD_RELATED PROBLEM. > > > > Scott M. Stingel > > Emerging Voice Technology Inc. > > Palo Alto, California and London, England > > > > URL:www.evtmedia.com > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can't get caller id?
On Fri, 2003-11-21 at 08:17, C M wrote: > --- Steven Critchfield <[EMAIL PROTECTED]> wrote: > > On Fri, 2003-11-21 at 07:14, C M wrote: > > > not with *. > > > > > > i guess its the country issue. like i saw som > > posts > > > with callerid issues in uk. i am in nepal. how can > > i > > > configure * AGI to make it compatible with my > > country. > > > has anyone written a script? > > > > AGI is not configurable, and also would not help you > > here. What you need > > to do is track down what form of CallerId you have > > and when it is sent > > on the line. Then see about tweaking callerid.c to > > support it. > > > > ok > > how can i the track callerid information? ask the > telco? how does yours work? i want to know how urs > happens so that i can figure out how diff is mine. Callerid in the Us is signaled with FSK between the first and second ring. For more information, you will need to consult the source as it will be definitive. So far, I think it has been established that in the UK, the callerid is sent after a line reversal and before the first ring. I think it has also been established that some locations have callerid as either dtmf or mf. I don't usually pay attention to this as much because I can't help them. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can't get caller id?
--- Steven Critchfield <[EMAIL PROTECTED]> wrote: > On Fri, 2003-11-21 at 07:14, C M wrote: > > not with *. > > > > i guess its the country issue. like i saw som > posts > > with callerid issues in uk. i am in nepal. how can > i > > configure * AGI to make it compatible with my > country. > > has anyone written a script? > > AGI is not configurable, and also would not help you > here. What you need > to do is track down what form of CallerId you have > and when it is sent > on the line. Then see about tweaking callerid.c to > support it. > ok how can i the track callerid information? ask the telco? how does yours work? i want to know how urs happens so that i can figure out how diff is mine. cm > > --- Chris Hirsch <[EMAIL PROTECTED]> > wrote: > > > C M wrote: > > > > > > >hi, > > > > > > > >i can get callerid in my phone directly > connected > > > to > > > >the pstn line. when i cannoect it o * it doen't > > > give > > > >me callerid. i have set usecallerid=yes in > > > zapata.conf > > > >file/ > > > > > > > >whaat could have happened? > > > > > > > > > > > > > > > Has it ever worked? I had soemthing similar with > a > > > new FXO card...as it > > > turns out it was bad... > > > > > > -- > > > Only in America...do we leave cars worth > thousands > > > of dollars in the > > > driveway and leave useless things and junk in > boxes > > > in the garage > > > > > > > > > http://ccicolorado.org > > > Exceptional Dogs for Exceptional People - Help > Out > > > Today! > > > > > > > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > = > > Designs > > > > __ > > Do you Yahoo!? > > Free Pop-Up Blocker - Get it now > > http://companion.yahoo.com/ > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users = Designs __ Do you Yahoo!? Free Pop-Up Blocker - Get it now http://companion.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Solved! Snom 200 Busy signal
-Original Message- >From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Lawson >Sent: Thursday, November 20, 2003 7:03 PM >To: [EMAIL PROTECTED] >Subject: [Asterisk-Users] Solved! Snom 200 Busy signal >As a follow up to my earlier posting, the problem with the Snom 200 Busy >signal was the firmware! I reverted back to 1.16x and everything's OK. >That made today pretty complicated, since I already had a new kernel and >a new Asterisk build I was trying all at once >Hopefully someone else will benefit... >- Matt I had the same type of problem you did when I first upgraded to 2.02t. But I did a CVS update on 11/19/03, and everything is fine now. What I did notice (and I have a customer that saw the same thing) is that that version 2.02t has added a new SIP "Publish" message. I still haven't been able to determine what they're publishing with it. But Asterisk didn't seem to like it until I did a CVS update. It didn't seem to respond to it. Now Asterisk sends back a "Status: 405 Method Not Allowed" message, and everything is fine. I can't definitively say this was the cause because I didn't go back to the old Asterisk code and verify. But I can tell you that my Snom 200 phones with version 2.02t work fine now. Jerry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can't get caller id?
On Fri, 2003-11-21 at 07:14, C M wrote: > not with *. > > i guess its the country issue. like i saw som posts > with callerid issues in uk. i am in nepal. how can i > configure * AGI to make it compatible with my country. > has anyone written a script? AGI is not configurable, and also would not help you here. What you need to do is track down what form of CallerId you have and when it is sent on the line. Then see about tweaking callerid.c to support it. > --- Chris Hirsch <[EMAIL PROTECTED]> wrote: > > C M wrote: > > > > >hi, > > > > > >i can get callerid in my phone directly connected > > to > > >the pstn line. when i cannoect it o * it doen't > > give > > >me callerid. i have set usecallerid=yes in > > zapata.conf > > >file/ > > > > > >whaat could have happened? > > > > > > > > > > > Has it ever worked? I had soemthing similar with a > > new FXO card...as it > > turns out it was bad... > > > > -- > > Only in America...do we leave cars worth thousands > > of dollars in the > > driveway and leave useless things and junk in boxes > > in the garage > > > > > > http://ccicolorado.org > > Exceptional Dogs for Exceptional People - Help Out > > Today! > > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > = > Designs > > __ > Do you Yahoo!? > Free Pop-Up Blocker - Get it now > http://companion.yahoo.com/ > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SAY NUMBER in AGI?
I am trying to use the SAY NUMBER command from an AGI script but it does not seem to be working.. If I use "EXEC SayNumber 2" and execute the asterisk command from the AGI it works and I hear the 2 said on the phone.. If I use "SAY NUMBER 2" I see "-- Playing 'digits/2' (language 'en')" on the console but I don't hear the number said on the phone.. I would prefer to use the AGI commands from within the AGI script.. :) Is this a bug or am I doing something wrong?? Thanks.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is Asterisk suitable for this use?
Anton L. Kapela wrote: Jeremy McNamara said: Do you really want all those spans going down cause someone tripped over a power cable or your hard drive nukes itself? How's this worse than an as5300? I could install ata-flash and get high-ish end pc hardware (rcc serverworks boards, etc). Heck, if it was such an evil thing, I'm sure Voicepulse (along with every other clec) and friends wouldn't be doing it! --Tk I think what Jeremy is saying is that by having multiple servers you inherintly have higher fault tolerence and a lower chance of system failure.. You could suffer a single server failure and your system would still run and service you users.. You will not be in a panic to get that new part in 30 seconds because the rest of the servers will be able to take over the load for the failed one.. Thats all.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] channel banks
We have 8 z-plex and they are fairly new! We replaced them with what is the best so far. adtran 750. We got them on ebay for around $ 500.00 new! With all the 24 fxs ports! - Original Message - From: "Alex Pavlovic" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, November 20, 2003 7:15 PM Subject: [Asterisk-Users] channel banks > Hi, > > Recently we started experiencing problems with our zhone > z-plex 10-24S channel bank. Currently we are looking to get > a replacement. Just wondering if anyone can comment > on their personal experience or maybe give some pointers. > We are looking for a reliable product to support a call center > and live operators. > > Cheers. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is Asterisk suitable for this use?
Jeremy McNamara said: > > Do you really want all those spans going down cause someone tripped > over > a power cable or your hard drive nukes itself? How's this worse than an as5300? I could install ata-flash and get high-ish end pc hardware (rcc serverworks boards, etc). Heck, if it was such an evil thing, I'm sure Voicepulse (along with every other clec) and friends wouldn't be doing it! --Tk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can't get caller id?
not with *. i guess its the country issue. like i saw som posts with callerid issues in uk. i am in nepal. how can i configure * AGI to make it compatible with my country. has anyone written a script? cm --- Chris Hirsch <[EMAIL PROTECTED]> wrote: > C M wrote: > > >hi, > > > >i can get callerid in my phone directly connected > to > >the pstn line. when i cannoect it o * it doen't > give > >me callerid. i have set usecallerid=yes in > zapata.conf > >file/ > > > >whaat could have happened? > > > > > > > Has it ever worked? I had soemthing similar with a > new FXO card...as it > turns out it was bad... > > -- > Only in America...do we leave cars worth thousands > of dollars in the > driveway and leave useless things and junk in boxes > in the garage > > > http://ccicolorado.org > Exceptional Dogs for Exceptional People - Help Out > Today! > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users = Designs __ Do you Yahoo!? Free Pop-Up Blocker - Get it now http://companion.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Upgrade CISCO 7960 Question
Hello, My Cisco phone has software: Boot Load: PC030300 Ver: 3.2(7.0) And I want to upgrade it to SIP 6.0 Is it possible or I have to upgrade to ealier then 6.0 and then to 6.0 ? bart
Re: [Asterisk-Users] can't get caller id?
C M wrote: hi, i can get callerid in my phone directly connected to the pstn line. when i cannoect it o * it doen't give me callerid. i have set usecallerid=yes in zapata.conf file/ whaat could have happened? Has it ever worked? I had soemthing similar with a new FXO card...as it turns out it was bad... -- Only in America...do we leave cars worth thousands of dollars in the driveway and leave useless things and junk in boxes in the garage http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk SIP implementation
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 What does Asterisk SIP support? What I mean is, is it just for VoIP calls or i can send message to someone? As i understand with SIP i can contact [EMAIL PROTECTED] and communicate via voice, video or whatever. Can i use Astersik for anything else beside voice telephony? There is format_jpeg and app_image, can anybody direct me how to try sending picture? Dario - -- You single-handedly fought your way into this hopeless mess. -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQE/vgrdGWimCEGw9AoRArZ4AKCsI0ewcNs0xm+grXqhGazhEsA50gCdHNFt E3Pbl2Zg9OjL6zYAylTLrQo= =7q4h -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tunnel iax via gnophone with ssh?
Fearghas McKay wrote: Don't use CIPE, it has holes in it and is breakable. Use IPsec either FreeS/WAN or if you are running 2.6 kernel you can use its IPsec stack. Or an appliance to provide tunnels if you don't want to fiddle with kernels. I have FreeS/WAN setup for my wireless...I guess its time to set it up so its available from the outside...it would solve some other problems I've been having too... Whats the deal with teh IPsec stack in 2.6? Is it just FreeS/WAN and I don't have to patch any more? -- Software is like entropy. It is difficult to grasp, weighs nothing, and obeys the Second Law of Thermodynamics;i.e., it always increases. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Proxy issues
I'm trying to have the following small architecture: PSTN | Linux box with an public IP and Asterisk | Internet | Firewall/NAT/Router | SIP Proxy | IP Phones (or soft phone like X-Lite) Firt of all, can I do such a schema ? Let assume that the previous answer is yes... So if anybody that succeed in such a thing can hel me a bit I would really appreciate... By the way, I can put the sip proxy on the firewall or forward some ports. And I'd like to register on asterisk... Thanks a lot for your help, chris -- Christophe Sauthier <[EMAIL PROTECTED]> Eikonex - Open Source Engineering (http://www.eikonex.net) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tunnel iax via gnophone with ssh?
Il ven, 2003-11-21 alle 11:29, Fearghas McKay ha scritto: > At 15:38 -0500 20/11/03, Billy Huddleston wrote: > >Use CIPE, It's a UDP based VPN solution. > > Don't use CIPE, it has holes in it and is breakable. I've started testing OpenVPN and it doesn't seem to be that bad. Its main weakness, IMHO, is that it does not scale, but if you have a handful of sites to connect, it could do the job. -- Emanuele ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-oh323 v0.5.7 bugfix release
Senad Jordanovic wrote: Does this implementation of H323 for * terminates and originates calls successfully from Cisco 5300? The H.323 driver that ships with Asterisk most certainly inter-operates with As5300. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Which ISDM BRI Card for Asterisk?
I'm testing the Eicon Server BRI/PRI cards for the moment and they are very satisfactory because they have some interesting features onboard like echo cancelation () and onboard encoding see www.eicon.com Van: Daniel ANDRE [mailto:[EMAIL PROTECTED] Verzonden: vr 21/11/2003 9:48 Aan: [EMAIL PROTECTED] Onderwerp: [Asterisk-Users] Which ISDM BRI Card for Asterisk? Hello all, I wonder to have some feedback on using ISDN BRI Cards with Asterisk and the Echo problem. I have tried a simple BRI card with i4l driver and encounter huge echo problem. I have tried to solve it with a Sw chocanceller without success. What I'd like to know is wether some of you have used other BRI Cards (I have seen reference to Eicon cards on this list) and if the echo disappeares with these cards? Best regards, Daniel ANDRE -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users <>
