Re: [Asterisk-Users] Re: Re: quadbri NT_mode S-Bus Problem
I'm currently looking for a good solid solution that works here in the US with BRI-U NI-1 off a DMS100 or 5ESS. bkw - Original Message - From: "Ben Bosshardt" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, July 19, 2004 11:18 PM Subject: [Asterisk-Users] Re: Re: quadbri NT_mode S-Bus Problem > >Are you using Santis Siemens ISDN NT1 ? If yes , we have the same from > >Siemens Switzerland , What I've done is to get one cable from ISDN NT > >-- > ISDN MODEM in * Machine ( HFC - S Modem Euro 30 - 40 ) and then > >used bristuff ( google for it ) , and used that , it just works! . I > >can send you my configs if you need som ehlp > > I gladly look at your config files to see what I have done wrong. At the > moment the setup is hooked up that I can make inbound and outbound calls > (from ISDN and SIP clients), just with the limitations as below : > > 1. On outbound calls, I get the normal rining call progress tone althought > the the other party has not even been reached. This then changes from normal > ringing suddenly to busy when the other party is sending a busy signal. I'd > rather have the call progress send a busy signal right away. > > 2. Internal calls between two ISDN client phones on the S-bus is not > possible. The phone rings but the call is dropped as soon as it is answered. > > Kind Regards, > Ben > > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: spa-3000 review?
> Set Admin-Advanced->Line1->DialPlan-> > ([2-9]xx<:@gw0>|[3469]11|0|00|[2-9]xx|1xxx[2-9]xxS0|.) > > "[2-9]xx<:@gw0>" will send 211,311,411...911 to gw0 which is the local > pots port. this is not what i expected. i expected something like ([49]11<:@gw0>|rest of dial plan) randy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem Starting RC1
On Mon, 19 Jul 2004, Nathan Martinez wrote: > Hello, > > I was running a very simple test setup with * HEAD 7/15/2004 on Fedora > Core 2 and things were working fine. Today I upgraded to RC1 and my > asterisk service will no longer start. I downloaded the tarball, > extracted, ran 'make', ran 'service asterisk stop', ran 'make install', > removed all files in /etc/asterisk, ran 'make samples' and then ran > 'service asterisk start'. You need to go to your modules.conf and replace res_parking.so with res_features.so. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo on a PRI
On Mon, 19 Jul 2004, David Goldfein wrote: > Hi, > I recently set up the following in a production system (2.8 GHZ Xeon, 1 Gig > Memory, Dell 2650). > > Telco - PRI - Asterisk - T1 - PBX > > I am getting an occasional noticeable echo on some of the phone lines > (random inbound and outbound). Everyone I ask keeps telling me that I can't > be having echo since I am on a PRI, which is a digital circuit. Ok, so I > can't be having echo, but I am! Does anyone have any ideas of what might be > causing the echo in this situation? If you've got any analogue anywhere in the call patch then echo is a possibility. Specifically 2-wire analogue. So if one or both of your callers are on analogue phones. Its not usually an issue with your sort of setup though because the Zaptel driver has echo cancellation (have you enabled it?) and the overall call delay is small. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 500 Voicemail
My Polycom Message button goes straight to voicemail. Here's how: 1) Use the latest firmware, available on the Wiki 2) In your phone.cfg file (for each phone) set 3) In your extensions.conf, have something like: exten => 76,1,VoiceMailMain2([EMAIL PROTECTED]) (Let's assume your voice mailbox is the same as your extension) Then when you push the message button, asterisk will ask for your password! You're in! John Chris A. Icide wrote: On 04:28 PM 7/19/2004, Wiley E. Siler wrote: >Mine does the same. Once in Message center I can choose selection >1.Message Center and then soft key Select.Then I select the >registered line that I want to check voice mail on. That is no less than >4 key strokes just to get into your voice mail. Not many to me but tons >to an unskilled user. However, in the documentation regarding the >bypassInstantMessage value, supposedly, setting bypassInstantMessage to >1 is supposed to allow you to go right into voice mail without >navigating the Message Center. That is the big question on my mind at >this point. I have yet to get this to work and I also don't think I am >receiving any SIMPLE messages ti show me that I have messages waiting. > >Do you get a message waiting indicator? > I do get MWI, there are a few things you need to set, and I forget what off the top of my head, soon as I can look and post it here. I haven't tried the bypassInstantMessage value, but I'll take a look and see if I can get it to work. -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] quadbri NT_mode S-Bus Problem
>Are you using Santis Siemens ISDN NT1 ? If yes , we have the same from >Siemens Switzerland , What I've done is to get one cable from ISDN NT >-- > ISDN MODEM in * Machine ( HFC - S Modem Euro 30 - 40 ) and then >used bristuff ( google for it ) , and used that , it just works! . I >can send you my configs if you need som ehlp I gladly look at your config files to see what I have done wrong. At the moment the setup is hooked up that I can make inbound and outbound calls (from ISDN and SIP clients), just with the limitations as below : 1. On outbound calls, I get the normal rining call progress tone althought the the other party has not even been reached. This then changes from normal ringing suddenly to busy when the other party is sending a busy signal. I'd rather have the call progress send a busy signal right away. 2. Internal calls between two ISDN client phones on the S-bus is not possible. The phone rings but the call is dropped as soon as it is answered. Kind Regards, Ben ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to launch asterisk and connect to console. ?????
run safe_asterisk and then asterisk -rc (add v's to your liking). You probably weren't able to connect to remote asterisk because none was running. safe_asterisk is a script which re-starts asterisk in the event that it segfaults, dies, or otherwise implodes.. Greg On Mon, 19 Jul 2004, James wrote: > Any ideas? > > Thanks. > > > > [EMAIL PROTECTED] root]# asterisk -r > Unable to connect to remote asterisk > [EMAIL PROTECTED] root]# asterisk -vgcd > Parsing /etc/asterisk/asterisk.conf > Asterisk 0.7.0, Copyright (C) 1999-2001 Linux Support Services, Inc. > Written by Mark Spencer <[EMAIL PROTECTED]> [snip] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: quadbri NT_mode S-Bus Problem
>Are you using Santis Siemens ISDN NT1 ? If yes , we have the same from >Siemens Switzerland , What I've done is to get one cable from ISDN NT >-- > ISDN MODEM in * Machine ( HFC - S Modem Euro 30 - 40 ) and then >used bristuff ( google for it ) , and used that , it just works! . I >can send you my configs if you need som ehlp I gladly look at your config files to see what I have done wrong. At the moment the setup is hooked up that I can make inbound and outbound calls (from ISDN and SIP clients), just with the limitations as below : 1. On outbound calls, I get the normal rining call progress tone althought the the other party has not even been reached. This then changes from normal ringing suddenly to busy when the other party is sending a busy signal. I'd rather have the call progress send a busy signal right away. 2. Internal calls between two ISDN client phones on the S-bus is not possible. The phone rings but the call is dropped as soon as it is answered. Kind Regards, Ben ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] isdn cli
> hi! > > I need to pass the CLI for my outgoing ISDN PRI call from * box. > > here's the ISDN protocol debug. > > Q.931 > Calling Number (len=10) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) > Presentation: Presentation permitted, user > number passed network screening (1) '123123' ] > > Called Number (len=12) [ Ext: 1 TON: Unknown Number Type (0) NPI: > Unknown Number Plan (0) '818818' ] > Sending Complete (len= 0) > > But the CLI is not seen at the end mobile. Instead a fix number is seen. Is this a problem with * ISDN > driivers ? > > but123 No, I cannot see a problem here... Welcome to the PSTN world. Please read the past messages of this list. You cannot send some number as you wish. It must be according with your DID, and in the range specified for your provider. Also, you provider must give you "ANI for internal number" or "AMA for internal number". Regards, Gus P.D: I'm tired of this: "TON: Unknown". It must be local. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] isdn cli
hi! I need to pass the CLI for my outgoing ISDN PRI call from * box. here's the ISDN protocol debug. Q.931 > Calling Number (len=10) [ Ext: 0 TON: Unknown Number Type (0) NPI:Unknown Number Plan (0)> Presentation: Presentation permitted, usernumber passed network screening (1) '123123' ]> Called Number (len=12) [ Ext: 1 TON: Unknown Number Type (0) NPI:Unknown Number Plan (0) '818818' ] Sending Complete (len= 0) But the CLI is not seen at the end mobile. Instead a fix number is seen. Is this a problem with * ISDN driivers ? but123
Re: [Asterisk-Users] Echo on a PRI
> I am getting an occasional noticeable echo on some of the phone lines > (random inbound and outbound). Everyone I ask keeps telling me that I > can't be having echo since I am on a PRI, which is a digital circuit. > Ok, so I can't be having echo, but I am! > Does anyone have any ideas of what might be > causing the echo in this situation? There are many far more knowledgeable than I about echo but in the interim I will contribute from my limited understanding. While echo will not be generated at the interface between your Asterisk server and the PRI trunk, echo still can be generated elsewhere in the network (i.e. at the other end of the call where your "digital" call is converted into analog to interface with the two copper wires running to the POTS phone). Normally end-to-end latency for local calls is so short that the echo appears (if it appears at all) as reverb rather than echo. The problem is that if you are running VoIP calls through your Asterisk box you can introduce enough latency so that the echo already present in the system becomes noticeable. Hence the need to enable echo cancelling. Read the wiki and play with the parameters and #defines in the echo can code. Good luck! George Pajari netVOICE communications ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem Starting RC1
Remove res_parking.so from /usr/lib/asterisk/modules bkw - Original Message - From: "Nathan Martinez" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, July 19, 2004 7:53 PM Subject: [Asterisk-Users] Problem Starting RC1 > Hello, > > I was running a very simple test setup with * HEAD 7/15/2004 on Fedora > Core 2 and things were working fine. Today I upgraded to RC1 and my > asterisk service will no longer start. I downloaded the tarball, > extracted, ran 'make', ran 'service asterisk stop', ran 'make install', > removed all files in /etc/asterisk, ran 'make samples' and then ran > 'service asterisk start'. > > > I get the following errors logged to /var/log/asterisk/messages each > time I try to start: > > Jul 19 17:32:26 WARNING[1076227072]: Command 'showparkedcalls' already > registered (or something close enough) > Jul 19 17:32:26 WARNING[1076227072]: Already have an application > 'ParkedCall' > Jul 19 17:32:26 WARNING[1076227072]: res_parking.so: load_module failed, > returning -1 > Jul 19 17:32:26 WARNING[1076227072]: Soft unload failed, > 'res_parking.so' has use count 1 > Jul 19 17:32:26 WARNING[1076227072]: Loading module res_parking.so > failed! > > > Any ideas would be great. > > Thank you, > Nathan > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] quadbri NT_mode S-Bus Problem
Are you using Santis Siemens ISDN NT1 ? If yes , we have the same from
Siemens Switzerland , What I've done is to get one cable from ISDN NT
-- > ISDN MODEM in * Machine ( HFC - S Modem Euro 30 - 40 ) and then
used bristuff ( google for it ) , and used that , it just works! . I
can send you my configs if you need som ehlp
regards
~uppal
On Sun, 18 Jul 2004 23:47:47 +0200, Ben Bosshardt
<[EMAIL PROTECTED]> wrote:
> >What type is your "ISDN house telephone system"?
> >Without more specific information all we can do is guess...
>
> Our system is a just the basic subscription to SWISSCOM, which is the main
> phone company in Switzerland. We have BRI with 2 Channels which can be used
> simulaniously and a Siemens NT that has only the function of feeding our
> S-bus with 4 phones connected.
>
> >For a sollution to 1 ... drop the "r" option of dial...
> >exten => _X.,1,Dial(Zap/g1/${EXTEN})
>
> I will give it a try.
>
> >You might need pridialplan/prilocaldialplan set to local for local
> >calls... or both to unknown... just experiment with those values.
>
> I am still looking for any documentation regarding the use of
> pridialplan/prilocaldialplan. I don't know how to find out what SWISSCOM
> requires.
>
> Thanks for your help.
> Ben
>
> ___
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>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] Numbering Plan and Siemens EWSD
> Hi all, > > We're trying to hook up our Asterisk config (Card: TE410P) with a > Siemens EWSD switch. The link is ok on both ends (green), with no errors. > > The problem is when we try to make a call from our side (via call > files), we get the pri/E1 error >Ext: 1 Cause: Temporary failure (41), class = Network Congestion (2) > > Our Telecom partner (they checked with Siemens) mentioned that we need > to configure a dialplan as > > " numbering plan (Rec. E.164) > The stands for ISDN (Telephony), ISDN (Speech), etc" > > This is what they told us, but the closest we can configure in Asterisk > is the pridialplan (unknown, private, local, national, international). > > We tried all of them, with no difference. > We also tried them with callerid set, no advance. > > > Anyone familiar with this other dialplan, or with the integration of > Asterisk/E1 with a Siemens EWSD switch. > > > pri debug log of the call below (this was with pridialplan set to 'unknown') > and without callerid. > > -- Making new call for cr 32780 > > Protocol Discriminator: Q.931 (8) len=32 > > Call Ref: len= 2 (reference 12/0xC) (Originator) > > Message type: SETUP (5) > > Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) > > Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) > > Ext: 1 User information layer 1: A-Law (35) > > Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: > 0 As Bruno said, check that you are using euroisdn. If you are not using ISDN equipment to dial thru pri, SETUP message is wrong. And please... change you pridial to local. > > Called Number (len=12) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) 'x' ] Anyway, can you send a SIGNTRAC, or maybe a LTGTRAC (better to view more deeply)? Regards, Gus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem Starting RC1
>I was running a very simple test setup with * HEAD 7/15/2004 on Fedora >Core 2 and things were working fine. Today I upgraded to RC1 and my >asterisk service will no longer start. I downloaded the tarball, >extracted, ran 'make', ran 'service asterisk stop', ran 'make install', try make clean install sound like modules (.so from previous install) are getting loaded ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem Starting RC1
This worked great! Thank you, Nathan -Original Message- From: Bruce Komito [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 5:58 PM To: Nathan Martinez Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Problem Starting RC1 I had the same problem. Before you "make install" from the asterisk directory, try removing all the files in /usr/lib/asterisk/modules . That should resolve any potential conflicts from stuff left over from the last build. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Mon, 19 Jul 2004, Nathan Martinez wrote: > Hello, > > I was running a very simple test setup with * HEAD 7/15/2004 on Fedora > Core 2 and things were working fine. Today I upgraded to RC1 and my > asterisk service will no longer start. I downloaded the tarball, > extracted, ran 'make', ran 'service asterisk stop', ran 'make > install', removed all files in /etc/asterisk, ran 'make samples' and > then ran 'service asterisk start'. > > > I get the following errors logged to /var/log/asterisk/messages each > time I try to start: > > Jul 19 17:32:26 WARNING[1076227072]: Command 'showparkedcalls' already > registered (or something close enough) Jul 19 17:32:26 > WARNING[1076227072]: Already have an application 'ParkedCall' > Jul 19 17:32:26 WARNING[1076227072]: res_parking.so: load_module > failed, returning -1 Jul 19 17:32:26 WARNING[1076227072]: Soft unload > failed, 'res_parking.so' has use count 1 Jul 19 17:32:26 > WARNING[1076227072]: Loading module res_parking.so failed! > > > Any ideas would be great. > > Thank you, > Nathan > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo on a PRI
Might as well come join the * SIG [EMAIL PROTECTED] bare your sole there ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem Starting RC1
Sounds like you'd need to delete res_parking.so since that was replaced with res_features.so if I've followed the mailing list correctly.. - andrewg On Mon, Jul 19, 2004 at 05:53:08PM -0700, Nathan Martinez wrote: > Hello, > > I was running a very simple test setup with * HEAD 7/15/2004 on Fedora > Core 2 and things were working fine. Today I upgraded to RC1 and my > asterisk service will no longer start. I downloaded the tarball, > extracted, ran 'make', ran 'service asterisk stop', ran 'make install', > removed all files in /etc/asterisk, ran 'make samples' and then ran > 'service asterisk start'. > > > I get the following errors logged to /var/log/asterisk/messages each > time I try to start: > > Jul 19 17:32:26 WARNING[1076227072]: Command 'showparkedcalls' already > registered (or something close enough) > Jul 19 17:32:26 WARNING[1076227072]: Already have an application > 'ParkedCall' > Jul 19 17:32:26 WARNING[1076227072]: res_parking.so: load_module failed, > returning -1 > Jul 19 17:32:26 WARNING[1076227072]: Soft unload failed, > 'res_parking.so' has use count 1 > Jul 19 17:32:26 WARNING[1076227072]: Loading module res_parking.so > failed! > > > Any ideas would be great. > > Thank you, > Nathan > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem Starting RC1
I had the same problem. Before you "make install" from the asterisk directory, try removing all the files in /usr/lib/asterisk/modules . That should resolve any potential conflicts from stuff left over from the last build. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Mon, 19 Jul 2004, Nathan Martinez wrote: > Hello, > > I was running a very simple test setup with * HEAD 7/15/2004 on Fedora > Core 2 and things were working fine. Today I upgraded to RC1 and my > asterisk service will no longer start. I downloaded the tarball, > extracted, ran 'make', ran 'service asterisk stop', ran 'make install', > removed all files in /etc/asterisk, ran 'make samples' and then ran > 'service asterisk start'. > > > I get the following errors logged to /var/log/asterisk/messages each > time I try to start: > > Jul 19 17:32:26 WARNING[1076227072]: Command 'showparkedcalls' already > registered (or something close enough) > Jul 19 17:32:26 WARNING[1076227072]: Already have an application > 'ParkedCall' > Jul 19 17:32:26 WARNING[1076227072]: res_parking.so: load_module failed, > returning -1 > Jul 19 17:32:26 WARNING[1076227072]: Soft unload failed, > 'res_parking.so' has use count 1 > Jul 19 17:32:26 WARNING[1076227072]: Loading module res_parking.so > failed! > > > Any ideas would be great. > > Thank you, > Nathan > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Phone recommendation
Hello Yiannis, I have an ipDialog SipTone II sitting right beside me. Overall it is an excellent phone but lacks codecs. It only has ulaw, alaw, and g729. The speakerphone is adequate for most things, call transferring works, holding, volume controller, conferencing, 2 lines, it pretty much all works. The interesting thing about the phone though is that it runs Linux. Thanks to ipDialog sending me the firmware I have been able to modify it slightly to get a telnet prompt available. I can't release the firmware though, who knows what trouble I could get into... but below is a snippet of info. Oh, be on the watch... I may end up selling the phone when my Ciscos come. - Joshua Colp. /proc> cat version Linux version 2.4.10-uc2 ([EMAIL PROTECTED]) (gcc version 2.95.3 20010315 (release)) #1 Fri Mar 21 12:39:17 PST 2003 /proc> cat cpuinfo Processor : STMicro STLC1502 rev 0 (v3l) BogoMIPS: 6.55 Hardware: STMicro STLC1502 Revision: Serial : > On Monday 19 July 2004 12:04 pm, Yiannis Costopoulos wrote: > Hi, > > I am looking for some affordable IP Phones. Any experiences with the > SipToneII by ipDialog? > > What about soft phones? Any recommendations there (for Windoze and > Linux)? > > Thanks, > Yiannis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem Starting RC1
Hello, I was running a very simple test setup with * HEAD 7/15/2004 on Fedora Core 2 and things were working fine. Today I upgraded to RC1 and my asterisk service will no longer start. I downloaded the tarball, extracted, ran 'make', ran 'service asterisk stop', ran 'make install', removed all files in /etc/asterisk, ran 'make samples' and then ran 'service asterisk start'. I get the following errors logged to /var/log/asterisk/messages each time I try to start: Jul 19 17:32:26 WARNING[1076227072]: Command 'showparkedcalls' already registered (or something close enough) Jul 19 17:32:26 WARNING[1076227072]: Already have an application 'ParkedCall' Jul 19 17:32:26 WARNING[1076227072]: res_parking.so: load_module failed, returning -1 Jul 19 17:32:26 WARNING[1076227072]: Soft unload failed, 'res_parking.so' has use count 1 Jul 19 17:32:26 WARNING[1076227072]: Loading module res_parking.so failed! Any ideas would be great. Thank you, Nathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs - Advantages
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 19 July 2004 01:43 pm, [EMAIL PROTECTED] wrote: > Hi, > I'm planning to use a Asterisk with Digium E1 cards, I understand that > using a codec such as G.729 can be very CPU demanding. What are the real > advantages of using a codec such as G.729 ? Bandwidth only ? Using no > compression wouldn't increase the scalability of my asterisk PBX ? This is > considering I have no bandwidth issues in my network. > > Thanks Actually besides from the best sound quality it's also not heavy on the CPU or bandwidth. It's actually better than any of the other. - -- Steve "They that would give up essential liberty for temporary safety deserve neither liberty nor safety." Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFA/GxjljK16xgETzkRAqf7AJ92M97CYwKYTYAOM843xafpl5pD4gCeJCPp Hwt5G6/tnw9Eq6T0+/2vCj0= =OcuB -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Phone recommendation
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 19 July 2004 12:04 pm, Yiannis Costopoulos wrote: > Hi, > > I am looking for some affordable IP Phones. Any experiences with the > SipToneII by ipDialog? > > What about soft phones? Any recommendations there (for Windoze and Linux)? > > Thanks, > Yiannis One very nice phone is the Snom 200. Pricing is in the mid to high $200 and worth every penny. This phone is built for flexibility. I can be a five line phone, actually seven. Works well with * and is running Linux. You can have handset, speakerphone, special headset ala $135 and a normal $30 computer headset. At the same time! To be a bit nit picky: it does have an unusual cradle. It does not "fall" or "slip" into it, but it works OK. If you then get a stranded patch cable and apply power over ethernet (PoE) you got a very nice combo! - -- Steve "They that would give up essential liberty for temporary safety deserve neither liberty nor safety." Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFA/GtxljK16xgETzkRAqiPAJ4rp5kVkCMgvdIaQ/cFFJl+Gt0hKACg2TMT WL6bq0h+fEUqSVYMP+BFqDA= =6McY -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hospitality Industry
Anyone connected Asterisk to hospitality packages such as: Micros Fidelio Visual One Jonas We'd be interested in providing bounty on providing a connection to one or more (depending upon what the client selects) if our proposal goes through. Ultimately, about 300 to 600 stations will be provided. Ray. - This mail sent through IMP: http://horde.org/imp/ -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP to H323 call timeout
Fred Lee a écrit :
Hi all,
I have the following setup:
UAs SER -- ASTERISK --GNUGK - GWs
SER is configured to route call requests from UAs to Asterisk.
Asterisk is configured to receive the call on SIP channel and dial out
to GNUGK over H323 channel. The problem I'm facing is that asterisk
sends out the call request to GNUGK and times out immediately, so call
setup is never completed. On GNUGK the call request comes in followed
by a normal call drop.
Any ideas on what could be the problem ??
Do you use the h323 - Nufone? Is it a recent installation? If so, could
be the problem that GW need FastStart and the * h323 don't send it.
My asterisk configuration, debug and console output are as follow :
SIP.CONF
==
[general]
port = 5080
bindaddr = 10.10.1.170
context = to_GNUGK
disallow=all
allow=g729
H323.CONF
===
[general]
port = 1720
allow = g729
gatekeeper = 64.80.103.12
allowgkrouted = yes
context = to_SER
EXTENSIONS.CONF
[general]
static = yes
writeprotect = yes
[to_GNUGK]]
exten => _.,1,Dial(h323/[EMAIL PROTECTED]:1720,60,C)
[to_SER]
exten => _.,1,Dial(SIP/[EMAIL PROTECTED]:5060,60)
DEBUG File
==
Jul 15 16:14:10 DEBUG[65541]: Check for res for
Jul 15 16:14:10 DEBUG[65541]: is not a local user
Jul 15 16:14:10 DEBUG[65541]: build_route: Record-Route hop:
Jul 15 16:14:10 DEBUG[65541]: build_route: Contact hop:
Jul 15 16:14:10 DEBUG[311316]: SIMPLE DIAL (NO URL)
Jul 15 16:14:10 DEBUG[311316]: type=h323, format=256,
[EMAIL PROTECTED]:1720.
Jul 15 16:14:10 DEBUG[311316]: Host: 10.10.1.12:1720 Username:
15613021234
Jul 15 16:14:10 DEBUG[311316]: [EMAIL PROTECTED]:1720,
timeout=0.
Jul 15 16:14:13 DEBUG[213006]: Cleaning up our mess
Jul 15 16:14:23 DEBUG[311316]: SIMPLE DIAL (NO URL)
Jul 15 16:14:23 DEBUG[311316]: type=h323, format=256,
[EMAIL PROTECTED]:1720.
Jul 15 16:14:23 DEBUG[311316]: Host: 10.10.1.12:1720 Username: t
Jul 15 16:14:23 DEBUG[311316]: [EMAIL PROTECTED]:1720, timeout=0.
Jul 15 16:14:24 DEBUG[213006]: Cleaning up our mess
Jul 15 16:14:31 DEBUG[311316]: SIMPLE DIAL (NO URL)
Jul 15 16:14:31 DEBUG[311316]: type=h323, format=256,
[EMAIL PROTECTED]:1720.
Jul 15 16:14:31 DEBUG[311316]: Host: 10.10.1.12:1720 Username: h
Jul 15 16:14:31 DEBUG[311316]: find_user() - decrement inUse counter
Jul 15 16:14:31 DEBUG[311316]: is not a local user
Jul 15 16:14:31 DEBUG[65541]: Stopping retransmission on
'[EMAIL PROTECTED]' of Response 1: Found
CONSOLE Output
==
*CLI> -- Executing Dial("SIP/-08121388",
"h323/[EMAIL PROTECTED]:1720|60|C") in new stack
-- Called [EMAIL PROTECTED]:1720
== No one is available to answer at this time
-- Timeout on SIP/-08121388
== CDR updated on SIP/-08121388
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RE: [Asterisk-Users] Echo on a PRI
I think I saw a reference to a similar problem and it regarded IRQ issues on the machine in question. IF there was IRQ sharing, cagey things happened. But if the T1 card had a static IRQ, it resolved the issue. Does your T1 card have a dedicated IRQ? I am sure someone will be able to explain further and possibly give you some validation on your Mobo too? Thanks, Wiley -Original Message- From: David Goldfein [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 5:13 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Echo on a PRI Hi, I recently set up the following in a production system (2.8 GHZ Xeon, 1 Gig Memory, Dell 2650). Telco - PRI - Asterisk - T1 - PBX I am getting an occasional noticeable echo on some of the phone lines (random inbound and outbound). Everyone I ask keeps telling me that I can't be having echo since I am on a PRI, which is a digital circuit. Ok, so I can't be having echo, but I am! Does anyone have any ideas of what might be causing the echo in this situation? Thanks, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Suscription
Name: Carlos Clemares ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo on a PRI
Hi, I recently set up the following in a production system (2.8 GHZ Xeon, 1 Gig Memory, Dell 2650). Telco - PRI - Asterisk - T1 - PBX I am getting an occasional noticeable echo on some of the phone lines (random inbound and outbound). Everyone I ask keeps telling me that I can't be having echo since I am on a PRI, which is a digital circuit. Ok, so I can't be having echo, but I am! Does anyone have any ideas of what might be causing the echo in this situation? Thanks, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 500 Voicemail
On 04:28 PM 7/19/2004, Wiley E. Siler wrote: >Mine does the same. Once in Message center I can choose selection >1.Message Center and then soft key Select.Then I select the >registered line that I want to check voice mail on. That is no less than >4 key strokes just to get into your voice mail. Not many to me but tons >to an unskilled user. However, in the documentation regarding the >bypassInstantMessage value, supposedly, setting bypassInstantMessage to >1 is supposed to allow you to go right into voice mail without >navigating the Message Center. That is the big question on my mind at >this point. I have yet to get this to work and I also don't think I am >receiving any SIMPLE messages ti show me that I have messages waiting. > >Do you get a message waiting indicator? > I do get MWI, there are a few things you need to set, and I forget what off the top of my head, soon as I can look and post it here. I haven't tried the bypassInstantMessage value, but I'll take a look and see if I can get it to work. -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 500 Voicemail
Thank you so much! That was exactly what I needed to know! Cheersm Wiley -Original Message- From: Tor Roberts [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 3:35 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Polycom IP 500 Voicemail Wiley, I don't have any 500s, but I use 600s, which use the same file I think. Here is my digitmap: What this says is that if I dial 9, then a 7 digit local number, I don't need to hit send. If I dial 91, then 10 digit long distance number, I don't need to hit send. If I dial extension 85 plus any 2 digits ex., 8523, I don't need to hit send. If I dial extension 5,6, or 7 plus 2 digits, ex. 635, I don't need to hit send. And if I dial 9411, or 9911 (info or emergency) I don't need to hit send. Hope this helps. -Tor Wiley E. Siler wrote: >I read the administrator document repeatedly. I have not been able to >find a wiki that applied to digitmap feature at all and I have searched >repeatedly and read several of the wikis regarding Polycoms. The >administrators guide doesn't have enough context explanation to make the >use of the digitmap understandable. > >That is the basis of my request for a digitmap explanation. I am not >asking someone to write mine for me. I am asking to see an example and >an explanation that gives context so I can write my own and know I have >done it properly. My PBX is Asterisk and the setup is about as generic >as generic can be. Polycoms over SIP to the PBX. > >If you know where the wiki is for digitmaps please send it. If you feel >inspired, a short explanation of the relevance and context of digitmaps >would be greatly appreciated. I know everyone has to take their own >time to answer these emails and I truly appreciate that. That is why I >do my research until I hit a wall, then I will ask here. I appreciate >whatever you can spare time for. > >Thanks! >Wiley > > > >-Original Message- >From: Brent Franks [mailto:[EMAIL PROTECTED] >Sent: Monday, July 19, 2004 10:26 AM >To: [EMAIL PROTECTED] >Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail > > > >>Thank you! >> >>Can you tell me more about the dial plan feature? How do you setup >> >> >the > > >>correct digitmap? >> >> >> > >Check the Administrator's Document. You can find it on the Wiki, under >IP Phones.. Polycom. Did you try to look up the digitmap feature before >sending this post? If not, you should be able to understand it when you >read it, it's relatively straight forward. > >No one can setup a correct digitmap for you, as it will vary greatly on >how you have setup your PBX. > >- Brent > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE:RE: [Asterisk-Users] Codecs - Advantages
Is this bascially setting your bandwith value = high inside of iax.conf? Or is there another place to designate the codec? Thanks, Wiley -Original Message- From: Senad Jordanovic [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 2:11 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RE:RE: [Asterisk-Users] Codecs - Advantages Yes, I don't have any bandwidth issues, so I could use 2 Mbps for 30 calls. I do have issues with processing CPU capacity. Is g711 CPU intensive as g729 ? I understand g729 is very CPU intensive. >>>... Forgive me, but what you just wrote tells you EXACTLY what you should use! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MWI - Config Stupidity or Notify Issues?
I am having a problem with the message waiting indicator. We are currently using the ast_data modules for both our sip configuration and our voicemail configuration. In the mailbox field I have tried using both [EMAIL PROTECTED] and simply mailboxnumber. Yet so far I am still not getting a MWI on my 7905's or on my 7960's. My assumption would be that I am still missing something, but at this point I can't figure it out. I have recently seen a message that Notify is not working properly with CVS HEAD. Thanks for you help in advance. Robert Jackson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] uip200 clips audio prompts
On July 19, 2004 10:19 pm, [EMAIL PROTECTED] wrote: > On Mon, 19 Jul 2004, Ryan Courtnage wrote: > > > This happens with my 7940s as well. I have found that using and Answer, > > > and a Wait(1) before playing back prompts works well. Prevents Alisson > > > from saying "Assword?" when dialing VoicemailMail(20). > > > > Thanks for your reply. I have been able to use this method to eliminate > > some of the problems, but from within the voicemail application, I don't > > beleive there is a way to set a delay between each prompt? > > > > ie: I'll hear: "Press 0 for New messages, ... for old messages, ... for > > work message ". The "Press x.." is cut off of the beginning of the > > prompts. > > > > I only see this problem with uip200s. BT102s, handytones, sipuras, etc > > work just fine. > > Could it be silence supression? Perhaps. If the phone does support silence suppression, it isn't advertised - and neither are the config parameters needed to adjust it / turn it off. I'll check with Uniden. Thanks Ran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 500 Voicemail
Mine does the same. Once in Message center I can choose selection 1.Message Center and then soft key Select.Then I select the registered line that I want to check voice mail on. That is no less than 4 key strokes just to get into your voice mail. Not many to me but tons to an unskilled user. However, in the documentation regarding the bypassInstantMessage value, supposedly, setting bypassInstantMessage to 1 is supposed to allow you to go right into voice mail without navigating the Message Center. That is the big question on my mind at this point. I have yet to get this to work and I also don't think I am receiving any SIMPLE messages ti show me that I have messages waiting. Do you get a message waiting indicator? W -Original Message- From: Chris A. Icide [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 3:03 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail On 12:40 PM 7/19/2004, Wiley E. Siler wrote: >My Polycom is on loan as a demo and I assume it is one of the first >revision models. In fact it shows as Rev A on the back of the phone. > >I have all the same buttons you listed save for the Messages button. >The 3rd from the bottom on the right column of buttons sayd Voice Mail >on my version. That corresponds to the location of your button that >says Messages. I assume this was changed by Polycom since their phone >has other messaging capability (isntant message for instance) and it was >easier to use Messages and unify the meaning instead of Voice Mail and >lock it into one type of messaging. > >Does your Messages button dump you right into voice mail or do you have >to navigate a menu first? > >Thanks, >Wiley My messages button dumps me right to message center, which I then have to use soft buttons. My IP500 is Rev. C -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration
Date: Mon, 19 Jul 2004 14:54:44 -0500 From: "Christopher L. Wade" <[EMAIL PROTECTED]> Organization: Unistar-Sparco Computers, Inc. To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration Reply-To: [EMAIL PROTECTED] Would the TLI(2)-U10 ETU work as well? That is a 2 port analog tie line card, I don't think that Digium has a card that can be set up as an analog 4W E&M trunk. bad idea anyway, the t-1 will be a much better interface and if you ever press the eject on the IPK you could use the t-1 as a PSTN interface. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Cisco 7960 SIP V6 and distinctive ring.
On Mon, Jul 19, 2004 at 02:09:34PM -0700, asteriskstuff @ ziplip. com wrote: > Thanks..it's a numeric value!! in the wiki it refers to a text field!! The wiki is also correct... I have: exten => 101,1,SetVar(ALERT_INFO=Bellcore-dr1) And that works fine. What was the error message you were getting? -- -- Sam Tilders [EMAIL PROTECTED] (Move to Jupiter) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk
Thanks Wayne. P > -Original Message- > From: Wayne [mailto:[EMAIL PROTECTED] > Sent: Monday, July 19, 2004, 3:48 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk > > Hiya! > Looks like you have the same problem as I had... found the answer by > doing a 'debug sip-messages' by telnet'ing into one of my cisco phones... > > The short answer is 'its your "callerid=" line' > you need to remove the quotes around the text part. The cisco's cant > handle it. > eg > where you have for [phone1] in your Sip.conf > callerid="Lounge1" <1> > > what you should have is > callerid=Lounge1 <1> > > etc... > > Threw me for a while but the debug options on the cisco's helped out > there... I think the docs read like you should have the text in quotes - > but as I said - my cisco's didnt like it :) > > anyways - hope this helps :) > Wayne! > > > > > > [EMAIL PROTECTED] wrote: > > >Hi Sean > > > >Both phones are set for context=sip in the sip.conf file. > > > >As I say the phones will both call out OK (I can dial the 500 test number and > successfully connect to the remote PBX through my firewall). It's just that > when I'm trying to call from phone to phone I'm getting the 404 not found > error in the asteris verbose dialog. > > > >If anyone has a documented example of their 7960 config sipdefault.cnf and > sipxipadd.cnf files together with their sip.conf and extensions.conf files > I could have to test directly on my system I'd be appreciative to test them on > my system. > > > >While the WiKi's are very useful as example files it would be great (and I > may do it myself!!) if there was an up to date example file with all the > options for each filed and a verbose description for the rational behind it > (although I recognise that this is an 'in development' product and therefore > the docs have to be done at the end!!). > > > >Part of the problem is there are so many dependencies that can affect the > system including how the dhpcd server serves IP address's and associated files > (for example the files have to be structured in a particular order on the > tftpd server for the cisco's to pick them up correctly). Given this level of > dependency I'm not sure where the break could be. > > > >The one thing I have noticed from the show sip peers field is that it's > showing the phones as having a netmask of 255.255.255.255 although they're > actually configyred for 255.255.255.0. > > > >P > > > > > > > > > >>-Original Message- > >>From: Sean Cheesman [mailto:[EMAIL PROTECTED] > >>Sent: Sunday, July 18, 2004, 11:37 AM > >>To: [EMAIL PROTECTED] > >>Subject: RE: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk > >> > >>It doesn't look like you have a context set for phone1. Try putting > >>context=sip in the phone1 section like you have in phone2. That'll put > >>both in the same context of your extensions.conf file and should allow > >>interaction between the two. > >> > >>-Original Message- > >>From: [EMAIL PROTECTED] > >>[mailto:[EMAIL PROTECTED] On Behalf Of > >>[EMAIL PROTECTED] > >>Sent: Sunday, July 18, 2004 7:13 AM > >>To: [EMAIL PROTECTED] > >>Subject: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk > >> > >> > >>Hi All > >> > >>Total noob on the list so all help appreciated > >> > >>I've successfully installed Asterisk on an IBM A30P Thinkpad using > >>fedora Core 2 (I'm looking at having a mobile PBX for conferences and > >>shows). > >> > >>I've plugged in two Cisco 7960 phones > >> > >>The phones register with the Asterisk correctly and I can run the demo's > >>and even the AIX demo through to digium works correctly... > >> > >>but I cannot get the phones to dial each other :( > >> > >>Initially I was getting a "extension not found in local" message (when > >>dialling from console...from phone just engaged (busy) tone. > >> > >>when I add extension from console I now get a "not found 404" > >>messageI see that there was an earlier thread on the list that > >>discussed removing the proxy forwarding from the phone settings and I've > >>tried that from SIPDefault.cnf but it doesn't fix the problem. > >> > >>I've obviously missed something but am too inexperienced to spot it. P > >> > >>my files are as follows:- > >> > >> > >> > >>sipxx.cnf > >> > >> > >># Lounge Phone Settings > >> > >># Line 1 Settings > >>line1_name: "11"; Line 1 Extension\User ID > >>line1_displayname: "Lounge1"; Line 1 Display Name > >>line1_authname: "lounge11" ; Line 1 Registration Authentication > >>line1_password: "lounge"; Line 1 Registration Password > >> > >>- > >> > >>sipdefault.cnf > >> > >># Image Version > >> > >>image_version: P0S3-06-3-00 > >> > >># Proxy Server > >> > >>proxy1_address: ipaddress of A30P ; Can be dotted IP or FQDN > >> > >>proxy1_port: > >>5060 > >># Proxy Registration (0-disabl
Re: [Asterisk-Users] Polycom IP 500 Voicemail
Wiley, I don't have any 500s, but I use 600s, which use the same file I think. Here is my digitmap: What this says is that if I dial 9, then a 7 digit local number, I don't need to hit send. If I dial 91, then 10 digit long distance number, I don't need to hit send. If I dial extension 85 plus any 2 digits ex., 8523, I don't need to hit send. If I dial extension 5,6, or 7 plus 2 digits, ex. 635, I don't need to hit send. And if I dial 9411, or 9911 (info or emergency) I don't need to hit send. Hope this helps. -Tor Wiley E. Siler wrote: I read the administrator document repeatedly. I have not been able to find a wiki that applied to digitmap feature at all and I have searched repeatedly and read several of the wikis regarding Polycoms. The administrators guide doesn't have enough context explanation to make the use of the digitmap understandable. That is the basis of my request for a digitmap explanation. I am not asking someone to write mine for me. I am asking to see an example and an explanation that gives context so I can write my own and know I have done it properly. My PBX is Asterisk and the setup is about as generic as generic can be. Polycoms over SIP to the PBX. If you know where the wiki is for digitmaps please send it. If you feel inspired, a short explanation of the relevance and context of digitmaps would be greatly appreciated. I know everyone has to take their own time to answer these emails and I truly appreciate that. That is why I do my research until I hit a wall, then I will ask here. I appreciate whatever you can spare time for. Thanks! Wiley -Original Message- From: Brent Franks [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 10:26 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail Thank you! Can you tell me more about the dial plan feature? How do you setup the correct digitmap? Check the Administrator's Document. You can find it on the Wiki, under IP Phones.. Polycom. Did you try to look up the digitmap feature before sending this post? If not, you should be able to understand it when you read it, it's relatively straight forward. No one can setup a correct digitmap for you, as it will vary greatly on how you have setup your PBX. - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs - Advantages
Hmmm, remember though that G.711u/a sends 64Kbit Frames, however it is not actually 64Kbit/call... We're not doing Circuit-switching here, we're doing packet switching. If you figure on IP overhead as well as the RTP information and of course SIP messages, then you add the 64Kbit of G.711u/a Payload you get around 80Kbit/sec Per user... that means on a Full T1 you only get 19 simultaneous calls, not 24... and on an E1 you would get around 25 simultaneous calls, not 32... And that's assuming that you have a very good SLA (or european equivalent) and that your latency is VERY low... I hope to god you use QoS on your router or your calls will sound like absolute crap... It's totally possible, don't be scared off by what I said, I'm even using * on a cable connection at home and it works just fine, you just need to do some preparation, it's a big change especially for the end users of the system... -Chris - Original Message - From: "brian" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, July 19, 2004 11:44 AM Subject: RE: [Asterisk-Users] Codecs - Advantages > If you have the bandwidth then use ulaw :) > > bkw > > > -Original Message- > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] > > Sent: Monday, July 19, 2004 12:44 PM > > To: [EMAIL PROTECTED] > > Subject: [Asterisk-Users] Codecs - Advantages > > > > Hi, > > I'm planning to use a Asterisk with Digium E1 cards, I understand that > > using a codec such as G.729 can be very CPU demanding. What are the real > > advantages of using a codec such as G.729 ? Bandwidth only ? Using no > > compression wouldn't increase the scalability of my asterisk PBX ? This is > > considering I have no bandwidth issues in my network. > > > > Thanks > > __ > > > > Todavía no tenés tu Ciudad Internet Mail? Obtenelo ahora! - > > http://webmail.ciudad.com.ar > > > > Descargá Gratis el nuevo Internet Explorer 6.0, el mejor software para > > actualizar tu PC. > > http://www.ciudad.com.ar/ar/servicios/ie/ > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Affordable SIP Phone - Stiil a Myth?
Hi! > Also in sip configuration , disable auto attendant , enable vad vad is not supported by Asterisk Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] uip200 clips audio prompts
On Mon, 19 Jul 2004, Ryan Courtnage wrote: > > This happens with my 7940s as well. I have found that using and Answer, > > and a Wait(1) before playing back prompts works well. Prevents Alisson > > from saying "Assword?" when dialing VoicemailMail(20). > > Thanks for your reply. I have been able to use this method to eliminate some > of the problems, but from within the voicemail application, I don't beleive > there is a way to set a delay between each prompt? > > ie: I'll hear: "Press 0 for New messages, ... for old messages, ... for work > message ". The "Press x.." is cut off of the beginning of the prompts. > > I only see this problem with uip200s. BT102s, handytones, sipuras, etc work > just fine. Could it be silence supression? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (Asterisk-Users] Affordable SIP Phone - Stiil a Myth?
Steve, Here is the config, I pulled from my server, that works with D'Link Phones: Main Menu SIP.CONF [general] port = 5060 ; Port to bind to (SIP is 5060) ;bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) bindaddr = 67.109.153.236 disallow=all ;allow=ilbc allow=gsm allow=ulaw allow=alaw allow=g729 ;allow=g723 jitterbuffer=no localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks localnet=10.0.0.0/255.0.0.0 ; Also RFC1918 localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation localnet=169.254.0.0/255.255.0.0 ;Zero conf local network context = bogon-calls ; Send SIP callers that we don't know about here [2000] type=friend ; This device takes and makes calls username=2000 ; Username on device secret=2000 ; Password for device host=dynamic ; This host is not on the same IP addr every time context=from-sip ; Inbound calls from this host go here mailbox=100 ; Activate the message waiting light if this ;dtmfmod=inband ; voicemailbox has messages in it reinvite=no canreinvite=no nat=yes qualify=4000 callerid=Mr. Mirchandani <2000> [2001] ; Duplicate of 2000, except with different auth data type=friend username=2001 secret=2001 host=dynamic context=from-sip mailbox=101 ;dtmfmod=inband nat=yes reinvite=no canreinvite=no callerid=Mr. Mandar <2001> [2002] type=friend host=dynamic callerid=William Suffill <2002> username=2002 secret=2002 context=from-sip nat=yes mailbox=2002 [2003] ; Duplicate of 2000, except with different auth data type=friend username=2003 secret=2003 host=dynamic context=from-sip mailbox=103 ;dtmfmod=inband reinvite=no canreinvite=no callerid=Mr.Seshu <2003> [2004] ; Duplicate of 2000, except with different auth data type=friend username=2004 ;secret=2004 secret=2004 host=dynamic context=from-sip mailbox=103 ;dtmfmod=inband reinvite=no canreinvite=no nat=yes callerid=D-link ID1 <2004> [2005] ; Duplicate of 2000, except with different auth data type=friend username=2005 ;secret=2005 secret=2005 host=dynamic context=from-sip mailbox=104 ;dtmfmod=inband reinvite=no canreinvite=no nat=yes callerid=D-link ID2 <2005> [2006] ; Duplicate of 2000, except with different auth data type=friend username=2006 secret=2006 host=dynamic context=from-sip mailbox=105 ;dtmfmod=inband reinvite=no canreinvite=no ;nat=yes callerid=Pacenet ID1 <2006> [2007] ; Duplicate of 2000, except with different auth data type=friend username=2007 secret=2007 host=dynamic context=from-sip mailbox=106 ;dtmfmod=inband reinvite=no canreinvite=no ;nat=yes callerid=Pacenet ID1 <2007> [2008] ; Duplicate of 2000, except with different auth data type=friend username=2008 ;secret=2008 secret=2008 host=dynamic context=from-sip mailbox=107 ;dtmfmod=inband reinvite=no canreinvite=no ;nat=yes callerid=D-link ID3 <2008> [2009] ; Duplicate of 2000, except with different auth data type=friend username=2009 secret=2009 host=dynamic context=from-sip mailbox=108 ;dtmfmod=inband reinvite=no canreinvite=no ;nat=yes callerid=D-link ID4 <2009> [2010] ; Duplicate of 2000, except with different auth data type=friend username=2010 secret=2010 host=dynamic context=from-sip mailbox=109 ;dtmfmod=inband reinvite=no canreinvite=no ;nat=yes callerid=USA-ID10<2010> [2011] ; Duplicate of 2000, except with different auth data type=friend username=2011 secret=2011 host=dynamic context=from-sip mailbox=110 ;dtmfmod=inband reinvite=no canreinvite=no ;nat=yes callerid=USA-ID11<2011> [3001] ; Duplicate of 3000, except with different auth data type=friend username=3001 secret=3001 host=dynamic context=for-dlink mailbox=109 ;dtmfmod=inband reinvite=no canreinvite=no ;nat=yes callerid=D-link Contact1 <3001> [3002] ; Duplicate of 3000, except with different auth data type=friend username=3002 secret=3002 host=dynamic context=for-dlink mailbox=110 ;dtmfmod=inband reinvite=no canreinvite=no ;nat=yes callerid=D-link Contact2 <3002> [3003] ; Duplicate of 3000, except with different auth data type=friend username=3003 secret=3003 host=dynamic context=for-dlink mailbox=111 ;dtmfmod=inband reinvite=no canreinvite=no ;nat=yes callerid=D-link Contact3 <3003> [3004] ; Duplicate of 3000, except with different auth data type=friend username=3004 secret=3004 host=dynamic context=for-dlink mailbox=112 ;dtmfmod=inband reinvite=no canreinvite=no ;nat=yes callerid=D-link Contact4 <3004> [4001] ; Duplicate of 3000, except with different auth data type=friend username=4001 secret=4001 host=dynamic context=for-NetWeb mailbox=109 ;dtmfmod=inband reinvite=no canreinvite=no ;nat=yes callerid=NetWeb1 <4001> [4002] ; Duplicate of 3000, except with different auth data type=friend username=4002 secret=4002 host=dynami
Re: [Asterisk-Users] FATAL: Module zaptel not found.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 19 July 2004 10:01 am, PBX Portela wrote: > Dear Sirs, > > I'm running an Asterisk 0.9.1 in a Fedora Core 2 box. > I installed a X100P card on my box and when i try to load modules i am > rejected. > When i type modprobe zaptel my Fedora respond : "FATAL: Module zaptel not > found." . The same uccurs when i type "modprobe wcfxo" > > May you help me. > > Thank you in advance > > Juanjo FC2 has a 2.6 kernel. So it zaptel needs to be compiled against it with a make linux26, if I recall correctly. See wiki. - -- Steve "They that would give up essential liberty for temporary safety deserve neither liberty nor safety." Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFA/EbGljK16xgETzkRAsZiAKC3e6YEx6RYgcLkSecfBtU/0ASqOwCgss6e DcLa6dxM2ZS1/qSlSEOHPfs= =7Iuk -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Adding voice mail box
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 19 July 2004 08:49 am, Eric Wieling wrote: > On Mon, 2004-07-19 at 02:19, Steve wrote: > > On Monday 19 July 2004 01:23 am, Brian K. West wrote: > > > Dont have to.. just add it to the voicemail.conf and it will auto do > > > everything for you. > > > > > > bkw > > > > Well, after having restarted * a few times, and rebooted once, I can say > > that no mailboxes were created automatically. I'm running a week old > > HEAD. > > > > Brian, what version were you running when you observed this nice feature? > > Pretty much anything in the last year. Edit voicemail.conf, issue a > "reload" and then LEAVE A VOICEMAIL. The mailbox won't actually be > created until it needs to record a message. Indeed it does. I was stuck on the idea that it had to be done manually! Thanks! - -- Steve "They that would give up essential liberty for temporary safety deserve neither liberty nor safety." Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFA/EZgljK16xgETzkRAo08AJ9k1GhgLOXMVwOv7Z7PITPTZ5EufwCaAptR YipjSMrGX+feR+ErTw5MJJQ= =cCaD -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 500 Voicemail
On 12:40 PM 7/19/2004, Wiley E. Siler wrote: >My Polycom is on loan as a demo and I assume it is one of the first >revision models. In fact it shows as Rev A on the back of the phone. > >I have all the same buttons you listed save for the Messages button. >The 3rd from the bottom on the right column of buttons sayd Voice Mail >on my version. That corresponds to the location of your button that >says Messages. I assume this was changed by Polycom since their phone >has other messaging capability (isntant message for instance) and it was >easier to use Messages and unify the meaning instead of Voice Mail and >lock it into one type of messaging. > >Does your Messages button dump you right into voice mail or do you have >to navigate a menu first? > >Thanks, >Wiley My messages button dumps me right to message center, which I then have to use soft buttons. My IP500 is Rev. C -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cheap PoE switches/injectors?
Guys Just to clarify REAL POE is 802.3af compliant, anything else and you're taking a risk with your kit!! (Cisco have something called inline power that is only 24v...but they're switching to 802.3af now). P > -Original Message- > From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] > Sent: Monday, July 19, 2004, 2:02 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Cheap PoE switches/injectors? > > Greg Hill wrote: > > > officially, a POE capable switch/etc is supposed to do a discovery routine > > to detect, when a device is plugged into it, whether that device requires > > POE. Right? And the single-port POE injectors are usually nothing more > > than two RJ45 packs with a dc power connector, right? That could be the > > difference in price there: the detection circuitry. Or am I way off? > > No, even the single-port injectors have to have that circuitry, I > believe. Otherwise power on the extra pins would be live all the time, > which could be damaging to any non-PoE equipment you plugged in there. > Also, the PoE device itself wants to negotiate its power usage with the > power supplier, and if the injector didn't respond I don't know if the > PoE device would ever come up. > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Cisco 7960 SIP V6 and distinctive ring.
Thanks..it's a numeric value!! in the wiki it refers to a text field!! P > -Original Message- > From: Brian Buhrow [mailto:[EMAIL PROTECTED] > Sent: Monday, July 19, 2004, 1:38 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Re: Cisco 7960 SIP V6 and distinctive ring. > > [Try this again...] > > Hello. Here is what my extension which uses distinctive ring on a > Cisco 7960 running V6.2 firmware looks like. Note that the distinctive > ring tones are changes in cadence, rather than changes in ringing sounds on > the 7960. Also, if you adjust the ringer volume wile the distinctive ring > is sounding, the phone will revert to the non-distinctive ring cadence. > -Brian > exten => 2135551212,1,setvar(ALERT_INFO=4) > exten => 2135551212,2,Dial(SIP/100&SIP/401&SIP/403|20|tr) > exten => 2135551212,3,Voicemail,u401 > > } Hi > } > } Can anyone with distinctive ring on their 7960's possibly post how they've > got it to work? > } > } I understand that the ALERT_INFO variable is involved but using the examples > for the variable value from the WiKi I'm just getting an error message from > the Asterisk concole. > } > } Thanks in advance. > } > } P > } > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk RC1 and advanced voice mail.
Hi, Sorry if this has already been answered but I could find the answer. Does the patch for advanced voice mail, advanced10full.tar.gz, need to be applied to Asterisk 1.0 RC1? David ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk
Hiya! Looks like you have the same problem as I had... found the answer by doing a 'debug sip-messages' by telnet'ing into one of my cisco phones... The short answer is 'its your "callerid=" line' you need to remove the quotes around the text part. The cisco's cant handle it. eg where you have for [phone1] in your Sip.conf callerid="Lounge1" <1> what you should have is callerid=Lounge1 <1> etc... Threw me for a while but the debug options on the cisco's helped out there... I think the docs read like you should have the text in quotes - but as I said - my cisco's didnt like it :) anyways - hope this helps :) Wayne! [EMAIL PROTECTED] wrote: Hi Sean Both phones are set for context=sip in the sip.conf file. As I say the phones will both call out OK (I can dial the 500 test number and successfully connect to the remote PBX through my firewall). It's just that when I'm trying to call from phone to phone I'm getting the 404 not found error in the asteris verbose dialog. If anyone has a documented example of their 7960 config sipdefault.cnf and sipxipadd.cnf files together with their sip.conf and extensions.conf files I could have to test directly on my system I'd be appreciative to test them on my system. While the WiKi's are very useful as example files it would be great (and I may do it myself!!) if there was an up to date example file with all the options for each filed and a verbose description for the rational behind it (although I recognise that this is an 'in development' product and therefore the docs have to be done at the end!!). Part of the problem is there are so many dependencies that can affect the system including how the dhpcd server serves IP address's and associated files (for example the files have to be structured in a particular order on the tftpd server for the cisco's to pick them up correctly). Given this level of dependency I'm not sure where the break could be. The one thing I have noticed from the show sip peers field is that it's showing the phones as having a netmask of 255.255.255.255 although they're actually configyred for 255.255.255.0. P -Original Message- From: Sean Cheesman [mailto:[EMAIL PROTECTED] Sent: Sunday, July 18, 2004, 11:37 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk It doesn't look like you have a context set for phone1. Try putting context=sip in the phone1 section like you have in phone2. That'll put both in the same context of your extensions.conf file and should allow interaction between the two. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, July 18, 2004 7:13 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk Hi All Total noob on the list so all help appreciated I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm looking at having a mobile PBX for conferences and shows). I've plugged in two Cisco 7960 phones The phones register with the Asterisk correctly and I can run the demo's and even the AIX demo through to digium works correctly... but I cannot get the phones to dial each other :( Initially I was getting a "extension not found in local" message (when dialling from console...from phone just engaged (busy) tone. when I add extension from console I now get a "not found 404" messageI see that there was an earlier thread on the list that discussed removing the proxy forwarding from the phone settings and I've tried that from SIPDefault.cnf but it doesn't fix the problem. I've obviously missed something but am too inexperienced to spot it. P my files are as follows:- sipxx.cnf # Lounge Phone Settings # Line 1 Settings line1_name: "11" ; Line 1 Extension\User ID line1_displayname: "Lounge1" ; Line 1 Display Name line1_authname: "lounge11"; Line 1 Registration Authentication line1_password: "lounge" ; Line 1 Registration Password - sipdefault.cnf # Image Version image_version: P0S3-06-3-00 # Proxy Server proxy1_address: ipaddress of A30P ; Can be dotted IP or FQDN proxy1_port: 5060 # Proxy Registration (0-disable (default), 1-enable) proxy_register: 0 # Phone Registration Expiration [1-3932100 sec] (Default - 3600) timer_register_expires: 3600 # Codec for media stream (g711ulaw (default), g711alaw, g729a) preferred_codec: g711ulaw # TOS bits in media stream [0-5] (Default - 5) tos_media: 5 # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: 1 # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: avt # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: 3 # SIP Timers timer_t1: 500 ; Default 500 msec timer_t2: 4000 ; Default 4 sec sip_retx: 10 ; Default 10 sip_invite_retx: 6 ; De
Re: [Asterisk-Users] Cheap PoE switches/injectors?
You're spot on!! 802.3af has all kinds of protection circuitary built in. P > -Original Message- > From: Greg Hill [mailto:[EMAIL PROTECTED] > Sent: Monday, July 19, 2004, 1:28 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Cheap PoE switches/injectors? > > On Mon, 19 Jul 2004, Kevin P. Fleming wrote: > > > Scott Laird wrote: > > > > > So $1600 for 24 ports. That's not *too* bad. HP seems to have a > > > similar model (2626-PWR) for a similar price. 3com also seems to have a > > > 24-port injector for $800. > > > > I still don't understand why I can buy single-port injectors for $20, > > but multi-port models are $30 per port and up. You'd think that having a > > single combined power supply and other bits would reduce the cost, not > > increase it. > > > officially, a POE capable switch/etc is supposed to do a discovery routine > to detect, when a device is plugged into it, whether that device requires > POE. Right? And the single-port POE injectors are usually nothing more > than two RJ45 packs with a dc power connector, right? That could be the > difference in price there: the detection circuitry. Or am I way off? > > Greg > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to launch asterisk and connect to console. ?????
Any ideas? Thanks. [EMAIL PROTECTED] root]# asterisk -rUnable to connect to remote asterisk[EMAIL PROTECTED] root]# asterisk -vgcdParsing /etc/asterisk/asterisk.confAsterisk 0.7.0, Copyright (C) 1999-2001 Linux Support Services, Inc.Written by Mark Spencer <[EMAIL PROTECTED]>=Parsing /etc/asterisk/logger.confAsterisk Event Logger Started /var/log/asterisk/event_log == Manager registered action Ping == Manager registered action Logoff == Manager registered action Hangup == Manager registered action Status == Manager registered action Redirect == Manager registered action Originate == Manager registered action MailboxStatus == Manager registered action Command == Manager registered action ExtensionState == Manager registered action AbsoluteTimeout == Manager registered action MailboxCount == RTP Allocating from port range 1 -> 2Asterisk PBX Core InitializingRegistering builtin applications: [AbsoluteTimeout] == Registered application 'AbsoluteTimeout' [Answer] == Registered application 'Answer' [BackGround] == Registered application 'BackGround' [Busy] == Registered application 'Busy' [Congestion] == Registered application 'Congestion' [DigitTimeout] == Registered application 'DigitTimeout' [Goto] == Registered application 'Goto' [GotoIf] == Registered application 'GotoIf' [GotoIfTime] == Registered application 'GotoIfTime' [Hangup] == Registered application 'Hangup' [NoOp] == Registered application 'NoOp' [Prefix] == Registered application 'Prefix' [ResetCDR] == Registered application 'ResetCDR' [ResponseTimeout] == Registered application 'ResponseTimeout' [Ringing] == Registered application 'Ringing' [SayNumber] == Registered application 'SayNumber' [SayDigits] == Registered application 'SayDigits' [SetAccount] == Registered application 'SetAccount' [SetGlobalVar] == Registered application 'SetGlobalVar' [SetLanguage] == Registered application 'SetLanguage' [SetVar] == Registered application 'SetVar' [StripMSD] == Registered application 'StripMSD' [Suffix] == Registered application 'Suffix' [Wait] == Registered application 'Wait'Asterisk Dynamic Loader Starting: [chan_modem.so] => (Generic Voice Modem Driver) == Loading modem driver chan_modem_aopen.so => (A/Open (Rockwell Chipset) ITU-2 VoiceModem Driver) [res_musiconhold.so] => (Music On Hold Resource) == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' [res_adsi.so] => (ADSI Resource) [res_features.so] => (Call Parking Resource) == Registered application 'ParkedCall'asterisk: relocation error: /usr/lib/asterisk/modules/res_features.so: undefined symbol: ast_manager_register2
[Asterisk-Users] Cant compile Zaptel at all
I have been trying to compile Zaptel 1.0-RC1 that I just downloaded via tarball on my debian 3.0 system running a 2.4.26 kernel. I have all the headers, libraries and sources installed for the kernel along with the latest versions of GCC. I dont know what else to do to trouble shoot this so I have included the entire output below. Thanks a lot! -James Freire linux1:/usr/src/zaptel-1.0-RC1# make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DSTANDALONE_ZAPATA -c zaptel.c In file included from /usr/src/linux/include/linux/kernel.h:13, from zaptel.c:42: /usr/src/linux/include/linux/types.h:21: error: parse error before "dev_t" /usr/src/linux/include/linux/types.h:21: warning: type defaults to `int' in declaration of `dev_t' /usr/src/linux/include/linux/types.h:21: warning: data definition has no type or storage class In file included from /usr/include/asm/math_emu.h:4, from /usr/include/asm/processor.h:11, from /usr/src/linux/include/linux/prefetch.h:13, from /usr/src/linux/include/linux/list.h:6, from /usr/src/linux/include/linux/module.h:12, from zaptel.c:44: /usr/include/asm/sigcontext.h:79: error: parse error before '*' token /usr/include/asm/sigcontext.h:82: error: parse error before '}' token In file included from /usr/include/asm/processor.h:11, from /usr/src/linux/include/linux/prefetch.h:13, from /usr/src/linux/include/linux/list.h:6, from /usr/src/linux/include/linux/module.h:12, from zaptel.c:44: /usr/include/asm/math_emu.h:6: error: parse error before '*' token /usr/include/asm/math_emu.h:7: error: parse error before '*' token In file included from /usr/src/linux/include/linux/prefetch.h:13, from /usr/src/linux/include/linux/list.h:6, from /usr/src/linux/include/linux/module.h:12, from zaptel.c:44: /usr/include/asm/processor.h:421: error: parse error before '*' token /usr/include/asm/processor.h:427: error: parse error before '}' token In file included from zaptel.c:44: /usr/src/linux/include/linux/module.h:21:34: linux/modversions.h: No such file or directory In file included from /usr/src/linux/include/linux/fs.h:19, from /usr/src/linux/include/linux/capability.h:17, from /usr/src/linux/include/linux/binfmts.h:5, from /usr/src/linux/include/linux/sched.h:9, from /usr/src/linux/include/linux/mm.h:4, from /usr/src/linux/include/linux/slab.h:14, from /usr/src/linux/include/linux/proc_fs.h:5, from zaptel.c:45: /usr/src/linux/include/linux/dcache.h: In function `d_drop': /usr/src/linux/include/linux/dcache.h:149: warning: implicit declaration of function `spin_lock' /usr/src/linux/include/linux/dcache.h:152: warning: implicit declaration of function `spin_unlock' In file included from /usr/src/linux/include/linux/capability.h:17, from /usr/src/linux/include/linux/binfmts.h:5, from /usr/src/linux/include/linux/sched.h:9, from /usr/src/linux/include/linux/mm.h:4, from /usr/src/linux/include/linux/slab.h:14, from /usr/src/linux/include/linux/proc_fs.h:5, from zaptel.c:45: /usr/src/linux/include/linux/fs.h: At top level: /usr/src/linux/include/linux/fs.h:426: error: parse error before "dev_t" /usr/src/linux/include/linux/fs.h:426: warning: no semicolon at end of struct or union /usr/src/linux/include/linux/fs.h:429: error: parse error before '}' token /usr/src/linux/include/linux/fs.h:435: error: parse error before "dev_t" /usr/src/linux/include/linux/fs.h:435: warning: no semicolon at end of struct or union /usr/src/linux/include/linux/fs.h:440: error: parse error before '}' token In file included from /usr/src/linux/include/linux/reiserfs_fs_sb.h:8, from /usr/src/linux/include/linux/fs.h:731, from /usr/src/linux/include/linux/capability.h:17, from /usr/src/linux/include/linux/binfmts.h:5, from /usr/src/linux/include/linux/sched.h:9, from /usr/src/linux/include/linux/mm.h:4, from /usr/src/linux/include/linux/slab.h:14, from /usr/src/linux/include/linux/proc_fs.h:5, from zaptel.c:45: /usr/src/linux/include/linux/tqueue.h: In function `queue_task': /usr/src/linux/include/linux/tqueue.h:107: warning: implicit declaration of function `typecheck' /usr/src/linux/include/linux/tqueue.h:107: e
Re: [Asterisk-Users] uip200 clips audio prompts
On July 19, 2004 08:57 pm, [EMAIL PROTECTED] wrote: > On Mon, 19 Jul 2004, Ryan Courtnage wrote: > > Hi all, > > > > We find that our UIP200s clip off the 1st second of audio prompts from * > > (ie: the beginning of voicemail prompts). > > > > Has anyone found a way around this? > > > > Running CVS-D2004.06.29.15.30.00 on WBEL 3.0 > > This happens with my 7940s as well. I have found that using and Answer, > and a Wait(1) before playing back prompts works well. Prevents Alisson > from saying "Assword?" when dialing VoicemailMail(20). Thanks for your reply. I have been able to use this method to eliminate some of the problems, but from within the voicemail application, I don't beleive there is a way to set a delay between each prompt? ie: I'll hear: "Press 0 for New messages, ... for old messages, ... for work message ". The "Press x.." is cut off of the beginning of the prompts. I only see this problem with uip200s. BT102s, handytones, sipuras, etc work just fine. Thanks Ryan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PSTN gateway implementation?
This is an upgrade from a previous system. The old one didn't handle PRI, so they had analog phone lines as trunks. Management won't invest the money right now to get a PRI circuit. Any suggestions? > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith > Sent: Monday, July 19, 2004 4:59 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] PSTN gateway implementation? > > > -I have a TE405P board and only one T1 worth of phone lines (24) > > connected to it using an Adtran TA750 channel bank. > > Any particular reason against using PRI from your telco? > > > Is Asterisk capable of handling multiple incoming VoIP calls arriving > > from the same source (IP) or do I need to get something else to take the > > incoming traffic and pass it on to Asterisk? (I've read about using SER > > as a SIP proxy, but it's not clear to me wheather I need it or not). Can > > I use the OpenH.323 module to take care of the incoming VoIP traffic? > > Asterisk can handle multiple calls from the same IP without any worry. > Your > main worry is the lack of real billing since you're terminating to analog > PSTN instead of using PRI -- you have no way of actually knowing if the > call > was answered or not, so he'll be billed on every call. I doubt you want > to > try and work with callprogress=yes. > > -A. > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE:RE: [Asterisk-Users] Codecs - Advantages
Yes, I don't have any bandwidth issues, so I could use 2 Mbps for 30 calls. I do have issues with processing CPU capacity. Is g711 CPU intensive as g729 ? I understand g729 is very CPU intensive. >>>... Forgive me, but what you just wrote tells you EXACTLY what you should use! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Occationally SIP ext apparently is busy and goes to VM
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 This occurs every now and then even though there's no one on the phone. I sit there by the desk, and all off a sudden the VM light goes off. I pick up the phone and I can make calls. No indication on how it would be busy. I don't see any entries on the * console, other than the caller is told it's not available. It happens to a Grandstream 102 connected to a week old HEAD (of cabbage). - -- Steve "They that would give up essential liberty for temporary safety deserve neither liberty nor safety." Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFA/Dh0ljK16xgETzkRAoGGAKDLu0l4id9SpiYmNE7i8HWYtjSXMwCeJreg R3LOGNbp5A+F0jF3ZzU2WaY= =JFb6 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dropping g729 frames
I'm getting this error continuously when sending to a cisco 5300: frame.c:120 ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end The connection is highly intermittent, sometimes there's a ring, other times there is not. Is there a way to completely disable vad support in *? -g ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Affordable SIP Phone - Stiil a Myth?
Folks! This is to let all of you know that I am making D'Link make an all out effort to make D'Link Phone DPH80 and DPH100 work with Asterisk. I have provided the Asterisk Platform to D'Link's R&D Division located in Goa, India, where their IP phone's SIP Bios is undergoing modifications based on my recommendations/suggestions. I have also provided the test bed & manpower for the tests. After 2 months of sleepless nights, the phones are finally working. Some of the initial problems we found were as under: 1)These phones auto Un-register dfrom Asterisk after 30 seconds 2)No provison to give the NAT IP address for STUN Server 3)Line rings after connect to Asterisk Extension but the Phone does not pickup up the line ...etc. etc. This phone would soon be available in India(by next week or so, first branded asr "Netweb Phone") and USA in the next couple of months as "NetwebPhone" and could be priced around $65/-(tentative) See the communication from D'Link below. Seshu Kanuri - Original Message - From: Mandar Pise (Netweb India Ltd) To: Abhijit M ( Voip Dept.) Sent: Monday, July 19, 2004 1:51 PM Subject: Re: D-link ( DPH-80 ),Sip Firmware Upgrade . Dear Mr. Abhijit, As per our teleconversation with you, I am forwarding you the sip server logs attached along with this email. Thank you, Regards, Mandar Pise "Abhijit M ( Voip Dept.)" <[EMAIL PROTECTED]> wrote: Dear Raghvendra / Mandar , Forwarding to you the Dph-80 , upgrade of firmware which now works with your Sip Server ( 67.109.153.236). Please remember to factory reset once the upgrade through TFTP server is over before any further configuration. Now the user name , passwords for accessing the phones through web are kept blank. Factory reset can be done through keys using *789*# . Also enable log server in Network settings with your sip server ip ( 67.109.153.236). Please inform us what port you are using for log server . Also in sip configuration , disable auto attendant , enable vad , keep user=phone enabled . Always click submit on every page you are configuring and at last click = save and restart . Now the phones will only be accessible by there new ips , and should give a dial tone . I have tested this in-between two DPH-80s and from Mr.Mandars side at pune in-between soft phone to our hard phone . with G.729 and G.711 enabled , its working perfectly well and getting registered on your server . Mr.Mandar please forward me all the server logs for today between 2006 & 2008 nos that I have configured at my end . Thanks & regards, Abhijit M. VoIP Dept. D-Link India Ltd. Mumbai. Phone: +91-22-2652 6696/56902210, Ext-194 Fax : +91-22-26528914 Log Jul 15 01:45:42 WARNING[-151058624]: Unable to open /dev/dsp: No such device Jul 15 01:45:42 WARNING[-151058624]: Invalid localnet keyword: 192.168.0.0/255.255.0.0 Jul 15 01:45:52 WARNING[-151058624]: Invalid localnet keyword: 10.0.0.0/255.0.0.0 Jul 15 01:46:02 WARNING[-151058624]: Invalid localnet keyword: 172.16.0.0/12 Jul 15 01:46:13 WARNING[-151058624]: Invalid localnet keyword: 169.254.0.0/255.255.0.0 Jul 15 01:46:13 VERBOSE[-151058624]: -- SIP Seeding '8612312342' at [EMAIL PROTECTED]:5061 for 1800 Jul 15 01:46:13 WARNING[-151058624]: Ignoring port for now Jul 15 01:46:13 WARNING[-151058624]: MySQL database sock file not specified. Using default Jul 15 01:46:20 WARNING[-203732048]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) Jul 15 01:55:50 WARNING[-288244816]: MySQL database sock file not specified. Using default Jul 15 01:55:50 WARNING[-288244816]: Ignoring port for now Jul 15 01:55:50 NOTICE[-288244816]: Removed default indication country 'us' Jul 15 01:56:00 WARNING[-203732048]: Invalid localnet keyword: 192.168.0.0/255.255.0.0 Jul 15 01:56:10 WARNING[-203732048]: Invalid localnet keyword: 10.0.0.0/255.0.0.0 Jul 15 01:56:20 WARNING[-203732048]: Invalid localnet keyword: 172.16.0.0/12 Jul 15 01:56:30 WARNING[-203732048]: Invalid localnet keyword: 169.254.0.0/255.255.0.0 Jul 15 01:56:31 WARNING[-298812496]: No entry in voicemail config file for '8612312344' Jul 15 01:56:36 WARNING[-203732048]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 57664 (Response) Jul 15 02:00:05 WARNING[-288244816]: No entry in voicemail config file for '8612312344' Jul 15 02:02:42 WARNING[-288244816]: No entry in voicemail config file for '8612312344' Jul 15 02:04:01 WARNING[-288244816]: No entry in voicemail config file for '8612312344' Jul 15 02:07:20 WARNING[-203732048]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) Jul 15 02:07:20 WARNING[-288244816]: No entry in voicemail config file for '8612312344' Jul 15 02:07:26 WARNING[-203732048]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Reque
RE: [Asterisk-Users] Mac OS X installer: missing files fix
Benjamin, Did you try using the post-install script to do the cat and copy operations in lieu of the Mac package installer doing it? The package installer will run any shell command you put in the post-install script? I once used it to overcome a similar problem with PostgreSQL on Panther. That might do the trick. I am going to try out your * AppleScripts as soon as possible! Thanks, Ted -Original Message- From: Sunrise Ltd [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 3:39 PM To: astusr Subject: Re: [Asterisk-Users] Mac OS X installer: missing files fix Wallingford, Ted wrote: >I've paraphrased the OS X installer developer's comments: there's a >bug in Installer that is preventing the archive from working right. >Below is the fix for the problem. Apple may have a reputation for attention to detail and perfectionism, but their PackageMaker utillity -- which is what you use to create those install packages -- does most definitely NOT share any of these virtues. It's one of the worst examples of sloppiness I have seen. PackageMaker simply refuses to include files targeted at /usr (and below) into the bills of materials file (Archive.bom). The files are all there, nicely shrinkwrapped into the archive itself (Archive.pax.gz), but no matter what you do, they won't show up in the BOM file. As a result, the installer will not install but ignore them. I am now going to change the target to /private/tmp and then run a postinstall script (luckily this feature actually works) to move the files into /usr. I will also provide a patching utility for those who have been hit by this. Folks, I am very sorry about the inconvenience. I have tested the installer on various systems beforehand, but I must have missed to wipe everything at some point. My sincerest apologies. >First (obviously) run the installer. Since the executables are in the >archive.pax.gz file in the installer package, first do a "show package >contents" on the package file, then unstuff the enclosed archive.pax.gz >file to the desktop... Then open up a shell, >CD to the desktop, and run the following: > > cat Archive.pax | pax -r > sudo cp -R usr/* /usr Thanks for your follow up. Yes, this will work, but I guess that it is a bit too crude for many Mac folks, they like to just click on things to make stuff work. That was the whole point of creating the install package in the first place. I have failed the Mac folks miserably in this regard. So, please hang on there for a little while, I'll get back to you with a new install package and a patching utility. Meanwhile, we've released a few clickable AppleScript script apps for basic control of Asterisk (start/stop/reload/show version). You can download them (as a zip archive) from http:/www.astmasters.net/stuff/AsteriskApplescripts.zip regards benjamin -- Sunrise Telephone Systems Ltd 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan __ Do You Yahoo!? http://bb.yahoo.co.jp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Wallingford, Ted.vcf Description: Binary data
Re: [Asterisk-Users] PSTN gateway implementation?
> -I have a TE405P board and only one T1 worth of phone lines (24) > connected to it using an Adtran TA750 channel bank. Any particular reason against using PRI from your telco? > Is Asterisk capable of handling multiple incoming VoIP calls arriving > from the same source (IP) or do I need to get something else to take the > incoming traffic and pass it on to Asterisk? (I've read about using SER > as a SIP proxy, but it's not clear to me wheather I need it or not). Can > I use the OpenH.323 module to take care of the incoming VoIP traffic? Asterisk can handle multiple calls from the same IP without any worry. Your main worry is the lack of real billing since you're terminating to analog PSTN instead of using PRI -- you have no way of actually knowing if the call was answered or not, so he'll be billed on every call. I doubt you want to try and work with callprogress=yes. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] uip200 clips audio prompts
On Mon, 19 Jul 2004, Ryan Courtnage wrote: > Hi all, > > We find that our UIP200s clip off the 1st second of audio prompts from * (ie: > the beginning of voicemail prompts). > > Has anyone found a way around this? > > Running CVS-D2004.06.29.15.30.00 on WBEL 3.0 This happens with my 7940s as well. I have found that using and Answer, and a Wait(1) before playing back prompts works well. Prevents Alisson from saying "Assword?" when dialing VoicemailMail(20). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: LAN Switch w/ QoS
> For those on a low budget compex (http://cpx.com) has some very low cost > switches that support QoS. > > http://www.cpx.com/proddetail.asp?c=Switches&e=109 > > Bought a few of these myself, seem to work well. They are only > manageable through an rs-232 console though, and don't have some other > features of the high end switches like Spanning Tree protocol. Sounds like its a secure switch too. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE:RE: [Asterisk-Users] Codecs - Advantages
Yes, I don't have any bandwidth issues, so I could use 2 Mbps for 30 calls. I do have issues with processing CPU capacity. Is g711 CPU intensive as g729 ? I understand g729 is very CPU intensive. -- Mensaje Original -- Enviado por: Sebastian Nocetti <[EMAIL PROTECTED]> Fecha: 19/07/2004 18:29:27 Para: <[EMAIL PROTECTED]> Título: RE: [Asterisk-Users] Codecs - Advantages If you dont have bandwith issues, use g711, with 2 mb bandwith you can pass 30 calls, aprox. G729 compress from g711 64 kbps to g729 8 kbps -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: Lunes, 19 de Julio de 2004 02:44 p.m. Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] Codecs - Advantages Hi, I'm planning to use a Asterisk with Digium E1 cards, I understand that using a codec such as G.729 can be very CPU demanding. What are the real advantages of using a codec such as G.729 ? Bandwidth only ? Using no compression wouldn't increase the scalability of my asterisk PBX ? This is considering I have no bandwidth issues in my network. Thanks __ Todavía no tenés tu Ciudad Internet Mail? Obtenelo ahora! - http://webmail.ciudad.com.ar Descargá Gratis el nuevo Internet Explorer 6.0, el mejor software para actualizar tu PC. http://www.ciudad.com.ar/ar/servicios/ie/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Todavía no tenés tu Ciudad Internet Mail? Obtenelo ahora! - http://webmail.ciudad.com.ar Descargá Gratis el nuevo Internet Explorer 6.0, el mejor software para actualizar tu PC. http://www.ciudad.com.ar/ar/servicios/ie/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cheap PoE switches/injectors?
Greg Hill wrote: officially, a POE capable switch/etc is supposed to do a discovery routine to detect, when a device is plugged into it, whether that device requires POE. Right? And the single-port POE injectors are usually nothing more than two RJ45 packs with a dc power connector, right? That could be the difference in price there: the detection circuitry. Or am I way off? No, even the single-port injectors have to have that circuitry, I believe. Otherwise power on the extra pins would be live all the time, which could be damaging to any non-PoE equipment you plugged in there. Also, the PoE device itself wants to negotiate its power usage with the power supplier, and if the injector didn't respond I don't know if the PoE device would ever come up. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PSTN gateway implementation?
Hello, I need help in creating a simple PSTN Gateway. This is the scenario: -I have one client sending me VoIP traffic (they don’t have asterisk, so IAX is out of the picture for me) and I need to validate that traffic (only accept calls coming from his IP). After that I would terminate the calls to the PSTN network and keep logs for billing purposes. -I have a TE405P board and only one T1 worth of phone lines (24) connected to it using an Adtran TA750 channel bank. Is Asterisk capable of handling multiple incoming VoIP calls arriving from the same source (IP) or do I need to get something else to take the incoming traffic and pass it on to Asterisk? (I’ve read about using SER as a SIP proxy, but it’s not clear to me wheather I need it or not). Can I use the OpenH.323 module to take care of the incoming VoIP traffic? I am totally new to all this. Any help will be really appreciated. Please give me your input on which is the best implementation of a VoIP->PSTN gateway and the some sample configuration files for the plattforms involved (Asterisk, SER, OpenH323, etc.) Thanks in advance. Alejandro Sosa.
Re: [Asterisk-Users] Cheap PoE switches/injectors?
Some certainly are dumb power. For instance the Linksys POE modules. I wasn't aware of that being an issue with the 3com module. The way to know for sure is to ask if it supports 802.3af which is the standard that includes the autodetection to pretect non-POE equipment. Did a quick search and it looks like you are correct about the 3com single line injectors, which does indeed explain the price differential. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting Greg Hill <[EMAIL PROTECTED]>: > On Mon, 19 Jul 2004, Kevin P. Fleming wrote: > > > Scott Laird wrote: > > > > > So $1600 for 24 ports. That's not *too* bad. HP seems to have a > > > similar model (2626-PWR) for a similar price. 3com also seems to have a > > > 24-port injector for $800. > > > > I still don't understand why I can buy single-port injectors for $20, > > but multi-port models are $30 per port and up. You'd think that having a > > single combined power supply and other bits would reduce the cost, not > > increase it. > > > officially, a POE capable switch/etc is supposed to do a discovery routine > to detect, when a device is plugged into it, whether that device requires > POE. Right? And the single-port POE injectors are usually nothing more > than two RJ45 packs with a dc power connector, right? That could be the > difference in price there: the detection circuitry. Or am I way off? > > Greg > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] uip200 clips audio prompts
Hi all, We find that our UIP200s clip off the 1st second of audio prompts from * (ie: the beginning of voicemail prompts). Has anyone found a way around this? Running CVS-D2004.06.29.15.30.00 on WBEL 3.0 Thanks -- .. Ryan Courtnage Coalescent Systems Inc 403.244.8089 www.voxbox.ca ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration
Would the TLI(2)-U10 ETU work as well? Jason Kawakami wrote: From: "Christopher L. Wade" <[EMAIL PROTECTED]> Organization: Unistar-Sparco Computers, Inc. To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration Reply-To: [EMAIL PROTECTED] Exactly which NEC T1 interface did you use? I'm looking at the DTI-U20, I don't think I'll need the U30, but I'm not entirely sure. The U20 will be fine. The U30 adds MF receivers for Feature Group D/E911. The T-1 is just set up as E&M tie line. Good Luck Jason Kawakami Open Telephony Labs, LLC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Phone recommendation
I really wish a few users would help me with my Avaya 4602SW SIP phone as I believe I am the first to try it with Asterisk. Its a very modern looking phone, I've seen it on froogle for about $135, and it has a decent set of features, and Avaya is really good quality and has a good name in telephony. The Avaya 4602 has been out for a while, but they just released the SIP firmware for it about 1 week ago. The only issue so far that I have with it is that it stops working after I get a "-- Got SIP response 481 "Call Does Not Exist" back from my.home.ip.address" which occurs about 15 or 20 minutes after I plug the phone in, if I reset the phone it, again, will work for about 15-20 mins. On Mon, 19 Jul 2004 09:25:48 -0700, Harry McGregor <[EMAIL PROTECTED]> wrote: > On Mon, 2004-07-19 at 09:04, Yiannis Costopoulos wrote: > > Hi, > > > > I am looking for some affordable IP Phones. Any experiences with the > > SipToneII by ipDialog? > > So far our experience with the IP Dialog SipToneII is not good. It > locks up after hang up on us, and just does not play nice. If anyone > has any suggestions on how to get it working, we are all ears. > > The IP Dialog phone is running $200, while the Zip 4x4 is running > $280-300 (depending on qty). We are deploying ~60 phones. Originally > we were going to try and do 20 Uniden UIP200 and 40 Zip 4x4. We were > unable to get our hands on a Uniden, and found that it would not even be > available for an august deployment, so we decided to try the IP Dialog > phone. The Uniden would have been a very worth while cost savings, as > it's $150 and the Zip is $280 for our qty, but the $80 savings of the IP > Dialog is not worth it to us. > >Harry > > > What about soft phones? Any recommendations there (for Windoze and Linux)? > > Have not tried it but what about PhoneGaim? > > > Thanks, > > Yiannis > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > -- > Harry McGregor, Computing Manager > Tucson Support Group - U.S. Geological Survey > University of Arizona - Environment and Natural Resource Building > 520-670-5574 (office) - [EMAIL PROTECTED] > 520-661-7875 (Cell) - [EMAIL PROTECTED] > > The opinions/statements expressed herein are my own and should > not be taken as a position, opinion, or endorsement of the > University of Arizona or the U.S. Geological Survey. > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Cisco 7960 SIP V6 and distinctive ring.
[Try this again...] Hello. Here is what my extension which uses distinctive ring on a Cisco 7960 running V6.2 firmware looks like. Note that the distinctive ring tones are changes in cadence, rather than changes in ringing sounds on the 7960. Also, if you adjust the ringer volume wile the distinctive ring is sounding, the phone will revert to the non-distinctive ring cadence. -Brian exten => 2135551212,1,setvar(ALERT_INFO=4) exten => 2135551212,2,Dial(SIP/100&SIP/401&SIP/403|20|tr) exten => 2135551212,3,Voicemail,u401 } Hi } } Can anyone with distinctive ring on their 7960's possibly post how they've got it to work? } } I understand that the ALERT_INFO variable is involved but using the examples for the variable value from the WiKi I'm just getting an error message from the Asterisk concole. } } Thanks in advance. } } P } ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 500 Voicemail
And it is throughly convoluted in the admin guide. What is the T for? Pipe obviously separates entries. X = any digit one would assume? I am just luooking for a brief explanation. Thanks. Here is the excerpt from the manual. Attribute dialplan.digitmap Permitted Values string compatible with the digit map feature of MGCP described in 2.1.5 of RFC 3435. String is limited to 512 bytes and 20 segments; a comma is also allowed; when reached in the digit map, a comma will turn dial tone back on. [2-9]11|0T| 011xxx.T| [0-1][2- 9]x| [2-9]x| [2-9]xxxT Default Interpretation When this attribute is present, number-only dialing during the setup phase of new calls will be compared against the patterns therein and if a match is found, the call will be initiated automatically eliminating the need to press Send. Attribute Permitted Values Default Interpretation -Original Message- From: Eric Wieling [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 11:50 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail On Mon, 2004-07-19 at 12:42, Wiley E. Siler wrote: > Thank you! > > Can you tell me more about the dial plan feature? How do you setup the > correct digitmap? It is all in the Admin Guide you can download from the Polycom web site. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss." ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 500 Voicemail
My Polycom is on loan as a demo and I assume it is one of the first revision models. In fact it shows as Rev A on the back of the phone. I have all the same buttons you listed save for the Messages button. The 3rd from the bottom on the right column of buttons sayd Voice Mail on my version. That corresponds to the location of your button that says Messages. I assume this was changed by Polycom since their phone has other messaging capability (isntant message for instance) and it was easier to use Messages and unify the meaning instead of Voice Mail and lock it into one type of messaging. Does your Messages button dump you right into voice mail or do you have to navigate a menu first? Thanks, Wiley -Original Message- From: Chris A. Icide [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 11:46 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail Strange, I have an IP500 right out of the new-plastic-gadget-smell box, and it doesn't have a button labelled Voicemail. On the left side are the blue speaker, red mute, and blue headset buttons, then next to them top to bottom are the three Line buttons (clear covers for putting your own labels), Directories, Services, Call Lists, Conference, Transfer, and Redial. On the right of the system, top side are the 4 way selection pad with select and delete, then below that are Menu, Messages, and Do Not Disturb, and finally Hold. In the middle are the 12 keypad keys, 4 soft keys, and volume + and - buttons. No where am I able to find a hard voicemail button. -Chris On 10:42 AM 7/19/2004, Wiley E. Siler wrote: >Thank you! > >Can you tell me more about the dial plan feature? How do you setup the >correct digitmap? > >W > >-Original Message- >From: Russ Beaupre, P.E. [mailto:[EMAIL PROTECTED] >Sent: Monday, July 19, 2004 4:56 AM >To: [EMAIL PROTECTED] >Subject: Re: [Asterisk-Users] Polycom IP 500 Voicemail > >Wiley E. Siler wrote: > >> I have a solution that allows me to assign a soft key with no >problems. >> However, it seems like a waste the the hard button labeled Voice Mail >> is not dialing right into voice mail. Is there a known way yo do >> this? I have tried everything in the manual but it doesn't seem to >> work. I have IP 500s and I want to be able to use all three display >> lines for just lines on the phone. >> >I think that feature is inly available on the 1.2.0 sip firmware. It >works on ours but when you press it, you still have to pick a line, then >connect. The line button goes right to the voicemail. > >> Also, do you know if it is possible to program the buttons along the >> bottom of the screen like normal soft buttons? >> >Probably, but I haven't looked into it enough > >> And finally... >> Is there a way to make the system dial without having to hit the Send >> key after dialing a number? >> >look at the digitmap in sip.cfg > >-rb > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mac OS X installer: missing files fix
Wallingford, Ted wrote: (B (B>I've paraphrased the OS X installer developer's comments: (Bthere's a (B>bug in Installer that is preventing the archive from (Bworking right. (B>Below is the fix for the problem. (B (BApple may have a reputation for attention to detail and (Bperfectionism, but their PackageMaker utillity -- which is (Bwhat you use to create those install packages -- does most (Bdefinitely NOT share any of these virtues. It's one of the (Bworst examples of sloppiness I have seen. (B (BPackageMaker simply refuses to include files targeted at (B/usr (and below) into the bills of materials file (B(Archive.bom). The files are all there, nicely (Bshrinkwrapped into the archive itself (Archive.pax.gz), (Bbut no matter what you do, they won't show up in the BOM (Bfile. As a result, the installer will not install but (Bignore them. (B (BI am now going to change the target to /private/tmp and (Bthen run a postinstall script (luckily this feature (Bactually works) to move the files into /usr. I will also (Bprovide a patching utility for those who have been hit by (Bthis. (B (BFolks, I am very sorry about the inconvenience. I have (Btested the installer on various systems beforehand, but I (Bmust have missed to wipe everything at some point. My (Bsincerest apologies. (B (B>First (obviously) run the installer. Since the (Bexecutables are in the (B>archive.pax.gz file in the installer package, first do a (B"show package (B>contents" on the package file, then unstuff the enclosed (B>archive.pax.gz file to the desktop... Then open up a (Bshell, (B>CD to the desktop, and run the following: (B> (B> cat Archive.pax | pax -r (B> sudo cp -R usr/* /usr (B (BThanks for your follow up. (B (BYes, this will work, but I guess that it is a bit too (Bcrude for many Mac folks, they like to just click on (Bthings to make stuff work. That was the whole point of (Bcreating the install package in the first place. I have (Bfailed the Mac folks miserably in this regard. (B (BSo, please hang on there for a little while, I'll get back (Bto you with a new install package and a patching utility. (B (BMeanwhile, we've released a few clickable AppleScript (Bscript apps for basic control of Asterisk (B(start/stop/reload/show version). You can download them (B(as a zip archive) from (Bhttp:/www.astmasters.net/stuff/AsteriskApplescripts.zip (B (Bregards (Bbenjamin (B (B-- (BSunrise Telephone Systems Ltd (B9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan (B (B__ (BDo You Yahoo!? (Bhttp://bb.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 500 Voicemail
I read the administrator document repeatedly. I have not been able to find a wiki that applied to digitmap feature at all and I have searched repeatedly and read several of the wikis regarding Polycoms. The administrators guide doesn't have enough context explanation to make the use of the digitmap understandable. That is the basis of my request for a digitmap explanation. I am not asking someone to write mine for me. I am asking to see an example and an explanation that gives context so I can write my own and know I have done it properly. My PBX is Asterisk and the setup is about as generic as generic can be. Polycoms over SIP to the PBX. If you know where the wiki is for digitmaps please send it. If you feel inspired, a short explanation of the relevance and context of digitmaps would be greatly appreciated. I know everyone has to take their own time to answer these emails and I truly appreciate that. That is why I do my research until I hit a wall, then I will ask here. I appreciate whatever you can spare time for. Thanks! Wiley -Original Message- From: Brent Franks [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 10:26 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail > Thank you! > > Can you tell me more about the dial plan feature? How do you setup the > correct digitmap? > Check the Administrator's Document. You can find it on the Wiki, under IP Phones.. Polycom. Did you try to look up the digitmap feature before sending this post? If not, you should be able to understand it when you read it, it's relatively straight forward. No one can setup a correct digitmap for you, as it will vary greatly on how you have setup your PBX. - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Flash Zap trunk from a Sipura
I think this has been an ongoing issue. When you figure out a solution let me know. The only solution I could come up with isnt feasible for everyone. I have another pbx that provides the dialtone for my asterisk box. I put two zap cards in my asterisk. On my other switch I set it so that if line 1 is busy it rolls to line 2, this way when my second call comes in I can switch over to it from my Sipura. It works for me, but isnt possible for everyone. On Mon, 19 Jul 2004 10:09:30 -0700, Trevor Peirce <[EMAIL PROTECTED]> wrote: > Hello, > > In my quest to create several proof of concepts for what can be done > with Asterisk, I've run into a bit of a problem. I have a pair of > SPA-2000's acting as off premise extensions for an analog line. When a > call waiting call comes in, the caller id information makes it though > the ULAW codec and displays on the caller id box, however asterisk > doesn't seem to want to pick up the hook-flash sent by the Sipura to > answer that second call. > > I have configured the Sipura to send hook-flash messages to asterisk, > and it does, but asterisk doesn't seem to know what to do with them. > I've searched the wiki and google with no success. > > Thanks, > Trevor > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] collect calls
Hi, Does anybody knows where can I change timing for collect calls? tks Oz _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cheap PoE switches/injectors?
On Mon, 19 Jul 2004, Kevin P. Fleming wrote: > Scott Laird wrote: > > > So $1600 for 24 ports. That's not *too* bad. HP seems to have a > > similar model (2626-PWR) for a similar price. 3com also seems to have a > > 24-port injector for $800. > > I still don't understand why I can buy single-port injectors for $20, > but multi-port models are $30 per port and up. You'd think that having a > single combined power supply and other bits would reduce the cost, not > increase it. officially, a POE capable switch/etc is supposed to do a discovery routine to detect, when a device is plugged into it, whether that device requires POE. Right? And the single-port POE injectors are usually nothing more than two RJ45 packs with a dc power connector, right? That could be the difference in price there: the detection circuitry. Or am I way off? Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Cisco 7960 SIP V6 and distinctive ring.
Hello. Here is what my extension which uses distinctive ring on a Cisco 7960 running V6.2 firmware looks like. Note that the distinctive ring tones are changes in cadence, rather than changes in ringing sounds on the 7960. Also, if you adjust the ringer volume wile the distinctive ring is sounding, the phone will revert to the non-distinctive ring cadence. -Brian } Hi } } Can anyone with distinctive ring on their 7960's possibly post how they've got it to work? } } I understand that the ALERT_INFO variable is involved but using the examples for the variable value from the WiKi I'm just getting an error message from the Asterisk concole. } } Thanks in advance. } } P } ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PSTN gateway implementation?
you can do that, in my experiencie, using oh323 I could not handle more than 30 active calls, doing g729 passthru... I dont know how to do IP limitation for restrict ip access use iptables I did basic dialpeers like this: exten -> 1305.,1,dial(OH323/) exten -> 1305.,2,congestion I am right? De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Alejandro SosaEnviado el: Lunes, 19 de Julio de 2004 03:25 p.m.Para: [EMAIL PROTECTED]Asunto: [Asterisk-Users] PSTN gateway implementation? Hello, I need help in creating a simple PSTN Gateway. This is the scenario: -I have one client sending me VoIP traffic (they don’t have asterisk, so IAX is out of the picture for me) and I need to validate that traffic (only accept calls coming from his IP). After that I would terminate the calls to the PSTN network and keep logs for billing purposes. -I have a TE405P board and only one T1 worth of phone lines (24) connected to it using an Adtran TA750 channel bank. Is Asterisk capable of handling multiple incoming VoIP calls arriving from the same source (IP) or do I need to get something else to take the incoming traffic and pass it on to Asterisk? (I’ve read about using SER as a SIP proxy, but it’s not clear to me wheather I need it or not). Can I use the OpenH.323 module to take care of the incoming VoIP traffic? I am totally new to all this. Any help will be really appreciated. Please give me your input on which is the best implementation of a VoIP->PSTN gateway and the some sample configuration files for the plattforms involved (Asterisk, SER, OpenH323, etc.) Thanks in advance. Alejandro Sosa.
Re: [Asterisk-Users] Cheap PoE switches/injectors?
You can pick up 3c17205's on ebay for usually around $500-$700 new in the box (non on there at the moment). They come up about 3-4 a month although it's summer at the moment so a bit quiet. P > -Original Message- > From: Scott Laird [mailto:[EMAIL PROTECTED] > Sent: Monday, July 19, 2004, 11:42 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Cheap PoE switches/injectors? > > > On Jul 19, 2004, at 9:03 AM, [EMAIL PROTECTED] wrote: > > > Look out for 3c17205 switches from 3com and read the QOS thread > > posting here at the moment. > > > > So $1600 for 24 ports. That's not *too* bad. HP seems to have a > similar model (2626-PWR) for a similar price. 3com also seems to have > a 24-port injector for $800. > > > Scott > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration
From: "Christopher L. Wade" <[EMAIL PROTECTED]> Organization: Unistar-Sparco Computers, Inc. To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration Reply-To: [EMAIL PROTECTED] Exactly which NEC T1 interface did you use? I'm looking at the DTI-U20, I don't think I'll need the U30, but I'm not entirely sure. The U20 will be fine. The U30 adds MF receivers for Feature Group D/E911. The T-1 is just set up as E&M tie line. Good Luck Jason Kawakami Open Telephony Labs, LLC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 500 Voicemail
On Mon, 2004-07-19 at 12:42, Wiley E. Siler wrote: > Thank you! > > Can you tell me more about the dial plan feature? How do you setup the > correct digitmap? It is all in the Admin Guide you can download from the Polycom web site. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss." ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rotary phones? (No, I'm serious)
> I don't know how serious pulse dial is, but it is supported in asterisk. > You will not likely find any device that converts pulse to tone though. > Although it might be possible if it went through a channel that doesn't > use pulse dialing like sip. Hp So a channel bank w/ FXS ports will pass the rotary data thru to Asterisk? Neat. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re:FATAL: Module zaptel not found.
The output of my lsmod doesn´t seems to have the zaptel and wcfxo modules, And the output of cat /proc/interrupts is: CPU0 0: 103001742 XT-PIC timer 1: 1134 XT-PIC i8042 2: 0 XT-PIC cascade 5: 37511 XT-PIC Ensoniq AudioPCI 8: 1 XT-PIC rtc 9: 0 XT-PIC ohci_hcd 10:4979359 XT-PIC eth0 11: 10285 XT-PIC aic7xxx 12: 3070 XT-PIC i8042 14: 824226 XT-PIC ide0 15:485 XT-PIC ide1 NMI: 0 ERR: 0 And the kudzu detected the card as a modem. When i made "make" in /usr/src/zaptel gave me error. May you help me. Kind regards, Juanjo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 500 Voicemail
Strange, I have an IP500 right out of the new-plastic-gadget-smell box, and it doesn't have a button labelled Voicemail. On the left side are the blue speaker, red mute, and blue headset buttons, then next to them top to bottom are the three Line buttons (clear covers for putting your own labels), Directories, Services, Call Lists, Conference, Transfer, and Redial. On the right of the system, top side are the 4 way selection pad with select and delete, then below that are Menu, Messages, and Do Not Disturb, and finally Hold. In the middle are the 12 keypad keys, 4 soft keys, and volume + and - buttons. No where am I able to find a hard voicemail button. -Chris On 10:42 AM 7/19/2004, Wiley E. Siler wrote: >Thank you! > >Can you tell me more about the dial plan feature? How do you setup the >correct digitmap? > >W > >-Original Message- >From: Russ Beaupre, P.E. [mailto:[EMAIL PROTECTED] >Sent: Monday, July 19, 2004 4:56 AM >To: [EMAIL PROTECTED] >Subject: Re: [Asterisk-Users] Polycom IP 500 Voicemail > >Wiley E. Siler wrote: > >> I have a solution that allows me to assign a soft key with no >problems. >> However, it seems like a waste the the hard button labeled Voice Mail >> is not dialing right into voice mail. Is there a known way yo do >> this? I have tried everything in the manual but it doesn't seem to >> work. I have IP 500s and I want to be able to use all three display >> lines for just lines on the phone. >> >I think that feature is inly available on the 1.2.0 sip firmware. It >works on ours but when you press it, you still have to pick a line, then >connect. The line button goes right to the voicemail. > >> Also, do you know if it is possible to program the buttons along the >> bottom of the screen like normal soft buttons? >> >Probably, but I haven't looked into it enough > >> And finally... >> Is there a way to make the system dial without having to hit the Send >> key after dialing a number? >> >look at the digitmap in sip.cfg > >-rb > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Codecs - Advantages
If you have the bandwidth then use ulaw :) bkw > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] > Sent: Monday, July 19, 2004 12:44 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Codecs - Advantages > > Hi, > I'm planning to use a Asterisk with Digium E1 cards, I understand that > using a codec such as G.729 can be very CPU demanding. What are the real > advantages of using a codec such as G.729 ? Bandwidth only ? Using no > compression wouldn't increase the scalability of my asterisk PBX ? This is > considering I have no bandwidth issues in my network. > > Thanks > __ > > Todavía no tenés tu Ciudad Internet Mail? Obtenelo ahora! - > http://webmail.ciudad.com.ar > > Descargá Gratis el nuevo Internet Explorer 6.0, el mejor software para > actualizar tu PC. > http://www.ciudad.com.ar/ar/servicios/ie/ > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 500 Voicemail
> Thank you! > > Can you tell me more about the dial plan feature? How do you setup the > correct digitmap? > Check the Administrator's Document. You can find it on the Wiki, under IP Phones.. Polycom. Did you try to look up the digitmap feature before sending this post? If not, you should be able to understand it when you read it, it's relatively straight forward. No one can setup a correct digitmap for you, as it will vary greatly on how you have setup your PBX. - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cheap PoE switches/injectors?
Scott Laird wrote: So $1600 for 24 ports. That's not *too* bad. HP seems to have a similar model (2626-PWR) for a similar price. 3com also seems to have a 24-port injector for $800. I still don't understand why I can buy single-port injectors for $20, but multi-port models are $30 per port and up. You'd think that having a single combined power supply and other bits would reduce the cost, not increase it. I'd like to see someone make a single-port injector that fits into a keystone jack, so I can insert 24 of them into a 24-port keystone rack-mount panel, and then power them with a stock Valcom 48V A-battery power supply. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MAC OS X Panther :?
On Mon, 2004-07-19 at 12:02, Francisco Perez-Landaeta wrote: > Just wondering if anyone has tried MAC OS X and panther. > I will like to do SIP to H323, not sure if this will be possible on the MAC > because of the Libraries PWlib and OPenh32 for Linux.. Is this for running asterisk on OS X? or for a soft phone? If it's a soft phone, then there shouldn't be a problem. > > Just curious.. > > Anyway, anyone has an easy guide (step by step) to setup oh323 with > asterisk. I saw a guide but i am not very savy on linux. > thanks, > Francisco > > - Original Message - > From: <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Monday, July 19, 2004 12:25 PM > Subject: Asterisk-Users digest, Vol 1 #4598 - 14 msgs > > > > Send Asterisk-Users mailing list submissions to > > [EMAIL PROTECTED] > > > > To subscribe or unsubscribe via the World Wide Web, visit > > http://lists.digium.com/mailman/listinfo/asterisk-users > > or, via email, send a message with subject or body 'help' to > > [EMAIL PROTECTED] > > > > You can reach the person managing the list at > > [EMAIL PROTECTED] > > > > When replying, please edit your Subject line so it is more specific > > than "Re: Contents of Asterisk-Users digest..." > > > > > > Today's Topics: > > > >1. Re: STILL NO AUDIO (Michael Manousos) > >2. Re: TDM400P Internal Extenion Config (Nick Cobley) > >3. Re: ZyXEL 2000W (Jason Williams) > >4. Channel banks, voicemail, and immediate=no (Chris A. Icide) > >5. RE: STILL NO AUDIO (Eric Wieling) > >6. Re: STILL NO AUDIO (Holger Schurig) > >7. RE: Mac OS X installer for Asterisk (Wallingford, Ted) > >8. Re: PhoneGaim? ([EMAIL PROTECTED]) > >9. Re: BroadVoice problems? (Chris Shaw) > > 10. RE: STILL NO AUDIO (Sebastian Nocetti) > > 11. Re: TDM400P Internal Extenion Config (Jason Williams) > > 12. IP Phone recommendation (Yiannis Costopoulos) > > 13. Re: Cheap PoE switches/injectors? ([EMAIL PROTECTED]) > > 14. RE: STILL NO AUDIO (Sebastian Nocetti) > > > > --__--__-- > > > > Message: 1 > > Date: Mon, 19 Jul 2004 18:24:39 +0300 > > From: Michael Manousos <[EMAIL PROTECTED]> > > Organization: inAccess Networks > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] STILL NO AUDIO > > Reply-To: [EMAIL PROTECTED] > > > > > > Why don't you use asterisk-oh323? > > > > Michael. > > > > Sebastian Nocetti wrote: > > > I WANT TO USE G729, I HAVE TO USE IT... > > > > > > -Mensaje original- > > > De: [EMAIL PROTECTED] > > > [mailto:[EMAIL PROTECTED] En nombre de Eric Wieling > > > Enviado el: Lunes, 19 de Julio de 2004 11:46 a.m. > > > Para: [EMAIL PROTECTED] > > > Asunto: Re: [Asterisk-Users] STILL NO AUDIO > > > > > > I suspect it will be solved when you put disallow=all and allow=ulaw in > > > sip.conf and h323.conf (and NO OTHER ALLOW= LINES) > > > > > > On Mon, 2004-07-19 at 09:25, Sebastian Nocetti wrote: > > > > > >>I cant do SIP - CHAN_H323 transmit audio!!! I can hear rings, but when > > >>connected, NOTHING > > >> > > >> > > >> > > >>It happened in both: SIP -> CHAN_H323 and CHAN_H323 -> SIP... > > >> > > >> > > >> > > >>when it will be solved? > > > > > > --__--__-- > > > > Message: 2 > > Date: Mon, 19 Jul 2004 23:26:06 +0800 > > From: Nick Cobley <[EMAIL PROTECTED]> > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] TDM400P Internal Extenion Config > > Reply-To: [EMAIL PROTECTED] > > > > Thanks Steve, > > > > The SIP handsets are working find as I can make calls to other handsets > > as well as receive incoming calls via the FXO module. So all is good > there. > > > > Cheers > > Nick > > > > Steven Critchfield wrote: > > > > >On Mon, 2004-07-19 at 07:13, Nick Cobley wrote: > > > > > > > > > > > >>If I dial the extension I just get a 404 error on the phone > > >>(Grandstream), but no errors at all on the console. I am using > > >>CVS-HEAD-07/14/04. Here is a snippet of what I have in the various > > >>config files. > > >> > > >> > > > > > >Welcome to SIP. Dialtone is local to your phone and is not dependent on > > >proper config. Hope that helps put you on the correct step to fix that > > >problem. > > > > > > > > > > > > --__--__-- > > > > Message: 3 > > Date: Mon, 19 Jul 2004 16:26:26 +0100 > > From: Jason Williams <[EMAIL PROTECTED]> > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] ZyXEL 2000W > > Reply-To: [EMAIL PROTECTED] > > > > On Fri, 16 Jul 2004 01:23:54 +1000, Andrew Yager <[EMAIL PROTECTED]> > wrote: > > > Does anyone have the call hold feature working? If you do... how did > > > you make it work? The instructions say to press the left button to > > > place the call on hold, and the right button to take it off - except > > > when I am in a call, these keys have no effect. > > > > > > I've tried teh 000c firmware, the 000e firmware and the Pulver 0011 > > > firmware - but none work, so I'm wondering if this feature just simply > > > isn't implemented, or if there is likely to be something wrong with my > > > asterisk config. > > >
RE: [Asterisk-Users] Codecs - Advantages
If you dont have bandwith issues, use g711, with 2 mb bandwith you can pass 30 calls, aprox. G729 compress from g711 64 kbps to g729 8 kbps -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: Lunes, 19 de Julio de 2004 02:44 p.m. Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] Codecs - Advantages Hi, I'm planning to use a Asterisk with Digium E1 cards, I understand that using a codec such as G.729 can be very CPU demanding. What are the real advantages of using a codec such as G.729 ? Bandwidth only ? Using no compression wouldn't increase the scalability of my asterisk PBX ? This is considering I have no bandwidth issues in my network. Thanks __ Todavía no tenés tu Ciudad Internet Mail? Obtenelo ahora! - http://webmail.ciudad.com.ar Descargá Gratis el nuevo Internet Explorer 6.0, el mejor software para actualizar tu PC. http://www.ciudad.com.ar/ar/servicios/ie/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Codecs - Advantages
Hi, I'm planning to use a Asterisk with Digium E1 cards, I understand that using a codec such as G.729 can be very CPU demanding. What are the real advantages of using a codec such as G.729 ? Bandwidth only ? Using no compression wouldn't increase the scalability of my asterisk PBX ? This is considering I have no bandwidth issues in my network. >> If you have no bandwidth issues use G711 as it will provide you with the best quality! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PSTN gateway implementation?
Hello, I need help in creating a simple PSTN Gateway. This is the scenario: -I have one client sending me VoIP traffic (they don’t have asterisk, so IAX is out of the picture for me) and I need to validate that traffic (only accept calls coming from his IP). After that I would terminate the calls to the PSTN network and keep logs for billing purposes. -I have a TE405P board and only one T1 worth of phone lines (24) connected to it using an Adtran TA750 channel bank. Is Asterisk capable of handling multiple incoming VoIP calls arriving from the same source (IP) or do I need to get something else to take the incoming traffic and pass it on to Asterisk? (I’ve read about using SER as a SIP proxy, but it’s not clear to me wheather I need it or not). Can I use the OpenH.323 module to take care of the incoming VoIP traffic? I am totally new to all this. Any help will be really appreciated. Please give me your input on which is the best implementation of a VoIP->PSTN gateway and the some sample configuration files for the plattforms involved (Asterisk, SER, OpenH323, etc.) Thanks in advance. Alejandro Sosa.
Re: [Asterisk-Users] Cheap PoE switches/injectors?
Has anyone tried the new dlink powered switches? I remember seeing an online voip store selling these as a good option for providing power in a voip application. They were price at 1100 for a 24 port model. The lowest cost solution I have seen are the individual 3com power injectors which can be had for between $16-$25. I have done some minimal testing with one for use with wireless access points and it seems workable, although not a good solution for a high density environment. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting Scott Laird <[EMAIL PROTECTED]>: > > On Jul 19, 2004, at 9:03 AM, [EMAIL PROTECTED] wrote: > > > Look out for 3c17205 switches from 3com and read the QOS thread > > posting here at the moment. > > > > So $1600 for 24 ports. That's not *too* bad. HP seems to have a > similar model (2626-PWR) for a similar price. 3com also seems to have > a 24-port injector for $800. > > > Scott > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CTR21/CTR37 Gigaset phones and GS HT286
I'm having no end of trouble with some Siemens Gigaset phones and GS HT286s. Gigaset 100 and 3010 phones work perfectly, but a 4010 only rings once then it goes off and then just flashes it's LEDs and displays "incoming call" on the LCD with no further ringing. According to the manual it is CTR37 but the only setting on the GSs is CTR21, I've tried different cables but some actually make it worse i.e. no ring at all. -- Dave Cotton <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP V6 and distinctive ring.
On Jul 19, 2004, at 9:29 AM, [EMAIL PROTECTED] wrote:
Hi
Can anyone with distinctive ring on their 7960's possibly post how
they've got it to work?
I understand that the ALERT_INFO variable is involved but using the
examples for the variable value from the WiKi I'm just getting an
error message from the Asterisk concole.
I'm setting it to 'Bellcore-dr1' through 'Bellcore-dr4'. I'm grabbing
the value out of Asterisk's database and sticking it into ALERT_INFO
like this:
[macro-setalertinfo]
exten => s,1,DBGet(ALERT_INFO=distinctivering/${CALLERIDNUM})
Works fine for me. You should also be able to do
'SetVar(ALERT_INFO=Bellcore-dr1)' without problems. Can you show us
the line that's generating errors?
Scott
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RE: [Asterisk-Users] Polycom IP 500 Voicemail
Thank you! Can you tell me more about the dial plan feature? How do you setup the correct digitmap? W -Original Message- From: Russ Beaupre, P.E. [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 4:56 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Polycom IP 500 Voicemail Wiley E. Siler wrote: > I have a solution that allows me to assign a soft key with no problems. > However, it seems like a waste the the hard button labeled Voice Mail > is not dialing right into voice mail. Is there a known way yo do > this? I have tried everything in the manual but it doesn't seem to > work. I have IP 500s and I want to be able to use all three display > lines for just lines on the phone. > I think that feature is inly available on the 1.2.0 sip firmware. It works on ours but when you press it, you still have to pick a line, then connect. The line button goes right to the voicemail. > Also, do you know if it is possible to program the buttons along the > bottom of the screen like normal soft buttons? > Probably, but I haven't looked into it enough > And finally... > Is there a way to make the system dial without having to hit the Send > key after dialing a number? > look at the digitmap in sip.cfg -rb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codecs - Advantages
Hi, I'm planning to use a Asterisk with Digium E1 cards, I understand that using a codec such as G.729 can be very CPU demanding. What are the real advantages of using a codec such as G.729 ? Bandwidth only ? Using no compression wouldn't increase the scalability of my asterisk PBX ? This is considering I have no bandwidth issues in my network. Thanks __ Todavía no tenés tu Ciudad Internet Mail? Obtenelo ahora! - http://webmail.ciudad.com.ar Descargá Gratis el nuevo Internet Explorer 6.0, el mejor software para actualizar tu PC. http://www.ciudad.com.ar/ar/servicios/ie/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration
Exactly which NEC T1 interface did you use? I'm looking at the DTI-U20, I don't think I'll need the U30, but I'm not entirely sure. Thanks, Chris Tony Nichols wrote: On Fri, 2004-07-16 at 16:34, Christopher L. Wade wrote: Hi, I'm am currently in the process of trying to integrate an * box with an NEC Electra Elite IPK. Currently, we have 7 POTS lines coming into our building. These lines are plugged into our NEC using the appropriate analog line interface card from NEC. The NEC effectively has NO configuration done to it, other than to make all the internal phones ring when a call comes in. We also have voicemail and an extremely simple auto attendant setup to deal with calls during off hours. Due to the cost of all the components/software/consulting needed to make the NEC do everything it needs to do, we are hoping to 'merge' the NEC with an * box. In my 'working' * box, I have a wctdm11b (asterisk dev-kit) with 1 FXO and 1 FXS card. I say working, because I have everything setup if I totally bypass the NEC. As per an email conversation with Digium, we are connecting our POTS line to the FXS card, and the NEC to the FXO card. My current dilemma is that when I plug the * box and the NEC together, I cannot get the * box to 'dial' a particular extension on the NEC. It is my belief that this is due to some configuration changes needing to be made on the NEC. Unfortunately, this is the exact thing I needed to avoid and the reason for changing from the NEC to * in the first place. I know some changes to the NEC need to be made, but I am unsure as to exactly what, and how to do it. Any input on how to get this working would be greatly appreciated. If more information is required, please let me know. Please don't flame me for possibly being off-top, I don't think I need baby stepping through this, I simply need to know where to start looking. Thanks, Chris I know what ya mean I've spent nearly $800 in tech time for the Nec guy to help me get mine going. I have the Eletra 192 functioning right now, still have some bugs left but working. I used an Nec T1 card in the electra, and a digium t100p in my * box. Let me know if I can be of any help. When I get the last of the bugs worked out I plan to write down the details and put it on the wikki. t o n y ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Gui client
Hi All, Please checkout the following GUI web panels, which have been created and installed from the source code available in this forum. http://67.109.153.236/*web/ It edits extensions.conf after some customization.However unable to update sip.conf. http://67.109.153.236/asterisk-stat/cdr.php Link to the CDR Tool. http://67.109.153.236/cgi-bin/am/am-main.pl The perl based Asterisk GUI Management system. Help is available online in same panel. This code is a bit cumbersome and I am not going to attempt developing this. PHP is much more preferrable. http://67.109.153.236/cgi-bin/astcc/astcc-admin.cgi Calling card application is installed. Uses database `asteriskcc`. Unable to get make it run though, to check it's technical functionality. Once the code reaches some useful level, I am going to post the source code back here, through a download link. Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James Freire Sent: Friday, July 16, 2004 5:13 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk Gui client I have installed the Asterisk gui client that is available off of sourceforge.net. I was curious if anybody here has used it and what experiences they have had with it. I am having a problem with it, I am able to use the admin page except when I try to submit information to the server to add phones I get an error, "The requested URL /astguiclient/method=POST was not found on this server." The directory /astguiclient does exist and works because that is where the php files are located and running from. The URL for this command, so you can see what its submiting, is: http://172.16.200.80/astguiclient/method=POST?ADD=2&extension=&dialplan_number=&voicemail_id=&phone_ip=&computer_ip=&server_ip=&login=&pass=&status=ACTIVE&active=Y&phone_type=&fullname=&company=&picture=&submit=submit I am running Apache/1.3.29 with php installed also. My guess is that there is a bug somewhere in the php code but I do not know php well enough to troubleshoot it. Thanks a lot for any help, James Freire ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cheap PoE switches/injectors?
On Jul 19, 2004, at 9:03 AM, [EMAIL PROTECTED] wrote: Look out for 3c17205 switches from 3com and read the QOS thread posting here at the moment. So $1600 for 24 ports. That's not *too* bad. HP seems to have a similar model (2626-PWR) for a similar price. 3com also seems to have a 24-port injector for $800. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
