[Asterisk-Users] Question about DID
Hello Group I have a Asterisk server running with 2 Digium T1 cards installed. 1 card connects to Telco via a PRI. The 2nd card is connected to a fax server via Digi DataFire RAS 24 PT1 Adapter (Digi0001). The idea is to have Asterisk route the calls based on DID or FAX tones. Everything is working great so far. The only problem is the Fax server does not see the DID. How can I tell if Asterisk it passing the DID and CallerID info to the server? I seen this was done with HylaFax. Any help would be great!! Here is my configs cat zaptel.conf#PRI to Telco span=1,1,0,esf,b8zsbchan=1-23dchan=24 # PRI to Fax serverspan=2,0,0,esf,b8zsbchan=25-47dchan=48 zapata.conf[channels]context=from-analogsignalling=pri_cpeswitchtype=dms100group=1usecallerid=yeshidecallerid=norestrictcid=nousecallingpres=nouseincomingcalleridonzaptransfer=yescallerid=asreceivedfaxdetect=nomusiconhold=defaultchannel = 1-23 context=from-sip-internalswitchtype=dms100signalling=pri_netgroup=2overlapdial=yesusecallerid=yeshidecallerid=norestrictcid=nousecallingpres=nouseincomingcalleridonzaptransfer=yescallerid=asreceivedfaxdetect=nomusiconhold=default channel = 25-47 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Question about DID
I have is like so
exten = 6149233422,1,Dial(Zap/g2/9233422)
Also I found some config file that ask about the following.. This is not an
Asterisk problem but I can't think of a better group of people to help with
this problem...
Address Type (International, National, Network, Subscriber, Abbreviated)
Numbering Plan (ISDN, Data, Telex, National, Private)
Subaddress Type (NSAP, User)
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Crocker
Sent: Friday, February 11, 2005 11:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Question about DID
How are you telling Asterisk to send the call to the fax group? You should
have something in extensions.conf like
exten = _4135551234,1,Dial($FAXTRUNKS/${EXTEN})
Asterisk should send the EXTEN down as a DID to the fax server
-Matt
On Feb 11, 2005, at 11:05 AM, Eric Hall wrote:
Hello Group
I have a Asterisk server running with 2 Digium T1 cards installed. 1
card connects to Telco via a PRI. The 2nd card is connected to a fax
server via Digi DataFire RAS 24 PT1 Adapter (Digi0001). The idea is to
have Asterisk route the calls based on DID or FAX tones. Everything is
working great so far. The only problem is the Fax server does not see
the DID. How can I tell if Asterisk it passing the DID and CallerID
info to the server? I seen this was done with HylaFax.
Any help would be great!!
Here is my configs
cat zaptel.conf
#PRI to Telco
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
# PRI to Fax server
span=2,0,0,esf,b8zs
bchan=25-47
dchan=48
zapata.conf
[channels]
context=from-analog
signalling=pri_cpe
switchtype=dms100
group=1
usecallerid=yes
hidecallerid=no
restrictcid=no
usecallingpres=no
useincomingcalleridonzaptransfer=yes
callerid=asreceived
faxdetect=no
musiconhold=default
channel = 1-23
context=from-sip-internal
switchtype=dms100
signalling=pri_net
group=2
overlapdial=yes
usecallerid=yes
hidecallerid=no
restrictcid=no
usecallingpres=no
useincomingcalleridonzaptransfer=yes
callerid=asreceived
faxdetect=no
musiconhold=default
channel = 25-47
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[Asterisk-Users] Question about wildcard T1 card
Group Can I have 2 wildcard T1 cards in the same box? I was thinking the first card would have channels 1-24 and the second card would have 25-48 Does this sound correct? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Question about wildcard T1 card
I have a system up and running now with 1 card. I need to add a second card for connection from asterisk to my fax server. So the best thing to do is just try it and if it does not work order a 4 port card from Digium. Thanks for all your help!!! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Thursday, February 03, 2005 1:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Question about wildcard T1 card On Thu, 2005-02-03 at 13:16 -0500, Eric Hall wrote: Group Can I have 2 wildcard T1 cards in the same box? I was thinking the first card would have channels 1-24 and the second card would have 25-48 Does this sound correct? You could, but you increase the interupts and might have a system problem at that point. Also the cost difference from 2 T100Ps is not too bad to go to a TE4xxP card and you get 2 more spans along with fewer interupts. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Question about wildcard T1 card
You should be a salesman!! Ha Ha I have 1 T100P card in the server and I have a spare card already paid for and is sitting in an antistatic bag. If Digium will take it back on a trade then I'm all for it. :) The final step will be to get the 4 port card and put it in our prod system when we start selling this! You have been a great help and if money was not so tight for testing I would do that tomorrow. Thanks again!!! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Thursday, February 03, 2005 3:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Question about wildcard T1 card On Thu, 2005-02-03 at 13:54 -0500, Eric Hall wrote: I have a system up and running now with 1 card. I need to add a second card for connection from asterisk to my fax server. So the best thing to do is just try it and if it does not work order a 4 port card from Digium. The following is just suggestions. I am not a Digium reseller and I will only benefit from the following suggestions from by helping Digium and by putting some new ideas in others minds. Okay so your option is to choose an additional $500 card or possibly an additional $1500 card. I can say the extra $1000 is no longer small change, but I would still suggest it as a better option for the following set of reasons. 1. With the spare card sitting in an antistatic bag, you could partially recover from catastrophic failure relatively quickly. You are only a small machine install configure away from getting your original single span back up and running. 2. Development/testing environment. With a 4 span card in your primary gateway machine, you can use 1 for your current usage, 1 for your fax server, and a final span to cross connect to your backup system installed with your legacy T100P card. Makes it much less dangerous to test new CVS checkouts when it isn't the primary machine that helps pay the bills. Number 2 also builds on number 1 as it wouldn't be that difficult to put a copy of the deployed code on your primary machine on the testing machine and keep current copies of the dialplan/configs too. That would lower your response time to a failure of any nature. Of course when I started te response, I thought I had a few more ideas to share, but they seem to have escaped me now. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Thursday, February 03, 2005 1:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Question about wildcard T1 card On Thu, 2005-02-03 at 13:16 -0500, Eric Hall wrote: Group Can I have 2 wildcard T1 cards in the same box? I was thinking the first card would have channels 1-24 and the second card would have 25-48 Does this sound correct? You could, but you increase the interupts and might have a system problem at that point. Also the cost difference from 2 T100Ps is not too bad to go to a TE4xxP card and you get 2 more spans along with fewer interupts. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Realtime voicemail question
Group I'm using realtime for voicemail the it works great.. The only problem I have is I'm not able to use directory or vmail.cgi Does anyone have a solution for this problem? Asterisk CVS-HEAD-01/24/05-07:36:37 RedHat 9.0 Any help would be great Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Upgrade to the newest cvs now asterisk will not start
Hello group I just update to the newest CVS now I'm not able to get asterisk to start. No error during the make or make install I did a make clean before the make;make install Any help would be great Here is the output asterisk -vgcd Parsing /etc/asterisk/asterisk.conf Parsing /etc/asterisk/extconfig.conf == Binding realtime_ext to mysql/realtime/extensions_table == Binding voicemail to mysql/realtime/voicemail_users == Binding sipfriends to mysql/realtime/sip_buddies Asterisk CVS-HEAD-01/23/05-19:38:48, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer [EMAIL PROTECTED] = Asterisk Event Logger Started /var/log/asterisk/event_log == Manager registered action Ping == Manager registered action Events == Manager registered action Logoff == Manager registered action Hangup == Manager registered action Status == Manager registered action Setvar == Manager registered action Getvar == 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 == Manager registered action ListCommands Asterisk Management interface listening on port 5038 == RTP Allocating from port range 1 - 2 Asterisk PBX Core Initializing Registering 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' [Progress] == Registered application 'Progress' [ResetCDR] == Registered application 'ResetCDR' [ResponseTimeout] == Registered application 'ResponseTimeout' [Ringing] == Registered application 'Ringing' [SayNumber] == Registered application 'SayNumber' [SayDigits] == Registered application 'SayDigits' [SayAlpha] == Registered application 'SayAlpha' [SayPhonetic] == Registered application 'SayPhonetic' [SetAccount] == Registered application 'SetAccount' [SetAMAFlags] == Registered application 'SetAMAFlags' [SetGlobalVar] == Registered application 'SetGlobalVar' [SetLanguage] == Registered application 'SetLanguage' [SetVar] == Registered application 'SetVar' [ImportVar] == Registered application 'ImportVar' [StripMSD] == Registered application 'StripMSD' [Suffix] == Registered application 'Suffix' [Wait] == Registered application 'Wait' [WaitExten] == Registered application 'WaitExten' Asterisk Dynamic Loader Starting: [chan_modem.so] = (Generic Voice Modem Driver) [res_musiconhold.so] = (Music On Hold Resource) == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' Junk at the beginning 49443302 Warning, flexibel rate not heavily tested! Junk at the beginning 49443303 Warning, flexibel rate not heavily tested! [res_adsi.so] = (ADSI Resource) [res_features.so] = (Call Parking Resource) == Registered application 'ParkedCall' == Registered application 'Park' == Manager registered action ParkedCalls [res_crypto.so] = (Cryptographic Digital Signatures) -- Loaded PUBLIC key 'iaxtel' -- Loaded PUBLIC key 'freeworlddialup' [res_indications.so] = (Indications Configuration) -- Registered indication country 'cl' -- Registered indication country 'tw' -- Registered indication country 'us' -- Registered indication country 'au' -- Registered indication country 'fr' -- Registered indication country 'de' -- Registered indication country 'nl' -- Registered indication country 'uk' -- Registered indication country 'fi' -- Registered indication country 'no' -- Registered indication country 'br' -- Registered indication country 'za' -- Registered indication country 'it' -- Registered indication country 'us-o' -- Registered indication country 'gr' -- Registered indication country 'ru' -- Registered indication country 'nz' -- Setting default indication country to 'us' == Registered application 'Playtones' == Registered application 'StopPlaytones' [res_monitor.so] = (Call Monitoring Resource) == Registered application 'Monitor' == Registered application 'StopMonitor' == Registered application 'ChangeMonitor' == Manager registered action Monitor == Manager registered action StopMonitor == Manager registered
RE: [Asterisk-Users] asterisk one number service
I have it setup to dial my sip phone and my cell at the same time. Is this what you are looking for? If so just add after your dial sip command (sip/123456789zap/g1/6145551212) This works for me -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ashling O'Driscoll Sent: Tuesday, January 11, 2005 5:47 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] asterisk one number service I wonder does anyone have any thoughts or can give me some direction on the following: I have an asterisk testbed environment set up. My task is to make a personal number service available whereby users would be given one number (perhaps a voip number) and this number would enable them to be reached via the pstn, pots, gsm etc Does anyone have ideas where I could start looking at sites to research this or how asterisk might fit into this?. It would be great if someone could maybe point me in the right direction. Thanks, Aisling. ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using SPANDSP for faxes
I installed spandsp on our asterisk server to get faxes. It works however the images are a little off. Sometimes a few pages will be together, pages missing and sentence missing. Is this normal for this program? Any input would be great. Thank You Eric ___ 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] [OT] Adit 600 Question
I think its print config -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Stewart Sent: Thursday, December 09, 2004 8:27 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] [OT] Adit 600 Question Hi, I'm using an Adit 600 Channel Bank with *. I love it and it works really great for my FXS lines. One problem that I have with it (It's really not a problem yet, but it's a potential one) is that I've scoured the manaual for the Adit to see if there's a way to dump out a config file from the bank so in the event of a power and battery failure I don't have to type in the configuration commands, just load a file. Is there a way to get a config from the Adit 600 and load it back in again? Thanks, Jason Stewart ___ 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] ASTCC Question
Hello group I just installed ASTCC and it was VERY easy to get running. I have a question about Pattern Via the web page I click the Routes link and everything makes sense to me but the pattern part. I tried _NXXNXX with the idea that everything would match this. Well it doesn't work... Does anyone have a good how-to? Thanks for all your help!! ___ 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] ASTCC and Pattern question
Hello group I just installed ASTCC and it was VERY easy to get running. I have a question about Pattern Via the web page I click the Routes link and everything makes sense to me but the pattern part. I tried _NXXNXX with the idea that everything would match this. Well it doesn't work... Does anyone have a good how-to? Thanks for all your help!! ___ 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] ASTCC and Pattern question
Sorry for the double post!!! Not sure what happen! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Hall Sent: Tuesday, November 30, 2004 10:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] ASTCC and Pattern question Hello group I just installed ASTCC and it was VERY easy to get running. I have a question about Pattern Via the web page I click the Routes link and everything makes sense to me but the pattern part. I tried _NXXNXX with the idea that everything would match this. Well it doesn't work... Does anyone have a good how-to? Thanks for all your help!! ___ 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] chan_capi on 2.6 - impossible?
That Did it Thank You.. Now I know what works I can start learning why!!! I have the Redex Coach so its just time Thanks again to the list!!! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Sent: Tuesday, November 30, 2004 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] chan_capi on 2.6 - impossible? On Tue, 2004-11-30 at 14:19 +0100, Tomasz Chmielewski wrote: [snip] Before I go investigating - is it possible to compile chan_capi on 2.6 kernels? [snip] Yes it is possible to compile chan_capi on 2.6. Use kernel 2.6.9 or later due to capi, eicon fixes and additions. Afaik chan_capi only works with the stable branch (1.0.x) of asterisk. So make sure you first have zaptel, libpri and asterisk (all version 1.0.x) installed before building/installing chan_capi. Regards, Patrick ___ 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] Spandsp and Asterisk
Does anyone have an update patch file to get Spandsp installed? I'm running asterisk CVS-HEAD-11/19/04-21:53:37 on redhat 9.0 I installed spandsp-0.0.2 when runnig the patch I get patching file MakefileHunk #1 FAILED at 41.Hunk #2 FAILED at 69.2 out of 2 hunks FAILED -- saving rejects to file Makefile.rej ___ 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] Spandsp and Asterisk
I did that [EMAIL PROTECTED] apps]# patch Makefile.patch patching file Makefile Hunk #1 succeeded at 52 with fuzz 2 (offset 11 lines). Hunk #2 succeeded at 88 with fuzz 2 (offset 19 lines). When back to the top-level and did a make I get this make[1]: *** [app_rxfax.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/apps' make: *** [subdirs] Error 1 [EMAIL PROTECTED] asterisk]# -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Seth Remington Sent: Tuesday, November 23, 2004 9:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Spandsp and Asterisk On Tue, 2004-11-23 at 09:00, Eric Hall wrote: Does anyone have an update patch file to get Spandsp installed? I'm running asterisk CVS-HEAD-11/19/04-21:53:37 on redhat 9.0 I installed spandsp-0.0.2 when runnig the patch I get patching file Makefile Hunk #1 FAILED at 41. Hunk #2 FAILED at 69. 2 out of 2 hunks FAILED -- saving rejects to file Makefile.rej Make sure you are trying to patch the Makefile in the apps directory, not the top-level Makefile. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] Spandsp and Asterisk
Still getting errors make[1]: Entering directory `/usr/src/asterisk/apps' Makefile:103: warning: overriding commands for target `app_rxfax.so' Makefile:85: warning: ignoring old commands for target `app_rxfax.so' Makefile:106: warning: overriding commands for target `app_rxfax.o' Makefile:88: warning: ignoring old commands for target `app_rxfax.o' Makefile:109: warning: overriding commands for target `app_txfax.so' Makefile:91: warning: ignoring old commands for target `app_txfax.so' Makefile:112: warning: overriding commands for target `app_txfax.o' Makefile:94: warning: ignoring old commands for target `app_txfax.o' Makefile:115: warning: overriding commands for target `app_dtmftotext.so' Makefile:97: warning: ignoring old commands for target `app_dtmftotext.so' Makefile:118: warning: overriding commands for target `app_dtmftotexto' Makefile:100: warning: ignoring old commands for target `app_dtmftotexto' make[1]: *** [app_rxfax.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/apps' make: *** [subdirs] Error 1 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peer Oliver Schmidt Sent: Tuesday, November 23, 2004 10:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Spandsp and Asterisk Eric Hall wrote: When back to the top-level and did a make I get this make[1]: *** [app_rxfax.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/apps' make: *** [subdirs] Error 1 [EMAIL PROTECTED] asterisk]# I just fought a battle with spandsp/rxfax and won. My winning strategy can be found at http://www.voip-info.org/tiki-index.php?page=Asterisk%20spandsp hth rgds pos ___ 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] Little off topic
I'm going to get 2 T100P cards. One for our Asterisk server and one for the HylaFax Server. Will this work? My next question is can I have Asterisk detect fax tone and route the call to an extension. You call 555-1212 and it's a voice call it goes to his SIP phone. If it's a fax route call to 555-1213. Thanks for your great help -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Friday, November 19, 2004 4:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Little off topic Martin List-Petersen wrote: You can't, the T100P is a unchannelized T1 card. This is 100% wrong. The T100P supports Channelized Voice T-1 (aka CT1) If you want to use it with HylaFax you need either SpanDSP OR an analog port on Asterisk in addition to the T100P. A search of the mailing lists would have told you this. ___ 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] Little off topic
Here is what I was trying to do Telco gives me a PRI It connects to my Asterisk via Wildcard T100P using a 2nd Wildcard T100P I would like to connect with a X-Cable my HylaFax server. My question is will a Wildcard T100P work in a Hylafax server? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Friday, November 19, 2004 9:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Little off topic Martin List-Petersen wrote: Citat Eric Wieling [EMAIL PROTECTED]: Martin List-Petersen wrote: You can't, the T100P is a unchannelized T1 card. This is 100% wrong. The T100P supports Channelized Voice T-1 (aka CT1) If you want to use it with HylaFax you need either SpanDSP OR an analog port on Asterisk in addition to the T100P. Might be that i'm wrong on the unchannelized bit, but i don't see, where the analog port will help you ? The guy wants to do Hylafax directly on a T100P w/o Asterisk or Asterisk as middleware, which i don't see working. SpanDSP on the other side works well, but that is basically a softmodem emulation, something Hylafax can't do. I have not seen any applications for spandsp outside Asterisk, yet. *nod* I mist have missed the part about doing it all within Asterisk. I think I wrote that message before my 2nd cup of coffee. An analog port would allow you to plug a modem into the Asterisk box and run Hylafax using that. T-1- Asterisk - Analog - Modem. ___ 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] PRI NI2 and callerID name
I have a PRI and I'm trying to send Name + Number to Telco. The number is just fine but the name is not being passed. In my sip.conf file I have callerid=Name of person Phone Number Am I missing something or will asterisk not send callerid name out? On a side note sip to sip I see the name and number. ___ 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] Little off topic
Does anyone know if you can use a Wildcard T100P with HylaFAX? I'm trying to setup trunking from our asterisk server to a Fax server. Any help would be great! ___ 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
