[Asterisk-Users] SIP peer vs. user-- how is the USER ever selected?
Here's a real simple question for the Asterisk Venerable and Wise Ones: Help me understand how to name my section for the SIP user, so there is any hope of it ever being used in my sip.conf file. The Wiki says that it tries to match the user name from the From: header in the INVITE packet. If no match is found in sip.conf, it will look thru the PEERS for a matching IP. The From line from an INVITE looks like this, with CALLERID info contained there (my email packages folds the single line into 3): From: MURPHY STEVE ZZ sip:[EMAIL PROTECTED]:5060;transport=udp;isup- oli=23;tag=SDp6an001-voip.ipprovider.net+1+1f2a0f+361a4f13 (and, of couse, the caller id info will vary from one caller to another!) I have tried to have a [ipprovider] type=user host=voip.ipprovider.net ... and [ipprovider_out] type=peer host=voip.ipprovider.net md5secret=abcdefabcdefabcdefabcdefabcdefabcdef ... But, all incoming calls go to the peer definition, as well as the outgoing calls, and I can't authenticate just outgoing calls. What should I rename the [ipprovider] to, so that it will be used for incoming SIP calls? murf -- Steve Murphy murf at e-tools.com Electronic Tools Company ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] send SMS via own SMS Service
I've used Kannel (www.kannel.org) for long time; it works out Asterisk, but it's a good solution sometimes Use de latest CVS version! Best regards hi list, does anyone know how to configure asterisk to be able sending and receiving SMS over my own SMS gateway? it is connected via a serial (V24) cable on the asterisk server. i know that i have to use COM1 and our tel.number; (ServiceCenterAddress) but don't know where toconfigure it. another question: if i can receive some SMS, can asterisk check which first letters are used and then redirect the SMS to a specific mailbox? thanks in advance Andrew -- Alejandro Alfonso Fernndez Dpto de Sistemas. rea Corporativa [EMAIL PROTECTED] http://www.telecyl.com/ Procin 7, Portales 1-2 Edificio Amrica II 28023 Madrid Tfn: 91 452 18 00 - Fax: 91 452 18 08 Juan Garca Hortelano, 43 Edificio Telecyl 47014 Valladolid Tfn: 983 428 200 - Fax: 983 428 223 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi AVM C2
context=capi-in devices=2 This is just one section which two sets of options. You need to define two sections with [...]. See README. Armin no in or out call if i do that (with or without [interfaces]): [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 ;[interfaces] [contr1] blah... [contr2] blah in readme i don't read section [] for each controller can you post your capi.conf plz ? -- Stephane Plichon | HASGARD jabber: [EMAIL PROTECTED] ~ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi AVM C2
On Wed, 14 Dec 2005, stéphane plichon wrote: context=capi-in devices=2 This is just one section which two sets of options. You need to define two sections with [...]. See README. Armin no in or out call if i do that (with or without [interfaces]): [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 ;[interfaces] [contr1] blah... [contr2] blah in readme i don't read section [] for each controller can you post your capi.conf plz ? Send me a verbose log level 5 with 'capi debug', if the following does not work. Armin [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [AVM1] isdnmode=msn incomingmsn=* controller=1 context=capi-in devices=2 [AVM2] isdnmode=msn incomingmsn=* controller=2 context=capi-in devices=2 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] capi.conf - AVM C4 P2P or P2MP
Quick question, I have an AVM C4 connected to a Mehrgeräteanschluss. What should I put into the /etc/isdn/capi.conf? Putting P2P works, but I think is wrong. P2MP does not work (CAPI modules load, but capiinfo says no CAPI installed). Any help is greatly appreciated. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PSTN gateway Asterisk - Virtual Switchboard???
Hi all, My imaginary scenario is the following one: I have a PSTN gateway called TotalControl1000, and I want to know, if connecting it to a visible server with asterisk with public IP I could config it to offer customers services of virtual switchboard. The customer would save the cost of a telephone switchboard and would benefit from the advantages of the use of voice over IP technology. Questions are: - Asterisk could offer same functionality as standard telephone switchboard? Capture of calls. Deflection of calls. Voice Mailbox. Groups of Calls. Automatic answering menus. - If anyone worked with TotalControl1000 PTSN gateway. Could it work with asterisk to develop proposed scene and give VOIP clients make calls to anyone on the public switched telephony network?. Thanks Rafael Ledesma Serrano Administrador de Sistemas Palmanet Networking Services [EMAIL PROTECTED] http://www.palmanet.net Tel +34 957649199 Fax +34 957644926 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi AVM C2
Armin Schindler wrote: On Wed, 14 Dec 2005, stéphane plichon wrote: context=capi-in devices=2 This is just one section which two sets of options. You need to define two sections with [...]. See README. Armin no in or out call if i do that (with or without [interfaces]): [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 ;[interfaces] [contr1] blah... [contr2] blah in readme i don't read section [] for each controller can you post your capi.conf plz ? Send me a verbose log level 5 with 'capi debug', if the following does not work. Armin nothing, tut-tut-tut signal nothing in the log -- Stephane Plichon | HASGARD jabber: [EMAIL PROTECTED] ~ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P and SPANDSP
Hi, I also experienced broken page receiving fax on asterisk + spandsp with Digium TE410P. I also tried diff. versions of spandsp and asterisk, still no luck. I had no issues using the same asterisk + spandsp config with TE110P. Any ideas? At 09:21 AM 11/24/2005, you wrote: Hi, All Does any one has successful experience use te410p and spandsp together? Could they work well with all 120 channels receive/send fax at the same time? My practice is that rxfax always get broken fax page. Help! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Join when empty problem, in queue
Hi all, when calling to a queue that has no agents logged in we expect to hang up, here is the extensions.conf queue configuration. exten= 2020,1,Answer exten= 2020,2,Ringing exten= 2020,3,Wait(2) exten= 2020,4,Queue(gestoria) exten= 2020,5,Hangup But althougth there isn't any agent it let us enter in the queue. Any idea? Here is the queues.conf: [gestoria] musiconhold = default strategy = ringall servicelevel = 40 context = default timeout = 25 retry = 10 ;weight=0 ;wrapuptime=15 maxlen = 0 announce-frequency = 120 periodic-announce-frequency=60 announce-holdtime = no announce-round-seconds = 10 monitor-format = gsm monitor-join = no joinempty = no leavewhenempty = no eventwhencalled = no eventmemberstatusoff = yes reportholdtime = yes memberdelay = 0 timeoutrestart = no member = Agent/1001 member = Agent/1002 We are using the asterisk from svn repository. __ Renovamos el Correo Yahoo! Nuevos servicios, más seguridad http://correo.yahoo.es ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi AVM C2
On Wed, 14 Dec 2005, stéphane plichon wrote: Armin Schindler wrote: On Wed, 14 Dec 2005, stéphane plichon wrote: context=capi-in devices=2 This is just one section which two sets of options. You need to define two sections with [...]. See README. Armin no in or out call if i do that (with or without [interfaces]): [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 ;[interfaces] [contr1] blah... [contr2] blah in readme i don't read section [] for each controller can you post your capi.conf plz ? Send me a verbose log level 5 with 'capi debug', if the following does not work. Armin nothing, tut-tut-tut signal nothing in the log If you don't get anything from capi log with 'capi debug' and 'set verbose 5' then maybe your extensions.conf is wrong. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] capi.conf - AVM C4 P2P or P2MP
On Wed, 14 Dec 2005, Peer Oliver Schmidt wrote: Quick question, I have an AVM C4 connected to a Mehrgeräteanschluss. What should I put into the /etc/isdn/capi.conf? isdnmode=msn Putting P2P works, but I think is wrong. P2MP does not work (CAPI modules load, but capiinfo says no CAPI installed). Is the card loaded with firmware? Correct permissions to /dev/capi20 ? Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P and SPANDSP
TE405p and spandsp works good. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: extension seen as busy when it is not
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Every few days our receptionist's phone will not take calls on one of the extensions. We have an extension 118 going to the first two lines of her phone and extension 101 going to the other. If we try to dial 118 it goes to voicemail even though she is not on the phone. Asterisk is thinking she is not logged on or something because the message in the log stays there is congestions calling that extension: I head the same problem. I didn't solve it, but I know why did it happend to me. I was trying to make videocalls and wen I put line videosupport=yes in sip.conf incoming call goes directly to voicemail. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wildcard TDM2400P: comments
Hi all, we have the need for alot of plain analog lines. We thinking of buying the new Wildcard TDM2400P. Does anybody have any comments with using this card, with any version of Asterisk, (maybe ill make this one Asterisk 1.2.x). I have had some stabilty issues using the 4 TDM400P. What about this new TDM2400P??? thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] format_mp3 uninstalling mpg123
GK How did you install mpg123? If you installed it with the package GK management system, then use the package management system on your GK OS to remove it. If you installed it manually, you'll need to remove GK it manually. GK To actually allow format_mp3 to work you also need to change GK musiconhold.conf from mode=quietmp3 to mode=files. Regarding this issue: anyone knows how to setup streaming music on hold (from webradios) with the new native syntax ? Previously I was using this as suggested by the wiki: radiowazee= mp3:/var/lib/asterisk/sounds/pbx/webradio,http://grace.fast-serv.com:9206/ where in the webradio dir there was just a dummy mp3 file I would like to reproduce this using native mp3 ... any idea ? Tnx ! -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wildcard TDM2400P: comments
Can you define a LOT of pots line? Have you considered a channel bank. Here I'm running an ADTRAN 750. It's painless. You just need 1 T1 interface card for 24 lines. Jacques yusuf wrote: Hi all, we have the need for alot of plain analog lines. We thinking of buying the new Wildcard TDM2400P. Does anybody have any comments with using this card, with any version of Asterisk, (maybe ill make this one Asterisk 1.2.x). I have had some stabilty issues using the 4 TDM400P. What about this new TDM2400P??? thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] capi.conf - AVM C4 P2P or P2MP
Hello Armin, thanks for the quick response. I have an AVM C4 connected to a Mehrgeräteanschluss. What should I put into the /etc/isdn/capi.conf? isdnmode=msn isdnmode=msn is in /etc/asterisk/capi.conf, but what about the /etc/isdn/capi.conf --- the configuration file for the capi modules? Putting P2P works, but I think is wrong. P2MP does not work (CAPI modules load, but capiinfo says no CAPI installed). Is the card loaded with firmware? Correct permissions to /dev/capi20 ? Yes. As said, putting p2p into the /etc/isdn/capi.conf works. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail boxes
HI I am new to asterisk. Could anyone pls tell me how do i create voicemail boxes Best Regards Vinod Yahoo! Shopping Find Great Deals on Holiday Gifts at Yahoo! Shopping ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial multiple destinations
Hey, When users call my phone at the office, I also want my mobile to ring.. This works fine using dial(..), but there may be some problems with the cdr's generated from this. There is only one cdr generated (for the first destination). I need to see if the call is answered by the mobile or by the office phone (different rating). Is this a bug, or is it a feature? Morten Tryfoss ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail boxes
After installing Asterisk, the first thing you'll need to do is add an extension. In the process of adding the extension, you can activate whether you want that extension to have voicemail or not. Have Fun Dakota - Original Message - From: Vinod To: asterisk-users@lists.digium.com Sent: Wednesday, December 14, 2005 6:26 AM Subject: [Asterisk-Users] voicemail boxes HII am new to asterisk.Could anyone pls tell me how do i create voicemail boxes Best RegardsVinod Yahoo! ShoppingFind Great Deals on Holiday Gifts at Yahoo! Shopping ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Join when empty problem, in queue
Hi, try: joinempty = strict Morten - Original Message - From: Xavier Gil [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, December 14, 2005 10:19 AM Subject: [Asterisk-Users] Join when empty problem, in queue Hi all, when calling to a queue that has no agents logged in we expect to hang up, here is the extensions.conf queue configuration. exten= 2020,1,Answer exten= 2020,2,Ringing exten= 2020,3,Wait(2) exten= 2020,4,Queue(gestoria) exten= 2020,5,Hangup But althougth there isn't any agent it let us enter in the queue. Any idea? Here is the queues.conf: [gestoria] musiconhold = default strategy = ringall servicelevel = 40 context = default timeout = 25 retry = 10 ;weight=0 ;wrapuptime=15 maxlen = 0 announce-frequency = 120 periodic-announce-frequency=60 announce-holdtime = no announce-round-seconds = 10 monitor-format = gsm monitor-join = no joinempty = no leavewhenempty = no eventwhencalled = no eventmemberstatusoff = yes reportholdtime = yes memberdelay = 0 timeoutrestart = no member = Agent/1001 member = Agent/1002 We are using the asterisk from svn repository. __ Renovamos el Correo Yahoo! Nuevos servicios, más seguridad http://correo.yahoo.es ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [helpp] Problem in astersik
Hi Guys
After your guies replies now i have changed the machine .But this time i
get little different problem
i made following chnages in sip.conf
[901]
context=fromsip
type=friend
username=901
secret=901
callerid=Test2 901
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=rfc2833
callgroup=3
pickupgroup=3
qualify=1000
;[902]
;context=fromsip
;type=friend
;username=902
;secret=902
;callerid=Test3 902
;host=dynamic
;nat=yes
;canreinvite=no
;disallow=all
;allow=ulaw
;dtmfmode=info
;callgroup=3
;pickupgroup=3
;qualify=1000
in extension.conf
[fromsip]
exten = s,1,Answer( )
exten = _9XX,1,Dial(SIP/${EXTEN},100,tr)
exten = _5XX,1,Dial(SIP/${EXTEN},100,tr)
exten = h,1,Hangup
exten = t,1,Hangup
exten = i,1,Hangup
Now
Asterisk 1.0.9, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
[ BootingDec 14 15:23:05 WARNING[3478]: chan_oss.c:257
sound_thread: Read error on sound device: Resource temporarily
unavailable
.Dec
14 15:23:07 WARNING[3478]: chan_skinny.c:2587 reload_config: Unable to get our
IP address, Skinny disabled
]
Asterisk Ready.
*CLI
Now from xpro lite software after configuring it for my machine when i
try to connect to my machine i am unable to get connection it says
unable to connect contact your network administratot.Althoug i am the
network admin
Plz tell me what to do
Regard
Talat
On Mon, 2005-12-12 at 06:40 -0500, Steven wrote:
/var/log/asterisk/full text file may give you a more specific error.
--
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[Fwd: Re: [Asterisk-Users] Re: [helpp] Problem in astersik]
---BeginMessage---
Hi Guys
After your guies replies now i have changed the machine .But this time i
get little different problem
i made following chnages in sip.conf
[901]
context=fromsip
type=friend
username=901
secret=901
callerid=Test2 901
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=rfc2833
callgroup=3
pickupgroup=3
qualify=1000
;[902]
;context=fromsip
;type=friend
;username=902
;secret=902
;callerid=Test3 902
;host=dynamic
;nat=yes
;canreinvite=no
;disallow=all
;allow=ulaw
;dtmfmode=info
;callgroup=3
;pickupgroup=3
;qualify=1000
in extension.conf
[fromsip]
exten = s,1,Answer( )
exten = _9XX,1,Dial(SIP/${EXTEN},100,tr)
exten = _5XX,1,Dial(SIP/${EXTEN},100,tr)
exten = h,1,Hangup
exten = t,1,Hangup
exten = i,1,Hangup
Now
Asterisk 1.0.9, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
[ BootingDec 14 15:23:05 WARNING[3478]: chan_oss.c:257
sound_thread: Read error on sound device: Resource temporarily
unavailable
.Dec
14 15:23:07 WARNING[3478]: chan_skinny.c:2587 reload_config: Unable to get our
IP address, Skinny disabled
]
Asterisk Ready.
*CLI
Now from xpro lite software after configuring it for my machine when i
try to connect to my machine i am unable to get connection it says
unable to connect contact your network administratot.Althoug i am the
network admin
Plz tell me what to do
Regard
Talat
On Mon, 2005-12-12 at 06:40 -0500, Steven wrote:
/var/log/asterisk/full text file may give you a more specific error.
--
---End Message---
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Re: [Asterisk-Users] voicemail boxes
Hi But that does not create the voice mail boxes. Is there any script which does it as mentioned in this link below http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=3 Regards VinodDakota [EMAIL PROTECTED] wrote: After installing Asterisk, the first thing you'll need to do is add an extension. In the process of adding the extension, you can activate whether you want that extension to have voicemail or not. Have Fun Dakota- Original Message -From:Vinod To: asterisk-users@lists.digium.com Sent: Wednesday, December 14, 2005 6:26AM Subject: [Asterisk-Users] voicemailboxes HII am new to asterisk.Could anyone plstell me how do i create voicemail boxes Best RegardsVinod Yahoo! ShoppingFind Great Deals on Holiday Gifts at Yahoo!Shopping ___--Bandwidth andColocation provided by Easynews.com --Asterisk-Users mailinglistTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Shopping Find Great Deals on Holiday Gifts at Yahoo! Shopping ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] capi.conf - AVM C4 P2P or P2MP
On Wed, 14 Dec 2005, Peer Oliver Schmidt wrote: Hello Armin, thanks for the quick response. I have an AVM C4 connected to a Mehrgeräteanschluss. What should I put into the /etc/isdn/capi.conf? isdnmode=msn isdnmode=msn is in /etc/asterisk/capi.conf, but what about the /etc/isdn/capi.conf --- the configuration file for the capi modules? Putting P2P works, but I think is wrong. P2MP does not work (CAPI modules load, but capiinfo says no CAPI installed). Is the card loaded with firmware? Correct permissions to /dev/capi20 ? Yes. As said, putting p2p into the /etc/isdn/capi.conf works. Sorry, I thought you mean capi.conf of chan_capi. I don't know anything about the AVM stuff... Armin___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial multiple destinations
well, guess that's the way it is(a feature? i don't know)... but you can help ourself with the cmd SetCDRUserField respectively AppendCDRUserField (see http://www.voip-info.org/wiki/view/Asterisk+cmd+SetCDRUserField) for dialing multiple destinations, maybe follow_me could be an interesting patch for you. it'd be nice for confirmation(confirm with # before call is connected, for mobiles where you don't want the mailbox to pickup), though i had no possibility to add it yet... - http://bugs.digium.com/view.php?id=5574 regards christian On Wed, 14 Dec 2005 11:31:17 +0100 Morten Tryfoss [EMAIL PROTECTED] wrote: Hey, When users call my phone at the office, I also want my mobile to ring.. This works fine using dial(..), but there may be some problems with the cdr's generated from this. There is only one cdr generated (for the first destination). I need to see if the call is answered by the mobile or by the office phone (different rating). Is this a bug, or is it a feature? Morten Tryfoss ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [help] problem in astersik
Hi Guys
After your guies replies now i have changed the machine .But this time i
get little different problem
i made following chnages in sip.conf
[901]
context=fromsip
type=friend
username=901
secret=901
callerid=Test2 901
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=rfc2833
callgroup=3
pickupgroup=3
qualify=1000
;[902]
;context=fromsip
;type=friend
;username=902
;secret=902
;callerid=Test3 902
;host=dynamic
;nat=yes
;canreinvite=no
;disallow=all
;allow=ulaw
;dtmfmode=info
;callgroup=3
;pickupgroup=3
;qualify=1000
in extension.conf
[fromsip]
exten = s,1,Answer( )
exten = _9XX,1,Dial(SIP/${EXTEN},100,tr)
exten = _5XX,1,Dial(SIP/${EXTEN},100,tr)
exten = h,1,Hangup
exten = t,1,Hangup
exten = i,1,Hangup
Now
Asterisk 1.0.9, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
[ BootingDec 14 15:23:05 WARNING[3478]: chan_oss.c:257
sound_thread: Read error on sound device: Resource temporarily
unavailable
.Dec
14 15:23:07 WARNING[3478]: chan_skinny.c:2587 reload_config: Unable to get our
IP address, Skinny disabled
]
Asterisk Ready.
*CLI
Now from xpro lite software after configuring it for my machine when i
try to connect to my machine i am unable to get connection it says
unable to connect contact your network administratot.Althoug i am the
network admin
Plz tell me what to do
Regard
Talat
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[Asterisk-Users] Help:asterisk 1.2.1 release compile
Hi all, I downloaded version asterisk 1.2.1 on my Debian. When i did make, i met this error: build_tools/make_version_hinclude/asterisk/version.h.tmp /bin/sh: line 1: build_tools/make_version_h: permisson non accordee make: ** [include/asterisk/version.h] erreur 126. It's not an error of asterisk. But i don't know how to solve it. Many thank for you help. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial multiple destinations
Thanks, This sounds interesting, but it may cause some delay before it rings on my cell..? It is possible to implement a ringall strategy in the patch? Morten - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, December 14, 2005 12:31 PM Subject: Re: [Asterisk-Users] Dial multiple destinations well, guess that's the way it is(a feature? i don't know)... but you can help ourself with the cmd SetCDRUserField respectively AppendCDRUserField (see http://www.voip-info.org/wiki/view/Asterisk+cmd+SetCDRUserField) for dialing multiple destinations, maybe follow_me could be an interesting patch for you. it'd be nice for confirmation(confirm with # before call is connected, for mobiles where you don't want the mailbox to pickup), though i had no possibility to add it yet... - http://bugs.digium.com/view.php?id=5574 regards christian On Wed, 14 Dec 2005 11:31:17 +0100 Morten Tryfoss [EMAIL PROTECTED] wrote: Hey, When users call my phone at the office, I also want my mobile to ring.. This works fine using dial(..), but there may be some problems with the cdr's generated from this. There is only one cdr generated (for the first destination). I need to see if the call is answered by the mobile or by the office phone (different rating). Is this a bug, or is it a feature? Morten Tryfoss ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] quadbri, isnd, netherlands: callerid not working
hello, We installed an asterisk box last weekend dealing with 2 incoming groups of ISDN lines, and some 15 polycom phones. Works quite OK so far. But we have some problems that somehow I cannot resolve so far :-( The most urging issue is getting callerID to work. When I check logfiles, I see lines like 'Activating voice calls from '065MYNUMBER' to '010OFFICE' But when I dump the variables (call Macro dumpvars in [EMAIL PROTECTED], or NoOp in the extension plan), the CALLERID variables just remain empty. The box is an [EMAIL PROTECTED] 1.5 install, with the junghanns bristuff installed (rc8o) and a quadbri card. Configuration has the things I could find for solving callerID issues (immediate=no, usecallerid=yes, hidecallerid=no). The provider is KPN Netherlands. What more can I do to resolve this issue? Thanks in advance, Mark -- They say if you play a Micro$oft CD backwards, you hear satanic messages... That's nothing, because if you play it forwards, it installs Windows! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to find key
Is this normal? Can I ignore this messages? Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key 'MGCP/aaln/[EMAIL PROTECTED]' in family 'cfb' Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key 'MGCP/aaln/[EMAIL PROTECTED]' in family 'cfb' [...] -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] subscription
___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extension seen as busy when it is not
Every few days our receptionist's phone will not take calls on one of the extensions. We have an extension 118 going to the first two lines of her phone and extension 101 going to the other. If we try to dial 118 it goes to voicemail even though she is not on the phone. Asterisk is thinking she is not logged on or something because the message in the log stays there is congestions calling that extension: dialparties.agi: extnum: 118 dialparties.agi: exthascw: 1 dialparties.agi: exthascfb: 0 dialparties.agi: extcfb: dialparties.agi: Extension 118 has call waiting enabled2 dialparties.agi: get_dial_string: extnum=[118] -- dialparties.agi: get dial string 118, SIP/118 -- dialparties.agi: DbSet CALLTRACE/118 to 101 -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial(SIP/101-dc56, SIP/118|25|tTwWr) in new stack == Everyone is busy/congested at this time (1:0/0/1) -- Executing GotoIf(SIP/101-dc56, 0?s-CHANUNAVAIL|1) in new stack -- Executing GotoIf(SIP/101-dc56, 0?s-CHANUNAVAIL|1) in new stack -- Executing NoOp(SIP/101-dc56, Sending to Voicemail box 118) in new stack What can I look at to see why this is happening? I'd start by looking for registration timeout issues associated with the sip phone. Might also check /var/log/asterisk/messages (or where ever your log files are located) to see if there might be some indications. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Exceptionally long queue in SIP Channel
Hi, Started getting a bombardment of these messages on the Asterisk console this morning (20+ a second): Dec 14 10:00:30 WARNING[17006]: channel.c:588 ast_queue_frame: Exceptionally long queue length queuing to SIP/bluecity29-a5cfDec 14 10:00:30 WARNING[17006]: channel.c:603 ast_queue_frame: Unable to write to alert pipe on SIP/bluecity29-a5cf, frametype/subclass 5/0 (qlen = 173842): Resource temporarily unavailable! Had to restart Asterisk to get rid of them. Has anybody seen this before? Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Subscription Storage Location
On 12/14/05, Brian Capouch [EMAIL PROTECTED] wrote: Douglas Garstang wrote: I can't understand why it was implemented this way (lack of design maybe?). Yep, that's it. Asterisk was designed by a bunch of fools who never even gave the first thought to what they were coding up. Yore kinda quick to knock over the china, pardner. It's not as easy sip registrations. Should we assume that the phone is going be OK and isn't going to get confused if we just pickup and start sending state notification that may conflict with where we left off when Asterisk exited or sip reloaded? I agree with you that this is something we should implement, but it's not a trivial matter to do so. Will you make systems and yourself available for testing once it is implemented by the dev team? -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bonded ethernet ports and *
Rich - Even though I mentioned ethernet failover, I might have made it still a little too broad. The linux ethernet bonding module has been around for years, and there are several modes the linux bonding module can use which include: mode=0 (balance-rr) Round-robin policy: Transmit packets in sequential order from the first available slave through the last. This mode provides load balancing and fault tolerance. mode=1 (active-backup) Active-backup policy: Only one slave in the bond is active. A different slave becomes active if, and only if, the active slave fails. The bond's MAC address is externally visible on only one port (network adapter) to avoid confusing the switch. This mode provides fault tolerance. The primary option affects the behavior of this mode. mode=2 (balance-xor) XOR policy: Transmit based on [(source MAC address XOR'd with destination MAC address) modulo slave count]. This selects the same slave for each destination MAC address. This mode provides load balancing and fault tolerance. mode=3 (broadcast) Broadcast policy: transmits everything on all slave interfaces. This mode provides fault tolerance. mode=4 (802.3ad) IEEE 802.3ad Dynamic link aggregation. Creates aggregation groups that share the same speed and duplex settings. Utilizes all slaves in the active aggregator according to the 802.3ad specification. What I was talking about was simply mode 1, active-backup. Some of our past equipment's network interfaces had some issues with link up/down which could only be traced back to the ethernet port itself, so using bonding to use two ports for active/backup failover works very smoothly. Our policy is 500ms mii monitor for link status, and then a wait of 500ms before actually failing over for a total of about 1s of possible down time. This also benefits us as we use redundant switches in our distribution layer so that if one of the switches goes down, it automatically switches over. My question was really more of the bonding module than anything else, and how much more overhead it puts on. Most of the other modes(except 0) typically require trunk ports or special switch setup, since my issues are not bandwidth related, I've stayed away from them. I'd agree that nics are the least concerning, but if you have an extra eth port, and aren't using it for something already, why not make it a failover port.. Best regards, Rolf On 12/13/05 4:14 PM, Rich Adamson [EMAIL PROTECTED] wrote: Hey all - I'm sure this has been done before, but I'm curious about how well it works.. Typically we have all our servers setup for dual fast/gig ethernet failover... I.e. bond0 slaves eth0 and eth1 and fails over between the two. This together with dual p/s and raid1'd(at least) drives provides for a pretty safe solution(aside from building up a second server). So I'm courious thoughts/expectations/issues with doing network failover... Probably is a moot point, but I thought I'd ask. I've done profession network assessments for a large number of companies throughout the US and I've never ever seen bonded nics work as the implementor expected them to work. If you think seriously about how well the underlying OS and drivers function, the length of the code path that must be executed to move packets from the application layer all the way through to the nic card, you'll find that most OS's are pressed very hard to keep a 1 gig interface running at max smoke. Combine that with the overhead of tcp (not udp), latency, and the typical tcp windowing, and its even worse. I'd also be checking exactly how the bonding function works in the primary/backup arrangement as several implementations that I've seen do not handle shared mac addresses very well. That translates into arp table timeout issues that essentially negates the expected benefits (eg, session failures). Could there be some good implementations? Probably, but just haven't seen any persoanlly as yet. From a VoIP perspective, a 100 meg nic interface can (in theory) handle 1,176 simultanous g711 (or about 3,000 g729) conversations. That is significantly greater then what can be handled from a processing perspective (assuming all conversations pass through asterisk code). If all conversations essentially involves canreinvite=yes, a 100 meg nic is still not the bottleneck. Last, the bonding of two nics at the server level _requires_ the associated switch interface to support the exact same bonding algorithm. Historically, that has been a problem for many switch vendors. Short answer... I'd never do it. Long answer... think in terms of high availability systems; the nic card is the least concerning. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and
[Asterisk-Users] '#' (fast foward) and '*' (Rewind) not working in VoicemailMain
Hi, '#' (fast forward) and '*' (Rewind) not working in VoicemailMain with Asterisk 1.2.1 Do I have to do something in the dialplan to make this work? I have '##' set as a blind transfer and '*2' set as a attended transfer in features.conf. Per the Wiki Voicemailmain has the following settings: * *1* Read voicemail messages o *3* Advanced options (with option to reply; introduced in Asterisk CVS Head April 28, 2004 with 'enhanced voicemail') + *1* Reply + *2* Call back(1) + *3* Envelope + *4* Outgoing call(1) o *4* Play previous message o *5* Repeat current message o *6* Play next message o *7* Delete current message o *8* Forward message to another mailbox o *9* Save message in a folder o *** Help; during msg playback: Rewind o *#* Exit; during msg playback: Skip forward * *2* Change folders * *0* Mailbox options o *1* Record your unavailable message o *2* Record your busy message o *3* Record your name o *4* Record your temporary message (new in Asterisk v1.2) o *5* Change your password o *** Return to the main menu * *** Help * *#* Exit * After recording a message (incoming message, busy/unavail greeting, or name) o 1 - Accept o 2 - Review o 3 - Re-record o 0 - Reach operator(1) (not available when recording greetings/name) Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to find key
I think this is normal if you don't have call-forward-busy enabled. They key is deleted when it is disabled and added when enabled. - James Alejandro Vargas wrote: Is this normal? Can I ignore this messages? Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key 'MGCP/aaln/[EMAIL PROTECTED]' in family 'cfb' Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key 'MGCP/aaln/[EMAIL PROTECTED]' in family 'cfb' [...] -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail boxes
Vinod wrote: Is there any script which does it as mentioned in this link below http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=3 A script is no longer needed. Just edit your voicemail.conf file and add a line for each voicemail box. The first time someone drops into voicemail for that user, the voicemail box will be created. Don Pobanz ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MGCP Unable to find key
Although I do not have an answer I changed the title so maybe someone with MGCP experience may notice it. Is this normal? Can I ignore this messages? Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key 'MGCP/aaln/[EMAIL PROTECTED]' in family 'cfb' Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key 'MGCP/aaln/[EMAIL PROTECTED]' in family 'cfb' [...] -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fax and voice
Hello, I wish to configure Hylafax in order to send either fax or voice to Asterisk I've got a TDM400P (1FXS/1FXO) . What' s the best way to check the line to send fax or voice for incoming or outgoing ? Thanks for help H.G ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voicemail boxes
-Original Message- From: Vinod [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 14, 2005 5:26 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] voicemail boxes HI I am new to asterisk. Could anyone pls tell me how do i create voicemail boxes Best Regards Vinod RTFM ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need help with sipura
I need helphow to config sipura 3000send and receive calls please. Thanks-- [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need help with Sipura 3000
Hello all, Please help me with the sipura 3000 how the Asterisk config need send and receive calls from Sipura 3000 What is Asterisk config need to input Thanks [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Video calls (MS Messenger, Tandberg)
Hi, According to http://www.voip-info.org/wiki-Asterisk+video it should be possible to place video calls using asterisk. So far I managed to get both Microsoft Messenger and a video conference system from Tandberg to register with asterisk. Voice calls between both stations work perfectly (using ulaw codec). Video calls fail with asterisk putting the tandberg system on hold (playing Music-on-hold). Despite both clients claim H261/H263 codecs, SDP negotiation results: Capabilities: us - 0xc020e(GSM|ULAW|ALAW|SPEEX|H261|H263), peer - audio=0xe(GSM|ULAW|ALAW)/video 0x0(EMPTY), combined - 0xe(GSM|ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) So no common video codec was negotiated (thus connections is voice-only) Any ideas? Do I need to active the H261/H263 codecs somewhere? I tried forcing theses codecs in sip.conf, but no luck either. regards, Jens SIP/SDP Debug for Sip read: INVITE sip:52800 SIP/2.0 Via: SIP/2.0/UDP 129.247.XXX.XXX:5060;branch=z9hG4bK2375534000-17264122 Max-Forwards: 70 From: 5sip:[EMAIL PROTECTED];epid=82052805FAC6AK;tag=plcm_237552 -17264121 To: sip:52800 Call-ID: 2375519000-17264119 CSeq: 2 INVITE Session-Expires: 90 Supported: timer Contact: sip:129.247.XXX.XXX:5060;transport=udp Content-Type: application/sdp Proxy-Authorization: Digest username=5,realm=asterisk,nonce=6f505027,uri=sip:52800,respo nse=d260786708039bed1a05af94a4a69fb3,algorithm=md5 User-Agent: Polycom VSX 7000A Release 8.0.3 - 06Oct2005 13:49 Content-Length: 990 v=0 o=DLR-KN 1353514857 0 IN IP4 129.247.173.207 s=- c=IN IP4 129.247.XXX.XXX b=AS:384 t=0 0 m=audio 49184 RTP/AVP 99 98 97 102 101 103 9 15 0 8 18 a=rtpmap:99 SIREN14/16000 a=fmtp:99 bitrate=48000 a=rtpmap:98 SIREN14/16000 a=fmtp:98 bitrate=32000 a=rtpmap:97 SIREN14/16000 a=fmtp:97 bitrate=24000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:101 G7221/16000 a=fmtp:101 bitrate=24000 a=rtpmap:103 G7221/16000 a=fmtp:103 bitrate=16000 a=rtpmap:9 G722/8000 a=rtpmap:15 G728/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729A/8000 a=fmtp:18 annexb=no a=sendrecv m=video 49186 RTP/AVP 109 34 96 31 b=TIAS:384000 a=rtpmap:109 H264/9 a=fmtp:109 profile-level-id=42800c; max-mbps=1; max-fs=1792; max-br=775 a=rtpmap:34 H263/9 a=rtpmap:96 H263-1998/9 a=fmtp:96 CIF4=2;CIF=1;QCIF=1;SQCIF=1;F;J;T a=rtp 14 headers, 34 lines Using latest request as basis request Sending to 129.247.XXX.XXX : 5060 (NAT) Found RTP audio format 99 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 102 Found RTP audio format 101 Found RTP audio format 103 Found RTP audio format 9 Found RTP audio format 15 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Peer audio RTP is at port 129.247.XXX.XXX:49184 Found description format SIREN14 Found description format SIREN14 Found description format SIREN14 Found description format G7221 Found description format G7221 Found description format G7221 Found description format G722 Found description format G728 Found description format PCMU Found description format PCMA Found description format G729A Found description format H264 Found description format H263 Found description format H263-1998 Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x50c(ULAW|ALAW|G729A|ILBC)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found user '5' Looking for 52800 in sip_dlrpbx list_route: hop: sip:129.247.XXX.XXX:5060;transport=udp Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 129.247.XXX.XXX:5060;branch=z9hG4bK2375534000-17264122;received=129.247. XXX.XXX;rport=5060 From: 5sip:[EMAIL PROTECTED];epid=82052805FAC6AK;tag=plcm_237552 -17264121 To: sip:52800;tag=as7a4b0ce2 Call-ID: 2375519000-17264119 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hint Priority for Polycom Phones
Title: Re: [Asterisk-Users] Hint Priority for Polycom Phones Hello Doug, I assume you have subscribecontext set in sip.conf right? Also,I have a 601/sidecar and have the hints working fine on the first registration. On my second server registration they are not yet coming through. I am not sure if I can use the hint on the first server registration, and have the server point the hint to an iax connection, which is what I have started to try doing. Greg From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas GarstangSent: Tuesday, December 06, 2005 11:10 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Hint Priority for Polycom Phones Dang. I must be missing something then. I've modified the contacts directory, set bw and bb, can see the buddy on the second appeareance, have tried every imaginable combination of the hint command in extensions.conf and nada! :( The buddy never updates to show busy/not busy. I thought it was interesting too... in the Polycom admin guide it says on page 51 "Notification when a change in monitored status will be available in a subsequent release". That's for SIP version 1.6.x, dated July 2005. Beats the heck out of me how it works when Polycom says it doesn't! Do you phones send SUBSCRIBE messages to Asterisk on boot? Do you see anything if you do a 'sip show subscriptions' for the phone? Doug -Original Message- From: Adam Goryachev [mailto:[EMAIL PROTECTED] Sent: Tue 12/6/2005 9:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Hint Priority for Polycom Phones On Tue, 2005-12-06 at 21:41 -0600, Jerry Jones wrote: Just in the process of figuring this out myself. i do have it working on an IP601 with a sidecar. Here are my notes. On the polycom Create a contact directory entry for the extension you wish to monitor. Yes the contact must match the exten= statement in your dialplan. Note: It must reside within the same context as the last configured button on your telephone. I have a test phone and had to swap my test extension which is in a test context with my office extension which is in the context with my office phones I wanted to monitor. Had to have the test number register on button one and the office number register on button 2.Nope, this isn't needed... I have an IP600 which registers to asteriskon button 1, another asterisk on button2, a third asterisk on button 3and then has a buddy/hint/monitoring on buttons 4 and 5 which areworking against the first asterisk on button 1. Finally I have a speeddial on button 6...Then again, I've not got this working on a polycom IP 300 as yet...Regards,Adam___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] appradius
Hello all. Has anybody works with appradius? where can I find documentation? Regards, Jsalas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as client to PortaBilling
I am trying to register my Asterisk to a Portabilling system. My asterisk registers with no problems, but when i try to send calls to portabilling I get the response: Dec 14 15:23:31 WARNING[420]: chan_sip.c:727 retrans_pkt: Maximum retries exceed ed on call [EMAIL PROTECTED] for seqno 104 (Non-cri tical Request) -- Executing SetCIDName(SIP/101-436a, ) in new stack -- Executing Dial(SIP/101-436a, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] Dec 14 15:23:51 NOTICE[420]: chan_sip.c:7177 handle_response: Failed to authenticate on INVITE to ' sip:[EMAIL PROTECTED];tag=as1e4ccdd7' Has anyone managed to successfully send calls from Asterisk to Porta? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Testing 10.0.0.203 with 10.0.0.0
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... FC4, Asterisk 1.0.9 and SjPhone softphone. On CLI I get this message every 20 sec. # Testing 10.0.0.203 with 10.0.0.0 10.0.0.203 is the IP of softphone and 10.0.0.0 is the network defind in sip.conf. Asterisk server is on 10.0.0.26 address. Why do I get this message? No body knows this one? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Partial PRI pass thru?
I'd recommend the Digium dual port cards - generation 2 card are excellent and the support we receive superb. And it supports Digiums support and development of Asterisk - Sangomas contribution is token if any. Unfortunately I cant speak for the 2nd gen. cards as we haven't used them. The 1st gen. TE4xxP cause us a lot headache. HDLC errors and other strange behaviour wich we - at least until now - don't experience with the Sangomas. We made some not-so-good experiences with the Digium support but that may be not the ususal case. Thanks to the great community we solved much of our problems but still our Digium installations are far from stable. I usually prefer to support smaller companies and it may be true that buying at Digium helps improving Asterisk. Thats why I would even pay a higher price for a Digium card that works just as good for us as the Sangomas. But - at least in our installations - they don't and I can not afford the hours and hours of solving problems anymore. Of course we will test the 2nd gen cards and check if they work better for us if we get the opportunity. Chris ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bonded ethernet ports and *
Rich - Even though I mentioned ethernet failover, I might have made it still a little too broad. The linux ethernet bonding module has been around for years, and there are several modes the linux bonding module can use which include: mode=0 (balance-rr) Round-robin policy: Transmit packets in sequential order from the first available slave through the last. This mode provides load balancing and fault tolerance. mode=1 (active-backup) Active-backup policy: Only one slave in the bond is active. A different slave becomes active if, and only if, the active slave fails. The bond's MAC address is externally visible on only one port (network adapter) to avoid confusing the switch. This mode provides fault tolerance. The primary option affects the behavior of this mode. mode=2 (balance-xor) XOR policy: Transmit based on [(source MAC address XOR'd with destination MAC address) modulo slave count]. This selects the same slave for each destination MAC address. This mode provides load balancing and fault tolerance. mode=3 (broadcast) Broadcast policy: transmits everything on all slave interfaces. This mode provides fault tolerance. mode=4 (802.3ad) IEEE 802.3ad Dynamic link aggregation. Creates aggregation groups that share the same speed and duplex settings. Utilizes all slaves in the active aggregator according to the 802.3ad specification. What I was talking about was simply mode 1, active-backup. Some of our past equipment's network interfaces had some issues with link up/down which could only be traced back to the ethernet port itself, so using bonding to use two ports for active/backup failover works very smoothly. Our policy is 500ms mii monitor for link status, and then a wait of 500ms before actually failing over for a total of about 1s of possible down time. This also benefits us as we use redundant switches in our distribution layer so that if one of the switches goes down, it automatically switches over. My question was really more of the bonding module than anything else, and how much more overhead it puts on. Most of the other modes(except 0) typically require trunk ports or special switch setup, since my issues are not bandwidth related, I've stayed away from them. I'd agree that nics are the least concerning, but if you have an extra eth port, and aren't using it for something already, why not make it a failover port.. Cool... just test the implementation to ensure what you are expecting is truly what happens with no assumptions. The majority of my previous comments were oriented around that thought process and see'ing a large number of system admin's that assume all documentation, etc, is 100% accurate. Typically its not. A fairly common assumption is the failover happens in xxx milliseconds, but due to nic card design (etc) a different MAC address is used in the failover condition. That confuses the hell out of the layer-3 boxes and negates the value of the failover. (All documentation, etc, is correct but actual implementation in this example is limited by the nic card's inability to use a different MAC address from what's programmed into it. There are a large number of current nic cards like that.) I'd suggest that your comment about ...traced back to the ethernet port... and using the failover approach is sort of like saying rebooting the box fixed the problem. No, it bypassed the problem; what was the root cause of the problem? I'd certainly agree with comments relative to high availability and redundancy, and it sounds like you've done the technical research (and probably testing) to validate the implementation. That's excellent, but I can assure you that's not the norm for the majority of implementations that I've seen. (Then again, we are not typically contracted into a business where their network and system resources are working well. ;) As far as the added overhead, I've never attempted to quantify it. But, it shouldn't be all that difficult to measure its impact from a throughput and failover perspective. Best guess: probably insignificant overhead. Rich ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need help with sipura
I need help how to config sipura 3000 send and receive calls please. Go to www.voxilla.com and run their config wizard. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Blind transferred user does not hear phone ring while waiting for phone to be picked up.
Hi,
Please excuse the double post but I am about to report this as a bug and
I want to verify that others are having the same problem. Also I have
seen numerous bugs reported that are not bugs but just misconfiguration,
etc. and I do not want to burden the developers with a frivolus bug
report if the problem is mine. I have found several postings addressing
this issue but no solution. I have done a partial work around but I do
not like the results. Here is the problem - when I blind transfer a user
the transferred user does not here the phone ringing despite adding the
'r' option to the Dial statement (I will provide all of my files in a
moment..). I have also tried the dial statement without the 'r' option
and I get the same results. If I place a the 'm' option in the dial
statement the transferred user does here musiconhold but this also means
that users doing inter office calls hear musiconhold when calling one
another user instead of ringing (thus my work around that is not
desirable). I also am using a macro to handle dialing and voicemail and
perhaps there is a problem here. In my menus I created a separate macro
that does use the 'm' option as it does seem appropriate here. There is
nothing in the CLI output that appears to show a problem so that further
confuses the issue. Here are my files:
extensions.conf
[general]
#include macros.incl
#include incoming-home.incl
#include extensions-home.incl
#include phrase.incl
#include menu.incl
#include outgoing.incl
[globals]
OUTBOUNDTRUNK=Zap/g1
PSTN1=Zap/1
PSTN2=Zap/2
PHONE1=Zap/3
PHONE2=Zap/4
*extensions-hone.incl
[extensions-home]
;Operator queue, Operator Console, and Receptionist Phone
exten = s,1,Answer()
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout(5)
exten = s,4,ResponseTimeout(15)
exten = s,5,Queue(extensions-home|tr|||20)
exten = s,6,Goto(mainmenu,s,1)
include = mainmenu
;Office Personnel
exten = _30[0-1],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _31[0-4],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _399,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _40[0-1],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _499,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
;Spa Personnel
exten = _500,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _51[0-2],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _590,1,Macro(novmail,${EXTEN},ZAP/3)
;Voicemail Main
exten = 800,1,Answer
exten = 800,2,VoicemailMain(@extensions-home)
;Agent Login
exten = 801,1,AgentCallbackLogin(||@extensions-home)
;Recording Interface
exten = 820,1,Goto(phrase-menu,s,1)
;Voice Conferencing
exten = _85X,1,Answer
exten = _85X,2,MeetMe(${EXTEN})
;Music on Hold
exten = 870,1,Answer
exten = 870,2,SetMusicOnHold(default)
exten = 870,3,WaitMusicOnHold(420)
exten = 870,4,Hangup
macros.incl
[macro-stdexten]
exten = s,1,Set(DYNAMIC_FEATURES=automon)
exten = s,2,Dial(${ARG2},20,Ttrw)
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail([EMAIL PROTECTED])
exten = s-NOANSWER,2,Playback(thank-you-for-callinggoodbye)
exten = s-NOANSWER,3,Hangup
exten = s-BUSY,1,Voicemail([EMAIL PROTECTED])
exten = s-BUSY,2,Playback(thank-you-for-callinggoodbye)
exten = s-BUSY,3,Hangup
exten = s-CHANUNAVAIL,1,Voicemail([EMAIL PROTECTED])
exten = s-CHANUNAVAIL,2,Playback(thank-you-for-callinggoodbye)
exten = s-CHANUNAVAIL,3,Hangup
exten = _s-.,1,Goto(s-NOANSWER,1)
[macro-menuexten]
exten = s,1,Set(DYNAMIC_FEATURES=automon)
exten = s,2,Dial(${ARG2},20,Ttmw)
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail([EMAIL PROTECTED])
exten = s-NOANSWER,2,Playback(thank-you-for-callinggoodbye)
exten = s-NOANSWER,3,Hangup
exten = s-BUSY,1,Voicemail([EMAIL PROTECTED])
exten = s-BUSY,2,Playback(thank-you-for-callinggoodbye)
exten = s-BUSY,3,Hangup
exten = s-CHANUNAVAIL,1,Voicemail([EMAIL PROTECTED])
exten = s-CHANUNAVAIL,2,Playback(thank-you-for-callinggoodbye)
exten = s-CHANUNAVAIL,3,Hangup
exten = _s-.,1,Goto(s-NOANSWER,1)
[macro-novmail]
exten = s,1,Dial(${ARG2},20,Ttrw)
exten = s,2,Playback(thank-you-for-callinggoodbye)
exten = s,3,Hangup
exten = s,102,Playback(thank-you-for-callinggoodbye)
exten = s,103,Hangup
menu.incl
[mainmenu]
exten = s,1,Answer()
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout(5)
exten = s,4,ResponseTimeout(15)
exten = s,5,Background(custom/welcome-main)
exten = 2,1,Goto(spa,s,1)
exten = 3,1,Goto(ageless,s,1)
exten = 4,1,Directory(extensions-home,extensions-home,f)
exten = 5,1,Directory(extensions-home,extensions-home)
exten = t,1,Playback(please-try-again)
exten = t,2,Goto(mainmenu,s,1)
exten = i,1,Playback(pbx-invalid)
exten = i,2,Goto(mainmenu,s,1)
exten = 0,1,Goto(operator,s,1)
[operator]
exten = s,1,Answer()
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout(5)
exten = s,4,ResponseTimeout(15)
exten = s,5,Background(custom/operator)
exten = s,6,Macro(menuexten,300,SIP/300)
exten = t,1,Playback(please-try-again)
exten
RE: [Asterisk-Users] Partial PRI pass thru?
\ I'd recommend the Digium dual port cards - generation 2 card are excellent and the support we receive superb. And it supports Digiums support and development of Asterisk - Sangomas contribution is token if any. Unfortunately I cant speak for the 2nd gen. cards as we haven't used them. The 1st gen. TE4xxP cause us a lot headache. HDLC errors and other strange behaviour wich we - at least until now - don't experience with the Sangomas. We made some not-so-good experiences with the Digium support but that may be not the ususal case. Thanks to the great community we solved much of our problems but still our Digium installations are far from stable. I usually prefer to support smaller companies and it may be true that buying at Digium helps improving Asterisk. Thats why I would even pay a higher price for a Digium card that works just as good for us as the Sangomas. But - at least in our installations - they don't and I can not afford the hours and hours of solving problems anymore. Of course we will test the 2nd gen cards and check if they work better for us if we get the opportunity. Chris This has not been my experience at all with any of Digium's T1/E1 products. Analog boards are a different story but I am not sure of another solution for these cards. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_addon_mysql can't find libmysqlclient.so
Hi! Dec 13 12:19:29 WARNING[4112]: loader.c:325 __load_resource: libmysqlclient.so.15: cannot open shared object file: No such file or directory Mine is in /usr/lib/libmysqlclient.so, so how about just adding a symlink? Philipp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to find key
Hi! Is this normal? Can I ignore this messages? Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key 'MGCP/aaln/[EMAIL PROTECTED]' in family 'cfb' Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key 'MGCP/aaln/[EMAIL PROTECTED]' in family 'cfb' [...] Look like you have enabled call-forward-on-busy... so try to disable that...? Philipp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best way to automatically stop and start Asterisk nightly
Hi, I am planning on restarting asterisk nightly as I seem to be experiencing some sort of memory leak (Asterisk slows down over time). I have reviewed the Asterisk suggestions for management and one item is the routine rebooting of Asterisk. Since I have Asterisk 1.2.1 what is the recommend way to implement an automatic stop and start of asterisk (there are changes in 1.2 with reload and restart) and is this enough or should I restart the hardware as well?? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Gateway crashes when transferring to external lines
I'm using an Asterisk system with a Zultys MX250 as a media gateway for our PSTN. When I configured this originally, I couldn't make or receive any outside calls - the MX250 would actually crash and restart itself. Very bad. As I was troubleshooting, I found a tip in the Asterisk wiki that recommended disabling reinvites for buggy gateway. So, I tried that and it fixed the problem. But now I'm seeing some a similar issue when I receive an outside call to an Asterisk extension, then that extension does a manual transfer to another outside number. The receiving line will ring, but as soon as it is answered the MX250 will crash. Is it possible that Asterisk is attempting a reinvite when I do this, or could it be a completely unrelated problem? Here's the sip.conf entry that I'm using for the MX: [mx250] context=mx250incoming type=friend host=192.168.1.10 dtmfmode=rfc2833 canreinvite=no disallow=all allow=ulaw - .Dustin Wenz ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Blind transferred user does not hear phone ring while waiting for phone to be picked up.
OOPs I forgot to mention I am using Asterisk 1.2.1 and I had the same
problem with 1.0.9 and 1.2.0
Chuck Bunn wrote:
Hi,
Please excuse the double post but I am about to report this as a bug
and I want to verify that others are having the same problem. Also I
have seen numerous bugs reported that are not bugs but just
misconfiguration, etc. and I do not want to burden the developers with
a frivolus bug report if the problem is mine. I have found several
postings addressing this issue but no solution. I have done a partial
work around but I do not like the results. Here is the problem - when
I blind transfer a user the transferred user does not here the phone
ringing despite adding the 'r' option to the Dial statement (I will
provide all of my files in a moment..). I have also tried the dial
statement without the 'r' option and I get the same results. If I
place a the 'm' option in the dial statement the transferred user does
here musiconhold but this also means that users doing inter office
calls hear musiconhold when calling one another user instead of
ringing (thus my work around that is not desirable). I also am using a
macro to handle dialing and voicemail and perhaps there is a problem
here. In my menus I created a separate macro that does use the 'm'
option as it does seem appropriate here. There is nothing in the CLI
output that appears to show a problem so that further confuses the
issue. Here are my files:
extensions.conf
[general]
#include macros.incl
#include incoming-home.incl
#include extensions-home.incl
#include phrase.incl
#include menu.incl
#include outgoing.incl
[globals]
OUTBOUNDTRUNK=Zap/g1
PSTN1=Zap/1
PSTN2=Zap/2
PHONE1=Zap/3
PHONE2=Zap/4
*extensions-hone.incl
[extensions-home]
;Operator queue, Operator Console, and Receptionist Phone
exten = s,1,Answer()
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout(5)
exten = s,4,ResponseTimeout(15)
exten = s,5,Queue(extensions-home|tr|||20)
exten = s,6,Goto(mainmenu,s,1)
include = mainmenu
;Office Personnel
exten = _30[0-1],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _31[0-4],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _399,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _40[0-1],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _499,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
;Spa Personnel
exten = _500,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _51[0-2],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _590,1,Macro(novmail,${EXTEN},ZAP/3)
;Voicemail Main
exten = 800,1,Answer
exten = 800,2,VoicemailMain(@extensions-home)
;Agent Login
exten = 801,1,AgentCallbackLogin(||@extensions-home)
;Recording Interface
exten = 820,1,Goto(phrase-menu,s,1)
;Voice Conferencing
exten = _85X,1,Answer
exten = _85X,2,MeetMe(${EXTEN})
;Music on Hold
exten = 870,1,Answer
exten = 870,2,SetMusicOnHold(default)
exten = 870,3,WaitMusicOnHold(420)
exten = 870,4,Hangup
macros.incl
[macro-stdexten]
exten = s,1,Set(DYNAMIC_FEATURES=automon)
exten = s,2,Dial(${ARG2},20,Ttrw)
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail([EMAIL PROTECTED])
exten = s-NOANSWER,2,Playback(thank-you-for-callinggoodbye)
exten = s-NOANSWER,3,Hangup
exten = s-BUSY,1,Voicemail([EMAIL PROTECTED])
exten = s-BUSY,2,Playback(thank-you-for-callinggoodbye)
exten = s-BUSY,3,Hangup
exten = s-CHANUNAVAIL,1,Voicemail([EMAIL PROTECTED])
exten = s-CHANUNAVAIL,2,Playback(thank-you-for-callinggoodbye)
exten = s-CHANUNAVAIL,3,Hangup
exten = _s-.,1,Goto(s-NOANSWER,1)
[macro-menuexten]
exten = s,1,Set(DYNAMIC_FEATURES=automon)
exten = s,2,Dial(${ARG2},20,Ttmw)
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail([EMAIL PROTECTED])
exten = s-NOANSWER,2,Playback(thank-you-for-callinggoodbye)
exten = s-NOANSWER,3,Hangup
exten = s-BUSY,1,Voicemail([EMAIL PROTECTED])
exten = s-BUSY,2,Playback(thank-you-for-callinggoodbye)
exten = s-BUSY,3,Hangup
exten = s-CHANUNAVAIL,1,Voicemail([EMAIL PROTECTED])
exten = s-CHANUNAVAIL,2,Playback(thank-you-for-callinggoodbye)
exten = s-CHANUNAVAIL,3,Hangup
exten = _s-.,1,Goto(s-NOANSWER,1)
[macro-novmail]
exten = s,1,Dial(${ARG2},20,Ttrw)
exten = s,2,Playback(thank-you-for-callinggoodbye)
exten = s,3,Hangup
exten = s,102,Playback(thank-you-for-callinggoodbye)
exten = s,103,Hangup
menu.incl
[mainmenu]
exten = s,1,Answer()
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout(5)
exten = s,4,ResponseTimeout(15)
exten = s,5,Background(custom/welcome-main)
exten = 2,1,Goto(spa,s,1)
exten = 3,1,Goto(ageless,s,1)
exten = 4,1,Directory(extensions-home,extensions-home,f)
exten = 5,1,Directory(extensions-home,extensions-home)
exten = t,1,Playback(please-try-again)
exten = t,2,Goto(mainmenu,s,1)
exten = i,1,Playback(pbx-invalid)
exten = i,2,Goto(mainmenu,s,1)
exten = 0,1,Goto(operator,s,1)
[operator]
exten = s,1,Answer()
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout(5)
exten = s,4,ResponseTimeout(15)
Re: [Asterisk-Users] IP Phone Recommendation
Kris, I highly recommend the snom 320. Very easy to configure, and very easy to setup line appearances. As has already been mentioned, the idea of lines is a bit dated. For more information, read this: http://forums.digium.com/viewtopic.php?t=891. Sean Duracom ISP Lists wrote: We are going to replace our existing PBX system with an Asterisks box. I have 7 phone lines that come in and I need to get a phone that would support that many lines at minimum. Do you guys recommend any phones that you have used that work well. Kris ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hardware echo cancellation for TDM card
Hi Just checking, Is there any hardware echo cancellation card available for the digium TDM400P card I read the archives and could not find any. I think I need the TDM2400 card for this Thanks Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Asterisk to Avaya IP Office
Title: Re: [Asterisk-Users] TDM01B vs. X100P I fixed this by making the extensions on the asterisk box DIDs . So when the IPO passed the 3 digit extension number then the asterisk box looked at that and sent it on to the extension. Asterisk seemed to see all H323 calls as outside calls inbound (incoming calls) so I treated it that way . From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giles Coochey Sent: Friday, November 04, 2005 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] RE: Asterisk to Avaya IP Office Hi Chris, I have this more or less working, I can dial the IP Office extensions directly from Asterisk. How do I configure being able to dial Asterisk VoIP extensions directly from IP Office phones?? Currently I have a short code to dial Asterisk with a prompt for an extension, but it means that external callers can't seamlessly call VoIP extensions... Any ideas? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Clauss, Chris Sent: 31 October 2005 14:00 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] RE: Asterisk to Avaya IP Office On the IP Office, try making sure that fast start is off on the h.323 trunk links. Also, look in Monitor on the IP Office, see what errors are coming up. Kind regards, Chris Clauss Avaya Certified Expert; Cisco CCDA; Microsoft MCSE Strategic Products and Services AVAYA 2003 Business Partner of the Year 3 Wing Drive Cedar Knolls, NJ 07927 973-359-8557 Voice 973-944-5800 Fax From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Rahn Sent: Sunday, October 30, 2005 10:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Asterisk to Avaya IP Office has anyone had any luck connecting * to IPOFFICE via h323 trunk I can call * from IPO but don't get a connection the other way the * box is sending packets to the ipoffice I see the Call hit the IPOFFICE as an H323 event but it doesn't actually connect a call thanks NOTICE: This e-mail message and all attachments transmitted with it may contain legally privileged and confidential information intended solely for the use of the addressee. If the reader of this message is not the intended recipient, you are hereby notified that any reading, dissemination, distribution, copying, or other use of this message or its attachments, hyperlinks, or any other files of any kind is strictly prohibited. If you have received this message in error, please notify the sender immediately by telephone (+44-1865-265500) or by a reply to this electronic mail message and delete this message and all copies and backups thereof. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Join when empty problem, in queue
in queues.conf change joinempty = no leavewhenempty = no to joinempty = strict leavewhenempty = strict From: Xavier Gil [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Join when empty problem, in queue Date: Wed, 14 Dec 2005 10:19:08 +0100 (CET) MIME-Version: 1.0 Hi all, when calling to a queue that has no agents logged in we expect to hang up, here is the extensions.conf queue configuration. exten= 2020,1,Answer exten= 2020,2,Ringing exten= 2020,3,Wait(2) exten= 2020,4,Queue(gestoria) exten= 2020,5,Hangup But althougth there isn't any agent it let us enter in the queue. Any idea? Here is the queues.conf: [gestoria] musiconhold = default strategy = ringall servicelevel = 40 context = default timeout = 25 retry = 10 ;weight=0 ;wrapuptime=15 maxlen = 0 announce-frequency = 120 periodic-announce-frequency=60 announce-holdtime = no announce-round-seconds = 10 monitor-format = gsm monitor-join = no joinempty = no leavewhenempty = no eventwhencalled = no eventmemberstatusoff = yes reportholdtime = yes memberdelay = 0 timeoutrestart = no member = Agent/1001 member = Agent/1002 We are using the asterisk from svn repository. __ Renovamos el Correo Yahoo! Nuevos servicios, más seguridad http://correo.yahoo.es ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T1 to T1 dialout problem
I need a few minutes of time to work out a dial out problem. I'm willing to pay for your time. What I have is a system that connect 2 external VMS systems to one of two Telco T1's. Mainly the Telco T1's route inbound calls to one of the two external VM systems depending on the DNIS. This parts works correctly. These are connected using TE410P cards using standard em wink start, D4 T1's. The problem is, one External VM systems needs to be able to dial out to one of two Telco T1's. I tried to setup a context that will allow this but it's not working. I'll get congestion and something about context. What should happen: 1. VMS comes off hook and hears dial tone from asterisk. (Problem 1 - EM don't provide dial tone, maybe could play fake one in Background?) 2. VMS dials the telephone number (10 digits), pauses for 2 second, then send a 4 digit billing account code. (A tone comes from Telco when ready for code) 3. Asterisk then routes the call to a ZAP trunk group 7 for all area codes except 714 or 800 or group 3 for 714 800 - Pauses and then sends 4 digit account code. 4. After dialed party answers, the VM dials additional digits to system that was called and asterisk should ignore these 5. Then the VM terminates the call. Fairly simple - huh? I can get you SSH access and will detail more what the problem is when I hear from you Bart [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WIFI Phones
rossi.tek wrote: I'm looking for iax2 wifi phones, do you know where i can buy them? Yes. Nowhere. :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: Linux on treo 650
http://www.engadget.com/entry/1234000497072377/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need help with sipura
Really? Did you try reading classes? On 12/14/05, Talkvoip Telecom Canada [EMAIL PROTECTED] wrote: I need help how to config sipura 3000 send and receive calls please. Thanks -- [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hardware echo cancellation for TDM card
On 12/14/05, Patrick Fortin [EMAIL PROTECTED] wrote: Hi Just checking, Is there any hardware echo cancellation card available for the digium TDM400P card I read the archives and could not find any. I think I need the TDM2400 card for this No. Not at this time. You will need indeed need the TDM2400 series if you'd like hardware echo cans. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hardware echo cancellation for TDM card
Patrick Fortin wrote: Is there any hardware echo cancellation card available for the digium TDM400P card No. Software echo cancellation is fine for small density applications like 4-8 ports, unless you are using a very low performance CPU. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 501 remapping keys
What I really want to be able todo is use the services button or any of the other buttons that serve no purpose right now. I would like to have it start a page (which on my * box is just dialing a particular extension), I have this working on my Polycom 501's using the 3rd line appearance, however I would rather keep that a line appearance. I would also like to be able to use a button to park a call. My users can usually get it right at this point, but they still mess it up far too often. Matt Bill Gibbs wrote: Yeah I just got in a 301 to test and I can configure a key (for example in sip.cfg key.IP_300.2.function.prim=Messages/ and then when I hit the line 2 key it drops me right into VM (since I have that configured too) Still playing around, I noticed that if you get the soft keys (the menu ones under the LCD) then it ALWAYS is that function... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Sent: Friday, December 09, 2005 9:06 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom 501 remapping keys There has been a fair amount of converstaion about this, but I'm not sure anyone really has this working. I had exactly the same problem that the button got remapped to a volume up function. The only button remapping I got working was to map the Transfer button to the # key so that when you hit Transfer it started and Asterisk based transfer. I would love to hear from someone who has this working. Matthew O'Connor [EMAIL PROTECTED] wrote: I've tried to configure the services-key on my Polycom 501 to run a SpeedDial-entry in [MACADRESS]-directory.xml (which would call a asterisk-extension that starts SayUnixTime) but i have not been able to accomplish my goal. Whe configuring the SpeedDial-function in sip.cfg VolUp is started when i press the Services-Key. Also some other possible functions listed under 4.6.1.15 in the SIP 1.6 Administrator Guide fail. Some of them were working with the expected function, some where not giving any response at all but some where starting totally different functions, e.g. configuring Redial as the function starts Settings, function Messages starts Redial, SpeedDialMenu starts VolUp, VolUp starts Line1 :-[ I've seen that other failed as well (http://lists.digium.com/pipermail/asterisk-users/2005-October/130129.ht ml) - anyone ever got this working? Maybe with BootROM 3.0/3.1? Or should i downgrade to 1.5 where there was a ipmid-file for remapping-info...? I'm running Firmware 1.6.2.0041/BootROM 2.6.2.0032 regards Christian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7940 Time Source
Does anyone have an idea of where cisco 7940's get their time from? Up until monday (when our dns server crapped out so we killed it), our phones all had time... now they only show time when they're just rebooted and it's only for a few minutes. Any ideas? Aaron Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Headset Phones?
I am looking for IP phones that are headset and then a phone that clips on your belt. I have been looking at wi-fi phones, but I'm not sure about headset capabilities. I'm using these with *, so compatibility with * is required. Can anyone offer any suggestions? ~kurth -- Kurth Bemis ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WIFI Phones
combine an iaxy and a plain old cordless phone. It works great for me around the workplace. I think there are even four-line, five-handset cordless systems, although you'd need four iaxys as well. Moj rossi.tek wrote: I'm looking for iax2 wifi phones, do you know where i can buy them? Thanks Mario ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WIFI Phones
On Wed, 2005-12-14 at 17:33 +0100, Matt Riddell wrote: rossi.tek wrote: I'm looking for iax2 wifi phones, do you know where i can buy them? Yes. Nowhere. :) not entirely true if you expand your definition :P I have an ipaq which is capable of acting like a soft phone (and it does, although I use a sip client) and it has integrated wifi. There are some really cheap pdas out there now with integrated wifi, in some cases cheaper than some of the wifi phones sold. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] traffic shaping
Hi all, Has anyone a good piece of advice on using traffic shaping embeded with *? As in our case it is not possible to configure it in the ADSL router we would like to implement some kind of bandwidth reservation policy in *. What about using * with 2 network cards betwen the LAN and ADSL router and giving preference to VoIP traffic over web surfing? Thanks, jose ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Background() followed by Read - something wrong?
Hi, I'm using Asterisk 1.2.1, and have been trying to sue the Background() command followed by Read() (for a screening app, but that's beside the point) I did the following s,1,Background(blablabla) s,2,Read(VARIABLE||1) ; accepting only one digit (1 to accept call, anything else to hangup) ... My problem is that when using Background, the following happens: 1) When I wait until the file has finished playing, the VARIABLE is read according to input. Good! 2) If I press a key while the sound file is playing, it seems not to go into the VARIABLE as its value, but go to the extension pressed. NOT good. What I want to do is simply play a file but accept a Read() value while the file is playing. What am I missing? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail boxes
Dakota wrote: After installing Asterisk, the first thing you'll need to do is add an extension. In the process of adding the extension, you can activate whether you want that extension to have voicemail or not. Have Fun Dakota Sounds like someone is using [EMAIL PROTECTED] (or at least AMP): [EMAIL PROTECTED] != Asterisk AMP != Asterisk With Asterisk, you have to edit voicemail.conf. There are some good examples there. There is also a ton of documentation on the subject. Did you google Asterisk voicemail, or some combination of? -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sharing a line w/multiple extensions
I'd like to configure Asterisk so that incoming calls from one POTS line are shared amongst multiple extensions. i.e. If one SIP phone answers the call, another SIP extension phone can pick up and join the conversation. How do I configure this in extensions.conf? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940 Time Source
If you change your config for your phones to use unicast for your SNTP Mode, the time should stay. I had the same problems until I changed it to unicast.On 12/14/05, Aaron Daniel [EMAIL PROTECTED] wrote: Does anyone have an idea of where cisco 7940's get their time from?Upuntil monday (when our dns server crapped out so we killed it), ourphones all had time... now they only show time when they're justrebooted and it's only for a few minutes. Any ideas?Aaron Daniel___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tad Heckaman ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 translation to zap (ISDN) doesn´t work
I thougt i have some problems with ztdummy and removed that # in front of ztdummy in the zaptel Makefile before compiling. But still no change. I even tried it with another Phone, a Planet VIP-150T. Still the same Problem, i don´t hear anything from the SIP Phone on the ISDN Phone, but i hear everything fine the other way. Any Ideas? Thanks a lot for help. regards Klaus Peras Klaus Peras schrieb: Hi, i just figured out, that there is also a problem by going in a conference with the sip phone that runs the g729a codec. Could it be, that i have timing problems? I don´t have digium hardware installed, but i have ztdummy: asterisk3:/etc/asterisk# lsmod | grep ztdummy ztdummy 3748 0 zaptel225540 24 ztdummy,qozap Does anybody have a advice for me? Mit freundlichen Grüßen With kind regards Klaus Peras Klaus Peras schrieb: Hi Asterisk Users, i have a bristuffed-0.2.0-RC8q Asterisk 1.0.9 System running on a Debian 3.1. With a quadbri card installad, wich is running on the bristuff drivers. Everything seems to be fine so far. but now i wanted to use the g.729A Codec. I bought 5 licences and installed them: asterisk3*CLI show g729 0/0 encoders/decoders of 5 licensed channels are currently in use When i do sip to sip calls, everything is working fine (from a snom 190 wich is running with that codec to a sip phone with g.711a), asterisk is translating correct. the output on the CLI is: asterisk3*CLI show g729 1/0 encoders/decoders of 5 licensed channels are currently in use But if i try to call a zap channel from that sip phone (snom 190) wich runs that g729 Codec, i don´t hear anything on the ISDN Phone. the output on the CLI: asterisk3*CLI show g729 1/1 encoders/decoders of 5 licensed channels are currently in use Here is the output of the show channel command for the SIP Channel and the ZAP Channel: asterisk3*CLI show channel SIP/71-d293 -- General -- Name: SIP/71-d293 Type: SIP UniqueID: asterisk-2204-1134137006.49 Caller ID: 30071 DNID Digits: 329 State: Up (6) Rings: 0 NativeFormat: 256 WriteFormat: 256 ReadFormat: 64 1st File Descriptor: 31 Frames in: 7949 Frames out: 7956 Time to Hangup: 0 Elapsed Time: 0h2m39s -- PBX -- Context: default Extension: 329 Priority: 2 Call Group: 0 Pickup Group: 0 Application: Dial Data: Zap/g1/329 Stack: 0 Blocking in: ast_waitfor_nandfds asterisk3*CLI show channel Zap/1-1 -- General -- Name: Zap/1-1 Type: Zap UniqueID: asterisk-2204-1134137006.50 Caller ID: 30071 DNID Digits: 329 State: Up (6) Rings: 0 NativeFormat: 72 WriteFormat: 64 ReadFormat: 256 1st File Descriptor: 13 Frames in: 8255 Frames out: 8246 Time to Hangup: 0 Elapsed Time: 0h0m0s -- PBX -- Context: default Extension: s Priority: 1 Call Group: 0 Pickup Group: 0 Application: Bridged Call Data: SIP/71-d293 Stack: -1 Blocking in: ast_waitfor_nandfds I don´t know what i can do on this problem and would be pleased to get some help. Thank you very much! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Klaus Peras n:Peras;Klaus org:HOB;Netzwerk Support adr;quoted-printable:;;Schwaderm=C3=BChlstrasse 3;Cadolzburg;Bayern;90556;Germany email;internet:[EMAIL PROTECTED] tel;work:09103 / 715 - 329 url:http://www.hob.de version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Headset Phones?
Kurth: The UTStarcom F1000 has an ear bud for it. I believe the Hitachi IPF-5000 also has a jack for headset/ ear bud use. Garrett Smith [EMAIL PROTECTED] 716-250-3408 Direct 716-903-9495 Cell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurth Bemis Sent: Wednesday, December 14, 2005 12:11 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Headset Phones? I am looking for IP phones that are headset and then a phone that clips on your belt. I have been looking at wi-fi phones, but I'm not sure about headset capabilities. I'm using these with *, so compatibility with * is required. Can anyone offer any suggestions? ~kurth -- Kurth Bemis ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WIFI Phones
and that ipaq can do iax2 ?? Guess not cheers klaus trixter aka Bret McDanel schrieb: On Wed, 2005-12-14 at 17:33 +0100, Matt Riddell wrote: rossi.tek wrote: I'm looking for iax2 wifi phones, do you know where i can buy them? Yes. Nowhere. :) not entirely true if you expand your definition :P I have an ipaq which is capable of acting like a soft phone (and it does, although I use a sip client) and it has integrated wifi. There are some really cheap pdas out there now with integrated wifi, in some cases cheaper than some of the wifi phones sold. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Klaus Peras n:Peras;Klaus org:HOB;Netzwerk Support adr;quoted-printable:;;Schwaderm=C3=BChlstrasse 3;Cadolzburg;Bayern;90556;Germany email;internet:[EMAIL PROTECTED] tel;work:09103 / 715 - 329 url:http://www.hob.de version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk + H323 + 723
Hi I am using asterisk 1.2.1, does any one has any luck with asterisk and h323. I want to use the codecs 723 and 729 with it. I am having one way audio issues with oh323 with I receive a call to asterieks through 723 . is there a successful implementation ? regards kani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] traffic shaping
http://www.krisk.org/astlinux/misc/astshape kicks butt hth -Original Message- From: Jose Limeres [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 14, 2005 10:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] traffic shaping Hi all, Has anyone a good piece of advice on using traffic shaping embeded with *? As in our case it is not possible to configure it in the ADSL router we would like to implement some kind of bandwidth reservation policy in *. What about using * with 2 network cards betwen the LAN and ADSL router and giving preference to VoIP traffic over web surfing? Thanks, jose ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sharing a line w/multiple extensions
Look at meetme, also FOP (www.asternic.org) can do that for you. On 12/14/05, Robert La Ferla [EMAIL PROTECTED] wrote: I'd like to configure Asterisk so that incoming calls from one POTS line are shared amongst multiple extensions. i.e. If one SIP phone answers the call, another SIP extension phone can pick up and join the conversation. How do I configure this in extensions.conf? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: {Scanned} Re[2]: [Asterisk-Users] format_mp3 uninstalling mpg123
Alessio Focardi wrote: GK How did you install mpg123? If you installed it with the package GK management system, then use the package management system on your GK OS to remove it. If you installed it manually, you'll need to remove GK it manually. GK To actually allow format_mp3 to work you also need to change GK musiconhold.conf from mode=quietmp3 to mode=files. Regarding this issue: anyone knows how to setup streaming music on hold (from webradios) with the new native syntax ? Previously I was using this as suggested by the wiki: radiowazee= mp3:/var/lib/asterisk/sounds/pbx/webradio,http://grace.fast-serv.com:9206/ where in the webradio dir there was just a dummy mp3 file I would like to reproduce this using native mp3 ... any idea ? Tnx ! google for icecast and you will also need ices. i also found a howto on the wiki it runs great on my 1.0.12 box . hope this help Tom Tom -- This message has been scanned for viruses and dangerous content by Cache Communications, and is believed to be clean. Thank You For Choosing Cache Communications ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sharing a line w/multiple extensions
Let me revise this a little: I'd like to configure Asterisk so an incoming call from one POTS line is shared amongst multiple extensions - both SIP and analog. i.e. If one SIP phone answers the call, another SIP or analog extension phone can pick up and join the conversation. How do I configure this? Is it all in extensions.conf? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] traffic shaping
look into linux advanced routing and traffic control lartc Casey Boone Jose Limeres wrote: Hi all, Has anyone a good piece of advice on using traffic shaping embeded with *? As in our case it is not possible to configure it in the ADSL router we would like to implement some kind of bandwidth reservation policy in *. What about using * with 2 network cards betwen the LAN and ADSL router and giving preference to VoIP traffic over web surfing? Thanks, jose ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] traffic shaping
Jose, I don't know what everyone else uses, but I use wondershaper. It's a bit rough, but it does the job well for what I need ( prioritize voip traffic over everything else ). Google should bring it up for you. Sean Jose Limeres wrote: Hi all, Has anyone a good piece of advice on using traffic shaping embeded with *? As in our case it is not possible to configure it in the ADSL router we would like to implement some kind of bandwidth reservation policy in *. What about using * with 2 network cards betwen the LAN and ADSL router and giving preference to VoIP traffic over web surfing? Thanks, jose ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Headset Phones?
Maybe a bit simplistic... ATA with any cordless phone and walmart headset... Garrett Smith wrote: Kurth: The UTStarcom F1000 has an ear bud for it. I believe the Hitachi IPF-5000 also has a jack for headset/ ear bud use. Garrett Smith [EMAIL PROTECTED] 716-250-3408 Direct 716-903-9495 Cell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurth Bemis Sent: Wednesday, December 14, 2005 12:11 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Headset Phones? I am looking for IP phones that are headset and then a phone that clips on your belt. I have been looking at wi-fi phones, but I'm not sure about headset capabilities. I'm using these with *, so compatibility with * is required. Can anyone offer any suggestions? ~kurth -- Kurth Bemis ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: Linux on treo 650
But can they run Asterisk and create an IAX trunk back to your PBX while running a softphone? I thought not. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Wednesday, December 14, 2005 8:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] OT: Linux on treo 650 http://www.engadget.com/entry/1234000497072377/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HOWOT transfer call from mobile back to extension?
Cheers all. I hope I did not miss this in my quick searching for this, and if I did, apologies, please just a minor scolding as you point me at the URL. But, if I did not miss it, then is there any one out there who has figured-out how to skin this cat. I have a few people's mobile phone numbers in call queues, or in follow-me type set-up's, when we want to transfer that call from the mobile phone back to an extension at the office, how best to do that? I happen to (today) be on a Treo650 on the carrier referred to as Stinkular, if it matters. I do not think I can create a three-way call from the mobile, one leg to original caller, and one leg of new call to PBX then enter extension, then get them on phone, and hang-up, since that will drop both legs. Do I need to get Stinkular to add Centrex to my mobile? :) Any all ideas/suggestions/tips/tricks of any kind are very appreciated. (if this grows to a large enough list of tips/tricks, I will distill post to wiki for us all) Thanks very much, Sjobeck www.voip-info.org/tiki-index.php?page=UserPagesjobeck ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sharing a line w/multiple extensions
also you can ring multiple extensions: Dial(SIP/101SIP/102SIP/103) C F wrote: Look at meetme, also FOP (www.asternic.org) can do that for you. On 12/14/05, Robert La Ferla [EMAIL PROTECTED] wrote: I'd like to configure Asterisk so that incoming calls from one POTS line are shared amongst multiple extensions. i.e. If one SIP phone answers the call, another SIP extension phone can pick up and join the conversation. How do I configure this in extensions.conf? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sharing a line w/multiple extensions
Sean Cook wrote: also you can ring multiple extensions: Dial(SIP/101SIP/102SIP/103) I have that but once one extension picks up, others can't join in. Well, at least when I tried it with mixed SIP and Zap, it didn't work. Maybe all SIP does but I need it to work for all phones SIP and analog (via Zap). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Background() followed by Read - something wrong?
I did the following s,1,Background(blablabla) s,2,Read(VARIABLE||1) ; accepting only one digit (1 to accept call, anything else to hangup) That's not the right approach. Do something like his: [confirmcall] exten = s,1,Background(blablabla) exten = 1,1,Goto(accept_call_context,s,1) exten = t,1,Hangup exten = i,1,Hangup ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] traffic shaping
a simple m0n0wall or pfsense system running on a sub $100 pc makes a GREAT router for this and allows you to use multiple internet connections in concurrency for speed increases other than that, yes, 2 NICs and some creative networking and you're done From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jose LimeresSent: Wednesday, December 14, 2005 12:21 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] traffic shaping Hi all,Has anyone a good piece of advice on using traffic shaping embeded with *? As in our case it is not possible to configure it in the ADSL router we would like to implement some kind of bandwidth reservation policy in *. What about using * with 2 network cards betwen the LAN and ADSL router and giving preference to VoIP traffic over web surfing?Thanks, jose ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ChanIsAvail() and SIP
Hi everyone, I have started trying to use ChanIsAvail() to detect when a phone is in use (on any call) and my results are disappointing. Here are some examples out output to the console followed by the meaning of the return status code based on what I have found in the comments on this page: http://www.voip-info.org/wiki/index.php? page=Asterisk+cmd+ChanIsAvail Test using a real extension (224) that I know is in use at the time of the test. Calling from 227: -- Executing Playback(SIP/227-c825, silence/1) in new stack -- Playing 'silence/1' (language 'en') -- Executing ChanIsAvail(SIP/227-c825, SIP/224|sj) in new stack -- Executing NoOp(SIP/227-c825, SIP/224-08ce|SIP/224|0) in new stack -- Executing Dial(SIP/227-c825, SIP/224|10) in new stack -- Called 224 -- SIP/224-4fc4 is ringing /* 0 AST_DEVICE_UNKNOWN */ Unknown, /* Valid, but unknown state */ Test using a fake extension (333) that doesn't exist and is not defined anywhere. Calling from 227: -- Executing Playback(SIP/227-e4d2, sales) in new stack -- Playing 'sales' (language 'en') -- Executing ChanIsAvail(SIP/227-e4d2, SIP/333|sj) in new stack -- Executing NoOp(SIP/227-e4d2, ||4) in new stack -- Executing Hangup(SIP/227-e4d2, ) in new stack /* 4 AST_DEVICE_INVALID */ Invalid, /* Invalid - not known to Asterisk */ Test using a real extension (206) that is defined, but not registered. Calling from 227: -- Executing Playback(SIP/227-8a76, sales) in new stack -- Playing 'sales' (language 'en') -- Executing ChanIsAvail(SIP/227-8a76, SIP/206|sj) in new stack -- Executing NoOp(SIP/227-8a76, ||5) in new stack -- Executing Hangup(SIP/227-8a76, ) in new stack /* 5 AST_DEVICE_UNAVAILABLE */ Unavailable, /* Unavailable (not registred) */ This all seems to be fine, except for the 1st example where I am testing a known, registered, in use Polycom 501. Does anyone have any idea why Asterisk is returning 0 for that test? Is anyone else using ChanIsAvail() successfully? This is with Asterisk 1.2.0. - Scott ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 1.2.1 Compile Error
I'm trying to compile 1.2.1 for the first time and I am getting a compile error that I can't figure out. I've compiled 1.0.x many times without this issue, although it has been on different boxes. The error is configure: error: termcap support not found, which is odd because when I do a rpm -qa, termcap and libtermcap are both there. Any ideas? Below is the output of the make and the rpm. I'm using Trustix linux, which is the same version I've used on all of my other installs. Thanks. [EMAIL PROTECTED] /usr/src/asterisk/asterisk-1.2.1# make build_tools/make_version_h include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp if cmp -s .cleancount .lastclean ; then echo ; else \ make clean; cp -f .cleancount .lastclean;\ fi build_tools/make_defaults_h defaults.h.tmp if cmp -s defaults.h.tmp defaults.h ; then echo ; else \ mv defaults.h.tmp defaults.h ; \ fi rm -f defaults.h.tmp for x in res channels pbx apps codecs formats agi cdr funcs utils stdtime; do make -C $x depend || exit 1 ; done make[1]: Entering directory `/usr/src/asterisk/asterisk-1.2.1/res' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2.1/res' make[1]: Entering directory `/usr/src/asterisk/asterisk-1.2.1/channels' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2.1/channels' make[1]: Entering directory `/usr/src/asterisk/asterisk-1.2.1/pbx' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2.1/pbx' /bin/sh: line 1: curl-config: command not found make[1]: Entering directory `/usr/src/asterisk/asterisk-1.2.1/apps' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2.1/apps' make[1]: Entering directory `/usr/src/asterisk/asterisk-1.2.1/codecs' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2.1/codecs' make[1]: Entering directory `/usr/src/asterisk/asterisk-1.2.1/formats' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2.1/formats' make[1]: Entering directory `/usr/src/asterisk/asterisk-1.2.1/agi' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2.1/agi' make[1]: Entering directory `/usr/src/asterisk/asterisk-1.2.1/cdr' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2.1/cdr' make[1]: Entering directory `/usr/src/asterisk/asterisk-1.2.1/funcs' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2.1/funcs' make[1]: Entering directory `/usr/src/asterisk/asterisk-1.2.1/utils' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2.1/utils' make[1]: Entering directory `/usr/src/asterisk/asterisk-1.2.1/stdtime' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2.1/stdtime' cd editline unset CFLAGS LIBS test -f config.h || ./configure loading cache ./config.cache checking for gcc... gcc checking whether the C compiler (gcc ) works... yes checking whether the C compiler (gcc ) is a cross-compiler... no checking whether we are using GNU C... yes checking whether gcc accepts -g... yes checking how to run the C preprocessor... gcc -E checking host system type... i586-pc-linux-gnu checking for a BSD compatible install... install checking for ranlib... ranlib checking for ar... /usr/bin/ar checking for tgetent in -ltermcap... no checking for tgetent in -ltinfo... no checking for tgetent in -lcurses... no checking for tgetent in -lncurses... no configure: error: termcap support not found make: *** [editline/libedit.a] Error 1 [EMAIL PROTECTED] /usr/src/asterisk/asterisk-1.2.1# [EMAIL PROTECTED] /usr/src/asterisk/asterisk-1.2.1# [EMAIL PROTECTED] /usr/src/asterisk/asterisk-1.2.1# rpm -qa | grep termcap libtermcap-2.0.8-27tr termcap-11.0.1-7tr [EMAIL PROTECTED] /usr/src/asterisk/asterisk-1.2.1# -- Network stuff you didn't know http://www.networkoblivion.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: 1.2.1 Compile Error
Oops, my bad. 5 minutes after I sent it, I realized I was missing ncurses-devel. Peder @ NetworkOblivion wrote: I'm trying to compile 1.2.1 for the first time and I am getting a compile error that I can't figure out. I've compiled 1.0.x many times without this issue, although it has been on different boxes. The error is configure: error: termcap support not found, which is odd because when I do a rpm -qa, termcap and libtermcap are both there. Any ideas? Below is the output of the make and the rpm. I'm using Trustix linux, which is the same version I've used on all of my other installs. Thanks. [EMAIL PROTECTED] /usr/src/asterisk/asterisk-1.2.1# make build_tools/make_version_h include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp if cmp -s .cleancount .lastclean ; then echo ; else \ make clean; cp -f .cleancount .lastclean;\ fi build_tools/make_defaults_h defaults.h.tmp if cmp -s defaults.h.tmp defaults.h ; then echo ; else \ mv defaults.h.tmp defaults.h ; \ fi rm -f defaults.h.tmp for x in res channels pbx apps codecs formats agi cdr funcs utils stdtime; do make -C $x depend || exit 1 ; done make[1]: Entering directory `/usr/src/asterisk/asterisk-1.2.1/res' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2.1/res' make[1]: Entering directory `/usr/src/asterisk/asterisk-1.2.1/channels' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2.1/channels' make[1]: Entering directory `/usr/src/asterisk/asterisk-1.2.1/pbx' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2.1/pbx' /bin/sh: line 1: curl-config: command not found make[1]: Entering directory `/usr/src/asterisk/asterisk-1.2.1/apps' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2.1/apps' make[1]: Entering directory `/usr/src/asterisk/asterisk-1.2.1/codecs' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2.1/codecs' make[1]: Entering directory `/usr/src/asterisk/asterisk-1.2.1/formats' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2.1/formats' make[1]: Entering directory `/usr/src/asterisk/asterisk-1.2.1/agi' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2.1/agi' make[1]: Entering directory `/usr/src/asterisk/asterisk-1.2.1/cdr' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2.1/cdr' make[1]: Entering directory `/usr/src/asterisk/asterisk-1.2.1/funcs' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2.1/funcs' make[1]: Entering directory `/usr/src/asterisk/asterisk-1.2.1/utils' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2.1/utils' make[1]: Entering directory `/usr/src/asterisk/asterisk-1.2.1/stdtime' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2.1/stdtime' cd editline unset CFLAGS LIBS test -f config.h || ./configure loading cache ./config.cache checking for gcc... gcc checking whether the C compiler (gcc ) works... yes checking whether the C compiler (gcc ) is a cross-compiler... no checking whether we are using GNU C... yes checking whether gcc accepts -g... yes checking how to run the C preprocessor... gcc -E checking host system type... i586-pc-linux-gnu checking for a BSD compatible install... install checking for ranlib... ranlib checking for ar... /usr/bin/ar checking for tgetent in -ltermcap... no checking for tgetent in -ltinfo... no checking for tgetent in -lcurses... no checking for tgetent in -lncurses... no configure: error: termcap support not found make: *** [editline/libedit.a] Error 1 [EMAIL PROTECTED] /usr/src/asterisk/asterisk-1.2.1# [EMAIL PROTECTED] /usr/src/asterisk/asterisk-1.2.1# [EMAIL PROTECTED] /usr/src/asterisk/asterisk-1.2.1# rpm -qa | grep termcap libtermcap-2.0.8-27tr termcap-11.0.1-7tr [EMAIL PROTECTED] /usr/src/asterisk/asterisk-1.2.1# -- Network stuff you didn't know http://www.networkoblivion.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HOWOT transfer call from mobile back to extension?
I use this at work. You have to make sure you use the right T / t options when dialing the mobile, then just use the standard # transfer. I changed ours to ##. - James [EMAIL PROTECTED] wrote: Cheers all. I hope I did not miss this in my quick searching for this, and if I did, apologies, please just a minor scolding as you point me at the URL. But, if I did not miss it, then is there any one out there who has figured-out how to skin this cat. I have a few people's mobile phone numbers in call queues, or in follow-me type set-up's, when we want to transfer that call from the mobile phone back to an extension at the office, how best to do that? I happen to (today) be on a Treo650 on the carrier referred to as Stinkular, if it matters. I do not think I can create a three-way call from the mobile, one leg to original caller, and one leg of new call to PBX then enter extension, then get them on phone, and hang-up, since that will drop both legs. Do I need to get Stinkular to add Centrex to my mobile? :) Any all ideas/suggestions/tips/tricks of any kind are very appreciated. (if this grows to a large enough list of tips/tricks, I will distill post to wiki for us all) Thanks very much, Sjobeck www.voip-info.org/tiki-index.php?page=UserPagesjobeck ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
