[Asterisk-Users] Forward call without answer
Hi, I want to forward a incoming E1 PRI call to an external phone number. Is possible to do this without answering the call first? Is important that the incoming call not be answered until * establish a channel to external number. Thanks, David. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Forward call without answer
David Acacio a écrit : Hi, I want to forward a incoming E1 PRI call to an external phone number. Is possible to do this without answering the call first? Yes. Just make sure you don't use Answer(). ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER+ASTERISK
Hello, I wish to setup this scheme: ser-0.9.4 asterisk-1.2-bêta polycom ip300 phones asterisk 5050-- |firewall+nat|-192.168. ser 5060--- My ip phones use ser as outbound sip proxy and asterisk as sip registrar server. Ser Forward REGISTER requests to asterisk however when a phone try to send an invite message then asterisk send icmp to private ip (host=dynamic in sip.conf) Is it possible to solve this problem ? Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Every SIP on its own FXO
I have two SIP phones and two FXO ports. I would like to be able to define that every SIP phone can call POTS on specific FXO port. How to do that? Right now I have done it on this way. In sip.conf for every sip phone (every user) I have defined different context in extensions.conf. And there, by using includes, I call different external contexts (external_sip1 and external_sip2). It works but it doesn't look nice. Imagine how it would look like if I needed to specify 100 lines. Question. Can I define one context for all SIP phones and then in dial plan chose - for SIP1 dial on FXO1; for SIP2 dial on FXO2... I hope I have make my self clear :)) Thank you for your time. -- Tomislav Parčina Lama d.o.o. www.lama.hr tparcina#lama.hr ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hold Music is breaking up
Dear All, I have installed a Mediatrix 1204 on a client site and I'm sending calls between my Asterisk Server and the clients PBX over a VPN. The call quality is very good when I'm speaking with the staff and there are no breakups. The only problem I'm running into is the hold music is 'choppy'. If I call in over the T1 or using my Softphone there are no such problems. I was wondering if anyone could point me in the right direction. Many Thanks Shad Mortazavi -- Nexus Group Technical Manager n|m Nexus Management Inc ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2.FWDNET.NET not responding?
Hi all, Since a few days my (*) no longer seems to log in to FWD through IAX2. IAX2 DEBUG only shows outbound registration requests, but no replies from FWD: Tx-Frame Retry[003] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 00014ms SCall: 6 DCall: 0 [65.39.205.121:4569] USERNAME: 715749 REFRESH : 60 It apparently doesn't reply to the lag requests either: Tx-Frame Retry[000] -- OSeqno: 004 ISeqno: 000 Type: IAX Subclass: LAGRQ Timestamp: 30013ms SCall: 6 DCall: 0 [65.39.205.121:4569] Tx-Frame Retry[003] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: LAGRQ Timestamp: 10013ms SCall: 6 DCall: 0 [65.39.205.121:4569] Tx-Frame Retry[001] -- OSeqno: 004 ISeqno: 000 Type: IAX Subclass: LAGRQ Timestamp: 30013ms SCall: 6 DCall: 0 [65.39.205.121:4569] Tx-Frame Retry[002] -- OSeqno: 002 ISeqno: 000 Type: IAX Subclass: PING Timestamp: 20013ms SCall: 6 DCall: 0 [65.39.205.121:4569] Tx-Frame Retry[002] -- OSeqno: 003 ISeqno: 000 Type: IAX Subclass: LAGRQ Timestamp: 20016ms SCall: 6 DCall: 0 [65.39.205.121:4569] (Note the increasing retries) It *does* connect to VoipBuster and GoIAX... Is this most likely to be an issue at FWD, my account or something else? TIA! -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Called number (Destination Number)
Hi, I have E1 PRI, When I have an incoming call, how can I know the called number (or the destination number) before answer the call? My provider say that he send it. E1 PRI 900XX 9XXX -- Asterisk It appears in some event under the Asterisk Manager API? Thanks, David ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel: Hz != 1000 causing ztdummy compilation error?
Hi all, When I compile zaptel from today's cvs HEAD on an updated FC4 box it fails with the following message: CC [M] /home/patrick/redhat/BUILD/zaptel/ztdummy.o /home/patrick/redhat/BUILD/zaptel/ztdummy.c:103:2: error: #error ztdummy requires 1000 hz jiffies If I comment out the code causing that error the compilation goes fine but I guess it's there for a reason :) After some googling I tried the following but that did not solve the issue: echo 1000 /proc/sys/dev/rtc/max-user-freq compile still fails echo 1024 /proc/sys/dev/rtc/max-user-freq compile also fails Anyone have a pointer how I solve this error? Thanks and regards, Patrick ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel: Hz != 1000 causing ztdummy compilationerror?
A jiffy is a kernel timer, this affects many thing in the kernel. Linux for as long as I know uses 1000hz. I am really surprised this failed on fc4. Ztdummy uses this as a base for timing, particularly with meetme and tdmoe. If its not high enough quality may be degraded. As for what you tried, you tried to adjust the realtime clock, which is slightly different. What kernel version are you using? -Original Message- From: Patrick[EMAIL PROTECTED] Sent: 11/4/05 2:34:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: [Asterisk-Users] Zaptel: Hz != 1000 causing ztdummy compilationerror? Hi all, When I compile zaptel from today's cvs HEAD on an updated FC4 box it fails with the following message: CC [M] /home/patrick/redhat/BUILD/zaptel/ztdummy.o /home/patrick/redhat/BUILD/zaptel/ztdummy.c:103:2: error: #error ztdummy requires 1000 hz jiffies If I comment out the code causing that error the compilation goes fine but I guess it's there for a reason :) After some googling I tried the following but that did not solve the issue: echo 1000 /proc/sys/dev/rtc/max-user-freq compile still fails echo 1024 /proc/sys/dev/rtc/max-user-freq compile also fails Anyone have a pointer how I solve this error? Thanks and regards, Patrick ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list [Message truncated. Tap Edit-Mark for Download to get remaining portion.] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Called number (Destination Number)
On Fri, November 4, 2005 11:27, David Acacio said: Hi, I have E1 PRI, When I have an incoming call, how can I know the called number (or the destination number) before answer the call? My provider say that he send it. E1 PRI 900XX 9XXX -- Asterisk It appears in some event under the Asterisk Manager API? Thanks, David Log in to the CLI (if not on your main system, use 'asterisk -vr') and watch for the incoming call. If you want to do DID's you may have to put 'immediate=no' and 'overlapdial=yes' in the zap channel definition (zapata.conf) to ensure that it waits to receive the DID info and put it in the appropriate variable. Do not forget to restart after changing zapata.conf. An 'asterisk -rx reload' does NOT reload zapata conf! Once it works, you should see something like this in the CLI: -- Extension '0793429193' in context 'from-pstn' from '0174287114' does not exist. Rejecting call on channel 0/1, span 1 After that all you need to do is define an incoming extension with the correct DID data, like: [ext-did] exten = 0123456788,1,SetVar(FROM_DID=0123456788) ; exten = 0123456788,2,Goto(ext-local,200,1) ; exten = 0123456789,1,SetVar(FROM_DID=0123456789) ; exten = 0123456789,2,Goto(aa_1,s,1); HTH! -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel: Hz != 1000 causing ztdummy compilationerror?
On Tue, 1980-01-01 at 09:11 -0800, Trixter http://www.0xdecafbad.com/ wrote: A jiffy is a kernel timer, this affects many thing in the kernel. Linux for as long as I know uses 1000hz. I am really surprised this failed on fc4. Ztdummy uses this as a base for timing, particularly with meetme and tdmoe. If its not high enough quality may be degraded. As for what you tried, you tried to adjust the realtime clock, which is slightly different. What kernel version are you using? He is probably using 2.6.14 in which you can opt for 100hz (servers), 250hz (mixed) and 1000hz (desktops). I discussed this with Kevin when the 2.6.14 series started, have a look in the archives. CVS ztdummy certainly compiles correctly with 100hz. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Called number (Destination Number)
Hello David, Friday, November 4, 2005, 11:27:51 AM, you wrote: DA Hi, DA I have E1 PRI, When I have an incoming call, how can I know the called DA number (or the destination number) before answer the call? DA My provider say that he send it. DA E1 PRI DA 900XX 9XXX -- Asterisk Maybe you have immediate=yes in zapata.conf and all calls are coming in to s extension. Try to set immediate=no in zapata.conf for the span: you should be able to see on the cli the called number. Then you will have to create the relative extensions in the incoming context ... just s will not work anymore. Hope it helps! -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Every SIP on its own FXO
Assuming SIP users dial 9 to get FXO lines, and callerid's for them are set
as 11 and 12 in sip.conf, and Zap lines are also 1 and 2, you could do it
easily:
exten = 9,1,Dial(Zap/${CALLERIDNUM:1})
You can manipulate dial string further with arithmetic expressions too.
Hope this helps,
Soner
- Original Message -
From: Tomislav Parčina [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, November 04, 2005 10:33 AM
Subject: [Asterisk-Users] Every SIP on its own FXO
I have two SIP phones and two FXO ports. I would like to be able to define
that every SIP phone can call POTS on specific FXO port. How to do that?
Right now I have done it on this way.
In sip.conf for every sip phone (every user) I have defined different
context in extensions.conf. And there, by using includes, I call different
external contexts (external_sip1 and external_sip2). It works but it doesn't
look nice. Imagine how it would look like if I needed to specify 100 lines.
Question.
Can I define one context for all SIP phones and then in dial plan chose -
for SIP1 dial on FXO1; for SIP2 dial on FXO2... I hope I have make my self
clear :))
Thank you for your time.
--
Tomislav Parčina
Lama d.o.o.
www.lama.hr
tparcina#lama.hr
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RE: [Asterisk-Users] Every SIP on its own FXO
Soner,
Thank you! This is what I needed.
Tomislav
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Soner Tari
Sent: 4. studeni 2005 12:28
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Every SIP on its own FXO
Assuming SIP users dial 9 to get FXO lines, and callerid's
for them are set as 11 and 12 in sip.conf, and Zap lines are
also 1 and 2, you could do it
easily:
exten = 9,1,Dial(Zap/${CALLERIDNUM:1})
You can manipulate dial string further with arithmetic
expressions too.
Hope this helps,
Soner
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Re: [Asterisk-Users] Multiple zaphfc cards (for ISDN BRI) in a singlemachine
On 3 Nov 2005, at 11:36, Chris Bagnall wrote: I would suggest using a pair of 4-port cards. The interrupts alone from 5 PCI cards would kill most boxes. There is also an octo-card, but I have no personal experience of that. Hmm... the price is something of an obstacle - given that single BRI cards can be had for sub-£20, justifying £425 on a 4-port card onto which there'd need to be another single BRI anyway might be a challenge. Are there any other options worth considering here? How about a board with 2 PCI buses (e.g. one PCI, one PCI-X) ? I know this isn't what you asked, but, I'd think hard about moving to PRI instead. I find that around at around 8 lines it is generally cheaper (and far easier) to switch to partial E1. All the UK providers will offer you a free install on a 10 channel PRI if you sign up for long enough or commit to enough spend. You can run a partial E1 on a lowpowered 1U server (1Ghz 512Mb), so you could save on the hardware bigtime. Tim. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk connected with CAPI
Hi all, i'm trying to install a EICON DIVA 4BRI (on CentOS 4.1 2.6.9-22.0.1.EL) using latest package from sourceforge (chan_capi-cm-0.6.tar.gz). I have installed divactrl_2.1.tar.gz and untared protocols_all.tar.bz2 in /usr/share/eicon. --- lsmod gives me the following... Module Size Used by divacapi 157937 0 capi 18177 0 capifs 5961 2 capi kernelcapi 44641 2 divacapi,capi md5 4033 1 ipv6 234881 12 lp 12077 0 autofs423237 0 i2c_dev11329 0 i2c_core 22081 1 i2c_dev sunrpc159269 1 microcode 6881 0 button 6481 0 battery 8901 0 ac 4805 0 uhci_hcd 31065 0 parport_pc 24577 0 parport37129 2 lp,parport_pc divas 76345 0 divadidd 13081 2 divacapi,divas e100 41793 0 mii 4673 1 e100 floppy 58481 0 dm_snapshot16901 0 dm_zero 2369 0 dm_mirror 27825 0 ext3 116809 2 jbd71385 1 ext3 dm_mod 56661 6 dm_snapshot,dm_zero,dm_mirror --- Starting divactrl --- [EMAIL PROTECTED] asterisk]# divactrl load -c 1 -f ETSI Start adapter Nr:1 - 'Diva Server 4BRI-8M 2.0 PCI', SN: 7113 ... OK [EMAIL PROTECTED] asterisk]# but the /var/log/asterisk/messages gives me following errors when i try to start asterisk: Nov 4 12:25:45 WARNING[2658]: CAPI not installed, CAPI disabled! Nov 4 12:25:45 WARNING[2658]: chan_capi.so: load_module failed, returning -1 Nov 4 12:25:45 WARNING[2658]: Loading module chan_capi.so failed! Is CAPI really not installed or have i forgotten something? Here my capi.conf and modules.conf ; ; CAPI config ; ; ; general section [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [EICON] controller=1,2,3,4 isdnmode=msn incomingmsn=* softdtmf=on relaxdtmf=on accountcode= context=incoming echocancel=yes devices=2 group=1 ; ; Asterisk configuration file ; ; Module Loader configuration file ; [modules] autoload=yes noload = pbx_gtkconsole.so noload = pbx_kdeconsole.so noload = app_intercom.so load = chan_modem.so load = res_musiconhold.so load = chan_capi.so noload = chan_alsa.so [global] chan_modem.so=yes chan_capi.so=yes thx in advance __ Start your day with Yahoo! - Make it your home page! http://www.yahoo.com/r/hs ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Timestamps in Console?
Might be worth it to read the stuff in /usr/src/asterisk/doc and in particular the README.asterisk.conf file. Lots of other good stuff in that directory as well. (Not much need to read the source now.) Thank you. I don't mind being told to RTFM, if you can point out the FM I'm supposed to R. There is no FM to read. The above reference is to the directory that comes with cvs-head. If you don't use cvs-head, download it anyway and take a look. You misunderstand. I realized you had *already* told me where the information was located. I wasn't *asking* you to point me to the manual, I was *thanking* you for already pointing me to the manual. But thank you again! :) Ops, sorry. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel: Hz != 1000 causing ztdummy compilationerror?
On Tue, 1980-01-01 at 09:11 -0800, Trixter http://www.0xdecafbad.com/ wrote: A jiffy is a kernel timer, this affects many thing in the kernel. Linux for as long as I know uses 1000hz. I am really surprised this failed on fc4. Ztdummy uses this as a base for timing, particularly with meetme and tdmoe. If its not high enough quality may be degraded. As for what you tried, you tried to adjust the realtime clock, which is slightly different. What kernel version are you using? The kernelversion is 2.6.13-1.1526_FC4 on x86_64. I just tried to build the same zaptel on a i686 FC4 box (same kernel version) and it built just fine. The only difference between the two (besides the obvious) is that the x86_64 is booted with no_timer_check to prevent the clock/time from going way faster than it should. When booting the kernel there is a messages that says 8254 timer not connected to IO-APIC. Any suggestions? Regards, Patrick ps. the date on your email said Jan 1, 1980. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk connected with CAPI
On Fri, 2005-11-04 at 03:55 -0800, richard Coco wrote:
Hi all,
i'm trying to install a EICON DIVA 4BRI (on CentOS 4.1
2.6.9-22.0.1.EL) using latest package from
sourceforge (chan_capi-cm-0.6.tar.gz).
I have installed divactrl_2.1.tar.gz and untared
protocols_all.tar.bz2 in /usr/share/eicon.
---
lsmod gives me the following...
Module Size Used by
divacapi 157937 0
capi 18177 0
capifs 5961 2 capi
kernelcapi 44641 2 divacapi,capi
md5 4033 1
ipv6 234881 12
lp 12077 0
autofs423237 0
i2c_dev11329 0
i2c_core 22081 1 i2c_dev
sunrpc159269 1
microcode 6881 0
button 6481 0
battery 8901 0
ac 4805 0
uhci_hcd 31065 0
parport_pc 24577 0
parport37129 2 lp,parport_pc
divas 76345 0
divadidd 13081 2 divacapi,divas
e100 41793 0
mii 4673 1 e100
floppy 58481 0
dm_snapshot16901 0
dm_zero 2369 0
dm_mirror 27825 0
ext3 116809 2
jbd71385 1 ext3
dm_mod 56661 6
dm_snapshot,dm_zero,dm_mirror
---
Starting divactrl
---
[EMAIL PROTECTED] asterisk]# divactrl load -c 1 -f ETSI
Start adapter Nr:1 - 'Diva Server 4BRI-8M 2.0 PCI',
SN: 7113 ... OK
[EMAIL PROTECTED] asterisk]#
but the /var/log/asterisk/messages gives me following
errors when i try to start asterisk:
Nov 4 12:25:45 WARNING[2658]: CAPI not installed,
CAPI disabled!
Nov 4 12:25:45 WARNING[2658]: chan_capi.so:
load_module failed, returning -1
Nov 4 12:25:45 WARNING[2658]: Loading module
chan_capi.so failed!
Is CAPI really not installed or have i forgotten
something? Here my capi.conf and modules.conf
;
; CAPI config
;
;
; general section
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[EICON]
controller=1,2,3,4
isdnmode=msn
incomingmsn=*
softdtmf=on
relaxdtmf=on
accountcode=
context=incoming
echocancel=yes
devices=2
group=1
;
; Asterisk configuration file
;
; Module Loader configuration file
;
[modules]
autoload=yes
noload = pbx_gtkconsole.so
noload = pbx_kdeconsole.so
noload = app_intercom.so
load = chan_modem.so
load = res_musiconhold.so
load = chan_capi.so
noload = chan_alsa.so
[global]
chan_modem.so=yes
chan_capi.so=yes
thx in advance
Iirc CentOS 4.1 uses udev so you have to add the proper udev rules so
the capi devices are properly created Stick the following lines in a
file called e.g. 10-capi.rules and add it to /etc/udev/rules.d:
SYSFS{dev}=68:0, NAME=capi20
SYSFS{dev}=191:[0-9]*,NAME=capi/%n
Use tabs between the , and NAME!
Once you have done this as root do udevstart. Unload all the capi
modules and load them again. With capiinfo you can check if it all went
well (it should give output). If it doesn't work then reboot the box.
Regards,
Patrick
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Re: [Asterisk-Users] IAX2.FWDNET.NET not responding?
Since a few days my (*) no longer seems to log in to FWD through IAX2. IAX2 DEBUG only shows outbound registration requests, but no replies from FWD: Tx-Frame Retry[003] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 00014ms SCall: 6 DCall: 0 [65.39.205.121:4569] USERNAME: 715749 REFRESH : 60 Its a FWD problem that has been going on for a month or so. It also seems to be somewhat intermitant. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk connected with CAPI
richard Coco ha scritto: i'm trying to install a EICON DIVA 4BRI (on CentOS 4.1 2.6.9-22.0.1.EL) using latest package from Maybe you just need to check for the libcapi20 and /dev/capi20 device Anyway you can compile a fresh libcapi from here ftp://ftp.isdn4linux.de/pub/isdn4linux/CVS-Snapshots/isdn4k-utils-CVS-2005-10-28.tar.bz2 tar xjf isdn4k-utils-CVS-2005-10-28.tar.bz2 cd isdn4k* ./configure make clean make install ldconfig Sergio ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hold Music is breaking up
I have installed a Mediatrix 1204 on a client site and I'm sending calls between my Asterisk Server and the clients PBX over a VPN. The call quality is very good when I'm speaking with the staff and there are no breakups. The only problem I'm running into is the hold music is 'choppy'. If I call in over the T1 or using my Softphone there are no such problems. I was wondering if anyone could point me in the right direction. Make sure you are not using silence suppression on your phone and the 1204. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 2 Dial plan questions
I have two questions about a dial plan I'd like to try: 1) How do you put together a dial plan that includes a call transfer that first asked the called person to accept this call press 1, to refuse it press 2? 2) I know how you can switch a dial plan from one behavior to anothr based on who is calling (callerID) but how do you do this based on which line was called? (let's say I have a T1 line with 23 phone numbers) Regards, Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Route call based on CallerID
I need to send calls to a choice of DIDs based on the CallerID. I thought of some kind of lookup table but I would think this would require an AGI, is that correct or is there an easy way to do this? -- Chris Mason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CVS HEAD Broken? app_muxmon.so
Nov 4 07:39:01 VERBOSE[32012] logger.c: == Registered file format vox, extension(s) vox Nov 4 07:39:01 VERBOSE[32012] logger.c: [app_muxmon.so]Nov 4 07:39:01 WARNING[32012] loader.c: /usr/lib/asterisk/modules/app_muxmon.so: undefined symbol: ast_parseoptions Nov 4 07:39:01 WARNING[32012] loader.c: Loading module app_muxmon.so failed! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2.FWDNET.NET not responding?
On Fri, November 4, 2005 13:09, Rich Adamson said: Since a few days my (*) no longer seems to log in to FWD through IAX2. IAX2 DEBUG only shows outbound registration requests, but no replies from FWD: Tx-Frame Retry[003] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 00014ms SCall: 6 DCall: 0 [65.39.205.121:4569] USERNAME: 715749 REFRESH : 60 Its a FWD problem that has been going on for a month or so. It also seems to be somewhat intermitant. Ok, thanks! I'll stop worrying about that one then... ;-) Just gotta figure out how to forward my DID then, but that's another issue... -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk connected with CAPI
Hi Patrick,
i've absolutly no idea what these magic lines do but
it WORKS!!! ;-)))
so thx for you input...
--- Patrick [EMAIL PROTECTED] wrote:
On Fri, 2005-11-04 at 03:55 -0800, richard Coco
wrote:
Hi all,
i'm trying to install a EICON DIVA 4BRI (on CentOS
4.1
2.6.9-22.0.1.EL) using latest package from
sourceforge (chan_capi-cm-0.6.tar.gz).
I have installed divactrl_2.1.tar.gz and untared
protocols_all.tar.bz2 in /usr/share/eicon.
---
lsmod gives me the following...
Module Size Used by
divacapi 157937 0
capi 18177 0
capifs 5961 2 capi
kernelcapi 44641 2 divacapi,capi
md5 4033 1
ipv6 234881 12
lp 12077 0
autofs423237 0
i2c_dev11329 0
i2c_core 22081 1 i2c_dev
sunrpc159269 1
microcode 6881 0
button 6481 0
battery 8901 0
ac 4805 0
uhci_hcd 31065 0
parport_pc 24577 0
parport37129 2 lp,parport_pc
divas 76345 0
divadidd 13081 2 divacapi,divas
e100 41793 0
mii 4673 1 e100
floppy 58481 0
dm_snapshot16901 0
dm_zero 2369 0
dm_mirror 27825 0
ext3 116809 2
jbd71385 1 ext3
dm_mod 56661 6
dm_snapshot,dm_zero,dm_mirror
---
Starting divactrl
---
[EMAIL PROTECTED] asterisk]# divactrl load -c 1 -f
ETSI
Start adapter Nr:1 - 'Diva Server 4BRI-8M 2.0
PCI',
SN: 7113 ... OK
[EMAIL PROTECTED] asterisk]#
but the /var/log/asterisk/messages gives me
following
errors when i try to start asterisk:
Nov 4 12:25:45 WARNING[2658]: CAPI not installed,
CAPI disabled!
Nov 4 12:25:45 WARNING[2658]: chan_capi.so:
load_module failed, returning -1
Nov 4 12:25:45 WARNING[2658]: Loading module
chan_capi.so failed!
Is CAPI really not installed or have i forgotten
something? Here my capi.conf and modules.conf
;
; CAPI config
;
;
; general section
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[EICON]
controller=1,2,3,4
isdnmode=msn
incomingmsn=*
softdtmf=on
relaxdtmf=on
accountcode=
context=incoming
echocancel=yes
devices=2
group=1
;
; Asterisk configuration file
;
; Module Loader configuration file
;
[modules]
autoload=yes
noload = pbx_gtkconsole.so
noload = pbx_kdeconsole.so
noload = app_intercom.so
load = chan_modem.so
load = res_musiconhold.so
load = chan_capi.so
noload = chan_alsa.so
[global]
chan_modem.so=yes
chan_capi.so=yes
thx in advance
Iirc CentOS 4.1 uses udev so you have to add the
proper udev rules so
the capi devices are properly created Stick the
following lines in a
file called e.g. 10-capi.rules and add it to
/etc/udev/rules.d:
SYSFS{dev}=68:0,NAME=capi20
SYSFS{dev}=191:[0-9]*,NAME=capi/%n
Use tabs between the , and NAME!
Once you have done this as root do udevstart. Unload
all the capi
modules and load them again. With capiinfo you can
check if it all went
well (it should give output). If it doesn't work
then reboot the box.
Regards,
Patrick
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[Asterisk-Users] voxby.com $29.95 unlimited...and no catch in T C is anyone using them
Hi IS anyone using them with asterisk, it sounds too good to be true, even with the $1000 for life connection, I could route all my calls to them :-) Iqbal ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CVS HEAD Broken? app_muxmon.so
Nov 4 07:39:01 VERBOSE[32012] logger.c: == Registered file format vox, extension(s) vox Nov 4 07:39:01 VERBOSE[32012] logger.c: [app_muxmon.so]Nov 4 07:39:01 WARNING[32012] loader.c: /usr/lib/asterisk/modules/app_muxmon.so: undefined symbol: ast_parseoptions Nov 4 07:39:01 WARNING[32012] loader.c: Loading module app_muxmon.so failed! noload in modules.conf is a temp fix. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] One Touch Record in 1.2
I have been trying to find more information on the One Touch Record
feature in 1.2 (features.conf) but have not been very successful.
Basically, I've been trying to get more information as to:
1) Do I need to specify any particular option in the Dial command
yes
w W (for enablig caller calle)
2) How can I customize the location of the recorded file(s)
I don't know if you can change the location, I think not. You can
somewhat customize the file name setting the variable TOUCH_MONITOR.
You can set the format setting TOUCH_MONITOR_FORMAT, by default is
.wav
The name of the file will be
auto-{TIMESTAMP}-{CALLER-CLID}-{CALLEE-CLID} by default and
auto-{TIMESTAMP}-{TOUCH_MONITOR} if TOUCH_MONITOR is set.
3) Will the files be soxmix'ed together or not
yes
4) How to use it in general
Just dial the sequence specified in features.conf to start/stop the
recording. By default is *1.
--
Nicolás Gudiño
Buenos Aires - Argentina
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Re: [Asterisk-Users] CVS HEAD Broken? app_muxmon.so
ast_parseoptions is a relatively new call introduced for a cleaner way of parsing API arguments within the code. If you're getting this you've probably got some stale modules /usr/lib/asterisk/modules. Probably best to clean out that directory, do a full make clean, make update, and then rebuild and make install. On 11/4/05, asterisk [EMAIL PROTECTED] wrote: Nov 4 07:39:01 VERBOSE[32012] logger.c: == Registered file format vox, extension(s) vox Nov 4 07:39:01 VERBOSE[32012] logger.c: [app_muxmon.so]Nov 4 07:39:01 WARNING[32012] loader.c: /usr/lib/asterisk/modules/app_muxmon.so: undefined symbol: ast_parseoptions Nov 4 07:39:01 WARNING[32012] loader.c: Loading module app_muxmon.so failed! noload in modules.conf is a temp fix. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2.FWDNET.NET not responding?
Francesco Peeters wrote: Hi all, Since a few days my (*) no longer seems to log in to FWD through IAX2. Use freevoip instead: http://freevoip.gedameurope.com (It links into FWD when FWD is up) -- 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 sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Route call based on CallerID
If you're not using realtime to do your dial plan, why not just do exten = did/callerid,priority,Goto(specialroutefordidwhencid,ext,n) ? On 11/4/05, Chris Mason (Lists) [EMAIL PROTECTED] wrote: I need to send calls to a choice of DIDs based on the CallerID. I thought of some kind of lookup table but I would think this would require an AGI, is that correct or is there an easy way to do this? -- Chris Mason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] one way audio on oh323 channel, there's no rtp traffic
Hi all, i'm experiencing a one way call only between a ipPhone and an analog one through a oh323 channel between my asterisk and a Nortel GK. Doing some sniffing and some debug with ethereal and tcpump i can say (i hope, as newby to say the right thing) that i can't see any rtp traffic between the asterisk and the nortel. In the analog phone (in the outside telecom world) i can't ear nothing said in the ipPhone. Viceversa in the ipPhone (Mitel one) i can ear the voice comming from the outside world. In my sip.conf [419] callerid=0432281316 TEST test 419 type=friend username=419 secret=password host=dynamic nat=yes canreinvite=no reinvite=no disallow=all allow=ulaw allow=gsm ;allow=alaw dtmfmode=rfc2833 context=out callgroup=1 pickupgroup=1 There's no rtp traffic from the phone or from the asterisk to the GK. The GK stays on the intranet even if it has a internet looking ip. ipPhone 10.24.3.40 asterisk 10.24.2.253 GK 80.74.178.196 Issuing on asterisk rtp debug [2]WrapH323EndPoint::AnswerCall: Request to answer call ip$80.74.178.196:34404/1169 Got RTP packet from 10.24.3.40:20012 (type 0, seq 14, ts -1120604096, len 160) [2]WrapH323EndPoint::AnswerCall: Call answered [ip$80.74.178.196:34404/1169] Got RTP packet from 10.24.3.40:20012 (type 0, seq 15, ts -1120603936, len 160) Got RTP packet from 10.24.3.40:20012 (type 0, seq 16, ts -1120603776, len 160) [2]WrapH323Connection::OnReceivedFacility: Received FACILITY message [ip$80.74.178.196:34404/1169] [2]WrapH323Connection::OnReceivedFacility: Received FACILITY message [ip$80.74.178.196:34404/1169] [2]WrapH323Connection::OnReceivedFacility: Received FACILITY message [ip$80.74.178.196:34404/1169] [3]WrapH323EndPoint::OpenAudioChannel: Direction = RECODER, Buffer = 320 [2]WrapH323EndPoint::OpenAudioChannel: Media format: FrameSize 8, FrameTime 8, TimeUnits 8 [2]WrapH323EndPoint::OpenAudioChannel: Codec info: FrameRate 160 [2]WrapH323EndPoint::OpenAudioChannel: Packet size: 160 [2]WrapH323EndPoint::OpenAudioChannel: Frames per packet: 20 [2]WrapH323EndPoint::OpenAudioChannel: LID Codec G.711-uLaw-64k [3]WrapH323EndPoint::OpenAudioChannel: The sound channel with the application is asterisk-oh323(fd=42) [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. [4]PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized. [3]PAsteriskSoundChannel::Open: os_handle 42, mediaFormat 0, frameTime 1 ms, frameNum 20, packetSize 160 [3]WrapH323EndPoint::OpenAudioChannel: Opened sound channel Asterisk for recording using 1x320 byte buffers. [3]WrapH323Connection::OnEstablished: WrapH323Connection [ip$80.74.178.196:34404/1169] established (FastStartDisabled/H245Tunneling) [3]WrapH323EndPoint::OnConnectionEstablished: Connection [ip$80.74.178.196:34404/1169] established. [3]WrapH323EndPoint::GetConnectionInfo: [ip$80.74.178.196:34404/1169] RTP Media: 10.24.2.253:21002-0.0.0.0:0 Got RTP packet from 10.24.3.40:20012 (type 0, seq 17, ts -1120603616, len 160) Got RTP packet from 10.24.3.40:20012 (type 0, seq 18, ts -1120603456, len 160) [2]WrapH323Connection::OnReceivedFacility: Received FACILITY message [ip$80.74.178.196:34404/1169] Got RTP packet from 10.24.3.40:20012 (type 0, seq 19, ts -1120603296, len 160) [3]WrapH323EndPoint::OpenAudioChannel: Direction = PLAYER, Buffer = 320 [2]WrapH323EndPoint::OpenAudioChannel: Media format: FrameSize 8, FrameTime 8, TimeUnits 8 [2]WrapH323EndPoint::OpenAudioChannel: Codec info: FrameRate 160 [2]WrapH323EndPoint::OpenAudioChannel: Packet size: 160 [2]WrapH323EndPoint::OpenAudioChannel: Frames per packet: 20 [2]WrapH323EndPoint::OpenAudioChannel: LID Codec G.711-uLaw-64k [3]WrapH323EndPoint::OpenAudioChannel: The sound channel with the application is asterisk-oh323(fd=40) [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. [4]PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized. [3]PAsteriskSoundChannel::Open: os_handle 40, mediaFormat 0, frameTime 1 ms, frameNum 20, packetSize 160 [3]WrapH323EndPoint::OpenAudioChannel: Opened sound channel Asterisk for playing using 1x320 byte buffers. [5]PAsteriskSoundChannel::Write: Written [160 bytes] Sent RTP packet to 10.24.3.40:20012 (type 0, seq 26203, ts 160, len 160) [2]WrapH323Connection::OnReceivedFacility: Received FACILITY message [ip$80.74.178.196:34404/1169] [5]PAsteriskSoundChannel::Read: Data read [320 bytes] [4]PAsteriskSoundChannel::Read: Timeout [0 bytes] [4]PAsteriskSoundChannel::Read: Timeout [0 bytes] Got RTP packet from 10.24.3.40:20012 (type 0, seq 20, ts -1120603136, len 160) [5]PAsteriskSoundChannel::Write: Written [160 bytes] Sent RTP packet to 10.24.3.40:20012 (type 0, seq 26204, ts 320, len 160) [5]PAsteriskSoundChannel::Read: Data read [320 bytes] Got RTP packet from 10.24.3.40:20012 (type 0, seq 21, ts -1120602976, len 160) [5]PAsteriskSoundChannel::Write: Written [160 bytes] Sent RTP packet to
RE: [Asterisk-Users] libpri
You don't need it, but in Asterisk The Future of Telephony they recommend to install it. --Tomislav ParinaLama Computers SplitStinice 12, 21000 SplitTel.: +385(21)393447e-mail: tparcina#lama.hrhttp://www.lama.hr From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael BielickiSent: 30. listopad 2005 15:49To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] libpri no, libpri is only needed for pri trunks On 10/30/05, Mark Quitoriano [EMAIL PROTECTED] wrote: do i need to install libpri? my only setup is Digium TDM400P with 2 fxo port. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Route call based on CallerID
BJ Weschke wrote: If you're not using realtime to do your dial plan, why not just do exten = did/callerid,priority,Goto(specialroutefordidwhencid,ext,n) ? I'm not sure I understand. Let's say the CallerID = 497 and I want that to dial 222-222-, but if it is 497 I want it to go to 222-222-2223 How would I write that? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Route call based on CallerID
exten = s/497,1,Dial(SIP/[EMAIL PROTECTED]) exten = s/497,1,Dial(SIP/[EMAIL PROTECTED]) exten = s,1,Dial(SIP/[EMAIL PROTECTED]) exten = s,2,Hangup On 11/4/05, Chris Mason (Lists) [EMAIL PROTECTED] wrote: BJ Weschke wrote: If you're not using realtime to do your dial plan, why not just do exten = did/callerid,priority,Goto(specialroutefordidwhencid,ext,n) ? I'm not sure I understand. Let's say the CallerID = 497 and I want that to dial 222-222-, but if it is 497 I want it to go to 222-222-2223 How would I write that? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_agent.c fails to compile
It may compile, but there's no assurances from any of the dev team that you're not going to have other wierd stuff going on from a build that was built with 3.0 gcc. There are multiple areas in the code that now use = 3.0 gcc optimizations. It's important that use a compliant compiler for not only being able to build correctly, but also to make sure that the performance and functionality ends up being what we intended it to be. On 11/4/05, Dinesh Nair [EMAIL PROTECTED] wrote: On 11/04/05 03:26 BJ Weschke said the following: gcc 3.0 and up is now a minimum requirement to build Asterisk. This is most likely your problem. On 11/3/05, Matt Hess [EMAIL PROTECTED] wrote: gcc version 2.95.3 20010125 (prerelease, propolice) on OpenBSD 3.6. which was the same problem i faced when i tryed to compile asterisk cvs head on freebsd 4.x as well. a simple patch (attached) to chan_agent.c fixes this problem and allows a clean compile with gcc 2.95. CUT HERE --- --- ./channels/chan_agent.c.origMon Oct 31 16:30:28 2005 +++ ./channels/chan_agent.c Mon Oct 31 16:34:03 2005 @@ -1680,7 +1680,7 @@ AST_APP_ARG(agent_id); AST_APP_ARG(options); AST_APP_ARG(extension); - ); + ) char *tmpoptions = NULL; char *context = NULL; int play_announcement = 1; CUT HERE --- -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] One Touch Record in 1.2
Nicolás Gudiño wrote:
2) How can I customize the location of the recorded file(s)
I don't know if you can change the location, I think not.
Well, I'd like them to drop in my voicemail when done recording - maybe
in a separate recordings folder but I'd like to use the same interface
to play them back.
You can
somewhat customize the file name setting the variable TOUCH_MONITOR.
You can set the format setting TOUCH_MONITOR_FORMAT, by default is
.wav
The name of the file will be
auto-{TIMESTAMP}-{CALLER-CLID}-{CALLEE-CLID} by default and
auto-{TIMESTAMP}-{TOUCH_MONITOR} if TOUCH_MONITOR is set.
3) Will the files be soxmix'ed together or not
yes
4) How to use it in general
Just dial the sequence specified in features.conf to start/stop the
recording. By default is *1.
--
Nicolás Gudiño
Buenos Aires - Argentina
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Re: [Asterisk-Users] How to detect AGI script failure?
Alex Hutton wrote:
Hello,
I'm new to the list so I hope I'm asking the question in the right
place. In our extensions.conf, we call an AGI script using the AGI
command.
e.g.
exten = 11,1,Answer
exten = 11,2,Wait(0.5)
exten = 11,3,Playback(welcome1)
exten = 11,4,agi(agi://192.168.1.88/hello.agi?src=test|${CALLERID})
If for some reason, the AGI script fails to run (e.g. our AGI prog isn't
running), can we detect it and direct the call to a pre-recorded message?
What I personally would do is first set a variable before you run the agi
(i.e. completionstatus to beforerun) then run the AGI. Once inside the AGI,
set the variable for completion status. I.E. you could have ran well, failed
with x etc etc. Then on the next priority, you can check this variable and
via gotoif for the various statuses (including beforerun which would mean
that the AGI didn't run at all).
While this doesn't exactly answer your question, it is the best way to use
multiple statuses.
Make sense?
--
Cheers,
Matt Riddell
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[Asterisk-Users] SCCP: ServiceURL and Mailbox Notification
Hi. I tried to configure the ServiceURL on the asterisk inside the xml but i can't get it ro work i always get the errror hos tnot found and the ServiceURL field in the telephone is empty. I tried to put it in den SEPxx AND XmlDedault config without success. This is the url: http://phone-xml.berbee.com/menu.xml In my old 7960 i always get a lettersymbol at my line when i got a mailboxmessage via SIP but this won'z be with the sccp protocol? Or how cna i have this symbols there? I have new voicemessages on my asterisk but the telephone is saying nothing about that. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AMP and voicemail passwords
Anyone here using AMP and having problems with users chaning their voicemail passwords? They stick until I go into AMP and make changes then reload. The AMP settings contain the old password and are overwriting the new one saved by the user. What am I doing wrong or what is the correct way to do it? - James ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Why n s priority in CVS but not in release?
Kevin P. Fleming [EMAIL PROTECTED] wrote: actual release that will contain these features will be 1.2.0, scheduled for release within the next two weeks. Oh my God! A Date! :) Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to detect AGI script failure?
How about agi debug on the CLI?
Alex Hutton wrote:
Hello,
I'm new to the list so I hope I'm asking the question in the right
place. In our extensions.conf, we call an AGI script using the AGI
command.
e.g.
exten = 11,1,Answer
exten = 11,2,Wait(0.5)
exten = 11,3,Playback(welcome1)
exten = 11,4,agi(agi://192.168.1.88/hello.agi?src=test|${CALLERID})
If for some reason, the AGI script fails to run (e.g. our AGI prog isn't
running), can we detect it and direct the call to a pre-recorded
message?
What I personally would do is first set a variable before you run the agi
(i.e. completionstatus to beforerun) then run the AGI. Once inside the
AGI,
set the variable for completion status. I.E. you could have ran well,
failed
with x etc etc. Then on the next priority, you can check this variable
and
via gotoif for the various statuses (including beforerun which would
mean
that the AGI didn't run at all).
While this doesn't exactly answer your question, it is the best way to use
multiple statuses.
Make sense?
--
Cheers,
Matt Riddell
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RE: [Asterisk-Users] How to configure Asterisk through webmin
The Thirdlane PBX Manager solution is just a few perl scripts. This is no better than what you can do by directly modifying the Asterisk Config files or many Open Source GUIs like Phonecall etc you have out there. Infact Areski's A2Billing has a good extension configurator in the solution. So that may be something you can consider. Seshu Kanuri From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of PikoroSent: Thursday, November 03, 2005 7:09 PMTo: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] How to configure Asterisk through webmin I tried the third lane asterisk manager thingy for webmin and let me tell you, it did not work. Only made things harder and i had to result to making the configuration by hand in order to get asterisk to work. Going to email them today and ask for a refund.That webmin module by third lane looks like a good solution, but the thing i noticed by reading the manual was that there are quite a few references to "you'll have to change that in the config file" type lines. Basically, it's good for creating extensions, but nothing more.AaronStefan-Michael. Guenther (in-put GbR) wrote: On Thu, November 3, 2005 17:46, nr k said: Hi all I configured asterisk and webmin.i dont know how to integrate webmin with asterisk and how to access asterisk through webmin.pls do the needful. regards ramakrishnan.n Asterisk is not managed through webmin. Webmin is a tool to help administer the rest of the server. and Asterisk, too: Have a look at THIRD LANE ASTERISK PBX MANAGER http://www.thirdlane.com/opensource.htm#manager Stefan NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SIP Disconnect Supervision
Steve Blair [EMAIL PROTECTED] wrote: I have a case where a call from the PSTN to our SER proxy goes unanswered. As a result it is relayed to our Asterisk server for voicemail. However before the greeting plays the caller hangs up. This results in an empty message being created and emailed or the mwi gets activated. I saw a few posts about Disconnect Supervision and Disconnect Supervision with inbound SIP connections but I did not see any resolution to the SIP question. Does this sound like a SIP Disconnect Supervision issue? If not what seems to be the issue? Also does anyone have any suggestions on how to stop these messages from being created? I think you may be misunderstanding this a bit. Disconnect supervision is a PSTN issue not a SIP issue. Presumably your problem occurs because the caller hangs up and the telco doesn't signal this to you in a timely fashion. Therefore your device doesn't go on hook and it doesn't tear-down the SIP session. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP 600/601 microbrowser specs
I've seen a couple of threads on the Polycom IP 600/601 microbrowser and have it up and working for simple pages. Does anyone know how to get more detailed specs on the browser and what it actually supports? And particularly if there are any custom capabilities in the browser. When I worked with WAP and WML on cell phones, there were tags that allowed you do do phone specific things, such as dialing a number, etc. It would be nice to dial a number by selecting a link. Thanks, Mike Clark ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: SIP Disconnect Supervision
Doug Meredith wrote: Steve Blair [EMAIL PROTECTED] wrote: I have a case where a call from the PSTN to our SER proxy goes unanswered. As a result it is relayed to our Asterisk server for voicemail. However before the greeting plays the caller hangs up. This results in an empty message being created and emailed or the mwi gets activated. I saw a few posts about Disconnect Supervision and Disconnect Supervision with inbound SIP connections but I did not see any resolution to the SIP question. Does this sound like a SIP Disconnect Supervision issue? If not what seems to be the issue? Also does anyone have any suggestions on how to stop these messages from being created? I think you may be misunderstanding this a bit. Disconnect supervision is a PSTN issue not a SIP issue. Presumably your problem occurs because the caller hangs up and the telco doesn't signal this to you in a timely fashion. Therefore your device doesn't go on hook and it doesn't tear-down the SIP session. Thanks. My issue is a SIP issue. Doug -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] COREDUMP in actual CVS
Actual cvs is impossible to start get coredump: == Registered application 'SetRDNIS'[app_alarmreceiver.so] = (Alarm Receiver for Asterisk) == Parsing '/etc/asterisk/alarmreceiver.conf': Found == Registered application 'AlarmReceiver'[codec_a_mu.so] = (A-law and Mulaw direct Coder/Decoder) == Registered translator 'alawtoulaw' from format alaw to ulaw, cost 1 == Registered translator 'ulawtoalaw' from format ulaw to alaw, cost 1[app_math.so] = (Basic Math Functions) == Registered application 'Math'[skipping chan_modem_i4l.so][app_sendtext.so] = (Send Text Applications) == Registered application 'SendText'[app_muxmon.so]Ouch ... error while writing audio data: : Broken pipeOuch ... error while writing audio data: : Broken pipeOuch ... error while writing audio data: : Broken pipe ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: Attempted to delete nonexistent schedule entry...
I am having this same issue. But after I get the message all IAX processing stops. I have to reload Asterisk to get IAX peers back up and running. When I capture IAX traffic with ethereal there is no IAX messages being transmitted by Asterisk. - Dustin - On 10/15/2005, J. Iddings jeff at iddings.us http://lists.digium.com/mailman/listinfo/asterisk-users wrote: /I'm also having this issue. Everything seems to work, but it's an //unnerving error. Any thoughts? // //Jimmy wrote: // I just upgraded my test Asterisk box to the latest CVS HEAD. show // version only shows Asterisk CVS HEAD built by rootetc, with no // date or version number. I downloaded this version on Monday, Oct 3. // About once every minute, I get this while at the CLI prompt: // // sched.c:296 ast_sched_del: Attempted to delete nonexistent schedule // entry 1! // // This only appeared after updating. All functions seem normal, other // than these messages. Phones work, auto-attendant works, voicemail works, // etc. What's going on? / OK - I been wrong so many times this week - it ain't funny... But - I think - this part of the scheduling change to the registration stuff. In one update, when a remote phone/system stopped responding to qualify attempts, the system would stop trying to verify the connection. Forever. Not exactly a 'good thing'. It would tell you that by saying Forever but still not good. Then an update added some stuff to ?iax.conf? like: ;qualify=yes ;qualifysmoothing = yes ;qualifyfreqok = 12 ;qualifyfreqnotok = 3 to modify how and when the system would retry these connections. During the time between the first and second update, I would get these messages when I did an iax2 reload. It had stopped trying to qualify the connection - and then the reload would start it backup. It would 'inform' me with the 'attempted to delete nonexistant schedule entry' because the time of the next scheduled event was no longer active. So in essence - it is a warning and not an error. On 10/15/2005, J. Iddings jeff at iddings.us http://lists.digium.com/mailman/listinfo/asterisk-users wrote: /I'm also having this issue. Everything seems to work, but it's an //unnerving error. Any thoughts? // //Jimmy wrote: // I just upgraded my test Asterisk box to the latest CVS HEAD. show // version only shows Asterisk CVS HEAD built by rootetc, with no // date or version number. I downloaded this version on Monday, Oct 3. // About once every minute, I get this while at the CLI prompt: // // sched.c:296 ast_sched_del: Attempted to delete nonexistent schedule // entry 1! // // This only appeared after updating. All functions seem normal, other // than these messages. Phones work, auto-attendant works, voicemail works, // etc. What's going on? / OK - I been wrong so many times this week - it ain't funny... But - I think - this part of the scheduling change to the registration stuff. In one update, when a remote phone/system stopped responding to qualify attempts, the system would stop trying to verify the connection. Forever. Not exactly a 'good thing'. It would tell you that by saying Forever but still not good. Then an update added some stuff to ?iax.conf? like: ;qualify=yes ;qualifysmoothing = yes ;qualifyfreqok = 12 ;qualifyfreqnotok = 3 to modify how and when the system would retry these connections. During the time between the first and second update, I would get these messages when I did an iax2 reload. It had stopped trying to qualify the connection - and then the reload would start it backup. It would 'inform' me with the 'attempted to delete nonexistant schedule entry' because the time of the next scheduled event was no longer active. So in essence - it is a warning and not an error. Brett Brett ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2.FWDNET.NET not responding?
Does freevoip support other codecs other than GSM?On 11/4/05, Matt Riddell [EMAIL PROTECTED] wrote:Francesco Peeters wrote: Hi all, Since a few days my (*) no longer seems to log in to FWD through IAX2. Use freevoip instead:http://freevoip.gedameurope.com(It links into FWD when FWD is up)--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 sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER+ASTERISK
the ser an asterisk run in the same box??? redirect host + port :) 2005/11/4, harry gaillac [EMAIL PROTECTED]: Hello,I wish to setup this scheme:ser-0.9.4asterisk-1.2-bêtapolycom ip300 phonesasterisk 5050-- |firewall+nat|-192.168.ser 5060---My ip phones use ser as outbound sip proxy and asterisk as sip registrar server.Ser Forward REGISTER requests to asterisk however whena phone try to send an invite message then asterisksend icmp to private ip (host=dynamic in sip.conf)Is it possible to solve this problem ? RegardsHarry___Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! MessengerTéléchargez cette version sur http://fr.messenger.yahoo.com___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial in via pstn , out over IP
Hi, all I'm trying to setup a [EMAIL PROTECTED] box so that I can dial in to the pstn, be offered a menu then literally dial an option that will put the line into the from-internal context, so i can then dial out over any of my trunks. the menu isnt a problem, but does anyone know how i can achieve the change of context. Thanks in advance Bails ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Does AEL support arrays?
Hello all, Does anyone know whether there's any support in AEL for arrays, and if so, how one would go about implementing a shift statement? Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited Tel: (01604) 808408 Mobile: (07811) 332969 Skype: minotaur-uk ICQ: 13350579 AIM: MinotaurUK MSN: [EMAIL PROTECTED] Y!: Minotaur_Chris This email is made from 100% recycled electrons ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CVS HEAD Broken? app_muxmon.so
BJ Weschke wrote: ast_parseoptions is a relatively new call introduced for a cleaner way of parsing API arguments within the code. If you're getting this you've probably got some stale modules /usr/lib/asterisk/modules. Probably best to clean out that directory, do a full make clean, make update, and then rebuild and make install. Exactly the situation. When the OP did 'make install', it surely told him that app_muxmon.so was a module _not_ installed during that run (since the module name has been changed), and that he needed to ensure it was compatible with this version of Asterisk before trying to load it. Apparently the OP did not pay any attention to that message :-) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP phones supporting early dial
Hello all, Is there a list of phones that reliably support SIP early dial? One of the really nice things I've noticed about the 7960 (SCCP) is that each digit is sent straight to asterisk, so when the number has been completed, connection is almost instantaneous. I've tried early dial on both the GXP2000s and the HT486/488 units, none of which seem to work reliably. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited Tel: (01604) 808408 Mobile: (07811) 332969 Skype: minotaur-uk ICQ: 13350579 AIM: MinotaurUK MSN: [EMAIL PROTECTED] Y!: Minotaur_Chris This email is made from 100% recycled electrons ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial in via pstn , out over IP
bails wrote: Hi, all I'm trying to setup a [EMAIL PROTECTED] box so that I can dial in to the pstn, be offered a menu then literally dial an option that will put the line into the from-internal context, so i can then dial out over any of my trunks. the menu isnt a problem, but does anyone know how i can achieve the change of context. Thanks in advance vop-info.org search for 'disa' -- Christopher L. Wade, CCNA, CCDA, CQS-CIPTES, CQS-CWLSS ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2.FWDNET.NET not responding?
Tom Vile wrote: Does freevoip support other codecs other than GSM? Not at the moment. What would you like to see? -- 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 sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does AEL support arrays?
Chris Bagnall wrote: Does anyone know whether there's any support in AEL for arrays, and if so, how one would go about implementing a shift statement? No, it does not provide any types of variables that are not available already in the dialplan. Technically, it is only a new _syntax_, not a new dialplan language. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP phones supporting early dial
Chris Bagnall wrote: Hello all, Is there a list of phones that reliably support SIP early dial? One of the really nice things I've noticed about the 7960 (SCCP) is that each digit is sent straight to asterisk, so when the number has been completed, connection is almost instantaneous. I've tried early dial on both the GXP2000s and the HT486/488 units, none of which seem to work reliably. Not that I know of, but the Polycoms support their own digitmaps, which pretty much makes sure the call is dialed when you are done dialing. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does AEL support arrays?
Kevin P. Fleming wrote:
Chris Bagnall wrote:
Does anyone know whether there's any support in AEL for arrays, and if
so,
how one would go about implementing a shift statement?
No, it does not provide any types of variables that are not available
already in the dialplan. Technically, it is only a new _syntax_, not a
new dialplan language.
If that's the case then the following could be easily converted to AEL.
Notice the fake subscripts I used.
[macro-whatever]
exten =
s,1,AGI(callerid-fixup.agi,${CALLERIDNUM}${MACRO_EXTEN}00)
exten = s,2,Noop(AGI(set-ring))
exten = s,3,GotoIf($[${LEN(${FAX_DEST})} = 0]?9:4)
exten = s,4,Cut(TECHNOLOGY=CHANNEL,/,1)
exten = s,5,GotoIf($[${TECHNOLOGY} = Zap]?6:9)
exten = s,6,Answer
exten = s,7,Ringing
exten = s,8,NVFaxDetect(4,d)
exten = s,9,Goto(${MACRO_EXTEN},1)
exten = _,1,GotoIf($[${LEN(${DIAL_DEST[1]})} = 0]?2:4)
exten = _,2,GotoIf($[${LEN(${DIAL_DEST})} = 0]?14:3)
exten = _,3,SetVar(DIAL_DEST[1]=${DIAL_DEST})
exten = _,4,SetVar(INDEX=1)
exten = _,5,GotoIf($[${LEN(${DIAL_TIMEOUT[${INDEX}]})} = 0]?6:7)
exten = _,6,SetVar(DIAL_TIMEOUT[${INDEX}]=20)
exten =
_,7,Dial(${DIAL_DEST[${INDEX}]},${DIAL_TIMEOUT[${INDEX}]},${DIAL_OPTS[${INDEX}]}g)
exten = _,8,GotoIf($[${DIALSTATUS} = BUSY | ${DIALSTATUS} =
CHANUNAVAIL | ${DIALSTATUS} = CONGESTION]?12:9)
exten = _,9,GotoIf($[${DIALSTATUS} = NOANSWER]?14:10)
exten = _,10,Noop(DIALSTATUS=${DIALSTATUS})
exten = _,11,Hangup
exten = _,12,SetVar(INDEX=$[${INDEX} + 1])
exten = _,13,GotoIf($[${LEN(${DIAL_DEST[${INDEX}]})} = 0]?14:5)
exten = _,14,GotoIf($[${LEN(${VOICE_MAILBOX})} = 0]?19:15)
exten = _,15,Voicemail(${VOICE_MAILBOX})
exten = _,16,Wait(2)
exten = _,17,Hangup
exten = _,18,GotoIf($[${DIALSTATUS} = NOANSWER]?19:22)
exten = _,19,Voicemail(u${EXTEN})
exten = _,20,Wait(2)
exten = _,21,Hangup
exten = _,22,Voicemail(b${EXTEN})
exten = _,23,Wait(2)
exten = _,24,Hangup
exten = _,116,AbsoluteTimeout(30)
exten = _,117,Playback(sorry-mailbox-full)
exten = _,118,Wait(2)
exten = _,119,Congestion
exten = _,120,Goto(116)
exten = _,123,Goto(116)
exten = talk,1,Goto(${MACRO_EXTEN},1)
exten = fax,1,Cut(FAX_TECH=FAX_DEST,/,1)
exten = fax,2,GotoIf($[${FAX_TECH} = Zap]?3:7)
exten = fax,3,Dial(${FAX_DEST},20)
exten = fax,4,AbsoluteTimeout(30)
exten = fax,5,Wait(2)
exten = fax,6,Congestion
exten = fax,7,RxFax(/tmp/fax-${UNIQUEID}.tiff)
exten =
fax,8,DeadAGI(/usr/local/bin/fax2email.pl,/tmp/fax-${UNIQUEID}.tiff)
exten = fax,9,Hangup
exten = fax,104,AbsoluteTimeout(30)
exten = fax,105,Busy
exten = a,1,Playback(/etc/asterisk/directvm)
exten = a,2,VoicemailMain()
exten = a,3,Wait(.5)
exten = a,4,Goto(1)
exten = o,1,GotoIf($[${LEN(${OPER_DEST})} = 0]?2:4)
exten = o,2,Goto(extensions,0,1)
exten = o,3,Hangup
exten = o,4,GotoIf($[${OPER_TIMEOUT} = 0]?5:6)
exten = o,5,SetVar(OPER_TIMEOUT=)
exten = o,6,GotoIf($[${LEN(${OPER_MESSAGE})} = 0]?8:7)
exten = o,7,Playback(${OPER_MESSAGE})
exten = o,8,Dial(${OPER_DEST},${OPER_TIMEOUT},${OPER_FLAGS})
exten = o,9,Voicemail(u${MACRO_EXTEN})
exten = o,10,Hangup
___
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Re: [Asterisk-Users] AMP and voicemail passwords
James Armstrong wrote: Anyone here using AMP and having problems with users chaning their voicemail passwords? They stick until I go into AMP and make changes then reload. The AMP settings contain the old password and are overwriting the new one saved by the user. What am I doing wrong or what is the correct way to do it? Please post to the amportal-users mailing list: http://lists.sourceforge.net/lists/listinfo/amportal-users and/or Help forum: http://sourceforge.net/forum/forum.php?forum_id=414452 Please include in your post info such as version of AMP, where the users are trying to change their passwords (i.e. phone (0 - 5) or ARI), etc. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can´t compile asterisk1.2bet a2
Hi ... .. . gcc -shared -Xlinker -x -o chan_modem_bestdata.so chan_modem_bestdata.o gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -Wno-missing-prototypes -Wno-missing-declarations -DZAPATA_PRI -DIAX_TRUNKING -DCRYPTO -fPIC -c -o chan_agent.o chan_agent.c chan_agent.c: In function `__login_exec': chan_agent.c:1684: parse error before `char' chan_agent.c:1701: `agent_goodbye' undeclared (first use in this function) chan_agent.c:1701: (Each undeclared identifier is reported only once chan_agent.c:1701: for each function it appears in.) chan_agent.c:1708: `tmpoptions' undeclared (first use in this function) chan_agent.c:1714: `update_cdr' undeclared (first use in this function) chan_agent.c:1732: `context' undeclared (first use in this function) chan_agent.c:1737: `play_announcement' undeclared (first use in this function) chan_agent.c:1864: `filename' undeclared (first use in this function) make[1]: *** [chan_agent.o] Error 1 make[1]: Leaving directory `/var/root/astbillFiles/asterisk/channels' make: *** [subdirs] Error 1 any idea??? thanks Rafael ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can´t compile asterisk1.2beta2
This has already been discussed, you need to upgrade your GCC to 3 or higher. Joshua Colp From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rafael R. GV Sent: Friday, November 04, 2005 11:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Can´t compile asterisk1.2beta2 Hi ... .. . gcc -shared -Xlinker -x -o chan_modem_bestdata.so chan_modem_bestdata.o gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -Wno-missing-prototypes -Wno-missing-declarations -DZAPATA_PRI -DIAX_TRUNKING -DCRYPTO -fPIC -c -o chan_agent.o chan_agent.c chan_agent.c: In function `__login_exec': chan_agent.c:1684: parse error before `char' chan_agent.c:1701: `agent_goodbye' undeclared (first use in this function) chan_agent.c:1701: (Each undeclared identifier is reported only once chan_agent.c:1701: for each function it appears in.) chan_agent.c:1708: `tmpoptions' undeclared (first use in this function) chan_agent.c:1714: `update_cdr' undeclared (first use in this function) chan_agent.c:1732: `context' undeclared (first use in this function) chan_agent.c:1737: `play_announcement' undeclared (first use in this function) chan_agent.c:1864: `filename' undeclared (first use in this function) make[1]: *** [chan_agent.o] Error 1 make[1]: Leaving directory `/var/root/astbillFiles/asterisk/channels' make: *** [subdirs] Error 1 any idea??? thanks Rafael ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Looking por a provider to work with asterisk
I know about broadvoice.com But are they the only solution? I want to have two lines with Asterisk. This is just a home install. Believe it or not I have been using Vonage for about 2 ½ years and now I want to get rid of them to Use and learn Asterisk. Any help would be appreciated. -Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Asterisk to Avaya IP Office
Title: Re: [Asterisk-Users] TDM01B vs. X100P 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, ChrisSent: 31 October 2005 14:00To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [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 ClaussAvaya Certified Expert; Cisco CCDA; Microsoft MCSE Strategic Products and ServicesAVAYA 2003 Business Partner of the Year 3 Wing DriveCedar Knolls, NJ 07927 973-359-8557 Voice973-944-5800 Fax From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David RahnSent: Sunday, October 30, 2005 10:20 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: 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 sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking por a provider to work with asterisk
TelaSIP works great for me.On 11/3/05, Jason Brashear [EMAIL PROTECTED] wrote: I know about broadvoice.com But are they the only solution? I want to have two lines with Asterisk. This is just a home install. Believe it or not I have been using Vonage for about 2 ½ years and now I want to get rid of them to Use and learn Asterisk. Any help would be appreciated. -Jason ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony www.baldwintechsolutions.comPhone: 518-631-2855 x205Phone: 978-203-3848 x205Fax: 518-631-2856 Signup for Telasip at my link at baldwintechsolutions.com/phoneservice.php and receive 2 Phone numbers in the area codes of your choice. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Beta2 problems with DTMF with T option in Dial Command
I was running CVS HEAD from 2005/07/31 until the day that beta2 came out. I installed beta2 on a number of servers without touching anything in /etc/asterisk. Most everything has been working well. One thing that is not is remote DTMF, more specifically, the # key. When I dial voicemail from DIAX, connected directly to the asterisk machine, I can retrieve voicemail. If I have DIAX connected to another asterisk, and dial the extension that connects me back to voicemail on that first box, then after I type the box number, it complains about an incorrect password on the first number that I type, no matter what that is. This is _not_ just a voicemail problem. If I have a DISA statement, with a hard-coded PIN, if DIAX is connected to the box directly, DISA works correctly. If I go through a remote asterisk, DISA fails every time. It _never_ recognizes the #, so it thinks the password has timed out every time. A little digging seems to show that the problem is in the T option to the Dial command which connects the two asterisk boxes. My features.conf file has blindxfer = #7 and atxfer = ##. A single # has been passed through correctly for months. Now, if I remove the T from the Dial command, then the remote voicemail (or DISA) works correctly. A few details: 1) all boxes in this experiment are running 1.2 beta2. 2) all boxes force ULAW codec only 3) if dtmfmode is ever referenced, it is always set to inband. 4) all of this worked in CVS HEAD as of July 31st, 2005. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Looking por a provider to work with asterisk
You can also try http://www.terracall.com I have been using them with good results lately Seshu Kanuri From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason BrashearSent: Thursday, November 03, 2005 11:03 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Looking por a provider to work with asterisk I know about broadvoice.com But are they the only solution? I want to have two lines with Asterisk. This is just a home install. Believe it or not I have been using Vonage for about 2 years and now I want to get rid of them to Use and learn Asterisk. Any help would be appreciated. -Jason NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme: Sending DTMF to other users in a conference
Hi, I would like to know the possibility of sending DTMF to other users in a meetme. I'm looking at inviting a participant from within the conference, here the participant is another conference bridge. So we need to send PIN to this conference bridge. How can I bypass the IVR detect menu and send DTMF to the other participants. Does careful_write in case of frametype is AST_FRAME_DTMF will work ? Final aim here is to bridge asterisk's meetme and another conference bridge. This I need to do from within the conference. Another usage, say if we are inviting some person from within the conference, if this lands in the company's IVr then there should be some way to send DTMF to that IVR to reach that person. Anybody came across such a scenario ? Thanks, ~Vamsi ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking por a provider to work with asterisk
http://freevoip.gedameurope.com Jason Brashear wrote: I know about broadvoice.com But are they the only solution? I want to have two lines with Asterisk. This is just a home install. Believe it or not I have been using Vonage for about 2 years and now I want to get rid of them to Use and learn Asterisk. Any help would be appreciated. -Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- BRUNO DE LUCA Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] lucent TNT h323/sip config?
Asterisk cannot act as a H.323 gatekeeper for TNT to register. We need a gatekeeper like Lucent MVAM for TNT to register to. Asterisk will register to MVAM as a gateway. ~VamsiOn 10/31/05, Armand Sulter [EMAIL PROTECTED] wrote: Does anyone have an example of a lucentTNT h323 config to work with asterisk ?I'd like to use sip but it's not supported in theTAOS we have, if anyone has TAOS 10.x or laterthat would be awsome as well, we have the examples for a sip config.thx- Armand___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meetme: Sending DTMF to other users in a conference
Hello, We wrote a small AGI script to do just this. We just drop it's exten into the conference room and pass it the digits you want played and it will play the audio files of DTMF digits to all participants in the meetme room. It works great for us and we've been using it for over 2 years now. MATT--- On 11/4/05, Vamsi Pottangi [EMAIL PROTECTED] wrote: Hi, I would like to know the possibility of sending DTMF to other users in a meetme. I'm looking at inviting a participant from within the conference, here the participant is another conference bridge. So we need to send PIN to this conference bridge. How can I bypass the IVR detect menu and send DTMF to the other participants. Does careful_write in case of frametype is AST_FRAME_DTMF will work ? Final aim here is to bridge asterisk's meetme and another conference bridge. This I need to do from within the conference. Another usage, say if we are inviting some person from within the conference, if this lands in the company's IVr then there should be some way to send DTMF to that IVR to reach that person. Anybody came across such a scenario ? Thanks, ~Vamsi ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to configure Asterisk through webmin
Hi Seshu, I would be happy to walk you (or anyone else who may be interested) through the Thirdlane PBX Manager features, to explain that while it wont magically configure Asterisk for you, it does help quite a bit. It is all really about the expectations and the target audience what is a good tool for some is too limiting for the others, and whatever is not limiting may appear too complex and not immediately useful. Please contact me off list at [EMAIL PROTECTED], or even better, we could spend a half an hour on the phone that may change your opinion. Best regards, Alex From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: Friday, November 04, 2005 6:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] How to configure Asterisk through webmin The Thirdlane PBX Manager solution is just a few perl scripts. This is no better than what you can do by directly modifying the Asterisk Config files or many Open Source GUIs like Phonecall etc you have out there. Infact Areski's A2Billing has a good extension configurator in the solution. So that may be something you can consider. Seshu Kanuri From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pikoro Sent: Thursday, November 03, 2005 7:09 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to configure Asterisk through webmin I tried the third lane asterisk manager thingy for webmin and let me tell you, it did not work. Only made things harder and i had to result to making the configuration by hand in order to get asterisk to work. Going to email them today and ask for a refund. That webmin module by third lane looks like a good solution, but the thing i noticed by reading the manual was that there are quite a few references to you'll have to change that in the config file type lines. Basically, it's good for creating extensions, but nothing more. Aaron Stefan-Michael. Guenther (in-put GbR) wrote: On Thu, November 3, 2005 17:46, nr k said: Hi allI configured asterisk and webmin.i dont know how tointegrate webmin with asterisk and how to accessasteriskthrough webmin.pls do the needful.regardsramakrishnan.n Asterisk is not managed through webmin. Webmin is a tool to helpadminister the rest of the server. and Asterisk, too:Have a look at THIRD LANE ASTERISK PBX MANAGERhttp://www.thirdlane.com/opensource.htm#managerStefan NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Looking por a provider to work with asterisk
If you are in europe we can provide you sip and iax for asterisk Best regards Thierry [EMAIL PROTECTED] Tel : +33 (0)3 90 40 06 75 Fax: +33 (0)3 90 40 06 76 http://www.widevoip.com -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Jason Brashear Envoyé : jeudi 3 novembre 2005 17:03 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] Looking por a provider to work with asterisk I know about broadvoice.com But are they the only solution? I want to have two lines with Asterisk. This is just a home install. Believe it or not I have been using Vonage for about 2 ½ years and now I want to get rid of them to Use and learn Asterisk. Any help would be appreciated. -Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Looking por a provider to work with asterisk
I am in the US. Texas. -J (= -Original Message- From: WideVOIP [mailto:[EMAIL PROTECTED] Sent: Friday, November 04, 2005 10:39 AM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Looking por a provider to work with asterisk If you are in europe we can provide you sip and iax for asterisk Best regards Thierry [EMAIL PROTECTED] Tel : +33 (0)3 90 40 06 75 Fax: +33 (0)3 90 40 06 76 http://www.widevoip.com -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Jason Brashear Envoyé : jeudi 3 novembre 2005 17:03 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] Looking por a provider to work with asterisk I know about broadvoice.com But are they the only solution? I want to have two lines with Asterisk. This is just a home install. Believe it or not I have been using Vonage for about 2 ½ years and now I want to get rid of them to Use and learn Asterisk. Any help would be appreciated. -Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to configure Asterisk through webmin
Alex We paid some one for your product they did a install for us but later we found that it was a demo. The $295.00 Was paid directly to this person that Said that they were an affiliate of yours. Is there anything that we can do? I would love to talk to you off line about this. -Jason Austin Texas From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Epshteyn Sent: Friday, November 04, 2005 10:36 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] How to configure Asterisk through webmin Hi Seshu, I would be happy to walk you (or anyone else who may be interested) through the Thirdlane PBX Manager features, to explain that while it wont magically configure Asterisk for you, it does help quite a bit. It is all really about the expectations and the target audience what is a good tool for some is too limiting for the others, and whatever is not limiting may appear too complex and not immediately useful. Please contact me off list at [EMAIL PROTECTED], or even better, we could spend a half an hour on the phone that may change your opinion. Best regards, Alex From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: Friday, November 04, 2005 6:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] How to configure Asterisk through webmin The Thirdlane PBX Manager solution is just a few perl scripts. This is no better than what you can do by directly modifying the Asterisk Config files or many Open Source GUIs like Phonecall etc you have out there. Infact Areski's A2Billing has a good extension configurator in the solution. So that may be something you can consider. Seshu Kanuri From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pikoro Sent: Thursday, November 03, 2005 7:09 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to configure Asterisk through webmin I tried the third lane asterisk manager thingy for webmin and let me tell you, it did not work. Only made things harder and i had to result to making the configuration by hand in order to get asterisk to work. Going to email them today and ask for a refund. That webmin module by third lane looks like a good solution, but the thing i noticed by reading the manual was that there are quite a few references to you'll have to change that in the config file type lines. Basically, it's good for creating extensions, but nothing more. Aaron Stefan-Michael. Guenther (in-put GbR) wrote: On Thu, November 3, 2005 17:46, nr k said: Hi allI configured asterisk and webmin.i dont know how tointegrate webmin with asterisk and how to accessasteriskthrough webmin.pls do the needful.regardsramakrishnan.n Asterisk is not managed through webmin. Webmin is a tool to helpadminister the rest of the server. and Asterisk, too:Have a look at THIRD LANE ASTERISK PBX MANAGERhttp://www.thirdlane.com/opensource.htm#managerStefan NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Uninstall AMP
Hi! How do I uninstall AMP and FOP from my Asterisk? Regards Anders Svensson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PAP2 and ringing issues
Hi Aaron, I tried the progressinband=no and it worked great. Thanks for the tip. Humberto -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Humberto Aicardi Sent: 01 November 2005 17:17 To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] PAP2 and ringing issues Hi, I currently have several PAP2-NA units configured to an Asterisk box, everything works fine except from the fact that after dialing a number I can hear ringing tones. When I connect to the same Asterisk box using XLite or EyeBeam I hear only one, any ideas on what may be wrong on the PAP units? Hi Humberto, We had this problem with calls being sent to a PRI. The two ringtones were due to both an RTP audio stream being generated from the PRI (this is the one we wanted) and also a SIP 180 ringing response being sent by the same Asterisk server. I'm not sure why both are getting sent, in 1.0.7 I'm pretty sure they weren't. The fix was simply to set progressinband=no in sip.conf on the Asterisk server with the PRI. The reason you only get the doble ring on one UA and not others seems to be entirely down to the UA. In our case the Linksys units act passed on both ringing indications where as Cisco IP Phones disregarded the SIP 180 and just passed on the RTP. Hth. Aaron ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] One Touch Record in 1.2
Hello.
The one touch record features only work en asterisk 1.2? Because I
tryed in asterisk 1.0.9 and I can't make it works.
Best regards.
El vie, 04-11-2005 a las 08:04 -0600, Tim Litwiller escribió:
Nicolás Gudiño wrote:
2) How can I customize the location of the recorded file(s)
I don't know if you can change the location, I think not.
Well, I'd like them to drop in my voicemail when done recording - maybe
in a separate recordings folder but I'd like to use the same interface
to play them back.
You can
somewhat customize the file name setting the variable TOUCH_MONITOR.
You can set the format setting TOUCH_MONITOR_FORMAT, by default is
.wav
The name of the file will be
auto-{TIMESTAMP}-{CALLER-CLID}-{CALLEE-CLID} by default and
auto-{TIMESTAMP}-{TOUCH_MONITOR} if TOUCH_MONITOR is set.
3) Will the files be soxmix'ed together or not
yes
4) How to use it in general
Just dial the sequence specified in features.conf to start/stop the
recording. By default is *1.
--
Nicolás Gudiño
Buenos Aires - Argentina
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José Luis Gómez
Qualis Information Technology
Av. Rivadavia 2553, PB Of. 43 EP
0342-4565684 int 102
Bs. As. 011-51990896
www.qualis.com.ar
Soporte 0810-8880022
Santa Fe - Argentina
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Re: [Asterisk-Users] Te100 Digital vs Analog
On Friday 04 November 2005 11:57, Matt wrote: I have a Digium TE100 that I will be connecting to a T1. The T1 provider is asking whether the T1 voice circuits are T1 analog or T1 digital. ?? T1 is digital. There is no analog. I think what the provider is asking is whether you want CAS T1 or CCS T1 (PRI). -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking por a provider to work with asterisk
I've been using Teliax.com. Chris - Original Message - From: Jason Brashear [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, November 03, 2005 10:03 AM Subject: [Asterisk-Users] Looking por a provider to work with asterisk I know about broadvoice.com But are they the only solution? I want to have two lines with Asterisk. This is just a home install. Believe it or not I have been using Vonage for about 2 ½ years and now I want to get rid of them to Use and learn Asterisk. Any help would be appreciated. -Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Uninstall AMP
rm rf / ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anders Svensson Sent: Friday, November 04, 2005 11:54 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Uninstall AMP Hi! How do I uninstall AMP and FOP from my Asterisk? Regards Anders Svensson This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER+ASTERISK
Hello Walter, The ser an asterisk run in the same box. What do you mean redirect host + port :) Sip agents send sip requests to ser (outbound proxy) and this one to asterisk ! sip agents are both registered on ser and asterisk. Please to explain me how asterisk redirect the requests. Regards Harry --- Walter Willis [EMAIL PROTECTED] a écrit : the ser an asterisk run in the same box??? redirect host + port :) 2005/11/4, harry gaillac [EMAIL PROTECTED]: Hello, I wish to setup this scheme: ser-0.9.4 asterisk-1.2-bêta polycom ip300 phones asterisk 5050-- |firewall+nat|-192.168. ser 5060--- My ip phones use ser as outbound sip proxy and asterisk as sip registrar server. Ser Forward REGISTER requests to asterisk however when a phone try to send an invite message then asterisk send icmp to private ip (host=dynamic in sip.conf) Is it possible to solve this problem ? Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com-- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SCCP: ServiceURL and Mailbox Notification
I had the same issue. Here is a full config from a 4.1.3SR1 CCM for a
7970 - let me knwo if you need any others and I will tftp them off.
Thanks,
Greg
#
[EMAIL PROTECTED] ~ $ cat SEP00127F027D17.cnf.xml
device xsi:type=axl:XIPPhone ctiid=581916804
uuid={0B7DCA2C-453E-4F01-908 A-A3E877A707D2}
devicePool uuid={1B1B9EB6-7803-11D3-BDF0-00108302EAD1}
nameDefault/name
dateTimeSetting uuid={9EC4850A-7748-11D3-BDF0-00108302EAD1}
nameCMLocal/name
dateTemplateM/D/Y/dateTemplate
timeZoneCentral Standard/Daylight Time/timeZone
/dateTimeSetting
callManagerGroup
members
member priority=0
callManager
ports
analogAccessPort2002/analogAccessPort
digitalAccessPort2001/digitalAccessPort
ethernetPhonePort2000/ethernetPhonePort
mgcpPorts
listen2427/listen
keepAlive2428/keepAlive
/mgcpPorts
/ports
processNodeName192.168.2.10/processNodeName
/callManager
/member
member priority=1
callManager
ports
analogAccessPort2002/analogAccessPort
digitalAccessPort2001/digitalAccessPort
ethernetPhonePort2000/ethernetPhonePort
mgcpPorts
listen2427/listen
keepAlive2428/keepAlive
/mgcpPorts
/ports
processNodeName192.168.2.11/processNodeName
/callManager
/member
/members
/callManagerGroup
srstInfo uuid={CD241E11-4A58-4D3D-9661-F06C912A18A3}
nameDisable/name
srstOptionDisable/srstOption
userModifiablefalse/userModifiable
ipAddr1/ipAddr1
port12000/port1
ipAddr2/ipAddr2
port22000/port2
ipAddr3/ipAddr3
port32000/port3
isSecurefalse/isSecure
/srstInfo
mlppDomainId-1/mlppDomainId
mlppIndicationStatusDefault/mlppIndicationStatus
preemptionDefault/preemption
connectionMonitorDuration120/connectionMonitorDuration
/devicePool
loadInformationTERM70.7-0-2-0S/loadInformation
versionStamp{21ECCF08-13DB-4EC5-8BCE-B177569C489B}/versionStamp
userLocale
nameEnglish_United_States/name
uid1/uid
langCodeen/langCode
version4.1(3)/version
winCharSetiso-8859-1/winCharSet
/userLocale
networkLocaleUnited_States/networkLocale
networkLocaleInfo
nameUnited_States/name
uid64/uid
version4.1(3)/version
/networkLocaleInfo
deviceSecurityMode1/deviceSecurityMode
idleTimeout0/idleTimeout
authenticationURLhttp://192.168.2.10/CCMCIP/authenticate.asp/authenticationUR
L
directoryURLhttp://192.168.2.10/CCMCIP/xmldirectory.asp/directoryURL
idleURL/idleURL
informationURLhttp://192.168.2.10/CCMCIP/GetTelecasterHelpText.asp/informatio
nURL
messagesURL/messagesURL
proxyServerURL/proxyServerURL
servicesURLhttp://192.168.2.20/CiscoServices/fetchPhoneObject/servicesURL
dscpForCm2Dvce96/dscpForCm2Dvce
dscpForSCCPPhoneConfig96/dscpForSCCPPhoneConfig
dscpForSCCPPhoneServices0/dscpForSCCPPhoneServices
capfAuthMode1/capfAuthMode
capfList
capf
phonePort3804/phonePort
processNodeName192.168.2.10/processNodeName
/capf
/capfList
/device
#
On Fri, 2005-11-04 at 15:14 +0100, René Enskat [Teamware GmbH] wrote:
Hi.
I tried to configure the ServiceURL on the asterisk inside the xml but
i can't get it ro work i always get the errror hos tnot found and the
ServiceURL field in the telephone is empty.
I tried to put it in den SEPxx AND XmlDedault config without success.
This is the url:
http://phone-xml.berbee.com/menu.xml
In my old 7960 i always get a lettersymbol at my line when i got a
mailboxmessage via SIP but this won'z be with the sccp protocol?
Or how cna i have this symbols there?
I have new voicemessages on my asterisk but the telephone is saying
nothing about that.
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Re: [Asterisk-Users] SCCP: ServiceURL and Mailbox Notification
Forgot to mention - it is 7.0.2-0S firmware
On Fri, 2005-11-04 at 11:35 -0600, Greg Oliver wrote:
I had the same issue. Here is a full config from a 4.1.3SR1 CCM for a
7970 - let me knwo if you need any others and I will tftp them off.
Thanks,
Greg
#
[EMAIL PROTECTED] ~ $ cat SEP00127F027D17.cnf.xml
device xsi:type=axl:XIPPhone ctiid=581916804
uuid={0B7DCA2C-453E-4F01-908 A-A3E877A707D2}
devicePool uuid={1B1B9EB6-7803-11D3-BDF0-00108302EAD1}
nameDefault/name
dateTimeSetting uuid={9EC4850A-7748-11D3-BDF0-00108302EAD1}
nameCMLocal/name
dateTemplateM/D/Y/dateTemplate
timeZoneCentral Standard/Daylight Time/timeZone
/dateTimeSetting
callManagerGroup
members
member priority=0
callManager
ports
analogAccessPort2002/analogAccessPort
digitalAccessPort2001/digitalAccessPort
ethernetPhonePort2000/ethernetPhonePort
mgcpPorts
listen2427/listen
keepAlive2428/keepAlive
/mgcpPorts
/ports
processNodeName192.168.2.10/processNodeName
/callManager
/member
member priority=1
callManager
ports
analogAccessPort2002/analogAccessPort
digitalAccessPort2001/digitalAccessPort
ethernetPhonePort2000/ethernetPhonePort
mgcpPorts
listen2427/listen
keepAlive2428/keepAlive
/mgcpPorts
/ports
processNodeName192.168.2.11/processNodeName
/callManager
/member
/members
/callManagerGroup
srstInfo uuid={CD241E11-4A58-4D3D-9661-F06C912A18A3}
nameDisable/name
srstOptionDisable/srstOption
userModifiablefalse/userModifiable
ipAddr1/ipAddr1
port12000/port1
ipAddr2/ipAddr2
port22000/port2
ipAddr3/ipAddr3
port32000/port3
isSecurefalse/isSecure
/srstInfo
mlppDomainId-1/mlppDomainId
mlppIndicationStatusDefault/mlppIndicationStatus
preemptionDefault/preemption
connectionMonitorDuration120/connectionMonitorDuration
/devicePool
loadInformationTERM70.7-0-2-0S/loadInformation
versionStamp{21ECCF08-13DB-4EC5-8BCE-B177569C489B}/versionStamp
userLocale
nameEnglish_United_States/name
uid1/uid
langCodeen/langCode
version4.1(3)/version
winCharSetiso-8859-1/winCharSet
/userLocale
networkLocaleUnited_States/networkLocale
networkLocaleInfo
nameUnited_States/name
uid64/uid
version4.1(3)/version
/networkLocaleInfo
deviceSecurityMode1/deviceSecurityMode
idleTimeout0/idleTimeout
authenticationURLhttp://192.168.2.10/CCMCIP/authenticate.asp/authenticationUR
L
directoryURLhttp://192.168.2.10/CCMCIP/xmldirectory.asp/directoryURL
idleURL/idleURL
informationURLhttp://192.168.2.10/CCMCIP/GetTelecasterHelpText.asp/informatio
nURL
messagesURL/messagesURL
proxyServerURL/proxyServerURL
servicesURLhttp://192.168.2.20/CiscoServices/fetchPhoneObject/servicesURL
dscpForCm2Dvce96/dscpForCm2Dvce
dscpForSCCPPhoneConfig96/dscpForSCCPPhoneConfig
dscpForSCCPPhoneServices0/dscpForSCCPPhoneServices
capfAuthMode1/capfAuthMode
capfList
capf
phonePort3804/phonePort
processNodeName192.168.2.10/processNodeName
/capf
/capfList
/device
#
On Fri, 2005-11-04 at 15:14 +0100, René Enskat [Teamware GmbH] wrote:
Hi.
I tried to configure the ServiceURL on the asterisk inside the xml but
i can't get it ro work i always get the errror hos tnot found and the
ServiceURL field in the telephone is empty.
I tried to put it in den SEPxx AND XmlDedault config without success.
This is the url:
http://phone-xml.berbee.com/menu.xml
In my old 7960 i always get a lettersymbol at my line when i got a
mailboxmessage via SIP but this won'z be with the sccp protocol?
Or how cna i have this symbols there?
I have new voicemessages on my asterisk but the telephone is saying
nothing about that.
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[Asterisk-Users] Problem on Data-Connections through Asterisk
Hello, i'm using Asterisk as an intermediary between another PBX (Teles) and the E1 of our Carrier. We us a TE205P Dual Span Card from Digium. All the analog connections are running fine, but digital data calls from local ISDN (BRI @ Teles PBX) to a remote syncPPP-dialup via the E1 doesn't work well. It looks that the connection is answered ok, but no data is going over this connection, so the handshake for the PPP-Login doesn't go to the server. I've also tested a rawip connection over ISDN (isdnx Interfaces of Linux) and the effect is the same. The connection will be made but no data goes through it. When i conect the Teles-PBX directly to the E1 the calls and all data is going through the line. I've played with some options i've found in the Mailinglist in the Dial-Tag but without any success. Do someone have any idea if this fault can be removed or is there no chance to get this working? Thanks a lot Hans-Peter Straub -- ---* I-NetPartner GmbH Hans-Peter Straub Seewiesenstrasse 12 D-73054 Eislingen -- Phone: +49 7161 9849955 Fax: +49 7161 9849956 -- eMail: [EMAIL PROTECTED] Web: http://www.I-NetPartner.de ---* ** Informieren Sie Sich über ** -- GigaLan -- ** das Funknetz im Filstal ** http://www.GigaLan.de ---* -- PGP-ID: 24557EED PGP-Key: http://www.i-netpartner.de/hps.asc PGP-Fingerprint: 51F2 31E4 4361 1B7F 8648 60D9 FC1A 68D2 2455 7EED -- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Looking por a provider to work with asterisk
Very happy with nufone. http://www.nufone.net From: Jason Brashear [mailto:[EMAIL PROTECTED] Sent: Thursday, November 03, 2005 10:03 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Looking por a provider to work with asterisk I know about broadvoice.com But are they the only solution? I want to have two lines with Asterisk. This is just a home install. Believe it or not I have been using Vonage for about 2 ½ years and now I want to get rid of them to Use and learn Asterisk. Any help would be appreciated. -Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] One Touch Record in 1.2
Correct. It is not a feature that works in the 1.0 tree.
On 11/4/05, José Luis Gómez [EMAIL PROTECTED] wrote:
Hello.
The one touch record features only work en asterisk 1.2? Because I
tryed in asterisk 1.0.9 and I can't make it works.
Best regards.
El vie, 04-11-2005 a las 08:04 -0600, Tim Litwiller escribió:
Nicolás Gudiño wrote:
2) How can I customize the location of the recorded file(s)
I don't know if you can change the location, I think not.
Well, I'd like them to drop in my voicemail when done recording - maybe
in a separate recordings folder but I'd like to use the same interface
to play them back.
You can
somewhat customize the file name setting the variable TOUCH_MONITOR.
You can set the format setting TOUCH_MONITOR_FORMAT, by default is
.wav
The name of the file will be
auto-{TIMESTAMP}-{CALLER-CLID}-{CALLEE-CLID} by default and
auto-{TIMESTAMP}-{TOUCH_MONITOR} if TOUCH_MONITOR is set.
3) Will the files be soxmix'ed together or not
yes
4) How to use it in general
Just dial the sequence specified in features.conf to start/stop the
recording. By default is *1.
--
Nicolás Gudiño
Buenos Aires - Argentina
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RE: [Asterisk-Users] Looking por a provider to work with asterisk
Thank you Gleim I will look into that. -Jason From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gleim, Jason Sent: Friday, November 04, 2005 11:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Looking por a provider to work with asterisk Jason, Back in August there was a post of a sip.conf and extensions.conf that would setup Asterisk to work with Vonage. I havent tried it yet but the user that posted reported success. Search the archive for Asterisk and Vonage and you should be able to find it or e-mail me off-list and Ill send you a copy. J From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Brashear Sent: Thursday, November 03, 2005 11:03 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Looking por a provider to work with asterisk I know about broadvoice.com But are they the only solution? I want to have two lines with Asterisk. This is just a home install. Believe it or not I have been using Vonage for about 2 ½ years and now I want to get rid of them to Use and learn Asterisk. Any help would be appreciated. -Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] R2-Digital (Q.421)
Does anyone know how to make this work with Asterisk? (R2-Digital (Q.421)) I have MFCR2 configured but I'm told that outgoing calls are to use Q421 R2 Digital signalling. Any help is appreciated. Jesus Mogollon ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Uninstall AMP
Claudio Canseco wrote: Really is that the way to uninstall FOP and AMP?, thank you i've been looking for an answer about it. Regards Claudio. No Claudio, That will wipe your system. He's being a smartass. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Uninstall AMP
I agree. It is like we newbie's on Asterisk is just trouble for the list members. Pity there is no newbie list. But all were newbie's in the beginning and not so pompous as some on this list Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Ferrell Sent: den 4 november 2005 19:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Uninstall AMP Claudio Canseco wrote: Really is that the way to uninstall FOP and AMP?, thank you i've been looking for an answer about it. Regards Claudio. No Claudio, That will wipe your system. He's being a smartass. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] R2-Digital (Q.421)
Jesus Mogollon wrote: Does anyone know how to make this work with Asterisk? (R2-Digital (Q.421)) I have MFCR2 configured but I'm told that outgoing calls are to use Q421 R2 Digital signalling. Any help is appreciated. Jesus Mogollon See http://www.soft-switch.org Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Uninstall AMP
I want to give the benefit of doubt to the suggestion as I think there is a misunderstanding of the suggested method of removal of AMP. I guess that he was suggesting to remove it from your linux installation by using the rm -rf command as under cd /var/www rm -rf * which will efectively remove all the web pages associated with the AMP instalation. -Seshu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anders Svensson Sent: Friday, November 04, 2005 1:35 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Uninstall AMP I agree. It is like we newbie's on Asterisk is just trouble for the list members. Pity there is no newbie list. But all were newbie's in the beginning and not so pompous as some on this list Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Ferrell Sent: den 4 november 2005 19:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Uninstall AMP Claudio Canseco wrote: Really is that the way to uninstall FOP and AMP?, thank you i've been looking for an answer about it. Regards Claudio. No Claudio, That will wipe your system. He's being a smartass. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meetme: Sending DTMF to other users in a conference
The script is part of the astGUIclient package(http://astguiclient.sf.net) Here's a direct link to the agi script itself: http://astguiclient.sf.net/experimental_code/agi-dtmf.agi ; this is used for sending DTMF signals within conference calls ;sends the digits to be played in the callerID field ;sound files must be placed in /var/lib/asterisk/sounds exten = 8500998,1,Answer exten = 8500998,2,AGI(agi-dtmf.agi) exten = 8500998,3,Hangup I use a manager API Action call to trigger the agi: In this example 78600051 is the exten to silently enter the meetme conf Action: Originate Channel: local/[EMAIL PROTECTED] Context: demo Exten: 78600051 Priority: 1 Callerid: 123,,456 Hope this helps, MATT--- On 11/4/05, Vamsi Pottangi [EMAIL PROTECTED] wrote: Hi Matt, Do you mind sharing that AGI script and the exact procedure in detail with me. I would be very thankful to you. Thanks, ~Vamsi -- Forwarded message -- From: Matt Florell [EMAIL PROTECTED] Date: Nov 4, 2005 10:03 PM Subject: Re: [Asterisk-Users] Meetme: Sending DTMF to other users in a conference To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hello, We wrote a small AGI script to do just this. We just drop it's exten into the conference room and pass it the digits you want played and it will play the audio files of DTMF digits to all participants in the meetme room. It works great for us and we've been using it for over 2 years now. MATT--- On 11/4/05, Vamsi Pottangi [EMAIL PROTECTED] wrote: Hi, I would like to know the possibility of sending DTMF to other users in a meetme. I'm looking at inviting a participant from within the conference, here the participant is another conference bridge. So we need to send PIN to this conference bridge. How can I bypass the IVR detect menu and send DTMF to the other participants. Does careful_write in case of frametype is AST_FRAME_DTMF will work ? Final aim here is to bridge asterisk's meetme and another conference bridge. This I need to do from within the conference. Another usage, say if we are inviting some person from within the conference, if this lands in the company's IVr then there should be some way to send DTMF to that IVR to reach that person. Anybody came across such a scenario ? Thanks, ~Vamsi ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Moments of silence
We have also experienced momentary periods of silence in the middle of phone calls. I'm wondering if this could be related to the IAX peers becoming unreachable? Has anyone experienced moments of silence during a call, and do you know what the causes might be? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem on Data-Connections through Asterisk
Hello again,
I've found the problem. I wrote some entries to the extensions.conf like
--
exten = _5X,1,Dial(Zap/g2/${EXTEN},120,rt)
--
but it seems that Asterisk don't like the timeout and/or option entries (i.e.
120,rt) on digital calls. When i remove this entries like
--
exten = _5X,1,Dial(Zap/g2/${EXTEN})
--
all calls going fine through Asterisk :-)
Thanks
Hans-Peter Straub
i'm using Asterisk as an intermediary between another PBX (Teles) and the
E1 of our Carrier. We us a TE205P Dual Span Card from Digium. All the
analog connections are running fine, but digital data calls from local ISDN
(BRI @ Teles PBX) to a remote syncPPP-dialup via the E1 doesn't work well.
It looks that the connection is answered ok, but no data is going over this
connection, so the handshake for the PPP-Login doesn't go to the server.
I've also tested a rawip connection over ISDN (isdnx Interfaces of Linux)
and the effect is the same. The connection will be made but no data goes
through it.
When i conect the Teles-PBX directly to the E1 the calls and all data is
going through the line.
I've played with some options i've found in the Mailinglist in the Dial-Tag
but without any success.
Do someone have any idea if this fault can be removed or is there no chance
to get this working?
Thanks a lot
Hans-Peter Straub
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RE: [Asterisk-Users] Uninstall AMP
That doesn't solve much. What I want to do is to stop using AMP and FOP. Best way perhaps is to alter start and stop script but I can't find any info about how to do that Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: den 4 november 2005 19:48 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Uninstall AMP I want to give the benefit of doubt to the suggestion as I think there is a misunderstanding of the suggested method of removal of AMP. I guess that he was suggesting to remove it from your linux installation by using the rm -rf command as under cd /var/www rm -rf * which will efectively remove all the web pages associated with the AMP instalation. -Seshu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anders Svensson Sent: Friday, November 04, 2005 1:35 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Uninstall AMP I agree. It is like we newbie's on Asterisk is just trouble for the list members. Pity there is no newbie list. But all were newbie's in the beginning and not so pompous as some on this list Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Ferrell Sent: den 4 november 2005 19:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Uninstall AMP Claudio Canseco wrote: Really is that the way to uninstall FOP and AMP?, thank you i've been looking for an answer about it. Regards Claudio. No Claudio, That will wipe your system. He's being a smartass. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SCCP: ServiceURL and Mailbox Notification
Hmm i tried your config but the service url ist still not working.
i have the 7.1 images on the phone.
and the message waiting icon is nothing there too but i have a new message on
the server
On Fri, 04 Nov 2005 11:35:32 -0600
Greg Oliver [EMAIL PROTECTED] wrote:
I had the same issue. Here is a full config from a 4.1.3SR1 CCM for a
7970 - let me knwo if you need any others and I will tftp them off.
Thanks,
Greg
#
[EMAIL PROTECTED] ~ $ cat SEP00127F027D17.cnf.xml
device xsi:type=axl:XIPPhone ctiid=581916804
uuid={0B7DCA2C-453E-4F01-908 A-A3E877A707D2}
devicePool uuid={1B1B9EB6-7803-11D3-BDF0-00108302EAD1}
nameDefault/name
dateTimeSetting uuid={9EC4850A-7748-11D3-BDF0-00108302EAD1}
nameCMLocal/name
dateTemplateM/D/Y/dateTemplate
timeZoneCentral Standard/Daylight Time/timeZone
/dateTimeSetting
callManagerGroup
members
member priority=0
callManager
ports
analogAccessPort2002/analogAccessPort
digitalAccessPort2001/digitalAccessPort
ethernetPhonePort2000/ethernetPhonePort
mgcpPorts
listen2427/listen
keepAlive2428/keepAlive
/mgcpPorts
/ports
processNodeName192.168.2.10/processNodeName
/callManager
/member
member priority=1
callManager
ports
analogAccessPort2002/analogAccessPort
digitalAccessPort2001/digitalAccessPort
ethernetPhonePort2000/ethernetPhonePort
mgcpPorts
listen2427/listen
keepAlive2428/keepAlive
/mgcpPorts
/ports
processNodeName192.168.2.11/processNodeName
/callManager
/member
/members
/callManagerGroup
srstInfo uuid={CD241E11-4A58-4D3D-9661-F06C912A18A3}
nameDisable/name
srstOptionDisable/srstOption
userModifiablefalse/userModifiable
ipAddr1/ipAddr1
port12000/port1
ipAddr2/ipAddr2
port22000/port2
ipAddr3/ipAddr3
port32000/port3
isSecurefalse/isSecure
/srstInfo
mlppDomainId-1/mlppDomainId
mlppIndicationStatusDefault/mlppIndicationStatus
preemptionDefault/preemption
connectionMonitorDuration120/connectionMonitorDuration
/devicePool
loadInformationTERM70.7-0-2-0S/loadInformation
versionStamp{21ECCF08-13DB-4EC5-8BCE-B177569C489B}/versionStamp
userLocale
nameEnglish_United_States/name
uid1/uid
langCodeen/langCode
version4.1(3)/version
winCharSetiso-8859-1/winCharSet
/userLocale
networkLocaleUnited_States/networkLocale
networkLocaleInfo
nameUnited_States/name
uid64/uid
version4.1(3)/version
/networkLocaleInfo
deviceSecurityMode1/deviceSecurityMode
idleTimeout0/idleTimeout
authenticationURLhttp://192.168.2.10/CCMCIP/authenticate.asp/authenticationUR
L
directoryURLhttp://192.168.2.10/CCMCIP/xmldirectory.asp/directoryURL
idleURL/idleURL
informationURLhttp://192.168.2.10/CCMCIP/GetTelecasterHelpText.asp/informatio
nURL
messagesURL/messagesURL
proxyServerURL/proxyServerURL
servicesURLhttp://192.168.2.20/CiscoServices/fetchPhoneObject/servicesURL
dscpForCm2Dvce96/dscpForCm2Dvce
dscpForSCCPPhoneConfig96/dscpForSCCPPhoneConfig
dscpForSCCPPhoneServices0/dscpForSCCPPhoneServices
capfAuthMode1/capfAuthMode
capfList
capf
phonePort3804/phonePort
processNodeName192.168.2.10/processNodeName
/capf
/capfList
/device
#
On Fri, 2005-11-04 at 15:14 +0100, René Enskat [Teamware GmbH] wrote:
Hi.
I tried to configure the ServiceURL on the asterisk inside the xml but
i can't get it ro work i always get the errror hos tnot found and the
ServiceURL field in the telephone is empty.
I tried to put it in den SEPxx AND XmlDedault config without success.
This is the url:
http://phone-xml.berbee.com/menu.xml
In my old 7960 i always get a lettersymbol at my line when i got a
mailboxmessage via SIP but this won'z be with the sccp protocol?
Or how cna i have this symbols there?
I have new voicemessages on my asterisk but the telephone is saying
nothing about that.
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