Re: [asterisk-users] Request for help with DISA (Not taking my input number correctly?)
Hi Steve,
Thank you for your response. As you said, i tried. But, no result. Here I am
sending my configuration file.
Contents in Zapata.conf:
[trunkgroups]
[channels]
language=en
context=from-zaptel
signalling=fxs_ks
rxwink=300
usecallerid=yes
relaxdtmf=yes
dtmfmode=rfc2833
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no
busydetect=yes
busycount=6
group=1
Please tell me, if there are any modifications in config. files. So that, i
will test it again. Looking forward to your response. Thank you.
Regards,
Chandra.
Steve Totaro [EMAIL PROTECTED] wrote: Crazy Boy wrote:
Hi,
Thank you for response. I configured DISA and its working sometimes
and not working sometimes. Here I am sending the configuration and
output on Asterisk server console:
Extensions.conf file content:
[custom-CLID]
exten = s,1,Answer
exten = s,2,DigitTimeout(5)
exten = s,3,ResponseTimeout(10)
exten = s,4,Authenticate(1234)
exten = s,5,DISA(no-password|disa-ext)
[disa-ext]
exten = _X.,1,DIAL(IAX2/[EMAIL PROTECTED]/${EXTEN},30,tr)
Output on server console:
-- Playing 'custom/v1' (language 'en')
== CDR updated on Zap/1-1
-- Executing Goto(Zap/1-1, custom-CLID|s|1) in new stack
-- Goto (custom-CLID,s,1)
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing DigitTimeout(Zap/1-1, 5) in new stack
-- Set Digit Timeout to 5
-- Executing ResponseTimeout(Zap/1-1, 10) in new stack
-- Set Response Timeout to 10
-- Executing Authenticate(Zap/1-1, 1234) in new stack
-- Playing 'agent-pass' (language 'en')
-- Playing 'auth-thankyou' (language 'en')
-- Executing DISA(Zap/1-1, no-password|disa-ext) in new stack
-- Executing Dial(Zap/1-1,
IAX2/[EMAIL PROTECTED]/187773456|30|tr) in new stack
-- Called [EMAIL PROTECTED]/187773456
-- Hungup 'IAX2/teliax-1'
== Everyone is busy/congested at this time (1:0/0/1)
-- Hungup 'Zap/1-1'
What is happening is:
1) I called my zap number from my mobile
2) My IVR is responding
3) I entered a extension number to access DISA
4) Asterisk asked the secret (PIN) code to access DISA
5) I entered password of DISA
6) After validating the password, its giving Dial tone to dial a USA
number
7) I entered 17187773456 (This is a toll free number) to test
8) Call is going sometimes and call is not going sometimes. If we
observe on server console, its not taking my input number properly and
taking my input phone number wrongly.
9) I tested from other mobiles also. But, its not taking my input
number as i entered sometimes.
9) What is the wrong?
Please tell me. Looking forward to your response. Thank you.
Regards,
Chandra.
Do you have relaxdtmf set in your zap conf file? Try from a landline
phone and see if you have the same issue. If you have relaxdtmf=yes try
no and test again, or do the opposite. It is obvious that asterisk is
getting dtmf but it is mixing it up.
Also try dialing other companies IVRs and navigating the menus, maybe
your cell phone is just screwed up?
Is the call going through any other boxes that may have DTMF settings
misconfigured? I have seen DTMF come out in doubles (ie you press 911
and asterisk sees 99111).
Thanks,
Steve Totaro
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Re: [asterisk-users] Card don't hangup but Asterisk hangup
Still failing :( 2006/11/25, Leo Ann Boon [EMAIL PROTECTED]: Jesus Jimenez wrote: Hi , I have a problem with a X100, i do a external call to the asterisk server . The dialplan its simple answer and hangup.. when it's done , the telephone which i did the call , is in line but asterisk server is finish. I'll apreciate all your suggestion. Greetings, txus. The asterisk output: -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'Zap/1-1' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' zapata.conf [channels] language=es context=from-pstn signalling=fxs_ks Is your PSTN line really kwelstart? If it is loopstart, please use fxs_ls and busydetect. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FREE DOWNLOAD - PRI / T1 Circuit monitoring
On Thu, Nov 23, 2006 at 01:59:25PM -0500, Paul wrote: I have not created my final web site, but rather put together a quick one which will contain more free Asterisk software and tips as time permits. http://www.siliconvp.us For those who didn't notice it, this is a glorified 'asterisk -rx show span 1' script. Suggestions: * Parse zapata.conf to check which spans are configured * What happens if 'asterisk -rx' itself times out? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Card don't hangup but Asterisk hangup
On Sat, Nov 25, 2006 at 01:38:42AM +0100, Jesus Jimenez wrote: Hi , I have a problem with a X100, i do a external call to the asterisk server . The dialplan its simple answer and hangup.. when it's done , the telephone which i did the call , is in line but asterisk server is finish. What do you mean by finish? Could you verify that Asterisk actually got there using a NoOp in the dialplan? What do you see in 'za show channel 1' ? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Correct syntax to access a shell variable?
The right syntax should be externip=${ENV(MYIP)} but I **think** variables
are only allowed in extensions.* and not in sip.conf.
---
Dominique Dartois
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Larry Alkoff
Envoyé : samedi 25 novembre 2006 04:01
À : Asterisk-users
Objet : [asterisk-users] Correct syntax to access a shell variable?
I would like to access my shell environment variable MYIP from within
sip.conf to put in externip.
I've tried some variations of syntax after reading The Future of Telephony
but it's not working yet.
Should it be
externip=${ENV{$MYIP}}
or some other syntax??
Larry
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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Re: [asterisk-users] Correct syntax to access a shell variable?
On Sat, Nov 25, 2006 at 11:58:01AM +0100, Dominique Dartois wrote:
The right syntax should be externip=${ENV(MYIP)} but I **think** variables
are only allowed in extensions.* and not in sip.conf.
Right, they are.
As a workaround, use a trivial shell script (with sed -i) to rewrite the
IP address in the place you currently want to pass it through the
environment.
In sip.conf :
#include sip-externip.conf
sip-externip.conf:
externip = 1.2.3.4
A command to rewrite it:
sed -i /^externip/s/=.*/= $MYIP/ /etc/asterisk/sip-externip.conf
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Error uninstalling freepbx-panel
Hi I had some backlog on asterisk-users. Anyway: my answer from the users list at xorcom: http://xorcom.com/pipermail/users_xorcom.com/2006-November/000328.html -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Card don't hangup but Asterisk hangup
Hi, I mean that the server finish the action == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' I'm trying to design a mobile-parking infrastructure, (It's for a Finish University Project) I made more test .. When I make a call from a mobile to mi home phone , and hang up the home phone . The line still up. The problem will not belong to asterisk malfunction or bad configuration, it's my telephone line . I'll call to my phone provider and ask Thanks to all !!! P.D.: sorry for my English El Sábado, 25 de Noviembre de 2006 10:50, Tzafrir Cohen escribió: On Sat, Nov 25, 2006 at 01:38:42AM +0100, Jesus Jimenez wrote: Hi , I have a problem with a X100, i do a external call to the asterisk server . The dialplan its simple answer and hangup.. when it's done , the telephone which i did the call , is in line but asterisk server is finish. What do you mean by finish? Could you verify that Asterisk actually got there using a NoOp in the dialplan? What do you see in 'za show channel 1' ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Modem and TDM400P
I have a need to use a standard analog modem to call out in where asterisk and a TDM400P are in use. Thru asterisk and the TDM400P, in other words. Is this even possible? There seem to be some differing opinions. Or is it only reliably possible to run separate copper for this modem, and punch it down at the PSTN interface? Bypass asterisk and the TDM400P in other words. joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to park calls on a specific extension
The valet system gets us partway from what I read, but it still uses the
arbitrary number slots. It still requires the user know to transfer a
call to the valet.
no you can park to a specific number (lotname)
exten =
_6XX,1,ValetParkCall(auto|8${EXTEN:1:2}|180|${CALLEDEXTEN}|1|internal)
; Valet unpark from an extension
exten = _5XX,1,Playback(beep)
exten = _5XX,n,ValetUnParkCall(filo|8${EXTEN:1:2})
[Synopsis]
Valet Park Call
[Description]
ValetParkCall(exten|lotname|timeout[|return_ext][|return_pri][|return_context])
Park Call at exten in lotname until someone calls ValetUnparkCall on the
same exten + lotname
set exten to 'auto' to auto-choose the slot.
[Synopsis]
Valet UnPark Call
[Description]
ValetUnparkCall(exten|lotname)
Un-Park the call at exten in lot lotname use 'fifo' or 'filo' for
auto-ordered Un-Park.
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[asterisk-users] VOIP Consultants wanted to build a Scalable ITSP Architecture Using OpenSource Softwares
Hello Guys We are looking for VOIP Cosultants who can successfully build A Scalable ITSP Architecture Using OpenSource Softwares something like http://www.skyyconsulting.com/itsp_voip_asterisk.php. we are looking for some body who can design build a scallable highly redundant sollution with billing which can handle 2000-5000 calls per second. we use ILBC for SIP-SIP GSM for PSTN. Most of our PSTN hardware is Quintums Cisco Hardware. Servers in use Dell 1850,1950,2950. we need u to give us the complete source code of what u r deploying on our servers(Very Important). UA can be a Softphone or a IP Phone or a PDA with VOIP(wifi). If u think u can a provide a good sollution for our requirement at a low cost please contact me offlist with ur sollution pricing.. Thanx Regards Kishore Chowdary ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to do Call barging with SIP channel
Hello Users I'm planning to do Call Barging and Call snooping , I saw this Feature in asterisk.org. This Barging and Snooping are test for is Agents are replying the Answer or not that I'm guessing Can anybody help me... this Feature .. How to do Call Barging and snooping in SIP Channels , I;m not using any Zaptel Card -- Thanks and Regards Ravi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 28, Issue 131
I will be out of the office until Tuesday December 5th. , I will checking my email late in the evenings and will try to respond the next day. Thank you, Doug Leber ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Rewriting caller ID from database?
Anselm: Try using smartCID (www.generationd.com). You'll get the benefit of ranges of numbers mapping to single ID's (good for corporate blocks), action field for blocking/accepting calls, etc). MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anselm Martin Hoffmeister Sent: Friday, November 24, 2006 4:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: Rewriting caller ID from database? Am Freitag, den 24.11.2006, 22:22 +0100 schrieb Vincent Delporte: At 22:07 22/11/2006 -0700, Marco Mouta wrote: You can do it using AstDB, just load the database with callerid names and numbers and then include a lookup on database in all incoming calls, so you can override whatever you wanted:) Thanks everyone. Indeed, it seems like using the embedded BerkeleyDB-based AstDB is good enough for what I'm trying to do. However, asterisk barfs on the following script that I used to import data: #/bin/bash asterisk -rx database put cidname 1234567 'Me - cellular' asterisk -rx database put cidname 1234567 'Me - home' etc. Do try asterisk -rx database put cidname 12345676 \Me - cellular\ or asterisk -rx 'database put cidname 3871263 Me - home' These quotations seem to work. Hth Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Rewriting caller ID from database?
I use some custom scripts to do database lookups and rewrite CallerID information. Everything works fine with regard to the CID name, however my Cisco 7960 and Linksys SPA-942 phones do not display the calling number. Instead, they display the called number. This makes the phone's call return feature not work. The calling number and name are both properly displayed on all of the softphone clients that I've tried. Here's the format I'm using to set the CallerID. SET CALLERID JONES DARYL A6508701826 If you're using Asterisk 1.2, see this page : http://www.voip-info.org/wiki/view/Setting+Callerid hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk and UK ISDN 30
Thanks Steve, that's helpful. I use Cologne HFC card to connect 2-channel ISDN2e to my PBX. Do I just use the same card and give it 30 channels instead? Neil safeharbour IT Ltd Your IT Department tel: 0845 644 3607 fax: 0845 867 2891 mob: 07812 114784 voip: [EMAIL PROTECTED] email: [EMAIL PROTECTED] web: www.safeharbourit.co.uk The information in this e-mail is confidential and may be legally privileged. It is intended solely for the addressee. Access to this e-mail by anyone else is unauthorised. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. When addressed to our clients, any opinions or advice contained in this e-mail are subject to the terms and conditions expressed in any applicable governing terms of business. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy Sent: 24 November 2006 23:57 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk and UK ISDN 30 On Fri, Nov 24, 2006 at 11:51:41PM -, Neil Tancock wrote: Anyone know if Asterisk will work with ISDN 30 and what sort of device I'll need to connect it? It will work with UK ISDN, but ensure it's EuroISDN and NOT UK ISDN (it's set in the telco switch and can generally be changed). UK ISDN is v85 and EuroISDN v110. ISDN2e (as in basic rate) is the Euro variety. Modern PRI lines should be Euro, but some telcos still provision the older UK variant. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and UK ISDN 30
On 25 Nov 2006, at 13:34, Neil Tancock wrote: Thanks Steve, that's helpful. I use Cologne HFC card to connect 2-channel ISDN2e to my PBX. Do I just use the same card and give it 30 channels instead? Neil No, you will need an E1 capable card. I use one from Digium, but there are others see: http://www.voip-info.org/wiki/view/Asterisk+hardware Tim Panton www.mexuar.net www.westhawk.co.uk/industries/voip.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with sound quality
Hi, I have installed Asterisk with a 4 port digium card. It is working fine but eventhough the sound is clear, the volume is not loud enough. I have tweaked the txgain and rxgain values but it did not make much difference. Please let me know if there are any settings that could help. Regards Thanks, Ullas.G ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Card don't hangup but Asterisk hangup
Hello, The X100P, don't support reverse polarity, I have same problem, then I bougth a TDM. Regards On 11/25/06, txus [EMAIL PROTECTED] wrote: Hi, I mean that the server finish the action == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' I'm trying to design a mobile-parking infrastructure, (It's for a Finish University Project) I made more test .. When I make a call from a mobile to mi home phone , and hang up the home phone . The line still up. The problem will not belong to asterisk malfunction or bad configuration, it's my telephone line . I'll call to my phone provider and ask Thanks to all !!! P.D.: sorry for my English El Sábado, 25 de Noviembre de 2006 10:50, Tzafrir Cohen escribió: On Sat, Nov 25, 2006 at 01:38:42AM +0100, Jesus Jimenez wrote: Hi , I have a problem with a X100, i do a external call to the asterisk server . The dialplan its simple answer and hangup.. when it's done , the telephone which i did the call , is in line but asterisk server is finish. What do you mean by finish? Could you verify that Asterisk actually got there using a NoOp in the dialplan? What do you see in 'za show channel 1' ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DID Provider
I am using DIDx.net as my DID provider but they don't seem to get their act together. A lot of times the phone numbers don't work. How can provide my own DID, my asterisk server is being hosted at a Data center and has a reliable vendor that does my termination and do SIP to SIP and have no T1 channels.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dialing with different speed
Hi all,
I have a VOIP phone with the PA1688 chip; my firmware is V1.42.028.
This IP phone is registered in an Asterisk PBX and I've a problem when I
dialing internal number.
If I dial an internal number, like for example 102, the IP phone takes 35
seconds to send the number to Asterisk; here below the debug output
192.168.0.75: first
192.168.0.75: 1
192.168.0.75: 10
192.168.0.75: 102
192.168.0.75: b3 ce 00 00 00 00 00 00 00 00 06 0d 06 07 69 70
70 68 6f 6e 65 13 02 00 3c
If I dial an external number, the IP phone send the number to Asterisk
immediatly.
I checked all parameters in the firmware but I did not find any solution.
I'm thinking it could depend from my Asterisk PBX, herewith my Asterisk
configuration
Thanks
Emiliano
iax.conf:
bandwidth=low
disallow=all ; same as bandwidth=high
disallow=ulaw
disallow=alaw
allow=gsm
allow=iLBC
allow=Speex
jitterbuffer=yes
dropcount=2
maxjitterbuffer=500
maxexccessbuffer=400
tos=throughput
mailboxdetail=yes
[first]
type=friend
username=first
secret=first
auth=md5
host=dynamic
context=fullaccess
mailbox=101
callerid=first101
[second]
type=friend
username=second
secret=second
auth=md5
host=dynamic
context=fullaccess
mailbox=102
callerid=second102
[megavista]
type=peer
username=account
secret=password
disallow=all
allow=gsm
allow=ilbc
host=217.221.182.66
extenxion.conf:
[general]
static=yes
writeprotect=yes
[globals]
[macro-stdiax]
exten = s,1,Dial(IAX2/${ARG1}|20|Ttr)
[fullaccess]
include = local
[local]
exten = 101,1,Macro(stdiax,first,${EXTEN})
exten = 102,1,Macro(stdiax,second,${EXTEN})
exten = _XX,1,Dial(ZAP/1/${EXTEN})
exten = _X,1,Dial(ZAP/1/${EXTEN})
exten = _XXX,1,Dial(ZAP/1/${EXTEN})
exten = _011.,1,Dial(IAX2/megavista/${EXTEN:3})___
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[asterisk-users] 1.4 svn voicemail bug / crash
I cannot access my voicemail and get the following warning in my console: [Nov 25 10:26:43] WARNING[5628]: app.c:935 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/8900/Old': File exists I have also noticed that Asterisk will crash several minutes later after this warning message. I am using the latest SVN 1.4 branch of Asterisk (Revision 48007) and Zaptel (Revision 1640) on Fedora Core 5 (2.6.18-1.2239.fc5) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialing with different speed
On Saturday 25 November 2006 09:38 am, Androtech wrote: Hi all, I have a VOIP phone with the PA1688 chip; my firmware is V1.42.028. This IP phone is registered in an Asterisk PBX and I've a problem when I dialing internal number. If I dial an internal number, like for example 102, the IP phone takes 35 seconds to send the number to Asterisk Rather look to your phone settings. I have a AT-320. Or better dial your local number and hit the 'call' button which bypasses the internal dial plan of the phone and dials the number. Also - the current version for PA1688 is something like 1.55.XXX Brett ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura phone does not ring
I think it is wrong. You should specify the next hop with some like this S0:[EMAIL PROTECTED] 2006/11/23, Larry Alkoff [EMAIL PROTECTED]: Problem: SPA3000 phone does not ring for incoming PSTN call although I can dial out. I set up my Sipura with the Voxilla Wizard which is pretty good but leaves out some important details. The Voxilla Wizard for Supura SPA3000 gave me a setting for PSTN Tab - Dial Plans - Dial Plan 8 (S0:66610) Should I put extension [66610] in sip.conf with a context in extensions.conf that will contain dialing instructions? Can someone please tell me what the entries under [66610] and the associated context would look like? Or just tell me how to handle this - I'm been stuck for some time with this. The Wizard was nice enough to give detailed settings for sip.conf and extensions.conf but nothing about to handle Dial Plan 8 except You'll need to enter the extension you wish to forward all incoming PSTN calls to on your Asterisk server. I don't understand how to do that. Larry -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 28, Issue 132
I will be out of the office until Tuesday December 5th. , I will checking my email late in the evenings and will try to respond the next day. Thank you, Doug Leber ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CallerID number not being displayed on SIP phones
I'm having trouble with Cisco 7960 and Linksys SPA-942 SIP phones not displaying the Caller-ID number. The Caller-ID name is displayed, but not the number. Instead, the phones always display the value that's set in the fromuser= parameter in sip.conf. If fromuser= is not set, then the literal asterisk is displayed in the calling number field on the telephone sets. Can I dynamically set the fromuser= value to the CallerID number in extensions.conf? How can I solve this problem? Asterisk v1.2.9 Cisco 7960 firmware P0S-3-07-4-00. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linking Asterisk Servers using SIP instead of IAX
I posted a new article on linking Asterisk Servers via SIP instead of IAX on my web site. It is newbie driven, but I think useful for many since the information is in one place. Just search 'Linking Asterisk Servers' and all you will come up with is IAX configurations. http://www.siliconvp.us Regards, Paul Norris attachment: winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to park calls on a specific extension
On Sat, Nov 25, 2006 at 07:17:47AM -0500, marvin horst wrote:
The valet system gets us partway from what I read, but it still uses the
arbitrary number slots. It still requires the user know to transfer a
call to the valet.
no you can park to a specific number (lotname)
exten =
_6XX,1,ValetParkCall(auto|8${EXTEN:1:2}|180|${CALLEDEXTEN}|1|internal)
; Valet unpark from an extension
exten = _5XX,1,Playback(beep)
exten = _5XX,n,ValetUnParkCall(filo|8${EXTEN:1:2})
Definitely better, and I will install this add-on, but still not at the
UI I think most people want, which is, just put the call on hold, and
go somewhere else and push a button to pick it up.
That is not only an eaiser interface, it's actually a more powerful
one, because it gives you the ability to put a call on hold to go
away to check on something, and then decided after the fact that you
want to pick it up from another extension. Indeed, you could create
an interface if you wanted to so that you could pick up the call from
any pstn phone (ie. cell phone) by dialing a magic number and entering
a code, without having decided to explicitly park it first.
The other reason it's superior is that call transfer differs on various
phones, and sometimes transfer to the parking lot doesn't work right,
it's one more thing to go wrong. However, almost all phones have the
same interface for putting a call on hold (a hold button) and it is
more likely to work.
The only downside to the implicit park UI is that somebody else can grab a call
you put on hold that you didn't intend to park. That's not an issue
in a house or small office, though. And it's low risk. For example,
you can wait 5 seconds to put a call into implicit park so that if you
are putting them on hold to do attended transfer, this can be spotted
so that there is no implicit park. Implicit park would require you put
the call on hold and do nothing do it for some amount of time if you
like.
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[asterisk-users] MeetMe, background agi and playing sounds
Hello everyone! I have created an background agi which responds to dtmf 0-9, each key should playback a sound, and it does, but here is the problem. The sound which is played is just played to the person who touches the key, not to everyone else in the conference, does anyone know how i can do so the sound is heard by everyone? One more thing, is there a way to mute/unmute a channel within the AGI? Thank you! -- Jan Eirik Sandnes ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sip reinvite
If canreinvite=yes is specified in sip.conf for 2 sip extensions and call recording is disabled in asterisk, both legs have same codec . Doesit always does native bridging . I am using freepbx . How can i know if a call is going through asterisk or they are bridged directly to each other ? Does sip reinvite gives problems in billing ? Is there any cli command to know that ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DID Provider
I have the same problem. Also, the web interface is really awkward, they don't have DIDs in the countries where I need them (Chile, for example), and the quality of the sound is from bad to unusable, even from the US phone they provide you for free. If I would have the chance, I would have them refund me the money I spent on that service. I am using other services based in US (for example, rapidvox), they work fine and have no hassles like signing NDAs, bad quality, etc. If you know of any other DID wholesale provider, please tell me. Regards, Alex I am using DIDx.net as my DID provider but they don't seem to get their act together. A lot of times the phone numbers don't work. How can provide my own DID, my asterisk server is being hosted at a Data center and has a reliable vendor that does my termination and do SIP to SIP and have no T1 channels. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID Provider
Thanks Alex, I'll try the rapidvox also. I regret ever using didx.net. -- Original message -- From: Alex [EMAIL PROTECTED] I have the same problem. Also, the web interface is really awkward, they don't have DIDs in the countries where I need them (Chile, for example), and the quality of the sound is from bad to unusable, even from the US phone they provide you for free. If I would have the chance, I would have them refund me the money I spent on that service. I am using other services based in US (for example, rapidvox), they work fine and have no hassles like signing NDAs, bad quality, etc. If you know of any other DID wholesale provider, please tell me. Regards, Alex I am using DIDx.net as my DID provider but they don't seem to get their act together. A lot of times the phone numbers don't work. How can provide my own DID, my asterisk server is being hosted at a Data center and has a reliable vendor that does my termination and do SIP to SIP and have no T1 channels.---BeginMessage--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End Message--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FREE DOWNLOAD - PRI / T1 Circuit monitoring
I would think an external program that tried to make a sip call and try diffrent routes etc. would be better or maybe he can add it on. - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, November 25, 2006 11:45 AM Subject: Re: [asterisk-users] FREE DOWNLOAD - PRI / T1 Circuit monitoring On Thu, Nov 23, 2006 at 01:59:25PM -0500, Paul wrote: I have not created my final web site, but rather put together a quick one which will contain more free Asterisk software and tips as time permits. http://www.siliconvp.us For those who didn't notice it, this is a glorified 'asterisk -rx show span 1' script. Suggestions: * Parse zapata.conf to check which spans are configured * What happens if 'asterisk -rx' itself times out? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re:VOIP Consultants wanted to build a Scalable ITSP Architecture
pls visit www.inspiresoftbd.com -- Regards -- M Emran E-mail: [EMAIL PROTECTED] [EMAIL PROTECTED] Web: www.inspiresoftbd.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4 svn voicemail bug / crash
I retested this with 1.4.0-beta3 and I still can't access my voicemail. I dial the voicemail extension and I just get silence for a few seconds and it hangs up. HELP! I have 295 messages in my old mailbox and I want to retrieve my new messages. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: cisco 7961 , asterisk and busy lamp : solved
Max Bergmann schrieb: How can i programming a Cisco 7961 to be used as busy lamp field? my configs : sccp.conf : [devices] type= 7961 tzoffset= 0 autologin = 601 speeddial = *31, Hanna -- other SIP telefon extensions.conf : exten = *31,hint,SIP/hanna exten = *34,hint,SCCP/601 on SIP Telefon ( SNOM 360 ) everything functions good and i have busy lamp when cisco telefon Offhook, but differently does not function any idea ? Any input is greatly appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have solved my problem, thank you for your help ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisknow
Hello, Anyone saw asterisknow, ? Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium Iaxy S100 Factory Default?
Hi All, I have two old S100 units (the blue ones, not the newer black ones). I am trying to reset these to factory default using the following instructions, but it is not working. Does anyone have any other suggestions to reset this model of the adapter? Tried this: 1. Remove all of the cables, except for the power cable. 2. The factory reset button is next to the RJ45 jack. Press and hold the factory reset button for 10 seconds. Do not release the factory reset button until the very last step. 3. While holding the factory reset button, remove the power cable. 4. Continue holding the factory reset button for an additional 5 seconds after the power cable has been removed. 5. Continue holding the factory reset button and plug the power cable back into the s101i (IAXy). 6. Continue holding the factory reset button for an additional 5 seconds. 7. Now you may release the factory reset button. And This: # Unplug phone and network cables from IAXy device # Use a ballpoint pen to press and hold in the recessed reset button on back of unit # Wait 5 seconds # Unplug IAXy but keep reset button depressed # Wait 5 seconds # Reconnect power (only) to IAXy # Wait 5 seconds # Release reset button # Wait 5 seconds # Disconnect power cord # Connect phone and network cables to IAXy device # Reconnect power cord to IAXy I've also been watching the network - I see no DHCP packets, nothing coming from either of them cept for one errored packet - unfortunatly no tcpdump on the router so I can't dig deeper yet.. Thanks, Diwelf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Passing PRI traffic to remote * over IAX
We are moving our office, but our PRI isn't moving for a while yet. I'd like to setup a box at the old office to receive -ALL-- PRI traffic and send it over an IAX trunk to another Trixbox install at the new office. Everything should go, period. Any ideas on a simple dialplan to make this happen? Thanks, -Darren ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Passing PRI traffic to remote * over IAX
Something like this should do (assuming you get 4 digits for DIDs):
oldoffice:
exten = _,1,Dial(IAX2/whatever/${EXTEN})
exten = _,2,Busy();if you get here then something is wrong with
the connection, so busy out.
newoffice:
exten = _,1,Noop(we got this call from the old office)
On 11/25/06, Darren Wright [EMAIL PROTECTED] wrote:
We are moving our office, but our PRI isn't moving for a while yet.
I'd like to setup a box at the old office to receive –ALL-- PRI traffic and
send it over an IAX trunk to another Trixbox install at the new office.
Everything should go, period.
Any ideas on a simple dialplan to make this happen?
Thanks,
-Darren
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[asterisk-users] SOLVED - 1.4 svn voicemail bug / crash
There was a stale lock file in the mailbox directory. This is a bug though. Asterisk should clean up all lock files on startup. Lastly, I can't explain the intermittent crash and wasn't able to catch it using gdb either. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Passing PRI traffic to remote * over IAX
On Sat, Nov 25, 2006 at 08:46:27PM -0500, Darren Wright wrote:
We are moving our office, but our PRI isn't moving for a while yet.
I'd like to setup a box at the old office to receive -ALL-- PRI traffic
and send it over an IAX trunk to another Trixbox install at the new
office. Everything should go, period.
Any ideas on a simple dialplan to make this happen?
Send PRI calls to the context 'pri' (context=pri in zapata.conf), define
an IAX peer to connect to the new box in iax.conf (say, [newboxpeer]),
and use something along the lines of:
[pri]
exten = ._,1,Dial(IAX2/newboxpeer/${EXTEN})
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] 1.4 svn voicemail bug / crash
On Sat, Nov 25, 2006 at 10:57:18AM -0500, Robert La Ferla wrote: I cannot access my voicemail and get the following warning in my console: [Nov 25 10:26:43] WARNING[5628]: app.c:935 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/8900/Old': File exists Dandling lock file? (can that be?) ls -al /var/spool/asterisk/voicemail/default/8900/Old Do you have a problem with a specific mailbox or with all of them? I have also noticed that Asterisk will crash several minutes later after this warning message. I am using the latest SVN 1.4 branch of Asterisk (Revision 48007) and Zaptel (Revision 1640) on Fedora Core 5 (2.6.18-1.2239.fc5) Could you run asterisk with -g and try to get a backtrace of the generated core file? http://www.asterisk.org/doxygen/AstDebug.html -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 issues on 1.4 beta 3
Jason Adams wrote: I just upgraded to the latest beta version and I am running into one problem. We purchased g729a licenses from digium and they aren't loading anymore. If I roll back asterisk to 1.2.10 the codecs work fine. I've downloaded the new 1.4 version of the codec from their website and re-registerd everything with no luck. Here is the error message: error loading module 'codec_g729a.so': /usr/lib/asterisk/modules/codec_g729a.so: undefined symbol: ast_translator_activate I have tried i686, i386, athlon, and athlon-xp versions of the codec but none of them have loaded. Any help would be appreciated. If you use the latest code in the 1.4 branch, this issue should be resolved. It will be fixed in the next beta release (beta4). svn checkout http://svn.digium.com/svn/asterisk/branches/1.4 asterisk-1.4 -- Russell Bryant Software Engineer Digium, Inc. begin:vcard fn:Russell Bryant n:Bryant;Russell org:Digium, Inc. adr:;;150 West Park Loop;Huntsville;AL;35806;USA email;internet:[EMAIL PROTECTED] title:Software Engineer tel;work:+1-256-428-6000 x-mozilla-html:FALSE url:http://www.digium.com version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
