Re: [asterisk-users] Finding difficulty in installing Asterisk
Before installing ensure selinux is disabled. Check the link below to understand Selinux in Redhat/Fedora. http://www.redhat.com/docs/manuals/enterprise/RHEL-5-manual/Deployment_Guide-en-US/ch-selinux.html Check below link to disable selinux in Fedora, or google around for ur version of fedora. http://docs.fedoraproject.org/selinux-faq-fc3/ -- Deepak [EMAIL PROTECTED] wrote: Hi Dave, I did make clean and then make. But then when I am giving make install its giving error AVC access denied. I am using Fedora. What may be the problem? Help me.. Thanking you, Preeta Pandey -Original Message- From: [EMAIL PROTECTED] on behalf of Dave Cotton Sent: Fri 1/25/2008 1:39 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Finding difficulty in installing Asterisk On Friday 25 January 2008 05:25:57 Lyle Giese wrote: You need to do a 'make' before the 'make install'. make install will do all that is necessary to install a program including making any files necessary. -- Dave Cotton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Support the World Aids Awareness campaign this month with Yahoo! for Good___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE210P issues
I use TE212P, it shoudl work without errors. I use it with Asterisk 1.2.18 + zaptel-1.2.17.1 On RHEL 4.4 On Dell PowerEdge 850 It may be that the card is bad, try contacting Asterisk support. I had one bad card when I first got it, the 2nd one worked . -- Deepak Jerry Geis [EMAIL PROTECTED] wrote: I have a box with a TE210P. Things work for a while then stop when making call files. I get NOANSWER as the return code (right away). I am running asterisk 1.2.12.1, libpri 1.2.3 and zap 1.2.9.1 When I try to update to newer zaptel the machine locks when loading the zaptel drivers. I tried to manually load the wct1xxp module (I think that is the one for the dual T1 card???) and the machine locks. I am in a remote location so I cannot see if anything is on the console. I tried jumping to 1.4 and the same thing happens. I have updated quite a few asterisk boxes remotely and never had this issue before. Last thing I tried was chkconfig zaptel off, reboot, then try loading in new version and the same thing happened. It locked up. After rebooting I put back the old zaptel and it works again for awhile. What shall I try? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Yahoo! Answers - Get better answers from someone who knows. Tryit now.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] First Time T1 Questions
We switched to T1(PRI) for high bandwidth voice quality, echo
I am using TE212P(which is a dual span Echo Chancellor hardware DTMF). I
have only one PRI connection from PSTN, but I implemented this 6 months agao
when there were no single span cards.
Sangoma just came with one in April, but I didnt wanted to go with that bcos I
havent seen review that the drivers are old but the card is great.
Now when I have a nice setup of PRI with 95 SIP extension to Asterisk. I
recently got A101D(which has Echo cancellor hardware DTMF) for my standby
asterisk.
Bot of these with their current drivers work great for Echo Voice Quality.
But my system(config) had a big issue with DTMF detection, which means when
someone calls main line then trys to punch my extension(123) the asterisk
think its 112 dials that person or a wrong # like 111 which is not an
extension. SO I had to resolvbe this with Digium by enabling hardware DTMF 6
months ago from software DTMF(I am not sure wthere this was asterisk issue of
DTMF, anyways I enabled hardware DTMF in Digium card it worked fine.
But now the new Sangoma card which I bough for backup didnt have the drivers
compatible to enabled the hardware DTMF. SO had songoma give me a custom drive
for their hardwrae DTMF they did within 20-25 days it works.
But you wouldnt find that driver sin Sangoma site, bcos they are still working
on them(for me they fixed for my model-- A101D)
So in my view both are great unless they work. Atleast I have been using
Digium TE212P for 6 months.
Also note your Network QoS is also important, we have seperate switches to
avoid QoS it depends uto organisation wish funding.
Also the type of Desktop VoIP phones you have.
I think I have said lot, let me know if this was helpful or I was just barking
... ha ha ha...
--
Deepak
Michael J. Liberatore [EMAIL PROTECTED] wrote: Hi all, i have been
using asterisk for a few years but i am about to do my first t1 setup. After
terrible quality issues between two business locations, we have decided to
purchase a point to point t1 from the local phone co. The internet is too
crappy, too much lag, queing and jitter. Most calls were dropped.
I was about to order two cisco routers with csu cards and remembered our
wonderful asterisk supports direct t1. I remembered digium and sangoma both
make these cards.
After some problems with a digium fxo card, i just ordered a sangoma a200
with echo cancellation. I was also leaning towards getting the single t1
sangoma card that is $499 from voip supply. But i know digium also makes
one. I was wondering if the digium card works better or much easier with
asterisk? The digium description says you can split the t1 for voice and data
which sounds nice since i will only be using probably 4 channels max of the
t1. Does the sangoma card also do this? I noticed the sangoma card has a 5
year warranty which is nice since i have had multiple digium fxo cards die.
Is there any other reason to get or the other?
Thank you all for your help. I am hoping this opens up a whole new world in
asterisk for me.
-Mike
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Re: [asterisk-users] Receptionists Phone suggestions? (Not Snom370)
I hope 2 things need to be clear.
1) One call per line, needs to be set on the VoIP.
2)We user Polycom 501 for all Desktop Polycom 601 for reception.
http://media.polycom.com/usa/en/products/voice/soundpoint_ip/601/demo/index.html
OK, what I mean by one call per line
-- Polycom of SIP Phones usually comes with 3,6 etc line display for extensions.
-- And each line display can accept/call/hold total of 8 active phone calls per
line. This will cause problem if all is on the same line feed.
--So one needs to accept only one call per line in the VoIP phones config file.
I am not sure how ur line feeds are setup. I just wanted to let u know that
there can be aproblem with transfer if u have multiple calls comming on same
line display.
Or, may be I am wrong in understanding ur email.
--
Deepak
Russell Brown [EMAIL PROTECTED] wrote:
Does anyone have any suggestions for a decent receptionists phone?
Aastra? Grandstream?
Something with (potentially) lots of BLFs, large(ish) screen, headset
and most importantly the ability to transfer calls?
I've installed five Snom 370s that seemed ideal but my client is very
very unhappy as the Snom 370 can't transfer a call correctly if there's
another call coming in (details below if you/re interested). I've
verified this problem with Snom who's response is that the receptionist
should answer all of the incoming calls before trying to do a transfer -
That's just Bonkers!
So... any suggestions?
Details of Snom 370 problem for the record:
Snom370 gets a Call (Call A).
Snom370 answers Call A. Call A wants to be transferred to Phone C.
Snom370 has another call ringing (Call B).
Snom370 presses HOLD button gets Dialtone. Call A is on Hold, Call B
still ringing.
Snom370 Dials Phone C (Call C).
Snom370 talks to Call C.
Snom370 presses TRANSFER.
The display shows:
CallA
CallB
The soft keys now show and . Pressing them does nothing.
When the TRANSFER button is pressed again, CallA is connected to CallB
(the original caller is now talking to the previously unanswered party)
not what one wanted to happen!
It's not difficult to see why my client is throwing their toys out of
the pram and I'm going to have to replace the Snoms at my expense :-(
--
Regards,
Russell
| Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com |
| Peterborough, England | WWW Play: http://www.ruffle.me.uk |
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[asterisk-users] Paging to external speaker like in airports etc...
Hi, I have a production asterisk-1.2.8 system with FreePBX PRI Digium card.
I am looking for a paging system to an external speaker. I can page to
internal Polycom 501 VoIP.
But, what hardware or system do I need to integrate with the asterisk to have
this acheived.
--
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Re: [asterisk-users] Royalty for On Hold Music ?
Thanks Jared, Yes I am using with Asterisk only. So I am using the inbuilt music from Asterisk for onhold. -- Deepak Jared Smith [EMAIL PROTECTED] wrote: On Tue, 2007-07-31 at 06:36 +0100, Deepak Naidu wrote: I think we need to pay for the later, but I am not sure if we need to pay for the inbuilt asterisk(freepbx) on hold music. I'm no lawyer, but here's what I understand. (Please consult with an attorney in your area, and don't consider this legal advice.) The hold music that comes with Asterisk is provided by Digium under license from Freeplay Music Corporation for use in conjunction with the Asterisk software only. It's my understanding that you don't have to pay any kind of royalties to use it, as long as you're using it with Asterisk. You *do* have to pay royalties on music (or MP3 files) by commercial artists. These royalties vary by country. Using commercial music as hold music is considered broadcasting the music, which requires different licensing arrangements with the copyright holder. In the United States, you can buy a license from ASCAP (the American Society of Composers, Authors, and Publishers) to be able to broadcast music from the major record labels. There are also several other places you can get royalty-free music for hold music. I've had good luck looking online, especially at sites like MagnaTune. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your freeaccount today.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE212 or TE220
I am using TE212P with asterisk-1.2.18. It has echo DTMF in hardware to support. I use it on Dell Power Edge 85 no IRQ's ... Ya, just make sure that u get a good card I got the a broken card first time which ddnt work for echo cancellor then RMA'ed it with new one. -- Deepak fateme fatah [EMAIL PROTECTED] wrote: Hi: I want to have conference call with asterisknow and need 2 ports E1.Which Digium card is better?TE212 or TE220.I haven't problem with motherboard. Regards. - Get the Yahoo! toolbar and be alerted to new email wherever you're surfing. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Yahoo! Answers - Get better answers from someone who knows. Tryit now.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Royalty for On Hold Music ?
Hi, Is there any Royalty one needs to pay when using the inbuilt exisimg asterisk on hold music or when using any other mp3 from a music album. I think we need to pay for the later, but I am not sure if we need to pay for the inbuilt asterisk(freepbx) on hold music. -- Deepak - Yahoo! Answers - Get better answers from someone who knows. Tryit now.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuring Sangoma A101D with Asterisk 1.2.18 zaptel-1.2.17.1
It would help to know exactly what Dell Poweredge you were considering. They do vary. I have Dell Power Edge 850 Also how do I enable DTMF hardware detection. There are no drivers which support it. I have the lastest Beta drivers installed, they seem to show yes in the logs, but the hardware DTMF didnt work, so I wrote a mail, to the developer of the drivers he said they are still working in the lab probably have one within a week. Stephen Bosch [EMAIL PROTECTED] wrote: Deepak Naidu wrote: Hi, I have a Dell Power Edge server planning yo buy Sangoma A101D card. To configure with my Asterisk 1.2.18 zaptel-1.2.17.1 Free-PBX setup. It would help to know exactly what Dell Poweredge you were considering. They do vary. If you compile your kernel with SMP and IO-APIC support, you shouldn't have any problems. The Sangoma cards are very tolerant. So I wanted to know the steps any issue which I may come accross if any. I have googled have some docs handy wrt Trixbox-2.2. Just wanted to get some notes from user with custom install setup when used with Asterisk+freepbx+Sangoma. On the hardware side, the experience shouldn't be any different. Also how do I enable DTMF hardware detection. As far as I know, that is the default. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your freeaccount today.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Configuring Sangoma A101D with Asterisk 1.2.18 zaptel-1.2.17.1
Hi,
I have a Dell Power Edge server planning yo buy Sangoma A101D card.
To configure with my Asterisk 1.2.18 zaptel-1.2.17.1 Free-PBX setup.
So I wanted to know the steps any issue which I may come accross if any.
I have googled have some docs handy wrt Trixbox-2.2. Just wanted to get some
notes from user with custom install setup when used with
Asterisk+freepbx+Sangoma.
Also how do I enable DTMF hardware detection.
--
Deepak
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Re: [asterisk-users] Very bad TDMF tone !
OK, tel me put my share of experience. I have a PRI TE212P which had onboard VMP echo cancellation. I am using Asterisk-1.2-18. the DTMF issue was really bad. It rang the wrong extension some tings rang invalid extension. In the begining I didnt knew that it was an DTMF issue. Then to resolve this I enbled hardware DTMF in my TE212P card which worked fine, but still had issue with detecting DTMF Down 'f' signals during voice calls, due to which calls got dropped ? was the calls dropped... this was when during conversation DTMF Down 'f' signal was detected then a FAX line was initiated bcos it saw a 'f' signal, this was bcos in zapata.conf I had allowed FAX. I had to disable FAX in zapata.conf to resolve that issue. But daily I see at least more than 50 counts of DTMF Down 'f' signal getting detected in voice calls. So my question is the common issue which I see in this posting and mine is Asterisk 1.2-18 version, do anyone have same issue. Its bcos of hardware DTMF I am able to use Astersik, else its not worthy of that version to detect DTMF properly changed the DMF relaxed many other options, recompiled the sources to decrease the DTMF threashold value for DTMF(this is for hardware DTMF). -- Deepak Noah Miller [EMAIL PROTECTED] wrote: i am using tdm400P in my office. i tested that TDMF generated by asterisk is so bad. the sound is very soft and quality is so bad. i am using asterisk 1.2.18. most of time, the # key can not be detected correctly. Does anyone has that problem? please give me a hit for that problem! The only time I've heard of that problem is when VoIP is involved. I've never heard of this problem when the call is all analog, like with a TDM400P. 1. If asterisk is detecting DTMF, the parameter relaxdtmf= can affect DTMF detection. 2. Have you checked your handsets on both ends of the call? Some handsets try to filter out DTMF tones. 3. Is voice quiet on your calls, too, or is it just DTMF? It's possible to affect overall signal levels in zapata.conf. Can you post the relevant portion of your zapata.conf? - Noah ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - New Yahoo! Mail is the ultimate force in competitive emailing. Find out more at the Yahoo! Mail Championships. Plus: play games and win prizes.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query
I am not sure what exactly you wish to achieve. Just a basic SIP--to--SIP call
or ?
I am not much into the configs, but ya I can tell you that you can try using
FreePBX or Trixbox kind of setup to write ur Asterisk config file, rather then
u editing them, as it has macros, context etc... which is too high to me. But
the browser interface help a lot understanding the config files later once
configured via FreePBX.
FreePBX -- Its a tool(software which is wrapper over asterisk which gives a
web based interface to manage configure ur asterisk configuration files with
easy understanding.
tixbox-- Its a kind of Asterisk solution which is combination of
asterisk+freepbx+linux+crm tools etc.. for quick Asterisk deployment.
I am not sure whether u know all these if yes, hen excuse me.. but ur mail
sounded u might need this info needed.
[EMAIL PROTECTED] wrote:
Hi,
I am trying to establish call through sip phone between two PC connected to
linux box on which asterisk server is running
1st PC is having IP Adress : 192.168.1.149
2nd PC is having IP Adress : 192.168.1.53
Now, I am tying to dial from 1st PC to 2nd PC
I am trying to dial from 1st PC to 2nd PC through asterisk server
The problem is 1st PC is calling directly to 2nd PC not through asterisk server
I am doing the following additions in configuration files
1) sip.conf
[general]
context=sip
bindport=5060
bindaddr=0.0.0.0
[phone1]
type=friend
host=192.168.1.149
port=5060
nat=yes
dtmfmode=rfc2833
context=sip
[phone2]
type=friend
host=192.168.1.53
port=5060
nat=yes
dtmfmode=rfc2833
context=sip
2) extensions.conf
exten = 11,1,Dial(SIP/phone2,20,tr)
Now, I am calling from sip phone1 by name [EMAIL PROTECTED]
It is not being called through asterisk server running on linux m/c. It is
calling directly. As, I am running sip debub but no packet dumping is taking
place. Can anybody will tell me the error I am doing.
Thanx and regards
sanchal
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Re: [asterisk-users] Ring the second line when 1st line is busy
Do any one any clue.
This is what I need.
I have a Polycom 501 phone, which support multiple lines ie on the LCD you can
see the extensions asssigned to a user as.
555 --- Line 1 -- Extensions which registers with SIP(Asterisk) -- User A
8555 --- Line 2 -- Extensions which registers with SIP(Asterisk) -- User A
So when now someone calls one Extension 555 to User A, scenario as below.
1) If he is busy on the first line Ext 555, then ring Line 2 Ext 8555, if no
response send to voicemail
2) If he doesnt picks the line 1 Ext 555, then send to voicemail rather then
ringging on his second line ie Ext 8555.
This is what I need, if I can dow it with Follow me, then how, if through ring
group how.
Deepak Naidu [EMAIL PROTECTED] wrote: Hi,
I ma using Asterisk 1.2.18 FreePBX 2.2.1. I have assigned every
users in office with Polycom with 2 extensions as below
555
8555
I have configured Follow-me to ring when the users doesn't picks the phone on
line 1(555) after 10 seconds then ring the line 2(8555). But this is not a
standard telephony which I have been advised to change like below.
If someone calls Ext 555 its busy(means on a phone with someone), then only
ring the second line ie 8555 even if that is busy send on voicemail.
If the first line 555 is free no one picks up then let it go on the
voicemail not second line, bcos now no one has picked the phone nor busy.
I could find anyway to do in FreePBX, so was wondering how about doing this.
Thanx for any input.
exten = 555,1,Macro(exten-vm,555,555)
exten = 555,n,Hangup
exten = 555,hint,SIP/555
exten = ${VM_PREFIX}555,1,Macro(vm,555,DIRECTDIAL)
exten = ${VM_PREFIX}555,n,Hangup
--
Deepak
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[asterisk-users] Ring the second line when 1st line is busy
Hi,
I ma using Asterisk 1.2.18 FreePBX 2.2.1. I have assigned every
users in office with Polycom with 2 extensions as below
555
8555
I have configured Follow-me to ring when the users doesn't picks the phone on
line 1(555) after 10 seconds then ring the line 2(8555). But this is not a
standard telephony which I have been advised to change like below.
If someone calls Ext 555 its busy(means on a phone with someone), then only
ring the second line ie 8555 even if that is busy send on voicemail.
If the first line 555 is free no one picks up then let it go on the
voicemail not second line, bcos now no one has picked the phone nor busy.
I could find anyway to do in FreePBX, so was wondering how about doing this.
Thanx for any input.
exten = 555,1,Macro(exten-vm,555,555)
exten = 555,n,Hangup
exten = 555,hint,SIP/555
exten = ${VM_PREFIX}555,1,Macro(vm,555,DIRECTDIAL)
exten = ${VM_PREFIX}555,n,Hangup
--
Deepak
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Re: [asterisk-users] Query
The best person to check with is Digium support. They have support matrix for
Kernel hardware on which ur card will perform.
Please check the compatibility matrix. Should work fine with
http://www.digium.com/en/supportcenter/documentation/viewdocs/TE120P
Digium support. 256-428-6000
[EMAIL PROTECTED] wrote: Hi all,
Can anybody tell me that wether I should install DIGIUM-TE120P card on
redhat9.0 2.4.20-8 or 2.6.18. I am using kernel 2.6.18 but facing problem of
modutils and iptable.
Can anybody help me out of this.
Thanx and Regards
sanchal singh
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Re: [asterisk-users] DTMF detection -- Zaptel
So, I am not sure whether its a zaptel issue. It have TE212P card which has
echo based hardware cancellor.
--
Deepak
Deepak Naidu [EMAIL PROTECTED] wrote: Hi,
I have Asterisk-1.2.18 install with FreePBX more than 75 extnsion,
daily I come accross an issue try resolving them its either user learning
curve or my ignorance.
But, I dont know what to say regarding this issue.
I have my Dial Plan for internal users to have a 3 Digit Extensions.
So instance my Ext is 239 someone dials the main #, its gets the greeting
message to dial 3 digit ext. So when dialing its from my motorola Razor using
T-mobil I try to purposefully hold 2 button For more than a second then dial 3
9 which means dialing my extension 239. But I get an message saying Invalid
option, but in this case should ring my extension.
So I did the same thing running asterisk in debug mode. So there is see that
when dialing 239(that time when I hold 2 button for more than a second) its
sends 2 twice ie 22 then when I press 3 its 233, so I get Invalid option, bcos
there is no extension with 223.
I had to do this bcos I got feedback from many users saying that when
reaching their extension they get these invalid options, so using my phone was
the only way to replicate it. Further contacting Digium support they asked me
to enable. the relaxdtmf=yes option in zapata.conf. I did still the same
issue.
What is this a bug to live with or issue which has a solution.
==
Asterisk DEBUG Message
==
-- Playing 'custom/Greet1' (language 'en')
-- Invalid extension '223' in context 'ivr-2' on Zap/1-1
== CDR updated on Zap/1-1
-- Executing Playback(Zap/1-1, invalid) in new stack
-- Playing 'invalid' (language 'en')
-- Executing Goto(Zap/1-1, loop|1) in new stack
-- Goto (ivr-2,loop,1)
zapata.conf
; If you are having trouble with DTMF detection, you can relax the DTMF
; detection parameters. Relaxing them may make the DTMF detector more likely
; to have talkoff where DTMF is detected when it shouldn't be.
;
relaxdtmf=yes
---
Deepak
==
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[asterisk-users] Invalid DTMF detection -- Invalid Extension Bug or issue
Hi, I have Asterisk-1.2.18 install with FreePBX more than 75 extnsion, daily I come accross an issue try resolving them its either user learning curve or my ignorance. But, I dont know what to say regarding this issue. I have my Dial Plan for internal users to have a 3 Digit Extensions. So instance my Ext is 239 someone dials the main #, its gets the greeting message to dial 3 digit ext. So when dialing its from my motorola Razor using T-mobil I try to purposefully hold 2 button For more than a second then dial 3 9 which means dialing my extension 239. But I get an message saying Invalid option, but in this case should ring my extension. So I did the same thing running asterisk in debug mode. So there is see that when dialing 239(that time when I hold 2 button for more than a second) its sends 2 twice ie 22 then when I press 3 its 233, so I get Invalid option, bcos there is no extension with 223. I had to do this bcos I got feedback from many users saying that when reaching their extension they get these invalid options, so using my phone was the only way to replicate it. Further contacting Digium support they asked me to enable. the relaxdtmf=yes option in zapata.conf. I did still the same issue. What is this a bug to live with or issue which has a solution. == Asterisk DEBUG Message == -- Playing 'custom/Greet1' (language 'en') -- Invalid extension '223' in context 'ivr-2' on Zap/1-1 == CDR updated on Zap/1-1 -- Executing Playback(Zap/1-1, invalid) in new stack -- Playing 'invalid' (language 'en') -- Executing Goto(Zap/1-1, loop|1) in new stack -- Goto (ivr-2,loop,1) zapata.conf ; If you are having trouble with DTMF detection, you can relax the DTMF ; detection parameters. Relaxing them may make the DTMF detector more likely ; to have talkoff where DTMF is detected when it shouldn't be. ; relaxdtmf=yes --- Deepak == - Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your freeaccount today.___ --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] SOLVED -- Accessing Voicemail(Asterisk) -- Remotely either from Cell or Landline
Ya, its not to fancy way but just the easy way which was hidden from my
thoughts/
To access Vm remotely, dial in the number the punch in ur ext at the IVR
greet, the it rings ur ext, then when u get ur VM greet sorry xyz not
available etc.. punch in *followed by VM password, thats it
--
Deepak
Deepak Naidu [EMAIL PROTECTED] wrote:
Hi, I was wondering if we can check the voicemails remotely from a cell or a
landline number.
We have SIP 3 Digit Extensions connected to Asterisk server.
If users are away from Desk need to access voicemails can they dial in to
Asterisk PBX check their messages.
I know one can check through web link even have mailed. Aslo I have checked
regarding DISA, but I am not kind of OK in using DISA now for just voicemails.
Is their any other ways. I am using Free PBX so can I do any thing from
FreePBX to manager it, if not backend configs are fine.
--
Deepak
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[asterisk-users] Accessing Voicemail(Asterisk) -- Remotely either from Cell or Landline
Hi, I was wondering if we can check the voicemails remotely from a cell or a landline number. We have SIP 3 Digit Extensions connected to Asterisk server. If users are away from Desk need to access voicemails can they dial in to Asterisk PBX check their messages. I know one can check through web link even have mailed. Aslo I have checked regarding DISA, but I am not kind of OK in using DISA now for just voicemails. Is their any other ways. I am using Free PBX so can I do any thing from FreePBX to manager it, if not backend configs are fine. -- Deepak - What kind of emailer are you? Find out today - get a free analysis of your email personality. Take the quiz at the Yahoo! Mail Championship.___ --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] Bad Echo between SIP calls
I like the way people replied to this message of mine. It seems this thread is going back to the hybrid echo issue(no this is not the problem). As said by many ZAP is not in picture for SIP--SIP ie Ext-Ext internal calls. To put my inputs I did tons of QA on this issue to ground on whats the source. Its not just the phone or only the network but may be both. I am not sure how Asterisk would contribute to this. At time for a given 2 internal extension there was no echo but suddenly turned up. People dialing on my phone have echo but not on other at the same time I have few phones which I dial no echo. So ya dont know whats wrong. Thanks all for your inputs sharing ur experience. -- Deepak Darryl Dunkin [EMAIL PROTECTED] wrote: This should only be for TDM to TDM calls, SIP to SIP calls don't use the zaptel driver. - From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Tuesday, June 12, 2007 16:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bad Echo between SIP calls I don't see this listed anywhere here in the replies so. In your zapata.conf file try changing: echocancelwhenbridged=no to: echocancelwhenbridged=yes ___ --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 - What kind of emailer are you? Find out today - get a free analysis of your email personality. Take the quiz at the Yahoo! Mail Championship.___ --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] Bad Echo between SIP calls
Sounds crazy right? even was I, more over support guy logged in unloaded the zap modules to test them, still an echo. Ya, I was clear saying that we have SIP--- SIP issue ie internal extension echo problem. It seems the echo with SIP--SIP has many factors. I am just curios to eliminate any possibility of Asterisk failing to cancel the echo. OK, one question here howz the call flow when a SIP---SIP call is established ie. is the connection between 2 phones when an Internal call is made or does the SIP call goes via Asterisk once the SIP--SIP call is establised. -- Deepak Matthew Fredrickson [EMAIL PROTECTED] wrote: On Jun 8, 2007, at 6:00 PM, Deepak Naidu wrote: Hi, We have a PRI connection when its was on test networks we had echo problems withoutside line. So I bought a TE212P card resolve the echo problem. Which did to an extent. Its using asterisk 1.2.18 RHEL4-Update 4. But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear. I am not sure whats the problem. Also there's slight echo when calling Digium support. Totally lost Digium says we need to remove the echo module to resolve SIP echo problems. Then ? the heck we pay for.. Are you sure that they understood that you were having this problem between 2 SIP endpoints? That advice only makes sense to test if one side is Zap and the other side is SIP. --- Matthew Fredrickson Software Engineer Digium, Inc. ___ --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 - Yahoo! Answers - Get better answers from someone who knows. Tryit now.___ --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] which Wifi SIP phones are the good ones
We are planning to buy a wifi SIP phones to work with Asterisk 1.2-18 setup. I would like to get feedback views regarding Linksys WIP300 WIFI IP Phone or any other wifi phones which has been stable. Thanx for any updates. -- Deepak - The all-new Yahoo! Mail goes wherever you go - free your email address from your Internet provider.___ --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] Bad Echo between SIP calls
Hey thanx for sharing your troubleshooting. Ya over days I kind of did some QA. There are SIP--SIP echo's between random phones. We have 75 phones of Polycom 501. I think might be the network or combination of network polycom creating this. Do you have the backup of old setup without this card, which you can install and check what exactly the settings were before. This is an entie new setup by me, the old one was using 1.4 build I am using 1.2 build both are different server. -- Deepak Zeeshan Zakaria [EMAIL PROTECTED] wrote: Once upon a time I used to have a lot of SIP-SIP calls issues, which not always but sometimes included echo problems. There were no zap devices on the server. Googling and struggling to fix it, I found out that it was because of timing issues and ztdummy was not working properly. It had to do something with the kernel and USB modules and something needed to be fixed in BIOS and zaptel settings somewhere (not in zapata or zaptel confs) so that it can have a properly working timing source. I don't remember the details now but I remember I managed to fix it by building a different kernel version on that server after installaing some other version of zaptel, disabling USB modules on the motherboard, fixing something in zaptel Makefile, disabling unused modules in /etc/sysconfig/zaptel. I don't remember what else I did. but echo and other problems disappeared after whatever I did. It was about 2 years ago and I remember how frustrating it was. Anyways, I guess once you upgraded your hardware, something changed in zaptel settings somewhere which is now effecting the SIP-SIP calls and resulting in echo. Do you have the backup of old setup without this card, which you can install and check what exactly the settings were before. Also I recommend going with Sangoma. I hear a lot of bad stories about digium cards imcompatibility with certain motherboards and conflicts with USB modules on the motherboard, and conflicts with IRQs. Thats why When I went for PRI, I used Sangoma. I've used their A101c and A101d cards, and there have never been any issues. ___ --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 - New Yahoo! Mail is the ultimate force in competitive emailing. Find out more at the Yahoo! Mail Championships. Plus: play games and win prizes.___ --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] Bad Echo between SIP calls
Steve I understand your theory. We have Poycom 501 phones. Prior upgrading to PRI we were till date using 4 analog lines connected with TDM card from digium no echo for pure SIP to SIP lines. Now I have TE212P which had onboard echo cancellor. I am trying make myself clear before I blame on any network. B'cos for sure we have a spegati of networks no QoS. Also the intresting thing is if I call from one extension to other dialing the main line then extension the call is crystal clear. but when dialing a direct extension its a hell of echo. -- Deepak Stephen Davies [EMAIL PROTECTED] wrote: On 09/06/07, Deepak Naidu wrote: Ya, I have done that, below is zapata.conf. Also we had an TMP card with analog lines. SIP cals were great on them. now when we switched over. SIP calls have echo.. which shouldnt be at all. If you are getting echo on pure SIP to SIP calls, there's no point in fiddling around with your zapta.conf. That file is for configuring chan_zap, which is used to talk to Zap/ channels. Your calls are SIP to SIP so the zap channel and your PRI aren't being used at all. SIP calls are pure digital 4 wire lines so no electrical (Hybrid) echo will be present. The phones should not generate echo. If they are, they are presumably nasty phones (what kind are they?) and you should get properly made phones. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Yahoo! Answers - Get better answers from someone who knows. Tryit now.___ --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] Bad Echo between SIP calls
Yeah I have made sure its the correct port. We have 75 polycoms currently. ? the SIP-to-SIP echo is there. -- Deepak Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote: Deepak Naidu wrote: Steve I understand your theory. We have Poycom 501 phones. Prior upgrading to PRI we were till date using 4 analog lines connected with TDM card from digium no echo for pure SIP to SIP lines. Now I have TE212P which had onboard echo cancellor. I am trying make myself clear before I blame on any network. B'cos for sure we have a spegati of networks no QoS. Also the intresting thing is if I call from one extension to other dialing the main line then extension the call is crystal clear. but when dialing a direct extension its a hell of echo. Make SURE you have the handset plugged into the handset port of the phone, not the headset port of the phone. ___ --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 - Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your freeaccount today.___ --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] Bad Echo between SIP calls
reinvite is disabled. Also its a Dell PowerEdge 850 server running asterisk
connected to a Cisco switch. other network in company have Cisco Switch.
Also we have approx 75 Polycoms all over.
canreinvite=no
--
Deepak
Steve Totaro [EMAIL PROTECTED] wrote:
v\:* {behavior:url(#default#VML);} o\:* {behavior:url(#default#VML);}
w\:* {behavior:url(#default#VML);} .shape {behavior:url(#default#VML);}
st1\:*{behavior:url(#default#ieooui) }Do you have reinvites
enabled? Are you running this over a linksys four port SoHo router/switch or
something?
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Deepak Naidu
Sent: Saturday, June 09, 2007 4:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bad Echo between SIP calls
Steve I understand your theory. We have Poycom 501 phones. Prior
upgrading to PRI we were till date using 4 analog lines connected with TDM card
from digium no echo for pure SIP to SIP lines.
Now I have TE212P which had onboard echo cancellor.
I am trying make myself clear before I blame on any network. B'cos for
sure we have a spegati of networks no QoS. Also the intresting thing is if I
call from one extension to other dialing the main line then extension the
call is crystal clear. but when dialing a direct extension its a hell of echo.
--
Deepak
Stephen Davies [EMAIL PROTECTED] wrote:
On 09/06/07, Deepak Naidu wrote:
Ya, I have done that, below is zapata.conf. Also we had an TMP card with
analog lines. SIP cals were great on them. now when we switched over.
SIP calls have echo.. which shouldnt be at all.
If you are getting echo on pure SIP to SIP calls, there's no point in
fiddling around with your zapta.conf. That file is for configuring
chan_zap, which is used to talk to Zap/ channels. Your calls are SIP
to SIP so the zap channel and your PRI aren't being used at all.
SIP calls are pure digital 4 wire lines so no electrical (Hybrid)
echo will be present. The phones should not generate echo. If they
are, they are presumably nasty phones (what kind are they?) and you
should get properly made phones.
Steve
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Re: [asterisk-users] Bad Echo between SIP calls
The sip config firmware are the supported one for the existing firmware. If you have any stable working Polycom 501 SIP without echo between SIP--SIP wouldnt mind to share the sip.cfg, sip.ld bootrom would be great, bcos I have not got concreate resolution for this issue. Hope I can resolve this mess. Feels bad when one does best in aggregating things some louzy device screws up... Oh my frustation is comming on mail : -- Deepak C F [EMAIL PROTECTED] wrote: Are the config files you are using with the phones what was meant with that firmware? or did you upgrade the firmware and reused the old config files? On 6/9/07, Steve Underwood wrote: Stephen Davies wrote: On 09/06/07, Deepak Naidu wrote: Ya, I have done that, below is zapata.conf. Also we had an TMP card with analog lines. SIP cals were great on them. now when we switched over. SIP calls have echo.. which shouldnt be at all. If you are getting echo on pure SIP to SIP calls, there's no point in fiddling around with your zapta.conf. That file is for configuring chan_zap, which is used to talk to Zap/ channels. Your calls are SIP to SIP so the zap channel and your PRI aren't being used at all. SIP calls are pure digital 4 wire lines so no electrical (Hybrid) echo will be present. The phones should not generate echo. If they are, they are presumably nasty phones (what kind are they?) and you should get properly made phones. By this measure most phones are nasty. The handset should be echo cancelled, to prevent leakage of the earpiece into the mike. It is getting less and less common to do this, now. Polycoms, Sipuras, Snoms, you name it, they do it badly. Many are not too annoying until someone turns the volume up. Call someone a little hard of hearing and you will hear echo. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 - New Yahoo! Mail is the ultimate force in competitive emailing. Find out more at the Yahoo! Mail Championships. Plus: play games and win prizes.___ --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] Bad Echo between SIP calls
Hi, We have a PRI connection when its was on test networks we had echo problems withoutside line. So I bought a TE212P card resolve the echo problem. Which did to an extent. Its using asterisk 1.2.18 RHEL4-Update 4. But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear. I am not sure whats the problem. Also there's slight echo when calling Digium support. Totally lost Digium says we need to remove the echo module to resolve SIP echo problems. Then ? the heck we pay for... Has anyone come through this issue. -- Deepak - Yahoo! Answers - Get better answers from someone who knows. Tryit now.___ --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] Bad Echo between SIP calls
Ya, I have done that, below is zapata.conf. Also we had an TMP card with analog lines. SIP cals were great on them. now when we switched over. SIP calls have echo.. which shouldnt be at all. [channels] language=en #include zapata_additional.conf context=from-pstn switchtype=national pridialplan=national signalling=pri_cpe faxdetect=incoming usecallerid=yes echocancel=yes callerid=asreceived echocancelwhenbridged=no echotraining=128 ;rxgain=-3.0 ;txgain=-7.0 group=0 channel=1-23 -- Deepak Alex Balashov [EMAIL PROTECTED] wrote: On Sat, 9 Jun 2007, Deepak Naidu wrote: But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear. My impression is that the transcoding that takes place between two purely software SIP calls never goes through the TE212P card. There are probably echo cancellation options you can enable that are relevant to software channels. I distantly recall there even being some stuff youc an uncomment in the source. Totally lost Digium says we need to remove the echo module to resolve SIP echo problems. Then ? the heck we pay for... Not sure why Digium would say that. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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 - All New Yahoo! Mail Tired of unwanted email come-ons? Let our SpamGuard protect you.___ --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] reset Polycom phones remotely
We have Polycom 501 SIP phones. To remotely reboot, one can use the below command. # curl -d ntp=ntp --user Polycom:456 --url http://IP Address of the Polycom/form-submit NOTE: You can get the IP address of Polycom by using the asterisk -rv -x sip show peers command then parse for the IP address(use grep /or awk). -- Deepak Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: An answer to your original question: if you can get someone _to_ the phones, on the polycom 30x's you would hold the VolDn, VolUp, DND, and Hold buttons for a while to reboot. For anyone with the 50x or 60x, you would hold the VolDn, VolUp, Messages, and Hold buttons. Moj Forum wrote: I have provisioned a bunch of Polycom 301 phones to get the config files from my ftp server. Out of the 4 phones 2 get the config file however the other 2 cannot contact the boot server. I have reboot the phones a number of times remotely (the client is 400 km away) through vnc and logging onto the web config internally. No matter what I change on the web config page it is not saved. I feel I need to reset or reformat the phones - if so how can I do this remotely? Can anyone think of a reason why these 2 phones cannot contact the boot server when the other 2 can? Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 - Inbox full of unwanted email? Get leading protection and 1GB storage with All New Yahoo! Mail.___ --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] Issue installing TE212P -- Echo Cancellor not working -- VPM450: Not Present
Hi, I have installed TE212P. Loaded the zaptel modules wc2xxp module for TE212P. The span are up I can make a call, but the echo issue exists, so its same like my old TE110P card. So I called Digium support. They said that the card may be bad or the modules are not loaded for Hardware echo cancellor. He said one should see Octasia VPM successfull message for the hardware echo cancellor to be working. I get this is dmesg(which means hardware echo cancellor module is not loaded. VPM400: Not Present VPM450: Not Present But I dont see any, I just see the below in dmesg. TE2XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE210P (3rd Gen) eth0: no IPv6 routers present About to enter spanconfig! Done with spanconfig! Registered tone zone 0 (United States / North America) About to enter startup! TE2XXP: Span 1 configured for ESF/B8ZS wct2xxp: Setting yellow alarm on span 1 SPAN 1: Primary Sync Source VPM400: Not Present VPM450: Not Present Completed startup! wct2xxp: Clearing yellow alarm on span 1 Zaptel Transcoder support loaded Has any one had this issue with RHEL4-Update 4. Please let me know your views. -- Deepak - New Yahoo! Mail is the ultimate force in competitive emailing. Find out more at the Yahoo! Mail Championships. Plus: play games and win prizes.___ --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] TE212P octastic initialization failure
I think the best way is to conact Digium Hardware support. it seems there may be an IRQ problem. -- Deepak Francois Deppierraz [EMAIL PROTECTED] wrote: Hi, I'm trying to get a TE212 working on a Dell PowerEdge 1850 running Debian etch using the latest release of libpri (1.4.0), zaptel (1.4.2.1) and asterisk (1.4.4). The initilization of the Octasic echo canceller seems to fail when the wct4xxp module is loaded. [...] VPM450: echo cancellation for 64 channels Failed to open chip, code 00103017! VPM450: Failed to initialize [...] By looking in the zaptel code, this error value (0x00103017) means cOCT6100_ERR_OPEN_EXTERNAL_MEM_BIST_FAILED. Is anyone familiar with that problem ? Thanks for your help. --- TE212P card: jumpers are set to E1 mode and nothing is connected to that card at the moment. # uname -a Linux ditti-voipa-serv-1 2.6.18-4-amd64 #1 SMP Fri May 4 00:37:33 UTC 2007 x86_64 GNU/Linux # cat /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 span=2,1,0,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-62 # cat /proc/interrupts CPU0 CPU1 0: 42385 0 IO-APIC-edge timer 6: 3 0 IO-APIC-edge floppy 8: 1 0 IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 14: 64 0 IO-APIC-edge ide0 169: 0 0 IO-APIC-level uhci_hcd:usb1 177: 0 0 IO-APIC-level uhci_hcd:usb2 185: 0 0 IO-APIC-level uhci_hcd:usb3 193: 19 0 IO-APIC-level ehci_hcd:usb4 201: 2148 0 IO-APIC-level ioc0 217: 1153 0 IO-APIC-level eth1 225: 160247 0 IO-APIC-level wct2xxp NMI: 64 42 LOC: 42340 42317 ERR: 0 MIS: 0 # dmesg [...] Found TE2XXP at base address fe7ffc00, remapped to c2004c00 TE2XXP version c01a016a, burst OFF, slip debug: OFF Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x7daa5400 Reg 1: 0x7daa5000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0101 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1300 Reg 8: 0x Reg 9: 0x00ff0001 Reg 10: 0x004a TE2XXP: Launching card: 0 TE2XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE210P (3rd Gen) About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! About to enter startup! TE2XXP: Span 1 configured for CCS/HDB3/CRC4 wct2xxp: Setting yellow alarm on span 1 timing source auto card 0! VPM400: Not Present VPM450: echo cancellation for 64 channels Failed to open chip, code 00103017! VPM450: Failed to initialize Completed startup! About to enter startup! TE2XXP: Span 2 configured for CCS/HDB3/CRC4 wct2xxp: Setting yellow alarm on span 2 timing source auto card 0! SPAN 2: Primary Sync Source VPM400: Not Present Failed to get chip capacity, code 0010305e! Unsupported channel capacity found on VPM module (0). Completed startup! [...] # ztcfg -v Zaptel Version: 1.4.2.1 Echo Canceller: MG2 Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) 62 channels configured. # ___ --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 - New Yahoo! Mail is the ultimate force in competitive emailing. Find out more at the Yahoo! Mail Championships. Plus: play games and win prizes.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hardware Echo cancellor Digium Wildcard TE212P
Hi, I am currently using TE110P Digium card on a PRI card. Basically the echo is so much that one can disticntly identify that. I have tried all the combination if tuning configuration seen in forums etc. I am using MG2 cancellor algorithm also tuned the RX TX gains, still there is an echo. So I am thing to purchase an hardware based echo cancellor like Digium Wildcard TE212P. So in this regards I would like to get some view whether its worth to buy a hardwrae based echo cancellor. Will this resolve the issue, or will be just waste of money. I am using Asterisk 1.2.18 latest version of zaptel drivers. Hope if someone had the same issue, I what has done to resolve it would be much appreciable. -- Deepak - Yahoo! Answers - Got a question? Someone out there knows the answer. Tryit now.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware Echo cancellor Digium Wildcard TE212P
So Steven, did the echo problem stopped once the Hardware echo cancellor card was installed out of the box, or you needed to do some configuration changes like Rx Tx etc. Thanks for sharing your experience. -- Deepak »Steven Ringwald« [EMAIL PROTECTED] wrote: Deepak Naidu wrote: Hi, I am currently using TE110P Digium card on a PRI card. Basically the echo is so much that one can disticntly identify that. I have tried all the combination if tuning configuration seen in forums etc. I am using MG2 cancellor algorithm also tuned the RX TX gains, still there is an echo. So I am thing to purchase an hardware based echo cancellor like Digium Wildcard TE212P. So in this regards I would like to get some view whether its worth to buy a hardwrae based echo cancellor. Will this resolve the issue, or will be just waste of money. I am using Asterisk 1.2.18 latest version of zaptel drivers. Hope if someone had the same issue, I what has done to resolve it would be much appreciable. In my experience, it is well worth the money. After installing several for customers, we never bought the non-HWEC cards again... Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your freeaccount today.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SIP Problems continue...
A small way to make little easy, I dont know it people are ok to that, try integrating freepbx asterisk so you know what the sip configs should look like when things are all well. Things might stop working if there is a bug or change in configs. -- Deepak Ken Williams [EMAIL PROTECTED] wrote: I mean that SIP phones cannot answer incoming calls or make outgoing calls. When a call comes in on ZAP, it actually rings all the phones like normal, but when you try to answer no one is there. In addition, when you try to dial out you eventually get a message on the phones saying unable to communicate with the server. So there is some traffic still traveling on the SIP channel (the server's dialing extensions from an incoming ZAP call) but no further communication...almost as if it's a one way street of communication. The server can send data out on SIP but isn't receiving any. As for your issue, we haven't really had that (thankfully), so I don't think you're heading down the horrible spot we're in right now. Tonight I'm going to remove all aspects of Asterisk and reinstall fresh, if that fails I'll format reinstall the entire box. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Moffett Sent: Wednesday, May 09, 2007 12:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Problems continue... I also get the mysterious SIP INVITE channels. 10.101.2.204 xxx 748e8b0a625 00102/0 unkn No Init: INVITE And I also am running 1.4.4 on CentOS4. Is that a pattern or just coincidence? The other symptom you mention is this ...the SIP phones couldn't communicate with the server, though there was no error message on the server and everything appeared fine on the server. Do you mean no calls in or out until you reboot? I don't have that thankfully, but I do have a guy telling me that incoming audio just goes away for a few seconds at a time. He says also that it sometimes goes away for long enough time that he was mistaking it for a dropped call. But if he waits long enough it pretty generally always comes back. I have consistent solid network performance from the asterisk server to the ATA (and believe me, I've looked very hard for a network problem), and I don't know what to look at next. Incidentally, the guy hasn't called me since I rebooted last week. Is this similar to how your situation started? * Adam Moffett Plexicomm, LLC [EMAIL PROTECTED] ph: 866-759-4678x104 * ___ --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 - Yahoo! Answers - Got a question? Someone out there knows the answer. Tryit now.___ --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] Vista compatibilty in SIP softphones
I have Vista on my new HP laptop X-lite soft phone works like charm with it,
I tried sjphone, I couldnt get that working, its gets hung.
--
Deepak
Chris Bagnall [EMAIL PROTECTED] wrote:
Greetings list,
I've noticed over the last couple of weeks that, unsurprisingly, nearly every
new PC seems to be coming with Vista these days. I expect it'll only be a
matter of time for all of us before clients start needing Vista-compatible
softphones (if it's not already happened).
So, what's the story with Vista compatibility amongst the softphones currently
out there? Ideally, I'd like to find a decent open-source Vista-compatible
softphone, but free, even if closed-source would do the job for the time being.
What are your experiences with SIP softphones under Vista?
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it/chris.html
This email is made from 100% recycled electrons
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[asterisk-users] Re: [Dundi] Dial Plan for Multi-Location Support Queue
Can anyone help on this. -- Deepak Deepak Naidu [EMAIL PROTECTED] wrote: Hi, I am in the process of planning a dial plan, In regards to the requirement, I am confused how to go about the dial plan. The scenario is like below. BRANCH - A - (COMPANY) Line 1 -- Extension 239 Line 2 -- Extension 8239 BRANCH - B - (COMPANY) Line 1 -- Extension 239 Line 2 -- Extension 8239 Now what I need is that if a user in Branch - A wants to dial Branch - B, he just needs to use 88xxx(extension of Branch - B) Similarly, if a user in Branch - B wants to dial Branch - A, he just needs to use 89xxx(extension of Branch - A) In this regards, I am not sure how do I achieve inter brach connection using asterisk to fit my 88 89 prefix dial plan for multi-location. More over, said that, we will have a support Queue in Branch - A(extension 700), users from Branch - B should be able to join the Queue(extension 700) to accept support calls vice-versa, I dont know how this is possible what would my dial plans be. It would be much appreciated if someone can help me resolve this dial plan support issue. Thannks, Deepak - Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your free account today.___ Dundi mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/dundi - New Yahoo! Mail is the ultimate force in competitive emailing. Find out more at the Yahoo! Mail Championships. Plus: play games and win prizes.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial Plan for Multi-Location Support Queue
Hi, I am in the process of planning a dial plan, In regards to the requirement, I am confused how to go about the dial plan. The scenario is like below. BRANCH - A - (COMPANY) Line 1 -- Extension 239 Line 2 -- Extension 8239 BRANCH - B - (COMPANY) Line 1 -- Extension 239 Line 2 -- Extension 8239 Now what I need is that if a user in Branch - A wants to dial Branch - B, he just needs to use 88xxx(extension of Branch - B) Similarly, if a user in Branch - B wants to dial Branch - A, he just needs to use 89xxx(extension of Branch - A) In this regards, I am not sure how do I achieve inter brach connection using asterisk to fit my 88 89 prefix dial plan for multi-location. More over, said that, we will have a support Queue in Branch - A(extension 700), users from Branch - B should be able to join the Queue(extension 700) to accept support calls vice-versa, I dont know how this is possible what would my dial plans be. It would be much appreciated if someone can help me resolve this dial plan support issue. Thannks, Deepak - Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your freeaccount today.___ --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
