[asterisk-users] Outgoing phone calls muffled
Hello, Several people report that outgoing phone calls to our clients sound muffled, like they are talking underwater. Reported for both the Snom 870, and the polycom ip650. Incoming calls sound ok. Could this be a codec problem? My dialplan looks like: [general] port = 5060 bindaddr = 0.0.0.0 srvlookup = no tos_sip = cs7 tos_audio = ef registertimeout = 1 relaxdtmf = yes context = testofidea disallow = all ;allow=gsm allow = ulaw allow = alaw allow = g722 dtmfmode=rfc2833 ;; allows use of pushbuttoms ;dtmfmode = inband nat = no localnet = 10.0.0.0/255.0.0.0 canreinvite = no Thanks for any help. Best Eddie -- Eddie H. Mikell Senior Systems Engineer RKG Office: 434.970.1010 x 124 Email: emik...@rimmkaufman.com -- http://www.rimmkaufman.com http://twitter.com/rimmkaufman http://www.linkedin.com/company/85385 http://plus.google.com/104980442218952272663/posts http://www.facebook.com/rimmkaufman http://www.RKGblog.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outgoing phone calls muffled
sip show channels shows some info about active sip channels, the current codec included. What does it say? jg jg, sip show channels reports the Format as being ulaw for 17 active calls. Holds - no Peer User/ANR Call ID Format Hold Last MessageExpiry Peer xx kbrown (ulaw) No Rx: ACK219 mpetulla xxx (ulaw) No Tx: ACK240 xx xx 0035bf5711b0186 (ulaw) No Rx: ACKia.ntelos. etc. Thanks, Eddie -- Eddie H. Mikell Senior Systems Engineer RKG Office: 434.970.1010 x 124 Email: emik...@rimmkaufman.com -- http://www.rimmkaufman.com http://twitter.com/rimmkaufman http://www.linkedin.com/company/85385 http://plus.google.com/104980442218952272663/posts http://www.facebook.com/rimmkaufman http://www.RKGblog.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Tired of dropouts and garbled phone calls - where to go next?
All, The users in our organization are well, quite frankly, sick of phone service that is being provided. The choppy phone calls, and drop outs are detrimental to our sales force. I've tried about everything I can think of. Moved the asterisk server from VM machine to dedicated machine More than enough bandwidth Setting 802.1p = 7 Set Dedicated voice traffic 35% of bandwidth. Not sure what option would be the best Put analog lines in the conference room to avoid the dropouts - leave the sip lines in place for day to day use Hire a consultant Ditch the system and buy a pre-packaged system - RingCentral or some such. There are no local asterisk professionals who can help, and we are a little leery of opening up our system to outside consultants. Anyone else face the above, and finally abandoned Asterisk for a commercial system? We have 167 users. I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the conference rooms. Suggestions welcome. Best Eddie -- Eddie H. Mikell Senior Systems Engineer RKG Office: 434.970.1010 x 124 Email: emik...@rimmkaufman.com -- http://www.rimmkaufman.com http://twitter.com/rimmkaufman http://www.linkedin.com/company/85385 http://plus.google.com/104980442218952272663/posts http://www.facebook.com/rimmkaufman http://www.RKGblog.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] l2tp phones - only in China?
All, I'm looking for sip phones that support something other than openvpn. There are a lot of vendors in China (mainly Alibaba) that sell l2tp VPN phones. Are there any American vendors that support l2tp? Thanks, -- Eddie H. Mikell Senior Systems Engineer RKG Office: 434.970.1010 x 124 Email: emik...@rimmkaufman.com -- http://www.rimmkaufman.com http://twitter.com/rimmkaufman http://www.linkedin.com/company/85385 http://plus.google.com/104980442218952272663/posts http://www.facebook.com/rimmkaufman http://www.RKGblog.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk consultant needed in Charlottesville, VA
All: RKG needs an asterisk consultant to help us track down issues we are having with our system. Mainly dropouts and dropped calls. If you have experience in troubleshooting these issues, please contact me at email attached to this messages. Regards, Eddie -- Eddie H. Mikell Senior Systems Engineer RKG Office: 434.970.1010 x 124 Email: emik...@rimmkaufman.com -- http://www.rimmkaufman.com http://twitter.com/rimmkaufman http://www.linkedin.com/company/85385 http://plus.google.com/104980442218952272663/posts http://www.facebook.com/rimmkaufman http://www.RKGblog.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] using ${EXTEN} with waitexten
All:
Some of the people who dial into to our system will press the pound key
when entering an extension for the directory key. When waitexten gets
that, I get an error messages as, for example 123# doesn't match any
extension.
I was going to use ${EXTEN} to just use the first three numbers, but I'm
not sure how to use this with WaitExten.
so I have
exten = 4349701010,1,Answer()
exten = 4349701010,2,ringing
exten = 4349701010,3,wait(8)
exten = 4349701010,4,Background(asterisk-recording)
exten = 4349701010,5,WaitExten(9,m)
exten = 4349701010,6,Dial(SIP/100SIP/123SIP/132SIP/134SIP/149,20)
exten = 4349701010,7,VoiceMail(100@default,u)
exten = 4349701010,8,Playback(vm-goodbye)
exten = 4349701010,9,Hangup()
Where could I check for the extra # keystroke?
Thanks for your help.
eddie
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[asterisk-users] Two asterisk servers, two different service providers
All: I am looking to install another asterisk server in an office located in a different part of the country. I think I can configure the sip and extension conf files, so that the internal phones at the two locations can call each other. My question is this, how do I properly configure the sip file for a different provider at the new location? Can I use a different register statement for the provider at the new location? Can someone point me to some sample conf files that do this? Thanks for all help, AND non smart aleck RTFM answers. Eddie -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Directory routing to wrong extension if dial tones are pressed too quick.
Hi All, Have completely moved off the old ESI system, and things have been going pretty good with the new server. I have one issue, which has been reported by several of our customers. I've tested it, and it does indeed seem to be a problem. When the customer is asked to dial in the first three letters of the person they are trying to reach, they will be routed to the wrong extension. The problem seems to revolve around how quickly the keys are pressed. So if 645 for Mikell is pressed very quickly, they end up being routed to Sarah Fish. But if they take their time, say 2 seconds between each keystroke, everything works ok. Are they any settings that can be adjusted for this? Thanks, Eddie Mikell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk un-registering from provider
All: Starting switching over my phone lines. I got phone line 1 switched. Everyone working. I switched the second phone line, and it worked about an hour, then I started getting errors from the cli saying the server could not register with the providing. I restarted the system, and it worked ok for about 30 minutes, and then started giving he same errors. The error is [Jul 13 11:21:14] NOTICE[27331]: chan_sip.c:10169 sip_reg_timeout:-- Registration for '4342201...@ia.ntelos.net' timed out, trying again (Attempt #19) doing dnsmgr_lookup for 'ia.ntelos.net' It keeps doing this until I restart asterisk. No, the password hasn't changed - the system works fine for anywhere from 5 minutes to 30 minutes, but again, I suspect it depends on the call load. I suspect it has something to do with the call load, but there really wasn't that many calls in progress - maybe 5 at the time of the failure. I have since switched the last phone back to the old system (it handles the volume of our client calls). Anyone else experience this? Where should I start looking - at my server, or at the provider? Thanks, Eddie -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What is the voicemail u option
All: Still trying to get Grandstream to play personal greetings recorded by user - no luck. Someone mentioned the u option. What is that? Something in voicemail.conf? Eddie -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to get asterisk to playback personal greetings using grandstream gxp-2000
All: I am using the standard voicemail in asterisk. Everything works well, except, if a users wants to record their own personal greeting, it doesn't playback. I can see the soundfile being created. I suspect it is a setting in the voicemail.conf, or an option I am over-looking on the grandstream, but if anyone can point me in the write direction, I would certainly appreciate the help. Also, I would like for the user to be able to set up their own password. I set the initial password the same as the extension, so that forces voicemail to prompt for a new password. The problem is, I can see where asterisk is trying to write the password in the voicemail.conf file, but it is denied because the user doesn't have permission. I hate to open /etc/asterisk directory to the incorrect permissions. What would be the best way to enable the user to be able to change their password? Thanks, Eddie -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pattern matching - how to ignore numbers after 10 digits
All:
Yesterday I discovered something interesting. I dialed 1800ANCESTRY
from the asterisk system I am testing and got the number doesn't exist
message. I then dialed the same number from our old system and it went
through.
I realized that the Y in ancestry made the number too long, and went
back to my dialplan.
How do I ignore numbers that are too long? Obviously, I've done
something wrong in my pattern matching.
outgoing part of extensions.conf
exten = _91XX!,1,DIAL(SIP/${EXTEN:1...@ia.ntelos.net) ; long distance
exten = _9765XXX,1,DIAL(SIP/${EXTEN:1...@ia.ntelos.net) ; local
exten = _9XXX,1,DIAL(SIP/${EXTEN:1...@ia.ntelos.net) ; local
exten = _9011XXX!,1,DIAL(SIP/${EXTEN:1...@ia.ntelos.net) ; international
exten = _911,1,DIAL(SIP/${ext...@ia.ntelos.net) ; emergency
Thanks!
Eddie Mikell
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[asterisk-users] Stress Test new system
All: Getting ready to put the system in production. Any suggestions on stress testing the system? I'd like to initiate say 10 sip phone calls to make sure the provider has the bandwidth. Can you do that in CLI? I've called 4 numbers simultaneously with the hard phones I currently have and am thinking of adding 6 or so soft-phones to various pc's to make a total of ten outgoing calls at the same time. Any thing else that can be tested before we go live (total of 60 users)? Thanks, Eddie Mikell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] More clarification on outbound sip channels.
Jim, and all:
Thanks for the response.
If I can repeat what you are saying: you don't have to define the multiple
lines in sip.conf?
For comparison, with my current esi setup, we have 10 outgoing lines. If one
line is busy, then the service just rolls to the next number. This is set up
with the phone service.
That doesn't have to done with outgoing sip lines? Does the dialstatus have to
be checked when a user dials out?
I understand the incoming lines - we will have a block of DID numbers, and I
can check those and send to appropriate extensions.
Thanks all for helping to clarify. I have gotten a couple of users who haven't
been able to call out, and wasn't sure if I wasn't rolling over the sip lines
properly.
Best,
Eddie Mikell
From: Jim Dickensondicken...@cfmc.com
Subject: Re: [asterisk-users] Multiple SIP lines.
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:eda8102c-b255-46e0-940d-1ef217566...@cfmc.com
Content-Type: text/plain; charset=us-ascii
I think it is typical to have some limited number of outbound channels to your
SIP provider. You send all calls, up to your limit, to the same place. The
phone numbers your provider gave you are used to route inbound calls to your
asterisk box. You will typically have some limited number of inbound channels.
All people could call the same number, again controlled by the number of
channels your provider allows. A reason to have multiple inbound (DID) numbers
is so you can route each number to a specific dialplan extension. You might
route one number to the CEO of the company and the other to a voice tree that
allows the caller to specify the person's extension they want to talk with.
-- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On
May 7, 2010, at 11:17 AM, Eddie Mikell wrote:
All:
Still experimenting with the asterisk server for the company.
My local phone company has given me two sip numbers to experiment with,
say 444-456-1234 444-456-5678
Calling in and out works, and I've spread a couple of the phones out
with some co-workers.
My question is this: Do I have to define multiple sip lines in either
the sip.conf or the extensions.conf?
Now when I dial out, I just use
exten = _9.,1,DIAL(SIP/${EXTEN:1...@xx.tracfone.net).
How does it know which sip channel to use?
Hope that is clear.
Thanks for all the help.
Eddie Mikell
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[asterisk-users] Multiple SIP lines.
All:
Still experimenting with the asterisk server for the company.
My local phone company has given me two sip numbers to experiment with,
say 444-456-1234 444-456-5678
Calling in and out works, and I've spread a couple of the phones out
with some co-workers.
My question is this: Do I have to define multiple sip lines in either
the sip.conf or the extensions.conf?
Now when I dial out, I just use
exten = _9.,1,DIAL(SIP/${EXTEN:1...@xx.tracfone.net).
How does it know which sip channel to use?
Hope that is clear.
Thanks for all the help.
Eddie Mikell
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[asterisk-users] Bridging old system (ESI IVX E) with new Asterisk server - it is robbery!
All, Thanks for the suggestions, but the system is a plan non-sip, non-ip, non pri setup. It's pretty much a closed box setup. And the prices for the card and support are robbery - which is why we aren't going to go with another setup like that. While it has been reliable - I don't think there has ever been an issue with it, expansion is expensive. The local company was gouging us with $200 per incident (ie add an extension) service calls, until I found an installation manual on google, and downloaded. They griped because I was using it, but hey, it wasn't that hard to figure out. So might as well jump off the cliff and go full scale asterisk! Thanks, Eddie -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bridging old system (ESI IVX E) with new Asterisk server
All: My company has an existing ESI IVX E-class system with 45 phones. I can add one more card, to expand it another 6 phones, but it's $8000, and then the system will have to be replaced. I have the Asterisk server up and running, with 2 sip lines from the local phone service. (Thanks to you guys, it is working great!). I'm pretty sure this is the way the company will move, and I've have installed 8 phones for different people to test. My question is this: Is there some way I can bridge the two systems together, even temporarily? So if Jake is at extension 120 on the ESI system, and Regis is at extensions 155 on the asterisk server, Jake can call Regis and vice versa. I've pondered on this over the week-end, but don't see an easy way to handle this. Thanks! Eddie Mikell Senior Systems Engineer The Rimm-Kaufman Group -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] A matter of Context
Message: 15 Date: Mon, 19 Apr 2010 17:46:46 -0400 From: Ryan Bullockrrb3...@gmail.com Subject: Re: [asterisk-users] A matter of context To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: g2y3ceb22f71004191446x23bb96c3x5a3848b06ada4...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Have you tried 'type = friend', might also want to make sure 'allowguest' is set to 'no', as this may be putting guest calls into your default context. -- next part -- An HTML attachment was scrubbed... URL:http://lists.digium.com/pipermail/asterisk-users/attachments/20100419/ab623dd1/attachment-0001.htm -- No luck on those. In fact, if I set allowguest to no, it shuts incoming calls down completely. I can get things to work, but am puzzled why things have to go under the default context. I haven't received any other replies, so it is either so obvious, or no one else knows either best Eddie -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] I figured it out!!
If you do not put a context in the beginning of the sip.conf file, the default is, ta da, default in extensions.conf. Putting a context=testof idea in sip.conf got things moving: sip.conf [general] port=5060 bindaddr=0.0.0.0 ;10.8.0.34 *context=testofidea* srvlookup=yes disallow=all ;read somewhere you have to disallow everything first allow=ulaw allow=alaw allow=gsm dtmfmode=rfc2833 ;; allows use of pushbuttoms nat=no externip=64.4.127.106 localnet=10.0.0.0/255.0.0.0 canreinvite=no extensions.conf [general] autofallthrough=no [default] [testofidea] ;exten = br549,1,Dial(SIP/151,20) exten = br549,1,Answer() exten = br549,2,Background(tt-somethingwrong) If you don't put that, it defaults to, well default! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] A matter of context
All:
I've starting building an asterisk system for our company, which has
about 60 users. I am new to asterisk, so thank you for your patience.
I've stripped the sip.conf and the extensions.conf down to the bare minimum:
Here is my extensions.conf file
[globals]
[general]
autofallthrough=no
[default]
[fromprovider]
exten = YY,1,Dial(SIP/151,20)
[phones]
exten = 150,1,Dial(SIP/150)
exten = 151,1,Dial(SIP/151)
exten = _X.,1,DIAL(SIP/${ext...@xx.xx.net)
and the matching sip.conf:
[general]
port=5060
bindaddr=0.0.0.0 ;10.8.0.34
srvlookup=yes
disallow=all ;read somewhere you have to disallow everything first
allow=ulaw
allow=alaw
allow=gsm
dtmfmode=rfc2833 ;; allows use of push buttons on grandstream
nat=no
externip=64.4.127.106
localnet=10.0.0.0/255.0.0.0
canreinvite=no
;; register sip service
;
;register = Y:xxx:4...@ia.ntelos.net/YY
; This register statement works also.
;
register = YY:xxx:yyy...@ia.ntelos.net
;
;;;
[151] ; define extension number for grandstreams
type=friend
context=phones
username=eddie
host=dynamic
[150]
type=friend
context=phones
username=regis
host=dynamic
; define sip service
[xx.x.net]
type= peer
username= YY
fromuser= YY
user= phone
host= xx.x.net
fromdomain = xx.x.net
outboundproxy = .x.net
secret = secret
context = fromprovider
I can make outgoing phones ok.
Here's the problem. When I make an incoming phone call, I get a sip
error message stating extension not found.
If I comment out the [fromprovider] context, and leave exten =
YY,1,Dial(SIP/151,20) in the
default context, everything works fine.
Why do the incoming phone calls work ok when defined in the default
context and not in the fromprovider context.
I hope that is clear.
Thanks for any help and tips. And thanks for everything I have gleaned
from others who have answered previous newbie questions.
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