Okay everyone - I'm moving away from using sipura 841 phones. I'm starting to test with Polycom IP 501 phones. We plan to upgrade our server to a dual t1 but, for now we have a digium tdm400P with 4 analog lines coming into it. So my question is will upgrading the IP phones with my existing digium tdm400 card be enough to satisfy my users ? or is it really a combo deal needing to upgrade the TDM card and the phones? Basically, my users say the phone system is unusable as is. The sound quality is choppy and they can't understand people speaking on the other end. I don't want to swap out their IP phones and then find out they are seeing the same issues with that. Any help as always is greatly appreciated everyone. Thanks ! Nora Lavelle -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060227/8b6a5579/attachment.htm
Nora, If you have issues with choppy calls, most likely your issue isn't with your phones or TDM400, but it sounds like you have some issues with your voip trunks and/or network connectivity issues. ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nora Lavelle Sent: Monday, February 27, 2006 1:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] TDM400P digium card Okay everyone - I'm moving away from using sipura 841 phones. I'm starting to test with Polycom IP 501 phones. We plan to upgrade our server to a dual t1 but, for now we have a digium tdm400P with 4 analog lines coming into it. So my question is will upgrading the IP phones with my existing digium tdm400 card be enough to satisfy my users ? or is it really a combo deal needing to upgrade the TDM card and the phones? Basically, my users say the phone system is unusable as is. The sound quality is choppy and they can't understand people speaking on the other end. I don't want to swap out their IP phones and then find out they are seeing the same issues with that. Any help as always is greatly appreciated everyone. Thanks ! Nora Lavelle -------------------------------------------------------------------------------------------- CONFIDENTIALITY NOTICE: This email and any attachments are intended only for the designated recipients. Superior IT Solutions prohibits use, distribution or transmittal by or to an inintended recipient without Superior IT Solution's express written approval. If you are not the intended recipient, please delete this email and notify Superior IT Solutions (support@superior-it.com) This message (And any attachment) has been scanned by F-Secure and Norton Anti-Virus before leaving our mail server. ----------------------------------------------------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060227/542ad761/attachment.htm
>I'm moving away from using sipura 841 phones. I'm starting to test with >Polycom IP 501 phones. We plan to upgrade our server to a dual t1 but, >for now we have a digium tdm400P with 4 analog lines coming into it. So >my question is will upgrading the IP phones with my existing digium >tdm400 card be enough to satisfy my users ? or is it really a combo >deal needing to upgrade the TDM card and the phones? Basically, my users >say the phone system is unusable as is. The sound quality is choppy and >they can't understand people speaking on the other end. I don't want to >swap out their IP phones and then find out they are seeing the same >issues with that.We use IP501's with dual TDM400P's (6 x FXO) here and so far, everyone has been satisfied with performance. No big complaints other than occasional minor echo but nothing's a show stopper. Voice quality has been rated as "very good" rather than "great" as with our old Avaya PBX but for the price, I can't complain too much. I don't even have QoS setup yet so I expect performance to be even better once I get a new switch in.
Thanks dewey. Any feedback on how to debug this issue ? -nora ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dewey Straughn Sent: Monday, February 27, 2006 11:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] TDM400P digium card Nora, If you have issues with choppy calls, most likely your issue isn't with your phones or TDM400, but it sounds like you have some issues with your voip trunks and/or network connectivity issues. ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nora Lavelle Sent: Monday, February 27, 2006 1:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] TDM400P digium card Okay everyone - I'm moving away from using sipura 841 phones. I'm starting to test with Polycom IP 501 phones. We plan to upgrade our server to a dual t1 but, for now we have a digium tdm400P with 4 analog lines coming into it. So my question is will upgrading the IP phones with my existing digium tdm400 card be enough to satisfy my users ? or is it really a combo deal needing to upgrade the TDM card and the phones? Basically, my users say the phone system is unusable as is. The sound quality is choppy and they can't understand people speaking on the other end. I don't want to swap out their IP phones and then find out they are seeing the same issues with that. Any help as always is greatly appreciated everyone. Thanks ! Nora Lavelle ------------------------------------------------------------------------ -------------------- CONFIDENTIALITY NOTICE: This email and any attachments are intended only for the designated recipients. Superior IT Solutions prohibits use, distribution or transmittal by or to an inintended recipient without Superior IT Solution's express written approval. If you are not the intended recipient, please delete this email and notify Superior IT Solutions (support@superior-it.com) This message (And any attachment) has been scanned by F-Secure and Norton Anti-Virus before leaving our mail server. ------------------------------------------------------------------------ ----------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060227/f3b9d93b/attachment.htm
What is your setup? There are a lot of variables. How many VOIP trunks do you have? What is your Internet connection? Are you using G.729 for your voip trunks to cut down on bandwidth usage? Anytime you implement a phone system and are using more then just POTS for calls (IE. Voip trunks, remote extensions, etc.), you need to calculate your bandwidth requirements for your Internet connection. Obviously, if you have a slower connection such as xDSL, Cable, T1, you can't have someone on your network file sharing across the Internet and expect good quality VOIP calls. -Dewey ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nora Lavelle Sent: Monday, February 27, 2006 2:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] TDM400P digium card Thanks dewey. Any feedback on how to debug this issue ? -nora -------------------------------------------------------------------------------------------- CONFIDENTIALITY NOTICE: This email and any attachments are intended only for the designated recipients. Superior IT Solutions prohibits use, distribution or transmittal by or to an inintended recipient without Superior IT Solution's express written approval. If you are not the intended recipient, please delete this email and notify Superior IT Solutions (support@superior-it.com) This message (And any attachment) has been scanned by F-Secure and Norton Anti-Virus before leaving our mail server. ----------------------------------------------------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060227/44731264/attachment.htm
Hi Dewey -
So as those who read this list know I'm very new to voip software. So as
embarrassed as I am to say it. I don't know how to answer all of your
questions. I have no idea how many voip trunks I have or if I'm using
G.729. We have a DSL connection currently. I have 4 analog phone lines
connected to a digium card that's plugged into a dell Here's my
Zapata.conf and extensions.conf file. I'm definitely confused here. Can
y'all tell ? ;-)
zapata.conf:
;
; Zapata telephony interface
;
; Configuration file
[channels]
language=en
context=default
switchtype=national
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=0.0
txgain=0.0
group=1
immediate=yes
channel => 1,2,3,4
extensions.conf:
[incoming]
exten => s,1,Answer();
exten => s,2,Background(ssn-greeting);
exten => *,1,Directory(default)
exten => 205,1,Wait(2)
exten => 205,2,Record(/tmp/asterisk-recording:gsm)
exten => 205,3,Wait(2)
exten => 205,4,Playback(/tmp/asterisk-recording)
exten => 205,5,Wait(2)
exten => 205,6,Hangup
[internal]
exten => 101,1,Macro(stdexten,SIP/101)
exten => 102,1,Macro(stdexten,SIP/102)
exten => 103,1,Macro(stdexten,SIP/103)
exten => 123,1,Macro(stdexten,SIP/123)
exten => 124,1,Macro(stdexten,SIP/124)
exten => 125,1,Macro(stdexten,SIP/125)
exten => 126,1,Macro(stdexten,SIP/126)
exten => 127,1,Macro(stdexten,SIP/127)
exten => 128,1,Macro(stdexten,SIP/128)
exten => 129,1,Macro(stdexten,SIP/129)
exten => 130,1,Macro(stdexten,SIP/130)
exten => 135,1,Macro(stdexten,SIP/135)
exten => 117,1,Macro(stdexten,SIP/117)
exten => 201,1,Macro(stdexten,SIP/201)
; Please begin new extensions here
exten => 250,1,Macro(stdexten,SIP/250)
[voicemail]
exten => 300,1,Ringing
exten => 300,2,Wait(2)
exten => 300,3,System(/var/spool/asterisk/vm/fix_volume.pl)
exten => 300,4,VoicemailMain(ssn-voicemail-greeting)
exten => 300,5,Hangup
[local]
exten => _9NXXXXXX,1,Dial(Zap/g1/${EXTEN:1})
exten => _9NXXXXXX,2,Congestion
[longdistance]
exten => _91NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN:1})
exten => _91NXXNXXXXXX,2,Congestion
; exten => s,103,Hangup
[macro-stdexten]
exten => s,1,Dial(${ARG1},20)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})
exten => s-NOANSWER,2,Goto(default,s,1)
exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN})
exten => s-BUSY,2,Goto(default,s,1)
exten => s-CONGESTION,1,Voicemail(b${MACRO_EXTEN})
exten => s-CONGESTION,2,Goto(default,s,1)
exten => s-.,1,Goto(s-NOANSWER,1)
exten => a,1,VoicemailMain(${MACRO_EXTEN})
[default]
include => incoming
include => internal
include => voicemail
include => local
include => longdistance
________________________________
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dewey
Straughn
Sent: Monday, February 27, 2006 11:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] TDM400P digium card
What is your setup? There are a lot of variables. How many VOIP trunks
do you have? What is your Internet connection? Are you using G.729 for
your voip trunks to cut down on bandwidth usage? Anytime you implement a
phone system and are using more then just POTS for calls (IE. Voip
trunks, remote extensions, etc.), you need to calculate your bandwidth
requirements for your Internet connection. Obviously, if you have a
slower connection such as xDSL, Cable, T1, you can't have someone on
your network file sharing across the Internet and expect good quality
VOIP calls.
-Dewey
________________________________
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nora
Lavelle
Sent: Monday, February 27, 2006 2:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] TDM400P digium card
Thanks dewey. Any feedback on how to debug this issue ?
-nora
------------------------------------------------------------------------
--------------------
CONFIDENTIALITY NOTICE:
This email and any attachments are intended only for the designated
recipients. Superior IT Solutions prohibits use, distribution or
transmittal by or to an inintended recipient without Superior IT
Solution's express written approval. If you are not the intended
recipient, please delete this email and notify Superior IT Solutions
(support@superior-it.com)
This message (And any attachment) has been scanned by F-Secure and
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Unless I am missing something, it looks like you only use pots (Plain Old
Telephone Service) lines for making and receiving calls correct? It doesn't
appear you are using and VOIP termination (Making outbound calls) or origination
(Receiving inbound calls) provider. If you are getting choppy calls and your
extensions are not outside your LAN, you need to troubleshoot you lan and
Asterisk server. Make sure you can ping it with no packet loss or high latency.
That's were I would start. Using a basic configuration (IE. POTS lines,
TDM400, all lan extensions), you really shouldn't have any issues to deal
with. It's pretty straight forward. Is any of this wirelessly connected?
-Dewey
________________________________
From: asterisk-users-bounces@lists.digium.com on behalf of Nora Lavelle
Sent: Mon 2/27/2006 5:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] TDM400P digium card
Hi Dewey -
So as those who read this list know I'm very new to voip software. So as
embarrassed as I am to say it. I don't know how to answer all of your
questions. I have no idea how many voip trunks I have or if I'm using G.729.
We have a DSL connection currently. I have 4 analog phone lines connected to a
digium card that's plugged into a dell Here's my Zapata.conf and
extensions.conf file. I'm definitely confused here. Can y'all tell ?
;-)
zapata.conf:
;
; Zapata telephony interface
;
; Configuration file
[channels]
language=en
context=default
switchtype=national
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=0.0
txgain=0.0
group=1
immediate=yes
channel => 1,2,3,4
extensions.conf:
[incoming]
exten => s,1,Answer();
exten => s,2,Background(ssn-greeting);
exten => *,1,Directory(default)
exten => 205,1,Wait(2)
exten => 205,2,Record(/tmp/asterisk-recording:gsm)
exten => 205,3,Wait(2)
exten => 205,4,Playback(/tmp/asterisk-recording)
exten => 205,5,Wait(2)
exten => 205,6,Hangup
[internal]
exten => 101,1,Macro(stdexten,SIP/101)
exten => 102,1,Macro(stdexten,SIP/102)
exten => 103,1,Macro(stdexten,SIP/103)
exten => 123,1,Macro(stdexten,SIP/123)
exten => 124,1,Macro(stdexten,SIP/124)
exten => 125,1,Macro(stdexten,SIP/125)
exten => 126,1,Macro(stdexten,SIP/126)
exten => 127,1,Macro(stdexten,SIP/127)
exten => 128,1,Macro(stdexten,SIP/128)
exten => 129,1,Macro(stdexten,SIP/129)
exten => 130,1,Macro(stdexten,SIP/130)
exten => 135,1,Macro(stdexten,SIP/135)
exten => 117,1,Macro(stdexten,SIP/117)
exten => 201,1,Macro(stdexten,SIP/201)
; Please begin new extensions here
exten => 250,1,Macro(stdexten,SIP/250)
[voicemail]
exten => 300,1,Ringing
exten => 300,2,Wait(2)
exten => 300,3,System(/var/spool/asterisk/vm/fix_volume.pl)
exten => 300,4,VoicemailMain(ssn-voicemail-greeting)
exten => 300,5,Hangup
[local]
exten => _9NXXXXXX,1,Dial(Zap/g1/${EXTEN:1})
exten => _9NXXXXXX,2,Congestion
[longdistance]
exten => _91NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN:1})
exten => _91NXXNXXXXXX,2,Congestion
; exten => s,103,Hangup
[macro-stdexten]
exten => s,1,Dial(${ARG1},20)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})
exten => s-NOANSWER,2,Goto(default,s,1)
exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN})
exten => s-BUSY,2,Goto(default,s,1)
exten => s-CONGESTION,1,Voicemail(b${MACRO_EXTEN})
exten => s-CONGESTION,2,Goto(default,s,1)
exten => s-.,1,Goto(s-NOANSWER,1)
exten => a,1,VoicemailMain(${MACRO_EXTEN})
[default]
include => incoming
include => internal
include => voicemail
include => local
include => longdistance
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"Nora Lavelle" <nora@silverspringnet.com> wrote:> Thanks dewey. Any feedback on how to debug this issue ?Regarding sound quality issues with Sipura SPA-841 phones and Asterisk, please check out this page I created on the voip-info Wiki: http://www.voip-info.org/wiki/view/Asterisk+SPA-841+Sound+Quality Specifically, if you have not set the SPA-841's "RTP packet size" and "Silence Suppression" parameters correctly, then you _will_ have sound quality issues. After that: Are your phones sharing the same network segments as your non-VoIP ethernet data? Do you have a lot of ethernet traffic? We found that even on a fully switched network, if the SPA-841's received excessive ARP traffic (which is broadcast to all switch segments, even though most other network packets are suppressed), we had periodic "robot voice" sound issues. Check this out, and see if it helps. Thanks! Alan Ferrency pair Networks, Inc. alan@pair.com