RE: [Asterisk-Users] TDM400P digium card
Title: RE: [Asterisk-Users] TDM400P digium card
Thank you so much. we aren't using wireless. just our LAN. So it sounds like network connectivity and possibly my asterisk server. Thanks for helping to pointme in the right direction I so apprciate it.
-nora
-Original Message-
From: [EMAIL PROTECTED] on behalf of Dewey Straughn
Sent: Mon 2/27/2006 4:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] TDM400P digium card
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: [EMAIL PROTECTED] 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 => _9NXX,1,Dial(Zap/g1/${EXTEN:1})
exten => _9NXX,2,Congestion
[longdistance]
exten => _91NXXNXX,1,Dial(Zap/g1/${EXTEN:1})
exten => _91NXXNXX,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
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 ([EMAIL PROTECTED])This message (And any attachment) has been scanned by F-Secure and NortonAnti-Vir
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 =>
_9NXX,1,Dial(Zap/g1/${EXTEN:1})
exten => _9NXX,2,Congestion
[longdistance]
exten => _91NXXNXX,1,Dial(Zap/g1/${EXTEN:1})
exten => _91NXXNXX,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:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] 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:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] 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
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RE: [Asterisk-Users] TDM400P digium card
Thanks dewey. Any feedback on how to debug this issue ? -nora From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 ([EMAIL PROTECTED]) This message (And any attachment) has been scanned by F-Secure and Norton Anti-Virus before leaving our mail server. - ___ --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] 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk vs. Traditional PBX
Thanks everyone ! One more question if I do go with a PRI line. Will my existing TDM card from digium work or do I need to purchase a different card to handle this ? Thanks Nora Lavelle -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Friday, February 10, 2006 4:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk vs. Traditional PBX I am sorry. I thought you wrote Dell 850. Should have looked closer. The machine should do just fine. However it would not hurt to ptu in another gig. Also see if anyone else on the list has used a 650 and what expiriences they have had. Regards, Dovid --- Nora Lavelle <[EMAIL PROTECTED]> wrote: > > Hi Dovid, > > Thank you for the book. I'm already reading it. > > I have a dell 650 server, 1Gig of memory, 1 CPU > (3.07Ghz). What > hardware would you recommend for the 200 users w/ > about 20 concurrent > calls ? > > As always I thank you so much for your help. > > Nora Lavelle > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On > Behalf Of Dovid > Bender > Sent: Thursday, February 09, 2006 2:02 PM > To: Asterisk Users Mailing List - Non-Commercial > Discussion > Subject: Re: [Asterisk-Users] Asterisk vs. > Traditional PBX > > I think your problem is the Dell 650. What are the > specs on it ? If you want a system that can support > 200 users you will need to do a lot better than > that. > Also you will be dealing with T1's/E1's and not POTS > lines. I think a good place to start (if you havent > already) is the book that has come out a while back. > I > have it on my server at > http://www.h6315.com/ast_book/ > > Regards, > Dovid > (I posted my server and not from the publisher > becuase > I do not know thier URL and I have email access only > now.) > > --- Nora Lavelle <[EMAIL PROTECTED]> wrote: > > > > > Hi everyone ! > > > > So here's my question of the day ! I need to make > a > > decision on whether or not to go to a voip > solution > > or configure an existing pbx (norstar) that my > > company has available. We are a small startup. > I'm > > wanting a solution that will support up to about > 200 > > people, with direct dial-in capability, up to > about > > 30 concurrent phone calls and good voice quality. > > Right now I have an asterisk deployment with about > > 15 people on it. We have sipura 841 phones. The > > biggest issue currently is voice quality. lot of > > complaints there. I have a dell 650 poweredge > > (single processory system), with a digium tdm400 > > card and 4 analog lines plugged into it. > > > > So here are my questions: > > > > * Is asterisk a good solution for my company ? or > > should I just install the traditional pbx and look > > to move to asterisk in a couple of years ? (I > > personally would prefer asterisk cuz I'm a unix > > person not a phone person so from a manageability > > perspective i would love this ) > > > > * If I were to go to an asterisk solution to > support > > about 200 people with the requirements above what > > hardware platform would you recommend ? I'm > > guessing I'd need a PRI line and a different > digium > > card? Also would a 1cpu poweredge dell be enough ? > > or would that have to be upgraded too ? > > > > If anyone is running an environment similar to > this > > that can provide help I would really appreciate > > this. I'm having a hard time making this decision > > and would love to hear anybody's experience in a > > real time environment. > > > > Thanks again this list ROCKS! > > Nora Lavelle > > > > > > > ___ > > --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 > > > > > __ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam > protection around > http://mail.yahoo.com > ___ > --Bandwidth and Colocation provided by Easynews.com > -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > _
RE: [Asterisk-Users] Asterisk vs. Traditional PBX
Hi Dovid, Thank you for the book. I'm already reading it. I have a dell 650 server, 1Gig of memory, 1 CPU (3.07Ghz). What hardware would you recommend for the 200 users w/ about 20 concurrent calls ? As always I thank you so much for your help. Nora Lavelle -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Thursday, February 09, 2006 2:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk vs. Traditional PBX I think your problem is the Dell 650. What are the specs on it ? If you want a system that can support 200 users you will need to do a lot better than that. Also you will be dealing with T1's/E1's and not POTS lines. I think a good place to start (if you havent already) is the book that has come out a while back. I have it on my server at http://www.h6315.com/ast_book/ Regards, Dovid (I posted my server and not from the publisher becuase I do not know thier URL and I have email access only now.) --- Nora Lavelle <[EMAIL PROTECTED]> wrote: > > Hi everyone ! > > So here's my question of the day ! I need to make a > decision on whether or not to go to a voip solution > or configure an existing pbx (norstar) that my > company has available. We are a small startup. I'm > wanting a solution that will support up to about 200 > people, with direct dial-in capability, up to about > 30 concurrent phone calls and good voice quality. > Right now I have an asterisk deployment with about > 15 people on it. We have sipura 841 phones. The > biggest issue currently is voice quality. lot of > complaints there. I have a dell 650 poweredge > (single processory system), with a digium tdm400 > card and 4 analog lines plugged into it. > > So here are my questions: > > * Is asterisk a good solution for my company ? or > should I just install the traditional pbx and look > to move to asterisk in a couple of years ? (I > personally would prefer asterisk cuz I'm a unix > person not a phone person so from a manageability > perspective i would love this ) > > * If I were to go to an asterisk solution to support > about 200 people with the requirements above what > hardware platform would you recommend ? I'm > guessing I'd need a PRI line and a different digium > card? Also would a 1cpu poweredge dell be enough ? > or would that have to be upgraded too ? > > If anyone is running an environment similar to this > that can provide help I would really appreciate > this. I'm having a hard time making this decision > and would love to hear anybody's experience in a > real time environment. > > Thanks again this list ROCKS! > Nora Lavelle > > > > ___ > --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 > __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk vs. Traditional PBX
Thanks so much to all of you this has helped me out immensely ! Have a great day ! Nora -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of William Sent: Thursday, February 09, 2006 7:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk vs. Traditional PBX Kerry is right on. We use a similar config in dozens of installs with 200 users and it just cruises. Consider adding a duplicate server for failover at some point. -Original Message- From: "Kerry Garrison" <[EMAIL PROTECTED]> Date: Thu, 9 Feb 2006 06:57:32 To:"'Asterisk Users Mailing List - Non-Commercial Discussion'" Subject: RE: [Asterisk-Users] Asterisk vs. Traditional PBX Hi everyone ! So here's my question of the day ! I need to make a decision on whether or not to go to a voip solution or configure an existing pbx (norstar) that my company has available. We are a small startup. I'm wanting a solution that will support up to about 200 people, with direct dial-in capability, up to about 30 concurrent phone calls and good voice quality. Right now I have an asterisk deployment with about 15 people on it. We have sipura 841 phones. The biggest issue currently is voice quality. lot of complaints there. I have a dell 650 poweredge (single processory system), with a digium tdm400 card and 4 analog lines plugged into it. [Kerry Garrison sayeth] The 841 is fine for testing but I would never put one on a clients desk. The sound quality is bottom of the barrel. Combine that with the TDM400 card and its a wonder anyone will use the phone system at all. Move up to the Linksys SPA941 or SPA942 or the Polycom 501 and then use a different interface such as the Mediatrix 1204 or a PRI and your users will be singing your praises till the end of time. So here are my questions: * Is asterisk a good solution for my company ? or should I just install the traditional pbx and look to move to asterisk in a couple of years ? (I personally would prefer asterisk cuz I'm a unix person not a phone person so from a manageability perspective i would love this ) [Kerry Garrison sayeth] Asterisk is a great solution for your company and you will have many more benefits than the Northstar system. * If I were to go to an asterisk solution to support about 200 people with the requirements above what hardware platform would you recommend ? I'm guessing I'd need a PRI line and a different digium card? Also would a 1cpu poweredge dell be enough ? or would that have to be upgraded too ? [Kerry Garrison sayeth] You would want a beefier machine and at least one PRI. Its not the number of people, its the number of concurrent phone calls. I see businesses with 100 people and they average 5-7 concurrent calls and I have clients with 15 people that average 12-15 concurrent calls. If anyone is running an environment similar to this that can provide help I would really appreciate this. I'm having a hard time making this decision and would love to hear anybody's experience in a real time environment. [Kerry Garrison sayeth] My largest install is approaching 55 users, with the PRI and Polycom 501's they couldnt be happier. The system is on a nice 2.8ghz XEON system with 2gb of RAM and at peak times the server is basically idle. Thanks again this list ROCKS! Nora Lavelle ___ --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 Sent from my BlackBerry - please excuse any typos. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk vs. Traditional PBX
Title: Asterisk vs. Traditional PBX Hi everyone ! So here's my question of the day ! I need to make a decision on whether or not to go to a voip solution or configure an existing pbx (norstar) that my company has available. We are a small startup. I'm wanting a solution that will support up to about 200 people, with direct dial-in capability, up to about 30 concurrent phone calls and good voice quality. Right now I have an asterisk deployment with about 15 people on it. We have sipura 841 phones. The biggest issue currently is voice quality. lot of complaints there. I have a dell 650 poweredge (single processory system), with a digium tdm400 card and 4 analog lines plugged into it. So here are my questions: * Is asterisk a good solution for my company ? or should I just install the traditional pbx and look to move to asterisk in a couple of years ? (I personally would prefer asterisk cuz I'm a unix person not a phone person so from a manageability perspective i would love this ) * If I were to go to an asterisk solution to support about 200 people with the requirements above what hardware platform would you recommend ? I'm guessing I'd need a PRI line and a different digium card? Also would a 1cpu poweredge dell be enough ? or would that have to be upgraded too ? If anyone is running an environment similar to this that can provide help I would really appreciate this. I'm having a hard time making this decision and would love to hear anybody's experience in a real time environment. Thanks again this list ROCKS! Nora Lavelle ___ --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] Calls fading in and out
Hi John - Good call. yes I'm using spa841s. Did you end up replacing them or did you find a fix ? thanks so much ! nora -Original Message- From: [EMAIL PROTECTED] on behalf of John Cianfarani Sent: Fri 2/3/2006 2:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Calls fading in and out What model phones are you using? I've noticed this before on spa841's. John From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nora Lavelle Sent: Friday, February 03, 2006 3:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Calls fading in and out Hi again everyone ! Was wondering if anyone had any pointers on how to debug voice quality issues in asterisk. I've got a user who either can't be heard on her phone calls (outgoing and incoming) and today someone that called her said that her voice was coming in and out. Any pointers or suggestions are appreciated! Thanks so much again ! This list has been so helpful to me. Nora Lavelle <>___ --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] Calls fading in and out
Hi again everyone ! Was wondering if anyone had any pointers on how to debug voice quality issues in asterisk. I’ve got a user who either can’t be heard on her phone calls (outgoing and incoming) and today someone that called her said that her voice was coming in and out. Any pointers or suggestions are appreciated! Thanks so much again ! This list has been so helpful to me. Nora Lavelle ___ --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] extension to extension dialing
Hmm.. I definitely have type=friend in the sip.conf and I added
qualify=yes but, I think that's default anyways.. When I call from the
outside and enter his extension it goes through to him fine but, when I
go extension to extension it automatically goes to voicemail.. Here are
the messages from the console:
-- Executing Macro("SIP/130-58df", "stdexten|SIP/124") in new stack
-- Executing Dial("SIP/130-58df", "SIP/124|20") in new stack
-- Called 124
Jan 27 10:27:10 WARNING[28243]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Critical Request)
== No one is available to answer at this time
-- Executing Goto("SIP/130-58df", "s-NOANSWER|1") in new stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing VoiceMail("SIP/130-58df", "u124") in new stack
-- Playing 'voicemail/default/124/greet' (language 'en')
Jan 27 10:27:10 NOTICE[28243]: sched.c:290 ast_sched_del: Attempted to
delete non-existant schedule entry 22838!
-- Playing 'vm-isunavail' (language 'en')
-- Playing 'vm-intro' (language 'en')
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary
Richardson
Sent: Thursday, January 26, 2006 6:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] extension to extension dialing
In your sip.conf, make sure these phones have a Type=Friend entry and
a qualify=yes. I don't think the qualify=yes is required, but it helps
in debuging.
About the port, I'm not too sure about sipura and snom phones (I only
have Cisco phones :(). That could have something to do with it..
On 1/26/06, Nora Lavelle <[EMAIL PROTECTED]> wrote:
>
> Hi there gary. thanks so much for your help. we're using sipura-841
and snom 320s.
>
> Here's the sip show peers.. that's weird that extension 130 has port
2057.. could that be the problem ?
>
> -nora
>
> Name/usernameHostDyn Nat ACL Mask Port
Status
>
> 201/201 10.200.0.56 D 255.255.255.255 5060
Unmonitor
> ed
> 130/130 10.200.0.10 D 255.255.255.255 2057
Unmonitor
> ed
> 129/129 10.200.0.5 D 255.255.255.255 5060
Unmonitor
> ed
> 127/127 10.201.0.30 D 255.255.255.255 5060
Unmonitor
> ed
> 126/126 10.201.0.29 D 255.255.255.255 5060
Unmonitor
> ed
> 125/125 10.201.0.35 D 255.255.255.255 5060
Unmonitor
> ed
> 124/124 10.201.0.31 D 255.255.255.255 5060
Unmonitor
> ed
> 102/102 10.200.0.48 D 255.255.255.255 5060
Unmonitor
> ed
>
> -Original Message-
> From: [EMAIL PROTECTED] on behalf of Gary
Richardson
> Sent: Thu 1/26/2006 5:18 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] extension to extension dialing
>
> Check your error messages in you asterisk console. Perhaps your sip
> secret or caller id is broken?
>
> What type of phones are you using?
>
> On 1/26/06, Nora Lavelle <[EMAIL PROTECTED]> wrote:
> >
> >
> >
> > Sorry for all the newbie questions. I really appreciate everyone's
help
> > today.
> >
> >
> >
> > Okay I've got outgoing and incoming calls working with no echo. yay!
Now I'm
> > having an issue with SIP extension to extension calling. Any time I
dial
> > another extension it goes right into voice mail. My extensions.conf
is
> > pretty small and rough but, here's what I have right now. Most of it
was
> > taken from the voip-info website. Any help as always VERY
appreciated.
> >
> >
> >
> > Thanks again!
> >
> > Nora Lavelle
> >
> >
> >
> > # cat 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)
> >
> > exte
RE: [Asterisk-Users] extension to extension dialing
Hi there gary. thanks so much for your help. we're using sipura-841 and snom
320s.
Here's the sip show peers.. that's weird that extension 130 has port 2057..
could that be the problem ?
-nora
Name/usernameHostDyn Nat ACL Mask Port Status
201/201 10.200.0.56 D 255.255.255.255 5060 Unmonitor
ed
130/130 10.200.0.10 D 255.255.255.255 2057 Unmonitor
ed
129/129 10.200.0.5 D 255.255.255.255 5060 Unmonitor
ed
127/127 10.201.0.30 D 255.255.255.255 5060 Unmonitor
ed
126/126 10.201.0.29 D 255.255.255.255 5060 Unmonitor
ed
125/125 10.201.0.35 D 255.255.255.255 5060 Unmonitor
ed
124/124 10.201.0.31 D 255.255.255.255 5060 Unmonitor
ed
102/102 10.200.0.48 D 255.255.255.255 5060 Unmonitor
ed
-Original Message-
From: [EMAIL PROTECTED] on behalf of Gary Richardson
Sent: Thu 1/26/2006 5:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] extension to extension dialing
Check your error messages in you asterisk console. Perhaps your sip
secret or caller id is broken?
What type of phones are you using?
On 1/26/06, Nora Lavelle <[EMAIL PROTECTED]> wrote:
>
>
>
> Sorry for all the newbie questions. I really appreciate everyone's help
> today.
>
>
>
> Okay I've got outgoing and incoming calls working with no echo. yay! Now I'm
> having an issue with SIP extension to extension calling. Any time I dial
> another extension it goes right into voice mail. My extensions.conf is
> pretty small and rough but, here's what I have right now. Most of it was
> taken from the voip-info website. Any help as always VERY appreciated.
>
>
>
> Thanks again!
>
> Nora Lavelle
>
>
>
> # cat 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)
>
>
>
> [voicemail]
>
> exten => 300,1,Answer
>
> exten => 300,2,VoicemailMain(ssn-voicemail-greeting)
>
> exten => 300,3,Hangup
>
>
>
> [local]
>
> exten => _9NXX,1,Dial(Zap/g1/${EXTEN:1})
>
> exten => _9NXX,2,Congestion
>
>
>
> [longdistance]
>
> exten => _91NXXNXX,1,Dial(Zap/g1/${EXTEN:1})
>
> exten => _91NXXNXX,2,Congestion
>
>
>
> [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-.,1,Goto(s-NOANSWER,1)
>
> exten => a,1,VoicemailMain(${MACRO_EXTEN})
>
>
>
> [default]
>
> include => incoming
>
> include => internal
>
> include => voicemail
>
> include => local
>
> include => longdistance
> ___
> --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
>
>
>
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<>___
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RE: [Asterisk-Users] extension to extension dialing
Here's what I get in the the log this is when extension 130 dials
extension 129. Thanks again !
nora
-- Executing Macro("SIP/130-a644", "stdexten|SIP/129") in new stack
-- Executing Dial("SIP/130-a644", "SIP/129|20") in new stack
-- Called 129
Jan 26 17:20:48 WARNING[28243]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Critical Request)
== No one is available to answer at this time
-- Executing Goto("SIP/130-a644", "s-NOANSWER|1") in new stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing VoiceMail("SIP/130-a644", "u129") in new stack
-- Playing 'voicemail/default/129/unavail' (language 'en')
== Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on
'SIP/130-a644' in macro 'stdexten'
== Spawn extension (default, 129, 1) exited non-zero on 'SIP/130-a644'
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary
Richardson
Sent: Thursday, January 26, 2006 5:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] extension to extension dialing
Check your error messages in you asterisk console. Perhaps your sip
secret or caller id is broken?
What type of phones are you using?
On 1/26/06, Nora Lavelle <[EMAIL PROTECTED]> wrote:
>
>
>
> Sorry for all the newbie questions. I really appreciate everyone's
help
> today.
>
>
>
> Okay I've got outgoing and incoming calls working with no echo. yay!
Now I'm
> having an issue with SIP extension to extension calling. Any time I
dial
> another extension it goes right into voice mail. My extensions.conf
is
> pretty small and rough but, here's what I have right now. Most of it
was
> taken from the voip-info website. Any help as always VERY appreciated.
>
>
>
> Thanks again!
>
> Nora Lavelle
>
>
>
> # cat 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)
>
>
>
> [voicemail]
>
> exten => 300,1,Answer
>
> exten => 300,2,VoicemailMain(ssn-voicemail-greeting)
>
> exten => 300,3,Hangup
>
>
>
> [local]
>
> exten => _9NXX,1,Dial(Zap/g1/${EXTEN:1})
>
> exten => _9NXX,2,Congestion
>
>
>
> [longdistance]
>
> exten => _91NXXNXX,1,Dial(Zap/g1/${EXTEN:1})
>
> exten => _91NXXNXX,2,Congestion
>
>
>
> [macro-stdexten]
>
> exten => s,1,Dial(${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${MACRO_EXTEN})
>
> exten => s-BUSY,2,Goto(default,s,1)
>
> exten => s-.,1,Goto(s-NOANSWER,1)
>
> exten => a,1,VoicemailMain(${MACRO_EXTEN})
>
>
>
> [default]
>
> include => incoming
>
> include => internal
>
> include => voicemail
>
> include => local
>
> include => longdistance
> ___
> --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
>
>
>
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[Asterisk-Users] extension to extension dialing
Sorry for all the newbie questions. I really appreciate
everyone’s help today.
Okay I’ve got outgoing and incoming calls working with
no echo. yay! Now I’m having an issue with SIP extension to extension
calling. Any time I dial another extension it goes right into voice mail. My
extensions.conf is pretty small and rough but, here’s what I have right
now. Most of it was taken from the voip-info website. Any help as always VERY
appreciated.
Thanks again!
Nora Lavelle
# cat 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)
[voicemail]
exten => 300,1,Answer
exten => 300,2,VoicemailMain(ssn-voicemail-greeting)
exten => 300,3,Hangup
[local]
exten => _9NXX,1,Dial(Zap/g1/${EXTEN:1})
exten => _9NXX,2,Congestion
[longdistance]
exten => _91NXXNXX,1,Dial(Zap/g1/${EXTEN:1})
exten => _91NXXNXX,2,Congestion
[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-.,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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RE: [Asterisk-Users] snom 320 echo problems
Thanks so much for your help. The version information is attached below. Looks like it is 4.4 Also can you tell me which exact Setting under Audio in Advanced Options I should change ? Here are the options I see. Audio: Mute Microphone: OFF Disable Casing Speaker: OFF DTMF echo on Speaker Phone: ON Keytones: OFF Call Released Notification: OFF Silence Supression: OFF Casing Mic Volume (1-8): 4 Handset Mic Volume (1-8): 4 Headset Mic Volume (1-8): 4 Version Information – Phone Type:snom320-SIP MAC-Address:0004132425F6 IP-Address:10.200.0.10 Version-Code:snom320-SIP 4.4 Bootloader: Firmware:http://snom.com/download/share/snom320-4.4-SIP-j.bin Production Information:Mac:0004132425F6;Version:Standard;Hardware:snom320 (MB V1.0_K7,KB V1.0_L4-NC);Lot: 11/05 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: Thursday, January 26, 2006 10:35 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] snom 320 echo problems Turn down mic gains in the web mgmt interface under Advanced, also make sure you have the latest firmware. -Original Message----- From: Nora Lavelle [mailto:[EMAIL PROTECTED] Sent: Thursday, January 26, 2006 11:07 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] snom 320 echo problems Hi there - I'm having some echo problems on my snom 320 phones. Anybody experience this before ? I don't have any issues with the sipura 841s I have though. Any help is greatly appreciated. Thanks ! Nora Lavelle ___ --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] snom 320 echo problems
Hi there – I’m having some echo problems on my snom 320 phones. Anybody experience this before ? I don’t have any issues with the sipura 841s I have though. Any help is greatly appreciated. Thanks ! Nora Lavelle ___ --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] Sipura 841 echo cancel question
Hi there, I’m new to asterisk and hoping you can help out. I have a small deployment of asterisk running. 4 sipura 841 phones and a linux box with a digium TDM400P. When a user makes a call there is usually echo for about 15 seconds and then it goes away. I have read through all the echo stuff and to be honest totally confused. Not sure what to set or how to test. Any guidance totally appreciated ! Thanks in advance ! Nora ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
