[Asterisk-Users] need help asterisk and AS5300
hi All Any body already setup asteriks call routing to Cisco AS5300 with SIP Server ? i need informations sample config for that, or can show how to route docs . thanks Dirgan Meet your soulmate! Yahoo! Asia presents Meetic - where millions of singles gather ___ --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] chan_capi - B3 Error
Let e guess, you have an AVM card?
It is a known issue. In some cases the driver does no accept the B connect
request, which would be okay, but any try later when it must be possible, it
is rejected too.
I have this on my todo list, but in the meantime it should work fine when
you switch to overlap dial using options 'bo' instead of only 'b' in the
CAPI dial string.
Armin
On Tue, 24 Jan 2006, Nathan Alberti wrote:
> I seem to be having a problem with B3 on my ISDN line, as you can see from the
> dial string I am having to have asterisk generate ringing else there is no
> progress indication.
>
>
> -- Executing Dial("SIP/0014A8ACCB83-fd9f", "CAPI/g1/142392203000/b|40|r")
> in new stack
> -- Called g1/142392203000/b
> -- CAPI/ISDN1/142392203000-0 is proceeding passing it to SIP/
> 0014A8ACCB83-fd9f
> Jan 24 07:38:56 WARNING[10609]: chan_capi.c:3385 show_capi_conf_error: ISDN1:
> conf_error 0x2001 PLCI=0x101 Command=CONNECT_B3_CONF,0x8487
> -- CAPI/ISDN1/142392203000-0 answered SIP/0014A8ACCB83-fd9f
>
> This issue was only introduced after and upgrade to chan_capi-cm-0.6.1 and
> continues on to chan_capi-cm-0.6.3, my capi.conf is as follows;
>
>
> [general]
> nationalprefix=0
> internationalprefix=0
> rxgain=0.8
> txgain=0.8
>
> [ISDN1]
> isdnmode=msn
> controller=1
> group=1
> softdtmf=0
> relaxdtmf=on
> context=pstn_in
> callgroup=1
> devices=2
>
>
> Regards,
>
> Nathan.
> ___
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Re: [Asterisk-Users] Hardware recommendations
Hmmmdo you mean that the system needs 4 lines? Or that you need a phone that can make 4 concurrent calls? PaulH - Original Message - From: "Dane Reugger" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, January 24, 2006 4:32 PM Subject: Re: [Asterisk-Users] Hardware recommendations > I need 2 concurrent connections but prefer 4 - we spend a lot of time on > the phone here. Once things recover in New Orleans we will probably > build our staff up to 7 or 8 quickly. > > -Dane > > [EMAIL PROTECTED] wrote: > > Needing a 4 line phone is going to decrease your choices of phones. > > > > Why do you need 4 lines? > > > > PaulH > > > > - Original Message - > > From: "Dane Reugger" <[EMAIL PROTECTED]> > > To: > > Sent: Tuesday, January 24, 2006 2:45 PM > > Subject: [Asterisk-Users] Hardware recommendations > > > > > > > >> We would like to test Asterisk in our small office - 5 users. We are a > >> small computer shop in New Orleans and would like to offer VoIP and > >> Asterisk to our clients but we are very new to VoIP and Asterisk. We > >> feel the best way to learn is to jump in. > >> > >> We've signed up w/ Teliax and setup a D-link phone that works OK - but > >> our goal is an Asterisk PBX. We would like to avoid as many costly > >> mistakes as possible. We plan on keeping 2 analog lines for emergencies, > >> VoIP down, 911, credit card machine, and Fax machine as we understand > >> Fax and CC machines are very unreliable w/ VoIP but plan on integrating > >> them in to the Asterisk with an FXO card > >> > >> We are looking for recommendations for VoIP phones and a 1 or 2 Line > >> FXO(?) card. I suspect the first is kinda vague and the latter is a > >> Digium card. Just looking for solutions, brands, and even vendors that > >> are known to work well. > >> > >> Phone needs 4 lines, Hold, VM, Caller ID > >> > >> Any advice appreciated > >> > >> Thanks, > >> > >> Dane > >> > >> > >> ___ > >> --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 > > > > ___ > --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] (no subject)
Hi, This is test 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
RE: [Asterisk-Users] Bug in attended transfer or as expected?
> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] > Sent: 24 January 2006 05:53 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Bug in attended transfer or as expected? > > > > > > The problem is when reception is busy she doesn't always wait for > > > someone to answer the call, however hanging up a ringing transfer on > > > attended also hangs up the caller. > > Its the phone that is responsible for hanging up both calls, not Asterisk. > > On the SNOM phones you can disable "disconnect on on-hook" to stop the > phone from doing that. > > Steve > Sorry think you misunderstand, I don't really want the phones to have to do the attended transfer by merging two lines locally to the phone. Asterisk 1.2.x now supports attended transfer natively (well kind of supports it :P) Thanks though Alex Information contained in this e-mail and any attachments are intended for the use of the addressee only, and may contain confidential information of Ubiquity Software Corporation. All unauthorized use, disclosure or distribution is strictly prohibited. If you are not the addressee, please notify the sender immediately and destroy all copies of this email. Unless otherwise expressly agreed in writing signed by an officer of Ubiquity Software Corporation, nothing in this communication shall be deemed to be legally binding. Thank 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] Bug in attended transfer or as expected?
> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Moises Silva > Sent: 23 January 2006 15:35 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Bug in attended transfer or as expected? > > > The problem is when reception is busy she doesn't always wait for > > someone to answer the call, however hanging up a ringing transfer on > > attended also hangs up the caller. > > If you have enabled "Disconnect Call" feature, then you can hangup > with "*0" for example, that will hangup only the current call, not the > call on hold. > Just so I understand is this the expected call flow? - Receptionist picks up a call and wants to transfer - Dials "*1" (attended transfer key) - Transfer extension starts ringing - New call comes in so receptionist decides to answer that one - Receptionist dials "*0" to hang up the current call (expecting the person on hold to be connected to the ringing extension) - Original caller either gets answered or continues with the dial plan for that extension (in my case that is forward back to the reception queue after 30seconds) If the receptionist decides to stop waiting for a ringing transfer, to get the caller back she can dial "*2" which I think is good. I still think that this should be much simpler and that hanging up an attended transfer "mid transfer" should change it to being a blind transfer. Cutting off the caller instead is pretty terrible. Thanks for the help Alex Information contained in this e-mail and any attachments are intended for the use of the addressee only, and may contain confidential information of Ubiquity Software Corporation. All unauthorized use, disclosure or distribution is strictly prohibited. If you are not the addressee, please notify the sender immediately and destroy all copies of this email. Unless otherwise expressly agreed in writing signed by an officer of Ubiquity Software Corporation, nothing in this communication shall be deemed to be legally binding. Thank 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] weird zttest result
Is this result indicates no problem at all? 8192 samples in 27554 sample intervals -136.352539% Regards, Stevanus C F wrote: These are actulay not strange, but good results. On 1/23/06, stevanus <[EMAIL PROTECTED]> wrote: Hi, I have these strange results : 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 27554 sample intervals -136.352539% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% Anyone has any idea why this happens? Regards, Stevanus ___ --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 ___ --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] MOH Server
Has anyone managed to set up a moh server for Asterisk? Reason would be to offload processing off the asterisk box, onto another system. The wiki is a bit light on details. If anyone managed to get it up and working, what software did you use on the server side, and what client app did you use? Mpg123? Mpg321? Madplayer? Something else? Also, putting legal ramifications aside, it'd be nifty to do something similar and stream the audio from an online radio station... just for kicks. Thanks. ___ --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 recommendations
Do the linksys phones support BLF? A lot of businesses require/expect BLF. Do the linksys phones support Asterisk setting the ring-type to auto answer so that you can do paging and intercom? Businesses expect this too. -Original Message- From: Cory Andrews [mailto:[EMAIL PROTECTED] Sent: Mon 1/23/2006 9:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Hardware recommendations Dane - I might suggest the following. (5) Linksys SPA-841, SPA-941 or SPA-942 (All work very well with Asterisk, and support 4 line appearances) Not sure what your broadband is in the office, with 5 users I am guessing you are utilizing DSL or Cable broadband. You might want to consider purchasing a firewall with QOS capabilities, like a the Sonicwall TZ170, which is relatively inexpensive. This will also give you remote VPN capabilities and if you want to set up remote extensions off your Asterisk PBX this comes in handy. Here is a good article on the TZ170 firewall. http://www.voiploop.com/blogs/product-review-sonicwall-firewall-tz170-2.htm For dual FXO you'll want a Digium TDM02B or you could purchase an external, 2 port FXO gateway. Also, once you have your Asterisk server up and running, determine the power load of the server, your LAN switch, and any related equipment, and invest in a decent UPS like a Tripplite or APC unit. Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 ++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] AIM - B2CORY - Original Message - From: "Dane Reugger" <[EMAIL PROTECTED]> To: Sent: Monday, January 23, 2006 10:45 PM Subject: [Asterisk-Users] Hardware recommendations > We would like to test Asterisk in our small office - 5 users. We are a > small computer shop in New Orleans and would like to offer VoIP and > Asterisk to our clients but we are very new to VoIP and Asterisk. We > feel the best way to learn is to jump in. > > We've signed up w/ Teliax and setup a D-link phone that works OK - but > our goal is an Asterisk PBX. We would like to avoid as many costly > mistakes as possible. We plan on keeping 2 analog lines for emergencies, > VoIP down, 911, credit card machine, and Fax machine as we understand > Fax and CC machines are very unreliable w/ VoIP but plan on integrating > them in to the Asterisk with an FXO card > > We are looking for recommendations for VoIP phones and a 1 or 2 Line > FXO(?) card. I suspect the first is kinda vague and the latter is a > Digium card. Just looking for solutions, brands, and even vendors that > are known to work well. > > Phone needs 4 lines, Hold, VM, Caller ID > > Any advice appreciated > > Thanks, > > Dane > > > ___ > --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 ___ --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] canreinvite always =no * no matter what we try :-(
On Mon, 23 Jan 2006, Steve Gladden wrote: > been testing with a rather simple setup. > > The mission is to actually get a reinvite to work on the lan. > > I am trying with two sipura phones G.711 codec forced on both > both on the lan no nat no fancy options suchs as tT or H > > No matter what we do asterisk hangs on to the media path, how > in the world do I get a reinvite to work where the media path > is actually handled by the two phones on the lan? > > Any pointers greatly appreciated! Remove from your Dial command all options that require Asterisk to hear the media stream. (T, t etc) 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
Re: [Asterisk-Users] canreinvite always =no * no matter what we try :-(
> How are you testing if asterisk is in the media path? Two ways: One phone on a hub with ethereal on a laptop and watching the rtp packets, pretty obvious that asterisk is staying in the media path. and that the rtp i not coming from the other phone. Way two, in the middle of an active/established call unplugging the ethernet cable from the asterisk box audio instantly dies on both phones when this occurs. plug asterisk box back into it's ethernet termination audio comes right back. Seems odd that these reinvites are supposed to magically occur (from what I gather) and it only happens when the sun is shining and everything is just right... I'd like a way to force it or KNOW that it should be occuring versus just expecting it to 'possibly' occur automatically if all conditions are met and automatically detected. Or maybe I have this all worng :-) Thanks! Steve > please turn on all the debug, warning, error etc messages in the > console, see logger.conf, then type sip peer debug and sip > peer debug to see the SIP messages. > > How are you testing if asterisk is in the media path? > > Regards > > On 1/23/06, Steve Gladden <[EMAIL PROTECTED]> wrote: >> been testing with a rather simple setup. >> >> The mission is to actually get a reinvite to work on the lan. >> >> I am trying with two sipura phones G.711 codec forced on both >> both on the lan no nat no fancy options suchs as tT or H >> >> No matter what we do asterisk hangs on to the media path, how >> in the world do I get a reinvite to work where the media path >> is actually handled by the two phones on the lan? >> >> Any pointers greatly appreciated! >> >> Steve >> >> >> Pretty simple extensions, on lan no nat >> >> >> [4785] >> >> type=friend >> username=4785 >> secret=test >> host=dynamic >> canreinvite=yes >> >> [4786] >> >> type=friend >> username=4786 >> secret=tesst >> host=dynamic >> canreinvite=yes >> >> >> exten => 4785,1,Dial(SIP/4785,66) >> exten => 4785,3,hangup >> >> exten => 4786,1,Dial(SIP/4786,66) >> exten => 4786,3,hangup >> >> ___ >> --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 >> > > > -- > "Su nombre es GNU/Linux, no solamente Linux, mas info en > http://www.gnu.org"; > ___ > --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] Hardware recommendations
Polycom SoundPoint 601 has 4 'lines'. :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Mon 1/23/2006 9:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Hardware recommendations Needing a 4 line phone is going to decrease your choices of phones. Why do you need 4 lines? PaulH - Original Message - From: "Dane Reugger" <[EMAIL PROTECTED]> To: Sent: Tuesday, January 24, 2006 2:45 PM Subject: [Asterisk-Users] Hardware recommendations > We would like to test Asterisk in our small office - 5 users. We are a > small computer shop in New Orleans and would like to offer VoIP and > Asterisk to our clients but we are very new to VoIP and Asterisk. We > feel the best way to learn is to jump in. > > We've signed up w/ Teliax and setup a D-link phone that works OK - but > our goal is an Asterisk PBX. We would like to avoid as many costly > mistakes as possible. We plan on keeping 2 analog lines for emergencies, > VoIP down, 911, credit card machine, and Fax machine as we understand > Fax and CC machines are very unreliable w/ VoIP but plan on integrating > them in to the Asterisk with an FXO card > > We are looking for recommendations for VoIP phones and a 1 or 2 Line > FXO(?) card. I suspect the first is kinda vague and the latter is a > Digium card. Just looking for solutions, brands, and even vendors that > are known to work well. > > Phone needs 4 lines, Hold, VM, Caller ID > > Any advice appreciated > > Thanks, > > Dane > > > ___ > --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 <>___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Polycom FW
They had those in the focus groups too. I think it was Snom, Sipura, Polycom and Cisco. One of the reasons the Polycom's won, was that comlex star codes where not required to perform most of the functions of the phone. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Mon 1/23/2006 10:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [Asterisk-Users] Re: Polycom FW On Mon, 23 Jan 2006, Douglas Garstang wrote: > We conducted focus groups, looking at several different vendors, before > we decided to go with the Polycom. From the user interface perspective, > the Polycom's won hands down. I was never involved with it, but > apparently to configure the Cisco's you need to be converting hex??? > Yuk! SNOM? That would definitely be my favourite. 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
Re: [Asterisk-Users] Bug in attended transfer or as expected?
> > The problem is when reception is busy she doesn't always wait for > > someone to answer the call, however hanging up a ringing transfer on > > attended also hangs up the caller. Its the phone that is responsible for hanging up both calls, not Asterisk. On the SNOM phones you can disable "disconnect on on-hook" to stop the phone from doing that. 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
RE: [Asterisk-Users] Re: Polycom FW
On Mon, 23 Jan 2006, Douglas Garstang wrote: > We conducted focus groups, looking at several different vendors, before > we decided to go with the Polycom. From the user interface perspective, > the Polycom's won hands down. I was never involved with it, but > apparently to configure the Cisco's you need to be converting hex??? > Yuk! SNOM? That would definitely be my favourite. 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
Re: [Asterisk-Users] Hardware recommendations
I'm considering the Linksys - just don't really trust Cisco (parent company) good stuff but they don't like the little guy - I cant buy these phones from distribution - I'm told the are focused on VoIP providers. That said they were already at the top of the list. Using 6Mb/764Kb Speakeasy DSL behind an IPCOP firewall - it has QoS but I understand it not the best for VoIP - I like the SonicWalls very much - problem is besides the 8 or 10 computers we are using we may have as many as 15 customer computers hooked to the network - creates havoc on their licensing. Any advantages between TDM02B or 2 port FXO gateway? Thanks Again -Dane Cory Andrews wrote: > Dane - I might suggest the following. > > (5) Linksys SPA-841, SPA-941 or SPA-942 (All work very well with > Asterisk, and support 4 line appearances) > > Not sure what your broadband is in the office, with 5 users I am > guessing you are utilizing DSL or Cable broadband. You might want to > consider purchasing a firewall with QOS capabilities, like a the > Sonicwall TZ170, which is relatively inexpensive. This will also give > you remote VPN capabilities and if you want to set up remote > extensions off your Asterisk PBX this comes in handy. Here is a good > article on the TZ170 firewall. > > http://www.voiploop.com/blogs/product-review-sonicwall-firewall-tz170-2.htm > > > For dual FXO you'll want a Digium TDM02B or you could purchase an > external, 2 port FXO gateway. > > Also, once you have your Asterisk server up and running, determine the > power load of the server, your LAN switch, and any related equipment, > and invest in a decent UPS like a Tripplite or APC unit. > > Cory J Andrews > > VOIPSupply.com > 454 Sonwil Drive > Buffalo, NY 14225 > ++ > voice - 716.630.1555 X22 > email - [EMAIL PROTECTED] > AIM - B2CORY > - Original Message - From: "Dane Reugger" <[EMAIL PROTECTED]> > To: > Sent: Monday, January 23, 2006 10:45 PM > Subject: [Asterisk-Users] Hardware recommendations > > >> We would like to test Asterisk in our small office - 5 users. We are a >> small computer shop in New Orleans and would like to offer VoIP and >> Asterisk to our clients but we are very new to VoIP and Asterisk. We >> feel the best way to learn is to jump in. >> >> We've signed up w/ Teliax and setup a D-link phone that works OK - but >> our goal is an Asterisk PBX. We would like to avoid as many costly >> mistakes as possible. We plan on keeping 2 analog lines for emergencies, >> VoIP down, 911, credit card machine, and Fax machine as we understand >> Fax and CC machines are very unreliable w/ VoIP but plan on integrating >> them in to the Asterisk with an FXO card >> >> We are looking for recommendations for VoIP phones and a 1 or 2 Line >> FXO(?) card. I suspect the first is kinda vague and the latter is a >> Digium card. Just looking for solutions, brands, and even vendors that >> are known to work well. >> >> Phone needs 4 lines, Hold, VM, Caller ID >> >> Any advice appreciated >> >> Thanks, >> >> Dane >> >> >> ___ >> --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 ___ --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] RTCP XR support (RFC 3611)
Hi list, Can you recommend any VoIP device or phone which supports RTCP XR (RFC 3611) ? Does asterisk intend to support it in the future? Thanks, Arsen. __ 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] Hardware recommendations
I need 2 concurrent connections but prefer 4 - we spend a lot of time on the phone here. Once things recover in New Orleans we will probably build our staff up to 7 or 8 quickly. -Dane [EMAIL PROTECTED] wrote: > Needing a 4 line phone is going to decrease your choices of phones. > > Why do you need 4 lines? > > PaulH > > - Original Message - > From: "Dane Reugger" <[EMAIL PROTECTED]> > To: > Sent: Tuesday, January 24, 2006 2:45 PM > Subject: [Asterisk-Users] Hardware recommendations > > > >> We would like to test Asterisk in our small office - 5 users. We are a >> small computer shop in New Orleans and would like to offer VoIP and >> Asterisk to our clients but we are very new to VoIP and Asterisk. We >> feel the best way to learn is to jump in. >> >> We've signed up w/ Teliax and setup a D-link phone that works OK - but >> our goal is an Asterisk PBX. We would like to avoid as many costly >> mistakes as possible. We plan on keeping 2 analog lines for emergencies, >> VoIP down, 911, credit card machine, and Fax machine as we understand >> Fax and CC machines are very unreliable w/ VoIP but plan on integrating >> them in to the Asterisk with an FXO card >> >> We are looking for recommendations for VoIP phones and a 1 or 2 Line >> FXO(?) card. I suspect the first is kinda vague and the latter is a >> Digium card. Just looking for solutions, brands, and even vendors that >> are known to work well. >> >> Phone needs 4 lines, Hold, VM, Caller ID >> >> Any advice appreciated >> >> Thanks, >> >> Dane >> >> >> ___ >> --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 > ___ --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 recommendations
Sounds like good advice - I will. But would prefer to settle on Debian - I have a how two somewhere around here, Thanks, Dean Collins wrote: > Dane, install an [EMAIL PROTECTED] cd and look at how it is configured as a > first step. > > Cheers, > > Dean > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Dane > Reugger > Sent: Monday, 23 January 2006 10:45 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Hardware recommendations > > We would like to test Asterisk in our small office - 5 users. We are a > small computer shop in New Orleans and would like to offer VoIP and > Asterisk to our clients but we are very new to VoIP and Asterisk. We > feel the best way to learn is to jump in. > > We've signed up w/ Teliax and setup a D-link phone that works OK - but > our goal is an Asterisk PBX. We would like to avoid as many costly > mistakes as possible. We plan on keeping 2 analog lines for emergencies, > VoIP down, 911, credit card machine, and Fax machine as we understand > Fax and CC machines are very unreliable w/ VoIP but plan on integrating > them in to the Asterisk with an FXO card > > We are looking for recommendations for VoIP phones and a 1 or 2 Line > FXO(?) card. I suspect the first is kinda vague and the latter is a > Digium card. Just looking for solutions, brands, and even vendors that > are known to work well. > > Phone needs 4 lines, Hold, VM, Caller ID > > Any advice appreciated > > Thanks, > > Dane > > > ___ > --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 > ___ --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] Jumping on the asterisk bandwagon
After two weeks of reading about asterisk and joining this mailing list, I finally decided jumping on the asterisk bandwagon... Asterisk rocks!!! I have a www.Stanaphone.com SIP (free) for incoming line and a www.VOIPJET.com IAX line for outbound. I also have a www.Vonage.com line (gives me 500 outbound minutes) and a Cingular cell phone (gives me 800 minutes) and I also use Skype fairly regularly. Not sure if there is a skype to asterisk gateway in software (probably won't happen because of intellectual property reasons) so I needed three FXO ports and one FXS port. Below is how I plan to connect all these. FXO #1 for Vonage ATA RT31P2 www.vonage.com FXO #2 for Cellphone connected via Doc-N-Talk from www.phonelabs.com FXO #3 for Skype to RJ11 adapter (www.echostore.com) and skype http://www.skype.com/products/skype/linux/ and skypemate for linux http://www.yealink.com/en/download/install-SkypeMate.zip and the only FXS for the existing RJ11 house wiring that has been safely disconnected from PSTN several years ago since Vonage started business. All other extensions that I plan will be IP Phones (hardphone/softphone) that will hook into my home gigabit ethernet. So I ordered two x100p from ebay (gets me one FXO each), bought a SIPURA SPA-3000 (gets me one FXO and one FXS), reformatted my 7 year old pc and installed [EMAIL PROTECTED] for use in my home. I tried my first setup of [EMAIL PROTECTED] ISO v2.2 and I am in love with this already. I am looking for ideas and examples, specifically those that support the following scenarios: 1. Optimize outbound minutes usage: First use up all 500 Vonage minutes and then switchover to cellphone to use up 800 outbound any time minutes and then switch over to VOIPJET if needed. I have a small script that uses CURL and grabs remaining vonage minutes. I want to be able to extract that information from vonage page and use that intelligently to switch over to using my cellphone (via doc-n-talk) for outbound calls. Similarly, I want to be able to extract remaining minutes information from www.cingular.com/ocs and switch over to VOIPJET...and all of this needs to happen without my intervention. For example: I have succeed in getting to the vonage billing page via the following script but I still need a way to parse the resulting page via a script to extract the remaining vonage minutes. debugfile="/root/vonage_$username" curl -d "username=$username" -d "password=$password" \ -c "/tmp/von_cookie_$username" \ https://secure.vonage.com/vonage-web/public/login.htm 2>&1 > $debugfile curl -b "/tmp/von_cookie_$username" \ "https://secure.vonage.com/vonage-web/billing/index.htm"; 2>&1 >> $debugfile 2. Presense Detection: I have used instructions from http://www.mundy.org/blog/ and enabled a trivial script that detecs my (cell phone's) presense via bluetooth. I want to be able to use this presense information to route my calls intelligently based on where I am. 3. Least Cost Routing: VOIPJET offers good international rates. I also make international calls using www.relianceindiacall.com I need a least cost routing mechanism to manage the lowest cost option by default. I am pretty sure there must be some one of this list that must have already tried these scenarios and may already have a working configuration that they may be willing to share their example configuration files for their setup with asterisk newbies like me. Thanks for your help. - Nilesh ___ --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] weird zttest result
These are actulay not strange, but good results. On 1/23/06, stevanus <[EMAIL PROTECTED]> wrote: > Hi, > > I have these strange results : > > 8192 samples in 8192 sample intervals 100.00% > 8192 samples in 8191 sample intervals 99.987793% > 8192 samples in 8192 sample intervals 100.00% > 8192 samples in 8192 sample intervals 100.00% > 8192 samples in 8191 sample intervals 99.987793% > 8192 samples in 8192 sample intervals 100.00% > 8192 samples in 8192 sample intervals 100.00% > 8192 samples in 8191 sample intervals 99.987793% > 8192 samples in 8192 sample intervals 100.00% > 8192 samples in 8191 sample intervals 99.987793% > 8192 samples in 27554 sample intervals -136.352539% > 8192 samples in 8191 sample intervals 99.987793% > 8192 samples in 8192 sample intervals 100.00% > 8192 samples in 8192 sample intervals 100.00% > 8192 samples in 8191 sample intervals 99.987793% > 8192 samples in 8192 sample intervals 100.00% > 8192 samples in 8192 sample intervals 100.00% > 8192 samples in 8192 sample intervals 100.00% > 8192 samples in 8192 sample intervals 100.00% > 8192 samples in 8192 sample intervals 100.00% > > Anyone has any idea why this happens? > > Regards, > > Stevanus > ___ > --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] Polycom 501 horrible echo
> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Jeff Herring > I have the following situation: > > Asterisk 1.2.1 > 25+ Polycom 501 telephones. Bootrom 2.6.2.0032 Application 1.6.2.0041 > Some 501's local to my network, some across the great INTERNET divide. > PRI connected to Sangoma card. I've got the exact same setup, boot ROM and application versions too, except I'm running Asterisk 1.2.2 and am experiencing the same issue. At first I thought I'd broken the mike by seeing if the mike hole up front was for the power adapter (didn't even think to check out the included network cable). > Issue: horrible echo (and squeals, and "underwater-like" sound) on speaker > phone when calling from extension to extension. I can only attest to the echo, squeal and variance in volume going from the handset to an FXO connection, but it's there. Also, using the built in hardware check show a surprising amount of background noise from my PC, more so than my 7960. > All gains, etc. are as listed in the Polycom Admin Guide. I'm using pretty much the defaults too as I'm still correcting and moving entries to the sip.cfg and phone1.cfg files. The only thing I can think of is to play with the AGI settings. > Anyone with thoughts of where to start? Same here please! Regards, --- Gavin ___ --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] Fw: setting outgoing caller ID by the queue anextension is logged into
Use different prefixes for different outgoing calls? (I know that's a nuisance though) PaulH - Original Message - From: Franklin Webb To: asterisk-users@lists.digium.com Sent: Tuesday, January 24, 2006 7:34 AM Subject: [Asterisk-Users] Fw: setting outgoing caller ID by the queue anextension is logged into Greetings fellow list members, I am trying to add some tricky functionality to Asterisk dialplan and I was curious if anyone else has come up with a solution to something like this. Basically I have phone representatives that log into one of several queues (not using chan Agent, we log in by the extension), and frequently these agents have to make attended transfer calls to outside numbers. This transfer basically amounts to a new outgoing call. I have been asked to set the caller ID for these outgoing calls based on the queue the phone representative is currently logged in to. Unfortunetly I cannot think of a way to do this. The incomming and outgoing calls are two different calls. I have considered using DBPut and DBGet to store this information in a database. This might work, but I am also concerned about the overhead involved. I cannot think of a way to do this using global variables since I need to store a seperate value for each extension. Has anyone run into an issue like this and come up with a solution? Any thoughts are much appreciated. Thank you, Franklin Webb Assistant IT Project Leader Inter Media Marketing Solutions ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Hardware recommendations
Dane - I might suggest the following. (5) Linksys SPA-841, SPA-941 or SPA-942 (All work very well with Asterisk, and support 4 line appearances) Not sure what your broadband is in the office, with 5 users I am guessing you are utilizing DSL or Cable broadband. You might want to consider purchasing a firewall with QOS capabilities, like a the Sonicwall TZ170, which is relatively inexpensive. This will also give you remote VPN capabilities and if you want to set up remote extensions off your Asterisk PBX this comes in handy. Here is a good article on the TZ170 firewall. http://www.voiploop.com/blogs/product-review-sonicwall-firewall-tz170-2.htm For dual FXO you'll want a Digium TDM02B or you could purchase an external, 2 port FXO gateway. Also, once you have your Asterisk server up and running, determine the power load of the server, your LAN switch, and any related equipment, and invest in a decent UPS like a Tripplite or APC unit. Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 ++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] AIM - B2CORY - Original Message - From: "Dane Reugger" <[EMAIL PROTECTED]> To: Sent: Monday, January 23, 2006 10:45 PM Subject: [Asterisk-Users] Hardware recommendations We would like to test Asterisk in our small office - 5 users. We are a small computer shop in New Orleans and would like to offer VoIP and Asterisk to our clients but we are very new to VoIP and Asterisk. We feel the best way to learn is to jump in. We've signed up w/ Teliax and setup a D-link phone that works OK - but our goal is an Asterisk PBX. We would like to avoid as many costly mistakes as possible. We plan on keeping 2 analog lines for emergencies, VoIP down, 911, credit card machine, and Fax machine as we understand Fax and CC machines are very unreliable w/ VoIP but plan on integrating them in to the Asterisk with an FXO card We are looking for recommendations for VoIP phones and a 1 or 2 Line FXO(?) card. I suspect the first is kinda vague and the latter is a Digium card. Just looking for solutions, brands, and even vendors that are known to work well. Phone needs 4 lines, Hold, VM, Caller ID Any advice appreciated Thanks, Dane ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Zaptel issues
Yeah, recompiling the kernel is a bit over my head, but I don't want to install an older gcc, so I'll just have to await some hand-holding from the people that put my kernel together (OpenVZ). Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: <[EMAIL PROTECTED]> To: Sent: Monday, January 23, 2006 8:02 PM Subject: Asterisk-Users Digest, Vol 18, Issue 143 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than "Re: Contents of Asterisk-Users digest..." -- Message: 15 Date: Mon, 23 Jan 2006 22:35:28 -0300 From: Facundo Ameal <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Zaptel issues To: Asterisk Users Mailing List - Non-Commercial Discussion Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1 I think you have comiled your kernel with a version of gcc and zaptel with another one, Compile zaptel drivers with gcc-3.3 and you will solve it, otherwise, you cas recompile your kernel with the new version of gcc. i also had that problem. 2006/1/23, Mike Hammett <[EMAIL PROTECTED]>: [EMAIL PROTECTED] ~]# which modprobe /sbin/modprobe [EMAIL PROTECTED] ~]# modprobe --version module-init-tools version 3.1-pre5 [EMAIL PROTECTED] ~]# dmesg | tail zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' ztdummy: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' ztdummy: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' ztdummy: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' It looks like my gcc versions are different from the one that made the kernel and the one that made the zaptel stuff. So then of the zt lines, do I only need: install ztdummy /sbin/modprobe --ignore-install ztdummy && /sbin/ztcfg Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: <[EMAIL PROTECTED]> To: Sent: Friday, January 13, 2006 4:49 AM Subject: Asterisk-Users Digest, Vol 18, Issue 82 > -- > > Message: 12 > Date: Fri, 13 Jan 2006 11:52:20 +0200 > From: Tzafrir Cohen <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] Zaptel issues > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=us-ascii > > On Thu, Jan 12, 2006 at 09:39:18AM -0600, Mike Hammett wrote: >> On a side note: When poking around, I noticed in the zaptel Makefile >> that there is a section talking about ztdummy automatically being >> included on 2.6 kernels. Is this correct? >> >> On to the main topic: Any ideas for troubleshooting this? >> >> [EMAIL PROTECTED] zaptel-1.2.1]# /etc/rc.d/init.d/zaptel start >> Loading zaptel framework: FATAL: Error inserting zaptel >> (/lib/modules/2.6.8-022stab050.1/extra/zaptel.ko): Invalid module >> format >>[FAILED] >> Waiting for zap to come online...Error: missing /dev/zap! >> >> >> [EMAIL PROTECTED] libpri-1.2.1]# modprobe ztdummy >> WARNING: Error inserting zaptel >> (/lib/modules/2.6.8-022stab050.1/extra/zaptel.ko): Invalid module >> format >> WARNING: Error inserting zaptel >> (/lib/modules/2.6.8-022stab050.1/extra/zaptel.ko): Invalid module >> format >> FATAL: Error inserting ztdummy >> (/lib/modules/2.6.8-022stab050.1/extra/ztdummy.ko): Invalid module >> format >> FATAL: Error running install command for ztdummy > > Could you please provide the output of following: > > which modprobe > modprobe --version > > To make things simpler, do away with the stuff that the zaptel install > puts in /etc/modprobe.d/zaptel (or /etc/modprobe.conf ). > > (ztdummy needs no ztcfg run after it) > > Also, plea
Re: [Asterisk-Users] Hardware recommendations
Needing a 4 line phone is going to decrease your choices of phones. Why do you need 4 lines? PaulH - Original Message - From: "Dane Reugger" <[EMAIL PROTECTED]> To: Sent: Tuesday, January 24, 2006 2:45 PM Subject: [Asterisk-Users] Hardware recommendations > We would like to test Asterisk in our small office - 5 users. We are a > small computer shop in New Orleans and would like to offer VoIP and > Asterisk to our clients but we are very new to VoIP and Asterisk. We > feel the best way to learn is to jump in. > > We've signed up w/ Teliax and setup a D-link phone that works OK - but > our goal is an Asterisk PBX. We would like to avoid as many costly > mistakes as possible. We plan on keeping 2 analog lines for emergencies, > VoIP down, 911, credit card machine, and Fax machine as we understand > Fax and CC machines are very unreliable w/ VoIP but plan on integrating > them in to the Asterisk with an FXO card > > We are looking for recommendations for VoIP phones and a 1 or 2 Line > FXO(?) card. I suspect the first is kinda vague and the latter is a > Digium card. Just looking for solutions, brands, and even vendors that > are known to work well. > > Phone needs 4 lines, Hold, VM, Caller ID > > Any advice appreciated > > Thanks, > > Dane > > > ___ > --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] Hardware recommendations
Dane, install an [EMAIL PROTECTED] cd and look at how it is configured as a first step. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dane Reugger Sent: Monday, 23 January 2006 10:45 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Hardware recommendations We would like to test Asterisk in our small office - 5 users. We are a small computer shop in New Orleans and would like to offer VoIP and Asterisk to our clients but we are very new to VoIP and Asterisk. We feel the best way to learn is to jump in. We've signed up w/ Teliax and setup a D-link phone that works OK - but our goal is an Asterisk PBX. We would like to avoid as many costly mistakes as possible. We plan on keeping 2 analog lines for emergencies, VoIP down, 911, credit card machine, and Fax machine as we understand Fax and CC machines are very unreliable w/ VoIP but plan on integrating them in to the Asterisk with an FXO card We are looking for recommendations for VoIP phones and a 1 or 2 Line FXO(?) card. I suspect the first is kinda vague and the latter is a Digium card. Just looking for solutions, brands, and even vendors that are known to work well. Phone needs 4 lines, Hold, VM, Caller ID Any advice appreciated Thanks, Dane ___ --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] weird zttest result
Hi, I have these strange results : 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 27554 sample intervals -136.352539% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% Anyone has any idea why this happens? Regards, Stevanus ___ --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 recommendations
We would like to test Asterisk in our small office - 5 users. We are a small computer shop in New Orleans and would like to offer VoIP and Asterisk to our clients but we are very new to VoIP and Asterisk. We feel the best way to learn is to jump in. We've signed up w/ Teliax and setup a D-link phone that works OK - but our goal is an Asterisk PBX. We would like to avoid as many costly mistakes as possible. We plan on keeping 2 analog lines for emergencies, VoIP down, 911, credit card machine, and Fax machine as we understand Fax and CC machines are very unreliable w/ VoIP but plan on integrating them in to the Asterisk with an FXO card We are looking for recommendations for VoIP phones and a 1 or 2 Line FXO(?) card. I suspect the first is kinda vague and the latter is a Digium card. Just looking for solutions, brands, and even vendors that are known to work well. Phone needs 4 lines, Hold, VM, Caller ID Any advice appreciated Thanks, Dane ___ --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] Polycom 501 horrible echo
I have had the same issue. It has a lot to do with the acoustics, as well as gain. Before I messed with the config files it sounded great, then I fussed with them and upgraded to the latest sip, and now I also notice this on speaker. I would go totally default, local configure and see how they sound... Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Monday, January 23, 2006 9:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Polycom 501 horrible echo You aren't making calls from one phone to another, with them right next to each other on the same desk are you? Doug. -Original Message- From: Jeff Herring [mailto:[EMAIL PROTECTED] Sent: Mon 1/23/2006 6:46 PM To: asterisk-users@lists.digium.com Cc: Subject: [Asterisk-Users] Polycom 501 horrible echo I have the following situation: Asterisk 1.2.1 25+ Polycom 501 telephones. Bootrom 2.6.2.0032 Application 1.6.2.0041 Some 501's local to my network, some across the great INTERNET divide. PRI connected to Sangoma card. Issue: horrible echo (and squeals, and "underwater-like" sound) on speaker phone when calling from extension to extension. echo not present when calling outbound using PRI or when receiving calls from PRI. echo not present when using handset or headset in any case. All gains, etc. are as listed in the Polycom Admin Guide. Not specific to any phone, or its location on our network. I suspect the issue is related to the echo cancelation HW in the speaker phone, but I'm not sure...The unfortunate thing is these phones were purchased because of their excellent speaker phones which now appear to be worse than the Grandstreams! Anyone with thoughts of where to start? TIA - Jeff H. ___ --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] Config File Storage
Haven't used it, but I imagine it doesn't have anywhere near the flexibility I'm looking for. -Original Message- From: Jeff Herring [mailto:[EMAIL PROTECTED] Sent: Mon 1/23/2006 6:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Config File Storage At 08:42 PM 1/23/2006, Douglas Garstang wrote: >Content-Class: urn:content-classes:message >Content-Type: text/plain; > charset="UTF-8" > >I'm trying to think of a way to store/represent the Asterisk .conf files. >One method is to store them in MySQL in some format, and then write some >scripts to query MySQL and generate the conf files before doing a reload. Asterisk at Home? >MySQL is pretty heavy handed though. I'm looking for something a bit more >lightweight, maybe some sort of XML based database for Linux, where >the config files could be stored in XML format? Doesn't seem like it would >be too hard to represent them this way. > >Trying to find a way to store them so they can be accessed easier from a >web interface. > >Thanks, Doug > > >___ >--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 -- Jeff Herring / [EMAIL PROTECTED] Seacoast Laboratory Data Systems, Inc. Voice: 603 431 4114 x14 FAX:603 431 2112 -- CONFIDENTIALITY NOTICE: This e-mail transmission, and any documents, files or previous e-mail messages attached to it contain confidential and/or privileged information meant for the listed recipient(s) only. You many not distribute or share this correspondence without written authorization from the above author. If you are not the intended recipient, or a person responsible for delivering it to the intended recipient, you are hereby notified that any disclosure, copying, distribution or use of any of the information contained in or attached to this transmission is STRICTLY PROHIBITED. If you have received this transmission in error, please immediately notify me by reply e-mail and destroy the original transmission and its attachments without saving them in any manner. ___ --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] Home Test!
i am using a win-modem as a X100P clone. It has an especial motorola chiset which is detailed here: http://www.voip-info.org/wiki/view/X100P+clone it was really hard for me to get this modem. sorry, but I can't help you. If you come here you have to go to every store you see and ask, because it's very difficult to get them. i am part of a LUG (Linux User Group) and i am the only one who could manage to get this specific modem. Sorry. 2006/1/23, Maxi Belino <[EMAIL PROTECTED]>: > Hi, Facundo i'm from Uruguay, i'm plannig to visit Argentina and i would > like to know where i can get there the X100p Clone Card and some other VoIP > stuff. Is there a website you could recommend me? do you have a phone > number of this store ? name or address? Thanks gracias ! > saludos ! > Maxi > > 2006/1/24, Facundo Ameal <[EMAIL PROTECTED]>: > > > > I haven't said it but if someone believes there's a better choice > > than buying a sipura or a grandstream ht, please tell me, I considered > > thaat two because, here, they are popular. > > > > 2006/1/23, Facundo Ameal < [EMAIL PROTECTED]>: > > > Hi Michael, so which is your opinion about Sipura and what do you > > > think about Grandstream? I'm looking for opinions of whom has tested > > > the devices and has more experience, not to waste my money. Do you > > > deliver them to Argentina? > > > Erick: ya se que solamente se puede postear en ingles, por > > > eso segui con el dialogo en ingles > > > I'm new into this so I appreciate all the recomendations you are giving > me. > > > I'm between buying a Sipura 2002 (I didn't know Sipura 200 was > > > replaced) nad a GrandStream HT 486 (or any other model). I have > > > already obtained an FXO port by buying an X100P Clone (here they cost > > > USD10 aprox.), so I want only FXS ports. > > > > > > thanks. > > > > > > > > > 2006/1/23, The VoIP Connection > <[EMAIL PROTECTED] >: > > > > We have sold thousands of these with no reports of echo problems. > Perhaps > > > > the reviews were referring to a different Grandstream product? Some > of the > > > > phones have had some echo issues. BTW, the Sipura 2000 has been > replaced by > > > > the 2002. > > > > > > > > Michael Crown > > > > Managing Partner > > > > www.thevoipconnection.com > > > > 321.989.6728 ext. 611 > > > > sip:[EMAIL PROTECTED] > > > > > > > > > > > > > -Original Message- > > > > > From: Facundo Ameal [mailto: [EMAIL PROTECTED] > > > > > Sent: Monday, January 23, 2006 1:08 PM > > > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > > > Subject: [Asterisk-Users] Home Test! > > > > > > > > > > Hi everybody! > > > > > I'm from Argentina, so you'll have to sorry me for my English. > > > > > I have a Linux box with asterisk and want to buy an ATA. > > > > > Fist, I thought about the Grandstream HandyTone but I read > > > > > some reviews which says that it has a lot of echo. Some > > > > > people recommended me Sipura 2000 but I don't know what to > > > > > do. Now I just to make some tests at home and see what > > > > > happens and if it works ok, then I-m planning to install it > > > > > in other places. > > > > > > > > > > thank you in advance. > > > > > > > > > > regards, > > > > > -- > > > > > Facundo Ameal. > > > > > famealgmailcom > > > > > Linux User #395088 > > > > > > > > > > Open your mind, use open source. > > > > > > > > > > > > > > > > > > ___ > > > > --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 > > > > > > > > > > > > > -- > > > Facundo Ameal. > > > famealgmailcom > > > Linux User #395088 > > > > > > Open your mind, use open source. > > > > > > > > > -- > > Facundo Ameal. > > famealgmailcom > > Linux User #395088 > > > > Open your mind, use open source. > > ___ > > --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 > > > -- Facundo Ameal. famealgmailcom Linux User #395088 Open your mind, use open source. ___ --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] canreinvite always =no * no matter what we try :-(
Steve, > The mission is to actually get a reinvite to work on the lan. There isn't anything special to get this working... normally. I trust you verified the traffic flow with a network monitor tool (tcpdump?), correct? Does SIP debug give you any info (i.e., does it match the right peer) -- you don't show if you allow reinvites globally? What about the nat= setting? Couple pointers I can give you to get you excited: 1) Reinvites work quite reliably, I use them between the PTSN gateway and the end user's ATA, all the way across the Internet -- nicely reduces latency. 2) If you use RFC2833 for DTMF you can issue an reinvite and still use t/T for transfer. NOTE that you have to modify the source to make asterisk reinvite even when it needs to listen to DTMFs. I give no guarantees how well it will work for you but it does work. See "AST_BRIDGE_DTMF_CHANNEL_0 | AST_BRIDGE_DTMF_CHANNEL_1" in rtp.c. 3) Reinvites *can* work even if both ends are behind NAT. It really depends on the NATing router and the ATA. Sipura's and good NAT routers work, but I would not call it "reliable" -- it's really pushing it a bit... So if you really want to see why your Reinvites do not work, then you probably will have to make your hands dirty and analyze where ast_rtp_bridge() in rtp.c bails out. But since you are on a LAN it makes the situation a lot easier. --Luki ___ --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] canreinvite always =no * no matter what we try :-(
please turn on all the debug, warning, error etc messages in the console, see logger.conf, then type sip peer debug and sip peer debug to see the SIP messages. How are you testing if asterisk is in the media path? Regards On 1/23/06, Steve Gladden <[EMAIL PROTECTED]> wrote: > been testing with a rather simple setup. > > The mission is to actually get a reinvite to work on the lan. > > I am trying with two sipura phones G.711 codec forced on both > both on the lan no nat no fancy options suchs as tT or H > > No matter what we do asterisk hangs on to the media path, how > in the world do I get a reinvite to work where the media path > is actually handled by the two phones on the lan? > > Any pointers greatly appreciated! > > Steve > > > Pretty simple extensions, on lan no nat > > > [4785] > > type=friend > username=4785 > secret=test > host=dynamic > canreinvite=yes > > [4786] > > type=friend > username=4786 > secret=tesst > host=dynamic > canreinvite=yes > > > exten => 4785,1,Dial(SIP/4785,66) > exten => 4785,3,hangup > > exten => 4786,1,Dial(SIP/4786,66) > exten => 4786,3,hangup > > ___ > --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 > -- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org"; ___ --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] Port forwarding on a DLink Di-604
Thanks everyone! Sorted now. H On 1/19/06, Scott DesBles <[EMAIL PROTECTED]> wrote: > Select Advanced, then Firewall on the left. Create a rule that has the > range of ports you want. > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of hugolivude > Sent: Thursday, January 19, 2006 9:42 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Port forwarding on a DLink Di-604 > > Anyone know how to set up port forwarding of multiple ports on a DLink > DI-604? > > I successfully portforward the SIP port on the Advanced|Virtual Server > page. It works because I can register a SIP client, but it's a single > port - 5060. > > The DLink doesn't seem to provide an obvious way of portfarding the > 1 - 2 ports needed for RTP. > > Any ideas? > Hugh > ___ > --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 > ___ --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] Home Test!
Hi, Facundo i'm from Uruguay, i'm plannig to visit Argentina and i would like to know where i can get there the X100p Clone Card and some other VoIP stuff. Is there a website you could recommend me? do you have a phone number of this store ? name or address? Thanks gracias ! saludos ! Maxi2006/1/24, Facundo Ameal <[EMAIL PROTECTED]>: I haven't said it but if someone believes there's a better choicethan buying a sipura or a grandstream ht, please tell me, I consideredthaat two because, here, they are popular.2006/1/23, Facundo Ameal < [EMAIL PROTECTED]>:> Hi Michael, so which is your opinion about Sipura and what do you> think about Grandstream? I'm looking for opinions of whom has tested> the devices and has more experience, not to waste my money. Do you > deliver them to Argentina?> Erick: ya se que solamente se puede postear en ingles, por> eso segui con el dialogo en ingles > I'm new into this so I appreciate all the recomendations you are giving me. > I'm between buying a Sipura 2002 (I didn't know Sipura 200 was> replaced) nad a GrandStream HT 486 (or any other model). I have> already obtained an FXO port by buying an X100P Clone (here they cost > USD10 aprox.), so I want only FXS ports.>> thanks.>>> 2006/1/23, The VoIP Connection <[EMAIL PROTECTED] >:> > We have sold thousands of these with no reports of echo problems. Perhaps> > the reviews were referring to a different Grandstream product? Some of the> > phones have had some echo issues. BTW, the Sipura 2000 has been replaced by > > the 2002.> >> > Michael Crown> > Managing Partner> > www.thevoipconnection.com> > 321.989.6728 ext. 611> > sip:[EMAIL PROTECTED]> >> >> > > -Original Message-> > > From: Facundo Ameal [mailto: [EMAIL PROTECTED]]> > > Sent: Monday, January 23, 2006 1:08 PM> > > To: Asterisk Users Mailing List - Non-Commercial Discussion> > > Subject: [Asterisk-Users] Home Test! > > >> > > Hi everybody!> > > I'm from Argentina, so you'll have to sorry me for my English.> > > I have a Linux box with asterisk and want to buy an ATA.> > > Fist, I thought about the Grandstream HandyTone but I read > > > some reviews which says that it has a lot of echo. Some> > > people recommended me Sipura 2000 but I don't know what to> > > do. Now I just to make some tests at home and see what > > > happens and if it works ok, then I-m planning to install it> > > in other places.> > >> > > thank you in advance.> > >> > > regards,> > > -- > > > Facundo Ameal.> > > famealgmailcom> > > Linux User #395088> > >> > > Open your mind, use open source.> > >> > > > >> > ___> > --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> >>> > --> Facundo Ameal.> famealgmailcom> Linux User #395088>> Open your mind, use open source.>--Facundo Ameal.famealgmailcom Linux User #395088Open your mind, use open source.___--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] Polycom 501 horrible echo
You aren't making calls from one phone to another, with them right next to each other on the same desk are you? Doug. -Original Message- From: Jeff Herring [mailto:[EMAIL PROTECTED] Sent: Mon 1/23/2006 6:46 PM To: asterisk-users@lists.digium.com Cc: Subject: [Asterisk-Users] Polycom 501 horrible echo I have the following situation: Asterisk 1.2.1 25+ Polycom 501 telephones. Bootrom 2.6.2.0032 Application 1.6.2.0041 Some 501's local to my network, some across the great INTERNET divide. PRI connected to Sangoma card. Issue: horrible echo (and squeals, and "underwater-like" sound) on speaker phone when calling from extension to extension. echo not present when calling outbound using PRI or when receiving calls from PRI. echo not present when using handset or headset in any case. All gains, etc. are as listed in the Polycom Admin Guide. Not specific to any phone, or its location on our network. I suspect the issue is related to the echo cancelation HW in the speaker phone, but I'm not sure...The unfortunate thing is these phones were purchased because of their excellent speaker phones which now appear to be worse than the Grandstreams! Anyone with thoughts of where to start? TIA - Jeff H. ___ --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] Config File Storage
At 08:42 PM 1/23/2006, Douglas Garstang wrote: Content-Class: urn:content-classes:message Content-Type: text/plain; charset="UTF-8" I'm trying to think of a way to store/represent the Asterisk .conf files. One method is to store them in MySQL in some format, and then write some scripts to query MySQL and generate the conf files before doing a reload. Asterisk at Home? MySQL is pretty heavy handed though. I'm looking for something a bit more lightweight, maybe some sort of XML based database for Linux, where the config files could be stored in XML format? Doesn't seem like it would be too hard to represent them this way. Trying to find a way to store them so they can be accessed easier from a web interface. Thanks, Doug ___ --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 -- Jeff Herring / [EMAIL PROTECTED] Seacoast Laboratory Data Systems, Inc. Voice: 603 431 4114 x14 FAX:603 431 2112 -- CONFIDENTIALITY NOTICE: This e-mail transmission, and any documents, files or previous e-mail messages attached to it contain confidential and/or privileged information meant for the listed recipient(s) only. You many not distribute or share this correspondence without written authorization from the above author. If you are not the intended recipient, or a person responsible for delivering it to the intended recipient, you are hereby notified that any disclosure, copying, distribution or use of any of the information contained in or attached to this transmission is STRICTLY PROHIBITED. If you have received this transmission in error, please immediately notify me by reply e-mail and destroy the original transmission and its attachments without saving them in any manner. ___ --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] Dial() Jumping behaviour and Vesrsion 1.2
Thanks so much for your comments and for directing me back to the
sample extensions.conf. With all the examples floating around, I
sometimes forget to just go back to the soiurce!
H
On 1/22/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> see inline
> > The version 1.2 Dial() command does not use the n+101 jumping
> > behaviour by default. I know about the j option and setting
> > "priorityjumping=yes" as described here:
> >
> > http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial
> >
> > But if I use the default behaviour does that mean I have to check the
> > DIALSTATUS to determine whether or not to go to voicemail?
> No but is good idea
> > For example I used to do this:
> >
> > exten => s,1,Dial(SIP/[EMAIL PROTECTED],20,t)
> > exten => s,2,Voicemail(u${EXTEN})
> > exten => s,3,Goto(s,200)
> > ;
> > exten => s,102,Voicemail(b${EXTEN})
> > exten => s,103,Goto(s,200)
> > ;
> > exten => s,200,Playback(CallAgainRealSoon)
> > exten => s,201,Hangup
> > ;
> > exten => h,1,Hangup
> >
> > So in 1.2 would I do the following or am I missing something?
>
> No but is good idea
> or you can use "n" instead of "s"
> or
> exten => 1234,1,Dial(SIP/[EMAIL PROTECTED],20,t)
> exten => 1234,n,Voicemail(u${EXTEN)
> exten => 1234,dial+101,Voicemail(b${EXTEN})
> or even better use a macro (since your way is very close to that)
> (see below)
>
> > exten => s,1,Dial(SIP/[EMAIL PROTECTED],20,t)
> > exten => s,2,GotoIf($["${DIALSTATUS }" = "BUSY"]?10)
> > exten => s,3,GotoIf($["${DIALSTATUS }" = "NOANSWER"]?20)
> > ;
> > exten => s,10,Voicemail(b${EXTEN})
> > exten => s,11,Goto(s,100)
> > ;
> > exten => s,20,Voicemail(u${EXTEN})
> > exten => s,21,Goto(s,100)
> > ;
> > exten => s,100,Playback(CallAgainRealSoon)
> > exten => s,101,Hangup
> > ;
> > exten => h,1,Hangup
> [your-internal-context]
> exten =>1234,1,Macro(stdexten,${EXTEN},SIP/${EXTEN},20,t)
>
> [macro-stdexten]; straight from the extensionsconf sample
> ;; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
> ;; ${ARG2} - Device(s) to ring
>
> ;exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds
> maximum
> ;exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on
> status(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
> ;
> ;exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send to
> voicemail
> w/ unavail announce
> ;exten => s-NOANSWER,2,Goto(s,1); If they press #, return to
> start
> ;exten => s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to voicemail w/ busy
> announce
> ;exten => s-BUSY,2,Goto(s,1); If they press #, return to start
> ;
> ;exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no
> answer
> ;exten => a,1,VoicemailMain(${ARG1})
> benchev
> ___
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Re: [Asterisk-Users] Home Test!
I haven't said it but if someone believes there's a better choice than buying a sipura or a grandstream ht, please tell me, I considered thaat two because, here, they are popular. 2006/1/23, Facundo Ameal <[EMAIL PROTECTED]>: > Hi Michael, so which is your opinion about Sipura and what do you > think about Grandstream? I'm looking for opinions of whom has tested > the devices and has more experience, not to waste my money. Do you > deliver them to Argentina? > Erick: ya se que solamente se puede postear en ingles, por > eso segui con el dialogo en ingles > I'm new into this so I appreciate all the recomendations you are giving me. > I'm between buying a Sipura 2002 (I didn't know Sipura 200 was > replaced) nad a GrandStream HT 486 (or any other model). I have > already obtained an FXO port by buying an X100P Clone (here they cost > USD10 aprox.), so I want only FXS ports. > > thanks. > > > 2006/1/23, The VoIP Connection <[EMAIL PROTECTED]>: > > We have sold thousands of these with no reports of echo problems. Perhaps > > the reviews were referring to a different Grandstream product? Some of the > > phones have had some echo issues. BTW, the Sipura 2000 has been replaced by > > the 2002. > > > > Michael Crown > > Managing Partner > > www.thevoipconnection.com > > 321.989.6728 ext. 611 > > sip:[EMAIL PROTECTED] > > > > > > > -Original Message- > > > From: Facundo Ameal [mailto:[EMAIL PROTECTED] > > > Sent: Monday, January 23, 2006 1:08 PM > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > Subject: [Asterisk-Users] Home Test! > > > > > > Hi everybody! > > > I'm from Argentina, so you'll have to sorry me for my English. > > > I have a Linux box with asterisk and want to buy an ATA. > > > Fist, I thought about the Grandstream HandyTone but I read > > > some reviews which says that it has a lot of echo. Some > > > people recommended me Sipura 2000 but I don't know what to > > > do. Now I just to make some tests at home and see what > > > happens and if it works ok, then I-m planning to install it > > > in other places. > > > > > > thank you in advance. > > > > > > regards, > > > -- > > > Facundo Ameal. > > > famealgmailcom > > > Linux User #395088 > > > > > > Open your mind, use open source. > > > > > > > > > > ___ > > --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 > > > > > -- > Facundo Ameal. > famealgmailcom > Linux User #395088 > > Open your mind, use open source. > -- Facundo Ameal. famealgmailcom Linux User #395088 Open your mind, use open source. ___ --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] Polycom 501 horrible echo
I have the following situation: Asterisk 1.2.1 25+ Polycom 501 telephones. Bootrom 2.6.2.0032 Application 1.6.2.0041 Some 501's local to my network, some across the great INTERNET divide. PRI connected to Sangoma card. Issue: horrible echo (and squeals, and "underwater-like" sound) on speaker phone when calling from extension to extension. echo not present when calling outbound using PRI or when receiving calls from PRI. echo not present when using handset or headset in any case. All gains, etc. are as listed in the Polycom Admin Guide. Not specific to any phone, or its location on our network. I suspect the issue is related to the echo cancelation HW in the speaker phone, but I'm not sure...The unfortunate thing is these phones were purchased because of their excellent speaker phones which now appear to be worse than the Grandstreams! Anyone with thoughts of where to start? TIA - Jeff H. ___ --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 SIP phones to Cisco Unity via CCM 4.0 SIP Trunk
You can run a SIP image on a 7940. [EMAIL PROTECTED] has pretty good support for it. Check the voip-info.org wiki for instructions on switching the firmware. Hopefully that will take a step out of the plan -- you could completely ditch your Cisco system :) On 1/23/06, sys read <[EMAIL PROTECTED]> wrote: > > Hi, > > I've got a CCM ( Cisco Call Manager ), with a Cisco Unity VM server and > about 45 SCCP phones on the ccm, and 200 users on unity. we want to > migrate all users to IP Phones to ditch our ancient phone system. I would > love to get Linksys-Sipura SPA-941s for the 150 users not on IP phones yet > and run sip to an asterisk server, but have their voicemail on Unity. > > these phones are $150 each, the alternative is cisco 7940s ( around $250 ) > running SCCP through the CCM. at the quantities I'm talking about, $100 is > significant. > > Does anyone have any idea how to get this done? > > I've tried this: > > exten => 123,1,Dial(SIP/sipphone,20) > exten => 123,2,Dial(SIP/ccm/3040) > > where 3040 is our VM pilot for ccm. but all it does is take us to the main > greeting. > > we have smartnet, but they haven't been helpful at all > > I called digium to see if they could help if we paid, but they said they've > never heard of cisco unity > > help? > > thanks. > > ___ > --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] Config File Storage
I'm trying to think of a way to store/represent the Asterisk .conf files. One method is to store them in MySQL in some format, and then write some scripts to query MySQL and generate the conf files before doing a reload. MySQL is pretty heavy handed though. I'm looking for something a bit more lightweight, maybe some sort of XML based database for Linux, where the config files could be stored in XML format? Doesn't seem like it would be too hard to represent them this way. Trying to find a way to store them so they can be accessed easier from a web interface. Thanks, Doug ___ --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 & SER for dummies ?
Are, Are you using a common database for SER and Asterisk? How are you keeping the accounts synced? Does this setup cause any complications with AstBill? regards, David On 10/25/05, Are <[EMAIL PROTECTED]> wrote: > Good Question. > > We have tested it with any combination we can think about and it is working > safely. There is no way (we know about) that you can pass toll free calls. > :-) > > Basically SER is configured to only accept clients that have the same > callerid as account numbers so SER refuse to pass the call if you try to be > smart. Asterisk only passes the call if you have a valid account and the > request is handed over from the SER server. Asterisk determine the max > length of the call based on the Users Account balance in AstBill. > > Are Casilla -- > http://astartelecom.com - Independent VOIP Telecoms Broker. Asterisk and > Drupal Consultants > http://astbill.com - Billing, Routing and Management software for Asterisk > and VOIP > AstBill DEMO: http://demo.astbill.com > > > > On 10/25/05, trixter aka Bret McDanel <[EMAIL PROTECTED]> wrote: > > > > On Tue, 2005-10-25 at 08:27 +0100, Are wrote: > > > The authentication in Asterisk is done using ANI/CLI. > > > > > Same way as broadvoice, wonder if using that setup if I set my caller id > > to someone else will it cause the INVITE that broadvoice does > > (broadvoice will invite the person registered as that account if you try > > to make a call on their CID, asterisk ignores that invite, I am not so > > sure if all devices will) > > > > -- > > Trixter http://www.0xdecafbad.com Bret McDanel > > UK +44 870 340 4605 Germany +49 801 777 555 3402 > > US +1 360 207 0479 or +1 516 687 5200 > > FreeWorldDialup: 635378 > > > > > > -BEGIN PGP SIGNATURE- > > Version: GnuPG v1.4.1 (GNU/Linux) > > > > > iD8DBQBDXeNg+1olxlzQw5cRApWJAJ4sXCutFLLuAk26jzumrS/ioMiZ3ACfa8zZ > > IBWJRwuEQ1RN9EqRvajQG/c= > > =DzJ5 > > -END PGP SIGNATURE- > > > > > > ___ > > --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 > > > > > > > ___ > --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 > > ___ --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] SPA-3000 - the party's over :-(
I'll be reading this in Om Malik's blog tomorrow morning. Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 ++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] AIM - B2CORY - Original Message - From: "C F" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, January 23, 2006 8:31 PM Subject: Re: [Asterisk-Users] SPA-3000 - the party's over :-( Cory, please hold off, it's still not Aprils first :) On 1/23/06, Cory Andrews <[EMAIL PROTECTED]> wrote: Anyone have a conspiracy theory or two to roll into this thread? Cisco is actually coming out with a new line of gateways that only support IAX, and will be porting their entire Callmanager platform to IAX. The best thing about these gateways is that they will actually be running an embedded version of Microsoft Windows 98 SE (Second Edition). Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 ++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] AIM - B2CORY - Original Message - From: "Rich Adamson" <[EMAIL PROTECTED]> To: "Asterisk-users-list" Sent: Monday, January 23, 2006 6:50 PM Subject: RE: [Asterisk-Users] SPA-3000 - the party's over :-( > >> > > I can't speculate as to why their sales of Linksys/Sipura products >> > > have >> > > been restricted, but as a Linksys VAD I can say we are not under >> > > any >> > > such restriction at present. >> > >> > its pretty obvious, linksys/sipura are shifting to selling primarily >> > to >> > service providers who would provide service-locked ATAs to end users. >> > >> > sipura telegraphed their intent a long while back by withholding >> > auto-provisioning documentation from anyone except service providers, >> > and >> > now they have completed the move by no longer allowing sales to end >> > users >> > at all. > > That seems to be right in line with Chamber's objective to be a major > player in the home market. He's certainly not going approach that > objective > by selling one/two devices at a time, so it makes sense he'd change the > sales/marketing approach to focus/lock-in higher volume > customers/resellers > regardless of what the rest of us think. > > That certainly isn't the last shoe to drop in the voip market; wait till > the next level(s) of announcements from Cisco. > > If I were going to bet a couple bucks on this, I'd suggest the spa3000 > will > disappear alltogether, and a replacement in the form of a linksys box > with > a faster processor is not far behind. > > > ___ > --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 ___ --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: RE : [Asterisk-Users] make linux26
i compiled it with make linux26 and had no trouble. try it like that. 2006/1/23, Mike Hammett <[EMAIL PROTECTED]>: > Yeah, that's where I saw contradicting what I saw elsewhere. > > > > Mike Hammett > Intelligent Computing Solutions > http://www.ics-il.com > > > - Original Message - > From: <[EMAIL PROTECTED]> > To: > Sent: Monday, January 23, 2006 3:48 PM > Subject: Asterisk-Users Digest, Vol 18, Issue 141 > > > > Send Asterisk-Users mailing list submissions to > > asterisk-users@lists.digium.com > > > > To subscribe or unsubscribe via the World Wide Web, visit > > http://lists.digium.com/mailman/listinfo/asterisk-users > > or, via email, send a message with subject or body 'help' to > > [EMAIL PROTECTED] > > > > You can reach the person managing the list at > > [EMAIL PROTECTED] > > > > When replying, please edit your Subject line so it is more specific > > than "Re: Contents of Asterisk-Users digest..." > > > > > > -- > > > > Message: 23 > > Date: Mon, 23 Jan 2006 22:37:55 +0100 > > From: <[EMAIL PROTECTED]> > > Subject: RE : [Asterisk-Users] make linux26 > > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > > > > Message-ID: > > > > > > Content-Type: text/plain; charset="iso-8859-1" > > > > Hi Mike, > > > > You must continue - for zaptel only - to "make linux26", as it is > > described > > in the companion file "README.Linux26" in the Zaptel folder > > (/usr/src/zaptel). > > Read the text from this file, as suggested in its title : > > > > To build for Linux 2.6, first you must be sure that you have a > > symlink to your linux-2.6 sources in /usr/src/linux-2.6. The 2.6 > > kernel no longer needs the full sourcecode to build against it. You > > can create the symlink to /lib/modules/`uname -r`/build/ and then > > you can type: > > > > # make linux26 > > # make install > > > > Note that you will also need CRC-CCITT functions compiled > > with your kernel or as a kernel module. These can be > > selected from the "Library Routines" submenu during kernel > > configuration via "make menuconfig" > > > > It is a good habit to read all this "README..." files before to do > > something, as it is important to read any user manual for any sofisticated > > equipment ;-) > > > > Good luck ! > > > > Best Regards, > > Francois BERGERET, > > France. > > > > > > -Message d'origine- > > De : [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] De la part de Mike > > Hammett > > Envoyé : lundi 23 janvier 2006 22:10 > > À : asterisk-users@lists.digium.com > > Objet : [Asterisk-Users] make linux26 > > > > > > I thought I read somewhere that you no longer have to do a special make > > command for the 2.6 kernel. Is this true, or should I still make linux26? > > I'm having problems getting anything zaptel to load properly. > > > > > > > > Mike Hammett > > Intelligent Computing Solutions > > http://www.ics-il.com > > > > > > > > -- next part -- > > An HTML attachment was scrubbed... > > URL: > > http://lists.digium.com/pipermail/asterisk-users/attachments/20060123/9b097a35/attachment-0001.htm > > > > ___ > --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 > -- Facundo Ameal. famealgmailcom Linux User #395088 Open your mind, use open source. ___ --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] Zaptel issues
I think you have comiled your kernel with a version of gcc and zaptel with another one, Compile zaptel drivers with gcc-3.3 and you will solve it, otherwise, you cas recompile your kernel with the new version of gcc. i also had that problem. 2006/1/23, Mike Hammett <[EMAIL PROTECTED]>: > [EMAIL PROTECTED] ~]# which modprobe > /sbin/modprobe > [EMAIL PROTECTED] ~]# modprobe --version > module-init-tools version 3.1-pre5 > [EMAIL PROTECTED] ~]# dmesg | tail > zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be > '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' > zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be > '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' > zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be > '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' > ztdummy: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be > '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' > zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be > '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' > zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be > '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' > ztdummy: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be > '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' > zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be > '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' > zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be > '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' > ztdummy: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be > '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' > > > It looks like my gcc versions are different from the one that made the > kernel and the one that made the zaptel stuff. > > So then of the zt lines, do I only need: > > install ztdummy /sbin/modprobe --ignore-install ztdummy && /sbin/ztcfg > > > > > Mike Hammett > Intelligent Computing Solutions > http://www.ics-il.com > > > - Original Message - > From: <[EMAIL PROTECTED]> > To: > Sent: Friday, January 13, 2006 4:49 AM > Subject: Asterisk-Users Digest, Vol 18, Issue 82 > > > > -- > > > > Message: 12 > > Date: Fri, 13 Jan 2006 11:52:20 +0200 > > From: Tzafrir Cohen <[EMAIL PROTECTED]> > > Subject: Re: [Asterisk-Users] Zaptel issues > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > > Message-ID: <[EMAIL PROTECTED]> > > Content-Type: text/plain; charset=us-ascii > > > > On Thu, Jan 12, 2006 at 09:39:18AM -0600, Mike Hammett wrote: > >> On a side note: When poking around, I noticed in the zaptel Makefile > >> that there is a section talking about ztdummy automatically being > >> included on 2.6 kernels. Is this correct? > >> > >> On to the main topic: Any ideas for troubleshooting this? > >> > >> [EMAIL PROTECTED] zaptel-1.2.1]# /etc/rc.d/init.d/zaptel start > >> Loading zaptel framework: FATAL: Error inserting zaptel > >> (/lib/modules/2.6.8-022stab050.1/extra/zaptel.ko): Invalid module format > >>[FAILED] > >> Waiting for zap to come online...Error: missing /dev/zap! > >> > >> > >> [EMAIL PROTECTED] libpri-1.2.1]# modprobe ztdummy > >> WARNING: Error inserting zaptel > >> (/lib/modules/2.6.8-022stab050.1/extra/zaptel.ko): Invalid module format > >> WARNING: Error inserting zaptel > >> (/lib/modules/2.6.8-022stab050.1/extra/zaptel.ko): Invalid module format > >> FATAL: Error inserting ztdummy > >> (/lib/modules/2.6.8-022stab050.1/extra/ztdummy.ko): Invalid module format > >> FATAL: Error running install command for ztdummy > > > > Could you please provide the output of following: > > > > which modprobe > > modprobe --version > > > > To make things simpler, do away with the stuff that the zaptel install > > puts in /etc/modprobe.d/zaptel (or /etc/modprobe.conf ). > > > > (ztdummy needs no ztcfg run after it) > > > > Also, please provide the latest relevant kernel log messages: > > > > dmesg | tail > > > > -- > > Tzafrir Cohen | [EMAIL PROTECTED] | VIM is > > http://tzafrir.org.il | | a Mutt's > > [EMAIL PROTECTED] | | best > > ICQ# 16849755 | | friend > > > > > > ___ > --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 > -- Facundo Ameal. famealgmailcom Linux User #395088 Open your mind, use open source. ___ --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] Video Conferencing.
I'm looking for point to point Video Conferencing , just because, like I said in other post, I'm doing some tests at homeand I want to try *almost* every feature asterisk has. THank you, I 'll read about it. I also would like to develop for asterisk (it's not for the bounty) but I just don't know much about C or ANSI C. 2006/1/23, Dean Collins <[EMAIL PROTECTED]>: > It's possible to do point to point but not point to multipoint. > > I tried to get development for this some time ago and no one responded, > check out my Video Conference Bounty on www.voip-info.org, since then we > have developed our own solution that we have decided to market, it will > cost $2,000 for up to 10 users that uses the Macromedia communications > server. > > Regards, > > > Dean Collins > Cognation Pty Ltd > [EMAIL PROTECTED] > +1-212-203-4357 > +61-2-9016-5642 (Sydney in-dial). > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Facundo > Ameal > Sent: Monday, 23 January 2006 2:48 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Video Conferencing. > > I have a doubt... is it posible to do Video Conferencing using asterisk? > > -- > Facundo Ameal. > famealgmailcom > Linux User #395088 > > Open your mind, use open source. > ___ > --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 > -- Facundo Ameal. famealgmailcom Linux User #395088 Open your mind, use open source. ___ --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] SPA-3000 - the party's over :-(
Cory, please hold off, it's still not Aprils first :) On 1/23/06, Cory Andrews <[EMAIL PROTECTED]> wrote: > Anyone have a conspiracy theory or two to roll into this thread? Cisco is > actually coming out with a new line of gateways that only support IAX, and > will be porting their entire Callmanager platform to IAX. The best thing > about these gateways is that they will actually be running an embedded > version of Microsoft Windows 98 SE (Second Edition). > > Cory J Andrews > > VOIPSupply.com > 454 Sonwil Drive > Buffalo, NY 14225 > ++ > voice - 716.630.1555 X22 > email - [EMAIL PROTECTED] > AIM - B2CORY > - Original Message - > From: "Rich Adamson" <[EMAIL PROTECTED]> > To: "Asterisk-users-list" > Sent: Monday, January 23, 2006 6:50 PM > Subject: RE: [Asterisk-Users] SPA-3000 - the party's over :-( > > > > > >> > > I can't speculate as to why their sales of Linksys/Sipura products > >> > > have > >> > > been restricted, but as a Linksys VAD I can say we are not under any > >> > > such restriction at present. > >> > > >> > its pretty obvious, linksys/sipura are shifting to selling primarily to > >> > service providers who would provide service-locked ATAs to end users. > >> > > >> > sipura telegraphed their intent a long while back by withholding > >> > auto-provisioning documentation from anyone except service providers, > >> > and > >> > now they have completed the move by no longer allowing sales to end > >> > users > >> > at all. > > > > That seems to be right in line with Chamber's objective to be a major > > player in the home market. He's certainly not going approach that > > objective > > by selling one/two devices at a time, so it makes sense he'd change the > > sales/marketing approach to focus/lock-in higher volume > > customers/resellers > > regardless of what the rest of us think. > > > > That certainly isn't the last shoe to drop in the voip market; wait till > > the next level(s) of announcements from Cisco. > > > > If I were going to bet a couple bucks on this, I'd suggest the spa3000 > > will > > disappear alltogether, and a replacement in the form of a linksys box with > > a faster processor is not far behind. > > > > > > ___ > > --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 > ___ --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] Home Test!
Hi Michael, so which is your opinion about Sipura and what do you think about Grandstream? I'm looking for opinions of whom has tested the devices and has more experience, not to waste my money. Do you deliver them to Argentina? Erick: ya se que solamente se puede postear en ingles, por eso segui con el dialogo en ingles I'm new into this so I appreciate all the recomendations you are giving me. I'm between buying a Sipura 2002 (I didn't know Sipura 200 was replaced) nad a GrandStream HT 486 (or any other model). I have already obtained an FXO port by buying an X100P Clone (here they cost USD10 aprox.), so I want only FXS ports. thanks. 2006/1/23, The VoIP Connection <[EMAIL PROTECTED]>: > We have sold thousands of these with no reports of echo problems. Perhaps > the reviews were referring to a different Grandstream product? Some of the > phones have had some echo issues. BTW, the Sipura 2000 has been replaced by > the 2002. > > Michael Crown > Managing Partner > www.thevoipconnection.com > 321.989.6728 ext. 611 > sip:[EMAIL PROTECTED] > > > > -Original Message- > > From: Facundo Ameal [mailto:[EMAIL PROTECTED] > > Sent: Monday, January 23, 2006 1:08 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [Asterisk-Users] Home Test! > > > > Hi everybody! > > I'm from Argentina, so you'll have to sorry me for my English. > > I have a Linux box with asterisk and want to buy an ATA. > > Fist, I thought about the Grandstream HandyTone but I read > > some reviews which says that it has a lot of echo. Some > > people recommended me Sipura 2000 but I don't know what to > > do. Now I just to make some tests at home and see what > > happens and if it works ok, then I-m planning to install it > > in other places. > > > > thank you in advance. > > > > regards, > > -- > > Facundo Ameal. > > famealgmailcom > > Linux User #395088 > > > > Open your mind, use open source. > > > > > > ___ > --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 > -- Facundo Ameal. famealgmailcom Linux User #395088 Open your mind, use open source. ___ --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] G729a Pass-Through and Recording/Monitoring
Steve Totaro wrote: Hello, I am wondering about the ability of a server that is simply passing G729 through it to have the ability to record the calls. I know for voicemail, meetme, and things like that to work, a G729 license must be installed on the machine since there is transcoding going on. Is this also true for recording of calls? Will I require licensing for each recorded call? Will the server see a big performance hit in this setup whether or not a license is required? The way I understanding it - a license is required if the media has to be processed by the pbx core. Somebody please correct me if I'm wrong. To record the call, the pbx core will transcode the incoming stream to the its native format (SLINEAR?) and then write the stream out in your recording format (xlaw or GSM). In short, you'll need a G.729 license. ___ --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] Polycom phones and dynamic IP for NAT
Where do you have to set the public IP? We use dhcp Poly behind firewalls daily. Just set nat=yes in sip.conf On Jan 23, 2006, at 3:26 PM, Bill Gibbs wrote: I know the Polycoms work with NAT, but you have to specify the public IP. Is there anyway for it to discover the external IP automatically? I like the phones (been playing with a 301) but for some of our clients who have a dynamic IP (and no hope of getting a static ie cable or residential DSL) I’d be afraid to use them since you have to specify the IP. What about the Cisco phones? Is the IP hard set? Are there any good “dynamic IP” compatible SIP phones that aren’t crap? Bill ___ --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] SPA-3000 - the party's over :-(
I heard it was actually Longhorn Embedded VoIP version with free automatic updates if buy 500 CALS and software assurance. -D On 1/23/06, Cory Andrews <[EMAIL PROTECTED]> wrote: > Anyone have a conspiracy theory or two to roll into this thread? Cisco is > actually coming out with a new line of gateways that only support IAX, and > will be porting their entire Callmanager platform to IAX. The best thing > about these gateways is that they will actually be running an embedded > version of Microsoft Windows 98 SE (Second Edition). > > Cory J Andrews > > VOIPSupply.com > 454 Sonwil Drive > Buffalo, NY 14225 > ++ > voice - 716.630.1555 X22 > email - [EMAIL PROTECTED] > AIM - B2CORY > - Original Message - > From: "Rich Adamson" <[EMAIL PROTECTED]> > To: "Asterisk-users-list" > Sent: Monday, January 23, 2006 6:50 PM > Subject: RE: [Asterisk-Users] SPA-3000 - the party's over :-( > > > > > >> > > I can't speculate as to why their sales of Linksys/Sipura products > >> > > have > >> > > been restricted, but as a Linksys VAD I can say we are not under any > >> > > such restriction at present. > >> > > >> > its pretty obvious, linksys/sipura are shifting to selling primarily to > >> > service providers who would provide service-locked ATAs to end users. > >> > > >> > sipura telegraphed their intent a long while back by withholding > >> > auto-provisioning documentation from anyone except service providers, > >> > and > >> > now they have completed the move by no longer allowing sales to end > >> > users > >> > at all. > > > > That seems to be right in line with Chamber's objective to be a major > > player in the home market. He's certainly not going approach that > > objective > > by selling one/two devices at a time, so it makes sense he'd change the > > sales/marketing approach to focus/lock-in higher volume > > customers/resellers > > regardless of what the rest of us think. > > > > That certainly isn't the last shoe to drop in the voip market; wait till > > the next level(s) of announcements from Cisco. > > > > If I were going to bet a couple bucks on this, I'd suggest the spa3000 > > will > > disappear alltogether, and a replacement in the form of a linksys box with > > a faster processor is not far behind. > > > > > > ___ > > --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 > ___ --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 fax to pdf, blank pdfs?
Hello all, I'm working on asterisk fax>pdf/email & have a problem. I can see that the faxes are received (via the cli) and I get the fax pdf in my email, but they are always blank, any idea what is causing this? I'm using AAH 2.0 & have installed the fax/pdf (via install-pdf from the command line). Any ideas? Thanks Dan ___ --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] TE110P + PRI incoming + outgoing extensions question
You need to have an extension defined for each number comig in. They may be 4 digit if that is how your circuit is ordered. You then need to create a dialplan to tell the call what to do. Yes you could create a group in zapata to use for outdial The pri will automatically allow up to 23 calls for one number as long as channels are available On Jan 21, 2006, at 9:53 AM, Dan Sully wrote: * Doug Lytle shaped the electrons to say... exten => 1153,1,Answer I can get the incoming call. If I try and do: exten => s,1,Answer Why would an incoming call have a destination of 1153? My incoming don't have a destination until the end user selects something from and IVR or and operator sends them on to an extension. The destination is the last 4 digits of the number I dial. It sounds like something isn't configured quite correctly at XO then. I wasn't able to find much useful information on the Wiki. You really didn't look that hard then, took me all but 10 seconds doing a search on zapata.conf That really was addressing my first question, not the second - which was easy to find - I just needed some confirmation. Thanks -D -- Ya gotta love UNIX, where else do you wonder whether you can kill a zombie spawned by a daemon's fork? ___ --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] SPA-3000 - the party's over :-(
Anyone have a conspiracy theory or two to roll into this thread? Cisco is actually coming out with a new line of gateways that only support IAX, and will be porting their entire Callmanager platform to IAX. The best thing about these gateways is that they will actually be running an embedded version of Microsoft Windows 98 SE (Second Edition). Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 ++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] AIM - B2CORY - Original Message - From: "Rich Adamson" <[EMAIL PROTECTED]> To: "Asterisk-users-list" Sent: Monday, January 23, 2006 6:50 PM Subject: RE: [Asterisk-Users] SPA-3000 - the party's over :-( > > I can't speculate as to why their sales of Linksys/Sipura products > > have > > been restricted, but as a Linksys VAD I can say we are not under any > > such restriction at present. > > its pretty obvious, linksys/sipura are shifting to selling primarily to > service providers who would provide service-locked ATAs to end users. > > sipura telegraphed their intent a long while back by withholding > auto-provisioning documentation from anyone except service providers, > and > now they have completed the move by no longer allowing sales to end > users > at all. That seems to be right in line with Chamber's objective to be a major player in the home market. He's certainly not going approach that objective by selling one/two devices at a time, so it makes sense he'd change the sales/marketing approach to focus/lock-in higher volume customers/resellers regardless of what the rest of us think. That certainly isn't the last shoe to drop in the voip market; wait till the next level(s) of announcements from Cisco. If I were going to bet a couple bucks on this, I'd suggest the spa3000 will disappear alltogether, and a replacement in the form of a linksys box with a faster processor is not far behind. ___ --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] DTMF not working on overseas cellphone calls
I thought I sent this earlier this week, but I didn't see it. If I missed it, I apologize for the resend. We are running asterisk 1.2.2 with a TDM04B connected to PSTN lines. On incoming calls from cellphones located overseas, DTMF is not recognized - we have many single-digit choices in our menu so the problem isn't that some digits aren't working, it's not listening at all. Works fine from domestic landlines and cellphones and from overseas landlines. I know the cellphones don't have a problem with DTMF, they work with other IVRs. I've placed overseas calls (I'm currently in a different country from the asterisk machine) from both landlines and cellphones, and can't hear a difference in quality. Could playing with rxgain help? Is there any chance that I could cause the calls that don't have a problem to either be too loud or get distorted due to clipping? ___ --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] SPA-3000 - the party's over :-(
> If I were going to bet a couple bucks on this, I'd suggest the spa3000 > will > disappear alltogether, and a replacement in the form of a linksys box with > a faster processor is not far behind. One of the features I'd like to see in such a box is TDMoE support (eg it would be a 'mini' channel bank), and IAX support. Previously, with the 'unlocked' version available there wasn't too much incentive to try and 'unlock' them (although it has been done). Now I guess there will be more effort on such things, which means we can get them cheaper as they will have been subsidised by the Telco... James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Newer version of Zaptel with 1.0 branch of *
You can use the newer Zaptel, but your LibPRI must match your Asterisk. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- "Chris Earle (CBL)" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] > Is it possible to run the CVS-HEAD/Stable version of Zaptel (1.2 whatever) > with an older version of Asterisk? I'm running 1.09, but I was wondering if > I could get at the newer echo cancellers like KB1 and MG2 without upgrading > to Asterisk 1.2? > > > I'm going out on a limb here to try and fix a serious echo problem on a TDM > + BT PSTN line in the UK > > > Thanks for your suggestions everyone > > > -- > Chris Earle > System Solutions Specialist, > > > -- > This message has been scanned for viruses and dangerous content by > MailScanner, and is believed to be clean. > > ___ > --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] SPA-3000 - the party's over :-(
> > I can't speculate as to why their sales of Linksys/Sipura products have > > been restricted, but as a Linksys VAD I can say we are not under any > > such restriction at present. > > its pretty obvious, linksys/sipura are shifting to selling primarily to > service providers who would provide service-locked ATAs to end users. > > sipura telegraphed their intent a long while back by withholding > auto-provisioning documentation from anyone except service providers, and > now they have completed the move by no longer allowing sales to end users > at all. That seems to be right in line with Chamber's objective to be a major player in the home market. He's certainly not going approach that objective by selling one/two devices at a time, so it makes sense he'd change the sales/marketing approach to focus/lock-in higher volume customers/resellers regardless of what the rest of us think. That certainly isn't the last shoe to drop in the voip market; wait till the next level(s) of announcements from Cisco. If I were going to bet a couple bucks on this, I'd suggest the spa3000 will disappear alltogether, and a replacement in the form of a linksys box with a faster processor is not far behind. ___ --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] SPA-3000 - the party's over :-(
> > > I can't speculate as to why their sales of Linksys/Sipura products have > > > been restricted, but as a Linksys VAD I can say we are not under any > > > such restriction at present. > > > > its pretty obvious, linksys/sipura are shifting to selling primarily to > > service providers who would provide service-locked ATAs to end users. > > > > sipura telegraphed their intent a long while back by withholding > > auto-provisioning documentation from anyone except service providers, and > > now they have completed the move by no longer allowing sales to end users > > at all. That seems to be right in line with Chamber's objective to be a major player in the home market. He's certainly not going approach that objective by selling one/two devices at a time, so it makes sense he'd change the sales/marketing approach to focus/lock-in higher volume customers/resellers regardless of what the rest of us think. That certainly isn't the last shoe to drop in the voip market; wait till the next level(s) of announcements from Cisco. If I were going to bet a couple bucks on this, I'd suggest the spa3000 will disappear alltogether, and a replacement in the form of a linksys box with a faster processor is not far behind. ___ --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] chan_capi - B3 Error
I seem to be having a problem with B3 on my ISDN line, as you can see
from the dial string I am having to have asterisk generate ringing
else there is no progress indication.
-- Executing Dial("SIP/0014A8ACCB83-fd9f", "CAPI/g1/142392203000/
b|40|r") in new stack
-- Called g1/142392203000/b
-- CAPI/ISDN1/142392203000-0 is proceeding passing it to SIP/
0014A8ACCB83-fd9f
Jan 24 07:38:56 WARNING[10609]: chan_capi.c:3385
show_capi_conf_error: ISDN1: conf_error 0x2001 PLCI=0x101
Command=CONNECT_B3_CONF,0x8487
-- CAPI/ISDN1/142392203000-0 answered SIP/0014A8ACCB83-fd9f
This issue was only introduced after and upgrade to chan_capi-
cm-0.6.1 and continues on to chan_capi-cm-0.6.3, my capi.conf is as
follows;
[general]
nationalprefix=0
internationalprefix=0
rxgain=0.8
txgain=0.8
[ISDN1]
isdnmode=msn
controller=1
group=1
softdtmf=0
relaxdtmf=on
context=pstn_in
callgroup=1
devices=2
Regards,
Nathan.
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Re: [Asterisk-Users] OFF TOPIC: Core router upgrade for a voip colocation center
Josh,You are in a position that many of our customers have found themselves in. For ISPs/Colo operations starting out Cisco 720x, Foundry BigIron, RiverStone, and Extreme switches all present an aggressive price point for the performance. However once you pass this point Foundry, RiverStone and Extreme all start to become exponentially expensive due to the lack of parts on the resale market. I would advise you against the 7206/12008 upgrade for a couple of different reasons. 1) Like you said they are nearing their EOL date,2) Processor performance is limited based on today's standards (unless you want to shell out for the NPE-G1 or the PRP), 3) They have a limited number of fixed interface ports, and additional line cards are expensive. With the exception of sites running Sonet the 6500 platform is the only way to go. We have sites running the SUP 720 3BXL cards with over 20 full BGP sessions pushing Gigs worth of traffic through them. When you look at processor/memory utilization you wouldn't even know the switch was being used. For your configuration I would recommend a 6503/6504 with a Sup 32 (WS-SUP32-GE-3B)supervisor module... 1) The Sup 32 comes with enough processor/memory to handle BGP in real world situations (256MB standard, upgradable to 1GB),2) It has 8 SFP ports on the supervisor module which is enough for most mid tier applications, 3) 32GB shared bus,4) 15 million packets per second, and my favorite reason, 5) it runs IOS My company is based in Los Angeles, give me a call and I will be more than happy to go over all of this with you. Best, MaxMax ClarkCreative Thought, Inc.(866)231-7371 x 3874(213)784-3874 Direct(866)369-0953 24/7 SupportIT should facilitate business, we can help.On 1/23/06, josh harrington < [EMAIL PROTECTED]> wrote:> Hello, hope this isn't too far offtopic here but being a troller for a long> time here I've realized there is a great knowledge base so I wanted to at> least see if i could get some tips. I help run a small colocation company > in California and I am in the middle of recommending a new 'core router'> platform for our network. We offer mainly colo and dedicated servers, and> several of our clients use our space for VOIP services so quality even under > high peak usage is a must. We are not huge, but as we have had near 200%> growth in the past 12 months and need to expand our network asap to keep up.> Simply put, I'd love to hear feedback and/or suggestions from any of you > guys who have gone through this already.> > Our network map is real simple:> > [Carrier 7609] --> 100 mbit --> Our cisco 7206 --> 100 mbit --> racks> > [the racks on our end are a series of switches, mainly 2948gl3's] > > We push about 60 mbit to/from our (1) carrier at peak right now, and the> router keeps up fine [its a cisco 7206 npe 150 btw, very low end on the 7206> line], and at peak we have under 50,000 packets per second, and our 7206 > File router-choices.txt not changed so no update needed.> [EMAIL PROTECTED] Hardware]$ cat router-choices.txt> #1 -> http://list.linux-vserver.org/archive/vserver/ > #2 -> webhostingtalk.com: jharington68/adam123> http://www.webhostingtalk.com/forumdisplay.php?f=44 > #3 -> asterisk mail list> http://lists.digium.com/mailman/listinfo/asterisk-users> #4 -> cisco mail list?> > HOTMAIL [EMAIL PROTECTED] pw/adam123> > > Hello, hope this isn't too far offtopic here but being a troller for a long> time here I've realized there is a great knowledge base so I wanted to at > least see if i could get some tips. I help run a small colocation company> in California and I am in the middle of recommending a new 'core router'> platform for our network. We offer mainly colo and dedicated servers, and > several of our clients use our space for VOIP services so quality even under> high peak usage is a must. We are not huge, but as we have had near 200%> growth in the past 12 months and need to expand our network asap to keep up. > Simply put, I'd love to hear feedback and/or suggestions from any of you> guys who have gone through this already.> > Our network map is real simple:> > [Carrier 7609] --> 100 mbit --> Our cisco 7206 --> 100 mbit --> racks > > [the racks on our end are a series of switches, mainly 2948gl3's]> > We push about 60 mbit to/from our (1) carrier at peak right now, and the> router keeps up fine [its a cisco 7206 npe 150 btw, very low end on the 7206 > line], and at peak we have under 50,000 packets per second, and our 7206> has little/no features enabled [just static routes and passing all traffic> between 2 Ethernet 100 mbit interfaces].> > To date we have had 2 problems, both were DOS attacks launched FROM one of> our customer's servers flooding a full 100 mbit wire with more packets per> second than the router could handle (the 2948gl3's spiked to about 50% cpu > load during the attack but the 7200 literally just died for 3 minutes as the> interface(s) all rebooted]. So our main goal to grow is something that can> handle a lo
[Asterisk-Users] canreinvite always =no * no matter what we try :-(
been testing with a rather simple setup. The mission is to actually get a reinvite to work on the lan. I am trying with two sipura phones G.711 codec forced on both both on the lan no nat no fancy options suchs as tT or H No matter what we do asterisk hangs on to the media path, how in the world do I get a reinvite to work where the media path is actually handled by the two phones on the lan? Any pointers greatly appreciated! Steve Pretty simple extensions, on lan no nat [4785] type=friend username=4785 secret=test host=dynamic canreinvite=yes [4786] type=friend username=4786 secret=tesst host=dynamic canreinvite=yes exten => 4785,1,Dial(SIP/4785,66) exten => 4785,3,hangup exten => 4786,1,Dial(SIP/4786,66) exten => 4786,3,hangup ___ --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] background SayDigits()?
Why not?
here is an example:
exten => s,1,Set(LCNT=${LEN(${CALLERID(num)})})
exten => s,2,Set(TCNT=0)
exten => s,3,Goto(10);this is where we start the actual saydigits
exten => s,10,GotoIf($[${TCNT} = ${LCNT}]?200);if the value is the
same then there is nothing more to say
exten => s,11,Background(digits/${CALLERID(num):${TCNT}:1}
exten => s,12,Set(TTCNT=${TCNT})
exten => s,13,Set(TCNT=$[${TTCNT} + 1])
exten => s,14,Goto(10)
exten => s,200,Hangup()
Hope this helps
On 1/23/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> Is it possible to background SayDigits()?
>
> I know you can manually Background() each digit individually, but this
> does not solve the problem when you need to do something like
> SayDigits(${EXTEN}) or SayDigits(${CALLERID(number)})
>
> -Dan
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RE: [Asterisk-Users] Firewall/Embeded System/CF/etc
Manny, You really need to try Astlinux. See www.astlinux.org. It does pretty much what you desire. Also see my recent article about Aslinux embedded on a Soekris Net4801 (http://www.tomsnetworking.com/Sections-article153.php) Michael Graves Sr Product Specialist Pixel Power Inc [EMAIL PROTECTED] o(713) 861-4005 o(800) 905-6412 f(713) 864-8668 c(713) 201-1262 > Original Message > Subject: [Asterisk-Users] Firewall/Embeded System/CF/etc > From: "Manny A. Wise" <[EMAIL PROTECTED]> > Date: Mon, January 23, 2006 11:37 am > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > > > I am trying to build an silent non moving parts (fans,HD.etc) embedded > system...Firewall/Asterisk/FXo/FXs/CF/etc > > Looking for anyone running asterisk with Coyote, IPcop, m0n0wal, Shorewall, > etc in the same system/box!!! > > Offlist please... > > Thanks in advance!! > > Manny > > ___ > --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] SIP and NAT - best practices?
Apart of what everyone writes with the NAT=YES I would suggest using canreinvite=no as well as normally asterisk cans the reinvite and this might cause the audio not to get through the NAT and cause dead air for the users specially if the users are behind 2 seperate NAT servers eg. different private networks. By using canreinvite=no and nat=yes most of the NAT problems go away. In this scenario the example would look like this: [2201] user=blah secret=blah auth=blah allow=blah host=dynamic *nat=yes canreinvite=no* Mark Phillips wrote: Most often the simple addition of nat=yes in the relevant sip.conf stanza is all that's required to make a remote SIP phone work from behind a firewall. for example [2201] user=blah secret=blah auth=blah allow=blah host=dynamic nat=yes I've been running 4 remote SIP phones across the internet from my families houses all over the world in this manner. The only issues I get are those of bandwidth availability or rather occasional lack of it. Hosted PBX's are no different. The hosting service should be providing a similar mechanism (although it might not be Asterisk based). Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Michaël Gaudette wrote: Thanks Moises. I was kind of hoping that, at least if I hosted my Asterisk server somewhere where there was no NAT for the * box that the SIP phones wouldn't create any issues. How do you people with Hosted PBX handle the deployment of SIP phones behind NAT firewalls? Is it just elbow grease and configuring every single phone for the customer, or is there a way? Mike you can redirect the ports of the router as well. Or you can configure your SIP phone to use a STUN server. Please read in voip-info.org about SIP NAT, there are good suggestions. regards On 1/20/06, Michakl Gaudette <[EMAIL PROTECTED]> wrote: Hello, I'm a bit new to SIP, and I've set up a SIP line with Asterisk and my wholesale provider. That worked, fine. I ahd to open up the ports on my router, forward them to the correct box, again fine. Now, if I get one of my customers to connect his SIP phone to my Asterisk box, and HE'S behind a NAT firewall, does he have to go through the same process, or is it just the Asterisk box that needs to translate the SIP and RTP port? In other words: if my SIP phone is behind a Linksys router, do I need to configure the Router for any reason? Mike ___ --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 ___ --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] GXP-2000 fw 1.0.1.13 and NTP (SOLVED)
Tony Hoyle wrote: Philip Edelbrock wrote: 18 17.161118 Grandstr_05:a9:bf -> BroadcastARP Who has 206.228.191.144? Gratuitous ARP 19 17.609869 3com_96:2f:eb -> Grandstr_05:a9:bf ARP 206.228.191.144 is at 00:10:4b:96:2f:eb 20 20.155260 206.228.191.144 -> 206.228.191.7 DHCP DHCP Decline - Transaction ID 0xced0 It looks like your DHCP server is in fact broken. It's passing out duplicate addresses - the device 00:10:4b:96:2f:eb already has 206.228.191.144, so the Grandstream (correctly) declines the offer. The server then tries to send the same address *again* instead of selecting a new one, and the same sequence ensues. It should give a different address if the original one is declined. Ah, you are close! I figured it out (*hurray!*). It was in fact a misconfiguration on my part. 144 isn't the end of my subnet, 143 is. So, packet 18 is the phone confirming that it owns IP 144. Packet 19 is from the router saying, "no you don't, I own that" (this is a proxy arp setup). So, the phone declines and requests a new IP. The head scratcher was that for the next request, it requests 144 again, so the DHCP server says (again) "OK, you got it" and the loop continues. Once I adjusted my dhcp config to end my dynamic pool at 143 instead of 144, all was well. Additionally, I noticed that the phone requests these pieces of info in the dhcp response: - Subnet - Router - DNS server(s) - Time Server(s) <--- !! So, I additionally put in the dhcp config a time server (the ip for time.nist.gov for now). And after the first reboot, the phone gets an IP, pings the dhcp server once, registers, sets it's time, checks for firmware updates, and seems perfectly happy. Hurray! Phil ___ --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] OFF TOPIC: Core router upgrade for a voip colocation center
Hello, hope this isn't too far offtopic here but being a troller for a long time here I've realized there is a great knowledge base so I wanted to at least see if i could get some tips. I help run a small colocation company in California and I am in the middle of recommending a new 'core router' platform for our network. We offer mainly colo and dedicated servers, and several of our clients use our space for VOIP services so quality even under high peak usage is a must. We are not huge, but as we have had near 200% growth in the past 12 months and need to expand our network asap to keep up. Simply put, I'd love to hear feedback and/or suggestions from any of you guys who have gone through this already. Our network map is real simple: [Carrier 7609] --> 100 mbit --> Our cisco 7206 --> 100 mbit --> racks [the racks on our end are a series of switches, mainly 2948gl3's] We push about 60 mbit to/from our (1) carrier at peak right now, and the router keeps up fine [its a cisco 7206 npe 150 btw, very low end on the 7206 line], and at peak we have under 50,000 packets per second, and our 7206 File router-choices.txt not changed so no update needed. [EMAIL PROTECTED] Hardware]$ cat router-choices.txt #1 -> http://list.linux-vserver.org/archive/vserver/ #2 -> webhostingtalk.com: jharington68/adam123 http://www.webhostingtalk.com/forumdisplay.php?f=44 #3 -> asterisk mail list http://lists.digium.com/mailman/listinfo/asterisk-users #4 -> cisco mail list? HOTMAIL [EMAIL PROTECTED] pw/adam123 Hello, hope this isn't too far offtopic here but being a troller for a long time here I've realized there is a great knowledge base so I wanted to at least see if i could get some tips. I help run a small colocation company in California and I am in the middle of recommending a new 'core router' platform for our network. We offer mainly colo and dedicated servers, and several of our clients use our space for VOIP services so quality even under high peak usage is a must. We are not huge, but as we have had near 200% growth in the past 12 months and need to expand our network asap to keep up. Simply put, I'd love to hear feedback and/or suggestions from any of you guys who have gone through this already. Our network map is real simple: [Carrier 7609] --> 100 mbit --> Our cisco 7206 --> 100 mbit --> racks [the racks on our end are a series of switches, mainly 2948gl3's] We push about 60 mbit to/from our (1) carrier at peak right now, and the router keeps up fine [its a cisco 7206 npe 150 btw, very low end on the 7206 line], and at peak we have under 50,000 packets per second, and our 7206 has little/no features enabled [just static routes and passing all traffic between 2 Ethernet 100 mbit interfaces]. To date we have had 2 problems, both were DOS attacks launched FROM one of our customer's servers flooding a full 100 mbit wire with more packets per second than the router could handle (the 2948gl3's spiked to about 50% cpu load during the attack but the 7200 literally just died for 3 minutes as the interface(s) all rebooted]. So our main goal to grow is something that can handle a lot more in this arena against a DOS, and handle our future growth. In then next 12 months we plan to add a 2nd carrier, at t3, 100mbit, or possibly oc3 speed, and possibly upgrade our main carrier to a GigE connection. Probably maxing combined in the 300 mbit range, more likely closer to half that in 12 months. Problems/Requirements - Budget is in the $5k to $20k range ($20k if its going to outlast me even past my 12 month projections) - must not 'collapse' under simple packet flow DOS attack - must handle BGP4 from 2 carriers with full route tables - We plan to buy used, prices below are based on USED, 30 day warranty ebay postings = Choices/Options that we have looked at: Option #1: Cisco VXR 7206 [$4k to $12k] Option #2: Cisco 12008 [$7k to $14k] Option #3: Cisco 6509 [$10k to $15k] Here are the 3 main options, broken down a bit more in depth. [I have not ruled out juniper all together, but not enough experience with them and lots of experience with cisco, makes cisco our better option i think, especially since its easier to find used cisco gear than it is to find used juniper gear at a decent price]. [option #1 - Cisco 7206 VXR] Estimated: $4,000 [$6,000 with 400 mhz, $12,000 with the 1 ghz cpu upgrade] 1 Cisco 7206 VXR NPE 300 mhz w/max ram 2 AC Power 2 Fast Ethernet Adapters (1 included on the NPE) + lots of experience on this unit + lots of spare cards (most compatible) + can keep old 7200 as a hot standby, minimizing long term downtime - END OF LIFE/sale/support on most of the 7200 product line over 5 years ago! The VXR model is darn close to end of life i suspect - minimal horse power here for the money, prone to death by packet attack [option #2 - Cisco GSR (12008)] Estimated: $7,000 to $14,000 [varies if I start with GigE or j
Re: [Asterisk-Users] bug in Authenticate application ?
I can not get this to work either. Here is an except from my extensions.conf exten => 123,1,Answer exten => 123,2,Authenticate(1|j) exten => 123,3,SayDigits(3) exten => 123,4,Hangup exten => 123,102,SayDigits(102) exten => 123,103,SayDigits(103) exten => 123,104,SayDigits(104) After dialing 123 and entering 3 invalid Authenticate values, I get the congestion tone. I would expect to hear 'one zero three'. I am running SVN-branch-1.2-r7231 which was downloaded on November 30, 2005. Is this a bug? Don Pobanz aki toku wrote: > I'm Japanese. Sorry,English is not so understood,Please let me question > by items. > In Asterisk-1.2.1 and 1.2.2,I cannnot understand the operation of > Authenticate application's 'j' option. > > exten => 123,1,Answer() > exten => 123,2,Authenticate(789,j) > exten => 123,3,Playback(pin-number-accepted) > exten => 123,4,SayDigits(111) > exten => 123,103,SayDigits(999) > > In this case,When I fail in the authentication ,priority is '1 →2'. > > exten => 123,1,Answer() > exten => 123,2,Authenticate(789,j) > exten => 123,3,Playback(pin-number-accepted) > exten => 123,4,SayDigits(111) > > In this case,When I fail in the authentication ,priority is '1 →2→3→4'. > > exten => 123,1,Answer() > exten => 123,2,Authenticate(789,j) > exten => 123,3,Playback(pin-number-accepted) > exten => 123,104,SayDigits(111) > > In this case ,When I fail in the authentication ,priority is '1 →2→3→4'. > > Is this operation a bug? > Is writing a bug? > > toku ___ --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] SPA-3000 - the party's over :-(
On Mon, 23 Jan 2006, Cory Andrews wrote: I can't speculate as to why their sales of Linksys/Sipura products have been restricted, but as a Linksys VAD I can say we are not under any such restriction at present. its pretty obvious, linksys/sipura are shifting to selling primarily to service providers who would provide service-locked ATAs to end users. sipura telegraphed their intent a long while back by withholding auto-provisioning documentation from anyone except service providers, and now they have completed the move by no longer allowing sales to end users at all. -Dan ___ --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] make linux26
Yeah, that's where I saw contradicting what I saw elsewhere. Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: <[EMAIL PROTECTED]> To: Sent: Monday, January 23, 2006 3:48 PM Subject: Asterisk-Users Digest, Vol 18, Issue 141 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than "Re: Contents of Asterisk-Users digest..." -- Message: 23 Date: Mon, 23 Jan 2006 22:37:55 +0100 From: <[EMAIL PROTECTED]> Subject: RE : [Asterisk-Users] make linux26 To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Message-ID: Content-Type: text/plain; charset="iso-8859-1" Hi Mike, You must continue - for zaptel only - to "make linux26", as it is described in the companion file "README.Linux26" in the Zaptel folder (/usr/src/zaptel). Read the text from this file, as suggested in its title : To build for Linux 2.6, first you must be sure that you have a symlink to your linux-2.6 sources in /usr/src/linux-2.6. The 2.6 kernel no longer needs the full sourcecode to build against it. You can create the symlink to /lib/modules/`uname -r`/build/ and then you can type: # make linux26 # make install Note that you will also need CRC-CCITT functions compiled with your kernel or as a kernel module. These can be selected from the "Library Routines" submenu during kernel configuration via "make menuconfig" It is a good habit to read all this "README..." files before to do something, as it is important to read any user manual for any sofisticated equipment ;-) Good luck ! Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Mike Hammett Envoyé : lundi 23 janvier 2006 22:10 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] make linux26 I thought I read somewhere that you no longer have to do a special make command for the 2.6 kernel. Is this true, or should I still make linux26? I'm having problems getting anything zaptel to load properly. Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060123/9b097a35/attachment-0001.htm ___ --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] Caller ID and Sipura Router
I had this similar problem I have an extension (Sipura 1001) ext 2000 I call the phone, and i have an AGI lookup the incoming CID in a database, and reset the CIDname to be that from the database However, In my Mysql table i had callerid 2000 as the value. so.. if it was my cell calling it would be Ben Cell 2000 on the caller id. I removed the Callerid value of 2000 from the mysql table, and now it passes in the callerid number from the incoming call. I dont use the callerid field in the system any more because of this. If i am dialing out, i just set the callerid number to that of the calling extension... I dont dial my internal extensions that often, but it works. So.. I would Remove the callerid line from the configs, and test. > On Saturday 21 January 2006 20:30, Conrad Beckert wrote: >> Could anyone please help me with that: >> >> I have an analog telephone connected to Asterisk using a Sipura 2002 >> ATA. >> When calling the extension, the caller ID presented is always the number >> of >> that extension rather than the number of the calling one. >> >> While I learned that this is the new standard behaviour (?) of Asterisk, >> I >> want to show the original caller ID. >> >> I tried the options o and f in the dial command - e.g. > Don't know about "f" but "o" is "Operator extension, used for operator > exit by > pressing zero in voicemail " >> exten => 1002,4,dial(sip/2999,20,o) >> >> no avail. The phone rings and shows 2999 instead of the calling party! >> >> The SIPURA seems to be ok: when I connect to Sipgate/Nikotel etc. >> directly, >> everything is ok >> >> What's wrong? My Asterisk Version is 1.2.1 > sip.conf > [2999] > type=friend > secret=x > callerid=Analog Phone <1002> > regexten=1002 > etc... > > exten => 1002,1,Dial(sip/2999,20) > exten => 1002,2,Hangup > > Does this give you a clue? > benchev > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Connection TDM400P to UK PSTN
Thanks for that, insightful The quality on both ends with those settings is quite good -- but the extension side still has a ridiculous -- almost duplicate -- echo! If I turn the txgain right down, I lose all sound.of course the signals in ztmonitor show up perfect then, but can't hear anything (DTMF tones too low or whatever...) Still wondering if it's an impedance issue...or something along those lines/chipset..etc I'm going to attempt upgrading Zaptel now, without upgrading the asterisk Chris - Original Message - From: "Chris Bagnall" <[EMAIL PROTECTED]> To: "'Chris Earle (CBL)'" <[EMAIL PROTECTED]>; "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Sunday, January 22, 2006 9:13 AM Subject: RE: [Asterisk-Users] Re: Connection TDM400P to UK PSTN > > Successfully got the adapters to allow the BT phones to ring > > on lines coming out of a TDM.. but now my latest > > problem is echo. > > Suggestions / Experiences in UK appreciated > > Most of our clients with BT lines tend to have ISDN BRIs, but we do have one > in Northampton running 3 analogue lines from a TDM400. > > zaptel.conf is as follows: > fxsks = 1-3 > loadzone= uk > defaultzone = uk > > The TDM driver is loaded with opermode=UK and the output from dmesg confirms > this. > > Relevant settings from zapata.conf are as follows: > echocancel=yes > echocancelwhenbridged=yes > echotraining=800 > rxgain=8.0 > txgain=-4.0 > busydetect=yes > group = 1 > context = inbound > channel => 1-2 > group = 2 > context = inbound > channel => 3 > > They're running Asterisk/Zaptel 1.0.10. There were major echo issues when we > first deployed the system back in September, but some careful tweaking of > rxgain and txgain seems to have largely resolved the situation. Certainly my > experience has been that rxgain and txgain have far more impact on echo > reduction than any of the echo-specific settings. Get the gains right first, > then play with the echo-specific settings. > > Regards, > > Chris > -- > C.M. Bagnall, Director, Minotaur I.T. Limited > This email is made from 100% recycled electrons > > > > -- > This message has been scanned for viruses and dangerous content by > MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Zaptel issues
[EMAIL PROTECTED] ~]# which modprobe /sbin/modprobe [EMAIL PROTECTED] ~]# modprobe --version module-init-tools version 3.1-pre5 [EMAIL PROTECTED] ~]# dmesg | tail zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' ztdummy: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' ztdummy: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' zaptel: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' ztdummy: version magic '2.6.8-022stab061.1 686 4KSTACKS gcc-3.4' should be '2.6.8-022stab061.1 686 4KSTACKS gcc-3.3' It looks like my gcc versions are different from the one that made the kernel and the one that made the zaptel stuff. So then of the zt lines, do I only need: install ztdummy /sbin/modprobe --ignore-install ztdummy && /sbin/ztcfg Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: <[EMAIL PROTECTED]> To: Sent: Friday, January 13, 2006 4:49 AM Subject: Asterisk-Users Digest, Vol 18, Issue 82 -- Message: 12 Date: Fri, 13 Jan 2006 11:52:20 +0200 From: Tzafrir Cohen <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Zaptel issues To: Asterisk Users Mailing List - Non-Commercial Discussion Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=us-ascii On Thu, Jan 12, 2006 at 09:39:18AM -0600, Mike Hammett wrote: On a side note: When poking around, I noticed in the zaptel Makefile that there is a section talking about ztdummy automatically being included on 2.6 kernels. Is this correct? On to the main topic: Any ideas for troubleshooting this? [EMAIL PROTECTED] zaptel-1.2.1]# /etc/rc.d/init.d/zaptel start Loading zaptel framework: FATAL: Error inserting zaptel (/lib/modules/2.6.8-022stab050.1/extra/zaptel.ko): Invalid module format [FAILED] Waiting for zap to come online...Error: missing /dev/zap! [EMAIL PROTECTED] libpri-1.2.1]# modprobe ztdummy WARNING: Error inserting zaptel (/lib/modules/2.6.8-022stab050.1/extra/zaptel.ko): Invalid module format WARNING: Error inserting zaptel (/lib/modules/2.6.8-022stab050.1/extra/zaptel.ko): Invalid module format FATAL: Error inserting ztdummy (/lib/modules/2.6.8-022stab050.1/extra/ztdummy.ko): Invalid module format FATAL: Error running install command for ztdummy Could you please provide the output of following: which modprobe modprobe --version To make things simpler, do away with the stuff that the zaptel install puts in /etc/modprobe.d/zaptel (or /etc/modprobe.conf ). (ztdummy needs no ztcfg run after it) Also, please provide the latest relevant kernel log messages: dmesg | tail -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] SPA-3000 - the party's over :-(
[EMAIL PROTECTED] a écrit : The party's over folks, the new official cisco/linksys/sipura policy is to no longer sell SPA-3000's to end users. Oh well. There's plenty other interesting ATAs around... their loss! Their support was pretty horrible in my experience anyways. Could this will be a boon to IAX-compliant ATAs? Cheers, Jean-Michel. ___ --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] Newer version of Zaptel with 1.0 branch of *
Has upgrading to the newer Zaptel allowed you to use the newer improvements in it? (sorry if that was implied) Thanks for the speedy reply Chris - Original Message - From: "Adam Robins" <[EMAIL PROTECTED]> To: "Chris Earle (CBL)" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, January 23, 2006 5:01 PM Subject: RE: [Asterisk-Users] Newer version of Zaptel with 1.0 branch of * > I have done this successfully with Asterisk 1.07 and Zaptel 1.09 and > 1.2.1 for the same reasons as you. > > However, if you ever need to go recompile Asterisk, then you will first > need to recompile the old Zaptel, compile Asterisk and the new Zaptel > again. > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Chris > Earle (CBL) > Sent: Monday, January 23, 2006 4:41 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Newer version of Zaptel with 1.0 branch of * > > Is it possible to run the CVS-HEAD/Stable version of Zaptel (1.2 > whatever) with an older version of Asterisk? I'm running 1.09, but I was > wondering if I could get at the newer echo cancellers like KB1 and MG2 > without upgrading to Asterisk 1.2? > > > I'm going out on a limb here to try and fix a serious echo problem on a > TDM > + BT PSTN line in the UK > > > Thanks for your suggestions everyone > > > -- > Chris Earle > System Solutions Specialist, > > > -- > This message has been scanned for viruses and dangerous content by > MailScanner, and is believed to be clean. > > ___ > --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 > > The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. > > > > -- > This message has been scanned for viruses and dangerous content by > MailScanner, and is believed to be clean. > -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Newer version of Zaptel with 1.0 branch of *
I have done this successfully with Asterisk 1.07 and Zaptel 1.09 and 1.2.1 for the same reasons as you. However, if you ever need to go recompile Asterisk, then you will first need to recompile the old Zaptel, compile Asterisk and the new Zaptel again. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Earle (CBL) Sent: Monday, January 23, 2006 4:41 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Newer version of Zaptel with 1.0 branch of * Is it possible to run the CVS-HEAD/Stable version of Zaptel (1.2 whatever) with an older version of Asterisk? I'm running 1.09, but I was wondering if I could get at the newer echo cancellers like KB1 and MG2 without upgrading to Asterisk 1.2? I'm going out on a limb here to try and fix a serious echo problem on a TDM + BT PSTN line in the UK Thanks for your suggestions everyone -- Chris Earle System Solutions Specialist, -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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 The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --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] SPA-3000 - the party's over :-(
I can't speculate as to why their sales of Linksys/Sipura products have been restricted, but as a Linksys VAD I can say we are not under any such restriction at present. Cory Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 + voice @ 800.398.VOIP Ext 22 email @ [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, January 23, 2006 4:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SPA-3000 - the party's over :-( On Mon, 23 Jan 2006, Cory Andrews wrote: > Curious where you came by this bit of information? My contacts within > the industry cannot confirm this. you need better contacts... http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-34702223616 .htm read the red box... -Dan ___ --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] SPA-3000 - the party's over :-(
On Mon, 23 Jan 2006 13:28:33 -0800 (PST) [EMAIL PROTECTED] wrote: > > On Mon, 23 Jan 2006, Cory Andrews wrote: > > Curious where you came by this bit of information? My contacts > > within the industry cannot confirm this. > > you need better contacts... > > http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-34702223616.htm > > read the red box... I heard about this several months ago. I was looking to set up a pbx at home, but was waiting to buy the spa3000 until I was ready to set up the server (no spare time). I ended up buying the spa3000 several months early for this very reason. Regards, Ozz. pgpGAzsvO2y8J.pgp Description: PGP signature ___ --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] Installing the none commercial intel g729codecsinto [EMAIL PROTECTED] 2.2?
Dear Charles Wang and all Asterisk users and supporters! Thank Charles for giving me this intruction and it works greate for me after I copy the into the modules directory. I didn't know if that easy. However, after I copy and did the reload command, it didn't work until I have to do the restart now. Again, Thank Charles! and hope you guys enjoy the new version of asterisk. Regards, Lan. - Original Message - From: "Charles Wang" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Sunday, January 22, 2006 6:02 AM Subject: Re: [Asterisk-Users] Installing the none commercial intel g729codecsinto [EMAIL PROTECTED] 2.2? I have the same problem too. I install the G.729 (IPP) to asterisk 1.0.x, and it works well. When I change asterisk from 1.0.x to 1.2.x, and G.729 seems work fine. I can use "show translation" and find it too. But when I make a call using G.729. The asterisk (1.2.1) crashed. If i mark the line "allow=g729" from sip.conf. And asterisk works fine. 2006/1/22, Guillermo Salas M <[EMAIL PROTECTED]>: > Con fecha 21/1/2006, "Francesco Peeters (Asterisk)" > <[EMAIL PROTECTED]> escribió: > > >On Sat, January 21, 2006 23:21, Franz Bräuer said: > >> Hi, > >> > >> MapsAir wrote: > >>> Has anyone successfully Installing the none commercial intel g729 codecs > >>> into [EMAIL PROTECTED] 2.2? > > I'm using g723.1 and works very well. > > >> > >> Installed them today. Installing from source didn't work for me (Debian, > >> Asterisk 1.2 from svn) but just adding the binaries (see the wiki on > >> voip.org) did the job. Have you already tried the binaries? > >> > > > >Kewl! Those work like a treat! > > > >As my testbox is a PII-750 running [EMAIL PROTECTED] 2.2 I did: > > > >cd /usr/lib/asterisk/modules/ > >wget http://kvin.lv/pub/Linux/Asterisk/codec_g723-gcc-pentium2.so > >wget http://kvin.lv/pub/Linux/Asterisk/codec_g729-gcc-pentium2.so > > > >After reloading, 'show translation' gives: > > Translation times between formats (in milliseconds) > > Source Format (Rows) Destination Format(Columns) > > > > g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc > > g723 -22 8 817 8 724 115 19897 > >gsm 151 - 7 716 7 623 114 19796 > > ulaw 14616 - 111 2 118 109 19291 > > alaw 14616 1 -11 2 118 109 19291 > > g726 154241010 -10 926 117 20099 > > adpcm 14616 2 211 - 118 109 19291 > > slin 14515 1 110 1 -17 108 19190 > > lpc10 161311717261716 - 124 207 106 > > g729 16939252534252441 - 215 114 > > speex 16030161625161532 123 - 105 > > ilbc 17343292938292845 136 219 - > > > >Jolly good show, old chap! > > > >-- > >F Peeters > > PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch > > 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 > >Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. > > AMD Duron 1GHz - 1GB - * 1.2.1 > > 2 Sweex HFC-PCI cards > >___ > >--Bandwidth and Colocation provided by Easynews.com -- > > > >Asterisk-Users mailing list > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-user > ___ > --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 > -- Best Regards Charles ___ --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] GXP-2000 fw 1.0.1.13 and NTP
Philip Edelbrock wrote: 18 17.161118 Grandstr_05:a9:bf -> BroadcastARP Who has 206.228.191.144? Gratuitous ARP 19 17.609869 3com_96:2f:eb -> Grandstr_05:a9:bf ARP 206.228.191.144 is at 00:10:4b:96:2f:eb 20 20.155260 206.228.191.144 -> 206.228.191.7 DHCP DHCP Decline - Transaction ID 0xced0 It looks like your DHCP server is in fact broken. It's passing out duplicate addresses - the device 00:10:4b:96:2f:eb already has 206.228.191.144, so the Grandstream (correctly) declines the offer. The server then tries to send the same address *again* instead of selecting a new one, and the same sequence ensues. It should give a different address if the original one is declined. Tony ___ --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] Newer version of Zaptel with 1.0 branch of *
Is it possible to run the CVS-HEAD/Stable version of Zaptel (1.2 whatever) with an older version of Asterisk? I'm running 1.09, but I was wondering if I could get at the newer echo cancellers like KB1 and MG2 without upgrading to Asterisk 1.2? I'm going out on a limb here to try and fix a serious echo problem on a TDM + BT PSTN line in the UK Thanks for your suggestions everyone -- Chris Earle System Solutions Specialist, -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] make linux26
Title: Message Hi Mike, You must continue - for zaptel only - to "make linux26", as it is described in the companion file "README.Linux26" in the Zaptel folder (/usr/src/zaptel). Read the text from this file, as suggested in its title : To build for Linux 2.6, first you must be sure that you have asymlink to your linux-2.6 sources in /usr/src/linux-2.6. The 2.6kernel no longer needs the full sourcecode to build against it. Youcan create the symlink to /lib/modules/`uname -r`/build/ and thenyou can type: # make linux26# make install Note that you will also need CRC-CCITT functions compiledwith your kernel or as a kernel module. These can beselected from the "Library Routines" submenu during kernelconfiguration via "make menuconfig" It is a good habit to read all this "README..." files before to do something, as it is important to read any user manual for any sofisticated equipment ;-) Good luck ! Best Regards, Francois BERGERET, France. -Message d'origine-De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Mike HammettEnvoyé : lundi 23 janvier 2006 22:10À : asterisk-users@lists.digium.comObjet : [Asterisk-Users] make linux26 I thought I read somewhere that you no longer have to do a special make command for the 2.6 kernel. Is this true, or should I still make linux26? I'm having problems getting anything zaptel to load properly. Mike HammettIntelligent Computing Solutionshttp://www.ics-il.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] make linux26
Mike Hammett wrote: I thought I read somewhere that you no longer have to do a special make command for the 2.6 kernel. Is this true, or should I still make linux26? I'm having problems getting anything zaptel to load properly. I've found that doing a make causes it to do a make linux26 if the kernel is detected. I still do it out of habbit. Doug ___ --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] SPA-3000 - the party's over :-(
On Mon, 23 Jan 2006, Cory Andrews wrote: Curious where you came by this bit of information? My contacts within the industry cannot confirm this. you need better contacts... http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-34702223616.htm read the red box... -Dan ___ --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] Polycom phones and dynamic IP for NAT
I know the Polycoms work with NAT, but you have to specify the public IP. Is there anyway for it to discover the external IP automatically? I like the phones (been playing with a 301) but for some of our clients who have a dynamic IP (and no hope of getting a static ie cable or residential DSL) I’d be afraid to use them since you have to specify the IP. What about the Cisco phones? Is the IP hard set? Are there any good “dynamic IP” compatible SIP phones that aren’t crap? Bill ___ --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] GXP-2000 fw 1.0.1.13 and NTP
Kristof Hardy wrote: Was there a resolution to this issue? The GXP-2000 seems to be a very popular phone, so I can't imagine others on the list not experiencing this? Or is this part of a batch with unresolvable problems that I need to send back to the seller? Well, I'm using dozens of these phones without this problem. What kind of DHCP/ntp server are you using? I'm using dnsmasq on a Debian box, together with the ntp-server. I'm using a mixture of 1.0.1.13 beta and .12 firmwares, both working correct. The DHCP server is on the same 100BaseT switch as the phone right now (they are literally just a few feet away from each other). DHCP server is on Fedora 3 Linux "Internet Systems Consortium DHCP Server V3.0.1" (from the rpm: dhcp-3.0.1-44_FC3). Packet sniffer shows the phone getting in some sort of fight with the dhcp server. I attached a text dump of the sniff. You can see a repeating conversation from packet 20 to 40, and it continues on and on like that. And, my logs are filling up with gazillions of these (pattern repeats every 3 seconds): Jan 23 12:06:41 DrTheopolis dhcpd: DHCPDISCOVER from 00:0b:82:05:a9:bf via eth0 Jan 23 12:06:41 DrTheopolis dhcpd: DHCPOFFER on 206.228.191.144 to 00:0b:82:05:a9:bf via eth0 Jan 23 12:06:41 DrTheopolis dhcpd: DHCPREQUEST for 206.228.191.144 (206.228.191.7) from 00:0b:82:05:a9:bf via eth0 Jan 23 12:06:41 DrTheopolis dhcpd: DHCPACK on 206.228.191.144 to 00:0b:82:05:a9:bf via eth0 While I was thinking of logs, I set up remote syslog for the phone, but all I see while it is set to dhcp is a single log noting the firmware versions on the phone. With a static IP it logs info about registering w/ * (which it does successfully and I can make calls). Phil 1 0.00 0.0.0.0 -> 255.255.255.255 DHCP DHCP Discover - Transaction ID 0xaabbccdd 2 0.727622 206.228.191.7 -> 206.228.191.144 DHCP DHCP Offer- Transaction ID 0xaabbccdd 3 0.746653 0.0.0.0 -> 255.255.255.255 DHCP DHCP Request - Transaction ID 0xaabbccde 4 0.749231 206.228.191.7 -> 206.228.191.144 DHCP DHCP ACK - Transaction ID 0xaabbccde 5 0.766593 Grandstr_05:a9:bf -> BroadcastARP Who has 206.228.191.1? Tell 206.228.191.144 6 0.997865 3com_96:2f:eb -> Grandstr_05:a9:bf ARP 206.228.191.1 is at 00:10:4b:96:2f:eb 7 1.308918 206.228.191.144 -> 206.228.191.7 DHCP DHCP Release - Transaction ID 0xaabbccdf 8 14.164223 0.0.0.0 -> 255.255.255.255 DHCP DHCP Discover - Transaction ID 0xcecb 9 14.164531 206.228.191.7 -> 206.228.191.144 DHCP DHCP Offer- Transaction ID 0xcecb 10 14.166809 0.0.0.0 -> 255.255.255.255 DHCP DHCP Request - Transaction ID 0xcecc 11 14.172534 206.228.191.7 -> 206.228.191.144 DHCP DHCP ACK - Transaction ID 0xcecc 12 14.175408 Grandstr_05:a9:bf -> BroadcastARP Who has 206.228.191.144? Gratuitous ARP 13 14.339375 3com_96:2f:eb -> Grandstr_05:a9:bf ARP 206.228.191.144 is at 00:10:4b:96:2f:eb 14 17.155641 206.228.191.144 -> 255.255.255.255 DHCP DHCP Discover - Transaction ID 0xcece 15 17.155975 206.228.191.7 -> 206.228.191.144 DHCP DHCP Offer- Transaction ID 0xcece 16 17.158134 206.228.191.144 -> 255.255.255.255 DHCP DHCP Request - Transaction ID 0xcecf 17 17.159263 206.228.191.7 -> 206.228.191.144 DHCP DHCP ACK - Transaction ID 0xcecf 18 17.161118 Grandstr_05:a9:bf -> BroadcastARP Who has 206.228.191.144? Gratuitous ARP 19 17.609869 3com_96:2f:eb -> Grandstr_05:a9:bf ARP 206.228.191.144 is at 00:10:4b:96:2f:eb 20 20.155260 206.228.191.144 -> 206.228.191.7 DHCP DHCP Decline - Transaction ID 0xced0 21 20.155760 206.228.191.144 -> 255.255.255.255 DHCP DHCP Discover - Transaction ID 0xced1 22 20.155981 206.228.191.7 -> 206.228.191.144 DHCP DHCP Offer- Transaction ID 0xced1 23 20.158255 206.228.191.144 -> 255.255.255.255 DHCP DHCP Request - Transaction ID 0xced2 24 20.159714 206.228.191.7 -> 206.228.191.144 DHCP DHCP ACK - Transaction ID 0xced2 25 20.161242 Grandstr_05:a9:bf -> BroadcastARP Who has 206.228.191.144? Gratuitous ARP 26 20.640088 3com_96:2f:eb -> Grandstr_05:a9:bf ARP 206.228.191.144 is at 00:10:4b:96:2f:eb 27 23.165159 206.228.191.144 -> 206.228.191.7 DHCP DHCP Decline - Transaction ID 0xced3 28 23.165658 206.228.191.144 -> 255.255.255.255 DHCP DHCP Discover - Transaction ID 0xced4 29 23.165879 206.228.191.7 -> 206.228.191.144 DHCP DHCP Offer- Transaction ID 0xced4 30 23.168148 206.228.191.144 -> 255.255.255.255 DHCP DHCP Request - Transaction ID 0xced5 31 23.170237 206.228.191.7 -> 206.228.191.144 DHCP DHCP ACK - Transaction ID 0xced5 32 23.172210 Grandstr_05:a9:bf -> BroadcastARP Who has 206.228.191.144? Gratuitous ARP 33 23.180374 3com_96:2f:eb -> Grandstr_05:a9:bf ARP 206.228.191.144 is at 00:10:4b:96:2f:eb 34 26.165097 206.228.191.144 -> 206.228.191.7 DHCP DHCP Decline - T
Re: [Asterisk-Users] Announcing PodMail 1.0 (GPL)
Sorry about this - I hit send by accident while I was still writing the email. Pretend it never happened. PaulH - Original Message - From: <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, January 24, 2006 7:26 AM Subject: Re: [Asterisk-Users] Announcing PodMail 1.0 (GPL) > Cute? > > But it can use LDAP... > > PaulH > > - Original Message - > From: "Ben Klang" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Tuesday, January 24, 2006 3:58 AM > Subject: [Asterisk-Users] Announcing PodMail 1.0 (GPL) > > > > Hello Asterisk Community. > > > > While sitting at lunch the other day I had a typical napkin-prototype > idea: > > What if I could make my Asterisk Voicemail accessible as a Podcast in > iTunes? > > Three hours later with the help of two friends I had a working proof of > > concept. Now we are releasing the polished version of this idea as > PodMail > > 1.0 > > > > PodMail brings together open-source telephony and Podcasting to create a > new, > > useful way of accessing voicemail and podcasting. > > > > PodMail integrates with Asterisk to provide a secure podcast of your > > voicemail. Supporting authentication directly against voicemail.conf or > using > > an LDAP directory, PodMail allows you to subscribe to your own voicemail > box. > > Each time you dock your iPod, your new voicemails will sync right along. > > Listen to your voicemail at your convenience and without using cell > minutes. > > > > PodMail also allows for a brand new type of PodCasting. Unchain > Podcasting > > from the computer! Configure PodMail for public access and you have a > > ready-to-run PodCast. Updating your Podcast is as easy as phone call. > > Moblogging has never been so easy or flexible. > > > > Live Demo: > > Do not miss out our live demo at http://podmail.alkaloid.net/ > > Leave us a message in one of our mailboxes, subscribe to one of the > PodMail > > Podcasts, then see and hear your message immediately! > > > > Check out the PodMail Documentation and Installation Notes at > > http://projects.alkaloid.net. PodMail is released under the terms of the > > GPL. > > > > Enjoy! > > /BAK/ > > -- > > Ben Klang > > Alkaloid Networks > > http://projects.alkaloid.net > > [EMAIL PROTECTED] > > 404.475.4850 > > > > ___ > > --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 > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Polycom FW
Thread hijack - grr! On 23-Jan-06, at 7:37 AM, Brian Capouch wrote: Doug Lytle wrote: > Douglas Garstang wrote: > >> We conducted focus groups, looking at several different vendors, >> before we decided to go with the Polycom. From the user interface >> perspective, the Polycom's won hands down. I was never involved with >> it, but apparently to configure the Cisco's you need to be converting >> hex??? Yuk! >> >> >> > > This is not correct. The Polycom and Cisco phone configuration is very > similar. > Does anyone know whether the reports of the errors in the Asterisk book wrt to Dundi were correct or not? Anytime I read a technical posting that is written with such a harsh tone, I wonder if it has any meat to it. . . B. ___ --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] make linux26
I thought I read somewhere that you no longer have to do a special make command for the 2.6 kernel. Is this true, or should I still make linux26? I'm having problems getting anything zaptel to load properly. Mike HammettIntelligent Computing Solutionshttp://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with bad audio using MPC..
I sent the below message out last Friday when the list seemed to be having issues. Never got any responses and not sure if it just no one knows or if it did not get through. Please don't flog me too bad for reposting... :-) Hi all, I am having some audio quality issues with a provider under sip. The issue I am having is that the audio seems to be acting like a simplex connection. I have tested my setup with a second provider and the audio quality to them is great. Checked network type issues, latency, packet loss, etc. and all seems to be ok. What I did find was a difference in the RTP debugs. Here is a capture from both providers: RTP Debug from Teliax SIP connection w/ good audio: Sent RTP packet to 208.139.204.228:10102 (type 0, seq 9473, ts 135520, len 160) Sent RTP packet to 208.139.204.228:10102 (type 0, seq 9474, ts 135680, len 160) Got RTP packet from 208.139.204.228:10102 (type 0, seq 4467, ts 149600, len 160) Got RTP packet from 208.139.204.228:10102 (type 0, seq 4468, ts 149760, len 160) RTP Debug from MPC connection w/ bad audio: Sent RTP packet to 66.128.8.234:61414 (type 0, seq 3506, ts 51040, len 160) Sent RTP packet to 66.128.8.234:61414 (type 0, seq 3507, ts 51200, len 160) Got RTP packet from 66.128.8.234:61414 (type 0, seq 23701, ts 52480, len 80) Got RTP packet from 66.128.8.234:61414 (type 0, seq 23702, ts 52560, len 80) Got RTP packet from 66.128.8.234:61414 (type 0, seq 23703, ts 52640, len 80) Got RTP packet from 66.128.8.234:61414 (type 0, seq 23704, ts 52720, len 80) Notice that the lengths are different in the MPC packet capture. I am getting two packets from them to every one of mine. I was askied by them to set my packet size to 20ms but do not know where to do that or if it can be done. They also stated that the packet size should be negotiated in the SIP INVITE and 200 OK messages. Can someone point me in the right direction? Even just what to look for here. I am currently running version 1.2.2, but had the same issues with 1.09 and 1.2. Thanks, Robert ___ --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] app_rxfax.so and app_txfax.so
Loic, May be there are mixed modules from distinct asterisk versions in the asterisk/modules folder. Try burn everything and restart from scratch. If you are applyng patches and compiling, try older asterisk tarballs, some patches are very attached to a given asterisk version. Good luck, -- Paulo Scardine Support Internet.net wrote: Hi, I search in the archives and I don't find that case. I'm wanted to do Asterisk+spandsp working. I have installed spandsp and apply the patch without any errors. I have recompiled Asterisk and When I try to start it, the output say : [app_txfax.so]Jan 23 15:17:12 WARNING[3022]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_txfax.so: undefined symbol: span_set_message_handler If somebody can help me it would be appreciate, Loic Foucault ___ --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] SPA-3000 - the party's over :-(
Curious where you came by this bit of information? My contacts within the industry cannot confirm this. Cory Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 + voice @ 800.398.VOIP Ext 22 email @ [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, January 23, 2006 3:22 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] SPA-3000 - the party's over :-( The party's over folks, the new official cisco/linksys/sipura policy is to no longer sell SPA-3000's to end users. Buy them while you still can :-( -Dan ___ --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] Announcing PodMail 1.0 (GPL)
>Supporting authentication directly against voicemail.conf or using > an LDAP directory, -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: January 23, 2006 3:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Announcing PodMail 1.0 (GPL) Cute? But it can use LDAP... PaulH - Original Message - From: "Ben Klang" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, January 24, 2006 3:58 AM Subject: [Asterisk-Users] Announcing PodMail 1.0 (GPL) > Hello Asterisk Community. > > While sitting at lunch the other day I had a typical napkin-prototype idea: > What if I could make my Asterisk Voicemail accessible as a Podcast in iTunes? > Three hours later with the help of two friends I had a working proof of > concept. Now we are releasing the polished version of this idea as PodMail > 1.0 > > PodMail brings together open-source telephony and Podcasting to create a new, > useful way of accessing voicemail and podcasting. > > PodMail integrates with Asterisk to provide a secure podcast of your > voicemail. Supporting authentication directly against voicemail.conf or using > an LDAP directory, PodMail allows you to subscribe to your own voicemail box. > Each time you dock your iPod, your new voicemails will sync right along. > Listen to your voicemail at your convenience and without using cell minutes. > > PodMail also allows for a brand new type of PodCasting. Unchain Podcasting > from the computer! Configure PodMail for public access and you have a > ready-to-run PodCast. Updating your Podcast is as easy as phone call. > Moblogging has never been so easy or flexible. > > Live Demo: > Do not miss out our live demo at http://podmail.alkaloid.net/ > Leave us a message in one of our mailboxes, subscribe to one of the PodMail > Podcasts, then see and hear your message immediately! > > Check out the PodMail Documentation and Installation Notes at > http://projects.alkaloid.net. PodMail is released under the terms of the > GPL. > > Enjoy! > /BAK/ > -- > Ben Klang > Alkaloid Networks > http://projects.alkaloid.net > [EMAIL PROTECTED] > 404.475.4850 > > ___ > --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 ___ --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 SIP phones to Cisco Unity via CCM 4.0 SIP Trunk
Hi,I've got a CCM ( Cisco Call Manager ), with a Cisco Unity VM server and about 45 SCCP phones on the ccm, and 200 users on unity. we want to migrate all users to IP Phones to ditch our ancient phone system. I would love to get Linksys-Sipura SPA-941s for the 150 users not on IP phones yet and run sip to an asterisk server, but have their voicemail on Unity. these phones are $150 each, the alternative is cisco 7940s ( around $250 ) running SCCP through the CCM. at the quantities I'm talking about, $100 is significant.Does anyone have any idea how to get this done? I've tried this:exten => 123,1,Dial(SIP/sipphone,20)exten => 123,2,Dial(SIP/ccm/3040)where 3040 is our VM pilot for ccm. but all it does is take us to the main greeting.we have smartnet, but they haven't been helpful at all I called digium to see if they could help if we paid, but they said they've never heard of cisco unityhelp?thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [asterisk-dev] dial out and message playback
On Tue, 24 Jan 2006, Danish Samad wrote: Hi, -users questions In a normal PBX environment a user usually calls in and IVR's are played according to a predefined dialplan. Iam trying to develop an application where asterisk dials out to a user and initiates an IVR instead (please note that the IVR is not static and may vary according to different condtions). Can someone guide me how this is possible using Asterisk. Do I need to write some sort of AGI or application? use .call files in /var/spool/asterisk/outgoing I have looked into the autodial out feature but I am thinking of a more flexible or optimal solution. Any help will be appreciated. Regards, Danish - wasim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --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] app_rxfax.so and app_txfax.so
What version of SpanDSP are you running? You should be running -pre21 -Original Message-From: Support Internet.net [mailto:[EMAIL PROTECTED]Sent: Monday, January 23, 2006 1:18 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] app_rxfax.so and app_txfax.so Hi, I search in the archives and I don't find that case. I'm wanted to do Asterisk+spandsp working. I have installed spandsp and apply the patch without any errors. I have recompiled Asterisk and When I try to start it, the output say : [app_txfax.so]Jan 23 15:17:12 WARNING[3022]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_txfax.so: undefined symbol: span_set_message_handler If somebody can help me it would be appreciate, Loic Foucault ___ --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] Fw: setting outgoing caller ID by the queue an extension is logged into
Greetings fellow list members, I am trying to add some tricky functionality to Asterisk dialplan and I was curious if anyone else has come up with a solution to something like this. Basically I have phone representatives that log into one of several queues (not using chan Agent, we log in by the extension), and frequently these agents have to make attended transfer calls to outside numbers. This transfer basically amounts to a new outgoing call. I have been asked to set the caller ID for these outgoing calls based on the queue the phone representative is currently logged in to. Unfortunetly I cannot think of a way to do this. The incomming and outgoing calls are two different calls. I have considered using DBPut and DBGet to store this information in a database. This might work, but I am also concerned about the overhead involved. I cannot think of a way to do this using global variables since I need to store a seperate value for each extension. Has anyone run into an issue like this and come up with a solution? Any thoughts are much appreciated. Thank you, Franklin Webb Assistant IT Project Leader Inter Media Marketing Solutions ___ --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] app_rxfax.so and app_txfax.so
Support Internet.net wrote: I'm wanted to do Asterisk+spandsp working. I have installed spandsp and apply the patch without any errors. I have recompiled Asterisk and When I try to start it, the output say : [app_txfax.so]Jan 23 15:17:12 WARNING[3022]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_txfax.so: undefined symbol: span_set_message_handler Just a guess, You've installed more then one version of spandsp. Remove all modules and libraries and re-install spandsp. Doug ___ --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
