Re: [Asterisk-Users] What is the best IP phone?
i like the pingtel phones. www.pingtel.com - Original Message - From: Surajee Ratnayake To: [EMAIL PROTECTED] Sent: Saturday, September 06, 2003 4:40 AM Subject: [Asterisk-Users] What is the best IP phone? hi, Can anybody suggest me a good, reliable, robust, SIP supported hardware IP phone? Surajee
Re: [Asterisk-Users] What is the best IP phone?
can the pingtel phonebe used smoothly with asterisk? - Original Message - From: Steve Totaro To: [EMAIL PROTECTED] Sent: Saturday, September 06, 2003 11:15 PM Subject: Re: [Asterisk-Users] What is the best IP phone? i like the pingtel phones. www.pingtel.com - Original Message - From: Surajee Ratnayake To: [EMAIL PROTECTED] Sent: Saturday, September 06, 2003 4:40 AM Subject: [Asterisk-Users] What is the best IP phone? hi, Can anybody suggest me a good, reliable, robust, SIP supported hardware IP phone? Surajee
Re: [Asterisk-Users] What is the best IP phone?
how about the call transfer feature? is it working fine? and can u pls let me know the price too.. - Original Message - From: Steve Totaro To: [EMAIL PROTECTED] Sent: Sunday, September 07, 2003 12:16 AM Subject: Re: [Asterisk-Users] What is the best IP phone? i setup asterisk for the first time yesterday morning and had two pingtel phones working by noon (with message waiting indicators) the phones have a nice web interface for config and speed dials. its a java phone too. they are a little pricey and maybe too funky looking for some people though - Original Message - From: Surajee Ratnayake To: [EMAIL PROTECTED] Sent: Saturday, September 06, 2003 6:08 AM Subject: Re: [Asterisk-Users] What is the best IP phone? can the pingtel phonebe used smoothly with asterisk? - Original Message - From: Steve Totaro To: [EMAIL PROTECTED] Sent: Saturday, September 06, 2003 11:15 PM Subject: Re: [Asterisk-Users] What is the best IP phone? i like the pingtel phones. www.pingtel.com - Original Message - From: Surajee Ratnayake To: [EMAIL PROTECTED] Sent: Saturday, September 06, 2003 4:40 AM Subject: [Asterisk-Users] What is the best IP phone? hi, Can anybody suggest me a good, reliable, robust, SIP supported hardware IP phone? Surajee
Re: [Asterisk-Users] What is the best IP phone?
This is another option that a lot of people have used with Asterisk. http://www.sysconfig.com/ipsietph.html - Original Message - From: Surajee Ratnayake To: [EMAIL PROTECTED] Sent: Saturday, September 06, 2003 7:40 AM Subject: [Asterisk-Users] What is the best IP phone? hi, Can anybody suggest me a good, reliable, robust, SIP supported hardware IP phone? Surajee
Re: [Asterisk-Users] What is the best IP phone?
I am pretty sure that the price on the Pingtel phones recently changed so they are a bit more affordable now. You look like you are reselling. I will email you dealer prices on Monday for the Pingtel and IPDialog phones. Sean - Original Message - From: Surajee Ratnayake [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, September 06, 2003 9:43 AM Subject: Re: [Asterisk-Users] What is the best IP phone? i asked this because i have very bad experience with SNOM phones..(infact SNOM 200) i have given them to some telephone opearators (running asterisk pbx) and they are giving so many problems to me. Thats y i am asking for a good IP, SIP phone which can be used as a operator phone too.. - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, September 06, 2003 9:38 PM Subject: Re: [Asterisk-Users] What is the best IP phone? I guess you didn't look in the archives before you asked the question.. It seems that the same question is being asked evey two or three days.. Common phones used by the Asterisk community are : (In Approximate price order) Grandstream (www.grandstream.com) Snom (www.snom.com) Cisco (www.cisco.com) Pingtel (www.pingtel.com) All work with Asterisk.. Later.. hi, Can anybody suggest me a good, reliable, robust, SIP supported hardware IP phone? Surajee -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PCI master Abort
thanks Tjardick, when i changed the IRQ... the problem is solved. the PCI master message disapeared. Thanks. On Thu, 4 Sep 2003, Tjardick van der Kraan wrote: Hi, I have had the same problem. Turned out to be that my digium card was sharing irq with a different card allready in the machine. If you give your T100P it's own dedicated IP, i bet the problem will be solved. Greetings, Tj - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 04, 2003 12:28 PM Subject: [Asterisk-Users] PCI master Abort i had a T100P working perfectly when on my asterisk box. but to meet our current requirements i added another T100P to set up another channel bank to use with it. after a couple of minutes of operation i get the multiple PCI Master Abort messages on the screen and asterisk freezes. is there any way out this state? please i will know if anyone has a solution to this problem. My email is :- [EMAIL PROTECTED] -- Olaifa Augustine General Data Engineering Services Ltd 18b oshin road,kongi bodija p.o.box 29460, secretariate, ibadan. tel:- 234-2-8105156 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Olaifa Augustine General Data Engineering Services Ltd 18b oshin road,kongi bodija p.o.box 29460, secretariate, ibadan. tel:- 234-2-8105156 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: Creating documentation using a web interface
Hello all, I would like to document some things I am doing with asterisk, but would prefer to do this from a web interface. I am unfamiliar with any software that allows you to create online documentation from a web interface. Ideally I will be able to create documentation online from a browser, which then when saved, is immediately ready to be read online. Perhaps I can setup different authors who are also allowed to create documentation, or have a section where users to the site can create their own documentation and submit it for inclusion. A section to submit documentation edits would be nice, as well as maybe a history timeline or something like that? Just some thoughts. If you know of something like this, please let me know. In the meantime, I'll be googling some more. Thanks, -- Leif Madsen - Telecommunications Technology Sheridan College - Trafalgar Campus ICQ: 3445119 FWD: 18924 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: Creating documentation using a web interface
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Leif Madsen Sent: Saturday, September 06, 2003 10:42 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] OT: Creating documentation using a web interface Hello all, I would like to document some things I am doing with asterisk, but would prefer to do this from a web interface. I am unfamiliar with any software that allows you to create online documentation from a web interface. Ideally I will be able to create documentation online from a browser, which then when saved, is immediately ready to be read online. Perhaps I can setup different authors who are also allowed to create documentation, or have a section where users to the site can create their own documentation and submit it for inclusion. A section to submit documentation edits would be nice, as well as maybe a history timeline or something like that? Just some thoughts. If you know of something like this, please let me know. In the meantime, I'll be googling some more. Hm http://www.interactivetools.com/products/docbuilder/ This looks kind of what I want, but I am looking for a free version of something preferably. I would pay for something, but at this time I am unable to because of commitments to having to pay for tuition (still in school). Getting closer though... back to the google :) Thanks, Leif Madsen. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: Creating documentation using a web interface
Hm http://www.interactivetools.com/products/docbuilder/ This looks kind of what I want, but I am looking for a free version of something preferably. I would pay for something, but at this time I am unable to because of commitments to having to pay for tuition (still in school). Getting closer though. .. back to the google :) Jac Kersing mentioned to me to try twiki. Has anyone else used this with success to create documentation? Installing now as a test platform. Thanks, Leif Madsen. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialplan question
Hi,
Dialplan Question
I'm in holland and I have:
[naarbuiten]
ignorepat = 0
; interlocaal
exten = _00[1-9],1,Dial(Modem/g1:${EXTEN})
exten = _00[1-9],2,Congestion
; locaal
exten = _0[1-9]XX,1,Dial(Modem/g1:${EXTEN})
exten = _0[1-9]XX,2,Congestion
And sometimes I can get out, most of the time however I get a busy
signal halfway throu the number.
It works more often if I change
Early Dial: No
Yes (use
"Yes" only if proxy supports 484 response)
to No. In the Budgetone 100 phone.
regards,
fredrik chabot
---
*CLI show dialplan
[ Context 'default' created by 'pbx_config' ]
Include = 'demo'
[pbx_config]
[ Context 'demo' created by 'pbx_config' ]
'#' = 1. Playback(demo-thanks)
[pbx_config]
2. Hangup()
[pbx_config]
'100' = 1. Dial(SIP/100)
[pbx_config]
'101' = 1. Dial(SIP/101)
[pbx_config]
'190' = 1. Dial(Modem/g1:006400)
[pbx_config]
'8500' = 1. VoicemailMain()
[pbx_config]
2. Goto(s|6)
[pbx_config]
'i' = 1. Playback(invalid)
[pbx_config]
's' = 1. Wait(1)
[pbx_config]
2. Answer()
[pbx_config]
3. DigitTimeout(5)
[pbx_config]
4. ResponseTimeout(10)
[pbx_config]
5. BackGround(demo-congrats)
[pbx_config]
6. BackGround(demo-instruct)
[pbx_config]
't' = 1. Goto(#|1)
[pbx_config]
Include = 'naarbuiten'
[pbx_config]
[ Context 'naarbuiten' created by 'pbx_config' ]
'_00[1-9]' = 1.
Dial(Modem/g1:${EXTEN}) [pbx_config]
2. Congestion()
[pbx_config]
'_0[1-9]XX' = 1. Dial(Modem/g1:${EXTEN})
[pbx_config]
2. Congestion()
[pbx_config]
Ignore pattern = '0'
[pbx_config]
[ Context 'vanbuiten' created by 'pbx_config' ]
's' = 1. Wait(1)
[pbx_config]
2. Answer()
[pbx_config]
3. DigitTimeout(5)
[pbx_config]
4. ResponseTimeout(10)
[pbx_config]
5. Playback(tt-weasels)
[pbx_config]
6. Dial(SIP/100|4)
[pbx_config]
7. Dial(SIP/100SIP/101|10)
[pbx_config]
8.
Dial(SIP/100SIP/101Modem/g1:0064000) [pbx_config]
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[Asterisk-Users] NOTICE[98311]: File chan_sip.c, Line 5070 (handle_request): Registration from 'sip:100@192.168.123.2' failed for '192.168.123.110'
Hello, Is there any way to get rid of this message. NOTICE[98311]: File chan_sip.c, Line 5070 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.123.110' There where some pointer earlier in this list like avoiding dynamic ip's etc. And right after changing that this message was gone for about 2 day's. Its back however. [100] type=friend secret= host=192.168.123.110 username=100 dtmfmode=inband ; Choices are inband, rfc2833, or info mailbox=1234,2345 ; Mailbox for message waiting indicator [101] type=friend secret= username=101 host=192.168.123.106 dtmfmode=inband ; Choices are inband, rfc2833, or info mailbox=1234,2345 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NOTICE[98311]: File chan_sip.c, Line 5070 (handle_request): Registration from 'sip:100@192.168.123.2' failed for '192.168.123.110'
comment out register = user:[EMAIL PROTECTED] from sip.conf Martin On Sat, 6 Sep 2003, fredrik chabot wrote: Hello, Is there any way to get rid of this message. NOTICE[98311]: File chan_sip.c, Line 5070 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.123.110' There where some pointer earlier in this list like avoiding dynamic ip's etc. And right after changing that this message was gone for about 2 day's. Its back however. [100] type=friend secret= host=192.168.123.110 username=100 dtmfmode=inband ; Choices are inband, rfc2833, or info mailbox=1234,2345 ; Mailbox for message waiting indicator [101] type=friend secret= username=101 host=192.168.123.106 dtmfmode=inband ; Choices are inband, rfc2833, or info mailbox=1234,2345 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bug in my head or bug in the code?
What does this step show on the CLI ?
exten = 1,1,SetVar(FOO=123**)
exten = 1,2,SetVar(CHECK=${FOO:-1:1})
? If you're going to see CHECK=* then there is a bug in = operator ...
Martin
On Fri, 5 Sep 2003, John Todd wrote:
I am having Yet Another Regular Expression problem, but this one
might not be my fault, or at least it might not be obviously my
fault. :-)
exten = 2212,1,SetVar(FOO=123456**)
exten = 2212,2,SetVar(BAR=$[${FOO:-1:1} = *])
This script continues with a value of 0 in BAR.
Similarly, none of the following changes made a difference in that
result, which is expected since the * is not listed in
README.variables as a character that must be escaped:
exten = 2212,2,SetVar(BAR=$[${FOO:-1:1} = *])
exten = 2212,2,SetVar(BAR=$[${FOO:-1:1} = \*])
exten = 2212,2,SetVar(BAR=$[${FOO:-1:1} = \*])
I have also tried setting the variable ${BAZ}=* and then using that
in my comparison, with the same unexpected results.
Oddly enough, this almost-identical example below has different, but
normal, results: BAR=1
exten = 2212,1,SetVar(FOO=123456##)
exten = 2212,2,SetVar(BAR=$[${FOO:-1:1} = #])
What gives? Am I colliding with a problem that is the result of the
* character being used in expr evaluations and somehow not being
handled correctly, or am I simply not implementing the syntax
correctly?
JT
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RE: [Asterisk-Users] VONAGE or IP Dialtone
-Original Message- From: Brian Capouch [mailto:[EMAIL PROTECTED] Sent: Saturday, September 06, 2003 12:53 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VONAGE or IP Dialtone Not necessarily preposterous; I would certainly allow that its optimality is arguable. [several very good point deleted] Thank you. Well stated, and you saved me the typing ;) Find me SIP termination with unlimited minutes at a reasonable flat rate to US destinations that works natively with * and I'll dump Vonage tomorrow (and deal with the rest). Seriouslyplease? Daryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NuFone.net Was:VONAGE or IP Dialtone
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Saturday, September 06, 2003 8:39 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VONAGE or IP Dialtone Thanks for the great feedback on these options. I am fairly new at this and not familiar with the IAX/IAX2 capabilities offered by Nufone. Could you expand on this and is Nufone inbound and outbound from the PSTN? Personal and recent NuFon.net experience: The are a great group of people, who resond to email very quickly. You tell them you want termination and a DID for an * box, and they'll get it set up and send you configuration snippets to get it working. They are still working on their billing system, so they just email you invoices, etc. at this point, but they tell me they are going to have a new system done soon where you can check your balance online, etc. At the moment, I believe they have DIDs in Michigan (and 800) only. I am told they are working on agreements for more locations. My vonage # forwarded to the Michigan DID seems to work just fine ;). Daryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ser vs Asterisk?
Could someone give me a 10,000 foot view of what the differences are between Ser and Asterisk? I'd like to implement one or the other handle a small number of local ip phones, tie a couple of asterisk (or ser) machines together across the Internet, implement a couple of FX gateways (to handle incoming pstn calls, and for outgoing pstn calls), and use features mostly common to pbx's. No immediate need for CDR. Voice mail, callerid, etc, are wanted. Would like to accept incoming sip calls from anyone on the Internet that might choose to call. Would Ser or Asterisk be the most appropriate choice? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Forwarding
I am trying to setup call forwarding, I have 2 x100p cards.I want the calls forwarded to a static number. Is this done using meetme? Any help or HOWTOs would be great Get 10MB of e-mail storage! Sign up for Hotmail Extra Storage. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail Indications
Is there anyway in asterisk using the tdmcard, to make the dial tone "sound funny" when you have voicemail. The local bell does this here, and I was thinking that mabye asterisk did it Use custom emotions -- try MSN Messenger 6.0! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ser vs Asterisk?
asterisk would be appropriate choice. I don't think Ser has the ability to interface with the PSTN On Sat, Sep 06, 2003 at 12:03:02PM -0600, Rich Adamson wrote: Could someone give me a 10,000 foot view of what the differences are between Ser and Asterisk? I'd like to implement one or the other handle a small number of local ip phones, tie a couple of asterisk (or ser) machines together across the Internet, implement a couple of FX gateways (to handle incoming pstn calls, and for outgoing pstn calls), and use features mostly common to pbx's. No immediate need for CDR. Voice mail, callerid, etc, are wanted. Would like to accept incoming sip calls from anyone on the Internet that might choose to call. Would Ser or Asterisk be the most appropriate choice? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: Creating documentation using a web interface
here is program I would recommend: www.helpandmanual.com really easy to use, full of the features. Senad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Leif Madsen Sent: 06 September 2003 17:02 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] OT: Creating documentation using a web interface Hm http://www.interactivetools.com/products/docbuilder/ This looks kind of what I want, but I am looking for a free version of something preferably. I would pay for something, but at this time I am unable to because of commitments to having to pay for tuition (still in school). Getting closer though. .. back to the google :) Jac Kersing mentioned to me to try twiki. Has anyone else used this with success to create documentation? Installing now as a test platform. Thanks, Leif Madsen. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MOH other than mp3 ??
is there a clean way to have MOH (Music on Hold) source its audio from say a MS-ASF streem ??? got a radio station that wants to have their MOH come from their ASF based netbroadcast ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Phone - Asterisk - SIP LD Provider question
I'm asking because I have Asterisk running behind a NAT firewall along with an IP Phone (software) and I'm trying to get it working with Iconnecthere (ICH). I am able to register, connect , but no audio. I have ports opened up on the firewall, but they point to the Asterisk machine and not the IP phone machine. In this scenario, any audio traffic would have to go through the asterisk box to reach the IP phone. Is that how it works? I was using a sniffer a few minutes ago to identify an issue between a cisco 7960 and ata186. The call setup occurs between the phones and asterisk on udp 5060 (both source and destination ports), but the actual conversation was directly between the phones (in at least this one example) on udp ports 23570 and 1, with 180 byte data payloads occuring approximately every 20 milliseconds. Another call between XLite and a Snom 200 used udp ports 8000 and 10018 directly between the phones. The above is only intended to point out the NATing issues associated with using voip through a firewall. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VONAGE or IP Dialtone
-Original Message- From: Brian Capouch [mailto:[EMAIL PROTECTED] Sent: Saturday, September 06, 2003 12:53 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VONAGE or IP Dialtone Not necessarily preposterous; I would certainly allow that its optimality is arguable. [several very good point deleted] Thank you. Well stated, and you saved me the typing ;) Find me SIP termination with unlimited minutes at a reasonable flat rate to US destinations that works natively with * and I'll dump Vonage tomorrow (and deal with the rest). Seriouslyplease? Daryl (Let me preface this: I applaud Vonage for their product; it's opening up VoIP in the residential market, and that's great news for all of us. I think they've done a great marketing job, and they will undoubtedly be successful in the residential space.) Daryl - You will probably never find a flat-rate provider that offers that service without major handcuffs. My comments about Vonage being preposterous are due to not only their restrictions on giving out a userid/password that have been paid for; let me elaborate: Most of my customers are business customers. Even those people who are not business customers (friends, family, etc.) are tele-commuters sporadically. Looking at Vonage's terms of service, any type of business use is prohibited, which eliminates them as a possible candidate for almost 100% of the people who have asked me to recommend a VoIP provider. Preposterous is an adequate word to describe their limitations and contractual wording. I suspect Vonage will eliminate their flat rate pricing sometime in the near future, but that is purely speculative. They're harvesting the most abusive customers as their base, and that will come back around to bite them. Queries to Vonage about giving a business rate (minute-based at some low price) have been like talking to a brick wall. If they changed some of their contract wording, allowed me to use their service with Asterisk, and opened up a per-minute plan for business users, I'd be interested. But that's not their market, and that's fine - they know where the money is, so they should not try to satisfy markets in which they don't feel there is significant return. Let me give a brief overview of why I won't recommend Vonage to even residential users: From Vonage's terms of service: Firstly, why businesses shouldn't use it: 1.2 Residential Use of Service and Device The Service and Device are provided to you as a residential user, for your personal, residential, non-business and non-professional use. This means that you are not using them for any commercial or governmental activities, profit-making or non-profit, including but not limited to home office, business, sales, tele-commuting, tele-marketing, continuous autodialing, fax broadcast, fax blasting or any other activity that would be inconsistent with normal residential usage patterns. This also means that you are not to resell or transfer the Service or the Device to any other person for any purpose, or make any charge for the use of the Service, without express written permission from Vonage in advance. You agree that your use of the Service and/or Device, or the use of the Service and/or Device provided to you by any other person for any commercial or governmental purpose will obligate you to pay Vonage's higher rates for commercial service on account of all periods, including past periods, in which you use, or used, the Service for commercial or governmental purposes. Vonage reserves the right to immediately terminate or modify the Service, if Vonage determines, in its sole discretion, that Customer's Service is being used for non-residential or commercial use. Secondly, why people who value their rights shouldn't use it: 1.3.1 Prohibited Uses: You agree to use the Service and Device only for lawful purposes. This means that you agree not to use them for transmitting or receiving any illegal, harmful, threatening, abusive, harassing, defamatory, obscene, sexually explicit, profane, racially or ethnically disparaging remarks or otherwise objectionable material of any kind, including but not limited to any material that encourages conduct that would constitute a criminal offense, give rise to a civil liability, or otherwise violate any applicable local, state, national or international law. Vonage reserves the right to terminate your service immediately and without advance notice if Vonage, in its sole discretion, believes that you have violated the above restrictions, leaving you responsible for the full month's charges to the end of the current term, including without limitation unbilled charges, plus a disconnect fee, all of which immediately become due and payable. You are liable for any and all use of the Service and/or Device by any person making use of the Service or Device provided to you. If Vonage, in its sole discretion believes that
[Asterisk-Users] Limiting the number of SIP/IAX lines
Is it possible to limit the number of lines provided by a given SIP/IAX connection? For example: I want to limit SIP extensions to only a single incoming line, even the phone itself can handle three. Or, I might want to prevent extensions from making more than one outgoing call at a time. Or, I might want to protect my bandwidth/call quality by limiting outgoing calls through NuFone to only three calls at a time. Any thoughts? Thanks, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limiting the number of SIP/IAX lines
Is it possible to limit the number of lines provided by a given SIP/IAX connection? For example: I want to limit SIP extensions to only a single incoming line, even the phone itself can handle three. Or, I might want to prevent extensions from making more than one outgoing call at a time. Or, I might want to protect my bandwidth/call quality by limiting outgoing calls through NuFone to only three calls at a time. Any thoughts? Thanks, --Ernest There is a patch already in the bugtracker for outgoing SIP: http://bugs.digium.com/bug_view_page.php?bug_id=098 I imagine that it would be a good idea to also implement the same thing for IAX. Submit a feature request (and hopefully a patch, too!) via the bugtracker, and maybe it will get implemented. :) You can already handle this, sort of, by using incrementing variables stored in the local database (DBPut, DBGet) or by using an AGI script to keep a counter on each outgoing peer. That's not a particularly clean way of doing it, but it certainly would work. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] google search of asterisk archives?
Prefix your search with site:lists.digium.com So if you wanted to search the list archives for SIP you would enter site:lists.digium.com sip -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Saturday, September 06, 2003 4:46 PM To: Asterisk-users-list Subject: [Asterisk-Users] google search of asterisk archives? A while back someone posted a method for using google to search the asterisk archives. Could someone repeat that method please? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] google search of asterisk archives?
A while back someone posted a method for using google to search the asterisk archives. Could someone repeat that method please? http://www.google.com/custom?sitesearch=lists.digium.com JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NOTICE[98311]: File chan_sip.c, Line 5070 (handle_request): Registration from 'sip:100@192.168.123.2' failed for '192.168.123.110'
Martin Pycko wrote: comment out register = user:[EMAIL PROTECTED] I have no register statement in my sip.conf. Any other suggestions are very welcome. regards fredrik. BTW: My conpleet sip.conf -- [EMAIL PROTECTED] asterisk]# grep -v "^;" sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls [100] type=friend secret= host=192.168.123.110 username=100 dtmfmode=inband ; Choices are inband, rfc2833, or info mailbox=1234,2345 ; Mailbox for message waiting indicator [101] type=friend secret= username=101 host=192.168.123.106 dtmfmode=inband ; Choices are inband, rfc2833, or info mailbox=1234,2345 ; Mailbox for message waiting indicator [EMAIL PROTECTED] asterisk]# -- from sip.conf Martin On Sat, 6 Sep 2003, fredrik chabot wrote: Hello, Is there any way to get rid of this message. NOTICE[98311]: File chan_sip.c, Line 5070 (handle_request): Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.123.110' There where some pointer earlier in this list like avoiding dynamic ip's etc. And right after changing that this message was gone for about 2 day's. Its back however. [100] type=friend secret= host=192.168.123.110 username=100 dtmfmode=inband ; Choices are inband, rfc2833, or info mailbox=1234,2345 ; Mailbox for message waiting indicator [101] type=friend secret= username=101 host=192.168.123.106 dtmfmode=inband ; Choices are inband, rfc2833, or info mailbox=1234,2345 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ser vs Asterisk?
Rich Adamson wrote: Could someone give me a 10,000 foot view of what the differences are between Ser and Asterisk? Asterisk is a PBX that you can use to connect SIP clients to the PSTN or voicemail /IVR applications. SER is a SIP proxy that connects SIP clients to each other. Asterisk handles all media streams through the asterisk server, SER handles no media streams at all - it's up to the SIP clients to set up the media between themselves. So in an installation with many users, my preferred choice would be to have both - they're solving different problems. My 10 öre :-) /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: Creating documentation using a web interface
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Leif Madsen Sent: Saturday, September 06, 2003 9:42 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] OT: Creating documentation using a web interface Hello all, I would like to document some things I am doing with asterisk, but would prefer to do this from a web interface. I am unfamiliar with any software that allows you to create online documentation from a web interface. Ideally I will be able to create documentation online from a browser, which then when saved, is immediately ready to be read online. Perhaps I can setup different authors who are also allowed to create documentation, or have a section where users to the site can create their own documentation and submit it for inclusion. A section to submit documentation edits would be nice, as well as maybe a history timeline or something like that? Just some thoughts. If you know of something like this, please let me know. In the meantime, I'll be googling some more. Thanks, -- Leif Madsen - Telecommunications Technology Sheridan College - Trafalgar Campus ICQ: 3445119 FWD: 18924 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Check out Zope. It can be really complex or fairly simple. http://www.zope.org Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ser vs Asterisk?
Could someone give me a 10,000 foot view of what the differences are between Ser and Asterisk? Asterisk is a PBX that you can use to connect SIP clients to the PSTN or voicemail /IVR applications. SER is a SIP proxy that connects SIP clients to each other. Asterisk handles all media streams through the asterisk server, SER handles no media streams at all - it's up to the SIP clients to set up the media between themselves. So in an installation with many users, my preferred choice would be to have both - they're solving different problems. Can they both exist on the same system and communicate with some degree of stability? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] digium dev kit - X100P TDM400P
Hello. Well I finally rx'd my dev kit (new batch of TDM's apparently. I'm on Mandrake 9.1 There were no hardware install instructions, it would have been nice to know whether the 4-way power connector was to be used or was for some other future or expansion purpose. It came with a floppy disc, no label and it wasn't even write protected. The only readme file was 'README.DevKitLite' and sure enough, it was an explanation how to install the dev lite kit, which I DON't have. I tried it anyway ignoring the references to S100U. There is a astinstaller, that failed complaining of no openssl-devel. Astericks has been running for the past week with no obvious problems or alerts. I am informed that mandrake has renamed it to libopenssl-blah-blah, so I look at the astinstaller entry and change it. Still fails so I remove that check. It now builds and completes. It complains [EMAIL PROTECTED] etc]# ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. ZT_CHANCONFIG failed on channel 2: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? Another configuration error with the software (or being generous, perhaps it's because it's the wrong software for the kit). [EMAIL PROTECTED] asterisk]# asterisk -vvvr ERROR[16384]: File asterisk.c, Line 1323 (main): Unable to connect to remote asterisk Hmm...that wasn't happening before I did the build using the supplied disc. Well, if anyone has the correct install disc, can they copy and zip/tgz it for me. Otherwise, anyone have the X100P TDM400P cards and point me in the right direction. The digium FAQ :- http://www.digium.com/index.php?menu=faq#Configuration_1 didn't fix the problem either. I removed /etc/asterisk etc. and have started fresh a number of times with no success. Hints appreciated. -- Benson's Dogma: ASCII is our god, and Unix is his profit. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MOH other than mp3 ??
You didn't just say MS-ASF MP3's good.. bkw On Sat, 6 Sep 2003, John Brown wrote: is there a clean way to have MOH (Music on Hold) source its audio from say a MS-ASF streem ??? got a radio station that wants to have their MOH come from their ASF based netbroadcast ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MOH other than mp3 ??
On Sat, Sep 06, 2003 at 04:56:22PM -0500, Brian West wrote: You didn't just say MS-ASF Yup I did, ducking under the table... Customer requirement MP3's good.. They rock, but the customer is doing something different and wants to insert ASF formated tunes. bkw On Sat, 6 Sep 2003, John Brown wrote: is there a clean way to have MOH (Music on Hold) source its audio from say a MS-ASF streem ??? got a radio station that wants to have their MOH come from their ASF based netbroadcast ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] digium dev kit - X100P TDM400P
On Saturday 06 September 2003 16:54, marrandy wrote: Hello. Well I finally rx'd my dev kit (new batch of TDM's apparently. I'm on Mandrake 9.1 There were no hardware install instructions, it would have been nice to know whether the 4-way power connector was to be used or was for some other future or expansion purpose. Yes, it's to be used. It is there because of the power limitations of the PCI bus. It complains [EMAIL PROTECTED] etc]# ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. ZT_CHANCONFIG failed on channel 2: Invalid argument (22) Did you modprobe wcfxo or wcfxs first? If you did modprobe wcfxo first, then your channel definitions are backwards. It's also possible that you forgot to modprobe one or the other, in which case the second card will certainly not be detected. See the output of 'lsmod' to see if either is not loaded. Also possible is that your lack of connecting the power connector may make your computer unable to detect the TDM400. [EMAIL PROTECTED] asterisk]# asterisk -vvvr ERROR[16384]: File asterisk.c, Line 1323 (main): Unable to connect to remote asterisk Hmm...that wasn't happening before I did the build using the supplied disc. You can only remote connect (-r) if an instance of Asterisk is already running. Drop the -r if not (and possibly add -c, for console). -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MOH other than mp3 ??
You didn't just say MS-ASF Yup I did, ducking under the table... Customer requirement Gotta love those customers eh? Life'd be so much easier without 'em sometimes ;-) They rock, but the customer is doing something different and wants to insert ASF formated tunes. Is there no way to take raw audio in, like on the Line In of their soundcard or with a bridging device on a Zap channel? Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] digium dev kit - X100P TDM400P
On Saturday 06 September 2003 06:19 pm, Tilghman Lesher wrote: It complains [EMAIL PROTECTED] etc]# ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. ZT_CHANCONFIG failed on channel 2: Invalid argument (22) Did you modprobe wcfxo or wcfxs first? As per the only directions digium sent me, modprobe wcfxo was first If you did modprobe wcfxo first, then your channel definitions are backwards. It's also possible that you forgot to modprobe one or the other, in which case the second card will certainly not be detected. No, modprobe wcfxs was after the modprobe wcfxo as per digiums supplied instructions See the output of 'lsmod' to see if either is not loaded. wcfxo 9440 0 (unused) wcfxs 19168 0 (unused) zaptel182880 0 [wcfxo wcfxs] Also possible is that your lack of connecting the power connector may make your computer unable to detect the TDM400. That's connected [EMAIL PROTECTED] asterisk]# asterisk -vvvr ERROR[16384]: File asterisk.c, Line 1323 (main): Unable to connect to remote asterisk Hmm...that wasn't happening before I did the build using the supplied disc. You can only remote connect (-r) if an instance of Asterisk is already running. Drop the -r if not (and possibly add -c, for console). -Tilghman Do have the correct install disk ? -- Power is the finest token of affection. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MOH other than mp3 ??
Just tell em its ASF.. like the would know the diffrence. bkw On Sat, 6 Sep 2003, John Brown wrote: On Sat, Sep 06, 2003 at 04:56:22PM -0500, Brian West wrote: You didn't just say MS-ASF Yup I did, ducking under the table... Customer requirement MP3's good.. They rock, but the customer is doing something different and wants to insert ASF formated tunes. bkw On Sat, 6 Sep 2003, John Brown wrote: is there a clean way to have MOH (Music on Hold) source its audio from say a MS-ASF streem ??? got a radio station that wants to have their MOH come from their ASF based netbroadcast ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MOH other than mp3 ??
On Sat, Sep 06, 2003 at 03:32:00PM -0700, Paul Crick wrote: They rock, but the customer is doing something different and wants to insert ASF formated tunes. Is there no way to take raw audio in, like on the Line In of their soundcard or with a bridging device on a Zap channel? They have 5 different stations under one company. The idea is to have 5 different MOH sources (URL based) and depending on who/what/when/where placed on hold they would get the right stations current broadcast. So I don't think the cost/value is there for multiple Zap-Channel devices, or multiple sound cards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MOH other than mp3 ??
On Sat, Sep 06, 2003 at 05:40:28PM -0500, Brian West wrote: Just tell em its ASF.. like the would know the diffrence. The system they use to interface with the web is a pre-made system for the station and we can't touch it. Output is ASF and we can't make it something else. bkw On Sat, 6 Sep 2003, John Brown wrote: On Sat, Sep 06, 2003 at 04:56:22PM -0500, Brian West wrote: You didn't just say MS-ASF Yup I did, ducking under the table... Customer requirement MP3's good.. They rock, but the customer is doing something different and wants to insert ASF formated tunes. bkw On Sat, 6 Sep 2003, John Brown wrote: is there a clean way to have MOH (Music on Hold) source its audio from say a MS-ASF streem ??? got a radio station that wants to have their MOH come from their ASF based netbroadcast ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fwd: Asterisk BoF: Boston, Sept _2[2-4] - interested?
Hello all - Second-to-last notice to collect RSVP's for interested parties for the Asterisk Birds of a Feather dinner/beer meeting. There are five people signed up for this at the moment. Looks like it's going to be the 23rd of September, since that fits well with the VON conference schedule. Please email me directly for details. JT Date: Thu, 21 Aug 2003 01:52:52 -0700 To: asterisk-users-lists.digium.com From: John Todd [EMAIL PROTECTED] Subject: Asterisk BoF: Boston, Sept _2[2-4] - interested? There was some talk on the IRC channel about getting a convention/conference together for Asterisk users. While I do not have the authority (or time) to make such a proposition, I'd like to see if I can gather some support for a BOF (Birds Of a Feather) meeting on a somewhat smaller scale and in a shorter timeframe. The VON (Voice On the Net - http://www.pulver.com/von/) show is in Boston, from September 22nd through the 25th. This seems like a place where perhaps some Asterisk-inclined people might be congregating, and it is also fairly reasonably close to large urban population centers that maybe there will be some indigenous Asterisk users that might be able to stop by for an evening at a pub somewhere. I will take it upon myself to find a pub or other meeting place that is suitable for a group of tech people if there is suitable interest. Maybe some bar around MIT, but probably something close to the convention center (Back Bay) area. Please send me a note off-list if you would be interested in attending, and if there are any dates between the evening of the 22nd and the evening of the 24th that are unworkable (the 25th is probably out of bounds since people will be flying home starting at the end of the show in the afternoon.) JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MP3 streams for MOH: idea
[thread change, different topic] is there a clean way to have MOH (Music on Hold) source its audio from say a MS-ASF streem ??? got a radio station that wants to have their MOH come from their ASF based netbroadcast I have run out of time for fooling around today, but I started to think about this one but have no time (or ability) to build this little glue program. Perhaps something that uses named pipes would allow this to happen. Do you know of any tools that convert ASF to mp3? How about a little tiny program that connects to a remote host, grabs the contents of an MP3 stream, and pushes it into a FIFO locally? It would be a raw TCP-to-FIFO stream, so mpg123 would be able to digest it as if it was a local file. The program would take two arguments: remote hostname/IP and port, and then the file to which the output would be sent. I don't know how mpg123 handles blocking... Let's say you have a streaming music source in .mp3 format, on 10.0.3.1:8000 mkfifo /var/lib/asterisk/stream1.mp3 ./backtoback 10.0.3.1:8000 /var/lib/asterisk/stream1.mp3 Then... ; musiconhold.conf default = quietmp3:/var/lib/asterisk/mohmp3 JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NOTICE[98311]: File chan_sip.c, Line 5070 (handle_request): Registration from 'sip:100@192.168.123.2' failed for '192.168.123.110'
and set host=dynamic for that same user. On Saturday 06 September 2003 19:54, Andres wrote: remove the secret line for user 100 in you sip.conf.It should now register without problems. Then figure out what the password issue is. On Saturday 06 September 2003 11:28, fredrik chabot wrote: !DOCTYPE html PUBLIC -//W3C//DTD HTML 4.01 Transitional//EN html head meta http-equiv=Content-Type content=text/html;charset=windows-1252 title/title /head body text=#00 bgcolor=#ff font size=2font face=ArialHello,br br Is there any way to get rid of this message.br br NOTICE[98311]: File chan_sip.c, Line 5070 (handle_request): Registration from 'a class=moz-txt-link-rfc2396E href=mailto:sip:[EMAIL PROTECTED]lt;sip:[EMAIL PROTECTED]gt;/a' failed for '192.168.123.110'br /font/fontbr There where some pointer earlier in this list like avoiding dynamic ip's etc. And right after changing that this message was gone for about 2 day's. Its back however.br br [100]br type=friendbr secret=br host=192.168.123.110br username=100br dtmfmode=inband ; Choices are inband, rfc2833, or infobr mailbox=1234,2345 ; Mailbox for message waiting indicatorbr br [101]br type=friendbr secret=br username=101br host=192.168.123.106br dtmfmode=inband ; Choices are inband, rfc2833, or infobr mailbox=1234,2345 br /body /html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco IP Phone 7905G
Andrew Gillham wrote: [EMAIL PROTECTED] wrote: Has anyone had any success using a Cisco 7905G phone with Asterisk? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Mine works terrific. My 7940 (non-G) phones work nicely as well. What kind of issues are you having? -Andrew Ok, I was confused here, I read '7960G' not 7905G. Sorry about that. :) -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MP3 streams for MOH: idea
[thread change, different topic] is How about a little tiny program that connects to a remote host, grabs the contents of an MP3 stream, and pushes it into a FIFO locally? It would be a raw TCP-to-FIFO stream, so mpg123 would be able to digest it as if it was a local file. The program would take two arguments: remote hostname/IP and port, and then the file to which the output would be sent. I don't know how mpg123 handles blocking... Is there any particular reason (rather than not having time to code one and embed it into *) why we can't have our own in-thread connection to an MP3 stream or file, rather than spawning off a process (fork() is expensive as compared to pthread_create()) of mpg123 to play the stream/file? It seems that this spawning/hoping the process dies cleanly is a thorn in a few people's side. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZAPATA_R2
Hi All, I´m trying to compile * for R2 but seems that the source code tree doesn´t have the libmfcr2.h file (needed by chan_zap.c) Any clue of where can I find it? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MP3 streams for MOH: idea
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of John Todd Sent: Saturday, September 06, 2003 8:57 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] MP3 streams for MOH: idea [thread change, different topic] is there a clean way to have MOH (Music on Hold) source its audio from say a MS-ASF streem ??? got a radio station that wants to have their MOH come from their ASF based netbroadcast I have run out of time for fooling around today, but I started to think about this one but have no time (or ability) to build this little glue program. Perhaps something that uses named pipes would allow this to happen. Do you know of any tools that convert ASF to mp3? How about a little tiny program that connects to a remote host, grabs the contents of an MP3 stream, and pushes it into a FIFO locally? It would be a raw TCP-to-FIFO stream, so mpg123 would be able to digest it as if it was a local file. The program would take two arguments: remote hostname/IP and port, and then the file to which the output would be sent. I don't know how mpg123 handles blocking... Let's say you have a streaming music source in .mp3 format, on 10.0.3.1:8000 mkfifo /var/lib/asterisk/stream1.mp3 ./backtoback 10.0.3.1:8000 /var/lib/asterisk/stream1.mp3 Then... ; musiconhold.conf default = quietmp3:/var/lib/asterisk/mohmp3 You are thinking along the lines of using a shoutcast radio stream? I was thinking about trying something like that the other day, haven't had the time to sit down and implement it yet... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GrandStream Phones... White,Black or Green?
Just in case you guys haven't been paying attention Grandstream sliped in some diffrent colors on the IP phones and looks like they released the ATA-286 (Cisco is gonna have kittens I suspect) bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie IVR question
Steven Critchfield wrote: On Tue, 2003-09-02 at 10:00, [EMAIL PROTECTED] wrote: php is not just a web scripting language anymore. it has been used in other ways for quite a while now. it works nicely from the command line, can be used with ncurses and with gtk. there are several well-known respectable large projects out there built upon php. i usually find that php's biggest critics are those who know the least about the language. however that holds true with pretty much any technology. linux suffers from the same type of critics. Just to point out, I am a php developer. I actually am employed to create and maintain a large webapp in php. I like the fact that I can take my php or perl scripts and not have to change much to them to work in the other language. Well if they are simple enough. There is enough well known documented problems with php. Such as? Just saying that because it is used in large projects doesn't change whether it is suited to the task. There are enough people on this planet, that statistically you will find enough people who refuse to admit the are using a square peg for the round hole. If we go back to PERL's roots, we find that it was never intended as a general, all-purpose language, but one for extracting and formatting data. Now it seems as though it's being touted as the cure-all for *anything* that requires scripting. PHP's intent, on the other hand was a bit more sophisticated. Being a web-based scripting languange, it, by necessity, had to interface with other components (and do it efficiently) in order to acquire, manipulate, and pass data between the user and any backend processes. I'm more curious to know what exactly it is about AGI scripting that would make PHP an inappropriate choice. [snip] Regards, Tom ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
