Re: [Asterisk-Users] IBM to Run VoIP On Linux
Asterisk has got to be about the best kept secret in telephony. I've seen numerous articles on slashdot about VoIP, even in relation to Linux and only *once* has the post even mentioned Asterisk. Am I missing something, or is Asterisk clearly a good potential player in any kind of linux-based soft-switch idea? Mark On Sat, 8 Nov 2003, Dave Cotton wrote: For those who don't wake up at 5.00 am and start reading /. http://searchnetworking.techtarget.com/originalContent/0,289142,sid7_gci935769,00.html -- Dave Cotton [EMAIL PROTECTED] ___ 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] IBM to Run VoIP On Linux
Yes, it is a well-kept secret, which is a shame since it obviously fits so many different requirements. Here are some late-night musings as to why new users coming to Asterisk is only a stream when it should be a river: 1) No 1.0 release. In fact, no release structure at all really. (Hold your flames: I know this is to be remedied soon, along with backtrack patches for security/stability.) 2) No books (yet.) This also is going to be remedied soon. 3) Advocates fall (generally) into two camps: a) IT staff who have much more on their minds than being VoIP advocates, and who normally are told what to do. Even if they have experience with * in testbed situations, the larger vendors come in and throw whitepapers/jargon/FUD at executive staff, who make telephony decisions, thus overruling clueful staff. b) CLEC or other telephony-oriented people who will try very hard to prevent anyone from knowing what they use, or how they use it, since that is a competitive disadvantage if others should start to use the same software-driven architectures. There are some obvious exceptions to this, but you'll very rarely see (ever?) any posts by the two or three major IPCSP's that use Asterisk as part of their core systems. There are of course others who do not fall into one of these two camps, and those are the people being the zealots getting conversions to Asterisk. Personally, as an example, I have over two dozen institutions, companies, and very clueful individuals that I've introduced to Asterisk simply based on chatting with them. (excluding clients, who already have intentions on installing Asterisk.) The time it takes to explain why Asterisk is so useful is quite labor-intensive, actually, and the educational process takes some time even with the most clueful engineering types, simply because there are so MANY things to take into consideration with Asterisk and any telephony questions in general. 4) Hardware vendors are still blowing enough QOS issues around that it obscures open-source VoIP solutions. VoIP won't work is still a claim I hear EVERY DAY, until I disagree and tell that person that I'm disagreeing with them over a VoIP call that crosses a continent twice, across the public Internet (and three carriers.) This is obviously not Asterisk-specific, but it's certainly an issue that scares people away from OSS solutions that don't include magic hardware. 5) I would say that it's becoming less of a secret, so don't give up hope. The almost-unmanageable flood of newbie posts to the Asterisk lists in the last two months or so is evidence that success is sometimes more of a headache than one would want. In short, nothing in the above 4 worry items scares me, and Asterisk is and will become the telephony platform of choice for a large percentage of conversions to VoIP in the coming years. Fret not: you'll be the apache of VoIP soon enough. JT Asterisk has got to be about the best kept secret in telephony. I've seen numerous articles on slashdot about VoIP, even in relation to Linux and only *once* has the post even mentioned Asterisk. Am I missing something, or is Asterisk clearly a good potential player in any kind of linux-based soft-switch idea? Mark On Sat, 8 Nov 2003, Dave Cotton wrote: For those who don't wake up at 5.00 am and start reading /. http://searchnetworking.techtarget.com/originalContent/0,289142,sid7_gci935769,00.html -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM to Run VoIP On Linux
I'm new to Asterisk and I completely agree with Mark. Asterisks is the best kept secret in telephony. I cannot recall how I originally found Asterisk, but I do remember spending far too much time surfing before I did. One place I found seems to mention nearly all products, except Asterisk. http://www.voiptimes.com/research/products/ivr_systems/ If I'm out of place in the following suggestions, I'm sure others will tell me grin - Create a clean SDK of the wonderful IAX2 protocol for Win32 and Mac to gain exposure everywhere - Push, entice, bribe IP phone designers to support the IAX2 protocol based on the clean and easy to use SDK - Someone once suggested an Asterisk logo program, excellent idea My comments are meant in good faith. The effort done here is insanely great!! It would be a shame to watch this gem get passed over. I'm sure there is a number of ways Digium can reap the rewards they deserve for Asterisk. I'd be willing to assist on an IAX2 sdk for Win32. I think there was a thread about Java support. Contraversial, but if developers are building applications in Java, VB, DotNet, then supporting those environment certainly couldn't hurt the exposure of Asterisk. Like I said earlier, I'm new to Asterisk, so I don't know the history developed here. Hopefully I didn't offend anyone. Maybe we'll see Mark's mug shot on the cover of Wired next year g. Cheers, Darren - Original Message - From: Mark Spencer [EMAIL PROTECTED] To: Asterisk List [EMAIL PROTECTED] Sent: Friday, November 07, 2003 10:59 PM Subject: Re: [Asterisk-Users] IBM to Run VoIP On Linux Asterisk has got to be about the best kept secret in telephony. I've seen numerous articles on slashdot about VoIP, even in relation to Linux and only *once* has the post even mentioned Asterisk. Am I missing something, or is Asterisk clearly a good potential player in any kind of linux-based soft-switch idea? Mark On Sat, 8 Nov 2003, Dave Cotton wrote: For those who don't wake up at 5.00 am and start reading /. http://searchnetworking.techtarget.com/originalContent/0,289142,sid7_gci935769,00.html -- Dave Cotton [EMAIL PROTECTED] ___ 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] IBM to Run VoIP On Linux
Asterisk would need scalability and redundancy on the voip side to play in the soft-switch area. The biggest issue stopping Asterisk having redundancy and scalability using sip is the inability to work with just about any sip device without canreinvite turn off. If Asterisk could handled reinvites correctly you could setup fallback and/or redundant gateways to the PSTN network. Making it a shoe in for large installs. As it is Asterisk just can not scale from a Voip perspective. SER would need to have some kind of PSTN trans-coding. But it can scale! Vocal has the redundancy and scalability, but no real PSTN trans-coding. also Vocal also has serious quality control issues. So of the big three free, yeah Asterisk would be a good place to start. Although Vocal on paper is a little better though out. Asterisk has a lot more working features. But I would bet money IBM uses none of the above. smile Mark Spencer wrote: Asterisk has got to be about the best kept secret in telephony. I've seen numerous articles on slashdot about VoIP, even in relation to Linux and only *once* has the post even mentioned Asterisk. Am I missing something, or is Asterisk clearly a good potential player in any kind of linux-based soft-switch idea? Mark On Sat, 8 Nov 2003, Dave Cotton wrote: For those who don't wake up at 5.00 am and start reading /. http://searchnetworking.techtarget.com/originalContent/0,289142,sid7_gci935769,00.html -- Dave Cotton [EMAIL PROTECTED] ___ 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] IBM to Run VoIP On Linux
Besides you got list four times since May!smile http://slashdot.org/search.pl?query=asterisk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM to Run VoIP On Linux
Can I add to this and say that another thing that could be hindering the takup is Single System VoIP scalability and a certain amout of Enterprise flexibility.. Let me explain those two.. Before you start reading these and thinking This guy is mad!! let me just say that I love Asterisk and use it every day, but if M$ and IBM are getting into the game there is cause for concern.. The features I am going to talk about are very much Future Dreams becasue to impliment them would probably mean re-creating the entire code base from the groud up so I don't expest to see the features ant time soon.. I do think that these sorts of featured will be in the IBM and M$ IP PBX's and that is why I think Asterisk needs them.. So lets get started.. I know that many Asterisk servers can be connected together to scale the size of the system but this is still a problem because it is a headache to manage.. What is needed to get the big enterprise players on board is the ability to manage the PBX as a single entity no matter how many servers there are.. servers should simply be add on modules to the overall PBX to improve its VoIP call volume handling power.. I think the only way to achive this would be to make Asterisk a clustered software that sits a level above the servers.. The VoIP phones will see one Asterisk Server that listens on a single IP address per subnet on the network but behind that single system image could be one, two or fifty servers providing the processing power for all the calls, and as power is needed you simply have to add servers.. If you need more PRI lines just add a Digium card to a server and enable that server as a gateway node in the cluster.. With in this model the voicepath between the servers in the cluster needs to be dynamic so the shortest path is always used (IAX can probably handle this quite well already), and CDR must be accurate maybe one or two of the nodes needs to allocated the task of being the CDR server and all other servers will feed back to the central server with the call logging information.. In Enterprise flexibility I am taking about user and phone management and services.. On the phone management side (and I know many don't seem to like the idea) but a platform independent full featured management interface is needed.. If its done in Java or web based running on the Asterrisk sever itself, similar to how webmin has its own web server, does not matter but we live in a world now where admins like GUI management tools.. Leading on from that is an Operator Interface for receptionists and phone operators to be able to manage calls.. See which lines are busy, connect calls and the various other things that these interfaces do.. Next a monitoring interface (somthing similar MRTG would probably do it..) showing server loads and statistics so system management and upgrading is easy to see and plan for.. Then the need to support hot desking.. By this I mean that the phone and the user need to be seperate entities on the system.. then the user can sit down at any phone on any desk run through a login procedure (either on the phone or in some easily accesible interface) and all their calls will then be routed to that phone.. I know there are hacks and work arounds to getting this kind of functionality using queues and the Asterisk DB and various other options but it needs to be a standard working system.. Finally an automatic provisioning system.. New user joins the company, click a button on the management interface and give them their extension number and extension password.. no editing files and restarting servers or anything like that its all done behind the scene.. So did I just thumb suck these concepst out of this air?? not totally.. Last year I did a contract at a large comnpany in London and was working on a user provisioning system.. This company has thousands of users in a single building (and a single PBX) in London, and thousands more accross the country.. It was a provisioning system so I needed to talk to the telecoms guys to see if we could automatically provision the phone extensions from the central application.. So a lot of my ideas here come from what I saw they had and things they said they would like to have.. Anyway I will stop rambling on now.. I still think Asterisk is great for SOHO and medium businesses, and when the Digium multiport analog or a BRI card (I know ISDN cards can be used but it would be nice to have one that provided Zaptel timing and one that would probably be a lot cheaper than the current active ISDN options.) comes out it will be great for the small companies as well.. Later.. John Todd wrote: Yes, it is a well-kept secret, which is a shame since it obviously fits so many different requirements. Here are some late-night musings as to why new users coming to Asterisk is only a stream when it should be a river: 1) No 1.0 release. In fact, no release structure at
RE: [Asterisk-Users] Asterisk over VPN.
You need to add nat=yes for the sip phones in sip.conf, IMHO From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Arnt Sent: Saturday, November 08, 2003 1:30 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk over VPN. Hi People, Let's take a look in this diagram : Part A - Server running VPN IP ie.192.168.10.1 Part B - Client running over the VPN with internal IP ie. 192.168.10.2 -- From network A i can reach B. Use all programs - Share Printers , aplications, using Netmeeting etc.. Then i make this in the same server of the VPN i put Asterisk PBX. (Network A) Running SIP in the same network (the network below the server, all machines can login etc perfeclty and talk with each others). In the Network B . All machines can't connect to Asterisk ... Just if i point all to the External Address of the VPN Server that has asterisk... In the log i can only see the registrations using the external address of the client (VPN) not the Internal one . Ie. Using 200.300.200.100 not 192.168.10.2 Well i'm using X-lite to talk and works great . I make the same test using two machines one server with Asterisk and the other just a Windows CLient. If i point the Windows client to take the external address he login very well. If i try over the Internal address i can ... My question is , in the VPN Rules all TCP and UDP ports are open. I can even share printers and files etc in both machines, why i cant then use Asterisk to talk with my computer in this case inside the VPN ??? Thanks alot for helping . Carlos . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Streaming MOH
Hi, Thanks for info, Didn't know the mails were sent as HTML, will check the email settings. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Eric Wieling Sent: 08 November 2003 02:03 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Streaming MOH It takes 5 seconds or less to log into that site as an anonymous user. Just for the record I generally delete HTML mail on a mailing list without reading it. On Fri, 2003-11-07 at 19:50, Michael Koehler wrote: 1. can someone please quote the text from this restricted page which is linked below to the list. could be helpful for some. 2. just for the stats, i prefer html John Todd wrote: Hi All, I keep asking things as they come into my head. Is there any way to grab an audio stream and pipe it out as the MOH? I am a helper at a local Charity Hospital Radio Station and thought it would be nice to pipe the studio output to waiting callers. Dave Dave - 1) Please don't post HTML to the list. Some people appreciate the formats less than you might think. 2) http://bugs.digium.com/bug_view_page.php?bug_id=413 JT ___ 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 -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ 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] Putting call on hold
Is there a way to put a call on hold and play music on hold with out using the park app? Yes there is. ;-) Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Streaming MOH
Hi! Is there any way to grab an audio stream and pipe it out as the MOH? I am a helper at a local Charity Hospital Radio Station and thought it would be nice to pipe the studio output to waiting callers. Look here: http://bugs.digium.com/bug_view_page.php?bug_id=413 Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM to Run VoIP On Linux
Hi! 1) No 1.0 release. In fact, no release structure at all really. (Hold your flames: I know this is to be remedied soon, along with backtrack patches for security/stability.) With that comes a changelog and some basic documentation. I still find it amazing that coders are permitted to add features and introduce patches without ANY kind of documentation. ;- Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Skinny (SCCP) help
On Thu, Nov 06, 2003 at 12:57:49AM -0500, William Carlson wrote: Ok I see the confusion. I actually do have a TFTP server running on the asterisk machine but it does not have any Skinny stuff just ringtones and logos for my SIP 7960's. The id is found under settings then model information just add SEP in front of the MAC address. Thanks, Will Where do you get the 7960 ring tones and logos? sorry to cut in like this; very new to * and skinny phones; do you mean, all i need to install is *; no need to activate linux's tftp daemon? I have mine working w/o the tftp server running on my * machine. I just set the dhcpd option for the tftp server to the ip addr of my * machine. also, is the device name something i make up or burned in the phone's rom ; is so, where can i find the device name? This is just `SEP` . mac address [general] dateFormat = M-D-Y ; M,D,Y in any order (5 chars max) keepAlive = 120 I had to put this in to get the voice to go from the 7960 to *: bindaddr = 192.168.254.179 ; Address to bind to -- DataCrest, Inc. -- Technically Superior ** Walker Haddock http://www.datacrest.com DataCrest, Inc.e-mail: [EMAIL PROTECTED] 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Skinny (SCCP) help
This is where I got the ringtones.
http://www.loligo.com/asterisk/sounds/
- Original Message -
From: Walker Haddock [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, November 08, 2003 9:25 AM
Subject: Re: [Asterisk-Users] Skinny (SCCP) help
On Thu, Nov 06, 2003 at 12:57:49AM -0500, William Carlson wrote:
Ok I see the confusion. I actually do have a TFTP server running on the
asterisk machine but it does not have any Skinny stuff just ringtones
and
logos for my SIP 7960's. The id is found under settings then model
information just add SEP in front of the MAC address.
Thanks,
Will
Where do you get the 7960 ring tones and logos?
sorry to cut in like this; very new to * and skinny phones;
do you mean, all i need to install is *; no need to activate linux's
tftp daemon?
I have mine working w/o the tftp server running on my * machine. I just
set
the dhcpd option for the tftp server to the ip addr of my * machine.
also, is the device name something i make up or burned in the phone's
rom ; is so, where can i find the device name?
This is just `SEP` . mac address
[general]
dateFormat = M-D-Y ; M,D,Y in any order (5 chars max)
keepAlive = 120
I had to put this in to get the voice to go from the 7960 to *:
bindaddr = 192.168.254.179 ; Address to bind to
--
DataCrest, Inc. -- Technically Superior **
Walker Haddock http://www.datacrest.com
DataCrest, Inc.e-mail: [EMAIL PROTECTED]
1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216 fax: 1-205-823-7838
***
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___
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Re: [Asterisk-Users] Skinny (SCCP) help
woops I ment
http://www.loligo.com/asterisk/Cisco/79xx/current/
- Original Message -
From: Walker Haddock [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, November 08, 2003 9:25 AM
Subject: Re: [Asterisk-Users] Skinny (SCCP) help
On Thu, Nov 06, 2003 at 12:57:49AM -0500, William Carlson wrote:
Ok I see the confusion. I actually do have a TFTP server running on the
asterisk machine but it does not have any Skinny stuff just ringtones
and
logos for my SIP 7960's. The id is found under settings then model
information just add SEP in front of the MAC address.
Thanks,
Will
Where do you get the 7960 ring tones and logos?
sorry to cut in like this; very new to * and skinny phones;
do you mean, all i need to install is *; no need to activate linux's
tftp daemon?
I have mine working w/o the tftp server running on my * machine. I just
set
the dhcpd option for the tftp server to the ip addr of my * machine.
also, is the device name something i make up or burned in the phone's
rom ; is so, where can i find the device name?
This is just `SEP` . mac address
[general]
dateFormat = M-D-Y ; M,D,Y in any order (5 chars max)
keepAlive = 120
I had to put this in to get the voice to go from the 7960 to *:
bindaddr = 192.168.254.179 ; Address to bind to
--
DataCrest, Inc. -- Technically Superior **
Walker Haddock http://www.datacrest.com
DataCrest, Inc.e-mail: [EMAIL PROTECTED]
1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216 fax: 1-205-823-7838
***
___
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[EMAIL PROTECTED]
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___
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Re: [Asterisk-Users] IBM to Run VoIP On Linux
Engineers of all kinds can be a bit lax about documentation. However, the documentation police are rightly held in a regard usually reserved for lawyers, realtors, used car salesmen and serial killers. There isn't a single thing to stop anyone that really loves documentation actually producing some. This includes documentation for configuration management. However, I've yet to find a documentation whiner prepared to do anything useful. The ones who are genuine don't whine - they produce something. Regards, Steve Philipp von Klitzing wrote: Hi! 1) No 1.0 release. In fact, no release structure at all really. (Hold your flames: I know this is to be remedied soon, along with backtrack patches for security/stability.) With that comes a changelog and some basic documentation. I still find it amazing that coders are permitted to add features and introduce patches without ANY kind of documentation. ;- Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 6.0 image for Cisco 7960's?
The 6.0 image is available for download from Cisco TAC. The 6.0 image does support auto answer (Intercom.) Paul Mahler mail:[EMAIL PROTECTED] phone: 650.207.9855 fax: 877.408.0105 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: Thursday, November 06, 2003 1:37 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] 6.0 image for Cisco 7960's? Has anyone managed to get their hands on a 6.0 image for their 7960's yet? Or is it still in beta? Rumor (official rumor, from Cisco) is that it supports paging and intercom. I'm anxious to start working with those features, if they've been implemented sanely. What would be just as nice would be NOTIFY messages for pushing XML URL's to the phones, but sadly that feature request has gone uncommented-upon by Cisco, so I will assume the worst... JT ___ 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] diax request
Hi, - Original Message - From: Jon Pounder [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, November 08, 2003 12:30 AM Subject: [Asterisk-Users] diax request First of all great job on diax. I downloaded it and tried it, could not connect, got an authentication rejected,but I have not had a chance to figure out why yet - tried with a working gnophone setup in the configuration files. Check to see if you have a section like that in iax.conf file [yourusername] type=friend username=yourusername secret=yourpassword auth=plaintext host=dynamic context=yourcontext callerid=Your Full Nameyourextension Then you'll be able to register too. Is there any way to pass command line arguements to the program ? Where I see a real niche for a lightweight softphone is being able to serve the thing from a webpage, configured for whatever user is logged into the webserver. eg: I am at someone's office and want to make a call from the pbx, so I just login to the webserver and download my pbx extension (download the exe file complete with the configuration information passed in the command line to execute it -or take the config from a remote url) Not for the moment. For this purpose an ActiveX version will be available which can be integrated in a Web page, You can now take the app on a diskette (or flash disk) with you and launch it from there (just 130KB) Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 6.0 image for Cisco 7960's?
Nice this image lets my flakey 7960 run the SIP software :) Thanks, Will - Original Message - From: Paul Mahler [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, November 08, 2003 10:09 AM Subject: RE: [Asterisk-Users] 6.0 image for Cisco 7960's? The 6.0 image is available for download from Cisco TAC. The 6.0 image does support auto answer (Intercom.) Paul Mahler mail:[EMAIL PROTECTED] phone: 650.207.9855 fax: 877.408.0105 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: Thursday, November 06, 2003 1:37 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] 6.0 image for Cisco 7960's? Has anyone managed to get their hands on a 6.0 image for their 7960's yet? Or is it still in beta? Rumor (official rumor, from Cisco) is that it supports paging and intercom. I'm anxious to start working with those features, if they've been implemented sanely. What would be just as nice would be NOTIFY messages for pushing XML URL's to the phones, but sadly that feature request has gone uncommented-upon by Cisco, so I will assume the worst... JT ___ 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] IBM to Run VoIP On Linux
On Fri, 7 Nov 2003 23:50:06 -0800, Darren Martz [EMAIL PROTECTED] wrote: If I'm out of place in the following suggestions, I'm sure others will tell me grin - Create a clean SDK of the wonderful IAX2 protocol for Win32 and Mac to gain exposure everywhere - Push, entice, bribe IP phone designers to support the IAX2 protocol based on the clean and easy to use SDK - Someone once suggested an Asterisk logo program, excellent idea Darren, Take a look at iaxclient.sourceforge.net The current CVS version supports IAX or IAX2, and works on Win32, ia386Linux and Macs. There are also a few working crossplatform softphones there. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom200 MWI..
Is any one else having problems with the Snom 200 MWI?? If flashes and shows me there is a message then I go and listen to the message but the MWI does not clear.. The only way I have found to clear the MWI is to reboot the phone.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM to Run VoIP On Linux
Asterisk has got to be about the best kept secret in telephony. I've seen numerous articles on slashdot about VoIP, even in relation to Linux and only *once* has the post even mentioned Asterisk. Am I missing something, or is Asterisk clearly a good potential player in any kind of linux-based soft-switch idea? There are probably as many opinions on the topic as there are members of this list. Personal opinion based on many years of professional I/T and telecom experience is: - no marketing plan that list members can internalize (eg, no one knows where the project is heading, what's on the planning plate, when something will be released, or when problems will be corrected). - no formal/stable releases and associated components (as others have already noted) - poor documentation for a very sophisticated product Others have mentioned the product is lacking some feature or function, however those are probably related more to single instances of not being able to sell the product into a certain account, loosing a sale because some competitor sold a customer something that asterisk didn't have at the moment, or, inadequate skills to find a programming solution to some customer-needed function. I doubt the feature/function issue has anything to do with world acceptance of asterisk (except for better nat support). The poor documentation is evident by the number of how-to postings that have been occurring, the many arrogant responses from a select _few_ to newbie questions, and helpful examples that are so dispersed that finding them tends to consume inordinate amounts of time. The Handbook does a good job from an introduction perspective, but between it and getting to a basic working system is a significant and very time consuming problem (and that's coming from a person with 20+ years of central office, pbx, and transmission engineering experience within a telephone company that had over 8,000 employees.) If a newbie can't read C-code and hasn't been involved with any form of pbx from a technical perspective, the frustration of getting even a SOHO system operational is extremely high. John Todd's sample config's have been a good first step for newbies, but the average newbie doesn't have clue where to find them (as one example only) until after burning up the list with questions. After internalizing those configs and then looking at someone else's config for some specific function, it is not obvious in many cases how to integrate the two as: a) the second set of config's are not used in the same context as the first set, and, b) there is no base set of config's that would provide such an integrated understanding. If I were a trade rag evaluator/writer/publisher, I'd have to give up on trying to do much with this product as I'd run out of time trying to locate enough info to make it work in some acceptable pbx fashion. Although I have a small SOHO system running in production (with fully functional internet-based nat devices), I'm hesitant to suggest/recommend an asterisk-based system to clients right now because of the above items. Having been in the technical consulting business for over ten years now, I know without a doubt what it takes to support something like this, and I think several people on this list have already hinted that its far more difficult/time-consuming then most readers would anticipate. The more knowledgable list members have sufficient experience to find ways to address 99.9% up time, their own set of integrated feature/function configs, etc. But those few are not going to be the ones driving digium hardware sales upwards at a level needed for ongoing support. I'd suggest that a few not-so-time-consuming steps would lead to a significant increase in hardware and support sales, and therefore system acceptability and exposure. (Eg, 1000 newbies buying one/two x100p's have a greater financial impact then selling a few TE410's; 1000 newbies will create more industry exposure/acceptance then 10 highly skilled asterisk people that support their customer base.) I'd suggest something like the following to improve its acceptance and thus hardware sales (listed in priority order) and exposure: 1. Stop all new development for 30 days, fix existing problems, apply outstanding patches, and document (implementation/user, not developer) 2. get a stable release approach that includes a fully functional base system where 95%+ of the features actually work (with no echo, with nat working for newbie's, etc). 3. include in the stable release sufficient base-level config's that a newbie has a reasonable chance at implementation without having to post questions to the list or dig through 1000's of google items. (think about this very carefully). Might even include a url at the top of each config file where to look for sample config help, etc. 4. take a list of typical pbx features, develop the integrated asterisk configs and
Re: [Asterisk-Users] Snom200 MWI..
Is any one else having problems with the Snom 200 MWI?? If flashes and shows me there is a message then I go and listen to the message but the MWI does not clear.. The only way I have found to clear the MWI is to reboot the phone.. Gus, Works correct for me. Running v2.02q software on the 200, and just finished testing it. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM to Run VoIP On Linux
Asterisk has got to be about the best kept secret in telephony. I've seen numerous articles on slashdot about VoIP, even in relation to Linux and only *once* has the post even mentioned Asterisk. Am I missing something, or is Asterisk clearly a good potential player in any kind of linux-based soft-switch idea? There are probably as many opinions on the topic as there are members of this list. Personal opinion based on many years of professional I/T [snip] I'd suggest something like the following to improve its acceptance and thus hardware sales (listed in priority order) and exposure: 1. Stop all new development for 30 days, fix existing problems, apply outstanding patches, and document (implementation/user, not developer) 2. get a stable release approach that includes a fully functional base system where 95%+ of the features actually work (with no echo, with nat working for newbie's, etc). 3. include in the stable release sufficient base-level config's that a newbie has a reasonable chance at implementation without having to post questions to the list or dig through 1000's of google items. (think about this very carefully). Might even include a url at the top of each config file where to look for sample config help, etc. 4. take a list of typical pbx features, develop the integrated asterisk configs and scripts necessary to implement those features, and publish those in some common web or distro directory. 5. improve the printed documentation shipped with the hardware. (Those single-sheet instructions are missing several required steps, and have zero examples.) 6. document the basic user-oriented functions (eg, where is the list of *72-type functions). A user-guide would be nice. 7. publish, at a minimum, a TO-DO list that has some form of list prioritization of feature/function/problem-resolutions and estimated release timing. 8. Put together a marketing/sales plan for support and publish it on your web site. What's included; how to contact; options that might address an annual contract, per-call support, feature implementation, off-hour support, flat-annual-fee based on number of phones, etc. I, for one, would be interested in a commercial support plan based on a single 700-number or email address to reach help for certain items including configuration assistance. There certainly are a number of list members, including myself, that would be willing to help with the effort, but someone has to take a lead role and establish some common direction that does not exist today. Rich While I normally despise me-too posts, I think this one has enough weight that I must add my support for pretty much everything Rich has said above. I will assist with #'s 1, 3, 4, and 7. Task #1 is a bit larger than 30 days will allow, if you want real documentation, so I'd suggest only fix/patch/basic documentation. This requires, however, significant time investment from Digium and their staff (i.e.: Mark and Martin mostly) and we cannot know their schedule or enforce any type of time committment from them, as they have a business to run. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom200 MWI..
Works correct for me. Running v2.02q software on the 200, and just finished testing it. Thanks I will have to play a little more then.. What date CVS of Asterisk are you running? Gus, CLI show version Asterisk CVS-10/25/03-13:22:42 built by [EMAIL PROTECTED] on a i686 running Linux CLI sip.conf file: [3002] type=friend username=3002 secret=mypassword host=dynamic context=from-sip mailbox=3002 Snom 200 Config: S/W: 2.02q Basic out-of-the-box from Snom, dhcp, basic sip def's, no dialplan. (I did have to do a complete phone reset about a month ago when one of the beta releases was having problems.) Give me a call at 700-434-5395 if you have questions. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM to Run VoIP On Linux
On Sat, 2003-11-08 at 09:20, Rich Adamson wrote: John Todd's sample config's have been a good first step for newbies, but the average newbie doesn't have clue where to find them (as one example only) until after burning up the list with questions. After internalizing those configs and then looking at someone else's config for some specific function, it is not obvious in many cases how to integrate the two as: a) the second set of config's are not used in the same context as the first set, and, b) there is no base set of config's that would provide such an integrated understanding. Digium recently added a Unofficial Links section to the Documentation page on their web site. Doesn't seem like a lot of people know about it yet. I believe John Todd's page is on there, as well as my own and many links to other sites with information on Asterisk and products that talk to Asterisk (like X-Lite). Obviously it would be better to have this information in the main Asterisk Handbook, but it's a VERY good start and the Documentation page is now a great place to get Digium docs on Asterisk AND 3rd party docs for Asterisk. I've stopped referring people directly to my Asterisk site and instead refer them to the Unofficial Links page at Digium. --Eric -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom200 MWI..
Look in your mailbox somewhere in /var/spool/asterisk. If there is a gap in the MSG sequence numbers, or if there's a stray file in there it will make Asterisk think you have new messages even if you don't have new messages when you check your voicemail. On Sat, 2003-11-08 at 12:48, Rich Adamson wrote: Works correct for me. Running v2.02q software on the 200, and just finished testing it. Thanks I will have to play a little more then.. What date CVS of Asterisk are you running? Gus, CLI show version Asterisk CVS-10/25/03-13:22:42 built by [EMAIL PROTECTED] on a i686 running Linux CLI sip.conf file: [3002] type=friend username=3002 secret=mypassword host=dynamic context=from-sip mailbox=3002 Snom 200 Config: S/W: 2.02q Basic out-of-the-box from Snom, dhcp, basic sip def's, no dialplan. (I did have to do a complete phone reset about a month ago when one of the beta releases was having problems.) Give me a call at 700-434-5395 if you have questions. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: IAX2 trunking on one side only.
I sent this yesterday, but for some reawson it did not go through. Yes, ASTERISK1 = 2x TDM400P ASTERISK2 = 3x X100P I still cannot get it working past that. Is there something screwey with the wcfxs drivers and Linux? - Original Message - From: Louis-David Mitterrand [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, November 07, 2003 1:59 AM Subject: [Asterisk-Users] Re: IAX2 trunking on one side only. On Thu, Nov 06, 2003 at 10:41:15PM -0500, Brian Schrock wrote: Hello, I have searched google, read everything on the mailing list, read /usr/src/asterisk/README.iax and /usr/src/asterisk/doc/iax.txt(?), asked on the IRC channel and I cannot figure out what is wrong with my IAX2 trunk. Only asterisk2 of an ASTERISK1--LAN--ASTERISK2--PSTN will use IAX2 trunking. If I do an iax2 show trunk on asterisk1 it says 0 calls on trunk Do you have a zaptel device on each side? AFAIR zaptel timing is required for trunking to work. -- If Galileo is the spark that lights up the gas giant Jupiter, turning it into a second sun, that will be the last straw. We will then have no choice but to make safety the number one priority at NASA. -- falsification on /. ___ 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] IBM to Run VoIP On Linux
As we know market has thousands of great free source products, but somehow most of companies are buying commercial software and paying a lot of $$$ . Question becomes why they need to pay so much money for something what can be taken for free ? Also, why all these software products are so expensive ? For example voip billing software from MIND CTI, it cost more then 100K . However you can find a lot of individuals on internet (read message boards, posts, forums) who can create normal billing software and it will cost you ... 1K.. Yes , 1 K . So why not to buy it for 1K ? Why to spend thousands of dollars ?? Software itself cost 0$ if it has no support, like in case of example of individuals who is selling 1K programs. If you spend 100K to software you know that it has : support, normal documentation, updates, patches, and in most cases it is Ready-to-Go solution. If you need some kind of customization you can get one from manufacture too, sometimes it will cost more $$$, but still you will get what you need. In order to provide support, documentation, marketing materials, etc software companies are offering high cost for software. Every company has 2 major departments sales/marketing and technical. We are in Open Source World (OSW) here, I would say we belong to technical department in this case. We just don't have sales/marketing department for Asterisk software in OSW. Let's take a look , who are members of Asterisk community : programmers, developers, engineers,sys. admins, net admins - 99.9% Sales/marketing guys they don't know what is UNIX/Linux, Open Source, Zaptel driver, XP100, Kernel . They have no idea about it. So, they are coming here let's say and look into mailing list archive. What they see ? kernel, driver compilation problem, zaptel info, etc. Yes, nothing else. Where is asterisk-business mailing list ? Where it is ?? When I started unofficial asterisk forums, I thougth people will come to discuss business issues and solutions. Take a look, we have 99.9% technia questions postings ...(http://asterisk.xvoip.com) Let's take a look into our engineers, developers, programmers .. They are just great guys , with excellent knowledge, experience .. mostly in technical field. It is hard for regular developer to create proposal about IP PBX/Asterisk and to bring it to management for discussions/implementation. In 99% management will refuse proposal, even without reading it. But if they will start reading it, what they see ? Guess .. zaptel driver, kernel, unix, xp100.. Proposal will go to garbage can. Everybody has its own responsibilities within company. Technical guys/developers has nothing to do with Marketing/Sales and this is why we don't see Asterisk today in slashdot or in New York Times. We are missing marketing/sales actions to bring Asterisk to community, to companies. And I agree 200% with Adam and John about it. For Linus Tovalds it took couple years to make people to believe into Linux . Today, we can see his results, IBM, Dell, Compaq everybody is using Linux, it is known , very well known and end result of his Free Open Source campaign for Red Hat Linux is next: Red Hat becomes commercial software (commercial support per workstation.etc). Asterisk IP PBX project is great !! It has so many features, it is fantastic, but ... it is missing normal business/marketing/sales part. Without it , it will be very hard to bring it to community, to make it very well known. We need Asterisk Pro commercial package to be developed. We need to create nice documentation, faq, knowledgebase. Good software packaging. One more example about packaging, you can buy so called clone for X100p card and imagine it has beautiful colored box, CD , 30 pages colored nice manual and all these for 10$. If I will take X100p card from Digium and clone card ... I will choose clone card. So others will do. Asterisk Project is free source project and for someone who is using it at home environment and who is developer it is fine, they don't need anything else. But majority of people here represent companies, they can bring Asterisk to company as a solution. They only need help to do it. We can help them to do it, it will increase Digiums sales and will bring some profit to community participants. I think it is time to start commercial Pro version (not expensive !!!) of Asterisk. In my company we already made decision to do it, to offer people ready-to-go solution. But is is hard to do anykind of such product without Digium and Mark's support. Mark I think you are very overloaded with all projects, maybe we can help with Asterisk project. Asterisk Basic will stay as it is now, but we will be developing Asterisk Pro. This community has excellent talented people, just go to IRC and participate in chat .. You will see how helpful are people. And everything is free.. Some of guys spends couple hours to fix someone's problem, to write to someone AGI script, etc.. and everything is
Re: [Asterisk-Users] Re: IAX2 trunking on one side only.
Steven Critchfield wrote: On Fri, 2003-11-07 at 16:04, Olle E. Johansson wrote: Steven Critchfield wrote: We have to rename Zaptel timing to Asterisk timer, which is more correct since there are several ways of getting a timer to work, only one of them is by using Zaptel cards. http://www.voip-info.org/tiki-index.php?page=Asterisk+timer Actually it needs to be zapata timing. The other drivers just make a zapata compatible timing source that asterisk can use. Is the zapata timer superior to the ztdummy and ztrtc drivers, or are they, from an Asterisk point of view, compatible? From asterisk point of view they should be equivalent. The point is these all are a class of drivers providing similar functionality. The X100P, T100P, T400P, TE410P, S100U and such are zapata devices that provide timing and channel access. ztdummy and ztrtc don't provide a full zapata device only because they don't make a telephony interface, but the implement the timing needed for asterisk. Well, going back to my first suggestion, wouldn't it be easier to call it an Asterisk timer and to explain that ztdummy, ztrtc and the drivers for zapata devices all support the Asterisk timer. The zapata drivers in addition to the timer also support zapata (ZAP) channel access. I think it could be confusing to continue along the road on zapata compatible timers since there propably will be FreeBSD Asterisk timers and other drivers delivering timers in the future, and the Zapata driver will propably only be one of all possible drivers delivering a timer to Asterisk. It's not important to me, just trying to discuss semantics in order to be able to explain it to newcomers in what I believe to be an easy to understand way. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Callgroups and Pickupgroups in Console/dsp
Hi all. I've made a patch for chan_oss.c to enable callgroups and pickupgroups in it (since wasn't enabled). I needed it for a special use of the console (pickup calls arriving to the console from another phone) btw, If someone is interested, I can submit a patch to the bugtracker. I won't do it until that's usefult for someone... since is a very special features that probably no one will ever use lemme know. Matteo. -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 911 IAX(2): [EMAIL PROTECTED] - ext 911 Iaxtel: 1-700-56-62458 - ext 911 I assume this doesn't work with a console that is auto-answer? That's where I see some value, but I'm not sure from your description if that's what you've implemented. I would like to have an overhead paging system auto-answer a call (Bob, if you're in the garage, pick up a phone and dial *) and then allow a user pick up the call from their SIP extension by dialing a pickup group ID. This is good for large areas where there is normally a lot of background noise, or where the users of the system aren't sitting near phones most of the time and where carrying a portable is more hassle than it's worth (auto garages come to mind.) Having an overhead paging system connected to the console sound card, and then being able to yank a call from that channel onto another channel seems to be a very worthwhile feature. This is slightly different than callgroup/pickupgroups use that I've used in the past, which only work on channels that are still _ringing_ and have not been answered. Does your patch work on console channels that have already been answered? If so, please submit it to the bugtracker since that would be very useful. In fact, you should submit it anyway, since adding similar functionality to all channels is a Good Thing. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Rewriting asterisk to a full-scale system? (WAS: IBM to Run VoIP On Linux)
I beleive there's an (at least below) unmentioned argument why Asterisk may fail getting really big: Since Digium/Mark refuses to include any code that isn't copyrighted Digium/Mark, I'm afraid quite a few developers may be effectively excluded. By requiring a 'giveaway' to Digium/Mark, we stand the chance that the project one day will change its license to a (more) closed license (as just happened to MySQL, for instance). We already have channel drivers not included in the asterisk CVS because of this (chan_oh323 and chan_capi). I am not saying this license change will happen, but there's always a chance, and for some the requirement of giviing away the copyright to their own code may be a hinder to write it in the first place. roy On Saturday, Nov 8, 2003, at 10:16 Europe/Oslo, WipeOut wrote: Can I add to this and say that another thing that could be hindering the takup is Single System VoIP scalability and a certain amout of Enterprise flexibility.. Let me explain those two.. Before you start reading these and thinking This guy is mad!! let me just say that I love Asterisk and use it every day, but if M$ and IBM are getting into the game there is cause for concern.. The features I am going to talk about are very much Future Dreams becasue to impliment them would probably mean re-creating the entire code base from the groud up so I don't expest to see the features ant time soon.. I do think that these sorts of featured will be in the IBM and M$ IP PBX's and that is why I think Asterisk needs them.. So lets get started.. I know that many Asterisk servers can be connected together to scale the size of the system but this is still a problem because it is a headache to manage.. What is needed to get the big enterprise players on board is the ability to manage the PBX as a single entity no matter how many servers there are.. servers should simply be add on modules to the overall PBX to improve its VoIP call volume handling power.. I think the only way to achive this would be to make Asterisk a clustered software that sits a level above the servers.. The VoIP phones will see one Asterisk Server that listens on a single IP address per subnet on the network but behind that single system image could be one, two or fifty servers providing the processing power for all the calls, and as power is needed you simply have to add servers.. If you need more PRI lines just add a Digium card to a server and enable that server as a gateway node in the cluster.. With in this model the voicepath between the servers in the cluster needs to be dynamic so the shortest path is always used (IAX can probably handle this quite well already), and CDR must be accurate maybe one or two of the nodes needs to allocated the task of being the CDR server and all other servers will feed back to the central server with the call logging information.. In Enterprise flexibility I am taking about user and phone management and services.. On the phone management side (and I know many don't seem to like the idea) but a platform independent full featured management interface is needed.. If its done in Java or web based running on the Asterrisk sever itself, similar to how webmin has its own web server, does not matter but we live in a world now where admins like GUI management tools.. Leading on from that is an Operator Interface for receptionists and phone operators to be able to manage calls.. See which lines are busy, connect calls and the various other things that these interfaces do.. Next a monitoring interface (somthing similar MRTG would probably do it..) showing server loads and statistics so system management and upgrading is easy to see and plan for.. Then the need to support hot desking.. By this I mean that the phone and the user need to be seperate entities on the system.. then the user can sit down at any phone on any desk run through a login procedure (either on the phone or in some easily accesible interface) and all their calls will then be routed to that phone.. I know there are hacks and work arounds to getting this kind of functionality using queues and the Asterisk DB and various other options but it needs to be a standard working system.. Finally an automatic provisioning system.. New user joins the company, click a button on the management interface and give them their extension number and extension password.. no editing files and restarting servers or anything like that its all done behind the scene.. So did I just thumb suck these concepst out of this air?? not totally.. Last year I did a contract at a large comnpany in London and was working on a user provisioning system.. This company has thousands of users in a single building (and a single PBX) in London, and thousands more accross the country.. It was a provisioning system so I needed to talk to the telecoms guys to see if we could
Re: [Asterisk-Users] IBM to Run VoIP On Linux
Take a look at iaxclient.sourceforge.net The current CVS version supports IAX or IAX2, and works on Win32, ia386Linux and Macs. There are also a few working crossplatform softphones there. ...and iaxclient is probably not one of them. Working softphones for me includes stability, intuitive user interface at first. AFAIK, iaxclient lacks both roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM to Run VoIP On Linux
I've stopped referring people directly to my Asterisk site and instead refer them to the Unofficial Links page at Digium. --Eric -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ ;-) /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Rate in CDR
So what do people think about adding the call rate to the CDR structure?? This would allow you to detail a call with the rate that was in affect for that call. When you come back later and do the billing for the customer you would have the actual per min rate in the record. I think this solves an issue when you have changing rates and multiple providers. If one provider is down, and you use a back up you can track the rate better. In fact as I type this, I think we should have cost_rate the rate you where charged for this call cust_rate the rate you charge your customer for this call or is their a better way to do this. john brown, ceo chagres technologies, inc (US) chagres technologies, BV (EMEA) opening soon Providers of VoIP hardware http://www.chagres.net/products/voip/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM to Run VoIP On Linux
I actually meant on IRC. I had forgotten about my .sig, LOL! Thanks for pointing it out. --Eric On Sat, 2003-11-08 at 14:59, Olle E. Johansson wrote: I've stopped referring people directly to my Asterisk site and instead refer them to the Unofficial Links page at Digium. --Eric -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ ;-) -- Digium Documentation and links to 3rd party Asterisk pages: http://www.digium.com/index.php?menu=documentation BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Rate in CDR
So what do people think about adding the call rate to the CDR structure?? This would allow you to detail a call with the rate that was in affect for that call. When you come back later and do the billing for the customer you would have the actual per min rate in the record. I think this solves an issue when you have changing rates and multiple providers. If one provider is down, and you use a back up you can track the rate better. In fact as I type this, I think we should have cost_rate the rate you where charged for this call cust_rate the rate you charge your customer for this call or is their a better way to do this. john brown, ceo chagres technologies, inc (US) chagres technologies, BV (EMEA) opening soon Providers of VoIP hardware http://www.chagres.net/products/voip/ This is already more-or-less supported with the custom variable fields in the CDR, and is awaiting integration into CVS. There are two versions of the patch: one that has multiple custom fields, and the second version that has only one field, into which you'll need to create your own separators (ugly, but for some reason that was what was decided upon as the final version.) See http://bugs.digium.com/bug_view_page.php?bug_id=442 JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Rate in CDR
I personally think the right place to deal with billing is in your billing application. Your billing application should know about special rates for different customers, time of day rates, destination rates, etc. There is already enough information in the CDR to know which provider you are going thru. i.e. fields 6 and 8 in the CDR logs. These are Call Detail Record logs, not Call Billing Logs. 8-) If your billing application can't handle this, it should be pretty easy to build a small preprocessing application for the log files. Why invent something new when you can reuse existing tools to do all or most of the job? On Sat, 2003-11-08 at 15:08, John Brown (CV) wrote: So what do people think about adding the call rate to the CDR structure?? This would allow you to detail a call with the rate that was in affect for that call. When you come back later and do the billing for the customer you would have the actual per min rate in the record. I think this solves an issue when you have changing rates and multiple providers. If one provider is down, and you use a back up you can track the rate better. In fact as I type this, I think we should have cost_rate the rate you where charged for this call cust_rate the rate you charge your customer for this call or is their a better way to do this. john brown, ceo chagres technologies, inc (US) chagres technologies, BV (EMEA) opening soon Providers of VoIP hardware http://www.chagres.net/products/voip/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Now with links to Unofficial Asterisk pages! http://www.digium.com/index.php?menu=documentation BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Rate in CDR
At 01:08 PM 11/8/2003, you wrote: So what do people think about adding the call rate to the CDR structure?? Sounds great, but there's one problem. How does asterisk know what the current rate in effect is? I can think of several ways to do this, but they all involve some fairly significant C coding. or is their a better way to do this. I just use a perl script. The CDR record tells me the number dialed and the channel that handled the far end of the call. Based on that I can tell which provider was used, what number was called, how long the call lasted and when the call was made. That's more than enough information to calculate rates. Parse that file once an hour/day and there you go. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Rate in CDR
He did write the new one where you could append say value1:value2 into that field. Still not pretty but functional. bkw On Sat, 8 Nov 2003, John Todd wrote: So what do people think about adding the call rate to the CDR structure?? This would allow you to detail a call with the rate that was in affect for that call. When you come back later and do the billing for the customer you would have the actual per min rate in the record. I think this solves an issue when you have changing rates and multiple providers. If one provider is down, and you use a back up you can track the rate better. In fact as I type this, I think we should have cost_rate the rate you where charged for this call cust_rate the rate you charge your customer for this call or is their a better way to do this. john brown, ceo chagres technologies, inc (US) chagres technologies, BV (EMEA) opening soon Providers of VoIP hardware http://www.chagres.net/products/voip/ This is already more-or-less supported with the custom variable fields in the CDR, and is awaiting integration into CVS. There are two versions of the patch: one that has multiple custom fields, and the second version that has only one field, into which you'll need to create your own separators (ugly, but for some reason that was what was decided upon as the final version.) See http://bugs.digium.com/bug_view_page.php?bug_id=442 JT ___ 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] Getting Started
What is the best way in getting started evaluating Asterisk? Are there recommendations on the types of card I should be using for an initial eval? Thanks Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Rate in CDR
As rates change of the course of a month, it would be nice to know what the rate was at the time of the call. some detail records :) On Sat, Nov 08, 2003 at 03:37:13PM -0600, Eric Wieling wrote: I personally think the right place to deal with billing is in your billing application. Your billing application should know about special rates for different customers, time of day rates, destination rates, etc. There is already enough information in the CDR to know which provider you are going thru. i.e. fields 6 and 8 in the CDR logs. These are Call Detail Record logs, not Call Billing Logs. 8-) If your billing application can't handle this, it should be pretty easy to build a small preprocessing application for the log files. Why invent something new when you can reuse existing tools to do all or most of the job? On Sat, 2003-11-08 at 15:08, John Brown (CV) wrote: So what do people think about adding the call rate to the CDR structure?? This would allow you to detail a call with the rate that was in affect for that call. When you come back later and do the billing for the customer you would have the actual per min rate in the record. I think this solves an issue when you have changing rates and multiple providers. If one provider is down, and you use a back up you can track the rate better. In fact as I type this, I think we should have cost_rate the rate you where charged for this call cust_rate the rate you charge your customer for this call or is their a better way to do this. john brown, ceo chagres technologies, inc (US) chagres technologies, BV (EMEA) opening soon Providers of VoIP hardware http://www.chagres.net/products/voip/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Now with links to Unofficial Asterisk pages! http://www.digium.com/index.php?menu=documentation BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ 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] Getting Started
download the code, complile the code, start bashing on configs. :) if you want to glue to the PSTN, i'd recommend getting a FXO (WC-X100P) and FXS (TDM-10B) card and some cheap SIP / VoIP phones (grandstream or snom) john brown, ceo chagres technologies, inc Providers of VoIP hardware http://www.chagres.net/products/voip/ On Sat, Nov 08, 2003 at 05:23:57PM -0500, Peter A. Solomon wrote: What is the best way in getting started evaluating Asterisk? Are there recommendations on the types of card I should be using for an initial eval? Thanks Peter ___ 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] IBM to Run VoIP On Linux
On Sat, 8 Nov 2003 21:59:43 +0100, Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote: Take a look at iaxclient.sourceforge.net The current CVS version supports IAX or IAX2, and works on Win32, ia386Linux and Macs. There are also a few working crossplatform softphones there. ...and iaxclient is probably not one of them. Working softphones for me includes stability, intuitive user interface at first. AFAIK, iaxclient lacks both Could you offer some constructive feedback? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] contact
Sorry to do this to the list, but I have no choice . Walker, I've been trying to send you an email off-list for the last couple of weeks, but one of my mail-hops is failing, do you have alternative address that I can try ??? Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Rate in CDR
On Sat, 8 Nov 2003 15:30:14 -0700, John Brown (CV) [EMAIL PROTECTED] wrote: As rates change of the course of a month, it would be nice to know what the rate was at the time of the call. some detail records :) Then tell your billing package that, effective at 5:23pm on October 24, the price of a call to Zambia went up 2 cents per minute for bronze-pak customers and 1.25 cents per minute for silver and gold customers. :-) Call rating is a relatively complex task, traditionally done by people over in the business office. Stuffing rating code into Asterisk is not going to be all that useful for all but the most simple applications. Our billing is complex enough that we outsource it to a third-party company, and we only handle tens of thousands of calls per month. -rt -- Ryan Tucker Network Engineer NetAccess, Inc. 1159 Pittsford-Victor Road Bldg. 5, Suite 140 Pittsford, New York 14534 585-419-8200 www.netacc.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM to Run VoIP On Linux
I think it is time to start commercial Pro version (not expensive !!!) of Asterisk. In my company we already made decision to do it, to offer people ready-to-go solution. But is is hard to do anykind of such product without Digium and Mark's support. Mark I think you are very overloaded with all projects, maybe we can help with Asterisk project. Asterisk Basic will stay as it is now, but we will be developing Asterisk Pro. Correct me if I am wrong, but unless you have a license from Digium directly then you must sell your Pro version software under GPL. What you do for documentation/packaging is probalby not covered under GPL. You make some good points but I think that the solution is not to commercialize everything. There is starting to be a trend of businesses (and governments) turning away from commercialization (ever so slowly but it is in that direction). Pick something that is missing and contribute that to the community. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Rate in CDR
Ryan Tucker wrote: Call rating is a relatively complex task, traditionally done by people over in the business office. Stuffing rating code into Asterisk is not going to be all that useful for all but the most simple applications. Our billing is complex enough that we outsource it to a third-party company, and we only handle tens of thousands of calls per month. -rt Apparently you haven't figured out the true power of Asterisk, yet. The publicly available CDR backends are not dealing with CDRs properly or completely. We do an insane amount of traffic and the real-time rating engine I have created and integrated into Asterisk doesn't even break a sweat. It is all in how you tackle the problem. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM to Run VoIP On Linux
Robert, You are right about licensing issues. But I have mentioned in my email that we need direct support from Mark and Digium. It might be direct license or something else. But we need Digium and Mark to participate in this project. Now about commercialization : Idea is to create enhanced version of *, which will include customer support, documentations, manuals, new additional services (like voice termination), etc. Asterisk as it is now will stay as it is.. But enhanced version will be commercial, and again we need to calculate all fees and expenses involved. But commercial version will help community and Digum and all of us to make nice famous product. For someone who can't afford fees for technical support we will open Asterisk Knowledgebase .. my idea is to have it in normal way and open... But to create one, to make it nice and informative we need funds. It is clear that Digium is not going to sponsor the project (maybe I am wrong, this is why I need some feedback from Mark). so we need to find funding by ourselves. My company is willing to participate in project and ready to bring some parts into it. Once again, I want to make clear one thing. Nobody is talking about complete commercialization of Asterisk, I am talking only about enhanced version. I can bet that we will have customers willing to pay money to have system with enhanced features, support ,etc. We just need to start it. If any questions ,let's discuss it. Also thank you Robert that you published your info on forums, for someone who is not part of mailing list it bring information :) Thanks, Alexander Unofficial Asterisk Forums URL : http://asterisk.xvoip.com Registration is : http://asterisk.xvoip.com/profile.php?mode=register New XVOIP network , get your +1 777 number today. [EMAIL PROTECTED] - Original Message - From: rnc Info Lists [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, November 08, 2003 5:55 PM Subject: Re: [Asterisk-Users] IBM to Run VoIP On Linux I think it is time to start commercial Pro version (not expensive !!!) of Asterisk. In my company we already made decision to do it, to offer people ready-to-go solution. But is is hard to do anykind of such product without Digium and Mark's support. Mark I think you are very overloaded with all projects, maybe we can help with Asterisk project. Asterisk Basic will stay as it is now, but we will be developing Asterisk Pro. Correct me if I am wrong, but unless you have a license from Digium directly then you must sell your Pro version software under GPL. What you do for documentation/packaging is probalby not covered under GPL. You make some good points but I think that the solution is not to commercialize everything. There is starting to be a trend of businesses (and governments) turning away from commercialization (ever so slowly but it is in that direction). Pick something that is missing and contribute that to the community. Robert ___ 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] SIP, Sipura SPA-2000, and Voicemail2
I figured out what was going on with the lack of/stuck on stuttered dial tone. Apparently, there are two voicemail directories being referenced: /var/spool/asterisk/voicemail/default, and /var/spool/asterisk/voicemail/local. The sip phones were using /var/spool/asterisk/voicemail/local to dump VM messages into, yet the MWI looks at /var/spool/asterisk/voicemail/default. Does anyone know why two different directories are being used? The context of the sip phones is not default, it is house-local, and I'm curious as to why the 'local' directory is used, and how it is derived from the SIP phone context. I can correct the problem by making a symlink from local to default, but this does not appear to be the best way to solve this problem. Thanks, Steve. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom 200
I've seen that myself - both on Snom 100 and Snom 200 devices using the latest beta firmware. I did however suspect this was a bug in the Snom devices. Generally whatever comes out of the speaker sounds crappy - even ringing sometimes. Also it seems to come an go and wasn't like this a couple of firmware versions ago - so I haven't even bothered considering if this may be an Asterisk related problem. Might be worth asking Snom about this. Regards, Lars... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mark Evans Sent: 07 November 2003 21:23 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Snom 200 Hi All I have a snom 200 phone here which works perfectly when using the handset to playback the voicemail messages etc. However when I play back the voice using the speakerphone it sounds choppy. Anyone had this problem before? Regards Mark ___ 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] Eicon Diva Server 4BRI
Hi Everybody, Has anybody tried the above (or indeed any other 4XBRI cards) successfully with Asterisk. As far as I can see the above mentioned card is an active ISDN card but supported by it's own I4L driver. This leads to interesting questions particularly regarding echo cancellations (which usually doesn't work on the cheap passive cards with one exception as far as I can see). Regards, Lars... -- Lars Boegild Thomsen Technical Director JustIT Sdn. Bhd. Cell Phone (MY): +60 (16) 323 1999 ICQ: 6478559 Yahoo Chat: [EMAIL PROTECTED] MSN Chat: [EMAIL PROTECTED] http://www.justit.ws ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP, Sipura SPA-2000, and Voicemail2
I figured out what was going on with the lack of/stuck on stuttered dial tone. Apparently, there are two voicemail directories being referenced: /var/spool/asterisk/voicemail/default, and /var/spool/asterisk/voicemail/local. The sip phones were using /var/spool/asterisk/voicemail/local to dump VM messages into, yet the MWI looks at /var/spool/asterisk/voicemail/default. Does anyone know why two different directories are being used? The context of the sip phones is not default, it is house-local, and I'm curious as to why the 'local' directory is used, and how it is derived from the SIP phone context. I can correct the problem by making a symlink from local to default, but this does not appear to be the best way to solve this problem. Pure guess is something to do with voicemail vs voicemail2. I just swapped from voicemail to voicemail2 a couple of days ago, and the 'local' directory on my machine is datestamped from exactly when I did the conversion. (The only voicemail box within the 'local' directory is for an extension that I was using for testing during the conversion. All other extensions reside in the 'default' directory, and voicemail2 is working fine with 7960's and a Snom 200.) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom 200
Don't know if you seen my earlier post, but my 200 has been working just great since v2.02q was installed a couple of days ago. Speakerphone sounds fine, MWI works fine, etc. This release seems to be the best I've seen for a while. Rich I've seen that myself - both on Snom 100 and Snom 200 devices using the latest beta firmware. I did however suspect this was a bug in the Snom devices. Generally whatever comes out of the speaker sounds crappy - even ringing sometimes. Also it seems to come an go and wasn't like this a couple of firmware versions ago - so I haven't even bothered considering if this may be an Asterisk related problem. Might be worth asking Snom about this. Regards, Lars... -Original Message- Hi All I have a snom 200 phone here which works perfectly when using the handset to playback the voicemail messages etc. However when I play back the voice using the speakerphone it sounds choppy. Anyone had this problem before? Regards Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hum on z-plex 10
I have been getting disconnects on my pots lines from bellsouth recently. The BS repairman determined that their was an audible 60 HZ hum on all of my fxo ports. We also measured the following impedances between tip and ring: T-R 35K ohms R-R 90K ohms T-T 16K ohms The two latter measurements were between ports. The bellsouth tech kept saying we had a short, I had to get him to put his meater on the lines and read the impedances. My guess is that there is a bad capacitor in the zhone. Does anyone have any experience with the zplex-10 and what are the typical impedances and DC characteristics? Is there any problem with opening them up and replacing the components? Thanks, Walker Haddock -- DataCrest, Inc. -- Technically Superior ** Walker Haddock http://www.datacrest.com DataCrest, Inc.e-mail: [EMAIL PROTECTED] 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Rate in CDR
John, The only thing that should be in the call data record (CDR) is data about the call. You want to do all your rating of calls in the billing system. That way you offload the additional processing to a batch activity. You have enough data in the CDR to combine it with data in the billing system to generate bills and do PL calculations (within certain limits of course). THX/BDH On Sat, 2003-11-08 at 16:08, John Brown (CV) wrote: So what do people think about adding the call rate to the CDR structure?? This would allow you to detail a call with the rate that was in affect for that call. When you come back later and do the billing for the customer you would have the actual per min rate in the record. I think this solves an issue when you have changing rates and multiple providers. If one provider is down, and you use a back up you can track the rate better. In fact as I type this, I think we should have cost_rate the rate you where charged for this call cust_rate the rate you charge your customer for this call or is their a better way to do this. john brown, ceo chagres technologies, inc (US) chagres technologies, BV (EMEA) opening soon Providers of VoIP hardware http://www.chagres.net/products/voip/ ___ 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] Snom 200
Well - problem with this fast moving technology. No - my last update was about 10 days old. My phones are downloading as I write this :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson Sent: 09 November 2003 10:34 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Snom 200 Don't know if you seen my earlier post, but my 200 has been working just great since v2.02q was installed a couple of days ago. Speakerphone sounds fine, MWI works fine, etc. This release seems to be the best I've seen for a while. Rich I've seen that myself - both on Snom 100 and Snom 200 devices using the latest beta firmware. I did however suspect this was a bug in the Snom devices. Generally whatever comes out of the speaker sounds crappy - even ringing sometimes. Also it seems to come an go and wasn't like this a couple of firmware versions ago - so I haven't even bothered considering if this may be an Asterisk related problem. Might be worth asking Snom about this. Regards, Lars... -Original Message- Hi All I have a snom 200 phone here which works perfectly when using the handset to playback the voicemail messages etc. However when I play back the voice using the speakerphone it sounds choppy. Anyone had this problem before? Regards Mark ___ 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] Snom 200
And it seemed to have moved even faster than I imagined. My phone is now at 2.02r and not q. Wonder how that's going to work. I might have missed the only working version without even having had the pleasure of trying it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson Sent: 09 November 2003 10:34 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Snom 200 Don't know if you seen my earlier post, but my 200 has been working just great since v2.02q was installed a couple of days ago. Speakerphone sounds fine, MWI works fine, etc. This release seems to be the best I've seen for a while. Rich I've seen that myself - both on Snom 100 and Snom 200 devices using the latest beta firmware. I did however suspect this was a bug in the Snom devices. Generally whatever comes out of the speaker sounds crappy - even ringing sometimes. Also it seems to come an go and wasn't like this a couple of firmware versions ago - so I haven't even bothered considering if this may be an Asterisk related problem. Might be worth asking Snom about this. Regards, Lars... -Original Message- Hi All I have a snom 200 phone here which works perfectly when using the handset to playback the voicemail messages etc. However when I play back the voice using the speakerphone it sounds choppy. Anyone had this problem before? Regards Mark ___ 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] Re: SIP, Sipura SPA-2000, and Voicemail2
Solution: The context used in voicemail.conf has to match the default context in sip.conf. Sip.conf: [general] port=5060 bindaddr=192.168.17.2 tos=lowdelay disallow=all allow=ulaw context=default ; Note: this must match voicemail.conf ; ; SIP Entry for sipura line 1 ; This phone is allowed to dial extensions and local phone numbers ; [101] type=friend host=dynamic context=house-toll reinvite=no canreinvite=no qualify=300 secret=x callerid=Sipura Line 1 101 username=101 mailbox=101 nat=0 ; Sample for sipura line 2 ; This phone is allowed to dial extensions and local phone numbers ; [102] type=friend host=dynamic context=house-toll reinvite=no canreinvite=no qualify=300 secret=y callerid=Sipura Line 2 102 username=102 Voicemail.conf: [general] format=wav maxmessage=180 [default] ; Note: this was [local] ; ; format: password, name, email address for attached voicemail msgs ; 101 = ,Steve Rodgers,[EMAIL PROTECTED] 102 = ,Karen Rodgers,[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
