Re: [Asterisk-Users] Important: The Asterisk Mailing list (newsubject)
G'day Darren, On Fri, 19 Mar 2004, Darren Nickerson wrote: > I give people much more credit than you do, as does the author of that > essay. So do most experienced list-owners out there. Anyone who wants a > post to go to the list will use the 'reply all' feature of their mailer. > They'll understand that they're doing this, they'll know why they're posting > it to the whole list, and they'll probably also expect their post to be > reflected back to them from the list and go hunting for it if it's not. You give too much credit to people, indeed. I cannot say about this list, but most lists I use have high corporate populations, where the users *have* to use mailers like Outlook or (cringe) Notes. For mailing list admins to expect users of these mailers to try and find the functions referred to in the article is ludicrous (and yes, I know I just said in a prior note "use the function of your mailer", but I was referring to the standard Reply function -- if you have ever tried to use a mailing list with Lotus Notes you will bless the list admin who maintains the status-quo and munges "Reply-to"). > Some very compelling arguments are very clearly explained in the essay I > pointed you to, and replying with "use your mail filters or unsubscribe" is > not a very productive response (even if it COULD fix the problem). Nor is "List owner, action my request or resign" ;-) I did read the article you pointed to, and neither you or the article's author have convinced me that this is a good thing. When the entire world is using My Favourite Mailer, let's talk again. Until then, let's stick with what works for -- and is understood by -- the vast majority of people. Cheers, Vic Cross ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Speex
Carlos Chavez wrote: I have been trying out Asterisk with the speex codec with X-lite as a client. I applied the REG patch on my windows machine that is recommended in Voip-info.org. Every time I make a call I get the following error: codec_speex.c:167 speextolin_framein: Out of buffer space If I do not hang up before 30 seconds, my machine then slows down and it can take up to 10 minutes to shut down. Is speex worth the trouble? My personal opinion is that you would be better off using GSM or iLBC.. I don't think Speex has any advantage over these codecs and is always a PITA.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Important: The Asterisk Mailing list (new subject)
As I started this trend I take the right to end it. I just want us to follow John Postel's rule for how to act on the Internet (I think he defined it for TCP/IP software, but it can be applied here too.) Be strict in what you send Be generous in what you accept Sending a reply to the list might not always be the best choice if it's a personal matter, a quote for services or something. But we still want all tips and tricks, solutions and suggestions flowing on the list. The list archives is a valuable resource for those of us trying to write documentation, find bugs and solve bugs. We don't want those messages to go in private channels. This case is closed. However, there *A LOT* of open cases on http://bugs.digium.com that we need your help in solving. If you do not feel that you can assist in bug-hunting, you can assist in testing additions to Asterisk (patches) and make comments. Keep bugs.digium.com cooking. We need to kill bugs in order to reache the 1.0 release soon. If you have questions and need help navigating bugs, find a bug marshal on irc. They're listed on the front page of the bug tracker. Thank you for your help and assistance in getting closer to a 1.0 release! /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MOH: Copyright issues?
mattf [EMAIL PROTECTED] wrote: > Every time we get close to having old works fall > into the public domain, the large hollywood lobby spreads it's cash > around and buys enough votes to extend copyrights yet again. > The U.S. Senate -- white male millionaires working for YOU! -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Festival
Justin Carlson wrote: I am sorry if this is a silly question but I can not seem to locate the festival binaries. does this come with asterisk or is it another project? No question is silly. This is a good time to remind the list of the FAQ http://www.voip-info.org/tiki-index.php?page=Asterisk%20faq You'll find pointers to Festival and other third party softwares on that page. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Registration from failed for 'xxx'
Thomas Gallaway wrote:
Here is my problem. I have 2 phones (Grandstream Budge Tone-100)
loosing the sip registration
every 4 hours. I can not find out why.
It seems like the registration fails, then a few minutes after
registers sucessfull.
Mar 19 14:06:14 NOTICE[147466]: Registration from
'' failed for '192.168.1.114'
-- Executing NoOp("SIP/114-5d35", "") in new stack
-- Registered SIP '114' at 192.168.1.114 port 5060 expires 180
sip show peers show's me this:
116/116 192.168.1.116 (D) 255.255.255.255 5060
Unmonitored
114/114 (Unspecified) (D) 255.255.255.255 0
Unmonitored
113/113 192.168.1.113 (D) 255.255.255.255 5060
Unmonitored
Thanks for any input on this.
I guess the SIP show peer changed shortly after this.
The first message "Registration failed" means that your UA tried to register but
couldn't
authenticate. This is how SIP works, first you send a REGISTER without auth, to get an
authentication challenge from the server. The next line "Registred SIP '114' indicates
that the UA took the challenge and sent another REGISTER, this time *with*
authentication
and successfully registred.
A SIP ua that registers need to do this from time to time. You set interval in the
phone
or in the sip.conf file. The sip.conf file takes precedence.
/Olle
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RE: [Asterisk-Users] Important: The Asterisk Mailing list (newsubject)
Darren Nickerson [EMAIL PROTECTED] wrote: > I strongly support removing the current reply-to-list setting, and you > should too. > > Like many new list admins, I once thought the reply-to was kewel. Requests > to remove it kept coming up, ... usually around the same time someone > embarrassed themselves by posting a personal reply/flame to the list. > Someone, in frustration, finally pointed me to the following URL: > Every mail list I subscribe to works in the same way, and every single one has had the requisite 1.5 users moaning about the reply-To header. It's not difficult - either reply to the list (the default, and rightly so) or, if you feel the need to reply privately, modify the To address. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Important: The Asterisk Mailing list (newsubject)
Kevin Walsh wrote: Darren Nickerson [EMAIL PROTECTED] wrote: I strongly support removing the current reply-to-list setting, and you should too. Like many new list admins, I once thought the reply-to was kewel. Requests to remove it kept coming up, ... usually around the same time someone embarrassed themselves by posting a personal reply/flame to the list. Someone, in frustration, finally pointed me to the following URL: Every mail list I subscribe to works in the same way, and every single one has had the requisite 1.5 users moaning about the reply-To header. It's not difficult - either reply to the list (the default, and rightly so) or, if you feel the need to reply privately, modify the To address. Or just use the ``Reply All'' feature. I hate to keep a dead topic alive, but mailman has a neat setting that prevents you from getting duplicate emails across multiple lists _and_ if your address is also in the to/cc. Really, this is a user issue, not a software/software settings issue. Learn to use your mail client and your settings or don't use lists. I don't mean to come across harshly and this is directed towards nobody in particular. --Devon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voip terminations in Australia and New Zealand
Greetings from downunder, Does anybody know of any organization providing reasonably priced voip call terminations in Australia and New Zealand ?? Does anybody know of any reasonably priced DID providers in Australia and New Zealand ??. Please feel free to contact me off list. cr ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Speex
WipeOut wrote: Carlos Chavez wrote: I have been trying out Asterisk with the speex codec with X-lite as a client. I applied the REG patch on my windows machine that is recommended in Voip-info.org. Every time I make a call I get the following error: codec_speex.c:167 speextolin_framein: Out of buffer space If I do not hang up before 30 seconds, my machine then slows down and it can take up to 10 minutes to shut down. Is speex worth the trouble? My personal opinion is that you would be better off using GSM or iLBC.. I don't think Speex has any advantage over these codecs and is always a PITA.. Sorry but I disagree. I am using SPEEX and voice quality is much better than GSM and it consumes less bandwidth. Using Speex and Linux is just a "make; make install". Although, we MUST encourage OpenSource initiatives or we will pay Licenses forever. Take a look at G.723 or G.729, for example. ITU released this protocols many years ago and we still have to pay royalties. About this kind of license there is another thread talking about... Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Store caller IP in CDR
How can I configure * to store the caller and called Party IP Address in the CDR file. Thanks for support Craeck _ Fotos - MSN Fotos das virtuelle Fotoalbum. Allen Freunden zeigen oder einfach ausdrucken: http://www.msn.de/antispam/prevention/junkmailfilter Schluß mit Spam - MSN hilft Ihnen hier weiter. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Store caller IP in CDR
Fritz Müller wrote: How can I configure * to store the caller and called Party IP Address in the CDR file. Depends on the channel, not all channels are IP based. Check the CDRuserfield - it's a free field in the CDR you set in the dialplan or from a script. Without knowing why you want this, I can't see an easy way to do this in SIP, my playground. In SIP calls, the caller sometimes use one private IP address and one public address, so you have to decide which one you want. Is it really the IP address you're after or the URI:s? In chan_sip2, my playground, I've added some variables that track call ID and the From: user. The To: User is already there. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Qualify statement
Does anyone know if "qualify=XXX" should be used ONLY for user agents behind NAT. I tried to use it on a devices (ATA 186 and SPA 2000) on a public IP, and * goes to segmentation fault every time it starts. If it is meant to be used just behind NAT fine, but what and how does * monitor user agent status? Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Important: The Asterisk Mailing list (newsubject)
On Fri, Mar 19, 2004 at 11:23:53PM -0500, Darren Nickerson wrote: > I strongly support removing the current reply-to-list setting, and you > should too. I would agree with this too, when replying to a post, the reply should be to the sender, if the receipient wants to reply to everyone, then they should specifically request that. Most sensible mail user agents have mechanisms for specificially replying to lists. Also mail addresses from this list are being SPAM harvested (just FYI). Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voip terminations in Australia and New Zealand
Hi Craig, Someone mentioned packet8 to me earlier, having reasonable international calls from australia, I'd assume they could terminate to it. On Sat, Mar 20, 2004 at 10:14:44PM +1130, Craig wrote: > Greetings from downunder, > > Does anybody know of any organization providing reasonably priced voip call > terminations in Australia and New Zealand ?? > > Does anybody know of any reasonably priced DID providers in Australia and > New Zealand ??. > > Please feel free to contact me off list. > > cr > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users HTH, Andrew Griffiths -- Since email addresses here are being harvested, [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Qualify statement
Hi, Senad Jordanovic wrote: Does anyone know if "qualify=XXX" should be used ONLY for user agents behind NAT. No, you can use it if you want to monitor the agent. I tried to use it on a devices (ATA 186 and SPA 2000) on a public IP, and * goes to segmentation fault every time it starts. If it is meant to be used just behind NAT fine, but what and how does * monitor user agent status? Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Store caller IP in CDR
Fritz Müller wrote: How can I configure * to store the caller and called Party IP Address in the CDR file. Smells like you need to re-think your billing process. There is absolutely no reason to key on the IP address for billing purposes. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voip terminations in Australia and New Zealand
G'day Craig, On Sat, 20 Mar 2004, Craig wrote: > Does anybody know of any organization providing reasonably priced voip call > terminations in Australia and New Zealand ?? > > Does anybody know of any reasonably priced DID providers in Australia and > New Zealand ??. http://www.oztell.com I just signed up with this mob for ADSL, and have connected to their OZphone VoIP service. They offer SIP termination into their internal VoIP network, as well as offering a pretty thorough "voice application gateway" service. DIDs in Sydney and Melbourne (other capitals coming, and I guess they've got an office in Auckland for a reason) are A$9 per month (last time I checked). AFAIK, you don't need to be on their ADSL to get the voice services. Hmmm... Selecting Telephone Services from the above web page tells us only about their calling card service... Try poking around on http://ozforums.oztralia.com and see whether you think they fit you -- they are probably fairly immature corporate-wise, but seem to have a clue on the technology side. Cheers, Vic Cross ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Qualify statement
Senad Jordanovic wrote: Does anyone know if "qualify=XXX" should be used ONLY for user agents behind NAT. No, you can use it to qualify any address. Qualification means that Asterisk regurlarly sends SIP messages with the OPTION method and the UA answers. We clock the time and if the client takes too much time to answer or doesn't answer, it's set as UNREACHABLE until next time it answers. Calls will not be placed to unreachable peers. I tried to use it on a devices (ATA 186 and SPA 2000) on a public IP, and * goes to segmentation fault every time it starts. Does it crash even if you remove Qualify= from sip.conf? TO help you we need to get more information, a core file and your configuration. Check how to debug asterisk http://www.voip-info.org/wiki-Asterisk+debugging Report on the bug tracker http://bugs.digium.com If it is meant to be used just behind NAT fine, but what and how does * monitor user agent status? Even if it was, Asterisk shouldn't crash. Never ever. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Speex
Daniel Bichara wrote: WipeOut wrote: Carlos Chavez wrote: I have been trying out Asterisk with the speex codec with X-lite as a client. I applied the REG patch on my windows machine that is recommended in Voip-info.org. Every time I make a call I get the following error: codec_speex.c:167 speextolin_framein: Out of buffer space If I do not hang up before 30 seconds, my machine then slows down and it can take up to 10 minutes to shut down. Is speex worth the trouble? My personal opinion is that you would be better off using GSM or iLBC.. I don't think Speex has any advantage over these codecs and is always a PITA.. Sorry but I disagree. I am using SPEEX and voice quality is much better than GSM and it consumes less bandwidth. Using Speex and Linux is just a "make; make install". Although, we MUST encourage OpenSource initiatives or we will pay Licenses forever. Take a look at G.723 or G.729, for example. ITU released this protocols many years ago and we still have to pay royalties. About this kind of license there is another thread talking about... Daniel Agreed, speex is fine when going * to *.. UA to * over speex is usually a nightmare, and the number of UA's with speex support are very few.. Maybe if there was more support for it in the industry it would be worth using.. We will just have to wait and see what the future holds.. Anyway personal favorite is iLBC but there is not much support in the UA's for that either.. :( So for now its GSM or G.711 from UA to *.. What a choice!! Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Qualify statement
>> >> I tried to use it on a devices (ATA 186 and SPA 2000) on a public IP, >> and * goes to segmentation fault every time it starts. > > Does it crash even if you remove Qualify= from sip.conf? No it does not... Only when: Host=dynamic OR host=$PUBLIC IP AND qualify=YES > > TO help you we need to get more information, a core file and your > configuration. > > Check how to debug asterisk > http://www.voip-info.org/wiki-Asterisk+debugging > Report on the bug tracker http://bugs.digium.com > I did just that... :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voip terminations in Australia and New Zealand
Hi Craig, Packet8 doesn't allow asterisk terminations, you have to use their TA though I haven't looked yet I sure someone must have worked out a way to fake the info provided by TA. Costs $50 a month for unlimited calls into the USA, Australia and about 6 asian countries. BTW if you type "source" into their signup page it will knock $20 off the signup fee. I've been using their service from Sydney, Australia to make USA and Aust terminating calls for about 4 days and works great. The other alternative is I understand Commindico will be approving the Asterisk box for connection to their network however they have a $250pm minimum spend. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Sent: Saturday, 20 March 2004 9:45 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] voip terminations in Australia and New Zealand Greetings from downunder, Does anybody know of any organization providing reasonably priced voip call terminations in Australia and New Zealand ?? Does anybody know of any reasonably priced DID providers in Australia and New Zealand ??. Please feel free to contact me off list. cr ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] LipZ4 Sip Soft Phone
Thanks a lot I might give it a try. Any specific instructions for running it with asterisk? AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Important: The Asterisk Mailing list (new subject)
> EVERY other mailing list I use sets Reply-To to the list address. If the > asterisk lists change, then I'll be increasing the chance of so-called > "embarrassing gaffes" by not remembering how the list I'm posting to this > minute operates. Besides, the times I've seen such gaffes from mailing > list users, it serves to remind that user to take more care when directing > messages (yes, including when I've done it myself). That's an interesting data point Vic, thanks. I just did a quick survey of the multitude of mailing lists I'm on (a lot of technical discussion forums, like IETF etc), and NONE of them implement Reply-to munging. Perhaps we run in different circles ;-) Anyway, I'll let this thread die now. Thanks for your feedback. -Darren -- Darren Nickerson Senior Sales & Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Important: The Asterisk Mailing list (newsubject)
> You give too much credit to people, indeed. I cannot say about this list, > but most lists I use have high corporate populations, where the users > *have* to use mailers like Outlook or (cringe) Notes. Outlook and Outlook express implement Reply, and Reply All, which works well without needing reply-to munging. > > Some very compelling arguments are very clearly explained in the essay I > > pointed you to, and replying with "use your mail filters or unsubscribe" is > > not a very productive response (even if it COULD fix the problem). > > Nor is "List owner, action my request or resign" ;-) I agree ... that was very much tongue in cheek, and the smiley was there. > I did read the article you pointed to, and neither you or the article's > author have convinced me that this is a good thing. You don't have to agree with us. Thanks for reading it, at least! > When the entire world is using My Favourite Mailer, let's talk again. > Until then, let's stick with what works for -- and is understood by -- the > vast majority of people. While I don't agree that the convention of this list reflects the vast majority of well-run mailing lists, and I contend that it is counter to what is understood by the majority, I am happy to let the thread die. -Darren -- Darren Nickerson Senior Sales & Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AS5300 Firmware and H323 configuration
The AS5300 will NOT work as a gatekeeper. None of the IOS images support gatekeeper or IP-IP gateway functionality. It will NOT do IP to IP... it will do T1/E1 to IP or IP to T1/E1. The AS5300 will accept traffic from the VIP-400's but will not be able to forward them except to it's T1/E1 ports( IE: TDM to PSTN). If you want to use the AS5300, I would suggest that you get IOS 12.2(15)T or greater. Don't forget that you also need to obtain VCWare (for the DSPs) that is compatible with the IOS image (v10.2x or higher for the 12.2 or v10.3x for 12.3). good luck, Derek Bruce Calgary Telecom [EMAIL PROTECTED] - Original Message - From: "Christian Hoffmeyer" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, March 19, 2004 11:39 PM Subject: [Asterisk-Users] AS5300 Firmware and H323 configuration > I have a customer who wants to use an AS5300 - VOIP to trunk to Planet > VIP-400's H323 and also to trunk into their international long distance > provider with H323. > > Here's my problem. My client has an AS5300 with 4 e1's and it's never been > used. I know nothing about configuring these devices and am looking for > someone to make an image recommendation so we can do SIP and H323 and have > it operate as a client and gatekeeper. I also would like someone with > experience with these devices configure it to work. It would really save me > a lot of time. > > Any help will be greatly appreciated! > > Christian Hoffmeyer > YottaDot Solutions > Huntsville, AL > > (w) 256.859.4508 > (c)256.655.0321 > (iax) 700.859.4508 > > Ask me about Asterisk > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Speex
At 12:41 AM 3/20/2004 -0600, Carlos Chavez wrote: I have been trying out Asterisk with the speex codec with X-lite as a client. I applied the REG patch on my windows machine that is recommended in Voip-info.org. Every time I make a call I get the following error: codec_speex.c:167 speextolin_framein: Out of buffer space If I do not hang up before 30 seconds, my machine then slows down and it can take up to 10 minutes to shut down. Is speex worth the trouble? I'm using Asterisk CVS-03/10/04-17:49:23 with Speex 1.0.3 on RedHat Linux 8.0. I get exactly the same error. I am using Asterisk CVS-03/17/04-20:59:14 on Fedora Core 1 with Speex version 1.0.3. My ears say Speex is worth the trouble. One of my other careers is audio engineering; as a result, I'm very sensitive to compression artifacts. To me, if you can afford a little bit of extra bandwidth for Speex, it sounds a *lot* better than gsm. (And IMHO gsm sounds a little bit better than G.729.) I can't stand to listen to iLBC for more than 30 seconds. However, I'm just getting started with *, and digging into the code to find out what's wrong here has not yet risen to the top of my priority list. Until it does, I'm using gsm. -- Carlos Chavez Computer Engineer, CCNA Corporativo Lacer S.A. de C.V. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Basic authentication
How can I settup a way for Asterisk doesn´t make any use of DIGEST AUTHENTICATION method? I don t want ASTERISK to check out any username or password of my users. Thank you Joao Carlos Moura ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Basic authentication
Try adding 'insecure=yes' in sip.conf. Regards, Gus - Original Message - From: "Joao Carlos Moura" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, March 20, 2004 12:02 PM Subject: [Asterisk-Users] Basic authentication > How can I settup a way for Asterisk doesn´t make any use of DIGEST > AUTHENTICATION method? > I don t want ASTERISK to check out any username or password of my users. > > > Thank you > > Joao Carlos Moura > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problems with FWD solved maybe?
OK, I think I have an idea as to why it get the private number error. I do have externip=63.88.139.198 but its not being passed over to FWD. I think the problem is that I don't use the localnet and localmask statements. On my network I have an old 38.349.233.0/24 series that we used to use back when my predecessor didn't know squat about networking. Problem is that this number range is now so entrenched into my network that I couldn't change it if I wanted to. I have tried turning on the localnet/localmask but my 38 series sip clients get kicked off instantly. I'm guessing that the externip thing is starting to work which in turn is why I think this is my FWD problem/solution. Is there a way of defining more than one localnet and localmask? Perhaps something like; localnet=38.249.233.0,192.168.0,0 localmask=255.255.255.0,255.255.255.0 Folks? -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with Vegastream 50 BRI
Here is a sip log from my vegastream 50BRI to my asterisk box and i can't figure out why the call doesn't go trough ... sip.conf extract : [gw001] type=friend host=dynamic defaultip=192.168.0.12 nat=no dtmfmode=rfc2833 canreinvite=yes qualify=no context=tlsgw extensions.conf extract (from the contact [tlsgw]) : exten => 57228047,Dial(SIP/cs001,40,tr) ... Sip read: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.12:5060;branch=z9hG4bK-vega1-000A-0001-0002-1A36AF7D From: 478758923 ;tag=-0002-3A81BAD9 To: Max-Forwards: 70 Call-ID: [EMAIL PROTECTED] CSeq: 83606 INVITE Contact: Supported: replaces Allow: INVITE,ACK,BYE,CANCEL,INFO,NOTIFY,OPTIONS,REFER Accept-Language: en Content-Type: application/sdp Remote-Party-ID: 478758923 ;party=calling;screen=no;privacy=off Content-Length: 178 v=0 o=Vega50 3 1 IN IP4 192.168.0.12 s=Sip Call t=0 0 m=audio 10004 RTP/AVP 8 0 18 c=IN IP4 192.168.0.12 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 14 headers, 9 lines Using latest request as basis request Sending to 192.168.0.12 : 5060 (non-NAT) Found audio format ALAW Found audio format UNKN Found audio format UNKN Found description format PCMA Found description format PCMU Found description format G729 Capabilities: us - 524302, them - 268/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 Looking for 57228047 in sip Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.12:5060;branch=z9hG4bK-vega1-000A-0001-0002-1A36AF7D From: 478758923 ;tag=-0002-3A81BAD9 To: ;tag=as1fa83a23 Call-ID: [EMAIL PROTECTED] CSeq: 83606 INVITE ser-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.0.12:5060 dkmapbx*CLI> Sip read: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.12:5060;branch=z9hG4bK-vega1-000A-0001-0002-1A36AF7D From: 478758923 ;tag=-0002-3A81BAD9 To: Max-Forwards: 70 Call-ID: [EMAIL PROTECTED] CSeq: 83606 ACK Contact: Content-Length: 0 9 headers, 0 lines DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Message waiting indicators
A post in 2002 refered to Mike Sandman as a source for inexpensive (cheap) message waiting indicators. I called Mike but he doesn't know what Asterisk is (!) and wants to know what type of phone system I have or what protocol it uses so that he can send me a compatible indicator. I tried these acronyms on him: ADSI, MDMF, SDMF but he doesn't recognize them. What can I tell him so that I can order the right part? Or which popular switch is Asterisk compatible with? The post to which I refer: http://www.marko.net/asterisk/archives/0209/0588.html The style I'm interested in: Modular #WAL2D ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Important: The Asterisk Mailing list (newsubject)
[EMAIL PROTECTED] wrote on 03/20/2004 02:58:21 AM: > You give too much credit to people, indeed. I cannot say about this list, > but most lists I use have high corporate populations, where the users > *have* to use mailers like Outlook or (cringe) Notes. For mailing list > admins to expect users of these mailers to try and find the functions > referred to in the article is ludicrous (and yes, I know I just said in a > prior note "use the function of your mailer", but I was referring to the > standard Reply function -- if you have ever tried to use a mailing list > with Lotus Notes you will bless the list admin who maintains the > status-quo and munges "Reply-to"). It's odd: I use Lotus Notes, and while I prefer the current Reply-to action, having to click the Reply To All button right next to the Reply To button is not exactly a hardship... I do miss the way my old mailer (PM Mail 2000) prompted to either Reply to one or all, but Notes doesn't exactly make it hard.. Way off-topic, I know, but I had to defend Notes! :) Tim Massey ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need an example of using the directory command
Does someone please have a sample that shows how to use the directory command in extensions.conf? Thanks! Paul Mahler mail:[EMAIL PROTECTED] phone: 650.207.9855 fax: 877.408.0105
[Asterisk-Users] Just a question
Hello, Did anybody make Adtran TSU600 work with T100P? I cannot find anything in archives. I want to buy AdtranTSU600 and T100P but I am not sure if this is going to work. Bart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Need an example of using the directory command
Go on www.voip-info.org an search for IVR examples ... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paul Mahler Sent: Saturday, March 20, 2004 5:35 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Need an example of using the directory command Does someone please have a sample that shows how to use the directory command in extensions.conf? Thanks! Paul Mahler mail:[EMAIL PROTECTED] phone: 650.207.9855 fax: 877.408.0105 DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK BT caller ID revisted
Hi all, Does anyone know the procedure for adding a serial output to a cheap caller display unit. If I can find a way of doing this then I'm sure there will be away for linux to take the CallerID info, write it to a file, * to open that file an read the number from it. TIA Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UK BT caller ID revisted
John Lawrence wrote >Hi all, >Does anyone know the procedure for adding a serial output to a cheap caller >display unit. If I can find a way of doing this then I'm sure there will be >away for linux to take the CallerID info, write it to a file, * to open that >file an read the number from it. >TIA >Jon I am going to a Radio Rally tommorow and I will buy a couple of stand alone Caller Display units to strip and get a serial output from them. Watch this space. Regards Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Just a question
Come on, man! Take a look at all of the wonderful resources available before asking questions. http://www.voip-info.org is your friend. Start there, and take a few days to read over everything. Then you will find this: http://www.voip-info.org/wiki-Asterisk+Hardware. The mailing list is a wonderful resource, but you have to try and help yourself first. Otherwise you'll just get ignored. Sean -Original Message- From: Bartosz Jozwiak [mailto:[EMAIL PROTECTED] Sent: Saturday, March 20, 2004 11:38 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Just a question Hello, Did anybody make Adtran TSU600 work with T100P? I cannot find anything in archives. I want to buy AdtranTSU600 and T100P but I am not sure if this is going to work. Bart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: UK BT caller ID revisted
> Does anyone know the procedure for adding a serial output to a cheap caller > display unit. If I can find a way of doing this then I'm sure there will be > away for linux to take the CallerID info, write it to a file, * to open that > file an read the number from it. Sorry I never got round to answering your previous request - I will write up what I have done with asterisk in due course. In the meantime, there's some good info on hacking CID boxes here: http://www.automatedhome.co.uk/modules.php?name=News&file=print&sid=1207 Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Integration with Evolution.
Everyone-- I filed a bug with ximian against Evolution, in the form of an enhancement request for integration with Asterisk. Have a look: http://bugzilla.ximian.com/show_bug.cgi?id=55854 It wouldn't hurt to "pile on"! Please, add your own comments, suggestions, disagreements and clarifications. Or just plain vote for/against it there. A couple hundred comments might signal some real interest in such a feature? I did a bug search for Asterisk, and didn't find any reference to Asterisk at all there. What a shame, I thought. I outlined a "minimum" scenario I'd like to see implemented, and perhaps you all might violently disagree. I didn't mention anything about using databases of users, and how that might be used in the process of making a call. That's for someone with a clearer vision of how that might work. murf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Just a question
Useful Asterisk Docs (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the "Unofficial Links") and http://www.voip-info.org/wiki-Asterisk and http://www.fnords.org/~eric/asterisk/ (my site) and http://asteriskdocs.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Integration with Evolution.
On Sat, 2004-03-20 at 13:12, Steve Murphy wrote: > I filed a bug with ximian against Evolution, in the form of an > enhancement request for integration with Asterisk. > > Have a look: http://bugzilla.ximian.com/show_bug.cgi?id=55854 Clever. However rather than needing to actually know what the channel name is, why not use chan_local? That way the dial plan will figure out what channel to use, just like when you dial a phone connected to Asterisk. chan_local is VERY, VERY useful. You can do things like Dial(Local/95551212) and Asterisk will route the call just as though you daled the number directly. This way AGI scripts, .call files and stuff like that don't need to know what interfaces you have or be told what interface to route calls matching specific pattern should go to. You just Dial(Local/blah) and your dial plan will route the call. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: UK BT caller ID revisted
On Saturday 20 March 2004 18:51, Patrick Lidstone (Personal E-mail) wrote: > > In the meantime, there's some good info on hacking CID boxes here: > http://www.automatedhome.co.uk/modules.php?name=News&file=print&sid=1207 > Cheers. That'll do the job. No to rip apart a few Caller ID units I've got lying around and fit out what chips are in them :) Jkn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Integration with Evolution.
> Have a look: http://bugzilla.ximian.com/show_bug.cgi?id=55854 Since you don't want Jane magicaly making John dial Claire, there would need to be individule login authentication that would only allow Jane to dial and connect her channel. So, this is not just Evolution hacking. Asterisk needs a DB + API for enforcing triple A security. (Authentication, Authorization, Accounting) -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Basic authentication
Joao Carlos Moura wrote: How can I settup a way for Asterisk doesn´t make any use of DIGEST AUTHENTICATION method? I don t want ASTERISK to check out any username or password of my users. Set no secret in sip.conf our use autocreatepeer /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Basic authentication
CW_ASN wrote: Try adding 'insecure=yes' in sip.conf. insecure=yes doesn't help in regards to authentication, or? Please explain more. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with Vegastream 50 BRI
Michael Devenijn wrote: Here is a sip log from my vegastream 50BRI to my asterisk box and i can't figure out why the call doesn't go trough ... sip.conf extract : [gw001] type=friend host=dynamic defaultip=192.168.0.12 nat=no dtmfmode=rfc2833 canreinvite=yes qualify=no context=tlsgw extensions.conf extract (from the contact [tlsgw]) : exten => 57228047,Dial(SIP/cs001,40,tr) ... Looking for 57228047 in sip Transmitting (no NAT): SIP/2.0 404 Not Found Asterisk isn't looking in the context tlsgw, for some reason it checks in the "sip" context. If this is your default context, Asterisk doesn't connect the incoming call with gw001. You have host=dynamic - is the gateway registred with Asterisk at all? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Packet8
Hey all!! I know this issue has been address before, but I can not find someone who has the answer. I am trying to get my * server to authenticate directly to packet8. I was very close to them actually giving me the information and possibly using them for my SIP -> PSTN termination, but that fell through. They didn't think they had enough bandwidth. (LOL) There are a few questions that I would like to know answers to. -> Does anyone currently have a working implementation in which asterisk authenticates to pakcet8? (Making and receiving calls via packet8) If so, could you please share? -> Does anyone know how to get into the advanced configuration on the DTA-310? Packet8 seems to have password protected it. -> Has anyone been able to connect to Vonage? Thanks in advance for your help!! Zac - Zac Amsler <[EMAIL PROTECTED]> Computer Consulting Group ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 transfers - it's great!!!!
I have a network of IAX servers connecting to each other. I just realized that IAX does some clever magic by itself. Let me explain: --- Let's say you have three servers: A, B and Q A calls B with IAX2 B connects the call to Q with IAX2 B realizes that A can speak directly with Q and transfers the call so A speaks directly to Q without any further involvement of B. So you can easily build a mesh network with IAX2 where every call takes the optimal route without you bothering with it. Sometimes stuff just works. - If this is documented on the Wiki, I must spend time reading the Wiki again, because I wasn't aware of this cleverness in IAX2. It was a very pleasant surprise. And yes, there's a config in iax.conf so you can turn it off if you for some reason want to bother B with staying in the middle of the call. Good luck IAX'ing :-) /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] can't get the full callerid php/agi
Hi
folks,
I need some help
from php/agi experts out there;
I am having
difficulties in extracting the callerid number from php. My script is given
below;
#!/usr/local/bin/php -q
//environment dump
ob_implicit_flush(true);set_time_limit(0);
$err=fopen("php://stderr","w");$in =
fopen("php://stdin","r");while (!feof($in)) {$temp =
str_replace("\n","",fgets($in,4096));
echo "VERBOSE \"$temp\"\n";
if (($temp == "") || ($temp == "\n"))
{break;}}
?>
And the Response
is:
getenv.agi:
agi_request: getenv.agigetenv.agi: agi_channel: SIP/-081524c0getenv.agi:
agi_language: engetenv.agi: agi_type: SIPgetenv.agi: agi_uniqueid:
1079819757.97getenv.agi: agi_callerid: Sathya
<=getenv.agi: agi_dnid: unknowngetenv.agi:
agi_rdnis: unknowngetenv.agi: agi_context: default-ingetenv.agi:
agi_extension: 91234getenv.agi: agi_priority: 1getenv.agi: agi_enhanced:
0.0getenv.agi: agi_accountcode:
Actual caller ID is
; "Sathya Weerasooriya"<1001>, but what I get from this PHP is only first
part.
When I run the
example test script in agi-bin directory, I get the correct
callerid.
I am using PHP
4.3.5CR xx.
Any help is
appreciated.
Sathya
Re: [Asterisk-Users] Basic authentication
Yes, you're right. * sends 407 anyway. I'll try your method. Gus - Original Message - From: "Olle E. Johansson" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, March 20, 2004 6:56 PM Subject: Re: [Asterisk-Users] Basic authentication > CW_ASN wrote: > > > Try adding 'insecure=yes' in sip.conf. > insecure=yes doesn't help in regards to authentication, or? > Please explain more. > > /O > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Use of Alert_Info with C7960?
Using Asterisk CVS-03/20/04-11:54:56 with 7960 v6.2 and playing around
with distinctive ringing, trying to make it work. Extensions.conf looks like:
exten => 3010,1,SetVar(ALERT_INFO=3) ; selects Ringer
exten => 3010,2,Dial(SIP/3010,15)
exten => 3010,3,Voicemail2(u3010)
exten => 3010,102,Voicemail2(b3010)
exten => 3010,103,Hangup
Calling that extension, the CLI indicates:
-- Executing SetVar("SIP/3002-39d1", "ALERT_INFO=3") in new stack
-- Executing Dial("SIP/3002-39d1", "SIP/3010|15") in new stack
-- Called 3010
-- SIP/3010-f848 is ringing
On the phone, Settings/Ring Type indicates: Chirp 1, Chirp 2, Old Style
and Synth Low. The first three choices produce different ringing sounds
when selected from the display.
I expected Alert_Info=3 to cause the C7960 to ring with the Old Style
ringer, but it doesn't and setting it to 2 or 3 doesn't make any difference.
Am I doing something wrong?
Rich
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Re: [Asterisk-Users] Use of Alert_Info with C7960?
- Original Message - From: "Rich Adamson" <[EMAIL PROTECTED]> To: "Asterisk-a-users-list" <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Use of Alert_Info with C7960? > Using Asterisk CVS-03/20/04-11:54:56 with 7960 v6.2 and playing around > with distinctive ringing, trying to make it work. Extensions.conf looks like: > > exten => 3010,1,SetVar(ALERT_INFO=3) ; selects Ringer > exten => 3010,2,Dial(SIP/3010,15) > exten => 3010,3,Voicemail2(u3010) > exten => 3010,102,Voicemail2(b3010) > exten => 3010,103,Hangup > > On the phone, Settings/Ring Type indicates: Chirp 1, Chirp 2, Old Style > and Synth Low. The first three choices produce different ringing sounds > when selected from the display. > > I expected Alert_Info=3 to cause the C7960 to ring with the Old Style > ringer, but it doesn't and setting it to 2 or 3 doesn't make any difference. > > Am I doing something wrong? > > Rich A search on the mailing list returned this: http://lists.digium.com/pipermail/asterisk-users/2004-February/036747.html Try using: exten => 555,1,SetVar(ALERT_INFO=)Best regards,Nicolas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Use of Alert_Info with C7960?
> - Original Message - > From: "Rich Adamson" <[EMAIL PROTECTED]> > To: "Asterisk-a-users-list" <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] Use of Alert_Info with C7960? > > > > Using Asterisk CVS-03/20/04-11:54:56 with 7960 v6.2 and playing around > > with distinctive ringing, trying to make it work. Extensions.conf looks > like: > > > > exten => 3010,1,SetVar(ALERT_INFO=3) ; selects Ringer > > exten => 3010,2,Dial(SIP/3010,15) > > exten => 3010,3,Voicemail2(u3010) > > exten => 3010,102,Voicemail2(b3010) > > exten => 3010,103,Hangup > > > > On the phone, Settings/Ring Type indicates: Chirp 1, Chirp 2, Old Style > > and Synth Low. The first three choices produce different ringing sounds > > when selected from the display. > > > > I expected Alert_Info=3 to cause the C7960 to ring with the Old Style > > ringer, but it doesn't and setting it to 2 or 3 doesn't make any > difference. > > > > Am I doing something wrong? > > > > A search on the mailing list returned this: > > http://lists.digium.com/pipermail/asterisk-users/2004-February/036747.html > > Try using: > exten => 555,1,SetVar(ALERT_INFO=)Best regards,Nicolas The wiki indicates Alert_Info can be set to a number, and implies that number is the ringer type listed on the phone. Is there a way to select one of the internal ringer types via Alert_Info? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2 transfers - it's great!!!!
> And yes, there's a config in iax.conf so you can turn it off if you > for some reason want to bother B with staying in the middle of the > call. Yap. Great stuff :) Just so everyone knows the config is: notransfer=yes It would be good to know what happens with cdr records and call control? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Use of Alert_Info with C7960?
- Original Message - From: "Rich Adamson" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, March 20, 2004 8:55 PM Subject: Re: [Asterisk-Users] Use of Alert_Info with C7960? > > > On the phone, Settings/Ring Type indicates: Chirp 1, Chirp 2, Old Style > > > and Synth Low. The first three choices produce different ringing sounds > > > when selected from the display. > > > > > > I expected Alert_Info=3 to cause the C7960 to ring with the Old Style > > > ringer, but it doesn't and setting it to 2 or 3 doesn't make any > > difference. > > > > > > Am I doing something wrong? > > > > > > > A search on the mailing list returned this: > > > > http://lists.digium.com/pipermail/asterisk-users/2004-February/036747.html > > > > Try using: > > exten => 555,1,SetVar(ALERT_INFO=)Best regards,Nicolas > > The wiki indicates Alert_Info can be set to a number, and implies that > number is the ringer type listed on the phone. Is there a way to select > one of the internal ringer types via Alert_Info? > Hi Rich, The different ring tones are features of the sip phone/adapter. I dont have any Ciscos, but I do have the Sipura SPA-2000. I'm using ALERT_INFO to set distinctive rings and it works great. But the name of the ringtone is different from the one I quoted for the Cisco: exten => 12,2,SetVar(ALERT_INFO=Bellcore-r3) Is the phone/adapter job to interpret the alert info and play the acording ring tone. If the phone expects Bellcore-dr3, you should send that. Best regards, Nicolas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T100P T1 problem (Avaya -> asterisk IVR)
Hello all, I'm having a problem with a T1 connection to a Avaya PBX (asterisk is an IVR). I could not get pri working and now I'm simply trying to get asterisk working with fxs_ks. Questions: 1. Is there anyway to troubleshoot or see what is being sent on the T1. zttool shows no errors, and the Avaya rings the t1 if wct1xxp is loaded, and gives busy if it is not loaded -- so I think something is being sent down the line, I just don't know what it is (asterisk -vc shows nothing) 2. Is there anyone with Avaya PBX -> asterisk experience on the list? I'm remote, but I can login -- I just don't know what commands can troubleshoot the connection. Config: zaptel.conf (relevant section) span=1,1,0,esf,b8zs fxsks=1-24 zapata.conf [channels] context = demo switchtype = national signalling = fxs_ks group = 1 channel => 1-24 Thanks for any time and help Jeb Campbell [EMAIL PROTECTED] Cell: 865-385-1437 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T100P T1 problem (Avaya -> asterisk IVR)
Check out : http://www.voip-info.org/wiki-Asterisk+Avaya Depending upon the card you are using in the Avaya, you should set it up as a tie trunk on the Avaya side. James- - Original Message - From: "Jeb Campbell" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, March 20, 2004 7:36 PM Subject: [Asterisk-Users] T100P T1 problem (Avaya -> asterisk IVR) > Hello all, > I'm having a problem with a T1 connection to a Avaya PBX (asterisk is > an IVR). > I could not get pri working and now I'm simply trying to get asterisk > working with fxs_ks. > > Questions: > 1. Is there anyway to troubleshoot or see what is being sent on the T1. > zttool shows no errors, and the Avaya rings the t1 if wct1xxp is > loaded, and gives > busy if it is not loaded -- so I think something is being sent down > the line, I just > don't know what it is (asterisk -vc shows nothing) > > 2. Is there anyone with Avaya PBX -> asterisk experience on the list? > I'm remote, but I can > login -- I just don't know what commands can troubleshoot the > connection. > > Config: > zaptel.conf (relevant section) > span=1,1,0,esf,b8zs > fxsks=1-24 > > zapata.conf > [channels] > context = demo > switchtype = national > signalling = fxs_ks > group = 1 > channel => 1-24 > > Thanks for any time and help > > Jeb Campbell > [EMAIL PROTECTED] > Cell: 865-385-1437 > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Use of Alert_Info with C7960?
Unfortunately even though it would seem the phone should support the ability
to play custom ring tones, at present it only supports the internal tones
which are:-
Bellcore-BusyVerify
Bellcore-Stutter
Bellcore-MsgWaiting
Bellcore-dr1
Bellcore-dr2
Bellcore-dr3
Bellcore-dr4
Bellcore-dr5
--- Extract from the SIP 6.0 Firmware Release Notes ---
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp79
60/addprot/sip/relnote/phnrn60s.htm
In RFC-3261, the Alert-Info header is specified as a URL. When the
Alert-Info header is received, the phone downloads the file from the URL and
plays it as the alternate ring tone. This release does not support any
external ringers. Only the tones and ring patterns that are already internal
to the phone can be selected and played as an alternate ring tone.
In this release, the Alert-Info header consists of a name of an internal
tone or ringing pattern that can be played, as shown in the following
example:
Alert-Info:
There is no need to add a file extension (.au, .wav) to these names because
the names are internal to the phone. When an Alert-Info header is received,
the software scans the list of known tones and ringing patterns to find a
match. If the software finds a match, the phone plays that tone or ringing
pattern. If the software does not find a match, the phone plays the alert
ringing pattern as it does today.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Rich Adamson
> Sent: Sunday, 21 March 2004 9:28 AM
> To: Asterisk-a-users-list
> Subject: [Asterisk-Users] Use of Alert_Info with C7960?
>
> Using Asterisk CVS-03/20/04-11:54:56 with 7960 v6.2 and playing around
> with distinctive ringing, trying to make it work. Extensions.conf looks
> like:
>
> exten => 3010,1,SetVar(ALERT_INFO=3) ; selects Ringer
> exten => 3010,2,Dial(SIP/3010,15)
> exten => 3010,3,Voicemail2(u3010)
> exten => 3010,102,Voicemail2(b3010)
> exten => 3010,103,Hangup
>
> Calling that extension, the CLI indicates:
> -- Executing SetVar("SIP/3002-39d1", "ALERT_INFO=3") in new stack
> -- Executing Dial("SIP/3002-39d1", "SIP/3010|15") in new stack
> -- Called 3010
> -- SIP/3010-f848 is ringing
>
> On the phone, Settings/Ring Type indicates: Chirp 1, Chirp 2, Old Style
> and Synth Low. The first three choices produce different ringing sounds
> when selected from the display.
>
> I expected Alert_Info=3 to cause the C7960 to ring with the Old Style
> ringer, but it doesn't and setting it to 2 or 3 doesn't make any
> difference.
>
> Am I doing something wrong?
>
> Rich
>
>
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RE: [Asterisk-Users] Use of Alert_Info with C7960?
Then what's the point of being able to upload custom ring tones?
(as shown in
http://www.loligo.com/asterisk/Cisco/79xx/current/RINGLIST.DAT )
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Christopher Lee
> Sent: Saturday, March 20, 2004 8:50 PM
> To: Asterisk Users
> Subject: RE: [Asterisk-Users] Use of Alert_Info with C7960?
>
> Unfortunately even though it would seem the phone should
> support the ability to play custom ring tones, at present it
> only supports the internal tones which are:-
>
> Bellcore-BusyVerify
> Bellcore-Stutter
> Bellcore-MsgWaiting
> Bellcore-dr1
> Bellcore-dr2
> Bellcore-dr3
> Bellcore-dr4
> Bellcore-dr5
>
> --- Extract from the SIP 6.0 Firmware Release Notes ---
> http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon
> /english/ipp79
> 60/addprot/sip/relnote/phnrn60s.htm
>
> In RFC-3261, the Alert-Info header is specified as a URL.
> When the Alert-Info header is received, the phone downloads
> the file from the URL and plays it as the alternate ring
> tone. This release does not support any external ringers.
> Only the tones and ring patterns that are already internal to
> the phone can be selected and played as an alternate ring tone.
>
> In this release, the Alert-Info header consists of a name of
> an internal tone or ringing pattern that can be played, as
> shown in the following
> example:
>
> Alert-Info:
>
> There is no need to add a file extension (.au, .wav) to these
> names because the names are internal to the phone. When an
> Alert-Info header is received, the software scans the list of
> known tones and ringing patterns to find a match. If the
> software finds a match, the phone plays that tone or ringing
> pattern. If the software does not find a match, the phone
> plays the alert ringing pattern as it does today.
>
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Rich Adamson
> > Sent: Sunday, 21 March 2004 9:28 AM
> > To: Asterisk-a-users-list
> > Subject: [Asterisk-Users] Use of Alert_Info with C7960?
> >
> > Using Asterisk CVS-03/20/04-11:54:56 with 7960 v6.2 and
> playing around
> > with distinctive ringing, trying to make it work. Extensions.conf
> > looks
> > like:
> >
> > exten => 3010,1,SetVar(ALERT_INFO=3) ; selects Ringer exten =>
> > 3010,2,Dial(SIP/3010,15) exten => 3010,3,Voicemail2(u3010) exten =>
> > 3010,102,Voicemail2(b3010) exten => 3010,103,Hangup
> >
> > Calling that extension, the CLI indicates:
> > -- Executing SetVar("SIP/3002-39d1", "ALERT_INFO=3") in
> new stack
> > -- Executing Dial("SIP/3002-39d1", "SIP/3010|15") in new stack
> > -- Called 3010
> > -- SIP/3010-f848 is ringing
> >
> > On the phone, Settings/Ring Type indicates: Chirp 1, Chirp 2, Old
> > Style and Synth Low. The first three choices produce
> different ringing
> > sounds when selected from the display.
> >
> > I expected Alert_Info=3 to cause the C7960 to ring with the
> Old Style
> > ringer, but it doesn't and setting it to 2 or 3 doesn't make any
> > difference.
> >
> > Am I doing something wrong?
> >
> > Rich
> >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> ___
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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RE: [Asterisk-Users] Use of Alert_Info with C7960?
The custom ring tones are selectable through the Ring Type option in the
Settings menu. When the phone rings, it will play that custom ring tone.
Perhaps it's a memory limitation or an issue with the way Cisco are
implementing the SIP firmware as to why you can't select a custom ring tone,
you'd be better asking a Cisco engineer (open a TAC case)
The phone will only download the currently selected custom ring tone
(defined in RINGLIST.DAT) from the TFTP server each time it boots, or when
you select another custom ring tone from the Ring Type option.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Matthew Marlowe
> Sent: Sunday, 21 March 2004 11:53 AM
> To: Asterisk Users
> Subject: RE: [Asterisk-Users] Use of Alert_Info with C7960?
>
> Then what's the point of being able to upload custom ring tones?
>
> (as shown in
> http://www.loligo.com/asterisk/Cisco/79xx/current/RINGLIST.DAT )
>
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > Christopher Lee
> > Sent: Saturday, March 20, 2004 8:50 PM
> > To: Asterisk Users
> > Subject: RE: [Asterisk-Users] Use of Alert_Info with C7960?
> >
> > Unfortunately even though it would seem the phone should
> > support the ability to play custom ring tones, at present it
> > only supports the internal tones which are:-
> >
> > Bellcore-BusyVerify
> > Bellcore-Stutter
> > Bellcore-MsgWaiting
> > Bellcore-dr1
> > Bellcore-dr2
> > Bellcore-dr3
> > Bellcore-dr4
> > Bellcore-dr5
> >
> > --- Extract from the SIP 6.0 Firmware Release Notes ---
> > http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon
> > /english/ipp79
> > 60/addprot/sip/relnote/phnrn60s.htm
> >
> > In RFC-3261, the Alert-Info header is specified as a URL.
> > When the Alert-Info header is received, the phone downloads
> > the file from the URL and plays it as the alternate ring
> > tone. This release does not support any external ringers.
> > Only the tones and ring patterns that are already internal to
> > the phone can be selected and played as an alternate ring tone.
> >
> > In this release, the Alert-Info header consists of a name of
> > an internal tone or ringing pattern that can be played, as
> > shown in the following
> > example:
> >
> > Alert-Info:
> >
> > There is no need to add a file extension (.au, .wav) to these
> > names because the names are internal to the phone. When an
> > Alert-Info header is received, the software scans the list of
> > known tones and ringing patterns to find a match. If the
> > software finds a match, the phone plays that tone or ringing
> > pattern. If the software does not find a match, the phone
> > plays the alert ringing pattern as it does today.
> >
> > > -Original Message-
> > > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > > [EMAIL PROTECTED] On Behalf Of Rich Adamson
> > > Sent: Sunday, 21 March 2004 9:28 AM
> > > To: Asterisk-a-users-list
> > > Subject: [Asterisk-Users] Use of Alert_Info with C7960?
> > >
> > > Using Asterisk CVS-03/20/04-11:54:56 with 7960 v6.2 and
> > playing around
> > > with distinctive ringing, trying to make it work. Extensions.conf
> > > looks
> > > like:
> > >
> > > exten => 3010,1,SetVar(ALERT_INFO=3) ; selects Ringer exten =>
> > > 3010,2,Dial(SIP/3010,15) exten => 3010,3,Voicemail2(u3010) exten =>
> > > 3010,102,Voicemail2(b3010) exten => 3010,103,Hangup
> > >
> > > Calling that extension, the CLI indicates:
> > > -- Executing SetVar("SIP/3002-39d1", "ALERT_INFO=3") in
> > new stack
> > > -- Executing Dial("SIP/3002-39d1", "SIP/3010|15") in new stack
> > > -- Called 3010
> > > -- SIP/3010-f848 is ringing
> > >
> > > On the phone, Settings/Ring Type indicates: Chirp 1, Chirp 2, Old
> > > Style and Synth Low. The first three choices produce
> > different ringing
> > > sounds when selected from the display.
> > >
> > > I expected Alert_Info=3 to cause the C7960 to ring with the
> > Old Style
> > > ringer, but it doesn't and setting it to 2 or 3 doesn't make any
> > > difference.
> > >
> > > Am I doing something wrong?
> > >
> > > Rich
> > >
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > To UNSUBSCRIBE or update options visit:
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> ___
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> [EMAIL PROTECTED]
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
_
RE: [Asterisk-Users] Use of Alert_Info with C7960?
That's right, you can still select a different ring tone for ALL lines.
Ok.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Christopher Lee
> Sent: Saturday, March 20, 2004 9:04 PM
> To: Asterisk Users
> Subject: RE: [Asterisk-Users] Use of Alert_Info with C7960?
>
> The custom ring tones are selectable through the Ring Type
> option in the Settings menu. When the phone rings, it will
> play that custom ring tone.
>
> Perhaps it's a memory limitation or an issue with the way
> Cisco are implementing the SIP firmware as to why you can't
> select a custom ring tone, you'd be better asking a Cisco
> engineer (open a TAC case)
>
> The phone will only download the currently selected custom
> ring tone (defined in RINGLIST.DAT) from the TFTP server each
> time it boots, or when you select another custom ring tone
> from the Ring Type option.
>
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Matthew Marlowe
> > Sent: Sunday, 21 March 2004 11:53 AM
> > To: Asterisk Users
> > Subject: RE: [Asterisk-Users] Use of Alert_Info with C7960?
> >
> > Then what's the point of being able to upload custom ring tones?
> >
> > (as shown in
> > http://www.loligo.com/asterisk/Cisco/79xx/current/RINGLIST.DAT )
> >
> > > -Original Message-
> > > From: [EMAIL PROTECTED]
> > > [mailto:[EMAIL PROTECTED] On Behalf Of
> > > Christopher Lee
> > > Sent: Saturday, March 20, 2004 8:50 PM
> > > To: Asterisk Users
> > > Subject: RE: [Asterisk-Users] Use of Alert_Info with C7960?
> > >
> > > Unfortunately even though it would seem the phone should
> support the
> > > ability to play custom ring tones, at present it only
> supports the
> > > internal tones which are:-
> > >
> > > Bellcore-BusyVerify
> > > Bellcore-Stutter
> > > Bellcore-MsgWaiting
> > > Bellcore-dr1
> > > Bellcore-dr2
> > > Bellcore-dr3
> > > Bellcore-dr4
> > > Bellcore-dr5
> > >
> > > --- Extract from the SIP 6.0 Firmware Release Notes ---
> > > http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon
> > > /english/ipp79
> > > 60/addprot/sip/relnote/phnrn60s.htm
> > >
> > > In RFC-3261, the Alert-Info header is specified as a URL.
> > > When the Alert-Info header is received, the phone
> downloads the file
> > > from the URL and plays it as the alternate ring tone.
> This release
> > > does not support any external ringers.
> > > Only the tones and ring patterns that are already internal to the
> > > phone can be selected and played as an alternate ring tone.
> > >
> > > In this release, the Alert-Info header consists of a name of an
> > > internal tone or ringing pattern that can be played, as
> shown in the
> > > following
> > > example:
> > >
> > > Alert-Info:
> > >
> > > There is no need to add a file extension (.au, .wav) to
> these names
> > > because the names are internal to the phone. When an Alert-Info
> > > header is received, the software scans the list of known
> tones and
> > > ringing patterns to find a match. If the software finds a
> match, the
> > > phone plays that tone or ringing pattern. If the software
> does not
> > > find a match, the phone plays the alert ringing pattern
> as it does
> > > today.
> > >
> > > > -Original Message-
> > > > From: [EMAIL PROTECTED]
> > > > [mailto:asterisk-users- [EMAIL PROTECTED] On
> Behalf Of Rich
> > > > Adamson
> > > > Sent: Sunday, 21 March 2004 9:28 AM
> > > > To: Asterisk-a-users-list
> > > > Subject: [Asterisk-Users] Use of Alert_Info with C7960?
> > > >
> > > > Using Asterisk CVS-03/20/04-11:54:56 with 7960 v6.2 and
> > > playing around
> > > > with distinctive ringing, trying to make it work.
> Extensions.conf
> > > > looks
> > > > like:
> > > >
> > > > exten => 3010,1,SetVar(ALERT_INFO=3) ; selects Ringer exten =>
> > > > 3010,2,Dial(SIP/3010,15) exten =>
> 3010,3,Voicemail2(u3010) exten
> > > > =>
> > > > 3010,102,Voicemail2(b3010) exten => 3010,103,Hangup
> > > >
> > > > Calling that extension, the CLI indicates:
> > > > -- Executing SetVar("SIP/3002-39d1", "ALERT_INFO=3") in
> > > new stack
> > > > -- Executing Dial("SIP/3002-39d1", "SIP/3010|15")
> in new stack
> > > > -- Called 3010
> > > > -- SIP/3010-f848 is ringing
> > > >
> > > > On the phone, Settings/Ring Type indicates: Chirp 1,
> Chirp 2, Old
> > > > Style and Synth Low. The first three choices produce
> > > different ringing
> > > > sounds when selected from the display.
> > > >
> > > > I expected Alert_Info=3 to cause the C7960 to ring with the
> > > Old Style
> > > > ringer, but it doesn't and setting it to 2 or 3 doesn't
> make any
> > > > difference.
> > > >
> > > > Am I doing something wrong?
> > > >
> > > > Rich
> > > >
> > > >
> > > > ___
> > > > Asterisk-Users mailing list
> > > > [EMAIL PROTECTED]
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > > To UNSUBSCRIBE
[Asterisk-Users] Snom 200
Greetings All I'm busy trying out my new snom 200(s) I have it connected and * CLI tells me registered 1) I pick up the handset and hear the dial tone 2) Dial and Ext, that says Date & Time (13) 3) * CLI scrolls that the call is connected and time is being spoken YET the handset is quite and silent? WHY ? Also if I dial for voicemailmain ext (8) * CLI says connected vm-login Yet again the handset it silent, ? What have I not configured? sip.conf has disallow = all allow = ulaw The snom 200 is set to G.711u codec What is wrong here, please anyone? Thanks in advance Barry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Store caller IP in CDR
From: "Olle E. Johansson" <[EMAIL PROTECTED]> snip > Check the CDRuserfield - it's a free field in the CDR you set in the > dialplan or from a script. How would you set the CDRuserfield from the dialplan exten => ? Thanks in advance B ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 200
Barry, I also just got a new snom 200. Still discovering features, but as a whole it is working fine. Please include the sip.conf entry for the phone you have .. Also, from your comments I assume that the snom 200 is on the same LAN as the [*] box? On the snom web interface, does it show that line 1 (which I assume you are using) is 'registered'? Willy - Original Message Follows - > Greetings All > > I'm busy trying out my new snom 200(s) > I have it connected and * CLI tells me registered > > 1) I pick up the handset and hear the dial tone > 2) Dial and Ext, that says Date & Time (13) > 3) * CLI scrolls that the call is connected and time is > being spoken >YET the handset is quite and silent? WHY ? > > > Also if I dial for voicemailmain ext (8) > * CLI says connected vm-login > Yet again the handset it silent, ? > > What have I not configured? > sip.conf has > disallow = all > allow = ulaw > > The snom 200 is set to G.711u codec > > What is wrong here, please anyone? > Thanks in advance > > Barry > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Need an example of using the directory command
Thanks! I was searching for the wrong thing. Paul Paul Mahler [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Devenijn Sent: Saturday, March 20, 2004 9:30 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Need an example of using the directory command Go on www.voip-info.org an search for IVR examples ... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paul Mahler Sent: Saturday, March 20, 2004 5:35 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Need an example of using the directory command Does someone please have a sample that shows how to use the directory command in extensions.conf? Thanks! Paul Mahler mail:[EMAIL PROTECTED] phone: 650.207.9855 fax: 877.408.0105 DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Use of Alert_Info with C7960?
The 7960 will absolutely play custom ringtones.
Paul Mahler
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christopher Lee
Sent: Saturday, March 20, 2004 5:50 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Use of Alert_Info with C7960?
Unfortunately even though it would seem the phone should support the ability
to play custom ring tones, at present it only supports the internal tones
which are:-
Bellcore-BusyVerify
Bellcore-Stutter
Bellcore-MsgWaiting
Bellcore-dr1
Bellcore-dr2
Bellcore-dr3
Bellcore-dr4
Bellcore-dr5
--- Extract from the SIP 6.0 Firmware Release Notes ---
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp79
60/addprot/sip/relnote/phnrn60s.htm
In RFC-3261, the Alert-Info header is specified as a URL. When the
Alert-Info header is received, the phone downloads the file from the URL and
plays it as the alternate ring tone. This release does not support any
external ringers. Only the tones and ring patterns that are already internal
to the phone can be selected and played as an alternate ring tone.
In this release, the Alert-Info header consists of a name of an internal
tone or ringing pattern that can be played, as shown in the following
example:
Alert-Info:
There is no need to add a file extension (.au, .wav) to these names because
the names are internal to the phone. When an Alert-Info header is received,
the software scans the list of known tones and ringing patterns to find a
match. If the software finds a match, the phone plays that tone or ringing
pattern. If the software does not find a match, the phone plays the alert
ringing pattern as it does today.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Rich Adamson
> Sent: Sunday, 21 March 2004 9:28 AM
> To: Asterisk-a-users-list
> Subject: [Asterisk-Users] Use of Alert_Info with C7960?
>
> Using Asterisk CVS-03/20/04-11:54:56 with 7960 v6.2 and playing around
> with distinctive ringing, trying to make it work. Extensions.conf
> looks
> like:
>
> exten => 3010,1,SetVar(ALERT_INFO=3) ; selects Ringer exten =>
> 3010,2,Dial(SIP/3010,15) exten => 3010,3,Voicemail2(u3010) exten =>
> 3010,102,Voicemail2(b3010) exten => 3010,103,Hangup
>
> Calling that extension, the CLI indicates:
> -- Executing SetVar("SIP/3002-39d1", "ALERT_INFO=3") in new stack
> -- Executing Dial("SIP/3002-39d1", "SIP/3010|15") in new stack
> -- Called 3010
> -- SIP/3010-f848 is ringing
>
> On the phone, Settings/Ring Type indicates: Chirp 1, Chirp 2, Old
> Style and Synth Low. The first three choices produce different ringing
> sounds when selected from the display.
>
> I expected Alert_Info=3 to cause the C7960 to ring with the Old Style
> ringer, but it doesn't and setting it to 2 or 3 doesn't make any
> difference.
>
> Am I doing something wrong?
>
> Rich
>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] Snom 200
From: <[EMAIL PROTECTED]> > Please include the sip.conf entry for the phone you have .. SIP Configuration for Asterisk ; [general] port = 5060 bindaddr = 192.168.0.15 externip = 24.73.215.62 localnet = 192.168.0.0 localmask = 255.255.255.0 tos = lowdelay disallow = all allow = ulaw allow = all context = INVALID [4403] type= friend username= 4403 secret = 1234 nat = yes host= dynamic context = toll-access accountcode = barry mailbox = 4403 [4401] type= friend username= 4401 secret = 1234 nat = yes host= dynamic context = local-access accountcode = mark mailbox = 4401 > Also, from your comments I assume that the snom 200 is on > the same LAN as the [*] box? No they are not on the same LAN > On the snom web interface, does it show that line 1 (which I > assume you are using) is 'registered'? Not sure where you see this, First page has Outgoing line: [EMAIL PROTECTED] Sip Line Pages has Name: Phone1 Account: 4405 Registrar: 24.73.215.62 Mailbox: 4405Ringer: Ringer2 For some reason MWI, wants to dial [EMAIL PROTECTED], I have not exten or account "asterisk" ???, can't even find where this is set ? Thanks again Barry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Use of Alert_Info with C7960?
Yes, I know it will definitely play custom ringtones, I was even using the 24ctu.raw ring tone for a while (I've gone back to Chirp 1 for now). But all incoming calls get the currently selected ring tone. I should have clarified on an earlier statement I made:- "Perhaps it's a memory limitation or an issue with the way Cisco are implementing the SIP firmware as to why you can't select a custom ring tone, you'd be better asking a Cisco engineer (open a TAC case)" This was in reference to playing a custom ring tone with Alert_Info (ie: Distinctive Ring). If you know how to make the 7960 play a different ring tone by setting the Alert_Info variable, please tell! > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Paul Mahler > Sent: Sunday, 21 March 2004 1:17 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Use of Alert_Info with C7960? > > The 7960 will absolutely play custom ringtones. > > > > Paul Mahler > [EMAIL PROTECTED] > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Christopher > Lee > Sent: Saturday, March 20, 2004 5:50 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Use of Alert_Info with C7960? > > Unfortunately even though it would seem the phone should support the > ability > to play custom ring tones, at present it only supports the internal tones > which are:- > > Bellcore-BusyVerify > Bellcore-Stutter > Bellcore-MsgWaiting > Bellcore-dr1 > Bellcore-dr2 > Bellcore-dr3 > Bellcore-dr4 > Bellcore-dr5 > > --- Extract from the SIP 6.0 Firmware Release Notes --- > http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp > 79 > 60/addprot/sip/relnote/phnrn60s.htm > > In RFC-3261, the Alert-Info header is specified as a URL. When the > Alert-Info header is received, the phone downloads the file from the URL > and > plays it as the alternate ring tone. This release does not support any > external ringers. Only the tones and ring patterns that are already > internal > to the phone can be selected and played as an alternate ring tone. > > In this release, the Alert-Info header consists of a name of an internal > tone or ringing pattern that can be played, as shown in the following > example: > > Alert-Info: > > There is no need to add a file extension (.au, .wav) to these names > because > the names are internal to the phone. When an Alert-Info header is > received, > the software scans the list of known tones and ringing patterns to find a > match. If the software finds a match, the phone plays that tone or ringing > pattern. If the software does not find a match, the phone plays the alert > ringing pattern as it does today. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Subject: Re: [Asterisk-Users] firefly softphone
You need to update the registry and take out the profile you created. This will clear up the problem. In my opinion just dump firefly and use something that works. I did. Message: 3 Date: Sat, 20 Mar 2004 13:07:15 +1100 From: Adam Hart <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] firefly softphone Reply-To: [EMAIL PROTECTED] Simon Brown wrote: I had exactly the same problem. I tried removing and reinstalling several times but it always crashed. I sent an email to verbiage asking for help and all I got in response was "Have you got it working yet?" from them. I have been unable to get a reply since. Simon Brown Are you using http://www.virbiage.com/firefly/download/firefly-dev.exe If possible, could you get the Hex address, which XP stores under Event Viewer (in admin tools) _ All the action. All the drama. Get NCAA hoops coverage at MSN Sports by ESPN. http://msn.espn.go.com/index.html?partnersite=espn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Use of Alert_Info with C7960?
Okay, give me a clue how to do it. The phone (sip v6.2) currently has
a ringer called "Old Style" that plays from the front panel. How do I
code * to play that ringer? (I can't seem to make anything other then
the stuff work.)
Rich
> The 7960 will absolutely play custom ringtones.
>
>
>
> Paul Mahler
> [EMAIL PROTECTED]
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Christopher Lee
> Sent: Saturday, March 20, 2004 5:50 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] Use of Alert_Info with C7960?
>
> Unfortunately even though it would seem the phone should support the ability
> to play custom ring tones, at present it only supports the internal tones
> which are:-
>
> Bellcore-BusyVerify
> Bellcore-Stutter
> Bellcore-MsgWaiting
> Bellcore-dr1
> Bellcore-dr2
> Bellcore-dr3
> Bellcore-dr4
> Bellcore-dr5
>
> --- Extract from the SIP 6.0 Firmware Release Notes ---
> http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp79
> 60/addprot/sip/relnote/phnrn60s.htm
>
> In RFC-3261, the Alert-Info header is specified as a URL. When the
> Alert-Info header is received, the phone downloads the file from the URL and
> plays it as the alternate ring tone. This release does not support any
> external ringers. Only the tones and ring patterns that are already internal
> to the phone can be selected and played as an alternate ring tone.
>
> In this release, the Alert-Info header consists of a name of an internal
> tone or ringing pattern that can be played, as shown in the following
> example:
>
> Alert-Info:
>
> There is no need to add a file extension (.au, .wav) to these names because
> the names are internal to the phone. When an Alert-Info header is received,
> the software scans the list of known tones and ringing patterns to find a
> match. If the software finds a match, the phone plays that tone or ringing
> pattern. If the software does not find a match, the phone plays the alert
> ringing pattern as it does today.
>
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Rich Adamson
> > Sent: Sunday, 21 March 2004 9:28 AM
> > To: Asterisk-a-users-list
> > Subject: [Asterisk-Users] Use of Alert_Info with C7960?
> >
> > Using Asterisk CVS-03/20/04-11:54:56 with 7960 v6.2 and playing around
> > with distinctive ringing, trying to make it work. Extensions.conf
> > looks
> > like:
> >
> > exten => 3010,1,SetVar(ALERT_INFO=3) ; selects Ringer exten =>
> > 3010,2,Dial(SIP/3010,15) exten => 3010,3,Voicemail2(u3010) exten =>
> > 3010,102,Voicemail2(b3010) exten => 3010,103,Hangup
> >
> > Calling that extension, the CLI indicates:
> > -- Executing SetVar("SIP/3002-39d1", "ALERT_INFO=3") in new stack
> > -- Executing Dial("SIP/3002-39d1", "SIP/3010|15") in new stack
> > -- Called 3010
> > -- SIP/3010-f848 is ringing
> >
> > On the phone, Settings/Ring Type indicates: Chirp 1, Chirp 2, Old
> > Style and Synth Low. The first three choices produce different ringing
> > sounds when selected from the display.
> >
> > I expected Alert_Info=3 to cause the C7960 to ring with the Old Style
> > ringer, but it doesn't and setting it to 2 or 3 doesn't make any
> > difference.
> >
> > Am I doing something wrong?
> >
> > Rich
> >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
---End of Original Message-
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Re: [Asterisk-Users] T100P T1 problem (Avaya -> asterisk IVR)
Yes, I followed those directions when I attempted to use pri, but I had no luck. (Note, I could not verify that those instructions were followed -- I have no experience with Avaya's). If you know the commands to verify those instructions on the Avaya side, I would appreciate the tips (Google was no help). If you are positive those directions are correct I will try again -- I ask because on irc #asterisk, some people thought those were wrong (like to use esf framing). Thanks for the response, Jeb Campbell [EMAIL PROTECTED] Cell: 865-385-1437 On Mar 20, 2004, at 8:30 PM, James Coberly wrote: Check out : http://www.voip-info.org/wiki-Asterisk+Avaya Depending upon the card you are using in the Avaya, you should set it up as a tie trunk on the Avaya side. James- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 200
Here's another funny * CLI puts put "-- Registered SIP '4405' at IP.address Port 5060 Expires 3600 " and within seconds the snomm 200 beeps the MWI goes on the LCD and the light flashes a call from asterisk "Not Found" Willy if you could let me see you sip and config files, if you have yours working? I'm very sure it is not a LAN issue, but a config issue thanks in advance Barry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Use of Alert_Info with C7960?
The phone selects the ringtone, not asterisk. I don't know if SIP supports
asterisk selection of a different rington, I'll check it out.
Paul
Paul Mahler
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Saturday, March 20, 2004 7:41 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Use of Alert_Info with C7960?
Okay, give me a clue how to do it. The phone (sip v6.2) currently has a
ringer called "Old Style" that plays from the front panel. How do I code *
to play that ringer? (I can't seem to make anything other then the
stuff work.)
Rich
> The 7960 will absolutely play custom ringtones.
>
>
>
> Paul Mahler
> [EMAIL PROTECTED]
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Christopher Lee
> Sent: Saturday, March 20, 2004 5:50 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] Use of Alert_Info with C7960?
>
> Unfortunately even though it would seem the phone should support the
> ability to play custom ring tones, at present it only supports the
> internal tones which are:-
>
> Bellcore-BusyVerify
> Bellcore-Stutter
> Bellcore-MsgWaiting
> Bellcore-dr1
> Bellcore-dr2
> Bellcore-dr3
> Bellcore-dr4
> Bellcore-dr5
>
> --- Extract from the SIP 6.0 Firmware Release Notes ---
> http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english
> /ipp79
> 60/addprot/sip/relnote/phnrn60s.htm
>
> In RFC-3261, the Alert-Info header is specified as a URL. When the
> Alert-Info header is received, the phone downloads the file from the
> URL and plays it as the alternate ring tone. This release does not
> support any external ringers. Only the tones and ring patterns that
> are already internal to the phone can be selected and played as an
alternate ring tone.
>
> In this release, the Alert-Info header consists of a name of an
> internal tone or ringing pattern that can be played, as shown in the
> following
> example:
>
> Alert-Info:
>
> There is no need to add a file extension (.au, .wav) to these names
> because the names are internal to the phone. When an Alert-Info header
> is received, the software scans the list of known tones and ringing
> patterns to find a match. If the software finds a match, the phone
> plays that tone or ringing pattern. If the software does not find a
> match, the phone plays the alert ringing pattern as it does today.
>
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Rich Adamson
> > Sent: Sunday, 21 March 2004 9:28 AM
> > To: Asterisk-a-users-list
> > Subject: [Asterisk-Users] Use of Alert_Info with C7960?
> >
> > Using Asterisk CVS-03/20/04-11:54:56 with 7960 v6.2 and playing
> > around with distinctive ringing, trying to make it work.
> > Extensions.conf looks
> > like:
> >
> > exten => 3010,1,SetVar(ALERT_INFO=3) ; selects Ringer exten =>
> > 3010,2,Dial(SIP/3010,15) exten => 3010,3,Voicemail2(u3010) exten =>
> > 3010,102,Voicemail2(b3010) exten => 3010,103,Hangup
> >
> > Calling that extension, the CLI indicates:
> > -- Executing SetVar("SIP/3002-39d1", "ALERT_INFO=3") in new stack
> > -- Executing Dial("SIP/3002-39d1", "SIP/3010|15") in new stack
> > -- Called 3010
> > -- SIP/3010-f848 is ringing
> >
> > On the phone, Settings/Ring Type indicates: Chirp 1, Chirp 2, Old
> > Style and Synth Low. The first three choices produce different
> > ringing sounds when selected from the display.
> >
> > I expected Alert_Info=3 to cause the C7960 to ring with the Old
> > Style ringer, but it doesn't and setting it to 2 or 3 doesn't make
> > any difference.
> >
> > Am I doing something wrong?
> >
> > Rich
> >
> >
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Re: [Asterisk-Users] Snom 200
> Here's another funny > * CLI puts put > "-- Registered SIP '4405' at IP.address Port 5060 Expires 3600 " > and within seconds the snomm 200 beeps the MWI goes on the LCD and the > light flashes a call from asterisk "Not Found" > > Willy if you could let me see you sip and config files, if you have yours > working? I'm very sure it is not a LAN issue, but a config issue I've had a snom 200 working since about October last year. Currently running v2.03o with two extensions defined. It's not defined any different then a 7960 in *. Here's my sip.conf [3008] type=friend host=dynamic username=3008 ;Snom phone - line 2 secret=mysecret context=from-sip callgroup=2 pickupgroup=2 mailbox=3008 And in extensions.conf: exten => 3008,1,Dial(SIP/3008,15) exten => 3008,2,Voicemail2(u3008) exten => 3008,102,Voicemail2(b3008) exten => 3008,103,Hangup Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T100P T1 problem (Avaya -> asterisk IVR)
Jeb, What Avaya card are you using? What model of system? Definity, Merlin, etc? With this I should be able to send you the base commands to review the card slot settings for the PXB James- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Use of Alert_Info with C7960?
Rich Adamson wrote: The wiki indicates Alert_Info can be set to a number, and implies that number is the ringer type listed on the phone. Is there a way to select one of the internal ringer types via Alert_Info? My understanding is that: 1. 7940/7960 pre version 6 may support numeric values 1-5 (not tested). 2. 7940/7960 firmware version 6 supports textual ALERT_INFO (bellcore-dr2) etc. (see version 6.0 release notes) 3. You cannot specify the ringtone to use, only what I guess I'd call the 'cadence' - you'll notice dr1 through dr5 ring in different patterns. 4. Your current ringtone is used with the specified cadence. The cadences are mostly so similar as to be useless so I have resorted to having the 7960s register multiple line appearances so you can see which one is ringing through, rather than using distinctive ring. If anyone has successfully got a custom ring tone, do chime in. Similarly, if you know how to get VXML_URL to work on the 7960, let me know. This just appends stuff to the To: SIP header. I see no mention of this (or XML push) anywhere in the Cisco documentation, so I'm disinclined to believe the wiki/source that this field is actually for the Cisco phones. Maybe something else. David. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can't get the full callerid php/agi
Your script is receiving the data correctly, as you will see if you
actually dump that data to a file rather than back to the asterisk console.
The problem is actually in your VERBOSE statement. You are passing back
this string:
VERBOSE ""Sathya Weerasooriya" <1001>"
Naturally asterisk is confused by this quote nesting. Try this line instead:
echo "VERBOSE \"".str_replace("\"", "\\\"", $temp)."\"\n";
David
Sathya wrote:
Hi folks,
I need some help from php/agi experts out there;
I am having difficulties in extracting the callerid number from php. My
script is given below;
#!/usr/local/bin/php -q
//environment dump
ob_implicit_flush(true);
set_time_limit(0);
$err=fopen("php://stderr","w");
$in = fopen("php://stdin","r");
while (!feof($in)) {
$temp = str_replace("\n","",fgets($in,4096));
echo "VERBOSE \"$temp\"\n";
if (($temp == "") || ($temp == "\n")) {
break;
}
}
?>
And the Response is:
getenv.agi: agi_request: getenv.agi
getenv.agi: agi_channel: SIP/-081524c0
getenv.agi: agi_language: en
getenv.agi: agi_type: SIP
getenv.agi: agi_uniqueid: 1079819757.97
getenv.agi: agi_callerid: Sathya <=
getenv.agi: agi_dnid: unknown
getenv.agi: agi_rdnis: unknown
getenv.agi: agi_context: default-in
getenv.agi: agi_extension: 91234
getenv.agi: agi_priority: 1
getenv.agi: agi_enhanced: 0.0
getenv.agi: agi_accountcode:
Actual caller ID is ; "Sathya Weerasooriya"<1001>, but what I get from
this PHP is only first part.
When I run the example test script in agi-bin directory, I get the
correct callerid.
I am using PHP 4.3.5CR xx.
Any help is appreciated.
Sathya
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RE: [Asterisk-Users] Problem with Vegastream 50 BRI
sorry, the sip extract is from a previous test now i get the same problem but with looking for 57228047 in tlsgw and it's the same error, it searching in this direction : why are the 2 ast values 0 ?? Non-codec capabilities: us - 1, them - 0, combined - 0 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Olle E. Johansson Sent: Saturday, March 20, 2004 11:05 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Problem with Vegastream 50 BRI Michael Devenijn wrote: > Here is a sip log from my vegastream 50BRI to my asterisk box and i can't figure out > why the call doesn't go trough ... > > > sip.conf extract : > > [gw001] > type=friend > host=dynamic > defaultip=192.168.0.12 > nat=no > dtmfmode=rfc2833 > canreinvite=yes > qualify=no > context=tlsgw > > > > extensions.conf extract (from the contact [tlsgw]) : > > exten => 57228047,Dial(SIP/cs001,40,tr) > ... Looking for 57228047 in sip Transmitting (no NAT): SIP/2.0 404 Not Found Asterisk isn't looking in the context tlsgw, for some reason it checks in the "sip" context. If this is your default context, Asterisk doesn't connect the incoming call with gw001. You have host=dynamic - is the gateway registred with Asterisk at all? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
