RE: [Asterisk-Users] Will Adtran TSU 600 work with *?
People on the list tend to think you cant make many cards work on a regular desktop. If youre willing to wait a couple of week I might have an answer for you. From: Robert Augustyn [mailto:[EMAIL PROTECTED] Sent: Saturday, December 11, 2004 7:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Will Adtran TSU 600 work with *? Hi, I am looking at getting adtran tsu 600 p/n 1200.076L2 for my small office It comes with 6 FXS ports and I would use 2 X100Ps for FXO ports. Would that work ? Is there anything I would have to be aware of in such configuration? What would be a better solution? robert -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ 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] DIALSTATUS missing an important condition?
I have recently built my first asterisk system and am very impressed with
its capabilities.
However, I have run into one problem that hopefully someone can help me
with.
I am trying to use the DIALSTATUS function to route incoming calls to the
appropriate Voice Mail (busy or unavailable) or to an Unavailable Number
recording if the number is not assigned.
However, I find that DIALSTATUS seems to generate an CHANUNAVAIL status for
any 1 of 2 conditions:
1) the dialled user is not logged in (and hence no channel) or
2) the dialled user does not exist at all (ie the number is not assigned in
sip.conf) (and hence no channel)
Obviously for condition 1 the call should be sent to VM unavailable, whereas
for condition 2 I would like to send it to a number you have dialled is not
in service recording - with no Voice Mail involved.
I have managed to get this scenario working but I don't think my solution is
very elegant or even correct (although it seems to work).
Here are the relevant parts of my extensions.conf. My VM box numbers are
exactly the same as the phone number so I only use ARG1 in the macro. (2000
= 2000 etc.). I only have SIP phones at the moment and they are all
allocated in the 20XX numbering range.
[altea_extensions]
;This is a catchall for any 4 digit number dialled starting with 20
;Using it removes the need to provide a routing plan for each phone
exten = _20XX,1,ResponseTimeout,1 ; Response Timeout for non working
numbers
exten = _20XX,2,Macro(stdexten_sip,${EXTEN}) ;send to macro for processing
;following is needed if an extension is unassigned (ie not datafilled)
because
;DIALSTATUS cannot (?) differentiate between an unassigned # or 1 that is
not answered or not logged in
;an unassigned (non working number) causes a timeout in the std-extn macro
and it drops back here
;where I provide a not in service recording
exten = t,1,Macro(not_in_service);send to number not in service
recording
exten = t,2,Hangup
[macro-stdexten_sip]
; Standard extension macro for SIP phones (modified):
; ${ARG1} = Dialled number
;
exten = s,1,Dial(SIP/${ARG1},20,tT) ; Ring the interface, 20 secs maximum
exten = s,2,Goto(s-${DIALSTATUS},1); Jump on Status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(u${ARG1}) ; no answer = unavailable
exten = s-BUSY,1,Voicemail(b${ARG1}) ; busy
exten = s-CHANUNAVAIL,1,Voicemail(u${ARG1}) ;no channel (not logged in) =
unavailable
exten = s-CONGESTION,1,Macro(120_ipm) ;Don't know what this is but will
include anyway
Am I missing something here or should there be another condition such as
Unassigned? Asterisk seems to know that the number is unassigned because
it writes a No such host message into the log. Is there any way of
trapping this message in Call Processing to route this call correctly?
Or am I getting to deep here and there's a real simple way to do it that
I've missed?
chris
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[Asterisk-Users] Problems getting Asterisk Realtime to work
I have installed the CVS Head as of 12/12/04, as well as the asterisk-addons to ensure that /usr/lib/asterisk/modules/res_config_mysql.so exists. I have configured the following (after building a new DB with the appropriate SQL examples, with mods to drop the invalid keys, on the Wiki): - /etc/asterisk/res_mysql.conf [general] dbhost = 127.0.0.1 dbname = my_db dbuser = my_uname dbpass = my_secret dbport = 3306 dbsock = /tmp/mysql.sock - /etc/asterisk/extconfig.conf ; Extconfig.conf for realtime configuration voicemail = mysql,my_db,voicemail_users (Just want to try something simple such as voicemail for the initial testing.) I have removed the [default] section from my voicemail.conf. When I try to access voicemail after restarting Asterisk, no voicemail config is found. Anyone have any luck? - I notice I get this error at startup: parse error: No category context for line 1 of /etc/asterisk/extconfig.conf If I change my extconfig.conf to: ; Extconfig.conf for realtime configuration [default] voicemail = mysql,my_db,voicemail_users The error goes away, but the config still does not work. Can't find anything on the new Wiki pages on the subject though. - Also posted here: http://asterisk.xvoip.com/viewtopic.php?t=764start=0postdays=0postorder=aschighlight= ___ 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 TDM400P and cidstart=polarity
On Sat, 11 Dec 2004, Rickard Kristiansson wrote: I'm testing a TDM400P with FXO module to receive incoming calls from an analogue line and send it to a SIP device. To recieve callerid, I need to use cidsignalling=dtmf and cidstart=polarity. The problem is that when a call is finished, the TDM400P seems to require about 20 seconds to prepare for the next incoming call. If a new call comes in within 20 seconds after the previous call was hungup, the TDM400P answers with a modem carrier, sounding like you're calling a modem pool..! The caller hangs up and retries the call, and the next time everything is OK. If the second call comes in later than 20 seconds after the previous call was finished, or if I remove the cidstart=polarity (and don't get callerid) everything works fine. I can't see any difference in the Asterisk debug logs worth mentioning... Has anyone experienced anything similar..? It may be that the remote hangup supervision for your line is signalled via a polarity reversal as well and Asterisk misstankenly thinks that is the start of the callerid for the next call. The 20 seconds would then be the time it takes for Asterisk to abandon that phantom call. A similar issue has been discussed in http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002847 My comments there describe how the Swedish PSTN works. The signalling there is built around polarity reversals. At the moment I do not think Asterisk handles all cases. Since we are isdn based I have not looked further into it. Peter ___ 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] ACK from asterisk not matched to transaction by SER / LCS2005
Public Dump wrote: For reasons unknown to me, SER and subsequently a Microsoft Live Communcations Server 2005 seems to have problems, matching a SIP ACK request from asterisk to the ongoing SIP transaction, I have attached the complete log, but the essential lines are: That's a bug in Asterisk that is in the bug tracker and needs to be fixed. Asterisk is sending the ACK to the AOR in the invite, not to the contact address in the 200 OK. Regards, /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] How to Playback Mailbox Owners Name?
How do I Playback the Mailbox Owners Name? Ex.: I want a message saying I am sorry but + Mailbox Owner Name + has gone to lunch Thanks Thorben ___ 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] Caller ID info ZAP -- SIP??
Hi everyone, I've been toying with * for quite some time now. I've got two Cisco 7940's with the SIP firmware playing nice with *. I can also make outbound calls via IAXTel (toll-free calls only) and all other calls I have routed out my X100P-clone adapter. Here's my question... Is there a way to capture the inbound callerid from my phone line (coming in on the X100P) and have it appear on my SIP phones properly? Or maybe I'm just doing this wrong. I want all my IP phones to ring when someone from the outside calls my number. When this fails... send them to voicemail. Right now I just dial my IP phone when the outside line rings. If there is a different way about doing this -- please let me know! This is how my test config is right now: extensions.conf [incoming] exten = s,1,Dial(SIP/2002,20) The phone does ring... but for caller id info it just shows asterisk. Any help would be appreciated. -= EB =- ___ 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] Very Cool.........Asterisk Made Wired Magazine
On Fri, 10 Dec 2004, JR Richardson wrote: Hi Guys, The article They've Got Your number in the Dec 2004 issue of WIRED magazine mentions Asterisk PBX (on p.100). The article is about phone phreaks hijacking cell phones with Bluetooth technology along with spoofing CID to pull some clandestine hacks on the PSTN. Anyhow, Asterisk is mentioned as the PBX of choice for an outfit: Telephreaks.org out of Florida that has built their own free VoIP service. Quotes: Slestak, Da Beave, and GiD are the crew behind Florida-based Telephreaks.org, a free VoIP service that they've built to run on a roll-your-own, open source private branch exchange (PBX) system called Asterisk But with Asterisk, there's no need for the phone company to manage your phone lines anymore. Your can do it yourself. Well, it's good that Asterisk made WIRED magazine but really it should be on the front page with Mark's smiling mug on the cover like Linus' was on the Nov 2003 issue of WIRED. Well.. it took 11 years from the first Linux release for Linus to make it onto the cover. Mark has a way to go yet! ;) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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: Ethernet Channel Bank idea
On Fri, 10 Dec 2004, nik martin wrote: news.gmane.org wrote: Allied Telesyn VoIP Access Device http://www.alliedtelesyn.co.uk/site/files/documents/datasheet/VP624FXS_euro.pdf This is a 24-port FXS 1u device, conveniently presented as a single RJ-21 TELCO connector. yeah, but those are expensive as crap. i was thinking about something more competetive with a channel bank You know, if someone had some time on their hands, was good at hardware/software hacking and had the will, the old Livingston/Lucent PM3 platform would make an awesome 48 port IAX2 - PRI/T1 channel bank. Basically, the PM3 has 2 T1 ports that can be configured for ISDN PRI. The core of the system runs on an AMD x86 CPU. The plug in Modem cards have Lucent DSP's on them (up to 50 in a box). Flash size is 4 megs, and RAM is usually around 4 megs. That is still quite a bit of horsepower, and the boxes are under $400 now. The DSP's could be used for Codec Translation, if neccessary, or for echo cancellation. And, we can get access to the original Lucent ComOS Source code. Anyone game? :) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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 getting Asterisk Realtime to work
I have installed the CVS Head as of 12/12/04, as well as the asterisk-addons to ensure that /usr/lib/asterisk/modules/res_config_mysql.so exists. I have configured the following (after building a new DB with the appropriate SQL examples, with mods to drop the invalid keys, on the Wiki): - /etc/asterisk/res_mysql.conf [general] dbhost = 127.0.0.1 dbname = my_db dbuser = my_uname dbpass = my_secret dbport = 3306 dbsock = /tmp/mysql.sock - /etc/asterisk/extconfig.conf ; Extconfig.conf for realtime configuration voicemail = mysql,my_db,voicemail_users (Just want to try something simple such as voicemail for the initial testing.) I have removed the [default] section from my voicemail.conf. When I try to access voicemail after restarting Asterisk, no voicemail config is found. Anyone have any luck? - I notice I get this error at startup: parse error: No category context for line 1 of /etc/asterisk/extconfig.conf If I change my extconfig.conf to: ; Extconfig.conf for realtime configuration [default] voicemail = mysql,my_db,voicemail_users The error goes away, but the config still does not work. Can't find anything on the new Wiki pages on the subject though. - Also posted here: http://asterisk.xvoip.com/viewtopic.php?t=764start=0postdays=0postorder=aschighlight= ___ 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: Polycom 500 - Dialtone while connected
I'm interested too. Any chance to put the archive in a ftp site?. I am also interested in getting the 1.3.4 firmware. It annoys me that I can't just get it from Polycom's website, and forces me to rethink deploying their phones for customers. Send emails to the Polycom sales, support and other groups, and complain to them. Maybe if enough folks do that they will rethink their policy. They claim to be handling it the way they do because they want to maintain high quality customer support through certified dealers. That might be true for their more sophisticated products, but it certainly does not appear to be working for their IP phones. I'd bet a fair number of folks reselling their IP phones aren't certified and they are picking up the product through (back-door) distributors. (That's got to be part of the reason why resellers do not include copies of the required (license) software when shipping product, if when its stipulated on a purchase order.) ___ 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] How to Playback Mailbox Owners Name?
On Sun, 2004-12-12 at 21:45, Thorben G. Jensen wrote: How do I Playback the Mailbox Owners Name? Ex.: I want a message saying I am sorry but + Mailbox Owner Name + has gone to lunch You could get them to record their temp message in the voicemail services; option 0, IIRC. Thanks Thorben __ ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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: Ethernet Channel Bank idea
Allied Telesyn VoIP Access Device http://www.alliedtelesyn.co.uk/site/files/documents/datasheet/VP624FXS_euro.pdf This is a 24-port FXS 1u device, conveniently presented as a single RJ-21 TELCO connector. yeah, but those are expensive as crap. i was thinking about something more competetive with a channel bank You know, if someone had some time on their hands, was good at hardware/software hacking and had the will, the old Livingston/Lucent PM3 platform would make an awesome 48 port IAX2 - PRI/T1 channel bank. Basically, the PM3 has 2 T1 ports that can be configured for ISDN PRI. The core of the system runs on an AMD x86 CPU. The plug in Modem cards have Lucent DSP's on them (up to 50 in a box). Flash size is 4 megs, and RAM is usually around 4 megs. That is still quite a bit of horsepower, and the boxes are under $400 now. The DSP's could be used for Codec Translation, if neccessary, or for echo cancellation. And, we can get access to the original Lucent ComOS Source code. One of Livingston's developers use to hang around this list. Haven't seen him post for awhile so not sure if he's still hanging out or not. Maybe he'll read this and comment. ___ 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] gap in priorities - what happens
When I first saw the priority numbers in extensions.conf, I thought BASIC, if a number is missing, * will fall thru to the next number. I learned that this is not so, if you have nothing between 1 and 3, you don't ever get to 3. But I'm wondering what does happen? Hangup and wait for next offhook? Undefined? ___ 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] gap in priorities - what happens
Hi, Il giorno dom, 12-12-2004 alle 14:38 +0200, Warren Burstein ha scritto: When I first saw the priority numbers in extensions.conf, I thought BASIC, if a number is missing, * will fall thru to the next number. I learned that this is not so, if you have nothing between 1 and 3, you don't ever get to 3. that's true. But I'm wondering what does happen? Hangup and wait for next offhook? Undefined? Timeout is called. Ie if exists the exten t, after the timeout (default 5 secs, if I remember correctly) will be executed, otherwise hangup Matteo. ___ 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] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *
Welcome to the Asterisk users community! Asterisk.org is a fast moving project. New code is added every day. Asterisk is the leading Open Source Telephony platform, with support both for classical telephony and IP telephony. Our community is also growing fast and we're having a lot of interaction, on the IRC and on the mailing lists. It's great to have you participating in this Open Source project - building an Open Source PBX. Here are a few things to know and remember while working with the project. ** The mailing list is growing The lead programmer of Asterisk, Mark Spencer at Digium, inc, half a year ago wrote: The Asterisk community is growing at a remarkable pace. I know there are thousands of you out there -- in fact there are over eight *thousand* subscribers to asterisk-users alone, and almost one *thousand* registered users on the bug tracker. Today, we propably have over 10,000 readers. This means that everything anyone write to this mailing list, is sent to thousands of mailboxes that is already flowing over with messages. ** Think before sending a message, think twice I would like to stress the fact that you have to think before you send a message to such a big list. Do *not* send out personal replies on the list. If you offer services to someone, do *not* CC: or reply to the list, it will annoy more potential customers than get you new customers. If you send out a message by mistake, you don't have to apologize to all of us, we understand you're embarassed. We will get more annoyed by your apology than over your first message. ** Try finding the answer first, then ask the list The Asterisk Wiki at http://www.voip-info.org project is an important knowledge base for the project. Go there to find your answer first, then search the mailing list archives (Google or http://search.voip-forum.com) and then go to the IRC channel. The IRC channel is populated with Asterisk gurus around the clock (literally) and they'll help you move forward. * IRC info: http://www.asterisk.org/index.php?menu=support#irc * There's many links to Asterisk web pages on the documentation page at http://www.asterisk.org * The Asterisk FAQ is found on the wiki http://www.voip-info.org/wiki-Asterisk+FAQ * The Asterisk documentation project (which needs your help) is at http://www.asteriskdocs.org Their handbook The hitchhiker's guide to Asterisk is already well worth reading. Finally, if you don't find the answer elsewhere, try the list. ** Mailing lists For developers, there is a developer's list, asterisk-dev. For BSD users (FreeBSD, NetBSD, OpenBSD and OS/X) there's a list called asterisk-bsd. There is also a business list for those that want to ask for commercial services and inform their community about new services. You'll find all lists on http://lists.digium.com, which is the site where you manage your subscription to this list as well. Please, do not crosspost the same message to multiple mailing lists. It will not help you, it will only add to the mail flow and get people that read both lists irritated. ** Reporting bugs If you think you have found a bug, report it. We need bug reports. Read this document http://www.digium.com/bugtracker.html and then go to the bugtracker http://bugs.digium.com to file a report. If you are unsure, find a bug marshal on the IRC channel to help you. They're appointed to support you with how to handle bugs. Please check the bugtracker thoroughly before posting a new bug; often, your bug or feature already exists but is simply slowly making it's way through the system. Duplicate reports slow things down for everyone, so please spend a few minutes searching first. The bug tracker is also a place where you add your contribution to Asterisk. If you have coded extra functionality, make sure you give it back to the project so it can be added to the code base. This is how Asterisk grows, free contributions and consultants that are paid to add functionality on a case by case basis. ** Be a community member - contribute! The Asterisk software growth is very much based on user contributions. That's really how we all pay for the software - and get revenue back. If you develop custom functionality, you can rest assured that there is someone out there that wants it, needs it and will be helped by it. Don't forget to contribute. Open Source is both giving and taking. The financial model behind it all is really cooperative in some way. As one member to the community said to a contractor: Hey, I'm paying you to deliver code to me, then I'm giving it away to the community. How did this happen? It's the Open Source business model. And if it didn't work, we wouldn't have a lot of the software platforms that we all use in our business systems - Linux, Apache, MySQL, PostgreSQL and Asterisk. ** Remember: It's Open Source, it's voluntary Asterisk.org is a Open Source project. This means you can't request help from people, demand new functions or
[Asterisk-Users] Pattern-matching in the dial-plan
Hey all, I'm trying to add some logic to a dial-plan to allow the caller to terminate a number with a #, but also accept it without this terminator. While trying this, I noticed that, for example, extension _[*0-9]XXX.# always seems to match, whether the last digit dialled is a # or not. It's as if the parser assumes everything after the . will match and doesn't look any further. Is this expected behaviour? If so, what would be the best solution to my problem? I currently solved it by avoiding the . and matching every possible number-length seperately, both with and without the #-terminator. It works, but seems like it should be doable with just 2 matches. The box I'm trying this on is running the CVS HEAD of about a week ago. Thanks in advance for any suggestions. Grtz, Oliver ___ 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: Polycom 500 - Dialtone while connected
On Sun, Dec 12, 2004 at 05:50:16AM -0600, Rich Adamson arranged a set of bits into the following: I'm interested too. Any chance to put the archive in a ftp site?. I am also interested in getting the 1.3.4 firmware. It annoys me that I can't just get it from Polycom's website, and forces me to rethink deploying their phones for customers. Send emails to the Polycom sales, support and other groups, and complain to them. Maybe if enough folks do that they will rethink their policy. They claim to be handling it the way they do because they want to maintain high quality customer support through certified dealers. That might be true for their more sophisticated products, but it certainly does not appear to be working for their IP phones. I'd bet Doesn't work for their better phones either... The problem is that they assume that the reseller knows more then the customer, something that hasn't been true for a *long* time. Polycom are second only to Cisco in the shere stupidity of their managemnt (yes I haven't delt with Avaya or some of the more traditional companies who are apparently just as bad) with regards to VoIP, believing that their customers will want to buy their softswitches, and are not buying VoIP for say _the_flexibility_. I've tried telling Cisco that if we could simply have a copy of the Skinny protocol docs (which do exist and are distributed to some companies) that they would have increased sales due to the people who want features SIP can't provide, or the possibility of the integrated applications. But they don't listen and don't seem to care. Fortunatly the word I'm hearing (at least in .au large commercial) is that many large companies that have gone cisco are getting very annoyed at them for promising and not delivering, and if they continue their next upgrade will be explicitly *not* cisco. [Those are direct words from a few large scale PBX integrators I know, not myself] If they would just realise that if they had their products actually realise the potentional that VoIP offers they'd increase sales where it matters, on the equipment that's VISIBLE, with THEIR branding on it, not * or whoever makes the switch. And at least for * they dont need to do anything, just release the docs _they_already_have_. a fair number of folks reselling their IP phones aren't certified and they are picking up the product through (back-door) distributors. (That's got to be part of the reason why resellers do not include copies of the required (license) software when shipping product, if when its stipulated on a purchase order.) I assume you mean even when, and if a vendor did that I'd just return it and not pay the bill. If it's not what I ordered then it doesn't meet the contract and so I won't pay. Any vendor that still complained would never get my business, not that of anyone I know. pgpqdB3QeuM5p.pgp Description: PGP signature ___ 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: Voice Prompt Info
Ariel Batista wrote: Warren Burstein wrote: One more thing about prompts, it's better to say for sales press 5 than press 5 for sales, because by the time you hear sales you've already forgotten what number it was. If you add the sounds all you need is For Sales recorded the new sounds have press # already. So you don't need to get any additional recorded items except the one that says For Sales by Allison. If you want have her record Press as an additional recorded item. She already did a message containing just press, sounds/vm-press.gsm in asterisk-1.0.0 (not asterisk-sounds). I was thinking about a project I have to bring up very soon won't be time to wait for new recordings, so we are going to record the messages ourselves, but yeah, if we're making messages for everyone to use, we already have press and press 1. Anyway, the departments we need are support, sales, projects. systems, and operator. There already is %for-tech-support.gsm%For technical support %to-reach-operator.gsm%to reach an operator so all we would like to add is sales, projects, and systems. Maybe so all the messages are similar, add for an operator as well. Although we might decide to stick with our own recordings, in case later we need to change a message ASAP. ___ 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] 3com NBX and Asterisk Integration.
I sent this once already but it didnt show up in my mailbox so I am not sure if ever made it to the list. If it did, my apologies. I have an * system and an nbx. My plan is to use several grandstream 286s and plug thethe phone cords into the NBX's analog FXO card. It works fine but after I make a call, the port stays open on the NBX and I cannot make another call for around five minutes or I have to manually reset the port on the NBX. I assume the Grandstream is still sending a busy signal to the NBX so the port wont close. I know you can do things with Zap channels to monitor the line but is it possible with SIP? I am pretty sure that this needs to be implemented on the 286 or the NBX itself and not *. Here is what 3com says about it. Maybe someone can help without having to buy the graybar device listed in the article. SOLUTION OFFERING FOR Telco LINES NOT PROVIDING DISCONNECT:- This solution may not work in all implementations, and most likely does NOT work for PBX Analog Extensions coming into the NBX, as they are not actual phone company lines, they do not go into recordings or produce re-order tones- This solution is a workaround to be attempted, with the explicit understanding that 3Com Corporation does NOT endorse this product in any way, but prefers to provide a solution for continued use of the NBX system:The disconnect product is available through GrayBar (800 825 5517).Please shop around and get the unit you feel may be best for your situation.Product Manufacturer: Electronic Tele-Communications, Inc.Model: MAX Terminator Disconnect Unit"Scans for progress tones - dial tone, reorder tone, busy signal - and generates a disconnect signal within seconds of recognizing the tone. The disconnect signal is passed down the line to the peripheral equipment, allowing proper operation. Prevents voice mail systems from recording annoying telephone tones after a caller hangs up. Also helps alleviate tied up phone lines.- Provides disconnect detection for phone systems and peripherals- Helps maintain accurate call count statistics- Services 12 lines; up to 120 lines with rack-mount unit- Progress tone scanner- Stand alone chassis- Rack-mounted cards (card file accommodates up to 10 cards)- Easy installation"GrayBar Catalog # and Description (GrayBar 800 825 5517 for pricing and availability - ask for similar products too)ETC - 8208 - SN12 Standard Chassis Unit (12 lines) 11.2 " long x 8.5" wide x 1.5" deep or rack mount for more lines:ETC - 1144 - CN12 - Rack Card (12 lines per card)ETC - 1145 - AC - Rack Card File - accommodates up to 120 lines (10 cards) this thing costs over a grand! ___ 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] MeetMe performance
Hey folks, Using FreeBSD 5.2.1 and I've got the current zaptel driver installed from ports (0.8_1) and current ports asterisk (1.0.1). I've set options HZ=1000 in my kernel config, recompiled and rebooted and as far as I can tell, I've done everything right but when I try to use the conference, the audio is very delayed, choppy and segmented -- totally unusable. At the suggestion of someone on #asterisk, I cvsup'd * against digium and used that instead of ports, but that didn't seem to help either. FYI: When I said above when I try to use the conference I meant using two non-voip phones, specifically a cell phone and a land line. I'd dial the number for my asterix box which is in itself a b channel on a PRI answered by a T100P on a friend's * box and sent via IAX over to my * box. Not sure if that matters, but I figure I'd mention it anyway. Anyone have any ideas here? # meetme.conf [rooms] conf = 97531,24680 # extensions.conf [conf] exten = 1,1,Answer exten = 1,2,Wait(1) exten = 1,3,Authenticate(5447847) exten = 1,4,MeetMe(97531,Mas,24680) exten = 1,5,Playback(vm-goodbye) exten = 1,6,Hangup() exten = 2,1,MeetMe(97531,Ms,24680) [EMAIL PROTECTED]://~ ]$ kldstat Id Refs AddressSize Name 15 0xc040 5e16d8 kernel 24 0xc231e000 2f000zaptel.ko 31 0xc234f000 6000 wcfxo.ko 41 0xc2355000 a000 wcfxs.ko 51 0xc235f000 2000 ztdummy.ko [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
SV: [Asterisk-Users] How to Playback Mailbox Owners Name?
-Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Howard Lowndes Sendt: 12. december 2004 13:07 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] How to Playback Mailbox Owners Name? On Sun, 2004-12-12 at 21:45, Thorben G. Jensen wrote: How do I Playback the Mailbox Owners Name? Ex.: I want a message saying I am sorry but + Mailbox Owner Name + has gone to lunch You could get them to record their temp message in the voicemail services; option 0, IIRC. I understand that, but they all have recorded their name and I just would like to use that recording. ___ 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 a TDM400P FXS drop voltage on hangup?
I thought I had posted this, but I didnt see it in the archives, so I guess I hadnt. Ive got FXS lines going to a legacy IVR. When I Dial into one of these lines and then hang up, FXS plays the Congestion tone until the IVR drops voltage. I would like the IVR to hang up sooner. I could do this by either making the IVR recognize the standard Congestion tone, or changing the Congestion tone to be one that the IVR already recognizes (by the way, I was surprised to find that Zap tones are compiled in, not in indications.conf any thought of changing this (with backward compatability, of course)? I might be able to do this myself). But if I could get the FXS to drop voltage instead of play Congestion (or a second of Congestion in case a person is listening, and then drop voltage) that would be even simpler. But can I make that happen, and how? thanks ___ 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] Pattern-matching in the dial-plan
dialled is a # or not. It's as if the parser assumes everything after the . will match and doesn't look any further. Is this expected behaviour? Yes, the dot says match ANYTHING from here on AFAIK ___ 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] Digium Card Error
On Sun, 2004-12-12 at 00:36 -0800, Charles S. Antrim wrote: I have success installing and compiling, but if I reboot I have to modprobe again to get he drivers loaded for the module I am using. I am using rhes31 and a tdm card with one fxo and one fxs. This is where reading the mailing list is important. We just covered that for a person Friday. Look in /etc/modprobe.conf and add the modules you need. -- Steven Critchfield [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] can a TDM400P FXS drop voltage on hangup?
But if I could get the FXS to drop voltage instead of play Congestion (or a second of Congestion in case a person is listening, and then drop voltage) that would be even simpler. But can I make that happen, and how? I have the same setup at one of my sites, I tried to make the FXS drop voltage on a hangup , but I couldnt make it happen, seems like the talk voltage is always there on the fxs port. I ended up finding a setting in the legacy pbx in the CO line setup for the disconnect supervision and set it for not received, this seemed to help a lot. ___ 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] How to Playback Mailbox Owners Name?
Thorben G. Jensen wrote:
How do I Playback the Mailbox Owners Name?
Ex.: I want a message saying I am sorry but + Mailbox Owner Name +
has gone to lunch
Extension 999 in voicemail context internal
exten = 999,1,SetVar(VM_CONTEXT=internal)
exten = 999,2,Playback(im-sorry)
exten =
999,3,Playback(/var/spool/asterisk/voicemail/${VM_CONTEXT}/${EXTEN}/greet)
exten = 999,4,Playback(flagged-for-lea)
exten = 999,5,Hangup()
That should get you going.. :-)
/Sren
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[Asterisk-Users] TDM400P FXS polarity reversal?
Hello all, Is it possible to do a talk battery polarity reversal on a TDM400P FXS interface? Everything I can find seems to be referring to the procedure for detecting a battery reversal on a telephone company POTS line using the FXO interface, but not for actually generating one back to a station upon answer supervision. I would assume that VoicePulse and VoipJet provide a way of signaling far-end supervision back to the originating Asterisk PBX... Basically, my two questions are: (1) Is the hardware capable of even performing a reversal? (2) If the above is true, how would you make it happen in Asterisk? -- Strom Carlson http://www.stromcarlson.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
Re: SV: [Asterisk-Users] How to Playback Mailbox Owners Name?
On Mon, 2004-12-13 at 02:10, Thorben G. Jensen wrote: -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] P vegne af Howard Lowndes Sendt: 12. december 2004 13:07 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] How to Playback Mailbox Owners Name? On Sun, 2004-12-12 at 21:45, Thorben G. Jensen wrote: How do I Playback the Mailbox Owners Name? Ex.: I want a message saying I am sorry but + Mailbox Owner Name + has gone to lunch You could get them to record their temp message in the voicemail services; option 0, IIRC. I understand that, but they all have recorded their name and I just would like to use that recording. Well, record pre-greet.gsm and post-greet.gsm, then: cat pre-greet.gsm greet.gsm post-greet.gsm temp.gsm Bingo - your personalised temp greeting message. ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Will Adtran TSU 600 work with *?
Let me know how it works for you. Thanks robertShoval Tomer [EMAIL PROTECTED] wrote: People on the list tend to think you cant make many cards work on a regular desktop. If youre willing to wait a couple of week I might have an answer for you. From: Robert Augustyn [mailto:[EMAIL PROTECTED] Sent: Saturday, December 11, 2004 7:13 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Will Adtran TSU 600 work with *? Hi, I am looking at getting adtran tsu 600 p/n 1200.076L2 for my small office It comes with 6 FXS ports and I would use 2 X100Ps for FXO ports. Would that work ? Is there anything I would have to be aware of in such configuration? What would be a better solution? robert -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo 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] Totally LOST with dialplan and Extensions.
; outbound ; Firefly (Freshtel) [89280250] ; Firefly context=89280250 Where is this context? If you change it to default, it should work if the rest is right. Otherwise, post what you see as console messages when you try to dial. ___ 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] Many similar contexts - can I use Macro or some other template concept ?
Are there any other ways of context templates filled with data in dialplan ? Rob you really should read some of the beginning material to find this stuff out. Here is a great article for the basic concepts (including what macros are for IIRC): http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html Here is a site with documents that have answers to all the questions you are likely to ask in the next few days: http://asteriskdocs.org Read online or download the latest PDF and read it through a few times. That's why they spent all the time and effort writing, perfecting and putting this online. There are many, many other things you could read to get up to speed on this. ___ 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] TDM400P FXS polarity reversal?
Soren Rathje wrote: Specs for Si3210 (TDM400P FXS Module) says on page 93: --- Register 72. On-Hook Line Voltage Bit 6 VSGN On-Hook Line Voltage. The value written to this bit sets the on-hook line voltage polarity (VTIPVRING). 0 = VTIPVRING is positive 1 = VTIPVRING is negative --- The (missing) link.. https://www.mysilabs.com/public/documents/tpub_doc/dsheet/Wireline/ProSLIC/en/si3210.pdf Features ... ... * Software programmable signal generation and audio processing: - DTMF generation and decoding - 12 kHz/16 kHz pulse metering generation - Phase-continuous FSK (caller ID) generation - Dual audio tone generators - Smooth and abrupt polarity reversal == NB! - -Law/A-Law and 16-bit linear PCM audio ... ... ___ 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] [OT] Small SIP phones?
Hi. Does anyone know of any small SIP phones (and preferably have some experience of using them and happy to recommend them)? By 'small' I mean a single-piece phone, with dial buttons in the handset, so that it can be carried around easily in a laptop bag. Something like http://maplin.co.uk/images/Full/35493i0.jpg (which is unfortunately just a standard analogue telephone). Ideally I'd like something without a cradle, which can simply be put on a desk and answered by picking it up. Thanks, Antony. -- Linux is going to be part of the future. It's going to be like Unix was. - Peter Moore, Asia-Pacific general manager, Microsoft ___ 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] Digium Card Error
On Sun, 12 Dec 2004 09:00:50 -0600, Steven Critchfield [EMAIL PROTECTED] wrote: This is where reading the mailing list is important. We just covered that for a person Friday. Look in /etc/modprobe.conf and add the modules you need. Actually, the two fixes that worked to solve this were: 1. Add the following to /etc/rc.d/rc.local: /sbin/modprobe wcfxo or 2. cd to /usr/src/zaptel-1.0.3 and do a make config -- Lee ___ 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] [OT] Small SIP phones?
On Sunday 12 December 2004 20:12, Clay Reiche wrote: I don't know of a small phone, but you use a WorlACCXX TA200 device (pretty small) along with any standard analogue phone. http://www.worldaccxx.com I have one and carry it around in my laptop bag. Demensions are 6x4.5x1.25 Thanks. In fact I already have a Grandstream ATA-486, which I'm very pleased with: http://www.grandstream.com/y-ht486.htm This unit is even smaller - 105 x 75 x 25mm (or 4 x 2.75 x 1), however I'm just wondering if there's a neat all-in-one solution, instead of carrying around two items? Regards, Antony. -Original Message- Hi. Does anyone know of any small SIP phones (and preferably have some experience of using them and happy to recommend them)? By 'small' I mean a single-piece phone, with dial buttons in the handset, so that it can be carried around easily in a laptop bag. Something like http://maplin.co.uk/images/Full/35493i0.jpg (which is unfortunately just a standard analogue telephone). Ideally I'd like something without a cradle, which can simply be put on a desk and answered by picking it up. -- Never automate fully anything that does not have a manual override capability. Never design anything that cannot work under degraded conditions in emergency. Please reply to the list; please don't CC me. ___ 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] I'm stumped
I am trying to use the simple CID name management script on the wiki.
http://www.voip-info.org/wiki-Asterisk+tips+managing+CID+names I can not
see what is wrong. The values never get entered in the database. Here are
the files: I have asterisk running as the user asterisk also.
---cid-store.php
HTML
HEAD
TITLEStoring Asterisk CID data/TITLE
/HEAD
BODY
h1Asterisk phone book/h1
?php
set_time_limit(5);
if ($PhoneNumber$PhoneName ) {
system(sudo -u asterisk /usr/sbin/asterisk -rx .
escapeshellarg(database put cidname $PhoneNumber \$PhoneName\) .
/tmp/error);
print Successfully stored b$PhoneNumber/b as b$PhoneName/b.;
} else {
print Please enter both phone number and name!;
}
?
/BODY
/HTML
-My sudoers files-
asterisk ALL=(ALL) NOPASSWD: /usr/sbin/asterisk
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RE: [Asterisk-Users] [OT] Small SIP phones?
I don't know of a small phone, but you use a WorlACCXX TA200 device (pretty small) along with any standard analogue phone. http://www.worldaccxx.com I have one and carry it around in my laptop bag. Demensions are 6x4.5x1.25 Clay -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Antony Stone Sent: Sunday, December 12, 2004 3:01 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] [OT] Small SIP phones? Hi. Does anyone know of any small SIP phones (and preferably have some experience of using them and happy to recommend them)? By 'small' I mean a single-piece phone, with dial buttons in the handset, so that it can be carried around easily in a laptop bag. Something like http://maplin.co.uk/images/Full/35493i0.jpg (which is unfortunately just a standard analogue telephone). Ideally I'd like something without a cradle, which can simply be put on a desk and answered by picking it up. Thanks, Antony. -- Linux is going to be part of the future. It's going to be like Unix was. - Peter Moore, Asia-Pacific general manager, Microsoft ___ 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] Pattern-matching in the dial-plan
On Sun, 12 Dec 2004, Wilson Pickett wrote: dialled is a # or not. It's as if the parser assumes everything after the . will match and doesn't look any further. Is this expected behaviour? Yes, the dot says match ANYTHING from here on AFAIK To be precise it will match one or more digits, ognoring the rest of the pattern as well. There is no match zero-or-more wildcard and no wildcard that tries to continue to read the pattern after the wildcard. The former can easily be implemented, we had to do it to handle some cases of overlap dialing. We'll clean it up and submit it later. The latter case could probably be implemented in the ast_extension_match ast_extension_close EXTENSION_MATCH_CORE functions in pbx.c. Alternativly a regular or extended regexp could be added as a an extension switch, similar to pbx_loopback.c. Yet another option would be to integrate it with the res_perl in asterisk-addons. Peter ___ 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] Digium Card Error
Thanks Lee -Original Message- From: Lee [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk- [EMAIL PROTECTED] Date: Sun, 12 Dec 2004 12:26:15 -0800 Subject: Re: [Asterisk-Users] Digium Card Error On Sun, 12 Dec 2004 09:00:50 -0600, Steven Critchfield [EMAIL PROTECTED] wrote: This is where reading the mailing list is important. We just covered that for a person Friday. Look in /etc/modprobe.conf and add the modules you need. Actually, the two fixes that worked to solve this were: 1. Add the following to /etc/rc.d/rc.local: /sbin/modprobe wcfxo or 2. cd to /usr/src/zaptel-1.0.3 and do a make config -- Lee ___ 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] Cisco AS5XXX to asterisk debugging.
voice class codec 11 codec preference 1 g729br8 codec preference 2 g729r8 codec preference 3 gsmfr codec preference 4 g726r32 codec preference 6 g726r16 codec preference 7 g723r63 codec preference 8 g723r53 codec preference 9 g726r24 codec preference 10 g723ar63 codec preference 11 g723ar53 codec preference 12 g711ulaw codec preference 13 g711alaw codec preference 14 clear-channel Why so many codecs listed in class 11? Asterisk can only use ulaw,alaw, or gsm unless licensed. Try only listing g711ulaw for testing purposes. Make sure you have all of the correct ports for RTP listed in your rtp.conf file also. Just a thought or two. ___ 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] Cisco AS5XXX to asterisk debugging.
Hi,
thanks for your help .
here is the cisco config
GWSCZ01en
Password:
GWSCZ01#sh run
Building configuration...
Current configuration : 5053 bytes
!
! Last configuration change at 05:17:58 UTC Mon Apr 16 2001
! NVRAM config last updated at 12:06:13 UTC Sat Apr 14 2001
!
version 12.2
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname GWSCZ01
!
no boot startup-test
logging queue-limit 100
!
!
!
resource-pool disable
spe default-firmware spe-firmware-1
ip subnet-zero
ip cef
no ip domain lookup
!
isdn switch-type primary-net5
!
!
voice service voip
fax protocol t38 ls-redundancy 2 hs-redundancy 2 fallback none
sip
!
voice class codec 11
codec preference 1 g729br8
codec preference 2 g729r8
codec preference 3 gsmfr
codec preference 4 g726r32
codec preference 6 g726r16
codec preference 7 g723r63
codec preference 8 g723r53
codec preference 9 g726r24
codec preference 10 g723ar63
codec preference 11 g723ar53
codec preference 12 g711ulaw
codec preference 13 g711alaw
codec preference 14 clear-channel
!
!
!
!
!
!
!
no voice hpi capture buffer
no voice hpi capture destination
!
voice source-group cisco
access-list 8
carrier-id target cisco
!
!
!
fax interface-type fax-mail
mta receive maximum-recipients 0
!
!
!
controller E1 7/0
framing NO-CRC4
line-termination 75-ohm
ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled
cas-custom 0
country bolivia
!
controller E1 7/1
line-termination 75-ohm
pri-group timeslots 1-31
!
controller E1 7/2
line-termination 75-ohm
pri-group timeslots 1-31
description Embratel
--More--
!
!
interface FastEthernet0/0
ip address y.y.y.y 255.255.255.224
duplex auto
speed auto
no cdp enable
h323-gateway voip interface
h323-gateway voip id GK01 ipaddr y.y.y.z 1719
h323-gateway voip h323-id GWSCZ01
h323-gateway voip tech-prefix 2032#
!
!
ip classless
ip route 0.0.0.0 0.0.0.0 y.y.y.v
no ip http server
!
!
!
!
!
!
call rsvp-sync
!
voice-port 7/0:0
compand-type a-law
!
voice-port 7/1:D
!
voice-port 7/2:D
!
voice-port 7/3:0
compand-type a-law
!
voice-port 7/4:0
compand-type a-law
!
voice-port 7/5:0
!
!
mgcp profile default
!
dial-peer cor custom
!
!
!
dial-peer voice voip
destination-pattern 44T
voice-class codec 11
session protocol sipv2
session target sip-server
session transport udp
!
dial-peer voice pots
destination-pattern T
direct-inward-dial
port 7/0:0
!
sip-ua
retry invite 3
retry cancel 2
sip-server ipv4:x.x.x.x
!
-
Where
y.y.y.z = ip address of gk h323
y.y.y.v = ip default gateway
x.x.x.x = ip address of Astersik
y.y.y.y = ip address of Cisco
And this is the asterisk configuration
sip.conf
[general]
context=default
port=5060
bindaddr=x.x.x.x
srvlookup=yes
videosupport=no
[y.y.y.y]
type=user
host=y.y.y.y
canreinvite=no
context=fromsip
dtmfmode=rfc2833
disallow=all
allow=g729
allow=ulaw
allow=alaw
--
extensions.conf
[fromsip]
exten = _X.,1,Dial(Zap/g2/${EXTEN:2})
I'd be very grateful for your help on this matter
Best regards,
Jorge
On Fri, 2004-12-10 at 08:37, Tenorio, Leandro wrote:
Pls, post your Cisco and * config files.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jorge
Verastegui G
Sent: Friday, December 10, 2004 12:30 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Cisco AS5XXX to asterisk debugging.
Hi,
I have a serious problem to configure Cisco AS5XXX and Asterisk ,
I trying to use asterisk for
PSTN(A) Cisco AS5xxx ASteriskPSTN(B)
(No Nat, no Firewall)
I hear (on the PSTN(A)) clearly what the other person is saying, but the
other person (on the PSTN(B) side) hears nothing from PSTN(A).
I use tcpdump for debug de rtp trafic, and ouput contains
19:06:00.741293 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF],
proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP,
length 32
19:06:00.763133 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF],
proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP,
length 32
19:06:00.740415 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF],
proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP,
length 32
19:06:00.810312 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF],
proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP,
length 32
19:06:00.860314 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF],
proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP,
length 32
19:06:00.980351 IP (tos 0x0, ttl 64, id 180, offset 0, flags [DF],
proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP,
length 32
19:06:01.000313 IP (tos 0x0, ttl 64, id 181, offset 0, flags [DF],
proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP,
length 32
19:06:01.014822 IP (tos 0x68, ttl 255, id 1, offset 0, flags [none],
proto 17, length: 164) y.y.y.y.18975 x.x.x.x.19927: UDP,
[Asterisk-Users] IAXPeerGraph - a beta of another windows monitor app
We've just completed another Windows monitor app. This one has a scrolling taskman-like interface. Once again the zip file just contains the .exe file and the INSTALL.txt file. Oh, and by the way, the blue light that flashes next to the green connect light (it is black in the picture) toggles between blue and black whenever it receives a response from the simple manager proxy with IAX peer information inside of it. Note that if you have multiple clients connecting to the same server, the pulses sent out by IAXPeerGraph and IAXPeers will be received by all peer clients. You can't really tell from the picture, but the line will change colour depending on how high the ping is (i.e. if it is low, the colour will be green, med - yellow, bad red). This is kinda experimental because it is continuously variable - let me know how it goes for you. You can download it directly from http://www.sineapps.com/down/IAXPeerGraph.zip or via the main page of the daily news (as listed in my sig) If you need any help or would like some changes made, please don't hesitate to contact us. Also, if anyone could take a screen shot (press print screen key, paste into windows paint etc) of it running in Windows XP and mail it to me, it would be much appreciated. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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] Cisco AS5XXX to asterisk debugging.
Hi, I already tested those (ulow and g729) , and the rtp.conf [general] ; ; RTP start and RTP end configure start and end addresses ; rtpstart=16384 rtpend=2 ; ; Whether to enable or disable UDP checksums on RTP traffic ; ;rtpchecksums=no ~ Best regards, Jorge On Sun, 2004-12-12 at 16:54, Henry Devito wrote: voice class codec 11 codec preference 1 g729br8 codec preference 2 g729r8 codec preference 3 gsmfr codec preference 4 g726r32 codec preference 6 g726r16 codec preference 7 g723r63 codec preference 8 g723r53 codec preference 9 g726r24 codec preference 10 g723ar63 codec preference 11 g723ar53 codec preference 12 g711ulaw codec preference 13 g711alaw codec preference 14 clear-channel Why so many codecs listed in class 11? Asterisk can only use ulaw,alaw, or gsm unless licensed. Try only listing g711ulaw for testing purposes. Make sure you have all of the correct ports for RTP listed in your rtp.conf file also. Just a thought or two. ___ 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] TDM400P FXS polarity reversal?
Rich Adamson wrote: Is it possible to do a talk battery polarity reversal on a TDM400P FXS interface? Everything I can find seems to be referring to the procedure for detecting a battery reversal on a telephone company POTS line using the FXO interface, but not for actually generating one back to a station upon answer supervision. I would assume that VoicePulse and VoipJet provide a way of signaling far-end supervision back to the originating Asterisk PBX... Basically, my two questions are: (1) Is the hardware capable of even performing a reversal? (2) If the above is true, how would you make it happen in Asterisk? The Silicon Labs spec sheet does not specifically indicate generating a reversal is possible. Therefore, best guess is the integrated circuits on the card does not support it, therefore asterisk has no means of doing it. Specs for Si3210 (TDM400P FXS Module) says on page 93: --- Register 72. On-Hook Line Voltage Bit 6 VSGN On-Hook Line Voltage. The value written to this bit sets the on-hook line voltage polarity (VTIPVRING). 0 = VTIPVRING is positive 1 = VTIPVRING is negative --- I wonder if this can control Pol-Rev's.. ? /Soren ___ 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 capture -1 return on Dial command
following setup, and I want to be able to process the audio file after the outbound call has been done regardless how how it ends. would the hangup priority be appropriate for this? ___ 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] [OT] Small SIP phones?
I may be wrong, but if you always carry your laptop around, why don't purchase a USB handset? It'll give you a mic and phones in one handset, and installs as a sound card, so it can ring and you don't have to put it next to your ear (the problem with using head phones is that once they're on the desk and not on your head you might miss some phone calls). Some of these come with a softphone, and some can use your softphone of choice. As far as I know, their not cheaper the full fledged VOIP phones, but it'll be the smallest option -Original Message- From: Florian Overkamp [mailto:[EMAIL PROTECTED] Sent: Monday, December 13, 2004 12:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [OT] Small SIP phones? Hi On Sun, 2004-12-12 at 21:27, Antony Stone wrote: Thanks. In fact I already have a Grandstream ATA-486, which I'm very pleased with: http://www.grandstream.com/y-ht486.htm This unit is even smaller - 105 x 75 x 25mm (or 4 x 2.75 x 1), however I'm just wondering if there's a neat all-in-one solution, instead of carrying around two items? Three, in fact. The powersupply also adds to the required space. This is one of the biggest advantages of having an all-in-one solution, because you don't have to generate high voltage/high power ring signalling. Florian ___ 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 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ 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] PoE VOIP phones in Australia
The Polycom IP600 is fairly available in Australia (at least in Melbourne) Regards, PaulH -Original Message- From: James Andrewartha [mailto:[EMAIL PROTECTED] Sent: Friday, 10 December 2004 6:32 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] PoE VOIP phones in Australia Hi, Are there any resellers of phones that can take power over ethernet in Australia? All I can find for sale online is the BT-10[12], which is cheap but not featureful enough, and the Snom 190, which is about right, but neither of them support PoE. I'm particularly intereseted in the Snom 220 with the keypad expansion for our receptionist. Although, could you make a PoE split-out cable for the Snom 190? James Andrewartha DAA Sysadmin ___ 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 CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ 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] zaptel 0.9.1 compile problem
I'm using gentoo 2004.3, and when I emerge the zaptel driver, compile fails with the following output: CC [M] /var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.o In file included from /var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.c:40: /var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/zaptel.h:55:2: warning: #warning Zaptel doesn't support DEVFS in post 2.4 kernels. Disabling DEVFS in zaptel /var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.c: In function `ztdeth_rcv': /var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.c:89: error: union has no member named `ethernet' make[2]: *** [/var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.o] Error 1 make[1]: *** [_module_/var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.9-gentoo-r9' make: *** [linux26] Error 2 Any ideas why this would be? ~jay ___ 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] BRI Problem dialing out
Hi All, I have a slight problem when trying to dial out. When I dial any number out I get only a dial tone and the number is not dialed I have to then dial it manually. I have tried my extension.conf with my pstn box and it works fine but for some reason it wont with the isdn card. Im using the fritz pci card. Has any one else had this problem in the past??. I have also tried to set up an extension that will open the line then SendDTMF of a number to dial. no luck Can any one help Regards Michael Hatzis ___ 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] Cisco AS5XXX to asterisk debugging.
Excuse the insistence but I am more than one week with this problem, and I do not have any idea to solve it. You know if the configuration with GK in the Cisco, can be interfering with the RTP traffic? Thanks in advance On Fri, 2004-12-10 at 08:37, Tenorio, Leandro wrote: Pls, post your Cisco and * config files. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Verastegui G Sent: Friday, December 10, 2004 12:30 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco AS5XXX to asterisk debugging. Hi, I have a serious problem to configure Cisco AS5XXX and Asterisk , I trying to use asterisk for PSTN(A) Cisco AS5xxx ASteriskPSTN(B) (No Nat, no Firewall) I hear (on the PSTN(A)) clearly what the other person is saying, but the other person (on the PSTN(B) side) hears nothing from PSTN(A). I use tcpdump for debug de rtp trafic, and ouput contains 19:06:00.741293 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:00.763133 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:00.740415 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:00.810312 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:00.860314 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:00.980351 IP (tos 0x0, ttl 64, id 180, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.000313 IP (tos 0x0, ttl 64, id 181, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.014822 IP (tos 0x68, ttl 255, id 1, offset 0, flags [none], proto 17, length: 164) y.y.y.y.18975 x.x.x.x.19927: UDP, length 136 19:06:01.020312 IP (tos 0x0, ttl 64, id 182, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.040302 IP (tos 0x0, ttl 64, id 183, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.060343 IP (tos 0x0, ttl 64, id 184, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.083311 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.128314 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.130316 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.165318 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.186312 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 Where x.x.x.x = ip address of Astersik y.y.y.y = ip address of Cisco Two types of codecs were proven ( ulow, g729 ). When use the Asterisk with Sip phones everything works well. SipPhone--Asterisk---PSTN(B) The configurations, are the usual ones (from the wiki). the version of asterisk is 1.0.3, the linux is FC2. Please help me. ___ 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] Cisco AS5XXX to asterisk debugging.
What's the cisco box,52 / 53; version ios? can you post a config dump? Regards Michael Hatzis 0421 476 211 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Verastegui G Sent: Monday, 13 December 2004 10:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Cisco AS5XXX to asterisk debugging. Excuse the insistence but I am more than one week with this problem, and I do not have any idea to solve it. You know if the configuration with GK in the Cisco, can be interfering with the RTP traffic? Thanks in advance On Fri, 2004-12-10 at 08:37, Tenorio, Leandro wrote: Pls, post your Cisco and * config files. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Verastegui G Sent: Friday, December 10, 2004 12:30 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco AS5XXX to asterisk debugging. Hi, I have a serious problem to configure Cisco AS5XXX and Asterisk , I trying to use asterisk for PSTN(A) Cisco AS5xxx ASteriskPSTN(B) (No Nat, no Firewall) I hear (on the PSTN(A)) clearly what the other person is saying, but the other person (on the PSTN(B) side) hears nothing from PSTN(A). I use tcpdump for debug de rtp trafic, and ouput contains 19:06:00.741293 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:00.763133 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:00.740415 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:00.810312 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:00.860314 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:00.980351 IP (tos 0x0, ttl 64, id 180, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.000313 IP (tos 0x0, ttl 64, id 181, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.014822 IP (tos 0x68, ttl 255, id 1, offset 0, flags [none], proto 17, length: 164) y.y.y.y.18975 x.x.x.x.19927: UDP, length 136 19:06:01.020312 IP (tos 0x0, ttl 64, id 182, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.040302 IP (tos 0x0, ttl 64, id 183, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.060343 IP (tos 0x0, ttl 64, id 184, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.083311 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.128314 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.130316 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.165318 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:01.186312 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 Where x.x.x.x = ip address of Astersik y.y.y.y = ip address of Cisco Two types of codecs were proven ( ulow, g729 ). When use the Asterisk with Sip phones everything works well. SipPhone--Asterisk---PSTN(B) The configurations, are the usual ones (from the wiki). the version of asterisk is 1.0.3, the linux is FC2. Please help me. ___ 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] Using SPANDSP for faxes
I installed spandsp on our asterisk server to get faxes. It works however the images are a little off. Sometimes a few pages will be together, pages missing and sentence missing. Is this normal for this program? Any input would be great. Thank You Eric ___ 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] Using SPANDSP for faxes
Eric Hall wrote: I installed spandsp on our asterisk server to get faxes. It works however the images are a little off. Sometimes a few pages will be together, pages missing and sentence missing. Is this normal for this program? Yes it is with some fax machines. We had to make our own program that take the image and sets it correctly for viewing. It's not a GPL program it's one we got as a test. If all goes well we will post what we are doing to fix the problems. The programmer says it has to do with the libtiff library's. Any input would be great. Thank You Eric ___ 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] Re: Cant set H323 up
Hi Now I do have compiled all the libraries, and added the load = chan_h323.so in the modules.conf file. Actually, now asterisk is attempting to load the chan_h323.so module. The problem is that Im getting this error now: [chan_h323.so]Dec 13 02:24:01 WARNING[12023]: loader.c:258 ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object file: No such file or directory I've moved the libpt_linux_x86_r.so.1.5.2 file to /usr/lib, /usr/lib/asterisk, /usr/lib/asterisk/modules After each move, I ran ldconfig the error was always the same... does anyone know where does asterisk looks for this file? Or if the cause for this is another? Im using the H323 channel included in the Asterisk tree. Thanks, RODOLFO Corvin wrote: Rafael J. Risco G.V. wrote: On Sat, 11 Dec 2004 16:49:12 +, Corvin [EMAIL PROTECTED] wrote: Dnia sobota, 11 grudnia 2004 15:32, Rodolfo Grave napisa: Hi. I need to set up H323 on an Asterisk box. I've succesfuly compiled the asterisk oh323 (including of course all the dependencies: PWlib and OpenH323), and then compiled asterisk. However, asterisk doesn't report a registered H323 channel (when it starts, it reports IAX2, ZAP and SIP channels, however, the 323 word doesn't appear in the whole output). Is there anything I'm missing? I've read the documentation on the wiki, and none said nothing about editing a config file. I did noticed that they talked about the oh323.conf file, which I dont have. BTW. you should check direcory with oh323 a there should be asterisk-driver directory and there you find sample config. Then you sghould load module in modules.conf. BTW. I can't still compile any h323 driver :(((. Corvin ___ 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] Re: Ethernet Channel Bank idea
On Sun, 12 Dec 2004 06:22:14 -0500 (EST), Greg Boehnlein wrote: On Fri, 10 Dec 2004, nik martin wrote: news.gmane.org wrote: Allied Telesyn VoIP Access Device http://www.alliedtelesyn.co.uk/site/files/documents/datasheet/VP624FXS_euro.pdf This is a 24-port FXS 1u device, conveniently presented as a single RJ-21 TELCO connector. yeah, but those are expensive as crap. i was thinking about something more competetive with a channel bank You know, if someone had some time on their hands, was good at hardware/software hacking and had the will, the old Livingston/Lucent PM3 platform would make an awesome 48 port IAX2 - PRI/T1 channel bank. Basically, the PM3 has 2 T1 ports that can be configured for ISDN PRI. The core of the system runs on an AMD x86 CPU. The plug in Modem cards have Lucent DSP's on them (up to 50 in a box). Flash size is 4 megs, and RAM is usually around 4 megs. That is still quite a bit of horsepower, and the boxes are under $400 now. The DSP's could be used for Codec Translation, if neccessary, or for echo cancellation. And, we can get access to the original Lucent ComOS Source code. Anyone game? :) It is not such a dumb idea !! I am using The ericsson tigris platform for our few remaining PRI's for our dialup pools. We are currently hanging asterisk off these using drop insert (both ways)... Now if we could actually get some code to get the voip working to/from asterisk we could do away the the PRI cards Gary . ___ 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] zaptel 0.9.1 compile problem
Done, I got it compiled. Looks like there was some things changed in kernel 2.6.9 that the newer versions of the zaptel driver have been modified for. On Dec 12, 2004, at 8:03 PM, Kristian Kielhofner wrote: Jay Austad wrote: I'm using gentoo 2004.3, and when I emerge the zaptel driver, compile fails with the following output: CC [M] /var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.o In file included from /var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.c:40: /var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/zaptel.h:55:2: warning: #warning Zaptel doesn't support DEVFS in post 2.4 kernels. Disabling DEVFS in zaptel /var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.c: In function `ztdeth_rcv': /var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.c:89: error: union has no member named `ethernet' make[2]: *** [/var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.o] Error 1 make[1]: *** [_module_/var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.9-gentoo-r9' make: *** [linux26] Error 2 Any ideas why this would be? ~jay Jay, Portage is good for many things, but Asterisk and Zaptel is not one of them. Get them both from ftp.asterisk.org and you will be much better off. -- Kristian Kielhofner ___ 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] zaptel 0.9.1 compile problem
Jay Austad wrote: I'm using gentoo 2004.3, and when I emerge the zaptel driver, compile fails with the following output: CC [M] /var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.o In file included from /var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.c:40: /var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/zaptel.h:55:2: warning: #warning Zaptel doesn't support DEVFS in post 2.4 kernels. Disabling DEVFS in zaptel /var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.c: In function `ztdeth_rcv': /var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.c:89: error: union has no member named `ethernet' make[2]: *** [/var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1/ztd-eth.o] Error 1 make[1]: *** [_module_/var/tmp/portage/zaptel-0.9.1/work/zaptel-0.9.1] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.9-gentoo-r9' make: *** [linux26] Error 2 Any ideas why this would be? ~jay Jay, Portage is good for many things, but Asterisk and Zaptel is not one of them. Get them both from ftp.asterisk.org and you will be much better off. -- Kristian Kielhofner ___ 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] PoE VOIP phones in Australia
Adam Goryachev wrote: See the polycom IP 300/500/600 phones. There are many resellers of these phones in Australia. Note the 300/500 require an additional cable for PoE. Are there any that have online stores? I've searched fairly extensively and can only find brochureware sites. Thanks, James Andrewartha ___ 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] patton smartnode integration
Any have any success using a patton smartnode 4118/js/eiu fxs gateway with asterisk? We we're able to get the unit to register with asterisk, but when trying to place a call, no codec was compatible, even though I had all of the following enabled on the patton ... # G.711 A-Law/µ-Law (64kbps) # G.726 (ADPCM 40, 32, 24, 16 kpbs) # G.723.1 (5.3 or 6.3 kbps) # G.729ab (8kbps) the link to this product is : http://commerce.patton.com/pe_products.asp?category=51MiDAS_SessionID=e41363efa86e409caf79ab1fd9b32e49 ? thanks for any help, Michael Lyszczek ___ 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: Cant set H323 up
what os are you running? K. - Original Message - From: Rodolfo Grave [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, December 13, 2004 1:27 AM Subject: Re: [Asterisk-Users] Re: Cant set H323 up Hi Now I do have compiled all the libraries, and added the load = chan_h323.so in the modules.conf file. Actually, now asterisk is attempting to load the chan_h323.so module. The problem is that Im getting this error now: [chan_h323.so]Dec 13 02:24:01 WARNING[12023]: loader.c:258 ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object file: No such file or directory I've moved the libpt_linux_x86_r.so.1.5.2 file to /usr/lib, /usr/lib/asterisk, /usr/lib/asterisk/modules After each move, I ran ldconfig the error was always the same... does anyone know where does asterisk looks for this file? Or if the cause for this is another? Im using the H323 channel included in the Asterisk tree. Thanks, RODOLFO Corvin wrote: Rafael J. Risco G.V. wrote: On Sat, 11 Dec 2004 16:49:12 +, Corvin [EMAIL PROTECTED] wrote: Dnia sobota, 11 grudnia 2004 15:32, Rodolfo Grave napisa: Hi. I need to set up H323 on an Asterisk box. I've succesfuly compiled the asterisk oh323 (including of course all the dependencies: PWlib and OpenH323), and then compiled asterisk. However, asterisk doesn't report a registered H323 channel (when it starts, it reports IAX2, ZAP and SIP channels, however, the 323 word doesn't appear in the whole output). Is there anything I'm missing? I've read the documentation on the wiki, and none said nothing about editing a config file. I did noticed that they talked about the oh323.conf file, which I dont have. BTW. you should check direcory with oh323 a there should be asterisk-driver directory and there you find sample config. Then you sghould load module in modules.conf. BTW. I can't still compile any h323 driver :(((. Corvin ___ 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] 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] TDM400P FXS polarity reversal?
On December 12, 2004 09:59 pm, Rich Adamson wrote:
Can you translate that into * code?
do_hangup()
{
if(chan-signalling = FXO_KS)
{
if(!chan-reversed)
{
setreg(chan-port, funky_do_register, getreg(chan-port,
funky_do_register) | BATT_REVERSAL);
set_timer(chan-reversetime);
chan-reversed = 1;
}
else
{
setreg(chan-port, funky_do_register, getreg(chan-port,
funky_do_register) !BATT_REVERSAL);
chan-reversed = 0;
}
}
}
??
Oh, it has to work... :-)
-A.
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Re: [Asterisk-Users] TDM400P FXS polarity reversal?
Soren Rathje wrote: Specs for Si3210 (TDM400P FXS Module) says on page 93: --- Register 72. On-Hook Line Voltage Bit 6 VSGN On-Hook Line Voltage. The value written to this bit sets the on-hook line voltage polarity (VTIPVRING). 0 = VTIPVRING is positive 1 = VTIPVRING is negative --- The (missing) link.. https://www.mysilabs.com/public/documents/tpub_doc/dsheet/Wireline/ProSLIC/en/si3210.pdf Features ... ... * Software programmable signal generation and audio processing: - DTMF generation and decoding - 12 kHz/16 kHz pulse metering generation - Phase-continuous FSK (caller ID) generation - Dual audio tone generators - Smooth and abrupt polarity reversal == NB! - µ-Law/A-Law and 16-bit linear PCM audio ... Can you translate that into * code? ___ 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] Log's Message Codes
Hi All, Anybody knows where can I find more explanation about the log's message codes of Asterisk? By the way, anybody had this VERY ANNOYING warning flooding the logs? WARNING[23678]: Read error on sound device: File descriptor in bad state With the default config of logger.conf it can reach 2GB in a few hours. What can it be? Thanks for all! Raul. ___ 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] Log's Message Codes
noload = chan_oss.so in modules.conf bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Raúl Gómez Cabrera Sent: Sunday, December 12, 2004 9:43 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Log's Message Codes Hi All, Anybody knows where can I find more explanation about the log's message codes of Asterisk? By the way, anybody had this VERY ANNOYING warning flooding the logs? WARNING[23678]: Read error on sound device: File descriptor in bad state With the default config of logger.conf it can reach 2GB in a few hours. What can it be? Thanks for all! Raul. ___ 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] New PRI with DID in US?
On Fri, 10 Dec 2004 17:26:48 -0600, Rich Adamson [EMAIL PROTECTED] wrote:
Just turned up a new PRI with DID's in the US. I'm receiving 5 digits
of the DID numbers as I requested.
Assuming I have 100 DID numbers but only define 50 of those in
extensions.conf, is there an easy way to send the incoming calls
for the 20 undefined numbers to a common resource (ivr, operator,
or canned message) without having to define each one?
I handle my DIDs with a macro. A DBget fetches a target for Goto. If
the key doesn't exist, it jumps to a hangup macro that can either drop
with PRI_CAUSE=1 (invalid) or play Zapateller and ss-noservice.gsm
twice, then hang up with PRI_CAUSE=31, depending on how you want it to
work. Of course, don't answer first, and if you do Playback(),
remember the noanswer option and play a silence/1 first..
Some samples from database show (numbers changed to protect the
guilty, and I receive full 10 digits)
/DID/9001235900 : mainmenu|s
/DID/9001235904 : 104
/DID/9001235917 : 117
/DID/9001235939 : 139
/DID/9001235942 : 142
/DID/9001235970 : 170
/DID/9001235949 : disa|s
[from-pstn]
exten = _NXXNXX,1,Macro(did,${EXTEN});
[macro-did]
exten = s,1,DBget(target=DID/${ARG1})
exten = s,2,Goto(${target},1); db should not include the priority
exten = s,102,SetVar(PRI_CAUSE=1); tells telco to play discon message
exten = s,103,Hangup;
; target can be simply an extension: my stdexten does more DBget to find channel
exten = _1XX,1,Macro(stdexten,${EXTEN})
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Re: [Asterisk-Users] I'm stumped
the files: I have asterisk running as the user asterisk also. who is the web server running as? isn't the system() function seeing this: sudo -u asterisk /usr/sbin/asterisk -rx database put cidname $PhoneNumber \$PhoneName\ when it should be seeing this after the sudo -u asterisk : asterisk /usr/sbin/asterisk -rx database put cidname (etc) -- whole cmd in quotes Try typing this at the linux prompt: asterisk -rx database put cidname 123 Julius Ceasar I don't think it will work ___ 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] Follow Me Music on hold
OK, I have an extension setup with a follow me like so: ;Operator Going to Sue first, then Mary exten = 0,1,playback(pls-wait-connect-call) exten = 0,2,Dial(SIP/103,20,mTt) exten = 0,3,Dial(SIP/102,20,mTt) exten = 0,4,VoiceMail([EMAIL PROTECTED]) exten = 0,5,Goto,t|1 This works well except for the fact that the music on hold stops after the first timeout and starts over at the beginning of the next line. What I mean is that the music sort of skips a beat (so to speak) when * stops ring extension 103 and starts ringing extension 102. Can someone suggest a better/smoother way to do this so the music just continues to play until both extensions timeout? -- Start Your Own ISP! http://www.YourOwnISP.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
Re: [Asterisk-Users] Very Cool.........Asterisk Made Wired Magazine
I suppose Asterisk is first going to get press as an über-geek's home-brew PBX, but I sometimes wish this weren't so. It is such a legitimate technology that it should be getting press in more industry publications. Ah well, that'll come in good time I guess. You're right there, Jim: if it bleeds, it leads is the motto of TV news, that gives an idea of the press mentality. I always thought Wired was more of a yuppie lifestyle magazine than one to attract phrackers or real hackers. Still, there may be an influx of newbies looking to spoof cid. If this happens, I hope questions will be answered without scaring anyone who is really interested away. With the upward spiral of voIP popularity and buzz, asterisk should be talked about in something like PC Magazine where suits look at stuff to buiy for their offices. Maybe someone should try to gather some soho case studies and offer them to a widely-read serious computer magazine, serious meaning one read by people who are either small biz owners or who have decisional power for small and mid level business strategy. ___ 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] Follow Me Music on hold
Me wrote: OK, I have an extension setup with a follow me like so: ;Operator Going to Sue first, then Mary exten = 0,1,playback(pls-wait-connect-call) exten = 0,2,Dial(SIP/103,20,mTt) exten = 0,3,Dial(SIP/102,20,mTt) exten = 0,4,VoiceMail([EMAIL PROTECTED]) exten = 0,5,Goto,t|1 This works well except for the fact that the music on hold stops after the first timeout and starts over at the beginning of the next line. What I mean is that the music sort of skips a beat (so to speak) when * stops ring extension 103 and starts ringing extension 102. Can someone suggest a better/smoother way to do this so the music just continues to play until both extensions timeout? -- Start Your Own ISP! http://www.YourOwnISP.com What about calling them both at the same time, not sequentially: exten = 0,1,playback(pls-wait-connect-call) exten = 0,2,Dial(SIP/103SIP/102,20,mTt) exten = 0,3,VoiceMail([EMAIL PROTECTED]) exten = 0,4,Goto,t|1 asterisk -rx show application Dial would have told you this! -- Kristian Kielhofner ___ 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] Digium Card Error
I have success installing and compiling, but if I reboot I have to modprobe again to get he drivers loaded for the module I am using. I am using rhes31 and a tdm card with one fxo and one fxs. tia ___ 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] Digium Card Error
Hi, Il giorno dom, 12-12-2004 alle 00:36 -0800, Charles S. Antrim ha scritto: I have success installing and compiling, but if I reboot I have to modprobe again to get he drivers loaded for the module I am using. I am using rhes31 and a tdm card with one fxo and one fxs. perhaps you have to build a script that loads modules on boot? see in zaptel src dir, there's a zaptel.sysconfig zaptel.init demo examples. Not installed by default. Matteo. -- ___ 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 I re-write an incoming caller-id?
Eric Wieling aka ManxPower wrote:
Dan Weber wrote:
check out application SetCallerID
Thanks that worked, ... for some lines only.
I have still the problem that I cannot get the caller-id from the pstn
line. Maybe I do something wrong with the procedure.
I changed in zapata.conf to all combinations:
cidsignaling =bell, v23 and dtmf
cidstart = ring and polarity
after saving I reloaded in the running asterisk console and tried a call.
There is no caller id to see!!!
A parallel phone to the pstn line shows the caller-id though. So it must
be just a setting!!!
Since I use 9 as the prefix to dial the pstn line I setup my
extensions.conf
exten = s,1,NoOp(${CALLERIDNUM})
exten = s,2,Wait(1)
exten = s,3, SetCallerId(9${CALLERIDNUM})
...
and see now the 9 on my display but not the callerid of the caller.
bye
Ronald
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Re: [Asterisk-Users] RE: Polycom 500 - Dialtone while connected
On Thu, 9 Dec 2004, Jorge Mendoza wrote: Andrei, I'm interested too. Any chance to put the archive in a ftp site?. Jorge Mendoza I am also interested in getting the 1.3.4 firmware. It annoys me that I can't just get it from Polycom's website, and forces me to rethink deploying their phones for customers. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] Totally LOST with dialplan and Extensions.
Ok I have spent the last week working on getting my small PBX to work.
I will in the end only have 4 SIP extensions being either softphones of
IP phones. Currently only 1 SIP config for testing.
And at the this point it should be all fairly easy with all inbound and
outbound to PSTN will be going Via Firefly/Freshtel.net in Australia via
IAX. Inbound does work in it's current basic state.
There will be NO ZAP devices, so I have ztdummy running.
I would say that for the outbound dialing I have either missed out
something plainly obvious or a simple typo which would be the challange.
I would think that all the problems are in the extensions.conf file
which really has me confused and totally lost.
I don't expect answers just pointers in the correct direction so that I
can get it to work for the outbound calling to work, I have the inbound
working which was a task but I was able with some pointers to get it
working.
I would like to thank you all for your casting experianced eyes to look
over this. What ever is worked out I will make sure the info gets onto
the Wiki for Freshtel and for a SIP to IAX to PSTN config so that others
can look up the basic configs to do this type of setup. There does not
seem to be from what I can find this basic configs for IAX without FXS
FXO devices.
cheers
David
SIP.CONF
[general]
context=default
realm=monitor.diversified.com.au
bindaddr=203.29.98.221
srvlookup=yes
maxexpirey=180
defaultexpirey=160
disallow=all
allow=speex
allow=gsm
allow=ilbc
allow=ulaw
allow=ilbc
[801]
type=friend
regexten=801
username=801
secret=password
callerid=801
host=dynamic
nat=yes
canreinvite=no
qualify=yes
disallow=all
allow=gsm
allow=speex
allow=ulaw
allow=alaw
IAX.CONF
[general]
tos=lowdelay
jitterbuffer=no
disallow=all
allow=speex
allow=ilbc
allow=gsm
allow=adpcm
allow=alaw
register = 89280250:[EMAIL PROTECTED]
register = 89280250:[EMAIL PROTECTED]
[guest]
type=user
context=default
auth=none
;inbound
[firefly]
type=friend
host=cts-au.freshtel.net
context=default
; outbound
; Firefly (Freshtel)
[89280250] ; Firefly
context=89280250
qualify=no
username=89280250
secret=password
auth=md5
type=friend
host=gateway.freshtel.net
EXTENSIONS.CONF
[general]
static=yes
writeprotect=no
[globals]
SpeakingClock=123
[default]
exten = s,1,Wait,1
exten = s,n,Answer
exten = s,n,DigitTimeout,5
exten = s,n,ResponseTimeout,10
exten = s,n,WaitExten
exten = s,n,Dial(SIP/801)
exten = 13,1,DateTime()
exten = 13,2,Wait(1)
exten = 13,3,DateTime()
exten = 13,4,Hangup
exten = t,1,Goto(#,1)
exten = i,1,Playback(invalid)
exten = 600,1,Playback(demo-echotest)
exten = 600,n,Echo
exten = 600,n,Playback(demo-echodone)
exten = 600,n,Goto(s,6)
exten = ${SpeakingClock},1,Wait(1)
exten = ${SpeakingClock},2,setvar(FutureTime=$[${EPOCH} + 10])
exten = ${SpeakingClock},3,Wait,3
exten = ${SpeakingClock},4,SayUnixTime(${FutureTime},,R)
exten = ${SpeakingClock},5,playback(vm-and)
exten = ${SpeakingClock},6,SayUnixTime(${FutureTime},,S)
exten = ${SpeakingClock},7,playback(seconds)
exten = ${SpeakingClock},8,playback(beep)
exten = ${SpeakingClock},9,wait(2)
exten = ${SpeakingClock},10,goto(1)
exten = _394.,1,SetCallderId(89280250)
exten =
_394.,2,Dial(IAX2/89280250:[EMAIL PROTECTED]/${EXTEN:3},60,r)
[outgoing-firefly-peers]
exten = _62,1,Macro(outgoingfirefly,${EXTEN:2},70) ; Firefly
[macro-outgoingfirefly]
exten = s,1,SetCallerID(89280250 89280250)
exten =
s,2,Dial(IAX2/89280250:[EMAIL PROTECTED]/${ARG1},${ARG2},r)
exten = s,3,Congestion
[macro-outgoingfreshtel]
exten = s,1,SetCallerID(89280250 89280250)
exten =
s,2,Dial(IAX2/89280250:[EMAIL PROTECTED]/${ARG1},${ARG2},r)
exten = s,3,Congestion
___
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[Asterisk-Users] ACK from asterisk not matched to transaction by SER / LCS2005
Thanks for the info ! Is there any way to work around the bug, maybe by rewriting the SIP Message in SER ? Or some kind to temporary third party patch ? Chris. -- Message: 9 Date: Sun, 12 Dec 2004 10:34:53 +0100 From: Olle E. Johansson [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ACK from asterisk not matched to transaction by SER / LCS2005 To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Public Dump wrote: For reasons unknown to me, SER and subsequently a Microsoft Live Communcations Server 2005 seems to have problems, matching a SIP ACK request from asterisk to the ongoing SIP transaction, I have attached the complete log, but the essential lines are: That's a bug in Asterisk that is in the bug tracker and needs to be fixed. Asterisk is sending the ACK to the AOR in the invite, not to the contact address in the 200 OK. Regards, /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] TDM400P FXS polarity reversal?
Is it possible to do a talk battery polarity reversal on a TDM400P FXS interface? Everything I can find seems to be referring to the procedure for detecting a battery reversal on a telephone company POTS line using the FXO interface, but not for actually generating one back to a station upon answer supervision. I would assume that VoicePulse and VoipJet provide a way of signaling far-end supervision back to the originating Asterisk PBX... Basically, my two questions are: (1) Is the hardware capable of even performing a reversal? (2) If the above is true, how would you make it happen in Asterisk? The Silicon Labs spec sheet does not specifically indicate generating a reversal is possible. Therefore, best guess is the integrated circuits on the card does not support it, therefore asterisk has no means of doing it. ___ 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: Cant set H323 up
Rafael J. Risco G.V. wrote: On Sat, 11 Dec 2004 16:49:12 +, Corvin [EMAIL PROTECTED] wrote: Dnia sobota, 11 grudnia 2004 15:32, Rodolfo Grave napisa: Hi. I need to set up H323 on an Asterisk box. I've succesfuly compiled the asterisk oh323 (including of course all the dependencies: PWlib and OpenH323), and then compiled asterisk. However, asterisk doesn't report a registered H323 channel (when it starts, it reports IAX2, ZAP and SIP channels, however, the 323 word doesn't appear in the whole output). Is there anything I'm missing? I've read the documentation on the wiki, and none said nothing about editing a config file. I did noticed that they talked about the oh323.conf file, which I dont have. BTW. you should check direcory with oh323 a there should be asterisk-driver directory and there you find sample config. Then you sghould load module in modules.conf. BTW. I can't still compile any h323 driver :(((. Corvin ___ 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 capture -1 return on Dial command
following setup, and I want to be able to process the audio file after the outbound call has been done regardless how how it ends. would the hangup priority be appropriate for this? ___ 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] four wildcards in a single pc
[EMAIL PROTECTED] wrote: Hi. Please excuse me asking this again. But it really puzzles me. It is puzzling, no denying it. The development team is still struggling with these issues, and so far there has not been found a foolproof solution (at least I can't recall having seen one). We're installing asterisk at a branch office at NJ (HQ is at Petach-Tikva) It'll need to support 5 POTS lines, 11 analog extensions and four VOIP phones. I wanted to go with a T1 card from digium and a channel bank, but we have a dead line. It has to be up and running by January 1st. I don't have the time to start shopping at ebay, where you don't know what you'll get, and you need to install, under time pressure something you not familiar with. For sure, but you should also consider the experience of those who have been there before you. So I thought of installing a combination of four pci cards in the machine, and everybody on the list just keeps telling me it won't work. It _might_ work, but it is almost guaranteed not to work _well_. The Digium PCI cards are rather different from any PCI card you may have used in the past. I have installed successfully more then four cards in a machine before. I had a firewall with eight network interfaces (one quad card, one duo and two singles) I have machines with two dialogic boards, a pci display card, and a network interface. And I know I've had machines at home that had a display adaptor, modem, network, scsi, and soundblaster all together. Yep, so have we all. The thing is, just ONE of these Digium cards will request more interrupts all by its lonesome than every single PCI card you've ever installed in all the machines you've ever owned, all put together! Well, perhaps not, but seriously, these things work very differently from any PCI card you've ever seen before. Yet, people claim it won't work because of lack of IRQs, and that it's not related to Digium. That isn't strictly correct, but the problem does pertain to IRQs. OK, look, you _might_ be able to free up enough IRQs on a PIC-based motherboard -- if you disable the serial ports, mouse, parallel port and USB. It's not recommended, but it's theoretically possible. And if you have a MoBo that is APIC-compliant, you should be able to have all the IRQs you can handle, so lack of IRQs doesn't need to be an issue (make sure you have a BIOS and chipset that's up to the task). BUT . . . Getting dedicated IRQs for the cards is a minor problem compared to what happens when you have four cards hammering away mercilessly at the chipset and CPU of your motherboard; 1000 IRQs per second, per card. Nobody's really sure what's wrong, but it causes problems for pretty nearly everyone. What everyone here is saying is that we're all pretty sure you're gonna run into problems; problems that could easily be avoided by avoiding the whole TDM400 mess in the first place. What am I missing? The Digium cards are unique in the world of telephony, because instead of having an expensive DSP chip on board, they use the CPU to provide this functionality. The challenge comes from the fact that voice is intolerant of delay. In order to ensure that the voice processing that goes on in the CPU is handled with no perceivable delay, the zaptel cards have to establish a kind of pseudo-synchronous clocking with the CPU. Unfortunately, the signalling bus on a PC isn't synchronous, at least not in that way. The clock that the zaptel cards use is the IRQ of the card, literally requesting the CPU interrupt what it's doing and pay attention to it 1000 times per second, regardless of what it's doing. You are proposing the use of FOUR of these cards. Since this has caused trouble for nearly everyone who has tried it, everyone is suggesting that you might want to give the matter some careful thought. There _are_ less painful ways. Cheers, Jim. ___ 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] [OT] Small SIP phones?
Hi On Sun, 2004-12-12 at 21:27, Antony Stone wrote: Thanks. In fact I already have a Grandstream ATA-486, which I'm very pleased with: http://www.grandstream.com/y-ht486.htm This unit is even smaller - 105 x 75 x 25mm (or 4 x 2.75 x 1), however I'm just wondering if there's a neat all-in-one solution, instead of carrying around two items? Three, in fact. The powersupply also adds to the required space. This is one of the biggest advantages of having an all-in-one solution, because you don't have to generate high voltage/high power ring signalling. Florian ___ 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] [OT] Small SIP phones?
On Sunday 12 December 2004 23:08, Shoval Tomer wrote: I may be wrong, but if you always carry your laptop around, why don't purchase a USB handset? The main reason is that (I believe) the quality of audio with a soft phone is generally not as good as that from a real SIP phone? The other reason is that I want to be able to show VoIP in operation to clients (which is where I would be taking the phone with me), so a standalone phone, which is not dependent on any software installed on my laptop, is a much neater arrangement. -Original Message- From: Florian Overkamp [mailto:[EMAIL PROTECTED] On Sun, 2004-12-12 at 21:27, Antony Stone wrote: Thanks. In fact I already have a Grandstream ATA-486, which I'm very pleased with: http://www.grandstream.com/y-ht486.htm This unit is even smaller - 105 x 75 x 25mm (or 4 x 2.75 x 1), however I'm just wondering if there's a neat all-in-one solution, instead of carrying around two items? Three, in fact. The powersupply also adds to the required space. This is one of the biggest advantages of having an all-in-one solution, because you don't have to generate high voltage/high power ring signalling. True, you don't need the high voltage ringing, but with a standard SIP phone you still need a PSU for it. I couldn't rely on a client's network switch supporting PoE for when I wanted to plug one in. Regards, Antony. -- Wanted: telepath. You know where to apply. Please reply to the list; please don't CC me. ___ 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] Cisco AS5XXX to asterisk debugging.
Hi thanks for your help . I do not have direct access to the Cisco, but I believe that he is AS5300 The ios version is 12.2 and the cisco dum config is: GWSCZ01en Password: GWSCZ01#sh run Building configuration... Current configuration : 5053 bytes ! ! Last configuration change at 05:17:58 UTC Mon Apr 16 2001 ! NVRAM config last updated at 12:06:13 UTC Sat Apr 14 2001 ! version 12.2 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname GWSCZ01 ! no boot startup-test logging queue-limit 100 ! ! ! resource-pool disable spe default-firmware spe-firmware-1 ip subnet-zero ip cef no ip domain lookup ! isdn switch-type primary-net5 ! ! voice service voip fax protocol t38 ls-redundancy 2 hs-redundancy 2 fallback none sip ! voice class codec 11 codec preference 1 g729br8 codec preference 2 g729r8 codec preference 3 gsmfr codec preference 4 g726r32 codec preference 6 g726r16 codec preference 7 g723r63 codec preference 8 g723r53 codec preference 9 g726r24 codec preference 10 g723ar63 codec preference 11 g723ar53 codec preference 12 g711ulaw codec preference 13 g711alaw codec preference 14 clear-channel ! ! ! ! ! ! ! no voice hpi capture buffer no voice hpi capture destination ! voice source-group cisco access-list 8 carrier-id target cisco ! ! ! fax interface-type fax-mail mta receive maximum-recipients 0 ! ! ! controller E1 7/0 framing NO-CRC4 line-termination 75-ohm ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled cas-custom 0 country bolivia ! controller E1 7/1 line-termination 75-ohm pri-group timeslots 1-31 ! controller E1 7/2 line-termination 75-ohm pri-group timeslots 1-31 description Embratel --More-- ! ! interface FastEthernet0/0 ip address y.y.y.y 255.255.255.224 duplex auto speed auto no cdp enable h323-gateway voip interface h323-gateway voip id GK01 ipaddr y.y.y.z 1719 h323-gateway voip h323-id GWSCZ01 h323-gateway voip tech-prefix 2032# ! ! ip classless ip route 0.0.0.0 0.0.0.0 y.y.y.v no ip http server ! ! ! ! ! ! call rsvp-sync ! voice-port 7/0:0 compand-type a-law ! voice-port 7/1:D ! voice-port 7/2:D ! voice-port 7/3:0 compand-type a-law ! voice-port 7/4:0 compand-type a-law ! voice-port 7/5:0 ! ! mgcp profile default ! dial-peer cor custom ! ! ! dial-peer voice voip destination-pattern 44T voice-class codec 11 session protocol sipv2 session target sip-server session transport udp ! dial-peer voice pots destination-pattern T direct-inward-dial port 7/0:0 ! sip-ua retry invite 3 retry cancel 2 sip-server ipv4:x.x.x.x ! On Sun, 2004-12-12 at 20:07, Hatzis, Michael wrote: What's the cisco box,52 / 53; version ios? can you post a config dump? Regards Michael Hatzis 0421 476 211 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Verastegui G Sent: Monday, 13 December 2004 10:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Cisco AS5XXX to asterisk debugging. Excuse the insistence but I am more than one week with this problem, and I do not have any idea to solve it. You know if the configuration with GK in the Cisco, can be interfering with the RTP traffic? Thanks in advance On Fri, 2004-12-10 at 08:37, Tenorio, Leandro wrote: Pls, post your Cisco and * config files. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Verastegui G Sent: Friday, December 10, 2004 12:30 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco AS5XXX to asterisk debugging. Hi, I have a serious problem to configure Cisco AS5XXX and Asterisk , I trying to use asterisk for PSTN(A) Cisco AS5xxx ASteriskPSTN(B) (No Nat, no Firewall) I hear (on the PSTN(A)) clearly what the other person is saying, but the other person (on the PSTN(B) side) hears nothing from PSTN(A). I use tcpdump for debug de rtp trafic, and ouput contains 19:06:00.741293 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:00.763133 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:00.740415 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:00.810312 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:00.860314 IP (tos 0x0, ttl 64, id 179, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length 32 19:06:00.980351 IP (tos 0x0, ttl 64, id 180, offset 0, flags [DF], proto 17, length: 60) x.x.x.x.19926 y.y.y.y.18974: [no cksum] UDP, length
[Asterisk-Users] Any plans for video in oh323?
I did my happy first install of asterisk (cvs), and everything is working great so far, with one exception. Since I need h323 support, I first built chan_h323 with openh323 and pwlib pandora, and while the build went ok usage did not. More specifically, while asterisk would accept h323 calls, no voice was transmitted, hangup of the client was not recognized and the server didn't properly shut down any more. So I switched to oh323 with openh323 and pwlib janus, again the build went alright, and this time usage did too :) The only thing (obviously) missing from oh323 is video transport, hence my question: is this feature planned in the near resp. far future? I guess quite some people would be happy to see it included ... Thanks, Bruno. ___ 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] Sipura SPA-2000 won't ring
I had a Grandstream 286 at my home hitting my Asterisk box at the office, all worked well and I received phone calls fine until the device just up and died. I replaced this unit with an SPA-2000 because I have been impressed with the Sipura devices and decided to use them for most of my needs in the future. Problem is that my phone attached to the device rings shortly after power up of the device but seems to lose it's head after a period of time and stops ringing until I power cycle the unit or reboot it. My Asterisk config is the same regarding NAT for this extension and I have the Sipura registering with * so I am at a loss as to why Asterisk loses or stops ringing this device. I have dug around and can't seem to solve this issue so far, any help would be appreciated. -- Start Your Own ISP! http://www.YourOwnISP.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
RE: [Asterisk-Users] Very Cool.........Asterisk Made Wired Magazine
[EMAIL PROTECTED] wrote: Hi Guys, The article They've Got Your number in the Dec 2004 issue of WIRED magazine mentions Asterisk PBX (on p.100). The article is about phone phreaks hijacking cell phones with Bluetooth technology along with spoofing CID to pull some clandestine hacks on the PSTN. Anyhow, Asterisk is mentioned as the PBX of choice for an outfit: Telephreaks.org out of Florida that has built their own free VoIP service. I wish they had said a bit more about the usefulness of Asterisk, as opposed to talking about how some phrackers are using it to spoof CID. Then again, it's been said that there's no such thing as bad press. Quotes: Slestak, Da Beave, and GiD are the crew behind Florida-based Telephreaks.org, a free VoIP service that they've built to run on a roll-your-own, open source private branch exchange (PBX) system called Asterisk But with Asterisk, there's no need for the phone company to manage your phone lines anymore. Your can do it yourself. Well, it's good that Asterisk made WIRED magazine but really it should be on the front page with Mark's smiling mug on the cover like Linus' was on the Nov 2003 issue of WIRED. Ya, but that'll come in good time. There's no stopping Asterisk now! This is a good day for Asterisk because WIRED magazine has a huge subscriber base and this article will be read by a lot of people, and some of those people might have the gumption to check out Asterisk and see what it's all about. If you all see a lot more newbie question pop up in the next few weeks, be kind, help where you can and point to the Wiki for more info. Good advice at any time. Doubly so now. I'll be writing a letter to the editor, encouraging them to check out Asterisk as a possible full featured article and I encourage you all to do the same, if you're so inclined. Call me paranoid, but you might want to give some thought to the fact that the story you want them to write about Asterisk may not spin the way you think it will. Asterisk has been gaining a reputation as a Phrackers PBX, and that kind of reputation may increase interest, but not necessarily for the better. As Asterisk begins to take market share away from traditional Telephony platforms, the FUD will increase as well. The press serve their own ends, and if they decide that the better spin is a story about the dangers of hackers, they can use Asterisk's strengths against it. Remember that the press stole the term hacker away from the community and turned something good into something negative. Perhaps Wired is above all this, but security is a hotter topic than open-source, and FUD sells magazines. I suppose Asterisk is first going to get press as an über-geek's home-brew PBX, but I sometimes wish this weren't so. It is such a legitimate technology that it should be getting press in more industry publications. Ah well, that'll come in good time I guess. Cheers, Jim. ___ 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] Cisco AS5XXX to asterisk debugging.
I do not have direct access to the Cisco, but I believe that he is AS5300 A show ver will confirm this and the IOS release. Please publish that info as well. I will see if I can get a similar setup going in our lab. M ___ 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] DUNDi performance
Hello, I have a weird problem. My * server, a Pentium Celeron 1200 with 512 MB Ram and a Digium E100P card, is performing very well for IAX2, SIP and ZAP communication. There is no delay in transcoding, no packet loss etc etc. Now I added DUNDi, and I added +/- 8 peers in the dundi-test context and 1 peer in the GPA-bound e164 context. My server shows all but 1 peer as OK. DUNDi Ping times are between 20 and 200 ms. The Problem is, that no server but one can get a stable connection via DUNDi to my server. DUNDi ping times for my server are between 3000 an 7000 ms. Most servers have qualify of 2000ms, some even 500ms, so my server is quite always UNREACHABLE for those peers. When I activate DUNDi DEBUG, I can see that incoming DUNDi packets do take all long time before they show up in the * console, and they always show up with a whole bunch of others (filling 2-3 screens). But then the debug output stops in the middle of 1 debug packet, to continue over 20 seconds later (if it continues). The actual CPU load is load average: 0.00, 0.00, 0.00. I cannot find the problem, maybe someone over can help me! Thanks, Marc -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2727 3060 -- LuxAdmin powered service --- http://www.Gateway.lu Your gateway to the net Advantages of ADSL solutions by LuxAdmin: - price: cheap and clear - products: proven quality - support: friendly and helpful --- ___ 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
