[asterisk-users] voicemail issue
the last thing i was trying to do was change the default password to same as voicemail. i also tried reversing these changes but doesnt work. this is my log. i should probably mention that im running trixbox 1.21. when i connect to the voicemail system remotely, i enter the username, then a password and thats when this comes up. Core debug is at least 1 -- Executing Macro("Local/[EMAIL PROTECTED],2", "hangupcall") in new stack -- Executing ResetCDR("Local/[EMAIL PROTECTED],2", "w") in new stack -- Executing NoCDR("Local/[EMAIL PROTECTED],2", "") in new stack -- Executing Wait("Local/[EMAIL PROTECTED],2", "5") in new
stack -- Executing Hangup("Local/[EMAIL PROTECTED],2", "") in new stack
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Re: [asterisk-users] password for vm users
how about password strength? or remembering and not allowing password? or password duration?Marco Mouta <[EMAIL PROTECTED]> wrote: just set your initial password to be equal to vm-account number, and Voicemail application will do that for you and will request users to setup a new password! On 10/9/06, stan ford <[EMAIL PROTECTED]> wrote:how does one force mandatory password change on login? and a period of time to pass before mandating a password change? im using trixbox so if you have that info. that would be good, if you have it for asterisk, im sure i could figure it out as well thx. Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,Marco Mouta ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] password for vm users
easy enough. thanks!Marco Mouta <[EMAIL PROTECTED]> wrote: just set your initial password to be equal to vm-account number, and Voicemail application will do that for you and will request users to setup a new password! On 10/9/06, stan ford <[EMAIL PROTECTED]> wrote:how does one force mandatory password change on login? and a period of time to pass before mandating a password change? im using trixbox so if you have that info. that would be good, if you have it for asterisk, im sure i could figure it out as well thx. Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,Marco Mouta ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Get your email and more, right on the new Yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] password for vm users
how does one force mandatory password change on login? and a period of time to pass before mandating a password change? im using trixbox so if you have that info. that would be good, if you have it for asterisk, im sure i could figure it out as well thx. Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] commercial asterisk
anyone have experience with IntuitiveVoice's Asterisk system? Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: failed registration
what about the interval of the registration? is 2 minutes too often? Dovid B <[EMAIL PROTECTED]> wrote: Timed out from what I have seen comes from either a poor internet connection or a problem with your ITSP.- Original Message - From: stan ford To: asterisk-users@lists.digium.com Sent: Friday, October 06, 2006 4:42 AM Subject: [asterisk-users] Re: failed registration stan ford <[EMAIL PROTECTED]> wrote: i have this issue with failed registrations with my sip provider. it doesn't happen often, but it does happen. it also happens with 2 different vsp providers, so i dont think its them. this happens maybe 8 times a day, but then that doesn't sound too bad considering it registers itself every 2 minutes. im using trixbox 1.1 and have grandsream 101 SIP phones. thanks alot. Oct 4 06:12:02 NOTICE[2831] chan_sip.c: Failed to authenticate on REGISTER to '[EMAIL PROTECTED]' (Tries 3)Oct 4 06:12:19 NOTICE[2831] chan_sip.c: -- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #4) Stay in the know. Pulse on the new Yahoo.com. Check it out. __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail. Get your email and more, right on the new Yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: failed registration
stan ford <[EMAIL PROTECTED]> wrote:i have this issue with failed registrations with my sip provider. it doesn't happen often, but it does happen. it also happens with 2 different vsp providers, so i dont think its them. this happens maybe 8 times a day, but then that doesn't sound too bad considering it registers itself every 2 minutes. im using trixbox 1.1 and have grandsream 101 SIP phones. thanks alot. Oct 4 06:12:02 NOTICE[2831] chan_sip.c: Failed to authenticate on REGISTER to '[EMAIL PROTECTED]' (Tries 3)Oct 4 06:12:19 NOTICE[2831] chan_sip.c: -- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #4) Stay in the know. Pulse on the new Yahoo.com. Check it out. __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] failed registration
i have this issue with failed registrations with my sip provider. it doesn't happen often, but it does happen. it also happens with 2 different vsp providers, so i dont think its them. this happens maybe 8 times a day, but then that doesn't sound too bad considering it registers itself every 2 minutes. im using trixbox 1.1 and have grandsream 101 SIP phones. thanks alot. Oct 4 06:12:02 NOTICE[2831] chan_sip.c: Failed to authenticate on REGISTER to '[EMAIL PROTECTED]' (Tries 3)Oct 4 06:12:19 NOTICE[2831] chan_sip.c: -- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #4) Stay in the know. Pulse on the new Yahoo.com. Check it out. __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] t1 voip to failover pri
if i went with an SDSL line, don't those lines hook up to a common point, the DSLAM? i do like this idea of faling over not to a pri but another cheaper high speed line.adebayo omo-dare <[EMAIL PROTECTED]> wrote:I wouldn't first presume that there is any law that states you have to run VoIP over a PRI. Other technologies exist such as SDSL - with some providers speaking of crazy prices such as £65pcm for 2Mbs 5:1 contention ration and £100 for 1:1 (U.K) - should be even less in the states if that's where you are. One option, but not the only one, would be to drop your pri when your contract ends and take up SDSL - and voila an initial saving, in your case, of a 000 or more in the year. You could also have two SDSL lines for a little less than the price of the PRI. Both lines would not only serve for High Availability -possibly even better availability than single PRI- but could also, actively, both switch traffic, giving you 4Mbps of bandwidth for your VoIP, or if you choose, some other requirement while not required as failover - all for the price of less than one PRI. Then there is compression - 64k non negotiable, per channel for PRI, and flexible -i.e., less the 64k- for VoIP (International high quality Calls are transported at 16k), giving you the capacity to potentially service more traffic with less initial outlay. Other real cost efficiencies come in the form of the fact that IP-to-IP (local/national/international) calls are free. So if you have a lot of inter-branch communications, or communications you can switch on to IP, you can totally erradicate this cost - unlike with the PRI where you will still be subject to payment. Think like this - say I have two offices - one in london and the other New York. How much will I save by moving my calls on to VoIP with no per-time or call setup charges. Features related to OAM&P, can also be faster and cheaper with you having a lot more power in your hands. In real senses, and with regards to reliability, you should take in to consideration the great moves currently being made by telecom companies (incumbents most especially), with regards to a complete shift to NGNs, which have a strong focus on ToIP. With new fiber (FTTP), new technology, etc, a lot of networks are highly reliable at the present moment - I guess this would also depend on where you are. The thing about it is that complete IP networks in terms of telecom now look inevitable. And whether you do it yourself or it is done for you - it is the way things, many expect, are going to be in the next 5 or so years. stan ford <[EMAIL PROTECTED]> wrote:I'm confused with something, maybe someone can explain to me. if your currently on a pri and are considering moving over to VOIP, that means you would have to purchase a t1 or fractional t1 for a your voip connections. but then, voip connections aren't as reliable as PRI. so then you would probbaly have to get a PRI failover. but then having a PRI failover means that you now have to pay 400 for a T1, then another 400 for your PRI line. wouldn't have you have just defeated the cause of savig money by now having to have a PRI on standby? now costing you 800 a month? wouldn't it almost be the same price to stick with the PRI only? is anyone out there, using a VOIP only with no failover? __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Now you can scan emails quickly with a reading pane. Get the new Yahoo! Mail.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users All-new Yahoo! Mail - Fire up a more powerful email and get things done faster.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] t1 voip to failover pri
I'm confused with something, maybe someone can explain to me. if your currently on a pri and are considering moving over to VOIP, that means you would have to purchase a t1 or fractional t1 for a your voip connections. but then, voip connections aren't as reliable as PRI. so then you would probbaly have to get a PRI failover. but then having a PRI failover means that you now have to pay 400 for a T1, then another 400 for your PRI line. wouldn't have you have just defeated the cause of savig money by now having to have a PRI on standby? now costing you 800 a month? wouldn't it almost be the same price to stick with the PRI only? is anyone out there, using a VOIP only with no failover? __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] suggest a configuration
I have to setup a pbx system for a company, can someone suggest a configuration. Currently their phone bill is 1600 a month Currenlty 27 phone lines 1/2 of the calls are long distance I'd like the savings of a voip network, but also the reliability of a pstn/pri. How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pay as you go t.t38 fax termination and origination
no, i have to retain my fax numbers hmm. thanks anyways.John Kington <[EMAIL PROTECTED]> wrote: At 12:15 AM 9/28/2006 -0700, you wrote:>i can't for the life of me find a pay as you go termination and >origination service.>>there's garfachi, but they don't offer DID's in anywhere else other than >CA. Any suggestions? Thanks.They do offer tollfree DID. What about using one of those?Regards,John___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users All-new Yahoo! Mail - Fire up a more powerful email and get things done faster.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pstn failback
Lacy, can you confirm what i was saying about SIP Phones. if i fail from my voip connection to my pri, would i need to swap out my SIP phones with another type of digital phone?Lacy Moore - Aspendora <[EMAIL PROTECTED]> wrote: a couple things, if you guys could clear up for me. A) If i have a bunch of sip ip phones, and if i were to fail to my PRI. I should still be able to use my ip phones right? i assume the signal will be changed by my T1 card? and the reverse i would assume is true for incoming calls. Outgoing could be handled by the dial plan. Incoming would have to be something worked out with the providers. B) How about DID's, how would that be handled. is there a DID failover as well? I have my VOIP service with one company, if i had my PRI service with another. how would those DID's get failed to the other provider, if thats even possible at all in a timely manner. Not sure how this would be handled. C) also are failover pri's generally cheaper that their active counterparts? thanks alot. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] pstn failback
thanks for the responses a couple things, if you guys could clear up for me. A) If i have a bunch of sip ip phones, and if i were to fail to my PRI. I should still be able to use my ip phones right? i assume the signal will be changed by my T1 card? and the reverse i would assume is true for incoming calls. B) How about DID's, how would that be handled. is there a DID failover as well? I have my VOIP service with one company, if i had my PRI service with another. how would those DID's get failed to the other provider, if thats even possible at all in a timely manner. C) also are failover pri's generally cheaper that their active counterparts? thanks alot.Shawn Kelley <[EMAIL PROTECTED]> wrote: Stan,I agree with the comment below, we switched from analog lines to a PRI andit's not always as reliable as some people think. We are in a somewhat rurallocation and we have outages regularly. 1-4 hour outages every few monthsare not uncommon for us. Outages of 60 seconds or so are even more common.I'm told this is because the T1 line is running somewhat noisy/dirty andafter so many CRC errors the equipment is resetting.Make sure you negotiate a good SLA so that you can get credit when it doesgo down!You also have to be careful like mentioned below, if you get 2 PRI's, evenfrom different CLECS, the will normally still come out of the same CentralOffice and travel side by side on the cable. So it's likely if 1 goes downthen the other one will also. Summary: It's a good idea to have a few analog lines since they can take awhole lot more abuse than the digital T1 can handle. (Static/Noise doesn'tmake your call drop!)-Original Message-From: Lacy Moore [mailto:[EMAIL PROTECTED] Sent: Friday, September 29, 2006 4:23 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] pstn failback-BEGIN PGP SIGNED MESSAGE-Hash: SHA1stan ford wrote:> On fonalities web page, i see they offer pstn failback as a feature oftheir asterisk package. i've also heard before of failing back to a pri lineif your t1 voip line fails. my question is. in order to have pstn or prifailback, dont you basically have to have all the equipment there onstandby, a PRI line, TDM cards, PRI/T1 cards, a bunch of digital or analongphones. it just seems like a whole lot of hardware to be sitting therewaiting for a disaster. unless im just not understanding pstn/pri failback.can someone shed some light?> > also, if you've got a dedicated full t1 line for voice, and have a lowamount of users for that t1, is there really to worry about failing back toa pstn? seeing how reliable a t1 is. is anyone out here using full voiptelelphony solution only?> Having had our XO connection go down within a week or so of switching toa PRI, I can see how having a fallback would be good. It was down forabout 4 hours. However, that was back in May or June and hasn't beendown since. I couldn't justify having something on standby for ourbusiness. But, our clients can reach us by phone in the office or cell,and we can easily make outgoing calls on our cellphones. Our type ofbusiness is not really dependent on absolutely having the phones up 100%of the time.If you do get a fallback, get it from different providers. If your T1from provider X is down, chances are a PRI from provider X will befollowing the same path and be down as well.-BEGIN PGP SIGNATURE-Version: GnuPG v1.4.5 (MingW32)Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.orgiD8DBQFFHOX9VVXe/Qmwk9QRArSYAKDJN6Tf/L+L3ruXyXYcAeVbIyMxBwCgi2wMYwcV6yYYJX2cVly2z0dsdZ4==+Ik2-END PGP SIGNATURE-___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users All-new Yahoo! Mail - Fire up a more powerful email and get things done faster.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pstn failback
On fonalities web page, i see they offer pstn failback as a feature of their asterisk package. i've also heard before of failing back to a pri line if your t1 voip line fails. my question is. in order to have pstn or pri failback, dont you basically have to have all the equipment there on standby, a PRI line, TDM cards, PRI/T1 cards, a bunch of digital or analong phones. it just seems like a whole lot of hardware to be sitting there waiting for a disaster. unless im just not understanding pstn/pri failback. can someone shed some light? also, if you've got a dedicated full t1 line for voice, and have a low amount of users for that t1, is there really to worry about failing back to a pstn? seeing how reliable a t1 is. is anyone out here using full voip telelphony solution only? thanks. Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] t1-pri or sip trunk?
if you have to setup an office of 100 users now. would you rather setup a sip trunk,a t1-pri, or even a t1? and why? thx Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pay as you go t.t38 fax termination and origination
i can't for the life of me find a pay as you go termination and origination service. there's garfachi, but they don't offer DID's in anywhere else other than CA. Any suggestions? Thanks. Stay in the know. Pulse on the new Yahoo.com. Check it out. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: FXO module in TDM400P (UK, BT) - Hangup detection failing
On Tue, Nov 02, 2004 at 10:58:39PM +, StrUK wrote: > I guess my question is: does anyone have polarity reversal hangup > detection working on a BT line with an fxo module in a TDM400P? Testing with my fxo module shows that it takes about 8 seconds from pressing hangup on the mobile phone before hangup is detected in asterisk (maybe this relates to your 9 seconds). I'm not sure if this is when the local exchange sends the polarity reversal or if asterisk is detecting a hangup in some other way. I have busydetect=no and callprogress=no so I'm (AFAIK) not using any "fake disconnect functions" as per the keypad lights page[1]. If I pick up the phone during the 8 seconds I get a usable dialtone, which might be worrying if I didn't want people calling certain numbers. This seems to indicate that the local exchange knows that the mobile has hungup, as it wouldn't give dialtone in the middle of call. So perhaps I dont have polarity reversal hangup detection working, but HTH anyway. I'm not using the fxo callerid and asterisk is from cvs checkout date 14-Aug-2004 (as downloaded from bri-stuff.0.1.0-RC4a) my zapata config for the fxo module: group=1 context=fxo signalling=fxs_ks usecallerid=no callwaitingcallerid=no threewaycalling=no cancallforward=no transfer=no immediate=no callwaiting=no usecallingpres=no callreturn=no echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=2.0 busydetect=no callprogress=no channel=6 [1] http://www.voip-info.org/tiki-index.php?page=Asterisk%20Disconnect%20Supervision ___ 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] Hardware
Look for support by whatever operating system you plan on running. Henry Devito wrote: Hi guys I know this has been asked on the list before, but my hard drive crashed and I lost all of the past posts, I need to know what motherboard works ok for asterisk, I have no problems with the Dual and Quad Xeon processor boards I have used. Now I plan on building a Pentium 4 3.0 with hyper-threading. I looked through the wiki and could not find the recommended P4 board. Does anyone have any suggestions? 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 ___ 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] Sending broadcasts to all phones?
A friend of mine has a real panasonic PBX setup at his house, and is able to pick up the phone, dial an extension, and it broadcasts what he says over every phone in his house without the phones having to be picked up. What is this feature called? Would it be possible to set this up with Asterisk given the appropriate phones? (Cisco?) Thanks, Stan ___ 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] Cheap, Highquality IP Phones
Where do I buy one? Stan Matt Riddell wrote: James H. Thompson wrote: Link to Sipura Press Release http://www.sipura.com/Documents/SipuraPressRelease007.pdf I've put it up on the news page in HTML (just in case it takes anyone else as long as it takes me to open a PDF file!) The URL is: http://www.sineapps.com/news.php?rssid=230 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 ___ 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] Large Scale Asterisk Migration
Just as a what if... Lets say I have a 250 phone rollout. I have three incoming T1 lines (however thoes are usually setup) with say 1000 phone numbers available to me. Every phone is currently analog, but I would like to move to a VOIP based setup when the prices become comperable. What am I looking at for equipment here? How do I possibly provide 250 phone ports with the Digium 4 port pci cards? Wouldn't I need a ton of them? Thanks, Stan ___ 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] Unable to locate sample sounds
I am running the latest asterisk CVS.
[EMAIL PROTECTED] /etc/asterisk # locate demo-thanks
/var/lib/asterisk/sounds/demo-thanks.gsm
This directory has 150+ files.
I changed the [demo] section in extensions to [incoming] to play with
this included definition:
[incoming]
;
; We start with what to do when a call first comes in.
;
exten => s,1,Wait,1 ; Wait a second, just for fun
exten => s,2,Answer ; Answer the line
exten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten => s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
exten => s,5,BackGround(demo-congrats) ; Play a congratulatory message
exten => s,6,BackGround(demo-instruct) ; Play some instructions
exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
exten => 2,2,Goto(s,6)
When someone calls, I get this error:
-- Executing Wait("SIP/147.135.8.129-08146c60", "1") in new stack
-- Executing Answer("SIP/147.135.8.129-08146c60", "") in new stack
-- Executing DigitTimeout("SIP/147.135.8.129-08146c60", "5") in new
stack
-- Set Digit Timeout to 5
-- Executing ResponseTimeout("SIP/147.135.8.129-08146c60", "10") in
new stack
-- Set Response Timeout to 10
-- Executing BackGround("SIP/147.135.8.129-08146c60",
"demo-congrats") in new stack
Oct 10 21:48:38 WARNING[196619]: file.c:475 ast_openstream: File
demo-congrats does not exist in any format
Oct 10 21:48:38 WARNING[196619]: file.c:779 ast_streamfile: Unable to
open demo-congrats (format ULAW): No such file or directory
Oct 10 21:48:38 WARNING[196619]: pbx.c:4672 pbx_builtin_background:
ast_streamfile failed on SIP/147.135.8.129-08146c60 fro demo-congrats
Why can it not find the files?
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Re: [Asterisk-Users] Broadvoice registration timeout
This line was reprovisioned, and the password was changed. Stan Stan Brinkerhoff wrote: Hello, I have been trying to setup * with Broadvoice. I am using Gentoo Linux, and * 1.0.0, and now CVS. My current config looks like: (sip.conf) [general] port=5060 context=sip_incoming tos=lowdelay notifymimetype=text/plain allow=gsm allow=ulaw allow=alaw canreinvite=no nat=no register => 8027051441:[EMAIL PROTECTED] [broadvoice]; use for outbound and inbound proxy auth type=peer username=802705 fromuser=802705 secret=xx host=sip.broadvoice.com dtmfmode=inband fromdomain=sip.broadvoice.com canreinvite=no insecure=very srvlookup=yes [broadvoice-in1] ; use to match Broadvoice host 1 type=peer host=147.135.0.128 [broadvoice-in2] ; use to match Broadvoice host 2 type=peer host=147.135.8.128 When I start asterisk -c I get: Asterisk Ready. *CLI> Oct 10 19:46:49 WARNING[81925]: chan_sip.c:665 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Request) Oct 10 19:47:03 NOTICE[81925]: chan_sip.c:3978 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again These two errors keep repeating. I have tried every config I can find on the web, on the wiki, and in these Asterisk-user emails for the last 2 months. At one point I had a 404 error, but I can't get that to even show up anymore. Any help would be appreciated! ___ 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] Broadvoice registration timeout
Hello, I have been trying to setup * with Broadvoice. I am using Gentoo Linux, and * 1.0.0, and now CVS. My current config looks like: (sip.conf) [general] port=5060 context=sip_incoming tos=lowdelay notifymimetype=text/plain allow=gsm allow=ulaw allow=alaw canreinvite=no nat=no register => 8027051441:[EMAIL PROTECTED] [broadvoice]; use for outbound and inbound proxy auth type=peer username=802705 fromuser=802705 secret=xx host=sip.broadvoice.com dtmfmode=inband fromdomain=sip.broadvoice.com canreinvite=no insecure=very srvlookup=yes [broadvoice-in1] ; use to match Broadvoice host 1 type=peer host=147.135.0.128 [broadvoice-in2] ; use to match Broadvoice host 2 type=peer host=147.135.8.128 When I start asterisk -c I get: Asterisk Ready. *CLI> Oct 10 19:46:49 WARNING[81925]: chan_sip.c:665 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Request) Oct 10 19:47:03 NOTICE[81925]: chan_sip.c:3978 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again These two errors keep repeating. I have tried every config I can find on the web, on the wiki, and in these Asterisk-user emails for the last 2 months. At one point I had a 404 error, but I can't get that to even show up anymore. Any help would be appreciated! ___ 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] caller id over iax
Hi all - thanks for all your help. (not)
I've just sussed it.
I had a dial plan line..
;exten => s,5003,SetCIDNum(0${CALLERIDNUM})
The 0$ was the problem, remove the '0' and it works fine. So watch out
for this in extensions.conf - you could add or remove whatever you want
there for your Caller ID prefixes.
Obviously nobody has any idea about dial plans better than me here! (and
I'm a newbie).
On Thu, 2004-08-12 at 15:51, Stan Dard wrote:
> Hi
>
> I've 'inherited' an existing Asterisk with a number of users, and some
> pstn connections through its Zaptel card.
>
> I've recently set up another Asterisk which has no direct pstn access
> and I've connected the 2 systems with IAX. The original system has an
> extension number range 1xxx and the new asterisk has the number range
> 3xxx. I can call both ways, and when required can access the
> inbound/outbound pstn connections from both systems. Thats excellent -
> it works a treat!
>
> But my problem is with Caller ID.
> When I call from *1 to the new system the extension caller id's are
> correct. But when I call in the other direction I always get a leading
> '0' infront of the extension number.. ie 03003.
>
> I've almost googled myself insane searching for an answer & would really
> appreciate any suggestions anyone may have.
>
> Many Thanks
>
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[Asterisk-Users] caller id over iax
Hi I've 'inherited' an existing Asterisk with a number of users, and some pstn connections through its Zaptel card. I've recently set up another Asterisk which has no direct pstn access and I've connected the 2 systems with IAX. The original system has an extension number range 1xxx and the new asterisk has the number range 3xxx. I can call both ways, and when required can access the inbound/outbound pstn connections from both systems. Thats excellent - it works a treat! But my problem is with Caller ID. When I call from *1 to the new system the extension caller id's are correct. But when I call in the other direction I always get a leading '0' infront of the extension number.. ie 03003. I've almost googled myself insane searching for an answer & would really appreciate any suggestions anyone may have. Many 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] zaphfc - hfc pci based ISDN card : point2point & DDI
On Mon, Aug 09, 2004 at 11:52:51AM +0100, Nick Barnes wrote: > The reason I ask is that I installed a BRI system (Single Fritz! AVM card > using chan_CAPI) last week which refused to work - turned out that British > Telecom had provisioned the line as a point-to-point and not > point-to-multipoint as requested. Accepting that BT were going to take > several days to fix their cock-up, I tried to get the card to work in > point-to-point mode, but failed miserably. > > I could really do with getting point-to-point working on these cards as > they're cheap and in plentiful supply. > > Have I completely misunderstood the issue and am just being stoopid, or is > it not possible to do this with these cards? > they work in ptp mode (not espcially well last time I tried) with the mISDN drivers > PS If HFC is the best way to go, does anybody have any recommendations on > cheap HFC card suppliers in the UK? search for "billion isdn pci" on ebay or other HFC-S cards listed on http://www.voip-info.org/wiki-zaptelBRI. For new ones, i've seen them at http://www.komplett.co.uk/k/ki.asp?sku=119006&cks=PRL though not tried them from here myself. ___ 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] quadBRI and UK ISDN2e
On Wed, Apr 07, 2004 at 12:37:43PM +0100, Jon Fautley wrote: > Morning Asterikians, > > I've just got my nice shiny quadBRI card, and it seems to be working > very well - except for one little issue - CallerID. > > The card is currently connected to an ISDN2e line in P2P mode, and an S0 > adapter on our existing alcatel PBX. Is this an omnipcx? > The S0 connection recieves callerID > and displays it correctly - the 2e line doesn't, and BT have said that > CLID was enabled on the line two days ago. Does anyone have any pointers > on this? I assume the callerid is also being displayed on the alcatel handsets? or is this just callerid generated on internal calls? If it is then that should show the bt line is setup correctly. I have callerid working from a bt 2e line in ptp mode using zaphfc. So assuming the bri-stuff versions match the only difference would be the quadbri card. Not sure where that leads because I think/thought all the callerid stuff would be handled by libpri from a q.931 SETUP message on the d-channel and not be driver/card specific. I could only get my s0 box to operate in ptmp mode, so there would be change in the signalling line in zapata.conf, but then if you weren't changing that nothing tends to work rather than just callerid. I also note that you have to match the msn assigned to the s0 when dialling out through the omnipcx whereas bt doesn't seem to be as fussy, but again nothing todo with callerid. ___ 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] support for rfc3326 The Reason Header Field for SIP
A minor gripe with our current system (* + CP7960s) is calls answered by one handset showing as missed on others. RFC3326 seems to answer this problem but I see no support for it in the cisco phones or * (obviously less of a problem with the latter being OSS). Are cisco likely to add this? Do any other SIP phones support RFC3326? Anyone else have this gripe? ___ 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: cisco 7960G powered by 3com 3CNJPSE
On Mon, Mar 15, 2004 at 11:01:46PM -0500, Greg Boehnlein wrote: > On Tue, 16 Mar 2004, Dean Collins wrote: > > > Cisco have the terminals around the other way, this is a well known > > problem, do a search and you'll find what you need to do. > > > > Cheers, > > Dean > > Alright.. since I'm the one that posted about using the standard POE > injectors wih the 7960, I think I might need to revise the Wiki page. > > Am I to understand that the 7960G does -NOT- support the 802.3af Power > Over Ethernet standard? > I've built a cable according to the wiki page and this works with a G model. So it looks like the G models dont support 802.3af either. ___ 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: cisco 7960G powered by 3com 3CNJPSE
Is anyone using a 3com 3CNJPSE to power a 7960G? I have a couple of 7960Gs and 3CNJPSEs but no combination appears to work. Both phones work fine with a cisco power cube. I get a 47.6V reading across pins 5 and 8 on the patch cable coming from the 3CNJPSE. The network still works through the 3CNJPSE, just no power to the phone until I connect the power cube. Using two straight through wired patch cables, 7960G running sip firmware version 6.3. I've tried different patch cables with the same result. Note this is a G version of the phone which I understand is enable to work with 802.3af devices natively and hence I believe doesn't require a specially wired patch cable. I've looked for a setting to tell the phone to get power via the network without success, is there one? Any ideas? 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] XML Phone book software.
On Thu, Mar 11, 2004 at 04:06:41PM -0600, Brian R. Swan wrote: > I'm looking into writing a some phone book XML/PHP software for my Cisco > phones. Specifically, I'd like to be able to use a web interface (on the > computer) to maintain a contact list, and then dial from it on the phone. > Maybe using MySql on the back end or something (to be determined). Before I > start, and duplicate something else that exists, I wanted to see if anyone > has heard of software like that? Searches of Sourceforge, Freshmeat, and > Google didn't turn up much or anything. > see the cmxml software section of http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+cisco+79xx ___ 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] DSL (DMT) goes down when X100 plugged in
On Tue, Feb 24, 2004 at 06:17:27PM +, Chris Lee wrote: > I have been told that when I plug my X100p into the line I get a 36 Khom > loop condition and this may be affecting my ADSL connection (it keeps > dropping the line). > It may be to do with impedance differences here in the UK, But I know > very little about Such things. are you using "modprobe wcfxo opermode=1", as I believe is recommended for most european lines? ___ 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: Cisco 7940 Smartnet in the UK
On Wed, Feb 11, 2004 at 04:34:46PM -, Jason Ross wrote: > This is slightly off topic so sorry for the intrusion. > > I've got a couple of 7940 phones I'd like to put on Smartnet but I'm > looking for what I need to order, what it roughly costs and finally a > reseller in the UK who is easy to deal with. > > Preferably I'd like someone I can deal with online. > outside the UK there is http://www.ams.net/products/product_info.cfm?Product_ID=10891 at $6.90 - dont know if they'd deal with people outside the US. However, elsewhere that part number doesn't seem to mean much. A 7940 is a cateogory 1 device so product codes that seem to mean something to others are: CON-SW-VPKG1 (insight.com/uk - £39.99, tradeprice.co.uk - £31.99) telephone and web support only (search with "smartnet" on tradeprice, their search doesn't seem to like part numbers) CON-AR-VPKG1 - advanced replacement - normally cheaper than above but didn't provide web support so I didn't look into it CON-SNT-VPKG1 - 8x5xNBD version (8am-5pm, next business day) which I believe is when you can expect support and get hardware replacement respectively (insight.com/uk - £63.99, tradeprice.co.uk - £45.35) you can pay more for 24/7 support, 4hr response time or onsite engineer I ended up going with insight coz I didn't find tradeprice till after, so dont know what they're like Also, it seems to take ages for the contract to get registered (2+ weeks) with cisco once u finally get one HTH ___ 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
