Re: [asterisk-users] Sangoma card dying after 1hour
what kinda of cable are you using to connect to the e1. isit shielded or just the generic one they give with the card? On 1/28/07, Porier, Jeremy M. [EMAIL PROTECTED] wrote: Do you see anything in /var/log/messages? I am having a similar problem but I'm also getting some pci fatal error! messages. I had sangoma connect to the box and he couldn't find any config errors so we're leaning towards a hardware problem. - Jeremy -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Jon Schøpzinsky Sent: Friday, January 26, 2007 7:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Sangoma card dying after 1hour Asterisk is version 1.2.14, zaptel 1.2.12, libpri is whatever version was with zaptel 1.2.12 :) Jon -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Steve Davies Sent: 26. januar 2007 12:03 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sangoma card dying after 1hour Which asterisk versions etc etc? On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote: I am running the newest version, from the sangoma wiki. Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies Sent: 26. januar 2007 10:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sangoma card dying after 1hour On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote: Hello List I am having a rather big problem with a sangoma A104 card, I just installed to replace a Digium TE410 card, that was acting up. But now we have a problem with the sangoma card. It runs great after being started, and calls proceed as normal, but after about 1 hour, it stops being able to make and receive calls. If I run wanpipemon debug, can see that the card still receives packets from the ISDN, but when I make a call, I cant see it in wanpipemon, and asterisk just responds with a: NOTICE[17240] app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) I am pretty shure that this is a configuration issue, but are there anything I need to be aware of when moving from a Digium card to a sangoma card? Which wanpipe version? Anything lower than 2.3.4-4 should be upgraded as there are some resource leak fixes in that version. Regards, Steve ___ --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 ___ --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 ___ --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 ___ --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 ___ --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 ___ --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] Transfer on RTP timeout?
Hi all, We are looking at VoIP over Wifi and I was wondering if anybody had any ideas around automatically transfering calls after an RTP timeout? The idea is this: a user is on a call with their IP phone and the connection drops (e.g. user walks out of range of their Wifi AP). Using RTP timeout I was hoping rather than just dropping the call I could keep the other party on hold whilst transferring the call to another number (i.e. a PSTN number). Essentially, I would like to change the RTP timeout logic to lookup a 'forwarding number' in MySQL and then perform a blind transfer to that number. That way the call can stay up rather than the user having to redial. Is there a way of transferring back to the * dialplan on RTP timeout to perform some additional steps (instead of just hanging up?) Any suggestions very welcome. Rgds, Ray ___ --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] Response on dialin - no extension
I think, sip server even doesn't know, that user picks up handset, maybe with skinny or mgcp phone should it work because this phones are controled by signaling server PJ chester c young wrote: On a SIP phone is it possible to enter the dialplan when the user picks up the phone without having to wait for the user to press an extension? Is is possible to do something like [sip-test] s,1,Answer s,2,Playback(welcome) s,3,WaitExten(30) 1,1,Noop(exten 1) ... t,1,Goto[s,2] Be a PS3 game guru. Get your game face on with the latest PS3 news and previews at Yahoo! Games. http://us.rd.yahoo.com/evt=49936/*http://videogames.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 ___ --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] Response on dialin - no extension
chester c young wrote: On a SIP phone is it possible to enter the dialplan when the user picks up the phone without having to wait for the user to press an extension? You need a phone with a hotline function. Consult your phone's user manual. Leo ___ --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: How to exit from console?
On Thu, 2007-01-25 at 18:01 +0200, Tzafrir Cohen wrote: On Thu, Jan 25, 2007 at 01:37:50PM +0100, Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Try safe_asterisk , for an easy way to start asterisk in background, a plain 'asterisk' is even better and safer. asterisk -U asterisk . is better. /etc/init.d/asterisk start is similar. Why is this better than safe_asterisk? the init.d init scripts bundled with asterisk are using safe_asterisk and not calling the asterisk binary directly. E.g: because you have a valid PID file of the controlling process. If you actually want to kill it, you can. Granted, its a good idea. the init.d scripts bundled with asterisk kill safe_asterisk, which apparently works just as well (haven't looked at safe_asterisk code, but its probably killing its child when it is being killed, which should work well for any situation other then kill -9). And you don't need physical access to the system to get to the one and only real console. OTOH, if you do have physical access, you have full control of Asterisk, as you may inject custom dialplan. I wasn't aware that running asterisk -r on a physical tty has any advantages over running asterisk -r on a remote shell. -- Oded Arbel Atelis [EMAIL PROTECTED] Tel: +972-54-7340014 ::.. In this world, truth can wait; she's used to it. ___ --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] Asterisk 1.4 problem with ztdummy and MeetMe()
On Thu, 2007-01-25 at 18:40 +0100, Stefan Wintermeyer wrote: Hi, when I build zaptel-1.2 and asterisk-1.2 I can modprobe ztdummy and start asterisk to be able to use MeetMe(). When I build zaptel-1.4 and asterisk-1.4 I can modprobe ztdummy and start asterisk but I am not able to use MeetMe(). What do I miss? I'm not sure, because we missed the entire problem description, which I would imaging would have included log snippets and/or error message reports, but it was apparently removed from your e-mail. http://www.catb.org/~esr/faqs/smart-questions.html -- Oded Arbel Atelis [EMAIL PROTECTED] Tel: +972-54-7340014 ::.. He's dead, Jim. You grab his wallet, I'll grab his tricorder. ___ --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] Transfer on RTP timeout?
Hi, Ray Jackson wrote: transfer to that number. That way the call can stay up rather than the user having to redial. Is there a way of transferring back to the * dialplan on RTP timeout to perform some additional steps (instead of just hanging up?) Nokia seems to have done something like this in their E-series (E60 etc) with Avaya and Cisco. Anyone have a lowdown on the technical stuff there ? Florian ___ --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] Rxfax and Txfax on Asterisk 1.4
On Fri, 2007-01-26 at 07:06 -0500, Remzi Semsettin Turer wrote: Has anyone successfully installed spandsp and rxfax and txfax applications on 1.4.0 release of Asterisk? I tried the latest snapshot of spandsp, as well as couple other previous versions. I compiled it fine, downloaded the asterisk.patch, manually patched the asterisk files, run .configure, make clean, make menuselect and it shows app_txfax and app_rxfax as XX (unavailable). Each time I made sure no other spandsp versions are installed and put the proper path in /etc/ld.so.conf and run ldconfig, prior to compiling Asterisk. Still no luck. I don't remember how I got rxfax/txfax to be available in menuselect, but they won't compile - I couldn't get the app_rxfax.c and app_txfaxt.c to compile against 1.4 and I didn't have time to figure out how to get them running. AFAIK the current recommendation is to use HylaFax with something called iaxmodem. -- Oded Arbel Atelis [EMAIL PROTECTED] Tel: +972-54-7340014 ::.. A man with one watch will always know the time, A man with two watches will always be in doubt. ___ --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] AsteriskNow - H323 support for trunks
First of all, I was wondering if this is the right place to post questions and issues around AsteriskNow. If so, here's my question: I've been using Iphonecom.com's services for abour 3-4years now, to connect and call my brother who lives in the US and my parents, sister and me (all located in Belgium). We're OK with their services, but we wanted to be more free in call-numbers, options, ... So I decided to give Asterisk a go, and try to host these things myself (dedicated dsl line, dedicated server, ...) That way, we could also use my landline here to let my brother use the belgian phone system to call other relatives and friends, after-working-hours, which is free overhere. (for me that is). Now, Iphonecom.com only uses H323 as they mentioned 'they have too many problems with SIP thus sticking with H323 for the moment to guarantee everyone's service'... I was wondering if I can use this account in my Service Providers tab, and connect through my account, and receive my brothers call, before migrating him over to my Asterisk system. Just as a test-scenario... Anyone tried or did this before, pref'd with AsteriskNow. My knowledge about asterisk (especially configging) is rather basic, but I'm willing to learn ;) Thanks people! Great product, great user-base support! Love it!!! Bram ___ --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: How to exit from console?
On Sun, Jan 28, 2007 at 12:41:30PM +0200, Oded Arbel wrote:
On Thu, 2007-01-25 at 18:01 +0200, Tzafrir Cohen wrote:
On Thu, Jan 25, 2007 at 01:37:50PM +0100, Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Try safe_asterisk , for an easy way to start asterisk in background,
a plain 'asterisk' is even better and safer.
asterisk -U asterisk . is better.
/etc/init.d/asterisk start
is similar.
Why is this better than safe_asterisk?
the init.d init scripts bundled with asterisk are using safe_asterisk
and not calling the asterisk binary directly.
Those scripts will use either asterisk or safe asterisk. It depends if
safe_asterisk is executable (which implies that it was installed).
E.g: because you have a valid PID file of the controlling process. If
you actually want to kill it, you can.
Granted, its a good idea. the init.d scripts bundled with asterisk kill
safe_asterisk, which apparently works just as well (haven't looked at
safe_asterisk code, but its probably killing its child when it is being
killed, which should work well for any situation other then kill -9).
Let's look at the stop target of those robust scripts.
The Debian one:
stop)
echo -n Stopping $DESC:
$DAEMON -rx 'stop now' /dev/null 2 /dev/null echo -n $NAME
echo .
exit 0
;;
Cool. If asterisk goes bezerk, it will go bezerk as well. With proper timing of
concurrent calls to this script you can have some fun.
Here is the redhat one:
stop() {
# Stop daemons.
echo -n $Shutting down asterisk:
killproc asterisk
RETVAL=$?
[ $RETVAL -eq 0 ] rm -f /var/lock/subsys/asterisk
echo
return $RETVAL
}
And you don't need physical access to the system to get to the one and
only real console. OTOH, if you do have physical access, you have full
control of Asterisk, as you may inject custom dialplan.
I wasn't aware that running asterisk -r on a physical tty has any
advantages over running asterisk -r on a remote shell.
No. But if you're not going to use the local console anyway, why run it?
To allow intruders control of your Asterisk?
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Re: How to exit from console?
Oded Arbel wrote: And you don't need physical access to the system to get to the one and only real console. OTOH, if you do have physical access, you have full control of Asterisk, as you may inject custom dialplan. I wasn't aware that running asterisk -r on a physical tty has any advantages over running asterisk -r on a remote shell. In addition, if you include /dev/ttyx = notice,warning,error,verbose (or similar) in your logger.conf file, you get a console on ttyx as well. ___ --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] Enterprise quality SIP provider
I need to setup incoming (over an 800 number and some local DID's) and outgoing phone calls (all over the country) with an Asterisk server. This asterisk server has 20 Polycom 430 phones connecting to it. I need the best possible SIP provider out there. I have tried http://www.nufone.net and http://www.broadvoice.com and they do not even come close to the expected quality. Does ATT allow companies to connect to their backbone network using SIP ? Any suggestion of companies which provide enterprise quality SIP termination and origination. The office is in a building which has a data center in the basement and has DS3 coming into the data center. I can buy as much bandwidth as I want from the data center. Regards, -- Vikas http://www.stanford.edu/~vikask/ ___ --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] Mabe OT? What managed switch is best for VoIP application?
My Trendnet 26 port managed switch gave up on me so I'm shopping for a new switch. I learned the hard way NOT to trust marketing material from anyone so now I'm asking the list: what am I looking for in a managed, VoIP switch? P.S: For those that don't understand WHY I can't trust marketing material, let me tell you something about the Trendnet switch that's fast becoming garbidge. I wanted an managed switch so I boght the switch had Managed and Virtual LAN in the biggest possible letters. Later, after buying two Intel 1Gb Virtual Lan Enabled network cards, I discovered my Trendnet switch doesn't do standard VLan, it only does VLan if linked to an other Trendnet switch - not useful at all! Thanks, Cosmin Prund ___ --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] NAT: RTP Path Optimization
http://lisas.de/~patrick/temp/rtp-optimierung.png Everything is working fine in my Setup, but I want Extern1 to talk to Extern2 directly whitout going over Asterisk as the uplink is slow. When I set for Extern1/2 canreinvite=yes it works, but Intern-2-Extern doesn't work because Asteisk gives out the private IP-Adresses of Int1/2 I defined localnet=10.0.0.0/255.0.0.0 (Private LAN) but this doesn't help. Ideas, how to handle Extern-2-Extern (RTP bypass Asterisk)? Do I have to adjust nat somwhere? ___ --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] Show call coming back from Call Parking
On Friday 26 January 2007 5:44 pm, Eric ManxPower Wieling wrote:
[park-dial]
exten = _.,1,SetCIDName(Parking Timeout)
exten = _.,2,SetVar(__ALERT_INFO=Triplet)
exten = _.,3,Goto(extensions,3500,1)
I see your awesome little snippet and raise you a
exten = 700,1,Set(PARKRETURN=${CID(number)})
exten = 700,1,Park()
so your Goto() above can grab the ${PARKRETURN} and Dial() that back.
*Not tested, just thought of it for an enhancement so it didn't always come
back to the receptionist.
-A.
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Re: [asterisk-users] Show call coming back from Call Parking
Andrew Kohlsmith wrote:
On Friday 26 January 2007 5:44 pm, Eric ManxPower Wieling wrote:
[park-dial]
exten = _.,1,SetCIDName(Parking Timeout)
exten = _.,2,SetVar(__ALERT_INFO=Triplet)
exten = _.,3,Goto(extensions,3500,1)
I see your awesome little snippet and raise you a
exten = 700,1,Set(PARKRETURN=${CID(number)})
exten = 700,1,Park()
so your Goto() above can grab the ${PARKRETURN} and Dial() that back.
*Not tested, just thought of it for an enhancement so it didn't always come
back to the receptionist.
I will have to try that. The example I posted is a copy/paste from a
production dialplan so it *is* tested. In our environment, only the
receptionist can remember how to park a call.
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Re: [asterisk-users] Mabe OT? What managed switch is best for VoIP application?
Cosmin Prund schrieb: P.S: For those that don't understand WHY I can't trust marketing material, let me tell you something about the Trendnet switch that's fast becoming garbidge. I wanted an managed switch so I boght the switch had Managed and Virtual LAN in the biggest possible letters. Later, after buying two Intel 1Gb Virtual Lan Enabled network cards, I discovered my Trendnet switch doesn't do standard VLan, it only does VLan if linked to an other Trendnet switch - not useful at all! Standard Vlan = 802.1q Trendnet offered you only VLAN in the Switch, not 802.1q You have to look for the Protocol *802.1q* http://en.wikipedia.org/wiki/VLAN#Protocols_and_design ___ --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] Ringing oddity/stupidity
J. Oquendo wrote: Anyone experience ring oddities with extensions.conf rollovers? Let me summarize... One of my extensions.conf file is built to ring during the day, ring/go to voicemail after a certain time: [main-aa] exten = s,1,GotoIfTime(17:00-8:30|mon-fri|*|*|*?main-night-aa,s,1) exten = s,2,GotoIfTime(*|sat-sun|*|*|*?main-night-aa,s,1) ... [main-night-aa] exten = s,1,Answer exten = s,2,Background(/etc/asterisk/night) exten = s,3,Voicemail([EMAIL PROTECTED]) exten = s,4,Hangup When in night mode, if someone called, while Asterisk would show the phone as ringing (and INDEED the phone would ring) the caller wouldn't hear the phone ring. No music, no ringing no thing until the amount of time the rings ran out and then be transferred into voicemail. So... (un)Leet ASCII explanation: Caller (after hours) -- Dials in -- Press extension -- Asterisk makes transfer -- Caller hears dead air -- No one answers -- Voicemail -- Caller now hears voicemail prompts According to the dialplan, there should be no ring at all, it should go directly to voicemail. How long is the Caller hears dead air -- No one answers time? To comfort the caller you could add exten = s,1,ringing exten = s,2,wait(2) exten = s,3,answer() exten = s,4,Background(/etc/asterisk/night) exten = s,5,Voicemail([EMAIL PROTECTED]) Leif ___ --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] Mabe OT? What managed switch is best for VoIP application?
No one told me about 802.1q Vlan before I boght the switch. It was printed in big fat letters on the box. Now I *do* know about 802.1q but it's a little bit too late: I already have the switch. Fortunately (unfortunately) the switch is gone, it's dead. Now I want a better switch and I'm asking so I don't fall into the same trap *again*. Again, this big switch is not the only device I bought only to find out it doesn't exactly do what I want it to do. I also got a nice little ZyXEL VPN collecting dust in a drawer somewhere. I wanted a VPN router/firewall that allowed me to connect to my network from my Windows-based Laptop computer, using the tools available in the system. Guess what: I *can* connect to the ZyXEL using an paid-for client that costs almost as much as the firewall itself. I'm now running PopTop on my Linux Asterisk box and it works just fine, and it's a lot cheaper. And I did learn about a few other standards names in the process: AFTER I bought the hardware device. So the idea is very simple: I need a switch that does VoIP well, has lots of ports and does 802.1q VLAN. I also want it to be managed and have it's management tools help me diagnose problems. That's my biggest question right now: What *exactly* am I looking for? My Trendent switch has management and it's easy to use for what it does, but it would never help me diagnose a network problem. It took a number of disconected *local* LAN VoIP calls before I noticed the switch is flowed and needs to be replaced. Thanks, Cosmin Prund Patrick Cervicek wrote: Cosmin Prund schrieb: P.S: For those that don't understand WHY I can't trust marketing material, let me tell you something about the Trendnet switch that's fast becoming garbidge. I wanted an managed switch so I boght the switch had Managed and Virtual LAN in the biggest possible letters. Later, after buying two Intel 1Gb Virtual Lan Enabled network cards, I discovered my Trendnet switch doesn't do standard VLan, it only does VLan if linked to an other Trendnet switch - not useful at all! Standard Vlan = 802.1q Trendnet offered you only VLAN in the Switch, not 802.1q You have to look for the Protocol *802.1q* http://en.wikipedia.org/wiki/VLAN#Protocols_and_design ___ --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 ___ --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] Enterprise quality SIP provider
We LOVE Teliax. We're on a Time Warner business class fiber connection and avg 25ms latency from Ohio to Denver CO. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vikas Sent: Sunday, January 28, 2007 9:07 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Enterprise quality SIP provider I need to setup incoming (over an 800 number and some local DID's) and outgoing phone calls (all over the country) with an Asterisk server. This asterisk server has 20 Polycom 430 phones connecting to it. I need the best possible SIP provider out there. I have tried http://www.nufone.net and http://www.broadvoice.com and they do not even come close to the expected quality. Does ATT allow companies to connect to their backbone network using SIP ? Any suggestion of companies which provide enterprise quality SIP termination and origination. The office is in a building which has a data center in the basement and has DS3 coming into the data center. I can buy as much bandwidth as I want from the data center. Regards, -- Vikas http://www.stanford.edu/~vikask/ ___ --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 ___ --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] Channels Banks that support neon MWI
Anyone have suggestions for channel banks compatible with Trixbox that can set a MWI lamp on phones. We're a business, but have a lot of analog phones with the neon lamp on them and want to move them from a Mitel SX-200 to *. EKG ___ --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] Does X100P decode caller ID?
Yuan LIU wrote: From: Nilesh Londhe [EMAIL PROTECTED] On ebay, I have seen x100p (or clone) with two different chipsets; 1) has motorols chip 2) has something else I dont call. My experience says that the x100p/clone with motorola chipset shows caller id with default * settings. This one (SM56) is Motorola. I did get an authentic X100P from DigitNetworks that uses Intel chipset but haven't tested on this line. Yuan Liu Isn't the SM56 motorola a winmodem? signature.asc Description: OpenPGP digital signature ___ --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] Mabe OT? What managed switch is best for VoIP application?
For example: VPN ist not standarized. You have the choose between IPsec, PPtP, L2TP,... We are using OpenVPN, which is not a standard but is free of charge. And it works :-) Or managed switch is not standarized. You can manage a switch out-of-band e.g. serial console or you can manage a switch in-band with SSH, Telnet, WWW-Gui or SNMP. Please always look at the Specifications of a switch *before* you buy them ;-) example: http://www.hp.com/rnd/products/switches/ProCurve_Switch_2900_Series/specs.htm#Standards http://cisco.com/en/US/products/hw/switches/ps5023/products_data_sheet0900aecd80371991.html#wp9000360 By the way: Nobody can propose you a switch, if you don't know what you want. Do you want: 10, 100, 1000 Mbit/s? How much Ports? Quality-of-Service? (802.1p, Diff-Serv,...) Security-Features? (Port Security, Layer 3 ACLs,...) What VoIP Devices do you use? For snom, you don't need good switches as they don't support 802.1X or CDP. ... I worked with Cisco and HP and they should do what you are looking for. I even worked with cheap unmanaged switches ~20 Euro and they work with VoIP. ___ --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] Multiple parking lot
In order to use this patch, i have to download the complete version of SVN asterisk? I highly doubt this will work with the metermaid patch that allows the call park buttons to work with Snoms. Last time I let anyone share a PBX!! Any comments on this would be great! Thanks Brad On 1/26/07, Olle E Johansson [EMAIL PROTECTED] wrote: 25 jan 2007 kl. 08.26 skrev Darryl Dunkin: There is an SVN branch with this feature: http://svn.digium.com/view/asterisk/team/oej/multiparking/ I had hope this would be a feature added to Asterisk 1.4, but fail to see it on the changelog. It wasn't approved due to some architecture issues. I'll see if I get time to fix them for next release. /O ___ --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 ___ --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] Does X100P decode caller ID?
From: Derek Whitten [EMAIL PROTECTED] x100p/clone with motorola chipset shows caller id with default * settings. This one (SM56) is Motorola. I did get an authentic X100P from DigitNetworks that uses Intel chipset but haven't tested on this line. Isn't the SM56 motorola a winmodem? It is, and is identified by wcfxo as a Wildcard FXO: Wildcard X100P. So much for The DigitNetworks X100P is detected as an actual X101P card. On the other hand, I managed to hear audio from ztmonitor with SM56 - it sounded like pulses, and no chirp could be heard, whereas X101P does the opposite. I didn't have enough knowledge about rxgain, but I tested several settings with actual conversation on, so I figured callerid couldn't be fixed in SM56 within reasonable range. This may be one actual advantage DigitNetworks can claim. For now I'll just use SM56 when callerID is not required. Yuan Liu ___ --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: Delay in Call Distribution using the Queue Application
Thanks for the info, is there a patch available for version 1.2 that adds the autofill option? Thanks and Regards Markus Yes, I confirm the autofill option is present in 1.4, but must be enabled manually not to break compatibility with 1.2. l. On Fri, 19 Jan 2007 15:32:32 +0100, Tom Rymes [EMAIL PROTECTED] wrote: You may be running into the limitation in Asterisk 1.2 (It's fixed in 1.4, I think double check that) in how the queues distribute calls. Basically, the queue can only distribute one call at a time, so if you have two agents, both available, and two calls in the queue, asterisk will send call #1 to agent #1 first. Once that call is connected, Asterisk will then send call #2 to agent #2. In other words, until asterisk distributes the first call, it can't distribute any other calls waiting in line. ___ --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] OT: Admin manual for Linksys Sipura SPA-2102
take a look at http://spc.pifiu.com On 1/2/07, Erick Perez [EMAIL PROTECTED] wrote: Hi, Anyone knows where to get the admin (not the end user) manual for the linksys spa2102. This model is the 2 analog port+router. There are a lot of advanced options that I would like to see what they do. Thanks, -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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 ___ --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] ATCOM AT 468 manuals and firmware anyone?
Erick Perez wrote: Hi there, im looking for another place that provides manuals and firmware updates for the ATCOM AT 468 and their configuration with asterisk. the site www.atcom.com.cn has non functional download links. I suppose you mean the AG 468 If you can find somebody who still uses Internet Explorer, the links works. The download page used to have a link for a page which worked in Firefox, but not anymore. But anyway, here are the links. http://atcom.com.cn/en/down/userguide/EN/AG-468/AG468_User_ManualGuide.rar http://atcom.com.cn/en/down/program/en/ng_series/ag468_060119_vr41rls.zip Leif ___ --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 Wire Level Tapping
I am trying to do a wire level tap on T1 equipment using digum equipment. So far most call monitoring hardware for call centers try to stay on the analog side requiring a lot of rewiring. I have already posted to the list about T1 bridging using DAC's support in the zaptel drivers. I still don't know if I can spy on channel information since I don't have any digium hardware on me until the project begins. Anybody found a method of spying on a D-Channel and all voice channels using standard T1 equipment? I am making a rough assumption that if I can trick the zaptel drivers into operating without anything responding to a TX signal then I can do the following: S-T1 = T1 to Spy On T1-1 = Digium T1 card #1 T1-2 = Digium T1 card #2 Map S-T1(RX) to T1-1(RX) and S-T1(TX) to T1-2(RX) and decode the D-Channel where appropriate, should I be able to spy on the RX/TX channels enough to make a recording including CID information? This would help in situations where the monitoring system needs to be replaced or taken down without bothering in-progress calls. Shane ___ --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] PHP sip client
Hi all, I want to write a simit sip client in PHP with asterisk API, in this moment I'm able to compose a number on my browser and call between 2 hw sip phone. I digit a number, my phone ring and after hanging up the cornet the second phone ring. But I want to add a features I want to hang up the cornet of my phone, compose the number in my browser and call a second phone. In witch way I do this? Can i do this? quickly... I want to replicate the numeric keyboard of my hw phone! Thanks in advantage and sorry for my english. :( -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web: http://www.pasqualotto.org skype: epasqualotto ___ --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: Migration to Asterisk 1.4
Hi, I am trying to migrate my asterisk 1.0 to 1.4. I have downloaded Asterisk Now and installed. I am using the GUI that comes with Asterisk Now. I am trying to setup an internal phone system where I can make calls between extensions. I am able to create users using the interface. I am not able to figure out how to define the phones and assign them to the users. I have manually modified iax.conf and added my iaxcomm softphone and iaxy phone which seem to be registering. I am not able to figure out how to connect a user with one of these phones.I am not sure if this is the corret way to do. Any help is appreciated. Thanks, Naren ___ --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] Add current extension dynamically to template?
Is it possible for a template to know the extension number
in sip.conf? In the example extension below, I'd like to _not_ have to
type the lines
username=410
and
callerid=410
in each extension.
Instead, the template should be able to pick up the extension name from
the [410] extension specified.
In the [grandstream] template example below, I'd like to specify in the
template that it should use the udername=410.
This is the template:
[grandstream](!); template for Grandstream sip phones
;===
context=default ;Dials out to telasip-gw for grandstream
type=friend
qualify=yes
insecure=very
host=dynamic
canreinvite=no
nat=no
;add these lines to the template:
username=${extension I am now in}
callerid=${extension I am now in}
This is one of several extension used presently:
[410](grandstream) ; Karen Office
username=410
I want the extension to have just the line:
[410](grandstream)
Larry
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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[asterisk-users] Automating the setting/clearing of a flag
Is there any way I can set up a list of dates and times at which a flag on my Asterisk PBX is to be changed, thus making it change somewhat automatically? I used to have a flag that I set whenever it was office hours and I was working from home, to keep non-essential calls away from my phones. I have changed over to a flag I can set on the Sunday evening when my week on call starts and clear the next Sunday evening, with the dialplan knowing when office hours are and acting accordingly. The flag also enables calls on a special work-related number to ring. I'd like to be able to program the dates on which I am on call into Asterisk in advance, so that it knows somewhat automatically when to keep the non-essential calls away and allow the work calls through. -- Phil Reynolds o mail: [EMAIL PROTECTED] |L_ \ / Web: http://www.tinsleyviaduct.com/phil/ (_)- \/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95 ___ --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] Voicemail from sip phones
Hello, I'm having a problem in voicemail check attempts from SIP-based phones. I've searched a ton of docs but don't see anyone else having a similar issue. I have a TDM22B with two non-sip phones connected to it as well as several SIP phones including a GXP-2000 and some X-Lites. Users of the real phones in the same context can pick up and dial *8 to get to VoiceMailMain() just fine. However, SIP users cannot. When they dial *8 a message comes on stating The person you are calling is unavailable. I can get the MWIs to light up on the SIP phones when there's a voicemail waiting but there's no way to check it. SIP debugging through the CLI hasn't so far given any great hints. I've tried various settings of mailbox= and vmexten= combinations in the sip.conf. For someone with a mailbox of [EMAIL PROTECTED] and extension of 126 in the 'internal' context, what should the vmexten be? The real phones and the sip phones are all within the same 'internal' context in extensions.conf, as is the *8,1,VoiceMailMain() function call. Any hints or suggestions would be appreciated. Here's a sample of one of the Xlite SIP configurations that isn't working: [user] type=friend regexten=126 secret=greatsecret qualify=200 host=dynamic canreinvite=no context=internal [EMAIL PROTECTED] vmexten=126 ___ --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] Voicemail from sip phones
On Sun, Jan 28, 2007 at 05:16:38PM -0600, [EMAIL PROTECTED] wrote: Hello, I'm having a problem in voicemail check attempts from SIP-based phones. I've searched a ton of docs but don't see anyone else having a similar issue. I have a TDM22B with two non-sip phones connected to it as well as several SIP phones including a GXP-2000 and some X-Lites. Users of the real phones in the same context can pick up and dial *8 to get to VoiceMailMain() just fine. However, SIP users cannot. When they dial *8 a message comes on stating The person you are calling is unavailable. Have you got pedantic=yes in your sip.conf anywhere? -- Phil Reynolds o mail: [EMAIL PROTECTED] |L_ \ / Web: http://www.tinsleyviaduct.com/phil/ (_)- \/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95 ___ --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] Voicemail from sip phones
Thanks for the reply. It is there within the sample stuff but it's commented out. ;pedantic=yes ; Enable slow, pedantic checking for Pingtel ; and multiline formatted headers for strict ; SIP compatibility (defaults to no) I also should've included the version that I'm using which is SVN-trunk-r47770, being used because of the new distinctive ring range options that became available. On Sun, Jan 28, 2007 at 11:40:43PM +, Phil Reynolds wrote: On Sun, Jan 28, 2007 at 05:16:38PM -0600, [EMAIL PROTECTED] wrote: Hello, I'm having a problem in voicemail check attempts from SIP-based phones. I've searched a ton of docs but don't see anyone else having a similar issue. I have a TDM22B with two non-sip phones connected to it as well as several SIP phones including a GXP-2000 and some X-Lites. Users of the real phones in the same context can pick up and dial *8 to get to VoiceMailMain() just fine. However, SIP users cannot. When they dial *8 a message comes on stating The person you are calling is unavailable. Have you got pedantic=yes in your sip.conf anywhere? -- Phil Reynolds o mail: [EMAIL PROTECTED] |L_ \ / Web: http://www.tinsleyviaduct.com/phil/ (_)- \/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95 ___ --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 ___ --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] Voicemail from sip phones
On Sun, Jan 28, 2007 at 05:45:27PM -0600, [EMAIL PROTECTED] wrote: Thanks for the reply. It is there within the sample stuff but it's commented out. ;pedantic=yes ; Enable slow, pedantic checking for Pingtel ; and multiline formatted headers for strict ; SIP compatibility (defaults to no) If enabled, you may find you can use * from SIP phones - I had a problem with it being misinterpreted from both ekiga and SJphone. -- Phil Reynolds o mail: [EMAIL PROTECTED] |L_ \ / Web: http://www.tinsleyviaduct.com/phil/ (_)- \/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95 ___ --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] Voicemail from sip phones
Hmm. Nope. Still same thing. I added pedantic=yes both in the general context in sip.conf and in the user's context in sip.conf with no change. Just for fun, I also changed it to pedantic=no in each place with no luck either. (I stopped and started asterisk between each change). Other thoughts? On Sun, Jan 28, 2007 at 11:56:02PM +, Phil Reynolds wrote: If enabled, you may find you can use * from SIP phones - I had a problem with it being misinterpreted from both ekiga and SJphone. ___ --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] Zap channels staying offhook - restart required
Anyway, my question is, how do I get the offhook status to reset? So far only a server reboot works. I tried: - physically disconnecting the line from the socket - restarting asterisk - zap destroy channel and restarting asterisk Any suggestions on how to avoid a reboot? I tried the following: unload chan_zap.so load chan_zap.so That seemed to reset the offhook status without a reboot. How do I access this in a dialplan or via the Manager? Thanks Cameron ___ --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] Heartbeat on Digium T1 PCI cards?
I want to make sure that when an asterisk server dies that I am not left with a huge bill afterward for not hanging up a long distance call correctly. Are digium cards somehow set up to recieve a heartbeat from the drivers and if it skips a few beats it will take the t1 down in a way that would terminate the call? Shane ___ --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] Trouble with incoming calls
Hello everyone. I am having trouble receiving via my Linksys SPA-3102. I
have not problem dialing out. It is like asterisk never even sees the call.
I have 3 sip devices grandstream bt-100, spa-3102 fxs, and spa-3102 fxo. A
very simple setup, just getting familar with asterisk. Here are my relative
config files. let me know if you need more.
sip.conf
[general]
context=default
bind=0.0.0.0
bindport=5060
srvlookup=yes
[100] ;bt-100
type=friend
username=100
context=default
secret=secret
host=dynamic
dtmfmode=rfc2833
disallow=all
allow=ulaw
[EMAIL PROTECTED]
[101] ;fxs
type=friend
username=pots
context=default
secret=phone
host=dynamic
dtmfmode=rfc2833
disallow=all
allow=ulaw
[EMAIL PROTECTED]
[102] ;fxo
type=friend
context=default
secret=pstn
host=dynamic
dtmfmode=rfc2833
disallow=all
allow=ulaw
port=5061
extensions.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
context=default
[globals]
RINGGROUP1 = SIP/100SIP/101
[default]
; These next three lines are for testing, just to make sure I got the call,
but no good
exten = s,1,Answer
exten = s,2,System(touch $HOME/got_it)
exten = s,3,Hangup
;exten = s,1,Dial(SIP/100,10)
;exten = s,2,Hangup
exten = 97,1,Dial(${RINGGROUP1},10)
exten = 97,n,Hangup
exten = 98,1,Answer
exten = 98,n,AGI(agi-test.agi)
exten = 98,n,Hangup
exten = 99,1,Answer
exten = 99,n,Playback(hello-world)
exten = 99,n,Hangup
exten = 100,1,Answer
exten = 100,n,Dial(SIP/100,15)
exten = 100,n,VoiceMail([EMAIL PROTECTED])
exten = 100,n,Playback(vm-goodbye)
exten = 100,n,Hangup
exten = 101,1,Answer
exten = 101,n,Dial(SIP/101)
exten = 101,n,Hangup
exten = _XX,1,Dial(SIP/102/${EXTEN})
exten = _XX,n,Hangup
I appreciate your help
- Jim
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Re: [asterisk-users] Asterisk + Unicall + Telmex E1 MFC/R2 Argentina + Meridian
Hi Facundo, Were you able to match your phone's codec with the asterisk codec? Try to check and set them with the same codec. Also, try to adjust the rxgain txgain. Regards, Angel Facundo Ameal [EMAIL PROTECTED] wrote: Moises, I 've stated testing by raising all timers a bit. Everything went worse, now there are more failed calls. Can you give me a hint of which timers to modify and, if you know, a more extensive explanation of each one? I know it's documented into the file but I cannot catch the concept. Thanks you very much! Greets. On 1/21/07, Facundo Ameal wrote: Thanks Moises, I was trying to find some consistence, but the only similarity I could find is that much of the calls that fail are long distance ones or international. It fails in both, Telmex and Meridian link. I 'll try looping. I'll be posting results soon. I hope I can manage to get it work. Thanks for your help. Regards. On 1/19/07, Moises Silva wrote: Similar probles I had were fixed incrementing one of the timers, but if you have already done that, I have no idea of your problem, you require to debug the problem and try to find some consistence in the failures, find if the failure is on the Asterisk - telco Link, or in the Asterisk - meridian link? find if putting in loop your own asterisk still fails, etc etc. Kind Regards On 1/18/07, Facundo Ameal wrote: Thanks for your help, but I've already adjusted timers on the source code. I found your document a week ago and read it. Do you really think that is a matter of timers only? Greets! On 1/18/07, Moises Silva wrote: Sometimes timers need to be adjusted on the mfcr2 source code. Sometimes is missconfiguration. Anyway, may be this document can help you out to debug the problem: http://moy.ivsol.net/unicall/mfcr2-asterisk-unicall-0.2-english.pdf Kind Regards On 1/17/07, Facundo Ameal wrote: Hi everyone! I'm having some issue trying to place calls with asterisk connected to an E1 R2 from Telmex Argentina. The other E1 port is connected to a Meridian which also uses R2 protocol. Calls sometimes fail with different error messages such as: Unicall protocol error 32773, 32772, 32769. Some other calls fail saying: Far end disconnected(cause=Destination out of order [27]) Far end disconnected(cause=User alerting, no answer [19]) Far end disconnected(cause=Switching equipment congestion [42]) Far end disconnected(cause=User busy [17]) I don't think those causes are real, because if you use another line, yo establish the call. Could it be something about timing of ABCD bits? I'm using: Asterisk 1.2.6 Zaptel 1.2.5 libmfcr2 0.0.3 libunicall 0.0.3 libsupertone 0.0.2 spandsp-0.0.3 And this is my unicall.conf: [channels] loglevel=1023 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=no echocancelwhenbridged=no echotraining=no rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no musiconhold=default protocolclass=mfcr2 protocolvariant=ar,10,4,15 protocolend=cpe group=1 context=from-zaptel channel = 1-15 channel = 17-29 loglevel=0 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no protocolclass=mfcr2 protocolvariant=ar,0,12,12 protocolend=cpe group=2 context=hacia-afuera channel = 32-46 channel = 48-60 Thanks in advance! Greets! -- Facundo Ameal. famealgmailcom Linux User #395088 Share your knowledge, use free software. ___ --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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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 -- Facundo Ameal. famealgmailcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and
[asterisk-users] Test Hardware
Hello everybody Anyone, know TDM800 of yeastar? Anyone to test him with asterisk? Regards ___ --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] Asterisk very slow when internet down
Sadly, people have reported this fairly regularly. An option is to hard-code the server and IP address in your hosts file, but this can be even worse when the provider changes the IP address of the server... later, PaulH On Thu, 2007-01-25 at 22:27 +1030, Peter Mitchell wrote: Has anyone seen this issue with asterisk running like a dog when the internet is down ? Internal calls, incoming ISDN calls etc all seem to be affected. There is a local DNS server that is always available so I’m not sure why asterisk is so unresponsive. I’ve seen this on two different systems, and on 1 of them I commented out my SIP providers in sip.conf and it ran ok again. Thanks Peter. -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.410 / Virus Database: 268.17.10/651 - Release Date: 24/01/2007 ___ --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 ___ --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] Trouble with incoming calls
What appears on the Asterisk console?
PaulH
On Sun, 2007-01-28 at 20:06 -0500, James Caffrey wrote:
Hello everyone. I am having trouble receiving via my Linksys SPA-3102.
I have not problem dialing out. It is like asterisk never even sees
the call. I have 3 sip devices grandstream bt-100, spa-3102 fxs, and
spa-3102 fxo. A very simple setup, just getting familar with asterisk.
Here are my relative config files. let me know if you need more.
sip.conf
[general]
context=default
bind=0.0.0.0
bindport=5060
srvlookup=yes
[100] ;bt-100
type=friend
username=100
context=default
secret=secret
host=dynamic
dtmfmode=rfc2833
disallow=all
allow=ulaw
[EMAIL PROTECTED]
[101] ;fxs
type=friend
username=pots
context=default
secret=phone
host=dynamic
dtmfmode=rfc2833
disallow=all
allow=ulaw
[EMAIL PROTECTED]
[102] ;fxo
type=friend
context=default
secret=pstn
host=dynamic
dtmfmode=rfc2833
disallow=all
allow=ulaw
port=5061
extensions.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
context=default
[globals]
RINGGROUP1 = SIP/100SIP/101
[default]
; These next three lines are for testing, just to make sure I got the
call, but no good
exten = s,1,Answer
exten = s,2,System(touch $HOME/got_it)
exten = s,3,Hangup
;exten = s,1,Dial(SIP/100,10)
;exten = s,2,Hangup
exten = 97,1,Dial(${RINGGROUP1},10)
exten = 97,n,Hangup
exten = 98,1,Answer
exten = 98,n,AGI(agi-test.agi)
exten = 98,n,Hangup
exten = 99,1,Answer
exten = 99,n,Playback(hello-world)
exten = 99,n,Hangup
exten = 100,1,Answer
exten = 100,n,Dial(SIP/100,15)
exten = 100,n,VoiceMail([EMAIL PROTECTED])
exten = 100,n,Playback(vm-goodbye)
exten = 100,n,Hangup
exten = 101,1,Answer
exten = 101,n,Dial(SIP/101)
exten = 101,n,Hangup
exten = _XX,1,Dial(SIP/102/${EXTEN})
exten = _XX,n,Hangup
I appreciate your help
- Jim
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Re: [asterisk-users] T1 Wire Level Tapping
Shane Spencer wrote: I am trying to do a wire level tap on T1 equipment using digum equipment. So far most call monitoring hardware for call centers try to stay on the analog side requiring a lot of rewiring. I have already posted to the list about T1 bridging using DAC's support in the zaptel drivers. I still don't know if I can spy on channel information since I don't have any digium hardware on me until the project begins. There are a number of systems using ISDN digital taps. The proper way requires a high impedance bridge - you don't want to load the line that you're tapping. Anybody found a method of spying on a D-Channel and all voice channels using standard T1 equipment? I am making a rough assumption that if I can trick the zaptel drivers into operating without anything responding to a TX signal then I can do the following: You can directly bridge the 2 ports and extract what you need as you bridge - see pridump.c in libpri. You don't even need asterisk, just the zaptel and libpri. The only problem with this approach, is that the bridge becomes a point of failure. Your box down, your PRI goes down as well. S-T1 = T1 to Spy On T1-1 = Digium T1 card #1 T1-2 = Digium T1 card #2 Map S-T1(RX) to T1-1(RX) and S-T1(TX) to T1-2(RX) and decode the D-Channel where appropriate, should I be able to spy on the RX/TX channels enough to make a recording including CID information? This would help in situations where the monitoring system needs to be replaced or taken down without bothering in-progress calls. This is technically correct, but I don't know how well it works. Eicon recommends a similar technique to do monitoring with their Eicon Server cards. For the BRI, it's done this way. But for the PRI card, they actually suggest using a custom cable. Eicon cards have a special Hi-Z monitoring mode to support this application. http://www.eicon.com/worldwide/solutions/How_To_Call_Tapping_and_Monitoring_with_Diva_Server FYI, Voicetronix has a Hi-Z version of their OpenPRI card that work with an open-sourced voice logging application available from their site. Leo ___ --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] Does X100P decode caller ID?
It is, and is identified by wcfxo as a Wildcard FXO: Wildcard X100P. So much for The DigitNetworks X100P is detected as an actual X101P card. IIRC, there were 2 Digium single FXO cards - the X100P using the Motorola SM56 and the X101P with Intel/Ambient 537. The X101Ps have 2 RJ-11 jacks. Functionally, they're all Winmodems - effectively just DAAs connected to the PCI bus. The Zaptel driver is responsible for the caller ID and DTMF detection. Maybe you have a borked card or it could be due to impedance mismatch. I know that the X101P only works with FCC 600 Ohm impedance. For other parts of the world, YMMV. Leo ___ --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] Voicemail from sip phones
[EMAIL PROTECTED] wrote: Hmm. Nope. Still same thing. I added pedantic=yes both in the general context in sip.conf and in the user's context in sip.conf with no change. Just for fun, I also changed it to pedantic=no in each place with no luck either. (I stopped and started asterisk between each change). Other thoughts? check that your phone is not using *8 in its own dial plan. Also, do a sip debug and see that the phone is actually sending *8 to asterisk. Leo ___ --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] ATCOM AT 468 manuals and firmware anyone?
both not available. but thanks. On 1/28/07, Leif Neland [EMAIL PROTECTED] wrote: Erick Perez wrote: Hi there, im looking for another place that provides manuals and firmware updates for the ATCOM AT 468 and their configuration with asterisk. the site www.atcom.com.cn has non functional download links. I suppose you mean the AG 468 If you can find somebody who still uses Internet Explorer, the links works. The download page used to have a link for a page which worked in Firefox, but not anymore. But anyway, here are the links. http://atcom.com.cn/en/down/userguide/EN/AG-468/AG468_User_ManualGuide.rar http://atcom.com.cn/en/down/program/en/ng_series/ag468_060119_vr41rls.zip Leif ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Trouble outgoing VOIP Provider Calls
I have a weird problem
Asterisk 1.4
E100P connected to a Panasonic TDA phone system
Here is what I get
SIP Ext - Panasonic Extensions No Problems
Panasonic Ext - SIP Extensions No Problems
SIP Ext - VOIP Provider No Problems
Panasonic Ext - VOIP Provider Errors
-- Working SIP - VOIP
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/610-097aee60,
SIP/acevoip/03) in new stack
-- Called acevoip/03
-- SIP/acevoip-097b52c0 is making progress passing it to
SIP/610-097aee60
-- SIP/acevoip-097b52c0 is making progress passing it to
SIP/610-097aee60
== Spawn extension (from-sip, 903, 1) exited non-zero on
'SIP/610-097aee60'
-- Not Working Pana - VOIP
-- Executing [EMAIL PROTECTED]:1] Dial(Zap/31-1,
SIP/acevoip/03) in new stack
-- Called acevoip/03
[Jan 29 11:00:36] WARNING[20642]: chan_sip.c:11731
handle_response_invite: Received response: Forbidden from 'Unknown
sip:[EMAIL PROTECTED];tag=as3a292a14'
-- SIP/acevoip-097b1358 is circuit-busy
--
Both numbers dialled were exactly the same (9 is the leading number on
all calls in the system and is stripped before dialing), I just replaced
the numbers with .
Tested from several different sip phones and Pana handsets, and it is
only with outgoing calls to VOIP, incoming that go to a Pana extensions
work fine.
--- Extensions.conf
[dialstring]
exten = t,1,Dial(Zap/g1/100,60,tn)
exten = i,1,Dial(Zap/g1/100,60,tn)
[from-e100p]
include = dial-sip
include = out-voip
[dial-e100p]
exten = _1XX,1,System(mkdir /mnt/data/Recording/${CALLERID(num)})
exten =
_1XX,2,Set(CALLFILENAME=/mnt/data/Recording/${CALLERID(num)}/${CALLERID(
num)}-Called-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${EXTEN})
exten = _1XX,3,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0))
exten = _1XX,4,Dial(Zap/g1/${EXTEN},90,r)
exten = _91XX,1,System(mkdir /mnt/data/Recording/${CALLERID(num)})
exten =
_91XX,2,Set(CALLFILENAME=/mnt/data/Recording/${CALLERID(num)}/${CALLERID
(num)}-Called-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${EXTEN:1})
exten = _91XX,3,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0))
exten = _91XX,4,Dial(Zap/g1/${EXTEN:1},90,r)
exten = _9X.,1,System(mkdir /mnt/data/Recording/${CALLERID(num)})
exten =
_9X.,2,Set(CALLFILENAME=/mnt/data/Recording/${CALLERID(num)}/${CALLERID(
num)}-Called-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${EXTEN:1})
exten = _9X.,3,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0))
exten = _9X.,4,Dial(Zap/g1/${EXTEN},90,r)
exten = _9X.,5,Busy
exten = 000,1,System(mkdir /mnt/data/Recording/${CALLERID(num)})
exten =
000,2,Set(CALLFILENAME=/mnt/data/Recording/${CALLERID(num)}/${CALLERID(n
um)}-Called-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${EXTEN})
exten = 000,3,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0))
exten = 000,4,Dial(Zap/g1/000,60,r)
exten = 9000,1,System(mkdir /mnt/data/Recording/${CALLERID(num)})
exten =
9000,2,Set(CALLFILENAME=/mnt/data/Recording/${CALLERID(num)}/${CALLERID(
num)}-Called-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${EXTEN:1})
exten = 9000,3,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0))
exten = 9000,4,Dial(Zap/g1/000,60,r)
[out-voip]
exten = _902X.,1,Dial(SIP/acevoip/${EXTEN:1})
exten = _903X.,1,Dial(SIP/acevoip/${EXTEN:1})
exten = _905X.,1,Dial(SIP/acevoip/${EXTEN:1})
exten = _906X.,1,Dial(SIP/acevoip/${EXTEN:1})
exten = _908X.,1,Dial(SIP/acevoip/${EXTEN:1})
exten = _954X.,1,Dial(SIP/acevoip/${EXTEN:1})
exten = _955X.,1,Dial(SIP/acevoip/${EXTEN:1})
[from-acevoip]
include = dialstring
exten = 073...,1,Answer
exten = 073...,2,Dial(Zap/g1/100,60,tn)
exten = _073.XX,1,Answer
exten = _073.XX,2,System(mkdir
/mnt/data/Recording/${SIP_HEADER(TO):12:3})
exten =
_073.XX,3,Set(CALLFILENAME=/mnt/data/Recording/${SIP_HEADER(TO):12:3
}/${SIP_HEADER(TO):12:3}-Received-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-$
{CALLERID(num)})
exten = _073.XX,4,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0))
exten = _073.XX,5,Dial(SIP/${SIP_HEADER(TO):12:3},60,tn)
exten = _073.XX,6,Voicemail(${SIP_HEADER(TO):12:3}u)
exten = _073.XX,7,Hangup
exten = _073.XX,106,Voicemail(${SIP_HEADER(TO):12:3}u)
exten = _073.XX,107,Hangup
include = dial-sip
include = dial-e100p
[from-sip]
include = dialstring
include = dial-sip
include = out-voip
include = dial-e100p
[dial-sip]
exten = 600,1,Dial(Zap/g1/100,60,tr)
exten = 9600,1,Dial(Zap/g1/100,60,tr)
exten = _6XX,1,SetMusicOnHold(random)
exten = _6XX,2,System(mkdir /mnt/data/Recording/${EXTEN})
exten =
_6XX,3,Set(CALLFILENAME=/mnt/data/Recording/${EXTEN}/${EXTEN}-Received-$
{STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${CALLERID(num)}.wav49)
exten = _6XX,4,MixMonitor(${CALLFILENAME}|v(0)V(0))
exten = _6XX,5,Dial(SIP/${EXTEN},45,Ttr)
exten = _6XX,6,Voicemail(u${EXTEN})
exten = _6XX,7,Hangup
exten = _6XX,106,Voicemail(b${EXTEN})
exten = _6XX,107,Hangup
exten = _96XX,1,SetMusicOnHold(random)
exten =
[asterisk-users] Re: Via EPIA channel_find_locked: Avoided initial deadlock
I have tried compiling asterisk with -march 586 and 386 and the deadlocks minimizedin 386 but did not dissapear. Is this because of asterisk, my epia or centos? On 1/27/07, Erick Perez [EMAIL PROTECTED] wrote: In asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 My equipment is a Via EPIA minit-itx CN1 1.2ghz, 1gb ram and a tdm400p (4fxo). A call comes from zap, a SIP ulaw receives the call, talks for a while and when SIP users tries to park the call, then dozens of... WARNING[3853]: channel.c:781 channel_find_locked: Avoided initial deadlock for '0x91bb840', 10 retries! I use stock Centos 4.4 with kernel 2.6.9-42.0.3.EL i686. I guess also asterisk was compiled for i686. and the machine is completely unusable, I need to reboot. I posted the digium script output from autosupport. It is available at: http://pastebin.com/868590 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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: Delay in Call Distribution using the Queue Application
[EMAIL PROTECTED] wrote: Thanks for the info, is there a patch available for version 1.2 that adds the autofill option? Gavin Hamill has back ported some of the 1.4 queue features into 1.2. See his post to this list http://www.mail-archive.com/asterisk-users@lists.digium.com/msg171158.html Leo ___ --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] Sangoma card dying after 1hour
We made our own, but it isn't shielded. Is there something specific to sangoma regarding cabling? We've made our own for Digium and Nortel equipment and all is well. - Jeremy From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rosli Sukri Sent: Sunday, January 28, 2007 1:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sangoma card dying after 1hour what kinda of cable are you using to connect to the e1. isit shielded or just the generic one they give with the card? On 1/28/07, Porier, Jeremy M. [EMAIL PROTECTED] wrote: Do you see anything in /var/log/messages? I am having a similar problem but I'm also getting some pci fatal error! messages. I had sangoma connect to the box and he couldn't find any config errors so we're leaning towards a hardware problem. - Jeremy -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] On Behalf Of Jon Schøpzinsky Sent: Friday, January 26, 2007 7:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Sangoma card dying after 1hour Asterisk is version 1.2.14, zaptel 1.2.12, libpri is whatever version was with zaptel 1.2.12 :) Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies Sent: 26. januar 2007 12:03 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sangoma card dying after 1hour Which asterisk versions etc etc? On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote: I am running the newest version, from the sangoma wiki. Jon -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] On Behalf Of Steve Davies Sent: 26. januar 2007 10:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sangoma card dying after 1hour On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote: Hello List I am having a rather big problem with a sangoma A104 card, I just installed to replace a Digium TE410 card, that was acting up. But now we have a problem with the sangoma card. It runs great after being started, and calls proceed as normal, but after about 1 hour, it stops being able to make and receive calls. If I run wanpipemon debug, can see that the card still receives packets from the ISDN, but when I make a call, I cant see it in wanpipemon, and asterisk just responds with a: NOTICE[17240] app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) I am pretty shure that this is a configuration issue, but are there anything I need to be aware of when moving from a Digium card to a sangoma card? Which wanpipe version? Anything lower than 2.3.4-4 should be upgraded as there are some resource leak fixes in that version. Regards, Steve ___ --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 ___ --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 ___ --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 ___ --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 ___ --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 ___ --Bandwidth and Colocation provided by
[asterisk-users] Cordless SIP Phones
Can anyone recommend a good cordless user-configurable SIP hardphone that is readily available in the states and doesn't cost $300? There seem to be a plethora of decent and affordable corded phones (like from Grandstream) but the search for a cordless unit seems elusive. I purchased a vtech 8100 online only to discover after receiving it that it is locked to vonage service. Thank you. Ed ___ --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] Voicemail from sip phones
I don't see anything apparent in for the SIP phones that would indicate that *8 is anything but VoiceMailMain(). I looked in extensions.conf for it, as well as sip.conf (the config that I pasted in a previous e-mail. It does indeed appear that a literal *8 is being passed to asterisk. Here's the relevant context of extensions.conf: [internal] ;downstairs office exten = 7191,1,Dial(Zap/1) ;cordless exten = 7192,1,Dial(Zap/2) ;second cordless exten = 7193,1,Dial(SIP/ht3861) ;call a couple phones exten = 7194,1,Dial(SIP/gxp2SIP/user) ;goto voicemail exten = *8,1,VoiceMailMain() include = local include = ld The sip.conf bits are contained in my initial post. Here's the debug output from the console, it's somewhat long. Could the key line be (towards the bottom) this? [Jan 28 20:39:10] NOTICE[31924]: chan_sip.c:13519 handle_request_invite: Nothing to pick up for OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI. Here's the full output: --- SIP read from 192.168.1.165:5806 --- INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.165:5806;branch=z9hG4bK-d87543-42396923770b6e49-1--d87543-;rport Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:5806 To: *8sip:[EMAIL PROTECTED] From: SIP Usersip:[EMAIL PROTECTED];tag=2a71c861 Call-ID: OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 328 v=0 o=- 6 2 IN IP4 192.168.1.165 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.165 t=0 0 m=audio 35172 RTP/AVP 107 119 0 98 8 3 101 a=alt:1 1 : UGnBtE65 8rk0z5iz 192.168.1.165 35172 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:119 BV32-FEC/16000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv - --- (12 headers 13 lines) --- Sending to 192.168.1.165 : 5806 (NAT) Using INVITE request as basis request - OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI. Found user 'user' for 'user' ord*CLI --- Reliably Transmitting (no NAT) to 192.168.1.165:5806 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.165:5806;branch=z9hG4bK-d87543-42396923770b6e49-1--d87543-;received=192.168.1.165;rport=5806 From: SIP Usersip:[EMAIL PROTECTED];tag=2a71c861 To: *8sip:[EMAIL PROTECTED];tag=as31c6aff1 Call-ID: OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI. CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=69d017e6 Content-Length: 0 Scheduling destruction of SIP dialog 'OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI.' in 32000 ms (Method: INVITE) ord*CLI --- SIP read from 192.168.1.165:5806 --- ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.165:5806;branch=z9hG4bK-d87543-42396923770b6e49-1--d87543-;rport To: *8sip:[EMAIL PROTECTED];tag=as31c6aff1 From: SIP Usersip:[EMAIL PROTECTED];tag=2a71c861 Call-ID: OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI. CSeq: 1 ACK Content-Length: 0 - --- (7 headers 0 lines) --- ord*CLI --- SIP read from 192.168.1.165:5806 --- INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.165:5806;branch=z9hG4bK-d87543-dc51447ec8163e0a-1--d87543-;rport Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:5806 To: *8sip:[EMAIL PROTECTED] From: SIP Usersip:[EMAIL PROTECTED];tag=2a71c861 Call-ID: OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1006e stamp 34025 Authorization: Digest username=user,realm=asterisk,nonce=69d017e6,uri=sip:[EMAIL PROTECTED],response=b82d4700cc1f72ef2711df0b597e7184,algorithm=MD5 Content-Length: 328 v=0 o=- 6 2 IN IP4 192.168.1.165 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.165 t=0 0 m=audio 35172 RTP/AVP 107 119 0 98 8 3 101 a=alt:1 1 : UGnBtE65 8rk0z5iz 192.168.1.165 35172 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:119 BV32-FEC/16000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv - --- (13 headers 13 lines) --- Sending to 192.168.1.165 : 5806 (NAT) Using INVITE request as basis request - OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI. Found user 'user' for 'user' Found RTP audio format 107 Found RTP audio format 119 Found RTP audio format 0 Found RTP audio format 98 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.165:35172 Found description format BV32 for ID 107 Found description format BV32-FEC for ID 119 Found description format iLBC for ID 98 Found description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1
Re: [asterisk-users] Cordless SIP Phones
I share your frustration. Might I suggest a Grandstream HT-386 (or 486, etc) gateway to a regular cordless phone? On Sun, Jan 28, 2007 at 09:52:19PM -0500, Edward Halman wrote: Can anyone recommend a good cordless user-configurable SIP hardphone that is readily available in the states and doesn't cost $300? There seem to be a plethora of decent and affordable corded phones (like from Grandstream) but the search for a cordless unit seems elusive. I purchased a vtech 8100 online only to discover after receiving it that it is locked to vonage service. Thank you. Ed ___ --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 ___ --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] Voicemail from sip phones
[EMAIL PROTECTED] wrote: Here's the debug output from the console, it's somewhat long. Could the key line be (towards the bottom) this? [Jan 28 20:39:10] NOTICE[31924]: chan_sip.c:13519 handle_request_invite: Nothing to pick up for OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI. Ah ha - your features.conf has *8 (the default) for group pick up. Leo ___ --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] Queue Manager
Hi, I'm looking a queue manager compatible with queues.conf. It should allow me to change agents from one queue to another and change it's priority without poblem. Also it must be web based :). Does anyone know any program? Thank you in advance! Santiago del Castillo ___ --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] Sangoma card dying after 1hour
try it out with shielded... we got the same problem previously, we resorted to making a short cable and wrapping it up with cooking foil and the problem appears no more On 1/29/07, Porier, Jeremy M. [EMAIL PROTECTED] wrote: We made our own, but it isn't shielded. Is there something specific to sangoma regarding cabling? We've made our own for Digium and Nortel equipment and all is well. - Jeremy -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Rosli Sukri *Sent:* Sunday, January 28, 2007 1:04 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Sangoma card dying after 1hour what kinda of cable are you using to connect to the e1. isit shielded or just the generic one they give with the card? On 1/28/07, Porier, Jeremy M. [EMAIL PROTECTED] wrote: Do you see anything in /var/log/messages? I am having a similar problem but I'm also getting some pci fatal error! messages. I had sangoma connect to the box and he couldn't find any config errors so we're leaning towards a hardware problem. - Jeremy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Schøpzinsky Sent: Friday, January 26, 2007 7:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Sangoma card dying after 1hour Asterisk is version 1.2.14, zaptel 1.2.12, libpri is whatever version was with zaptel 1.2.12 :) Jon -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Steve Davies Sent: 26. januar 2007 12:03 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sangoma card dying after 1hour Which asterisk versions etc etc? On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote: I am running the newest version, from the sangoma wiki. Jon -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Steve Davies Sent: 26. januar 2007 10:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sangoma card dying after 1hour On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote: Hello List I am having a rather big problem with a sangoma A104 card, I just installed to replace a Digium TE410 card, that was acting up. But now we have a problem with the sangoma card. It runs great after being started, and calls proceed as normal, but after about 1 hour, it stops being able to make and receive calls. If I run wanpipemon debug, can see that the card still receives packets from the ISDN, but when I make a call, I cant see it in wanpipemon, and asterisk just responds with a: NOTICE[17240] app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) I am pretty shure that this is a configuration issue, but are there anything I need to be aware of when moving from a Digium card to a sangoma card? Which wanpipe version? Anything lower than 2.3.4-4 should be upgraded as there are some resource leak fixes in that version. Regards, Steve ___ --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 ___ --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 ___ --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 ___ --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 ___ --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 ___ --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 ___ --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] Does X100P decode caller ID?
From: Leo Ann Boon [EMAIL PROTECTED] It is, and is identified by wcfxo as a Wildcard FXO: Wildcard X100P. So much for The DigitNetworks X100P is detected as an actual X101P card. IIRC, there were 2 Digium single FXO cards - the X100P using the Motorola SM56 and the X101P with Intel/Ambient 537. The X101Ps have 2 RJ-11 jacks. Functionally, they're all Winmodems - effectively just DAAs connected to the PCI bus. The Zaptel driver is responsible for the caller ID and DTMF detection. Maybe you have a borked card or it could be due to impedance mismatch. I know that the X101P only works with FCC 600 Ohm impedance. For other parts of the world, YMMV. Leo Is DTMF pass-through and caller ID fundamentally different? This card does not seem to cause significant difficulty in DTMF detection. The high DC voltage during ringing could be one factor. Any easy way to test serious mismatch? The 600 Ohm is actually on-hook resistance, right? Or is this audio impedance? The card is indeed sold and used in North America. Yuan Liu ___ --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] International Carriers
Facundo, the company that I work for use Crossfone, www.crossfone.com.ar Best Regards, Martín On 1/26/07, Facundo Ameal [EMAIL PROTECTED] wrote: Hello everyone! I 've looking for carriers which can terminate my international calls. They must accept payments from Argentina and give me interconection to my Asterisk. I'd appreciate your help or recomendations. Regards. -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --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 ___ --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] Does X100P decode caller ID?
Yuan LIU wrote: From: Leo Ann Boon [EMAIL PROTECTED] It is, and is identified by wcfxo as a Wildcard FXO: Wildcard X100P. So much for The DigitNetworks X100P is detected as an actual X101P card. IIRC, there were 2 Digium single FXO cards - the X100P using the Motorola SM56 and the X101P with Intel/Ambient 537. The X101Ps have 2 RJ-11 jacks. Functionally, they're all Winmodems - effectively just DAAs connected to the PCI bus. The Zaptel driver is responsible for the caller ID and DTMF detection. Maybe you have a borked card or it could be due to impedance mismatch. I know that the X101P only works with FCC 600 Ohm impedance. For other parts of the world, YMMV. Leo Is DTMF pass-through and caller ID fundamentally different? This card does not seem to cause significant difficulty in DTMF detection. The high DC voltage As I mentioned all the DSP work is done in software. So there shouldn't be any fundamental difference. during ringing could be one factor. Any easy way to test serious mismatch? The 600 Ohm is actually on-hook resistance, right? Or is this audio impedance? The 600 Ohm is off-hook AC impedance for US and countries that follow the same specs - consult your local regulatory docs. card is indeed sold and used in North America. But, where in the world are you using it? Leo ___ --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] Does X100P decode caller ID?
From: Leo Ann Boon [EMAIL PROTECTED] Is DTMF pass-through and caller ID fundamentally different? This card does not seem to cause significant difficulty in DTMF detection. The high DC voltage As I mentioned all the DSP work is done in software. So there shouldn't be any fundamental difference. during ringing could be one factor. Any easy way to test serious mismatch? The 600 Ohm is actually on-hook resistance, right? Or is this audio impedance? The 600 Ohm is off-hook AC impedance for US and countries that follow the same specs - consult your local regulatory docs. card is indeed sold and used in North America. But, where in the world are you using it? Leo I'm in FCC zone (North America). Maybe the DAA has a defect under high voltage. The way I heard it, sounded like the 20Hz ringing was passed through to ADC. Possibly the ADC isn't able to recover from saturation in between rings. Yuan Liu ___ --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] Cordless SIP Phones
You can usually meet this by using an FXS adapter along with a standard cordless phone and thus achieve the same result. If you *really* want a voip-cordless phone, check this out: http://www.voiplink.com/Aastra_480i_CT_p/aastra-480i-ct.htm p.s. I've had good experiences with VoIP Link (no I don't work for them!) On 1/28/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I share your frustration. Might I suggest a Grandstream HT-386 (or 486, etc) gateway to a regular cordless phone? On Sun, Jan 28, 2007 at 09:52:19PM -0500, Edward Halman wrote: Can anyone recommend a good cordless user-configurable SIP hardphone that is readily available in the states and doesn't cost $300? There seem to be a plethora of decent and affordable corded phones (like from Grandstream) but the search for a cordless unit seems elusive. I purchased a vtech 8100 online only to discover after receiving it that it is locked to vonage service. Thank you. Ed ___ --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 ___ --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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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] Rxfax and Txfax on Asterisk 1.4
AFAIK the current recommendation is to use HylaFax with something called iaxmodem. After having been through a lot of problems with RxFax and TxFax I completely agree with this statement. Allthough the initial configuration is a bit complicated, once you have this running you'll get far better reliability. Kind regards, Ardjan Zwartjes. ___ --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] Cordless SIP Phones
Siemens Gigaset IP phones (C450-IP, S450-IP) are not that bad (gigaset.siemens.com). C450IP costs less than 100 USD (in Italy at least), S450 is slightly more expensive. Grandstream HT-286 works quite well with DECT handsets too. (I've deployed both and both are working with Asterisk). Alberto. [EMAIL PROTECTED] ha scritto: I share your frustration. Might I suggest a Grandstream HT-386 (or 486, etc) gateway to a regular cordless phone? On Sun, Jan 28, 2007 at 09:52:19PM -0500, Edward Halman wrote: Can anyone recommend a good cordless user-configurable SIP hardphone that is readily available in the states and doesn't cost $300? There seem to be a plethora of decent and affordable corded phones (like from Grandstream) but the search for a cordless unit seems elusive. I purchased a vtech 8100 online only to discover after receiving it that it is locked to vonage service. Thank you. Ed ___ --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
