Re: [Asterisk-Users] `hint` priority and Polycom 500
Olle E. Johansson wrote: Sean Kennedy wrote: Hi all, I'm trying to see if I can get the hint priority working with my polycom 500. So far I have 2 /reg entries with the same sip registration, one is labeled as private, the other as shared. I have set the hint priority before anything else in my dialplan for my extensions. As it stands, I have two registrations on the phone, one has a half greyed out phone icon, the other is a full icon. However, when I place a call to that phone, the shared line display doesn't change. Currently we don't support the shared line implementation in a lot of phones. I don't know how the Polycom implements shared lines yet, but if I get hold of one I'll take a look. Maybe someone on the list knows? /Olle Astricon - the Asterisk User's conference - Madrid June 15-17 http://www.astricon.net/europe/ - Register today! Olle, If you need a Polycom, let me know where I can send one of my IP 600's. Seriously, contact me off list! -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie with kphone and asterisk
hello all,
i have already configure sip.conf and dialplan.
i done the follow me script.
first problem:
i want to call(with kphone) someone at my extension, i
must dial the extension number.
i can't dial their username.
[EMAIL PROTECTED] (work)
[EMAIL PROTECTED] (call fail)
is it possible to do that??
second problem:
if i want to call another number (not my extension)
with my kphone also fail.
example if i call my mobile than it fail.
where i must configure so my asterisk can do that??
my sip.conf:
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw
[mustafa]
type=friend
secret=mustafa
host=dynamic
dtmfmode=inband
mailbox=1604
extension.conf:
exten = 20531604,1,Dial(SIP/mustafa,20)
exten = 20531604,2,playback(pls-wait-connect-call)
exten = 20531604,3,Setvar(NewCaller=${CALLERIDNUM})
exten = 20531604,4,SetCIDNum(0${CALLERIDNUM})
exten = 20531604,5,dial(${TRUNK}/0193041624,20,r)
exten = 20531604,6,SetCIDNum(${NewCaller})
exten = 20531604,7,VoiceMail(u1604)
exten = 20531604,8,Hangup
can anyone give an advise or some idea??
please..
thanks..
regard:
shahdan
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
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] Phone always busy after caller hangup
Hi, If you use digium card, then maybe you set wrong signaling on fxs... Best regards, Stevanus Tim P wrote: I have multiple sipura 2100 boxes connected to my * box and for some reason that i cannot figure out when making a call to one and answering it and then hanging up results in the line be permanently busy (the phone called is permanently busy until * is rebooted). Any idea where to start with this one? It seems to me that either the SPA2100 is not registering the end of the call or * isn't. I suspect the SPA2100 but see nothing in the logs or in the SPA config to indicate a fix. Any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Phone always busy after caller hangup
Hi, The problem is that the Sipura boxes don't do call progress monitoring. I saw this on a thread about a week ago. If the call is dropped at the * side, then the sip channel is droppeds and the sipura will drop the PSTN connection. However if the Sipura has trouble with the PSTN's start/stop process (Signalling) then the sipura may think that the call is still connected if the remote (PSTN) side terminates the call. Regards, T -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of stevanus Sent: Wednesday, 1 June 2005 5:37 PM To: Tim P; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Phone always busy after caller hangup Hi, If you use digium card, then maybe you set wrong signaling on fxs... Best regards, Stevanus Tim P wrote: I have multiple sipura 2100 boxes connected to my * box and for some reason that i cannot figure out when making a call to one and answering it and then hanging up results in the line be permanently busy (the phone called is permanently busy until * is rebooted). Any idea where to start with this one? It seems to me that either the SPA2100 is not registering the end of the call or * isn't. I suspect the SPA2100 but see nothing in the logs or in the SPA config to indicate a fix. Any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Built-In Transfer Questions
On Wednesday 01 June 2005 06:45, Jennifer Hales wrote: Hello Matthew, You need to put exten = o,1,Hangup underneath your voicemail macro, then if your dial zero the call will come back to you, however it does read back an error in your ear. It still works. ... or alternatively, if you add the 'h' option to the Dial command, you will be able to hang up by pressing the * key on your phone (or if you use CVS, any sequence you define as disconnect in the [featuremap] section of features.conf) Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dynamic IAX Server
Hi: I read many documents and I posted my question several times here without luck. I hope someone can help now please. Here is an example to demonstarte my problem: Suppose you manage the FWD server, how do you define an IAX client behind nat so he can receive calls from FWD. NAT client would register with FWD to let it know how to locate it. I just don't see how FWD finds the nat client. How is that translated in terms of IAX.conf contexts and what FWD dial in extensions.conf file. Regards; Chawki __ Discover Yahoo! Use Yahoo! to plan a weekend, have fun online and more. Check it out! http://discover.yahoo.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Phone always busy after caller hangup
hi, i m new to asterisk word, pl. help me for the below scenario i have installed TDM22B card. Module is - wcfxs i m in India so first of all wat zone is to specified is not defined? Zaptel.conf is - fxoks=1-2 fxsks=3-4 # ztcfg parses it cleanly. Zapata.conf contains- signalling=fxo_ks callerid=asreceived group=1 context=default ; points to the default context of extensions.conf file. channel=1-2 ;for FXS interfaces on the TDM22 card installed. signalling=fxs_ks ; to signal an internal FXO inferaces group=2 context=incoming channel=3-4 its just simple configuration. running asterisk gives error - unable to specify channel 1: no such device unable to register channel 1-2 On 6/1/05, stevanus [EMAIL PROTECTED] wrote: Hi,If you use digium card, then maybe you set wrong signaling on fxs...Best regards,StevanusTim P wrote:I have multiple sipura 2100 boxes connected to my * box and for some reason that i cannot figure out when making a call to one andanswering it and then hanging up results in the line be permanentlybusy (the phone called is permanently busy until * is rebooted).Any idea where to start with this one?It seems to me that either theSPA2100 is not registering the end of the call or * isn't.I suspectthe SPA2100 but see nothing in the logs or in the SPA config to indicate a fix.Any ideas?___Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Kulbir SinghEon Infotech LimitedSCO 315-316Sector 35-BChandigarh - 160 022 INDIAt: +91 172 2609849 f: +91 172 2615465m: 09872822266 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] When to use 'Answer' and when NOT to...
While everything seems to be working for the most part correctly in my mix-network of Zap and Sip phones, it occurred to me that every call, regardless of whether or not it was answered, is reporting ANSWERED in the cdr records on mysql. I was having problems with strange hang-ups the moment a call went off hook, and having Answer in the extensions.conf contexts made it all go away. Am I under-thinking the use of Answer()? Chris Coulthurst [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IVR Load
Hi, Thinking about an IVR application and trying to get a handle on the best way to structure it so that the maximum number of concurrent calls can be achieved.. If the voice prompts were stored in a GSM format and were being played out through an IAX trunk that uses GSM compression would asterisk do a decompress/compress on the audio or would it simply pass through the GSM encoding? Obviously if I could eliminate the decompress/compress activity is would make the server far more scalable.. Thanks.. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hardware questions
Hello! I would like to know which hardware I need, to use asterisk with up to 20 analog lines. Also I woul like to know if there is any card that suport both analog and isdn lines, and if there is any way to make the analog phones now I'm using work with asterisk. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: **POSSIBLE SPAM** [Asterisk-Users] AreskiCC - DOES IT REALLY WORK??????
I think we should be thankful that the authors are relasing the software, rather then crying out loud when you cannot get it to work. More people will be willing to help you that way. Be ashamed of yourself! Best Regards, == David Choo Sales Engineer Business Technology Division Engineered for Changing Businesses Espore Corp Pte Ltd 68 Kallang Pudding Rd #04-03 SYH Logistics Bldg Singapore 349327 Tel: 65-6842 2725, Ext - 404 Fax : 65-6842 2724 E-mail :[EMAIL PROTECTED] = Privileged/Confidential information may be contained in this message. If you are not the intended recipient, you must not copy it or use it for any purpose, nor deliver this message to anyone. Instead, please delete this message and destroy any other record of it immediately and kindly notify the sender by return email. Thank you for your co-operation. Internet communications cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, arrive late, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the context of this message nor can the sender guarantee that this message is virus free. [EMAIL PROTECTED] t.com Sent by: To asterisk-users-bo asterisk-users@lists.digium.com [EMAIL PROTECTED] cc m.com Subject **POSSIBLE SPAM** [Asterisk-Users] 31/05/2005 11:26 AreskiCC - DOES IT REALLY PMWORK?? Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com Hi all, I am quite disappointed at the application AreskiCC. I have installed everything following the instructions but the thing doesnt want to work. First of all, when I start the index.php page, any name/password logs in. After the login it takes me to a page with a single option LOGOUT We are monitoring the database and it seems like the application doesnt connect to it. Does anybody in this have made this work? Can someone help me please?? Thanks, Robson___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] UPS rating for SOHO asterisk box
Daryl G. Jurbala wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver Sent: Tuesday, May 31, 2005 5:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] UPS rating for SOHO asterisk box [...] Regarding this, I have done this hack yesterday: - Remove the battery from an existing UPS - Rewire the UPS onto biggest car lead acid battery (12v) you can find. Et voila! Bigger capacity. Put the batteries in parrallel and you do get monstruous UPS capacity... the only trouble with it is that re-charging the batteries may take some time. [...] Congratulationsyou've just given this part-time small town fire marshal and 14-year fire service veteran nightmares. Kidsdo NOT try this at home. The inverters in small UPSes are not designed to deal with runtimes that exceed the batteries in them. If you run this setup well past the time it was designed to run (by adding 3, 4, or more times that battery capacity it was ever designed to have) that chances of a catastrophic inverter failure (meaning flash, boom, fire) are very real and very likely. Ouch... In the test I have done, I replaced a HR 1224W F2F1 lead acid sealed battery by a fulmen heavy duty 95 amp/hours battery. The UPS flattened the battery out after 70 minutes instead of the original 15 minutes. However, charging *is* slow: it's been now 36 hours and it's still charging. Looks like I'll be better off buying a proper smart charger along with a decent inverter. I wouldn't want to fry the house :) -- Ykoz Un Max - La VoIP en pré-payé! Essayez gratuitement - 5 crédits offerts. --- http://ykoz.net/voip/max --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unreliable DTMF detection with DISA on incoming Zap channel on bristuffed * and GSM gateway
Hi, I'm getting unusable DTMF detection with DISA on incoming ZAP channel (bristuffed *) on quadbri from GSM gateway. DTMF detection works ok in normal ISDN incoming line. How can I check what's going on ? What settings to check ? Anyone with more experience on such scenarios ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BT101 new firmware problem (1.0.6.3)
Hello, We found out that after upgrading the firmware in our GrandStream BudgeTone phones, that we were not able to transfer calls anymore. We use the BT's own tranfering mechanisme. We can dial the phone where the call should be tranfered to. But after that, the original caller stays in music on hold on the server and there's no way to get the calling channel back again (not to the first receiver, nor to the transfering target). At first I was thinking it had something to do with the asterisk, because upgraded to version 1.0.7 a week ago. Though, the strange thing is that we also have some elmeg/snom190's and they do not have this transfer problem. Not being able to transfer calls is a major problem. I'm puzzled. I didn't thought upgrading a firware would distroy existing functionality. The new firware version in the BT's is 1.0.6.3, the old firmware was still one of the 1.5.x series. We are unable to downgrade the firmware. Something to do with missing files in the older versions compared to the new one. Is there anybody that has similar problems with the BT's? Is there anybody that might have an idee or advise how to fix this? Please regards; elwin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] UPS rating for SOHO asterisk box
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver Sent: Wednesday, 1 June 2005 6:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] UPS rating for SOHO asterisk box Daryl G. Jurbala wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver Sent: Tuesday, May 31, 2005 5:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] UPS rating for SOHO asterisk box [...] Regarding this, I have done this hack yesterday: - Remove the battery from an existing UPS - Rewire the UPS onto biggest car lead acid battery (12v) you can find. Et voila! Bigger capacity. Put the batteries in parrallel and you do get monstruous UPS capacity... the only trouble with it is that re-charging the batteries may take some time. [...] Congratulationsyou've just given this part-time small town fire marshal and 14-year fire service veteran nightmares. Kidsdo NOT try this at home. The inverters in small UPSes are not designed to deal with runtimes that exceed the batteries in them. If you run this setup well past the time it was designed to run (by adding 3, 4, or more times that battery capacity it was ever designed to have) that chances of a catastrophic inverter failure (meaning flash, boom, fire) are very real and very likely. Ouch... In the test I have done, I replaced a HR 1224W F2F1 lead acid sealed battery by a fulmen heavy duty 95 amp/hours battery. The UPS flattened the battery out after 70 minutes instead of the original 15 minutes. However, charging *is* slow: it's been now 36 hours and it's still charging. Looks like I'll be better off buying a proper smart charger along with a decent inverter. I wouldn't want to fry the house :) I have many sites that have a 35amp Charger with 2 x 400ah 900CCA deep cycle batteries (10 year warranty), and 1000VA inverters. The combination makes for perfect power and about 2.5 days run time with my network kit whish consists of several Dlink wifi access points, 1 xbox (hacked into a router/firewall) and a vsat system. Total cost for the power kit AUD$1400 all up, and not a single second of downtime in over a year. On the flipside, I have seen a ups flare when the transformer overheated and melted the varnish, nasty! Regards, T ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems hanging up PSTN line
I am having problems with * not hanging up an incoming PSTN line, if that line is not answered before the person calling in hangs up.The line hangs in various states, it has hung with a busy tone, with no tone at all. I am running [EMAIL PROTECTED] and have a digium 4port line card. This wasconfigured by the genzaptel command I then added trunks for each line. I also have a Pulver WiSip phone which I need to be able to transfercalls on by dialing a prefix. Eg *2ext Any help on either problem would be greatly appreciated, I'm stilllearning my way around the whole * [EMAIL PROTECTED] thing. Rick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Problems hanging up PSTN line
Hi, What version of Asterisk @ Home are you using? I had problems like that until I upgraded to version 1.0 The problem has not recurred since. Regards,T From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]Sent: Wednesday, 1 June 2005 7:24 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Problems hanging up PSTN line I am having problems with * not hanging up an incoming PSTN line, if that line is not answered before the person calling in hangs up.The line hangs in various states, it has hung with a busy tone, with no tone at all. I am running [EMAIL PROTECTED] and have a digium 4port line card. This wasconfigured by the genzaptel command I then added trunks for each line. I also have a Pulver WiSip phone which I need to be able to transfercalls on by dialing a prefix. Eg *2ext Any help on either problem would be greatly appreciated, I'm stilllearning my way around the whole * [EMAIL PROTECTED] thing. Rick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] UPS rating for SOHO asterisk box
I have many sites that have a 35amp Charger with 2 x 400ah 900CCA deep cycle batteries (10 year warranty), and 1000VA inverters. Those deep cycles batteries look quite appropriate... in which kind of store do you get them? The combination makes for perfect power and about 2.5 days run time with my network kit whish consists of several Dlink wifi access points, 1 xbox (hacked into a router/firewall) and a vsat system. Total cost for the power kit AUD$1400 all up, and not a single second of downtime in over a year. Looks pretty cool :) On the flipside, I have seen a ups flare when the transformer overheated and melted the varnish, nasty! Woops! Well, at the moment I have only 95 amp/hour and the power drain on the UPS is pretty low (about 130W). Still, it looks like I need to get a proper charger / inverter :-/ Cheers, Jean-Michel. -- Ykoz Un Max - La VoIP en pré-payé! Essayez gratuitement - 5 crédits offerts. --- http://ykoz.net/voip/max --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Problems hanging up PSTN line
On Wed, 2005-06-01 at 21:23, [EMAIL PROTECTED] wrote: I am having problems with * not hanging up an incoming PSTN line, if that line is not answered before the person calling in hangs up. The line hangs in various states, it has hung with a busy tone, with no tone at all. I am running [EMAIL PROTECTED] and have a digium 4port line card. This was configured by the genzaptel command I then added trunks for each line. I think that genzaptel uses fxs_ks as the default for the fxo devices. This need to be changed to fxs_ls. Then it should work fine. I also have a Pulver WiSip phone which I need to be able to transfer calls on by dialing a prefix. Eg *2ext Any help on either problem would be greatly appreciated, I'm still learning my way around the whole * [EMAIL PROTECTED] thing. Stay with it. It's worth the effort. Mike Rick __ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] send and receive MMS
Hello, did anyone already experience MMS? SMS works fine, but I can't find infos on how to send and receive MMS on a similar way with Asterisk. Thanks Daryan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Problems hanging up PSTN line
Im running 0.9 I will try upgrading thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Terry H. Gilsenan Sent: Wednesday, June 01, 2005 9:33 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Problems hanging up PSTN line Hi, What version of Asterisk @ Home are you using? I had problems like that until I upgraded to version 1.0 The problem has not recurred since. Regards, T From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, 1 June 2005 7:24 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Problems hanging up PSTN line I am having problems with * not hanging up an incoming PSTN line, if that line is not answered before the person calling in hangs up.The line hangs in various states, it has hung with a busy tone, with no tone at all. I am running [EMAIL PROTECTED] and have a digium 4port line card. This wasconfigured by the genzaptel command I then added trunks for each line. I also have a Pulver WiSip phone which I need to be able to transfercalls on by dialing a prefix. Eg *2ext Any help on either problem would be greatly appreciated, I'm stilllearning my way around the whole * [EMAIL PROTECTED] thing. Rick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] debugging zap channel
Hi, I cannot seem to establish what is causing my analogue line to be generating incoming calls, so I would like to do some debugging on my Zap channel. Can anyone confirm the syntax? I have tried; Debug channel Zap/2 Debug channel Zap/2-1 Debug channel zap/2 Debug channel zap/2-1 Debug channel zap 2 Debug channel zap 2-1 Debug channel zap 02 Debug channel 02 All of which just comes back with no such channel. Any help or tips would be greatly appreciated. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: R: R: R: [Asterisk-Users] AT-320 + supervised transfer
Hi Gavin, I'm testing atxfer and it looks work fine, but i have a small problem: A call B B answer, dial atxfer extension and then the new peer (C) If C does not answer the phone, A and B got hangup and cannot speak again I set canreinvite to no. Can u help me ? Thanks Giordano -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Gavin Hamill Inviato: martedì 31 maggio 2005 16.21 A: asterisk-users@lists.digium.com Oggetto: Re: R: R: R: [Asterisk-Users] AT-320 + supervised transfer On Tuesday 31 May 2005 14:41, Giordano Grandis wrote: Hi Gavin, But...how atxfer work ? Ehm, just the way I explained yesterday :) Just make sure you include the 't' option to the Dial application, in the same way you need for the old-style '#' blind-transfer to function. gdh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hang up a SIP channel
Hi all i been trying to manually hangup a sip channel which is inactive. Peer User/ANR Call ID Seq (Tx/Rx) Format x2.xx.xx.x5 6574260125 6f06bf400e9 00102/2 UNKN (d) i tried soft hangup callerID and User but asterisk said is not a channel. and i tried sip show channels User and CallerID as well. Non tell me which is the channel to soft hangup help is appreciated Discover Yahoo! Get on-the-go sports scores, stock quotes, news & more. Check it out!___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: R: R: R: [Asterisk-Users] AT-320 + supervised transfer
On Wednesday 01 June 2005 11:01, Giordano Grandis wrote: Hi Gavin, I'm testing atxfer and it looks work fine, but i have a small problem: A call B B answer, dial atxfer extension and then the new peer (C) If C does not answer the phone, A and B got hangup and cannot speak again I set canreinvite to no. Can u help me ? Hm, this is the same response as I've posted to the list earlier today :) if you add the 'h' option to the Dial command, you will be able to hang up by pressing the * key on your phone ... If C's phone does not answer, pressing * should return you to talking to A. You can change the '*' button by changing the 'disconnect' line in features.conf. Be sure to STOP NOW and restart asterisk when changing features.conf - a 'reload' is /not/ enough. Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with codec negotiation
Title: Message Hi everyone I am having trouble with codec negotiation. I have Asterisk running at the office and a SIP phone at home. In my sip.conf, I have allow ordered as follows: alaw ulaw g729 and gsm On all my office extensions, I have no allow, or disallow entries. My Cisco gateway is setup to do alaw ulaw g729 and gsm My home phone does g729 alaw and ulaw. In sip.conf, I have disallow all and allow g729. In all my extensions and cisco gateway I have canreinvite set to yes and my dial commands don't have the t option, so all sip endpoints can talk directly to each other (rtp). If I call from home to the office, calls go through fine. SIP show channels, shows that the call is g729 as one would expect. If I get a call from Office to home, or from PSTN (via Cisco) to home, the phone rings, but as soon as I answer it hangs up. Asterisk says: May 29 05:45:49 WARNING[7514]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/390-8a3b(256) to SIP/192.168.44.23-08acccf0( -- SIP/390-8a3b is ringing -- SIP/390-8a3b answered SIP/192.168.44.23-08acccf0 May 29 05:45:55 WARNING[7514]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/192.168.44.23-08acccf0( to SIP/390-8a3b(256) May 29 05:45:55 WARNING[7514]: app_dial.c:1006 dial_exec: Had to drop call because I couldn't make SIP/192.168.44.23-08acccf0 compatible with SIP/390-8a3b == Spawn extension (default, 390, 1) exited non-zero on 'SIP/192.168.44.23-08acccf0'SNIP. One week later. I have now purchased two 9.729 licences as I suspected Asterisk was not allowing direct endpoint negotiation. Now my home phone answers, but it is receiving on 9.729 and sending on g.711a This leads to delays building up on the g711a side. I want to calls coming to my office phones to use alaw as the prefered codec in sip.conf, but I want calls to my home and remote sites to be g.729. Asterisk seems to ignore the codec negotiation phase and insists on running two different codecs in two directions. Most sip servers will always use the same codec in both directions based on the first agreed codec. As my home phone is set in sip.conf to only allow g.729, then it should do g.729 in both directions. I see this as a bug. Anyone know how to make it work properly? Thanks to the gurus who might no the answer to this one. Cheers Mark P.S. Why do the real experts not use the users web forum? Much easier to manage than a mailing list. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] pbx - fiber - network media converter - wifi - network media converter - fiber - pbx ???
Please forgive the (almost?) OT post. (and the fact that I need a clue-bat) We've got a situation at one of our sites where a construction crew is likely to dig up our conduit which houses some data fiber and one pair of fiber used to tie a Definity 3gsi at a small office building to the rest of the phone system (school district). We're using a pair of Aeronets to the data network stays up, but haven't decided how to keep the phone system up yet. I wonder if it is possible to bridge what I guess it a telco t1 via fiber over wireless using standard media converters like we use for data networks? We're able to dedicate a set of radios to this if needed. Anyone ever tried this or know the basics well enough to know that it (will|will not) work? Any thoughts on how a guy might use * to save the day without having to hack the Definity or get fiber in and out of a * box on each end? Yes, you can use wireless to accomplish this. However, the aeronet won't be able to accomplish this without something to convert the datastream into IP-based dataflows (eg, two asterisk boxes with iax between). There are wireless boxes that will operate at 70 megabits/sec and will accept T1 interfaces, but those typically are in the $15k - $20k range. If you can estimate the true number of simultanous calls expected across the facility, using an asterisk box at both ends (each with a T1 card interfacing to the respective phone equipment) might work across the aeronets. If you really had 24 simultanous conversations going on, the likelihood of the aeronets providing acceptable service will be very low. The exact number of simultanous conversations will be 100% dependent on the codec used between the asterisk boxes, the quality of the signal between the aeronets, and the stability (including jitter) of the end- to-end wireless link. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Ztdummy usage
On Tue, May 31, 2005 at 12:35:32PM +0100, Gentian Bajraktari wrote: Then try to 'modprobe zaptel' and then 'modprobe ztdummy' 'modprobe ztdummy' should load zaptel as well. If ytou happen to use debian, add the line 'ztdummy' (without quotes) to the file /etc/modules to modprobe it at system boot. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] UPS rating for SOHO asterisk box
I have many sites that have a 35amp Charger with 2 x 400ah 900CCA deep cycle batteries (10 year warranty), and 1000VA inverters. Those deep cycles batteries look quite appropriate... in which kind of store do you get them? In the US just about any store that sells batteries including Sears, InterState Batteries, most automotive parts stores, etc, etc. Just ask them. You see a lot of the deep cycle batteries used in fishing boats where they power electric trolling motors. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A newbie question - SIP to Trunk
Hello, Firstly sorry for covering old ground - I'm new to this. . . . I've read that you have to be careful when configuring SIP phone extensions so that an incoming call can't be connected to the trunk. Anyone have some info on how this can happen and how to stop it? Next, Can anyone tell me (in outline) how to set up a wifi SIP phone so that when I'm in the office I call in/out over Asterisk and the trunk and when I go home I can still be called from the office and still use the office Asterisk for trunk calls. Of course the office Asterisk is behind a NAT/firewall. Thanks in advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Launching an application from within Asterisk
Hello, I need to run an application that sets a few Asterisk variables, that will be used by AGI scrpits. Therefore, I believe that application should be run somehow from within Asterisk, on startup. The application needs to be always running, since it may need to update those variables. Is there any simple way to do this, like running an AGI script on startup or do I need to compile my application as an Asterisk application? Thanks in avance, Paulo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Ztdummy usage
[EMAIL PROTECTED] wrote: On Tue, May 31, 2005 at 12:35:32PM +0100, Gentian Bajraktari wrote: Then try to 'modprobe zaptel' and then 'modprobe ztdummy' 'modprobe ztdummy' should load zaptel as well. I've seen this faul, when only modprobe zaptel first would help. (Debian sarge) -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: R: R: R: R: [Asterisk-Users] AT-320 + supervised transfer
No...maybe i don't explain u well. After that B call C andC not answer (go in timeout), B hear first the beeperr and then, together A the busy tone. Now i can't re-take the call :| Thanks Giordano -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Gavin Hamill Inviato: mercoledì 1 giugno 2005 12.34 A: asterisk-users@lists.digium.com Oggetto: Re: R: R: R: R: [Asterisk-Users] AT-320 + supervised transfer On Wednesday 01 June 2005 11:01, Giordano Grandis wrote: Hi Gavin, I'm testing atxfer and it looks work fine, but i have a small problem: A call B B answer, dial atxfer extension and then the new peer (C) If C does not answer the phone, A and B got hangup and cannot speak again I set canreinvite to no. Can u help me ? Hm, this is the same response as I've posted to the list earlier today :) if you add the 'h' option to the Dial command, you will be able to hang up by pressing the * key on your phone ... If C's phone does not answer, pressing * should return you to talking to A. You can change the '*' button by changing the 'disconnect' line in features.conf. Be sure to STOP NOW and restart asterisk when changing features.conf - a 'reload' is /not/ enough. Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Sipura 2000 behind NAT issue, Vonage is working
On Tue, May 31, 2005 at 12:06:55PM +0200, David Hajek wrote: Hi, I'm trying to configure Sipura 2000 (behind NAT) which connects to Asterisk (public IP, no NAT) and having interesting results. When Sipura is behind Linux/NAT firewall it works great and no special NAT settings on Sipura are necessary. The issue I'm having is when Sipura is behind Linksys broadband NAT router. Sipura gets registered with Asterisk just fine, but I can't hear the other party (to be more precise I can hear first two secs then nothing). So it must be the incoming RTP is blocked on Linksys. Here I think STUN server enters the game and give some help? I have installed Vovida STUN server and point Sipura to use it. But no luck, I still can't hear the other party. I've ended up with having Linksys to forward all ports to my Sipura (DMZ host) which works. What is interesting is that when I'm using Vonage service (Cisco ATA) it works just fine without touching the Linksys. How come they can get through it? Any hints? Do you have the NAT Enable and NAT keepalive set to Yes on the Sipura? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] AT-320 + supervised transfer
On Wednesday 01 June 2005 12:43, Giordano Grandis wrote: No...maybe i don't explain u well. After that B call C andC not answer (go in timeout), B hear first the beeperr and then, together A the busy tone. Now i can't re-take the call :| I'm afraid I don't have any more suggestions to offer - anyone else? Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] AT-320 + supervised transfer
Ok, thanks for all. Just a thingh: how do u set DTMF on your phones ? Giordano -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Gavin Hamill Inviato: mercoledì 1 giugno 2005 13.51 A: asterisk-users@lists.digium.com Oggetto: Re: [Asterisk-Users] AT-320 + supervised transfer On Wednesday 01 June 2005 12:43, Giordano Grandis wrote: No...maybe i don't explain u well. After that B call C andC not answer (go in timeout), B hear first the beeperr and then, together A the busy tone. Now i can't re-take the call :| I'm afraid I don't have any more suggestions to offer - anyone else? Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MOH Jittery Voice
Hi All, I am having trouble with MOH. I have downloaded the latest CVS head and when I try to call from PSTN side and play MOH on a queue then the voice breaks. However if I play the same file using Playback() application and listen to it through PSTN side then there is no problem. CVan somebody tell me how can i use Playbak or background application to be used as MOH player I am waiting for any response. Khan. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] AT-320 + supervised transfer
On Wednesday 01 June 2005 13:04, Giordano Grandis wrote: Ok, thanks for all. Just a thingh: how do u set DTMF on your phones ? We have them set to RFC2833. I think I've noticed some cases where the remote party hears the tones, but it's not an issue that bothers me :) Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Sipura 2000 behind NAT issue, Vonage is working
I have installed Vovida STUN server and point Sipura to use it. But no luck, I still can't hear the other party. I've ended up with having Linksys to forward all ports to my Sipura (DMZ host) which works. What is interesting is that when I'm using Vonage service (Cisco ATA) it works just fine without touching the Linksys. How come they can get through it? Any hints? Do you have the NAT Enable and NAT keepalive set to Yes on the Sipura? Yes, I do. I have find out that Sipura works when I set it as DMZ host on the Linksys firewall. Why Vonage can work without any special settings? -David http://hajek.net/blog ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] IVR Load
Dear All,
I was trying to enable call forwarding, following the steps of the link
on voip.org regarding this issue it doesn't work and the phone I am trying
to implement on is still ringing. below is my conf in extensions.conf and
the CLI output during the process.
My configuration is :
exten => _*5X.,1,DBput(CF/${CALLERIDNUM}=${EXTEN:2})
exten => _*5X.,2,Hangup
exten => *5,1,DBdel(CF/${CALLERIDNUM})
exten => *5,2,Hangup
[macro-stdexten]
;
; Standard extension macro (with call forwarding):
; ${ARG1} - Extension(we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten => s,1,DBget(temp=CF/${ARG1})
exten => s,2,Goto(${temp}|1)
exten => s,102,Goto(s|3)
exten => s,3,Dial(${ARG2},120)
exten => s,103,Goto(s|50)
exten => s,4,Voicemail(u${ARG1})
exten => s,5,Hangup
exten => s,104,Voicemail(b${ARG1}) ; busy
exten => s,105,Hangup
the output on the CLI during this process was:
*CLI>
-- Executing DBdel("SIP/777-a77c", "CF/777") in
new stack
-- DBdel: family=CF, key=777
Urgent handler
-- Executing Hangup("SIP/777-a77c", "") in new stack
Urgent handler
-- Executing DBput("SIP/777-ad46", "CF/777=888")
in new stack
-- DBput: family=CF, key=777, value=888
Urgent handler
-- Executing Hangup("SIP/777-ad46", "") in new stack
Urgent handler
*CLI>
*CLI>
-- Executing Dial("SIP/999-8f50", "SIP/777|7|tr")
in new stack
-- Called 777
Urgent handler
Urgent handler
-- SIP/777-82e9 is ringing
Urgent handler
Any Idea what's wrong
--
Thx
MAG
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk x PROLIANT ML 150 G2 SATA
Fedora core 3 supports SATA on that model. listas iPfone wrote: Hi All, I´m tryingo to install asterisk in an PROLIANT ML 150 G2 SATA and can´t make it work because linux cant recognize the Hd (HP 160 mb). No drivers for Centos ...Red Hat... i´t´s drivig me crazy.. Someone have a tip? if i make change it to SCSI i think it will work but not sure about. Thanks Miklos ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] gnugk
HI, I would like to know how can I check if gateway is registered with gnugk? Thank you, Mitja ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] UPS rating for SOHO asterisk box
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Terry H. Gilsenan Sent: Wednesday, June 01, 2005 5:05 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] UPS rating for SOHO asterisk box I have many sites that have a 35amp Charger with 2 x 400ah 900CCA deep cycle batteries (10 year warranty), and 1000VA inverters. The combination makes for perfect power and about 2.5 days run time with my network kit whish consists of several Dlink wifi access points, 1 xbox (hacked into a router/firewall) and a vsat system. Total cost for the power kit AUD$1400 all up, and not a single second of downtime in over a year. [...] Yepyou can (somewhat) build your own UPS with peoperly rated equipment. As a matter of fact, most telco installations don't have monolithic UPS's (like you'll see in most larger datacentersyou know..the big box that says Liebert on it), they use racks of batteries with separate charging circuits. Most of the equipment runs directly off of the battery voltage, but you will find places with some inverters as well. Of course, the room is properly designed (spaced, non-combustible racks, fire detection and supression systems, etc.) and, in most jurisdictions they also have to carry one or more operational permits (current Internation Fire Code requires permitting for stationar lead-acid battery systems exceeding 50 gallons liquid capacity). On the flipside, I have seen a ups flare when the transformer overheated and melted the varnish, nasty! I've seen completely unmodified (although not properly maintained) UPSes as large as 5000 Va completely melt down to the point where they destroyed their own chassis, damaged the rack they were sitting in, and activated the clean-agent supression system in the rooms they were in. This was actually a big problem with one of my customersthey hadn't been maintaining their UPSesthe replace battery lights had been lit for months (they had all been purchased at about the same time). Within a span of about 3 months, 4 of them melted down similarly. A quick call to APC revealed that the batteries in these units were rated for about 12 monts less than they had actually been in service, and a simple battery replacement would have prevented the problem (the chassis was rated for something like 3 sets of batteries...whatever the lifespan of the batteries was3 years I believe). So, don't do stupid things with high voltage, like modifying equipment that wasn't meant to be modified, using undersized equipment, failing to properly vent batteries, or storing your contraption on or near combustibles. It's just NOT worth the risk. Take it from someone who's pulled his share of bodies (of both the live and dead varities) out of buildings. I've seen way too many fires started by electrical system or device modifications similar to those described in previous posts. And most people who do things like this just never consider the life safety risk involved until its way too late. I'll get off my soap-box now and get back on topic. Daryl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] gnugk
telnet your-gnugk-ip 7000 use AllRgistrations command or limply ? or ?? for PrintAllRegistrationsVerbose Of course you have to configure your gnugk to allow you to use telnet on port 7000 ... but i think you can use it by default --- Micko [EMAIL PROTECTED] wrote: HI, I would like to know how can I check if gateway is registered with gnugk? Thank you, Mitja ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Discover Yahoo! Find restaurants, movies, travel and more fun for the weekend. Check it out! http://discover.yahoo.com/weekend.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Sipura 2000 behind NAT issue, Vonage is working
I have installed Vovida STUN server and point Sipura to use it. But no luck, I still can't hear the other party. I've ended up with having Linksys to forward all ports to my Sipura (DMZ host) which works. What is interesting is that when I'm using Vonage service (Cisco ATA) it works just fine without touching the Linksys. How come they can get through it? Any hints? Do you have the NAT Enable and NAT keepalive set to Yes on the Sipura? Yes, I do. I have find out that Sipura works when I set it as DMZ host on the Linksys firewall. Why Vonage can work without any special settings? I fired up a spa3000 behind a linksys wireless firewall (befw11s4 v2) and it works just fine with nat=yes and canreinvite=no in sip.conf. Registration functions fine, no timeouts after days of operation, and 2-way audio functions correctly regardless of where a call originates. * sip debug _should_ provide the clues needed to resolve the issue. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: R: [Asterisk-Users] AT-320 + supervised transfer
This is what happen when i call a peer that not answer: -- Executing Dial(SIP/401-4de6, SIP/402|60|Thtr) in new stack -- Called 402 -- SIP/402-fa23 is ringing -- SIP/402-fa23 answered SIP/401-4de6 -- Attempting native bridge of SIP/401-4de6 and SIP/402-fa23 -- Started music on hold, class 'default', on SIP/401-4de6 -- Playing 'pbx-transfer' (language 'it') -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/406|60|Tthr) in new stack -- Called 406 -- SIP/406-aa46 is ringing Warning, flexibel rate not heavily tested! Jun 1 13:45:57 WARNING[25325]: res_features.c:858 builtin_atxfer: Unable to create channel Local/[EMAIL PROTECTED]/n do you have chan_local? -- Stopped music on hold on SIP/401-4de6 == Spawn extension (local, 406, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' -- Playing 'beeperr' (language 'it') == Spawn extension (local, 402, 1) exited non-zero on 'SIP/401-4de6' It could some extensions.conf problem ? Thanks -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Gavin Hamill Inviato: mercoledì 1 giugno 2005 14.20 A: asterisk-users@lists.digium.com Oggetto: Re: R: [Asterisk-Users] AT-320 + supervised transfer On Wednesday 01 June 2005 13:04, Giordano Grandis wrote: Ok, thanks for all. Just a thingh: how do u set DTMF on your phones ? We have them set to RFC2833. I think I've noticed some cases where the remote party hears the tones, but it's not an issue that bothers me :) Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Dynamic IAX Server
You just need to read up on IAX a little. IAX has no trouble with firewalling. As long as the client registers to the IAX server, the path will be defined and connectivity will occur. It may look like an odd port if you don't have a static port forward in place but it will work. If you really want the mechanics of how then google up IAX. There is nothing special in iax.conf or extensions.conf. Just use the example given for a logical dialplan and all should work fine. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of chawki hammoud Sent: Wednesday, June 01, 2005 12:49 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Dynamic IAX Server Hi: I read many documents and I posted my question several times here without luck. I hope someone can help now please. Here is an example to demonstarte my problem: Suppose you manage the FWD server, how do you define an IAX client behind nat so he can receive calls from FWD. NAT client would register with FWD to let it know how to locate it. I just don't see how FWD finds the nat client. How is that translated in terms of IAX.conf contexts and what FWD dial in extensions.conf file. Regards; Chawki __ Discover Yahoo! Use Yahoo! to plan a weekend, have fun online and more. Check it out! http://discover.yahoo.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MOH Jittery Voice
Are you using custom music files? If so, how did you transfer them to the box? If you transferred via FTP, you need to be sure you set the tranfer type to Binary before sending. Tranferring using ASCII has always hosed mp3 files for me on the * box. The net result being similar to your description. Are you using the MOH definition that has normal volume? Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, June 01, 2005 7:52 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] MOH Jittery Voice Hi All, I am having trouble with MOH. I have downloaded the latest CVS head and when I try to call from PSTN side and play MOH on a queue then the voice breaks. However if I play the same file using Playback() application and listen to it through PSTN side then there is no problem. CVan somebody tell me how can i use Playbak or background application to be used as MOH player I am waiting for any response. Khan. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: R: [Asterisk-Users] AT-320 + supervised transfer
On Wednesday 01 June 2005 14:15, Giordano Grandis wrote: This is what happen when i call a peer that not answer: Jun 1 13:45:57 WARNING[25325]: res_features.c:858 builtin_atxfer: Unable to create channel Local/[EMAIL PROTECTED]/n do you have chan_local? I don't like the look of this part at all. Please try to rm /usr/lib/asterisk/modules/* then 'make clean; make install' on a fresh checkout of CVS HEAD :) Also, there should be no need for the 'r' option to Dial since SIP already supports all the progress indication necessary. gdh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FW: TellMe pay-as-you-go? - UPDATE
As some of you know Ive been trying to facilitate an involvement with www.tellme.com speech recognition tools and Asterisk. See www.studio.tellme.com There have been a number of people who are already integrating the two and utilizing Tellme as an ASP to deliver speech recognition to their asterisk applications. However I do need to update the asterisk list that it isnt proceeding as fast I would have originally hoped. My original intention was to have Tellme set up a website where anyone with a credit card could log in and purchase blocks of time in advance. Unfortunately Tellme have decided that they are only interested in taking commercial customers at this time (though have indicated that from Jan they would be in a position to relook at this situation). Below is an email between myself and Bryan which gives you an idea on what we were looking to develop. Its a great opportunity for commercial high volume applications to deliver speech without the outlay (if you are one of these contact me for details on the trials) however I cant say Im not disappointed in their decision not to offer this as a prepaid service similar to how the asterisk community is being serviced by the sms ASPs. If anyone has some alternative suggestions Im open to hearing it Cheers, Dean From: Bryan A. Pendleton [mailto:[EMAIL PROTECTED] Sent: Wednesday, 25 May 2005 2:16 PM To: Dean Collins Subject: Re: TellMe pay-as-you-go? Yeah, I understand. It'd have to land into the community support world, just as asterisk has for small-time users. I'd imagine that folks like yourself, and open-source developers who just think it'd be neat to have certain home apps would do the support/feeding of the little guys. TellMe would just need to provide a utility service. Anyway, assuming that TellMe would be willing to SIP redirect (ie, I write my app to redirect a caller to TellMe for the voice rec, but when I complete the call somewhere, I pull TellMe out of the loop, so I'm not paying them for termination. If that's the case, then probably something like $0.10-$0.15/minute is reasonable. Perhaps $0.20-$0.30 if billing were in fractional minutes (ie, 6 sec increments like the industry does for long distance). I dunno, I'm just making this up probably a monthly maintainence would also be ok. I'd like to see that be $5/mo, unless it's just a minimum charge, rather than a flat fee. Or, an entirely other direction is to just charge $5/mo for some big pseudo-unmetered-until-it's-abused usage. That would cover most home users, who'd probably only generate a couple of dozen minutes a month, but also make the billing system (and, correlated, number of disputes) a lot simpler to deal with. I'm a techie, though, not a business man, so I'm just guessing at what I and others doing similar things would find reasonable. I have a pay-as-you-go toll free number, a couple of free DIDs in different area codes, and a $0.013/minute long distance termination company that bills in 6s increments. If folks who tinker on the horizon I do are going to use these kinds of services, they need to be cheap. On 5/24/05, Dean Collins [EMAIL PROTECTED] wrote: Hey I totally agree with you about scaling down. One of my biggest arguments with a previous client was around this exact idea. Unfortunately I think the average personal user would want programming support for when things go wrong, what would you do in that situation? I don't think you could charge them $100 for the service call on an app they only pay $10 a month for? I agree with what you are saying, I've flicked the Tellme team an email about your idea, may they might take your app on as a trial. If they were to take it on it would be on the understanding that it may be cancelled at any time. That it would not involve support from their end. That it would a fully paid in advance basis in the initial trial. How much a minute per cycle do you think this is worth to you? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 +61-2-8307-3503 (Sydney in-dial) From: Bryan A. Pendleton [mailto:[EMAIL PROTECTED]] Sent: Tuesday, 24 May 2005 6:13 PM To: Dean Collins Subject: Re: TellMe pay-as-you-go? Well, for changes to the wiki, you might mention that the service is not likely to be for personal use. It's not totally clear from the entry. Also, I disagree. There's absolutely no reason that VXML hosting couldn't be scaled to providing individual users low-volume service. It's merely a matter of having a sufficient pool of resources, and removing all of the management cost-per-user, or making it very very small and mostly the burden of the user. In principle, this is exactly what VoIP termination companies like voipjet.com and voicepulse.com do, or, conversely, super-large-scale web hosting companies, only the resource being dolled out is more expensive. You've gotta' provide CPU cycles on
[Asterisk-Users] Asterisk Google API applications - $4500 bounties available
In conjunction with my last post on Tellme I want to write another suggestion for an application I had. I dont know if you guys have come across Google Gas http://www.ahding.com/cheapgas But basically it is an application that this guy has developed using the Google API to search an online database on gas prices in your area. One of my strong beliefs about how Asterisk is going to leave the Commercial IP-PBX vendors behind is by leveraging the open source community to write voice driven applications for Asterisk. The weather app written for [EMAIL PROTECTED] is great example. (http://sourceforge.net/forum/message.php?msg_id=3004652 the WAF on this was worth setting up asterisk alone, she checks this every morning for NY weather). I was also hoping that the www.tellme.com and www.studio.tellme.com tools would also stimulate this area. People should also check out www.angel.com for other ideas on best of breed speech applications. The suggestion I would like to make is that someone use the Google api to write code for a directions application. You could use Tellme to deliver the current address and the destination address into the Google API and then use text to speech to read back the directions. With enough finessing this could compete with any of the current commercial direction solutions that are out there and because its asterisk your cost base could be extremely minimal. Hell you might even get paid for it http://code.google.com/summerofcode.html Just a suggestion, any thoughts? Are there any other speech driven apps being used today? Cheers, Dean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SIP Soft Video phone for Asterisk usage
--- Ronald Wiplinger [EMAIL PROTECTED] wrote: Nardis Dome wrote: in your sip.conf: [general] videosupport=yes ; That helped a lot in your eyeBeam settings- try to enable all the h.263 codec. hope it helps.. However, I am still not there. I have installed eyeBeam on 612 and 617. While 612 gets the video of 617, 617 sees 612 as a picture, like a big spreadsheet with dots in each cell. Absolutely no picture to recognize. 612 sees itself clear. What could be still wrong? I have enabled all codecs on both Xten. In asterisk it has the same settings (realtime shows the same record - of course user and password is different) Sometimes i have the same probleme. I have to restart eyeBeam or reboot my PC... __ Discover Yahoo! Use Yahoo! to plan a weekend, have fun online and more. Check it out! http://discover.yahoo.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] pbx - fiber - network media converter - wifi - network media converter - fiber - pbx ???
Thanks for your reply. I wouldn't expect more than half a dozen concurrent calls. Also, we can do the bridge with proxims if needed (not the model with a telco t1 broken out). The reason I ask about the media converters is to save the trouble having to interface an * box to each Definity. Rich Adamson wrote: Please forgive the (almost?) OT post. (and the fact that I need a clue-bat) We've got a situation at one of our sites where a construction crew is likely to dig up our conduit which houses some data fiber and one pair of fiber used to tie a Definity 3gsi at a small office building to the rest of the phone system (school district). We're using a pair of Aeronets to the data network stays up, but haven't decided how to keep the phone system up yet. I wonder if it is possible to bridge what I guess it a telco t1 via fiber over wireless using standard media converters like we use for data networks? We're able to dedicate a set of radios to this if needed. Anyone ever tried this or know the basics well enough to know that it (will|will not) work? Any thoughts on how a guy might use * to save the day without having to hack the Definity or get fiber in and out of a * box on each end? Yes, you can use wireless to accomplish this. However, the aeronet won't be able to accomplish this without something to convert the datastream into IP-based dataflows (eg, two asterisk boxes with iax between). There are wireless boxes that will operate at 70 megabits/sec and will accept T1 interfaces, but those typically are in the $15k - $20k range. If you can estimate the true number of simultanous calls expected across the facility, using an asterisk box at both ends (each with a T1 card interfacing to the respective phone equipment) might work across the aeronets. If you really had 24 simultanous conversations going on, the likelihood of the aeronets providing acceptable service will be very low. The exact number of simultanous conversations will be 100% dependent on the codec used between the asterisk boxes, the quality of the signal between the aeronets, and the stability (including jitter) of the end- to-end wireless link. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users [Scanned for viruses] [Scanned for viruses] -- Ben Dugdale [EMAIL PROTECTED] Network Administrator Apache County Schools Business Consortium www.acsbc.net Apache County Arizona www.co.apache.az.us Work (928) 337-7507 Cell (928) 245-2754 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] pbx - fiber - network media converter - wifi - network media converter - fiber - pbx ???
Thanks for the reply. I'll get up there today and get more details on the Definity. Alexander Lopez wrote: We use Wireless b/w two office in Miami We are using the Proxim stuff and it is solid. Two Asterisk servers doing Iax b/w them should (will) work fine. What is the interface into the 3gsi?? Do you have a card part number to post, that would help in determining what you need to do. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Dugdale Sent: Tuesday, May 31, 2005 10:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] pbx - fiber - network media converter - wifi - network media converter - fiber - pbx ??? Please forgive the (almost?) OT post. (and the fact that I need a clue-bat) We've got a situation at one of our sites where a construction crew is likely to dig up our conduit which houses some data fiber and one pair of fiber used to tie a Definity 3gsi at a small office building to the rest of the phone system (school district). We're using a pair of Aeronets to the data network stays up, but haven't decided how to keep the phone system up yet. I wonder if it is possible to bridge what I guess it a telco t1 via fiber over wireless using standard media converters like we use for data networks? We're able to dedicate a set of radios to this if needed. Anyone ever tried this or know the basics well enough to know that it (will|will not) work? Any thoughts on how a guy might use * to save the day without having to hack the Definity or get fiber in and out of a * box on each end? Thanks!! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Setting up a TDM
Greetings to all! I have been writing a great new voice messaging application on Asterisk, and am getting to the point of moving it to my own hosting environment. I have been in discussions with service providers who can provide me with a TDM voice T1 line (analog?), but cannot provide a SIP-terminated line. My question is: what do I need to hook that TDM T1 line into the server where I'm running Asterisk? Is it simply a Digium card like the Wildcard TE410P? Many thanks! Hugo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 installation with Asterisk
Hi all, i am looking for informations about large installation with Asterisk (~3000 users). Has anybody experience with such a setup. Any comments, suggestions or problems would be appreciated. thx in advance... __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pass-through
In an order to save money, I would like to use a PRI that we have going to one of our dial-up modem banks (We are an ISP.) During business hours these channels are idle and during our peak internet times, we are closed. Sounds too good to be true, but I thought I would throw it out there. These are modem calls that if they would call our modem bank number, they would be bridged to the outbound zap channels??? And of course, if they dial our business number we would send them to the appropriate sip channels. I didnt know if this could be done with two T1 cards and asterisk Here is a primitive sketch. If anyone has information, please share. Thank You Adam Vocks CTI ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [q] About chan_misdn, latest mISDNuser and asterisk cvs
Hi. Where I can get chan_misdn that compiles with latest asterisk and mISDNuser cvs ? Or may be chan_misdn is already present in some asterisk cvs branch ? TIA Rus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Pass-through
This may or may not work due to timings slips that you may experiance with the Digium Cards. Your are correct in assuming this scenaro. I did the same (pre-asterisk) with an Adtran Atlas. It is rock solid and works great. What modem access bank are you using, there has been some talk about using the PM3 as an IAX gateway. (highly vaporware at this point), The Acend unit support SIP, the Cisco suport..., etc. etc, From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam VocksSent: Wednesday, June 01, 2005 11:07 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Pass-through In an order to save money, I would like to use a PRI that we have going to one of our dial-up modem banks (We are an ISP.) During business hours these channels are idle and during our peak internet times, we are closed. Sounds too good to be true, but I thought I would throw it out there. These are modem calls that if they would call our modem bank number, they would be bridged to the outbound zap channels??? And of course, if they dial our business number we would send them to the appropriate sip channels. I didnt know if this could be done with two T1 cards and asterisk Here is a primitive sketch. If anyone has information, please share. Thank You Adam Vocks CTI ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Pass-through
Adam Vocks wrote: In an order to save money, I would like to use a PRI that we have going to one of our dial-up modem banks (We are an ISP.) During business hours these channels are idle and during our peak internet times, we are closed. Sounds too good to be true, but I thought I would throw it out there. These are modem calls that if they would call our modem bank number, they would be bridged to the outbound zap channels??? And of course, if they dial our business number we would send them to the appropriate sip channels. I didnt know if this could be done with two T1 cards and asterisk Here is a primitive sketch. If anyone has information, please share. Thank You Adam Vocks CTI I've done the exact same thing. We had a 23-channel PRI that a client was using for voice, but had a small IVR for their banking application that had direct analog lines pointed to it. I ordered an Adtran Total Access 750 and an additional T1 (T100P) card. The TA750 had 24 analog lines, with one T1 interface. The asterisk server had 2 T100Ps one card was for the PRI, the second was a cross-over to the Adtran 750. Works great, don't see why it wouldn't work for you in the same method you are talking about for a modem pool. Drawing: inline: astModem.jpg___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Pass-through
Were still using Lucent PM3s Adam From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Wednesday, June 01, 2005 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Pass-through This may or may not work due to timings slips that you may experiance with the Digium Cards. Your are correct in assuming this scenaro. I did the same (pre-asterisk) with an Adtran Atlas. It is rock solid and works great. What modem access bank are you using, there has been some talk about using the PM3 as an IAX gateway. (highly vaporware at this point), The Acend unit support SIP, the Cisco suport..., etc. etc, From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Vocks Sent: Wednesday, June 01, 2005 11:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Pass-through In an order to save money, I would like to use a PRI that we have going to one of our dial-up modem banks (We are an ISP.) During business hours these channels are idle and during our peak internet times, we are closed. Sounds too good to be true, but I thought I would throw it out there. These are modem calls that if they would call our modem bank number, they would be bridged to the outbound zap channels??? And of course, if they dial our business number we would send them to the appropriate sip channels. I didnt know if this could be done with two T1 cards and asterisk Here is a primitive sketch. If anyone has information, please share. Thank You Adam Vocks CTI ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Pass-through
On Wed, 1 Jun 2005, Dustin Wildes wrote: Adam Vocks wrote: In an order to save money, I would like to use a PRI that we have going to one of our dial-up modem banks (We are an ISP.) During business hours these channels are idle and during our peak internet times, we are closed. Sounds too good to be true, but I thought I would throw it out there. These are modem calls that if they would call our modem bank number, they would be bridged to the outbound zap channels??? And of course, if they dial our business number we would send them to the appropriate sip channels. I didn?t know if this could be done with two T1 cards and asterisk? I've done the exact same thing. We had a 23-channel PRI that a client was using for voice, but had a small IVR for their banking application that had direct analog lines pointed to it. I ordered an Adtran Total Access 750 and an additional T1 (T100P) card. The TA750 had 24 analog lines, with one T1 interface. The asterisk server had 2 T100Ps one card was for the PRI, the second was a cross-over to the Adtran 750. Works great, don't see why it wouldn't work for you in the same method you are talking about for a modem pool. There is a better device if you have a PRI, an Adtran Atlas 550 is basically a full phone switch. You can put an entire dialplan on it to do router based on DID/DNIS. It will also do channelized T1 to PRI conversion each way. Very slick boxes, I'm about to set one up for another asterisk user to split 1 PRI to 12 pots lines for an older switch, 1 PRI for Asterisk, 1 channelized T1 for a modem bank, and some FXO ports for an older Brooktrout card. If anyone wants more info on them let me know. -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: R: R: [Asterisk-Users] AT-320 + supervised transfer
I did it...but with no good results. Could i see a example of peer in extensions.conf ? I'm trying everythinghs but i always have differenta results :| Thanks giordano -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Gavin Hamill Inviato: mercoledì 1 giugno 2005 15.31 A: asterisk-users@lists.digium.com Oggetto: Re: R: R: [Asterisk-Users] AT-320 + supervised transfer On Wednesday 01 June 2005 14:15, Giordano Grandis wrote: This is what happen when i call a peer that not answer: Jun 1 13:45:57 WARNING[25325]: res_features.c:858 builtin_atxfer: Unable to create channel Local/[EMAIL PROTECTED]/n do you have chan_local? I don't like the look of this part at all. Please try to rm /usr/lib/asterisk/modules/* then 'make clean; make install' on a fresh checkout of CVS HEAD :) Also, there should be no need for the 'r' option to Dial since SIP already supports all the progress indication necessary. gdh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Segmentation Fautl / Core Dump with G.729
Hello, Has anyone experienced a segmentation fault in asterisk for using G729 against an AS5300 in SIP ? I'm having this problem. Also, any other codec I use gives me incompatible media (can be read in SIP DEBUG messages). AS5300 configured for multiple codecs, so is Asterisk. Tried G711u/A G723 and G.729. Any clues ? Regards, Jorge A. Info: Asterisk ver 1.0.7 stable Using AMPortal 1.0.0.8 SIP.CONF --- ; Note: If your SIP devices are behind a NAT and your Asterisk ; server isn't, try adding nat=1 to each peer definition to ; solve translation problems. [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=alaw allow=g729 allow=g723 context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown language=es #include sip_nat.conf #include sip_custom.conf #include sip_additional.conf --- SIP_ADDITIONAL.CONF --- [as5300] type=peer qualify=yes host=xxx.xxx.xxx.xxx (AS5300 box) --- AS5300 relevant Config --- ... ! voice class codec 1010 codec preference 1 g711alaw codec preference 2 g711ulaw codec preference 3 g723ar63 codec preference 4 g723r63 ! ... ! dial-peer voice 1010 voip destination-pattern 85.. progress_ind setup enable 3 progress_ind progress enable 8 voice-class codec 1010 session protocol sipv2 session target ipv4:xxx.xxx.xxx.xxx (ASterisk Box) dtmf-relay cisco-rtp rtp-nte h245-signal h245-alphanumeric ! --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Pass-through
Would something as simple as this work? [InFromZap1] ;Context for incoming telco calls exten = 1234567890, 1, Dial(Zap/g2) ;g2 would be the second digium card connected to our Lucent PM3 with a crossover cable. Thanks Adam From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Vocks Sent: Wednesday, June 01, 2005 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Pass-through Were still using Lucent PM3s Adam From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Wednesday, June 01, 2005 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Pass-through This may or may not work due to timings slips that you may experiance with the Digium Cards. Your are correct in assuming this scenaro. I did the same (pre-asterisk) with an Adtran Atlas. It is rock solid and works great. What modem access bank are you using, there has been some talk about using the PM3 as an IAX gateway. (highly vaporware at this point), The Acend unit support SIP, the Cisco suport..., etc. etc, From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Vocks Sent: Wednesday, June 01, 2005 11:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Pass-through In an order to save money, I would like to use a PRI that we have going to one of our dial-up modem banks (We are an ISP.) During business hours these channels are idle and during our peak internet times, we are closed. Sounds too good to be true, but I thought I would throw it out there. These are modem calls that if they would call our modem bank number, they would be bridged to the outbound zap channels??? And of course, if they dial our business number we would send them to the appropriate sip channels. I didnt know if this could be done with two T1 cards and asterisk Here is a primitive sketch. If anyone has information, please share. Thank You Adam Vocks CTI ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ARESKICC - Another issue
Hi all, After finally making the web interface for AreskiCC work I am now running into new issues. 1 In Asterisk the manager doesnt seem to connect 2 When I try to create the file additional_areskicc_sip.conf it says Could not open buddy file /etc/asterisk/additional_areskicc_sip.conf Any clues? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Last of the servers forsale cheap
Ok guys, due to someone recently backing out I have a couple more servers left. These are tested, all freshly installed freebsd, double boxed and ready to ship. I need to get these shipped out by tommorow before I got out of town, so I need to k now today if anyone wants them. Make an offer on them if interested, I just want to get rid of them.. 1u Dual P3 800mhz 512megs memory, 2 x 9gb scsi U160 drives. Dual ethernet, Supermicro server, v ery nice. CDROM, FLoppy, etc $400 1u Dual P3 933mhz 1gb memory, 2 x 40gb IDE drives Dual Ethernet $400 1u 1.2ghz AMD servers, 512megs memory, 2 x 60gig hard drive $350 2u p4 2ghz 60gig drive, 512megs memory, CDROM $400 Make me an offer if these prices seem too high, would like to get rid of them before tommorow.. let me know.. Preston Garrison direct: 877-748-4142 fax: 310-774-3901 cell: 623-748-4140 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] rxfax problems - cont.
Well, my faxes passes through asterisk successfully, however I still have some problems about fax reception by rxfax. The softfax answers, and negotiates transmission, however then as some stage of communiation something is wrong. But I have nothing more but this log: Jun 2 00:10:21 DEBUG[16900]: chan_zap.c:4242 zt_read: DTMF digit: * on Zap/10-1 Jun 2 00:10:22 DEBUG[16900]: chan_zap.c:4242 zt_read: DTMF digit: 1 on Zap/10-1 Jun 2 00:10:22 DEBUG[16900]: chan_zap.c:4242 zt_read: DTMF digit: 0 on Zap/10-1 Jun 2 00:10:22 DEBUG[16900]: chan_zap.c:4242 zt_read: DTMF digit: 7 on Zap/10-1 Jun 2 00:10:23 DEBUG[16900]: chan_zap.c:4242 zt_read: DTMF digit: 8 on Zap/10-1 Jun 2 00:10:23 DEBUG[16900]: chan_zap.c:1384 zt_enable_ec: No echocancellation requested -- Executing SetVar(Zap/10-1, CALLEDFAX=*1078) in new stack -- Executing Answer(Zap/10-1, ) in new stack Jun 2 00:10:23 DEBUG[16900]: chan_zap.c:2484 zt_answer: Took Zap/10-1 off hook -- Executing Goto(Zap/10-1, fax|*1078|1) in new stack -- Goto (fax,*1078,1) -- Executing Macro(Zap/10-1, faxreceive) in new stack -- Executing SetVar(Zap/10-1, FAXFILE=/var/spool/asterisk/fax/*1078/asterisk-419-1117663820.32) in new stack -- Executing RxFAX(Zap/10-1, /var/spool/asterisk/fax/*1078/asterisk-419-1117663820.32.tif) in new stack Jun 2 00:10:37 DEBUG[16900]: app_rxfax.c:80 phase_e_handler: == Jun 2 00:10:37 DEBUG[16900]: app_rxfax.c:81 phase_e_handler: Fax successfully received. Jun 2 00:10:37 DEBUG[16900]: app_rxfax.c:82 phase_e_handler: Remote station id: Jun 2 00:10:37 DEBUG[16900]: app_rxfax.c:83 phase_e_handler: Local station id: Jun 2 00:10:37 DEBUG[16900]: app_rxfax.c:84 phase_e_handler: Pages transferred: 0 Jun 2 00:10:37 DEBUG[16900]: app_rxfax.c:85 phase_e_handler: Image resolution: 0 x 0 Jun 2 00:10:37 DEBUG[16900]: app_rxfax.c:86 phase_e_handler: Transfer Rate: 9600 Jun 2 00:10:37 DEBUG[16900]: app_rxfax.c:87 phase_e_handler: == Jun 2 00:10:41 DEBUG[16900]: chan_zap.c:3974 __zt_exception: Exception on 28, channel 10 Jun 2 00:10:41 DEBUG[16900]: chan_zap.c:3286 zt_handle_event: Got event On hook(1) on channel 10 (index 0) -- Executing System(Zap/10-1, /var/lib/asterisk/scripts/mailfax *1076 *1078 /var/spool/asterisk/fax/*1078/asterisk-419-1117663820.32 ) in new stack Jun 2 00:10:41 WARNING[16900]: app_system.c:70 system_exec_helper: Unable to execute '/var/lib/asterisk/scripts/mailfax *1076 *1078 /var/spool/asterisk/fax/*1078/asterisk-419-1117663820.32 ' Jun 2 00:10:41 WARNING[16900]: app_system.c:70 system_exec_helper: Unable to execute '/var/lib/asterisk/scripts/mailfax *1076 *1078 /var/spool/asterisk/fax/*1078/asterisk-419-1117663820.32 ' == Spawn extension (fax, h, 1) exited non-zero on 'Zap/10-1' Jun 2 00:10:41 DEBUG[445]: rate_engine.c:708 poster_worker: Rating Engine poster thread processing Jun 2 00:10:41 DEBUG[445]: rate_engine.c:770 poster_worker: Attempting to write queue entry to database Jun 2 00:10:41 DEBUG[16900]: chan_zap.c:2164 zt_hangup: Hangup: channel: 10 index = 0, normal = 28, callwait = -1, thirdcall = -1 Jun 2 00:10:41 DEBUG[16900]: chan_zap.c:2577 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/10-1 Jun 2 00:10:41 DEBUG[16900]: chan_zap.c:1352 update_conf: Updated conferencing on 10, with 0 conference users -- Hungup 'Zap/10-1' Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k libspandsp-dev 0.0.2pre17-1 libspandsp00.0.2pre17-1 txfax/rxfax from spandsp-0.0.2pre18 Physical test fax is Panasonic KX-FT908PD connected to TDM400P FXS port I also tried incomming fax through BRI interface - the same result, error code 41 (on Panasonic) Maybe someone meet this problem ? Regards Marcin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] VoiPSupply Dot Com
On May 31, 2005, at 4:30 PM, Karl J. Vesterling wrote: Garrett, evidently there is some verbage to that effect on the site. But just to let you know, no other business that we've done business with requires anything like that. Not a one. Also worthy of note is that the purchase was not a credit card order, so I'm rather surprised your terms regarding credit cards would apply. In retrospect, I guess I should have spent the 16 hours browsing your site looking for the fine print instead of waiting for a prompt shipment. But, alas... We found someone that knows how to do business with businesses. Since you're compelled to send us evidence of your other business dealings, why don't you send the list some pictures of yourself with some completely unimportant politicians to further validate your sense of self-righteousness? Listen: you obviously have no understanding of merchant accounts nor business risk management in general, so there's no amount of explaining a seller's right to uphold any and all terms in efforts of mitigating said risk. Go back to your workbench and sniff some solder fumes. /rg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] astapi memory errors?
Im using outlook 2003 on windows xp. [EMAIL PROTECTED] v 0.8 Is anyone else having issues with Astapi? About 50% of the time after I make a call and then terminate it I have a memory 0X093 error. Does anyone know what this is? Cheers, Dean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF not working
Im trying to configure voicemail, but asterisk doesnt respond to dtmf codes. I uses kphone with g711u codec (I've tryed the others one) and in sip.conf I configure dtmfmode=rfc2833 (I've tryied inband and info). Asterisk seems not to see the tones. Could somebody help me? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] HELP Cisco - can't find P0S3-07-4-00.sbn
What I need to do? Rename/Copy P003-07-4-00.bin to P0S3-07-4-00.sbn Regards, -- Ing CIP Alejandro Celi Mariátegui [EMAIL PROTECTED] El mar, 31-05-2005 a las 23:15, Andrew Latham escribió: sbn is a signed bin file P0S-xx-x-xx.sbn would be the format for the SIP image after version 5 P0S-xx-x-xx.bin would be the format for the SIP image before version 5 On 5/31/05, Ing CIP Alejandro Celi Mariátegui [EMAIL PROTECTED] wrote: I have a problem, I'm working with firmware SIP 6.3 installed on my Cisco phone and works fine, and I have the 7.4 firmware version to upgrade: [EMAIL PROTECTED]/home/alex/central/P0S3-07-4-00 ls -l total 2.3M -rw-r--r--1 root root 126K mar 10 15:33 P003-07-4-00.bin -rw-r--r--1 root root 578K mar 10 15:44 P0S3-07-4-00.bin -rw-r--r--1 root root 461 mar 10 16:01 P0S3-07-4-00.loads -rw-r--r--1 root root 579K mar 10 15:45 P0S3-07-4-00.sb2 -rw-r--r--1 root root 127K mar 10 15:33 P003-07-4-00.sbn -rw-r--r--1 root root 15 mar 10 15:33 OS79XX.TXT -rw-r--r--1 root root 895K abr 13 23:30 P0S3-07-4-00.zip When I try to upgrade to the 7.4 firmware I get this log: uploading OS79XX.TXT uploading P0S3-07-4-00.bin uploading P0S3-07-4-00.loads uploading P0S3-07-4-00.sb2 can't find P0S3-07-4-00.sbn - Aborted My phone is asking for a P0S3-07-4-00.sbn file, and can't find it in the Cisco distro. Perhaps a Cisco bug? Any idea? Regards, -- Ing CIP Alejandro Celi Mariátegui [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] list down?
List doesnt seem to be posting out still active here http://lists.digium.com/pipermail/asterisk-users/2005-June/date.html but not being received by email (time warner is the isp but other emails coming in every few minutes as per normal). Cheers, Dean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Astcc does not work - no repeat metering
I have installed xorcom and [EMAIL PROTECTED] on 2 different pc's, with astcc. It only registers the once of connection billing, and never again. I have tried everything. Am I doing something wrong? I will appreciate any help! -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 267.3.3 - Release Date: 31/05/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] HELP Cisco - can't find P0S3-07-4-00.sbn
OS79XX.TXT should contain: P003-07-4-00 _ Mobilcom http://www.mobilcom.net - Original Message - From: Ing CIP Alejandro Celi Mariátegui [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 31, 2005 11:59 PM Subject: [Asterisk-Users] HELP Cisco - can't find P0S3-07-4-00.sbn I have a problem, I'm working with firmware SIP 6.3 installed on my Cisco phone and works fine, and I have the 7.4 firmware version to upgrade: [EMAIL PROTECTED]/home/alex/central/P0S3-07-4-00 ls -l total 2.3M -rw-r--r--1 root root 126K mar 10 15:33 P003-07-4-00.bin -rw-r--r--1 root root 578K mar 10 15:44 P0S3-07-4-00.bin -rw-r--r--1 root root 461 mar 10 16:01 P0S3-07-4-00.loads -rw-r--r--1 root root 579K mar 10 15:45 P0S3-07-4-00.sb2 -rw-r--r--1 root root 127K mar 10 15:33 P003-07-4-00.sbn -rw-r--r--1 root root 15 mar 10 15:33 OS79XX.TXT -rw-r--r--1 root root 895K abr 13 23:30 P0S3-07-4-00.zip When I try to upgrade to the 7.4 firmware I get this log: uploading OS79XX.TXT uploading P0S3-07-4-00.bin uploading P0S3-07-4-00.loads uploading P0S3-07-4-00.sb2 can't find P0S3-07-4-00.sbn - Aborted My phone is asking for a P0S3-07-4-00.sbn file, and can't find it in the Cisco distro. Perhaps a Cisco bug? Any idea? Regards, -- Ing CIP Alejandro Celi Mariátegui [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] pbx - fiber - network media converter - wifi - network media converter - fiber - pbx ???
Check this out: http://www.engagecom.com/Products/iptube_T1.htm On 6/1/05, Rich Adamson [EMAIL PROTECTED] wrote: Please forgive the (almost?) OT post. (and the fact that I need a clue-bat) We've got a situation at one of our sites where a construction crew is likely to dig up our conduit which houses some data fiber and one pair of fiber used to tie a Definity 3gsi at a small office building to the rest of the phone system (school district). We're using a pair of Aeronets to the data network stays up, but haven't decided how to keep the phone system up yet. I wonder if it is possible to bridge what I guess it a telco t1 via fiber over wireless using standard media converters like we use for data networks? We're able to dedicate a set of radios to this if needed. Anyone ever tried this or know the basics well enough to know that it (will|will not) work? Any thoughts on how a guy might use * to save the day without having to hack the Definity or get fiber in and out of a * box on each end? Yes, you can use wireless to accomplish this. However, the aeronet won't be able to accomplish this without something to convert the datastream into IP-based dataflows (eg, two asterisk boxes with iax between). There are wireless boxes that will operate at 70 megabits/sec and will accept T1 interfaces, but those typically are in the $15k - $20k range. If you can estimate the true number of simultanous calls expected across the facility, using an asterisk box at both ends (each with a T1 card interfacing to the respective phone equipment) might work across the aeronets. If you really had 24 simultanous conversations going on, the likelihood of the aeronets providing acceptable service will be very low. The exact number of simultanous conversations will be 100% dependent on the codec used between the asterisk boxes, the quality of the signal between the aeronets, and the stability (including jitter) of the end- to-end wireless link. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Suppress Missed Calls 7960 SIP
On May 31, 2005, at 8:05 PM, Andy Hamilton wrote: On 5/31/05, Robert Goodyear [EMAIL PROTECTED] wrote: Does anyone know how to suppress the Missed Calls indication -- perhaps on a per-line basis -- on the 7960 running SIP? Reason: I've configured a group of extensions to ring for inbound calls and it seems pointless to accrue missed calls on those line presentations. /rg Rob: Not sure how to (though I agree it would be handy). If anything, it would be a Cisco thing. Have you checked their website to see if the have any tips? -Andy Yeah, no such luck. I'm guessing it would require a Firmware hack, which is CERTAINLY out of my realm. Which begs the question: I wonder if/when anyone will attempt to write FW for IP phones in the same vein as the openWRT / Sveasoft crowd. /rg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Pass-through
That should work but you need to have the asterisk box
setup to do pri-net on the connection to the PM3. I would add the did dialed so
that the PM3 knows about it for radius accounting..
exten = 1234567890,
1, Dial(Zap/g2/${EXTEN})
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
VocksSent: Wednesday, June 01, 2005 12:02 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: RE:
[Asterisk-Users] Pass-through
Would something as
simple as this work?
[InFromZap1]
;Context for incoming telco calls
exten = 1234567890,
1, Dial(Zap/g2) ;g2 would be the second digium card
connected to our Lucent PM3 with a crossover cable.
Thanks
Adam
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam VocksSent: Wednesday, June 01, 2005 10:24
AMTo: Asterisk Users Mailing
List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users]
Pass-through
Were still using
Lucent PM3s
Adam
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander LopezSent: Wednesday, June 01, 2005 10:24
AMTo: Asterisk Users Mailing
List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users]
Pass-through
This may or may not
work due to timings slips that you may experiance with the Digium Cards.
Your are correct in assuming this scenaro.
I did the same
(pre-asterisk) with an Adtran Atlas. It is rock solid and works
great. What modem access bank are you using, there has been some talk
about using the PM3 as an IAX gateway. (highly vaporware at this point), The
Acend unit support SIP, the Cisco suport..., etc.
etc,
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam VocksSent: Wednesday, June 01, 2005 11:07
AMTo: Asterisk Users Mailing
List - Non-Commercial DiscussionSubject: [Asterisk-Users]
Pass-through
In an order to save money, I would
like to use a PRI that we have going to one of our dial-up modem banks (We are
an ISP.) During business hours these channels are idle and during our peak
internet times, we are closed. Sounds too good to be true, but I thought I
would throw it out there. These are modem calls that if they would call
our modem bank number, they would be bridged to the outbound zap
channels??? And of course, if they dial our business number we would send
them to the appropriate sip channels. I didnt know if this could be done
with two T1 cards and asterisk
Here is a primitive
sketch.
If anyone has information, please
share.
Thank
You
Adam
Vocks
CTI
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
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] Unreliable DTMF detection with DISA on incomingZap channel on bristuffed * and GSM gateway
Hello, I'm getting unusable DTMF detection with DISA on incoming ZAP channel (bristuffed *) on quadbri from GSM gateway. DTMF detection works ok in normal ISDN incoming line. I'm having similar problems with a gsm gateway connected to x100p. The DTMF for 1, 4 and 7 are detected fine, but 2, 5 and 8 gets detected dupe and 3 6 and 9 aren't detected. Anyone has any idea? i've already tried relaxdtmf=yes with no success. Regards, Marcelo. --- Este mensaje está libre de virus - www.v2r.com.ar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] list down?
No problems here. 27 min behind according to your post time. Dean Collins wrote: List doesnt seem to be posting out still active here http://lists.digium.com/pipermail/asterisk-users/2005-June/date.html but not being received by email (time warner is the isp but other emails coming in every few minutes as per normal). Cheers, Dean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] HELP Cisco - can't find P0S3-07-4-00.sbn
On Jun 1, 2005, at 9:38 AM, Ing CIP Alejandro Celi Mariátegui wrote: What I need to do? Rename/Copy P003-07-4-00.bin to P0S3-07-4-00.sbn Regards, -- Ing CIP Alejandro Celi Mariátegui [EMAIL PROTECTED] No, renaming won't work, as it's a signed binary. Plus S versus O designates the application type. The file came with your firmware download from Cisco; it should have included: OS79XX.TXT POS3-07-4-00.bin POS3-07-4-00.loads POS3-07-4-00.sb2 POO3-07-4-00.bin POO3-07-4-00.sbn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax and codecs preferences to PSTN
Hi, I have an asterisk running with a passtrought conf with G729, when I try to send a fax from SIP to SIP the ATAs make a good codec negociation and the fax transmicion is OK, But when I try to send the fax to PSTN fax machine (SIP -- AS5400 -- PSTN) The ATA Device try to send the RTP with G711ulaw and the Cisco keep answereing with G729 a snip some part of my confs. sip.conf [general] port=5060 bindaddr=0.0.0.0 context=default allow=g729 . . . [22194007] type=friend host=dynamic secret=22194007 canreinvite=yes callerid=ATA Sipura FAX 22194007 . . . [as5400] type=friend host=XXX.XXX.XXX.XXX canreinvite=yes insecure=yes insecure=very qualify=yes /sip.conf as5400 dial-peer voice 999001 pots description PRUEBAS SIP max-conn 3 destination-pattern 65732.% progress_ind alert enable 8 port 7/5:D ! dial-peer voice 999000 voip description PRUEBAS SIP destination-pattern 2219400. session protocol sipv2 session target sip-server dtmf-relay h245-alphanumeric fax-relay ecm disable fax rate 9600 fax nsf 00 fax protocol pass-through g711alaw no vad /as5400 Thanks in advance -- René Mayorga Internet Data El Salvador Telecom S.A. de S.V. Tel:(503) 2247-7246 (503) 2247-7156 Cel:(503) 7962-8205 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ARESKICC - Another issue
Try manually creating the file first. --- [EMAIL PROTECTED] wrote: Hi all, After finally making the web interface for AreskiCC work I am now running into new issues. 1 - In Asterisk the manager doesn't seem to connect 2 - When I try to create the file additional_areskicc_sip.conf it says Could not open buddy file '/etc/asterisk/additional_areskicc_sip.conf' Any clues? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Julius Igugu SouthWork Co. Ltd. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P Channels stop answering after some time .
Hi Need help on bridging SIP with TDM400P(4 FXO Modules ) My setup is as follows US OFFICE -TDM400P(FXO) --SIP--- TDM400P(FXOs)INDIA OFFICE (DSL Line) Asterisk Asterisk PBX(Siemens) /DSL Line Server Server Everithing works fine for one or two calls or maximum 4 calls over the setup. Ie after some time zap channels are not ringing.Then I have to reload asterisk.Once restart everithing works fine for 2 or 3 calls over the setup then the same issue .I need to restart asterisk again . Is it the problem with TDM400P ? OR the problem with 2.6 Kernel ? or Problem with SIP and TDM Card ? How I can troubleshoot ? I am using Fedora core3 Kernel 2.6.9-1.667 My zaptel.conf on both systes: loadzone = us defaultzone=us fxsks=1-4 My zapdata.conf on both systems : signalling=fxs_ks rxwink=300 usecallingpres=yes transfer=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes relaxdtmf=yes rxgain=4.9 txgain=6.9 busydetect=yes callprogress=yes progzone=us musiconhold=default jitterbuffers=4 My sip.conf on both systems [pbx] type=friend username=pbx secret=pbx host=192.168.X.Y dtmfmode=info insecure=very qualify=no disallow=all allow=ulaw Do you want any more details ? thanks -Sandeep ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] VoiPSupply Dot Com
Give it a break you freakin Cry Baby Race the Tyrant Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Karl J. Vesterling Sent: Tuesday, May 31, 2005 8:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] VoiPSupply Dot Com Interesting... Seems the first portion of the message of the previous post got chopped... Garrett, evidently there is some verbage to that effect on the site. But just to let you know, no other business that we've done business with requires anything like that. Not a one. Also worthy of note is that the purchase was not a credit card order, so I'm rather surprised your terms regarding credit cards would apply. In retrospect, I guess I should have spent the 16 hours browsing your site looking for the fine print instead of waiting for a prompt shipment. But, alas... We found someone that knows how to do business with businesses. At 07:30 PM 5/31/2005, you wrote: www.myriadsupply.com -Original Message- From: Joe Scinta [mailto:[EMAIL PROTECTED]] Sent: Tuesday, May 31, 2005 12:55 PM To: [EMAIL PROTECTED] Subject: Re: Your request for a Cisco pricing quote from Myriad Supply great hilary and will we be able to accomplish terms (net1) with your company? Joseph A. Scinta President KEN-TON Electronics Inc. 187 Greenacres Rd. Tonawanda, NY 14150 (716-837-9168) PRODUCTION Facility: KEN-TON ELECTRONICS INC 2256 Sheridan Drive BUFFALO NY 14223 (716-875-5114) - Original Message - From: Hilary DeCourcey - Myriad Supply [EMAIL PROTECTED] To: 'joe scinta' [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Tuesday, May 31, 2005 12:46 PM Subject: RE: Your request for a Cisco pricing quote from Myriad Supply Hi Joe, Thank you for contacting Myriad Supply for your Cisco needs. This is what we have for you: CP-7960G @ $299 (New) - requires 2 week lead CP-PWR-CUBE @ $38/ea (New Cisco) CP-PWR-CUBE @ $22/ea (New OEM) All phones are backed by a One Year Warranty. Extended warranties and Cisco SMARTnets are also available. A standard Cisco SMARTnet for this phone would be $20. Please let me know if you have any questions or would like to place an order. Best, Hilary Myriad Supply Company, LLC 212.366.6996 phone x114 212.859.7329 fax [EMAIL PROTECTED] AIM: MyriadHilary www.myriadsupply.com -- END EXCERPT -- At 03:00 PM 5/31/2005, you wrote: Karl: http://www.voipsupply.com/credit_authorization/ If you read the second paragraph it explains our policy as it refers to the billing and shipping addresses. **PLEASE NOTE** If your billing address and shipping address are different you should call your credit card company and have your shipping address added as a valid address. If you do not do this your order may be delayed or possibly cancelled. This is where everyone is getting their information from. Thanks, Garrett Smith [EMAIL PROTECTED] 716-250-3408 Direct 716-903-9495 Cell From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Karl J. Vesterling Sent: Tuesday, May 31, 2005 12:30 PM To: C F; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VoiPSupply Dot Com I'm amazed that this thread keeps going... About the claim of Ship-To being on file with bank... CDW doesn't have a problem with it... Ingram Micro doesn't have a problem with it. Merisel doesn't have a problem with it. Digi-Key doesn't have a problem with it... Why would Voip-Supply??? We accept packages every day with the same Ship-To address specified to Voip-Supply... Additional comments dispursed throughout At 02:32 PM 5/27/2005, you wrote: On 5/27/05, Karl J. Vesterling [EMAIL PROTECTED] wrote: At 08:59 AM 5/27/2005, you wrote: [ snip for brevity ] I just wanted to clarify ... this isn't a voipsupply.com problem at all, but rather a courier screwup... which happens anywhere and at anytime... right? TWO screw ups in the shipment. 1.) It was shipped to the Bill-To address. Since there is no one there during the day I had to sit and wait for it lest it not be delivered. This screw up has to do with the person that ordered it, because they didn't have the ship to address on file with their bank. This was not a paypal transaction. The PO had BIG BOLD LETTERS - Ship To: I'm unaware of any practices with the bank that requiring Ship-To addresses to be on file with them. Perhaps your financial institution is a bit different? 2.) when an order is placed on a Tuesday AM (or) Monday PM, and it's priority overnight, and it's across town, and the tracking number was supplied on Wednesday one would expect that it would show up Thursday, not Friday. See above, again this is a screw up that happened because of the one that ordered it, by NOT having the ship to address on file with their bank.
[Asterisk-Users] Dell SC1425 and TE110P
Hi List I bought 1 Dell SC1425 server and 1 Digium TE110P T1/E1 card. I installed Asterisk from aah 1.0 In the CLI I type 'genzaptelconf -svd' as I have done with other servers and FXO cards to detect and configure the cards; this time it is not recognizing the T1 card. Any ideas why this might be happening?? Thanks! Oswaldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] tellme hiring VXML
Btw just in case someone is looking, maybe we can get someone on the inside to help out J http://www.tellme.com/job_voice-xml.html Service Production Engineering: Senior Engineer, Applications and Tools Tellme leads the industry in large-scale deployments of voice-enabled services, having answered over 1 billion calls to date. Radical improvements in voice recognition at scale and the rapid progress of VoIP for carrier integration are enabling a new generation of services at low cost. Tellme is rapidly expanding to address the opportunities in automated directory assistance, next-generation telephony services (Dial Tone 2.0), and the tools and infrastructure to support those services. Tellme seeks software engineers to lead this effort, working hand-in-hand with cross-functional teams of subject matter experts. Requirements: 3+ years software engineering and systems integration experience requiring first-rate problem solving and analytical skills Extensive web programming experience using _javascript_, cvs, Document Object Model, and perl at scale in UNIX environments Excellent internal cross-functional communication skills a must; client-facing professional services experience preferred Practical experience delivering budgetary estimates and familiarity with speech recognition, VoiceXML, _javascript_ interpreter, and telephony a plus Please send resumes to [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk compatible, hot swappable PRI card
Hmmm, You are going to price yourself out of the market if you go with hot swap. If I understand you correctly that is. Your residential gateway sits in a home and connects to the internet to do VoIP calls for the owner. What is your cost for this gateway? Doing hot swap is going to add, let's say, $500 to the cost. If the system goes down because of the hot swap card who is going to replace it? The customer? Well if the machine is down then a reboot isn't that big a deal vs. the extra cost. A service tech? Well if the machine is down then a reboot isn't that big a deal vs. the extra cost. Besides you would have to have a spare card at the customer's house. So every house has to have more expense and costs. Sorry if my assumption is wrong. I went through a similar exercise once. Race the tyrant Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Fortin Sent: Monday, May 30, 2005 10:59 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] asterisk compatible, hot swappable PRI card Hi We are in a project where we will use asterisk as a residential gateway for IP phone service. We are aiming to replace the primary phone line so the service must be up as long as possible so we are looking at ways to avoid shut downs. We are looking for a solution to allow us to add/remove PRI cards without shutting down the system Is there such a thing as an asterisk compatible hot-swappable PRI card and board ? Someone told me to look at the C-PCI technology, it seems that telecom company use this. Thanks Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Pass-through
It is likely possible. It's going to depend on getting * and your modem bank to play nice together. If your modem bank is collecting ANI or any kind of other carrier signaling info for normal operation, you might have an easier time doing EM wink between * and the modem bank if your modem bank support thats. The setup you're looking to put together here doesn't appear to be much different than folks who have hooked up an * device to an Avaya Definity or other PBX via a PRI tie line. On 6/1/05, Adam Vocks [EMAIL PROTECTED] wrote: In an order to save money, I would like to use a PRI that we have going to one of our dial-up modem banks (We are an ISP.) During business hours these channels are idle and during our peak internet times, we are closed. Sounds too good to be true, but I thought I would throw it out there. These are modem calls that if they would call our modem bank number, they would be bridged to the outbound zap channels??? And of course, if they dial our business number we would send them to the appropriate sip channels. I didn't know if this could be done with two T1 cards and asterisk Here is a primitive sketch. If anyone has information, please share. Thank You Adam Vocks CTI ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 99% cpu on asterisk with chan_unicall and low traffic
Hi, I made a full strace of the running Asterisk process during a high load 99% of cpu usage, aprox. ~800 MBytes of data was gathered and found lots of errors in this log. The errors started when * tried to open a /dev/zap/channel file (before this, there were other errors but I think there are harmless). Following, I include some pieces of the log and an url if you want to download the whole thing in gzipped format (25 Mbytes). .. 1085 open(/dev/zap/channel, O_RDWR|O_NONBLOCK unfinished ... 1091 ... fcntl64 resumed ) = -1 EBADF (Bad file descriptor) 1085 ... open resumed ) = 34 1091 fcntl64(3615, F_GETFL unfinished ... 1085 ioctl(34, 0x40044a26 unfinished ... 1091 ... fcntl64 resumed ) = -1 EBADF (Bad file descriptor) 1085 ... ioctl resumed , 0xbfffdb4c) = 0 1091 fcntl64(3616, F_GETFL unfinished ... 1085 ioctl(34, 0x80184a1c unfinished ... 1091 ... fcntl64 resumed ) = -1 EBADF (Bad file descriptor) 1085 ... ioctl resumed , 0xbfffdd60) = 0 1091 fcntl64(3617, F_GETFL unfinished ... 1085 ioctl(34, 0x40184a1b unfinished ... 1091 ... fcntl64 resumed ) = -1 EBADF (Bad file descriptor) When the cpu is at 99%, lots of write(34, \321\321\321\323\320\323\320\320\321\320\323\320\320\322..., 160) = -1 EAGAIN (Resour ce temporarily unavailable) write(34, \321\321\321\323\320\323\320\320\321\320\323\320\320\322..., 160) = -1 EAGAIN (Resour ce temporarily unavailable) ocurrs which is a FD for /dev/zap/channel. I think there is a problem with file descriptor 34 which is /dev/zap/channel. My zaptel.conf is as follows: # Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 HDB3/ span=1,1,0,cas,hdb3 cas=1-15:1101 cas=17-31:1101 # Span 2: WCTDM/0 Wildcard TDM400P REV E/F Board 1 fxols=32 fxols=33 fxols=34 fxols=35 and the MFCR2 proto configuration in unicall is as follows: protocolclass=mfcr2 protocolvariant=ve,10,7 protocolend=co group = 1 channel = 1-15 group = 2 channel = 17-31 This could give us more clues as to where the problem might be located. Your comments are welcome. The full gzipped 25 Mbytes log can be downloaded from http://www.iconos.com.ve/download/unicall/asterisk-kia-unicall-strace.out.gz Regards. Andres. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Alternate DID
I have 3 Asterisk systems that connected through IAX2 trunks. System 1 has a TE110P installed with a PRI and routes calls based on calling number to systems 2 and 3 through the IAX2 trunk. Systems 2 and 3 have TDM400P cards installed for failover and emergency/911. I am having problems configuring an alternate route if the IAX2 trunk is down to the destination system (2 or 3) from system 1. Ideally, I would like to be able to route the call over the PSTN to the TDM400P if the IAX2 trunk is down or even forward the call to an answering service. So far I am not having any luck so I was hoping that someone out there would be able to share a similar configuration. Thanks, Dave Lewis Sent via the ISCG Web Mail system at !--http://MAIL.iscg.net-- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P vs SIP3000 x2
Thank you very much for all answers. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cannot receive incoming calls via ISDN
I'm experimenting with asterisk. This is my environment:
- Debian sarge (vanilla kernel 2.4.29)
- Asterisk 1.0.7-BRIstuffed-0.2.0-RC8g
- Two sip phones (One cisco 7905 and one soft-phones X-Lite)
- Digi International Datafire Micro V (Europe) (rev 02) (zaphfc)
After two days of work now I can call between the two sip phones and
I can call from sip phones to outside numbers via ISDN card.
Now I'm not able to call one of the two numbers assigned to the ISDN
line and forward the call to the sip phones. When I try to call my two
assigned number I can't get any response.
This im my configuration:
***
ztcfg -vvv
Zaptel Configuration
==
SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
Channel map:
Channel 01: Individual Clear channel (Default) (Slaves: 01)
Channel 02: Individual Clear channel (Default) (Slaves: 02)
Channel 03: D-channel (Default) (Slaves: 03)
3 channels configured.
***
cat /etc/zaptel.conf
loadzone=it
defaultzone=it
span=1,1,3,ccs,ami
bchan=1-2
dchan=3
***
cat /etc/asterisk/zapata.conf
[channels]
switchtype = euroisdn
signalling = bri_cpe
context=incoming
usecallerid=yes
echocancel=yes
callprogress=yes
transfer=yes
setcallerid(${CALLERIDNUM})
overlapdial=yes
immediate=no
callgroup = 1
group = 1
channel = 1-2
language=it
***
*CLI zap show channels
Chan Extension Context Language MusicOnHold
pseudoincomingit
1incoming
2incoming
***
*CLI show dialplan
[ Context 'incoming' created by 'pbx_config' ]
'+390372xx' = 1. Dial(SIP/cisco2) [pbx_config]
'00372xx' = 1. Dial(SIP/cisco2) [pbx_config]
'xx' = 1. Dial(SIP/cisco2) [pbx_config]
'0372xx' = 1. Dial(SIP/cisco2) [pbx_config]
'390372xx' = 1. Dial(SIP/cisco2) [pbx_config]
'' = 1. Dial(SIP/cisco2) [pbx_config]
[ Context 'default' created by 'pbx_config' ]
'100' = 1. dial(SIP/topper) [pbx_config]
'103' = 1. dial(SIP/cisco2) [pbx_config]
'cisco2' = 1. goto(103|1)[pbx_config]
'topper' = 1. goto(100|1)[pbx_config]
'_0.' = 1. Dial(Zap/g1/${EXTEN}) [pbx_config]
***
I'm in stale: if I call cisco2 via sip it works, but when i try to
call from an external line the number 0372xx (xx is omitted)
nothing happens. What can I do to debug this?
Thanks in advance for you attention.
Igor
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pri restarting randomly (TE110P or TE405P)
Hi, we have a E1 pri from Citylink, (they are using Ericsson Engine exchange), that are restarting after 5 - 15 minutes, before and after that we can make calls in and out w/o problems. The cards have been tested in two computers (Atholon XP 2200+ and Celeron 2.6Ghz), are on there own IRQ, not showing any IRQ misses in zttool. It's all correctly configuered - zaptel and zapata (attached). We have tried with Asterisk 1.0.7 and CVS Head. Digium has done the loopbacktests w/o errors. Our question is, is this hardware, software or pri related? What can we do? The PRI is used to an Lucent pbx and is working fine, but when we connect it to asterisk we get this log (CVS HEAD): May 31 20:08:31 NOTICE[1522] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 May 31 20:08:31 NOTICE[1522] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 May 31 20:09:17 NOTICE[1522] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 May 31 20:09:17 NOTICE[1522] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 May 31 20:20:49 WARNING[1522] app_dial.c: Unable to forward frame May 31 20:21:37 NOTICE[1522] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 May 31 20:21:37 NOTICE[1522] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 May 31 20:21:47 NOTICE[1522] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 May 31 20:23:49 WARNING[1522] chan_zap.c: [Span 0 D-Channel 0] PRI: Write to 48 failed: Unknown error 500 May 31 20:23:49 WARNING[1522] chan_zap.c: [Span 0 D-Channel 0] PRI: Short write: 0/15 (Unknown error 500) May 31 20:23:49 WARNING[1522] chan_zap.c: Detected alarm on channel 3: Red Alarm May 31 20:23:49 WARNING[1522] chan_zap.c: [Span 0 D-Channel 0] PRI: Write to 48 failed: Unknown error 500 May 31 20:23:49 WARNING[1522] chan_zap.c: [Span 0 D-Channel 0] PRI: Short write: 0/15 (Unknown error 500) May 31 20:23:49 WARNING[1522] chan_zap.c: Detected alarm on channel 1: Red Alarm May 31 20:23:49 WARNING[1522] chan_zap.c: [Span 0 D-Channel 0] PRI: Write to 48 failed: Unknown error 500 May 31 20:23:49 WARNING[1522] chan_zap.c: [Span 0 D-Channel 0] PRI: Short write: 0/15 (Unknown error 500) May 31 20:23:49 WARNING[1522] chan_zap.c: Detected alarm on channel 2: Red Alarm May 31 20:23:49 WARNING[1522] chan_zap.c: [Span 0 D-Channel 0] PRI: Write to 48 failed: Unknown error 500 May 31 20:23:49 WARNING[1522] chan_zap.c: [Span 0 D-Channel 0] PRI: Short write: 0/15 (Unknown error 500) May 31 20:23:49 WARNING[1522] chan_zap.c: Detected alarm on channel 4: Red Alarm May 31 20:23:49 WARNING[1522] chan_zap.c: Detected alarm on channel 6: Red Alarm May 31 20:23:49 WARNING[1522] chan_zap.c: Unable to disable echo cancellation on channel 6 May 31 20:23:49 WARNING[1522] chan_zap.c: Detected alarm on channel 7: Red Alarm May 31 20:23:49 WARNING[1522] chan_zap.c: Unable to disable echo cancellation on channel 7 May 31 20:23:49 WARNING[1522] chan_zap.c: Detected alarm on channel 8: Red Alarm May 31 20:23:49 WARNING[1522] chan_zap.c: Unable to disable echo cancellation on channel 8 May 31 20:23:49 WARNING[1522] chan_zap.c: Detected alarm on channel 9: Red Alarm May 31 20:23:49 WARNING[1522] chan_zap.c: Unable to disable echo cancellation on channel 9 May 31 20:23:49 WARNING[1522] chan_zap.c: Detected alarm on channel 10: Red Alarm May 31 20:23:49 WARNING[1522] chan_zap.c: Unable to disable echo cancellation on channel 10 May 31 20:23:49 WARNING[1522] chan_zap.c: Detected alarm on channel 11: Red Alarm May 31 20:23:49 WARNING[1522] chan_zap.c: Unable to disable echo cancellation on channel 11 May 31 20:23:49 WARNING[1522] chan_zap.c: Detected alarm on channel 12: Red Alarm May 31 20:23:49 WARNING[1522] chan_zap.c: Unable to disable echo cancellation on channel 12 May 31 20:23:49 WARNING[1522] chan_zap.c: Detected alarm on channel 13: Red Alarm May 31 20:23:49 WARNING[1522] chan_zap.c: Unable to disable echo cancellation on channel 13 May 31 20:23:49 WARNING[1522] chan_zap.c: Detected alarm on channel 14: Red Alarm May 31 20:23:49 WARNING[1522] chan_zap.c: Unable to disable echo cancellation on channel 14 May 31 20:23:49 WARNING[1522] chan_zap.c: Detected alarm on channel 15: Red Alarm May 31 20:23:49 WARNING[1522] chan_zap.c: Unable to disable echo cancellation on channel 15 May 31 20:23:49 WARNING[1522] chan_zap.c: Detected alarm on channel 17: Red Alarm May 31 20:23:49 WARNING[1522] chan_zap.c: Unable to disable echo cancellation on channel 17 May 31 20:23:49 WARNING[1522] chan_zap.c: Detected alarm on channel 18: Red Alarm May 31 20:23:49 WARNING[1522] chan_zap.c: Unable to disable echo cancellation on channel 18 May 31 20:23:49 WARNING[1522] chan_zap.c: Detected alarm on channel 19: Red Alarm May 31 20:23:49 WARNING[1522] chan_zap.c: Unable to disable echo cancellation on channel 19 May 31 20:23:49 WARNING[1522] chan_zap.c: Detected alarm on
[Asterisk-Users] voice-coloring with asterisk
I was pondering of the best way to implement voice-coloring within Asterisk, e.g. pass a channel thru a multiband equalizer and modify it enough where it could be distinguished from other voices in a conference call. This could make conference calls much less confusing. Perhaps the easiest way would be to use sox as the equalizer but I am not familiar enough with * to know how to put a channel thru sox. Anyone? Scripthead ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A Way to Write DTMF Digits as text to CDR?
I've gotten my CDR working the way I like, but I am looking to customize it a bit. I have set up an IVR menu, which works great. I would like to be able to capture the prompted DTMF digits pressed by callers, to my CDR database but I don't see any AGI or Asterisk commands that allow one to customize the CDR contents. Am I thinking about this on the wrong track? If someone calls sales for instance, and presses 44364 for their PO number when prompted, I just want to have a text record of the digits they pressed in my CDR so I can easily view it. No trying to do database lookups or screen pops from it or anything fancy, I'm trying to eat an elephant one bite at a time. Anyone have a solution for that? I hope I'm not being a pest by asking a question every other day, but the responses I've gotten have been very helpful. I'm trying to learn as much as I can from the array of documentation, and I swear I'm only asking when I feel like I've exhausted what I could find. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] HELP Cisco - can't find P0S3-07-4-00.sbn
El mié, 01-06-2005 a las 12:34, Robert Goodyear escribió: No, renaming won't work, as it's a signed binary. Plus S versus O designates the application type. Yes, that's correct, S isn't the same to O My firmware version is 6.3. I check info on these files: cat OS79XX.TXT POS3-07-4-00 and SIPDefault.cnf # Image Version image_version: P0S3-07-4-00 The file came with your firmware download from Cisco; it should have included: OS79XX.TXT POS3-07-4-00.bin POS3-07-4-00.loads POS3-07-4-00.sb2 POO3-07-4-00.bin POO3-07-4-00.sbn I have the same files, and got this error messages: Connection received from 192.168.100.183 on port 50182 [06/01 00:08:55.080] Read request for file OS79XX.TXT. Mode octet [06/01 00:08:55.080] OS79XX.TXT: sent 1 blk, 15 bytes in 0 s. 0 blk resent [06/01 00:08:55.080] Connection received from 192.168.100.183 on port 50183 [06/01 00:08:55.080] Read request for file SIPDefault.cnf. Mode octet [06/01 00:08:55.080] SIPDefault.cnf: sent 12 blks, 6110 bytes in 0 s. 0 blk resent [06/01 00:08:55.130] Connection received from 192.168.100.183 on port 50184 [06/01 00:08:55.190] Read request for file SIP001319ACBD66.cnf. Mode octet [06/01 00:08:55.190] SIP001319ACBD66.cnf: sent 2 blks, 823 bytes in 0 s. 0 blk resent [06/01 00:08:55.190] Connection received from 192.168.100.183 on port 50185 [06/01 00:08:56.510] Read request for file P0S3-07-4-00.sbn. Mode octet [06/01 00:08:56.510] File P0S3-07-4-00.sbn : error 2 in system call CreateFile, Not found. [06/01 00:08:56.510] Connection received from 192.168.100.183 on port 50187 [06/01 00:08:57.110] Connection received from 192.168.100.183 on port 50188 [06/01 00:08:57.110] Read request for file RINGLIST.DAT. Mode octet [06/01 00:08:57.110] Read request for file dialplan.xml. Mode octet [06/01 00:08:57.110] RINGLIST.DAT: sent 1 blk, 44 bytes in 0 s. 0 blk resent [06/01 00:08:57.220] dialplan.xml: sent 3 blks, 1429 bytes in 0 s. 0 blk resent [06/01 00:08:57.220] Hope that you can help me... Regards, -- Ing CIP Alejandro Celi Mariátegui [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] list of settings
Dear all, Sorry to ask, but... Do you know where I can find a full list of configuration parameters and values for each of the .conf files? Do default .conf files include all options? Thanks Again ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] TE11OP - Mitel 200Sx??
Does anyone know the pinout to make a cable so that My Asterisk can talk to
my Mitel 200SX?
- Original Message -
From: Henry Devito [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, April 24, 2005 1:47 PM
Subject: Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
According to the Mitel manuals that version of SX-200D can only use a
regular 24 channel T1. It can not use a PRI interface. You are going to
have to configure * to use a standard T1 not a PRI D4/AMI is the correct
signaling.
- Original Message -
From: Scott Wolfe [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, April 24, 2005 11:09 AM
Subject: Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
Thanks Henry,
-Scott
- Original Message -
From: Henry Devito [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, April 23, 2005 11:05 PM
Subject: Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
I am trying to locate the manual for that level software. If it's not
here at home it is at my office and I will look everything up in the
morning.
- Original Message -
From: Scott Wolfe [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com; [EMAIL PROTECTED]
Sent: Saturday, April 23, 2005 9:00 PM
Subject: Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
The Switch is since 1995 and I get a SX-200 Digital G1005 ENH 672P /
F25.0 09-FEB1994 when I look up the software on the switch board so if
I
am reading what your telling me then I have to do D4/AMI. So does my
zaptel look correct? Maybe my cableing is off.
Thanks,
-Scott
- Original Message -
From: Henry Devito [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Friday, April 22, 2005 8:34 PM
Subject: Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
Of course there are exceptions to the rules. I see now on a couple
software releases where they do allow PRI with D4/AMI and PRI with
esf/b8zs. It's been a year or so since I messed with trunking on a
200, I've mostly been installing and maintaining the SX2000's and
3300's.
Henry
- Original Message -
From: Dennis Walker [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Friday, April 22, 2005 9:13 PM
Subject: RE: [Asterisk-Users] TE11OP - Mitel 200Sx??
I have done the same thing with an sx200 and a pri circuit
My sx200 can only do ami d4 and em channels
Here's parts of my config that takes the pri and converts it to em
with
ANI DNIS
zaptel.conf
# t1 connected to the PRI circuit
span=1,1,0,exf,b8zs
# t1 connected to SX200
# the t1 card on my sx200 did d4 ami and I supplied ANI and DNIS
through
the dial plan
span=2,0,0,d4,ami
bchan=1-23
dchan=24
em=25-47
-
zapata.conf
[channels]
echocancel=yes
echocancelwhenbridged=yes
echotraining=no
rxgain=0.0
txgain=0.0
useincomingcalleridonzaptransfer=yes
restrictcid=no
context=default
usecallingpres=yes
usercallerid=yes
hidecallerid=no
callerid=Company Name8005551212
signalling=pri_cpe
switchtype=dms100
group=1
channel = 1-23
group=2
signalling=em_w
emdigitwait=500
channel = 24-47
# I needed the emdigitwait=500 to wait long enough for the SX200 to
dial
out it's digits
--
extensions.conf
# our PRI circiut gave us the last 4 digits of the dialed number and
this
is how I passed
# *ANI*DNIS* to the SX200 for it to decode
# the first group were individual numbers that mapped to faxes and
modems
exten = 1234,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
exten = ,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
exten = ,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
# this set mapped our did 5000 - 5199 to the SX200
exten = _5[0-1]XX,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
The reset of the dial plan took what ever I set up in the sx200 ARS
to
dial
out and
sent out put Zap/G1
Hope this helps
--
From: Henry Devito[SMTP:[EMAIL PROTECTED]
Reply To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, April 22, 2005 8:56 PM
To: Scott Wolfe; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
File: ATT00262.htmlFile: ATT00263.txt
I was wrong. I just looked in my Mitel IM's. What level software
are you
on in the SX200? Up until a certain level 200's could only do
D4/AMI
T1's,
they could not do PRI's.
RE: [Asterisk-Users] Asterisk@Home 1.1b1 has been released
I don't see the SugarCRM being part of the install. How do you activate this? Seshu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, May 31, 2005 4:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] [EMAIL PROTECTED] 1.1b1 has been released We have replaced the simple contact management system in [EMAIL PROTECTED] with SugarCRM a full CRM system. This might seem like over kill for a home PBX but Sugar has the best contact management we have seen. With click to dial functionality and the ability to import data from other contact managers it's a great fit for [EMAIL PROTECTED] We have also added new version of the usual Asterisk software AMP and Flash operator panel. Download from http://asteriskathome.sourceforge.net For support please read the [EMAIL PROTECTED] Handbook http://asteriskathome.sourceforge.net/handbook/index.html and use our support forum at http://sourceforge.net/forum/?group_id=123387 __ Do you Yahoo!? Yahoo! Small Business - Try our new Resources site http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
