Re: [Asterisk-Users] Unable to create channel of type 'SIP'

if sip show peers didn't work, probably the phone u trying to dial isn't registered as was said before. In sip.conf check the username, secret, host, userid. You made a mistake somewhere along the line. some phones will only register if host is set to dynamic..._

[Asterisk-Users] Maybe not worded right, Answering a call

Can i set the time asterisk takes to answer a call. because it takes at least 10 seconds before it starts any dialplan activity.___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSU

Re: [Asterisk-Users] sip phones make connection but no-sound is heared

Do you have a firewall turned?___ 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] 3-Way Calling in Asterisk

I have to agree with eric, 3 way calling, in my situation is as easy as putting the caller on hold, flash, and take the call off hold. Of course the appropriate settings have to be on in the zapata.conf___ Asterisk-Users mailing list Asterisk-Users@lists.

Re: FW: [Asterisk-Users] polycom 500 help!!

WHen you say cannot communicate you mean it keeps giving you a busy signal when you try and dial? and could you post ur sip.conf along with the messages asterisk prints out. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.dig

Re: [Asterisk-Users] IRQ headaches

Have you tried moving it to a different pci slot? ___ 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/a

Re: [Asterisk-Users] Meetme2 compilation problem

I'm sure there was a patch for meetme2 regarding compilation... google for meetme2 + patch. It worked for me. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update

Re: [Asterisk-Users] meetme2 compilation

That doesn't work. I was trying to do it yesterday, there is a patch that fixes the problem. google for it or if ur too lazy: http://lists.digium.com/pipermail/asterisk-users/2004-August/059709.html ___ Asterisk-Users mailing list Asterisk-Users@list

[Asterisk-Users] Manager Redirect fails on Zap Channels

Attempting to redirect a Zap channel through the manager interface fails. SIP channels can successfully redirected, however, for Zap channels, the Manager API will report that the redirect was successful however the channel (Zap) is immediately hung up. Any experiences with redirecting Zap

Re: [Asterisk-Users] Voicetronix Tones

Hmmm, maybe the dtmfmode is incorrect. in your sip.conf what is dtmfmode set to? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://list

Re: Re: [Asterisk-Users] DyDNS + externip

So is this where SER comes into play? Letting ppl register thru SER and use asterisk behind it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

[Asterisk-Users] DyDNS + externip

Can i use a domain name instead of an IP address for externip (sip.conf) Because im using dynamic dns. Not sure what i'm trying to achieve as yet but, i want to know if it is possible? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http:/

Re: [Asterisk-Users] Searchable Asterisk-users archive available

Nice idea, good job ___ 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] I can't load modules (ztdummy, wcfxo.o)

Are u using zaptel-1.0.6 & what OS? cuz i got the same errors this morning when trying to modprobe ztdummy. Im using gentoo + 1.0.6, on redhat9 everything is working ok. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.

Re: [Asterisk-Users] A bit of a survey: What do do if you need morethan 4 C.O. lines

I'm was having a couple issues also, mainly callerid when turned on was crashing asterisk, but its was my fault still. But does any of the digium cards beside x100p offer redundancy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://

Re: [Asterisk-Users] SpanDSP - Still can't send

BTW what versions of libtiff & spandsp u using, cuz i can't recieve faxes at all. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lis

Re: [Asterisk-Users] Zaptel Red Alarm

could be that someone plugged out ur telephone line and plugged it back in. ___ 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.dig

[Asterisk-Users] Power failure + which card must i choose

If there is a power failure, which cards other than x100p and voicetronix openswitch provide "redundancy". ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update opt

Re: [Asterisk-Users] Trying to install X100p

Did you install the drivers for the x100p (zaptel) first and then install asterisk. and what version of asterisk you using ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRI

Re: [Asterisk-Users] Trying to install X100p

Never mind, i saw it futher down. I guessing that you've plugged a phone line from your telephone jack to the x100p (on the right side) then if you've loaded zaptel & wcfxo the in you dialplan add something like this: exten => 100,1,Dial(Zap/1/"any telephone number") with out the quotes. try that

Re: [Asterisk-Users] Trying to install X100p

Wait a second, whats the problem you having? ___ 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/asteri

Re: [Asterisk-Users] Asterisk and Sipura SPA-841 SIP phones

Nothing to do with your question, but by any chance, when you plugged the phone into the wall did you hear a dialtone or is this something generated by asterisk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailma

[Asterisk-Users] Spaces in config files??

In most configuration files i see that they comment lines instead of adding spaces. e.g. - correct way ; ;IAX configuration ; [general] blah blah blah.. ; register => .. Is this incorrect: ; ;IAX configuration [general] blah blah blah... register => I guess what i'm tryin

[Asterisk-Users] codec order, does it matter

Does the order in which you allow codecs matter? cuz i've found that somethings work better if you allow them in a particular order. Alot of warnings and errors can also be eliminated. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http:

[Asterisk-Users] echo's + cheap phones

Is it possible that cheap phones (Budgetone) cause echo's? I had a digium X100p and i managed to get rid of all the echo problems i was having. Recently i got a Voicetronix Openswitch12, and getting terrible noises when i use an IP phone (budgetone) to call analog phones or PSTN. I have tried al

Re: [Asterisk-Users] Who is in control Voicetronix OR Asterisk

Thanks for the link. will have to check it monday. Respect ___ 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/l

[Asterisk-Users] Who is in control Voicetronix OR Asterisk

>From what i've read i think that asterisk is suppose to generate the dialtone. But when i pickup my analog phone it seems that the Openswitch12 card is the one playng the dialtone. Im almost positive the card is playing the dialtone, can anyone confirm this? And if so how can i change the dialt

[Asterisk-Users] Am i in control after i dial?

I want to be able to interrupt a conversation and let festival say something like "you have 10 seconds left for this conversation". Can i interrupt a call after its been dialed. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.

Re: [Asterisk-Users] Network Test Tool?

www.testyourvoip.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] Dialing Delay

Like manxpower said, set DigitTimeout to 2 seconds or whatever u want. Visit voip-info and look for urself. All whats happening is that it waiting to see if u will press another number (pattern matching) by default digitstimeout is set to 6 seconds you might want to change your dialplan as well.

Re: [Asterisk-Users] Unable to create channel of type 'Zap'

Hmmm, when you plug the phone line into the x100p card slot (from PSTN) then asterisk should say something like: alarm cleared blah blah blah... then try the call again. If your not getting that message try changing the signalling in zaptel & zapata.conf to fxsks=1, signalling=fxs_ks respectively

Re: [Asterisk-Users] Jamaica DID

I thought i was the only one from jamaica testing asterisk. Nice to see others testing it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: h

Re: [Asterisk-Users] Re: Asterisk-Users] very loud scratchy noise!

you could have echo cancel set higher than 128. ___ 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/ast

Re: [Asterisk-Users] Cannot Hear at all

The asterisk server doesn't need a sound card. The machines that have the xlite softphone need to have a sound card. E.g. you have 2 pc's running xlite + the asterisk server. The asterisk server doesn't need a sound card but the 2 pc's will. (sounblaster live solved similar problems i was having)

Re: [Asterisk-Users] Firefly lockup in Win98

well i found out that when im not connected to the internet and try to run firefly it crashes at startup (WinXP) when i connect to the internet and try again it runs alright. (WinXP) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://l

Re: [Asterisk-Users] What is sip-friends.sql??????

Its a way of storing ur sip stuff in a database rather than using the flat files. Sip friends - extensions.conf stuff. Sip_buddies - sip.conf stuff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/as

Re: [Asterisk-Users] Asterisk on SuSE 9.1?

Never had a problem like that ionstalling asterisk on suse. maybe its the cvs version try using 1.0.1, or 1.02 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit

Re: [Asterisk-Users] Problem with Grandstream bt100

Can you still type the ip address of phone in the browser? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/aster

Re: [Asterisk-Users] Why, why, why???

Sure you have a zaptel.conf file. /etc/zaptel.conf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] ast_data causes pbx_realtime.so to fail

installing ast_data causes loading of pbx_realtime.so to fail. asterisk reports undefined symbol pri_dump_info is undefined then exits. Numerous versions of asterisk, libpri, and ast_data have been tried with the same error causing the same failure. ___ A

[Asterisk-Users] Module Failure

How can i get asterisk to still load, after a module has failed to load. Can i skip over some modules. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: ht

Re: [Asterisk-Users] Grandstream Ringtone

Thanks for the link ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] X100p and 6 second delay

I tried that but still getting a delay, do you think its the X100p card. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/l

Re: [Asterisk-Users] X100p and 6 second delay

Or can i set the Interval of ingorepat ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users