similar to: Mailing list down?

Hi list! I have a strange echo problem. Two days ago I setup * 1.0.6. at a friend of mine. Just an * server and for outbound calls wengo.fr was used to place calls via sip. He had a strange echo on the line I didn't experience on my setup. Today I upgraded my asterisk 1.0.5 to 1.0.6 and suddenly I have an echo too on sip calls thru wengo!! I already verified wengo was not the source of

Most people seem quite positive about Snom phones, I cannot share this opinion. The displays are dying quite often, and firmware is buggy. I have tried every firmware from 4.5 up to 5.x and 6.04 but keep having problems with phones locking up or rebooting during an ongoing conversation. REALLY annoying for a phone that is advertised / targeted as a business class phone

Is it possible in any way to use an Outlook contacts database as the source for the internal Asterisk database that is used for callerid lookups? Thanks!

Hi list! I'm sure the topic has been discussed but I could not find what I was looking for. What would be the best / easiest VPN software solution. I would like to install vpn software on the * server for roadwarriors to connect to with laptops running windows. Ideally the vpn solution will not require any additional software on the client side but will use IPSEC. (Ofcourse call quality

Hi list! I have some sip phones and Sipura ATA 2000's. However after dialling a number I need to dial a # to control a device. When I dial # Asterisk kicks in and puts the call on hold. How can I change this? Thx!! Remco

Hi, Snom phones firmware 5.0 is now out. Try it if you like: http://www.snom.com/wiki/index.php/Main_Page. Regards, --------------------------------------------------------------------- Usman Tahir snom technology AG www.snom.com --------------------------------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL:

-----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r Terry Wade Skickat: den 19 augusti 2005 07:08 Till: 'Asterisk Users Mailing List - Non-Commercial Discussion' ?mne: RE: [Asterisk-Users] Zaphfc.ko module error Hi Remco Thanks for the response. I am running Suse 9.3 kernel 2.6.11.4-20a-default. Will check

Hi list! What exactly is the meaning / function of the pridialplan & prilocaldialplan? I've been trying to find out what the different possibilities for these settings are but couldn't find a clear answer. The possible parameters I could find are are : local,unknown,dynamic,national,international and maybe there are more? Thanks!

Hi list ! Is it possible to show the used Useragent of a peer that registered with Asterisk? It's being saved obviously because the console says so when a phone is registering but sip show peers doesn't show it? Is there any other way to view it? Thanks!

Hi List! I installed Asterisk 1.0.3 stable on a RHEL rebuild. Due to problems with * modules refusing to build I replaced the RHEL kernel with stock 2.6.10. Asterisk seems to be working but when I dial voicemail I hear nothing. When I hangup I see a message on the console that the calller did not specify a mailbox number so I guess voicemail app is working. The phone(Grandstream BT100) is

This is a re-post as it was pointed out that I replied to a different thread instead of creating a new post. Sorry for the additional traffic. Mike Dear All, I understand the excitement surrounding a service like Asterisk, and how easy it is to jump in and ask a heap of questions. I also know how frustrating it can be dealing with a 200+ post per day mailing list as one of the question

Hi! I'm trying to setup a Wildcard TE110P with a PRI in The Netherlands. I get a Red Alarm on the line. Is there any way of debugging this? I've tried some configs that should work but without success. Is there any way of telling if the cabling is correct or what else the problem could be? Thanks!! Remco

Hello, It's been noticed that mailing list monthly rollovers didn't happen because the machine was down when the cronjob to do the rollover would normally have fired. I will trigger them by hand now... upshot is that I'm not bothering to refile the first five days of May in the correct slot. I sincerely hope that no one cares :-) Monty --- >8 ---- List archives:

Hello, It's been noticed that mailing list monthly rollovers didn't happen because the machine was down when the cronjob to do the rollover would normally have fired. I will trigger them by hand now... upshot is that I'm not bothering to refile the first five days of May in the correct slot. I sincerely hope that no one cares :-) Monty --- >8 ---- List archives:

Hello, It's been noticed that mailing list monthly rollovers didn't happen because the machine was down when the cronjob to do the rollover would normally have fired. I will trigger them by hand now... upshot is that I'm not bothering to refile the first five days of May in the correct slot. I sincerely hope that no one cares :-) Monty --- >8 ---- List archives:

Hello, It's been noticed that mailing list monthly rollovers didn't happen because the machine was down when the cronjob to do the rollover would normally have fired. I will trigger them by hand now... upshot is that I'm not bothering to refile the first five days of May in the correct slot. I sincerely hope that no one cares :-) Monty --- >8 ---- List archives:

Hello, It's been noticed that mailing list monthly rollovers didn't happen because the machine was down when the cronjob to do the rollover would normally have fired. I will trigger them by hand now... upshot is that I'm not bothering to refile the first five days of May in the correct slot. I sincerely hope that no one cares :-) Monty --- >8 ---- List archives:

I installed bristuff0.20-RC3 (which includes * 1.0.3 stable) It works fine until I disconnect the phone jack for the ISDN line. Even after plug it back in asterisk still reports that it could not create a zap channel when I try to dial out and the line gives an engaged tone when I try to dial. Re-starting asterisk doesn't solve this, I have to stop asterisk, unload the modules, reload

Hi list! I have some strange problems with Asterisk 1.0.5-BRIstuffed-0.2.0-RC5. Very regularly asterisk seems to lose connectivity with the ISDN line. If you try to call in you get the information tone that the number is not in use. Outbound calls do stil work however. Unloading the modules and reloading them and start/stop asterisk will solve the problem. Another problem that occurs

I have Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 up and running. Everything seems to be running fine but after some time asterisk just goes crazy (even withouth any incoming or outgoing call activity perviously). If I leave the box up for some time * goes haywire and the console is flooded with this message: Ouch ... error while writing audio data: : Broken pipe At that time I can see that there