similar to: Sip phones how to dial a # sign?

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

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

I use them with just the one NIC card. I don't use them as a router so the phones and my gateway are all on the same network. -Jonathan -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Remco Barende Sent: Tuesday, September 06, 2005 4:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion

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

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

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

Hi list! While looking for the Snom 190 I found another phone, the Elmeg IP 290 (www.elmeg.de). Looking at the pictures & the specs they seem to be very similar beasts but the firmware is supposedly not interchangeable. Does anyone know the difference between the 2, do they work with Asterisk? The weird thing is that Elmeg has similar phones with the Snom look but they are ISDN only (no

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!

Is it possible to rewrite caller id's? I would like to have sip phones appear by their local cid (like Henk <208>) but when they call out using the PRI I would like their full DID (MSN) to appear (like 0031201234567) I could ofcourse set callerid to the main phonenumber but surely there must be a better solution? Thanks!! Remco

I tried to get callerid working the normal way but the cid is never passed to the phone. It doesn't work untill I set SetCIDNum(0${PRI_NETWORK_CID}) in extensions.conf which I found in the wiki: http://www.voip-info.org/tiki-print.php?page=Asterisk+zaphfc Is this intended behaviour, or still a bug? It does work but it only shows one zero even though I have nationalprefix = 0

Hi list! I'm using Asterisk 1.2.7.1. with FreePBX 2.0.1 on a CentOS 3.7 box. On the * box I also have a samba share where our CRM app can dump call files and a cron script is moving the call files every second to the asterisk directory. Everything goes really quickly, the call file is placed on the samba share and very quickly moved to the asterisk dir, so far so good. But then the call

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

Hi list! I'm currently using a HFC-S card for my ISDN BRI line with bristuff. The instability is driving me crazy however. I'm having continuous problems where inbound calls will not work after some time of operation (the number then appears as not in use to the caller) or also outbound calls do not work. The solution is to unload the modules, stop asterisk, re-load the modules and

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! 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!

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

Just out of interest, why is there no zaptel script included in the tarballs of 1.0.3? I used to use the RPMS but they haven't been updated for some time but now I'm missing the zaptel init.d script. Or should I have the modules loaded another way? Cheers! Remco

Nicol?s Gudi?o <asternic@gmail.com> wrote: >>> Paul, 1.0.5 stable suffers from caller id issues as well, at least for >>> SIP channels. What fixed things for me was swapping in app_dial.c from >>> 1.0.2 stable (didn't try others). You could also just diff app_dial.c >>> between versions to find the problem but I took the lazy way out the >>>

When I use an analog phone connected to a Sipura SPA-2000 it takes about 3-4 seconds before the number is actually dialled. Very annoying especially if you are connecting an intercom to it. Can I change this behaviour and do I need to look at * config or the config of the SPA-2000? Thanks!

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: