similar to: Can I dial a number from handset to pickup voicemail?

Hello On a Sipura SPA-841 handset (and also at other end) you hear a sea wave like sound - it gets louder then softer and continually repeats. I don't remember hearing this when using other handsets. But what is this effect? How can I reduce it? Angus

Hello I have created an iax exten in my iax.conf file: [300] type=friend username=300 secret=*** context=default host=dynamic callerid="some name" <300> auth=md5 Then in my extensions.conf I have: exten => 300,1,Dial(IAX/${EXTEN},20) exten => 300,2,Hangup I can dial from iaxComm (a soft IAX client) and that works fine. But when I try to dial 300 get: WARNING[22077]:

Hello I have a Junghanns ISDN BRI card for incoming calls and use SIP Polycom IP300 phones. My config files look like this: features.conf pickupextn = *8 zapata.conf context=frompstnisdn group=1 callgroup=1 pickupgroup=1 I also edited sip.conf like this: group=1 callgroup=1 pickupgroup=1 But on internal and incoming calls if I dial *8 from any phone I cannot pickup. Do I need to add

Hello I am running Asterisk on SUSE Linux Professional 9.3 on a VIA Epia M Series motherboard - CPU runs at 1GHz. There is no fan - just a large heatsink. Currently system is running off standard IDE hard drive - because I couldn't get astlinux to run with my Digium TDM04B card (only PCI card in system). Strangely I also have the same system also running SUSE Linux running as a file

This is my first post, so please feel free to direct me to another list if needed. I am in the early stages of researching Asterisk. I administer a small Avaya Definity G3 switch (~400 users). Can anyone point my to resources/documents/actual implementation notes of using Asterisk's voicemail with an Avaya Definity switch? Many thanks, Brian Hudson --------------

I have 2 sip accounts setup - 200 and 202. If I do sip show peers I get: sip show peers Name/username Host Dyn Nat ACL Mask Port Status 202/202 192.168.0.6 D 255.255.255.255 5060 Unmonitored 201/201 (Unspecified) D 255.255.255.255 5060 Unmonitored 200/200 192.168.0.3 D 255.255.255.255 5060

Hello When I dial out from my Asterisk (using Digium analog TDM04B card over pstn line), calls appear to be from +34<rest of number> I am in UK which is +44 so cannot work out why seeing +34. In my zapata.conf I have: loadzone = uk defaultzone = uk I can't find any country specific stuff in any other conf files. Any ideas how I can correctly set so that calls from my asterisk do

Hello I am using a Snom 190 and the quality seems OK. Trouble is the volume is quite low and full volume on the Snom is still too low. Is there something I can do on the asterisk to increase the volume? Angus

Hello I want to setup an Asterisk in several offices with 4 BRI ISDN. I am looking for recommendations on hardware. Criteria would be ease of setup, reliability and cost. The Eicon 4 BRI cards seem fairly pricey. Shame Digium don't do a ISDN BRI card. Angus -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello I have been asked by a client to process a list of telephone numbers. Asterisk should call each number in turn and if the recipient of the call answers, play a message - eg from a wav. How would I go about doing that? Angus

Hi! Trying to figure out what I am doing wrong... 1 std SIP phone (0317998975) registered to an Asterisk (SVN-trunk-r234256) 1 Cell phone 00733025975 attached through H323. extensions.conf exten => 975,1,Goto(975-${DEVICE_STATE(SIP/0317998975)},1) exten => 975-INUSE,1,VoiceMail(0317998975 at inputinterior.se,bs) exten => 975-INUSE,2,Hangup() exten =>

Hello I see it is possible to buy Flash Disks up to 4GB now. Has anyone any experience of building an Asterisk system with a flash disk as the only storage device? Any brands you recommend? Is 2 or 4GB enough for an Asterisk installation? Typically how many MB is required for voicemail recording files for say a 10 user system? What about voicemail - I suppose files could be emailed and

i was reading posts on wiki and noticed lots of posts about Avaya 4610 handset having issue with MWI, Anyone has any more updates? Is this still the case? Any good tutorial for configuring these phones and Asterisk? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080318/3d397ca7/attachment.htm

Hi ! I am trying to connect Asterisk with Avaya Definity. I use this tutorial to do this http://cyril-constantin.blogspot.com/2008/04/howto-connect-avaya-to-asterisk.html The comunication between avaya and asterisk is fine but without sound. I can call from Asterisk to Avaya and extension ring or Avaya to Asterisk and extension ring too but I cant hear anything Example Asterisk ---> Avaya --

What an ACTIVE newsgroup! I'm in the early stages of researching Asterisk. My current environment is a small college (~1000 sets/~400 student sets), Avaya Definity G3si/Seimens Rolm Phonemail. As you can imagine, the maintenance, licensing, and equipment costs are HEFTY. So.. are there any small colleges/universities using PBX or Voicemail? If so, I'd be interested in your migration

Hello I have a Grandstream GXP2000 with latest firmware. When I use it holding the handpiece I don't hear any echo - neither does other end. However, if I use it handsfree, the other end notices echo when they speak - ie their voice is echoy. I hear their voice being a bit echoy. Is this purely down to the IP Phone? Is there anything I can do about it? I considered buying a more

Hello! We have asterisk connected over PRI no our phone network, so I'm avaya PBX user. Asterisk connects to another avaya system over h323. Connection can be shown as avaya--PRI-asterisk--h323-avaya When I do call as avaya user I see name of remote end avay user, i.e. connected line info. As I see in debug remote side send is as 14:07:29:758 Received H.2250 Message = { 14:07:29:758

So far I have not found a way that I can register the Avaya phone with Asterisk. From what I have found so far is that Avaya phone needs the Avaya Media Server and Avaya Gateway. Looking at the h.323.conf (in Asterisk) and the file 46xxsettings.txt (avaya file located in tftpboot) there are no settings to make the phone initialize. I have sent an email to the Asterisk Users Mailing List to see

Does VoicemailMan have to be installed ? Why not available. I have setup a mailbox in voicemail.conf and I can leave a voicemail - just cannot pickup up using *97. My *97 code in extensions.conf: exten => *97,1,Answer exten => *97,2,VoicemailMain(${CALLERIDNUM}@default) exten => *97,3,Hangup asterisk console: Verbosity was 8 and is now 12 -- Executing