similar to: Multiple Line Appearances / Why use this?

Displaying 20 results from an estimated 800 matches similar to: "Multiple Line Appearances / Why use this?"

2005 Sep 07
3
Extensions - Realtime
CVS HEAD/Asterisk 1.2: Is there a way to have the entire extensions.conffile coming from the realtime? It appears that RealTime for the extensions.conf file is on a context by context basis, but you have to create each new context in the extensions.conf file then add a "switch => Realtime" line (then reload). I want to be able to add phones without having to edit any files.
2005 Sep 07
3
Hosted PBX (vPBX) and Call/PickUP Groups
Hi, I'm working with this issue for a while, Now I already solve the dialplan issues, but I still have a question about the Callgroups, I read at www.voip-info.org that , there is a 63 limit of callgroups. And I'm wondering why?? and if the 1.2.0beta version supported more than 63 Groups?? (I did'nt find any Changelog for 1.2) or If not There is any unoficial patch for that ? Thanks
2005 Sep 07
1
Several SIP clients behind router register with the same IP, messing up call routing, any ideas?
Dear all, Has anyone seen this before and can suggest a solution? I've got 3 sip clients behind the router, and they all register with Asterisk using the same IP address. Now, wenn all are registered, all the calls get routed to the client that registered most recently, but not to the correct client. Also, there seems to be some problems registering all 3 clients simultaneously. Could
2005 Sep 08
1
Additional: Several SIP clients behind router registerwiththe same IP, messing up call routing, any ideas?
Hello, I'm still looking for any ideas on this problem: I've got 3 sip clients behind the router, and they all register with Asterisk using the same IP address. Now, wenn all are registered, all the calls get routed to the client that registered most recently, but not to the correct client. Also, there seems to be some problems registering all 3 clients simultaneously. I have
2005 Sep 08
2
play each person's voicemail
How do I set each extension to play it's own voicemail prompts? I have vm working in that it plays the standard "person at extension 1234 is not available....." and takes the message. I've recorded seperate .gsm files for each user but can not figure out how to use them. - Gary Edison Information Technologies www.EdisonInfo.com P.O. Box 554
2005 Sep 08
2
Distinctive ringing on Cisco 79xx
Greetings, I am trying to implement distinctive ringing on a Cisco 7960. I have tried setting alert_info to chirp1 or chirp2 before dialing the phone, but it has no affect. If you have successfully implemented distinctive ringing on a 7960, I would really appreciate seeing the snipit of code that works. Thanks in advance Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775)
2005 Sep 06
5
Good Polycom Dealer?
Could any of you provide me information on a good Polycom phone dealers to utilize. One who provides firmwares ..etc Thank you! Kenny ______________________________________________________ Click here to donate to the Hurricane Katrina relief effort. http://store.yahoo.com/redcross-donate3/
2005 Sep 07
1
asterisk.org blocked - rejecting connections
Address lookup canonical name asterisk.org. aliases addresses 216.27.40.102 Service scan FTP - 21 Error: TimedOut SMTP - 25 Error: ConnectionRefused HTTP - 80 Error: ConnectionRefused POP3 - 110 Error: TimedOut NNTP - 119 Error: TimedOut digium.com is ok though Address lookup canonical name digium.com. aliases addresses 216.207.245.1 Service scan FTP - 21 Error: TimedOut SMTP - 25
2005 Sep 08
1
(OT) Dialplan Standards for Business/Offices
Are there any standards for setting up pbx dialplans for businesses/offices? What I mean is that, which numbers are reserved for a specific use ex. 0 for operator ? Putting Zero for operator in the dialplan seems to be the common practice of businesses. If there is such a standard, * and # are used for what ? -------------- next part -------------- An HTML attachment was scrubbed... URL:
2005 Sep 06
1
Some problems (SendDTMF, Wait, Parked Calls)
Hi all! I would like to solve some problems: I have a sip provider that lets me make pstn calls after listening some stuff and entering a pin number: 1) How can I make Asterisk enter the pin number? Then wait 1 second and enter the phone number? I have in extensions.conf: exten => 6*,1,Dial,SIP/2002@myprovider,60,tr I have tried with w (like with ZAP channels) but it does not work, nor
2005 Oct 04
1
Strange Problem
Hi I am facing a strange problem. I have integrated speex codec's narrowband mode in my SIP based server. Then I tried to integrate the wideband mode. But the program crashes mysteriously. My encode and decode codes for wide band mode are exact similiar to that of narrowband, except the mode initialization, where I put "speex_wb_mode" instead of "speex_nb_mode". My
2005 Sep 10
1
PRI echo
Hi, My configuration is pri----*(te405p)---iaxclient. My * version is 1.0.7 running on tyan dual opteron board. I have several problems. 1) inbound echo For outbound call(iaxclient-->pri), there is almost no echo. But for inbound(pri-->iaxclient), I can hear distinct echo. Can Sangoma a104 or digium te406p help this problem? 2)Today i received te406p. I know T1/E1 jumper. But how can i
2005 Sep 07
1
Eeven Stranger - Asterisk BT100 Password Issue
Following on from my below email, things have taken another bizarre twist.. I have continued getting the error when 2092 tries to listen to messages by dialing 9999. --Executing VoiceMailMain ("SIP/2092-6918", "2092") in new stack --Playing 'vm-password' (language 'en') WARNING: app_voicemail.c:4922 vm_authentication: Unable to read password. Then I
2005 Sep 15
0
Transfering from a device to a queue crashesAsterisk
Hi David, I've got probably the same/a similar problem. Do you add the phones to the queue (AgentLogin/AddQueueMember)? If there are entries like: " Spawn extension (macro-dialout-trunk,s,21) exited non-zero..." in your * log file you might have the same problem like me. I suspect that something goes wrong with the nested macro calls within the AMP-generated dialplan, so what I
2005 Sep 29
0
Can't make outside call with SIP softphone
Hi, I am can make local extension to and from SIP X-Lite softphone, but I can't dial out using X-Lite but local analog works just fine. Here are my conf files any idea? Thanks, David my sip.conf [general] bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) allow=all [3000]
2005 Oct 05
3
IPComms Setup
Hey I just setup service with IPComms and they are telling me to setup such as this: iax.conf: [IPCommsNet] type=user host=69.15.xxx.xx context=voicepulse-in ;(changed by me) nat=yes dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=gsm When I'm calling once of my numbers it's giving me this though: Oct 5 12:11:06 NOTICE[49584]: chan_iax2.c:5476 socket_read: Rejected connect
2005 Sep 29
14
Draggables and overflow div''s revisited
I have two scrollable div''s (overflow:auto), one with a list of elements (the source) and the other is the drop target (dest). I''ve enabled ghosting so that the drag element gets out of the scrollable box (good). Interesting, at least on Firefox, the ghosted drag ends up going ''under'' the destination div when I drag it. No amount of z-order fidding seems to
2005 May 25
4
Polycom IP501
Hi All - I noticed that the Polycom IP501's are now shipping. Has anyone gotten one yet, and if so, what's different about the phone? Any UI improvements, or is it just better hardware? Thanks, Noah
2005 Oct 05
2
Trillian?
Has anybody got Trillian to work? If so, how? Dex -- -----BEGIN GEEK CODE BLOCK----- Version: 3.12 GCS d--(+)@ s-:+ a- C+++(++++) UL+>++++ P+>++ L+++>++++ E-- W++ N o? K- w--(---) !O M+ V- PS++(+) PE(-) Y+ PGP(-) t++(---)@ 5 X+(++) R+(++) tv--(+)@ b+(+++) DI+++ D G++ e* h>++ r%>* y? ------END GEEK CODE BLOCK------ http://www.againsttcpa.com - nothing fights like the opposition
2005 Sep 12
2
Callerid fails in any release after beta1 fails
I have 1 x100p. Caller id works fine with the beta1 release. Cvshead releases fail with a combination of checksum and ss_thread errors? I'm concerned when beta2 or the 1.2 release comes out it will not work. I have been through the configs I can't find and changes that need to be made to get CVSHEAD to work. Thanks John Hill