similar to: Grandstream Setup

Displaying 20 results from an estimated 1000 matches similar to: "Grandstream Setup"

2005 Jul 05
4
Asterisk on Linksys WRT54G
Hi all, Any one tried installing Asterisk on Linksys WRT54G? We have but facing problems with SIP to SIP calls. The phones ring and calls are established but we cannot hear any voice at all. I tried allow=all in the general section but did not work. So I forced ulaw. Can any one please check it out and let me know what is wrong? Here are the conf files: Asterisk Version: Asterisk
2003 Nov 15
2
Internal server error - cannot align media streams - help needed
Hi, I configured asterisk on redhat linux 9 box. I installed two different ip softphones (SJPHONE and X-PRO) and got them registered with asterisk. The call from one phone to another does get routed via asterisk, but there is one problem coming up. As soon as call is accepted by the end user , it is automatically disconnected with the error "cannot align media streams". If I enable SIP
2004 Jul 05
7
Calling an outside phone number as part of a hunt
I'm trying to see if this is even possible. When you dial ext 2000 I want it to ring my sip phone for 20 sec then call my cell and let it ring for 10 sec if I do not pick up the call on my cell I would like it to go back to * and leave a voice message for me. Here is what I have so far in my extensions.conf Everything works except the call will not go back to * after the 10 sec rule has
2005 Feb 24
2
softphone has problem to call out via X100P card
Hi all, I have the Asterisk set up and 2 softphone (Xlite) set up on two other PC. With the following configuration, I can use one softphone (2000) to call the other one (2001) and/or the voicemail at 2999. Here is my problem: 1. When I dial 9+nxxx-xxxx with one of the softphone to the PSTN via X100P card, I got busy tone. (i.e. I want to use the phone line which is connected to the
2004 May 04
3
g.729 - licenses and opinions
I have a few SIP phones, Cisco 7960s, and was looking into implementing some compression, ala G.729. I'm looking into purchasing a g729 licenses just to get an idea of performance and voice quality, over lans, wireless and single channel isdn. Does anyone have positive/negative experience w/ getting licenses/support from Digium? Hows the sound quality compared w/ g.711? Is 729 better
2003 Jul 28
1
Call Forwarding and DND conf
I have put together this call forwarding and dnd config: I'm sure it can be dome with macro's but I couldn't figure that out... anyone care to input. 74 Turns DND on my phone will not ring, drops caller to voicemail... 73 Turns DND off 72+ext forward your extension to another extension and voicemail is left at the forwarded extension. 71 turns off call forwarding. ; dnd Could
2004 May 20
4
x100p card + dailing out
I think I have it configured properly. ztcfg -vv shows it as channel 1 and zttool shows it as OK. But I can't dial out. When I try, it shows it arrive in teh right stack, but then issues the following errors: channel.c:1676 ast_request: No channel type registered for '{PSTN-1}' app_dial.c:554 dial_exec: Unable to create channel of type '{PSTN-1}' = = Everyone is busy at
2005 Jul 01
1
asterisk newbie and phones which don't want tocomunicate
hi do u have the sip phones extensions in the extension.conf and are they in the right context (sip-incoming)??? are the sip phone registering to asterisk?? try stop asterisk and reconnect as asterisk -vvvvvvvc to check see them registering... ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Sistemista WebSolvingJaa Sent: Fri 7/1/2005 6:43 PM
2005 Aug 17
4
Voicemail Retrival
Hi, I am very new to Asterisk. I wanted to know how to retrive the Voicemails. I could see some voicemails assosiated with some extensions. Any ideas?? --------------------------------- How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos. Get Yahoo! Photos -------------- next part -------------- An HTML attachment was scrubbed... URL:
2003 Oct 11
1
SIP / IAX over satellite
Hi all, ------ I tried to use * over satellite, but all my effort did not succeed. The Asterisk is behind the VSAT and is resposibel for alle the SIP clients in a field location. The clients are notebooks and PDA's running SJPhoen for Windows and PocketPC. Unfortunately I could not find any Linux Client wich worked satisfying. SJ LAbs promised a Linux Version at the end of August but they
2005 Mar 18
1
Broadvoice hangs-up / disconnects after about 30 deconds
I have just installed * from the latest CVS and I can make calls via X-Lite to outside numbers but for only 30 to 35 Seconds at a time then Broadvoice will hang up on me but X-Lite will not know it. Once I hang up X-Lite then I will get the following error message: Spawn extension (default, Number-I-Dialed, 1) exited non-zero on 'SIP/200-6a22' -- Got SIP response 404 "Not Found"
2005 Aug 10
1
asterisk query mysql problem or bug?
Hi; I have entries as below in DB, mysql> select * from sip_buddies; +----+------+----------+------------+---------+------------+--------+------- -----+------------+----------+------+ | id | name | context | defaultip | host | mailbox | type | regseconds | ipaddr | username | port | +----+------+----------+------------+---------+------------+--------+-------
2003 Nov 02
6
Asterisk behind LinkSys NAT Routing
Problem I have is this. outside firewall (extension 2003) can call me inside firewall (extension 2000) and all is fine. If I call from inside firewall (extension 2000) to outside firewall (extension 2003) I hear no ringing and person at other end can pick up and I hear for maybe a half second then I go to voicemail. If I add another extension on the outside then communication between outside
2005 Mar 18
2
No sound when calling in from pstn
I am just starting out with * so bear with me please. I have tdm400p with 4 fxo modules on it. When I call into the asterisk box from my mobile, I can see the asterisk console picks the call up and routes it to my computer with x-lite. There was no sound coming from either - just silence. I then decided to route it directly to voice mail to see if that would narrow the problem down, but it
2005 Feb 10
4
why asterisk is replying 404 Not Found
[3000] type=friend dtmfmode=INFO insecure=yes canreinvite=no auth=plaintext host=dynamic allow=ulaw [2000] type=friend dtmfmode=INFO insecure=yes canreinvite=no auth=plaintext host=dynamic allow=ulaw i have declared these two users 3000 and 2000. they are registering successfully. problem is that
2004 Jul 05
4
Question about x100P and zap
I have 2 X100P card and configured everything based on configs here http://www.onlamp.com/onlamp/2004/01/22/examples/config_files.txt I changed the area codes to match mine. When I try to dial out I get app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' A zap show channels gives me this Chan Extension Context Language MusicOnHold 1
2005 Jul 15
2
[Aserisk-Users]no audio inside the net
Hi list, i've problems with my * server and the 4 phones which are linked to it. i've 2 grandstream bt100 with the firmware upgraded to 101, a wi-fi phone (i don't know its brand) and another ip phone i don't know its brand. with this sip.conf : [general] port = 5060 bindaddr = 192.168.100.229 context = default ;x changed from default to sip localnet = 192.168.100.0/24
2005 Aug 26
1
Fw: IAX2 Softphone Quality & Network Cards
> Hi! > >> We are in the process of an Asterisk call center deployment using IAX2 >> G711 ulaw softphones. Outbound sound quality is terrible. RESPONSE: This is Bill McCready from PCPhoneline.com . To address your sound quality issues, you may want to check out our VTA1000 Skype+SIP+IAX2 Tri-Mode Gateway which plugs into the USB port of a Windows computer. You can use it
2005 Jan 30
4
Processing incoming calls with multiple contextst over PRI
So I have a problem. A customer of mine wants a PBX, owns an office building. I want to sell him on asterisk. He has 4 tenants. I am using my asterisk box to simulate it. My asterisk box has a TDM400P card, not a PRI card. Don't know if it makes any difference. Anyway, I want to route incoming phone calls to different contexts based on the phone number being called. Here is my
2005 Jan 12
4
Is this a $50 wifi or wireless USB VOIP phone ?
http://www.pcphoneline.com/skype "The VPT1000 is NOT a simple last generation USB phone audio device but is rather a next generation integrated gateway and USB phoneset with simultaneous dual mode Skype and SIP calling support. Skype is not forecast to have "SkypeIn" available until June 2005 but you can have the capability now via its built in SIP capabilities." Is this a