I have a home asterisk box which connects to the office asterisk, so I can just dial extensions. This used to work just fine. Im using 10.0-rc1 on the home box, 188.8.131.52 on the office. But it doesnt work now: [Nov 14 18:38:19] NOTICE: chan_sip.c:13..
, Im still having trouble using asterisk with the grandstream gxw4108, in the gxw4108 Im using 1 stage dialing in the profile1 I already type my asterisk server address 192.168.14.80 and my grandstream IP is 192.168.101.184 heres my asterisk config fi..
I am troubleshooting a SIP trunk problem. The system is Asterisk 1.8.5. The problem is cant make any outbound/inbound. It always get Number is not valid 701. I tried to figure out the reason the call got dropped and couldnt find out the solution. I noti..
We have a client that has sporadic problems with the *8 pickup facility. The server they are using is 1.8.5 and they are using Snom phones. Every now and then when they try to do a pickup from another phone they get a forbidden message on the phone ..
Good morning, I am writing a Asterisk dialplan from scratch (for learning and testing purposes), but im having trouble with a algorithm to dial a SIP group using round-robin. I want that asterisk dial the member of the group in a circular way, un..
, is there somebody how is able to help me here? Or at least to get more details why this occurs? best regards Christoph Am 08.06.2011 18:00, schrieb Christoph Timm: >, > > Im running into trouble, if I perform a yum update on my AsteriskNOW. > > Curren..
Hi. Im having trouble setting variables in channel dialplan and re-using them in Extension dialplan… Im using the following call file: Channel: Local/210332450@ZonNew-Outbound CallerID: ZonNew-Outbound:49:210332450: MaxRetries: 5 RetryTime: 10 WaitTi..
Steve. Thanks for the insight. I wont pretend to know what early-audio is, but I guess Im about to find out :-). Also, I believe that I have a nearly identical setup like this with the exact same SIP provider w/o any trouble. However, I think that sys..
I am having trouble running the command siptest:~# asterisk -rx dialplan reload most times it does what I expect and I get a response as below siptest:~# asterisk -rx dialplan reload Dialplan reloaded. every now and then I get no response i.e. siptest..
I am having trouble with no return audio on inbound calls. I have been working on this for 18 hours and even built a fresh server and moved everything over and I am getting the same results. I need someone that can help get this resolved tonight. I ..