Re: [asterisk-users] passing '302 moved temporarily' back to the SIP provider

2013-05-08 Thread Satish Barot
On 5/9/13, Satish Barot wrote: > On 5/9/13, Carlos Alvarez wrote: >> On Tue, May 7, 2013 at 10:05 PM, Satish Barot >> wrote: >> >>> >>> >>> promiscredir= yes in sip.conf should help you achieve your requirement. >>> >> >> I haven't been able to get that to work in a similar situation, except we >

Re: [asterisk-users] passing '302 moved temporarily' back to the SIP provider

2013-05-08 Thread Satish Barot
On 5/9/13, Carlos Alvarez wrote: > On Tue, May 7, 2013 at 10:05 PM, Satish Barot > wrote: > >> >> >> promiscredir= yes in sip.conf should help you achieve your requirement. >> > > I haven't been able to get that to work in a similar situation, except we > are the provider. It results in the new i

Re: [asterisk-users] passing '302 moved temporarily' back to the SIP provider

2013-05-08 Thread Carlos Alvarez
On Tue, May 7, 2013 at 10:05 PM, Satish Barot wrote: > > > promiscredir= yes in sip.conf should help you achieve your requirement. > I haven't been able to get that to work in a similar situation, except we are the provider. It results in the new invite being from the CLID of the original caller

[asterisk-users] No early media on 302 redirect via two servers

2013-05-08 Thread Carlos Alvarez
We have a situation where we get no early media in this call flow: VoIP origination provider Server1 (our server) Customer server Customer phone with call-forward set Server1 to dial the forward-to number Then there is no early media while the forward-to number is ringing. Our server is Asterisk

[asterisk-users] Transfer cmd via AsyncAGI

2013-05-08 Thread Dan Cropp
Hello, I am using Asterisk 11.0.1 and do not notice any changes regarding the Transfer on newer Asterisk 11.x versions. I am using AMI and controlling a channel via AsyncAGI. I send a Transfer cmd (such as the following) Action: AGI ActionID: C8 Channel: SIP/1004-0002 Command

Re: [asterisk-users] hwo to stok variable wiith menu

2013-05-08 Thread Salaheddine Elharit
hello list i would your help please regarding this issue with the below code i can store the call date and the callerid ,now i want to store also the sip phone called 223 could you please see the code and tell me how can i add the sip phone in my table 'Menu' exten => 506,1,Ringing() exten =>

[asterisk-users] Confbridge Dynamic video_mode

2013-05-08 Thread Rizwan Hisham
Hi All, I want to set the video_mode of the confbridge dynamically in the dialplan. SO say if 5 users join the conference with follow_talker mode, it should work like that (and it does). But if 6th user changes the video_mode to first_marked and gets marked in the dial plan and joins the conference

Re: [asterisk-users] Obtaining high voice quality in dahdi channel

2013-05-08 Thread jg
Asterisk uses "echo cancellation" to enhance audio. EC can be done in soft- or hardware. All major card manufacturers have variants of their cards with hardware echo cancellation support. Searching the internet gives a lot of information about this topic. If you install a Sangoma card without