[asterisk-users] Asterisk Call Forwarding

Hi I have to setup call forwarding. How do we setup Call forwarding in asterisk?. Eg. user dials a number and insert some mobile number for forwarding and dial another number to cancel the forwarding. thanks a lot. Best Regards, Madushan --

Re: [asterisk-users] RTP / NAT question ( pjsip )

Hi Joshua, Looking at the transmitted SIP packets from Asterisk, it looks like Asterisk is only sending it’s own internal IP (it is behind a NAT too, with proper port forwarding) . I did set in my transport the external_signaling_address and external_media_address , and I have now put

Re: [asterisk-users] RTP / NAT question ( pjsip )

Kevin Long wrote: Thank you for the response Joshua . I had rtp_symmetric=yes before I wrote the email, then I set it to no, restart asterisk, and tried to make the call from the remote endpoint again but still tcpdump is showing me the RTP packets are being sent from Asterisk to the

Re: [asterisk-users] RTP / NAT question ( pjsip )

Thank you for the response Joshua . I had rtp_symmetric=yes before I wrote the email, then I set it to no, restart asterisk, and tried to make the call from the remote endpoint again but still tcpdump is showing me the RTP packets are being sent from Asterisk to the private IP. tcpdump

Re: [asterisk-users] RTP / NAT question ( pjsip )

Kevin Long wrote: I am having trouble with RTP and NAT : Below is a SIP SDP invite from a remote endpoint which is trying to call extension 420 which is the ECHO application . As you can see, the public IP is where the request comes in from, but the SDP contains the private, internal IP in

[asterisk-users] RTP / NAT question ( pjsip )

I am having trouble with RTP and NAT : Below is a SIP SDP invite from a remote endpoint which is trying to call extension 420 which is the ECHO application . As you can see, the public IP is where the request comes in from, but the SDP contains the private, internal IP in numerous places.

Re: [asterisk-users] Dial your phone and contact phone from within outlook?

Hi Travis, Have a look at this: http://www.ipcom.at/en/telephony/siptapi/ I have used this in the past to do something similar, unless you have an Exchange Enterprise setup in which case I would suggest exploring unified messaging Thanks, Neeraj On Thu, Mar 3, 2016 at 8:22 AM, Ryan, Travis

[asterisk-users] How to install Huawei E153 in a Asterisk 11 or 13?

Hi everyone! I tried to install chan_dongle for Asterisk 11 in a Ubuntu 14.04, but my Huawei E153 is not working in my Asterisk. I fallow this rules http://blog.denisbondar.com/post/asterisk11-chan_dongle_e1550-ubuntu14 But not successes. Thanks in advanced, --

[asterisk-users] Dial your phone and contact phone from within outlook?

I am wondering what the best solution is for initiating a call from Outlook Contacts. I imagine something that would start the call from the outlook card (or similar) and then dial the user's extension and the contact's phone number and place them in a bridge. Anyone use something like this?

Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

I'm discovering WebRTC and I think it's a technology that is quite difficult to integrate with so many changing interfaces. I think this is typically the kind of subject where the community could positively contribute to keep wiki pages updated. As I'm quite interested in this topic, I'm

Re: [asterisk-users] DAHDI-Linux and DAHDI-Tools 2.11.1 Now Available

Hello; I have a question about Dahdi-Linux and Dahdi-Tools. If I'm using a particular version of Dahdi-Linux, say x.y.z, does Dahdi-Tools have to be the same version? Thanks; John V. -Original Message- From: asterisk-users-boun...@lists.digium.com

[asterisk-users] function SHARED and function IMPORT : 2 questions

Hello I am trying to use the functions SHARED and IMPORT to share variables across SIP-channels. During my use I encounter 2 problems/questions. Question 1. only 1 shared variable per channel ?? When I set 2 shared variables on a channel, and I read them a bit futher in the dialplan,

Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

Olivier wrote: 2016-02-19 12:01 GMT+01:00 Marek Červenka >: on my own server Today, I'm back from holidays trip. First of all, thanks for replying ! I'll try to use jssip as you suggested. Anyway, I'm still failing to understand if

Re: [asterisk-users] Abandoned SIP-TCP connection causes Asterisk to crash

Nasir Iqbal wrote: Hi All, We are using SIP over TCP transport but often we got an Asterisk crash with following error. [Mar 1 11:23:13] WARNING[1509]: chan_sip.c:3755 __sip_xmit: sip_xmit of 0x7f294000cac0 (len 680) to Soft.Phone.IP.Address:56780 returned -2: Interrupted system call

Re: [asterisk-users] Can't send 10 type frames with PJSIP

Carlos Chavez wrote: I am getting flooded with these messages: [Mar 1 12:25:29] WARNING[6962][C-005a]: chan_pjsip.c:712 chan_pjsip_write: Can't send 10 type frames with PJSIP [Mar 1 12:25:30] WARNING[6962][C-005a]: chan_pjsip.c:712 chan_pjsip_write: Can't send 10 type frames with PJSIP

Re: [asterisk-users] DTMF issues between Asterisk and Callmanager with Zoiper

Carlos Chavez wrote: I had an old Asterisk installation die recently and we decided to upgrade to Asterisk 13 to replace the old server. Everything seems to be working with PJSIP but there is one issue. Asterisk talks to a callmanager via a SIP trunk and send calls to PSTN (another country).