[asterisk-users] py-Asterisk or pyst?
Hi there, I need a Python interface to asterisk manager for my own project. The voip-info.org (http://www.voip-info.org/wiki/view/Asterisk+manager+Examples) lists 4 python projects for this purpose: Fats, py-Asterisk, pyst and StarPy. Because my project is rather small and I don't want to involve twisted in, the options left for me are py-Asterisk and pyst. So I want to ask your opinion: Which project is easier to get started? Pros and cons? Thank you! -- Ye Liu (AKA @jaux) http://jaux.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] echo when calling to the pstn
I'm assuming you haven't googled for solution, please go through http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation and extra links in that article. If that article were not helpful, please provide more information of you setup, such as what analog card are you using, are you using software or hardware echo canceller, how does your chan_dahdi.conf look like, etc. On Tue, Feb 8, 2011 at 3:11 PM, Vitor Carlos Flausino wrote: > Hello all. > > I have a asterisk implementation (asterisknow 1.7.1), with a card with 2 FXO > interfaces. > > When I call (or receive a call) from the pstn, I ear echo. This happens if I > use a softphone or IP phone, and does not happens if the call is internal. > > Can you help me with this issue? > > Best regards, > Vitor Flausino > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Ye Liu (AKA @jaux) http://jaux.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phone multi-registration
Yes, it is possible. Multiple peers can be registered by a single IP. I have Snom 300 which supports 4 identities. On Fri, Jan 21, 2011 at 2:28 PM, Warren Selby wrote: > On Fri, Jan 21, 2011 at 12:53 PM, Olivier wrote: >> >> Hi, >> >> Is it possible for a SIP to register twice to the same Asterisk server >> using 2 different ids ? >> Consulting this list archives gives mixed answers. >> > > Yes, I do this with my Polycom 550, and I've done it with other phones > before as well. Show us the CLI output and possibly a SIP Debug of the > registration attempts. > > -- > Thanks, > --Warren Selby, dCAP > http://www.selbytech.com > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Ye Liu (AKA @jaux) http://jaux.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Failed SIP registration kicks registered device off?
Hi folks, I'm currently running a modified version of Asterisk 1.6.1.1, I observed an unexpected behavior of my system today: 1. SIP device A successfully registered extension 100; 2. SIP device B tried to register extension 100 but with wrong password, so registration failed; 3. A then showed it was unregistered! Failed registration of device B shouldn't kick A off, I expect A stay online and work properly in this situation. Could anyone confirm this? Because my asterisk is modified, I'm not sure this behavior is in vanilla asterisk or it is caused by my own code. Thank you! -- Ye Liu (AKA @jaux) http://jaux.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detecting hook flash in asterisk
Hi Paul, On Sat, Jun 26, 2010 at 1:33 PM, Paul Belanger wrote: > On Sat, Jun 26, 2010 at 7:33 AM, Deepesh D wrote: >> Is it possible to do this action on hook flash? >> > Currently no. You would need to add logic to the channel driver. Or > use DTMF to initiate the hookflash: My PSTN line has call waiting, and I have to use zapflash application to answer the new incoming call. If I want to flash hook to switch calls, which channel driver do I need to look at? chan_dahdi? I noticed that I can use hook flash to switch between SIP calls, or even between a SIP call and a PSTN call, does this mean chan_sip has such hook flash detection logic so I can learn from there? > > extensions.conf > [globals] > DYNAMIC_FEATURES=>zapflash > > features.conf > [applicationmap] > zapflash => *0,callee,flash,() > > -- > Paul Belanger | dCAP > Polybeacon | Consultant > Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) > blog.polybeacon.com > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Ye Liu (AKA @jaux) http://jaux.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Callerid over IAX Trunks
Hi Alyed, Thank you for the response. I tried this solution, I got "Unknown" displayed instead of 999. Also, I tried both 200 and <200> as the CID number for the extension, but the results were the same. On Sat, Apr 10, 2010 at 2:10 PM, Alyed wrote: > Don't have a system to test this right now, but read somewhere this was a 2 > steps solution: > > 1) Leave the callerid in your tunk definition blank (in your example the 999 > username) > > 2) Use brakets around the callerid definition of your peers: callerid= <200> > (extension 200 in your example) > > Let us know if it worked. > > Alyed > > > 2010/4/9 Ye Liu >> >> Hello everyone, >> >> I'm fairly new to asterisk and this list. Currently I'm working on IAX >> trunks to send/receive calls between 2 asterisk boxes with asterisk >> 1.6.1.1+asterisk gui 2.0. After some work in the gui, two boxes can >> send/receive calls to/from the other just fine, the only problem I >> have is the caller id. >> >> Here is my setup: >> >> 1. on both boxes, I added an IAX user in the gui, say the extension >> and password are 999 >> 2. I then created IAX trunks for each box using 999 as username and >> password, hostname/IP was set to be other box's IP >> 3. when done, from the system status panel, I saw the trunks >> successfully registered to the other box >> 4. then I added "Outgoing Call Rules" to each box: >> for box1, _2XX --> to_box2_trunk >> for box2, _1XX --> to_box1_trunk >> >> This setup works ok, the only problem is caller id, i.e. when >> extension(200) from box2 calls to extension(100) from box1, the call >> can be made but the caller id displayed on 100 is 999 not 200. >> >> I have been on this problem for some time already, could anyone here >> give me a bit help please? >> -- >> Ye Liu (AKA @jaux) >> >> http://jaux.net >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > _________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Ye Liu (AKA @jaux) http://jaux.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Callerid over IAX Trunks
Hello everyone, I'm fairly new to asterisk and this list. Currently I'm working on IAX trunks to send/receive calls between 2 asterisk boxes with asterisk 1.6.1.1+asterisk gui 2.0. After some work in the gui, two boxes can send/receive calls to/from the other just fine, the only problem I have is the caller id. Here is my setup: 1. on both boxes, I added an IAX user in the gui, say the extension and password are 999 2. I then created IAX trunks for each box using 999 as username and password, hostname/IP was set to be other box's IP 3. when done, from the system status panel, I saw the trunks successfully registered to the other box 4. then I added "Outgoing Call Rules" to each box: for box1, _2XX --> to_box2_trunk for box2, _1XX --> to_box1_trunk This setup works ok, the only problem is caller id, i.e. when extension(200) from box2 calls to extension(100) from box1, the call can be made but the caller id displayed on 100 is 999 not 200. I have been on this problem for some time already, could anyone here give me a bit help please? -- Ye Liu (AKA @jaux) http://jaux.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
